Removed OpenAL backend

5 files changed, 12 insertions, 641 deletions

diff --git a/src/CMakeLists.txt b/src/CMakeLists.txt
index 005b22cb..1b1d0d06 100644
--- a/src/CMakeLists.txt
+++ b/src/CMakeLists.txt
@@ -45,13 +45,8 @@ endif()
 add_definitions("-DRAYLIB_CMAKE=1")
 
 if(USE_AUDIO)
-  if (NOT USE_OPENAL_BACKEND)
-    file(GLOB mini_al external/mini_al.c)
-    MESSAGE(STATUS "Audio Backend: mini_al")
-  else()
-    find_package(OpenAL REQUIRED)
-    MESSAGE(STATUS "Audio Backend: OpenAL")
-  endif()
+  file(GLOB mini_al external/mini_al.c)
+  MESSAGE(STATUS "Audio Backend: mini_al")
   file(GLOB stb_vorbis external/stb_vorbis.c)
   set(sources ${raylib_sources} ${mini_al} ${stb_vorbis})
 else()
diff --git a/src/CMakeOptions.txt b/src/CMakeOptions.txt
index d4ecb392..eee3f1a9 100644
--- a/src/CMakeOptions.txt
+++ b/src/CMakeOptions.txt
@@ -12,11 +12,6 @@ option(SHARED "Build raylib as a dynamic library" OFF)
 option(STATIC "Build raylib as a static library" ON)
 option(MACOS_FATLIB  "Build fat library for both i386 and x86_64 on macOS" OFF)
 option(USE_AUDIO "Build raylib with audio module" ON)
-if(${PLATFORM} MATCHES "Web")
-  cmake_dependent_option(USE_OPENAL_BACKEND "Link raylib with openAL instead of mini-al" ON "USE_AUDIO" OFF)
-else()
-  cmake_dependent_option(USE_OPENAL_BACKEND "Link raylib with openAL instead of mini-al" OFF "USE_AUDIO" OFF)
-endif()
 
 enum_option(USE_EXTERNAL_GLFW "OFF;IF_POSSIBLE;ON" "Link raylib against system GLFW instead of embedded one")
 if(UNIX AND NOT APPLE)
diff --git a/src/Makefile b/src/Makefile
index ac4b15b4..28b73beb 100644
--- a/src/Makefile
+++ b/src/Makefile
@@ -63,14 +63,6 @@ RAYLIB_BUILD_MODE    ?= RELEASE
 # NOTE: Some programs like tools could not require audio support
 INCLUDE_AUDIO_MODULE ?= TRUE
 
-# Use OpenAL Soft backend for audio
-USE_OPENAL_BACKEND   ?= FALSE
-
-# OpenAL Soft audio backend forced on HTML5 and OSX (see below)
-ifeq ($(PLATFORM),PLATFORM_WEB)
-    USE_OPENAL_BACKEND = TRUE
-endif
-
 # Use external GLFW library instead of rglfw module
 # TODO: Review usage of examples on Linux.
 USE_EXTERNAL_GLFW    ?= FALSE
@@ -154,13 +146,6 @@ endif
 #    RAYLIB_PATH       ?= /home/pi/raylib
 #endif
 
-# Force OpenAL Soft audio backend for OSX platform
-# NOTE 1: mini_al library does not support CoreAudio yet
-# NOTE 2: Required OpenAL libraries should be available on OSX
-ifeq ($(PLATFORM_OS),OSX)
-    USE_OPENAL_BACKEND = TRUE
-endif
-
 ifeq ($(PLATFORM),PLATFORM_WEB)
     # Emscripten required variables
     EMSDK_PATH          = C:/emsdk
@@ -343,11 +328,6 @@ ifeq ($(RAYLIB_LIBTYPE),SHARED)
     CFLAGS += -fPIC -DBUILD_LIBTYPE_SHARED
 endif
 
-# Use OpenAL Soft backend instead of mini_al
-ifeq ($(USE_OPENAL_BACKEND),TRUE)
-    CFLAGS += -DUSE_OPENAL_BACKEND
-endif
-
 # Use Wayland display on Linux desktop
 ifeq ($(PLATFORM),PLATFORM_DESKTOP)
     ifeq ($(PLATFORM_OS), LINUX)
@@ -426,9 +406,7 @@ endif
 ifeq ($(INCLUDE_AUDIO_MODULE),TRUE)
     OBJS += audio.o
     OBJS += stb_vorbis.o
-    ifeq ($(USE_OPENAL_BACKEND),FALSE)
-        OBJS += mini_al.o
-    endif
+    OBJS += mini_al.o
 endif
 
 ifeq ($(PLATFORM),PLATFORM_ANDROID)
diff --git a/src/audio.c b/src/audio.c
index fc65127b..a646c981 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -16,9 +16,6 @@
 *       Define to use the module as standalone library (independently of raylib).
 *       Required types and functions are defined in the same module.
 *
-*   #define USE_OPENAL_BACKEND
-*       Use OpenAL Soft audio backend
-*
 *   #define SUPPORT_FILEFORMAT_WAV
 *   #define SUPPORT_FILEFORMAT_OGG
 *   #define SUPPORT_FILEFORMAT_XM
@@ -82,25 +79,9 @@
     #include "utils.h"          // Required for: fopen() Android mapping
 #endif
 
-#if !defined(USE_OPENAL_BACKEND)
-    #define USE_MINI_AL 1       // Set to 1 to use mini_al; 0 to use OpenAL.
-#endif
-
-#include "external/mini_al.h"   // Implemented in mini_al.c. Cannot implement this here because it conflicts with Win32 APIs such as CloseWindow(), etc.
-
-#if !defined(USE_MINI_AL) || (USE_MINI_AL == 0)
-    #if defined(__APPLE__)
-        #include "OpenAL/al.h"          // OpenAL basic header
-        #include "OpenAL/alc.h"         // OpenAL context header (like OpenGL, OpenAL requires a context to work)
-    #else
-        #include "AL/al.h"              // OpenAL basic header
-        #include "AL/alc.h"             // OpenAL context header (like OpenGL, OpenAL requires a context to work)
-        //#include "AL/alext.h"         // OpenAL extensions header, required for AL_EXT_FLOAT32 and AL_EXT_MCFORMATS
-    #endif
-
-    // OpenAL extension: AL_EXT_FLOAT32 - Support for 32bit float samples
-    // OpenAL extension: AL_EXT_MCFORMATS - Support for multi-channel formats (Quad, 5.1, 6.1, 7.1)
-#endif
+#include "external/mini_al.h"   // mini_al audio library
+                                // NOTE: Cannot be implement here because it conflicts with 
+                                // Win32 APIs: Rectangle, CloseWindow(), ShowCursor(), PlaySoundA()
 
 #include <stdlib.h>             // Required for: malloc(), free()
 #include <string.h>             // Required for: strcmp(), strncmp()
@@ -147,15 +128,6 @@
 // In case of music-stalls, just increase this number
 #define AUDIO_BUFFER_SIZE        4096    // PCM data samples (i.e. 16bit, Mono: 8Kb)
 
-// Support uncompressed PCM data in 32-bit float IEEE format
-// NOTE: This definition is included in "AL/alext.h", but some OpenAL implementations
-// could not provide the extensions header (Android), so its defined here
-#if !defined(AL_EXT_float32)
-    #define AL_EXT_float32              1
-    #define AL_FORMAT_MONO_FLOAT32      0x10010
-    #define AL_FORMAT_STEREO_FLOAT32    0x10011
-#endif
-
 //----------------------------------------------------------------------------------
 // Types and Structures Definition
 //----------------------------------------------------------------------------------
@@ -233,8 +205,6 @@ void TraceLog(int msgType, const char *text, ...);              // Show trace lo
 //----------------------------------------------------------------------------------
 // mini_al AudioBuffer Functionality
 //----------------------------------------------------------------------------------
-#if USE_MINI_AL
-
 #define DEVICE_FORMAT       mal_format_f32
 #define DEVICE_CHANNELS     2
 #define DEVICE_SAMPLE_RATE  44100
@@ -487,7 +457,6 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 f
         }
     }
 }
-#endif
 
 //----------------------------------------------------------------------------------
 // Module Functions Definition - Audio Device initialization and Closing
@@ -495,7 +464,6 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 f
 // Initialize audio device
 void InitAudioDevice(void)
 {
-#if USE_MINI_AL
     // Context.
     mal_context_config contextConfig = mal_context_config_init(OnLog);
     mal_result result = mal_context_init(NULL, 0, &contextConfig, &context);
@@ -545,45 +513,11 @@ void InitAudioDevice(void)
     TraceLog(LOG_INFO, "Audio buffer size: %d", device.bufferSizeInFrames);
 
     isAudioInitialized = MAL_TRUE;
-#else
-    // Open and initialize a device with default settings
-    ALCdevice *device = alcOpenDevice(NULL);
-
-    if (!device) TraceLog(LOG_ERROR, "Audio device could not be opened");
-    else
-    {
-        ALCcontext *context = alcCreateContext(device, NULL);
-
-        if ((context == NULL) || (alcMakeContextCurrent(context) == ALC_FALSE))
-        {
-            if (context != NULL) alcDestroyContext(context);
-
-            alcCloseDevice(device);
-
-            TraceLog(LOG_ERROR, "Could not initialize audio context");
-        }
-        else
-        {
-            TraceLog(LOG_INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER));
-
-            // Listener definition (just for 2D)
-            alListener3f(AL_POSITION, 0.0f, 0.0f, 0.0f);
-            alListener3f(AL_VELOCITY, 0.0f, 0.0f, 0.0f);
-            alListener3f(AL_ORIENTATION, 0.0f, 0.0f, -1.0f);
-            
-            alListenerf(AL_GAIN, 1.0f);
-
-            if (alIsExtensionPresent("AL_EXT_float32")) TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 supported");
-            else TraceLog(LOG_INFO, "[EXTENSION] AL_EXT_float32 not supported");
-        }
-    }
-#endif
 }
 
 // Close the audio device for all contexts
 void CloseAudioDevice(void)
 {
-#if USE_MINI_AL
     if (!isAudioInitialized) 
     {
         TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
@@ -593,18 +527,6 @@ void CloseAudioDevice(void)
     mal_mutex_uninit(&audioLock);
     mal_device_uninit(&device);
     mal_context_uninit(&context);
-#else
-    ALCdevice *device;
-    ALCcontext *context = alcGetCurrentContext();
-
-    if (context == NULL) TraceLog(LOG_WARNING, "Could not get current audio context for closing");
-
-    device = alcGetContextsDevice(context);
-
-    alcMakeContextCurrent(NULL);
-    alcDestroyContext(context);
-    alcCloseDevice(device);
-#endif
 
     TraceLog(LOG_INFO, "Audio device closed successfully");
 }
@@ -612,20 +534,7 @@ void CloseAudioDevice(void)
 // Check if device has been initialized successfully
 bool IsAudioDeviceReady(void)
 {
-#if USE_MINI_AL
     return isAudioInitialized;
-#else
-    ALCcontext *context = alcGetCurrentContext();
-
-    if (context == NULL) return false;
-    else
-    {
-        ALCdevice *device = alcGetContextsDevice(context);
-
-        if (device == NULL) return false;
-        else return true;
-    }
-#endif
 }
 
 // Set master volume (listener)
@@ -634,17 +543,13 @@ void SetMasterVolume(float volume)
     if (volume < 0.0f) volume = 0.0f;
     else if (volume > 1.0f) volume = 1.0f;
  
-#if USE_MINI_AL
     masterVolume = volume;
-#else
-    alListenerf(AL_GAIN, volume);
-#endif
 }
 
 //----------------------------------------------------------------------------------
 // Module Functions Definition - Audio Buffer management
 //----------------------------------------------------------------------------------
-#if USE_MINI_AL
+
 // Create a new audio buffer. Initially filled with silence
 AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint32 sampleRate, mal_uint32 bufferSizeInFrames, AudioBufferUsage usage)
 {
@@ -843,7 +748,6 @@ void UntrackAudioBuffer(AudioBuffer *audioBuffer)
     
     mal_mutex_unlock(&audioLock);
 }
-#endif
 
 //----------------------------------------------------------------------------------
 // Module Functions Definition - Sounds loading and playing (.WAV)
@@ -909,7 +813,6 @@ Sound LoadSoundFromWave(Wave wave)
 
     if (wave.data != NULL)
     {
-#if USE_MINI_AL
         // When using mini_al 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:
         // 
@@ -931,61 +834,6 @@ Sound LoadSoundFromWave(Wave wave)
         if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");
 
         sound.audioBuffer = audioBuffer;
-#else
-        ALenum format = 0;
-
-        // The OpenAL format is worked out by looking at the number of channels and the sample size (bits per sample)
-        if (wave.channels == 1)
-        {
-            switch (wave.sampleSize)
-            {
-                case 8: format = AL_FORMAT_MONO8; break;
-                case 16: format = AL_FORMAT_MONO16; break;
-                case 32: format = AL_FORMAT_MONO_FLOAT32; break;  // Requires OpenAL extension: AL_EXT_FLOAT32
-                default: TraceLog(LOG_WARNING, "Wave sample size not supported: %i", wave.sampleSize); break;
-            }
-        }
-        else if (wave.channels == 2)
-        {
-            switch (wave.sampleSize)
-            {
-                case 8: format = AL_FORMAT_STEREO8; break;
-                case 16: format = AL_FORMAT_STEREO16; break;
-                case 32: format = AL_FORMAT_STEREO_FLOAT32; break;  // Requires OpenAL extension: AL_EXT_FLOAT32
-                default: TraceLog(LOG_WARNING, "Wave sample size not supported: %i", wave.sampleSize); break;
-            }
-        }
-        else TraceLog(LOG_WARNING, "Wave number of channels not supported: %i", wave.channels);
-
-        // Create an audio source
-        ALuint source;
-        alGenSources(1, &source);            // Generate pointer to audio source
-
-        alSourcef(source, AL_PITCH, 1.0f);
-        alSourcef(source, AL_GAIN, 1.0f);
-        alSource3f(source, AL_POSITION, 0.0f, 0.0f, 0.0f);
-        alSource3f(source, AL_VELOCITY, 0.0f, 0.0f, 0.0f);
-        alSourcei(source, AL_LOOPING, AL_FALSE);
-
-        // Convert loaded data to OpenAL buffer
-        //----------------------------------------
-        ALuint buffer;
-        alGenBuffers(1, &buffer);            // Generate pointer to buffer
-
-        unsigned int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;    // Size in bytes
-
-        // Upload sound data to buffer
-        alBufferData(buffer, format, wave.data, dataSize, wave.sampleRate);
-
-        // Attach sound buffer to source
-        alSourcei(source, AL_BUFFER, buffer);
-
-        TraceLog(LOG_INFO, "[SND ID %i][BUFR ID %i] Sound data loaded successfully (%i Hz, %i bit, %s)", source, buffer, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
-
-        sound.source = source;
-        sound.buffer = buffer;
-        sound.format = format;
-#endif
     }
 
     return sound;
@@ -1002,14 +850,7 @@ void UnloadWave(Wave wave)
 // Unload sound
 void UnloadSound(Sound sound)
 {
-#if USE_MINI_AL
     DeleteAudioBuffer((AudioBuffer *)sound.audioBuffer);
-#else
-    alSourceStop(sound.source);
-
-    alDeleteSources(1, &sound.source);
-    alDeleteBuffers(1, &sound.buffer);
-#endif
 
     TraceLog(LOG_INFO, "[SND ID %i][BUFR ID %i] Unloaded sound data from RAM", sound.source, sound.buffer);
 }
@@ -1018,8 +859,8 @@ void UnloadSound(Sound sound)
 // NOTE: data must match sound.format
 void UpdateSound(Sound sound, const void *data, int samplesCount)
 {
-#if USE_MINI_AL
     AudioBuffer *audioBuffer = (AudioBuffer *)sound.audioBuffer;
+    
     if (audioBuffer == NULL)
     {
         TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
@@ -1030,29 +871,6 @@ void UpdateSound(Sound sound, const void *data, int samplesCount)
 
     // 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*mal_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn));
-#else
-    ALint sampleRate, sampleSize, channels;
-    alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate);
-    alGetBufferi(sound.buffer, AL_BITS, &sampleSize);           // It could also be retrieved from sound.format
-    alGetBufferi(sound.buffer, AL_CHANNELS, &channels);         // It could also be retrieved from sound.format
-
-    TraceLog(LOG_DEBUG, "UpdateSound() : AL_FREQUENCY: %i", sampleRate);
-    TraceLog(LOG_DEBUG, "UpdateSound() : AL_BITS: %i", sampleSize);
-    TraceLog(LOG_DEBUG, "UpdateSound() : AL_CHANNELS: %i", channels);
-
-    unsigned int dataSize = samplesCount*channels*sampleSize/8;   // Size of data in bytes
-
-    alSourceStop(sound.source);                 // Stop sound
-    alSourcei(sound.source, AL_BUFFER, 0);      // Unbind buffer from sound to update
-    //alDeleteBuffers(1, &sound.buffer);          // Delete current buffer data
-    //alGenBuffers(1, &sound.buffer);             // Generate new buffer
-
-    // Upload new data to sound buffer
-    alBufferData(sound.buffer, sound.format, data, dataSize, sampleRate);
-
-    // Attach sound buffer to source again
-    alSourcei(sound.source, AL_BUFFER, sound.buffer);
-#endif
 }
 
 // Export wave data to file
@@ -1141,102 +959,48 @@ void ExportWave(Wave wave, const char *fileName)
 // Play a sound
 void PlaySound(Sound sound)
 {
-#if USE_MINI_AL
     PlayAudioBuffer((AudioBuffer *)sound.audioBuffer);
-#else
-    alSourcePlay(sound.source);        // Play the sound
-#endif
-
-    //TraceLog(LOG_INFO, "Playing sound");
-
-    // Find the current position of the sound being played
-    // NOTE: Only work when the entire file is in a single buffer
-    //int byteOffset;
-    //alGetSourcei(sound.source, AL_BYTE_OFFSET, &byteOffset);
-    //
-    //int sampleRate;
-    //alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate);    // AL_CHANNELS, AL_BITS (bps)
-
-    //float seconds = (float)byteOffset/sampleRate;      // Number of seconds since the beginning of the sound
-    //or
-    //float result;
-    //alGetSourcef(sound.source, AL_SEC_OFFSET, &result);   // AL_SAMPLE_OFFSET
 }
 
 // Pause a sound
 void PauseSound(Sound sound)
 {
-#if USE_MINI_AL
     PauseAudioBuffer((AudioBuffer *)sound.audioBuffer);
-#else
-    alSourcePause(sound.source);
-#endif
 }
 
 // Resume a paused sound
 void ResumeSound(Sound sound)
 {
-#if USE_MINI_AL
     ResumeAudioBuffer((AudioBuffer *)sound.audioBuffer);
-#else
-    ALenum state;
-
-    alGetSourcei(sound.source, AL_SOURCE_STATE, &state);
-
-    if (state == AL_PAUSED) alSourcePlay(sound.source);
-#endif
 }
 
 // Stop reproducing a sound
 void StopSound(Sound sound)
 {
-#if USE_MINI_AL
     StopAudioBuffer((AudioBuffer *)sound.audioBuffer);
-#else
-    alSourceStop(sound.source);
-#endif
 }
 
 // Check if a sound is playing
 bool IsSoundPlaying(Sound sound)
 {
-#if USE_MINI_AL
     return IsAudioBufferPlaying((AudioBuffer *)sound.audioBuffer);
-#else
-    bool playing = false;
-    ALint state;
-
-    alGetSourcei(sound.source, AL_SOURCE_STATE, &state);
-    if (state == AL_PLAYING) playing = true;
-
-    return playing;
-#endif
 }
 
 // Set volume for a sound
 void SetSoundVolume(Sound sound, float volume)
 {
-#if USE_MINI_AL
     SetAudioBufferVolume((AudioBuffer *)sound.audioBuffer, volume);
-#else
-    alSourcef(sound.source, AL_GAIN, volume);
-#endif
 }
 
 // Set pitch for a sound
 void SetSoundPitch(Sound sound, float pitch)
 {
-#if USE_MINI_AL
     SetAudioBufferPitch((AudioBuffer *)sound.audioBuffer, pitch);
-#else
-    alSourcef(sound.source, AL_PITCH, pitch);
-#endif
 }
 
 // Convert wave data to desired format
 void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
 {
-#if USE_MINI_AL
     mal_format formatIn  = ((wave->sampleSize == 8) ? mal_format_u8 : ((wave->sampleSize == 16) ? mal_format_s16 : mal_format_f32));
     mal_format formatOut = ((      sampleSize == 8) ? mal_format_u8 : ((      sampleSize == 16) ? mal_format_s16 : mal_format_f32));
 
@@ -1264,87 +1028,6 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
     wave->channels = channels;
     free(wave->data);
     wave->data = data;
-
-#else
-    // Format sample rate
-    // NOTE: Only supported 22050 <--> 44100
-    if (wave->sampleRate != sampleRate)
-    {
-        // TODO: Resample wave data (upsampling or downsampling)
-        // NOTE 1: To downsample, you have to drop samples or average them.
-        // NOTE 2: To upsample, you have to interpolate new samples.
-
-        wave->sampleRate = sampleRate;
-    }
-
-    // Format sample size
-    // NOTE: Only supported 8 bit <--> 16 bit <--> 32 bit
-    if (wave->sampleSize != sampleSize)
-    {
-        void *data = malloc(wave->sampleCount*wave->channels*sampleSize/8);
-
-        for (int i = 0; i < wave->sampleCount; i++)
-        {
-            for (int j = 0; j < wave->channels; j++)
-            {
-                if (sampleSize == 8)
-                {
-                    if (wave->sampleSize == 16) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float)(((short *)wave->data)[wave->channels*i + j])/32767.0f)*256);
-                    else if (wave->sampleSize == 32) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float *)wave->data)[wave->channels*i + j]*127.0f + 127);
-                }
-                else if (sampleSize == 16)
-                {
-                    if (wave->sampleSize == 8) ((short *)data)[wave->channels*i + j] = (short)(((float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f)*32767);
-                    else if (wave->sampleSize == 32) ((short *)data)[wave->channels*i + j] = (short)((((float *)wave->data)[wave->channels*i + j])*32767);
-                }
-                else if (sampleSize == 32)
-                {
-                    if (wave->sampleSize == 8) ((float *)data)[wave->channels*i + j] = (float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f;
-                    else if (wave->sampleSize == 16) ((float *)data)[wave->channels*i + j] = (float)(((short *)wave->data)[wave->channels*i + j])/32767.0f;
-                }
-            }
-        }
-
-        wave->sampleSize = sampleSize;
-        free(wave->data);
-        wave->data = data;
-    }
-
-    // Format channels (interlaced mode)
-    // NOTE: Only supported mono <--> stereo
-    if (wave->channels != channels)
-    {
-        void *data = malloc(wave->sampleCount*wave->sampleSize/8*channels);
-
-        if ((wave->channels == 1) && (channels == 2))       // mono ---> stereo (duplicate mono information)
-        {
-            for (int i = 0; i < wave->sampleCount; i++)
-            {
-                for (int j = 0; j < channels; j++)
-                {
-                    if (wave->sampleSize == 8) ((unsigned char *)data)[channels*i + j] = ((unsigned char *)wave->data)[i];
-                    else if (wave->sampleSize == 16) ((short *)data)[channels*i + j] = ((short *)wave->data)[i];
-                    else if (wave->sampleSize == 32) ((float *)data)[channels*i + j] = ((float *)wave->data)[i];
-                }
-            }
-        }
-        else if ((wave->channels == 2) && (channels == 1))  // stereo ---> mono (mix stereo channels)
-        {
-            for (int i = 0, j = 0; i < wave->sampleCount; i++, j += 2)
-            {
-                if (wave->sampleSize == 8) ((unsigned char *)data)[i] = (((unsigned char *)wave->data)[j] + ((unsigned char *)wave->data)[j + 1])/2;
-                else if (wave->sampleSize == 16) ((short *)data)[i] = (((short *)wave->data)[j] + ((short *)wave->data)[j + 1])/2;
-                else if (wave->sampleSize == 32) ((float *)data)[i] = (((float *)wave->data)[j] + ((float *)wave->data)[j + 1])/2.0f;
-            }
-        }
-
-        // TODO: Add/remove additional interlaced channels
-
-        wave->channels = channels;
-        free(wave->data);
-        wave->data = data;
-    }
-#endif
 }
 
 // Copy a wave to a new wave
@@ -1578,8 +1261,8 @@ void UnloadMusicStream(Music music)
 // Start music playing (open stream)
 void PlayMusicStream(Music music)
 {
-#if USE_MINI_AL
     AudioBuffer *audioBuffer = (AudioBuffer *)music->stream.audioBuffer;
+    
     if (audioBuffer == NULL)
     {
         TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer");
@@ -1595,61 +1278,25 @@ void PlayMusicStream(Music music)
     PlayAudioStream(music->stream); // <-- This resets the cursor position.
 
     audioBuffer->frameCursorPos = frameCursorPos;
-#else
-    alSourcePlay(music->stream.source);
-#endif
 }
 
 // Pause music playing
 void PauseMusicStream(Music music)
 {
-#if USE_MINI_AL
     PauseAudioStream(music->stream);
-#else
-    alSourcePause(music->stream.source);
-#endif
 }
 
 // Resume music playing
 void ResumeMusicStream(Music music)
 {
-#if USE_MINI_AL
     ResumeAudioStream(music->stream);
-#else
-    ALenum state;
-    alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state);
-
-    if (state == AL_PAUSED) 
-    {
-        TraceLog(LOG_INFO, "[AUD ID %i] Resume music stream playing", music->stream.source);
-        alSourcePlay(music->stream.source);
-    }
-#endif
 }
 
 // Stop music playing (close stream)
 // TODO: To clear a buffer, make sure they have been already processed!
 void StopMusicStream(Music music)
 {
-#if USE_MINI_AL
     StopAudioStream(music->stream);
-#else
-    alSourceStop(music->stream.source);
-    
-    /*
-    // Clear stream buffers
-    // WARNING: Queued buffers must have been processed before unqueueing and reloaded with data!!!
-    void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, 1);
-    
-    for (int i = 0; i < MAX_STREAM_BUFFERS; i++)
-    {
-        //UpdateAudioStream(music->stream, pcm, AUDIO_BUFFER_SIZE);       // Update one buffer at a time
-        alBufferData(music->stream.buffers[i], music->stream.format, pcm, AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, music->stream.sampleRate);
-    }
-
-    free(pcm);
-    */
-#endif
     
     // Restart music context
     switch (music->ctxType)
@@ -1677,7 +1324,6 @@ void StopMusicStream(Music music)
 // TODO: Make sure buffers are ready for update... check music state
 void UpdateMusicStream(Music music)
 {
-#if USE_MINI_AL
     bool streamEnding = false;
 
     unsigned int subBufferSizeInFrames = ((AudioBuffer *)music->stream.audioBuffer)->bufferSizeInFrames/2;
@@ -1761,139 +1407,24 @@ void UpdateMusicStream(Music music)
         // just make sure to play again on window restore
         if (IsMusicPlaying(music)) PlayMusicStream(music);
     }
-#else
-    ALenum state;
-    ALint processed = 0;
-
-    alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state);          // Get music stream state
-    alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed); // Get processed buffers
-
-    if (processed > 0)
-    {
-        bool streamEnding = false;
-
-        // NOTE: Using dynamic allocation because it could require more than 16KB
-        void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.sampleSize/8*music->stream.channels, 1);
-
-        int numBuffersToProcess = processed;
-        int samplesCount = 0;    // Total size of data steamed in L+R samples for xm floats, 
-                                 // individual L or R for ogg shorts
-
-        for (int i = 0; i < numBuffersToProcess; i++)
-        {
-            if (music->samplesLeft >= AUDIO_BUFFER_SIZE) samplesCount = AUDIO_BUFFER_SIZE;
-            else samplesCount = music->samplesLeft;
-
-            // TODO: Really don't like ctxType thingy...
-            switch (music->ctxType)
-            {
-                case MUSIC_AUDIO_OGG:
-                {
-                    // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
-                    int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, samplesCount*music->stream.channels);
-
-                } break;
-            #if defined(SUPPORT_FILEFORMAT_FLAC)
-                case MUSIC_AUDIO_FLAC:
-                {
-                    // NOTE: Returns the number of samples to process
-                    unsigned int numSamplesFlac = (unsigned int)drflac_read_s16(music->ctxFlac, samplesCount*music->stream.channels, (short *)pcm);
-
-                } break;
-            #endif
-            #if defined(SUPPORT_FILEFORMAT_MP3)
-                case MUSIC_AUDIO_MP3:
-                {
-                    // NOTE: Returns the number of samples to process
-                    unsigned int numSamplesMp3 = (unsigned int)drmp3_read_f32(&music->ctxMp3, samplesCount*music->stream.channels, (float *)pcm);
-                } break;
-            #endif
-            #if defined(SUPPORT_FILEFORMAT_XM)
-                case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, samplesCount); break;
-            #endif
-            #if defined(SUPPORT_FILEFORMAT_MOD)
-                case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, samplesCount, 0); break;
-            #endif
-                default: break;
-            }
-
-            UpdateAudioStream(music->stream, pcm, samplesCount);
-            music->samplesLeft -= samplesCount;
-
-            if (music->samplesLeft <= 0)
-            {
-                streamEnding = true;
-                break;
-            }
-        }
-        
-        // Free allocated pcm data
-        free(pcm);
-
-        // Reset audio stream for looping
-        if (streamEnding)
-        {
-            StopMusicStream(music);        // Stop music (and reset)
-            
-            // Decrease loopCount to stop when required
-            if (music->loopCount > 0)
-            {
-                music->loopCount--;        // Decrease loop count
-                PlayMusicStream(music);    // Play again
-            }
-            else
-            {
-                if (music->loopCount == -1)
-                {
-                    PlayMusicStream(music);
-                }
-            }
-        }
-        else
-        {
-            // NOTE: In case window is minimized, music stream is stopped,
-            // just make sure to play again on window restore
-            if (state != AL_PLAYING) PlayMusicStream(music);
-        }
-    }
-#endif
 }
 
 // Check if any music is playing
 bool IsMusicPlaying(Music music)
 {
-#if USE_MINI_AL
     return IsAudioStreamPlaying(music->stream);
-#else
-    bool playing = false;
-    ALint state;
-
-    alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state);
-
-    if (state == AL_PLAYING) playing = true;
-
-    return playing;
-#endif
 }
 
 // Set volume for music
 void SetMusicVolume(Music music, float volume)
 {
-#if USE_MINI_AL
     SetAudioStreamVolume(music->stream, volume);
-#else
-    alSourcef(music->stream.source, AL_GAIN, volume);
-#endif
 }
 
 // Set pitch for music
 void SetMusicPitch(Music music, float pitch)
 {
-#if USE_MINI_AL
     SetAudioStreamPitch(music->stream, pitch);
-#else
-    alSourcef(music->stream.source, AL_PITCH, pitch);
-#endif
 }
 
 // Set music loop count (loop repeats)
@@ -1939,8 +1470,6 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
         stream.channels = 1;  // Fallback to mono channel
     }
 
-
-#if USE_MINI_AL
     mal_format formatIn = ((stream.sampleSize == 8) ? mal_format_u8 : ((stream.sampleSize == 16) ? mal_format_s16 : mal_format_f32));
 
     // The size of a streaming buffer must be at least double the size of a period.
@@ -1957,52 +1486,6 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
 
     audioBuffer->looping = true;        // Always loop for streaming buffers.
     stream.audioBuffer = audioBuffer;
-#else
-    // Setup OpenAL format
-    if (stream.channels == 1)
-    {
-        switch (sampleSize)
-        {
-            case 8: stream.format = AL_FORMAT_MONO8; break;
-            case 16: stream.format = AL_FORMAT_MONO16; break;
-            case 32: stream.format = AL_FORMAT_MONO_FLOAT32; break;     // Requires OpenAL extension: AL_EXT_FLOAT32
-            default: TraceLog(LOG_WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break;
-        }
-    }
-    else if (stream.channels == 2)
-    {
-        switch (sampleSize)
-        {
-            case 8: stream.format = AL_FORMAT_STEREO8; break;
-            case 16: stream.format = AL_FORMAT_STEREO16; break;
-            case 32: stream.format = AL_FORMAT_STEREO_FLOAT32; break;   // Requires OpenAL extension: AL_EXT_FLOAT32
-            default: TraceLog(LOG_WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break;
-        }
-    }
-
-    // Create an audio source
-    alGenSources(1, &stream.source);
-    alSourcef(stream.source, AL_PITCH, 1.0f);
-    alSourcef(stream.source, AL_GAIN, 1.0f);
-    alSource3f(stream.source, AL_POSITION, 0.0f, 0.0f, 0.0f);
-    alSource3f(stream.source, AL_VELOCITY, 0.0f, 0.0f, 0.0f);
-
-    // Create Buffers (double buffering)
-    alGenBuffers(MAX_STREAM_BUFFERS, stream.buffers);
-
-    // Initialize buffer with zeros by default
-    // NOTE: Using dynamic allocation because it requires more than 16KB
-    void *pcm = calloc(AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, 1);
-
-    for (int i = 0; i < MAX_STREAM_BUFFERS; i++)
-    {
-        alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, stream.sampleRate);
-    }
-
-    free(pcm);
-
-    alSourceQueueBuffers(stream.source, MAX_STREAM_BUFFERS, stream.buffers);
-#endif
 
     TraceLog(LOG_INFO, "[AUD ID %i] Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.source, stream.sampleRate, stream.sampleSize, (stream.channels == 1) ? "Mono" : "Stereo");
 
@@ -2012,28 +1495,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
 // Close audio stream and free memory
 void CloseAudioStream(AudioStream stream)
 {
-#if USE_MINI_AL
     DeleteAudioBuffer((AudioBuffer *)stream.audioBuffer);
-#else
-    // Stop playing channel
-    alSourceStop(stream.source);
-
-    // Flush out all queued buffers
-    int queued = 0;
-    alGetSourcei(stream.source, AL_BUFFERS_QUEUED, &queued);
-
-    ALuint buffer = 0;
-
-    while (queued > 0)
-    {
-        alSourceUnqueueBuffers(stream.source, 1, &buffer);
-        queued--;
-    }
-
-    // Delete source and buffers
-    alDeleteSources(1, &stream.source);
-    alDeleteBuffers(MAX_STREAM_BUFFERS, stream.buffers);
-#endif
     
     TraceLog(LOG_INFO, "[AUD ID %i] Unloaded audio stream data", stream.source);
 }
@@ -2043,7 +1505,6 @@ void CloseAudioStream(AudioStream stream)
 // NOTE 2: To unqueue a buffer it needs to be processed: IsAudioBufferProcessed()
 void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
 {
-#if USE_MINI_AL
     AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer;
     if (audioBuffer == NULL)
     {
@@ -2054,6 +1515,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
     if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
     {
         mal_uint32 subBufferToUpdate;
+        
         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.
@@ -2073,6 +1535,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
         if (subBufferSizeInFrames >= (mal_uint32)samplesCount)
         {
             mal_uint32 framesToWrite = subBufferSizeInFrames;
+            
             if (framesToWrite > (mal_uint32)samplesCount) framesToWrite = (mal_uint32)samplesCount;
 
             mal_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
@@ -2080,6 +1543,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
 
             // Any leftover frames should be filled with zeros.
             mal_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
+            
             if (leftoverFrameCount > 0) 
             {
                 memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
@@ -2098,24 +1562,11 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
         TraceLog(LOG_ERROR, "Audio buffer not available for updating");
         return;
     }
-#else
-    ALuint buffer = 0;
-    alSourceUnqueueBuffers(stream.source, 1, &buffer);
-
-    // Check if any buffer was available for unqueue
-    if (alGetError() != AL_INVALID_VALUE)
-    {
-        alBufferData(buffer, stream.format, data, samplesCount*stream.sampleSize/8*stream.channels, stream.sampleRate);
-        alSourceQueueBuffers(stream.source, 1, &buffer);
-    }
-    else TraceLog(LOG_WARNING, "[AUD ID %i] Audio buffer not available for unqueuing", stream.source);
-#endif
 }
 
 // Check if any audio stream buffers requires refill
 bool IsAudioBufferProcessed(AudioStream stream)
 {
-#if USE_MINI_AL
     AudioBuffer *audioBuffer = (AudioBuffer *)stream.audioBuffer;
     if (audioBuffer == NULL)
     {
@@ -2124,92 +1575,46 @@ bool IsAudioBufferProcessed(AudioStream stream)
     }
 
     return audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1];
-#else
-    ALint processed = 0;
-
-    // Determine if music stream is ready to be written
-    alGetSourcei(stream.source, AL_BUFFERS_PROCESSED, &processed);
-
-    return (processed > 0);
-#endif
 }
 
 // Play audio stream
 void PlayAudioStream(AudioStream stream)
 {
-#if USE_MINI_AL
     PlayAudioBuffer((AudioBuffer *)stream.audioBuffer);
-#else
-    alSourcePlay(stream.source);
-#endif
 }
 
 // Play audio stream
 void PauseAudioStream(AudioStream stream)
 {
-#if USE_MINI_AL
     PauseAudioBuffer((AudioBuffer *)stream.audioBuffer);
-#else
-    alSourcePause(stream.source);
-#endif
 }
 
 // Resume audio stream playing
 void ResumeAudioStream(AudioStream stream)
 {
-#if USE_MINI_AL
     ResumeAudioBuffer((AudioBuffer *)stream.audioBuffer);
-#else
-    ALenum state;
-    alGetSourcei(stream.source, AL_SOURCE_STATE, &state);
-
-    if (state == AL_PAUSED) alSourcePlay(stream.source);
-#endif
 }
 
 // Check if audio stream is playing.
 bool IsAudioStreamPlaying(AudioStream stream)
 {
-#if USE_MINI_AL
     return IsAudioBufferPlaying((AudioBuffer *)stream.audioBuffer);
-#else
-    bool playing = false;
-    ALint state;
-
-    alGetSourcei(stream.source, AL_SOURCE_STATE, &state);
-
-    if (state == AL_PLAYING) playing = true;
-
-    return playing;
-#endif
 }
 
 // Stop audio stream
 void StopAudioStream(AudioStream stream)
 {
-#if USE_MINI_AL
     StopAudioBuffer((AudioBuffer *)stream.audioBuffer);
-#else
-    alSourceStop(stream.source);
-#endif
 }
 
 void SetAudioStreamVolume(AudioStream stream, float volume)
 {
-#if USE_MINI_AL
     SetAudioBufferVolume((AudioBuffer *)stream.audioBuffer, volume);
-#else
-    alSourcef(stream.source, AL_GAIN, volume);
-#endif
 }
 
 void SetAudioStreamPitch(AudioStream stream, float pitch)
 {
-#if USE_MINI_AL
     SetAudioBufferPitch((AudioBuffer *)stream.audioBuffer, pitch);
-#else
-    alSourcef(stream.source, AL_PITCH, pitch);
-#endif
 }
 
 //----------------------------------------------------------------------------------
diff --git a/src/config.h.in b/src/config.h.in
index 742067ce..b0e62480 100644
--- a/src/config.h.in
+++ b/src/config.h.in
@@ -1,7 +1,5 @@
 /* config.h.in */
 
-#cmakedefine USE_OPENAL_BACKEND 1
-
 // core.c
 /* Camera module is included (camera.h) and multiple predefined cameras are available: free, 1st/3rd person, orbital */
 #cmakedefine SUPPORT_CAMERA_SYSTEM 1