Updated audio library: mini_al -> miniaudio

projects/VS2015/raylib/raylib.vcxproj

@@ -246,7 +246,7 @@
     <ClInclude Include="..\..\..\src\external\glad.h" />
     <ClInclude Include="..\..\..\src\external\jar_mod.h" />
     <ClInclude Include="..\..\..\src\external\jar_xm.h" />
-    <ClInclude Include="..\..\..\src\external\mini_al.h" />
+    <ClInclude Include="..\..\..\src\external\miniaudio.h" />
     <ClInclude Include="..\..\..\src\external\stb_image.h" />
     <ClInclude Include="..\..\..\src\external\stb_image_resize.h" />
     <ClInclude Include="..\..\..\src\external\stb_image_write.h" />

projects/VS2017/raylib/raylib.vcxproj

@@ -173,7 +173,7 @@
     <ClInclude Include="..\..\..\src\external\glad.h" />
     <ClInclude Include="..\..\..\src\external\jar_mod.h" />
     <ClInclude Include="..\..\..\src\external\jar_xm.h" />
-    <ClInclude Include="..\..\..\src\external\mini_al.h" />
+    <ClInclude Include="..\..\..\src\external\miniaudio.h" />
     <ClInclude Include="..\..\..\src\external\stb_image.h" />
     <ClInclude Include="..\..\..\src\external\stb_image_resize.h" />
     <ClInclude Include="..\..\..\src\external\stb_image_write.h" />

src/raudio.c

@@ -1,6 +1,6 @@
 /**********************************************************************************************
 *
-*   raudio - A simple and easy-to-use audio library based on mini_al
+*   raudio - A simple and easy-to-use audio library based on miniaudio
 *
 *   FEATURES:
 *       - Manage audio device (init/close)
@@ -26,7 +26,7 @@
 *       supported by default, to remove support, just comment unrequired #define in this module
 *
 *   DEPENDENCIES:
-*       mini_al.h    - Audio device management lib (https://github.com/dr-soft/mini_al)
+*       miniaudio.h  - Audio device management lib (https://github.com/dr-soft/miniaudio)
 *       stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
 *       dr_mp3.h     - MP3 audio file loading (https://github.com/mackron/dr_libs)
 *       dr_flac.h    - FLAC audio file loading (https://github.com/mackron/dr_libs)
@@ -35,7 +35,7 @@
 *
 *   CONTRIBUTORS:
 *       David Reid (github: @mackron) (Nov. 2017):
-*           - Complete port to mini_al library
+*           - Complete port to miniaudio library
 *
 *       Joshua Reisenauer (github: @kd7tck) (2015)
 *           - XM audio module support (jar_xm)
@@ -77,11 +77,9 @@
     #include "utils.h"          // Required for: fopen() Android mapping
 #endif
 
-#define MAL_NO_SDL
+#define MA_NO_JACK
-#define MAL_NO_JACK
+#define MINIAUDIO_IMPLEMENTATION
-#define MAL_NO_OPENAL
+#include "external/miniaudio.h" // miniaudio library
-#define MINI_AL_IMPLEMENTATION
-#include "external/mini_al.h"   // mini_al audio library
 #undef PlaySound                // Win32 API: windows.h > mmsystem.h defines PlaySound macro
 
 #include <stdlib.h>             // Required for: malloc(), free()
@@ -208,9 +206,9 @@ void TraceLog(int msgType, const char *text, ...);              // Show trace lo
 #endif
 
 //----------------------------------------------------------------------------------
-// mini_al AudioBuffer Functionality
+// miniaudio AudioBuffer Functionality
 //----------------------------------------------------------------------------------
-#define DEVICE_FORMAT       mal_format_f32
+#define DEVICE_FORMAT       ma_format_f32
 #define DEVICE_CHANNELS     2
 #define DEVICE_SAMPLE_RATE  44100
 
@@ -220,7 +218,7 @@ typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioB
 // NOTE: Slightly different logic is used when feeding data to the playback device depending on whether or not data is streamed
 typedef struct rAudioBuffer rAudioBuffer;
 struct rAudioBuffer {
-    mal_dsp dsp;                    // Required for format conversion
+    ma_pcm_converter dsp;                    // Required for format conversion
     float volume;
     float pitch;
     bool playing;
@@ -239,26 +237,26 @@ struct rAudioBuffer {
 // NOTE: This system should probably be redesigned
 #define AudioBuffer rAudioBuffer
 
-// mini_al global variables
+// miniaudio global variables
-static mal_context context;
+static ma_context context;
-static mal_device device;
+static ma_device device;
-static mal_mutex audioLock;
+static ma_mutex audioLock;
-static bool isAudioInitialized = MAL_FALSE;
+static bool isAudioInitialized = MA_FALSE;
 static float masterVolume = 1.0f;
 
 // Audio buffers are tracked in a linked list
 static AudioBuffer *firstAudioBuffer = NULL;
 static AudioBuffer *lastAudioBuffer = NULL;
 
-// mini_al functions declaration
+// miniaudio functions declaration
-static void OnLog(mal_context *pContext, mal_device *pDevice, const char *message);
+static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message);
-static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameCount, void *pFramesOut);
+static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
-static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, void *pFramesOut, void *pUserData);
+static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 frameCount, void *pUserData);
-static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 frameCount, float localVolume);
+static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);
 
 // AudioBuffer management functions declaration
 // NOTE: Those functions are not exposed by raylib... for the moment
-AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint32 sampleRate, mal_uint32 bufferSizeInFrames, AudioBufferUsage usage);
+AudioBuffer *CreateAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, AudioBufferUsage usage);
 void DeleteAudioBuffer(AudioBuffer *audioBuffer);
 bool IsAudioBufferPlaying(AudioBuffer *audioBuffer);
 void PlayAudioBuffer(AudioBuffer *audioBuffer);
@@ -270,35 +268,34 @@ void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch);
 void TrackAudioBuffer(AudioBuffer *audioBuffer);
 void UntrackAudioBuffer(AudioBuffer *audioBuffer);
 
-
 // Log callback function
-static void OnLog(mal_context *pContext, mal_device *pDevice, const char *message)
+static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message)
 {
     (void)pContext;
     (void)pDevice;
 
-    TraceLog(LOG_ERROR, message);   // All log messages from mini_al are errors
+    TraceLog(LOG_ERROR, message);   // All log messages from miniaudio are errors
 }
 
 // Sending audio data to device callback function
-static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameCount, void *pFramesOut)
+static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount)
 {
     // This is where all of the mixing takes place.
     (void)pDevice;
 
     // Mixing is basically just an accumulation. We need to initialize the output buffer to 0.
-    memset(pFramesOut, 0, frameCount*pDevice->channels*mal_get_bytes_per_sample(pDevice->format));
+    memset(pFramesOut, 0, frameCount*pDevice->playback.channels*ma_get_bytes_per_sample(pDevice->playback.format));
 
     // Using a mutex here for thread-safety which makes things not real-time. This is unlikely to be necessary for this project, but may
     // want to consider how you might want to avoid this.
-    mal_mutex_lock(&audioLock);
+    ma_mutex_lock(&audioLock);
     {
         for (AudioBuffer *audioBuffer = firstAudioBuffer; audioBuffer != NULL; audioBuffer = audioBuffer->next)
         {
             // Ignore stopped or paused sounds.
             if (!audioBuffer->playing || audioBuffer->paused) continue;
 
-            mal_uint32 framesRead = 0;
+            ma_uint32 framesRead = 0;
             for (;;)
             {
                 if (framesRead > frameCount)
@@ -310,21 +307,21 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameC
                 if (framesRead == frameCount) break;
 
                 // Just read as much data as we can from the stream.
-                mal_uint32 framesToRead = (frameCount - framesRead);
+                ma_uint32 framesToRead = (frameCount - framesRead);
                 while (framesToRead > 0)
                 {
                     float tempBuffer[1024]; // 512 frames for stereo.
 
-                    mal_uint32 framesToReadRightNow = framesToRead;
+                    ma_uint32 framesToReadRightNow = framesToRead;
                     if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS)
                     {
                         framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS;
                     }
 
-                    mal_uint32 framesJustRead = (mal_uint32)mal_dsp_read(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, audioBuffer->dsp.pUserData);
+                    ma_uint32 framesJustRead = (ma_uint32)ma_pcm_converter_read(&audioBuffer->dsp, tempBuffer, framesToReadRightNow);
                     if (framesJustRead > 0)
                     {
-                        float *framesOut = (float *)pFramesOut + (framesRead*device.channels);
+                        float *framesOut = (float *)pFramesOut + (framesRead*device.playback.channels);
                         float *framesIn  = tempBuffer;
                         MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
 
@@ -357,18 +354,16 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameC
         }
     }
 
-    mal_mutex_unlock(&audioLock);
+    ma_mutex_unlock(&audioLock);
-
-    return frameCount;  // We always output the same number of frames that were originally requested.
 }
 
 // DSP read from audio buffer callback function
-static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, void *pFramesOut, void *pUserData)
+static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 frameCount, void *pUserData)
 {
     AudioBuffer *audioBuffer = (AudioBuffer *)pUserData;
 
-    mal_uint32 subBufferSizeInFrames = (audioBuffer->bufferSizeInFrames > 1)? audioBuffer->bufferSizeInFrames/2 : audioBuffer->bufferSizeInFrames;
+    ma_uint32 subBufferSizeInFrames = (audioBuffer->bufferSizeInFrames > 1)? audioBuffer->bufferSizeInFrames/2 : audioBuffer->bufferSizeInFrames;
-    mal_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
+    ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
 
     if (currentSubBufferIndex > 1)
     {
@@ -381,10 +376,10 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
     isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
     isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];
 
-    mal_uint32 frameSizeInBytes = mal_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)*audioBuffer->dsp.formatConverterIn.config.channels;
+    ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)*audioBuffer->dsp.formatConverterIn.config.channels;
 
     // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0.
-    mal_uint32 framesRead = 0;
+    ma_uint32 framesRead = 0;
     for (;;)
     {
         // We break from this loop differently depending on the buffer's usage. For static buffers, we simply fill as much data as we can. For
@@ -398,21 +393,21 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
             if (isSubBufferProcessed[currentSubBufferIndex]) break;
         }
 
-        mal_uint32 totalFramesRemaining = (frameCount - framesRead);
+        ma_uint32 totalFramesRemaining = (frameCount - framesRead);
         if (totalFramesRemaining == 0) break;
 
-        mal_uint32 framesRemainingInOutputBuffer;
+        ma_uint32 framesRemainingInOutputBuffer;
         if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
         {
             framesRemainingInOutputBuffer = audioBuffer->bufferSizeInFrames - audioBuffer->frameCursorPos;
         }
         else
         {
-            mal_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames * currentSubBufferIndex;
+            ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames * currentSubBufferIndex;
             framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
         }
 
-        mal_uint32 framesToRead = totalFramesRemaining;
+        ma_uint32 framesToRead = totalFramesRemaining;
         if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer;
 
         memcpy((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->buffer + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
@@ -437,7 +432,7 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
     }
 
     // Zero-fill excess.
-    mal_uint32 totalFramesRemaining = (frameCount - framesRead);
+    ma_uint32 totalFramesRemaining = (frameCount - framesRead);
     if (totalFramesRemaining > 0)
     {
         memset((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes);
@@ -453,14 +448,14 @@ static mal_uint32 OnAudioBufferDSPRead(mal_dsp *pDSP, mal_uint32 frameCount, voi
 
 // 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, mal_uint32 frameCount, float localVolume)
+static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume)
 {
-    for (mal_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
+    for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
     {
-        for (mal_uint32 iChannel = 0; iChannel < device.channels; ++iChannel)
+        for (ma_uint32 iChannel = 0; iChannel < device.playback.channels; ++iChannel)
         {
-                  float *frameOut = framesOut + (iFrame*device.channels);
+                  float *frameOut = framesOut + (iFrame*device.playback.channels);
-            const float *frameIn  = framesIn  + (iFrame*device.channels);
+            const float *frameIn  = framesIn  + (iFrame*device.playback.channels);
 
             frameOut[iChannel] += frameIn[iChannel]*masterVolume*localVolume;
         }
@@ -474,54 +469,64 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, mal_uint32 f
 void InitAudioDevice(void)
 {
     // Context.
-    mal_context_config contextConfig = mal_context_config_init(OnLog);
+    ma_context_config contextConfig = ma_context_config_init();
-    mal_result result = mal_context_init(NULL, 0, &contextConfig, &context);
+    contextConfig.logCallback = OnLog;
-    if (result != MAL_SUCCESS)
+    ma_result result = ma_context_init(NULL, 0, &contextConfig, &context);
+    if (result != MA_SUCCESS)
     {
         TraceLog(LOG_ERROR, "Failed to initialize audio context");
         return;
     }
 
     // Device. Using the default device. Format is floating point because it simplifies mixing.
-    mal_device_config deviceConfig = mal_device_config_init(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, OnSendAudioDataToDevice);
+    ma_device_config config = ma_device_config_init(ma_device_type_playback);
+    config.playback.pDeviceID = NULL;  // NULL for the default playback device.
+    config.playback.format    = DEVICE_FORMAT;
+    config.playback.channels  = DEVICE_CHANNELS;
+    config.capture.pDeviceID  = NULL;  // NULL for the default capture device.
+    config.capture.format     = ma_format_s16;
+    config.capture.channels   = 1;
+    config.sampleRate         = DEVICE_SAMPLE_RATE;
+    config.dataCallback       = OnSendAudioDataToDevice;
+    config.pUserData          = NULL;
 
-    result = mal_device_init(&context, mal_device_type_playback, NULL, &deviceConfig, NULL, &device);
+    result = ma_device_init(&context, &config, &device);
-    if (result != MAL_SUCCESS)
+    if (result != MA_SUCCESS)
     {
         TraceLog(LOG_ERROR, "Failed to initialize audio playback device");
-        mal_context_uninit(&context);
+        ma_context_uninit(&context);
         return;
     }
 
     // Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running
     // while there's at least one sound being played.
-    result = mal_device_start(&device);
+    result = ma_device_start(&device);
-    if (result != MAL_SUCCESS)
+    if (result != MA_SUCCESS)
     {
         TraceLog(LOG_ERROR, "Failed to start audio playback device");
-        mal_device_uninit(&device);
+        ma_device_uninit(&device);
-        mal_context_uninit(&context);
+        ma_context_uninit(&context);
         return;
     }
 
     // Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may
     // want to look at something a bit smarter later on to keep everything real-time, if that's necessary.
-    if (mal_mutex_init(&context, &audioLock) != MAL_SUCCESS)
+    if (ma_mutex_init(&context, &audioLock) != MA_SUCCESS)
     {
         TraceLog(LOG_ERROR, "Failed to create mutex for audio mixing");
-        mal_device_uninit(&device);
+        ma_device_uninit(&device);
-        mal_context_uninit(&context);
+        ma_context_uninit(&context);
         return;
     }
 
-    TraceLog(LOG_INFO, "Audio device initialized successfully: %s", device.name);
+    TraceLog(LOG_INFO, "Audio device initialized successfully");
-    TraceLog(LOG_INFO, "Audio backend: mini_al / %s", mal_get_backend_name(context.backend));
+    TraceLog(LOG_INFO, "Audio backend: miniaudio / %s", ma_get_backend_name(context.backend));
-    TraceLog(LOG_INFO, "Audio format: %s -> %s", mal_get_format_name(device.format), mal_get_format_name(device.internalFormat));
+    TraceLog(LOG_INFO, "Audio format: %s -> %s", ma_get_format_name(device.playback.format), ma_get_format_name(device.playback.internalFormat));
-    TraceLog(LOG_INFO, "Audio channels: %d -> %d", device.channels, device.internalChannels);
+    TraceLog(LOG_INFO, "Audio channels: %d -> %d", device.playback.channels, device.playback.internalChannels);
-    TraceLog(LOG_INFO, "Audio sample rate: %d -> %d", device.sampleRate, device.internalSampleRate);
+    TraceLog(LOG_INFO, "Audio sample rate: %d -> %d", device.sampleRate, device.playback.internalSampleRate);
-    TraceLog(LOG_INFO, "Audio buffer size: %d", device.bufferSizeInFrames);
+    TraceLog(LOG_INFO, "Audio buffer size: %d", device.playback.internalBufferSizeInFrames);
     
-    isAudioInitialized = MAL_TRUE;
+    isAudioInitialized = MA_TRUE;
 }
 
 // Close the audio device for all contexts
@@ -533,9 +538,9 @@ void CloseAudioDevice(void)
         return;
     }
 
-    mal_mutex_uninit(&audioLock);
+    ma_mutex_uninit(&audioLock);
-    mal_device_uninit(&device);
+    ma_device_uninit(&device);
-    mal_context_uninit(&context);
+    ma_context_uninit(&context);
 
     TraceLog(LOG_INFO, "Audio device closed successfully");
 }
@@ -560,9 +565,9 @@ void SetMasterVolume(float volume)
 //----------------------------------------------------------------------------------
 
 // 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)
+AudioBuffer *CreateAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, AudioBufferUsage usage)
 {
-    AudioBuffer *audioBuffer = (AudioBuffer *)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*mal_get_bytes_per_sample(format)), 1);
+    AudioBuffer *audioBuffer = (AudioBuffer *)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*ma_get_bytes_per_sample(format)), 1);
     if (audioBuffer == NULL)
     {
         TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to allocate memory for audio buffer");
@@ -570,7 +575,7 @@ AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint3
     }
 
     // We run audio data through a format converter.
-    mal_dsp_config dspConfig;
+    ma_pcm_converter_config dspConfig;
     memset(&dspConfig, 0, sizeof(dspConfig));
     dspConfig.formatIn = format;
     dspConfig.formatOut = DEVICE_FORMAT;
@@ -580,9 +585,10 @@ AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint3
     dspConfig.sampleRateOut = DEVICE_SAMPLE_RATE;
     dspConfig.onRead = OnAudioBufferDSPRead;
     dspConfig.pUserData = audioBuffer;
-    dspConfig.allowDynamicSampleRate = MAL_TRUE;    // <-- Required for pitch shifting.
+    dspConfig.allowDynamicSampleRate = MA_TRUE;    // <-- Required for pitch shifting.
-    mal_result resultMAL = mal_dsp_init(&dspConfig, &audioBuffer->dsp);
+    ma_result result = ma_pcm_converter_init(&dspConfig, &audioBuffer->dsp);
-    if (resultMAL != MAL_SUCCESS)
+    
+    if (result != MA_SUCCESS)
     {
         TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to create data conversion pipeline");
         free(audioBuffer);
@@ -716,16 +722,16 @@ void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch)
 
     // Pitching is just an adjustment of the sample rate. Note that this changes the duration of the sound - higher pitches
     // will make the sound faster; lower pitches make it slower.
-    mal_uint32 newOutputSampleRate = (mal_uint32)((float)audioBuffer->dsp.src.config.sampleRateOut / pitchMul);
+    ma_uint32 newOutputSampleRate = (ma_uint32)((float)audioBuffer->dsp.src.config.sampleRateOut / pitchMul);
     audioBuffer->pitch *= (float)audioBuffer->dsp.src.config.sampleRateOut / newOutputSampleRate;
 
-    mal_dsp_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate);
+    ma_pcm_converter_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate);
 }
 
 // Track audio buffer to linked list next position
 void TrackAudioBuffer(AudioBuffer *audioBuffer)
 {
-    mal_mutex_lock(&audioLock);
+    ma_mutex_lock(&audioLock);
 
     {
         if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer;
@@ -738,13 +744,13 @@ void TrackAudioBuffer(AudioBuffer *audioBuffer)
         lastAudioBuffer = audioBuffer;
     }
 
-    mal_mutex_unlock(&audioLock);
+    ma_mutex_unlock(&audioLock);
 }
 
 // Untrack audio buffer from linked list
 void UntrackAudioBuffer(AudioBuffer *audioBuffer)
 {
-    mal_mutex_lock(&audioLock);
+    ma_mutex_lock(&audioLock);
 
     {
         if (audioBuffer->prev == NULL) firstAudioBuffer = audioBuffer->next;
@@ -757,7 +763,7 @@ void UntrackAudioBuffer(AudioBuffer *audioBuffer)
         audioBuffer->next = NULL;
     }
 
-    mal_mutex_unlock(&audioLock);
+    ma_mutex_unlock(&audioLock);
 }
 
 //----------------------------------------------------------------------------------
@@ -828,7 +834,7 @@ Sound LoadSoundFromWave(Wave wave)
 
     if (wave.data != NULL)
     {
-        // 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
+        // When using miniaudio we need to do our own mixing. To simplify this we need convert the format of each sound to be consistent with
         // the format used to open the playback device. We can do this two ways:
         //
         //   1) Convert the whole sound in one go at load time (here).
@@ -836,16 +842,16 @@ Sound LoadSoundFromWave(Wave wave)
         //
         // I have decided on the first option because it offloads work required for the format conversion to the to the loading stage.
         // The downside to this is that it uses more memory if the original sound is u8 or s16.
-        mal_format formatIn  = ((wave.sampleSize == 8)? mal_format_u8 : ((wave.sampleSize == 16)? mal_format_s16 : mal_format_f32));
+        ma_format formatIn  = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
-        mal_uint32 frameCountIn = wave.sampleCount/wave.channels;
+        ma_uint32 frameCountIn = wave.sampleCount/wave.channels;
 
-        mal_uint32 frameCount = (mal_uint32)mal_convert_frames(NULL, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn);
+        ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn);
         if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to get frame count for format conversion");
 
         AudioBuffer* audioBuffer = CreateAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
         if (audioBuffer == NULL) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer");
 
-        frameCount = (mal_uint32)mal_convert_frames(audioBuffer->buffer, audioBuffer->dsp.formatConverterIn.config.formatIn, audioBuffer->dsp.formatConverterIn.config.channels, audioBuffer->dsp.src.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn);
+        frameCount = (ma_uint32)ma_convert_frames(audioBuffer->buffer, audioBuffer->dsp.formatConverterIn.config.formatIn, audioBuffer->dsp.formatConverterIn.config.channels, audioBuffer->dsp.src.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn);
         if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");
 
         sound.audioBuffer = audioBuffer;
@@ -885,7 +891,7 @@ void UpdateSound(Sound sound, const void *data, int samplesCount)
     StopAudioBuffer(audioBuffer);
 
     // TODO: May want to lock/unlock this since this data buffer is read at mixing time.
-    memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*mal_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn));
+    memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn));
 }
 
 // Export wave data to file
@@ -999,12 +1005,12 @@ void SetSoundPitch(Sound sound, float pitch)
 // Convert wave data to desired format
 void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
 {
-    mal_format formatIn  = ((wave->sampleSize == 8)? mal_format_u8 : ((wave->sampleSize == 16)? mal_format_s16 : mal_format_f32));
+    ma_format formatIn  = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32));
-    mal_format formatOut = ((      sampleSize == 8)? mal_format_u8 : ((      sampleSize == 16)? mal_format_s16 : mal_format_f32));
+    ma_format formatOut = ((      sampleSize == 8)? ma_format_u8 : ((      sampleSize == 16)? ma_format_s16 : ma_format_f32));
 
-    mal_uint32 frameCountIn = wave->sampleCount;  // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
+    ma_uint32 frameCountIn = wave->sampleCount;  // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
 
-    mal_uint32 frameCount = (mal_uint32)mal_convert_frames(NULL, formatOut, channels, sampleRate, NULL, formatIn, wave->channels, wave->sampleRate, frameCountIn);
+    ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, formatOut, channels, sampleRate, NULL, formatIn, wave->channels, wave->sampleRate, frameCountIn);
     if (frameCount == 0)
     {
         TraceLog(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion.");
@@ -1013,7 +1019,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
 
     void *data = malloc(frameCount*channels*(sampleSize/8));
 
-    frameCount = (mal_uint32)mal_convert_frames(data, formatOut, channels, sampleRate, wave->data, formatIn, wave->channels, wave->sampleRate, frameCountIn);
+    frameCount = (ma_uint32)ma_convert_frames(data, formatOut, channels, sampleRate, wave->data, formatIn, wave->channels, wave->sampleRate, frameCountIn);
     if (frameCount == 0)
     {
         TraceLog(LOG_ERROR, "WaveFormat() : Format conversion failed.");
@@ -1288,7 +1294,7 @@ void PlayMusicStream(Music music)
         //     // NOTE: In case window is minimized, music stream is stopped,
         //     // just make sure to play again on window restore
         //     if (IsMusicPlaying(music)) PlayMusicStream(music);
-        mal_uint32 frameCursorPos = audioBuffer->frameCursorPos;
+        ma_uint32 frameCursorPos = audioBuffer->frameCursorPos;
 
         PlayAudioStream(music->stream); // <-- This resets the cursor position.
 
@@ -1513,10 +1519,10 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
         stream.channels = 1;  // Fallback to mono channel
     }
 
-    mal_format formatIn = ((stream.sampleSize == 8)? mal_format_u8 : ((stream.sampleSize == 16)? mal_format_s16 : mal_format_f32));
+    ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32));
 
     // The size of a streaming buffer must be at least double the size of a period.
-    unsigned int periodSize = device.bufferSizeInFrames/device.periods;
+    unsigned int periodSize = device.playback.internalBufferSizeInFrames/device.playback.internalPeriods;
     unsigned int subBufferSize = AUDIO_BUFFER_SIZE;
     if (subBufferSize < periodSize) subBufferSize = periodSize;
 
@@ -1557,7 +1563,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
 
     if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
     {
-        mal_uint32 subBufferToUpdate;
+        ma_uint32 subBufferToUpdate;
 
         if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->isSubBufferProcessed[1])
         {
@@ -1571,21 +1577,21 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
             subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0])? 0 : 1;
         }
 
-        mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
+        ma_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
         unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
 
         // 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 >= (mal_uint32)samplesCount/stream.channels)
+        if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
         {
-            mal_uint32 framesToWrite = subBufferSizeInFrames;
+            ma_uint32 framesToWrite = subBufferSizeInFrames;
 
-            if (framesToWrite > ((mal_uint32)samplesCount/stream.channels)) framesToWrite = (mal_uint32)samplesCount/stream.channels;
+            if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
 
-            mal_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
+            ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
             memcpy(subBuffer, data, bytesToWrite);
 
             // Any leftover frames should be filled with zeros.
-            mal_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
+            ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
 
             if (leftoverFrameCount > 0)
             {

src/raudio.h

@@ -11,7 +11,7 @@
 *       - Manage raw audio context
 *
 *   DEPENDENCIES:
-*       mini_al.h    - Audio device management lib (https://github.com/dr-soft/mini_al)
+*       miniaudio.h  - Audio device management lib (https://github.com/dr-soft/miniaudio)
 *       stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
 *       dr_mp3.h     - MP3 audio file loading (https://github.com/mackron/dr_libs)
 *       dr_flac.h    - FLAC audio file loading (https://github.com/mackron/dr_libs)