Update external audio libraries.

1 files changed, 11 insertions, 8 deletions

diff --git a/src/audio.c b/src/audio.c
index 3ee1fe81..c7104dbc 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -296,7 +296,7 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameC
     (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_sample_size_in_bytes(pDevice->format));
+    memset(pFramesOut, 0, frameCount*pDevice->channels*mal_get_bytes_per_sample(pDevice->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.
@@ -334,7 +334,7 @@ static mal_uint32 OnSendAudioDataToDevice(mal_device *pDevice, mal_uint32 frameC
                     // last few samples.
                     bool flushDSP = !audioBuffer->looping;
 
-                    mal_uint32 framesJustRead = mal_dsp_read_frames_ex(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, flushDSP);
+                    mal_uint32 framesJustRead = mal_dsp_read_ex(&audioBuffer->dsp, framesToReadRightNow, tempBuffer, flushDSP, audioBuffer->dsp.pUserData);
                     if (framesJustRead > 0) 
                     {
                         float *framesOut = (float *)pFramesOut + (framesRead*device.channels);
@@ -394,7 +394,7 @@ 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_sample_size_in_bytes(audioBuffer->dsp.config.formatIn)*audioBuffer->dsp.config.channelsIn;
+    mal_uint32 frameSizeInBytes = mal_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;
@@ -640,7 +640,7 @@ 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 *audioBuffer = (AudioBuffer *)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*mal_get_sample_size_in_bytes(format)), 1);
+    AudioBuffer *audioBuffer = (AudioBuffer *)calloc(sizeof(*audioBuffer) + (bufferSizeInFrames*channels*mal_get_bytes_per_sample(format)), 1);
     if (audioBuffer == NULL)
     {
         TraceLog(LOG_ERROR, "CreateAudioBuffer() : Failed to allocate memory for audio buffer");
@@ -656,7 +656,10 @@ AudioBuffer *CreateAudioBuffer(mal_format format, mal_uint32 channels, mal_uint3
     dspConfig.channelsOut = DEVICE_CHANNELS;
     dspConfig.sampleRateIn = sampleRate;
     dspConfig.sampleRateOut = DEVICE_SAMPLE_RATE;
-    mal_result resultMAL = mal_dsp_init(&dspConfig, OnAudioBufferDSPRead, audioBuffer, &audioBuffer->dsp);
+    dspConfig.onRead = OnAudioBufferDSPRead;
+    dspConfig.pUserData = audioBuffer;
+    dspConfig.allowDynamicSampleRate = MAL_TRUE;    // <-- Required for pitch shifting.
+    mal_result resultMAL = mal_dsp_init(&dspConfig, &audioBuffer->dsp);
     if (resultMAL != MAL_SUCCESS) 
     {
         TraceLog(LOG_ERROR, "LoadSoundFromWave() : Failed to create data conversion pipeline");
@@ -791,7 +794,7 @@ 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.config.sampleRateOut / (float)audioBuffer->dsp.config.sampleRateIn) / pitch) * audioBuffer->dsp.config.sampleRateIn);
+    mal_uint32 newOutputSampleRate = (mal_uint32)((((float)audioBuffer->dsp.src.config.sampleRateOut / (float)audioBuffer->dsp.src.config.sampleRateIn) / pitch) * audioBuffer->dsp.src.config.sampleRateIn);
     mal_dsp_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate);
 }
 
@@ -913,7 +916,7 @@ Sound LoadSoundFromWave(Wave wave)
         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_convert_frames(audioBuffer->buffer, audioBuffer->dsp.config.formatIn, audioBuffer->dsp.config.channelsIn, audioBuffer->dsp.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn);
+        frameCount = 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);
         if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");
 
         sound.audioBuffer = audioBuffer;
@@ -1015,7 +1018,7 @@ 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.config.channelsIn*mal_get_sample_size_in_bytes(audioBuffer->dsp.config.formatIn));
+    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);