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Update external audio libraries.

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This commit is contained in:
David Reid
2018-04-21 17:34:56 +10:00
parent f5ebbfb6bc
commit 0febaa2446
4 changed files with 12908 additions and 3889 deletions
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19
src/audio.c
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@ -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);