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OpenAL: Don't request samples if buffers are full 

Makes the buffering code a bit more explicit (circular buffer, but
blocks until individual buffers get unqueued by OpenAL), and fixes a
bug in the startup of Super Mario Sunshine:
https://bugs.dolphin-emu.org/issues/9811
This commit is contained in:
Michael Maltese 2016-10-01 20:55:40 -07:00
parent 4eeb158b58
commit 567dffc1ee
1 changed files with 31 additions and 73 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

104
Source/Core/AudioCommon/OpenALStream.cpp
View File

@ -159,44 +159,15 @@ void OpenALStream::SoundLoop()
// Generate a Source to playback the Buffers
alGenSources(1, &uiSource);
// Short Silence
if (float32_capable)
memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_FLOAT);
else
memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_SHORT);
memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * FRAME_STEREO_SHORT);
for (int i = 0; i < numBuffers; i++)
{
if (surround_capable)
{
if (float32_capable)
alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT,
ulFrequency);
else
alBufferData(uiBuffers[i], AL_FORMAT_51CHN16, sampleBuffer, 4 * FRAME_SURROUND_SHORT,
ulFrequency);
}
else
{
alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * FRAME_STEREO_SHORT,
ulFrequency);
}
}
alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
alSourcePlay(uiSource);
// Set the default sound volume as saved in the config file.
alSourcef(uiSource, AL_GAIN, fVolume);
// TODO: Error handling
// ALenum err = alGetError();
ALint iBuffersFilled = 0;
ALint iBuffersProcessed = 0;
unsigned int nextBuffer = 0;
unsigned int numBuffersQueued = 0;
ALint iState = 0;
ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};
soundTouch.setChannels(2);
soundTouch.setSampleRate(ulFrequency);
@ -209,6 +180,28 @@ void OpenALStream::SoundLoop()
while (m_run_thread.IsSet())
{
// Block until we have a free buffer
int numBuffersProcessed;
alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
if (numBuffers == numBuffersQueued && !numBuffersProcessed)
{
soundSyncEvent.Wait();
continue;
}
// Remove the Buffer from the Queue.
if (numBuffersProcessed)
{
ALuint unqueuedBufferIds[OAL_MAX_BUFFERS];
alSourceUnqueueBuffers(uiSource, numBuffersProcessed, unqueuedBufferIds);
ALenum err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
}
numBuffersQueued -= numBuffersProcessed;
}
// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
const u32 stereo_16_bit_size = 4;
const u32 dma_length = 32;
@ -232,14 +225,6 @@ void OpenALStream::SoundLoop()
soundTouch.putSamples(dest, numSamples);
if (iBuffersProcessed == iBuffersFilled)
{
alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
iBuffersFilled = 0;
}
if (iBuffersProcessed)
{
double rate = (double)m_mixer->GetCurrentSpeed();
if (rate <= 0)
{
@ -263,18 +248,6 @@ void OpenALStream::SoundLoop()
if (nSamples <= minSamples)
continue;
// Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued
// Buffer)
if (iBuffersFilled == 0)
{
alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
ALenum err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
}
}
if (surround_capable)
{
float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
@ -291,7 +264,7 @@ void OpenALStream::SoundLoop()
if (float32_capable)
{
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2,
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
}
else
@ -300,7 +273,7 @@ void OpenALStream::SoundLoop()
for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
surround_short[i] = (short)((float)dpl2[i] * (1 << 15));
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN16, surround_short,
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
nSamples * FRAME_SURROUND_SHORT, ulFrequency);
}
@ -322,7 +295,7 @@ void OpenALStream::SoundLoop()
{
if (float32_capable)
{
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
nSamples * FRAME_STEREO_FLOAT, ulFrequency);
ALenum err = alGetError();
if (err == AL_INVALID_ENUM)
@ -334,7 +307,6 @@ void OpenALStream::SoundLoop()
ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
}
}
else
{
// Convert the samples from float to short
@ -342,28 +314,19 @@ void OpenALStream::SoundLoop()
for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo,
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
nSamples * FRAME_STEREO_SHORT, ulFrequency);
}
}
alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
alSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]);
ALenum err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
}
iBuffersFilled++;
if (iBuffersFilled == numBuffers)
{
alSourcePlay(uiSource);
err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
}
}
numBuffersQueued++;
nextBuffer = (nextBuffer + 1) % numBuffers;
alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)
@ -376,11 +339,6 @@ void OpenALStream::SoundLoop()
ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
}
}
}
else
{
soundSyncEvent.Wait();
}
}
}