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Merge pull request #4271 from ligfx/audiofix 

OpenAL: Don't request samples if buffers are full
This commit is contained in:
shuffle2 2016-10-02 21:00:10 -07:00 committed by GitHub
parent 25f983cb21 8fa79f3897
commit 7f4106646e
1 changed files with 109 additions and 151 deletions
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260
Source/Core/AudioCommon/OpenALStream.cpp
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@ -159,44 +159,15 @@ void OpenALStream::SoundLoop()
// Generate a Source to playback the Buffers // Generate a Source to playback the Buffers
alGenSources(1, &uiSource); 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. // Set the default sound volume as saved in the config file.
alSourcef(uiSource, AL_GAIN, fVolume); alSourcef(uiSource, AL_GAIN, fVolume);
// TODO: Error handling // TODO: Error handling
// ALenum err = alGetError(); // ALenum err = alGetError();
ALint iBuffersFilled = 0; unsigned int nextBuffer = 0;
ALint iBuffersProcessed = 0; unsigned int numBuffersQueued = 0;
ALint iState = 0; ALint iState = 0;
ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};
soundTouch.setChannels(2); soundTouch.setChannels(2);
soundTouch.setSampleRate(ulFrequency); soundTouch.setSampleRate(ulFrequency);
@ -209,6 +180,28 @@ void OpenALStream::SoundLoop()
while (m_run_thread.IsSet()) 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. // num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
const u32 stereo_16_bit_size = 4; const u32 stereo_16_bit_size = 4;
const u32 dma_length = 32; const u32 dma_length = 32;
@ -232,154 +225,119 @@ void OpenALStream::SoundLoop()
soundTouch.putSamples(dest, numSamples); soundTouch.putSamples(dest, numSamples);
if (iBuffersProcessed == iBuffersFilled) double rate = (double)m_mixer->GetCurrentSpeed();
if (rate <= 0)
{ {
alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed); Core::RequestRefreshInfo();
iBuffersFilled = 0; rate = (double)m_mixer->GetCurrentSpeed();
} }
if (iBuffersProcessed) // Place a lower limit of 10% speed. When a game boots up, there will be
// many silence samples. These do not need to be timestretched.
if (rate > 0.10)
{ {
double rate = (double)m_mixer->GetCurrentSpeed(); soundTouch.setTempo(rate);
if (rate <= 0) if (rate > 10)
{ {
Core::RequestRefreshInfo(); soundTouch.clear();
rate = (double)m_mixer->GetCurrentSpeed(); }
}
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
if (nSamples <= minSamples)
continue;
if (surround_capable)
{
float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
DPL2Decode(sampleBuffer, nSamples, dpl2);
// zero-out the subwoofer channel - DPL2Decode generates a pretty
// good 5.0 but not a good 5.1 output. Sadly there is not a 5.0
// AL_FORMAT_50CHN32 to make this super-explicit.
// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
for (u32 i = 0; i < nSamples; ++i)
{
dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
} }
// Place a lower limit of 10% speed. When a game boots up, there will be if (float32_capable)
// many silence samples. These do not need to be timestretched.
if (rate > 0.10)
{ {
soundTouch.setTempo(rate); alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
if (rate > 10) nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
{ }
soundTouch.clear(); else
} {
short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
surround_short[i] = (short)((float)dpl2[i] * (1 << 15));
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
nSamples * FRAME_SURROUND_SHORT, ulFrequency);
} }
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers); ALenum err = alGetError();
if (err == AL_INVALID_ENUM)
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); // 5.1 is not supported by the host, fallback to stereo
ALenum err = alGetError(); WARN_LOG(AUDIO,
if (err != 0) "Unable to set 5.1 surround mode. Updating OpenAL Soft might fix this issue.");
{ surround_capable = false;
ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
}
} }
else if (err != 0)
if (surround_capable)
{ {
float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS]; ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
DPL2Decode(sampleBuffer, nSamples, dpl2); }
}
// zero-out the subwoofer channel - DPL2Decode generates a pretty
// good 5.0 but not a good 5.1 output. Sadly there is not a 5.0
// AL_FORMAT_50CHN32 to make this super-explicit.
// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
for (u32 i = 0; i < nSamples; ++i)
{
dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
}
if (float32_capable)
{
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2,
nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
}
else
{
short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
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,
nSamples * FRAME_SURROUND_SHORT, ulFrequency);
}
else
{
if (float32_capable)
{
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
nSamples * FRAME_STEREO_FLOAT, ulFrequency);
ALenum err = alGetError(); ALenum err = alGetError();
if (err == AL_INVALID_ENUM) if (err == AL_INVALID_ENUM)
{ {
// 5.1 is not supported by the host, fallback to stereo float32_capable = false;
WARN_LOG(AUDIO,
"Unable to set 5.1 surround mode. Updating OpenAL Soft might fix this issue.");
surround_capable = false;
} }
else if (err != 0) else if (err != 0)
{ {
ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err); ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
} }
} }
else else
{ {
if (float32_capable) // Convert the samples from float to short
{ short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
nSamples * FRAME_STEREO_FLOAT, ulFrequency); stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
ALenum err = alGetError();
if (err == AL_INVALID_ENUM)
{
float32_capable = false;
}
else if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
}
}
else alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
{ nSamples * FRAME_STEREO_SHORT, ulFrequency);
// Convert the samples from float to short
short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo,
nSamples * FRAME_STEREO_SHORT, ulFrequency);
}
}
alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
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);
}
}
alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)
{
// Buffer underrun occurred, resume playback
alSourcePlay(uiSource);
err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
}
} }
} }
else
alSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]);
ALenum err = alGetError();
if (err != 0)
{ {
soundSyncEvent.Wait(); ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
}
numBuffersQueued++;
nextBuffer = (nextBuffer + 1) % numBuffers;
alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)
{
// Buffer underrun occurred, resume playback
alSourcePlay(uiSource);
err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
}
} }
} }
} }