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
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4eeb158b58
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567dffc1ee
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@ -159,44 +159,15 @@ void OpenALStream::SoundLoop()
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// Generate a Source to playback the Buffers
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// Generate a Source to playback the Buffers
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alGenSources(1, &uiSource);
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alGenSources(1, &uiSource);
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// Short Silence
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if (float32_capable)
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memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_FLOAT);
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else
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memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_SHORT);
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memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * FRAME_STEREO_SHORT);
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for (int i = 0; i < numBuffers; i++)
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{
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if (surround_capable)
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{
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if (float32_capable)
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alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT,
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ulFrequency);
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else
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alBufferData(uiBuffers[i], AL_FORMAT_51CHN16, sampleBuffer, 4 * FRAME_SURROUND_SHORT,
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ulFrequency);
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}
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else
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{
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alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * FRAME_STEREO_SHORT,
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ulFrequency);
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}
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}
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alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
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alSourcePlay(uiSource);
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// Set the default sound volume as saved in the config file.
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// Set the default sound volume as saved in the config file.
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alSourcef(uiSource, AL_GAIN, fVolume);
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alSourcef(uiSource, AL_GAIN, fVolume);
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// TODO: Error handling
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// TODO: Error handling
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// ALenum err = alGetError();
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// ALenum err = alGetError();
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ALint iBuffersFilled = 0;
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unsigned int nextBuffer = 0;
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ALint iBuffersProcessed = 0;
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unsigned int numBuffersQueued = 0;
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ALint iState = 0;
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ALint iState = 0;
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ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};
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soundTouch.setChannels(2);
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soundTouch.setChannels(2);
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soundTouch.setSampleRate(ulFrequency);
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soundTouch.setSampleRate(ulFrequency);
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@ -209,6 +180,28 @@ void OpenALStream::SoundLoop()
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while (m_run_thread.IsSet())
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while (m_run_thread.IsSet())
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{
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{
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// Block until we have a free buffer
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int numBuffersProcessed;
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alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
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if (numBuffers == numBuffersQueued && !numBuffersProcessed)
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{
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soundSyncEvent.Wait();
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continue;
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}
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// Remove the Buffer from the Queue.
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if (numBuffersProcessed)
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{
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ALuint unqueuedBufferIds[OAL_MAX_BUFFERS];
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alSourceUnqueueBuffers(uiSource, numBuffersProcessed, unqueuedBufferIds);
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ALenum err = alGetError();
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if (err != 0)
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{
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ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
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}
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numBuffersQueued -= numBuffersProcessed;
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}
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// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
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// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
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const u32 stereo_16_bit_size = 4;
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const u32 stereo_16_bit_size = 4;
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const u32 dma_length = 32;
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const u32 dma_length = 32;
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@ -232,14 +225,6 @@ void OpenALStream::SoundLoop()
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soundTouch.putSamples(dest, numSamples);
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soundTouch.putSamples(dest, numSamples);
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if (iBuffersProcessed == iBuffersFilled)
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{
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alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
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iBuffersFilled = 0;
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}
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if (iBuffersProcessed)
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{
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double rate = (double)m_mixer->GetCurrentSpeed();
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double rate = (double)m_mixer->GetCurrentSpeed();
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if (rate <= 0)
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if (rate <= 0)
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{
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{
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@ -263,18 +248,6 @@ void OpenALStream::SoundLoop()
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if (nSamples <= minSamples)
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if (nSamples <= minSamples)
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continue;
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continue;
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// Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued
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// Buffer)
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if (iBuffersFilled == 0)
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{
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alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
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ALenum err = alGetError();
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if (err != 0)
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{
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ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
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}
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}
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if (surround_capable)
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if (surround_capable)
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{
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{
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float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
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float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
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@ -291,7 +264,7 @@ void OpenALStream::SoundLoop()
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if (float32_capable)
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if (float32_capable)
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{
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{
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alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2,
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alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
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nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
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nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
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}
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}
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else
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else
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@ -300,7 +273,7 @@ void OpenALStream::SoundLoop()
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for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
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for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
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surround_short[i] = (short)((float)dpl2[i] * (1 << 15));
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surround_short[i] = (short)((float)dpl2[i] * (1 << 15));
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alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN16, surround_short,
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alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
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nSamples * FRAME_SURROUND_SHORT, ulFrequency);
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nSamples * FRAME_SURROUND_SHORT, ulFrequency);
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}
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}
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@ -322,7 +295,7 @@ void OpenALStream::SoundLoop()
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{
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{
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if (float32_capable)
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if (float32_capable)
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{
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{
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alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
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alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
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nSamples * FRAME_STEREO_FLOAT, ulFrequency);
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nSamples * FRAME_STEREO_FLOAT, ulFrequency);
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ALenum err = alGetError();
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ALenum err = alGetError();
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if (err == AL_INVALID_ENUM)
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if (err == AL_INVALID_ENUM)
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@ -334,7 +307,6 @@ void OpenALStream::SoundLoop()
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ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
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ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
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}
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}
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}
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}
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else
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else
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{
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{
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// Convert the samples from float to short
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// Convert the samples from float to short
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@ -342,28 +314,19 @@ void OpenALStream::SoundLoop()
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for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
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for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
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stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
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stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
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alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo,
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alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
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nSamples * FRAME_STEREO_SHORT, ulFrequency);
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nSamples * FRAME_STEREO_SHORT, ulFrequency);
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}
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}
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}
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}
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alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
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alSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]);
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ALenum err = alGetError();
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ALenum err = alGetError();
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if (err != 0)
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if (err != 0)
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{
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{
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ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
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ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
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}
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}
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iBuffersFilled++;
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numBuffersQueued++;
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nextBuffer = (nextBuffer + 1) % numBuffers;
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if (iBuffersFilled == numBuffers)
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{
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alSourcePlay(uiSource);
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err = alGetError();
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if (err != 0)
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{
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ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
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}
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}
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alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
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alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
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if (iState != AL_PLAYING)
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if (iState != AL_PLAYING)
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@ -376,11 +339,6 @@ void OpenALStream::SoundLoop()
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ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
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ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
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}
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}
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}
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}
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}
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else
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{
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soundSyncEvent.Wait();
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}
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}
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}
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}
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}
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