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Clarifying the OpenAL loop 

because it isn't as clear as it can be
This commit is contained in:
John Peterson 2013-06-09 09:43:27 +02:00
parent 3ddd24872b
commit adb83cfabe
2 changed files with 108 additions and 113 deletions
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202
Source/Core/AudioCommon/Src/OpenALStream.cpp
View File

@ -97,7 +97,7 @@ void OpenALStream::SetVolume(int volume)
fVolume = (float)volume / 100.0f;
if (uiSource)
alSourcef(uiSource, AL_GAIN, fVolume);
alSourcef(uiSource, AL_GAIN, fVolume);
}
void OpenALStream::Update()
@ -124,6 +124,15 @@ void OpenALStream::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - openal");
bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
#if defined(__APPLE__)
bool float32_capable = false;
// OSX does not have the alext AL_FORMAT_51CHN32 yet.
surround_capable = false;
#else
bool float32_capable = true;
#endif
u32 ulFrequency = m_mixer->GetSampleRate();
numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers
@ -136,22 +145,20 @@ void OpenALStream::SoundLoop()
alGenSources(1, &uiSource);
// Short Silence
memset(sampleBuffer, 0, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * numBuffers);
memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_FLOAT);
memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * FRAME_STEREO_SHORT);
for (int i = 0; i < numBuffers; i++)
{
#if !defined(__APPLE__)
if (Core::g_CoreStartupParameter.bDPL2Decoder)
alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
if (surround_capable)
alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT, ulFrequency);
else
#endif
alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, ulFrequency);
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);
alSourcef(uiSource, AL_GAIN, fVolume);
// TODO: Error handling
//ALenum err = alGetError();
@ -170,14 +177,7 @@ void OpenALStream::SoundLoop()
soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
#if defined(__APPLE__)
bool float32_capable = false;
#else
bool float32_capable = true;
#endif
while (!threadData)
while (!threadData)
{
// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
const u32 stereo_16_bit_size = 4;
@ -193,12 +193,9 @@ void OpenALStream::SoundLoop()
numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
// Convert the samples from short to float
float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
for (u32 i = 0; i < numSamples; ++i)
{
dest[i * 2 + 0] = (float)realtimeBuffer[i * 2 + 0] / (1 << 16);
dest[i * 2 + 1] = (float)realtimeBuffer[i * 2 + 1] / (1 << 16);
}
float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
dest[i] = (float)realtimeBuffer[i] / (1 << 16);
soundTouch.putSamples(dest, numSamples);
@ -230,101 +227,94 @@ void OpenALStream::SoundLoop()
}
}
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * SIZE_FLOAT * OAL_MAX_BUFFERS);
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
if (nSamples > minSamples)
if (nSamples < minSamples)
continue;
// Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued Buffer)
if (iBuffersFilled == 0)
{
// 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 defined(__APPLE__)
// OSX does not have the alext AL_FORMAT_51CHN32 yet.
surround_capable = false;
#else
if (surround_capable)
{
float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
dpl2decode(sampleBuffer, nSamples, dpl2);
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
ALenum err = alGetError();
if (err == AL_INVALID_ENUM)
{
// 5.1 is not supported by the host, fallback to stereo
WARN_LOG(AUDIO, "Unable to set 5.1 surround mode. Updating OpenAL Soft might fix this issue.");
surround_capable = false;
}
else if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
}
}
#endif
if (!surround_capable)
{
#if !defined(__APPLE__)
if (float32_capable)
{
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * 4 * 2, ulFrequency);
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);
}
}
#endif
if (!float32_capable)
{
// Convert the samples from float to short
short stereo[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples; ++i)
{
stereo[i * 2 + 0] = (short)((float)sampleBuffer[i * 2 + 0] * (1 << 16));
stereo[i * 2 + 1] = (short)((float)sampleBuffer[i * 2 + 1] * (1 << 16));
}
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo, nSamples * 2 * 2, ulFrequency);
}
}
alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
ALenum err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
}
iBuffersFilled++;
}
if (iBuffersFilled == numBuffers)
if (surround_capable)
{
float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
dpl2decode(sampleBuffer, nSamples, dpl2);
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
ALenum err = alGetError();
if (err == AL_INVALID_ENUM)
{
alSourcePlay(uiSource);
err = alGetError();
if (err != 0)
// 5.1 is not supported by the host, fallback to stereo
WARN_LOG(AUDIO, "Unable to set 5.1 surround mode. Updating OpenAL Soft might fix this issue.");
surround_capable = false;
}
else if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
}
}
else
{
if (float32_capable)
{
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer, nSamples * FRAME_STEREO_FLOAT, ulFrequency);
ALenum err = alGetError();
if (err == AL_INVALID_ENUM)
{
ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
float32_capable = false;
}
else if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
}
}
alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)
else
{
// Buffer underrun occurred, resume playback
alSourcePlay(uiSource);
err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
}
// 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 << 16));
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);
}
}
}

19
Source/Core/AudioCommon/Src/OpenALStream.h
View File

@ -30,11 +30,16 @@
#include <soundtouch/STTypes.h>
// 16 bit Stereo
#define SFX_MAX_SOURCE 1
#define OAL_MAX_BUFFERS 32
#define OAL_MAX_SAMPLES 256
#define SURROUND_CHANNELS 6 // number of channels in surround mode
#define SIZE_FLOAT 4 // size of a float in bytes
#define SFX_MAX_SOURCE 1
#define OAL_MAX_BUFFERS 32
#define OAL_MAX_SAMPLES 256
#define STEREO_CHANNELS 2
#define SURROUND_CHANNELS 6 // number of channels in surround mode
#define SIZE_SHORT 2
#define SIZE_FLOAT 4 // size of a float in bytes
#define FRAME_STEREO_SHORT STEREO_CHANNELS * SIZE_SHORT
#define FRAME_STEREO_FLOAT STEREO_CHANNELS * SIZE_FLOAT
#define FRAME_SURROUND_FLOAT SURROUND_CHANNELS * SIZE_FLOAT
#endif
class OpenALStream: public SoundStream
@ -61,8 +66,8 @@ private:
std::thread thread;
Common::Event soundSyncEvent;
short realtimeBuffer[OAL_MAX_SAMPLES * 2];
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
short realtimeBuffer[OAL_MAX_SAMPLES * STEREO_CHANNELS];
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
ALuint uiBuffers[OAL_MAX_BUFFERS];
ALuint uiSource;
ALfloat fVolume;