Added a latency setting to the audio settings.

Removed the Sample Rate setting.  It is now hardcoded to 48000hz (accurate audio timing).

Fixes issue 5672.
This commit is contained in:
skidau
2013-01-13 00:05:30 +11:00
parent 73140c7da7
commit 1c462a1eca
11 changed files with 57 additions and 39 deletions

View File

@ -46,6 +46,11 @@ bool OpenALStream::Start()
pContext = alcCreateContext(pDevice, NULL);
if (pContext)
{
// Used to determine an appropriate period size (2x period = total buffer size)
//ALCint refresh;
//alcGetIntegerv(pDevice, ALC_REFRESH, 1, &refresh);
//period_size_in_millisec = 1000 / refresh;
alcMakeContextCurrent(pContext);
thread = std::thread(std::mem_fun(&OpenALStream::SoundLoop), this);
bReturn = true;
@ -90,7 +95,7 @@ void OpenALStream::Stop()
// Clean up buffers and sources
alDeleteSources(1, &uiSource);
uiSource = 0;
alDeleteBuffers(OAL_NUM_BUFFERS, uiBuffers);
alDeleteBuffers(numBuffers, uiBuffers);
ALCcontext *pContext = alcGetCurrentContext();
ALCdevice *pDevice = alcGetContextsDevice(pContext);
@ -133,19 +138,20 @@ void OpenALStream::SoundLoop()
Common::SetCurrentThreadName("Audio thread - openal");
u32 ulFrequency = m_mixer->GetSampleRate();
numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers
memset(uiBuffers, 0, OAL_NUM_BUFFERS * sizeof(ALuint));
memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
uiSource = 0;
// Generate some AL Buffers for streaming
alGenBuffers(OAL_NUM_BUFFERS, (ALuint *)uiBuffers);
alGenBuffers(numBuffers, (ALuint *)uiBuffers);
// Generate a Source to playback the Buffers
alGenSources(1, &uiSource);
// Short Silence
memset(sampleBuffer, 0, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_NUM_BUFFERS);
memset(sampleBuffer, 0, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * numBuffers);
memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
for (int i = 0; i < OAL_NUM_BUFFERS; i++)
for (int i = 0; i < numBuffers; i++)
{
#if !defined(__APPLE__)
if (Core::g_CoreStartupParameter.bDPL2Decoder)
@ -154,7 +160,7 @@ void OpenALStream::SoundLoop()
#endif
alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, ulFrequency);
}
alSourceQueueBuffers(uiSource, OAL_NUM_BUFFERS, uiBuffers);
alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
alSourcePlay(uiSource);
// Set the default sound volume as saved in the config file.
@ -166,7 +172,7 @@ void OpenALStream::SoundLoop()
ALint iBuffersFilled = 0;
ALint iBuffersProcessed = 0;
ALint iState = 0;
ALuint uiBufferTemp[OAL_NUM_BUFFERS] = {0};
ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};
soundTouch.setChannels(2);
soundTouch.setSampleRate(ulFrequency);
@ -216,7 +222,7 @@ void OpenALStream::SoundLoop()
soundTouch.setSetting(SETTING_SEQUENCE_MS, (int)(1 / (rate * rate)));
soundTouch.setTempo(rate);
}
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_NUM_BUFFERS);
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS);
if (nSamples > 0)
{
// Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued Buffer)
@ -236,14 +242,14 @@ void OpenALStream::SoundLoop()
if (surround_capable)
{
// Convert the samples from short to float for the dpl2 decoder
float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_NUM_BUFFERS];
float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples; ++i)
{
dest[i * 2 + 0] = (float)sampleBuffer[i * 2 + 0] / (1<<16);
dest[i * 2 + 1] = (float)sampleBuffer[i * 2 + 1] / (1<<16);
}
float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_NUM_BUFFERS];
float dpl2[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
dpl2decode(dest, nSamples, dpl2);
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2, nSamples * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
@ -273,7 +279,7 @@ void OpenALStream::SoundLoop()
}
iBuffersFilled++;
if (iBuffersFilled == OAL_NUM_BUFFERS)
if (iBuffersFilled == numBuffers)
{
alSourcePlay(uiSource);
ALenum err = alGetError();