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Added a latency setting to the audio settings.

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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
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2
Source/Core/AudioCommon/Src/AudioCommonConfig.cpp
View File

@ -42,7 +42,6 @@ void AudioCommonConfig::Load()
#else
file.Get("Config", "Backend", &sBackend, BACKEND_NULLSOUND);
#endif
file.Get("Config", "Frequency", &iFrequency, 48000);
file.Get("Config", "Volume", &m_Volume, 100);
}
@ -55,7 +54,6 @@ void AudioCommonConfig::SaveSettings()
file.Set("Config", "EnableJIT", m_EnableJIT);
file.Set("Config", "DumpAudio", m_DumpAudio);
file.Set("Config", "Backend", sBackend);
file.Set("Config", "Frequency", iFrequency);
file.Set("Config", "Volume", m_Volume);
file.Save(File::GetUserPath(F_DSPCONFIG_IDX));

1
Source/Core/AudioCommon/Src/AudioCommonConfig.h
View File

@ -37,7 +37,6 @@ struct AudioCommonConfig
bool m_DumpAudio;
int m_Volume;
std::string sBackend;
int iFrequency;
// Load from given file
void Load();

28
Source/Core/AudioCommon/Src/OpenALStream.cpp
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();

10
Source/Core/AudioCommon/Src/OpenALStream.h
View File

@ -44,8 +44,8 @@
// 16 bit Stereo
#define SFX_MAX_SOURCE 1
#define OAL_NUM_BUFFERS 16
#define OAL_MAX_SAMPLES 512
#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
#endif
@ -75,10 +75,12 @@ private:
Common::Event soundSyncEvent;
short realtimeBuffer[OAL_MAX_SAMPLES * 2];
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_NUM_BUFFERS];
ALuint uiBuffers[OAL_NUM_BUFFERS];
soundtouch::SAMPLETYPE sampleBuffer[OAL_MAX_SAMPLES * SIZE_FLOAT * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
ALuint uiBuffers[OAL_MAX_BUFFERS];
ALuint uiSource;
ALfloat fVolume;
u8 numBuffers;
#else
public:
OpenALStream(CMixer *mixer, void *hWnd = NULL): SoundStream(mixer) {}