void OpenALStream::Stop()
{
m_run_thread.Clear();
// kick the thread if it's waiting
soundSyncEvent.Set();
soundTouch.clear();
thread.join();
alSourceStop(uiSource);
alSourcei(uiSource, AL_BUFFER, 0);
// Clean up buffers and sources
alDeleteSources(1, &uiSource);
uiSource = 0;
alDeleteBuffers(numBuffers, uiBuffers);
ALCcontext* pContext = alcGetCurrentContext();
ALCdevice* pDevice = alcGetContextsDevice(pContext);
alcMakeContextCurrent(nullptr);
alcDestroyContext(pContext);
alcCloseDevice(pDevice);
}
void OpenALStream::Clear(bool mute)
{
m_muted = mute;
if (m_muted)
{
soundTouch.clear();
alSourceStop(uiSource);
}
else
{
alSourcePlay(uiSource);
}
}
//
// AyuanX: Spec says OpenAL1.1 is thread safe already
//
bool OpenALStream::Start()
{
m_run_thread.Set();
bool bReturn = false;
ALDeviceList pDeviceList;
if (pDeviceList.GetNumDevices())
{
char* defDevName = pDeviceList.GetDeviceName(pDeviceList.GetDefaultDevice());
INFO_LOG(AUDIO, "Found OpenAL device %s", defDevName);
ALCdevice* pDevice = alcOpenDevice(defDevName);
if (pDevice)
{
ALCcontext* pContext = alcCreateContext(pDevice, nullptr);
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(&OpenALStream::SoundLoop, this);
bReturn = true;
}
else
{
alcCloseDevice(pDevice);
PanicAlertT("OpenAL: can't create context for device %s", defDevName);
}
}
else
{
PanicAlertT("OpenAL: can't open device %s", defDevName);
}
}
else
{
PanicAlertT("OpenAL: can't find sound devices");
}
// Initialize DPL2 parameters
DPL2Reset();
soundTouch.clear();
return bReturn;
}
void ApplySettings(soundtouch::SoundTouch &sndtouch)
{
sndtouch.setSetting(SETTING_SEQUENCE_MS, SequenceLenMS);
sndtouch.setSetting(SETTING_SEEKWINDOW_MS, SeekWindowMS);
sndtouch.setSetting(SETTING_OVERLAP_MS, OverlapMS);
}
void OpenALStream::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - openal");
bool surround_capable = SConfig::GetInstance().bDPL2Decoder;
bool float32_capable = false;
bool fixed32_capable = false;
#if defined(__APPLE__)
surround_capable = false;
#endif
u32 ulFrequency = m_mixer->GetSampleRate();
numBuffers = SConfig::GetInstance().iLatency + 2; // OpenAL requires a minimum of two buffers
memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
uiSource = 0;
if (alIsExtensionPresent("AL_EXT_float32"))
float32_capable = true;
// As there is no extension to check for 32-bit fixed point support
// and we know that only a X-Fi with hardware OpenAL supports it,
// we just check if one is being used.
if (strstr(alGetString(AL_RENDERER), "X-Fi"))
fixed32_capable = true;
// Clear error state before querying or else we get false positives.
ALenum err = alGetError();
// Generate some AL Buffers for streaming
alGenBuffers(numBuffers, (ALuint*)uiBuffers);
err = CheckALError("generating buffers");
// Generate a Source to playback the Buffers
alGenSources(1, &uiSource);
err = CheckALError("generating sources");
// Set the default sound volume as saved in the config file.
alSourcef(uiSource, AL_GAIN, fVolume);
// TODO: Error handling
// ALenum err = alGetError();
unsigned int nextBuffer = 0;
unsigned int numBuffersQueued = 0;
ALint iState = 0;
soundTouch.setChannels(2);
soundTouch.setSampleRate(ulFrequency);
soundTouch.setTempo(1.0);
soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
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);
err = CheckALError("unqueuing buffers");
numBuffersQueued -= numBuffersProcessed;
}
// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
const u32 stereo_16_bit_size = 4;
const u32 dma_length = 32;
const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
u64 audio_dma_period = SystemTimers::GetTicksPerSecond() /
(AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
u64 num_samples_to_render =
(audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
unsigned int numSamples = (unsigned int)num_samples_to_render;
unsigned int minSamples =
surround_capable ? 240 : 0; // DPL2 accepts 240 samples minimum (FWRDURATION)
numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
numSamples = m_mixer->Mix(realtimeBuffer, numSamples, false);
// Convert the samples from short to float
float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
dest[i] = (float)realtimeBuffer[i] / (1 << 15);
soundTouch.putSamples(dest, numSamples);
double rate = (double)m_mixer->GetCurrentSpeed();
if (rate <= 0)
{
Core::RequestRefreshInfo();
rate = (double)m_mixer->GetCurrentSpeed();
}
// 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)
{
soundTouch.setTempo(rate);
if (rate > 10)
{
soundTouch.clear();
}
}
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;
}
if (float32_capable)
{
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
}
else if (fixed32_capable)
{
int surround_int32[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
{
// For some reason the ffdshow's DPL2 decoder outputs samples bigger than 1.
// Most are close to 2.5 and some go up to 8. Hard clamping here, we need to
// fix the decoder or implement a limiter.
dpl2[i] = dpl2[i] * (INT64_C(1) << 31);
if (dpl2[i] > INT_MAX)
surround_int32[i] = INT_MAX;
else if (dpl2[i] < INT_MIN)
surround_int32[i] = INT_MIN;
else
surround_int32[i] = (int)dpl2[i];
}
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, surround_int32,
nSamples * FRAME_SURROUND_INT32, ulFrequency);
}
else
{
short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
{
dpl2[i] = dpl2[i] * (1 << 15);
if (dpl2[i] > SHRT_MAX)
surround_short[i] = SHRT_MAX;
else if (dpl2[i] < SHRT_MIN)
surround_short[i] = SHRT_MIN;
else
surround_short[i] = (int)dpl2[i];
}
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
nSamples * FRAME_SURROUND_SHORT, ulFrequency);
}
err = CheckALError("buffering data");
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 (float32_capable)
{
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
nSamples * FRAME_STEREO_FLOAT, ulFrequency);
err = CheckALError("buffering float32 data");
if (err == AL_INVALID_ENUM)
{
float32_capable = false;
}
}
else if (fixed32_capable)
{
// Clamping is not necessary here, samples are always between (-1,1)
int stereo_int32[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
stereo_int32[i] = (int)((float)sampleBuffer[i] * (INT64_C(1) << 31));
alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO32, stereo_int32,
nSamples * FRAME_STEREO_INT32, ulFrequency);
}
else
{
// 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(uiBuffers[nextBuffer], AL_FORMAT_STEREO16, stereo,
nSamples * FRAME_STEREO_SHORT, ulFrequency);
}
}
alSourceQueueBuffers(uiSource, 1, &uiBuffers[nextBuffer]);
err = CheckALError("queuing buffers");
numBuffersQueued++;
nextBuffer = (nextBuffer + 1) % numBuffers;
alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
if (iState != AL_PLAYING)
{
// Buffer underrun occurred, resume playback
alSourcePlay(uiSource);
err = CheckALError("occurred resuming playback");
}
}
}
void OpenALStream::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - openal");
bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
#if defined(__APPLE__)
bool float32_capable = false;
const ALenum AL_FORMAT_STEREO_FLOAT32 = 0;
// OSX does not have the alext AL_FORMAT_51CHN32 yet.
surround_capable = false;
const ALenum AL_FORMAT_51CHN32 = 0;
#else
bool float32_capable = true;
#endif
u32 ulFrequency = m_mixer->GetSampleRate();
numBuffers = Core::g_CoreStartupParameter.iLatency + 2; // OpenAL requires a minimum of two buffers
memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
uiSource = 0;
// Generate some AL Buffers for streaming
alGenBuffers(numBuffers, (ALuint *)uiBuffers);
// Generate a Source to playback the Buffers
alGenSources(1, &uiSource);
// Short Silence
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 (surround_capable)
alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT, 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.
alSourcef(uiSource, AL_GAIN, fVolume);
// TODO: Error handling
//ALenum err = alGetError();
ALint iBuffersFilled = 0;
ALint iBuffersProcessed = 0;
ALint iState = 0;
ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};
soundTouch.setChannels(2);
soundTouch.setSampleRate(ulFrequency);
soundTouch.setTempo(1.0);
soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
while (!threadData)
{
// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
const u32 stereo_16_bit_size = 4;
const u32 dma_length = 32;
const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
u64 audio_dma_period = SystemTimers::GetTicksPerSecond() / (AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
u64 num_samples_to_render = (audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
unsigned int numSamples = (unsigned int)num_samples_to_render;
unsigned int minSamples = surround_capable ? 240 : 0; // DPL2 accepts 240 samples minimum (FWRDURATION)
numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
numSamples = m_mixer->Mix(realtimeBuffer, numSamples, false);
// Convert the samples from short to float
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);
if (iBuffersProcessed == iBuffersFilled)
{
alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
iBuffersFilled = 0;
}
if (iBuffersProcessed)
{
float rate = m_mixer->GetCurrentSpeed();
if (rate <= 0)
{
Core::RequestRefreshInfo();
rate = m_mixer->GetCurrentSpeed();
}
// 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)
{
// Adjust SETTING_SEQUENCE_MS to balance between lag vs hollow audio
soundTouch.setSetting(SETTING_SEQUENCE_MS, (int)(1 / (rate * rate)));
soundTouch.setTempo(rate);
if (rate > 10)
{
soundTouch.clear();
}
}
unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
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);
ALenum err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
}
}
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)
{
// 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)
{
float32_capable = false;
}
else if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
}
}
else
{
// 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);
}
}
}
else
{
soundSyncEvent.Wait();
}
}
}