skidau 3632ce6df5 Removed the synchronisation between the CPU thread and the audio thread.
Added code to detect and resume from buffer underruns.
Disabled the ability to change the DPL2 option after the game has started.
Fixed a memory leak that occurred in the DPL2 decoder.
Fixed the OSX build.
2013-01-11 19:42:03 +11:00

305 lines
8.1 KiB
C++

// Copyright (C) 2003 Dolphin Project.
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, version 2.0.
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License 2.0 for more details.
// A copy of the GPL 2.0 should have been included with the program.
// If not, see http://www.gnu.org/licenses/
// Official SVN repository and contact information can be found at
// http://code.google.com/p/dolphin-emu/
#include "aldlist.h"
#include "OpenALStream.h"
#include "DPL2Decoder.h"
#if defined HAVE_OPENAL && HAVE_OPENAL
soundtouch::SoundTouch soundTouch;
//
// AyuanX: Spec says OpenAL1.1 is thread safe already
//
bool OpenALStream::Start()
{
ALDeviceList *pDeviceList = NULL;
ALCcontext *pContext = NULL;
ALCdevice *pDevice = NULL;
bool bReturn = false;
pDeviceList = new ALDeviceList();
if ((pDeviceList) && (pDeviceList->GetNumDevices()))
{
char *defDevName = pDeviceList->GetDeviceName(pDeviceList->GetDefaultDevice());
WARN_LOG(AUDIO, "Found OpenAL device %s", defDevName);
pDevice = alcOpenDevice(defDevName);
if (pDevice)
{
pContext = alcCreateContext(pDevice, NULL);
if (pContext)
{
alcMakeContextCurrent(pContext);
thread = std::thread(std::mem_fun(&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);
}
delete pDeviceList;
}
else
{
PanicAlertT("OpenAL: can't find sound devices");
}
// Initialise DPL2 parameters
dpl2reset();
soundTouch.clear();
return bReturn;
}
void OpenALStream::Stop()
{
threadData = 1;
// 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(OAL_NUM_BUFFERS, uiBuffers);
ALCcontext *pContext = alcGetCurrentContext();
ALCdevice *pDevice = alcGetContextsDevice(pContext);
alcMakeContextCurrent(NULL);
alcDestroyContext(pContext);
alcCloseDevice(pDevice);
}
void OpenALStream::SetVolume(int volume)
{
fVolume = (float)volume / 100.0f;
if (uiSource)
alSourcef(uiSource, AL_GAIN, fVolume);
}
void OpenALStream::Update()
{
soundSyncEvent.Set();
}
void OpenALStream::Clear(bool mute)
{
m_muted = mute;
if(m_muted)
{
soundTouch.clear();
alSourceStop(uiSource);
}
else
{
alSourcePlay(uiSource);
}
}
void OpenALStream::SoundLoop()
{
Common::SetCurrentThreadName("Audio thread - openal");
u32 ulFrequency = m_mixer->GetSampleRate();
memset(uiBuffers, 0, OAL_NUM_BUFFERS * sizeof(ALuint));
uiSource = 0;
// Generate some AL Buffers for streaming
alGenBuffers(OAL_NUM_BUFFERS, (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(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
for (int i = 0; i < OAL_NUM_BUFFERS; i++)
{
#if !defined(__APPLE__)
if (Core::g_CoreStartupParameter.bDPL2Decoder)
alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * SIZE_FLOAT * SURROUND_CHANNELS, ulFrequency);
else
#endif
alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, ulFrequency);
}
alSourceQueueBuffers(uiSource, OAL_NUM_BUFFERS, 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_NUM_BUFFERS] = {0};
soundTouch.setChannels(2);
soundTouch.setSampleRate(ulFrequency);
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);
bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
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;
numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
numSamples = m_mixer->Mix(realtimeBuffer, numSamples);
soundTouch.putSamples(realtimeBuffer, 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();
}
if (rate > 0)
{
// Adjust SETTING_SEQUENCE_MS to balance between lag vs hollow audio
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);
if (nSamples > 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)
{
// Convert the samples from short to float for the dpl2 decoder
float dest[OAL_MAX_SAMPLES * 2 * 2 * OAL_NUM_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];
dpl2decode(dest, 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)
{
alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, sampleBuffer, nSamples * 2 * 2, ulFrequency);
}
alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
ALenum err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
}
iBuffersFilled++;
if (iBuffersFilled == OAL_NUM_BUFFERS)
{
alSourcePlay(uiSource);
ALenum 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);
ALenum err = alGetError();
if (err != 0)
{
ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
}
}
}
}
else
{
soundSyncEvent.Wait();
}
}
}
#endif //HAVE_OPENAL