dolphin/Source/Core/AudioCommon/Src/OpenALStream.cpp
lioncash edd9d0e0ef Clean up more space/tab mismatches in AudioCommon, Common, and VideoCommon.
Not planning to touch Core since it's the most actively changed part of the project.
2013-03-19 21:51:12 -04:00

353 lines
9.5 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)
{
// 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;
}
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(numBuffers, 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();
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 * SIZE_FLOAT * SURROUND_CHANNELS * numBuffers);
memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * 4);
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);
else
#endif
alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * 2 * 2, 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);
bool surround_capable = Core::g_CoreStartupParameter.bDPL2Decoder;
#if defined(__APPLE__)
bool float32_capable = false;
#else
bool float32_capable = true;
#endif
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);
// 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);
}
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 * SIZE_FLOAT * OAL_MAX_BUFFERS);
if (nSamples > minSamples)
{
// 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]);
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();
}
}
}
#endif //HAVE_OPENAL