forked from dolphin-emu/dolphin
dca22e08eb
Replaces old and simple usages of std::atomic<bool> with Common::Flag (which was introduced after the initial usage), so it's clear that the variable is a flag and because Common::Flag is well tested. This also replaces the ready logic in WiimoteReal with Common::Event since it was basically just unnecessarily reimplementing Common::Event.
388 lines
11 KiB
C++
388 lines
11 KiB
C++
// Copyright 2008 Dolphin Emulator Project
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// Licensed under GPLv2+
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// Refer to the license.txt file included.
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#include <cstring>
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#include <thread>
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#include "AudioCommon/DPL2Decoder.h"
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#include "AudioCommon/OpenALStream.h"
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#include "AudioCommon/aldlist.h"
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#include "Common/Logging/Log.h"
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#include "Common/Thread.h"
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#include "Core/ConfigManager.h"
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#if defined HAVE_OPENAL && HAVE_OPENAL
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#ifdef _WIN32
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#pragma comment(lib, "openal32.lib")
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#endif
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static soundtouch::SoundTouch soundTouch;
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//
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// AyuanX: Spec says OpenAL1.1 is thread safe already
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//
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bool OpenALStream::Start()
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{
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m_run_thread.Set();
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bool bReturn = false;
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ALDeviceList pDeviceList;
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if (pDeviceList.GetNumDevices())
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{
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char* defDevName = pDeviceList.GetDeviceName(pDeviceList.GetDefaultDevice());
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WARN_LOG(AUDIO, "Found OpenAL device %s", defDevName);
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ALCdevice* pDevice = alcOpenDevice(defDevName);
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if (pDevice)
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{
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ALCcontext* pContext = alcCreateContext(pDevice, nullptr);
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if (pContext)
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{
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// Used to determine an appropriate period size (2x period = total buffer size)
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// ALCint refresh;
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// alcGetIntegerv(pDevice, ALC_REFRESH, 1, &refresh);
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// period_size_in_millisec = 1000 / refresh;
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alcMakeContextCurrent(pContext);
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thread = std::thread(&OpenALStream::SoundLoop, this);
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bReturn = true;
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}
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else
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{
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alcCloseDevice(pDevice);
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PanicAlertT("OpenAL: can't create context for device %s", defDevName);
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}
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}
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else
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{
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PanicAlertT("OpenAL: can't open device %s", defDevName);
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}
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}
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else
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{
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PanicAlertT("OpenAL: can't find sound devices");
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}
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// Initialize DPL2 parameters
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DPL2Reset();
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soundTouch.clear();
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return bReturn;
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}
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void OpenALStream::Stop()
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{
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m_run_thread.Clear();
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// kick the thread if it's waiting
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soundSyncEvent.Set();
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soundTouch.clear();
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thread.join();
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alSourceStop(uiSource);
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alSourcei(uiSource, AL_BUFFER, 0);
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// Clean up buffers and sources
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alDeleteSources(1, &uiSource);
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uiSource = 0;
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alDeleteBuffers(numBuffers, uiBuffers);
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ALCcontext* pContext = alcGetCurrentContext();
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ALCdevice* pDevice = alcGetContextsDevice(pContext);
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alcMakeContextCurrent(nullptr);
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alcDestroyContext(pContext);
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alcCloseDevice(pDevice);
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}
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void OpenALStream::SetVolume(int volume)
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{
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fVolume = (float)volume / 100.0f;
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if (uiSource)
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alSourcef(uiSource, AL_GAIN, fVolume);
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}
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void OpenALStream::Update()
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{
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soundSyncEvent.Set();
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}
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void OpenALStream::Clear(bool mute)
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{
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m_muted = mute;
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if (m_muted)
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{
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soundTouch.clear();
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alSourceStop(uiSource);
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}
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else
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{
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alSourcePlay(uiSource);
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}
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}
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void OpenALStream::SoundLoop()
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{
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Common::SetCurrentThreadName("Audio thread - openal");
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bool surround_capable = SConfig::GetInstance().bDPL2Decoder;
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#if defined(__APPLE__)
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bool float32_capable = false;
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const ALenum AL_FORMAT_STEREO_FLOAT32 = 0;
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// OS X does not have the alext AL_FORMAT_51CHN32 yet.
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surround_capable = false;
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const ALenum AL_FORMAT_51CHN32 = 0;
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const ALenum AL_FORMAT_51CHN16 = 0;
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#else
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bool float32_capable = true;
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#endif
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u32 ulFrequency = m_mixer->GetSampleRate();
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numBuffers = SConfig::GetInstance().iLatency + 2; // OpenAL requires a minimum of two buffers
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memset(uiBuffers, 0, numBuffers * sizeof(ALuint));
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uiSource = 0;
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// Checks if a X-Fi is being used. If it is, disable FLOAT32 support as this sound card has no
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// support for it even though it reports it does.
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if (strstr(alGetString(AL_RENDERER), "X-Fi"))
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float32_capable = false;
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// Generate some AL Buffers for streaming
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alGenBuffers(numBuffers, (ALuint*)uiBuffers);
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// Generate a Source to playback the Buffers
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alGenSources(1, &uiSource);
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// Short Silence
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if (float32_capable)
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memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_FLOAT);
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else
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memset(sampleBuffer, 0, OAL_MAX_SAMPLES * numBuffers * FRAME_SURROUND_SHORT);
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memset(realtimeBuffer, 0, OAL_MAX_SAMPLES * FRAME_STEREO_SHORT);
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for (int i = 0; i < numBuffers; i++)
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{
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if (surround_capable)
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{
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if (float32_capable)
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alBufferData(uiBuffers[i], AL_FORMAT_51CHN32, sampleBuffer, 4 * FRAME_SURROUND_FLOAT,
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ulFrequency);
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else
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alBufferData(uiBuffers[i], AL_FORMAT_51CHN16, sampleBuffer, 4 * FRAME_SURROUND_SHORT,
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ulFrequency);
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}
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else
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{
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alBufferData(uiBuffers[i], AL_FORMAT_STEREO16, realtimeBuffer, 4 * FRAME_STEREO_SHORT,
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ulFrequency);
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}
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}
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alSourceQueueBuffers(uiSource, numBuffers, uiBuffers);
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alSourcePlay(uiSource);
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// Set the default sound volume as saved in the config file.
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alSourcef(uiSource, AL_GAIN, fVolume);
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// TODO: Error handling
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// ALenum err = alGetError();
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ALint iBuffersFilled = 0;
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ALint iBuffersProcessed = 0;
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ALint iState = 0;
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ALuint uiBufferTemp[OAL_MAX_BUFFERS] = {0};
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soundTouch.setChannels(2);
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soundTouch.setSampleRate(ulFrequency);
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soundTouch.setTempo(1.0);
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soundTouch.setSetting(SETTING_USE_QUICKSEEK, 0);
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soundTouch.setSetting(SETTING_USE_AA_FILTER, 0);
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soundTouch.setSetting(SETTING_SEQUENCE_MS, 1);
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soundTouch.setSetting(SETTING_SEEKWINDOW_MS, 28);
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soundTouch.setSetting(SETTING_OVERLAP_MS, 12);
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while (m_run_thread.IsSet())
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{
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// num_samples_to_render in this update - depends on SystemTimers::AUDIO_DMA_PERIOD.
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const u32 stereo_16_bit_size = 4;
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const u32 dma_length = 32;
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const u64 ais_samples_per_second = 48000 * stereo_16_bit_size;
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u64 audio_dma_period = SystemTimers::GetTicksPerSecond() /
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(AudioInterface::GetAIDSampleRate() * stereo_16_bit_size / dma_length);
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u64 num_samples_to_render =
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(audio_dma_period * ais_samples_per_second) / SystemTimers::GetTicksPerSecond();
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unsigned int numSamples = (unsigned int)num_samples_to_render;
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unsigned int minSamples =
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surround_capable ? 240 : 0; // DPL2 accepts 240 samples minimum (FWRDURATION)
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numSamples = (numSamples > OAL_MAX_SAMPLES) ? OAL_MAX_SAMPLES : numSamples;
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numSamples = m_mixer->Mix(realtimeBuffer, numSamples, false);
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// Convert the samples from short to float
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float dest[OAL_MAX_SAMPLES * STEREO_CHANNELS];
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for (u32 i = 0; i < numSamples * STEREO_CHANNELS; ++i)
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dest[i] = (float)realtimeBuffer[i] / (1 << 15);
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soundTouch.putSamples(dest, numSamples);
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if (iBuffersProcessed == iBuffersFilled)
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{
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alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &iBuffersProcessed);
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iBuffersFilled = 0;
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}
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if (iBuffersProcessed)
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{
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double rate = (double)m_mixer->GetCurrentSpeed();
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if (rate <= 0)
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{
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Core::RequestRefreshInfo();
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rate = (double)m_mixer->GetCurrentSpeed();
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}
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// Place a lower limit of 10% speed. When a game boots up, there will be
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// many silence samples. These do not need to be timestretched.
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if (rate > 0.10)
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{
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soundTouch.setTempo(rate);
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if (rate > 10)
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{
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soundTouch.clear();
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}
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}
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unsigned int nSamples = soundTouch.receiveSamples(sampleBuffer, OAL_MAX_SAMPLES * numBuffers);
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if (nSamples <= minSamples)
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continue;
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// Remove the Buffer from the Queue. (uiBuffer contains the Buffer ID for the unqueued
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// Buffer)
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if (iBuffersFilled == 0)
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{
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alSourceUnqueueBuffers(uiSource, iBuffersProcessed, uiBufferTemp);
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ALenum err = alGetError();
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if (err != 0)
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{
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ERROR_LOG(AUDIO, "Error unqueuing buffers: %08x", err);
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}
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}
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if (surround_capable)
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{
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float dpl2[OAL_MAX_SAMPLES * OAL_MAX_BUFFERS * SURROUND_CHANNELS];
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DPL2Decode(sampleBuffer, nSamples, dpl2);
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// zero-out the subwoofer channel - DPL2Decode generates a pretty
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// good 5.0 but not a good 5.1 output. Sadly there is not a 5.0
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// AL_FORMAT_50CHN32 to make this super-explicit.
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// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
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for (u32 i = 0; i < nSamples; ++i)
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{
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dpl2[i * SURROUND_CHANNELS + 3 /*sub/lfe*/] = 0.0f;
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}
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if (float32_capable)
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{
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alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN32, dpl2,
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nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
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}
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else
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{
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short surround_short[OAL_MAX_SAMPLES * SURROUND_CHANNELS * OAL_MAX_BUFFERS];
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for (u32 i = 0; i < nSamples * SURROUND_CHANNELS; ++i)
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surround_short[i] = (short)((float)dpl2[i] * (1 << 15));
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alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_51CHN16, surround_short,
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nSamples * FRAME_SURROUND_SHORT, ulFrequency);
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}
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ALenum err = alGetError();
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if (err == AL_INVALID_ENUM)
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{
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// 5.1 is not supported by the host, fallback to stereo
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WARN_LOG(AUDIO,
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"Unable to set 5.1 surround mode. Updating OpenAL Soft might fix this issue.");
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surround_capable = false;
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}
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else if (err != 0)
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{
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ERROR_LOG(AUDIO, "Error occurred while buffering data: %08x", err);
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}
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}
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else
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{
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if (float32_capable)
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{
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alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
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nSamples * FRAME_STEREO_FLOAT, ulFrequency);
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ALenum err = alGetError();
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if (err == AL_INVALID_ENUM)
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{
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float32_capable = false;
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}
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else if (err != 0)
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{
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ERROR_LOG(AUDIO, "Error occurred while buffering float32 data: %08x", err);
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}
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}
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else
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{
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// Convert the samples from float to short
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short stereo[OAL_MAX_SAMPLES * STEREO_CHANNELS * OAL_MAX_BUFFERS];
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for (u32 i = 0; i < nSamples * STEREO_CHANNELS; ++i)
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stereo[i] = (short)((float)sampleBuffer[i] * (1 << 15));
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alBufferData(uiBufferTemp[iBuffersFilled], AL_FORMAT_STEREO16, stereo,
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nSamples * FRAME_STEREO_SHORT, ulFrequency);
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}
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}
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alSourceQueueBuffers(uiSource, 1, &uiBufferTemp[iBuffersFilled]);
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ALenum err = alGetError();
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if (err != 0)
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{
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ERROR_LOG(AUDIO, "Error queuing buffers: %08x", err);
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}
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iBuffersFilled++;
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if (iBuffersFilled == numBuffers)
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{
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alSourcePlay(uiSource);
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err = alGetError();
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if (err != 0)
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{
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ERROR_LOG(AUDIO, "Error occurred during playback: %08x", err);
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}
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}
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alGetSourcei(uiSource, AL_SOURCE_STATE, &iState);
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if (iState != AL_PLAYING)
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{
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// Buffer underrun occurred, resume playback
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alSourcePlay(uiSource);
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err = alGetError();
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if (err != 0)
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{
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ERROR_LOG(AUDIO, "Error occurred resuming playback: %08x", err);
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}
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}
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}
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else
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{
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soundSyncEvent.Wait();
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}
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}
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}
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#endif // HAVE_OPENAL
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