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https://github.com/dolphin-emu/dolphin.git
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412 lines
12 KiB
C++
412 lines
12 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 <climits>
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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/MsgHandler.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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INFO_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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static ALenum CheckALError(const char* desc)
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{
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ALenum err = alGetError();
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if (err != AL_NO_ERROR)
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{
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std::string type;
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switch (err)
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{
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case AL_INVALID_NAME:
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type = "AL_INVALID_NAME";
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break;
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case AL_INVALID_ENUM:
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type = "AL_INVALID_ENUM";
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break;
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case AL_INVALID_VALUE:
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type = "AL_INVALID_VALUE";
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break;
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case AL_INVALID_OPERATION:
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type = "AL_INVALID_OPERATION";
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break;
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case AL_OUT_OF_MEMORY:
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type = "AL_OUT_OF_MEMORY";
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break;
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default:
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type = "UNKNOWN_ERROR";
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break;
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}
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ERROR_LOG(AUDIO, "Error %s: %08x %s", desc, err, type.c_str());
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}
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return err;
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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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bool float32_capable = false;
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bool fixed32_capable = false;
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#if defined(__APPLE__)
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surround_capable = false;
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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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if (alIsExtensionPresent("AL_EXT_float32"))
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float32_capable = true;
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// As there is no extension to check for 32-bit fixed point support
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// and we know that only a X-Fi with hardware OpenAL supports it,
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// we just check if one is being used.
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if (strstr(alGetString(AL_RENDERER), "X-Fi"))
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fixed32_capable = true;
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// Clear error state before querying or else we get false positives.
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ALenum err = alGetError();
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// Generate some AL Buffers for streaming
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alGenBuffers(numBuffers, (ALuint*)uiBuffers);
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err = CheckALError("generating buffers");
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// Generate a Source to playback the Buffers
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alGenSources(1, &uiSource);
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err = CheckALError("generating sources");
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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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unsigned int nextBuffer = 0;
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unsigned int numBuffersQueued = 0;
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ALint iState = 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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// Block until we have a free buffer
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int numBuffersProcessed;
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alGetSourcei(uiSource, AL_BUFFERS_PROCESSED, &numBuffersProcessed);
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if (numBuffers == numBuffersQueued && !numBuffersProcessed)
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{
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soundSyncEvent.Wait();
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continue;
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}
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// Remove the Buffer from the Queue.
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if (numBuffersProcessed)
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{
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ALuint unqueuedBufferIds[OAL_MAX_BUFFERS];
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alSourceUnqueueBuffers(uiSource, numBuffersProcessed, unqueuedBufferIds);
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err = CheckALError("unqueuing buffers");
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numBuffersQueued -= numBuffersProcessed;
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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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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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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(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, dpl2,
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nSamples * FRAME_SURROUND_FLOAT, ulFrequency);
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}
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else if (fixed32_capable)
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{
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int surround_int32[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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{
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// For some reason the ffdshow's DPL2 decoder outputs samples bigger than 1.
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// Most are close to 2.5 and some go up to 8. Hard clamping here, we need to
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// fix the decoder or implement a limiter.
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dpl2[i] = dpl2[i] * (INT64_C(1) << 31);
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if (dpl2[i] > INT_MAX)
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surround_int32[i] = INT_MAX;
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else if (dpl2[i] < INT_MIN)
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surround_int32[i] = INT_MIN;
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else
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surround_int32[i] = (int)dpl2[i];
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}
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alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN32, surround_int32,
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nSamples * FRAME_SURROUND_INT32, 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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{
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dpl2[i] = dpl2[i] * (1 << 15);
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if (dpl2[i] > SHRT_MAX)
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surround_short[i] = SHRT_MAX;
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else if (dpl2[i] < SHRT_MIN)
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surround_short[i] = SHRT_MIN;
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else
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surround_short[i] = (int)dpl2[i];
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}
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alBufferData(uiBuffers[nextBuffer], AL_FORMAT_51CHN16, surround_short,
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nSamples * FRAME_SURROUND_SHORT, ulFrequency);
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}
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err = CheckALError("buffering data");
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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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}
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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(uiBuffers[nextBuffer], AL_FORMAT_STEREO_FLOAT32, sampleBuffer,
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nSamples * FRAME_STEREO_FLOAT, ulFrequency);
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err = CheckALError("buffering float32 data");
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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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}
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else if (fixed32_capable)
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{
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// Clamping is not necessary here, samples are always between (-1,1)
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int stereo_int32[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_int32[i] = (int)((float)sampleBuffer[i] * (INT64_C(1) << 31));
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alBufferData(uiBuffers[nextBuffer], AL_FORMAT_STEREO32, stereo_int32,
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nSamples * FRAME_STEREO_INT32, ulFrequency);
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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(uiBuffers[nextBuffer], 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, &uiBuffers[nextBuffer]);
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err = CheckALError("queuing buffers");
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numBuffersQueued++;
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nextBuffer = (nextBuffer + 1) % numBuffers;
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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 = CheckALError("occurred resuming playback");
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}
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}
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}
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#endif // HAVE_OPENAL
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