mirror of
https://github.com/qTox/qTox.git
synced 2024-03-22 14:00:36 +08:00
367 lines
9.9 KiB
C++
367 lines
9.9 KiB
C++
/*
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Copyright (C) 2014 by Project Tox <https://tox.im>
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This file is part of qTox, a Qt-based graphical interface for Tox.
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This program is libre software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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See the COPYING file for more details.
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*/
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// Output some extra debug info
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#define AUDIO_DEBUG 1
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// Fix a 7 years old openal-soft/alsa bug
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// http://blog.gmane.org/gmane.comp.lib.openal.devel/month=20080501
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// If set to 1, the capture will be started as long as the device is open
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#define FIX_SND_PCM_PREPARE_BUG 0
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#include "audio.h"
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#include "src/core/core.h"
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#include <QDebug>
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#include <QThread>
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#include <QMutexLocker>
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#include <cassert>
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std::atomic<int> Audio::userCount{0};
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Audio* Audio::instance{nullptr};
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QThread* Audio::audioThread{nullptr};
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QMutex* Audio::audioInLock{nullptr};
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QMutex* Audio::audioOutLock{nullptr};
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ALCdevice* Audio::alInDev{nullptr};
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ALCdevice* Audio::alOutDev{nullptr};
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ALCcontext* Audio::alContext{nullptr};
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ALuint Audio::alMainSource{0};
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float Audio::outputVolume{1.0};
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void audioDebugLog(QString msg)
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{
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#if (AUDIO_DEBUG)
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qDebug()<<"Audio: "<<msg;
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#endif
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}
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Audio& Audio::getInstance()
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{
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if (!instance)
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{
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instance = new Audio();
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audioThread = new QThread(instance);
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audioThread->setObjectName("qTox Audio");
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audioThread->start();
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audioInLock = new QMutex(QMutex::Recursive);
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audioOutLock = new QMutex(QMutex::Recursive);
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instance->moveToThread(audioThread);
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}
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return *instance;
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}
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Audio::~Audio()
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{
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qDebug() << "Deleting Audio";
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audioThread->exit(0);
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audioThread->wait();
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if (audioThread->isRunning())
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audioThread->terminate();
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delete audioThread;
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delete audioInLock;
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delete audioOutLock;
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}
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void Audio::suscribeInput()
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{
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if (!alInDev)
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{
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qWarning()<<"Audio::suscribeInput: input device is closed";
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return;
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}
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audioDebugLog("suscribing");
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QMutexLocker lock(audioInLock);
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if (!userCount++ && alInDev)
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{
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#if (!FIX_SND_PCM_PREPARE_BUG)
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audioDebugLog("starting capture");
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alcCaptureStart(alInDev);
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#endif
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}
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}
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void Audio::unsuscribeInput()
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{
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if (!alInDev)
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{
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qWarning()<<"Audio::unsuscribeInput: input device is closed";
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return;
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}
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audioDebugLog("unsuscribing");
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QMutexLocker lock(audioInLock);
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if (!--userCount && alInDev)
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{
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#if (!FIX_SND_PCM_PREPARE_BUG)
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audioDebugLog("stopping capture");
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alcCaptureStop(alInDev);
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#endif
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}
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}
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void Audio::openInput(const QString& inDevDescr)
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{
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audioDebugLog("Trying to open input "+inDevDescr);
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QMutexLocker lock(audioInLock);
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auto* tmp = alInDev;
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alInDev = nullptr;
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if (tmp)
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alcCaptureCloseDevice(tmp);
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int stereoFlag = av_DefaultSettings.audio_channels==1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
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if (inDevDescr.isEmpty())
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alInDev = alcCaptureOpenDevice(nullptr,av_DefaultSettings.audio_sample_rate, stereoFlag,
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(av_DefaultSettings.audio_frame_duration * av_DefaultSettings.audio_sample_rate * 4)
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/ 1000 * av_DefaultSettings.audio_channels);
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else
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alInDev = alcCaptureOpenDevice(inDevDescr.toStdString().c_str(),av_DefaultSettings.audio_sample_rate, stereoFlag,
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(av_DefaultSettings.audio_frame_duration * av_DefaultSettings.audio_sample_rate * 4)
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/ 1000 * av_DefaultSettings.audio_channels);
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if (!alInDev)
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qWarning() << "Audio: Cannot open input audio device";
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else
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qDebug() << "Audio: Opening audio input "<<inDevDescr;
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Core::getInstance()->resetCallSources(); // Force to regen each group call's sources
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// Restart the capture if necessary
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if (userCount.load() != 0 && alInDev)
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{
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alcCaptureStart(alInDev);
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}
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else
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{
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#if (FIX_SND_PCM_PREPARE_BUG)
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alcCaptureStart(alInDev);
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#endif
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}
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}
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void Audio::openOutput(const QString& outDevDescr)
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{
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audioDebugLog("Trying to open output "+outDevDescr);
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QMutexLocker lock(audioOutLock);
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auto* tmp = alOutDev;
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alOutDev = nullptr;
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if (outDevDescr.isEmpty())
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alOutDev = alcOpenDevice(nullptr);
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else
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alOutDev = alcOpenDevice(outDevDescr.toStdString().c_str());
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if (!alOutDev)
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{
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qWarning() << "Audio: Cannot open output audio device";
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}
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else
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{
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if (alContext && alcMakeContextCurrent(nullptr) == ALC_TRUE)
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alcDestroyContext(alContext);
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if (tmp)
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alcCloseDevice(tmp);
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alContext=alcCreateContext(alOutDev,nullptr);
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if (!alcMakeContextCurrent(alContext))
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{
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qWarning() << "Audio: Cannot create output audio context";
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alcCloseDevice(alOutDev);
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}
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else
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{
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alGenSources(1, &alMainSource);
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}
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qDebug() << "Audio: Opening audio output "<<outDevDescr;
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}
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Core::getInstance()->resetCallSources(); // Force to regen each group call's sources
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}
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void Audio::closeInput()
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{
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audioDebugLog("Closing input");
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QMutexLocker lock(audioInLock);
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if (alInDev)
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{
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if (alcCaptureCloseDevice(alInDev) == ALC_TRUE)
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{
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alInDev = nullptr;
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userCount = 0;
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}
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else
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{
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qWarning() << "Audio: Failed to close input";
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}
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}
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}
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void Audio::closeOutput()
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{
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audioDebugLog("Closing output");
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QMutexLocker lock(audioOutLock);
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if (alContext && alcMakeContextCurrent(nullptr) == ALC_TRUE)
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alcDestroyContext(alContext);
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if (alOutDev)
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{
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if (alcCloseDevice(alOutDev) == ALC_TRUE)
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alOutDev = nullptr;
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else
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qWarning() << "Audio: Failed to close output";
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}
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}
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void Audio::playMono16Sound(const QByteArray& data)
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{
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QMutexLocker lock(audioOutLock);
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if (!alOutDev)
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return;
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ALuint buffer;
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alGenBuffers(1, &buffer);
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alBufferData(buffer, AL_FORMAT_MONO16, data.data(), data.size(), 44100);
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alSourcei(alMainSource, AL_BUFFER, buffer);
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alSourcePlay(alMainSource);
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alDeleteBuffers(1, &buffer);
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}
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void Audio::playGroupAudioQueued(Tox*,int group, int peer, const int16_t* data,
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unsigned samples, uint8_t channels, unsigned sample_rate, void* core)
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{
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QMetaObject::invokeMethod(instance, "playGroupAudio", Qt::BlockingQueuedConnection,
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Q_ARG(int,group), Q_ARG(int,peer), Q_ARG(const int16_t*,data),
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Q_ARG(unsigned,samples), Q_ARG(uint8_t,channels), Q_ARG(unsigned,sample_rate));
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emit static_cast<Core*>(core)->groupPeerAudioPlaying(group, peer);
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}
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void Audio::playGroupAudio(int group, int peer, const int16_t* data,
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unsigned samples, uint8_t channels, unsigned sample_rate)
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{
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assert(QThread::currentThread() == audioThread);
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QMutexLocker lock(audioOutLock);
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ToxGroupCall& call = Core::groupCalls[group];
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if (!call.active || call.muteVol)
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return;
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if (!call.alSources.contains(peer))
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{
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alGenSources(1, &call.alSources[peer]);
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alSourcef(call.alSources[peer], AL_GAIN, outputVolume);
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}
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playAudioBuffer(call.alSources[peer], data, samples, channels, sample_rate);
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}
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void Audio::playAudioBuffer(ALuint alSource, const int16_t *data, int samples, unsigned channels, int sampleRate)
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{
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assert(channels == 1 || channels == 2);
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QMutexLocker lock(audioOutLock);
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ALuint bufid;
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ALint processed = 0, queued = 16;
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alGetSourcei(alSource, AL_BUFFERS_PROCESSED, &processed);
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alGetSourcei(alSource, AL_BUFFERS_QUEUED, &queued);
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alSourcei(alSource, AL_LOOPING, AL_FALSE);
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if (processed)
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{
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ALuint bufids[processed];
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alSourceUnqueueBuffers(alSource, processed, bufids);
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alDeleteBuffers(processed - 1, bufids + 1);
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bufid = bufids[0];
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}
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else if (queued < 16)
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{
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alGenBuffers(1, &bufid);
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}
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else
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{
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qDebug() << "Audio: Dropped frame";
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return;
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}
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alBufferData(bufid, (channels == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16, data,
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samples * 2 * channels, sampleRate);
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alSourceQueueBuffers(alSource, 1, &bufid);
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ALint state;
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alGetSourcei(alSource, AL_SOURCE_STATE, &state);
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alSourcef(alSource, AL_GAIN, outputVolume);
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if (state != AL_PLAYING)
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alSourcePlay(alSource);
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}
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bool Audio::isInputReady()
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{
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return (alInDev && userCount);
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}
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bool Audio::isOutputClosed()
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{
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return (alOutDev);
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}
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bool Audio::tryCaptureSamples(uint8_t* buf, int framesize)
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{
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QMutexLocker lock(audioInLock);
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ALint samples=0;
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alcGetIntegerv(Audio::alInDev, ALC_CAPTURE_SAMPLES, sizeof(samples), &samples);
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if (samples < framesize)
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return false;
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memset(buf, 0, framesize * 2 * av_DefaultSettings.audio_channels); // Avoid uninitialized values (Valgrind)
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alcCaptureSamples(Audio::alInDev, buf, framesize);
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return true;
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}
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#ifdef QTOX_FILTER_AUDIO
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#include "audiofilterer.h"
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/* include for compatibility with older versions of OpenAL */
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#ifndef ALC_ALL_DEVICES_SPECIFIER
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#include <AL/alext.h>
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#endif
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void Audio::getEchoesToFilter(AudioFilterer* filterer, int framesize)
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{
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#ifdef ALC_LOOPBACK_CAPTURE_SAMPLES
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ALint samples;
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alcGetIntegerv(Audio::alOutDev, ALC_LOOPBACK_CAPTURE_SAMPLES, sizeof(samples), &samples);
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if (samples >= framesize)
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{
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int16_t buf[framesize];
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alcCaptureSamplesLoopback(Audio::alOutDev, buf, framesize);
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filterer->passAudioOutput(buf, framesize);
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filterer->setEchoDelayMs(5); // This 5ms is configurable I believe
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}
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#else
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Q_UNUSED(filterer);
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Q_UNUSED(framesize);
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#endif
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}
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#endif
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