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mirror of https://github.com/qTox/qTox.git synced 2024-03-22 14:00:36 +08:00
qTox/src/audio.cpp
2015-04-24 15:43:57 +02:00

367 lines
9.9 KiB
C++

/*
Copyright (C) 2014 by Project Tox <https://tox.im>
This file is part of qTox, a Qt-based graphical interface for Tox.
This program is libre 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, either version 3 of the License, or
(at your option) any later version.
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 COPYING file for more details.
*/
// Output some extra debug info
#define AUDIO_DEBUG 1
// Fix a 7 years old openal-soft/alsa bug
// http://blog.gmane.org/gmane.comp.lib.openal.devel/month=20080501
// If set to 1, the capture will be started as long as the device is open
#define FIX_SND_PCM_PREPARE_BUG 0
#include "audio.h"
#include "src/core/core.h"
#include <QDebug>
#include <QThread>
#include <QMutexLocker>
#include <cassert>
std::atomic<int> Audio::userCount{0};
Audio* Audio::instance{nullptr};
QThread* Audio::audioThread{nullptr};
QMutex* Audio::audioInLock{nullptr};
QMutex* Audio::audioOutLock{nullptr};
ALCdevice* Audio::alInDev{nullptr};
ALCdevice* Audio::alOutDev{nullptr};
ALCcontext* Audio::alContext{nullptr};
ALuint Audio::alMainSource{0};
float Audio::outputVolume{1.0};
void audioDebugLog(QString msg)
{
#if (AUDIO_DEBUG)
qDebug()<<"Audio: "<<msg;
#endif
}
Audio& Audio::getInstance()
{
if (!instance)
{
instance = new Audio();
audioThread = new QThread(instance);
audioThread->setObjectName("qTox Audio");
audioThread->start();
audioInLock = new QMutex(QMutex::Recursive);
audioOutLock = new QMutex(QMutex::Recursive);
instance->moveToThread(audioThread);
}
return *instance;
}
Audio::~Audio()
{
qDebug() << "Deleting Audio";
audioThread->exit(0);
audioThread->wait();
if (audioThread->isRunning())
audioThread->terminate();
delete audioThread;
delete audioInLock;
delete audioOutLock;
}
void Audio::suscribeInput()
{
if (!alInDev)
{
qWarning()<<"Audio::suscribeInput: input device is closed";
return;
}
audioDebugLog("suscribing");
QMutexLocker lock(audioInLock);
if (!userCount++ && alInDev)
{
#if (!FIX_SND_PCM_PREPARE_BUG)
audioDebugLog("starting capture");
alcCaptureStart(alInDev);
#endif
}
}
void Audio::unsuscribeInput()
{
if (!alInDev)
{
qWarning()<<"Audio::unsuscribeInput: input device is closed";
return;
}
audioDebugLog("unsuscribing");
QMutexLocker lock(audioInLock);
if (!--userCount && alInDev)
{
#if (!FIX_SND_PCM_PREPARE_BUG)
audioDebugLog("stopping capture");
alcCaptureStop(alInDev);
#endif
}
}
void Audio::openInput(const QString& inDevDescr)
{
audioDebugLog("Trying to open input "+inDevDescr);
QMutexLocker lock(audioInLock);
auto* tmp = alInDev;
alInDev = nullptr;
if (tmp)
alcCaptureCloseDevice(tmp);
int stereoFlag = av_DefaultSettings.audio_channels==1 ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
if (inDevDescr.isEmpty())
alInDev = alcCaptureOpenDevice(nullptr,av_DefaultSettings.audio_sample_rate, stereoFlag,
(av_DefaultSettings.audio_frame_duration * av_DefaultSettings.audio_sample_rate * 4)
/ 1000 * av_DefaultSettings.audio_channels);
else
alInDev = alcCaptureOpenDevice(inDevDescr.toStdString().c_str(),av_DefaultSettings.audio_sample_rate, stereoFlag,
(av_DefaultSettings.audio_frame_duration * av_DefaultSettings.audio_sample_rate * 4)
/ 1000 * av_DefaultSettings.audio_channels);
if (!alInDev)
qWarning() << "Audio: Cannot open input audio device";
else
qDebug() << "Audio: Opening audio input "<<inDevDescr;
Core::getInstance()->resetCallSources(); // Force to regen each group call's sources
// Restart the capture if necessary
if (userCount.load() != 0 && alInDev)
{
alcCaptureStart(alInDev);
}
else
{
#if (FIX_SND_PCM_PREPARE_BUG)
alcCaptureStart(alInDev);
#endif
}
}
void Audio::openOutput(const QString& outDevDescr)
{
audioDebugLog("Trying to open output "+outDevDescr);
QMutexLocker lock(audioOutLock);
auto* tmp = alOutDev;
alOutDev = nullptr;
if (outDevDescr.isEmpty())
alOutDev = alcOpenDevice(nullptr);
else
alOutDev = alcOpenDevice(outDevDescr.toStdString().c_str());
if (!alOutDev)
{
qWarning() << "Audio: Cannot open output audio device";
}
else
{
if (alContext && alcMakeContextCurrent(nullptr) == ALC_TRUE)
alcDestroyContext(alContext);
if (tmp)
alcCloseDevice(tmp);
alContext=alcCreateContext(alOutDev,nullptr);
if (!alcMakeContextCurrent(alContext))
{
qWarning() << "Audio: Cannot create output audio context";
alcCloseDevice(alOutDev);
}
else
{
alGenSources(1, &alMainSource);
}
qDebug() << "Audio: Opening audio output "<<outDevDescr;
}
Core::getInstance()->resetCallSources(); // Force to regen each group call's sources
}
void Audio::closeInput()
{
audioDebugLog("Closing input");
QMutexLocker lock(audioInLock);
if (alInDev)
{
if (alcCaptureCloseDevice(alInDev) == ALC_TRUE)
{
alInDev = nullptr;
userCount = 0;
}
else
{
qWarning() << "Audio: Failed to close input";
}
}
}
void Audio::closeOutput()
{
audioDebugLog("Closing output");
QMutexLocker lock(audioOutLock);
if (alContext && alcMakeContextCurrent(nullptr) == ALC_TRUE)
alcDestroyContext(alContext);
if (alOutDev)
{
if (alcCloseDevice(alOutDev) == ALC_TRUE)
alOutDev = nullptr;
else
qWarning() << "Audio: Failed to close output";
}
}
void Audio::playMono16Sound(const QByteArray& data)
{
QMutexLocker lock(audioOutLock);
if (!alOutDev)
return;
ALuint buffer;
alGenBuffers(1, &buffer);
alBufferData(buffer, AL_FORMAT_MONO16, data.data(), data.size(), 44100);
alSourcei(alMainSource, AL_BUFFER, buffer);
alSourcePlay(alMainSource);
alDeleteBuffers(1, &buffer);
}
void Audio::playGroupAudioQueued(Tox*,int group, int peer, const int16_t* data,
unsigned samples, uint8_t channels, unsigned sample_rate, void* core)
{
QMetaObject::invokeMethod(instance, "playGroupAudio", Qt::BlockingQueuedConnection,
Q_ARG(int,group), Q_ARG(int,peer), Q_ARG(const int16_t*,data),
Q_ARG(unsigned,samples), Q_ARG(uint8_t,channels), Q_ARG(unsigned,sample_rate));
emit static_cast<Core*>(core)->groupPeerAudioPlaying(group, peer);
}
void Audio::playGroupAudio(int group, int peer, const int16_t* data,
unsigned samples, uint8_t channels, unsigned sample_rate)
{
assert(QThread::currentThread() == audioThread);
QMutexLocker lock(audioOutLock);
ToxGroupCall& call = Core::groupCalls[group];
if (!call.active || call.muteVol)
return;
if (!call.alSources.contains(peer))
{
alGenSources(1, &call.alSources[peer]);
alSourcef(call.alSources[peer], AL_GAIN, outputVolume);
}
playAudioBuffer(call.alSources[peer], data, samples, channels, sample_rate);
}
void Audio::playAudioBuffer(ALuint alSource, const int16_t *data, int samples, unsigned channels, int sampleRate)
{
assert(channels == 1 || channels == 2);
QMutexLocker lock(audioOutLock);
ALuint bufid;
ALint processed = 0, queued = 16;
alGetSourcei(alSource, AL_BUFFERS_PROCESSED, &processed);
alGetSourcei(alSource, AL_BUFFERS_QUEUED, &queued);
alSourcei(alSource, AL_LOOPING, AL_FALSE);
if (processed)
{
ALuint bufids[processed];
alSourceUnqueueBuffers(alSource, processed, bufids);
alDeleteBuffers(processed - 1, bufids + 1);
bufid = bufids[0];
}
else if (queued < 16)
{
alGenBuffers(1, &bufid);
}
else
{
qDebug() << "Audio: Dropped frame";
return;
}
alBufferData(bufid, (channels == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16, data,
samples * 2 * channels, sampleRate);
alSourceQueueBuffers(alSource, 1, &bufid);
ALint state;
alGetSourcei(alSource, AL_SOURCE_STATE, &state);
alSourcef(alSource, AL_GAIN, outputVolume);
if (state != AL_PLAYING)
alSourcePlay(alSource);
}
bool Audio::isInputReady()
{
return (alInDev && userCount);
}
bool Audio::isOutputClosed()
{
return (alOutDev);
}
bool Audio::tryCaptureSamples(uint8_t* buf, int framesize)
{
QMutexLocker lock(audioInLock);
ALint samples=0;
alcGetIntegerv(Audio::alInDev, ALC_CAPTURE_SAMPLES, sizeof(samples), &samples);
if (samples < framesize)
return false;
memset(buf, 0, framesize * 2 * av_DefaultSettings.audio_channels); // Avoid uninitialized values (Valgrind)
alcCaptureSamples(Audio::alInDev, buf, framesize);
return true;
}
#ifdef QTOX_FILTER_AUDIO
#include "audiofilterer.h"
/* include for compatibility with older versions of OpenAL */
#ifndef ALC_ALL_DEVICES_SPECIFIER
#include <AL/alext.h>
#endif
void Audio::getEchoesToFilter(AudioFilterer* filterer, int framesize)
{
#ifdef ALC_LOOPBACK_CAPTURE_SAMPLES
ALint samples;
alcGetIntegerv(Audio::alOutDev, ALC_LOOPBACK_CAPTURE_SAMPLES, sizeof(samples), &samples);
if (samples >= framesize)
{
int16_t buf[framesize];
alcCaptureSamplesLoopback(Audio::alOutDev, buf, framesize);
filterer->passAudioOutput(buf, framesize);
filterer->setEchoDelayMs(5); // This 5ms is configurable I believe
}
#else
Q_UNUSED(filterer);
Q_UNUSED(framesize);
#endif
}
#endif