kopia lustrzana https://gitlab.com/eliggett/wfview
391 wiersze
12 KiB
C++
391 wiersze
12 KiB
C++
/*
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This class handles both RX and TX audio, each is created as a separate instance of the class
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but as the setup/handling if output (RX) and input (TX) devices is so similar I have combined them.
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*/
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#include "audiohandler.h"
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#include "logcategories.h"
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#include "ulaw.h"
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audioHandler::audioHandler(QObject* parent) : QObject(parent)
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{
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Q_UNUSED(parent)
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}
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audioHandler::~audioHandler()
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{
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if (converterThread != Q_NULLPTR) {
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converterThread->quit();
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converterThread->wait();
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}
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if (isInitialized) {
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stop();
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}
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if (audioInput != Q_NULLPTR) {
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delete audioInput;
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audioInput = Q_NULLPTR;
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}
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if (audioOutput != Q_NULLPTR) {
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delete audioOutput;
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audioOutput = Q_NULLPTR;
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}
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}bool audioHandler::init(audioSetup setup)
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{
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if (isInitialized) {
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return false;
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}
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this->setup = setup;
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "audio handler starting:" << setup.name;
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if (setup.port.isNull())
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{
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "No audio device was found. You probably need to install libqt5multimedia-plugins.";
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return false;
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}
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qDebug(logAudio()) << "Creating" << (setup.isinput ? "Input" : "Output") << "audio device:" << setup.name <<
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", bits" << inFormat.sampleSize() <<
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", codec" << setup.codec <<
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", latency" << setup.latency <<
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", localAFGain" << setup.localAFgain <<
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", radioChan" << inFormat.channelCount() <<
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", resampleQuality" << setup.resampleQuality <<
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", samplerate" << inFormat.sampleRate() <<
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", uLaw" << setup.ulaw;
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inFormat = toQAudioFormat(setup.codec, setup.sampleRate);
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outFormat = setup.port.preferredFormat();
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Preferred Format: SampleSize" << outFormat.sampleSize() << "Channel Count" << outFormat.channelCount() <<
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"Sample Rate" << outFormat.sampleRate() << "Codec" << outFormat.codec() << "Sample Type" << outFormat.sampleType();
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if (outFormat.channelCount() > 2) {
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outFormat.setChannelCount(2);
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}
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else if (outFormat.channelCount() < 1)
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{
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qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << "No channels found, aborting setup.";
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return false;
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}
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if (outFormat.channelCount() == 1 && inFormat.channelCount() == 2) {
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outFormat.setChannelCount(2);
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if (!setup.port.isFormatSupported(outFormat)) {
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Cannot request stereo reverting to mono";
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outFormat.setChannelCount(1);
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}
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}
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if (outFormat.sampleRate() < 48000) {
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int tempRate=outFormat.sampleRate();
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outFormat.setSampleRate(48000);
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if (!setup.port.isFormatSupported(outFormat)) {
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qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << "Cannot request 48K, reverting to "<< tempRate;
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outFormat.setSampleRate(tempRate);
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}
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}
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if (outFormat.sampleType() == QAudioFormat::UnSignedInt && outFormat.sampleSize()==8) {
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outFormat.setSampleType(QAudioFormat::SignedInt);
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outFormat.setSampleSize(16);
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if (!setup.port.isFormatSupported(outFormat)) {
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qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << "Cannot request 16bit Signed samples, reverting to 8bit Unsigned";
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outFormat.setSampleType(QAudioFormat::UnSignedInt);
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outFormat.setSampleSize(8);
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}
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}
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/*
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if (outFormat.sampleType()==QAudioFormat::SignedInt) {
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outFormat.setSampleType(QAudioFormat::Float);
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outFormat.setSampleSize(32);
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if (!setup.port.isFormatSupported(outFormat)) {
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qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << "Attempt to select 32bit Float failed, reverting to SignedInt";
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outFormat.setSampleType(QAudioFormat::SignedInt);
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outFormat.setSampleSize(16);
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}
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}
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*/
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if (outFormat.sampleSize() == 24) {
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// We can't convert this easily so use 32 bit instead.
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outFormat.setSampleSize(32);
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if (!setup.port.isFormatSupported(outFormat)) {
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qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << "24 bit requested and 32 bit audio not supported, try 16 bit instead";
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outFormat.setSampleSize(16);
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}
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}
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Selected format: SampleSize" << outFormat.sampleSize() << "Channel Count" << outFormat.channelCount() <<
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"Sample Rate" << outFormat.sampleRate() << "Codec" << outFormat.codec() << "Sample Type" << outFormat.sampleType();
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// We "hopefully" now have a valid format that is supported so try connecting
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converter = new audioConverter();
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converterThread = new QThread(this);
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if (setup.isinput) {
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converterThread->setObjectName("audioConvIn()");
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}
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else {
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converterThread->setObjectName("audioConvOut()");
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}
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converter->moveToThread(converterThread);
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connect(this, SIGNAL(setupConverter(QAudioFormat,QAudioFormat,quint8,quint8)), converter, SLOT(init(QAudioFormat,QAudioFormat,quint8,quint8)));
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connect(converterThread, SIGNAL(finished()), converter, SLOT(deleteLater()));
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connect(this, SIGNAL(sendToConverter(audioPacket)), converter, SLOT(convert(audioPacket)));
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converterThread->start(QThread::TimeCriticalPriority);
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if (setup.isinput) {
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audioInput = new QAudioInput(setup.port, outFormat, this);
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connect(audioInput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State)));
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emit setupConverter(outFormat, inFormat, 7, setup.resampleQuality);
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connect(converter, SIGNAL(converted(audioPacket)), this, SLOT(convertedInput(audioPacket)));
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}
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else {
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audioOutput = new QAudioOutput(setup.port, outFormat, this);
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connect(audioOutput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State)));
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emit setupConverter(inFormat, outFormat, 7, setup.resampleQuality);
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connect(converter, SIGNAL(converted(audioPacket)), this, SLOT(convertedOutput(audioPacket)));
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}
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "thread id" << QThread::currentThreadId();
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underTimer = new QTimer();
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underTimer->setSingleShot(true);
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connect(underTimer, SIGNAL(timeout()), this, SLOT(clearUnderrun()));
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this->setVolume(setup.localAFgain);
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this->start();
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return true;
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}
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void audioHandler::start()
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{
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "start() running";
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if (setup.isinput) {
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//this->open(QIODevice::WriteOnly);
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//audioInput->start(this);
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#ifdef Q_OS_WIN
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audioInput->setBufferSize(inFormat.bytesForDuration(setup.latency * 100));
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#else
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audioInput->setBufferSize(inFormat.bytesForDuration(setup.latency * 1000));
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#endif
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audioDevice = audioInput->start();
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connect(audioInput, SIGNAL(destroyed()), audioDevice, SLOT(deleteLater()), Qt::UniqueConnection);
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connect(audioDevice, SIGNAL(readyRead()), this, SLOT(getNextAudioChunk()), Qt::UniqueConnection);
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//audioInput->setNotifyInterval(setup.blockSize/2);
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//connect(audioInput, SIGNAL(notify()), this, SLOT(getNextAudioChunk()), Qt::UniqueConnection);
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}
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else {
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// Buffer size must be set before audio is started.
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#ifdef Q_OS_WIN
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audioOutput->setBufferSize(outFormat.bytesForDuration(setup.latency * 100));
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#else
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audioOutput->setBufferSize(outFormat.bytesForDuration(setup.latency * 1000));
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#endif
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audioDevice = audioOutput->start();
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connect(audioOutput, SIGNAL(destroyed()), audioDevice, SLOT(deleteLater()), Qt::UniqueConnection);
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}
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if (!audioDevice) {
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio device failed to start()";
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return;
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}
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}
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void audioHandler::stop()
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{
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "stop() running";
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if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) {
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// Stop audio output
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audioOutput->stop();
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}
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if (audioInput != Q_NULLPTR && audioInput->state() != QAudio::StoppedState) {
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// Stop audio output
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audioInput->stop();
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}
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audioDevice = Q_NULLPTR;
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}
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void audioHandler::setVolume(unsigned char volume)
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{
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this->volume = audiopot[volume];
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "setVolume: " << volume << "(" << this->volume << ")";
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}
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void audioHandler::incomingAudio(audioPacket packet)
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{
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if (audioDevice != Q_NULLPTR && packet.data.size() > 0) {
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packet.volume = volume;
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emit sendToConverter(packet);
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}
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return;
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}
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void audioHandler::convertedOutput(audioPacket packet) {
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if (packet.data.size() > 0 ) {
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currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audioOutput->bufferSize() - audioOutput->bytesFree()) / 1000);
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if (audioDevice != Q_NULLPTR) {
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if (audioDevice->write(packet.data) < packet.data.size()) {
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Buffer full!";
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isOverrun=true;
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} else {
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isOverrun = false;
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}
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if (lastReceived.msecsTo(QTime::currentTime()) > 100) {
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Time since last audio packet" << lastReceived.msecsTo(QTime::currentTime()) << "Expected around" << setup.blockSize;
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}
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lastReceived = QTime::currentTime();
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}
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/*if ((packet.seq > lastSentSeq + 1) && (setup.codec == 0x40 || setup.codec == 0x80)) {
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Attempting FEC on packet" << packet.seq << "as last is" << lastSentSeq;
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lastSentSeq = packet.seq;
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incomingAudio(packet); // Call myself again to run the packet a second time (FEC)
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}
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*/
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lastSentSeq = packet.seq;
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amplitude = packet.amplitudePeak;
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computeLevels();
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emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun, isOverrun);
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}
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}
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void audioHandler::computeLevels()
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{
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if(levelMean)
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levelMean[(levelPosition++)%levelSize] = amplitude * 255;
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}
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void audioHandler::getNextAudioChunk()
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{
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if (audioDevice) {
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tempBuf.data.append(audioDevice->readAll());
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}
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if (tempBuf.data.length() >= outFormat.bytesForDuration(setup.blockSize * 1000)) {
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audioPacket packet;
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packet.time = QTime::currentTime();
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packet.sent = 0;
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packet.volume = volume;
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memcpy(&packet.guid, setup.guid, GUIDLEN);
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//QTime startProcessing = QTime::currentTime();
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packet.data.clear();
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packet.data = tempBuf.data.mid(0, outFormat.bytesForDuration(setup.blockSize * 1000));
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tempBuf.data.remove(0, outFormat.bytesForDuration(setup.blockSize * 1000));
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emit sendToConverter(packet);
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}
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/* If there is still enough data in the buffer, call myself again in 20ms */
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if (tempBuf.data.length() >= outFormat.bytesForDuration(setup.blockSize * 1000)) {
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QTimer::singleShot(setup.blockSize, this, &audioHandler::getNextAudioChunk);
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}
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return;
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}
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void audioHandler::convertedInput(audioPacket audio)
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{
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if (audio.data.size() > 0) {
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emit haveAudioData(audio);
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if (lastReceived.msecsTo(QTime::currentTime()) > 100) {
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Time since last audio packet" << lastReceived.msecsTo(QTime::currentTime()) << "Expected around" << setup.blockSize ;
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}
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lastReceived = QTime::currentTime();
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amplitude = audio.amplitudePeak;
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emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun, isOverrun);
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}
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}
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void audioHandler::changeLatency(const quint16 newSize)
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{
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setup.latency = newSize;
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if (!setup.isinput) {
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stop();
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start();
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}
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Configured latency: " << setup.latency << "Buffer Duration:" << outFormat.durationForBytes(audioOutput->bufferSize())/1000 << "ms";
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}
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int audioHandler::getLatency()
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{
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return currentLatency;
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}
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quint16 audioHandler::getAmplitude()
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{
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return static_cast<quint16>(amplitude * 255.0);
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}
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void audioHandler::stateChanged(QAudio::State state)
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{
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// Process the state
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switch (state)
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{
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case QAudio::IdleState:
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{
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isUnderrun = true;
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if (underTimer->isActive()) {
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underTimer->stop();
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}
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break;
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}
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case QAudio::ActiveState:
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{
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//qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio started!";
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if (!underTimer->isActive()) {
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underTimer->start(500);
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}
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break;
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}
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case QAudio::SuspendedState:
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{
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break;
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}
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case QAudio::StoppedState:
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{
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break;
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}
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default: {
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}
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break;
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}
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}
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void audioHandler::clearUnderrun()
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{
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isUnderrun = false;
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}
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