kopia lustrzana https://gitlab.com/eliggett/wfview
Try a different method
Phil Taylor
rodzic
e068bc5745
commit
0a711e5c1f
313
audiohandler.cpp
313
audiohandler.cpp
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@ -31,12 +31,6 @@ audioHandler::~audioHandler()
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audioOutput = Q_NULLPTR;
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}
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if (audioTimer != Q_NULLPTR) {
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audioTimer->stop();
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delete audioTimer;
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audioTimer = Q_NULLPTR;
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}
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if (resampler != Q_NULLPTR) {
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speex_resampler_destroy(resampler);
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@ -109,8 +103,6 @@ bool audioHandler::init(audioSetup setupIn)
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", uLaw" << setup.ulaw;
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tempBuf.sent = 0;
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if(!setup.isinput)
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{
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this->setVolume(setup.localAFgain);
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@ -135,9 +127,6 @@ bool audioHandler::init(audioSetup setupIn)
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if (setup.isinput) {
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audioInput = new QAudioInput(setup.port, format, this);
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Starting audio timer";
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audioTimer = new QTimer();
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audioTimer->setTimerType(Qt::PreciseTimer);
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connect(audioTimer, &QTimer::timeout, this, &audioHandler::getNextAudioChunk);
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connect(audioInput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State)));
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}
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@ -196,8 +185,8 @@ void audioHandler::start()
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if (setup.isinput) {
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audioDevice = audioInput->start();
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connect(audioInput, &QAudioOutput::destroyed, audioDevice, &QIODevice::deleteLater, Qt::UniqueConnection);
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audioTimer->start(setup.blockSize);
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connect(audioInput, &QAudioInput::destroyed, audioDevice, &QIODevice::deleteLater, Qt::UniqueConnection);
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connect(audioDevice, &QIODevice::readyRead, this, &audioHandler::getNextAudioChunk);
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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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@ -238,7 +227,7 @@ void audioHandler::setVolume(unsigned char volume)
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void audioHandler::incomingAudio(audioPacket inPacket)
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{
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audioPacket livePacket = inPacket;
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// Process uLaw.
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if (setup.ulaw)
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@ -408,175 +397,171 @@ void audioHandler::incomingAudio(audioPacket inPacket)
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void audioHandler::getNextAudioChunk()
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{
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tempBuf.data = tempBuf.data + audioDevice->readAll();
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if (tempBuf.data.length() <= format.bytesForDuration(setup.blockSize * 1000)) {
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return;
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}
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audioPacket livePacket;
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livePacket.time= QTime::currentTime();
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livePacket.sent = 0;
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memcpy(&livePacket.guid, setup.guid, GUIDLEN);
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if (audioInput != Q_NULLPTR && audioDevice != Q_NULLPTR) {
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if (setup.codec == 0x40 || setup.codec == 0x80) {
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livePacket.data = audioDevice->read(format.bytesForDuration(setup.blockSize*1000));
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}
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else
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livePacket.data = tempBuf.data.left(format.bytesForDuration(setup.blockSize*1000));
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tempBuf.data.remove(0, format.bytesForDuration(setup.blockSize * 1000));
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//qDebug(logAudio()) << "Sending audio len" << livePacket.data.length() << "remaining" << tempBuf.data.length();
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if (livePacket.data.length() > 0)
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{
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Eigen::VectorXf samplesF;
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if (format.sampleType() == QAudioFormat::SignedInt && format.sampleSize() == 32)
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{
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livePacket.data = audioDevice->readAll();
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VectorXint32 samplesI = Eigen::Map<VectorXint32>(reinterpret_cast<qint32*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(qint32)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint32>::max());
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}
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if (livePacket.data.length() > 0)
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else if (format.sampleType() == QAudioFormat::SignedInt && format.sampleSize() == 16)
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{
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Eigen::VectorXf samplesF;
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if (format.sampleType() == QAudioFormat::SignedInt && format.sampleSize() == 32)
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{
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VectorXint32 samplesI = Eigen::Map<VectorXint32>(reinterpret_cast<qint32*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(qint32)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint32>::max());
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}
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else if (format.sampleType() == QAudioFormat::SignedInt && format.sampleSize() == 16)
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{
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VectorXint16 samplesI = Eigen::Map<VectorXint16>(reinterpret_cast<qint16*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(qint16)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint16>::max());
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}
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else if (format.sampleType() == QAudioFormat::UnSignedInt && format.sampleSize() == 8)
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{
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VectorXuint8 samplesI = Eigen::Map<VectorXuint8>(reinterpret_cast<quint8*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(quint8)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<quint8>::max());;
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}
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else if (format.sampleType() == QAudioFormat::SignedInt && format.sampleSize() == 8)
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{
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VectorXint8 samplesI = Eigen::Map<VectorXint8>(reinterpret_cast<qint8*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(qint8)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint8>::max());;
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}
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else if (format.sampleType() == QAudioFormat::Float)
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{
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(float)));
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}
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else {
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unsupported Sample Type:" << format.sampleType() << format.sampleSize();
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}
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VectorXint16 samplesI = Eigen::Map<VectorXint16>(reinterpret_cast<qint16*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(qint16)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint16>::max());
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}
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else if (format.sampleType() == QAudioFormat::UnSignedInt && format.sampleSize() == 8)
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{
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VectorXuint8 samplesI = Eigen::Map<VectorXuint8>(reinterpret_cast<quint8*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(quint8)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<quint8>::max());;
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}
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else if (format.sampleType() == QAudioFormat::SignedInt && format.sampleSize() == 8)
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{
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VectorXint8 samplesI = Eigen::Map<VectorXint8>(reinterpret_cast<qint8*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(qint8)));
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samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint8>::max());;
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}
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else if (format.sampleType() == QAudioFormat::Float)
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{
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(float)));
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}
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else {
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unsupported Sample Type:" << format.sampleType() << format.sampleSize();
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}
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/* samplesF is now an Eigen Vector of the current samples in float format */
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/* samplesF is now an Eigen Vector of the current samples in float format */
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// Set the max amplitude found in the vector
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if (samplesF.size() > 0) {
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amplitude = samplesF.array().abs().maxCoeff();
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// Channel count should now match the device that audio is going to (rig)
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// Set the max amplitude found in the vector
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if (samplesF.size() > 0) {
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amplitude = samplesF.array().abs().maxCoeff();
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// Channel count should now match the device that audio is going to (rig)
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if (resampleRatio != 1.0) {
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if (resampleRatio != 1.0) {
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// We need to resample
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// We have a stereo 16bit stream.
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quint32 outFrames = ((samplesF.size() / format.channelCount()) * resampleRatio);
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quint32 inFrames = (samplesF.size() / format.channelCount());
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// We need to resample
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// We have a stereo 16bit stream.
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quint32 outFrames = ((samplesF.size() / format.channelCount()) * resampleRatio);
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quint32 inFrames = (samplesF.size() / format.channelCount());
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QByteArray outPacket(outFrames * format.channelCount() * sizeof(float), (char)0xff); // Preset the output buffer size.
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const float* in = (float*)samplesF.data();
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float* out = (float*)outPacket.data();
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QByteArray outPacket(outFrames * format.channelCount() * sizeof(float), (char)0xff); // Preset the output buffer size.
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const float* in = (float*)samplesF.data();
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float* out = (float*)outPacket.data();
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int err = 0;
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if (format.channelCount() == 1) {
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err = wf_resampler_process_float(resampler, 0, in, &inFrames, out, &outFrames);
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}
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else {
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err = wf_resampler_process_interleaved_float(resampler, in, &inFrames, out, &outFrames);
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}
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if (err) {
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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}
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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// If we need to drop one of the audio channels, do it now
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if (format.channelCount() == 2 && setup.format.channelCount() == 1) {
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Eigen::VectorXf samplesTemp(samplesF.size() / 2);
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samplesTemp = Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesF.data(), samplesF.size() / 2);
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samplesF = samplesTemp;
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}
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if (setup.codec == 0x40 || setup.codec == 0x80)
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{
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//Are we using the opus codec?
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float* in = (float*)samplesF.data();
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/* Encode the frame. */
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QByteArray outPacket(1275, (char)0xff); // Preset the output buffer size to MAXIMUM possible Opus frame size
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unsigned char* out = (unsigned char*)outPacket.data();
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int nbBytes = opus_encode_float(encoder, in, (samplesF.size()/setup.format.channelCount()), out, outPacket.length());
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if (nbBytes < 0)
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{
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus encode failed:" << opus_strerror(nbBytes) << "Num Samples:" << samplesF.size();
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return;
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}
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else {
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outPacket.resize(nbBytes);
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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}
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if (setup.format.sampleType() == QAudioFormat::SignedInt && setup.format.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint8>::max());
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VectorXint8 samplesI = samplesITemp.cast<qint8>();
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint8)));
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}
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else if (setup.format.sampleType() == QAudioFormat::UnSignedInt && setup.format.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<quint8>::max());
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VectorXuint8 samplesI = samplesITemp.cast<quint8>();
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(quint8)));
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}
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else if (setup.format.sampleType() == QAudioFormat::SignedInt && setup.format.sampleSize() == 16)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint16>::max());
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VectorXint16 samplesI = samplesITemp.cast<qint16>();
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint16)));
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}
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else if (setup.format.sampleType() == QAudioFormat::SignedInt && setup.format.sampleSize() == 32)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint32>::max());
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VectorXint32 samplesI = samplesITemp.cast<qint32>();
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint32)));
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}
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else if (setup.format.sampleType() == QAudioFormat::Float)
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{
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
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int err = 0;
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if (format.channelCount() == 1) {
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err = wf_resampler_process_float(resampler, 0, in, &inFrames, out, &outFrames);
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}
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else {
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unsupported Sample Type:" << format.sampleType();
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err = wf_resampler_process_interleaved_float(resampler, in, &inFrames, out, &outFrames);
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}
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/* Need to find a floating point uLaw encoder!*/
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if (setup.ulaw)
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{
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QByteArray outPacket((int)livePacket.data.length() / 2, (char)0xff);
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qint16* in = (qint16*)livePacket.data.data();
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for (int f = 0; f < outPacket.length(); f++)
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{
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qint16 sample = *in++;
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if (setup.ulaw) {
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int sign = (sample >> 8) & 0x80;
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if (sign)
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sample = (short)-sample;
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if (sample > cClip)
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sample = cClip;
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sample = (short)(sample + cBias);
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int exponent = (int)MuLawCompressTable[(sample >> 7) & 0xFF];
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int mantissa = (sample >> (exponent + 3)) & 0x0F;
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int compressedByte = ~(sign | (exponent << 4) | mantissa);
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outPacket[f] = (quint8)compressedByte;
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}
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}
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livePacket.data.clear();
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livePacket.data = outPacket; // Copy output packet back to input buffer.
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if (err) {
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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}
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emit haveAudioData(livePacket);
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//ret = livePacket.data;
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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}
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emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun);
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}
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// If we need to drop one of the audio channels, do it now
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if (format.channelCount() == 2 && setup.format.channelCount() == 1) {
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Eigen::VectorXf samplesTemp(samplesF.size() / 2);
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samplesTemp = Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesF.data(), samplesF.size() / 2);
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samplesF = samplesTemp;
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}
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if (setup.codec == 0x40 || setup.codec == 0x80)
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{
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//Are we using the opus codec?
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float* in = (float*)samplesF.data();
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/* Encode the frame. */
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QByteArray outPacket(1275, (char)0xff); // Preset the output buffer size to MAXIMUM possible Opus frame size
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unsigned char* out = (unsigned char*)outPacket.data();
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int nbBytes = opus_encode_float(encoder, in, (samplesF.size()/setup.format.channelCount()), out, outPacket.length());
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if (nbBytes < 0)
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{
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus encode failed:" << opus_strerror(nbBytes) << "Num Samples:" << samplesF.size();
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return;
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}
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else {
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outPacket.resize(nbBytes);
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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}
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if (setup.format.sampleType() == QAudioFormat::SignedInt && setup.format.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint8>::max());
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VectorXint8 samplesI = samplesITemp.cast<qint8>();
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint8)));
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}
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else if (setup.format.sampleType() == QAudioFormat::UnSignedInt && setup.format.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<quint8>::max());
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VectorXuint8 samplesI = samplesITemp.cast<quint8>();
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(quint8)));
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}
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else if (setup.format.sampleType() == QAudioFormat::SignedInt && setup.format.sampleSize() == 16)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint16>::max());
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VectorXint16 samplesI = samplesITemp.cast<qint16>();
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint16)));
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}
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else if (setup.format.sampleType() == QAudioFormat::SignedInt && setup.format.sampleSize() == 32)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint32>::max());
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VectorXint32 samplesI = samplesITemp.cast<qint32>();
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint32)));
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}
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else if (setup.format.sampleType() == QAudioFormat::Float)
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{
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livePacket.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
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}
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else {
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unsupported Sample Type:" << format.sampleType();
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}
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/* Need to find a floating point uLaw encoder!*/
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if (setup.ulaw)
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{
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QByteArray outPacket((int)livePacket.data.length() / 2, (char)0xff);
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qint16* in = (qint16*)livePacket.data.data();
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for (int f = 0; f < outPacket.length(); f++)
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{
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qint16 sample = *in++;
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if (setup.ulaw) {
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int sign = (sample >> 8) & 0x80;
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if (sign)
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sample = (short)-sample;
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if (sample > cClip)
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sample = cClip;
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sample = (short)(sample + cBias);
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int exponent = (int)MuLawCompressTable[(sample >> 7) & 0xFF];
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int mantissa = (sample >> (exponent + 3)) & 0x0F;
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int compressedByte = ~(sign | (exponent << 4) | mantissa);
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outPacket[f] = (quint8)compressedByte;
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}
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}
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livePacket.data.clear();
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livePacket.data = outPacket; // Copy output packet back to input buffer.
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}
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emit haveAudioData(livePacket);
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//ret = livePacket.data;
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}
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}
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emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun);
|
||||
|
||||
return;
|
||||
|
||||
|
|
|
|||
|
|
@ -115,7 +115,6 @@ private:
|
|||
QAudioOutput* audioOutput=Q_NULLPTR;
|
||||
QAudioInput* audioInput=Q_NULLPTR;
|
||||
QIODevice* audioDevice=Q_NULLPTR;
|
||||
QTimer* audioTimer = Q_NULLPTR;
|
||||
QAudioFormat format;
|
||||
QAudioDeviceInfo deviceInfo;
|
||||
SpeexResamplerState* resampler = Q_NULLPTR;
|
||||
|
|
|
|||
Ładowanie…
Reference in New Issue