kopia lustrzana https://gitlab.com/eliggett/wfview
515 wiersze
16 KiB
C++
515 wiersze
16 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)
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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 (isInitialized) {
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stop();
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}
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if (ringBuf != Q_NULLPTR) {
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delete ringBuf;
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}
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if (audioInput != Q_NULLPTR) {
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audioInput = Q_NULLPTR;
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delete audioInput;
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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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if (resampler != Q_NULLPTR) {
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speex_resampler_destroy(resampler);
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qDebug(logAudio()) << "Resampler closed";
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}
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if (encoder != Q_NULLPTR) {
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qInfo(logAudio()) << "Destroying opus encoder";
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opus_encoder_destroy(encoder);
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}
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if (decoder != Q_NULLPTR) {
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qInfo(logAudio()) << "Destroying opus decoder";
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opus_decoder_destroy(decoder);
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}
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}
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bool audioHandler::init(audioSetup setupIn)
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{
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if (isInitialized) {
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return false;
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}
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/*
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0x01 uLaw 1ch 8bit
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0x02 PCM 1ch 8bit
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0x04 PCM 1ch 16bit
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0x08 PCM 2ch 8bit
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0x10 PCM 2ch 16bit
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0x20 uLaw 2ch 8bit
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*/
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setup = setupIn;
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setup.format.setChannelCount(1);
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setup.format.setSampleSize(8);
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setup.format.setSampleType(QAudioFormat::UnSignedInt);
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "audio handler starting:" << setup.name;
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if (setup.codec == 0x01 || setup.codec == 0x20) {
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setup.ulaw = true;
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}
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if (setup.codec == 0x08 || setup.codec == 0x10 || setup.codec == 0x20 || setup.codec == 0x80) {
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setup.format.setChannelCount(2);
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}
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if (setup.codec == 0x04 || setup.codec == 0x10 || setup.codec == 0x40 || setup.codec == 0x80) {
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setup.format.setSampleSize(16);
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setup.format.setSampleType(QAudioFormat::SignedInt);
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}
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qDebug(logAudio()) << "Creating" << (setup.isinput ? "Input" : "Output") << "audio device:" << setup.name <<
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", bits" << setup.format.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" << setup.format.channelCount() <<
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", resampleQuality" << setup.resampleQuality <<
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", samplerate" << setup.format.sampleRate() <<
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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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}
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format.setSampleSize(16);
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format.setChannelCount(2);
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format.setSampleRate(INTERNAL_SAMPLE_RATE);
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format.setCodec("audio/pcm");
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format.setByteOrder(QAudioFormat::LittleEndian);
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format.setSampleType(QAudioFormat::SignedInt);
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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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else if (!setup.port.isFormatSupported(format))
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{
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Format not supported, choosing nearest supported format - which may not work!";
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format=setup.port.nearestFormat(format);
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}
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if (format.channelCount() > 2) {
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format.setChannelCount(2);
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}
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else if (format.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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devChannels = format.channelCount();
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nativeSampleRate = format.sampleRate();
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Internal: sample rate" << format.sampleRate() << "channel count" << format.channelCount();
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// We "hopefully" now have a valid format that is supported so try connecting
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if (setup.isinput) {
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audioInput = new QAudioInput(setup.port, format, this);
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//connect(audioInput, SIGNAL(notify()), SLOT(notified()));
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isInitialized = true;
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}
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else {
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audioOutput = new QAudioOutput(setup.port, format, this);
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audioOutput->setBufferSize(getAudioSize(setup.latency, format));
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isInitialized = true;
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}
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// Setup resampler and opus if they are needed.
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int resample_error = 0;
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int opus_err = 0;
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if (setup.isinput) {
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resampler = wf_resampler_init(devChannels, nativeSampleRate, setup.format.sampleRate(), setup.resampleQuality, &resample_error);
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if (setup.codec == 0x40 || setup.codec == 0x80) {
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// Opus codec
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encoder = opus_encoder_create(setup.format.sampleRate(), setup.format.channelCount(), OPUS_APPLICATION_AUDIO, &opus_err);
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opus_encoder_ctl(encoder, OPUS_SET_LSB_DEPTH(16));
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opus_encoder_ctl(encoder, OPUS_SET_INBAND_FEC(1));
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opus_encoder_ctl(encoder, OPUS_SET_DTX(1));
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opus_encoder_ctl(encoder, OPUS_SET_PACKET_LOSS_PERC(5));
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qInfo(logAudio()) << "Creating opus encoder: " << opus_strerror(opus_err);
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}
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}
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else {
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//resampBufs = new r8b::CFixedBuffer<double>[format.channelCount()];
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//resamps = new r8b::CPtrKeeper<r8b::CDSPResampler24*>[format.channelCount()];
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resampler = wf_resampler_init(devChannels, setup.format.sampleRate(), this->nativeSampleRate, setup.resampleQuality, &resample_error);
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if (setup.codec == 0x40 || setup.codec == 0x80) {
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// Opus codec
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decoder = opus_decoder_create(setup.format.sampleRate(), setup.format.sampleRate(), &opus_err);
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qInfo(logAudio()) << "Creating opus decoder: " << opus_strerror(opus_err);
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}
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}
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unsigned int ratioNum;
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unsigned int ratioDen;
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wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
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resampleRatio = static_cast<double>(ratioDen) / ratioNum;
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "wf_resampler_init() returned: " << resample_error << " resampleRatio: " << resampleRatio;
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "thread id" << QThread::currentThreadId();
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this->start();
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return isInitialized;
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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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audioDevice = audioInput->start();
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connect(audioInput, &QAudioOutput::destroyed, audioDevice, &QIODevice::deleteLater, Qt::UniqueConnection);
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connect(audioDevice, &QIODevice::destroyed, this, &QAudioInput::deleteLater, Qt::UniqueConnection);
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}
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else {
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audioDevice = audioOutput->start();
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connect(audioOutput, &QAudioOutput::destroyed, audioDevice, &QIODevice::deleteLater, Qt::UniqueConnection);
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connect(audioDevice, &QIODevice::destroyed, this, &QAudioOutput::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::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 inPacket)
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{
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// No point buffering audio until stream is actually running.
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// Regardless of the radio stream format, the buffered audio will ALWAYS be
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// 16bit sample interleaved stereo 48K (or whatever the native sample rate is)
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audioPacket livePacket = inPacket;
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if (setup.codec == 0x40 || setup.codec == 0x80) {
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/* Opus data */
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unsigned char* in = (unsigned char*)inPacket.data.data();
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/* Decode the frame. */
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QByteArray outPacket((setup.format.sampleRate() / 50) * sizeof(qint16) * setup.format.channelCount(), (char)0xff); // Preset the output buffer size.
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qint16* out = (qint16*)outPacket.data();
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int nSamples = opus_packet_get_nb_samples(in, livePacket.data.size(),setup.format.sampleRate());
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if (nSamples == -1) {
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// No opus data yet?
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return;
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}
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else if (nSamples != setup.format.sampleRate() / 50)
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{
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qDebug(logAudio()) << "Opus nSamples=" << nSamples << " expected:" << (setup.format.sampleRate() / 50);
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return;
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}
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if (livePacket.seq > lastSentSeq + 1) {
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nSamples = opus_decode(decoder, in, livePacket.data.size(), out, (setup.format.sampleRate() / 50), 1);
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}
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else {
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nSamples = opus_decode(decoder, in, livePacket.data.size(), out, (setup.format.sampleRate() / 50), 0);
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}
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if (nSamples < 0)
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{
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decode failed:" << opus_strerror(nSamples) << "packet size" << livePacket.data.length();
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return;
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}
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else {
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if (int(nSamples * sizeof(qint16) * setup.format.channelCount()) != outPacket.size())
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{
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decoder mismatch: nBytes:" << nSamples * sizeof(qint16) * setup.format.channelCount() << "outPacket:" << outPacket.size();
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outPacket.resize(nSamples * sizeof(qint16) * setup.format.channelCount());
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}
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//qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus decoded" << livePacket.data.size() << "bytes, into" << outPacket.length() << "bytes";
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livePacket.data.clear();
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livePacket.data = outPacket; // Replace incoming data with converted.
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}
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}
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// Process uLaw.
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if (setup.ulaw)
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{
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// Current packet is 8bit so need to create a new buffer that is 16bit
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QByteArray outPacket((int)livePacket.data.length() * 2, (char)0xff);
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qint16* out = (qint16*)outPacket.data();
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for (int f = 0; f < livePacket.data.length(); f++)
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{
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*out++ = ulaw_decode[(quint8)livePacket.data[f]];
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}
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livePacket.data.clear();
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livePacket.data = outPacket; // Replace incoming data with converted.
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setup.format.setSampleSize(16);
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setup.format.setSampleType(QAudioFormat::SignedInt);
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// Buffer now contains 16bit signed samples.
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}
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if (!livePacket.data.isEmpty()) {
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Eigen::VectorXf samplesF;
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if (setup.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
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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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// Set the max amplitude found in the vector
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amplitude = samplesF.array().abs().maxCoeff();
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// Set the volume
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samplesF *= volume;
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// Convert mono to stereo
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if (setup.format.channelCount() == 1) {
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Eigen::VectorXf samplesTemp(samplesF.size() * 2);
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Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data(), samplesF.size()) = samplesF;
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Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data() + 1, samplesF.size()) = samplesF;
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samplesF = samplesTemp;
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}
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if (format.sampleType() == QAudioFormat::SignedInt)
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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
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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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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 = ((livePacket.data.length() / 2 / devChannels) * resampleRatio);
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quint32 inFrames = (livePacket.data.length() / 2 / devChannels);
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QByteArray outPacket(outFrames * 4, (char)0xff); // Preset the output buffer size.
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const qint16* in = (qint16*)livePacket.data.constData();
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qint16* out = (qint16*)outPacket.data();
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int err = 0;
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err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
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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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livePacket.data.clear();
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livePacket.data = outPacket; // Replace incoming data with converted.
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}
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//qDebug(logAudio()) << "Adding packet to buffer:" << livePacket.seq << ": " << livePacket.data.length();
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currentLatency = livePacket.time.msecsTo(QTime::currentTime()) + getAudioDuration(audioOutput->bufferSize()-audioOutput->bytesFree(),format);
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if (audioDevice != Q_NULLPTR) {
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audioDevice->write(livePacket.data);
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}
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if ((inPacket.seq > lastSentSeq + 1) && (setup.codec == 0x40 || setup.codec == 0x80)) {
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Attempting FEC on packet" << inPacket.seq << "as last is" << lastSentSeq;
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lastSentSeq = inPacket.seq;
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incomingAudio(inPacket); // Call myself again to run the packet a second time (FEC)
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}
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lastSentSeq = inPacket.seq;
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}
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return;
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}
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void audioHandler::changeLatency(const quint16 newSize)
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{
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << setup.latency;
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setup.latency = newSize;
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if (!setup.isinput) {
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stop();
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audioOutput->setBufferSize(getAudioSize(setup.latency, format));
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start();
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}
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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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void audioHandler::getNextAudioChunk(QByteArray& ret)
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{
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audioPacket packet;
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packet.sent = 0;
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if (audioDevice != Q_NULLPTR) {
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packet.data = audioDevice->readAll();
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}
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if (packet.data.length() > 0)
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{
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// Packet will arrive as stereo interleaved 16bit 48K
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if (resampleRatio != 1.0)
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{
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quint32 outFrames = ((packet.data.length() / 2 / devChannels) * resampleRatio);
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quint32 inFrames = (packet.data.length() / 2 / devChannels);
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QByteArray outPacket((int)outFrames * 2 * devChannels, (char)0xff);
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const qint16* in = (qint16*)packet.data.constData();
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qint16* out = (qint16*)outPacket.data();
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int err = 0;
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err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
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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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packet.data.clear();
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packet.data = outPacket; // Copy output packet back to input buffer.
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}
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//qDebug(logAudio()) << "Now resampled, length" << packet.data.length();
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int tempAmplitude = 0;
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// Do we need to convert mono to stereo?
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if (setup.format.channelCount() == 1 && devChannels > 1)
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{
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// Strip out right channel?
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QByteArray outPacket(packet.data.length()/2, (char)0xff);
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const qint16* in = (qint16*)packet.data.constData();
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qint16* out = (qint16*)outPacket.data();
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for (int f = 0; f < outPacket.length()/2; f++)
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{
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tempAmplitude = qMax(tempAmplitude, (int)(abs(*in) / 256));
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*out++ = *in++;
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in++; // Skip each even channel.
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}
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packet.data.clear();
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packet.data = outPacket; // Copy output packet back to input buffer.
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}
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//qDebug(logAudio()) << "Now mono, length" << packet.data.length();
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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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qint16* in = (qint16*)packet.data.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(encoder, in, (setup.format.sampleRate() / 50), 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);
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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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packet.data.clear();
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packet.data = outPacket; // Replace incoming data with converted.
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}
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}
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else if (setup.format.sampleSize() == 8)
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{
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// Do we need to convert 16-bit to 8-bit?
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QByteArray outPacket((int)packet.data.length() / 2, (char)0xff);
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qint16* in = (qint16*)packet.data.data();
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|
for (int f = 0; f < outPacket.length(); f++)
|
|
{
|
|
qint16 sample = *in++;
|
|
if (setup.ulaw) {
|
|
int sign = (sample >> 8) & 0x80;
|
|
if (sign)
|
|
sample = (short)-sample;
|
|
if (sample > cClip)
|
|
sample = cClip;
|
|
sample = (short)(sample + cBias);
|
|
int exponent = (int)MuLawCompressTable[(sample >> 7) & 0xFF];
|
|
int mantissa = (sample >> (exponent + 3)) & 0x0F;
|
|
int compressedByte = ~(sign | (exponent << 4) | mantissa);
|
|
outPacket[f] = (quint8)compressedByte;
|
|
}
|
|
else {
|
|
int compressedByte = (((sample + 32768) >> 8) & 0xff);
|
|
outPacket[f] = (quint8)compressedByte;
|
|
}
|
|
tempAmplitude = qMax(tempAmplitude, abs(outPacket[f]));
|
|
}
|
|
packet.data.clear();
|
|
packet.data = outPacket; // Copy output packet back to input buffer.
|
|
}
|
|
amplitude = tempAmplitude;
|
|
|
|
ret = packet.data;
|
|
|
|
}
|
|
|
|
return;
|
|
|
|
}
|
|
|
|
|
|
void audioHandler::stop()
|
|
{
|
|
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "stop() running";
|
|
|
|
if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) {
|
|
// Stop audio output
|
|
audioOutput->stop();
|
|
}
|
|
|
|
if (audioInput != Q_NULLPTR && audioInput->state() != QAudio::StoppedState) {
|
|
// Stop audio output
|
|
audioInput->stop();
|
|
}
|
|
audioDevice = Q_NULLPTR;
|
|
}
|
|
|
|
|
|
quint16 audioHandler::getAmplitude()
|
|
{
|
|
return static_cast<quint16>(amplitude * 255.0);
|
|
}
|
|
|