kopia lustrzana https://gitlab.com/eliggett/wfview
Add tx audio
Phil Taylor
rodzic
f726073e22
commit
0c7892bd82
191
audiohandler.cpp
191
audiohandler.cpp
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@ -60,7 +60,6 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
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else {
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aParams.deviceId = audio.getDefaultOutputDevice();
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}
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aParams.nChannels = 2; // Internally this is always 2 channels
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aParams.firstChannel = 0;
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try {
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@ -99,6 +98,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
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if (isInput) {
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resampler = wf_resampler_init(radioChannels, INTERNAL_SAMPLE_RATE, samplerate, resampleQuality, &resample_error);
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try {
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aParams.nChannels = 1; // Internally this is always 2 channels
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audio.openStream(NULL, &aParams, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &this->chunkSize, &staticWrite, this);
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audio.startStream();
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}
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@ -112,6 +112,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
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resampler = wf_resampler_init(radioChannels, samplerate, INTERNAL_SAMPLE_RATE, resampleQuality, &resample_error);
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try {
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unsigned int length = chunkSize / 2;
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aParams.nChannels = 2; // Internally this is always 2 channels
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audio.openStream(&aParams, NULL, RTAUDIO_SINT16, INTERNAL_SAMPLE_RATE, &length, &staticRead, this);
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audio.startStream();
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}
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@ -131,6 +132,7 @@ bool audioHandler::init(const quint8 bits, const quint8 channels, const quint16
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void audioHandler::setVolume(unsigned char volume)
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{
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "setVolume: " << volume << "(" << (qreal)(volume/255.0) << ")";
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this->volume = (qreal)(volume / 255.0);
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}
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@ -217,47 +219,29 @@ int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int n
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int audioHandler::writeData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
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{
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int sentlen = 0;
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/*
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QMutexLocker locker(&mutex);
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audioPacket* current;
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while (sentlen < len) {
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if (!audioBuffer.isEmpty())
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unsigned int nBytes = nFrames * 2; // This is ALWAYS 2 bytes per sample and 1 channels
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const char* data = (const char*)inputBuffer;
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while (sentlen < nBytes) {
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if (tempBuf.sent != chunkSize)
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{
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if (audioBuffer.last().sent == chunkSize)
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int send = qMin((int)(nBytes - sentlen), (int)chunkSize - tempBuf.sent);
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tempBuf.data.append(QByteArray::fromRawData(data + sentlen, send));
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sentlen = sentlen + send;
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tempBuf.seq = 0; // Not used in TX
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tempBuf.time = QTime::currentTime();
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tempBuf.sent = tempBuf.data.length();
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}
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else {
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if (!ringBuf->try_write(tempBuf))
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{
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audioBuffer.append(audioPacket());
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audioBuffer.last().sent = 0;
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qDebug(logAudio()) << "outgoing audio buffer full!";
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}
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tempBuf.data.clear();
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tempBuf.sent = 0;
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}
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else
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{
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audioBuffer.append(audioPacket());
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audioBuffer.last().sent = 0;
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}
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current = &audioBuffer.last();
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int send = qMin((int)(len - sentlen), (int)chunkSize - current->sent);
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current->data.append(QByteArray::fromRawData(data + sentlen, send));
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sentlen = sentlen + send;
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current->seq = 0; // Not used in TX
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current->time = QTime::currentTime();
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current->sent = current->data.length();
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if (current->sent == chunkSize)
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{
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chunkAvailable = true;
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}
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else if (audioBuffer.length() <= 1 && current->sent != chunkSize) {
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chunkAvailable = false;
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}
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}
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*/
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return (sentlen); // Always return the same number as we received
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return 0;
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}
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qint64 audioHandler::bytesAvailable() const
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@ -338,21 +322,28 @@ int audioHandler::incomingAudio(audioPacket data)
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data.data = outPacket; // Replace incoming data with converted.
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}
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if (radioChannels == 1)
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{
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// Convert to stereo
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if (radioChannels == 1) {
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// Convert to stereo and set volume.
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QByteArray outPacket(data.data.length()*2, 0xff); // Preset the output buffer size.
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qint16* in = (qint16*)data.data.data();
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qint16* out = (qint16*)outPacket.data();
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for (int f = 0; f < data.data.length()/2; f++)
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{
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*out++ = *in;
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*out++ = *in++;
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*out++ = *in * volume;
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*out++ = *in++ * volume;
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}
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data.data.clear();
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data.data = outPacket; // Replace incoming data with converted.
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} else {
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// We already have two channels so just update volume.
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qint16* in = (qint16*)data.data.data();
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for (int f = 0; f < data.data.length() / 2; f++)
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{
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in[f] = in[f] * volume;
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}
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}
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if (!ringBuf->try_write(data))
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{
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qDebug(logAudio()) << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size();
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@ -378,90 +369,62 @@ bool audioHandler::isChunkAvailable()
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void audioHandler::getNextAudioChunk(QByteArray& ret)
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{
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/*
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if (!audioBuffer.isEmpty() && chunkAvailable)
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audioPacket packet;
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packet.sent = 0;
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if (ringBuf != Q_NULLPTR && ringBuf->try_read(packet))
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{
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QMutexLocker locker(&mutex);
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// Skip through audio buffer deleting any old entry.
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auto packet = audioBuffer.begin();
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while (packet != audioBuffer.end())
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if (ratioNum != 1)
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{
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if (packet->time.msecsTo(QTime::currentTime()) > 100) {
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//qInfo(logAudio()) << "TX Packet too old " << dec << packet->time.msecsTo(QTime::currentTime()) << "ms";
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packet = audioBuffer.erase(packet); // returns next packet
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// We need to resample (we are STILL 16 bit!)
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quint32 outFrames = ((packet.data.length() / 2) / ratioNum) / radioChannels;
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quint32 inFrames = (packet.data.length() / 2) / radioChannels;
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QByteArray inPacket((int)outFrames * 2 * radioChannels, (char)0xff);
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const qint16* in = (qint16*)packet.data.constData();
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qint16* out = (qint16*)inPacket.data();
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int err = 0;
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if (this->radioChannels == 1) {
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err = wf_resampler_process_int(resampler, 0, in, &inFrames, out, &outFrames);
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}
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else {
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if (packet->data.length() == chunkSize && ret.length() == 0)
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{
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// We now have an array of samples in the computer native format (48000)
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// If the radio sample rate is below 48000, we need to resample.
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err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
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}
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if (err) {
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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}
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//qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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//qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length();
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packet.data.clear();
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packet.data = inPacket; // Copy output packet back to input buffer.
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}
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if (ratioNum != 1)
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{
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// We need to resample (we are STILL 16 bit!)
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quint32 outFrames = ((packet->data.length() / 2) / ratioNum) / radioChannels;
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quint32 inFrames = (packet->data.length() / 2) / radioChannels;
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packet->dataout.resize(outFrames * 2 * radioChannels); // Preset the output buffer size.
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const qint16* in = (qint16*)packet->data.constData();
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qint16* out = (qint16*)packet->dataout.data();
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int err = 0;
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if (this->radioChannels == 1) {
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err = wf_resampler_process_int(resampler, 0, in, &inFrames, out, &outFrames);
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}
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else {
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err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
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}
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if (err) {
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qInfo(logAudio()) << (isInput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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}
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//qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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//qInfo(logAudio()) << "Resampler run inLen:" << packet->data.length() << " outLen:" << packet->dataout.length();
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if (radioSampleBits == 8)
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{
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packet->data = packet->dataout; // Copy output packet back to input buffer.
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packet->dataout.clear(); // Buffer MUST be cleared ready to be re-filled by the upsampling below.
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}
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}
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else if (radioSampleBits == 16) {
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// Only copy buffer if radioSampleBits is 16, as it will be handled below otherwise.
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packet->dataout = packet->data;
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}
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// Do we need to convert 16-bit to 8-bit?
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if (radioSampleBits == 8) {
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packet->dataout.resize(packet->data.length() / 2);
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qint16* in = (qint16*)packet->data.data();
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for (int f = 0; f < packet->dataout.length(); f++)
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{
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quint8 outdata = 0;
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if (isUlaw) {
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qint16 enc = qFromLittleEndian<quint16>(in + f);
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if (enc >= 0)
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outdata = ulaw_encode[enc];
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else
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outdata = 0x7f & ulaw_encode[-enc];
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}
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else {
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outdata = (quint8)(((qFromLittleEndian<qint16>(in + f) >> 8) ^ 0x80) & 0xff);
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}
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packet->dataout[f] = (char)outdata;
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}
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}
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ret = packet->dataout;
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packet = audioBuffer.erase(packet); // returns next packet
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// Do we need to convert 16-bit to 8-bit?
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if (radioSampleBits == 8) {
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QByteArray inPacket((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 < inPacket.length(); f++)
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{
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quint8 outdata = 0;
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if (isUlaw) {
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qint16 enc = qFromLittleEndian<quint16>(in + f);
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if (enc >= 0)
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outdata = ulaw_encode[enc];
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else
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outdata = 0x7f & ulaw_encode[-enc];
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}
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else {
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packet++;
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outdata = (quint8)(((qFromLittleEndian<qint16>(in + f) >> 8) ^ 0x80) & 0xff);
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}
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inPacket[f] = (char)outdata;
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}
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packet.data.clear();
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packet.data = inPacket; // Copy output packet back to input buffer.
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}
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ret = packet.data;
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}
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*/
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return;
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}
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@ -6,6 +6,7 @@
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#include <QtMultimedia/QAudioOutput>
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#include <QByteArray>
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#include <QMutex>
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#include <QtEndian>
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#include "rtaudio/RtAudio.h"
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typedef signed short MY_TYPE;
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@ -116,6 +117,7 @@ private:
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volatile bool ready = false;
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audioPacket tempBuf;
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quint16 currentLatency;
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qreal volume=1.0;
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};
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@ -858,10 +858,12 @@ void udpAudio::watchdog()
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void udpAudio::sendTxAudio()
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{
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if (txaudio && txaudio->isChunkAvailable()) {
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QByteArray audio;
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txaudio->getNextAudioChunk(audio);
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if (txaudio == Q_NULLPTR) {
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return;
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}
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QByteArray audio;
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txaudio->getNextAudioChunk(audio);
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if (audio.length() > 0) {
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int counter = 1;
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int len = 0;
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