Merge branch 'wfserver' into shuttle

half-duplex
Phil Taylor 2022-05-13 18:32:07 +01:00
commit 19c16daa06
5 zmienionych plików z 89 dodań i 46 usunięć

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@ -82,7 +82,7 @@ private:
bool isUnderrun = false;
bool isOverrun = true;
bool isOverrun = false;
bool isInitialized=false;
bool isReady = false;
bool audioBuffered = false;

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@ -38,7 +38,7 @@ bool paHandler::init(audioSetup setup)
this->setup = setup;
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "PortAudio handler starting:" << setup.name;
if (setup.portInt==-1)
if (setup.portInt == -1)
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "No audio device was found.";
return false;
@ -82,7 +82,7 @@ bool paHandler::init(audioSetup setup)
outFormat.setChannelCount(info->maxOutputChannels);
}
aParams.suggestedLatency = (float)setup.latency/1000.0f;
aParams.suggestedLatency = (float)setup.latency / 1000.0f;
outFormat.setSampleRate(info->defaultSampleRate);
aParams.sampleFormat = paFloat32;
outFormat.setSampleSize(32);
@ -91,11 +91,6 @@ bool paHandler::init(audioSetup setup)
outFormat.setCodec("audio/pcm");
if (!setup.isinput)
{
this->setVolume(setup.localAFgain);
}
if (outFormat.channelCount() > 2) {
outFormat.setChannelCount(2);
}
@ -114,7 +109,7 @@ bool paHandler::init(audioSetup setup)
if (outFormat.sampleRate() < 44100) {
outFormat.setSampleRate(48000);
}
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Selected format: SampleSize" << outFormat.sampleSize() << "Channel Count" << outFormat.channelCount() <<
"Sample Rate" << outFormat.sampleRate() << "Codec" << outFormat.codec() << "Sample Type" << outFormat.sampleType();
@ -139,10 +134,59 @@ bool paHandler::init(audioSetup setup)
aParams.hostApiSpecificStreamInfo = NULL;
// Per channel chunk size.
this->chunkSize = (outFormat.bytesForDuration(setup.blockSize*1000)/sizeof(float))*outFormat.channelCount();
this->chunkSize = (outFormat.bytesForDuration(setup.blockSize * 1000) / sizeof(float)) * outFormat.channelCount();
// Check the format is supported
if (setup.isinput) {
err = Pa_IsFormatSupported(&aParams, NULL, outFormat.sampleRate());
}
else
{
err = Pa_IsFormatSupported(NULL,&aParams, outFormat.sampleRate());
}
if (err != paNoError) {
if (err == paInvalidChannelCount)
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unsupported channel count" << aParams.channelCount;
if (aParams.channelCount == 2) {
aParams.channelCount = 1;
outFormat.setChannelCount(1);
}
else {
aParams.channelCount = 2;
outFormat.setChannelCount(2);
}
}
else if (err == paInvalidSampleRate)
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unsupported sample rate" << outFormat.sampleRate();
outFormat.setSampleRate(44100);
}
else if (err == paSampleFormatNotSupported)
{
aParams.sampleFormat = paInt16;
outFormat.setSampleType(QAudioFormat::SignedInt);
outFormat.setSampleSize(16);
}
if (setup.isinput) {
err = Pa_IsFormatSupported(&aParams, NULL, outFormat.sampleRate());
}
else
{
err = Pa_IsFormatSupported(NULL, &aParams, outFormat.sampleRate());
}
if (err != paNoError) {
qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << "Cannot find suitable format, aborting:" << Pa_GetErrorText(err);
return false;
}
}
if (setup.isinput) {
err = Pa_OpenStream(&audio, &aParams, 0, outFormat.sampleRate(), this->chunkSize, paNoFlag, &paHandler::staticWrite, (void*)this);
emit setupConverter(outFormat, inFormat, 7, setup.resampleQuality);
connect(converter, SIGNAL(converted(audioPacket)), this, SLOT(convertedInput(audioPacket)));
@ -173,7 +217,11 @@ bool paHandler::init(audioSetup setup)
void paHandler::setVolume(unsigned char volume)
{
#ifdef Q_OS_WIN
this->volume = audiopot[volume] * 5;
#else
this->volume = audiopot[volume];
#endif
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "setVolume: " << volume << "(" << this->volume << ")";
}
@ -202,6 +250,18 @@ int paHandler::writeData(const void* inputBuffer, void* outputBuffer,
packet.data.append((char*)inputBuffer, nFrames*inFormat.channelCount()*sizeof(float));
emit sendToConverter(packet);
if (status == paInputUnderflow) {
isUnderrun = true;
}
else if (status == paInputOverflow) {
isOverrun = true;
}
else
{
isUnderrun = false;
isOverrun = false;
}
return paContinue;
}
@ -217,44 +277,24 @@ void paHandler::convertedOutput(audioPacket packet) {
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Error writing audio!";
}
const PaStreamInfo* info = Pa_GetStreamInfo(audio);
//currentLatency = packet.time.msecsTo(QTime::currentTime()) + (info->outputLatency * 1000);
currentLatency = (info->outputLatency * 1000);
}
/*
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audioOutput->bufferSize() - audioOutput->bytesFree()) / 1000);
if (audioDevice != Q_NULLPTR) {
if (audioDevice->write(packet.data) < packet.data.size()) {
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Buffer full!";
isOverrun = true;
}
else {
isOverrun = false;
}
if (lastReceived.msecsTo(QTime::currentTime()) > 100) {
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Time since last audio packet" << lastReceived.msecsTo(QTime::currentTime()) << "Expected around" << setup.blockSize;
}
lastReceived = QTime::currentTime();
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (info->outputLatency * 1000);
}
lastSentSeq = packet.seq;
*/
amplitude = packet.amplitude;
emit haveLevels(getAmplitude(), setup.latency, currentLatency, false, false);
emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun, isOverrun);
}
}
void paHandler::convertedInput(audioPacket audio)
void paHandler::convertedInput(audioPacket packet)
{
if (audio.data.size() > 0) {
emit haveAudioData(audio);
amplitude = audio.amplitude;
emit haveLevels(getAmplitude(), setup.latency, currentLatency, false,false);
if (packet.data.size() > 0) {
emit haveAudioData(packet);
amplitude = packet.amplitude;
const PaStreamInfo* info = Pa_GetStreamInfo(audio);
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (info->inputLatency * 1000);
emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun, isOverrun);
}
}

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@ -88,6 +88,8 @@ private:
QAudioFormat outFormat;
audioConverter* converter = Q_NULLPTR;
QThread* converterThread = Q_NULLPTR;
bool isUnderrun = false;
bool isOverrun = false;
};
#endif // PAHANDLER_H

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@ -62,7 +62,7 @@ bool rtHandler::init(audioSetup setup)
", uLaw" << setup.ulaw;
#if !defined(Q_OS_MACX)
//options.flags = !RTAUDIO_HOG_DEVICE | RTAUDIO_MINIMIZE_LATENCY;
options.flags = ((!RTAUDIO_HOG_DEVICE) | (RTAUDIO_MINIMIZE_LATENCY));
//options.flags = RTAUDIO_MINIMIZE_LATENCY;
#endif
@ -303,17 +303,18 @@ void rtHandler::convertedOutput(audioPacket packet)
arrayBuffer.append(packet.data);
audioMutex.unlock();
amplitude = packet.amplitude;
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audio->getStreamLatency() * (outFormat.sampleSize() / 8) * outFormat.channelCount()) * 1000);
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audio->getStreamLatency() * (outFormat.sampleSize() / 8) * outFormat.channelCount())/1000);
emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun, isOverrun);
}
void rtHandler::convertedInput(audioPacket audio)
void rtHandler::convertedInput(audioPacket packet)
{
if (audio.data.size() > 0) {
emit haveAudioData(audio);
amplitude = audio.amplitude;
if (packet.data.size() > 0) {
emit haveAudioData(packet);
amplitude = packet.amplitude;
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audio->getStreamLatency() * (outFormat.sampleSize() / 8) * outFormat.channelCount())/1000);
emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun, isOverrun);
}
}

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@ -106,7 +106,7 @@ private:
QThread* converterThread = Q_NULLPTR;
QByteArray arrayBuffer;
bool isUnderrun = false;
bool isOverrun = true;
bool isOverrun = false;
QMutex audioMutex;
};