kopia lustrzana https://gitlab.com/eliggett/wfview
365 wiersze
10 KiB
C++
365 wiersze
10 KiB
C++
#include "rthandler.h"
|
|
|
|
#include "logcategories.h"
|
|
|
|
#if defined(Q_OS_WIN)
|
|
#include <objbase.h>
|
|
#endif
|
|
|
|
#define RT_EXCEPTION
|
|
|
|
rtHandler::rtHandler(QObject* parent)
|
|
{
|
|
Q_UNUSED(parent)
|
|
}
|
|
|
|
rtHandler::~rtHandler()
|
|
{
|
|
|
|
if (converterThread != Q_NULLPTR) {
|
|
converterThread->quit();
|
|
converterThread->wait();
|
|
}
|
|
|
|
if (isInitialized) {
|
|
|
|
#ifdef RT_EXCEPTION
|
|
try {
|
|
#endif
|
|
audio->abortStream();
|
|
audio->closeStream();
|
|
#ifdef RT_EXCEPTION
|
|
}
|
|
catch (RtAudioError& e) {
|
|
qInfo(logAudio()) << "Error closing stream:" << aParams.deviceId << ":" << QString::fromStdString(e.getMessage());
|
|
}
|
|
#endif
|
|
delete audio;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
bool rtHandler::init(audioSetup setup)
|
|
{
|
|
if (isInitialized) {
|
|
return false;
|
|
}
|
|
|
|
this->setup = setup;
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "RTAudio handler starting:" << setup.name;
|
|
|
|
if (setup.portInt==-1)
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "No audio device was found.";
|
|
return false;
|
|
}
|
|
|
|
inFormat = toQAudioFormat(setup.codec, setup.sampleRate);
|
|
|
|
qDebug(logAudio()) << "Creating" << (setup.isinput ? "Input" : "Output") << "audio device:" << setup.name <<
|
|
", bits" << inFormat.sampleSize() <<
|
|
", codec" << setup.codec <<
|
|
", latency" << setup.latency <<
|
|
", localAFGain" << setup.localAFgain <<
|
|
", radioChan" << inFormat.channelCount() <<
|
|
", resampleQuality" << setup.resampleQuality <<
|
|
", samplerate" << inFormat.sampleRate() <<
|
|
", uLaw" << setup.ulaw;
|
|
|
|
#if !defined(Q_OS_MACX)
|
|
options.flags = ((!RTAUDIO_HOG_DEVICE) | (RTAUDIO_MINIMIZE_LATENCY));
|
|
//options.flags = RTAUDIO_MINIMIZE_LATENCY;
|
|
#endif
|
|
|
|
#if defined(Q_OS_LINUX)
|
|
audio = new RtAudio(RtAudio::Api::LINUX_ALSA);
|
|
#elif defined(Q_OS_WIN)
|
|
audio = new RtAudio(RtAudio::Api::WINDOWS_WASAPI);
|
|
#elif defined(Q_OS_MACX)
|
|
audio = new RtAudio(RtAudio::Api::MACOSX_CORE);
|
|
#endif
|
|
|
|
options.numberOfBuffers = int(setup.latency/setup.blockSize);
|
|
|
|
if (setup.portInt > 0) {
|
|
aParams.deviceId = setup.portInt;
|
|
}
|
|
else if (setup.isinput) {
|
|
aParams.deviceId = audio->getDefaultInputDevice();
|
|
}
|
|
else {
|
|
aParams.deviceId = audio->getDefaultOutputDevice();
|
|
}
|
|
aParams.firstChannel = 0;
|
|
|
|
#ifdef RT_EXCEPTION
|
|
try {
|
|
#endif
|
|
info = audio->getDeviceInfo(aParams.deviceId);
|
|
#ifdef RT_EXCEPTION
|
|
}
|
|
catch (RtAudioError e) {
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Device exception:" << aParams.deviceId << ":" << QString::fromStdString(e.getMessage());
|
|
goto errorHandler;
|
|
}
|
|
#endif
|
|
if (info.probed)
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed";
|
|
|
|
RtAudioFormat sampleFormat;
|
|
outFormat.setByteOrder(QAudioFormat::LittleEndian);
|
|
outFormat.setCodec("audio/pcm");
|
|
|
|
if (info.nativeFormats == 0)
|
|
{
|
|
qCritical(logAudio()) << " No natively supported data formats!";
|
|
goto errorHandler;
|
|
}
|
|
else {
|
|
qDebug(logAudio()) << " Supported formats:" <<
|
|
(info.nativeFormats & RTAUDIO_SINT8 ? "8-bit int," : "") <<
|
|
(info.nativeFormats & RTAUDIO_SINT16 ? "16-bit int," : "") <<
|
|
(info.nativeFormats & RTAUDIO_SINT24 ? "24-bit int," : "") <<
|
|
(info.nativeFormats & RTAUDIO_SINT32 ? "32-bit int," : "") <<
|
|
(info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") <<
|
|
(info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : "");
|
|
|
|
qInfo(logAudio()) << " Preferred sample rate:" << info.preferredSampleRate;
|
|
if (setup.isinput) {
|
|
outFormat.setChannelCount(info.inputChannels);
|
|
}
|
|
else {
|
|
outFormat.setChannelCount(info.outputChannels);
|
|
}
|
|
|
|
qInfo(logAudio()) << " Channels:" << outFormat.channelCount();
|
|
|
|
if (outFormat.channelCount() > 2) {
|
|
outFormat.setChannelCount(2);
|
|
}
|
|
else if (outFormat.channelCount() < 1)
|
|
{
|
|
qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << "No channels found, aborting setup.";
|
|
goto errorHandler;
|
|
}
|
|
|
|
aParams.nChannels = outFormat.channelCount();
|
|
|
|
|
|
outFormat.setSampleRate(info.preferredSampleRate);
|
|
|
|
if (outFormat.sampleRate() < 44100) {
|
|
outFormat.setSampleRate(48000);
|
|
}
|
|
|
|
if (info.nativeFormats & RTAUDIO_FLOAT32) {
|
|
outFormat.setSampleType(QAudioFormat::Float);
|
|
outFormat.setSampleSize(32);
|
|
sampleFormat = RTAUDIO_FLOAT32;
|
|
}
|
|
else if (info.nativeFormats & RTAUDIO_SINT32) {
|
|
outFormat.setSampleType(QAudioFormat::SignedInt);
|
|
outFormat.setSampleSize(32);
|
|
sampleFormat = RTAUDIO_SINT32;
|
|
}
|
|
else if (info.nativeFormats & RTAUDIO_SINT16) {
|
|
outFormat.setSampleType(QAudioFormat::SignedInt);
|
|
outFormat.setSampleSize(16);
|
|
sampleFormat = RTAUDIO_SINT16;
|
|
}
|
|
else {
|
|
qCritical(logAudio()) << "Cannot find supported sample format!";
|
|
goto errorHandler;
|
|
}
|
|
}
|
|
|
|
|
|
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();
|
|
|
|
// We "hopefully" now have a valid format that is supported so try connecting
|
|
converter = new audioConverter();
|
|
converterThread = new QThread(this);
|
|
if (setup.isinput) {
|
|
converterThread->setObjectName("audioConvIn()");
|
|
}
|
|
else {
|
|
converterThread->setObjectName("audioConvOut()");
|
|
}
|
|
converter->moveToThread(converterThread);
|
|
|
|
connect(this, SIGNAL(setupConverter(QAudioFormat, QAudioFormat, quint8, quint8)), converter, SLOT(init(QAudioFormat, QAudioFormat, quint8, quint8)));
|
|
connect(converterThread, SIGNAL(finished()), converter, SLOT(deleteLater()));
|
|
connect(this, SIGNAL(sendToConverter(audioPacket)), converter, SLOT(convert(audioPacket)));
|
|
converterThread->start(QThread::TimeCriticalPriority);
|
|
|
|
|
|
// Per channel chunk size.
|
|
this->chunkSize = (outFormat.bytesForDuration(setup.blockSize * 1000) / (outFormat.sampleSize()/8) / outFormat.channelCount());
|
|
|
|
#ifdef RT_EXCEPTION
|
|
try {
|
|
#endif
|
|
if (setup.isinput) {
|
|
audio->openStream(NULL, &aParams, sampleFormat, outFormat.sampleRate(), &this->chunkSize, &staticWrite, this, &options);
|
|
emit setupConverter(outFormat, inFormat, 7, setup.resampleQuality);
|
|
connect(converter, SIGNAL(converted(audioPacket)), this, SLOT(convertedInput(audioPacket)));
|
|
}
|
|
else {
|
|
audio->openStream(&aParams, NULL, sampleFormat, outFormat.sampleRate(), &this->chunkSize, &staticRead, this , &options);
|
|
emit setupConverter(inFormat, outFormat, 7, setup.resampleQuality);
|
|
connect(converter, SIGNAL(converted(audioPacket)), this, SLOT(convertedOutput(audioPacket)));
|
|
}
|
|
audio->startStream();
|
|
isInitialized = true;
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "device successfully opened";
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "detected latency:" << audio->getStreamLatency();
|
|
#ifdef RT_EXCEPTION
|
|
}
|
|
catch (RtAudioError& e) {
|
|
qInfo(logAudio()) << "Error opening:" << QString::fromStdString(e.getMessage());
|
|
// Try again?
|
|
goto errorHandler;
|
|
}
|
|
#endif
|
|
}
|
|
else
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") could not be probed, check audio configuration!";
|
|
goto errorHandler;
|
|
}
|
|
|
|
this->setVolume(setup.localAFgain);
|
|
|
|
|
|
return isInitialized;
|
|
|
|
errorHandler:
|
|
if (retryConnectCount < 10) {
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "*** Attempting to reconnect to audio device in 500ms";
|
|
QTimer::singleShot(500, this, std::bind(&rtHandler::init, this, setup));
|
|
retryConnectCount++;
|
|
}
|
|
else
|
|
{
|
|
qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << "*** Retry count exceeded, giving up!";
|
|
}
|
|
return false;
|
|
|
|
}
|
|
|
|
|
|
void rtHandler::setVolume(unsigned char volume)
|
|
{
|
|
this->volume = audiopot[volume];
|
|
}
|
|
|
|
void rtHandler::incomingAudio(audioPacket packet)
|
|
{
|
|
packet.volume = volume;
|
|
emit sendToConverter(packet);
|
|
return;
|
|
}
|
|
|
|
int rtHandler::readData(void* outputBuffer, void* inputBuffer,
|
|
unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
|
|
{
|
|
Q_UNUSED(inputBuffer);
|
|
Q_UNUSED(streamTime);
|
|
int nBytes = nFrames * outFormat.channelCount() * (outFormat.sampleSize()/8);
|
|
|
|
//lastSentSeq = packet.seq;
|
|
if (arrayBuffer.length() >= nBytes) {
|
|
if (audioMutex.tryLock(0)) {
|
|
std::memcpy(outputBuffer, arrayBuffer.constData(), nBytes);
|
|
arrayBuffer.remove(0, nBytes);
|
|
audioMutex.unlock();
|
|
}
|
|
}
|
|
|
|
if (status == RTAUDIO_INPUT_OVERFLOW) {
|
|
isUnderrun = true;
|
|
}
|
|
else if (status == RTAUDIO_OUTPUT_UNDERFLOW) {
|
|
isOverrun = true;
|
|
}
|
|
else
|
|
{
|
|
isUnderrun = false;
|
|
isOverrun = false;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
|
|
int rtHandler::writeData(void* outputBuffer, void* inputBuffer,
|
|
unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
|
|
{
|
|
Q_UNUSED(outputBuffer);
|
|
Q_UNUSED(streamTime);
|
|
Q_UNUSED(status);
|
|
audioPacket packet;
|
|
packet.time = QTime::currentTime();
|
|
packet.sent = 0;
|
|
packet.volume = volume;
|
|
memcpy(&packet.guid, setup.guid, GUIDLEN);
|
|
packet.data.append((char*)inputBuffer, nFrames *outFormat.channelCount() * (outFormat.sampleSize()/8));
|
|
emit sendToConverter(packet);
|
|
if (status == RTAUDIO_INPUT_OVERFLOW) {
|
|
isUnderrun = true;
|
|
}
|
|
else if (status == RTAUDIO_OUTPUT_UNDERFLOW) {
|
|
isOverrun = true;
|
|
}
|
|
else
|
|
{
|
|
isUnderrun = false;
|
|
isOverrun = false;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
void rtHandler::convertedOutput(audioPacket packet)
|
|
{
|
|
audioMutex.lock();
|
|
arrayBuffer.append(packet.data);
|
|
audioMutex.unlock();
|
|
amplitude = packet.amplitudePeak;
|
|
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audio->getStreamLatency() * (outFormat.sampleSize() / 8) * outFormat.channelCount())/1000);
|
|
emit haveLevels(getAmplitude(), packet.amplitudeRMS, setup.latency, currentLatency, isUnderrun, isOverrun);
|
|
}
|
|
|
|
|
|
|
|
void rtHandler::convertedInput(audioPacket packet)
|
|
{
|
|
if (packet.data.size() > 0) {
|
|
emit haveAudioData(packet);
|
|
amplitude = packet.amplitudePeak;
|
|
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audio->getStreamLatency() * (outFormat.sampleSize() / 8) * outFormat.channelCount())/1000);
|
|
emit haveLevels(getAmplitude(), static_cast<quint16>(packet.amplitudeRMS * 255.0), setup.latency, currentLatency, isUnderrun, isOverrun);
|
|
}
|
|
}
|
|
|
|
|
|
|
|
void rtHandler::changeLatency(const quint16 newSize)
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << setup.latency;
|
|
}
|
|
|
|
int rtHandler::getLatency()
|
|
{
|
|
return currentLatency;
|
|
}
|
|
|
|
|
|
quint16 rtHandler::getAmplitude()
|
|
{
|
|
return static_cast<quint16>(amplitude * 255.0);
|
|
}
|