kopia lustrzana https://gitlab.com/eliggett/wfview
726 wiersze
21 KiB
C++
726 wiersze
21 KiB
C++
/*
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This class handles both RX and TX audio, each is created as a seperate 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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#if defined(RTAUDIO)
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try {
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audio->abortStream();
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audio->closeStream();
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}
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catch (RtAudioError& e) {
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qInfo(logAudio()) << "Error closing stream:" << aParams.deviceId << ":" << QString::fromStdString(e.getMessage());
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}
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delete audio;
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#elif defined(PORTAUDIO)
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#else
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stop();
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#endif
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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 (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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}
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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.radioChan = 1;
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setup.bits = 8;
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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) {
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setup.radioChan = 2;
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}
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if (setup.codec == 0x04 || setup.codec == 0x10) {
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setup.bits = 16;
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}
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ringBuf = new wilt::Ring<audioPacket>(100); // Should be customizable.
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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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#if defined(RTAUDIO)
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#if !defined(Q_OS_MACX)
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options.flags = ((!RTAUDIO_HOG_DEVICE) | (RTAUDIO_MINIMIZE_LATENCY));
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#endif
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#if defined(Q_OS_LINUX)
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audio = new RtAudio(RtAudio::Api::LINUX_ALSA);
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#elif defined(Q_OS_WIN)
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audio = new RtAudio(RtAudio::Api::WINDOWS_WASAPI);
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#elif defined(Q_OS_MACX)
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audio = new RtAudio(RtAudio::Api::MACOSX_CORE);
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#endif
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if (setup.port > 0) {
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aParams.deviceId = setup.port;
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}
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else if (setup.isinput) {
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aParams.deviceId = audio->getDefaultInputDevice();
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}
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else {
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aParams.deviceId = audio->getDefaultOutputDevice();
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}
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aParams.firstChannel = 0;
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try {
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info = audio->getDeviceInfo(aParams.deviceId);
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}
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catch (RtAudioError& e) {
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qInfo(logAudio()) << "Device error:" << aParams.deviceId << ":" << QString::fromStdString(e.getMessage());
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return isInitialized;
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}
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if (info.probed)
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{
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// if "preferred" sample rate is 44100, try 48K instead
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if (info.preferredSampleRate == (unsigned int)44100) {
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qDebug(logAudio()) << "Preferred sample rate 44100, trying 48000";
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this->nativeSampleRate = 48000;
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}
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else {
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this->nativeSampleRate = info.preferredSampleRate;
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}
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// Per channel chunk size.
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this->chunkSize = (this->nativeSampleRate / 50);
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") successfully probed";
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if (info.nativeFormats == 0)
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{
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qInfo(logAudio()) << " No natively supported data formats!";
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return false;
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}
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else {
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qDebug(logAudio()) << " Supported formats:" <<
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(info.nativeFormats & RTAUDIO_SINT8 ? "8-bit int," : "") <<
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(info.nativeFormats & RTAUDIO_SINT16 ? "16-bit int," : "") <<
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(info.nativeFormats & RTAUDIO_SINT24 ? "24-bit int," : "") <<
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(info.nativeFormats & RTAUDIO_SINT32 ? "32-bit int," : "") <<
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(info.nativeFormats & RTAUDIO_FLOAT32 ? "32-bit float," : "") <<
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(info.nativeFormats & RTAUDIO_FLOAT64 ? "64-bit float," : "");
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qInfo(logAudio()) << " Preferred sample rate:" << info.preferredSampleRate;
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if (setup.isinput) {
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devChannels = info.inputChannels;
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}
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else {
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devChannels = info.outputChannels;
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}
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qInfo(logAudio()) << " Channels:" << devChannels;
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if (devChannels > 2) {
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devChannels = 2;
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}
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aParams.nChannels = devChannels;
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}
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qInfo(logAudio()) << " chunkSize: " << chunkSize;
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try {
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if (setup.isinput) {
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audio->openStream(NULL, &aParams, RTAUDIO_SINT16, this->nativeSampleRate, &this->chunkSize, &staticWrite, this, &options);
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}
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else {
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audio->openStream(&aParams, NULL, RTAUDIO_SINT16, this->nativeSampleRate, &this->chunkSize, &staticRead, this, &options);
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}
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audio->startStream();
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isInitialized = true;
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "device successfully opened";
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "detected latency:" << audio->getStreamLatency();
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}
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catch (RtAudioError& e) {
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qInfo(logAudio()) << "Error opening:" << QString::fromStdString(e.getMessage());
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}
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}
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else
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{
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qCritical(logAudio()) << (setup.isinput ? "Input" : "Output") << QString::fromStdString(info.name) << "(" << aParams.deviceId << ") could not be probed, check audio configuration!";
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}
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#elif defined(PORTAUDIO)
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#else
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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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// chunk size is always relative to Internal Sample Rate.
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this->chunkSize = (nativeSampleRate / 50);
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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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connect(audioInput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State)));
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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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#ifdef Q_OS_MAC
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audioOutput->setBufferSize(chunkSize*4);
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#endif
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connect(audioOutput, SIGNAL(notify()), SLOT(notified()));
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connect(audioOutput, SIGNAL(stateChanged(QAudio::State)), SLOT(stateChanged(QAudio::State)));
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isInitialized = true;
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}
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#endif
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// Setup resampler if it is needed.
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int resample_error = 0;
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if (setup.isinput) {
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resampler = wf_resampler_init(devChannels, nativeSampleRate, setup.samplerate, setup.resampleQuality, &resample_error);
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}
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else {
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resampler = wf_resampler_init(devChannels, setup.samplerate, this->nativeSampleRate, setup.resampleQuality, &resample_error);
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}
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wf_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "wf_resampler_init() returned: " << resample_error << " ratioNum" << ratioNum << " ratioDen" << ratioDen;
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "thread id" << QThread::currentThreadId();
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#if !defined (RTAUDIO) && !defined(PORTAUDIO)
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if (isInitialized) {
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this->start();
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}
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#endif
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return isInitialized;
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}
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#if !defined (RTAUDIO) && !defined(PORTAUDIO)
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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 ((audioOutput == Q_NULLPTR || audioOutput->state() != QAudio::StoppedState) &&
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(audioInput == Q_NULLPTR || audioInput->state() != QAudio::StoppedState)) {
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return;
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}
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if (setup.isinput) {
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#ifdef Q_OS_MACX
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this->open(QIODevice::WriteOnly);
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#else
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this->open(QIODevice::WriteOnly | QIODevice::Unbuffered);
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#endif
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audioInput->start(this);
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}
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else {
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#ifdef Q_OS_MACX
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this->open(QIODevice::ReadOnly);
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#else
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this->open(QIODevice::ReadOnly | QIODevice::Unbuffered);
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#endif
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audioOutput->start(this);
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}
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}
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#endif
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void audioHandler::setVolume(unsigned char volume)
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{
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//this->volume = (qreal)volume/255.0;
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this->volume = audiopot[volume];
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "setVolume: " << volume << "(" << this->volume << ")";
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}
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/// <summary>
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/// This function processes the incoming audio FROM the radio and pushes it into the playback buffer *data
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/// </summary>
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/// <param name="data"></param>
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/// <param name="maxlen"></param>
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/// <returns></returns>
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#if defined(RTAUDIO)
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int audioHandler::readData(void* outputBuffer, void* inputBuffer, unsigned int nFrames, double streamTime, RtAudioStreamStatus status)
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{
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Q_UNUSED(inputBuffer);
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Q_UNUSED(streamTime);
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if (status == RTAUDIO_OUTPUT_UNDERFLOW)
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qDebug(logAudio()) << "Underflow detected";
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int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample and 2 channels
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quint8* buffer = (quint8*)outputBuffer;
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#elif defined(PORTAUDIO)
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#else
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qint64 audioHandler::readData(char* buffer, qint64 nBytes)
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{
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#endif
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// Calculate output length, always full samples
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int sentlen = 0;
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if (!isReady) {
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isReady = true;
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}
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if (ringBuf->size()>0)
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{
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// Output buffer is ALWAYS 16 bit.
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//qDebug(logAudio()) << "Read: nFrames" << nFrames << "nBytes" << nBytes;
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while (sentlen < nBytes)
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{
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audioPacket packet;
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if (!ringBuf->try_read(packet))
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{
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qDebug() << "No more data available but buffer is not full! sentlen:" << sentlen << " nBytes:" << nBytes ;
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break;
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}
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currentLatency = packet.time.msecsTo(QTime::currentTime());
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// This shouldn't be required but if we did output a partial packet
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// This will add the remaining packet data to the output buffer.
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if (tempBuf.sent != tempBuf.data.length())
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{
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int send = qMin((int)nBytes - sentlen, tempBuf.data.length() - tempBuf.sent);
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memcpy(buffer + sentlen, tempBuf.data.constData() + tempBuf.sent, send);
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tempBuf.sent = tempBuf.sent + send;
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sentlen = sentlen + send;
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if (tempBuf.sent != tempBuf.data.length())
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{
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// We still don't have enough buffer space for this?
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break;
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}
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//qDebug(logAudio()) << "Adding partial:" << send;
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}
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while (currentLatency > setup.latency) {
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qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Packet " << hex << packet.seq <<
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" arrived too late (increase output latency!) " <<
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dec << packet.time.msecsTo(QTime::currentTime()) << "ms";
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lastSeq = packet.seq;
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if (!ringBuf->try_read(packet))
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break;
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currentLatency = packet.time.msecsTo(QTime::currentTime());
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}
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int send = qMin((int)nBytes - sentlen, packet.data.length());
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memcpy(buffer + sentlen, packet.data.constData(), send);
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sentlen = sentlen + send;
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if (send < packet.data.length())
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{
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//qDebug(logAudio()) << "Asking for partial, sent:" << send << "packet length" << packet.data.length();
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tempBuf = packet;
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tempBuf.sent = tempBuf.sent + send;
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lastSeq = packet.seq;
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break;
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}
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if (packet.seq <= lastSeq) {
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Duplicate/early audio packet: " << hex << lastSeq << " got " << hex << packet.seq;
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}
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else if (packet.seq != lastSeq + 1) {
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qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Missing audio packet(s) from: " << hex << lastSeq + 1 << " to " << hex << packet.seq - 1;
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}
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lastSeq = packet.seq;
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}
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}
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//qDebug(logAudio()) << "looking for: " << nBytes << " got: " << sentlen;
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// fill the rest of the buffer with silence
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if (nBytes > sentlen) {
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memset(buffer+sentlen,0,nBytes-sentlen);
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}
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#if defined(RTAUDIO)
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return 0;
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#elif defined(PORTAUDIO)
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#else
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return nBytes;
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#endif
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}
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#if defined(RTAUDIO)
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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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Q_UNUSED(outputBuffer);
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Q_UNUSED(streamTime);
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Q_UNUSED(status);
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int nBytes = nFrames * devChannels * 2; // This is ALWAYS 2 bytes per sample and 2 channels
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const char* data = (const char*)inputBuffer;
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#elif defined(PORTAUDIO)
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#else
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qint64 audioHandler::writeData(const char* data, qint64 nBytes)
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{
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#endif
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if (!isReady) {
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isReady = true;
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}
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int sentlen = 0;
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//qDebug(logAudio()) << "nFrames" << nFrames << "nBytes" << nBytes;
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int chunkBytes = chunkSize * devChannels * 2;
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while (sentlen < nBytes) {
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if (tempBuf.sent != chunkBytes)
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{
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int send = qMin((int)(nBytes - sentlen), chunkBytes - 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.sent + send;
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}
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else {
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ringBuf->write(tempBuf);
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/*
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if (!ringBuf->try_write(tempBuf))
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{
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qDebug(logAudio()) << "outgoing audio buffer full!";
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break;
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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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}
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//qDebug(logAudio()) << "sentlen" << sentlen;
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#if defined(RTAUDIO)
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return 0;
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#elif defined(PORTAUDIO)
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#else
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return nBytes;
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#endif
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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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if (!isInitialized && !isReady)
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{
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qDebug(logAudio()) << "Packet received when stream was not ready";
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return;
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}
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//qDebug(logAudio()) << "Got" << radioSampleBits << "bits, length" << inPacket.data.length();
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// Incoming data is 8bits?
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if (setup.bits == 8)
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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)inPacket.data.length() * 2 * (devChannels / setup.radioChan), (char)0xff);
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qint16* out = (qint16*)outPacket.data();
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for (int f = 0; f < inPacket.data.length(); f++)
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{
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for (int g = setup.radioChan; g <= devChannels; g++)
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{
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if (isUlaw)
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*out++ = ulaw_decode[(quint8)inPacket.data[f]] * this->volume;
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else
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*out++ = (qint16)(((quint8)inPacket.data[f] << 8) - 32640 * this->volume);
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}
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}
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inPacket.data.clear();
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inPacket.data = outPacket; // Replace incoming data with converted.
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}
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else
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{
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// This is already a 16bit stream, do we need to convert to stereo?
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if (setup.radioChan == 1 && devChannels > 1) {
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// Yes
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QByteArray outPacket(inPacket.data.length() * 2, (char)0xff); // Preset the output buffer size.
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qint16* in = (qint16*)inPacket.data.data();
|
|
qint16* out = (qint16*)outPacket.data();
|
|
for (int f = 0; f < inPacket.data.length() / 2; f++)
|
|
{
|
|
*out++ = (qint16)*in * this->volume;
|
|
*out++ = (qint16)*in++ * this->volume;
|
|
}
|
|
inPacket.data.clear();
|
|
inPacket.data = outPacket; // Replace incoming data with converted.
|
|
}
|
|
else
|
|
{
|
|
// We already have the same number of channels so just update volume.
|
|
qint16* in = (qint16*)inPacket.data.data();
|
|
for (int f = 0; f < inPacket.data.length() / 2; f++)
|
|
{
|
|
*in = *in * this->volume;
|
|
in++;
|
|
}
|
|
}
|
|
|
|
}
|
|
|
|
/* We now have an array of 16bit samples in the NATIVE samplerate of the radio
|
|
If the radio sample rate is below 48000, we need to resample.
|
|
*/
|
|
//qDebug(logAudio()) << "Now 16 bit stereo, length" << inPacket.data.length();
|
|
|
|
if (ratioDen != 1) {
|
|
|
|
// We need to resample
|
|
// We have a stereo 16bit stream.
|
|
quint32 outFrames = ((inPacket.data.length() / 2 / devChannels) * ratioDen);
|
|
quint32 inFrames = (inPacket.data.length() / 2 / devChannels);
|
|
QByteArray outPacket(outFrames * 4, (char)0xff); // Preset the output buffer size.
|
|
|
|
const qint16* in = (qint16*)inPacket.data.constData();
|
|
qint16* out = (qint16*)outPacket.data();
|
|
|
|
int err = 0;
|
|
err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
|
|
if (err) {
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
|
|
}
|
|
inPacket.data.clear();
|
|
inPacket.data = outPacket; // Replace incoming data with converted.
|
|
}
|
|
|
|
//qDebug(logAudio()) << "Adding packet to buffer:" << inPacket.seq << ": " << inPacket.data.length();
|
|
|
|
if (!ringBuf->try_write(inPacket))
|
|
{
|
|
qDebug(logAudio()) << "Buffer full! capacity:" << ringBuf->capacity() << "length" << ringBuf->size();
|
|
}
|
|
return;
|
|
}
|
|
|
|
void audioHandler::changeLatency(const quint16 newSize)
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Changing latency to: " << newSize << " from " << setup.latency;
|
|
setup.latency = newSize;
|
|
}
|
|
|
|
int audioHandler::getLatency()
|
|
{
|
|
return currentLatency;
|
|
}
|
|
|
|
void audioHandler::getNextAudioChunk(QByteArray& ret)
|
|
{
|
|
audioPacket packet;
|
|
packet.sent = 0;
|
|
|
|
if (isInitialized && ringBuf != Q_NULLPTR && ringBuf->try_read(packet))
|
|
{
|
|
currentLatency = packet.time.msecsTo(QTime::currentTime());
|
|
|
|
while (currentLatency > setup.latency) {
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Packet " << hex << packet.seq <<
|
|
" arrived too late (increase output latency!) " <<
|
|
dec << packet.time.msecsTo(QTime::currentTime()) << "ms";
|
|
if (!ringBuf->try_read(packet))
|
|
break;
|
|
currentLatency = packet.time.msecsTo(QTime::currentTime());
|
|
}
|
|
|
|
//qDebug(logAudio) << "Chunksize" << this->chunkSize << "Packet size" << packet.data.length();
|
|
// Packet will arrive as stereo interleaved 16bit 48K
|
|
if (ratioNum != 1)
|
|
{
|
|
quint32 outFrames = ((packet.data.length() / 2 / devChannels) / ratioNum);
|
|
quint32 inFrames = (packet.data.length() / 2 / devChannels);
|
|
QByteArray outPacket((int)outFrames * 2 * devChannels, (char)0xff);
|
|
|
|
const qint16* in = (qint16*)packet.data.constData();
|
|
qint16* out = (qint16*)outPacket.data();
|
|
|
|
int err = 0;
|
|
err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
|
|
if (err) {
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
|
|
}
|
|
//qInfo(logAudio()) << "Resampler run " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
|
|
//qInfo(logAudio()) << "Resampler run inLen:" << packet->datain.length() << " outLen:" << packet->dataout.length();
|
|
packet.data.clear();
|
|
packet.data = outPacket; // Copy output packet back to input buffer.
|
|
}
|
|
|
|
//qDebug(logAudio()) << "Now resampled, length" << packet.data.length();
|
|
|
|
// Do we need to convert mono to stereo?
|
|
if (setup.radioChan == 1 && devChannels > 1)
|
|
{
|
|
// Strip out right channel?
|
|
QByteArray outPacket(packet.data.length()/2, (char)0xff);
|
|
const qint16* in = (qint16*)packet.data.constData();
|
|
qint16* out = (qint16*)outPacket.data();
|
|
for (int f = 0; f < outPacket.length()/2; f++)
|
|
{
|
|
*out++ = *in++;
|
|
in++; // Skip each even channel.
|
|
}
|
|
packet.data.clear();
|
|
packet.data = outPacket; // Copy output packet back to input buffer.
|
|
}
|
|
|
|
//qDebug(logAudio()) << "Now mono, length" << packet.data.length();
|
|
|
|
// Do we need to convert 16-bit to 8-bit?
|
|
if (setup.bits == 8) {
|
|
QByteArray outPacket((int)packet.data.length() / 2, (char)0xff);
|
|
qint16* in = (qint16*)packet.data.data();
|
|
for (int f = 0; f < outPacket.length(); f++)
|
|
{
|
|
quint8 outdata = 0;
|
|
if (isUlaw) {
|
|
qint16 enc = qFromLittleEndian<quint16>(in + f);
|
|
if (enc >= 0)
|
|
outdata = ulaw_encode[enc];
|
|
else
|
|
outdata = 0x7f & ulaw_encode[-enc];
|
|
}
|
|
else {
|
|
outdata = (quint8)(((qFromLittleEndian<qint16>(in + f) >> 8) ^ 0x80) & 0xff);
|
|
}
|
|
outPacket[f] = (char)outdata;
|
|
}
|
|
packet.data.clear();
|
|
packet.data = outPacket; // Copy output packet back to input buffer.
|
|
}
|
|
ret = packet.data;
|
|
//qDebug(logAudio()) << "Now radio format, length" << packet.data.length();
|
|
}
|
|
|
|
|
|
return;
|
|
|
|
}
|
|
|
|
|
|
#if !defined (RTAUDIO) && !defined(PORTAUDIO)
|
|
|
|
qint64 audioHandler::bytesAvailable() const
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
bool audioHandler::isSequential() const
|
|
{
|
|
return true;
|
|
}
|
|
|
|
void audioHandler::notified()
|
|
{
|
|
}
|
|
|
|
|
|
void audioHandler::stateChanged(QAudio::State state)
|
|
{
|
|
// Process the state
|
|
switch (state)
|
|
{
|
|
case QAudio::IdleState:
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio now in idle state: " << audioBuffer.size() << " packets in buffer";
|
|
if (audioOutput != Q_NULLPTR && audioOutput->error() == QAudio::UnderrunError)
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "buffer underrun";
|
|
//audioOutput->suspend();
|
|
}
|
|
break;
|
|
}
|
|
case QAudio::ActiveState:
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio now in active state: " << audioBuffer.size() << " packets in buffer";
|
|
break;
|
|
}
|
|
case QAudio::SuspendedState:
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio now in suspended state: " << audioBuffer.size() << " packets in buffer";
|
|
break;
|
|
}
|
|
case QAudio::StoppedState:
|
|
{
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Audio now in stopped state: " << audioBuffer.size() << " packets in buffer";
|
|
break;
|
|
}
|
|
default: {
|
|
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unhandled audio state: " << audioBuffer.size() << " packets in buffer";
|
|
}
|
|
}
|
|
}
|
|
|
|
void audioHandler::stop()
|
|
{
|
|
if (audioOutput != Q_NULLPTR && audioOutput->state() != QAudio::StoppedState) {
|
|
// Stop audio output
|
|
audioOutput->stop();
|
|
this->stop();
|
|
this->close();
|
|
delete audioOutput;
|
|
audioOutput = Q_NULLPTR;
|
|
}
|
|
|
|
if (audioInput != Q_NULLPTR && audioInput->state() != QAudio::StoppedState) {
|
|
// Stop audio output
|
|
audioInput->stop();
|
|
this->stop();
|
|
this->close();
|
|
delete audioInput;
|
|
audioInput = Q_NULLPTR;
|
|
}
|
|
isInitialized = false;
|
|
}
|
|
|
|
#endif
|
|
|
|
|