mirror of
https://github.com/overte-org/overte.git
synced 2025-08-04 05:03:31 +02:00
immediately send ICE heartbeat once data present
This commit is contained in:
Stephen Birarda
parent
603528eede
commit
69f1cd80e6
7 changed files with 167 additions and 121 deletions
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@ -68,6 +68,7 @@ void DatagramProcessor::processDatagrams() {
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Qt::QueuedConnection,
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Q_ARG(QByteArray, incomingPacket));
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} else {
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// qDebug() << "Received audio data packet at" << usecTimestampNow();
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QMetaObject::invokeMethod(DependencyManager::get<AudioClient>().data(), "addReceivedAudioToStream",
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Qt::QueuedConnection,
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Q_ARG(QByteArray, incomingPacket));
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@ -116,7 +116,7 @@ AudioClient::AudioClient() :
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{
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// clear the array of locally injected samples
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memset(_localProceduralSamples, 0, AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL);
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connect(&_receivedAudioStream, &MixedProcessedAudioStream::processSamples,
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this, &AudioClient::processReceivedSamples, Qt::DirectConnection);
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@ -127,7 +127,7 @@ AudioClient::AudioClient() :
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QTimer* updateTimer = new QTimer(this);
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connect(updateTimer, &QTimer::timeout, this, &AudioClient::checkDevices);
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updateTimer->start(DEVICE_CHECK_INTERVAL_MSECS);
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// create GVerb filter
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_gverb = createGverbFilter();
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configureGverbFilter(_gverb);
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@ -135,7 +135,7 @@ AudioClient::AudioClient() :
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AudioClient::~AudioClient() {
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stop();
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if (_gverb) {
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gverb_free(_gverb);
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}
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@ -148,7 +148,7 @@ void AudioClient::reset() {
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_toneSource.reset();
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_sourceGain.reset();
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_inputGain.reset();
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gverb_flush(_gverb);
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}
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@ -186,7 +186,7 @@ int numDestinationSamplesRequired(const QAudioFormat& sourceFormat, const QAudio
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int numSourceSamples) {
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float ratio = (float) destinationFormat.channelCount() / sourceFormat.channelCount();
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ratio *= (float) destinationFormat.sampleRate() / sourceFormat.sampleRate();
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return (numSourceSamples * ratio) + 0.5f;
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}
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@ -287,7 +287,7 @@ QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
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}
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qCDebug(audioclient) << "DEBUG [" << deviceName << "] [" << getNamedAudioDeviceForMode(mode, deviceName).deviceName() << "]";
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return getNamedAudioDeviceForMode(mode, deviceName);
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#endif
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@ -302,7 +302,7 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
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if (!audioDevice.isFormatSupported(desiredAudioFormat)) {
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qCDebug(audioclient) << "The desired format for audio I/O is" << desiredAudioFormat;
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qCDebug(audioclient, "The desired audio format is not supported by this device");
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if (desiredAudioFormat.channelCount() == 1) {
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adjustedAudioFormat = desiredAudioFormat;
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adjustedAudioFormat.setChannelCount(2);
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@ -313,17 +313,17 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
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adjustedAudioFormat.setChannelCount(1);
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}
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}
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const int FORTY_FOUR = 44100;
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adjustedAudioFormat = desiredAudioFormat;
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#ifdef Q_OS_ANDROID
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adjustedAudioFormat.setSampleRate(FORTY_FOUR);
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#else
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const int HALF_FORTY_FOUR = FORTY_FOUR / 2;
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if (audioDevice.supportedSampleRates().contains(AudioConstants::SAMPLE_RATE * 2)) {
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// use 48, which is a sample downsample, upsample
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adjustedAudioFormat.setSampleRate(AudioConstants::SAMPLE_RATE * 2);
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@ -335,7 +335,7 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
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adjustedAudioFormat.setSampleRate(FORTY_FOUR);
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}
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#endif
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if (adjustedAudioFormat != desiredAudioFormat) {
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// return the nearest in case it needs 2 channels
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adjustedAudioFormat = audioDevice.nearestFormat(adjustedAudioFormat);
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@ -357,18 +357,18 @@ bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationS
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for (uint i = 0; i < numSourceSamples; i += 2) {
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destinationSamples[i / 2] = (sourceSamples[i] / 2) + (sourceSamples[i + 1] / 2);
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}
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return true;
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} else if (sourceAudioFormat.channelCount() == 1 && destinationAudioFormat.channelCount() == 2) {
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// loop through the mono input audio and repeat each sample twice
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for (uint i = 0; i < numSourceSamples; ++i) {
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destinationSamples[i * 2] = destinationSamples[(i * 2) + 1] = sourceSamples[i];
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}
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return true;
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}
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return false;
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}
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@ -376,7 +376,7 @@ soxr_error_t possibleResampling(soxr_t resampler,
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const int16_t* sourceSamples, int16_t* destinationSamples,
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unsigned int numSourceSamples, unsigned int numDestinationSamples,
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const QAudioFormat& sourceAudioFormat, const QAudioFormat& destinationAudioFormat) {
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if (numSourceSamples > 0) {
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if (!resampler) {
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if (!sampleChannelConversion(sourceSamples, destinationSamples, numSourceSamples,
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@ -384,41 +384,41 @@ soxr_error_t possibleResampling(soxr_t resampler,
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// no conversion, we can copy the samples directly across
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memcpy(destinationSamples, sourceSamples, numSourceSamples * sizeof(int16_t));
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}
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return 0;
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} else {
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soxr_error_t resampleError = 0;
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if (sourceAudioFormat.channelCount() != destinationAudioFormat.channelCount()) {
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float channelCountRatio = (float) destinationAudioFormat.channelCount() / sourceAudioFormat.channelCount();
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int numChannelCoversionSamples = (int) (numSourceSamples * channelCountRatio);
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int16_t* channelConversionSamples = new int16_t[numChannelCoversionSamples];
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sampleChannelConversion(sourceSamples, channelConversionSamples,
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numSourceSamples,
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sourceAudioFormat, destinationAudioFormat);
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resampleError = soxr_process(resampler,
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channelConversionSamples, numChannelCoversionSamples, NULL,
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destinationSamples, numDestinationSamples, NULL);
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delete[] channelConversionSamples;
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} else {
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unsigned int numAdjustedSourceSamples = numSourceSamples;
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unsigned int numAdjustedDestinationSamples = numDestinationSamples;
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if (sourceAudioFormat.channelCount() == 2 && destinationAudioFormat.channelCount() == 2) {
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numAdjustedSourceSamples /= 2;
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numAdjustedDestinationSamples /= 2;
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}
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resampleError = soxr_process(resampler,
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sourceSamples, numAdjustedSourceSamples, NULL,
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destinationSamples, numAdjustedDestinationSamples, NULL);
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}
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return resampleError;
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}
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} else {
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@ -429,30 +429,30 @@ soxr_error_t possibleResampling(soxr_t resampler,
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soxr_t soxrResamplerFromInputFormatToOutputFormat(const QAudioFormat& sourceAudioFormat,
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const QAudioFormat& destinationAudioFormat) {
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soxr_error_t soxrError;
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// setup soxr_io_spec_t for input and output
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soxr_io_spec_t inputToNetworkSpec = soxr_io_spec(soxrDataTypeFromQAudioFormat(sourceAudioFormat),
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soxrDataTypeFromQAudioFormat(destinationAudioFormat));
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// setup soxr_quality_spec_t for quality options
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soxr_quality_spec_t qualitySpec = soxr_quality_spec(SOXR_MQ, 0);
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int channelCount = (sourceAudioFormat.channelCount() == 2 && destinationAudioFormat.channelCount() == 2)
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? 2 : 1;
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soxr_t newResampler = soxr_create(sourceAudioFormat.sampleRate(),
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destinationAudioFormat.sampleRate(),
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channelCount,
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&soxrError, &inputToNetworkSpec, &qualitySpec, 0);
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if (soxrError) {
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qCDebug(audioclient) << "There was an error setting up the soxr resampler -" << "soxr error code was " << soxrError;
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soxr_delete(newResampler);
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return NULL;
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}
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return newResampler;
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}
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@ -476,7 +476,7 @@ void AudioClient::start() {
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QAudioDeviceInfo outputDeviceInfo = defaultAudioDeviceForMode(QAudio::AudioOutput);
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qCDebug(audioclient) << "The default audio output device is" << outputDeviceInfo.deviceName();
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bool outputFormatSupported = switchOutputToAudioDevice(outputDeviceInfo);
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if (!inputFormatSupported) {
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qCDebug(audioclient) << "Unable to set up audio input because of a problem with input format.";
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qCDebug(audioclient) << "The closest format available is" << inputDeviceInfo.nearestFormat(_desiredInputFormat);
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@ -503,11 +503,11 @@ void AudioClient::stop() {
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_sourceGain.finalize();
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_noiseSource.finalize();
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_toneSource.finalize();
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// "switch" to invalid devices in order to shut down the state
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switchInputToAudioDevice(QAudioDeviceInfo());
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switchOutputToAudioDevice(QAudioDeviceInfo());
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if (_loopbackResampler) {
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soxr_delete(_loopbackResampler);
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_loopbackResampler = NULL;
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@ -543,7 +543,7 @@ ty_gverb* AudioClient::createGverbFilter() {
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_reverbOptions->getReverbTime(), _reverbOptions->getDamping(), _reverbOptions->getSpread(),
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_reverbOptions->getInputBandwidth(), _reverbOptions->getEarlyLevel(),
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_reverbOptions->getTailLevel());
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return filter;
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}
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@ -560,7 +560,7 @@ void AudioClient::configureGverbFilter(ty_gverb* filter) {
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void AudioClient::updateGverbOptions() {
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bool reverbChanged = false;
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if (_receivedAudioStream.hasReverb()) {
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if (_zoneReverbOptions.getReverbTime() != _receivedAudioStream.getRevebTime()) {
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_zoneReverbOptions.setReverbTime(_receivedAudioStream.getRevebTime());
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reverbChanged = true;
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@ -569,7 +569,7 @@ void AudioClient::updateGverbOptions() {
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_zoneReverbOptions.setWetLevel(_receivedAudioStream.getWetLevel());
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// Not part of actual filter config, no need to set reverbChanged to true
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}
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if (_reverbOptions != &_zoneReverbOptions) {
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_reverbOptions = &_zoneReverbOptions;
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reverbChanged = true;
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@ -578,7 +578,7 @@ void AudioClient::updateGverbOptions() {
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_reverbOptions = &_scriptReverbOptions;
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reverbChanged = true;
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}
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if (reverbChanged) {
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gverb_free(_gverb);
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_gverb = createGverbFilter();
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@ -588,7 +588,7 @@ void AudioClient::updateGverbOptions() {
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void AudioClient::setReverb(bool reverb) {
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_reverb = reverb;
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if (!_reverb) {
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gverb_flush(_gverb);
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}
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@ -620,7 +620,7 @@ void AudioClient::addReverb(ty_gverb* gverb, int16_t* samplesData, int16_t* reve
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QAudioFormat& audioFormat, bool noEcho) {
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float wetFraction = DB_CO(_reverbOptions->getWetLevel());
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float dryFraction = 1.0f - wetFraction;
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float lValue,rValue;
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for (int sample = 0; sample < numSamples; sample += audioFormat.channelCount()) {
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// Run GVerb
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@ -634,7 +634,7 @@ void AudioClient::addReverb(ty_gverb* gverb, int16_t* samplesData, int16_t* reve
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int lResult = glm::clamp((int)(samplesData[j] * dryFraction + lValue * wetFraction),
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AudioConstants::MIN_SAMPLE_VALUE, AudioConstants::MAX_SAMPLE_VALUE);
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samplesData[j] = (int16_t)lResult;
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if (noEcho) {
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reverbAlone[j] = (int16_t)lValue * wetFraction;
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}
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@ -643,7 +643,7 @@ void AudioClient::addReverb(ty_gverb* gverb, int16_t* samplesData, int16_t* reve
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int rResult = glm::clamp((int)(samplesData[j] * dryFraction + rValue * wetFraction),
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AudioConstants::MIN_SAMPLE_VALUE, AudioConstants::MAX_SAMPLE_VALUE);
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samplesData[j] = (int16_t)rResult;
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if (noEcho) {
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reverbAlone[j] = (int16_t)rValue * wetFraction;
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}
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@ -660,53 +660,53 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
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if (_muted || !_audioOutput || (!_shouldEchoLocally && !hasReverb)) {
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return;
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}
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// if this person wants local loopback add that to the locally injected audio
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// if there is reverb apply it to local audio and substract the origin samples
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if (!_loopbackOutputDevice && _loopbackAudioOutput) {
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// we didn't have the loopback output device going so set that up now
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_loopbackOutputDevice = _loopbackAudioOutput->start();
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if (!_loopbackOutputDevice) {
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return;
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}
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}
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// do we need to setup a resampler?
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if (_inputFormat.sampleRate() != _outputFormat.sampleRate() && !_loopbackResampler) {
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qCDebug(audioclient) << "Attemping to create a resampler for input format to output format for audio loopback.";
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_loopbackResampler = soxrResamplerFromInputFormatToOutputFormat(_inputFormat, _outputFormat);
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if (!_loopbackResampler) {
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return;
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}
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}
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static QByteArray reverbAlone; // Intermediary for local reverb with no echo
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static QByteArray loopBackByteArray;
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int numInputSamples = inputByteArray.size() / sizeof(int16_t);
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int numLoopbackSamples = numDestinationSamplesRequired(_inputFormat, _outputFormat, numInputSamples);
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reverbAlone.resize(numInputSamples * sizeof(int16_t));
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loopBackByteArray.resize(numLoopbackSamples * sizeof(int16_t));
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int16_t* inputSamples = reinterpret_cast<int16_t*>(inputByteArray.data());
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int16_t* reverbAloneSamples = reinterpret_cast<int16_t*>(reverbAlone.data());
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int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
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if (hasReverb) {
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updateGverbOptions();
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addReverb(_gverb, inputSamples, reverbAloneSamples, numInputSamples,
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_inputFormat, !_shouldEchoLocally);
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}
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possibleResampling(_loopbackResampler,
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(_shouldEchoLocally) ? inputSamples : reverbAloneSamples, loopbackSamples,
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numInputSamples, numLoopbackSamples,
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_inputFormat, _outputFormat);
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_loopbackOutputDevice->write(loopBackByteArray);
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}
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@ -726,7 +726,7 @@ void AudioClient::handleAudioInput() {
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int inputSamplesRequired = (int)((float)AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * inputToNetworkInputRatio);
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QByteArray inputByteArray = _inputDevice->readAll();
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// Add audio source injection if enabled
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if (!_muted && _audioSourceInjectEnabled) {
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int16_t* inputFrameData = (int16_t*)inputByteArray.data();
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@ -745,11 +745,11 @@ void AudioClient::handleAudioInput() {
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_sourceGain.render(_inputFrameBuffer); // post gain
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_inputFrameBuffer.copyFrames(1, inputFrameCount, inputFrameData, true /*copy out*/);
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}
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handleLocalEchoAndReverb(inputByteArray);
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_inputRingBuffer.writeData(inputByteArray.data(), inputByteArray.size());
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float audioInputMsecsRead = inputByteArray.size() / (float)(_inputFormat.bytesForDuration(USECS_PER_MSEC));
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_stats.updateInputMsecsRead(audioInputMsecsRead);
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@ -763,50 +763,50 @@ void AudioClient::handleAudioInput() {
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: AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL;
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if (!_muted) {
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// zero out the monoAudioSamples array and the locally injected audio
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memset(networkAudioSamples, 0, numNetworkBytes);
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// Increment the time since the last clip
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if (_timeSinceLastClip >= 0.0f) {
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_timeSinceLastClip += (float) numNetworkSamples / (float) AudioConstants::SAMPLE_RATE;
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}
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int16_t* inputAudioSamples = new int16_t[inputSamplesRequired];
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_inputRingBuffer.readSamples(inputAudioSamples, inputSamplesRequired);
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possibleResampling(_inputToNetworkResampler,
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inputAudioSamples, networkAudioSamples,
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inputSamplesRequired, numNetworkSamples,
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_inputFormat, _desiredInputFormat);
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delete[] inputAudioSamples;
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// only impose the noise gate and perform tone injection if we are sending mono audio
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if (!_isStereoInput && !_audioSourceInjectEnabled && _isNoiseGateEnabled) {
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_inputGate.gateSamples(networkAudioSamples, numNetworkSamples);
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// if we performed the noise gate we can get values from it instead of enumerating the samples again
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_lastInputLoudness = _inputGate.getLastLoudness();
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if (_inputGate.clippedInLastFrame()) {
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_timeSinceLastClip = 0.0f;
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}
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} else {
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float loudness = 0.0f;
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for (int i = 0; i < numNetworkSamples; i++) {
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int thisSample = std::abs(networkAudioSamples[i]);
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loudness += (float) thisSample;
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|
||||
if (thisSample > (AudioConstants::MAX_SAMPLE_VALUE * AudioNoiseGate::CLIPPING_THRESHOLD)) {
|
||||
_timeSinceLastClip = 0.0f;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
_lastInputLoudness = fabs(loudness / numNetworkSamples);
|
||||
}
|
||||
|
||||
|
||||
emit inputReceived(QByteArray(reinterpret_cast<const char*>(networkAudioSamples),
|
||||
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * sizeof(AudioConstants::AudioSample)));
|
||||
|
||||
|
|
@ -814,13 +814,13 @@ void AudioClient::handleAudioInput() {
|
|||
// our input loudness is 0, since we're muted
|
||||
_lastInputLoudness = 0;
|
||||
_timeSinceLastClip = 0.0f;
|
||||
|
||||
|
||||
_inputRingBuffer.shiftReadPosition(inputSamplesRequired);
|
||||
}
|
||||
|
||||
auto nodeList = DependencyManager::get<NodeList>();
|
||||
SharedNodePointer audioMixer = nodeList->soloNodeOfType(NodeType::AudioMixer);
|
||||
|
||||
|
||||
if (audioMixer && audioMixer->getActiveSocket()) {
|
||||
glm::vec3 headPosition = _positionGetter();
|
||||
glm::quat headOrientation = _orientationGetter();
|
||||
|
|
@ -852,7 +852,7 @@ void AudioClient::handleAudioInput() {
|
|||
// memcpy the three float positions
|
||||
memcpy(currentPacketPtr, &headPosition, sizeof(headPosition));
|
||||
currentPacketPtr += (sizeof(headPosition));
|
||||
|
||||
|
||||
// memcpy our orientation
|
||||
memcpy(currentPacketPtr, &headOrientation, sizeof(headOrientation));
|
||||
currentPacketPtr += sizeof(headOrientation);
|
||||
|
|
@ -864,7 +864,7 @@ void AudioClient::handleAudioInput() {
|
|||
// memcpy the three float positions
|
||||
memcpy(currentPacketPtr, &headPosition, sizeof(headPosition));
|
||||
currentPacketPtr += (sizeof(headPosition));
|
||||
|
||||
|
||||
// memcpy our orientation
|
||||
memcpy(currentPacketPtr, &headOrientation, sizeof(headOrientation));
|
||||
currentPacketPtr += sizeof(headOrientation);
|
||||
|
|
@ -875,6 +875,8 @@ void AudioClient::handleAudioInput() {
|
|||
|
||||
_stats.sentPacket();
|
||||
|
||||
// qDebug() << "Sending audio packet at" << usecTimestampNow();
|
||||
|
||||
int packetBytes = currentPacketPtr - audioDataPacket;
|
||||
nodeList->writeDatagram(audioDataPacket, packetBytes, audioMixer);
|
||||
_outgoingAvatarAudioSequenceNumber++;
|
||||
|
|
@ -890,7 +892,7 @@ void AudioClient::processReceivedSamples(const QByteArray& inputBuffer, QByteArr
|
|||
outputBuffer.resize(numDeviceOutputSamples * sizeof(int16_t));
|
||||
|
||||
const int16_t* receivedSamples = reinterpret_cast<const int16_t*>(inputBuffer.data());
|
||||
|
||||
|
||||
// copy the packet from the RB to the output
|
||||
possibleResampling(_networkToOutputResampler, receivedSamples,
|
||||
reinterpret_cast<int16_t*>(outputBuffer.data()),
|
||||
|
|
@ -900,20 +902,20 @@ void AudioClient::processReceivedSamples(const QByteArray& inputBuffer, QByteArr
|
|||
|
||||
void AudioClient::sendMuteEnvironmentPacket() {
|
||||
auto nodeList = DependencyManager::get<NodeList>();
|
||||
|
||||
|
||||
QByteArray mutePacket = nodeList->byteArrayWithPopulatedHeader(PacketTypeMuteEnvironment);
|
||||
int headerSize = mutePacket.size();
|
||||
|
||||
|
||||
const float MUTE_RADIUS = 50;
|
||||
|
||||
|
||||
glm::vec3 currentSourcePosition = _positionGetter();
|
||||
mutePacket.resize(mutePacket.size() + sizeof(glm::vec3) + sizeof(float));
|
||||
memcpy(mutePacket.data() + headerSize, ¤tSourcePosition, sizeof(glm::vec3));
|
||||
memcpy(mutePacket.data() + headerSize + sizeof(glm::vec3), &MUTE_RADIUS, sizeof(float));
|
||||
|
||||
|
||||
// grab our audio mixer from the NodeList, if it exists
|
||||
SharedNodePointer audioMixer = nodeList->soloNodeOfType(NodeType::AudioMixer);
|
||||
|
||||
|
||||
if (audioMixer) {
|
||||
// send off this mute packet
|
||||
nodeList->writeDatagram(mutePacket, audioMixer);
|
||||
|
|
@ -922,6 +924,7 @@ void AudioClient::sendMuteEnvironmentPacket() {
|
|||
|
||||
void AudioClient::addReceivedAudioToStream(const QByteArray& audioByteArray) {
|
||||
if (_audioOutput) {
|
||||
// qDebug() << "Adding received audio to stream at" << usecTimestampNow();
|
||||
// Audio output must exist and be correctly set up if we're going to process received audio
|
||||
_receivedAudioStream.parseData(audioByteArray);
|
||||
}
|
||||
|
|
@ -930,11 +933,11 @@ void AudioClient::addReceivedAudioToStream(const QByteArray& audioByteArray) {
|
|||
void AudioClient::parseAudioEnvironmentData(const QByteArray &packet) {
|
||||
int numBytesPacketHeader = numBytesForPacketHeader(packet);
|
||||
const char* dataAt = packet.constData() + numBytesPacketHeader;
|
||||
|
||||
|
||||
char bitset;
|
||||
memcpy(&bitset, dataAt, sizeof(char));
|
||||
dataAt += sizeof(char);
|
||||
|
||||
|
||||
bool hasReverb = oneAtBit(bitset, HAS_REVERB_BIT);;
|
||||
if (hasReverb) {
|
||||
float reverbTime, wetLevel;
|
||||
|
|
@ -956,13 +959,13 @@ void AudioClient::toggleMute() {
|
|||
void AudioClient::setIsStereoInput(bool isStereoInput) {
|
||||
if (isStereoInput != _isStereoInput) {
|
||||
_isStereoInput = isStereoInput;
|
||||
|
||||
|
||||
if (_isStereoInput) {
|
||||
_desiredInputFormat.setChannelCount(2);
|
||||
} else {
|
||||
_desiredInputFormat.setChannelCount(1);
|
||||
}
|
||||
|
||||
|
||||
// change in channel count for desired input format, restart the input device
|
||||
switchInputToAudioDevice(_inputAudioDeviceName);
|
||||
}
|
||||
|
|
@ -976,7 +979,7 @@ void AudioClient::selectAudioSourcePinkNoise() {
|
|||
_noiseSourceEnabled = true;
|
||||
_toneSourceEnabled = false;
|
||||
}
|
||||
|
||||
|
||||
void AudioClient::selectAudioSourceSine440() {
|
||||
_toneSourceEnabled = true;
|
||||
_noiseSourceEnabled = false;
|
||||
|
|
@ -986,24 +989,24 @@ bool AudioClient::outputLocalInjector(bool isStereo, AudioInjector* injector) {
|
|||
if (injector->getLocalBuffer()) {
|
||||
QAudioFormat localFormat = _desiredOutputFormat;
|
||||
localFormat.setChannelCount(isStereo ? 2 : 1);
|
||||
|
||||
|
||||
QAudioOutput* localOutput = new QAudioOutput(getNamedAudioDeviceForMode(QAudio::AudioOutput, _outputAudioDeviceName),
|
||||
localFormat,
|
||||
injector->getLocalBuffer());
|
||||
|
||||
|
||||
// move the localOutput to the same thread as the local injector buffer
|
||||
localOutput->moveToThread(injector->getLocalBuffer()->thread());
|
||||
|
||||
|
||||
// have it be stopped when that local buffer is about to close
|
||||
connect(injector->getLocalBuffer(), &AudioInjectorLocalBuffer::bufferEmpty, localOutput, &QAudioOutput::stop);
|
||||
connect(injector->getLocalBuffer(), &QIODevice::aboutToClose, localOutput, &QAudioOutput::stop);
|
||||
|
||||
|
||||
qCDebug(audioclient) << "Starting QAudioOutput for local injector" << localOutput;
|
||||
|
||||
|
||||
localOutput->start(injector->getLocalBuffer());
|
||||
return localOutput->state() == QAudio::ActiveState;
|
||||
}
|
||||
|
||||
|
||||
return false;
|
||||
}
|
||||
|
||||
|
|
@ -1015,7 +1018,7 @@ void AudioClient::outputFormatChanged() {
|
|||
|
||||
bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceInfo) {
|
||||
bool supportedFormat = false;
|
||||
|
||||
|
||||
// cleanup any previously initialized device
|
||||
if (_audioInput) {
|
||||
// The call to stop() causes _inputDevice to be destructed.
|
||||
|
|
@ -1030,7 +1033,7 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
|
|||
|
||||
_inputAudioDeviceName = "";
|
||||
}
|
||||
|
||||
|
||||
if (_inputToNetworkResampler) {
|
||||
// if we were using an input to network resampler, delete it here
|
||||
soxr_delete(_inputToNetworkResampler);
|
||||
|
|
@ -1040,36 +1043,36 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
|
|||
if (!inputDeviceInfo.isNull()) {
|
||||
qCDebug(audioclient) << "The audio input device " << inputDeviceInfo.deviceName() << "is available.";
|
||||
_inputAudioDeviceName = inputDeviceInfo.deviceName().trimmed();
|
||||
|
||||
|
||||
if (adjustedFormatForAudioDevice(inputDeviceInfo, _desiredInputFormat, _inputFormat)) {
|
||||
qCDebug(audioclient) << "The format to be used for audio input is" << _inputFormat;
|
||||
|
||||
|
||||
// we've got the best we can get for input
|
||||
// if required, setup a soxr resampler for this input to our desired network format
|
||||
if (_inputFormat != _desiredInputFormat
|
||||
&& _inputFormat.sampleRate() != _desiredInputFormat.sampleRate()) {
|
||||
qCDebug(audioclient) << "Attemping to create a soxr resampler for input format to network format.";
|
||||
_inputToNetworkResampler = soxrResamplerFromInputFormatToOutputFormat(_inputFormat, _desiredInputFormat);
|
||||
|
||||
|
||||
if (!_inputToNetworkResampler) {
|
||||
return false;
|
||||
}
|
||||
} else {
|
||||
qCDebug(audioclient) << "No resampling required for audio input to match desired network format.";
|
||||
}
|
||||
|
||||
|
||||
// if the user wants stereo but this device can't provide then bail
|
||||
if (!_isStereoInput || _inputFormat.channelCount() == 2) {
|
||||
if (!_isStereoInput || _inputFormat.channelCount() == 2) {
|
||||
_audioInput = new QAudioInput(inputDeviceInfo, _inputFormat, this);
|
||||
_numInputCallbackBytes = calculateNumberOfInputCallbackBytes(_inputFormat);
|
||||
_audioInput->setBufferSize(_numInputCallbackBytes);
|
||||
|
||||
|
||||
// how do we want to handle input working, but output not working?
|
||||
int numFrameSamples = calculateNumberOfFrameSamples(_numInputCallbackBytes);
|
||||
_inputRingBuffer.resizeForFrameSize(numFrameSamples);
|
||||
|
||||
|
||||
_inputDevice = _audioInput->start();
|
||||
|
||||
|
||||
if (_inputDevice) {
|
||||
connect(_inputDevice, SIGNAL(readyRead()), this, SLOT(handleAudioInput()));
|
||||
supportedFormat = true;
|
||||
|
|
@ -1079,7 +1082,7 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
|
|||
}
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
return supportedFormat;
|
||||
}
|
||||
|
||||
|
|
@ -1117,7 +1120,7 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
|
|||
// cleanup any previously initialized device
|
||||
if (_audioOutput) {
|
||||
_audioOutput->stop();
|
||||
|
||||
|
||||
delete _audioOutput;
|
||||
_audioOutput = NULL;
|
||||
|
||||
|
|
@ -1125,13 +1128,13 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
|
|||
delete _loopbackAudioOutput;
|
||||
_loopbackAudioOutput = NULL;
|
||||
}
|
||||
|
||||
|
||||
if (_networkToOutputResampler) {
|
||||
// if we were using an input to network resampler, delete it here
|
||||
soxr_delete(_networkToOutputResampler);
|
||||
_networkToOutputResampler = NULL;
|
||||
}
|
||||
|
||||
|
||||
if (_loopbackResampler) {
|
||||
// if we were using an input to output resample, delete it here
|
||||
soxr_delete(_loopbackResampler);
|
||||
|
|
@ -1141,24 +1144,24 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
|
|||
if (!outputDeviceInfo.isNull()) {
|
||||
qCDebug(audioclient) << "The audio output device " << outputDeviceInfo.deviceName() << "is available.";
|
||||
_outputAudioDeviceName = outputDeviceInfo.deviceName().trimmed();
|
||||
|
||||
|
||||
if (adjustedFormatForAudioDevice(outputDeviceInfo, _desiredOutputFormat, _outputFormat)) {
|
||||
qCDebug(audioclient) << "The format to be used for audio output is" << _outputFormat;
|
||||
|
||||
|
||||
// we've got the best we can get for input
|
||||
// if required, setup a soxr resampler for this input to our desired network format
|
||||
if (_desiredOutputFormat != _outputFormat
|
||||
&& _desiredOutputFormat.sampleRate() != _outputFormat.sampleRate()) {
|
||||
qCDebug(audioclient) << "Attemping to create a resampler for network format to output format.";
|
||||
_networkToOutputResampler = soxrResamplerFromInputFormatToOutputFormat(_desiredOutputFormat, _outputFormat);
|
||||
|
||||
|
||||
if (!_networkToOutputResampler) {
|
||||
return false;
|
||||
}
|
||||
} else {
|
||||
qCDebug(audioclient) << "No resampling required for network output to match actual output format.";
|
||||
}
|
||||
|
||||
|
||||
outputFormatChanged();
|
||||
|
||||
// setup our general output device for audio-mixer audio
|
||||
|
|
@ -1168,20 +1171,20 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
|
|||
connect(_audioOutput, &QAudioOutput::notify, this, &AudioClient::outputNotify);
|
||||
|
||||
qCDebug(audioclient) << "Output Buffer capacity in frames: " << _audioOutput->bufferSize() / sizeof(int16_t) / (float)_outputFrameSize;
|
||||
|
||||
|
||||
_audioOutputIODevice.start();
|
||||
_audioOutput->start(&_audioOutputIODevice);
|
||||
|
||||
// setup a loopback audio output device
|
||||
_loopbackAudioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this);
|
||||
|
||||
|
||||
|
||||
_timeSinceLastReceived.start();
|
||||
|
||||
supportedFormat = true;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
return supportedFormat;
|
||||
}
|
||||
|
||||
|
|
@ -1202,7 +1205,7 @@ void AudioClient::setOutputBufferSize(int numFrames) {
|
|||
|
||||
// The following constant is operating system dependent due to differences in
|
||||
// the way input audio is handled. The audio input buffer size is inversely
|
||||
// proportional to the accelerator ratio.
|
||||
// proportional to the accelerator ratio.
|
||||
|
||||
#ifdef Q_OS_WIN
|
||||
const float AudioClient::CALLBACK_ACCELERATOR_RATIO = 0.1f;
|
||||
|
|
@ -1226,7 +1229,7 @@ int AudioClient::calculateNumberOfInputCallbackBytes(const QAudioFormat& format)
|
|||
}
|
||||
|
||||
float AudioClient::calculateDeviceToNetworkInputRatio() const {
|
||||
float inputToNetworkInputRatio = (int)((_numInputCallbackBytes
|
||||
float inputToNetworkInputRatio = (int)((_numInputCallbackBytes
|
||||
* CALLBACK_ACCELERATOR_RATIO
|
||||
/ AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL) + 0.5f);
|
||||
|
||||
|
|
@ -1257,7 +1260,7 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
|
|||
int samplesRequested = maxSize / sizeof(int16_t);
|
||||
int samplesPopped;
|
||||
int bytesWritten;
|
||||
|
||||
|
||||
if ((samplesPopped = _receivedAudioStream.popSamples(samplesRequested, false)) > 0) {
|
||||
AudioRingBuffer::ConstIterator lastPopOutput = _receivedAudioStream.getLastPopOutput();
|
||||
lastPopOutput.readSamples((int16_t*)data, samplesPopped);
|
||||
|
|
|
|||
|
|
@ -106,6 +106,8 @@ bool AddressManager::handleUrl(const QUrl& lookupUrl) {
|
|||
|
||||
qCDebug(networking) << "Trying to go to URL" << lookupUrl.toString();
|
||||
|
||||
qDebug() << "Lookup of HF URL at" << usecTimestampNow();
|
||||
|
||||
// there are 4 possible lookup strings
|
||||
|
||||
// 1. global place name (name of domain or place) - example: sanfrancisco
|
||||
|
|
@ -163,6 +165,8 @@ void AddressManager::handleAPIResponse(QNetworkReply& requestReply) {
|
|||
QJsonObject responseObject = QJsonDocument::fromJson(requestReply.readAll()).object();
|
||||
QJsonObject dataObject = responseObject["data"].toObject();
|
||||
|
||||
qDebug() << "Go to address from API response at" << usecTimestampNow();
|
||||
|
||||
goToAddressFromObject(dataObject.toVariantMap(), requestReply);
|
||||
|
||||
emit lookupResultsFinished();
|
||||
|
|
|
|||
|
|
@ -135,10 +135,21 @@ void DomainHandler::setIceServerHostnameAndID(const QString& iceServerHostname,
|
|||
replaceableSockAddr->~HifiSockAddr();
|
||||
replaceableSockAddr = new (replaceableSockAddr) HifiSockAddr(iceServerHostname, ICE_SERVER_DEFAULT_PORT);
|
||||
|
||||
if (_iceServerSockAddr.getAddress().isNull()) {
|
||||
// connect to lookup completed for ice-server socket so we can request a heartbeat once hostname is looked up
|
||||
connect(&_iceServerSockAddr, &HifiSockAddr::lookupCompleted, this, &DomainHandler::completedIceServerHostnameLookup);
|
||||
} else {
|
||||
completedIceServerHostnameLookup();
|
||||
}
|
||||
|
||||
|
||||
// refresh our ICE client UUID to something new
|
||||
_iceClientID = QUuid::createUuid();
|
||||
|
||||
qCDebug(networking) << "ICE required to connect to domain via ice server at" << iceServerHostname;
|
||||
|
||||
qDebug() << "ICE server info set at" << usecTimestampNow();
|
||||
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -172,6 +183,14 @@ void DomainHandler::completedHostnameLookup(const QHostInfo& hostInfo) {
|
|||
qCDebug(networking, "Failed domain server lookup");
|
||||
}
|
||||
|
||||
void DomainHandler::completedIceServerHostnameLookup() {
|
||||
qDebug() << "ICE server socket is at" << _iceServerSockAddr;
|
||||
qDebug() << "ICE server socket lookup completed at" << usecTimestampNow();
|
||||
// emit our signal so we can send a heartbeat to ice-server immediately
|
||||
emit iceSocketAndIDReceived();
|
||||
|
||||
}
|
||||
|
||||
void DomainHandler::setIsConnected(bool isConnected) {
|
||||
if (_isConnected != isConnected) {
|
||||
_isConnected = isConnected;
|
||||
|
|
@ -270,6 +289,8 @@ void DomainHandler::processICEResponsePacket(const QByteArray& icePacket) {
|
|||
NetworkPeer packetPeer;
|
||||
iceResponseStream >> packetPeer;
|
||||
|
||||
qDebug() << "Recieved response for network peer from ICE server at" << usecTimestampNow();
|
||||
|
||||
if (packetPeer.getUUID() != _iceDomainID) {
|
||||
qCDebug(networking) << "Received a network peer with ID that does not match current domain. Will not attempt connection.";
|
||||
} else {
|
||||
|
|
|
|||
|
|
@ -85,7 +85,9 @@ public slots:
|
|||
|
||||
private slots:
|
||||
void completedHostnameLookup(const QHostInfo& hostInfo);
|
||||
void completedIceServerHostnameLookup();
|
||||
void settingsRequestFinished();
|
||||
|
||||
signals:
|
||||
void hostnameChanged(const QString& hostname);
|
||||
|
||||
|
|
@ -96,6 +98,7 @@ signals:
|
|||
void connectedToDomain(const QString& hostname);
|
||||
void disconnectedFromDomain();
|
||||
|
||||
void iceSocketAndIDReceived();
|
||||
void requestICEConnectionAttempt();
|
||||
|
||||
void settingsReceived(const QJsonObject& domainSettingsObject);
|
||||
|
|
|
|||
|
|
@ -64,6 +64,9 @@ NodeList::NodeList(char newOwnerType, unsigned short socketListenPort, unsigned
|
|||
// clear our NodeList when the domain changes
|
||||
connect(&_domainHandler, &DomainHandler::disconnectedFromDomain, this, &NodeList::reset);
|
||||
|
||||
// send an ICE heartbeat as soon as we get ice server information
|
||||
connect(&_domainHandler, &DomainHandler::iceSocketAndIDReceived, this, &NodeList::handleICEConnectionToDomainServer);
|
||||
|
||||
// handle ICE signal from DS so connection is attempted immediately
|
||||
connect(&_domainHandler, &DomainHandler::requestICEConnectionAttempt, this, &NodeList::handleICEConnectionToDomainServer);
|
||||
|
||||
|
|
@ -160,6 +163,7 @@ void NodeList::processNodeData(const HifiSockAddr& senderSockAddr, const QByteAr
|
|||
if (!_domainHandler.getSockAddr().isNull()) {
|
||||
// only process a list from domain-server if we're talking to a domain
|
||||
// TODO: how do we make sure this is actually the domain we want the list from (DTLS probably)
|
||||
qDebug() << "Processing domain server list at" << usecTimestampNow();
|
||||
processDomainServerList(packet);
|
||||
}
|
||||
break;
|
||||
|
|
@ -198,6 +202,8 @@ void NodeList::processNodeData(const HifiSockAddr& senderSockAddr, const QByteAr
|
|||
if (sendingNode) {
|
||||
sendingNode->setLastHeardMicrostamp(usecTimestampNow());
|
||||
|
||||
qDebug() << "Activating socket for node" << sendingNode->getUUID() << "at" << usecTimestampNow();
|
||||
|
||||
// activate the appropriate socket for this node, if not yet updated
|
||||
activateSocketFromNodeCommunication(packet, sendingNode);
|
||||
|
||||
|
|
@ -216,6 +222,8 @@ void NodeList::processNodeData(const HifiSockAddr& senderSockAddr, const QByteAr
|
|||
case PacketTypeUnverifiedPingReply: {
|
||||
qCDebug(networking) << "Received reply from domain-server on" << senderSockAddr;
|
||||
|
||||
qDebug() << "Received reply from domain server at" << usecTimestampNow();
|
||||
|
||||
// for now we're unsafely assuming this came back from the domain
|
||||
if (senderSockAddr == _domainHandler.getICEPeer().getLocalSocket()) {
|
||||
qCDebug(networking) << "Connecting to domain using local socket";
|
||||
|
|
@ -379,6 +387,8 @@ void NodeList::sendDomainServerCheckIn() {
|
|||
}
|
||||
}
|
||||
|
||||
qDebug() << "Sending domain server check in at" << usecTimestampNow();
|
||||
|
||||
if (!isUsingDTLS) {
|
||||
writeUnverifiedDatagram(domainServerPacket, _domainHandler.getSockAddr());
|
||||
}
|
||||
|
|
@ -504,6 +514,8 @@ void NodeList::handleICEConnectionToDomainServer() {
|
|||
|
||||
_domainHandler.getICEPeer().resetConnectionAttemps();
|
||||
|
||||
qDebug() << "Sending heartbeat to ice server at" << usecTimestampNow();
|
||||
|
||||
LimitedNodeList::sendHeartbeatToIceServer(_domainHandler.getICEServerSockAddr(),
|
||||
_domainHandler.getICEClientID(),
|
||||
_domainHandler.getICEDomainID());
|
||||
|
|
@ -511,6 +523,8 @@ void NodeList::handleICEConnectionToDomainServer() {
|
|||
qCDebug(networking) << "Sending ping packets to establish connectivity with domain-server with ID"
|
||||
<< uuidStringWithoutCurlyBraces(_domainHandler.getICEDomainID());
|
||||
|
||||
qDebug() << "Sending ping packet to domain server at" << usecTimestampNow();
|
||||
|
||||
// send the ping packet to the local and public sockets for this node
|
||||
QByteArray localPingPacket = constructPingPacket(PingType::Local, false, _domainHandler.getICEClientID());
|
||||
writeUnverifiedDatagram(localPingPacket, _domainHandler.getICEPeer().getLocalSocket());
|
||||
|
|
@ -596,6 +610,7 @@ void NodeList::sendAssignment(Assignment& assignment) {
|
|||
}
|
||||
|
||||
void NodeList::pingPunchForInactiveNode(const SharedNodePointer& node) {
|
||||
qDebug() << "Sending ping punch to node" << node->getUUID() << "at" << usecTimestampNow();
|
||||
|
||||
// send the ping packet to the local and public sockets for this node
|
||||
QByteArray localPingPacket = constructPingPacket(PingType::Local);
|
||||
|
|
|
|||
|
|
@ -76,6 +76,7 @@ signals:
|
|||
void limitOfSilentDomainCheckInsReached();
|
||||
private slots:
|
||||
void sendPendingDSPathQuery();
|
||||
void handleICEConnectionToDomainServer();
|
||||
private:
|
||||
NodeList() : LimitedNodeList(0, 0) { assert(false); } // Not implemented, needed for DependencyManager templates compile
|
||||
NodeList(char ownerType, unsigned short socketListenPort = 0, unsigned short dtlsListenPort = 0);
|
||||
|
|
@ -85,8 +86,6 @@ private:
|
|||
void sendSTUNRequest();
|
||||
bool processSTUNResponse(const QByteArray& packet);
|
||||
|
||||
void handleICEConnectionToDomainServer();
|
||||
|
||||
void processDomainServerAuthRequest(const QByteArray& packet);
|
||||
void requestAuthForDomainServer();
|
||||
void activateSocketFromNodeCommunication(const QByteArray& packet, const SharedNodePointer& sendingNode);
|
||||
|
|
|
|||
Loading…
Reference in a new issue