mirror of
https://github.com/overte-org/overte.git
synced 2025-04-21 08:04:01 +02:00
Just to make sure, writes data back to a WAV file
This commit is contained in:
Zach Fox
parent
53226e7924
commit
34befd4a52
1 changed files with 188 additions and 190 deletions
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@ -38,6 +38,7 @@
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#include <QtCore/QBuffer>
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#include <QtMultimedia/QAudioInput>
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#include <QtMultimedia/QAudioOutput>
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#include <PathUtils.h>
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#include <ThreadHelpers.h>
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#include <NodeList.h>
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@ -67,7 +68,7 @@ static const int CHECK_INPUT_READS_MSECS = 2000;
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static const int MIN_READS_TO_CONSIDER_INPUT_ALIVE = 10;
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#endif
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static const auto DEFAULT_POSITION_GETTER = []{ return Vectors::ZERO; };
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static const auto DEFAULT_POSITION_GETTER = [] { return Vectors::ZERO; };
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static const auto DEFAULT_ORIENTATION_GETTER = [] { return Quaternions::IDENTITY; };
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static const int DEFAULT_BUFFER_FRAMES = 1;
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@ -78,12 +79,11 @@ static const int OUTPUT_CHANNEL_COUNT = 2;
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static const bool DEFAULT_STARVE_DETECTION_ENABLED = true;
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static const int STARVE_DETECTION_THRESHOLD = 3;
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static const int STARVE_DETECTION_PERIOD = 10 * 1000; // 10 Seconds
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static const int STARVE_DETECTION_PERIOD = 10 * 1000; // 10 Seconds
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Setting::Handle<bool> dynamicJitterBufferEnabled("dynamicJitterBuffersEnabled",
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InboundAudioStream::DEFAULT_DYNAMIC_JITTER_BUFFER_ENABLED);
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Setting::Handle<int> staticJitterBufferFrames("staticJitterBufferFrames",
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InboundAudioStream::DEFAULT_STATIC_JITTER_FRAMES);
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InboundAudioStream::DEFAULT_DYNAMIC_JITTER_BUFFER_ENABLED);
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Setting::Handle<int> staticJitterBufferFrames("staticJitterBufferFrames", InboundAudioStream::DEFAULT_STATIC_JITTER_FRAMES);
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// protect the Qt internal device list
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using Mutex = std::mutex;
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@ -127,7 +127,7 @@ QAudioDeviceInfo AudioClient::getActiveAudioDevice(QAudio::Mode mode) const {
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if (mode == QAudio::AudioInput) {
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return _inputDeviceInfo;
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} else { // if (mode == QAudio::AudioOutput)
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} else { // if (mode == QAudio::AudioOutput)
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return _outputDeviceInfo;
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}
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}
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@ -137,14 +137,13 @@ QList<QAudioDeviceInfo> AudioClient::getAudioDevices(QAudio::Mode mode) const {
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if (mode == QAudio::AudioInput) {
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return _inputDevices;
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} else { // if (mode == QAudio::AudioOutput)
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} else { // if (mode == QAudio::AudioOutput)
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return _outputDevices;
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}
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}
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static void channelUpmix(int16_t* source, int16_t* dest, int numSamples, int numExtraChannels) {
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for (int i = 0; i < numSamples/2; i++) {
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for (int i = 0; i < numSamples / 2; i++) {
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// read 2 samples
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int16_t left = *source++;
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int16_t right = *source++;
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@ -159,8 +158,7 @@ static void channelUpmix(int16_t* source, int16_t* dest, int numSamples, int num
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}
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static void channelDownmix(int16_t* source, int16_t* dest, int numSamples) {
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for (int i = 0; i < numSamples/2; i++) {
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for (int i = 0; i < numSamples / 2; i++) {
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// read 2 samples
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int16_t left = *source++;
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int16_t right = *source++;
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@ -175,48 +173,22 @@ static inline float convertToFloat(int16_t sample) {
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}
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AudioClient::AudioClient() :
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AbstractAudioInterface(),
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_gate(this),
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_audioInput(NULL),
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_dummyAudioInput(NULL),
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_desiredInputFormat(),
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_inputFormat(),
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_numInputCallbackBytes(0),
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_audioOutput(NULL),
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_desiredOutputFormat(),
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_outputFormat(),
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_outputFrameSize(0),
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_numOutputCallbackBytes(0),
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_loopbackAudioOutput(NULL),
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_loopbackOutputDevice(NULL),
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_inputRingBuffer(0),
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_localInjectorsStream(0, 1),
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_receivedAudioStream(RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES),
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_isStereoInput(false),
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_outputStarveDetectionStartTimeMsec(0),
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_outputStarveDetectionCount(0),
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AbstractAudioInterface(), _gate(this), _audioInput(NULL), _dummyAudioInput(NULL), _desiredInputFormat(), _inputFormat(),
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_numInputCallbackBytes(0), _audioOutput(NULL), _desiredOutputFormat(), _outputFormat(), _outputFrameSize(0),
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_numOutputCallbackBytes(0), _loopbackAudioOutput(NULL), _loopbackOutputDevice(NULL), _inputRingBuffer(0),
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_localInjectorsStream(0, 1), _receivedAudioStream(RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES), _isStereoInput(false),
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_outputStarveDetectionStartTimeMsec(0), _outputStarveDetectionCount(0),
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_outputBufferSizeFrames("audioOutputBufferFrames", DEFAULT_BUFFER_FRAMES),
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_sessionOutputBufferSizeFrames(_outputBufferSizeFrames.get()),
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_outputStarveDetectionEnabled("audioOutputStarveDetectionEnabled", DEFAULT_STARVE_DETECTION_ENABLED),
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_lastInputLoudness(0.0f),
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_timeSinceLastClip(-1.0f),
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_muted(false),
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_shouldEchoLocally(false),
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_shouldEchoToServer(false),
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_isNoiseGateEnabled(true),
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_reverb(false),
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_reverbOptions(&_scriptReverbOptions),
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_inputToNetworkResampler(NULL),
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_networkToOutputResampler(NULL),
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_localToOutputResampler(NULL),
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_audioLimiter(AudioConstants::SAMPLE_RATE, OUTPUT_CHANNEL_COUNT),
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_outgoingAvatarAudioSequenceNumber(0),
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_audioOutputIODevice(_localInjectorsStream, _receivedAudioStream, this),
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_stats(&_receivedAudioStream),
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_lastInputLoudness(0.0f), _timeSinceLastClip(-1.0f), _muted(false), _shouldEchoLocally(false), _shouldEchoToServer(false),
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_isNoiseGateEnabled(true), _reverb(false), _reverbOptions(&_scriptReverbOptions), _inputToNetworkResampler(NULL),
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_networkToOutputResampler(NULL), _localToOutputResampler(NULL),
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_audioLimiter(AudioConstants::SAMPLE_RATE, OUTPUT_CHANNEL_COUNT), _outgoingAvatarAudioSequenceNumber(0),
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_audioOutputIODevice(_localInjectorsStream, _receivedAudioStream, this), _stats(&_receivedAudioStream),
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_positionGetter(DEFAULT_POSITION_GETTER),
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#if defined(Q_OS_ANDROID)
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_checkInputTimer(this),
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_isHeadsetPluggedIn(false),
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_checkInputTimer(this), _isHeadsetPluggedIn(false),
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#endif
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_orientationGetter(DEFAULT_ORIENTATION_GETTER) {
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// avoid putting a lock in the device callback
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@ -226,16 +198,20 @@ AudioClient::AudioClient() :
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{
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Setting::Handle<int>::Deprecated("maxFramesOverDesired", InboundAudioStream::MAX_FRAMES_OVER_DESIRED);
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Setting::Handle<int>::Deprecated("windowStarveThreshold", InboundAudioStream::WINDOW_STARVE_THRESHOLD);
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Setting::Handle<int>::Deprecated("windowSecondsForDesiredCalcOnTooManyStarves", InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_CALC_ON_TOO_MANY_STARVES);
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Setting::Handle<int>::Deprecated("windowSecondsForDesiredReduction", InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_REDUCTION);
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Setting::Handle<int>::Deprecated("windowSecondsForDesiredCalcOnTooManyStarves",
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InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_CALC_ON_TOO_MANY_STARVES);
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Setting::Handle<int>::Deprecated("windowSecondsForDesiredReduction",
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InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_REDUCTION);
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Setting::Handle<bool>::Deprecated("useStDevForJitterCalc", InboundAudioStream::USE_STDEV_FOR_JITTER);
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Setting::Handle<bool>::Deprecated("repetitionWithFade", InboundAudioStream::REPETITION_WITH_FADE);
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}
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connect(&_receivedAudioStream, &MixedProcessedAudioStream::processSamples,
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this, &AudioClient::processReceivedSamples, Qt::DirectConnection);
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connect(&_receivedAudioStream, &MixedProcessedAudioStream::processSamples, this, &AudioClient::processReceivedSamples,
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Qt::DirectConnection);
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connect(this, &AudioClient::changeDevice, this, [=](const QAudioDeviceInfo& outputDeviceInfo) {
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qCDebug(audioclient) << "got AudioClient::changeDevice signal, about to call switchOutputToAudioDevice() outputDeviceInfo: [" << outputDeviceInfo.deviceName() << "]";
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qCDebug(audioclient)
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<< "got AudioClient::changeDevice signal, about to call switchOutputToAudioDevice() outputDeviceInfo: ["
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<< outputDeviceInfo.deviceName() << "]";
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switchOutputToAudioDevice(outputDeviceInfo);
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});
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@ -244,20 +220,18 @@ AudioClient::AudioClient() :
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// initialize wasapi; if getAvailableDevices is called from the CheckDevicesThread before this, it will crash
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getAvailableDevices(QAudio::AudioInput);
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getAvailableDevices(QAudio::AudioOutput);
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// start a thread to detect any device changes
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_checkDevicesTimer = new QTimer(this);
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connect(_checkDevicesTimer, &QTimer::timeout, this, [this] {
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QtConcurrent::run(QThreadPool::globalInstance(), [this] { checkDevices(); });
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});
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connect(_checkDevicesTimer, &QTimer::timeout, this,
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[this] { QtConcurrent::run(QThreadPool::globalInstance(), [this] { checkDevices(); }); });
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const unsigned long DEVICE_CHECK_INTERVAL_MSECS = 2 * 1000;
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_checkDevicesTimer->start(DEVICE_CHECK_INTERVAL_MSECS);
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// start a thread to detect peak value changes
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_checkPeakValuesTimer = new QTimer(this);
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connect(_checkPeakValuesTimer, &QTimer::timeout, this, [this] {
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QtConcurrent::run(QThreadPool::globalInstance(), [this] { checkPeakValues(); });
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});
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connect(_checkPeakValuesTimer, &QTimer::timeout, this,
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[this] { QtConcurrent::run(QThreadPool::globalInstance(), [this] { checkPeakValues(); }); });
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const unsigned long PEAK_VALUES_CHECK_INTERVAL_MSECS = 50;
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_checkPeakValuesTimer->start(PEAK_VALUES_CHECK_INTERVAL_MSECS);
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@ -289,11 +263,11 @@ void AudioClient::customDeleter() {
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}
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void AudioClient::handleMismatchAudioFormat(SharedNodePointer node, const QString& currentCodec, const QString& recievedCodec) {
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qCDebug(audioclient) << __FUNCTION__ << "sendingNode:" << *node << "currentCodec:" << currentCodec << "recievedCodec:" << recievedCodec;
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qCDebug(audioclient) << __FUNCTION__ << "sendingNode:" << *node << "currentCodec:" << currentCodec
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<< "recievedCodec:" << recievedCodec;
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selectAudioFormat(recievedCodec);
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}
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void AudioClient::reset() {
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_receivedAudioStream.reset();
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_stats.reset();
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@ -321,7 +295,7 @@ void AudioClient::setAudioPaused(bool pause) {
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QAudioDeviceInfo getNamedAudioDeviceForMode(QAudio::Mode mode, const QString& deviceName) {
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QAudioDeviceInfo result;
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foreach(QAudioDeviceInfo audioDevice, getAvailableDevices(mode)) {
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foreach (QAudioDeviceInfo audioDevice, getAvailableDevices(mode)) {
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if (audioDevice.deviceName().trimmed() == deviceName.trimmed()) {
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result = audioDevice;
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break;
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@ -356,7 +330,8 @@ QString AudioClient::getWinDeviceName(wchar_t* guid) {
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HRESULT hr = S_OK;
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CoInitialize(nullptr);
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IMMDeviceEnumerator* pMMDeviceEnumerator = nullptr;
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CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator), (void**)&pMMDeviceEnumerator);
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CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator),
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(void**)&pMMDeviceEnumerator);
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IMMDevice* pEndpoint;
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hr = pMMDeviceEnumerator->GetDevice(guid, &pEndpoint);
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if (hr == E_NOTFOUND) {
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@ -380,34 +355,26 @@ QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
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if (getAvailableDevices(mode).size() > 1) {
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AudioDeviceID defaultDeviceID = 0;
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uint32_t propertySize = sizeof(AudioDeviceID);
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AudioObjectPropertyAddress propertyAddress = {
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kAudioHardwarePropertyDefaultInputDevice,
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kAudioObjectPropertyScopeGlobal,
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kAudioObjectPropertyElementMaster
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};
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AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDefaultInputDevice,
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kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
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if (mode == QAudio::AudioOutput) {
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propertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
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}
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OSStatus getPropertyError = AudioObjectGetPropertyData(kAudioObjectSystemObject,
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&propertyAddress,
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0,
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NULL,
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&propertySize,
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&defaultDeviceID);
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OSStatus getPropertyError =
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AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &defaultDeviceID);
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if (!getPropertyError && propertySize) {
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CFStringRef deviceName = NULL;
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propertySize = sizeof(deviceName);
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propertyAddress.mSelector = kAudioDevicePropertyDeviceNameCFString;
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getPropertyError = AudioObjectGetPropertyData(defaultDeviceID, &propertyAddress, 0,
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NULL, &propertySize, &deviceName);
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getPropertyError =
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AudioObjectGetPropertyData(defaultDeviceID, &propertyAddress, 0, NULL, &propertySize, &deviceName);
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if (!getPropertyError && propertySize) {
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// find a device in the list that matches the name we have and return it
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foreach(QAudioDeviceInfo audioDevice, getAvailableDevices(mode)) {
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foreach (QAudioDeviceInfo audioDevice, getAvailableDevices(mode)) {
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if (audioDevice.deviceName() == CFStringGetCStringPtr(deviceName, kCFStringEncodingMacRoman)) {
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return audioDevice;
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}
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@ -419,7 +386,7 @@ QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
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#ifdef WIN32
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QString deviceName;
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//Check for Windows Vista or higher, IMMDeviceEnumerator doesn't work below that.
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if (!IsWindowsVistaOrGreater()) { // lower then vista
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if (!IsWindowsVistaOrGreater()) { // lower then vista
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if (mode == QAudio::AudioInput) {
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WAVEINCAPS wic;
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// first use WAVE_MAPPER to get the default devices manufacturer ID
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@ -441,9 +408,11 @@ QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
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HRESULT hr = S_OK;
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CoInitialize(NULL);
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IMMDeviceEnumerator* pMMDeviceEnumerator = NULL;
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CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator), (void**)&pMMDeviceEnumerator);
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CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator),
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(void**)&pMMDeviceEnumerator);
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IMMDevice* pEndpoint;
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hr = pMMDeviceEnumerator->GetDefaultAudioEndpoint(mode == QAudio::AudioOutput ? eRender : eCapture, eMultimedia, &pEndpoint);
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hr = pMMDeviceEnumerator->GetDefaultAudioEndpoint(mode == QAudio::AudioOutput ? eRender : eCapture, eMultimedia,
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&pEndpoint);
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if (hr == E_NOTFOUND) {
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printf("Audio Error: device not found\n");
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deviceName = QString("NONE");
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@ -457,22 +426,22 @@ QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
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CoUninitialize();
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}
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qCDebug(audioclient) << "defaultAudioDeviceForMode mode: " << (mode == QAudio::AudioOutput ? "Output" : "Input")
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<< " [" << deviceName << "] [" << getNamedAudioDeviceForMode(mode, deviceName).deviceName() << "]";
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qCDebug(audioclient) << "defaultAudioDeviceForMode mode: " << (mode == QAudio::AudioOutput ? "Output" : "Input") << " ["
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<< deviceName << "] [" << getNamedAudioDeviceForMode(mode, deviceName).deviceName() << "]";
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return getNamedAudioDeviceForMode(mode, deviceName);
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#endif
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#if defined (Q_OS_ANDROID)
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#if defined(Q_OS_ANDROID)
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if (mode == QAudio::AudioInput) {
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Setting::Handle<bool> enableAEC(SETTING_AEC_KEY, false);
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bool aecEnabled = enableAEC.get();
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auto audioClient = DependencyManager::get<AudioClient>();
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bool headsetOn = audioClient? audioClient->isHeadsetPluggedIn() : false;
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bool headsetOn = audioClient ? audioClient->isHeadsetPluggedIn() : false;
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auto inputDevices = QAudioDeviceInfo::availableDevices(QAudio::AudioInput);
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for (auto inputDevice : inputDevices) {
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if (((headsetOn || !aecEnabled) && inputDevice.deviceName() == VOICE_RECOGNITION) ||
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((!headsetOn && aecEnabled) && inputDevice.deviceName() == VOICE_COMMUNICATION)) {
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((!headsetOn && aecEnabled) && inputDevice.deviceName() == VOICE_COMMUNICATION)) {
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return inputDevice;
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}
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}
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@ -486,11 +455,8 @@ bool AudioClient::getNamedAudioDeviceForModeExists(QAudio::Mode mode, const QStr
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return (getNamedAudioDeviceForMode(mode, deviceName).deviceName() == deviceName);
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}
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// attempt to use the native sample rate and channel count
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bool nativeFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
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QAudioFormat& audioFormat) {
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bool nativeFormatForAudioDevice(const QAudioDeviceInfo& audioDevice, QAudioFormat& audioFormat) {
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audioFormat = audioDevice.preferredFormat();
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audioFormat.setCodec("audio/pcm");
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@ -513,7 +479,6 @@ bool nativeFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
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bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
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const QAudioFormat& desiredAudioFormat,
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QAudioFormat& adjustedAudioFormat) {
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qCDebug(audioclient) << "The desired format for audio I/O is" << desiredAudioFormat;
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#if defined(Q_OS_ANDROID) || defined(Q_OS_OSX)
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@ -539,12 +504,11 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
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// Attempt the device sample rate and channel count in decreasing order of preference.
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//
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const int sampleRates[] = { 48000, 44100, 32000, 24000, 16000, 96000, 192000, 88200, 176400 };
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const int inputChannels[] = { 1, 2, 4, 6, 8 }; // prefer mono
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const int outputChannels[] = { 2, 4, 6, 8, 1 }; // prefer stereo, downmix as last resort
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const int inputChannels[] = { 1, 2, 4, 6, 8 }; // prefer mono
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const int outputChannels[] = { 2, 4, 6, 8, 1 }; // prefer stereo, downmix as last resort
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for (int channelCount : (desiredAudioFormat.channelCount() == 1 ? inputChannels : outputChannels)) {
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for (int sampleRate : sampleRates) {
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adjustedAudioFormat.setChannelCount(channelCount);
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adjustedAudioFormat.setSampleRate(sampleRate);
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@ -554,11 +518,14 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
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}
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}
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return false; // a supported format could not be found
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return false; // a supported format could not be found
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}
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bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationSamples, unsigned int numSourceSamples,
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const int sourceChannelCount, const int destinationChannelCount) {
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bool sampleChannelConversion(const int16_t* sourceSamples,
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int16_t* destinationSamples,
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unsigned int numSourceSamples,
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const int sourceChannelCount,
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const int destinationChannelCount) {
|
||||
if (sourceChannelCount == 2 && destinationChannelCount == 1) {
|
||||
// loop through the stereo input audio samples and average every two samples
|
||||
for (uint i = 0; i < numSourceSamples; i += 2) {
|
||||
|
|
@ -567,7 +534,6 @@ bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationS
|
|||
|
||||
return true;
|
||||
} else if (sourceChannelCount == 1 && destinationChannelCount == 2) {
|
||||
|
||||
// loop through the mono input audio and repeat each sample twice
|
||||
for (uint i = 0; i < numSourceSamples; ++i) {
|
||||
destinationSamples[i * 2] = destinationSamples[(i * 2) + 1] = sourceSamples[i];
|
||||
|
|
@ -580,32 +546,31 @@ bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationS
|
|||
}
|
||||
|
||||
void possibleResampling(AudioSRC* resampler,
|
||||
const int16_t* sourceSamples, int16_t* destinationSamples,
|
||||
unsigned int numSourceSamples, unsigned int numDestinationSamples,
|
||||
const int sourceChannelCount, const int destinationChannelCount) {
|
||||
|
||||
const int16_t* sourceSamples,
|
||||
int16_t* destinationSamples,
|
||||
unsigned int numSourceSamples,
|
||||
unsigned int numDestinationSamples,
|
||||
const int sourceChannelCount,
|
||||
const int destinationChannelCount) {
|
||||
if (numSourceSamples > 0) {
|
||||
if (!resampler) {
|
||||
if (!sampleChannelConversion(sourceSamples, destinationSamples, numSourceSamples,
|
||||
sourceChannelCount, destinationChannelCount)) {
|
||||
if (!sampleChannelConversion(sourceSamples, destinationSamples, numSourceSamples, sourceChannelCount,
|
||||
destinationChannelCount)) {
|
||||
// no conversion, we can copy the samples directly across
|
||||
memcpy(destinationSamples, sourceSamples, numSourceSamples * AudioConstants::SAMPLE_SIZE);
|
||||
}
|
||||
} else {
|
||||
|
||||
if (sourceChannelCount != destinationChannelCount) {
|
||||
|
||||
int numChannelCoversionSamples = (numSourceSamples * destinationChannelCount) / sourceChannelCount;
|
||||
int16_t* channelConversionSamples = new int16_t[numChannelCoversionSamples];
|
||||
|
||||
sampleChannelConversion(sourceSamples, channelConversionSamples, numSourceSamples,
|
||||
sourceChannelCount, destinationChannelCount);
|
||||
sampleChannelConversion(sourceSamples, channelConversionSamples, numSourceSamples, sourceChannelCount,
|
||||
destinationChannelCount);
|
||||
|
||||
resampler->render(channelConversionSamples, destinationSamples, numChannelCoversionSamples);
|
||||
|
||||
delete[] channelConversionSamples;
|
||||
} else {
|
||||
|
||||
unsigned int numAdjustedSourceSamples = numSourceSamples;
|
||||
unsigned int numAdjustedDestinationSamples = numDestinationSamples;
|
||||
|
||||
|
|
@ -621,7 +586,6 @@ void possibleResampling(AudioSRC* resampler,
|
|||
}
|
||||
|
||||
void AudioClient::start() {
|
||||
|
||||
// set up the desired audio format
|
||||
_desiredInputFormat.setSampleRate(AudioConstants::SAMPLE_RATE);
|
||||
_desiredInputFormat.setSampleSize(16);
|
||||
|
|
@ -710,7 +674,6 @@ void AudioClient::handleAudioDataPacket(QSharedPointer<ReceivedMessage> message)
|
|||
nodeList->flagTimeForConnectionStep(LimitedNodeList::ConnectionStep::ReceiveFirstAudioPacket);
|
||||
|
||||
if (_audioOutput) {
|
||||
|
||||
if (!_hasReceivedFirstPacket) {
|
||||
_hasReceivedFirstPacket = true;
|
||||
|
||||
|
|
@ -727,8 +690,8 @@ void AudioClient::handleAudioDataPacket(QSharedPointer<ReceivedMessage> message)
|
|||
}
|
||||
}
|
||||
|
||||
AudioClient::Gate::Gate(AudioClient* audioClient) :
|
||||
_audioClient(audioClient) {}
|
||||
AudioClient::Gate::Gate(AudioClient* audioClient) : _audioClient(audioClient) {
|
||||
}
|
||||
|
||||
void AudioClient::Gate::setIsSimulatingJitter(bool enable) {
|
||||
std::lock_guard<std::mutex> lock(_mutex);
|
||||
|
|
@ -781,7 +744,6 @@ void AudioClient::Gate::flush() {
|
|||
_index = 0;
|
||||
}
|
||||
|
||||
|
||||
void AudioClient::handleNoisyMutePacket(QSharedPointer<ReceivedMessage> message) {
|
||||
if (!_muted) {
|
||||
setMuted(true);
|
||||
|
|
@ -827,7 +789,6 @@ void AudioClient::handleSelectedAudioFormat(QSharedPointer<ReceivedMessage> mess
|
|||
}
|
||||
|
||||
void AudioClient::selectAudioFormat(const QString& selectedCodecName) {
|
||||
|
||||
_selectedCodecName = selectedCodecName;
|
||||
|
||||
qCDebug(audioclient) << "Selected Codec:" << _selectedCodecName << "isStereoInput:" << _isStereoInput;
|
||||
|
|
@ -845,12 +806,12 @@ void AudioClient::selectAudioFormat(const QString& selectedCodecName) {
|
|||
if (_selectedCodecName == plugin->getName()) {
|
||||
_codec = plugin;
|
||||
_receivedAudioStream.setupCodec(plugin, _selectedCodecName, AudioConstants::STEREO);
|
||||
_encoder = plugin->createEncoder(AudioConstants::SAMPLE_RATE, _isStereoInput ? AudioConstants::STEREO : AudioConstants::MONO);
|
||||
_encoder = plugin->createEncoder(AudioConstants::SAMPLE_RATE,
|
||||
_isStereoInput ? AudioConstants::STEREO : AudioConstants::MONO);
|
||||
qCDebug(audioclient) << "Selected Codec Plugin:" << _codec.get();
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
bool AudioClient::switchAudioDevice(QAudio::Mode mode, const QAudioDeviceInfo& deviceInfo) {
|
||||
|
|
@ -862,7 +823,7 @@ bool AudioClient::switchAudioDevice(QAudio::Mode mode, const QAudioDeviceInfo& d
|
|||
|
||||
if (mode == QAudio::AudioInput) {
|
||||
return switchInputToAudioDevice(device);
|
||||
} else { // if (mode == QAudio::AudioOutput)
|
||||
} else { // if (mode == QAudio::AudioOutput)
|
||||
return switchOutputToAudioDevice(device);
|
||||
}
|
||||
}
|
||||
|
|
@ -904,8 +865,8 @@ void AudioClient::configureReverb() {
|
|||
p.sampleRate = _outputFormat.sampleRate();
|
||||
p.wetDryMix = 100.0f;
|
||||
p.preDelay = 0.0f;
|
||||
p.earlyGain = -96.0f; // disable ER
|
||||
p.lateGain += _reverbOptions->getWetDryMix() * (24.0f/100.0f) - 24.0f; // -0dB to -24dB, based on wetDryMix
|
||||
p.earlyGain = -96.0f; // disable ER
|
||||
p.lateGain += _reverbOptions->getWetDryMix() * (24.0f / 100.0f) - 24.0f; // -0dB to -24dB, based on wetDryMix
|
||||
p.lateMixLeft = 0.0f;
|
||||
p.lateMixRight = 0.0f;
|
||||
|
||||
|
|
@ -915,7 +876,6 @@ void AudioClient::configureReverb() {
|
|||
void AudioClient::updateReverbOptions() {
|
||||
bool reverbChanged = false;
|
||||
if (_receivedAudioStream.hasReverb()) {
|
||||
|
||||
if (_zoneReverbOptions.getReverbTime() != _receivedAudioStream.getRevebTime()) {
|
||||
_zoneReverbOptions.setReverbTime(_receivedAudioStream.getRevebTime());
|
||||
reverbChanged = true;
|
||||
|
|
@ -1020,7 +980,8 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
|
|||
int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
|
||||
|
||||
// upmix mono to stereo
|
||||
if (!sampleChannelConversion(inputSamples, loopbackSamples, numInputSamples, _inputFormat.channelCount(), OUTPUT_CHANNEL_COUNT)) {
|
||||
if (!sampleChannelConversion(inputSamples, loopbackSamples, numInputSamples, _inputFormat.channelCount(),
|
||||
OUTPUT_CHANNEL_COUNT)) {
|
||||
// no conversion, just copy the samples
|
||||
memcpy(loopbackSamples, inputSamples, numInputSamples * AudioConstants::SAMPLE_SIZE);
|
||||
}
|
||||
|
|
@ -1028,17 +989,15 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
|
|||
// apply stereo reverb at the source, to the loopback audio
|
||||
if (!_shouldEchoLocally && hasReverb) {
|
||||
updateReverbOptions();
|
||||
_sourceReverb.render(loopbackSamples, loopbackSamples, numLoopbackSamples/2);
|
||||
_sourceReverb.render(loopbackSamples, loopbackSamples, numLoopbackSamples / 2);
|
||||
}
|
||||
|
||||
// if required, upmix or downmix to deviceChannelCount
|
||||
int deviceChannelCount = _outputFormat.channelCount();
|
||||
if (deviceChannelCount == OUTPUT_CHANNEL_COUNT) {
|
||||
|
||||
_loopbackOutputDevice->write(loopBackByteArray);
|
||||
|
||||
} else {
|
||||
|
||||
static QByteArray deviceByteArray;
|
||||
|
||||
int numDeviceSamples = (numLoopbackSamples * deviceChannelCount) / OUTPUT_CHANNEL_COUNT;
|
||||
|
|
@ -1074,7 +1033,7 @@ void AudioClient::handleAudioInput(QByteArray& audioBuffer) {
|
|||
}
|
||||
|
||||
int32_t loudness = 0;
|
||||
assert(numSamples < 65536); // int32_t loudness cannot overflow
|
||||
assert(numSamples < 65536); // int32_t loudness cannot overflow
|
||||
bool didClip = false;
|
||||
for (int i = 0; i < numSamples; ++i) {
|
||||
const int32_t CLIPPING_THRESHOLD = (int32_t)(AudioConstants::MAX_SAMPLE_VALUE * 0.9f);
|
||||
|
|
@ -1129,13 +1088,14 @@ void AudioClient::handleAudioInput(QByteArray& audioBuffer) {
|
|||
}
|
||||
|
||||
emitAudioPacket(encodedBuffer.data(), encodedBuffer.size(), _outgoingAvatarAudioSequenceNumber, _isStereoInput,
|
||||
audioTransform, avatarBoundingBoxCorner, avatarBoundingBoxScale,
|
||||
packetType, _selectedCodecName);
|
||||
audioTransform, avatarBoundingBoxCorner, avatarBoundingBoxScale, packetType, _selectedCodecName);
|
||||
_stats.sentPacket();
|
||||
}
|
||||
}
|
||||
|
||||
void AudioClient::processAudioAndAddToRingBuffer(QByteArray& inputByteArray, const uchar& channelCount, const qint32& bytesForDuration) {
|
||||
void AudioClient::processAudioAndAddToRingBuffer(QByteArray& inputByteArray,
|
||||
const uchar& channelCount,
|
||||
const qint32& bytesForDuration) {
|
||||
// input samples required to produce exactly NETWORK_FRAME_SAMPLES of output
|
||||
const int inputSamplesRequired =
|
||||
(_inputToNetworkResampler ? _inputToNetworkResampler->getMinInput(AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL)
|
||||
|
|
@ -1189,11 +1149,10 @@ void AudioClient::handleMicAudioInput() {
|
|||
}
|
||||
|
||||
void AudioClient::handleDummyAudioInput() {
|
||||
const int numNetworkBytes = _isStereoInput
|
||||
? AudioConstants::NETWORK_FRAME_BYTES_STEREO
|
||||
: AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL;
|
||||
const int numNetworkBytes =
|
||||
_isStereoInput ? AudioConstants::NETWORK_FRAME_BYTES_STEREO : AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL;
|
||||
|
||||
QByteArray audioBuffer(numNetworkBytes, 0); // silent
|
||||
QByteArray audioBuffer(numNetworkBytes, 0); // silent
|
||||
handleAudioInput(audioBuffer);
|
||||
}
|
||||
|
||||
|
|
@ -1202,13 +1161,59 @@ void AudioClient::handleRecordedAudioInput(const QByteArray& audio) {
|
|||
handleAudioInput(audioBuffer);
|
||||
}
|
||||
|
||||
int rawToWav(const char* rawData, const int& rawLength, const char* wavfn, long frequency) {
|
||||
long chunksize = 0x10;
|
||||
|
||||
struct {
|
||||
unsigned short wFormatTag;
|
||||
unsigned short wChannels;
|
||||
unsigned long dwSamplesPerSec;
|
||||
unsigned long dwAvgBytesPerSec;
|
||||
unsigned short wBlockAlign;
|
||||
unsigned short wBitsPerSample;
|
||||
} fmt;
|
||||
|
||||
long samplecount = rawLength / 2;
|
||||
long riffsize = samplecount * 2 + 0x24;
|
||||
long datasize = samplecount * 2;
|
||||
|
||||
FILE* wav = fopen(wavfn, "wb");
|
||||
if (!wav) {
|
||||
return -3;
|
||||
}
|
||||
|
||||
fwrite("RIFF", 1, 4, wav);
|
||||
fwrite(&riffsize, 4, 1, wav);
|
||||
fwrite("WAVEfmt ", 1, 8, wav);
|
||||
fwrite(&chunksize, 4, 1, wav);
|
||||
|
||||
fmt.wFormatTag = 1; // PCM
|
||||
fmt.wChannels = 1; // MONO
|
||||
fmt.dwSamplesPerSec = frequency * 1;
|
||||
fmt.dwAvgBytesPerSec = frequency * 1 * 2; // 16 bit
|
||||
fmt.wBlockAlign = 2;
|
||||
fmt.wBitsPerSample = 16;
|
||||
|
||||
fwrite(&fmt, sizeof(fmt), 1, wav);
|
||||
fwrite("data", 1, 4, wav);
|
||||
fwrite(&datasize, 4, 1, wav);
|
||||
fwrite(rawData, 1, rawLength, wav);
|
||||
fclose(wav);
|
||||
}
|
||||
|
||||
void AudioClient::handleTTSAudioInput(const QByteArray& audio) {
|
||||
QByteArray audioBuffer(audio);
|
||||
QVector<int16_t> audioBufferReal;
|
||||
|
||||
QString filename = QString::number(usecTimestampNow());
|
||||
QString path = PathUtils::getAppDataPath() + "Audio/" + filename + ".wav";
|
||||
rawToWav(audioBuffer.data(), audioBuffer.size(), path.toLocal8Bit(), 24000);
|
||||
|
||||
while (audioBuffer.size() > 0) {
|
||||
QByteArray part;
|
||||
part.append(audioBuffer.data(), AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
|
||||
audioBuffer.remove(0, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
|
||||
processAudioAndAddToRingBuffer(part, 1, 48);
|
||||
processAudioAndAddToRingBuffer(part, 1, 48);
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -1234,9 +1239,8 @@ void AudioClient::prepareLocalAudioInjectors(std::unique_ptr<Lock> localAudioLoc
|
|||
int bufferCapacity = _localInjectorsStream.getSampleCapacity();
|
||||
int maxOutputSamples = AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * AudioConstants::STEREO;
|
||||
if (_localToOutputResampler) {
|
||||
maxOutputSamples =
|
||||
_localToOutputResampler->getMaxOutput(AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL) *
|
||||
AudioConstants::STEREO;
|
||||
maxOutputSamples = _localToOutputResampler->getMaxOutput(AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL) *
|
||||
AudioConstants::STEREO;
|
||||
}
|
||||
|
||||
samplesNeeded = bufferCapacity - _localSamplesAvailable.load(std::memory_order_relaxed);
|
||||
|
|
@ -1259,7 +1263,7 @@ void AudioClient::prepareLocalAudioInjectors(std::unique_ptr<Lock> localAudioLoc
|
|||
if (_localToOutputResampler) {
|
||||
// resample to output sample rate
|
||||
int frames = _localToOutputResampler->render(_localMixBuffer, _localOutputMixBuffer,
|
||||
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
|
||||
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
|
||||
|
||||
// write to local injectors' ring buffer
|
||||
samples = frames * AudioConstants::STEREO;
|
||||
|
|
@ -1268,8 +1272,7 @@ void AudioClient::prepareLocalAudioInjectors(std::unique_ptr<Lock> localAudioLoc
|
|||
} else {
|
||||
// write to local injectors' ring buffer
|
||||
samples = AudioConstants::NETWORK_FRAME_SAMPLES_STEREO;
|
||||
_localInjectorsStream.writeSamples(_localMixBuffer,
|
||||
AudioConstants::NETWORK_FRAME_SAMPLES_STEREO);
|
||||
_localInjectorsStream.writeSamples(_localMixBuffer, AudioConstants::NETWORK_FRAME_SAMPLES_STEREO);
|
||||
}
|
||||
|
||||
_localSamplesAvailable.fetch_add(samples, std::memory_order_release);
|
||||
|
|
@ -1294,18 +1297,16 @@ bool AudioClient::mixLocalAudioInjectors(float* mixBuffer) {
|
|||
// the lock guarantees that injectorBuffer, if found, is invariant
|
||||
AudioInjectorLocalBuffer* injectorBuffer = injector->getLocalBuffer();
|
||||
if (injectorBuffer) {
|
||||
|
||||
static const int HRTF_DATASET_INDEX = 1;
|
||||
|
||||
int numChannels = injector->isAmbisonic() ? AudioConstants::AMBISONIC : (injector->isStereo() ? AudioConstants::STEREO : AudioConstants::MONO);
|
||||
int numChannels = injector->isAmbisonic() ? AudioConstants::AMBISONIC
|
||||
: (injector->isStereo() ? AudioConstants::STEREO : AudioConstants::MONO);
|
||||
size_t bytesToRead = numChannels * AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL;
|
||||
|
||||
// get one frame from the injector
|
||||
memset(_localScratchBuffer, 0, bytesToRead);
|
||||
if (0 < injectorBuffer->readData((char*)_localScratchBuffer, bytesToRead)) {
|
||||
|
||||
if (injector->isAmbisonic()) {
|
||||
|
||||
// no distance attenuation
|
||||
float gain = injector->getVolume();
|
||||
|
||||
|
|
@ -1322,11 +1323,10 @@ bool AudioClient::mixLocalAudioInjectors(float* mixBuffer) {
|
|||
float qz = relativeOrientation.y;
|
||||
|
||||
// Ambisonic gets spatialized into mixBuffer
|
||||
injector->getLocalFOA().render(_localScratchBuffer, mixBuffer, HRTF_DATASET_INDEX,
|
||||
qw, qx, qy, qz, gain, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
|
||||
injector->getLocalFOA().render(_localScratchBuffer, mixBuffer, HRTF_DATASET_INDEX, qw, qx, qy, qz, gain,
|
||||
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
|
||||
|
||||
} else if (injector->isStereo()) {
|
||||
|
||||
// stereo gets directly mixed into mixBuffer
|
||||
float gain = injector->getVolume();
|
||||
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i++) {
|
||||
|
|
@ -1334,7 +1334,6 @@ bool AudioClient::mixLocalAudioInjectors(float* mixBuffer) {
|
|||
}
|
||||
|
||||
} else {
|
||||
|
||||
// calculate distance, gain and azimuth for hrtf
|
||||
glm::vec3 relativePosition = injector->getPosition() - _positionGetter();
|
||||
float distance = glm::max(glm::length(relativePosition), EPSILON);
|
||||
|
|
@ -1342,19 +1341,17 @@ bool AudioClient::mixLocalAudioInjectors(float* mixBuffer) {
|
|||
float azimuth = azimuthForSource(relativePosition);
|
||||
|
||||
// mono gets spatialized into mixBuffer
|
||||
injector->getLocalHRTF().render(_localScratchBuffer, mixBuffer, HRTF_DATASET_INDEX,
|
||||
azimuth, distance, gain, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
|
||||
injector->getLocalHRTF().render(_localScratchBuffer, mixBuffer, HRTF_DATASET_INDEX, azimuth, distance, gain,
|
||||
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
|
||||
}
|
||||
|
||||
} else {
|
||||
|
||||
qCDebug(audioclient) << "injector has no more data, marking finished for removal";
|
||||
injector->finishLocalInjection();
|
||||
injectorsToRemove.append(injector);
|
||||
}
|
||||
|
||||
} else {
|
||||
|
||||
qCDebug(audioclient) << "injector has no local buffer, marking as finished for removal";
|
||||
injector->finishLocalInjection();
|
||||
injectorsToRemove.append(injector);
|
||||
|
|
@ -1373,7 +1370,6 @@ bool AudioClient::mixLocalAudioInjectors(float* mixBuffer) {
|
|||
}
|
||||
|
||||
void AudioClient::processReceivedSamples(const QByteArray& decodedBuffer, QByteArray& outputBuffer) {
|
||||
|
||||
const int16_t* decodedSamples = reinterpret_cast<const int16_t*>(decodedBuffer.data());
|
||||
assert(decodedBuffer.size() == AudioConstants::NETWORK_FRAME_BYTES_STEREO);
|
||||
|
||||
|
|
@ -1442,7 +1438,6 @@ void AudioClient::setNoiseReduction(bool enable, bool emitSignal) {
|
|||
}
|
||||
}
|
||||
|
||||
|
||||
bool AudioClient::setIsStereoInput(bool isStereoInput) {
|
||||
bool stereoInputChanged = false;
|
||||
if (isStereoInput != _isStereoInput && _inputDeviceInfo.supportedChannelCounts().contains(2)) {
|
||||
|
|
@ -1460,7 +1455,8 @@ bool AudioClient::setIsStereoInput(bool isStereoInput) {
|
|||
if (_encoder) {
|
||||
_codec->releaseEncoder(_encoder);
|
||||
}
|
||||
_encoder = _codec->createEncoder(AudioConstants::SAMPLE_RATE, _isStereoInput ? AudioConstants::STEREO : AudioConstants::MONO);
|
||||
_encoder = _codec->createEncoder(AudioConstants::SAMPLE_RATE,
|
||||
_isStereoInput ? AudioConstants::STEREO : AudioConstants::MONO);
|
||||
}
|
||||
qCDebug(audioclient) << "Reset Codec:" << _selectedCodecName << "isStereoInput:" << _isStereoInput;
|
||||
|
||||
|
|
@ -1500,7 +1496,7 @@ bool AudioClient::outputLocalInjector(const AudioInjectorPointer& injector) {
|
|||
|
||||
void AudioClient::outputFormatChanged() {
|
||||
_outputFrameSize = (AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * OUTPUT_CHANNEL_COUNT * _outputFormat.sampleRate()) /
|
||||
_desiredOutputFormat.sampleRate();
|
||||
_desiredOutputFormat.sampleRate();
|
||||
_receivedAudioStream.outputFormatChanged(_outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT);
|
||||
}
|
||||
|
||||
|
|
@ -1514,7 +1510,7 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo inputDeviceInf
|
|||
Lock lock(_deviceMutex);
|
||||
|
||||
#if defined(Q_OS_ANDROID)
|
||||
_shouldRestartInputSetup = false; // avoid a double call to _audioInput->start() from audioInputStateChanged
|
||||
_shouldRestartInputSetup = false; // avoid a double call to _audioInput->start() from audioInputStateChanged
|
||||
#endif
|
||||
|
||||
// cleanup any previously initialized device
|
||||
|
|
@ -1565,15 +1561,15 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo inputDeviceInf
|
|||
|
||||
// we've got the best we can get for input
|
||||
// if required, setup a resampler for this input to our desired network format
|
||||
if (_inputFormat != _desiredInputFormat
|
||||
&& _inputFormat.sampleRate() != _desiredInputFormat.sampleRate()) {
|
||||
if (_inputFormat != _desiredInputFormat && _inputFormat.sampleRate() != _desiredInputFormat.sampleRate()) {
|
||||
qCDebug(audioclient) << "Attemping to create a resampler for input format to network format.";
|
||||
|
||||
assert(_inputFormat.sampleSize() == 16);
|
||||
assert(_desiredInputFormat.sampleSize() == 16);
|
||||
int channelCount = (_inputFormat.channelCount() == 2 && _desiredInputFormat.channelCount() == 2) ? 2 : 1;
|
||||
|
||||
_inputToNetworkResampler = new AudioSRC(_inputFormat.sampleRate(), _desiredInputFormat.sampleRate(), channelCount);
|
||||
_inputToNetworkResampler =
|
||||
new AudioSRC(_inputFormat.sampleRate(), _desiredInputFormat.sampleRate(), channelCount);
|
||||
|
||||
} else {
|
||||
qCDebug(audioclient) << "No resampling required for audio input to match desired network format.";
|
||||
|
|
@ -1607,7 +1603,7 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo inputDeviceInf
|
|||
connect(_inputDevice, SIGNAL(readyRead()), this, SLOT(handleMicAudioInput()));
|
||||
supportedFormat = true;
|
||||
} else {
|
||||
qCDebug(audioclient) << "Error starting audio input -" << _audioInput->error();
|
||||
qCDebug(audioclient) << "Error starting audio input -" << _audioInput->error();
|
||||
_audioInput->deleteLater();
|
||||
_audioInput = NULL;
|
||||
}
|
||||
|
|
@ -1677,7 +1673,7 @@ void AudioClient::checkInputTimeout() {
|
|||
void AudioClient::setHeadsetPluggedIn(bool pluggedIn) {
|
||||
#if defined(Q_OS_ANDROID)
|
||||
if (pluggedIn == !_isHeadsetPluggedIn && !_inputDeviceInfo.isNull()) {
|
||||
QAndroidJniObject brand = QAndroidJniObject::getStaticObjectField<jstring>("android/os/Build", "BRAND");
|
||||
QAndroidJniObject brand = QAndroidJniObject::getStaticObjectField<jstring>("android/os/Build", "BRAND");
|
||||
// some samsung phones needs more time to shutdown the previous input device
|
||||
if (brand.toString().contains("samsung", Qt::CaseInsensitive)) {
|
||||
switchInputToAudioDevice(QAudioDeviceInfo(), true);
|
||||
|
|
@ -1715,8 +1711,8 @@ void AudioClient::outputNotify() {
|
|||
int newOutputBufferSizeFrames = setOutputBufferSize(oldOutputBufferSizeFrames + 1, false);
|
||||
|
||||
if (newOutputBufferSizeFrames > oldOutputBufferSizeFrames) {
|
||||
qCDebug(audioclient,
|
||||
"Starve threshold surpassed (%d starves in %d ms)", _outputStarveDetectionCount, dt);
|
||||
qCDebug(audioclient, "Starve threshold surpassed (%d starves in %d ms)", _outputStarveDetectionCount,
|
||||
dt);
|
||||
}
|
||||
|
||||
_outputStarveDetectionStartTimeMsec = now;
|
||||
|
|
@ -1730,7 +1726,8 @@ void AudioClient::outputNotify() {
|
|||
bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceInfo, bool isShutdownRequest) {
|
||||
Q_ASSERT_X(QThread::currentThread() == thread(), Q_FUNC_INFO, "Function invoked on wrong thread");
|
||||
|
||||
qCDebug(audioclient) << "AudioClient::switchOutputToAudioDevice() outputDeviceInfo: [" << outputDeviceInfo.deviceName() << "]";
|
||||
qCDebug(audioclient) << "AudioClient::switchOutputToAudioDevice() outputDeviceInfo: [" << outputDeviceInfo.deviceName()
|
||||
<< "]";
|
||||
bool supportedFormat = false;
|
||||
|
||||
// NOTE: device start() uses the Qt internal device list
|
||||
|
|
@ -1789,15 +1786,16 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceI
|
|||
|
||||
// we've got the best we can get for input
|
||||
// if required, setup a resampler for this input to our desired network format
|
||||
if (_desiredOutputFormat != _outputFormat
|
||||
&& _desiredOutputFormat.sampleRate() != _outputFormat.sampleRate()) {
|
||||
if (_desiredOutputFormat != _outputFormat && _desiredOutputFormat.sampleRate() != _outputFormat.sampleRate()) {
|
||||
qCDebug(audioclient) << "Attemping to create a resampler for network format to output format.";
|
||||
|
||||
assert(_desiredOutputFormat.sampleSize() == 16);
|
||||
assert(_outputFormat.sampleSize() == 16);
|
||||
|
||||
_networkToOutputResampler = new AudioSRC(_desiredOutputFormat.sampleRate(), _outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT);
|
||||
_localToOutputResampler = new AudioSRC(_desiredOutputFormat.sampleRate(), _outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT);
|
||||
_networkToOutputResampler =
|
||||
new AudioSRC(_desiredOutputFormat.sampleRate(), _outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT);
|
||||
_localToOutputResampler =
|
||||
new AudioSRC(_desiredOutputFormat.sampleRate(), _outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT);
|
||||
|
||||
} else {
|
||||
qCDebug(audioclient) << "No resampling required for network output to match actual output format.";
|
||||
|
|
@ -1809,7 +1807,9 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceI
|
|||
_audioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this);
|
||||
|
||||
int deviceChannelCount = _outputFormat.channelCount();
|
||||
int frameSize = (AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * deviceChannelCount * _outputFormat.sampleRate()) / _desiredOutputFormat.sampleRate();
|
||||
int frameSize =
|
||||
(AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * deviceChannelCount * _outputFormat.sampleRate()) /
|
||||
_desiredOutputFormat.sampleRate();
|
||||
int requestedSize = _sessionOutputBufferSizeFrames * frameSize * AudioConstants::SAMPLE_SIZE;
|
||||
_audioOutput->setBufferSize(requestedSize);
|
||||
|
||||
|
|
@ -1825,7 +1825,10 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceI
|
|||
_outputScratchBuffer = new int16_t[_outputPeriod];
|
||||
|
||||
// size local output mix buffer based on resampled network frame size
|
||||
int networkPeriod = _localToOutputResampler ? _localToOutputResampler->getMaxOutput(AudioConstants::NETWORK_FRAME_SAMPLES_STEREO) : AudioConstants::NETWORK_FRAME_SAMPLES_STEREO;
|
||||
int networkPeriod =
|
||||
_localToOutputResampler
|
||||
? _localToOutputResampler->getMaxOutput(AudioConstants::NETWORK_FRAME_SAMPLES_STEREO)
|
||||
: AudioConstants::NETWORK_FRAME_SAMPLES_STEREO;
|
||||
_localOutputMixBuffer = new float[networkPeriod];
|
||||
|
||||
// local period should be at least twice the output period,
|
||||
|
|
@ -1875,7 +1878,8 @@ int AudioClient::setOutputBufferSize(int numFrames, bool persist) {
|
|||
qCDebug(audioclient) << __FUNCTION__ << "numFrames:" << numFrames << "persist:" << persist;
|
||||
|
||||
numFrames = std::min(std::max(numFrames, MIN_BUFFER_FRAMES), MAX_BUFFER_FRAMES);
|
||||
qCDebug(audioclient) << __FUNCTION__ << "clamped numFrames:" << numFrames << "_sessionOutputBufferSizeFrames:" << _sessionOutputBufferSizeFrames;
|
||||
qCDebug(audioclient) << __FUNCTION__ << "clamped numFrames:" << numFrames
|
||||
<< "_sessionOutputBufferSizeFrames:" << _sessionOutputBufferSizeFrames;
|
||||
|
||||
if (numFrames != _sessionOutputBufferSizeFrames) {
|
||||
qCInfo(audioclient, "Audio output buffer set to %d frames", numFrames);
|
||||
|
|
@ -1906,10 +1910,10 @@ const float AudioClient::CALLBACK_ACCELERATOR_RATIO = 2.0f;
|
|||
#endif
|
||||
|
||||
int AudioClient::calculateNumberOfInputCallbackBytes(const QAudioFormat& format) const {
|
||||
int numInputCallbackBytes = (int)(((AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL
|
||||
* format.channelCount()
|
||||
* ((float) format.sampleRate() / AudioConstants::SAMPLE_RATE))
|
||||
/ CALLBACK_ACCELERATOR_RATIO) + 0.5f);
|
||||
int numInputCallbackBytes = (int)(((AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL * format.channelCount() *
|
||||
((float)format.sampleRate() / AudioConstants::SAMPLE_RATE)) /
|
||||
CALLBACK_ACCELERATOR_RATIO) +
|
||||
0.5f);
|
||||
|
||||
return numInputCallbackBytes;
|
||||
}
|
||||
|
|
@ -1931,10 +1935,9 @@ float AudioClient::azimuthForSource(const glm::vec3& relativePosition) {
|
|||
|
||||
float rotatedSourcePositionLength2 = glm::length2(rotatedSourcePosition);
|
||||
if (rotatedSourcePositionLength2 > SOURCE_DISTANCE_THRESHOLD) {
|
||||
|
||||
// produce an oriented angle about the y-axis
|
||||
glm::vec3 direction = rotatedSourcePosition * (1.0f / fastSqrtf(rotatedSourcePositionLength2));
|
||||
float angle = fastAcosf(glm::clamp(-direction.z, -1.0f, 1.0f)); // UNIT_NEG_Z is "forward"
|
||||
float angle = fastAcosf(glm::clamp(-direction.z, -1.0f, 1.0f)); // UNIT_NEG_Z is "forward"
|
||||
return (direction.x < 0.0f) ? -angle : angle;
|
||||
|
||||
} else {
|
||||
|
|
@ -1944,7 +1947,6 @@ float AudioClient::azimuthForSource(const glm::vec3& relativePosition) {
|
|||
}
|
||||
|
||||
float AudioClient::gainForSource(float distance, float volume) {
|
||||
|
||||
// attenuation = -6dB * log2(distance)
|
||||
// reference attenuation of 0dB at distance = 1.0m
|
||||
float gain = volume / std::max(distance, HRTF_NEARFIELD_MIN);
|
||||
|
|
@ -1952,8 +1954,7 @@ float AudioClient::gainForSource(float distance, float volume) {
|
|||
return gain;
|
||||
}
|
||||
|
||||
qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
|
||||
|
||||
qint64 AudioClient::AudioOutputIODevice::readData(char* data, qint64 maxSize) {
|
||||
// samples requested from OUTPUT_CHANNEL_COUNT
|
||||
int deviceChannelCount = _audio->_outputFormat.channelCount();
|
||||
int samplesRequested = (int)(maxSize / AudioConstants::SAMPLE_SIZE) * OUTPUT_CHANNEL_COUNT / deviceChannelCount;
|
||||
|
|
@ -1965,7 +1966,8 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
|
|||
|
||||
int networkSamplesPopped;
|
||||
if ((networkSamplesPopped = _receivedAudioStream.popSamples(samplesRequested, false)) > 0) {
|
||||
qCDebug(audiostream, "Read %d samples from buffer (%d available, %d requested)", networkSamplesPopped, _receivedAudioStream.getSamplesAvailable(), samplesRequested);
|
||||
qCDebug(audiostream, "Read %d samples from buffer (%d available, %d requested)", networkSamplesPopped,
|
||||
_receivedAudioStream.getSamplesAvailable(), samplesRequested);
|
||||
AudioRingBuffer::ConstIterator lastPopOutput = _receivedAudioStream.getLastPopOutput();
|
||||
lastPopOutput.readSamples(scratchBuffer, networkSamplesPopped);
|
||||
for (int i = 0; i < networkSamplesPopped; i++) {
|
||||
|
|
@ -1997,14 +1999,13 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
|
|||
samplesRequested = std::min(samplesRequested, samplesAvailable);
|
||||
if ((injectorSamplesPopped = _localInjectorsStream.appendSamples(mixBuffer, samplesRequested, append)) > 0) {
|
||||
_audio->_localSamplesAvailable.fetch_sub(injectorSamplesPopped, std::memory_order_release);
|
||||
qCDebug(audiostream, "Read %d samples from injectors (%d available, %d requested)", injectorSamplesPopped, _localInjectorsStream.samplesAvailable(), samplesRequested);
|
||||
qCDebug(audiostream, "Read %d samples from injectors (%d available, %d requested)", injectorSamplesPopped,
|
||||
_localInjectorsStream.samplesAvailable(), samplesRequested);
|
||||
}
|
||||
}
|
||||
|
||||
// prepare injectors for the next callback
|
||||
QtConcurrent::run(QThreadPool::globalInstance(), [this] {
|
||||
_audio->prepareLocalAudioInjectors();
|
||||
});
|
||||
QtConcurrent::run(QThreadPool::globalInstance(), [this] { _audio->prepareLocalAudioInjectors(); });
|
||||
|
||||
int samplesPopped = std::max(networkSamplesPopped, injectorSamplesPopped);
|
||||
int framesPopped = samplesPopped / AudioConstants::STEREO;
|
||||
|
|
@ -2038,7 +2039,6 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
|
|||
_audio->_audioFileWav.addRawAudioChunk(reinterpret_cast<char*>(scratchBuffer), bytesWritten);
|
||||
}
|
||||
|
||||
|
||||
int bytesAudioOutputUnplayed = _audio->_audioOutput->bufferSize() - _audio->_audioOutput->bytesFree();
|
||||
float msecsAudioOutputUnplayed = bytesAudioOutputUnplayed / (float)_audio->_outputFormat.bytesForDuration(USECS_PER_MSEC);
|
||||
_audio->_stats.updateOutputMsUnplayed(msecsAudioOutputUnplayed);
|
||||
|
|
@ -2075,7 +2075,6 @@ void AudioClient::loadSettings() {
|
|||
for (auto& plugin : codecPlugins) {
|
||||
qCDebug(audioclient) << "Codec available:" << plugin->getName();
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
void AudioClient::saveSettings() {
|
||||
|
|
@ -2088,7 +2087,6 @@ void AudioClient::setAvatarBoundingBoxParameters(glm::vec3 corner, glm::vec3 sca
|
|||
avatarBoundingBoxScale = scale;
|
||||
}
|
||||
|
||||
|
||||
void AudioClient::startThread() {
|
||||
moveToNewNamedThread(this, "Audio Thread", [this] { start(); }, QThread::TimeCriticalPriority);
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in a new issue