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Just to make sure, writes data back to a WAV file

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Zach Fox 2018-10-12 12:14:51 -07:00
parent 53226e7924
commit 34befd4a52
1 changed files with 188 additions and 190 deletions
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378
libraries/audio-client/src/AudioClient.cpp
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@ -38,6 +38,7 @@
#include <QtCore/QBuffer> #include <QtCore/QBuffer>
#include <QtMultimedia/QAudioInput> #include <QtMultimedia/QAudioInput>
#include <QtMultimedia/QAudioOutput> #include <QtMultimedia/QAudioOutput>
#include <PathUtils.h>
#include <ThreadHelpers.h> #include <ThreadHelpers.h>
#include <NodeList.h> #include <NodeList.h>
@ -67,7 +68,7 @@ static const int CHECK_INPUT_READS_MSECS = 2000;
static const int MIN_READS_TO_CONSIDER_INPUT_ALIVE = 10; static const int MIN_READS_TO_CONSIDER_INPUT_ALIVE = 10;
#endif #endif
static const auto DEFAULT_POSITION_GETTER = []{ return Vectors::ZERO; }; static const auto DEFAULT_POSITION_GETTER = [] { return Vectors::ZERO; };
static const auto DEFAULT_ORIENTATION_GETTER = [] { return Quaternions::IDENTITY; }; static const auto DEFAULT_ORIENTATION_GETTER = [] { return Quaternions::IDENTITY; };
static const int DEFAULT_BUFFER_FRAMES = 1; static const int DEFAULT_BUFFER_FRAMES = 1;
@ -78,12 +79,11 @@ static const int OUTPUT_CHANNEL_COUNT = 2;
static const bool DEFAULT_STARVE_DETECTION_ENABLED = true; static const bool DEFAULT_STARVE_DETECTION_ENABLED = true;
static const int STARVE_DETECTION_THRESHOLD = 3; static const int STARVE_DETECTION_THRESHOLD = 3;
static const int STARVE_DETECTION_PERIOD = 10 * 1000; // 10 Seconds static const int STARVE_DETECTION_PERIOD = 10 * 1000; // 10 Seconds
Setting::Handle<bool> dynamicJitterBufferEnabled("dynamicJitterBuffersEnabled", Setting::Handle<bool> dynamicJitterBufferEnabled("dynamicJitterBuffersEnabled",
InboundAudioStream::DEFAULT_DYNAMIC_JITTER_BUFFER_ENABLED); InboundAudioStream::DEFAULT_DYNAMIC_JITTER_BUFFER_ENABLED);
Setting::Handle<int> staticJitterBufferFrames("staticJitterBufferFrames", Setting::Handle<int> staticJitterBufferFrames("staticJitterBufferFrames", InboundAudioStream::DEFAULT_STATIC_JITTER_FRAMES);
InboundAudioStream::DEFAULT_STATIC_JITTER_FRAMES);
// protect the Qt internal device list // protect the Qt internal device list
using Mutex = std::mutex; using Mutex = std::mutex;
@ -127,7 +127,7 @@ QAudioDeviceInfo AudioClient::getActiveAudioDevice(QAudio::Mode mode) const {
if (mode == QAudio::AudioInput) { if (mode == QAudio::AudioInput) {
return _inputDeviceInfo; return _inputDeviceInfo;
} else { // if (mode == QAudio::AudioOutput) } else { // if (mode == QAudio::AudioOutput)
return _outputDeviceInfo; return _outputDeviceInfo;
} }
} }
@ -137,14 +137,13 @@ QList<QAudioDeviceInfo> AudioClient::getAudioDevices(QAudio::Mode mode) const {
if (mode == QAudio::AudioInput) { if (mode == QAudio::AudioInput) {
return _inputDevices; return _inputDevices;
} else { // if (mode == QAudio::AudioOutput) } else { // if (mode == QAudio::AudioOutput)
return _outputDevices; return _outputDevices;
} }
} }
static void channelUpmix(int16_t* source, int16_t* dest, int numSamples, int numExtraChannels) { static void channelUpmix(int16_t* source, int16_t* dest, int numSamples, int numExtraChannels) {
for (int i = 0; i < numSamples/2; i++) { for (int i = 0; i < numSamples / 2; i++) {
// read 2 samples // read 2 samples
int16_t left = *source++; int16_t left = *source++;
int16_t right = *source++; int16_t right = *source++;
@ -159,8 +158,7 @@ static void channelUpmix(int16_t* source, int16_t* dest, int numSamples, int num
} }
static void channelDownmix(int16_t* source, int16_t* dest, int numSamples) { static void channelDownmix(int16_t* source, int16_t* dest, int numSamples) {
for (int i = 0; i < numSamples/2; i++) { for (int i = 0; i < numSamples / 2; i++) {
// read 2 samples // read 2 samples
int16_t left = *source++; int16_t left = *source++;
int16_t right = *source++; int16_t right = *source++;
@ -175,48 +173,22 @@ static inline float convertToFloat(int16_t sample) {
} }
AudioClient::AudioClient() : AudioClient::AudioClient() :
AbstractAudioInterface(), AbstractAudioInterface(), _gate(this), _audioInput(NULL), _dummyAudioInput(NULL), _desiredInputFormat(), _inputFormat(),
_gate(this), _numInputCallbackBytes(0), _audioOutput(NULL), _desiredOutputFormat(), _outputFormat(), _outputFrameSize(0),
_audioInput(NULL), _numOutputCallbackBytes(0), _loopbackAudioOutput(NULL), _loopbackOutputDevice(NULL), _inputRingBuffer(0),
_dummyAudioInput(NULL), _localInjectorsStream(0, 1), _receivedAudioStream(RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES), _isStereoInput(false),
_desiredInputFormat(), _outputStarveDetectionStartTimeMsec(0), _outputStarveDetectionCount(0),
_inputFormat(),
_numInputCallbackBytes(0),
_audioOutput(NULL),
_desiredOutputFormat(),
_outputFormat(),
_outputFrameSize(0),
_numOutputCallbackBytes(0),
_loopbackAudioOutput(NULL),
_loopbackOutputDevice(NULL),
_inputRingBuffer(0),
_localInjectorsStream(0, 1),
_receivedAudioStream(RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES),
_isStereoInput(false),
_outputStarveDetectionStartTimeMsec(0),
_outputStarveDetectionCount(0),
_outputBufferSizeFrames("audioOutputBufferFrames", DEFAULT_BUFFER_FRAMES), _outputBufferSizeFrames("audioOutputBufferFrames", DEFAULT_BUFFER_FRAMES),
_sessionOutputBufferSizeFrames(_outputBufferSizeFrames.get()), _sessionOutputBufferSizeFrames(_outputBufferSizeFrames.get()),
_outputStarveDetectionEnabled("audioOutputStarveDetectionEnabled", DEFAULT_STARVE_DETECTION_ENABLED), _outputStarveDetectionEnabled("audioOutputStarveDetectionEnabled", DEFAULT_STARVE_DETECTION_ENABLED),
_lastInputLoudness(0.0f), _lastInputLoudness(0.0f), _timeSinceLastClip(-1.0f), _muted(false), _shouldEchoLocally(false), _shouldEchoToServer(false),
_timeSinceLastClip(-1.0f), _isNoiseGateEnabled(true), _reverb(false), _reverbOptions(&_scriptReverbOptions), _inputToNetworkResampler(NULL),
_muted(false), _networkToOutputResampler(NULL), _localToOutputResampler(NULL),
_shouldEchoLocally(false), _audioLimiter(AudioConstants::SAMPLE_RATE, OUTPUT_CHANNEL_COUNT), _outgoingAvatarAudioSequenceNumber(0),
_shouldEchoToServer(false), _audioOutputIODevice(_localInjectorsStream, _receivedAudioStream, this), _stats(&_receivedAudioStream),
_isNoiseGateEnabled(true),
_reverb(false),
_reverbOptions(&_scriptReverbOptions),
_inputToNetworkResampler(NULL),
_networkToOutputResampler(NULL),
_localToOutputResampler(NULL),
_audioLimiter(AudioConstants::SAMPLE_RATE, OUTPUT_CHANNEL_COUNT),
_outgoingAvatarAudioSequenceNumber(0),
_audioOutputIODevice(_localInjectorsStream, _receivedAudioStream, this),
_stats(&_receivedAudioStream),
_positionGetter(DEFAULT_POSITION_GETTER), _positionGetter(DEFAULT_POSITION_GETTER),
#if defined(Q_OS_ANDROID) #if defined(Q_OS_ANDROID)
_checkInputTimer(this), _checkInputTimer(this), _isHeadsetPluggedIn(false),
_isHeadsetPluggedIn(false),
#endif #endif
_orientationGetter(DEFAULT_ORIENTATION_GETTER) { _orientationGetter(DEFAULT_ORIENTATION_GETTER) {
// avoid putting a lock in the device callback // avoid putting a lock in the device callback
@ -226,16 +198,20 @@ AudioClient::AudioClient() :
{ {
Setting::Handle<int>::Deprecated("maxFramesOverDesired", InboundAudioStream::MAX_FRAMES_OVER_DESIRED); Setting::Handle<int>::Deprecated("maxFramesOverDesired", InboundAudioStream::MAX_FRAMES_OVER_DESIRED);
Setting::Handle<int>::Deprecated("windowStarveThreshold", InboundAudioStream::WINDOW_STARVE_THRESHOLD); Setting::Handle<int>::Deprecated("windowStarveThreshold", InboundAudioStream::WINDOW_STARVE_THRESHOLD);
Setting::Handle<int>::Deprecated("windowSecondsForDesiredCalcOnTooManyStarves", InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_CALC_ON_TOO_MANY_STARVES); Setting::Handle<int>::Deprecated("windowSecondsForDesiredCalcOnTooManyStarves",
Setting::Handle<int>::Deprecated("windowSecondsForDesiredReduction", InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_REDUCTION); InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_CALC_ON_TOO_MANY_STARVES);
Setting::Handle<int>::Deprecated("windowSecondsForDesiredReduction",
InboundAudioStream::WINDOW_SECONDS_FOR_DESIRED_REDUCTION);
Setting::Handle<bool>::Deprecated("useStDevForJitterCalc", InboundAudioStream::USE_STDEV_FOR_JITTER); Setting::Handle<bool>::Deprecated("useStDevForJitterCalc", InboundAudioStream::USE_STDEV_FOR_JITTER);
Setting::Handle<bool>::Deprecated("repetitionWithFade", InboundAudioStream::REPETITION_WITH_FADE); Setting::Handle<bool>::Deprecated("repetitionWithFade", InboundAudioStream::REPETITION_WITH_FADE);
} }
connect(&_receivedAudioStream, &MixedProcessedAudioStream::processSamples, connect(&_receivedAudioStream, &MixedProcessedAudioStream::processSamples, this, &AudioClient::processReceivedSamples,
this, &AudioClient::processReceivedSamples, Qt::DirectConnection); Qt::DirectConnection);
connect(this, &AudioClient::changeDevice, this, [=](const QAudioDeviceInfo& outputDeviceInfo) { connect(this, &AudioClient::changeDevice, this, [=](const QAudioDeviceInfo& outputDeviceInfo) {
qCDebug(audioclient) << "got AudioClient::changeDevice signal, about to call switchOutputToAudioDevice() outputDeviceInfo: [" << outputDeviceInfo.deviceName() << "]"; qCDebug(audioclient)
<< "got AudioClient::changeDevice signal, about to call switchOutputToAudioDevice() outputDeviceInfo: ["
<< outputDeviceInfo.deviceName() << "]";
switchOutputToAudioDevice(outputDeviceInfo); switchOutputToAudioDevice(outputDeviceInfo);
}); });
@ -244,20 +220,18 @@ AudioClient::AudioClient() :
// initialize wasapi; if getAvailableDevices is called from the CheckDevicesThread before this, it will crash // initialize wasapi; if getAvailableDevices is called from the CheckDevicesThread before this, it will crash
getAvailableDevices(QAudio::AudioInput); getAvailableDevices(QAudio::AudioInput);
getAvailableDevices(QAudio::AudioOutput); getAvailableDevices(QAudio::AudioOutput);
// start a thread to detect any device changes // start a thread to detect any device changes
_checkDevicesTimer = new QTimer(this); _checkDevicesTimer = new QTimer(this);
connect(_checkDevicesTimer, &QTimer::timeout, this, [this] { connect(_checkDevicesTimer, &QTimer::timeout, this,
QtConcurrent::run(QThreadPool::globalInstance(), [this] { checkDevices(); }); [this] { QtConcurrent::run(QThreadPool::globalInstance(), [this] { checkDevices(); }); });
});
const unsigned long DEVICE_CHECK_INTERVAL_MSECS = 2 * 1000; const unsigned long DEVICE_CHECK_INTERVAL_MSECS = 2 * 1000;
_checkDevicesTimer->start(DEVICE_CHECK_INTERVAL_MSECS); _checkDevicesTimer->start(DEVICE_CHECK_INTERVAL_MSECS);
// start a thread to detect peak value changes // start a thread to detect peak value changes
_checkPeakValuesTimer = new QTimer(this); _checkPeakValuesTimer = new QTimer(this);
connect(_checkPeakValuesTimer, &QTimer::timeout, this, [this] { connect(_checkPeakValuesTimer, &QTimer::timeout, this,
QtConcurrent::run(QThreadPool::globalInstance(), [this] { checkPeakValues(); }); [this] { QtConcurrent::run(QThreadPool::globalInstance(), [this] { checkPeakValues(); }); });
});
const unsigned long PEAK_VALUES_CHECK_INTERVAL_MSECS = 50; const unsigned long PEAK_VALUES_CHECK_INTERVAL_MSECS = 50;
_checkPeakValuesTimer->start(PEAK_VALUES_CHECK_INTERVAL_MSECS); _checkPeakValuesTimer->start(PEAK_VALUES_CHECK_INTERVAL_MSECS);
@ -289,11 +263,11 @@ void AudioClient::customDeleter() {
} }
void AudioClient::handleMismatchAudioFormat(SharedNodePointer node, const QString& currentCodec, const QString& recievedCodec) { void AudioClient::handleMismatchAudioFormat(SharedNodePointer node, const QString& currentCodec, const QString& recievedCodec) {
qCDebug(audioclient) << __FUNCTION__ << "sendingNode:" << *node << "currentCodec:" << currentCodec << "recievedCodec:" << recievedCodec; qCDebug(audioclient) << __FUNCTION__ << "sendingNode:" << *node << "currentCodec:" << currentCodec
<< "recievedCodec:" << recievedCodec;
selectAudioFormat(recievedCodec); selectAudioFormat(recievedCodec);
} }
void AudioClient::reset() { void AudioClient::reset() {
_receivedAudioStream.reset(); _receivedAudioStream.reset();
_stats.reset(); _stats.reset();
@ -321,7 +295,7 @@ void AudioClient::setAudioPaused(bool pause) {
QAudioDeviceInfo getNamedAudioDeviceForMode(QAudio::Mode mode, const QString& deviceName) { QAudioDeviceInfo getNamedAudioDeviceForMode(QAudio::Mode mode, const QString& deviceName) {
QAudioDeviceInfo result; QAudioDeviceInfo result;
foreach(QAudioDeviceInfo audioDevice, getAvailableDevices(mode)) { foreach (QAudioDeviceInfo audioDevice, getAvailableDevices(mode)) {
if (audioDevice.deviceName().trimmed() == deviceName.trimmed()) { if (audioDevice.deviceName().trimmed() == deviceName.trimmed()) {
result = audioDevice; result = audioDevice;
break; break;
@ -356,7 +330,8 @@ QString AudioClient::getWinDeviceName(wchar_t* guid) {
HRESULT hr = S_OK; HRESULT hr = S_OK;
CoInitialize(nullptr); CoInitialize(nullptr);
IMMDeviceEnumerator* pMMDeviceEnumerator = nullptr; IMMDeviceEnumerator* pMMDeviceEnumerator = nullptr;
CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator), (void**)&pMMDeviceEnumerator); CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator),
(void**)&pMMDeviceEnumerator);
IMMDevice* pEndpoint; IMMDevice* pEndpoint;
hr = pMMDeviceEnumerator->GetDevice(guid, &pEndpoint); hr = pMMDeviceEnumerator->GetDevice(guid, &pEndpoint);
if (hr == E_NOTFOUND) { if (hr == E_NOTFOUND) {
@ -380,34 +355,26 @@ QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
if (getAvailableDevices(mode).size() > 1) { if (getAvailableDevices(mode).size() > 1) {
AudioDeviceID defaultDeviceID = 0; AudioDeviceID defaultDeviceID = 0;
uint32_t propertySize = sizeof(AudioDeviceID); uint32_t propertySize = sizeof(AudioDeviceID);
AudioObjectPropertyAddress propertyAddress = { AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDefaultInputDevice,
kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
if (mode == QAudio::AudioOutput) { if (mode == QAudio::AudioOutput) {
propertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; propertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
} }
OSStatus getPropertyError =
OSStatus getPropertyError = AudioObjectGetPropertyData(kAudioObjectSystemObject, AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &defaultDeviceID);
&propertyAddress,
0,
NULL,
&propertySize,
&defaultDeviceID);
if (!getPropertyError && propertySize) { if (!getPropertyError && propertySize) {
CFStringRef deviceName = NULL; CFStringRef deviceName = NULL;
propertySize = sizeof(deviceName); propertySize = sizeof(deviceName);
propertyAddress.mSelector = kAudioDevicePropertyDeviceNameCFString; propertyAddress.mSelector = kAudioDevicePropertyDeviceNameCFString;
getPropertyError = AudioObjectGetPropertyData(defaultDeviceID, &propertyAddress, 0, getPropertyError =
NULL, &propertySize, &deviceName); AudioObjectGetPropertyData(defaultDeviceID, &propertyAddress, 0, NULL, &propertySize, &deviceName);
if (!getPropertyError && propertySize) { if (!getPropertyError && propertySize) {
// find a device in the list that matches the name we have and return it // find a device in the list that matches the name we have and return it
foreach(QAudioDeviceInfo audioDevice, getAvailableDevices(mode)) { foreach (QAudioDeviceInfo audioDevice, getAvailableDevices(mode)) {
if (audioDevice.deviceName() == CFStringGetCStringPtr(deviceName, kCFStringEncodingMacRoman)) { if (audioDevice.deviceName() == CFStringGetCStringPtr(deviceName, kCFStringEncodingMacRoman)) {
return audioDevice; return audioDevice;
} }
@ -419,7 +386,7 @@ QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
#ifdef WIN32 #ifdef WIN32
QString deviceName; QString deviceName;
//Check for Windows Vista or higher, IMMDeviceEnumerator doesn't work below that. //Check for Windows Vista or higher, IMMDeviceEnumerator doesn't work below that.
if (!IsWindowsVistaOrGreater()) { // lower then vista if (!IsWindowsVistaOrGreater()) { // lower then vista
if (mode == QAudio::AudioInput) { if (mode == QAudio::AudioInput) {
WAVEINCAPS wic; WAVEINCAPS wic;
// first use WAVE_MAPPER to get the default devices manufacturer ID // first use WAVE_MAPPER to get the default devices manufacturer ID
@ -441,9 +408,11 @@ QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
HRESULT hr = S_OK; HRESULT hr = S_OK;
CoInitialize(NULL); CoInitialize(NULL);
IMMDeviceEnumerator* pMMDeviceEnumerator = NULL; IMMDeviceEnumerator* pMMDeviceEnumerator = NULL;
CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator), (void**)&pMMDeviceEnumerator); CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator),
(void**)&pMMDeviceEnumerator);
IMMDevice* pEndpoint; IMMDevice* pEndpoint;
hr = pMMDeviceEnumerator->GetDefaultAudioEndpoint(mode == QAudio::AudioOutput ? eRender : eCapture, eMultimedia, &pEndpoint); hr = pMMDeviceEnumerator->GetDefaultAudioEndpoint(mode == QAudio::AudioOutput ? eRender : eCapture, eMultimedia,
&pEndpoint);
if (hr == E_NOTFOUND) { if (hr == E_NOTFOUND) {
printf("Audio Error: device not found\n"); printf("Audio Error: device not found\n");
deviceName = QString("NONE"); deviceName = QString("NONE");
@ -457,22 +426,22 @@ QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
CoUninitialize(); CoUninitialize();
} }
qCDebug(audioclient) << "defaultAudioDeviceForMode mode: " << (mode == QAudio::AudioOutput ? "Output" : "Input") qCDebug(audioclient) << "defaultAudioDeviceForMode mode: " << (mode == QAudio::AudioOutput ? "Output" : "Input") << " ["
<< " [" << deviceName << "] [" << getNamedAudioDeviceForMode(mode, deviceName).deviceName() << "]"; << deviceName << "] [" << getNamedAudioDeviceForMode(mode, deviceName).deviceName() << "]";
return getNamedAudioDeviceForMode(mode, deviceName); return getNamedAudioDeviceForMode(mode, deviceName);
#endif #endif
#if defined (Q_OS_ANDROID) #if defined(Q_OS_ANDROID)
if (mode == QAudio::AudioInput) { if (mode == QAudio::AudioInput) {
Setting::Handle<bool> enableAEC(SETTING_AEC_KEY, false); Setting::Handle<bool> enableAEC(SETTING_AEC_KEY, false);
bool aecEnabled = enableAEC.get(); bool aecEnabled = enableAEC.get();
auto audioClient = DependencyManager::get<AudioClient>(); auto audioClient = DependencyManager::get<AudioClient>();
bool headsetOn = audioClient? audioClient->isHeadsetPluggedIn() : false; bool headsetOn = audioClient ? audioClient->isHeadsetPluggedIn() : false;
auto inputDevices = QAudioDeviceInfo::availableDevices(QAudio::AudioInput); auto inputDevices = QAudioDeviceInfo::availableDevices(QAudio::AudioInput);
for (auto inputDevice : inputDevices) { for (auto inputDevice : inputDevices) {
if (((headsetOn || !aecEnabled) && inputDevice.deviceName() == VOICE_RECOGNITION) || if (((headsetOn || !aecEnabled) && inputDevice.deviceName() == VOICE_RECOGNITION) ||
((!headsetOn && aecEnabled) && inputDevice.deviceName() == VOICE_COMMUNICATION)) { ((!headsetOn && aecEnabled) && inputDevice.deviceName() == VOICE_COMMUNICATION)) {
return inputDevice; return inputDevice;
} }
} }
@ -486,11 +455,8 @@ bool AudioClient::getNamedAudioDeviceForModeExists(QAudio::Mode mode, const QStr
return (getNamedAudioDeviceForMode(mode, deviceName).deviceName() == deviceName); return (getNamedAudioDeviceForMode(mode, deviceName).deviceName() == deviceName);
} }
// attempt to use the native sample rate and channel count // attempt to use the native sample rate and channel count
bool nativeFormatForAudioDevice(const QAudioDeviceInfo& audioDevice, bool nativeFormatForAudioDevice(const QAudioDeviceInfo& audioDevice, QAudioFormat& audioFormat) {
QAudioFormat& audioFormat) {
audioFormat = audioDevice.preferredFormat(); audioFormat = audioDevice.preferredFormat();
audioFormat.setCodec("audio/pcm"); audioFormat.setCodec("audio/pcm");
@ -513,7 +479,6 @@ bool nativeFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice, bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
const QAudioFormat& desiredAudioFormat, const QAudioFormat& desiredAudioFormat,
QAudioFormat& adjustedAudioFormat) { QAudioFormat& adjustedAudioFormat) {
qCDebug(audioclient) << "The desired format for audio I/O is" << desiredAudioFormat; qCDebug(audioclient) << "The desired format for audio I/O is" << desiredAudioFormat;
#if defined(Q_OS_ANDROID) || defined(Q_OS_OSX) #if defined(Q_OS_ANDROID) || defined(Q_OS_OSX)
@ -539,12 +504,11 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
// Attempt the device sample rate and channel count in decreasing order of preference. // Attempt the device sample rate and channel count in decreasing order of preference.
// //
const int sampleRates[] = { 48000, 44100, 32000, 24000, 16000, 96000, 192000, 88200, 176400 }; const int sampleRates[] = { 48000, 44100, 32000, 24000, 16000, 96000, 192000, 88200, 176400 };
const int inputChannels[] = { 1, 2, 4, 6, 8 }; // prefer mono const int inputChannels[] = { 1, 2, 4, 6, 8 }; // prefer mono
const int outputChannels[] = { 2, 4, 6, 8, 1 }; // prefer stereo, downmix as last resort const int outputChannels[] = { 2, 4, 6, 8, 1 }; // prefer stereo, downmix as last resort
for (int channelCount : (desiredAudioFormat.channelCount() == 1 ? inputChannels : outputChannels)) { for (int channelCount : (desiredAudioFormat.channelCount() == 1 ? inputChannels : outputChannels)) {
for (int sampleRate : sampleRates) { for (int sampleRate : sampleRates) {
adjustedAudioFormat.setChannelCount(channelCount); adjustedAudioFormat.setChannelCount(channelCount);
adjustedAudioFormat.setSampleRate(sampleRate); adjustedAudioFormat.setSampleRate(sampleRate);
@ -554,11 +518,14 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
} }
} }
return false; // a supported format could not be found return false; // a supported format could not be found
} }
bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationSamples, unsigned int numSourceSamples, bool sampleChannelConversion(const int16_t* sourceSamples,
const int sourceChannelCount, const int destinationChannelCount) { int16_t* destinationSamples,
unsigned int numSourceSamples,
const int sourceChannelCount,
const int destinationChannelCount) {
if (sourceChannelCount == 2 && destinationChannelCount == 1) { if (sourceChannelCount == 2 && destinationChannelCount == 1) {
// loop through the stereo input audio samples and average every two samples // loop through the stereo input audio samples and average every two samples
for (uint i = 0; i < numSourceSamples; i += 2) { for (uint i = 0; i < numSourceSamples; i += 2) {
@ -567,7 +534,6 @@ bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationS
return true; return true;
} else if (sourceChannelCount == 1 && destinationChannelCount == 2) { } else if (sourceChannelCount == 1 && destinationChannelCount == 2) {
// loop through the mono input audio and repeat each sample twice // loop through the mono input audio and repeat each sample twice
for (uint i = 0; i < numSourceSamples; ++i) { for (uint i = 0; i < numSourceSamples; ++i) {
destinationSamples[i * 2] = destinationSamples[(i * 2) + 1] = sourceSamples[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, void possibleResampling(AudioSRC* resampler,
const int16_t* sourceSamples, int16_t* destinationSamples, const int16_t* sourceSamples,
unsigned int numSourceSamples, unsigned int numDestinationSamples, int16_t* destinationSamples,
const int sourceChannelCount, const int destinationChannelCount) { unsigned int numSourceSamples,
unsigned int numDestinationSamples,
const int sourceChannelCount,
const int destinationChannelCount) {
if (numSourceSamples > 0) { if (numSourceSamples > 0) {
if (!resampler) { if (!resampler) {
if (!sampleChannelConversion(sourceSamples, destinationSamples, numSourceSamples, if (!sampleChannelConversion(sourceSamples, destinationSamples, numSourceSamples, sourceChannelCount,
sourceChannelCount, destinationChannelCount)) { destinationChannelCount)) {
// no conversion, we can copy the samples directly across // no conversion, we can copy the samples directly across
memcpy(destinationSamples, sourceSamples, numSourceSamples * AudioConstants::SAMPLE_SIZE); memcpy(destinationSamples, sourceSamples, numSourceSamples * AudioConstants::SAMPLE_SIZE);
} }
} else { } else {
if (sourceChannelCount != destinationChannelCount) { if (sourceChannelCount != destinationChannelCount) {
int numChannelCoversionSamples = (numSourceSamples * destinationChannelCount) / sourceChannelCount; int numChannelCoversionSamples = (numSourceSamples * destinationChannelCount) / sourceChannelCount;
int16_t* channelConversionSamples = new int16_t[numChannelCoversionSamples]; int16_t* channelConversionSamples = new int16_t[numChannelCoversionSamples];
sampleChannelConversion(sourceSamples, channelConversionSamples, numSourceSamples, sampleChannelConversion(sourceSamples, channelConversionSamples, numSourceSamples, sourceChannelCount,
sourceChannelCount, destinationChannelCount); destinationChannelCount);
resampler->render(channelConversionSamples, destinationSamples, numChannelCoversionSamples); resampler->render(channelConversionSamples, destinationSamples, numChannelCoversionSamples);
delete[] channelConversionSamples; delete[] channelConversionSamples;
} else { } else {
unsigned int numAdjustedSourceSamples = numSourceSamples; unsigned int numAdjustedSourceSamples = numSourceSamples;
unsigned int numAdjustedDestinationSamples = numDestinationSamples; unsigned int numAdjustedDestinationSamples = numDestinationSamples;
@ -621,7 +586,6 @@ void possibleResampling(AudioSRC* resampler,
} }
void AudioClient::start() { void AudioClient::start() {
// set up the desired audio format // set up the desired audio format
_desiredInputFormat.setSampleRate(AudioConstants::SAMPLE_RATE); _desiredInputFormat.setSampleRate(AudioConstants::SAMPLE_RATE);
_desiredInputFormat.setSampleSize(16); _desiredInputFormat.setSampleSize(16);
@ -710,7 +674,6 @@ void AudioClient::handleAudioDataPacket(QSharedPointer<ReceivedMessage> message)
nodeList->flagTimeForConnectionStep(LimitedNodeList::ConnectionStep::ReceiveFirstAudioPacket); nodeList->flagTimeForConnectionStep(LimitedNodeList::ConnectionStep::ReceiveFirstAudioPacket);
if (_audioOutput) { if (_audioOutput) {
if (!_hasReceivedFirstPacket) { if (!_hasReceivedFirstPacket) {
_hasReceivedFirstPacket = true; _hasReceivedFirstPacket = true;
@ -727,8 +690,8 @@ void AudioClient::handleAudioDataPacket(QSharedPointer<ReceivedMessage> message)
} }
} }
AudioClient::Gate::Gate(AudioClient* audioClient) : AudioClient::Gate::Gate(AudioClient* audioClient) : _audioClient(audioClient) {
_audioClient(audioClient) {} }
void AudioClient::Gate::setIsSimulatingJitter(bool enable) { void AudioClient::Gate::setIsSimulatingJitter(bool enable) {
std::lock_guard<std::mutex> lock(_mutex); std::lock_guard<std::mutex> lock(_mutex);
@ -781,7 +744,6 @@ void AudioClient::Gate::flush() {
_index = 0; _index = 0;
} }
void AudioClient::handleNoisyMutePacket(QSharedPointer<ReceivedMessage> message) { void AudioClient::handleNoisyMutePacket(QSharedPointer<ReceivedMessage> message) {
if (!_muted) { if (!_muted) {
setMuted(true); setMuted(true);
@ -827,7 +789,6 @@ void AudioClient::handleSelectedAudioFormat(QSharedPointer<ReceivedMessage> mess
} }
void AudioClient::selectAudioFormat(const QString& selectedCodecName) { void AudioClient::selectAudioFormat(const QString& selectedCodecName) {
_selectedCodecName = selectedCodecName; _selectedCodecName = selectedCodecName;
qCDebug(audioclient) << "Selected Codec:" << _selectedCodecName << "isStereoInput:" << _isStereoInput; qCDebug(audioclient) << "Selected Codec:" << _selectedCodecName << "isStereoInput:" << _isStereoInput;
@ -845,12 +806,12 @@ void AudioClient::selectAudioFormat(const QString& selectedCodecName) {
if (_selectedCodecName == plugin->getName()) { if (_selectedCodecName == plugin->getName()) {
_codec = plugin; _codec = plugin;
_receivedAudioStream.setupCodec(plugin, _selectedCodecName, AudioConstants::STEREO); _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(); qCDebug(audioclient) << "Selected Codec Plugin:" << _codec.get();
break; break;
} }
} }
} }
bool AudioClient::switchAudioDevice(QAudio::Mode mode, const QAudioDeviceInfo& deviceInfo) { 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) { if (mode == QAudio::AudioInput) {
return switchInputToAudioDevice(device); return switchInputToAudioDevice(device);
} else { // if (mode == QAudio::AudioOutput) } else { // if (mode == QAudio::AudioOutput)
return switchOutputToAudioDevice(device); return switchOutputToAudioDevice(device);
} }
} }
@ -904,8 +865,8 @@ void AudioClient::configureReverb() {
p.sampleRate = _outputFormat.sampleRate(); p.sampleRate = _outputFormat.sampleRate();
p.wetDryMix = 100.0f; p.wetDryMix = 100.0f;
p.preDelay = 0.0f; p.preDelay = 0.0f;
p.earlyGain = -96.0f; // disable ER p.earlyGain = -96.0f; // disable ER
p.lateGain += _reverbOptions->getWetDryMix() * (24.0f/100.0f) - 24.0f; // -0dB to -24dB, based on wetDryMix p.lateGain += _reverbOptions->getWetDryMix() * (24.0f / 100.0f) - 24.0f; // -0dB to -24dB, based on wetDryMix
p.lateMixLeft = 0.0f; p.lateMixLeft = 0.0f;
p.lateMixRight = 0.0f; p.lateMixRight = 0.0f;
@ -915,7 +876,6 @@ void AudioClient::configureReverb() {
void AudioClient::updateReverbOptions() { void AudioClient::updateReverbOptions() {
bool reverbChanged = false; bool reverbChanged = false;
if (_receivedAudioStream.hasReverb()) { if (_receivedAudioStream.hasReverb()) {
if (_zoneReverbOptions.getReverbTime() != _receivedAudioStream.getRevebTime()) { if (_zoneReverbOptions.getReverbTime() != _receivedAudioStream.getRevebTime()) {
_zoneReverbOptions.setReverbTime(_receivedAudioStream.getRevebTime()); _zoneReverbOptions.setReverbTime(_receivedAudioStream.getRevebTime());
reverbChanged = true; reverbChanged = true;
@ -1020,7 +980,8 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data()); int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
// upmix mono to stereo // 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 // no conversion, just copy the samples
memcpy(loopbackSamples, inputSamples, numInputSamples * AudioConstants::SAMPLE_SIZE); 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 // apply stereo reverb at the source, to the loopback audio
if (!_shouldEchoLocally && hasReverb) { if (!_shouldEchoLocally && hasReverb) {
updateReverbOptions(); updateReverbOptions();
_sourceReverb.render(loopbackSamples, loopbackSamples, numLoopbackSamples/2); _sourceReverb.render(loopbackSamples, loopbackSamples, numLoopbackSamples / 2);
} }
// if required, upmix or downmix to deviceChannelCount // if required, upmix or downmix to deviceChannelCount
int deviceChannelCount = _outputFormat.channelCount(); int deviceChannelCount = _outputFormat.channelCount();
if (deviceChannelCount == OUTPUT_CHANNEL_COUNT) { if (deviceChannelCount == OUTPUT_CHANNEL_COUNT) {
_loopbackOutputDevice->write(loopBackByteArray); _loopbackOutputDevice->write(loopBackByteArray);
} else { } else {
static QByteArray deviceByteArray; static QByteArray deviceByteArray;
int numDeviceSamples = (numLoopbackSamples * deviceChannelCount) / OUTPUT_CHANNEL_COUNT; int numDeviceSamples = (numLoopbackSamples * deviceChannelCount) / OUTPUT_CHANNEL_COUNT;
@ -1074,7 +1033,7 @@ void AudioClient::handleAudioInput(QByteArray& audioBuffer) {
} }
int32_t loudness = 0; int32_t loudness = 0;
assert(numSamples < 65536); // int32_t loudness cannot overflow assert(numSamples < 65536); // int32_t loudness cannot overflow
bool didClip = false; bool didClip = false;
for (int i = 0; i < numSamples; ++i) { for (int i = 0; i < numSamples; ++i) {
const int32_t CLIPPING_THRESHOLD = (int32_t)(AudioConstants::MAX_SAMPLE_VALUE * 0.9f); 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, emitAudioPacket(encodedBuffer.data(), encodedBuffer.size(), _outgoingAvatarAudioSequenceNumber, _isStereoInput,
audioTransform, avatarBoundingBoxCorner, avatarBoundingBoxScale, audioTransform, avatarBoundingBoxCorner, avatarBoundingBoxScale, packetType, _selectedCodecName);
packetType, _selectedCodecName);
_stats.sentPacket(); _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 // input samples required to produce exactly NETWORK_FRAME_SAMPLES of output
const int inputSamplesRequired = const int inputSamplesRequired =
(_inputToNetworkResampler ? _inputToNetworkResampler->getMinInput(AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL) (_inputToNetworkResampler ? _inputToNetworkResampler->getMinInput(AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL)
@ -1189,11 +1149,10 @@ void AudioClient::handleMicAudioInput() {
} }
void AudioClient::handleDummyAudioInput() { void AudioClient::handleDummyAudioInput() {
const int numNetworkBytes = _isStereoInput const int numNetworkBytes =
? AudioConstants::NETWORK_FRAME_BYTES_STEREO _isStereoInput ? AudioConstants::NETWORK_FRAME_BYTES_STEREO : AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL;
: AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL;
QByteArray audioBuffer(numNetworkBytes, 0); // silent QByteArray audioBuffer(numNetworkBytes, 0); // silent
handleAudioInput(audioBuffer); handleAudioInput(audioBuffer);
} }
@ -1202,13 +1161,59 @@ void AudioClient::handleRecordedAudioInput(const QByteArray& audio) {
handleAudioInput(audioBuffer); 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) { void AudioClient::handleTTSAudioInput(const QByteArray& audio) {
QByteArray audioBuffer(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) { while (audioBuffer.size() > 0) {
QByteArray part; QByteArray part;
part.append(audioBuffer.data(), AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); part.append(audioBuffer.data(), AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
audioBuffer.remove(0, 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 bufferCapacity = _localInjectorsStream.getSampleCapacity();
int maxOutputSamples = AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * AudioConstants::STEREO; int maxOutputSamples = AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * AudioConstants::STEREO;
if (_localToOutputResampler) { if (_localToOutputResampler) {
maxOutputSamples = maxOutputSamples = _localToOutputResampler->getMaxOutput(AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL) *
_localToOutputResampler->getMaxOutput(AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL) * AudioConstants::STEREO;
AudioConstants::STEREO;
} }
samplesNeeded = bufferCapacity - _localSamplesAvailable.load(std::memory_order_relaxed); samplesNeeded = bufferCapacity - _localSamplesAvailable.load(std::memory_order_relaxed);
@ -1259,7 +1263,7 @@ void AudioClient::prepareLocalAudioInjectors(std::unique_ptr<Lock> localAudioLoc
if (_localToOutputResampler) { if (_localToOutputResampler) {
// resample to output sample rate // resample to output sample rate
int frames = _localToOutputResampler->render(_localMixBuffer, _localOutputMixBuffer, int frames = _localToOutputResampler->render(_localMixBuffer, _localOutputMixBuffer,
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
// write to local injectors' ring buffer // write to local injectors' ring buffer
samples = frames * AudioConstants::STEREO; samples = frames * AudioConstants::STEREO;
@ -1268,8 +1272,7 @@ void AudioClient::prepareLocalAudioInjectors(std::unique_ptr<Lock> localAudioLoc
} else { } else {
// write to local injectors' ring buffer // write to local injectors' ring buffer
samples = AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; samples = AudioConstants::NETWORK_FRAME_SAMPLES_STEREO;
_localInjectorsStream.writeSamples(_localMixBuffer, _localInjectorsStream.writeSamples(_localMixBuffer, AudioConstants::NETWORK_FRAME_SAMPLES_STEREO);
AudioConstants::NETWORK_FRAME_SAMPLES_STEREO);
} }
_localSamplesAvailable.fetch_add(samples, std::memory_order_release); _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 // the lock guarantees that injectorBuffer, if found, is invariant
AudioInjectorLocalBuffer* injectorBuffer = injector->getLocalBuffer(); AudioInjectorLocalBuffer* injectorBuffer = injector->getLocalBuffer();
if (injectorBuffer) { if (injectorBuffer) {
static const int HRTF_DATASET_INDEX = 1; 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; size_t bytesToRead = numChannels * AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL;
// get one frame from the injector // get one frame from the injector
memset(_localScratchBuffer, 0, bytesToRead); memset(_localScratchBuffer, 0, bytesToRead);
if (0 < injectorBuffer->readData((char*)_localScratchBuffer, bytesToRead)) { if (0 < injectorBuffer->readData((char*)_localScratchBuffer, bytesToRead)) {
if (injector->isAmbisonic()) { if (injector->isAmbisonic()) {
// no distance attenuation // no distance attenuation
float gain = injector->getVolume(); float gain = injector->getVolume();
@ -1322,11 +1323,10 @@ bool AudioClient::mixLocalAudioInjectors(float* mixBuffer) {
float qz = relativeOrientation.y; float qz = relativeOrientation.y;
// Ambisonic gets spatialized into mixBuffer // Ambisonic gets spatialized into mixBuffer
injector->getLocalFOA().render(_localScratchBuffer, mixBuffer, HRTF_DATASET_INDEX, injector->getLocalFOA().render(_localScratchBuffer, mixBuffer, HRTF_DATASET_INDEX, qw, qx, qy, qz, gain,
qw, qx, qy, qz, gain, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
} else if (injector->isStereo()) { } else if (injector->isStereo()) {
// stereo gets directly mixed into mixBuffer // stereo gets directly mixed into mixBuffer
float gain = injector->getVolume(); float gain = injector->getVolume();
for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i++) { for (int i = 0; i < AudioConstants::NETWORK_FRAME_SAMPLES_STEREO; i++) {
@ -1334,7 +1334,6 @@ bool AudioClient::mixLocalAudioInjectors(float* mixBuffer) {
} }
} else { } else {
// calculate distance, gain and azimuth for hrtf // calculate distance, gain and azimuth for hrtf
glm::vec3 relativePosition = injector->getPosition() - _positionGetter(); glm::vec3 relativePosition = injector->getPosition() - _positionGetter();
float distance = glm::max(glm::length(relativePosition), EPSILON); float distance = glm::max(glm::length(relativePosition), EPSILON);
@ -1342,19 +1341,17 @@ bool AudioClient::mixLocalAudioInjectors(float* mixBuffer) {
float azimuth = azimuthForSource(relativePosition); float azimuth = azimuthForSource(relativePosition);
// mono gets spatialized into mixBuffer // mono gets spatialized into mixBuffer
injector->getLocalHRTF().render(_localScratchBuffer, mixBuffer, HRTF_DATASET_INDEX, injector->getLocalHRTF().render(_localScratchBuffer, mixBuffer, HRTF_DATASET_INDEX, azimuth, distance, gain,
azimuth, distance, gain, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL);
} }
} else { } else {
qCDebug(audioclient) << "injector has no more data, marking finished for removal"; qCDebug(audioclient) << "injector has no more data, marking finished for removal";
injector->finishLocalInjection(); injector->finishLocalInjection();
injectorsToRemove.append(injector); injectorsToRemove.append(injector);
} }
} else { } else {
qCDebug(audioclient) << "injector has no local buffer, marking as finished for removal"; qCDebug(audioclient) << "injector has no local buffer, marking as finished for removal";
injector->finishLocalInjection(); injector->finishLocalInjection();
injectorsToRemove.append(injector); injectorsToRemove.append(injector);
@ -1373,7 +1370,6 @@ bool AudioClient::mixLocalAudioInjectors(float* mixBuffer) {
} }
void AudioClient::processReceivedSamples(const QByteArray& decodedBuffer, QByteArray& outputBuffer) { void AudioClient::processReceivedSamples(const QByteArray& decodedBuffer, QByteArray& outputBuffer) {
const int16_t* decodedSamples = reinterpret_cast<const int16_t*>(decodedBuffer.data()); const int16_t* decodedSamples = reinterpret_cast<const int16_t*>(decodedBuffer.data());
assert(decodedBuffer.size() == AudioConstants::NETWORK_FRAME_BYTES_STEREO); 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 AudioClient::setIsStereoInput(bool isStereoInput) {
bool stereoInputChanged = false; bool stereoInputChanged = false;
if (isStereoInput != _isStereoInput && _inputDeviceInfo.supportedChannelCounts().contains(2)) { if (isStereoInput != _isStereoInput && _inputDeviceInfo.supportedChannelCounts().contains(2)) {
@ -1460,7 +1455,8 @@ bool AudioClient::setIsStereoInput(bool isStereoInput) {
if (_encoder) { if (_encoder) {
_codec->releaseEncoder(_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; qCDebug(audioclient) << "Reset Codec:" << _selectedCodecName << "isStereoInput:" << _isStereoInput;
@ -1500,7 +1496,7 @@ bool AudioClient::outputLocalInjector(const AudioInjectorPointer& injector) {
void AudioClient::outputFormatChanged() { void AudioClient::outputFormatChanged() {
_outputFrameSize = (AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * OUTPUT_CHANNEL_COUNT * _outputFormat.sampleRate()) / _outputFrameSize = (AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * OUTPUT_CHANNEL_COUNT * _outputFormat.sampleRate()) /
_desiredOutputFormat.sampleRate(); _desiredOutputFormat.sampleRate();
_receivedAudioStream.outputFormatChanged(_outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT); _receivedAudioStream.outputFormatChanged(_outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT);
} }
@ -1514,7 +1510,7 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo inputDeviceInf
Lock lock(_deviceMutex); Lock lock(_deviceMutex);
#if defined(Q_OS_ANDROID) #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 #endif
// cleanup any previously initialized device // 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 // we've got the best we can get for input
// if required, setup a resampler for this input to our desired network format // if required, setup a resampler for this input to our desired network format
if (_inputFormat != _desiredInputFormat if (_inputFormat != _desiredInputFormat && _inputFormat.sampleRate() != _desiredInputFormat.sampleRate()) {
&& _inputFormat.sampleRate() != _desiredInputFormat.sampleRate()) {
qCDebug(audioclient) << "Attemping to create a resampler for input format to network format."; qCDebug(audioclient) << "Attemping to create a resampler for input format to network format.";
assert(_inputFormat.sampleSize() == 16); assert(_inputFormat.sampleSize() == 16);
assert(_desiredInputFormat.sampleSize() == 16); assert(_desiredInputFormat.sampleSize() == 16);
int channelCount = (_inputFormat.channelCount() == 2 && _desiredInputFormat.channelCount() == 2) ? 2 : 1; 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 { } else {
qCDebug(audioclient) << "No resampling required for audio input to match desired network format."; 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())); connect(_inputDevice, SIGNAL(readyRead()), this, SLOT(handleMicAudioInput()));
supportedFormat = true; supportedFormat = true;
} else { } else {
qCDebug(audioclient) << "Error starting audio input -" << _audioInput->error(); qCDebug(audioclient) << "Error starting audio input -" << _audioInput->error();
_audioInput->deleteLater(); _audioInput->deleteLater();
_audioInput = NULL; _audioInput = NULL;
} }
@ -1677,7 +1673,7 @@ void AudioClient::checkInputTimeout() {
void AudioClient::setHeadsetPluggedIn(bool pluggedIn) { void AudioClient::setHeadsetPluggedIn(bool pluggedIn) {
#if defined(Q_OS_ANDROID) #if defined(Q_OS_ANDROID)
if (pluggedIn == !_isHeadsetPluggedIn && !_inputDeviceInfo.isNull()) { 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 // some samsung phones needs more time to shutdown the previous input device
if (brand.toString().contains("samsung", Qt::CaseInsensitive)) { if (brand.toString().contains("samsung", Qt::CaseInsensitive)) {
switchInputToAudioDevice(QAudioDeviceInfo(), true); switchInputToAudioDevice(QAudioDeviceInfo(), true);
@ -1715,8 +1711,8 @@ void AudioClient::outputNotify() {
int newOutputBufferSizeFrames = setOutputBufferSize(oldOutputBufferSizeFrames + 1, false); int newOutputBufferSizeFrames = setOutputBufferSize(oldOutputBufferSizeFrames + 1, false);
if (newOutputBufferSizeFrames > oldOutputBufferSizeFrames) { if (newOutputBufferSizeFrames > oldOutputBufferSizeFrames) {
qCDebug(audioclient, qCDebug(audioclient, "Starve threshold surpassed (%d starves in %d ms)", _outputStarveDetectionCount,
"Starve threshold surpassed (%d starves in %d ms)", _outputStarveDetectionCount, dt); dt);
} }
_outputStarveDetectionStartTimeMsec = now; _outputStarveDetectionStartTimeMsec = now;
@ -1730,7 +1726,8 @@ void AudioClient::outputNotify() {
bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceInfo, bool isShutdownRequest) { bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceInfo, bool isShutdownRequest) {
Q_ASSERT_X(QThread::currentThread() == thread(), Q_FUNC_INFO, "Function invoked on wrong thread"); 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; bool supportedFormat = false;
// NOTE: device start() uses the Qt internal device list // 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 // we've got the best we can get for input
// if required, setup a resampler for this input to our desired network format // if required, setup a resampler for this input to our desired network format
if (_desiredOutputFormat != _outputFormat if (_desiredOutputFormat != _outputFormat && _desiredOutputFormat.sampleRate() != _outputFormat.sampleRate()) {
&& _desiredOutputFormat.sampleRate() != _outputFormat.sampleRate()) {
qCDebug(audioclient) << "Attemping to create a resampler for network format to output format."; qCDebug(audioclient) << "Attemping to create a resampler for network format to output format.";
assert(_desiredOutputFormat.sampleSize() == 16); assert(_desiredOutputFormat.sampleSize() == 16);
assert(_outputFormat.sampleSize() == 16); assert(_outputFormat.sampleSize() == 16);
_networkToOutputResampler = new AudioSRC(_desiredOutputFormat.sampleRate(), _outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT); _networkToOutputResampler =
_localToOutputResampler = new AudioSRC(_desiredOutputFormat.sampleRate(), _outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT); new AudioSRC(_desiredOutputFormat.sampleRate(), _outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT);
_localToOutputResampler =
new AudioSRC(_desiredOutputFormat.sampleRate(), _outputFormat.sampleRate(), OUTPUT_CHANNEL_COUNT);
} else { } else {
qCDebug(audioclient) << "No resampling required for network output to match actual output format."; 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); _audioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this);
int deviceChannelCount = _outputFormat.channelCount(); 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; int requestedSize = _sessionOutputBufferSizeFrames * frameSize * AudioConstants::SAMPLE_SIZE;
_audioOutput->setBufferSize(requestedSize); _audioOutput->setBufferSize(requestedSize);
@ -1825,7 +1825,10 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo outputDeviceI
_outputScratchBuffer = new int16_t[_outputPeriod]; _outputScratchBuffer = new int16_t[_outputPeriod];
// size local output mix buffer based on resampled network frame size // 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]; _localOutputMixBuffer = new float[networkPeriod];
// local period should be at least twice the output period, // 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; qCDebug(audioclient) << __FUNCTION__ << "numFrames:" << numFrames << "persist:" << persist;
numFrames = std::min(std::max(numFrames, MIN_BUFFER_FRAMES), MAX_BUFFER_FRAMES); 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) { if (numFrames != _sessionOutputBufferSizeFrames) {
qCInfo(audioclient, "Audio output buffer set to %d frames", numFrames); qCInfo(audioclient, "Audio output buffer set to %d frames", numFrames);
@ -1906,10 +1910,10 @@ const float AudioClient::CALLBACK_ACCELERATOR_RATIO = 2.0f;
#endif #endif
int AudioClient::calculateNumberOfInputCallbackBytes(const QAudioFormat& format) const { int AudioClient::calculateNumberOfInputCallbackBytes(const QAudioFormat& format) const {
int numInputCallbackBytes = (int)(((AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL int numInputCallbackBytes = (int)(((AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL * format.channelCount() *
* format.channelCount() ((float)format.sampleRate() / AudioConstants::SAMPLE_RATE)) /
* ((float) format.sampleRate() / AudioConstants::SAMPLE_RATE)) CALLBACK_ACCELERATOR_RATIO) +
/ CALLBACK_ACCELERATOR_RATIO) + 0.5f); 0.5f);
return numInputCallbackBytes; return numInputCallbackBytes;
} }
@ -1931,10 +1935,9 @@ float AudioClient::azimuthForSource(const glm::vec3& relativePosition) {
float rotatedSourcePositionLength2 = glm::length2(rotatedSourcePosition); float rotatedSourcePositionLength2 = glm::length2(rotatedSourcePosition);
if (rotatedSourcePositionLength2 > SOURCE_DISTANCE_THRESHOLD) { if (rotatedSourcePositionLength2 > SOURCE_DISTANCE_THRESHOLD) {
// produce an oriented angle about the y-axis // produce an oriented angle about the y-axis
glm::vec3 direction = rotatedSourcePosition * (1.0f / fastSqrtf(rotatedSourcePositionLength2)); 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; return (direction.x < 0.0f) ? -angle : angle;
} else { } else {
@ -1944,7 +1947,6 @@ float AudioClient::azimuthForSource(const glm::vec3& relativePosition) {
} }
float AudioClient::gainForSource(float distance, float volume) { float AudioClient::gainForSource(float distance, float volume) {
// attenuation = -6dB * log2(distance) // attenuation = -6dB * log2(distance)
// reference attenuation of 0dB at distance = 1.0m // reference attenuation of 0dB at distance = 1.0m
float gain = volume / std::max(distance, HRTF_NEARFIELD_MIN); float gain = volume / std::max(distance, HRTF_NEARFIELD_MIN);
@ -1952,8 +1954,7 @@ float AudioClient::gainForSource(float distance, float volume) {
return gain; return gain;
} }
qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) { qint64 AudioClient::AudioOutputIODevice::readData(char* data, qint64 maxSize) {
// samples requested from OUTPUT_CHANNEL_COUNT // samples requested from OUTPUT_CHANNEL_COUNT
int deviceChannelCount = _audio->_outputFormat.channelCount(); int deviceChannelCount = _audio->_outputFormat.channelCount();
int samplesRequested = (int)(maxSize / AudioConstants::SAMPLE_SIZE) * OUTPUT_CHANNEL_COUNT / deviceChannelCount; 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; int networkSamplesPopped;
if ((networkSamplesPopped = _receivedAudioStream.popSamples(samplesRequested, false)) > 0) { 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(); AudioRingBuffer::ConstIterator lastPopOutput = _receivedAudioStream.getLastPopOutput();
lastPopOutput.readSamples(scratchBuffer, networkSamplesPopped); lastPopOutput.readSamples(scratchBuffer, networkSamplesPopped);
for (int i = 0; i < networkSamplesPopped; i++) { for (int i = 0; i < networkSamplesPopped; i++) {
@ -1997,14 +1999,13 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
samplesRequested = std::min(samplesRequested, samplesAvailable); samplesRequested = std::min(samplesRequested, samplesAvailable);
if ((injectorSamplesPopped = _localInjectorsStream.appendSamples(mixBuffer, samplesRequested, append)) > 0) { if ((injectorSamplesPopped = _localInjectorsStream.appendSamples(mixBuffer, samplesRequested, append)) > 0) {
_audio->_localSamplesAvailable.fetch_sub(injectorSamplesPopped, std::memory_order_release); _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 // prepare injectors for the next callback
QtConcurrent::run(QThreadPool::globalInstance(), [this] { QtConcurrent::run(QThreadPool::globalInstance(), [this] { _audio->prepareLocalAudioInjectors(); });
_audio->prepareLocalAudioInjectors();
});
int samplesPopped = std::max(networkSamplesPopped, injectorSamplesPopped); int samplesPopped = std::max(networkSamplesPopped, injectorSamplesPopped);
int framesPopped = samplesPopped / AudioConstants::STEREO; 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); _audio->_audioFileWav.addRawAudioChunk(reinterpret_cast<char*>(scratchBuffer), bytesWritten);
} }
int bytesAudioOutputUnplayed = _audio->_audioOutput->bufferSize() - _audio->_audioOutput->bytesFree(); int bytesAudioOutputUnplayed = _audio->_audioOutput->bufferSize() - _audio->_audioOutput->bytesFree();
float msecsAudioOutputUnplayed = bytesAudioOutputUnplayed / (float)_audio->_outputFormat.bytesForDuration(USECS_PER_MSEC); float msecsAudioOutputUnplayed = bytesAudioOutputUnplayed / (float)_audio->_outputFormat.bytesForDuration(USECS_PER_MSEC);
_audio->_stats.updateOutputMsUnplayed(msecsAudioOutputUnplayed); _audio->_stats.updateOutputMsUnplayed(msecsAudioOutputUnplayed);
@ -2075,7 +2075,6 @@ void AudioClient::loadSettings() {
for (auto& plugin : codecPlugins) { for (auto& plugin : codecPlugins) {
qCDebug(audioclient) << "Codec available:" << plugin->getName(); qCDebug(audioclient) << "Codec available:" << plugin->getName();
} }
} }
void AudioClient::saveSettings() { void AudioClient::saveSettings() {
@ -2088,7 +2087,6 @@ void AudioClient::setAvatarBoundingBoxParameters(glm::vec3 corner, glm::vec3 sca
avatarBoundingBoxScale = scale; avatarBoundingBoxScale = scale;
} }
void AudioClient::startThread() { void AudioClient::startThread() {
moveToNewNamedThread(this, "Audio Thread", [this] { start(); }, QThread::TimeCriticalPriority); moveToNewNamedThread(this, "Audio Thread", [this] { start(); }, QThread::TimeCriticalPriority);
} }