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overte-lubosz/libraries/audio-client/src/AudioClient.cpp
Stephen Birarda 3a5701e02e first changes for new network API
2015-07-02 16:18:44 -07:00

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//
// AudioClient.cpp
// interface/src
//
// Created by Stephen Birarda on 1/22/13.
// Copyright 2013 High Fidelity, Inc.
//
// Distributed under the Apache License, Version 2.0.
// See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
//
#include <cstring>
#include <math.h>
#include <sys/stat.h>
#include <glm/glm.hpp>
#ifdef __APPLE__
#include <CoreAudio/AudioHardware.h>
#endif
#ifdef WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <Mmsystem.h>
#include <mmdeviceapi.h>
#include <devicetopology.h>
#include <Functiondiscoverykeys_devpkey.h>
#include <VersionHelpers.h>
#endif
#include <QtCore/QBuffer>
#include <QtMultimedia/QAudioInput>
#include <QtMultimedia/QAudioOutput>
#ifdef __GNUC__
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wdouble-promotion"
#endif
extern "C" {
#include <gverb/gverb.h>
#include <gverb/gverbdsp.h>
}
#ifdef __GNUC__
#pragma GCC diagnostic pop
#endif
#include <soxr.h>
#include <NodeList.h>
#include <PacketHeaders.h>
#include <PositionalAudioStream.h>
#include <SettingHandle.h>
#include <SharedUtil.h>
#include <UUID.h>
#include "AudioInjector.h"
#include "AudioConstants.h"
#include "PositionalAudioStream.h"
#include "AudioClientLogging.h"
#include "AudioClient.h"
static const int RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES = 100;
Setting::Handle<bool> dynamicJitterBuffers("dynamicJitterBuffers", DEFAULT_DYNAMIC_JITTER_BUFFERS);
Setting::Handle<int> maxFramesOverDesired("maxFramesOverDesired", DEFAULT_MAX_FRAMES_OVER_DESIRED);
Setting::Handle<int> staticDesiredJitterBufferFrames("staticDesiredJitterBufferFrames",
DEFAULT_STATIC_DESIRED_JITTER_BUFFER_FRAMES);
Setting::Handle<bool> useStDevForJitterCalc("useStDevForJitterCalc", DEFAULT_USE_STDEV_FOR_JITTER_CALC);
Setting::Handle<int> windowStarveThreshold("windowStarveThreshold", DEFAULT_WINDOW_STARVE_THRESHOLD);
Setting::Handle<int> windowSecondsForDesiredCalcOnTooManyStarves("windowSecondsForDesiredCalcOnTooManyStarves",
DEFAULT_WINDOW_SECONDS_FOR_DESIRED_CALC_ON_TOO_MANY_STARVES);
Setting::Handle<int> windowSecondsForDesiredReduction("windowSecondsForDesiredReduction",
DEFAULT_WINDOW_SECONDS_FOR_DESIRED_REDUCTION);
Setting::Handle<bool> repetitionWithFade("repetitionWithFade", DEFAULT_REPETITION_WITH_FADE);
AudioClient::AudioClient() :
AbstractAudioInterface(),
_audioInput(NULL),
_desiredInputFormat(),
_inputFormat(),
_numInputCallbackBytes(0),
_audioOutput(NULL),
_desiredOutputFormat(),
_outputFormat(),
_outputFrameSize(0),
_numOutputCallbackBytes(0),
_loopbackAudioOutput(NULL),
_loopbackOutputDevice(NULL),
_inputRingBuffer(0),
_receivedAudioStream(0, RECEIVED_AUDIO_STREAM_CAPACITY_FRAMES, InboundAudioStream::Settings()),
_isStereoInput(false),
_outputStarveDetectionStartTimeMsec(0),
_outputStarveDetectionCount(0),
_outputBufferSizeFrames("audioOutputBufferSize", DEFAULT_AUDIO_OUTPUT_BUFFER_SIZE_FRAMES),
_outputStarveDetectionEnabled("audioOutputStarveDetectionEnabled",
DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_ENABLED),
_outputStarveDetectionPeriodMsec("audioOutputStarveDetectionPeriod",
DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_PERIOD),
_outputStarveDetectionThreshold("audioOutputStarveDetectionThreshold",
DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_THRESHOLD),
_averagedLatency(0.0f),
_lastInputLoudness(0.0f),
_timeSinceLastClip(-1.0f),
_muted(false),
_shouldEchoLocally(false),
_shouldEchoToServer(false),
_isNoiseGateEnabled(true),
_audioSourceInjectEnabled(false),
_reverb(false),
_reverbOptions(&_scriptReverbOptions),
_gverb(NULL),
_inputToNetworkResampler(NULL),
_networkToOutputResampler(NULL),
_loopbackResampler(NULL),
_noiseSourceEnabled(false),
_toneSourceEnabled(true),
_outgoingAvatarAudioSequenceNumber(0),
_audioOutputIODevice(_receivedAudioStream, this),
_stats(&_receivedAudioStream),
_inputGate()
{
// clear the array of locally injected samples
memset(_localProceduralSamples, 0, AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL);
connect(&_receivedAudioStream, &MixedProcessedAudioStream::processSamples,
this, &AudioClient::processReceivedSamples, Qt::DirectConnection);
_inputDevices = getDeviceNames(QAudio::AudioInput);
_outputDevices = getDeviceNames(QAudio::AudioOutput);
const qint64 DEVICE_CHECK_INTERVAL_MSECS = 2 * 1000;
QTimer* updateTimer = new QTimer(this);
connect(updateTimer, &QTimer::timeout, this, &AudioClient::checkDevices);
updateTimer->start(DEVICE_CHECK_INTERVAL_MSECS);
// create GVerb filter
_gverb = createGverbFilter();
configureGverbFilter(_gverb);
}
AudioClient::~AudioClient() {
stop();
if (_gverb) {
gverb_free(_gverb);
}
}
void AudioClient::reset() {
_receivedAudioStream.reset();
_stats.reset();
_noiseSource.reset();
_toneSource.reset();
_sourceGain.reset();
_inputGain.reset();
gverb_flush(_gverb);
}
void AudioClient::audioMixerKilled() {
_hasReceivedFirstPacket = false;
_outgoingAvatarAudioSequenceNumber = 0;
_stats.reset();
emit disconnected();
}
QAudioDeviceInfo getNamedAudioDeviceForMode(QAudio::Mode mode, const QString& deviceName) {
QAudioDeviceInfo result;
foreach(QAudioDeviceInfo audioDevice, QAudioDeviceInfo::availableDevices(mode)) {
if (audioDevice.deviceName().trimmed() == deviceName.trimmed()) {
result = audioDevice;
break;
}
}
return result;
}
soxr_datatype_t soxrDataTypeFromQAudioFormat(const QAudioFormat& audioFormat) {
if (audioFormat.sampleType() == QAudioFormat::Float) {
return SOXR_FLOAT32_I;
} else {
if (audioFormat.sampleSize() == 16) {
return SOXR_INT16_I;
} else {
return SOXR_INT32_I;
}
}
}
int numDestinationSamplesRequired(const QAudioFormat& sourceFormat, const QAudioFormat& destinationFormat,
int numSourceSamples) {
float ratio = (float) destinationFormat.channelCount() / sourceFormat.channelCount();
ratio *= (float) destinationFormat.sampleRate() / sourceFormat.sampleRate();
return (numSourceSamples * ratio) + 0.5f;
}
QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
#ifdef __APPLE__
if (QAudioDeviceInfo::availableDevices(mode).size() > 1) {
AudioDeviceID defaultDeviceID = 0;
uint32_t propertySize = sizeof(AudioDeviceID);
AudioObjectPropertyAddress propertyAddress = {
kAudioHardwarePropertyDefaultInputDevice,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
if (mode == QAudio::AudioOutput) {
propertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
}
OSStatus getPropertyError = AudioObjectGetPropertyData(kAudioObjectSystemObject,
&propertyAddress,
0,
NULL,
&propertySize,
&defaultDeviceID);
if (!getPropertyError && propertySize) {
CFStringRef deviceName = NULL;
propertySize = sizeof(deviceName);
propertyAddress.mSelector = kAudioDevicePropertyDeviceNameCFString;
getPropertyError = AudioObjectGetPropertyData(defaultDeviceID, &propertyAddress, 0,
NULL, &propertySize, &deviceName);
if (!getPropertyError && propertySize) {
// find a device in the list that matches the name we have and return it
foreach(QAudioDeviceInfo audioDevice, QAudioDeviceInfo::availableDevices(mode)) {
if (audioDevice.deviceName() == CFStringGetCStringPtr(deviceName, kCFStringEncodingMacRoman)) {
return audioDevice;
}
}
}
}
}
#endif
#ifdef WIN32
QString deviceName;
//Check for Windows Vista or higher, IMMDeviceEnumerator doesn't work below that.
if (!IsWindowsVistaOrGreater()) { // lower then vista
if (mode == QAudio::AudioInput) {
WAVEINCAPS wic;
// first use WAVE_MAPPER to get the default devices manufacturer ID
waveInGetDevCaps(WAVE_MAPPER, &wic, sizeof(wic));
//Use the received manufacturer id to get the device's real name
waveInGetDevCaps(wic.wMid, &wic, sizeof(wic));
qCDebug(audioclient) << "input device:" << wic.szPname;
deviceName = wic.szPname;
} else {
WAVEOUTCAPS woc;
// first use WAVE_MAPPER to get the default devices manufacturer ID
waveOutGetDevCaps(WAVE_MAPPER, &woc, sizeof(woc));
//Use the received manufacturer id to get the device's real name
waveOutGetDevCaps(woc.wMid, &woc, sizeof(woc));
qCDebug(audioclient) << "output device:" << woc.szPname;
deviceName = woc.szPname;
}
} else {
HRESULT hr = S_OK;
CoInitialize(NULL);
IMMDeviceEnumerator* pMMDeviceEnumerator = NULL;
CoCreateInstance(__uuidof(MMDeviceEnumerator), NULL, CLSCTX_ALL, __uuidof(IMMDeviceEnumerator), (void**)&pMMDeviceEnumerator);
IMMDevice* pEndpoint;
hr = pMMDeviceEnumerator->GetDefaultAudioEndpoint(mode == QAudio::AudioOutput ? eRender : eCapture, eMultimedia, &pEndpoint);
if (hr == E_NOTFOUND) {
printf("Audio Error: device not found\n");
deviceName = QString("NONE");
} else {
IPropertyStore* pPropertyStore;
pEndpoint->OpenPropertyStore(STGM_READ, &pPropertyStore);
pEndpoint->Release();
pEndpoint = NULL;
PROPVARIANT pv;
PropVariantInit(&pv);
hr = pPropertyStore->GetValue(PKEY_Device_FriendlyName, &pv);
pPropertyStore->Release();
pPropertyStore = NULL;
deviceName = QString::fromWCharArray((wchar_t*)pv.pwszVal);
if (!IsWindows8OrGreater()) {
// Windows 7 provides only the 31 first characters of the device name.
const DWORD QT_WIN7_MAX_AUDIO_DEVICENAME_LEN = 31;
deviceName = deviceName.left(QT_WIN7_MAX_AUDIO_DEVICENAME_LEN);
}
qCDebug(audioclient) << (mode == QAudio::AudioOutput ? "output" : "input") << " device:" << deviceName;
PropVariantClear(&pv);
}
pMMDeviceEnumerator->Release();
pMMDeviceEnumerator = NULL;
CoUninitialize();
}
qCDebug(audioclient) << "DEBUG [" << deviceName << "] [" << getNamedAudioDeviceForMode(mode, deviceName).deviceName() << "]";
return getNamedAudioDeviceForMode(mode, deviceName);
#endif
// fallback for failed lookup is the default device
return (mode == QAudio::AudioInput) ? QAudioDeviceInfo::defaultInputDevice() : QAudioDeviceInfo::defaultOutputDevice();
}
bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
const QAudioFormat& desiredAudioFormat,
QAudioFormat& adjustedAudioFormat) {
if (!audioDevice.isFormatSupported(desiredAudioFormat)) {
qCDebug(audioclient) << "The desired format for audio I/O is" << desiredAudioFormat;
qCDebug(audioclient, "The desired audio format is not supported by this device");
if (desiredAudioFormat.channelCount() == 1) {
adjustedAudioFormat = desiredAudioFormat;
adjustedAudioFormat.setChannelCount(2);
if (audioDevice.isFormatSupported(adjustedAudioFormat)) {
return true;
} else {
adjustedAudioFormat.setChannelCount(1);
}
}
const int FORTY_FOUR = 44100;
adjustedAudioFormat = desiredAudioFormat;
#ifdef Q_OS_ANDROID
adjustedAudioFormat.setSampleRate(FORTY_FOUR);
#else
const int HALF_FORTY_FOUR = FORTY_FOUR / 2;
if (audioDevice.supportedSampleRates().contains(AudioConstants::SAMPLE_RATE * 2)) {
// use 48, which is a sample downsample, upsample
adjustedAudioFormat.setSampleRate(AudioConstants::SAMPLE_RATE * 2);
} else if (audioDevice.supportedSampleRates().contains(HALF_FORTY_FOUR)) {
// use 22050, resample but closer to 24
adjustedAudioFormat.setSampleRate(HALF_FORTY_FOUR);
} else if (audioDevice.supportedSampleRates().contains(FORTY_FOUR)) {
// use 48000, libsoxr will resample
adjustedAudioFormat.setSampleRate(FORTY_FOUR);
}
#endif
if (adjustedAudioFormat != desiredAudioFormat) {
// return the nearest in case it needs 2 channels
adjustedAudioFormat = audioDevice.nearestFormat(adjustedAudioFormat);
return true;
} else {
return false;
}
} else {
// set the adjustedAudioFormat to the desiredAudioFormat, since it will work
adjustedAudioFormat = desiredAudioFormat;
return true;
}
}
bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationSamples, unsigned int numSourceSamples,
const QAudioFormat& sourceAudioFormat, const QAudioFormat& destinationAudioFormat) {
if (sourceAudioFormat.channelCount() == 2 && destinationAudioFormat.channelCount() == 1) {
// loop through the stereo input audio samples and average every two samples
for (uint i = 0; i < numSourceSamples; i += 2) {
destinationSamples[i / 2] = (sourceSamples[i] / 2) + (sourceSamples[i + 1] / 2);
}
return true;
} else if (sourceAudioFormat.channelCount() == 1 && destinationAudioFormat.channelCount() == 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];
}
return true;
}
return false;
}
soxr_error_t possibleResampling(soxr_t resampler,
const int16_t* sourceSamples, int16_t* destinationSamples,
unsigned int numSourceSamples, unsigned int numDestinationSamples,
const QAudioFormat& sourceAudioFormat, const QAudioFormat& destinationAudioFormat) {
if (numSourceSamples > 0) {
if (!resampler) {
if (!sampleChannelConversion(sourceSamples, destinationSamples, numSourceSamples,
sourceAudioFormat, destinationAudioFormat)) {
// no conversion, we can copy the samples directly across
memcpy(destinationSamples, sourceSamples, numSourceSamples * sizeof(int16_t));
}
return 0;
} else {
soxr_error_t resampleError = 0;
if (sourceAudioFormat.channelCount() != destinationAudioFormat.channelCount()) {
float channelCountRatio = (float) destinationAudioFormat.channelCount() / sourceAudioFormat.channelCount();
int numChannelCoversionSamples = (int) (numSourceSamples * channelCountRatio);
int16_t* channelConversionSamples = new int16_t[numChannelCoversionSamples];
sampleChannelConversion(sourceSamples, channelConversionSamples,
numSourceSamples,
sourceAudioFormat, destinationAudioFormat);
resampleError = soxr_process(resampler,
channelConversionSamples, numChannelCoversionSamples, NULL,
destinationSamples, numDestinationSamples, NULL);
delete[] channelConversionSamples;
} else {
unsigned int numAdjustedSourceSamples = numSourceSamples;
unsigned int numAdjustedDestinationSamples = numDestinationSamples;
if (sourceAudioFormat.channelCount() == 2 && destinationAudioFormat.channelCount() == 2) {
numAdjustedSourceSamples /= 2;
numAdjustedDestinationSamples /= 2;
}
resampleError = soxr_process(resampler,
sourceSamples, numAdjustedSourceSamples, NULL,
destinationSamples, numAdjustedDestinationSamples, NULL);
}
return resampleError;
}
} else {
return 0;
}
}
soxr_t soxrResamplerFromInputFormatToOutputFormat(const QAudioFormat& sourceAudioFormat,
const QAudioFormat& destinationAudioFormat) {
soxr_error_t soxrError;
// setup soxr_io_spec_t for input and output
soxr_io_spec_t inputToNetworkSpec = soxr_io_spec(soxrDataTypeFromQAudioFormat(sourceAudioFormat),
soxrDataTypeFromQAudioFormat(destinationAudioFormat));
// setup soxr_quality_spec_t for quality options
soxr_quality_spec_t qualitySpec = soxr_quality_spec(SOXR_MQ, 0);
int channelCount = (sourceAudioFormat.channelCount() == 2 && destinationAudioFormat.channelCount() == 2)
? 2 : 1;
soxr_t newResampler = soxr_create(sourceAudioFormat.sampleRate(),
destinationAudioFormat.sampleRate(),
channelCount,
&soxrError, &inputToNetworkSpec, &qualitySpec, 0);
if (soxrError) {
qCDebug(audioclient) << "There was an error setting up the soxr resampler -" << "soxr error code was " << soxrError;
soxr_delete(newResampler);
return NULL;
}
return newResampler;
}
void AudioClient::start() {
// set up the desired audio format
_desiredInputFormat.setSampleRate(AudioConstants::SAMPLE_RATE);
_desiredInputFormat.setSampleSize(16);
_desiredInputFormat.setCodec("audio/pcm");
_desiredInputFormat.setSampleType(QAudioFormat::SignedInt);
_desiredInputFormat.setByteOrder(QAudioFormat::LittleEndian);
_desiredInputFormat.setChannelCount(1);
_desiredOutputFormat = _desiredInputFormat;
_desiredOutputFormat.setChannelCount(2);
QAudioDeviceInfo inputDeviceInfo = defaultAudioDeviceForMode(QAudio::AudioInput);
qCDebug(audioclient) << "The default audio input device is" << inputDeviceInfo.deviceName();
bool inputFormatSupported = switchInputToAudioDevice(inputDeviceInfo);
QAudioDeviceInfo outputDeviceInfo = defaultAudioDeviceForMode(QAudio::AudioOutput);
qCDebug(audioclient) << "The default audio output device is" << outputDeviceInfo.deviceName();
bool outputFormatSupported = switchOutputToAudioDevice(outputDeviceInfo);
if (!inputFormatSupported) {
qCDebug(audioclient) << "Unable to set up audio input because of a problem with input format.";
qCDebug(audioclient) << "The closest format available is" << inputDeviceInfo.nearestFormat(_desiredInputFormat);
}
if (!outputFormatSupported) {
qCDebug(audioclient) << "Unable to set up audio output because of a problem with output format.";
qCDebug(audioclient) << "The closest format available is" << outputDeviceInfo.nearestFormat(_desiredOutputFormat);
}
if (_audioInput) {
_inputFrameBuffer.initialize( _inputFormat.channelCount(), _audioInput->bufferSize() * 8 );
}
_inputGain.initialize();
_sourceGain.initialize();
_noiseSource.initialize();
_toneSource.initialize();
_sourceGain.setParameters(0.05f, 0.0f);
_inputGain.setParameters(1.0f, 0.0f);
}
void AudioClient::stop() {
_inputFrameBuffer.finalize();
_inputGain.finalize();
_sourceGain.finalize();
_noiseSource.finalize();
_toneSource.finalize();
// "switch" to invalid devices in order to shut down the state
switchInputToAudioDevice(QAudioDeviceInfo());
switchOutputToAudioDevice(QAudioDeviceInfo());
if (_loopbackResampler) {
soxr_delete(_loopbackResampler);
_loopbackResampler = NULL;
}
}
QString AudioClient::getDefaultDeviceName(QAudio::Mode mode) {
QAudioDeviceInfo deviceInfo = defaultAudioDeviceForMode(mode);
return deviceInfo.deviceName();
}
QVector<QString> AudioClient::getDeviceNames(QAudio::Mode mode) {
QVector<QString> deviceNames;
foreach(QAudioDeviceInfo audioDevice, QAudioDeviceInfo::availableDevices(mode)) {
deviceNames << audioDevice.deviceName().trimmed();
}
return deviceNames;
}
bool AudioClient::switchInputToAudioDevice(const QString& inputDeviceName) {
qCDebug(audioclient) << "DEBUG [" << inputDeviceName << "] [" << getNamedAudioDeviceForMode(QAudio::AudioInput, inputDeviceName).deviceName() << "]";
return switchInputToAudioDevice(getNamedAudioDeviceForMode(QAudio::AudioInput, inputDeviceName));
}
bool AudioClient::switchOutputToAudioDevice(const QString& outputDeviceName) {
qCDebug(audioclient) << "DEBUG [" << outputDeviceName << "] [" << getNamedAudioDeviceForMode(QAudio::AudioOutput, outputDeviceName).deviceName() << "]";
return switchOutputToAudioDevice(getNamedAudioDeviceForMode(QAudio::AudioOutput, outputDeviceName));
}
ty_gverb* AudioClient::createGverbFilter() {
// Initialize a new gverb instance
ty_gverb* filter = gverb_new(_outputFormat.sampleRate(), _reverbOptions->getMaxRoomSize(), _reverbOptions->getRoomSize(),
_reverbOptions->getReverbTime(), _reverbOptions->getDamping(), _reverbOptions->getSpread(),
_reverbOptions->getInputBandwidth(), _reverbOptions->getEarlyLevel(),
_reverbOptions->getTailLevel());
return filter;
}
void AudioClient::configureGverbFilter(ty_gverb* filter) {
// Configure the instance (these functions are not super well named - they actually set several internal variables)
gverb_set_roomsize(filter, _reverbOptions->getRoomSize());
gverb_set_revtime(filter, _reverbOptions->getReverbTime());
gverb_set_damping(filter, _reverbOptions->getDamping());
gverb_set_inputbandwidth(filter, _reverbOptions->getInputBandwidth());
gverb_set_earlylevel(filter, DB_CO(_reverbOptions->getEarlyLevel()));
gverb_set_taillevel(filter, DB_CO(_reverbOptions->getTailLevel()));
}
void AudioClient::updateGverbOptions() {
bool reverbChanged = false;
if (_receivedAudioStream.hasReverb()) {
if (_zoneReverbOptions.getReverbTime() != _receivedAudioStream.getRevebTime()) {
_zoneReverbOptions.setReverbTime(_receivedAudioStream.getRevebTime());
reverbChanged = true;
}
if (_zoneReverbOptions.getWetLevel() != _receivedAudioStream.getWetLevel()) {
_zoneReverbOptions.setWetLevel(_receivedAudioStream.getWetLevel());
// Not part of actual filter config, no need to set reverbChanged to true
}
if (_reverbOptions != &_zoneReverbOptions) {
_reverbOptions = &_zoneReverbOptions;
reverbChanged = true;
}
} else if (_reverbOptions != &_scriptReverbOptions) {
_reverbOptions = &_scriptReverbOptions;
reverbChanged = true;
}
if (reverbChanged) {
gverb_free(_gverb);
_gverb = createGverbFilter();
configureGverbFilter(_gverb);
}
}
void AudioClient::setReverb(bool reverb) {
_reverb = reverb;
if (!_reverb) {
gverb_flush(_gverb);
}
}
void AudioClient::setReverbOptions(const AudioEffectOptions* options) {
// Save the new options
_scriptReverbOptions.setMaxRoomSize(options->getMaxRoomSize());
_scriptReverbOptions.setRoomSize(options->getRoomSize());
_scriptReverbOptions.setReverbTime(options->getReverbTime());
_scriptReverbOptions.setDamping(options->getDamping());
_scriptReverbOptions.setSpread(options->getSpread());
_scriptReverbOptions.setInputBandwidth(options->getInputBandwidth());
_scriptReverbOptions.setEarlyLevel(options->getEarlyLevel());
_scriptReverbOptions.setTailLevel(options->getTailLevel());
_scriptReverbOptions.setDryLevel(options->getDryLevel());
_scriptReverbOptions.setWetLevel(options->getWetLevel());
if (_reverbOptions == &_scriptReverbOptions) {
// Apply them to the reverb instances
gverb_free(_gverb);
_gverb = createGverbFilter();
configureGverbFilter(_gverb);
}
}
void AudioClient::addReverb(ty_gverb* gverb, int16_t* samplesData, int16_t* reverbAlone, int numSamples,
QAudioFormat& audioFormat, bool noEcho) {
float wetFraction = DB_CO(_reverbOptions->getWetLevel());
float dryFraction = 1.0f - wetFraction;
float lValue,rValue;
for (int sample = 0; sample < numSamples; sample += audioFormat.channelCount()) {
// Run GVerb
float value = (float)samplesData[sample];
gverb_do(gverb, value, &lValue, &rValue);
// Mix, accounting for clipping, the left and right channels. Ignore the rest.
for (int j = sample; j < sample + audioFormat.channelCount(); j++) {
if (j == sample) {
// left channel
int lResult = glm::clamp((int)(samplesData[j] * dryFraction + lValue * wetFraction),
AudioConstants::MIN_SAMPLE_VALUE, AudioConstants::MAX_SAMPLE_VALUE);
samplesData[j] = (int16_t)lResult;
if (noEcho) {
reverbAlone[j] = (int16_t)lValue * wetFraction;
}
} else if (j == (sample + 1)) {
// right channel
int rResult = glm::clamp((int)(samplesData[j] * dryFraction + rValue * wetFraction),
AudioConstants::MIN_SAMPLE_VALUE, AudioConstants::MAX_SAMPLE_VALUE);
samplesData[j] = (int16_t)rResult;
if (noEcho) {
reverbAlone[j] = (int16_t)rValue * wetFraction;
}
} else {
// ignore channels above 2
}
}
}
}
void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
// If there is server echo, reverb will be applied to the recieved audio stream so no need to have it here.
bool hasReverb = _reverb || _receivedAudioStream.hasReverb();
if (_muted || !_audioOutput || (!_shouldEchoLocally && !hasReverb)) {
return;
}
// if this person wants local loopback add that to the locally injected audio
// if there is reverb apply it to local audio and substract the origin samples
if (!_loopbackOutputDevice && _loopbackAudioOutput) {
// we didn't have the loopback output device going so set that up now
_loopbackOutputDevice = _loopbackAudioOutput->start();
if (!_loopbackOutputDevice) {
return;
}
}
// do we need to setup a resampler?
if (_inputFormat.sampleRate() != _outputFormat.sampleRate() && !_loopbackResampler) {
qCDebug(audioclient) << "Attemping to create a resampler for input format to output format for audio loopback.";
_loopbackResampler = soxrResamplerFromInputFormatToOutputFormat(_inputFormat, _outputFormat);
if (!_loopbackResampler) {
return;
}
}
static QByteArray reverbAlone; // Intermediary for local reverb with no echo
static QByteArray loopBackByteArray;
int numInputSamples = inputByteArray.size() / sizeof(int16_t);
int numLoopbackSamples = numDestinationSamplesRequired(_inputFormat, _outputFormat, numInputSamples);
reverbAlone.resize(numInputSamples * sizeof(int16_t));
loopBackByteArray.resize(numLoopbackSamples * sizeof(int16_t));
int16_t* inputSamples = reinterpret_cast<int16_t*>(inputByteArray.data());
int16_t* reverbAloneSamples = reinterpret_cast<int16_t*>(reverbAlone.data());
int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
if (hasReverb) {
updateGverbOptions();
addReverb(_gverb, inputSamples, reverbAloneSamples, numInputSamples,
_inputFormat, !_shouldEchoLocally);
}
possibleResampling(_loopbackResampler,
(_shouldEchoLocally) ? inputSamples : reverbAloneSamples, loopbackSamples,
numInputSamples, numLoopbackSamples,
_inputFormat, _outputFormat);
_loopbackOutputDevice->write(loopBackByteArray);
}
void AudioClient::handleAudioInput() {
static char audioDataPacket[MAX_PACKET_SIZE];
static int numBytesPacketHeader = numBytesForPacketHeaderGivenPacketType(PacketTypeMicrophoneAudioNoEcho);
// NOTE: we assume PacketTypeMicrophoneAudioWithEcho has same size headers as
// PacketTypeMicrophoneAudioNoEcho. If not, then networkAudioSamples will be pointing to the wrong place for writing
// audio samples with echo.
static int leadingBytes = numBytesPacketHeader + sizeof(quint16) + sizeof(glm::vec3) + sizeof(glm::quat) + sizeof(quint8);
static int16_t* networkAudioSamples = (int16_t*)(audioDataPacket + leadingBytes);
float inputToNetworkInputRatio = calculateDeviceToNetworkInputRatio();
int inputSamplesRequired = (int)((float)AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * inputToNetworkInputRatio);
QByteArray inputByteArray = _inputDevice->readAll();
// Add audio source injection if enabled
if (!_muted && _audioSourceInjectEnabled) {
int16_t* inputFrameData = (int16_t*)inputByteArray.data();
const uint32_t inputFrameCount = inputByteArray.size() / sizeof(int16_t);
_inputFrameBuffer.copyFrames(1, inputFrameCount, inputFrameData, false /*copy in*/);
#if ENABLE_INPUT_GAIN
_inputGain.render(_inputFrameBuffer); // input/mic gain+mute
#endif
if (_toneSourceEnabled) { // sine generator
_toneSource.render(_inputFrameBuffer);
} else if(_noiseSourceEnabled) { // pink noise generator
_noiseSource.render(_inputFrameBuffer);
}
_sourceGain.render(_inputFrameBuffer); // post gain
_inputFrameBuffer.copyFrames(1, inputFrameCount, inputFrameData, true /*copy out*/);
}
handleLocalEchoAndReverb(inputByteArray);
_inputRingBuffer.writeData(inputByteArray.data(), inputByteArray.size());
float audioInputMsecsRead = inputByteArray.size() / (float)(_inputFormat.bytesForDuration(USECS_PER_MSEC));
_stats.updateInputMsecsRead(audioInputMsecsRead);
while (_inputRingBuffer.samplesAvailable() >= inputSamplesRequired) {
const int numNetworkBytes = _isStereoInput
? AudioConstants::NETWORK_FRAME_BYTES_STEREO
: AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL;
const int numNetworkSamples = _isStereoInput
? AudioConstants::NETWORK_FRAME_SAMPLES_STEREO
: AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL;
if (!_muted) {
// zero out the monoAudioSamples array and the locally injected audio
memset(networkAudioSamples, 0, numNetworkBytes);
// Increment the time since the last clip
if (_timeSinceLastClip >= 0.0f) {
_timeSinceLastClip += (float) numNetworkSamples / (float) AudioConstants::SAMPLE_RATE;
}
int16_t* inputAudioSamples = new int16_t[inputSamplesRequired];
_inputRingBuffer.readSamples(inputAudioSamples, inputSamplesRequired);
possibleResampling(_inputToNetworkResampler,
inputAudioSamples, networkAudioSamples,
inputSamplesRequired, numNetworkSamples,
_inputFormat, _desiredInputFormat);
delete[] inputAudioSamples;
// Remove DC offset
if (!_isStereoInput && !_audioSourceInjectEnabled) {
_inputGate.removeDCOffset(networkAudioSamples, numNetworkSamples);
}
// only impose the noise gate and perform tone injection if we are sending mono audio
if (!_isStereoInput && !_audioSourceInjectEnabled && _isNoiseGateEnabled) {
_inputGate.gateSamples(networkAudioSamples, numNetworkSamples);
// if we performed the noise gate we can get values from it instead of enumerating the samples again
_lastInputLoudness = _inputGate.getLastLoudness();
if (_inputGate.clippedInLastFrame()) {
_timeSinceLastClip = 0.0f;
}
} else {
float loudness = 0.0f;
for (int i = 0; i < numNetworkSamples; i++) {
int thisSample = std::abs(networkAudioSamples[i]);
loudness += (float) thisSample;
if (thisSample > (AudioConstants::MAX_SAMPLE_VALUE * AudioNoiseGate::CLIPPING_THRESHOLD)) {
_timeSinceLastClip = 0.0f;
}
}
_lastInputLoudness = fabs(loudness / numNetworkSamples);
}
emit inputReceived(QByteArray(reinterpret_cast<const char*>(networkAudioSamples),
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * sizeof(AudioConstants::AudioSample)));
} else {
// our input loudness is 0, since we're muted
_lastInputLoudness = 0;
_timeSinceLastClip = 0.0f;
_inputRingBuffer.shiftReadPosition(inputSamplesRequired);
}
auto nodeList = DependencyManager::get<NodeList>();
SharedNodePointer audioMixer = nodeList->soloNodeOfType(NodeType::AudioMixer);
if (audioMixer && audioMixer->getActiveSocket()) {
glm::vec3 headPosition = _positionGetter();
glm::quat headOrientation = _orientationGetter();
quint8 isStereo = _isStereoInput ? 1 : 0;
PacketType packetType;
if (_lastInputLoudness == 0) {
packetType = PacketTypeSilentAudioFrame;
} else {
if (_shouldEchoToServer) {
packetType = PacketTypeMicrophoneAudioWithEcho;
} else {
packetType = PacketTypeMicrophoneAudioNoEcho;
}
}
char* currentPacketPtr = audioDataPacket + nodeList->populatePacketHeader(audioDataPacket, packetType);
// pack sequence number
memcpy(currentPacketPtr, &_outgoingAvatarAudioSequenceNumber, sizeof(quint16));
currentPacketPtr += sizeof(quint16);
if (packetType == PacketTypeSilentAudioFrame) {
// pack num silent samples
quint16 numSilentSamples = numNetworkSamples;
memcpy(currentPacketPtr, &numSilentSamples, sizeof(quint16));
currentPacketPtr += sizeof(quint16);
// memcpy the three float positions
memcpy(currentPacketPtr, &headPosition, sizeof(headPosition));
currentPacketPtr += (sizeof(headPosition));
// memcpy our orientation
memcpy(currentPacketPtr, &headOrientation, sizeof(headOrientation));
currentPacketPtr += sizeof(headOrientation);
} else {
// set the mono/stereo byte
*currentPacketPtr++ = isStereo;
// memcpy the three float positions
memcpy(currentPacketPtr, &headPosition, sizeof(headPosition));
currentPacketPtr += (sizeof(headPosition));
// memcpy our orientation
memcpy(currentPacketPtr, &headOrientation, sizeof(headOrientation));
currentPacketPtr += sizeof(headOrientation);
// audio samples have already been packed (written to networkAudioSamples)
currentPacketPtr += numNetworkBytes;
}
_stats.sentPacket();
nodeList->flagTimeForConnectionStep(LimitedNodeList::ConnectionStep::SendAudioPacket);
int packetBytes = currentPacketPtr - audioDataPacket;
nodeList->writeDatagram(audioDataPacket, packetBytes, audioMixer);
_outgoingAvatarAudioSequenceNumber++;
}
}
}
void AudioClient::processReceivedSamples(const QByteArray& inputBuffer, QByteArray& outputBuffer) {
const int numNetworkOutputSamples = inputBuffer.size() / sizeof(int16_t);
const int numDeviceOutputSamples = numNetworkOutputSamples * (_outputFormat.sampleRate() * _outputFormat.channelCount())
/ (_desiredOutputFormat.sampleRate() * _desiredOutputFormat.channelCount());
outputBuffer.resize(numDeviceOutputSamples * sizeof(int16_t));
const int16_t* receivedSamples = reinterpret_cast<const int16_t*>(inputBuffer.data());
// copy the packet from the RB to the output
possibleResampling(_networkToOutputResampler, receivedSamples,
reinterpret_cast<int16_t*>(outputBuffer.data()),
numNetworkOutputSamples, numDeviceOutputSamples,
_desiredOutputFormat, _outputFormat);
}
void AudioClient::sendMuteEnvironmentPacket() {
auto nodeList = DependencyManager::get<NodeList>();
int dataSize = sizeof(glm::vec3) + sizeof(float);
NodeList::Packet mutePacket = nodeList->makePacket(PacketType::MuteEnvironment, dataSize);
const float MUTE_RADIUS = 50;
glm::vec3 currentSourcePosition = _positionGetter();
memcpy(mutePacket.payload().data(), &currentSourcePosition, sizeof(glm::vec3));
memcpy(mutePacket.payload() + sizeof(glm::vec3), &MUTE_RADIUS, sizeof(float));
// grab our audio mixer from the NodeList, if it exists
SharedNodePointer audioMixer = nodeList->soloNodeOfType(NodeType::AudioMixer);
if (audioMixer) {
// send off this mute packet
nodeList->sendPacket(mutePacket, audioMixer);
}
}
void AudioClient::addReceivedAudioToStream(const QByteArray& audioByteArray) {
DependencyManager::get<NodeList>()->flagTimeForConnectionStep(LimitedNodeList::ConnectionStep::ReceiveFirstAudioPacket);
if (_audioOutput) {
if (!_hasReceivedFirstPacket) {
_hasReceivedFirstPacket = true;
// have the audio scripting interface emit a signal to say we just connected to mixer
emit receivedFirstPacket();
}
// Audio output must exist and be correctly set up if we're going to process received audio
_receivedAudioStream.parseData(audioByteArray);
}
}
void AudioClient::parseAudioEnvironmentData(const QByteArray &packet) {
int numBytesPacketHeader = numBytesForPacketHeader(packet);
const char* dataAt = packet.constData() + numBytesPacketHeader;
char bitset;
memcpy(&bitset, dataAt, sizeof(char));
dataAt += sizeof(char);
bool hasReverb = oneAtBit(bitset, HAS_REVERB_BIT);;
if (hasReverb) {
float reverbTime, wetLevel;
memcpy(&reverbTime, dataAt, sizeof(float));
dataAt += sizeof(float);
memcpy(&wetLevel, dataAt, sizeof(float));
dataAt += sizeof(float);
_receivedAudioStream.setReverb(reverbTime, wetLevel);
} else {
_receivedAudioStream.clearReverb();
}
}
void AudioClient::toggleMute() {
_muted = !_muted;
emit muteToggled();
}
void AudioClient::setIsStereoInput(bool isStereoInput) {
if (isStereoInput != _isStereoInput) {
_isStereoInput = isStereoInput;
if (_isStereoInput) {
_desiredInputFormat.setChannelCount(2);
} else {
_desiredInputFormat.setChannelCount(1);
}
// change in channel count for desired input format, restart the input device
switchInputToAudioDevice(_inputAudioDeviceName);
}
}
void AudioClient::enableAudioSourceInject(bool enable) {
_audioSourceInjectEnabled = enable;
}
void AudioClient::selectAudioSourcePinkNoise() {
_noiseSourceEnabled = true;
_toneSourceEnabled = false;
}
void AudioClient::selectAudioSourceSine440() {
_toneSourceEnabled = true;
_noiseSourceEnabled = false;
}
bool AudioClient::outputLocalInjector(bool isStereo, AudioInjector* injector) {
if (injector->getLocalBuffer()) {
QAudioFormat localFormat = _desiredOutputFormat;
localFormat.setChannelCount(isStereo ? 2 : 1);
QAudioOutput* localOutput = new QAudioOutput(getNamedAudioDeviceForMode(QAudio::AudioOutput, _outputAudioDeviceName),
localFormat,
injector->getLocalBuffer());
// move the localOutput to the same thread as the local injector buffer
localOutput->moveToThread(injector->getLocalBuffer()->thread());
// have it be stopped when that local buffer is about to close
connect(localOutput, &QAudioOutput::stateChanged, this, &AudioClient::audioStateChanged);
connect(this, &AudioClient::audioFinished, localOutput, &QAudioOutput::stop);
connect(this, &AudioClient::audioFinished, injector, &AudioInjector::stop);
connect(injector->getLocalBuffer(), &QIODevice::aboutToClose, localOutput, &QAudioOutput::stop);
qCDebug(audioclient) << "Starting QAudioOutput for local injector" << localOutput;
localOutput->start(injector->getLocalBuffer());
return localOutput->state() == QAudio::ActiveState;
}
return false;
}
void AudioClient::outputFormatChanged() {
int outputFormatChannelCountTimesSampleRate = _outputFormat.channelCount() * _outputFormat.sampleRate();
_outputFrameSize = AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * outputFormatChannelCountTimesSampleRate / _desiredOutputFormat.sampleRate();
_receivedAudioStream.outputFormatChanged(outputFormatChannelCountTimesSampleRate);
}
bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceInfo) {
bool supportedFormat = false;
// cleanup any previously initialized device
if (_audioInput) {
// The call to stop() causes _inputDevice to be destructed.
// That in turn causes it to be disconnected (see for example
// http://stackoverflow.com/questions/9264750/qt-signals-and-slots-object-disconnect).
_audioInput->stop();
_inputDevice = NULL;
delete _audioInput;
_audioInput = NULL;
_numInputCallbackBytes = 0;
_inputAudioDeviceName = "";
}
if (_inputToNetworkResampler) {
// if we were using an input to network resampler, delete it here
soxr_delete(_inputToNetworkResampler);
_inputToNetworkResampler = NULL;
}
if (!inputDeviceInfo.isNull()) {
qCDebug(audioclient) << "The audio input device " << inputDeviceInfo.deviceName() << "is available.";
_inputAudioDeviceName = inputDeviceInfo.deviceName().trimmed();
if (adjustedFormatForAudioDevice(inputDeviceInfo, _desiredInputFormat, _inputFormat)) {
qCDebug(audioclient) << "The format to be used for audio input is" << _inputFormat;
// we've got the best we can get for input
// if required, setup a soxr resampler for this input to our desired network format
if (_inputFormat != _desiredInputFormat
&& _inputFormat.sampleRate() != _desiredInputFormat.sampleRate()) {
qCDebug(audioclient) << "Attemping to create a soxr resampler for input format to network format.";
_inputToNetworkResampler = soxrResamplerFromInputFormatToOutputFormat(_inputFormat, _desiredInputFormat);
if (!_inputToNetworkResampler) {
return false;
}
} else {
qCDebug(audioclient) << "No resampling required for audio input to match desired network format.";
}
// if the user wants stereo but this device can't provide then bail
if (!_isStereoInput || _inputFormat.channelCount() == 2) {
_audioInput = new QAudioInput(inputDeviceInfo, _inputFormat, this);
_numInputCallbackBytes = calculateNumberOfInputCallbackBytes(_inputFormat);
_audioInput->setBufferSize(_numInputCallbackBytes);
// how do we want to handle input working, but output not working?
int numFrameSamples = calculateNumberOfFrameSamples(_numInputCallbackBytes);
_inputRingBuffer.resizeForFrameSize(numFrameSamples);
_inputDevice = _audioInput->start();
if (_inputDevice) {
connect(_inputDevice, SIGNAL(readyRead()), this, SLOT(handleAudioInput()));
supportedFormat = true;
} else {
qCDebug(audioclient) << "Error starting audio input -" << _audioInput->error();
}
}
}
}
return supportedFormat;
}
void AudioClient::outputNotify() {
int recentUnfulfilled = _audioOutputIODevice.getRecentUnfulfilledReads();
if (recentUnfulfilled > 0) {
if (_outputStarveDetectionEnabled.get()) {
quint64 now = usecTimestampNow() / 1000;
int dt = (int)(now - _outputStarveDetectionStartTimeMsec);
if (dt > _outputStarveDetectionPeriodMsec.get()) {
_outputStarveDetectionStartTimeMsec = now;
_outputStarveDetectionCount = 0;
} else {
_outputStarveDetectionCount += recentUnfulfilled;
if (_outputStarveDetectionCount > _outputStarveDetectionThreshold.get()) {
_outputStarveDetectionStartTimeMsec = now;
_outputStarveDetectionCount = 0;
int oldOutputBufferSizeFrames = _outputBufferSizeFrames.get();
int newOutputBufferSizeFrames = oldOutputBufferSizeFrames + 1;
setOutputBufferSize(newOutputBufferSizeFrames);
newOutputBufferSizeFrames = _outputBufferSizeFrames.get();
if (newOutputBufferSizeFrames > oldOutputBufferSizeFrames) {
qCDebug(audioclient) << "Starve detection threshold met, increasing buffer size to " << newOutputBufferSizeFrames;
}
}
}
}
}
}
bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDeviceInfo) {
bool supportedFormat = false;
// cleanup any previously initialized device
if (_audioOutput) {
_audioOutput->stop();
delete _audioOutput;
_audioOutput = NULL;
_loopbackOutputDevice = NULL;
delete _loopbackAudioOutput;
_loopbackAudioOutput = NULL;
}
if (_networkToOutputResampler) {
// if we were using an input to network resampler, delete it here
soxr_delete(_networkToOutputResampler);
_networkToOutputResampler = NULL;
}
if (_loopbackResampler) {
// if we were using an input to output resample, delete it here
soxr_delete(_loopbackResampler);
_loopbackResampler = NULL;
}
if (!outputDeviceInfo.isNull()) {
qCDebug(audioclient) << "The audio output device " << outputDeviceInfo.deviceName() << "is available.";
_outputAudioDeviceName = outputDeviceInfo.deviceName().trimmed();
if (adjustedFormatForAudioDevice(outputDeviceInfo, _desiredOutputFormat, _outputFormat)) {
qCDebug(audioclient) << "The format to be used for audio output is" << _outputFormat;
// we've got the best we can get for input
// if required, setup a soxr resampler for this input to our desired network format
if (_desiredOutputFormat != _outputFormat
&& _desiredOutputFormat.sampleRate() != _outputFormat.sampleRate()) {
qCDebug(audioclient) << "Attemping to create a resampler for network format to output format.";
_networkToOutputResampler = soxrResamplerFromInputFormatToOutputFormat(_desiredOutputFormat, _outputFormat);
if (!_networkToOutputResampler) {
return false;
}
} else {
qCDebug(audioclient) << "No resampling required for network output to match actual output format.";
}
outputFormatChanged();
// setup our general output device for audio-mixer audio
_audioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this);
_audioOutput->setBufferSize(_outputBufferSizeFrames.get() * _outputFrameSize * sizeof(int16_t));
connect(_audioOutput, &QAudioOutput::notify, this, &AudioClient::outputNotify);
qCDebug(audioclient) << "Output Buffer capacity in frames: " << _audioOutput->bufferSize() / sizeof(int16_t) / (float)_outputFrameSize;
_audioOutputIODevice.start();
_audioOutput->start(&_audioOutputIODevice);
// setup a loopback audio output device
_loopbackAudioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this);
_timeSinceLastReceived.start();
supportedFormat = true;
}
}
return supportedFormat;
}
void AudioClient::setOutputBufferSize(int numFrames) {
numFrames = std::min(std::max(numFrames, MIN_AUDIO_OUTPUT_BUFFER_SIZE_FRAMES), MAX_AUDIO_OUTPUT_BUFFER_SIZE_FRAMES);
if (numFrames != _outputBufferSizeFrames.get()) {
qCDebug(audioclient) << "Audio output buffer size (frames): " << numFrames;
_outputBufferSizeFrames.set(numFrames);
if (_audioOutput) {
// The buffer size can't be adjusted after QAudioOutput::start() has been called, so
// recreate the device by switching to the default.
QAudioDeviceInfo outputDeviceInfo = defaultAudioDeviceForMode(QAudio::AudioOutput);
switchOutputToAudioDevice(outputDeviceInfo);
}
}
}
// The following constant is operating system dependent due to differences in
// the way input audio is handled. The audio input buffer size is inversely
// proportional to the accelerator ratio.
#ifdef Q_OS_WIN
const float AudioClient::CALLBACK_ACCELERATOR_RATIO = 0.1f;
#endif
#ifdef Q_OS_MAC
const float AudioClient::CALLBACK_ACCELERATOR_RATIO = 2.0f;
#endif
#ifdef Q_OS_LINUX
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);
return numInputCallbackBytes;
}
float AudioClient::calculateDeviceToNetworkInputRatio() const {
float inputToNetworkInputRatio = (int)((_numInputCallbackBytes
* CALLBACK_ACCELERATOR_RATIO
/ AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL) + 0.5f);
return inputToNetworkInputRatio;
}
int AudioClient::calculateNumberOfFrameSamples(int numBytes) const {
int frameSamples = (int)(numBytes * CALLBACK_ACCELERATOR_RATIO + 0.5f) / sizeof(int16_t);
return frameSamples;
}
float AudioClient::getInputRingBufferMsecsAvailable() const {
int bytesInInputRingBuffer = _inputRingBuffer.samplesAvailable() * sizeof(int16_t);
float msecsInInputRingBuffer = bytesInInputRingBuffer / (float)(_inputFormat.bytesForDuration(USECS_PER_MSEC));
return msecsInInputRingBuffer;
}
float AudioClient::getAudioOutputMsecsUnplayed() const {
if (!_audioOutput) {
return 0.0f;
}
int bytesAudioOutputUnplayed = _audioOutput->bufferSize() - _audioOutput->bytesFree();
float msecsAudioOutputUnplayed = bytesAudioOutputUnplayed / (float)_outputFormat.bytesForDuration(USECS_PER_MSEC);
return msecsAudioOutputUnplayed;
}
qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
int samplesRequested = maxSize / sizeof(int16_t);
int samplesPopped;
int bytesWritten;
if ((samplesPopped = _receivedAudioStream.popSamples(samplesRequested, false)) > 0) {
AudioRingBuffer::ConstIterator lastPopOutput = _receivedAudioStream.getLastPopOutput();
lastPopOutput.readSamples((int16_t*)data, samplesPopped);
bytesWritten = samplesPopped * sizeof(int16_t);
} else {
memset(data, 0, maxSize);
bytesWritten = maxSize;
}
int bytesAudioOutputUnplayed = _audio->_audioOutput->bufferSize() - _audio->_audioOutput->bytesFree();
if (bytesAudioOutputUnplayed == 0 && bytesWritten == 0) {
_unfulfilledReads++;
}
return bytesWritten;
}
void AudioClient::checkDevices() {
# ifdef Q_OS_LINUX
// on linux, this makes the audio stream hiccup
# else
QVector<QString> inputDevices = getDeviceNames(QAudio::AudioInput);
QVector<QString> outputDevices = getDeviceNames(QAudio::AudioOutput);
if (inputDevices != _inputDevices || outputDevices != _outputDevices) {
_inputDevices = inputDevices;
_outputDevices = outputDevices;
emit deviceChanged();
}
# endif
}
void AudioClient::loadSettings() {
_receivedAudioStream.setDynamicJitterBuffers(dynamicJitterBuffers.get());
_receivedAudioStream.setMaxFramesOverDesired(maxFramesOverDesired.get());
_receivedAudioStream.setStaticDesiredJitterBufferFrames(staticDesiredJitterBufferFrames.get());
_receivedAudioStream.setUseStDevForJitterCalc(useStDevForJitterCalc.get());
_receivedAudioStream.setWindowStarveThreshold(windowStarveThreshold.get());
_receivedAudioStream.setWindowSecondsForDesiredCalcOnTooManyStarves(
windowSecondsForDesiredCalcOnTooManyStarves.get());
_receivedAudioStream.setWindowSecondsForDesiredReduction(windowSecondsForDesiredReduction.get());
_receivedAudioStream.setRepetitionWithFade(repetitionWithFade.get());
}
void AudioClient::saveSettings() {
dynamicJitterBuffers.set(_receivedAudioStream.getDynamicJitterBuffers());
maxFramesOverDesired.set(_receivedAudioStream.getMaxFramesOverDesired());
staticDesiredJitterBufferFrames.set(_receivedAudioStream.getDesiredJitterBufferFrames());
windowStarveThreshold.set(_receivedAudioStream.getWindowStarveThreshold());
windowSecondsForDesiredCalcOnTooManyStarves.set(_receivedAudioStream.
getWindowSecondsForDesiredCalcOnTooManyStarves());
windowSecondsForDesiredReduction.set(_receivedAudioStream.getWindowSecondsForDesiredReduction());
repetitionWithFade.set(_receivedAudioStream.getRepetitionWithFade());
}
void AudioClient::audioStateChanged(QAudio::State state) {
if (state == QAudio::IdleState) {
emit audioFinished();
}
}
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