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handle input to output resampling

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This commit is contained in:
Stephen Birarda 2015-01-23 09:23:46 -08:00
parent 8544fccd84
commit 871d688db9
2 changed files with 51 additions and 37 deletions
libraries/audio-client/src
AudioClient.cppAudioClient.h

86
libraries/audio-client/src/AudioClient.cpp
View file

@ -85,6 +85,7 @@ AudioClient::AudioClient() :
_gverbLocal(NULL),
_gverb(NULL),
_inputToNetworkResampler(NULL),
_networkToOutputResampler(NULL),
_loopbackResampler(NULL),
_noiseSourceEnabled(false),
_toneSourceEnabled(true),
@ -96,7 +97,8 @@ AudioClient::AudioClient() :
// 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);
connect(&_receivedAudioStream, &MixedProcessedAudioStream::processSamples,
this, &AudioClient::processReceivedSamples, Qt::DirectConnection);
// Initialize GVerb
initGverb();
@ -149,6 +151,12 @@ soxr_datatype_t soxrDataTypeFromQAudioFormat(const QAudioFormat& audioFormat) {
}
}
float sampleFactorBetweenFormats(const QAudioFormat& sourceFormat, const QAudioFormat& destinationFormat) {
float ratio = (float) destinationFormat.channelCount() / sourceFormat.channelCount();
ratio *= (float) destinationFormat.sampleRate() / sourceFormat.sampleRate();
return ratio;
}
QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
#ifdef __APPLE__
if (QAudioDeviceInfo::availableDevices(mode).size() > 1) {
@ -294,7 +302,7 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
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 take every second sample
// loop through the stereo input audio samples and average every two samples
for (int i = 0; i < numSourceSamples; i += 2) {
destinationSamples[i / 2] = (sourceSamples[i] / 2) + (sourceSamples[i + 1] / 2);
}
@ -302,6 +310,11 @@ bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationS
return true;
} else if (sourceAudioFormat.channelCount() == 1 && destinationAudioFormat.channelCount() == 2) {
// loop through the mono input audio and repeat each sample twice
for (int i = 0; i < numSourceSamples; ++i) {
destinationSamples[i * 2] = destinationSamples[(i * 2) + 1] = sourceSamples[i];
}
return true;
}
@ -312,11 +325,12 @@ 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 (!resampler) {
if (!sampleChannelConversion(sourceSamples, destinationSamples, numSourceSamples,
sourceAudioFormat, destinationAudioFormat)) {
// no conversion, we can copy the samples directly across
memcpy(destinationSamples, sourceSamples, numSourceSamples);
memcpy(destinationSamples, sourceSamples, numSourceSamples * sizeof(int16_t));
}
return 0;
@ -324,7 +338,7 @@ soxr_error_t possibleResampling(soxr_t resampler,
soxr_error_t resampleError = 0;
if (sourceAudioFormat.channelCount() != destinationAudioFormat.channelCount()) {
float channelCountRatio = destinationAudioFormat.channelCount() / sourceAudioFormat.channelCount();
float channelCountRatio = (float) destinationAudioFormat.channelCount() / sourceAudioFormat.channelCount();
int numChannelCoversionSamples = (int) (numSourceSamples * channelCountRatio);
int16_t* channelConversionSamples = new int16_t[numChannelCoversionSamples];
@ -571,31 +585,31 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
// 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
static QByteArray loopBackByteArray;
if (!_loopbackOutputDevice && _loopbackAudioOutput) {
// we didn't have the loopback output device going so set that up now
_loopbackOutputDevice = _loopbackAudioOutput->start();
}
QByteArray loopBackByteArray(inputByteArray);
if (_inputFormat != _outputFormat) {
loopBackByteArray.resize(sampleFactorBetweenFormats(_inputFormat, _outputFormat) * inputByteArray.size());
// do we need to setup a resampler?
if (_inputFormat.sampleRate() != _outputFormat.sampleRate() && !_loopbackResampler) {
qDebug() << "Attemping to create a resampler for input format to output format for audio loopback.";
_loopbackResampler = soxrResamplerFromInputFormatToOutputFormat(_inputFormat, _outputFormat);
// do we need to setup a resampler?
if (_inputFormat.sampleRate() != _outputFormat.sampleRate() && !_loopbackResampler) {
qDebug() << "Attemping to create a resampler for input format to output format for audio loopback.";
_loopbackResampler = soxrResamplerFromInputFormatToOutputFormat(_inputFormat, _outputFormat);
if (!_loopbackResampler) {
return;
}
if (!_loopbackResampler) {
return;
}
possibleResampling(_loopbackResampler,
reinterpret_cast<int16_t*>(inputByteArray.data()),
reinterpret_cast<int16_t*>(loopBackByteArray.data()),
inputByteArray.size() / sizeof(int16_t), loopBackByteArray.size() / sizeof(int16_t),
_inputFormat, _outputFormat);
}
possibleResampling(_loopbackResampler,
reinterpret_cast<int16_t*>(inputByteArray.data()),
reinterpret_cast<int16_t*>(loopBackByteArray.data()),
inputByteArray.size() / sizeof(int16_t), loopBackByteArray.size() / sizeof(int16_t),
_inputFormat, _outputFormat);
if (hasLocalReverb) {
int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
int numLoopbackSamples = loopBackByteArray.size() / sizeof(int16_t);
@ -619,7 +633,7 @@ void AudioClient::handleAudioInput() {
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(_numInputCallbackBytes);
float inputToNetworkInputRatio = calculateDeviceToNetworkInputRatio();
int inputSamplesRequired = (int)((float)AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL * inputToNetworkInputRatio);
@ -657,8 +671,6 @@ void AudioClient::handleAudioInput() {
_stats.updateInputMsecsRead(audioInputMsecsRead);
while (_inputRingBuffer.samplesAvailable() >= inputSamplesRequired) {
int16_t* inputAudioSamples = NULL;
const int numNetworkBytes = _isStereoInput
? AudioConstants::NETWORK_FRAME_BYTES_STEREO
@ -677,14 +689,16 @@ void AudioClient::handleAudioInput() {
_timeSinceLastClip += (float) numNetworkSamples / (float) AudioConstants::SAMPLE_RATE;
}
int16_t* stereoInputAudioSamples = new int16_t[inputSamplesRequired];
_inputRingBuffer.readSamples(stereoInputAudioSamples, inputSamplesRequired);
int16_t* inputAudioSamples = new int16_t[inputSamplesRequired];
_inputRingBuffer.readSamples(inputAudioSamples, inputSamplesRequired);
possibleResampling(_inputToNetworkResampler,
stereoInputAudioSamples, networkAudioSamples,
inputAudioSamples, networkAudioSamples,
inputSamplesRequired, numNetworkSamples,
_inputFormat, _desiredInputFormat);
delete[] inputAudioSamples;
// only impose the noise gate and perform tone injection if we are sending mono audio
if (!_isStereoInput && !_audioSourceInjectEnabled && _isNoiseGateEnabled) {
_inputGate.gateSamples(networkAudioSamples, numNetworkSamples);
@ -782,8 +796,6 @@ void AudioClient::handleAudioInput() {
emit outputBytesToNetwork(packetBytes);
}
delete[] inputAudioSamples;
}
}
@ -794,12 +806,7 @@ void AudioClient::processReceivedSamples(const QByteArray& inputBuffer, QByteArr
outputBuffer.resize(numDeviceOutputSamples * sizeof(int16_t));
const int16_t* receivedSamples;
// copy the samples we'll resample from the ring buffer - this also
// pushes the read pointer of the ring buffer forwards
//receivedAudioStreamPopOutput.readSamples(receivedSamples, numNetworkOutputSamples);
receivedSamples = reinterpret_cast<const int16_t*>(inputBuffer.data());
const int16_t* receivedSamples = reinterpret_cast<const int16_t*>(inputBuffer.data());
// copy the packet from the RB to the output
possibleResampling(_networkToOutputResampler, receivedSamples,
@ -1038,6 +1045,12 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
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()) {
qDebug() << "The audio output device " << outputDeviceInfo.deviceName() << "is available.";
@ -1057,7 +1070,7 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
return false;
}
} else {
qDebug() << "No resampling required for audio input to match desired network format.";
qDebug() << "No resampling required for network output to match actual output format.";
}
outputFormatChanged();
@ -1075,6 +1088,7 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
// setup a loopback audio output device
_loopbackAudioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this);
_timeSinceLastReceived.start();
@ -1125,7 +1139,7 @@ int AudioClient::calculateNumberOfInputCallbackBytes(const QAudioFormat& format)
return numInputCallbackBytes;
}
float AudioClient::calculateDeviceToNetworkInputRatio(int numBytes) const {
float AudioClient::calculateDeviceToNetworkInputRatio() const {
float inputToNetworkInputRatio = (int)((_numInputCallbackBytes
* CALLBACK_ACCELERATOR_RATIO
/ AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL) + 0.5f);

2
libraries/audio-client/src/AudioClient.h
View file

@ -251,7 +251,7 @@ private:
// Callback acceleration dependent calculations
int calculateNumberOfInputCallbackBytes(const QAudioFormat& format) const;
int calculateNumberOfFrameSamples(int numBytes) const;
float calculateDeviceToNetworkInputRatio(int numBytes) const;
float calculateDeviceToNetworkInputRatio() const;
// Input framebuffer
AudioBufferFloat32 _inputFrameBuffer;