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fix for local loop resample

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This commit is contained in:
Stephen Birarda 2015-01-23 10:02:03 -08:00
parent 871d688db9
commit 39d9d99045
1 changed files with 48 additions and 39 deletions
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87
libraries/audio-client/src/AudioClient.cpp
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@ -151,10 +151,12 @@ soxr_datatype_t soxrDataTypeFromQAudioFormat(const QAudioFormat& audioFormat) {
}
}
float sampleFactorBetweenFormats(const QAudioFormat& sourceFormat, const QAudioFormat& destinationFormat) {
int numDestinationSamplesRequired(const QAudioFormat& sourceFormat, const QAudioFormat& destinationFormat,
int numSourceSamples) {
float ratio = (float) destinationFormat.channelCount() / sourceFormat.channelCount();
ratio *= (float) destinationFormat.sampleRate() / sourceFormat.sampleRate();
return ratio;
return (numSourceSamples * ratio) + 0.5f;
}
QAudioDeviceInfo defaultAudioDeviceForMode(QAudio::Mode mode) {
@ -326,39 +328,46 @@ soxr_error_t possibleResampling(soxr_t resampler,
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 * sizeof(int16_t));
}
return 0;
} else {
soxr_error_t resampleError = 0;
if (sourceAudioFormat.channelCount() != destinationAudioFormat.channelCount()) {
float channelCountRatio = (float) destinationAudioFormat.channelCount() / sourceAudioFormat.channelCount();
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));
}
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;
return 0;
} else {
resampleError = soxr_process(resampler,
sourceSamples, numSourceSamples, NULL,
destinationSamples, numDestinationSamples, NULL);
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);
qDebug() << "resample from" << sourceAudioFormat << "to" << destinationAudioFormat
<< "from" << numChannelCoversionSamples << "to" << numDestinationSamples;
resampleError = soxr_process(resampler,
channelConversionSamples, numChannelCoversionSamples, NULL,
destinationSamples, numDestinationSamples, NULL);
delete[] channelConversionSamples;
} else {
resampleError = soxr_process(resampler,
sourceSamples, numSourceSamples, NULL,
destinationSamples, numDestinationSamples, NULL);
}
return resampleError;
}
return resampleError;
} else {
return 0;
}
}
@ -375,7 +384,7 @@ soxr_t soxrResamplerFromInputFormatToOutputFormat(const QAudioFormat& sourceAudi
soxr_t newResampler = soxr_create(sourceAudioFormat.sampleRate(),
destinationAudioFormat.sampleRate(),
destinationAudioFormat.channelCount(),
1,
&soxrError, &inputToNetworkSpec, &qualitySpec, 0);
if (soxrError) {
@ -585,15 +594,11 @@ 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();
}
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.";
@ -604,6 +609,10 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
}
}
static QByteArray loopBackByteArray;
loopBackByteArray.resize(numDestinationSamplesRequired(_inputFormat, _outputFormat,
inputByteArray.size() / sizeof(int16_t)) * sizeof(int16_t));
possibleResampling(_loopbackResampler,
reinterpret_cast<int16_t*>(inputByteArray.data()),
reinterpret_cast<int16_t*>(loopBackByteArray.data()),
@ -963,7 +972,7 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
if (!inputDeviceInfo.isNull()) {
qDebug() << "The audio input device " << inputDeviceInfo.deviceName() << "is available.";
_inputAudioDeviceName = inputDeviceInfo.deviceName().trimmed();
if (adjustedFormatForAudioDevice(inputDeviceInfo, _desiredInputFormat, _inputFormat)) {
qDebug() << "The format to be used for audio input is" << _inputFormat;
@ -1133,7 +1142,7 @@ const float AudioClient::CALLBACK_ACCELERATOR_RATIO = 2.0f;
int AudioClient::calculateNumberOfInputCallbackBytes(const QAudioFormat& format) const {
int numInputCallbackBytes = (int)(((AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL
* format.channelCount()
* (format.sampleRate() / AudioConstants::SAMPLE_RATE))
* ((float) format.sampleRate() / AudioConstants::SAMPLE_RATE))
/ CALLBACK_ACCELERATOR_RATIO) + 0.5f);
return numInputCallbackBytes;