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In AudioClient, replace all SOXR with AudioSRC. Used for input-to-network, network-to-output, and loopback resampling.

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This commit is contained in:
Ken Cooke 2015-09-22 23:10:03 -07:00
parent 4b9c2ed6b6
commit c4bc5233b1
2 changed files with 39 additions and 104 deletions
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132
libraries/audio-client/src/AudioClient.cpp
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@ -47,8 +47,6 @@ extern "C" {
#pragma GCC diagnostic pop
#endif
#include <soxr.h>
#include <NodeList.h>
#include <udt/PacketHeaders.h>
#include <PositionalAudioStream.h>
@ -189,18 +187,6 @@ QAudioDeviceInfo getNamedAudioDeviceForMode(QAudio::Mode mode, const QString& de
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();
@ -350,7 +336,7 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
// use 22050, resample but closer to 24
adjustedAudioFormat.setSampleRate(HALF_FORTY_FOUR);
} else if (audioDevice.supportedSampleRates().contains(FORTY_FOUR)) {
// use 48000, libsoxr will resample
// use 48000, resample
adjustedAudioFormat.setSampleRate(FORTY_FOUR);
}
#endif
@ -391,10 +377,10 @@ bool sampleChannelConversion(const int16_t* sourceSamples, int16_t* destinationS
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) {
void possibleResampling(AudioSRC* 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) {
@ -403,32 +389,19 @@ soxr_error_t possibleResampling(soxr_t resampler,
// 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();
float channelCountRatio = (float)destinationAudioFormat.channelCount() / sourceAudioFormat.channelCount();
int numChannelCoversionSamples = (int) (numSourceSamples * channelCountRatio);
int numChannelCoversionSamples = (int)(numSourceSamples * channelCountRatio);
int16_t* channelConversionSamples = new int16_t[numChannelCoversionSamples];
sampleChannelConversion(sourceSamples, channelConversionSamples,
numSourceSamples,
sourceAudioFormat, destinationAudioFormat);
size_t numDestinationSamplesActual = 0;
resampleError = soxr_process(resampler,
channelConversionSamples, numChannelCoversionSamples, NULL,
destinationSamples, numDestinationSamples, &numDestinationSamplesActual);
// return silence instead of playing garbage samples
if (numDestinationSamplesActual < numDestinationSamples) {
unsigned int nBytes = (numDestinationSamples - numDestinationSamplesActual) * destinationAudioFormat.channelCount() * sizeof(int16_t);
memset(&destinationSamples[numDestinationSamplesActual * destinationAudioFormat.channelCount()], 0, nBytes);
qCDebug(audioclient) << "SOXR: padded with" << nBytes << "bytes of silence";
}
resampler->render(channelConversionSamples, destinationSamples, numChannelCoversionSamples);
delete[] channelConversionSamples;
} else {
@ -441,56 +414,12 @@ soxr_error_t possibleResampling(soxr_t resampler,
numAdjustedDestinationSamples /= 2;
}
size_t numAdjustedDestinationSamplesActual = 0;
resampleError = soxr_process(resampler,
sourceSamples, numAdjustedSourceSamples, NULL,
destinationSamples, numAdjustedDestinationSamples, &numAdjustedDestinationSamplesActual);
// return silence instead of playing garbage samples
if (numAdjustedDestinationSamplesActual < numAdjustedDestinationSamples) {
unsigned int nBytes = (numAdjustedDestinationSamples - numAdjustedDestinationSamplesActual) * destinationAudioFormat.channelCount() * sizeof(int16_t);
memset(&destinationSamples[numAdjustedDestinationSamplesActual * destinationAudioFormat.channelCount()], 0, nBytes);
qCDebug(audioclient) << "SOXR: padded with" << nBytes << "bytes of silence";
}
resampler->render(sourceSamples, destinationSamples, numAdjustedSourceSamples);
}
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
@ -546,7 +475,7 @@ void AudioClient::stop() {
switchOutputToAudioDevice(QAudioDeviceInfo());
if (_loopbackResampler) {
soxr_delete(_loopbackResampler);
delete _loopbackResampler;
_loopbackResampler = NULL;
}
}
@ -767,11 +696,12 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
// 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;
}
assert(_inputFormat.sampleSize() == 16);
assert(_outputFormat.sampleSize() == 16);
int channelCount = (_inputFormat.channelCount() == 2 && _outputFormat.channelCount() == 2) ? 2 : 1;
_loopbackResampler = new AudioSRC(_inputFormat.sampleRate(), _outputFormat.sampleRate(), channelCount);
}
static QByteArray reverbAlone; // Intermediary for local reverb with no echo
@ -1099,7 +1029,7 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
if (_inputToNetworkResampler) {
// if we were using an input to network resampler, delete it here
soxr_delete(_inputToNetworkResampler);
delete _inputToNetworkResampler;
_inputToNetworkResampler = NULL;
}
@ -1111,15 +1041,17 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
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 required, setup a 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);
qCDebug(audioclient) << "Attemping to create a resampler for input format to network format.";
assert(_inputFormat.sampleSize() == 16);
assert(_desiredInputFormat.sampleSize() == 16);
int channelCount = (_inputFormat.channelCount() == 2 && _desiredInputFormat.channelCount() == 2) ? 2 : 1;
_inputToNetworkResampler = new AudioSRC(_inputFormat.sampleRate(), _desiredInputFormat.sampleRate(), channelCount);
if (!_inputToNetworkResampler) {
return false;
}
} else {
qCDebug(audioclient) << "No resampling required for audio input to match desired network format.";
}
@ -1194,13 +1126,13 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
if (_networkToOutputResampler) {
// if we were using an input to network resampler, delete it here
soxr_delete(_networkToOutputResampler);
delete _networkToOutputResampler;
_networkToOutputResampler = NULL;
}
if (_loopbackResampler) {
// if we were using an input to output resample, delete it here
soxr_delete(_loopbackResampler);
delete _loopbackResampler;
_loopbackResampler = NULL;
}
@ -1212,15 +1144,17 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
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 required, setup a 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;
}
assert(_desiredOutputFormat.sampleSize() == 16);
assert(_outputFormat.sampleSize() == 16);
int channelCount = (_desiredOutputFormat.channelCount() == 2 && _outputFormat.channelCount() == 2) ? 2 : 1;
_networkToOutputResampler = new AudioSRC(_desiredOutputFormat.sampleRate(), _outputFormat.sampleRate(), channelCount);
} else {
qCDebug(audioclient) << "No resampling required for network output to match actual output format.";
}

11
libraries/audio-client/src/AudioClient.h
View file

@ -45,6 +45,7 @@
#include "AudioIOStats.h"
#include "AudioNoiseGate.h"
#include "AudioSRC.h"
#ifdef _WIN32
#pragma warning( push )
@ -72,7 +73,7 @@ static const quint64 DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_PERIOD = 10 * 1000; /
class QAudioInput;
class QAudioOutput;
class QIODevice;
struct soxr;
typedef struct ty_gverb ty_gverb;
typedef glm::vec3 (*AudioPositionGetter)();
@ -262,10 +263,10 @@ private:
AudioEffectOptions* _reverbOptions;
ty_gverb* _gverb;
// possible soxr streams needed for resample
soxr* _inputToNetworkResampler;
soxr* _networkToOutputResampler;
soxr* _loopbackResampler;
// possible streams needed for resample
AudioSRC* _inputToNetworkResampler;
AudioSRC* _networkToOutputResampler;
AudioSRC* _loopbackResampler;
// Adds Reverb
ty_gverb* createGverbFilter();