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use libsoxr for input to network resampling

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This commit is contained in:
Stephen Birarda 2015-01-22 16:04:57 -08:00
parent 102f076412
commit 0440576dc2
2 changed files with 103 additions and 13 deletions
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112
libraries/audio-client/src/AudioClient.cpp
View file

@ -34,6 +34,8 @@
#include <glm/glm.hpp>
#include <soxr.h>
#include <NodeList.h>
#include <PacketHeaders.h>
@ -82,6 +84,7 @@ AudioClient::AudioClient() :
_reverbOptions(&_scriptReverbOptions),
_gverbLocal(NULL),
_gverb(NULL),
_inputToNetworkResampler(NULL),
_noiseSourceEnabled(false),
_toneSourceEnabled(true),
_outgoingAvatarAudioSequenceNumber(0),
@ -624,9 +627,8 @@ void AudioClient::handleAudioInput() {
_stats.updateInputMsecsRead(audioInputMsecsRead);
while (_inputRingBuffer.samplesAvailable() >= inputSamplesRequired) {
int16_t* inputAudioSamples = new int16_t[inputSamplesRequired];
_inputRingBuffer.readSamples(inputAudioSamples, inputSamplesRequired);
int16_t* inputAudioSamples = NULL;
const int numNetworkBytes = _isStereoInput
? AudioConstants::NETWORK_FRAME_BYTES_STEREO
@ -635,20 +637,55 @@ void AudioClient::handleAudioInput() {
? AudioConstants::NETWORK_FRAME_SAMPLES_STEREO
: AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL;
// zero out the monoAudioSamples array and the locally injected audio
memset(networkAudioSamples, 0, numNetworkBytes);
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;
}
// we aren't muted, downsample the input audio
linearResampling((int16_t*) inputAudioSamples, networkAudioSamples,
inputSamplesRequired, numNetworkSamples,
_inputFormat, _desiredInputFormat);
// we might need to handle the special case where input has 2 channels, whereas our desired network format,
// unless we specifically want stereo input, uses 1
// we need to handle that before the resampler since it won't do it for us
if (_inputFormat.channelCount() == 2 && _desiredInputFormat.channelCount() == 1) {
inputAudioSamples = new int16_t[inputSamplesRequired / 2];
int16_t* stereoInputAudioSamples = new int16_t[inputSamplesRequired];
_inputRingBuffer.readSamples(stereoInputAudioSamples, inputSamplesRequired);
// loop through the stereo input audio samples and take every second sample
for (int i = 0; i < inputSamplesRequired; i += 2) {
inputAudioSamples[i / 2] = stereoInputAudioSamples[i];
}
qDebug() << "changed" << inputSamplesRequired << "into" << inputSamplesRequired / 2;
} else {
inputAudioSamples = new int16_t[inputSamplesRequired];
_inputRingBuffer.readSamples(inputAudioSamples, inputSamplesRequired);
}
// we aren't muted - do we need to use a resampler?
if (_inputToNetworkResampler) {
size_t idone, odone;
soxr_error_t resampleError = soxr_process(_inputToNetworkResampler,
inputAudioSamples, inputSamplesRequired, &idone,
networkAudioSamples, numNetworkSamples, &odone);
if (resampleError) {
qDebug() << "Error resampling from input format to network format - soxr error code" << resampleError;
qDebug() << "Sending a silent audio frame.";
memset(networkAudioSamples, 0, numNetworkSamples);
} else {
qDebug() << "Resampled" << idone << "to" << odone;
}
} else {
// we can copy the samples directly across
memcpy(networkAudioSamples, inputAudioSamples, numNetworkBytes);
}
// only impose the noise gate and perform tone injection if we are sending mono audio
if (!_isStereoInput && !_audioSourceInjectEnabled && _isNoiseGateEnabled) {
@ -674,15 +711,15 @@ void AudioClient::handleAudioInput() {
_lastInputLoudness = fabs(loudness / numNetworkSamples);
}
emit inputReceived(QByteArray(reinterpret_cast<const char*>(networkAudioSamples),
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL));
} else {
// our input loudness is 0, since we're muted
_lastInputLoudness = 0;
_timeSinceLastClip = 0.0f;
}
emit inputReceived(QByteArray(reinterpret_cast<const char*>(networkAudioSamples),
AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL));
auto nodeList = DependencyManager::get<NodeList>();
SharedNodePointer audioMixer = nodeList->soloNodeOfType(NodeType::AudioMixer);
@ -747,6 +784,7 @@ void AudioClient::handleAudioInput() {
emit outputBytesToNetwork(packetBytes);
}
delete[] inputAudioSamples;
}
}
@ -894,6 +932,18 @@ void AudioClient::outputFormatChanged() {
_receivedAudioStream.outputFormatChanged(outputFormatChannelCountTimesSampleRate);
}
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;
}
}
}
bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceInfo) {
bool supportedFormat = false;
@ -911,6 +961,12 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
_inputAudioDeviceName = "";
}
if (_inputToNetworkResampler) {
// if we were using an input to network resampler, delete it here
soxr_delete(_inputToNetworkResampler);
_inputToNetworkResampler = NULL;
}
if (!inputDeviceInfo.isNull()) {
qDebug() << "The audio input device " << inputDeviceInfo.deviceName() << "is available.";
@ -919,6 +975,36 @@ bool AudioClient::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceIn
if (adjustedFormatForAudioDevice(inputDeviceInfo, _desiredInputFormat, _inputFormat)) {
qDebug() << "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()) {
soxr_error_t soxrError;
// setup soxr_io_spec_t for input and output
soxr_io_spec_t inputToNetworkSpec = soxr_io_spec(soxrDataTypeFromQAudioFormat(_inputFormat),
soxrDataTypeFromQAudioFormat(_desiredInputFormat));
// setup soxr_quality_spec_t for quality options
soxr_quality_spec_t qualitySpec = soxr_quality_spec(SOXR_MQ, 0);
_inputToNetworkResampler = soxr_create(_inputFormat.sampleRate(),
_desiredInputFormat.sampleRate(),
_isStereoInput ? 2 : 1,
&soxrError,
&inputToNetworkSpec,
&qualitySpec,
0);
if (soxrError) {
qDebug() << "There was an error setting up the soxr resampler for input format to network format -"
<< "soxr error code - " << soxrError;
return false;
}
} else {
qDebug() << "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);

4
libraries/audio-client/src/AudioClient.h
View file

@ -65,6 +65,7 @@ static const quint64 DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_PERIOD = 10 * 1000; /
class QAudioInput;
class QAudioOutput;
class QIODevice;
struct soxr;
typedef glm::vec3 (*AudioPositionGetter)();
typedef glm::quat (*AudioOrientationGetter)();
@ -231,6 +232,9 @@ private:
AudioEffectOptions* _reverbOptions;
ty_gverb* _gverbLocal;
ty_gverb* _gverb;
// possible soxr streams needed for resample
soxr* _inputToNetworkResampler;
// Adds Reverb
void initGverb();