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add AudioConstants::SAMPLE_SIZE

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This commit is contained in:
Zach Pomerantz 2016-09-16 14:20:50 -07:00
parent bbbe070af1
commit b6d093889c
2 changed files with 14 additions and 13 deletions
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22
libraries/audio-client/src/AudioClient.cpp
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@ -441,7 +441,7 @@ void possibleResampling(AudioSRC* resampler,
if (!sampleChannelConversion(sourceSamples, destinationSamples, numSourceSamples, if (!sampleChannelConversion(sourceSamples, destinationSamples, numSourceSamples,
sourceAudioFormat, destinationAudioFormat)) { sourceAudioFormat, destinationAudioFormat)) {
// no conversion, we can copy the samples directly across // no conversion, we can copy the samples directly across
memcpy(destinationSamples, sourceSamples, numSourceSamples * sizeof(int16_t)); memcpy(destinationSamples, sourceSamples, numSourceSamples * AudioConstants::SAMPLE_SIZE);
} }
} else { } else {
@ -815,10 +815,10 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
static QByteArray loopBackByteArray; static QByteArray loopBackByteArray;
int numInputSamples = inputByteArray.size() / sizeof(int16_t); int numInputSamples = inputByteArray.size() / AudioConstants::SAMPLE_SIZE;
int numLoopbackSamples = numDestinationSamplesRequired(_inputFormat, _outputFormat, numInputSamples); int numLoopbackSamples = numDestinationSamplesRequired(_inputFormat, _outputFormat, numInputSamples);
loopBackByteArray.resize(numLoopbackSamples * sizeof(int16_t)); loopBackByteArray.resize(numLoopbackSamples * AudioConstants::SAMPLE_SIZE);
int16_t* inputSamples = reinterpret_cast<int16_t*>(inputByteArray.data()); int16_t* inputSamples = reinterpret_cast<int16_t*>(inputByteArray.data());
int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data()); int16_t* loopbackSamples = reinterpret_cast<int16_t*>(loopBackByteArray.data());
@ -826,7 +826,7 @@ void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
// upmix mono to stereo // upmix mono to stereo
if (!sampleChannelConversion(inputSamples, loopbackSamples, numInputSamples, _inputFormat, _outputFormat)) { if (!sampleChannelConversion(inputSamples, loopbackSamples, numInputSamples, _inputFormat, _outputFormat)) {
// no conversion, just copy the samples // no conversion, just copy the samples
memcpy(loopbackSamples, inputSamples, numInputSamples * sizeof(int16_t)); memcpy(loopbackSamples, inputSamples, numInputSamples * AudioConstants::SAMPLE_SIZE);
} }
// apply stereo reverb at the source, to the loopback audio // apply stereo reverb at the source, to the loopback audio
@ -942,7 +942,7 @@ void AudioClient::handleAudioInput() {
emitAudioPacket(encodedBuffer.constData(), encodedBuffer.size(), _outgoingAvatarAudioSequenceNumber, audioTransform, packetType, _selectedCodecName); emitAudioPacket(encodedBuffer.constData(), encodedBuffer.size(), _outgoingAvatarAudioSequenceNumber, audioTransform, packetType, _selectedCodecName);
_stats.sentPacket(); _stats.sentPacket();
int bytesInInputRingBuffer = _inputRingBuffer.samplesAvailable() * sizeof(int16_t); int bytesInInputRingBuffer = _inputRingBuffer.samplesAvailable() * AudioConstants::SAMPLE_SIZE;
float msecsInInputRingBuffer = bytesInInputRingBuffer / (float)(_inputFormat.bytesForDuration(USECS_PER_MSEC)); float msecsInInputRingBuffer = bytesInInputRingBuffer / (float)(_inputFormat.bytesForDuration(USECS_PER_MSEC));
_stats.updateInputMsUnplayed(msecsInInputRingBuffer); _stats.updateInputMsUnplayed(msecsInInputRingBuffer);
} }
@ -1025,7 +1025,7 @@ void AudioClient::processReceivedSamples(const QByteArray& decodedBuffer, QByteA
const int16_t* decodedSamples = reinterpret_cast<const int16_t*>(decodedBuffer.data()); const int16_t* decodedSamples = reinterpret_cast<const int16_t*>(decodedBuffer.data());
assert(decodedBuffer.size() == AudioConstants::NETWORK_FRAME_BYTES_STEREO); assert(decodedBuffer.size() == AudioConstants::NETWORK_FRAME_BYTES_STEREO);
outputBuffer.resize(_outputFrameSize * sizeof(int16_t)); outputBuffer.resize(_outputFrameSize * AudioConstants::SAMPLE_SIZE);
int16_t* outputSamples = reinterpret_cast<int16_t*>(outputBuffer.data()); int16_t* outputSamples = reinterpret_cast<int16_t*>(outputBuffer.data());
// convert network audio to float // convert network audio to float
@ -1283,7 +1283,7 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
// setup our general output device for audio-mixer audio // setup our general output device for audio-mixer audio
_audioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this); _audioOutput = new QAudioOutput(outputDeviceInfo, _outputFormat, this);
int osDefaultBufferSize = _audioOutput->bufferSize(); int osDefaultBufferSize = _audioOutput->bufferSize();
int requestedSize = _sessionOutputBufferSizeFrames *_outputFrameSize * sizeof(int16_t); int requestedSize = _sessionOutputBufferSizeFrames *_outputFrameSize * AudioConstants::SAMPLE_SIZE;
_audioOutput->setBufferSize(requestedSize); _audioOutput->setBufferSize(requestedSize);
connect(_audioOutput, &QAudioOutput::notify, this, &AudioClient::outputNotify); connect(_audioOutput, &QAudioOutput::notify, this, &AudioClient::outputNotify);
@ -1295,7 +1295,7 @@ bool AudioClient::switchOutputToAudioDevice(const QAudioDeviceInfo& outputDevice
_audioOutput->start(&_audioOutputIODevice); _audioOutput->start(&_audioOutputIODevice);
lock.unlock(); lock.unlock();
qCDebug(audioclient) << "Output Buffer capacity in frames: " << _audioOutput->bufferSize() / sizeof(int16_t) / (float)_outputFrameSize << qCDebug(audioclient) << "Output Buffer capacity in frames: " << _audioOutput->bufferSize() / AudioConstants::SAMPLE_SIZE / (float)_outputFrameSize <<
"requested bytes:" << requestedSize << "actual bytes:" << _audioOutput->bufferSize() << "requested bytes:" << requestedSize << "actual bytes:" << _audioOutput->bufferSize() <<
"os default:" << osDefaultBufferSize << "period size:" << _audioOutput->periodSize(); "os default:" << osDefaultBufferSize << "period size:" << _audioOutput->periodSize();
@ -1357,7 +1357,7 @@ int AudioClient::calculateNumberOfInputCallbackBytes(const QAudioFormat& format)
} }
int AudioClient::calculateNumberOfFrameSamples(int numBytes) const { int AudioClient::calculateNumberOfFrameSamples(int numBytes) const {
int frameSamples = (int)(numBytes * CALLBACK_ACCELERATOR_RATIO + 0.5f) / sizeof(int16_t); int frameSamples = (int)(numBytes * CALLBACK_ACCELERATOR_RATIO + 0.5f) / AudioConstants::SAMPLE_SIZE;
return frameSamples; return frameSamples;
} }
@ -1401,7 +1401,7 @@ float AudioClient::gainForSource(float distance, float volume) {
} }
qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) { qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
auto samplesRequested = maxSize / sizeof(int16_t); auto samplesRequested = maxSize / AudioConstants::SAMPLE_SIZE;
int samplesPopped; int samplesPopped;
int bytesWritten; int bytesWritten;
@ -1409,7 +1409,7 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
qCDebug(audiostream, "Read %d samples from buffer (%d available)", samplesPopped, _receivedAudioStream.getSamplesAvailable()); qCDebug(audiostream, "Read %d samples from buffer (%d available)", samplesPopped, _receivedAudioStream.getSamplesAvailable());
AudioRingBuffer::ConstIterator lastPopOutput = _receivedAudioStream.getLastPopOutput(); AudioRingBuffer::ConstIterator lastPopOutput = _receivedAudioStream.getLastPopOutput();
lastPopOutput.readSamples((int16_t*)data, samplesPopped); lastPopOutput.readSamples((int16_t*)data, samplesPopped);
bytesWritten = samplesPopped * sizeof(int16_t); bytesWritten = samplesPopped * AudioConstants::SAMPLE_SIZE;
} else { } else {
// nothing on network, don't grab anything from injectors, and just return 0s // nothing on network, don't grab anything from injectors, and just return 0s
// this will flood the log: qCDebug(audioclient, "empty/partial network buffer"); // this will flood the log: qCDebug(audioclient, "empty/partial network buffer");

5
libraries/audio/src/AudioConstants.h
View file

@ -23,15 +23,16 @@ namespace AudioConstants {
typedef int16_t AudioSample; typedef int16_t AudioSample;
const int SAMPLE_SIZE = sizeof(AudioSample);
inline const char* getAudioFrameName() { return "com.highfidelity.recording.Audio"; } inline const char* getAudioFrameName() { return "com.highfidelity.recording.Audio"; }
const int MAX_CODEC_NAME_LENGTH = 30; const int MAX_CODEC_NAME_LENGTH = 30;
const int MAX_CODEC_NAME_LENGTH_ON_WIRE = MAX_CODEC_NAME_LENGTH + sizeof(uint32_t); const int MAX_CODEC_NAME_LENGTH_ON_WIRE = MAX_CODEC_NAME_LENGTH + sizeof(uint32_t);
const int NETWORK_FRAME_BYTES_STEREO = 960; const int NETWORK_FRAME_BYTES_STEREO = 960;
const int NETWORK_FRAME_SAMPLES_STEREO = NETWORK_FRAME_BYTES_STEREO / sizeof(AudioSample); const int NETWORK_FRAME_SAMPLES_STEREO = NETWORK_FRAME_BYTES_STEREO / SAMPLE_SIZE;
const int NETWORK_FRAME_BYTES_PER_CHANNEL = NETWORK_FRAME_BYTES_STEREO / 2; const int NETWORK_FRAME_BYTES_PER_CHANNEL = NETWORK_FRAME_BYTES_STEREO / 2;
const int NETWORK_FRAME_SAMPLES_PER_CHANNEL = NETWORK_FRAME_BYTES_PER_CHANNEL / sizeof(AudioSample); const int NETWORK_FRAME_SAMPLES_PER_CHANNEL = NETWORK_FRAME_BYTES_PER_CHANNEL / SAMPLE_SIZE;
const float NETWORK_FRAME_SECS = (AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL / float(AudioConstants::SAMPLE_RATE)); const float NETWORK_FRAME_SECS = (AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL / float(AudioConstants::SAMPLE_RATE));
const float NETWORK_FRAME_MSECS = NETWORK_FRAME_SECS * 1000.0f; const float NETWORK_FRAME_MSECS = NETWORK_FRAME_SECS * 1000.0f;
const float NETWORK_FRAMES_PER_SEC = 1.0f / NETWORK_FRAME_SECS; const float NETWORK_FRAMES_PER_SEC = 1.0f / NETWORK_FRAME_SECS;