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copy all passed samples to AudioRingBuffer

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This commit is contained in:
Stephen Birarda 2013-12-12 10:15:43 -08:00
parent e27aa11216
commit b06af3cd6c
2 changed files with 125 additions and 60 deletions
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141
interface/src/Audio.cpp
View file

@ -167,22 +167,65 @@ bool adjustedFormatForAudioDevice(const QAudioDeviceInfo& audioDevice,
}
}
void nearestNeighborResampling(int16_t* sourceSamples, int16_t* destinationSamples,
int numDestinationSamples,
const QAudioFormat& sourceAudioFormat, const QAudioFormat& destinationAudioFormat) {
void linearResampling(int16_t* sourceSamples, int16_t* destinationSamples,
unsigned int numSourceSamples, unsigned int numDestinationSamples,
const QAudioFormat& sourceAudioFormat, const QAudioFormat& destinationAudioFormat) {
if (sourceAudioFormat == destinationAudioFormat) {
memcpy(destinationSamples, sourceSamples, numDestinationSamples * sizeof(int16_t));
memcpy(destinationSamples, sourceSamples, numSourceSamples * sizeof(int16_t));
} else {
float sourceMultiplier = sourceAudioFormat.sampleRate() / (float) destinationAudioFormat.sampleRate();
float sourceToDestinationFactor = numSourceSamples / numDestinationSamples;
// take into account the number of channels in source and destination
// accomodate for the case where have an output with > 2 channels
// this is the case with our HDMI capture
sourceMultiplier *= sourceAudioFormat.channelCount()
/ (float) (destinationAudioFormat.channelCount() > 2 ? 2 : destinationAudioFormat.channelCount());
for (int i = 0; i < numDestinationSamples; i++) {
destinationSamples[i] = sourceSamples[static_cast<int>(i * sourceMultiplier)];
if (sourceToDestinationFactor >= 2) {
// we need to downsample from 48 to 24
// for now this only supports a mono output - this would be the case for audio input
for (int i = 2; i < numSourceSamples; i += 4) {
if (i + 2 >= numSourceSamples) {
destinationSamples[(i - 2) / 4] = (sourceSamples[i - 2] / 2)
+ (sourceSamples[i] / 2);
} else {
destinationSamples[(i - 2) / 4] = (sourceSamples[i - 2] / 4)
+ (sourceSamples[i] / 2)
+ (sourceSamples[i + 2] / 4);
}
}
} else {
int numResultingDestinationSamples = numSourceSamples
* (destinationAudioFormat.sampleRate() / sourceAudioFormat.sampleRate())
* (destinationAudioFormat.channelCount() / sourceAudioFormat.channelCount());
// upsample from 24 to 48
for (int i = 0; i < numResultingDestinationSamples; i += destinationAudioFormat.channelCount()) {
destinationSamples[i] = sourceSamples[i / 4];
if (sourceAudioFormat.channelCount() == 1) {
destinationSamples[i + 1] = sourceSamples[i / 4];
} else {
destinationSamples[i + 1] = sourceSamples[(i / 4) + 1];
if (destinationAudioFormat.channelCount() > 2) {
// fill the rest of the channels with silence
for (int j = 2; j < destinationAudioFormat.channelCount(); j++) {
destinationSamples[i] = 0;
}
}
}
if (numResultingDestinationSamples < numDestinationSamples
&& i + destinationAudioFormat.channelCount() >= numResultingDestinationSamples) {
// make sure we don't leave a gap on the number of destination samples
for (int k = numResultingDestinationSamples; k < numDestinationSamples; k++) {
destinationSamples[k] = destinationSamples[k - destinationAudioFormat.channelCount()];
}
}
}
}
}
}
@ -249,19 +292,24 @@ void Audio::handleAudioInput() {
static int leadingBytes = numBytesPacketHeader + sizeof(glm::vec3) + sizeof(glm::quat) + NUM_BYTES_RFC4122_UUID;
static int16_t* monoAudioSamples = (int16_t*) (monoAudioDataPacket + leadingBytes);
QByteArray inputByteArray = _inputDevice->readAll();
static int inputToOutputRatio = _numOutputCallbackBytes / _numInputCallbackBytes;
static int inputToNetworkInputRatio = _numInputCallbackBytes * CALLBACK_ACCELERATOR_RATIO / BUFFER_LENGTH_BYTES_PER_CHANNEL;
int numResampledBytes = inputByteArray.size() / (_numInputCallbackBytes / BUFFER_LENGTH_BYTES_PER_CHANNEL);
QByteArray inputByteArray = _inputDevice->readAll();
int numResampledNetworkBytes = inputByteArray.size() / inputToNetworkInputRatio;
int numResampledNetworkSamples = numResampledNetworkBytes / sizeof(int16_t);
// zero out the monoAudioSamples array
memset(monoAudioSamples, 0, numResampledBytes);
memset(monoAudioSamples, 0, numResampledNetworkBytes);
if (Menu::getInstance()->isOptionChecked(MenuOption::EchoLocalAudio) && !_muted) {
_outputBuffer.resize(inputByteArray.size());
// if local loopback enabled, copy input to output
nearestNeighborResampling((int16_t*) inputByteArray.data(), (int16_t*) _outputBuffer.data(),
inputByteArray.size() / sizeof(int16_t),
_inputFormat, _outputFormat);
linearResampling((int16_t*) inputByteArray.data(), (int16_t*) _outputBuffer.data(),
inputByteArray.size() / sizeof(int16_t),
inputByteArray.size() * inputToOutputRatio / sizeof(int16_t),
_inputFormat, _outputFormat);
} else {
_outputBuffer.fill(0, inputByteArray.size());
}
@ -279,7 +327,7 @@ void Audio::handleAudioInput() {
NodeList* nodeList = NodeList::getInstance();
Node* audioMixer = nodeList->soloNodeOfType(NODE_TYPE_AUDIO_MIXER);
if (audioMixer) {
if (false) {
if (audioMixer->getActiveSocket()) {
MyAvatar* interfaceAvatar = Application::getInstance()->getAvatar();
@ -314,30 +362,45 @@ void Audio::handleAudioInput() {
_lastInputLoudness = loudness;
// we aren't muted - pull our input audio to send off to the mixer
nearestNeighborResampling((int16_t*) inputByteArray.data(),
monoAudioSamples, numResampledBytes / sizeof(int16_t),
_inputFormat, _desiredInputFormat);
linearResampling((int16_t*) inputByteArray.data(),
monoAudioSamples,
inputByteArray.size() / sizeof(int16_t),
numResampledNetworkSamples,
_inputFormat, _desiredInputFormat);
} else {
_lastInputLoudness = 0;
}
nodeList->getNodeSocket().writeDatagram(monoAudioDataPacket,
numResampledBytes + leadingBytes,
numResampledNetworkBytes + leadingBytes,
audioMixer->getActiveSocket()->getAddress(),
audioMixer->getActiveSocket()->getPort());
Application::getInstance()->getBandwidthMeter()->outputStream(BandwidthMeter::AUDIO)
.updateValue(BUFFER_LENGTH_BYTES_PER_CHANNEL + leadingBytes);
.updateValue(BUFFER_LENGTH_BYTES_PER_CHANNEL + leadingBytes);
} else {
nodeList->pingPublicAndLocalSocketsForInactiveNode(audioMixer);
}
}
// we aren't muted - pull our input audio to send off to the mixer
linearResampling((int16_t*) inputByteArray.data(),
monoAudioSamples,
inputByteArray.size() / sizeof(int16_t),
numResampledNetworkSamples,
_inputFormat, _desiredInputFormat);
linearResampling(monoAudioSamples,
(int16_t*) _outputBuffer.data(),
numResampledNetworkSamples,
inputByteArray.size() * inputToOutputRatio / sizeof(int16_t),
_desiredInputFormat, _outputFormat);
if (_outputDevice) {
// if there is anything in the ring buffer, decide what to do
if (_ringBuffer.getEndOfLastWrite()) {
if (false) {
if (_ringBuffer.isStarved() && _ringBuffer.diffLastWriteNextOutput() <
((_outputBuffer.size() / sizeof(int16_t)) + _jitterBufferSamples * (_ringBuffer.isStereo() ? 2 : 1))) {
// If not enough audio has arrived to start playback, keep waiting
@ -357,12 +420,16 @@ void Audio::handleAudioInput() {
}
int numOutputBufferSamples = _outputBuffer.size() / sizeof(int16_t);
if (_ringBuffer.getNextOutput() + numOutputBufferSamples > _ringBuffer.getBuffer() + RING_BUFFER_LENGTH_SAMPLES) {
qDebug() << _ringBuffer.getNextOutput() - _ringBuffer.getBuffer() << "\n";
numOutputBufferSamples = (_ringBuffer.getBuffer() + RING_BUFFER_LENGTH_SAMPLES) - _ringBuffer.getNextOutput();
}
// copy the packet from the RB to the output
nearestNeighborResampling(_ringBuffer.getNextOutput(),
(int16_t*) _outputBuffer.data(),
numOutputBufferSamples,
_desiredOutputFormat, _outputFormat);
// nearestNeighborResampling(_ringBuffer.getNextOutput(),
// (int16_t*) _outputBuffer.data(),
// numOutputBufferSamples,
// _desiredOutputFormat, _outputFormat);
_ringBuffer.setNextOutput(_ringBuffer.getNextOutput() + numOutputBufferSamples);
@ -409,17 +476,17 @@ void Audio::addReceivedAudioToBuffer(const QByteArray& audioByteArray) {
}
}
if (_ringBuffer.diffLastWriteNextOutput() + PACKET_LENGTH_SAMPLES >
PACKET_LENGTH_SAMPLES + (ceilf((float) (_jitterBufferSamples * 2) / PACKET_LENGTH_SAMPLES) * PACKET_LENGTH_SAMPLES)) {
// this packet would give us more than the required amount for play out
// discard the first packet in the buffer
_ringBuffer.setNextOutput(_ringBuffer.getNextOutput() + PACKET_LENGTH_SAMPLES);
if (_ringBuffer.getNextOutput() == _ringBuffer.getBuffer() + RING_BUFFER_LENGTH_SAMPLES) {
_ringBuffer.setNextOutput(_ringBuffer.getBuffer());
}
}
// if (_ringBuffer.diffLastWriteNextOutput() + PACKET_LENGTH_SAMPLES >
// PACKET_LENGTH_SAMPLES + (ceilf((float) (_jitterBufferSamples * 2) / PACKET_LENGTH_SAMPLES) * PACKET_LENGTH_SAMPLES)) {
// // this packet would give us more than the required amount for play out
// // discard the first packet in the buffer
//
// _ringBuffer.setNextOutput(_ringBuffer.getNextOutput() + PACKET_LENGTH_SAMPLES);
//
// if (_ringBuffer.getNextOutput() >= _ringBuffer.getBuffer() + RING_BUFFER_LENGTH_SAMPLES) {
// _ringBuffer.setNextOutput(_ringBuffer.getBuffer());
// }
// }
_ringBuffer.parseData((unsigned char*) audioByteArray.data(), audioByteArray.size());

44
libraries/audio/src/AudioRingBuffer.cpp
View file

@ -9,6 +9,8 @@
#include <cstring>
#include <math.h>
#include <QtCore/QDebug>
#include "PacketHeaders.h"
#include "AudioRingBuffer.h"
@ -43,30 +45,26 @@ int AudioRingBuffer::parseData(unsigned char* sourceBuffer, int numBytes) {
int AudioRingBuffer::parseAudioSamples(unsigned char* sourceBuffer, int numBytes) {
// make sure we have enough bytes left for this to be the right amount of audio
// otherwise we should not copy that data, and leave the buffer pointers where they are
int samplesToCopy = BUFFER_LENGTH_SAMPLES_PER_CHANNEL * (_isStereo ? 2 : 1);
if (numBytes == samplesToCopy * sizeof(int16_t)) {
if (!_endOfLastWrite) {
_endOfLastWrite = _buffer;
} else if (diffLastWriteNextOutput() > RING_BUFFER_LENGTH_SAMPLES - samplesToCopy) {
_endOfLastWrite = _buffer;
_nextOutput = _buffer;
_isStarved = true;
}
memcpy(_endOfLastWrite, sourceBuffer, numBytes);
_endOfLastWrite += samplesToCopy;
if (_endOfLastWrite >= _buffer + RING_BUFFER_LENGTH_SAMPLES) {
_endOfLastWrite = _buffer;
}
return numBytes;
} else {
return 0;
}
int samplesToCopy = numBytes / sizeof(int16_t);
if (!_endOfLastWrite) {
_endOfLastWrite = _buffer;
} else if (diffLastWriteNextOutput() > RING_BUFFER_LENGTH_SAMPLES - samplesToCopy) {
_endOfLastWrite = _buffer;
_nextOutput = _buffer;
_isStarved = true;
}
memcpy(_endOfLastWrite, sourceBuffer, numBytes);
_endOfLastWrite += samplesToCopy;
if (_endOfLastWrite >= _buffer + RING_BUFFER_LENGTH_SAMPLES) {
_endOfLastWrite = _buffer;
}
return numBytes;
}
int AudioRingBuffer::diffLastWriteNextOutput() const {