int InboundAudioStream::parseData(ReceivedMessage& message) {
// parse sequence number and track it
quint16 sequence;
message.readPrimitive(&sequence);
SequenceNumberStats::ArrivalInfo arrivalInfo = _incomingSequenceNumberStats.sequenceNumberReceived(sequence,
message.getSourceID());
QString codecInPacket = message.readString();
packetReceivedUpdateTimingStats();
int networkSamples;
// parse the info after the seq number and before the audio data (the stream properties)
int prePropertyPosition = message.getPosition();
int propertyBytes = parseStreamProperties(message.getType(), message.readWithoutCopy(message.getBytesLeftToRead()), networkSamples);
message.seek(prePropertyPosition + propertyBytes);
// handle this packet based on its arrival status.
switch (arrivalInfo._status) {
case SequenceNumberStats::Early: {
// Packet is early; write droppable silent samples for each of the skipped packets.
// NOTE: we assume that each dropped packet contains the same number of samples
// as the packet we just received.
int packetsDropped = arrivalInfo._seqDiffFromExpected;
writeSamplesForDroppedPackets(packetsDropped * networkSamples);
// fall through to OnTime case
}
case SequenceNumberStats::OnTime: {
// Packet is on time; parse its data to the ringbuffer
if (message.getType() == PacketType::SilentAudioFrame) {
// FIXME - Some codecs need to know about these silent frames... and can produce better output
writeDroppableSilentSamples(networkSamples);
} else {
// note: PCM and no codec are identical
bool selectedPCM = _selectedCodecName == "pcm" || _selectedCodecName == "";
bool packetPCM = codecInPacket == "pcm" || codecInPacket == "";
if (codecInPacket == _selectedCodecName || (packetPCM && selectedPCM)) {
auto afterProperties = message.readWithoutCopy(message.getBytesLeftToRead());
parseAudioData(message.getType(), afterProperties);
} else {
qDebug() << "Codec mismatch: expected" << _selectedCodecName << "got" << codecInPacket << "writing silence";
writeDroppableSilentSamples(networkSamples);
// inform others of the mismatch
auto sendingNode = DependencyManager::get<NodeList>()->nodeWithUUID(message.getSourceID());
emit mismatchedAudioCodec(sendingNode, _selectedCodecName, codecInPacket);
}
}
break;
}
default: {
// For now, late packets are ignored. It may be good in the future to insert the late audio packet data
// into the ring buffer to fill in the missing frame if it hasn't been mixed yet.
break;
}
}
int framesAvailable = _ringBuffer.framesAvailable();
// if this stream was starved, check if we're still starved.
if (_isStarved && framesAvailable >= _desiredJitterBufferFrames) {
_isStarved = false;
}
// if the ringbuffer exceeds the desired size by more than the threshold specified,
// drop the oldest frames so the ringbuffer is down to the desired size.
if (framesAvailable > _desiredJitterBufferFrames + _maxFramesOverDesired) {
int framesToDrop = framesAvailable - (_desiredJitterBufferFrames + DESIRED_JITTER_BUFFER_FRAMES_PADDING);
_ringBuffer.shiftReadPosition(framesToDrop * _ringBuffer.getNumFrameSamples());
_framesAvailableStat.reset();
_currentJitterBufferFrames = 0;
_oldFramesDropped += framesToDrop;
qCDebug(audiostream, "Dropped %d frames", framesToDrop);
qCDebug(audiostream, "Resetted current jitter frames");
}
framesAvailableChanged();
return message.getPosition();
}
int InboundAudioStream::parseData(ReceivedMessage& message) {
// parse sequence number and track it
quint16 sequence;
message.readPrimitive(&sequence);
SequenceNumberStats::ArrivalInfo arrivalInfo =
_incomingSequenceNumberStats.sequenceNumberReceived(sequence, message.getSourceID());
QString codecInPacket = message.readString();
packetReceivedUpdateTimingStats();
int networkFrames;
// parse the info after the seq number and before the audio data (the stream properties)
int prePropertyPosition = message.getPosition();
int propertyBytes = parseStreamProperties(message.getType(), message.readWithoutCopy(message.getBytesLeftToRead()), networkFrames);
message.seek(prePropertyPosition + propertyBytes);
// handle this packet based on its arrival status.
switch (arrivalInfo._status) {
case SequenceNumberStats::Unreasonable: {
lostAudioData(1);
break;
}
case SequenceNumberStats::Early: {
// Packet is early. Treat the packets as if all the packets between the last
// OnTime packet and this packet were lost. If we're using a codec this will
// also result in allowing the codec to interpolate lost data. Then
// fall through to the "on time" logic to actually handle this packet
int packetsDropped = arrivalInfo._seqDiffFromExpected;
lostAudioData(packetsDropped);
// fall through to OnTime case
}
case SequenceNumberStats::OnTime: {
// Packet is on time; parse its data to the ringbuffer
if (message.getType() == PacketType::SilentAudioFrame
|| message.getType() == PacketType::ReplicatedSilentAudioFrame) {
// If we recieved a SilentAudioFrame from our sender, we might want to drop
// some of the samples in order to catch up to our desired jitter buffer size.
writeDroppableSilentFrames(networkFrames);
} else {
// note: PCM and no codec are identical
bool selectedPCM = _selectedCodecName == "pcm" || _selectedCodecName == "";
bool packetPCM = codecInPacket == "pcm" || codecInPacket == "";
if (codecInPacket == _selectedCodecName || (packetPCM && selectedPCM)) {
auto afterProperties = message.readWithoutCopy(message.getBytesLeftToRead());
parseAudioData(message.getType(), afterProperties);
_mismatchedAudioCodecCount = 0;
} else {
_mismatchedAudioCodecCount++;
qDebug(audio) << "Codec mismatch: expected" << _selectedCodecName << "got" << codecInPacket;
if (packetPCM) {
// If there are PCM packets in-flight after the codec is changed, use them.
auto afterProperties = message.readWithoutCopy(message.getBytesLeftToRead());
_ringBuffer.writeData(afterProperties.data(), afterProperties.size());
} else {
// Since the data in the stream is using a codec that we aren't prepared for,
// we need to let the codec know that we don't have data for it, this will
// allow the codec to interpolate missing data and produce a fade to silence.
lostAudioData(1);
}
if (_mismatchedAudioCodecCount > MAX_MISMATCHED_AUDIO_CODEC_COUNT) {
_mismatchedAudioCodecCount = 0;
// inform others of the mismatch
auto sendingNode = DependencyManager::get<NodeList>()->nodeWithLocalID(message.getSourceID());
if (sendingNode) {
emit mismatchedAudioCodec(sendingNode, _selectedCodecName, codecInPacket);
qDebug(audio) << "Codec mismatch threshold exceeded, SelectedAudioFormat(" << _selectedCodecName << " ) sent";
}
}
}
}
break;
}
default: {
// For now, late packets are ignored. It may be good in the future to insert the late audio packet data
// into the ring buffer to fill in the missing frame if it hasn't been mixed yet.
break;
}
}
int framesAvailable = _ringBuffer.framesAvailable();
// if this stream was starved, check if we're still starved.
if (_isStarved && framesAvailable >= _desiredJitterBufferFrames) {
qCInfo(audiostream, "Starve ended");
_isStarved = false;
}
// if the ringbuffer exceeds the desired size by more than the threshold specified,
// drop the oldest frames so the ringbuffer is down to the desired size.
if (framesAvailable > _desiredJitterBufferFrames + MAX_FRAMES_OVER_DESIRED) {
int framesToDrop = framesAvailable - (_desiredJitterBufferFrames + DESIRED_JITTER_BUFFER_FRAMES_PADDING);
_ringBuffer.shiftReadPosition(framesToDrop * _ringBuffer.getNumFrameSamples());
_framesAvailableStat.reset();
_currentJitterBufferFrames = 0;
_oldFramesDropped += framesToDrop;
qCInfo(audiostream, "Dropped %d frames", framesToDrop);
qCInfo(audiostream, "Reset current jitter frames");
}
framesAvailableChanged();
return message.getPosition();
}
