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Fix audio glitches caused by initial burst of SelectedAudioFormat packets.

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Temporary codec mismatch is expected during audio codec startup, due to packets already in-flight.
This commit is contained in:
Ken Cooke 2017-10-16 11:41:40 -07:00
parent e4e37e65e6
commit a702c5a0e4
2 changed files with 26 additions and 6 deletions
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31
libraries/audio/src/InboundAudioStream.cpp
View file

@ -46,6 +46,10 @@ static const int STATS_FOR_STATS_PACKET_WINDOW_SECONDS = 30;
// _currentJitterBufferFrames is updated with the time-weighted avg and the running time-weighted avg is reset.
static const quint64 FRAMES_AVAILABLE_STAT_WINDOW_USECS = 10 * USECS_PER_SECOND;
// When the audio codec is switched, temporary codec mismatch is expected due to packets in-flight.
// A SelectedAudioFormat packet is not sent until this threshold is exceeded.
static const int MAX_MISMATCHED_AUDIO_CODEC_COUNT = 10;
InboundAudioStream::InboundAudioStream(int numChannels, int numFrames, int numBlocks, int numStaticJitterBlocks) :
_ringBuffer(numChannels * numFrames, numBlocks),
_numChannels(numChannels),
@ -134,6 +138,7 @@ int InboundAudioStream::parseData(ReceivedMessage& message) {
switch (arrivalInfo._status) {
case SequenceNumberStats::Unreasonable: {
lostAudioData(1);
qDebug(audio) << "Sequence Unreasonable (LOST)";
break;
}
case SequenceNumberStats::Early: {
@ -143,6 +148,7 @@ int InboundAudioStream::parseData(ReceivedMessage& message) {
// fall through to the "on time" logic to actually handle this packet
int packetsDropped = arrivalInfo._seqDiffFromExpected;
lostAudioData(packetsDropped);
qDebug(audio) << "Sequence Early (LOST)";
// fall through to OnTime case
}
@ -153,6 +159,8 @@ int InboundAudioStream::parseData(ReceivedMessage& message) {
// 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);
qDebug(audio) << "OnTime (SILENT)";
} else {
// note: PCM and no codec are identical
bool selectedPCM = _selectedCodecName == "pcm" || _selectedCodecName == "";
@ -160,11 +168,15 @@ int InboundAudioStream::parseData(ReceivedMessage& message) {
if (codecInPacket == _selectedCodecName || (packetPCM && selectedPCM)) {
auto afterProperties = message.readWithoutCopy(message.getBytesLeftToRead());
parseAudioData(message.getType(), afterProperties);
qDebug(audio) << "OnTime (DECODE:" << codecInPacket << afterProperties.size() << ")";
_mismatchedAudioCodecCount = 0;
} else {
_mismatchedAudioCodecCount++;
qDebug(audio) << "Codec mismatch: expected" << _selectedCodecName << "got" << codecInPacket;
if (packetPCM) {
// If there are PCM packets in-flight while the codec is changed, use them.
// 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 {
@ -174,12 +186,16 @@ int InboundAudioStream::parseData(ReceivedMessage& message) {
lostAudioData(1);
}
// inform others of the mismatch
auto sendingNode = DependencyManager::get<NodeList>()->nodeWithUUID(message.getSourceID());
if (sendingNode) {
emit mismatchedAudioCodec(sendingNode, _selectedCodecName, codecInPacket);
}
if (_mismatchedAudioCodecCount > MAX_MISMATCHED_AUDIO_CODEC_COUNT) {
_mismatchedAudioCodecCount = 0;
// inform others of the mismatch
auto sendingNode = DependencyManager::get<NodeList>()->nodeWithUUID(message.getSourceID());
if (sendingNode) {
emit mismatchedAudioCodec(sendingNode, _selectedCodecName, codecInPacket);
qDebug(audio) << "Codec mismatch threshold exceeded, SelectedAudioFormat(" << _selectedCodecName << " ) sent";
}
}
}
}
break;
@ -515,6 +531,7 @@ float calculateRepeatedFrameFadeFactor(int indexOfRepeat) {
}
void InboundAudioStream::setupCodec(CodecPluginPointer codec, const QString& codecName, int numChannels) {
qDebug(audio) << "Setup Codec:" << codecName;
cleanupCodec(); // cleanup any previously allocated coders first
_codec = codec;
_selectedCodecName = codecName;
@ -525,6 +542,8 @@ void InboundAudioStream::setupCodec(CodecPluginPointer codec, const QString& cod
void InboundAudioStream::cleanupCodec() {
// release any old codec encoder/decoder first...
qDebug(audio) << "Cleanup Codec:" << _selectedCodecName;
if (_codec) {
if (_decoder) {
_codec->releaseDecoder(_decoder);

1
libraries/audio/src/InboundAudioStream.h
View file

@ -186,6 +186,7 @@ protected:
CodecPluginPointer _codec;
QString _selectedCodecName;
Decoder* _decoder { nullptr };
int _mismatchedAudioCodecCount { 0 };
};
float calculateRepeatedFrameFadeFactor(int indexOfRepeat);