diff --git a/assignment-client/src/audio/AvatarAudioStream.cpp b/assignment-client/src/audio/AvatarAudioStream.cpp index 1b3ca9a8b1..24b14ac9e5 100644 --- a/assignment-client/src/audio/AvatarAudioStream.cpp +++ b/assignment-client/src/audio/AvatarAudioStream.cpp @@ -45,6 +45,7 @@ int AvatarAudioStream::parseStreamProperties(PacketType type, const QByteArray& : AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL); // restart the codec if (_codec) { + QMutexLocker lock(&_decoderMutex); if (_decoder) { _codec->releaseDecoder(_decoder); } diff --git a/libraries/audio/src/InboundAudioStream.cpp b/libraries/audio/src/InboundAudioStream.cpp index 8c5388e222..5ac3996029 100644 --- a/libraries/audio/src/InboundAudioStream.cpp +++ b/libraries/audio/src/InboundAudioStream.cpp @@ -10,6 +10,7 @@ // #include "InboundAudioStream.h" +#include "TryLocker.h" #include @@ -215,7 +216,7 @@ int InboundAudioStream::parseData(ReceivedMessage& message) { 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; @@ -247,10 +248,18 @@ int InboundAudioStream::lostAudioData(int numPackets) { QByteArray decodedBuffer; while (numPackets--) { + MutexTryLocker lock(_decoderMutex); + if (!lock.isLocked()) { + // an incoming packet is being processed, + // and will likely be on the ring buffer shortly, + // so don't bother generating more data + qCInfo(audiostream, "Packet currently being unpacked or lost frame already being generated. Not generating lost frame."); + return 0; + } if (_decoder) { _decoder->lostFrame(decodedBuffer); } else { - decodedBuffer.resize(AudioConstants::NETWORK_FRAME_BYTES_STEREO); + decodedBuffer.resize(AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL * _numChannels); memset(decodedBuffer.data(), 0, decodedBuffer.size()); } _ringBuffer.writeData(decodedBuffer.data(), decodedBuffer.size()); @@ -260,6 +269,12 @@ int InboundAudioStream::lostAudioData(int numPackets) { int InboundAudioStream::parseAudioData(PacketType type, const QByteArray& packetAfterStreamProperties) { QByteArray decodedBuffer; + + // may block on the real-time thread, which is acceptible as + // parseAudioData is only called by the packet processing + // thread which, while high performance, is not as sensitive to + // delays as the real-time thread. + QMutexLocker lock(&_decoderMutex); if (_decoder) { _decoder->decode(packetAfterStreamProperties, decodedBuffer); } else { @@ -278,16 +293,23 @@ int InboundAudioStream::writeDroppableSilentFrames(int silentFrames) { // case we will call the decoder's lostFrame() method, which indicates // that it should interpolate from its last known state down toward // silence. - if (_decoder) { - // FIXME - We could potentially use the output from the codec, in which - // case we might get a cleaner fade toward silence. NOTE: The below logic - // attempts to catch up in the event that the jitter buffers have grown. - // The better long term fix is to use the output from the decode, detect - // when it actually reaches silence, and then delete the silent portions - // of the jitter buffers. Or petentially do a cross fade from the decode - // output to silence. - QByteArray decodedBuffer; - _decoder->lostFrame(decodedBuffer); + { + // may block on the real-time thread, which is acceptible as + // writeDroppableSilentFrames is only called by the packet processing + // thread which, while high performance, is not as sensitive to + // delays as the real-time thread. + QMutexLocker lock(&_decoderMutex); + if (_decoder) { + // FIXME - We could potentially use the output from the codec, in which + // case we might get a cleaner fade toward silence. NOTE: The below logic + // attempts to catch up in the event that the jitter buffers have grown. + // The better long term fix is to use the output from the decode, detect + // when it actually reaches silence, and then delete the silent portions + // of the jitter buffers. Or petentially do a cross fade from the decode + // output to silence. + QByteArray decodedBuffer; + _decoder->lostFrame(decodedBuffer); + } } // calculate how many silent frames we should drop. @@ -338,10 +360,23 @@ int InboundAudioStream::popSamples(int maxSamples, bool allOrNothing) { popSamplesNoCheck(samplesAvailable); samplesPopped = samplesAvailable; } else { - // we can't pop any samples, set this stream to starved + // we can't pop any samples, set this stream to starved for jitter + // buffer calculations. setToStarved(); _consecutiveNotMixedCount++; - _lastPopSucceeded = false; + //Kick PLC to generate a filler frame, reducing 'click' + lostAudioData(allOrNothing ? (maxSamples - samplesAvailable) / _ringBuffer.getNumFrameSamples() : 1); + samplesPopped = _ringBuffer.samplesAvailable(); + if (samplesPopped) { + popSamplesNoCheck(samplesPopped); + } else { + // No samples available means a packet is currently being + // processed, so we don't generate lost audio data, and instead + // just wait for the packet to come in. This prevents locking + // the real-time audio thread at the cost of a potential (but rare) + // 'click' + _lastPopSucceeded = false; + } } } return samplesPopped; @@ -528,6 +563,7 @@ void InboundAudioStream::setupCodec(CodecPluginPointer codec, const QString& cod _codec = codec; _selectedCodecName = codecName; if (_codec) { + QMutexLocker lock(&_decoderMutex); _decoder = codec->createDecoder(AudioConstants::SAMPLE_RATE, numChannels); } } @@ -535,6 +571,7 @@ void InboundAudioStream::setupCodec(CodecPluginPointer codec, const QString& cod void InboundAudioStream::cleanupCodec() { // release any old codec encoder/decoder first... if (_codec) { + QMutexLocker lock(&_decoderMutex); if (_decoder) { _codec->releaseDecoder(_decoder); _decoder = nullptr; diff --git a/libraries/audio/src/InboundAudioStream.h b/libraries/audio/src/InboundAudioStream.h index 5ff9e2c84c..c10a86cb69 100644 --- a/libraries/audio/src/InboundAudioStream.h +++ b/libraries/audio/src/InboundAudioStream.h @@ -187,6 +187,7 @@ protected: CodecPluginPointer _codec; QString _selectedCodecName; + QMutex _decoderMutex; Decoder* _decoder { nullptr }; int _mismatchedAudioCodecCount { 0 }; }; diff --git a/libraries/audio/src/MixedProcessedAudioStream.cpp b/libraries/audio/src/MixedProcessedAudioStream.cpp index 082977246b..6510f0bfc9 100644 --- a/libraries/audio/src/MixedProcessedAudioStream.cpp +++ b/libraries/audio/src/MixedProcessedAudioStream.cpp @@ -11,6 +11,7 @@ #include "MixedProcessedAudioStream.h" #include "AudioLogging.h" +#include "TryLocker.h" MixedProcessedAudioStream::MixedProcessedAudioStream(int numFramesCapacity, int numStaticJitterFrames) : InboundAudioStream(AudioConstants::STEREO, AudioConstants::NETWORK_FRAME_SAMPLES_PER_CHANNEL, @@ -36,13 +37,20 @@ int MixedProcessedAudioStream::lostAudioData(int numPackets) { QByteArray outputBuffer; while (numPackets--) { + MutexTryLocker lock(_decoderMutex); + if (!lock.isLocked()) { + // an incoming packet is being processed, + // and will likely be on the ring buffer shortly, + // so don't bother generating more data + qCInfo(audiostream, "Packet currently being unpacked or lost frame already being generated. Not generating lost frame."); + return 0; + } if (_decoder) { _decoder->lostFrame(decodedBuffer); } else { decodedBuffer.resize(AudioConstants::NETWORK_FRAME_BYTES_STEREO); memset(decodedBuffer.data(), 0, decodedBuffer.size()); } - emit addedStereoSamples(decodedBuffer); emit processSamples(decodedBuffer, outputBuffer); @@ -55,6 +63,12 @@ int MixedProcessedAudioStream::lostAudioData(int numPackets) { int MixedProcessedAudioStream::parseAudioData(PacketType type, const QByteArray& packetAfterStreamProperties) { QByteArray decodedBuffer; + + // may block on the real-time thread, which is acceptible as + // parseAudioData is only called by the packet processing + // thread which, while high performance, is not as sensitive to + // delays as the real-time thread. + QMutexLocker lock(&_decoderMutex); if (_decoder) { _decoder->decode(packetAfterStreamProperties, decodedBuffer); } else {