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calculate unplayed ms on all streams/buffers as max

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This commit is contained in:
Zach Pomerantz 2016-09-15 17:54:41 -07:00
parent 9792d025fa
commit 177466e4c7
9 changed files with 102 additions and 92 deletions
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3
assignment-client/src/audio/AudioMixerClientData.cpp
View file

@ -270,6 +270,7 @@ QJsonObject AudioMixerClientData::getAudioStreamStats() {
downstreamStats["desired"] = streamStats._desiredJitterBufferFrames;
downstreamStats["available_avg_10s"] = streamStats._framesAvailableAverage;
downstreamStats["available"] = (double) streamStats._framesAvailable;
downstreamStats["unplayed"] = (double) streamStats._unplayedMs;
downstreamStats["starves"] = (double) streamStats._starveCount;
downstreamStats["not_mixed"] = (double) streamStats._consecutiveNotMixedCount;
downstreamStats["overflows"] = (double) streamStats._overflowCount;
@ -294,6 +295,7 @@ QJsonObject AudioMixerClientData::getAudioStreamStats() {
upstreamStats["desired_calc"] = avatarAudioStream->getCalculatedJitterBufferFrames();
upstreamStats["available_avg_10s"] = streamStats._framesAvailableAverage;
upstreamStats["available"] = (double) streamStats._framesAvailable;
upstreamStats["unplayed"] = (double) streamStats._unplayedMs;
upstreamStats["starves"] = (double) streamStats._starveCount;
upstreamStats["not_mixed"] = (double) streamStats._consecutiveNotMixedCount;
upstreamStats["overflows"] = (double) streamStats._overflowCount;
@ -323,6 +325,7 @@ QJsonObject AudioMixerClientData::getAudioStreamStats() {
upstreamStats["desired_calc"] = injectorPair.second->getCalculatedJitterBufferFrames();
upstreamStats["available_avg_10s"] = streamStats._framesAvailableAverage;
upstreamStats["available"] = (double) streamStats._framesAvailable;
upstreamStats["unplayed"] = (double) streamStats._unplayedMs;
upstreamStats["starves"] = (double) streamStats._starveCount;
upstreamStats["not_mixed"] = (double) streamStats._consecutiveNotMixedCount;
upstreamStats["overflows"] = (double) streamStats._overflowCount;

29
interface/src/ui/AudioStatsDialog.cpp
View file

@ -120,17 +120,14 @@ void AudioStatsDialog::renderStats() {
audioOutputBufferLatency = 0.0;
if (SharedNodePointer audioMixerNodePointer = DependencyManager::get<NodeList>()->soloNodeOfType(NodeType::AudioMixer)) {
mixerRingBufferFrames = (double)_stats->getMixerAvatarStreamStats()._framesAvailableAverage;
outputRingBufferFrames = (double)_stats->getMixerDownstreamStats()._framesAvailableAverage;
audioInputBufferLatency = (double)_stats->getAudioInputMsecsReadStats().getWindowAverage();
inputRingBufferLatency = (double)_stats->getInputRungBufferMsecsAvailableStats().getWindowAverage();
networkRoundtripLatency = (double) audioMixerNodePointer->getPingMs();
mixerRingBufferLatency = mixerRingBufferFrames * (double)AudioConstants::NETWORK_FRAME_MSECS;
outputRingBufferLatency = outputRingBufferFrames * (double)AudioConstants::NETWORK_FRAME_MSECS;
audioOutputBufferLatency = (double)_stats->getAudioOutputMsecsUnplayedStats().getWindowAverage();
audioInputBufferLatency = (double)_stats->getInputMsRead().getWindowMax();
inputRingBufferLatency = (double)_stats->getInputMsUnplayed().getWindowMax();
networkRoundtripLatency = (double)audioMixerNodePointer->getPingMs();
mixerRingBufferLatency = (double)_stats->getMixerAvatarStreamStats()._unplayedMs;
outputRingBufferLatency = (double)_stats->getMixerDownstreamStats()._unplayedMs;
audioOutputBufferLatency = (double)_stats->getOutputMsUnplayed().getWindowMax();
}
double totalLatency = audioInputBufferLatency + inputRingBufferLatency + mixerRingBufferLatency
+ outputRingBufferLatency + audioOutputBufferLatency + networkRoundtripLatency;
@ -142,20 +139,18 @@ void AudioStatsDialog::renderStats() {
_audioMixerStats.push_back(stats.arg(QString::number(inputRingBufferLatency, 'f', 0)));
stats = "Network (client->mixer):\t%1 ms";
_audioMixerStats.push_back(stats.arg(QString::number(networkRoundtripLatency / 2, 'f', 0)));
stats = "Mixer Ring:\t%1 ms (%2 frames)";
_audioMixerStats.push_back(stats.arg(QString::number(mixerRingBufferLatency, 'f', 0),
QString::number(mixerRingBufferFrames, 'f', 0)));
stats = "Mixer Ring:\t%1 ms";
_audioMixerStats.push_back(stats.arg(QString::number(mixerRingBufferLatency, 'f', 0)));
stats = "Network (mixer->client):\t%1 ms";
_audioMixerStats.push_back(stats.arg(QString::number(networkRoundtripLatency / 2, 'f', 0)));
stats = "Output Ring:\t%1 ms (%2 frames)";
_audioMixerStats.push_back(stats.arg(QString::number(outputRingBufferLatency, 'f', 0),
QString::number(outputRingBufferFrames, 'f', 0)));
stats = "Output Ring:\t%1 ms";
_audioMixerStats.push_back(stats.arg(QString::number(outputRingBufferLatency, 'f', 0)));
stats = "Output Read:\t%1 ms";
_audioMixerStats.push_back(stats.arg(QString::number(audioOutputBufferLatency, 'f', 0)));
stats = "TOTAL:\t%1 ms";
_audioMixerStats.push_back(stats.arg(QString::number(totalLatency, 'f', 0)));
const MovingMinMaxAvg<quint64>& packetSentTimeGaps = _stats->getPacketSentTimeGaps();
const MovingMinMaxAvg<quint64>& packetSentTimeGaps = _stats->getPacketTimegaps();
_upstreamClientStats.push_back("\nUpstream Mic Audio Packets Sent Gaps (by client):");

36
libraries/audio-client/src/AudioClient.cpp
View file

@ -123,12 +123,11 @@ AudioClient::AudioClient() :
_outputBufferSizeFrames("audioOutputBufferSizeFrames", DEFAULT_AUDIO_OUTPUT_BUFFER_SIZE_FRAMES),
_sessionOutputBufferSizeFrames(_outputBufferSizeFrames.get()),
_outputStarveDetectionEnabled("audioOutputBufferStarveDetectionEnabled",
DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_ENABLED),
DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_ENABLED),
_outputStarveDetectionPeriodMsec("audioOutputStarveDetectionPeriod",
DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_PERIOD),
DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_PERIOD),
_outputStarveDetectionThreshold("audioOutputStarveDetectionThreshold",
DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_THRESHOLD),
_averagedLatency(0.0f),
DEFAULT_AUDIO_OUTPUT_STARVE_DETECTION_THRESHOLD),
_lastInputLoudness(0.0f),
_timeSinceLastClip(-1.0f),
_muted(false),
@ -854,7 +853,7 @@ void AudioClient::handleAudioInput() {
_inputRingBuffer.writeData(inputByteArray.data(), inputByteArray.size());
float audioInputMsecsRead = inputByteArray.size() / (float)(_inputFormat.bytesForDuration(USECS_PER_MSEC));
_stats.updateInputMsecsRead(audioInputMsecsRead);
_stats.updateInputMsRead(audioInputMsecsRead);
const int numNetworkBytes = _isStereoInput
? AudioConstants::NETWORK_FRAME_BYTES_STEREO
@ -942,6 +941,10 @@ void AudioClient::handleAudioInput() {
emitAudioPacket(encodedBuffer.constData(), encodedBuffer.size(), _outgoingAvatarAudioSequenceNumber, audioTransform, packetType, _selectedCodecName);
_stats.sentPacket();
int bytesInInputRingBuffer = _inputRingBuffer.samplesAvailable() * sizeof(int16_t);
float msecsInInputRingBuffer = bytesInInputRingBuffer / (float)(_inputFormat.bytesForDuration(USECS_PER_MSEC));
_stats.updateInputMsUnplayed(msecsInInputRingBuffer);
}
}
@ -1358,22 +1361,6 @@ int AudioClient::calculateNumberOfFrameSamples(int numBytes) const {
return frameSamples;
}
float AudioClient::getInputRingBufferMsecsAvailable() const {
int bytesInInputRingBuffer = _inputRingBuffer.samplesAvailable() * sizeof(int16_t);
float msecsInInputRingBuffer = bytesInInputRingBuffer / (float)(_inputFormat.bytesForDuration(USECS_PER_MSEC));
return msecsInInputRingBuffer;
}
float AudioClient::getAudioOutputMsecsUnplayed() const {
if (!_audioOutput) {
return 0.0f;
}
int bytesAudioOutputUnplayed = _audioOutput->bufferSize() - _audioOutput->bytesFree();
float msecsAudioOutputUnplayed = bytesAudioOutputUnplayed / (float)_outputFormat.bytesForDuration(USECS_PER_MSEC);
return msecsAudioOutputUnplayed;
}
float AudioClient::azimuthForSource(const glm::vec3& relativePosition) {
// copied from AudioMixer, more or less
glm::quat inverseOrientation = glm::inverse(_orientationGetter());
@ -1430,8 +1417,11 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
bytesWritten = maxSize;
}
bool wasBufferStarved = _audio->_audioOutput->bufferSize() == _audio->_audioOutput->bytesFree();
if (wasBufferStarved) {
int bytesAudioOutputUnplayed = _audio->_audioOutput->bufferSize() - _audio->_audioOutput->bytesFree();
float msecsAudioOutputUnplayed = bytesAudioOutputUnplayed / (float)_audio->_outputFormat.bytesForDuration(USECS_PER_MSEC);
_audio->_stats.updateOutputMsUnplayed(msecsAudioOutputUnplayed);
if (bytesAudioOutputUnplayed == 0) {
_unfulfilledReads++;
}

4
libraries/audio-client/src/AudioClient.h
View file

@ -121,9 +121,6 @@ public:
const AudioIOStats& getStats() const { return _stats; }
float getInputRingBufferMsecsAvailable() const;
float getAudioOutputMsecsUnplayed() const;
int getOutputBufferSize() { return _outputBufferSizeFrames.get(); }
bool getOutputStarveDetectionEnabled() { return _outputStarveDetectionEnabled.get(); }
@ -284,7 +281,6 @@ private:
StDev _stdev;
QElapsedTimer _timeSinceLastReceived;
float _averagedLatency;
float _lastInputLoudness;
float _timeSinceLastClip;
int _totalInputAudioSamples;

76
libraries/audio-client/src/AudioIOStats.cpp
View file

@ -18,54 +18,73 @@
#include "AudioIOStats.h"
const int FRAMES_AVAILABLE_STATS_WINDOW_SECONDS = 10;
// This is called 5x/sec (see AudioStatsDialog), and we want it to log the last 5s
static const int INPUT_READS_WINDOW = 25;
static const int INPUT_UNPLAYED_WINDOW = 25;
static const int OUTPUT_UNPLAYED_WINDOW = 25;
const int APPROXIMATELY_30_SECONDS_OF_AUDIO_PACKETS = (int)(30.0f * 1000.0f / AudioConstants::NETWORK_FRAME_MSECS);
static const int APPROXIMATELY_30_SECONDS_OF_AUDIO_PACKETS = (int)(30.0f * 1000.0f / AudioConstants::NETWORK_FRAME_MSECS);
AudioIOStats::AudioIOStats(MixedProcessedAudioStream* receivedAudioStream) :
_receivedAudioStream(receivedAudioStream),
_audioInputMsecsReadStats(MSECS_PER_SECOND / (float)AudioConstants::NETWORK_FRAME_MSECS * AudioClient::CALLBACK_ACCELERATOR_RATIO, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS),
_inputRingBufferMsecsAvailableStats(1, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS),
_audioOutputMsecsUnplayedStats(1, FRAMES_AVAILABLE_STATS_WINDOW_SECONDS),
_lastSentAudioPacket(0),
_packetSentTimeGaps(1, APPROXIMATELY_30_SECONDS_OF_AUDIO_PACKETS)
_inputMsRead(0, INPUT_READS_WINDOW),
_inputMsUnplayed(0, INPUT_UNPLAYED_WINDOW),
_outputMsUnplayed(0, OUTPUT_UNPLAYED_WINDOW),
_lastSentPacketTime(0),
_packetTimegaps(0, APPROXIMATELY_30_SECONDS_OF_AUDIO_PACKETS)
{
}
AudioStreamStats AudioIOStats::getMixerDownstreamStats() const {
return _receivedAudioStream->getAudioStreamStats();
}
void AudioIOStats::reset() {
_receivedAudioStream->resetStats();
_inputMsRead.reset();
_inputMsUnplayed.reset();
_outputMsUnplayed.reset();
_packetTimegaps.reset();
_mixerAvatarStreamStats = AudioStreamStats();
_mixerInjectedStreamStatsMap.clear();
_audioInputMsecsReadStats.reset();
_inputRingBufferMsecsAvailableStats.reset();
_audioOutputMsecsUnplayedStats.reset();
_packetSentTimeGaps.reset();
}
void AudioIOStats::sentPacket() {
// first time this is 0
if (_lastSentAudioPacket == 0) {
_lastSentAudioPacket = usecTimestampNow();
if (_lastSentPacketTime == 0) {
_lastSentPacketTime = usecTimestampNow();
} else {
quint64 now = usecTimestampNow();
quint64 gap = now - _lastSentAudioPacket;
_packetSentTimeGaps.update(gap);
_lastSentAudioPacket = now;
quint64 gap = now - _lastSentPacketTime;
_lastSentPacketTime = now;
_packetTimegaps.update(gap);
}
}
void AudioIOStats::processStreamStatsPacket(QSharedPointer<ReceivedMessage> message, SharedNodePointer sendingNode) {
const MovingMinMaxAvg<float>& AudioIOStats::getInputMsRead() const {
_inputMsRead.currentIntervalComplete();
return _inputMsRead;
}
const MovingMinMaxAvg<float>& AudioIOStats::getInputMsUnplayed() const {
_inputMsUnplayed.currentIntervalComplete();
return _inputMsUnplayed;
}
const MovingMinMaxAvg<float>& AudioIOStats::getOutputMsUnplayed() const {
_outputMsUnplayed.currentIntervalComplete();
return _outputMsUnplayed;
}
const MovingMinMaxAvg<quint64>& AudioIOStats::getPacketTimegaps() const {
_packetTimegaps.currentIntervalComplete();
return _packetTimegaps;
}
const AudioStreamStats AudioIOStats::getMixerDownstreamStats() const {
return _receivedAudioStream->getAudioStreamStats();
}
void AudioIOStats::processStreamStatsPacket(QSharedPointer<ReceivedMessage> message, SharedNodePointer sendingNode) {
// parse the appendFlag, clear injected audio stream stats if 0
quint8 appendFlag;
message->readPrimitive(&appendFlag);
@ -92,14 +111,9 @@ void AudioIOStats::processStreamStatsPacket(QSharedPointer<ReceivedMessage> mess
}
void AudioIOStats::sendDownstreamAudioStatsPacket() {
auto audioIO = DependencyManager::get<AudioClient>();
// since this function is called every second, we'll sample for some of our stats here
_inputRingBufferMsecsAvailableStats.update(audioIO->getInputRingBufferMsecsAvailable());
_audioOutputMsecsUnplayedStats.update(audioIO->getAudioOutputMsecsUnplayed());
// also, call _receivedAudioStream's per-second callback
// call _receivedAudioStream's per-second callback
_receivedAudioStream->perSecondCallbackForUpdatingStats();
auto nodeList = DependencyManager::get<NodeList>();

32
libraries/audio-client/src/AudioIOStats.h
View file

@ -29,19 +29,20 @@ public:
void reset();
void updateInputMsecsRead(float msecsRead) { _audioInputMsecsReadStats.update(msecsRead); }
void updateInputMsRead(float ms) { _inputMsRead.update(ms); }
void updateInputMsUnplayed(float ms) { _inputMsUnplayed.update(ms); }
void updateOutputMsUnplayed(float ms) { _outputMsUnplayed.update(ms); }
void sentPacket();
AudioStreamStats getMixerDownstreamStats() const;
const MovingMinMaxAvg<float>& getInputMsRead() const;
const MovingMinMaxAvg<float>& getInputMsUnplayed() const;
const MovingMinMaxAvg<float>& getOutputMsUnplayed() const;
const MovingMinMaxAvg<quint64>& getPacketTimegaps() const;
const AudioStreamStats getMixerDownstreamStats() const;
const AudioStreamStats& getMixerAvatarStreamStats() const { return _mixerAvatarStreamStats; }
const QHash<QUuid, AudioStreamStats>& getMixerInjectedStreamStatsMap() const { return _mixerInjectedStreamStatsMap; }
const MovingMinMaxAvg<float>& getAudioInputMsecsReadStats() const { return _audioInputMsecsReadStats; }
const MovingMinMaxAvg<float>& getInputRungBufferMsecsAvailableStats() const { return _inputRingBufferMsecsAvailableStats; }
const MovingMinMaxAvg<float>& getAudioOutputMsecsUnplayedStats() const { return _audioOutputMsecsUnplayedStats; }
const MovingMinMaxAvg<quint64>& getPacketSentTimeGaps() const { return _packetSentTimeGaps; }
void sendDownstreamAudioStatsPacket();
public slots:
@ -49,17 +50,16 @@ public slots:
private:
MixedProcessedAudioStream* _receivedAudioStream;
MovingMinMaxAvg<float> _audioInputMsecsReadStats;
MovingMinMaxAvg<float> _inputRingBufferMsecsAvailableStats;
MovingMinMaxAvg<float> _audioOutputMsecsUnplayedStats;
mutable MovingMinMaxAvg<float> _inputMsRead;
mutable MovingMinMaxAvg<float> _inputMsUnplayed;
mutable MovingMinMaxAvg<float> _outputMsUnplayed;
quint64 _lastSentPacketTime;
mutable MovingMinMaxAvg<quint64> _packetTimegaps;
AudioStreamStats _mixerAvatarStreamStats;
QHash<QUuid, AudioStreamStats> _mixerInjectedStreamStatsMap;
quint64 _lastSentAudioPacket;
MovingMinMaxAvg<quint64> _packetSentTimeGaps;
};
#endif // hifi_AudioIOStats_h

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

@ -48,6 +48,7 @@ public:
quint32 _framesAvailable;
quint16 _framesAvailableAverage;
quint16 _unplayedMs;
quint16 _desiredJitterBufferFrames;
quint32 _starveCount;
quint32 _consecutiveNotMixedCount;

12
libraries/audio/src/InboundAudioStream.cpp
View file

@ -18,7 +18,10 @@
#include "InboundAudioStream.h"
#include "AudioLogging.h"
const int STARVE_HISTORY_CAPACITY = 50;
static const int STARVE_HISTORY_CAPACITY = 50;
// This is called 1x/s, and we want it to log the last 5s
static const int UNPLAYED_MS_WINDOW_SECS = 5;
InboundAudioStream::InboundAudioStream(int numFrameSamples, int numFramesCapacity, const Settings& settings) :
_ringBuffer(numFrameSamples, numFramesCapacity),
@ -46,6 +49,7 @@ InboundAudioStream::InboundAudioStream(int numFrameSamples, int numFramesCapacit
_starveHistory(STARVE_HISTORY_CAPACITY),
_starveThreshold(settings._windowStarveThreshold),
_framesAvailableStat(),
_unplayedMs(0, UNPLAYED_MS_WINDOW_SECS),
_currentJitterBufferFrames(0),
_timeGapStatsForStatsPacket(0, STATS_FOR_STATS_PACKET_WINDOW_SECONDS),
_repetitionWithFade(settings._repetitionWithFade),
@ -82,6 +86,7 @@ void InboundAudioStream::resetStats() {
_framesAvailableStat.reset();
_currentJitterBufferFrames = 0;
_timeGapStatsForStatsPacket.reset();
_unplayedMs.reset();
}
void InboundAudioStream::clearBuffer() {
@ -101,6 +106,7 @@ void InboundAudioStream::perSecondCallbackForUpdatingStats() {
_timeGapStatsForDesiredCalcOnTooManyStarves.currentIntervalComplete();
_timeGapStatsForDesiredReduction.currentIntervalComplete();
_timeGapStatsForStatsPacket.currentIntervalComplete();
_unplayedMs.currentIntervalComplete();
}
int InboundAudioStream::parseData(ReceivedMessage& message) {
@ -303,6 +309,9 @@ int InboundAudioStream::popFrames(int maxFrames, bool allOrNothing, bool starveI
}
void InboundAudioStream::popSamplesNoCheck(int samples) {
float unplayedMs = (_ringBuffer.samplesAvailable() / (float)_ringBuffer.getNumFrameSamples()) * AudioConstants::NETWORK_FRAME_MSECS;
_unplayedMs.update(unplayedMs);
_lastPopOutput = _ringBuffer.nextOutput();
_ringBuffer.shiftReadPosition(samples);
framesAvailableChanged();
@ -507,6 +516,7 @@ AudioStreamStats InboundAudioStream::getAudioStreamStats() const {
streamStats._framesAvailable = _ringBuffer.framesAvailable();
streamStats._framesAvailableAverage = _framesAvailableStat.getAverage();
streamStats._unplayedMs = (quint16)_unplayedMs.getWindowMax();
streamStats._desiredJitterBufferFrames = _desiredJitterBufferFrames;
streamStats._starveCount = _starveCount;
streamStats._consecutiveNotMixedCount = _consecutiveNotMixedCount;

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

@ -265,6 +265,7 @@ protected:
int _starveThreshold;
TimeWeightedAvg<int> _framesAvailableStat;
MovingMinMaxAvg<float> _unplayedMs;
// this value is periodically updated with the time-weighted avg from _framesAvailableStat. it is only used for
// dropping silent frames right now.