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application stats tracking and packing updated;

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This commit is contained in:
wangyix 2014-07-10 16:54:26 -07:00
parent 25f4f63a1e
commit a7ef7647ed
4 changed files with 17 additions and 56 deletions
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1
interface/src/Application.cpp
View file

@ -2158,7 +2158,6 @@ void Application::updateMyAvatar(float deltaTime) {
{
quint64 sinceLastNack = now - _lastSendDownstreamAudioStats;
const quint64 TOO_LONG_SINCE_LAST_SEND_DOWNSTREAM_AUDIO_STATS = 1 * USECS_PER_SECOND;
if (sinceLastNack > TOO_LONG_SINCE_LAST_SEND_DOWNSTREAM_AUDIO_STATS) {
_lastSendDownstreamAudioStats = now;

2
interface/src/Application.h
View file

@ -125,6 +125,8 @@ static const float MIRROR_REARVIEW_DISTANCE = 0.65f;
static const float MIRROR_REARVIEW_BODY_DISTANCE = 2.3f;
static const float MIRROR_FIELD_OF_VIEW = 30.0f;
static const quint64 TOO_LONG_SINCE_LAST_SEND_DOWNSTREAM_AUDIO_STATS = 1 * USECS_PER_SECOND;
class Application : public QApplication {
Q_OBJECT

57
interface/src/Audio.cpp
View file

@ -48,15 +48,18 @@ static const float AUDIO_CALLBACK_MSECS = (float) NETWORK_BUFFER_LENGTH_SAMPLES_
static const int NUMBER_OF_NOISE_SAMPLE_FRAMES = 300;
static const int AUDIO_STREAM_STATS_HISTORY_SIZE = 30;
// audio frames time gap stats (min/max/avg) for last ~30 seconds are recalculated every ~1 second
const int TIME_GAPS_STATS_INTERVAL_SAMPLES = USECS_PER_SECOND / BUFFER_SEND_INTERVAL_USECS;
const int TIME_GAP_STATS_WINDOW_INTERVALS = 30;
static const int TIME_GAPS_STATS_INTERVAL_SAMPLES = USECS_PER_SECOND / BUFFER_SEND_INTERVAL_USECS;
static const int TIME_GAP_STATS_WINDOW_INTERVALS = 30;
// incoming sequence number stats history will cover last 30s
static const int INCOMING_SEQ_STATS_HISTORY_LENGTH = INCOMING_SEQ_STATS_HISTORY_LENGTH_SECONDS /
(TOO_LONG_SINCE_LAST_SEND_DOWNSTREAM_AUDIO_STATS / USECS_PER_SECOND);
// Mute icon configration
static const int MUTE_ICON_SIZE = 24;
Audio::Audio(int16_t initialJitterBufferSamples, QObject* parent) :
AbstractAudioInterface(parent),
_audioInput(NULL),
@ -110,9 +113,8 @@ Audio::Audio(int16_t initialJitterBufferSamples, QObject* parent) :
_scopeOutputLeft(0),
_scopeOutputRight(0),
_audioMixerAvatarStreamAudioStats(),
_audioMixerAvatarStreamPacketStatsHistory(AUDIO_STREAM_STATS_HISTORY_SIZE),
_outgoingAvatarAudioSequenceNumber(0),
_incomingStreamPacketStatsHistory(AUDIO_STREAM_STATS_HISTORY_SIZE),
_incomingMixedAudioSequenceNumberStats(INCOMING_SEQ_STATS_HISTORY_LENGTH),
_interframeTimeGapStats(TIME_GAPS_STATS_INTERVAL_SAMPLES, TIME_GAP_STATS_WINDOW_INTERVALS),
_starveCount(0),
_consecutiveNotMixedCount(0)
@ -138,13 +140,8 @@ void Audio::reset() {
_audioMixerAvatarStreamAudioStats = AudioStreamStats();
_audioMixerInjectedStreamAudioStatsMap.clear();
_audioMixerAvatarStreamPacketStatsHistory.clear();
_audioMixerInjectedStreamPacketStatsHistoryMap.clear();
_outgoingAvatarAudioSequenceNumber = 0;
_incomingMixedAudioSequenceNumberStats.reset();
_incomingStreamPacketStatsHistory.clear();
}
QAudioDeviceInfo getNamedAudioDeviceForMode(QAudio::Mode mode, const QString& deviceName) {
@ -751,7 +748,6 @@ void Audio::parseAudioStreamStatsPacket(const QByteArray& packet) {
dataAt += sizeof(quint8);
if (!appendFlag) {
_audioMixerInjectedStreamAudioStatsMap.clear();
_audioMixerInjectedStreamPacketStatsHistoryMap.clear();
}
// parse the number of stream stats structs to follow
@ -766,20 +762,10 @@ void Audio::parseAudioStreamStatsPacket(const QByteArray& packet) {
if (streamStats._streamType == PositionalAudioRingBuffer::Microphone) {
_audioMixerAvatarStreamAudioStats = streamStats;
_audioMixerAvatarStreamPacketStatsHistory.insert(streamStats._packetStreamStats);
} else {
if (!_audioMixerInjectedStreamAudioStatsMap.contains(streamStats._streamIdentifier)) {
_audioMixerInjectedStreamPacketStatsHistoryMap.insert(streamStats._streamIdentifier,
RingBufferHistory<PacketStreamStats>(AUDIO_STREAM_STATS_HISTORY_SIZE));
}
_audioMixerInjectedStreamAudioStatsMap[streamStats._streamIdentifier] = streamStats;
_audioMixerInjectedStreamPacketStatsHistoryMap[streamStats._streamIdentifier].insert(streamStats._packetStreamStats);
}
}
// when an audio stream stats packet is received, also record the downstream packets stats in the history
// for calculating packet loss rates
_incomingStreamPacketStatsHistory.insert(_incomingMixedAudioSequenceNumberStats.getStats());
}
AudioStreamStats Audio::getDownstreamAudioStreamStats() const {
@ -803,12 +789,17 @@ AudioStreamStats Audio::getDownstreamAudioStreamStats() const {
stats._ringBufferSilentFramesDropped = 0;
stats._packetStreamStats = _incomingMixedAudioSequenceNumberStats.getStats();
stats._packetStreamWindowStats = _incomingMixedAudioSequenceNumberStats.getStatsForHistoryWindow();
return stats;
}
void Audio::sendDownstreamAudioStatsPacket() {
// push the current seq number stats into history, which moves the history window forward 1s
// (since that's how often pushStatsToHistory() is called)
_incomingMixedAudioSequenceNumberStats.pushStatsToHistory();
char packet[MAX_PACKET_SIZE];
// pack header
@ -1608,25 +1599,3 @@ int Audio::calculateNumberOfFrameSamples(int numBytes) {
int frameSamples = (int)(numBytes * CALLBACK_ACCELERATOR_RATIO + 0.5f) / sizeof(int16_t);
return frameSamples;
}
void Audio::calculatePacketLossRate(const RingBufferHistory<PacketStreamStats>& statsHistory,
float& overallLossRate, float& windowLossRate) const {
int numHistoryEntries = statsHistory.getNumEntries();
if (numHistoryEntries == 0) {
overallLossRate = 0.0f;
windowLossRate = 0.0f;
} else {
const PacketStreamStats& newestStats = *statsHistory.getNewestEntry();
overallLossRate = (float)newestStats._numLost / newestStats._numReceived;
if (numHistoryEntries == 1) {
windowLossRate = overallLossRate;
} else {
int age = std::min(numHistoryEntries-1, AUDIO_STREAM_STATS_HISTORY_SIZE-1);
const PacketStreamStats& oldestStats = *statsHistory.get(age);
windowLossRate = (float)(newestStats._numLost - oldestStats._numLost)
/ (newestStats._numReceived - oldestStats._numReceived);
}
}
}

13
interface/src/Audio.h
View file

@ -36,6 +36,8 @@
static const int NUM_AUDIO_CHANNELS = 2;
static const int INCOMING_SEQ_STATS_HISTORY_LENGTH_SECONDS = 30;
class QAudioInput;
class QAudioOutput;
class QIODevice;
@ -121,14 +123,8 @@ public slots:
const AudioStreamStats& getAudioMixerAvatarStreamAudioStats() const { return _audioMixerAvatarStreamAudioStats; }
const QHash<QUuid, AudioStreamStats>& getAudioMixerInjectedStreamAudioStatsMap() const { return _audioMixerInjectedStreamAudioStatsMap; }
const RingBufferHistory<PacketStreamStats>& getAudioMixerAvatarStreamPacketStatsHistory() const { return _audioMixerAvatarStreamPacketStatsHistory; }
const QHash<QUuid, RingBufferHistory<PacketStreamStats> >& getAudioMixerInjectedStreamPacketStatsHistoryMap() const {return _audioMixerInjectedStreamPacketStatsHistoryMap; }
const RingBufferHistory<PacketStreamStats>& getIncomingStreamPacketStatsHistory() const { return _incomingStreamPacketStatsHistory; }
const MovingMinMaxAvg<quint64>& getInterframeTimeGapStats() const { return _interframeTimeGapStats; }
void calculatePacketLossRate(const RingBufferHistory<PacketStreamStats>& statsHistory,
float& overallLossRate, float& windowLossRate) const;
signals:
bool muteToggled();
void preProcessOriginalInboundAudio(unsigned int sampleTime, QByteArray& samples, const QAudioFormat& format);
@ -266,14 +262,9 @@ private:
AudioStreamStats _audioMixerAvatarStreamAudioStats;
QHash<QUuid, AudioStreamStats> _audioMixerInjectedStreamAudioStatsMap;
RingBufferHistory<PacketStreamStats> _audioMixerAvatarStreamPacketStatsHistory;
QHash<QUuid, RingBufferHistory<PacketStreamStats> > _audioMixerInjectedStreamPacketStatsHistoryMap;
quint16 _outgoingAvatarAudioSequenceNumber;
SequenceNumberStats _incomingMixedAudioSequenceNumberStats;
RingBufferHistory<PacketStreamStats> _incomingStreamPacketStatsHistory;
MovingMinMaxAvg<quint64> _interframeTimeGapStats;
};