WebRTC RTCInboundRtpStreamStats
GPTKB entity
Statements (71)
| Predicate | Object |
|---|---|
| gptkbp:instanceOf |
WebRTC statistics dictionary
|
| gptkbp:definedIn |
gptkb:WebRTC_specification
|
| https://www.w3.org/2000/01/rdf-schema#label |
WebRTC RTCInboundRtpStreamStats
|
| gptkbp:partOf |
gptkb:WebRTC_API
|
| gptkbp:property |
jitter
kind mid audioLevel burstDiscardCount burstDiscardRate burstLossCount burstLossRate burstPacketsDiscarded burstPacketsLost bytesDiscardedOnReception bytesReceived codecId concealedSamples concealmentEvents decoderImplementation estimatedPlayoutTimestamp fecPacketsDiscarded fecPacketsReceived firCount frameBitDepth frameHeight frameWidth framesAssembledFromMultiplePackets framesDecoded framesDropped framesPerSecond framesReceived freezeCount gapDiscardRate gapLossRate headerBytesReceived insertedSamplesForDeceleration jitterBufferDelay jitterBufferEmittedCount jitterBufferMinimumDelay keyFramesDecoded lastPacketReceivedTimestamp nackCount packetsDiscarded packetsLost packetsReceived packetsRepaired pauseCount pliCount powerEfficientDecoder qpSum remoteId removedSamplesForAcceleration rid silentConcealedSamples sliCount ssrc totalAudioEnergy totalInterFrameDelay totalPacketSendDelay totalProcessingDelay totalSamplesDuration totalSamplesReceived totalSamplesSent totalSquaredInterFrameDelay trackId transportId voiceActivityFlag |
| gptkbp:usedFor |
collecting statistics about inbound RTP streams
|
| gptkbp:bfsParent |
gptkb:The_Web_Platform
|
| gptkbp:bfsLayer |
8
|