在用WebRTC进行通信时，可以通过RTCPeerConnection对象的getStats方法获取相关的连接统计信息，以此获取每秒帧数。
-- By Brisk yu
1 getStats的使用方法
const pc = new RTCPeerConnection() // 获取视频流对象 var selector = pc.getRemoteStreams()[0].getAudioTracks()[0] // 获取视频流通信信息 pc.getStats(selector).then(report => { report.forEach(stats => { console.log(stats.type); console.log(stats); }); }).catch(err => { console.error(err); });
2 getStats可以获取的信息
getStats可以获取的信息如下：
enum RTCStatsType { "codec", "inbound-rtp", "outbound-rtp", "remote-inbound-rtp", "remote-outbound-rtp", "media-source", "media-playout", "peer-connection", "data-channel", "stream", "track", "transport", "candidate-pair", "local-candidate", "remote-candidate", "certificate" };
其中"inbound-rtp"表示接收到的RTP流相关统计信息，其结构如下：
dictionary RTCInboundRtpStreamStats : RTCReceivedRtpStreamStats { required DOMString trackIdentifier; required DOMString kind; DOMString mid; DOMString remoteId; unsigned long framesDecoded; unsigned long keyFramesDecoded; unsigned long framesRendered; unsigned long framesDropped; unsigned long frameWidth; unsigned long frameHeight; double framesPerSecond; unsigned long long qpSum; double totalDecodeTime; double totalInterFrameDelay; double totalSquaredInterFrameDelay; unsigned long pauseCount; double totalPausesDuration; unsigned long freezeCount; double totalFreezesDuration; DOMHighResTimeStamp lastPacketReceivedTimestamp; unsigned long long headerBytesReceived; unsigned long long packetsDiscarded; unsigned long long fecPacketsReceived; unsigned long long fecPacketsDiscarded; unsigned long long bytesReceived; unsigned long nackCount; unsigned long firCount; unsigned long pliCount; double totalProcessingDelay; DOMHighResTimeStamp estimatedPlayoutTimestamp; double jitterBufferDelay; double jitterBufferTargetDelay; unsigned long long jitterBufferEmittedCount; double jitterBufferMinimumDelay; unsigned long long totalSamplesReceived; unsigned long long concealedSamples; unsigned long long silentConcealedSamples; unsigned long long concealmentEvents; unsigned long long insertedSamplesForDeceleration; unsigned long long removedSamplesForAcceleration; double audioLevel; double totalAudioEnergy; double totalSamplesDuration; unsigned long framesReceived; DOMString decoderImplementation; DOMString playoutId; boolean powerEfficientDecoder; unsigned long framesAssembledFromMultiplePackets; double totalAssemblyTime; };
其中framesPerSecond表示接收到的RTP流的最后一秒的帧数，可以用于展示每秒帧数。