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。
标签：DOMString,double,unsigned,long,每秒,rtp,速率,WebRTC,getStats From： https://www.cnblogs.com/brisk/p/17274025.html