RTCRtpReceiver
interface RTCRtpReceiver {
jitterBufferTarget: null | number;
track: MediaStreamTrack;
transform: null | RTCRtpScriptTransform;
transport: null | RTCDtlsTransport;
getContributingSources(): RTCRtpContributingSource[];
getParameters(): RTCRtpReceiveParameters;
getStats(): Promise<RTCStatsReport>;
getSynchronizationSources(): RTCRtpSynchronizationSource[];
}
jitterBufferTarget: null | number;
track: MediaStreamTrack;
transform: null | RTCRtpScriptTransform;
transport: null | RTCDtlsTransport;
getContributingSources(): RTCRtpContributingSource[];
getParameters(): RTCRtpReceiveParameters;
getStats(): Promise<RTCStatsReport>;
getSynchronizationSources(): RTCRtpSynchronizationSource[];
}
Index
Properties
Methods
Properties
jitter Buffer Target
jitterBufferTarget: null | number
Readonlytrack
transform
Readonlytransport
Methods
get Contributing Sources
getContributingSources(): RTCRtpContributingSource[]Returns RTCRtpContributingSource[]
get Parameters
getParameters(): RTCRtpReceiveParametersReturns RTCRtpReceiveParameters
get Stats
getStats(): Promise<RTCStatsReport>Returns Promise<RTCStatsReport>
get Synchronization Sources
getSynchronizationSources(): RTCRtpSynchronizationSource[]
This WebRTC API interface manages the reception and decoding of data for a MediaStreamTrack on an RTCPeerConnection.
MDN Reference