RTCRtpContributingSource
The RTCRtpContributingSource
dictionary of the WebRTC API is used by getContributingSources()
to provide information about a given contributing source (CSRC), including the most recent time a packet that the source contributed was played out.
The information provided is based on the last ten seconds of media received.
Instance properties
audioLevel
Optional-
A double-precision floating-point value between 0 and 1 specifying the audio level contained in the last RTP packet played from this source.
rtpTimestamp
Optional-
The RTP timestamp of the media played out at the time indicated by
timestamp
. This value is a source-generated time value which can be used to help with sequencing and synchronization. source
Optional-
A 32-bit unsigned integer value specifying the CSRC identifier of the contributing source.
timestamp
Optional-
A
DOMHighResTimeStamp
indicating the most recent time at which a frame originating from this source was delivered to the receiver'sMediaStreamTrack
Specifications
Specification |
---|
WebRTC: Real-Time Communication in Browsers # dom-rtcrtpcontributingsource |
Browser compatibility
BCD tables only load in the browser