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.
- A double-precision floating-point value between 0 and 1 specifying the audio level contained in the last RTP packet played from this source.
- 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.
- A 32-bit unsigned integer value specifying the CSRC identifier of the contributing source.
DOMHighResTimeStampindicating the most recent time at which a frame originating from this source was delivered to the receiver's
|WebRTC 1.0: Real-time Communication Between Browsers
The definition of 'RTCRtpContributingSource' in that specification.
|Candidate Recommendation||Initial definition.|
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