RTCRtpContributingSource dictionary of the 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.
While the published specification describes this as an interface, it has since been changed to a dictionary in follow-up drafts.
audioLevelRead only 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.
- A 32-bit unsigned integer value specifying the CSRC identifier of the contributing source.
DOMHighResTimeStampindicating the most recent time of playout of an RTP packet from the source.
|WebRTC 1.0: Real-time Communication Between Browsers
The definition of 'RTCRtpContributingSource' in that specification.
|Candidate Recommendation||Initial definition.|
|Feature||Android webview||Chrome for Android||Edge mobile||Firefox for Android||Opera Android||iOS Safari||Samsung Internet|
1. From version 59: this feature is behind the
media.peerconnection.rtpsourcesapi.enable preference (needs to be set to
true). To change preferences in Firefox, visit about:config.