292084c376
it for the AudioRtpReceiver. This method returns a vector of RtpSource(both CSRC source and SSRC source) which contains the ID of a source, the timestamp, the source type (SSRC or CSRC) and the audio level. The RtpSource objects are buffered and maintained by the RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, the info of the contributing source will be pulled along the object chain: AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> AudioReceiveStream -> voe::Channel -> RtpRtcp module Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource BUG=chromium:703122 TBR=stefan@webrtc.org, danilchap@webrtc.org Review-Url: https://codereview.webrtc.org/2770233003 Cr-Commit-Position: refs/heads/master@{#17591}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.