When using send-side bandwidth estimation, the inter-packet delay is reported back to the sender using RTCP TransportFeedback messages. Theis data needs to be translated into a format which the bandwidth estimator (now instantiated on the send side) can use, including looking up the local absolute send time from the send time history. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1329083005 Cr-Commit-Position: refs/heads/master@{#9929}
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.