be9d2a4549
Allows setting SSRCs for future simulcast layers even though no set send codec uses them. Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required for bitrate ramp-up, instead of send-side only (resolving issue 3078). This test was used to verify reserved modules' SSRCs are preserved correctly. To enable a multiple-stream end-to-end test test::CallTest was modified to work on a vector of receive streams instead of just one. BUG=3078 R=kjellander@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15859005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
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.