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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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