pbos@webrtc.org be9d2a4549 Reserve RTP/RTCP modules in SetSSRC.
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
2014-06-30 13:19:09 +00:00
2014-06-11 13:59:44 +00:00
2014-06-17 08:54:03 +00:00
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
BSD-3-Clause 446 MiB
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