Since the code for setting these up will set the codec before setting SSRCs for the streams, any frames sent in between will be sent on random-generated SSRCs. This part should be added back during work on issue 1695. BUG=1695 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4629004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5192 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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