This is useful for debugging h264 input when we don't have an h264 decoder, as the resulting file should be possible to play back using mplayer. It is also often convenient to dump rtp packets in an interleaved format where the size of a packet is inserted before the actual payload. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42139004 Cr-Commit-Position: refs/heads/master@{#8558} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8558 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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