3 Commits

Author	SHA1	Message	Date
stefan@webrtc.org	48ac226b9a	Add support for writing h264 decoder input to file and parsing interleaved length/packet RTP dumps. ... 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	2015-03-02 16:19:15 +00:00
stefan@webrtc.org	79c3359e67	Add end-to-end H.264 packetization test. ... Also correctly wires up H.264 packetization in the new Call api. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20079004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6835 4adac7df-926f-26a2-2b94-8c16560cd09d	2014-08-06 09:24:53 +00:00
pbos@webrtc.org	16e03b7bd8	Separate Call API/build files from video_engine/. ... BUG=2535 R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d	2013-10-28 16:32:01 +00:00