| commit | 79c3359e67bdbf3394a19f9251927c9482174043 | [log] [tgz] |
|---|---|---|
| author | stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Wed Aug 06 09:24:53 2014 |
| committer | stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Wed Aug 06 09:24:53 2014 |
| tree | 911e77f1295501648379e9902d2cb7cf4dccd156 | |
| parent | e415864a326ce48ff2d41d4c9a3faed26c033450 [diff] |
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