79c3359e67bdbf3394a19f9251927c9482174043 - src

commit: 79c3359e67bdbf3394a19f9251927c9482174043
[log]
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

webrtc/test/fake_decoder.cc[diff]
webrtc/test/fake_decoder.h[diff]
webrtc/test/fake_encoder.cc[diff]
webrtc/test/fake_encoder.h[diff]
webrtc/video/end_to_end_tests.cc[diff]
webrtc/video/video_send_stream.cc[diff]

6 files changed

tree: 911e77f1295501648379e9902d2cb7cf4dccd156