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 |
|
wu@webrtc.org
|
cd70119a10
|
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
|
2014-04-24 22:10:24 +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 |
|