stefan@webrtc.org e1c9caf6ee Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804. ...

TEST=buildtools/linux64/gn gen out/Default
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6805 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-07-31 15:07:59 +00:00

..

interface

Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.

2014-07-17 16:10:14 +00:00

mocks

Add H.264 packetization.

2014-07-31 14:59:24 +00:00

source

Add H.264 packetization.

2014-07-31 14:59:24 +00:00

test

Remove the send-side cname getter APIs from voice and video engine.

2014-07-11 09:55:30 +00:00

BUILD.gn

Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804.

2014-07-31 15:07:59 +00:00

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00