henrike@webrtc.org 0d55c8f96d Adding peerconnection_unittest. ...

Review URL: http://webrtc-codereview.appspot.com/226004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@757 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-10-17 21:12:45 +00:00

..

app/webrtc_dev

Adding peerconnection_unittest.

2011-10-17 21:12:45 +00:00

examples/peerconnection_client

The change will separate the media tracks based on media type. MediaStreamInterface currently will have list for audio and video. This way we don't need to check for the track type before converting to respective mediatrack.

2011-10-17 13:19:08 +00:00

p2p/client

more webrtc session changes. Transport and TransportChannel handling is complete. Need work on session state.

2011-10-03 20:33:06 +00:00

session/phone

session/phone/channel.cc updates after new push of libjingle revision.

2011-10-14 09:45:24 +00:00