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
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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