pbos@webrtc.org b74b96f487 Test multiple send/receive streams in Call.
Removes renderer in VideoReceiveStream as it wasn't properly
deregistered before. Makes sure that send/receive streams are properly
wired so that receive streams receive the expected stream.

BUG=2423
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2326004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4891 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-01 11:33:24 +00:00
..
2013-10-01 09:29:09 +00:00
2013-09-26 14:03:57 +00:00
2013-09-26 14:04:27 +00:00

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.