pbos@webrtc.org 6917e19ad4 Rename EngineTest to CallTest.
There's no real notion of VideoEngine left in these classes. They're
end-to-end tests built on Call, so CallTest makes more sense.

This also contains a modification to RtpRtcpObserver moving the
responsibility of creating the event that signals when the observation
is complete to RtpRtcpObserver. New tests are about to be introduced and
this will reduce code duplication.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4793 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-19 14:22:12 +00:00
..
2013-09-19 07:15:36 +00:00
2013-09-19 07:15:36 +00:00
2012-10-22 18:19:23 +00:00
2012-10-22 18:19:23 +00:00
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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.