kjellander@webrtc.org 918a8bf40c External transport is modified to never drop packets from the first frame.
Refactoring of FrameDropHandler: It now also tracks when frames are leaving the encoder and is being sent to external transport.

Previous 'Sent' state is now renamed to 'Created'.

NOTICE: The test seems to be a little flaky on Linux so it's not ready for buildbots yet. Since this might be caused by unstable production code further investigation should be performed to clear out the flakiness. I will file an issue for this when this CL is submitted (since I don't have any code to refer to before that). Usually the flakiness is caused by a decoded/rendered callback that is left out for the last frame, but I have seen other flaky failures too, which means it's not as simple as ignoring the last frame.
These errors occur even if 400kbps bit rate and 0% PL and 0 delay is configured.

BUG=
TEST=vie_auto_test --automated --gtest_filter="ViEVideoVerificationTest.RunsFullStackWithoutErrors" in Debug+Release on Linux, Mac and Windows.

Review URL: http://webrtc-codereview.appspot.com/339005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1597 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-03 12:40:28 +00:00
..
2012-02-03 12:33:50 +00:00
2012-01-17 12:45:47 +00:00
2011-11-02 09:31:39 +00:00
2012-02-01 23:04:48 +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.