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These high level tests were disabled over time. Since they depend on real-time results and the filesystem, they tended to be flaky on the bots. We now give it a very generous 1 second to start up all channels before verification and a further relaxed file length check. If we continue to see problems, I will up the startup delay. The restored tests would have caught the AGC bug fixed here: https://code.google.com/p/webrtc/source/detail?r=5454 Add a new "real audio" stress test to exercise more code paths. This would have caught the refactor bug fixed here: https://code.google.com/p/webrtc/source/detail?r=5437 BUG=2164,2844 TESTED=git try. Verified it would have caught the aforementioned bugs by reintroducing them. R=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5522 4adac7df-926f-26a2-2b94-8c16560cd09d
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.