8d15bd6dab
Reason for revert: The reverted commit didn't affect the tests, but the one before: https://codereview.webrtc.org/1385563005/ I've run the test that was failing (EndToEndTest.AssignsTransportSequenceNumbers) locally multiple times, and it works fine (finishes successfully in 150-170ms). Original issue's description: > Revert of Collecting encode_time_ms for each frame (patchset #13 id:220001 of https://codereview.webrtc.org/1374233002/ ) > > Reason for revert: > Breaks EndToEndTest.AssignsTransportSequenceNumbers in video_engine_tests > on several bots: > http://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5507 > http://build.chromium.org/p/client.webrtc/builders/Mac64%20Debug/builds/4815 > http://build.chromium.org/p/client.webrtc/builders/Win%20SyzyASan/builds/3272 > http://build.chromium.org/p/client.webrtc/builders/Linux%20Memcheck/builds/4414 > > It seems very unfortunate that it breaks on _exactly_ the bot configs that aren't covered by the CQ trybots. > > Original issue's description: > > Collecting encode_time_ms for each frame. > > > > Also, in Sample struct, replacing double with the original type. > > It makes more sense to save the original data as truthful as possible, and then > > convert it to double later if necessary (in the plot script). > > > > Committed: https://crrev.com/092b13384e57b33e2003d9736dfa1f491e76f938 > > Cr-Commit-Position: refs/heads/master@{#10184} > > TBR=sprang@webrtc.org,pbos@webrtc.org,mflodman@webrtc.org,asapersson@webrtc.org,ivica@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/810447972425e890bc7911af27f894b86e9b7e6f > Cr-Commit-Position: refs/heads/master@{#10185} TBR=sprang@webrtc.org,pbos@webrtc.org,mflodman@webrtc.org,asapersson@webrtc.org,kjellander@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1390163002 Cr-Commit-Position: refs/heads/master@{#10195}
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.