DescriptionRoll WebRTC 13783:13790 (7 commits)
Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/2c947a8..ecae934
$ git log 2c947a8..ecae934 --date=short --no-merges --format=%ad %ae %s
2016-08-16 noahric@chromium.org Add a gyp/gn option to use dummy audio file devices.
2016-08-16 honghaiz@webrtc.org Do not switch a connection if the new connection is not ready to send packets. There is no benefit of making such switches.
2016-08-16 zijiehe@chromium.org Currently there is not way to programmically test whether a ScreenCapturer implementation can accurately capture updated regions. Especially in ScreenCapturerWinDirectx, which has a specific updated region spreading logic and cannot be tested through regular code path. So we need a controllable ScreenDrawer to draw some basic shapes on the screen. And a platform independent test case can use the ScreenDrawer to test a ScreenCapturer.
2016-08-16 honghaiz@webrtc.org Change the default backup connection ping interval to 25 seconds.
2016-08-16 danilchap@webrtc.org Cleanup RtcpReceiver::TMMBRReceived function
2016-08-16 minyue@webrtc.org Revert of Adding audio to video_quality_test. (patchset #10 id:230001 of https://codereview.webrtc.org/2136573002/ )
2016-08-16 minyue@webrtc.org Adding audio to video_quality_test.
TBR=
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng
BUG=
Committed: https://crrev.com/df85217791a4245ef2d25de14bd482014e7f2745
Cr-Commit-Position: refs/heads/master@{#412481}
Patch Set 1 #Messages
Total messages: 5 (2 generated)
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