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Issue 2920983002: Roll WebRTC 18351:18408 (40 commits) (Closed)

Created:
3 years, 6 months ago by hbos_chromium
Modified:
3 years, 6 months ago
Reviewers:
CC:
chromium-reviews
Target Ref:
refs/heads/master
Project:
chromium
Visibility:
Public.

Description

Roll WebRTC 18351:18408 (40 commits) Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/7d76f48..c9e9080 $ git log 7d76f48..c9e9080 --date=short --no-merges --format=%ad %ae %s 2017-06-02 sakal@webrtc.org Call VideoCapturer.initialize directly from Java. 2017-06-02 kwiberg@webrtc.org SafeMin/SafeMax: Fix wrong return type when given two enum arguments 2017-06-02 eladalon@webrtc.org Prevent memory corruption by StreamId::Set 2017-06-02 magjed@webrtc.org Android: Add VideoFrame class 2017-06-02 jansson@webrtc.org Change all numerical string inputs to int and remove unused stderr 2017-06-01 danilchap@webrtc.org Rename rtp_header_extension.h to rtp_header_extension_map.h Move it to include path of the rtp_rtcp module to indicate it is ok to include it outside of the module. 2017-06-02 terelius@webrtc.org Overlay REMB in total bitrate graphs in visualization tool. 2017-06-02 brandtr@webrtc.org Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. 2017-06-01 braveyao@webrtc.org desktopCapture: scale the cursor image according to screen scale factor on OSX 2017-05-31 trnkumarchawla@gmail.com Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device. 2017-06-01 zstein@webrtc.org Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. 2017-06-01 mbonadei@webrtc.org enabling `gn check` on the whole WebRTC repo 2017-06-01 magjed@webrtc.org Update I420Buffer to new VideoFrameBuffer interface 2017-06-01 charujain@webrtc.org Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) 2017-06-01 henrik.lundin@webrtc.org Fix a bug in RtcEventLogSource 2017-06-01 holmer@chromium.org Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. 2017-06-01 sakal@webrtc.org Remove passing Android context to NetworkMonitor. 2017-06-01 henrika@webrtc.org Adds support for dynamic buffer size handling on recording side for iOS. Will also ensure that full-duplex audio now works on iOS simulators. 2017-06-01 nisse@webrtc.org Delete unneeded includes of system_wrappers/include/sleep.h 2017-06-01 terelius@webrtc.org Print configured header extensions and codecs in rtc_event_log2text. 2017-06-01 sakal@webrtc.org Switch from ScheduledExecutorService to ExecutorService. 2017-06-01 sakal@webrtc.org Remove deprecation warning from JVM::Initialize with the context parameter. 2017-06-01 mbonadei@webrtc.org Reland of Enabling `gn check` on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ ) 2017-06-01 mbonadei@webrtc.org Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ ) 2017-05-31 sakal@webrtc.org Remove native method VideoTrack.free which doesn't exist. 2017-06-01 brandtr@webrtc.org Re-enable EndToEndTest.PictureIdStateRetainedAfterReinitingVp8 on tsan. 2017-06-01 mbonadei@webrtc.org Enabling `gn check` on webrtc/test 2017-06-01 nisse@webrtc.org New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. 2017-06-01 terelius@webrtc.org Fix indexing error in event log analyzer. 2017-06-01 alessiob@webrtc.org In order to land https://codereview.webrtc.org/2790933002/ and due to the ongoing clean-up work (see https://codereview.webrtc.org/2887093002, https://codereview.webrtc.org/2894583002/ and https://codereview.webrtc.org/2891923002/), ReadDirectory() has been added in webrtc/test/testsupport/fileutils.h. 2017-06-01 asapersson@webrtc.org Use IsResolution(/Framerate)ScalingEnabled methods in more places. 2017-06-01 nisse@webrtc.org Delete class NullRtpData and function NullObjectRtpData. 2017-05-31 VladimirTechMan@gmail.com Definitions of video-codec name constants don't match their declarations 2017-05-31 VladimirTechMan@gmail.com Add media constraint-key constants for generating offers and answers 2017-05-31 jbauch@webrtc.org Support epoll in PhysicalSocketServer. 2017-05-31 eladalon@webrtc.org Create unit tests for RtpDemuxer 2017-05-31 kthelgason@webrtc.org Fix spelling mistake in field trial key name. 2017-05-31 brandtr@webrtc.org Disable flaky test EndToEndTest.TestFlexfecRtpStatePreservation on linux for now. 2017-05-31 nisse@webrtc.org Small cleanup of rtp_rtcp testAPI tests. 2017-05-31 kthelgason@webrtc.org Add back key for AGC field trial. TBR= CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng BUG= Review-Url: https://codereview.chromium.org/2920983002 Cr-Commit-Position: refs/heads/master@{#476643} Committed: https://chromium.googlesource.com/chromium/src/+/a8c5ece43442461d33c6de07d574cff22a971b81

Patch Set 1 #

Unified diffs Side-by-side diffs Delta from patch set Stats (+1 line, -1 line) Patch
M DEPS View 1 chunk +1 line, -1 line 0 comments Download

Messages

Total messages: 5 (3 generated)
commit-bot: I haz the power
CQ is trying da patch. Follow status at: https://chromium-cq-status.appspot.com/v2/patch-status/codereview.chromium.org/2920983002/1
3 years, 6 months ago (2017-06-02 12:38:12 UTC) #2
commit-bot: I haz the power
3 years, 6 months ago (2017-06-02 14:47:45 UTC) #5
Message was sent while issue was closed.
Committed patchset #1 (id:1) as
https://chromium.googlesource.com/chromium/src/+/a8c5ece43442461d33c6de07d574...

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