GitLab

https://webrtc.googlesource.com/src.git/+log/5f0ce99c04eb..c97933fb82f8


git log 5f0ce99c04eb..c97933fb82f8 --date=short --no-merges --format='%ad %ae %s'
2018-08-10 minyue@webrtc.org Clean up code regarding jitter buffer plot in event log visualizer.
2018-08-10 buildbot@webrtc.org Roll chromium_revision 86f8273e..f7e49bb7 (581886:582110)
2018-08-10 ivoc@webrtc.org Add command-line flag to enable the bugfix to postpone decoding after expand.
2018-08-10 terelius@webrtc.org Print timestamp-to-UTC map when event_log starts.
2018-08-10 nisse@webrtc.org Delete unused method RtpReceiver::CSRCs.
2018-08-10 mbonadei@webrtc.org Change visibility of some build targets that are publicly used.
2018-08-10 buildbot@webrtc.org Roll chromium_revision b0784bef..86f8273e (581665:581886)
2018-08-10 nisse@webrtc.org Reland "Refactor RtpVideoStreamReceiver without RtpReceiver."
2018-08-09 peah@webrtc.org AEC3: Ensure that the shadow filter is adapted at each block
2018-08-09 sprang@webrtc.org Experimental improvements for simulcast screenshare
2018-08-09 alessiob@webrtc.org APM: render pre-processor moved before echo detector queuing.
2018-08-09 sakal@webrtc.org Add extended header containing frame ID to the generic packetizer.
2018-08-09 aleloi@webrtc.org Optionally disable digital gain control in ExperimentalAgc.
2018-08-09 magjed@webrtc.org Android: Allow YuvConverter to be reused
2018-08-09 oprypin@webrtc.org cq_name is no longer used and can and should be removed
2018-08-09 phoglund@webrtc.org Roll chromium_revision 474eca05..b0784bef (581204:581665)
2018-08-09 oprypin@webrtc.org Add compile-only bots (used for binary size) to commit queue
2018-08-09 oprypin@webrtc.org Add post-submit builders without dcheck_always_on
2018-08-09 philipel@webrtc.org Remove RTPVideoHeader::vp9() accessors.
2018-08-09 danilchap@webrtc.org Remove raw extensions accessors from rtp packet
2018-08-09 sprang@webrtc.org SimulcastEncoderAdapter should not update maxQp for screencast
2018-08-09 titovartem@webrtc.org Remove old base64 header
2018-08-09 nisse@webrtc.org Add test CallPerfTest.PlaysOutAudioAndVideoInSyncWithoutClockDrift
2018-08-09 nisse@webrtc.org Delete unused constants from rtp_rtcp_config.h
2018-08-09 mbonadei@webrtc.org Making rtc_base:ptr_util and rtc_base:refcount public.


Created with:
  gclient setdep -r src/third_party/webrtc@c97933fb82f8

The AutoRoll server is located here: https://webrtc-chromium-roll.skia.org

Documentation for the AutoRoller is here:
https://skia.googlesource.com/buildbot/+/master/autoroll/README.md

If the roll is causing failures, please contact the current sheriff, who should
be CC'd on the roll, and stop the roller if necessary.

CQ_INCLUDE_TRYBOTS=luci.chromium.try:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng

BUG=chromium:None,chromium:None,chromium:None,chromium:None,chromium:872201,chromium:b/112386285,chromium:None,chromium:None,chromium:none,chromium:None,chromium:None,chromium:None,chromium:None
TBR=webrtc-chromium-sheriffs-robots@google.com

Change-Id: Ie0f047103689803c7eadfe059b10886153d22c2c
Reviewed-on: https://chromium-review.googlesource.com/1169939Reviewed-by: webrtc-chromium-autoroll <webrtc-chromium-autoroll@skia-buildbots.google.com.iam.gserviceaccount.com>
Commit-Queue: webrtc-chromium-autoroll <webrtc-chromium-autoroll@skia-buildbots.google.com.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#582173}