Commit 62d569e0 authored by chromium-autoroll's avatar chromium-autoroll Committed by Commit Bot

Roll src/third_party/webrtc b42aeaa3fb21..ccab06fb72be (53 commits)

https://webrtc.googlesource.com/src.git/+log/b42aeaa3fb21..ccab06fb72be

git log b42aeaa3fb21..ccab06fb72be --date=short --first-parent --format='%ad %ae %s'
2020-01-15 guidou@webrtc.org Revert "Replaces SynchronousMethodCall with rtc::Thread::Invoke."
2020-01-15 danilchap@webrtc.org Change H264 depacketizer to implement VideoRtpDepacketizer interface
2020-01-15 hbos@webrtc.org Move DegradationPreference logic to the encoder queue.
2020-01-15 eshr@google.com Don't pace audio by default
2020-01-15 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 65afcfa0..2638d764 (731779:731908)
2020-01-15 danilchap@webrtc.org Change Av1 depacketizer to implement VideoRtpDepacketizer interface
2020-01-15 srte@webrtc.org Revert "Using simulated rtc::Thread for peer connection scenario tests."
2020-01-15 srte@webrtc.org Using simulated rtc::Thread for peer connection scenario tests.
2020-01-15 mbonadei@webrtc.org Add data dependency to event_log_visualizer.
2020-01-15 srte@webrtc.org Allow overwriting current thread in ThreadManager.
2020-01-15 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision c85b2ddb..65afcfa0 (731677:731779)
2020-01-14 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision e61d470d..c85b2ddb (731529:731677)
2020-01-14 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 81b1889c..e61d470d (731328:731529)
2020-01-14 srte@webrtc.org Adds scenario test for transport wide feedback based retransmission.
2020-01-14 jonasolsson@webrtc.org Concatenate string literals at compile time.
2020-01-14 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision a989226e..81b1889c (731140:731328)
2020-01-14 saza@webrtc.org AEC3: Restrict default logging of some delay changes to VERBOSE
2020-01-14 srte@webrtc.org Use RTX SSRCs in scenario test framework.
2020-01-14 srte@webrtc.org Cleanup: Prepares for simulated time peer connection tests.
2020-01-14 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision d6f6958d..a989226e (731013:731140)
2020-01-14 yura.yaroshevich@gmail.com Fixed timeout overflow in sctp reliability test.
2020-01-14 jerome.humbert@microsoft.com Suppress C5041 constexpr warning for MSVC 2019
2020-01-14 phoglund@webrtc.org Extract an interface from the perf results logger.
2020-01-14 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 69c66e43..d6f6958d (730870:731013)
2020-01-13 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 0792dc5f..69c66e43 (730752:730870)
2020-01-13 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 210e7907..0792dc5f (730612:730752)
2020-01-13 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision b581de5b..210e7907 (730447:730612)
2020-01-13 natim@webrtc.org Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 hbos@webrtc.org Move DegradationPreference logic out of VideoSourceSinkController.
2020-01-13 yura.yaroshevich@gmail.com [iOS] Reset VT session when H264 decoder malfunction error happen
2020-01-13 srte@webrtc.org Cleanup: Merges Thread and MessageQueue.
2020-01-13 danilchap@webrtc.org Delete RtpGenericDepacketizer as no longer used
2020-01-13 saza@webrtc.org Change log level of AEC3 buffer info to VERBOSE
2020-01-13 danilchap@webrtc.org In TaskQueueWin fix race in canceling MutlimediaTimer
2020-01-13 saza@webrtc.org Add saza@ and peah@ to OWNERS of some audio files
2020-01-13 yvesg@webrtc.org More lenient double comparison for RunningStatistics.FullSimpleTest
2020-01-13 srte@webrtc.org Separates simulated TaskQueue and simulated ProcessThread.
2020-01-13 hbos@webrtc.org VideoStreamEncoder configuring source/sink with VideoSourceController.
2020-01-13 hbos@webrtc.org Introduce ResourceAdaptationModuleListener and VideoSourceRestrictions.
2020-01-13 jerome.humbert@microsoft.com Avoid [[nodiscard]] warning C4834 with MSVC 2019
2020-01-13 mbonadei@webrtc.org Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs""""
2020-01-13 jerome.humbert@microsoft.com Add missing header for dchecked_cast on UWP
2020-01-13 jonaso@webrtc.org Flip goog_ping_announce default to false
2020-01-12 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision d794106d..b581de5b (730346:730447)
2020-01-12 srte@webrtc.org Using tasks to process packets in FakeNetworkSocketServer.
2020-01-11 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision bd2395cd..d794106d (730226:730346)
2020-01-10 kron@webrtc.org Reland "Reland "Reland "Distinguish between send and receive video codecs"""
2020-01-10 shampson@webrtc.org Adding deadbeef to sctp/OWNERS and removing myself.
2020-01-10 srte@webrtc.org Replaces SynchronousMethodCall with rtc::Thread::Invoke.
2020-01-10 srte@webrtc.org Cleanup: Replace MessageQueue pointers with Thread pointers.
2020-01-10 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 54a7cb4b..bd2395cd (730109:730226)
2020-01-10 yvesg@webrtc.org Unflake P2PTransportChannelTest.TurnToTurnPresumedWritable.
2020-01-10 mbonadei@webrtc.org Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."

Created with:
  gclient setdep -r src/third_party/webrtc@ccab06fb72be

If this roll has caused a breakage, revert this CL and stop the roller
using the controls here:
https://autoroll.skia.org/r/webrtc-chromium-autoroll
Please CC webrtc-chromium-sheriffs-robots@google.com on the revert to ensure that a human
is aware of the problem.

To report a problem with the AutoRoller itself, please file a bug:
https://bugs.chromium.org/p/skia/issues/entry?template=Autoroller+Bug

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

Bug: chromium:1029452,chromium:1029737
Tbr: webrtc-chromium-sheriffs-robots@google.com
Change-Id: I0600cf2738cbd905c48476dbcca05dac5b5bbffc
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/2002969Reviewed-by: default avatarchromium-autoroll <chromium-autoroll@skia-public.iam.gserviceaccount.com>
Commit-Queue: chromium-autoroll <chromium-autoroll@skia-public.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#732030}
parent 961f1a5a
...@@ -1498,7 +1498,7 @@ deps = { ...@@ -1498,7 +1498,7 @@ deps = {
Var('chromium_git') + '/external/khronosgroup/webgl.git' + '@' + '4f3976e9b368ccfe7b9dd02014351936296dc72c', Var('chromium_git') + '/external/khronosgroup/webgl.git' + '@' + '4f3976e9b368ccfe7b9dd02014351936296dc72c',
'src/third_party/webrtc': 'src/third_party/webrtc':
Var('webrtc_git') + '/src.git' + '@' + 'b42aeaa3fb21d78e59c47d2a9916acb380494496', Var('webrtc_git') + '/src.git' + '@' + 'ccab06fb72be1fd8b165d865aefad9daeff8631f',
'src/third_party/libgifcodec': 'src/third_party/libgifcodec':
Var('skia_git') + '/libgifcodec' + '@'+ Var('libgifcodec_revision'), Var('skia_git') + '/libgifcodec' + '@'+ Var('libgifcodec_revision'),
......
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