Roll WebRTC from 1c5e63e54513 to 7acc2d9fe3a6 (48 revisions)

https://webrtc.googlesource.com/src.git/+log/1c5e63e54513..7acc2d9fe3a6

2021-01-14 mbonadei@webrtc.org Revert "Refactor rtc_base build targets."
2021-01-14 danilchap@webrtc.org Delete unused functions in ModuleRtpRtcpImpl
2021-01-14 auorion@microsoft.com Add ability to load CreateDirect3DDeviceFromDXGIDevice from d3d11.dll
2021-01-14 handellm@webrtc.org SetOfferedRtpHeaderExtensions: fix error code.
2021-01-14 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 189823ba75..42ab9dc8c8 (843026:843550)
2021-01-14 nisse@webrtc.org Update AudioDeviceBuffer to use C++ lambdas instead of rtc::Bind
2021-01-14 lndmrk@webrtc.org Include packetization in video codec string
2021-01-14 danilchap@webrtc.org Revert "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code"
2021-01-14 danilchap@webrtc.org Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code
2021-01-14 handellm@webrtc.org SetNegotiatedHeaderExtensions_w: Set list synchronously.
2021-01-14 sprang@webrtc.org Adds VideoDecoder::GetDecoderInfo()
2021-01-14 nisse@webrtc.org Delete unused VideoType enum values
2021-01-14 chunbo.hua@intel.com Fix typo in audio processing header.
2021-01-14 hta@webrtc.org Improve structuring of test for audio glitches.
2021-01-14 chunbo.hua@intel.com Fix AudioProcessing::Config::ToString() implementation.
2021-01-14 nisse@webrtc.org Add new empty build targets rtp_rtcp_legacy and video_legacy.
2021-01-14 landrey@webrtc.org Fix inconsistencies in network BUILD.gn file
2021-01-14 nisse@webrtc.org Change PeerConnectionE2EQualityTest to use lambdas instead of rtc::Bind
2021-01-14 alessiob@webrtc.org AGC2 minor code clean up
2021-01-13 jianj@google.com Add av1 to lower range IDs.
2021-01-13 natim@webrtc.org Reland "ChannelStatistics used for RTP stats in VoipStatistics."
2021-01-13 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision c0feffab8f..189823ba75 (842900:843026)
2021-01-13 mbonadei@webrtc.org Replace all uses of the word 'master' with 'builder_group' in //tools/mb
2021-01-13 sakal@webrtc.org Handle case createShader throws an exception.
2021-01-13 hta@webrtc.org Add a test to detect excessive audio delay during renegotiation.
2021-01-13 landrey@webrtc.org Fix cpu usage dvqa test on windows
2021-01-13 mbonadei@webrtc.org Remove CheckCompile method from MetaBuildWrapper.
2021-01-13 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision b24f1becea..c0feffab8f (842774:842900)
2021-01-13 webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Update WebRTC code version (2021-01-13T04:03:21).
2021-01-13 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision e35e978ee1..b24f1becea (842656:842774)
2021-01-12 aleloi@google.com Revert "ChannelStatistics used for RTP stats in VoipStatistics."
2021-01-12 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 0026868b2d..e35e978ee1 (842481:842656)
2021-01-12 natim@webrtc.org ChannelStatistics used for RTP stats in VoipStatistics.
2021-01-12 nisse@webrtc.org Revert "Add task queue to RtpRtcpInterface::Configuration."
2021-01-12 danilchap@webrtc.org Fill fps allocation by LibaomAv1Encoder::GetEncoderInfo
2021-01-12 terelius@webrtc.org Move logged types for RTC event log into event headers.
2021-01-12 ivoc@webrtc.org Correction for the calculation of the abs max value
2021-01-12 nisse@webrtc.org Add task queue to RtpRtcpInterface::Configuration.
2021-01-12 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision cbaaf9e4b2..0026868b2d (842367:842481)
2021-01-12 danilchap@webrtc.org Use rtc::CopyOnWriteBuffer::MutableData instead of data in fuzzers
2021-01-12 nisse@webrtc.org Delete KeepBufferRefs helpers, and use of rtc::Bind.
2021-01-12 nisse@webrtc.org Add thread annotation to IncomingVideoStream
2021-01-12 webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com Update WebRTC code version (2021-01-12T04:04:10).
2021-01-12 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 2059e3a759..cbaaf9e4b2 (842148:842367)
2021-01-11 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 78ea2c1bec..2059e3a759 (842019:842148)
2021-01-11 mbonadei@webrtc.org Refactor rtc_base build targets.
2021-01-11 philipel@webrtc.org Remove webrtc::VideoDecoder::PrefersLateDecoding.
2021-01-11 mbonadei@webrtc.org [build] Remove obsolete gn flag

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/+doc/master/autoroll/README.md

Bug: chromium:1051821,chromium:1095763,chromium:1117773,chromium:1146835,chromium:1158070,chromium:1161837,chromium:1164276
Tbr: webrtc-chromium-sheriffs-robots@google.com
Change-Id: I0b695f5946acc1216777701b894bfa60d78c8d37
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/2630487
Reviewed-by: chromium-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@{#843809}