Roll WebRTC from e39b378d4a92 to 59f1d1e36d26 (49 revisions)

https://webrtc.googlesource.com/src.git/+log/e39b378d4a92..59f1d1e36d26

2020-09-30 alessiob@webrtc.org AGC2 adaptive digital controller config: new params
2020-09-30 alessiob@webrtc.org AGC2 adaptive digital controller config: new param
2020-09-30 alessiob@webrtc.org AGC2: gain increase allowed once enough adjacent speech frames observed
2020-09-30 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision f9788a96..97f841d7 (811959:812293)
2020-09-30 mbonadei@webrtc.org Revert "Reduce the amount of howling reduction in AEC3"
2020-09-30 hta@webrtc.org Add a test for transceivers being removed when stopped.
2020-09-30 peah@webrtc.org Revert "Activating AVX2 support by default"
2020-09-30 peah@webrtc.org Reduce the amount of howling reduction in AEC3
2020-09-30 handellm@webrtc.org Mutex: remove mutex reentrancy crasher.
2020-09-30 gustaf@webrtc.org Remove unused enums and members
2020-09-30 alessiob@webrtc.org AGC2 Saturation Protector always on
2020-09-30 nisse@webrtc.org Delete stringize macros.
2020-09-30 peah@webrtc.org Revert "Clean up the AVX2 activation in the gni file"
2020-09-30 peah@webrtc.org Revert "Deactivating AVX2 support by default"
2020-09-30 alessiob@webrtc.org AGC2 AdaptiveDigitalGainApplier and AdaptiveAgc code improvements
2020-09-30 alessiob@webrtc.org AGC2 AdaptiveModeLevelEstimator min consecutive speech frames (3/3)
2020-09-30 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 01b26f76..f9788a96 (811724:811959)
2020-09-30 mbonadei@webrtc.org Revert "Begin implementing WGC CaptureFrame"
2020-09-30 alessiob@webrtc.org AGC2 AdaptiveModeLevelEstimator: minor code improvements
2020-09-30 mbonadei@webrtc.org Introduce RTC_NO_UNIQUE_ADDRESS.
2020-09-30 yunz@fb.com Don't trigger key frame when encoder is not reset during reconfigure
2020-09-30 kwiberg@webrtc.org Optimize RoboCaller::AddReceiver() for code size
2020-09-30 philipp.hancke@googlemail.com android: add rollback RTCSdpType
2020-09-29 henrik.lundin@webrtc.org Improve neteq_rtp_fuzzer
2020-09-29 perkj@webrtc.org Ensure FakeVp8Encoder::GetEncoderInfo() writes EncoderInfo.fps_allocation:
2020-09-29 auorion@microsoft.com Begin implementing WGC CaptureFrame
2020-09-29 philipp.hancke@googlemail.com red: ensure minimum amount of header bytes
2020-09-29 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 82e79a8a..01b26f76 (811607:811724)
2020-09-29 alessiob@webrtc.org AGC2 AdaptiveModeLevelEstimator: cache last level estimate
2020-09-29 jleconte@webrtc.org Create isolated output directory when creating the output file.
2020-09-29 hta@webrtc.org Break out separate compile targets for various classes
2020-09-29 titovartem@google.com Report sent_packets_queue_wait_time_us in PC level framework network debug mode
2020-09-29 titovartem@google.com Improve perf metrics plotter
2020-09-29 sprang@webrtc.org Default-enables WebRTC-DeferredFecGeneration.
2020-09-29 peah@webrtc.org Deactivating AVX2 support by default
2020-09-29 alessiob@webrtc.org AGC2 remove incorrect field trial parsing functions
2020-09-29 hta@webrtc.org Factor out the transceiver list into a separate object.
2020-09-29 tommi@webrtc.org Remove locks from BufferQueue (not needed).
2020-09-29 sprang@webrtc.org Deferred FEC: Prevents duplicate FEC addition of non-RTX retransmission.
2020-09-29 alessiob@webrtc.org AGC2 AdaptiveModeLevelEstimator min consecutive speech frames (2/3)
2020-09-29 ilnik@webrtc.org Add NV12 to libvpx wrappers output
2020-09-29 nisse@webrtc.org Delete macros RTC_DISALLOW_ASSIGN and RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
2020-09-29 gustaf@webrtc.org Remove echo suppression in transparent mode
2020-09-29 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 8276edf4..82e79a8a (811503:811607)
2020-09-29 nisse@webrtc.org Delete unused header file sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h
2020-09-29 eshr@google.com Accept NV12 frames into VP9
2020-09-29 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 5db2d14d..8276edf4 (811347:811503)
2020-09-28 natim@webrtc.org Start/stop ProcessThread as first/last audio channel is added/removed.
2020-09-28 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 3274067a..5db2d14d (811217:811347)

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:980879
Tbr: webrtc-chromium-sheriffs-robots@google.com
Change-Id: I6c6ff15eb7379088c467c8b3248780910ccd09a4
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/2440992Reviewed-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@{#812627}

DEPS

@@ -1486,7 +1486,7 @@ deps = {
   },
 
   'src/third_party/webrtc':
-    Var('webrtc_git') + '/src.git' + '@' + 'e39b378d4a921e9982f39ebb02a6145afb653956',
+    Var('webrtc_git') + '/src.git' + '@' + '59f1d1e36d262cc3dfbadcd8d0214fffb4664676',
 
   'src/third_party/libgifcodec':
      Var('skia_git') + '/libgifcodec' + '@'+  Var('libgifcodec_revision'),