Roll WebRTC from 3444a49d77b0 to e88c95e516b6 (28 revisions)

https://webrtc.googlesource.com/src.git/+log/3444a49d77b0..e88c95e516b6

2020-07-08 hbos@webrtc.org [Stats] Add more rtc::Thread::ScopedDisallowBlockingCalls to getStats().
2020-07-08 tommi@webrtc.org Remove unused GuardedAsyncInvoker class
2020-07-08 handellm@webrtc.org Revert "Migrate test/time_controller to webrtc::Mutex."
2020-07-08 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision b5359525..a69b9c61 (785857:786202)
2020-07-08 hbos@webrtc.org [Stats] Optimization: Minimize number of thread-invokes in getStats().
2020-07-08 mbonadei@webrtc.org Revert "Implement packets_(sent | received) for RTCTransportStats"
2020-07-08 handellm@webrtc.org Migrate modules/audio_device to webrtc::Mutex.
2020-07-08 handellm@webrtc.org VideoAnalyzer: remove lock recursions.
2020-07-08 handellm@webrtc.org Migrate test/pc/e2e to webrtc::Mutex.
2020-07-08 mbonadei@webrtc.org Add possibility to expand strings in the autoroller.
2020-07-08 handellm@webrtc.org Migrate test/time_controller to webrtc::Mutex.
2020-07-08 handellm@webrtc.org Migrate modules/congestion_controller to webrtc::Mutex.
2020-07-08 titovartem@webrtc.org Fix thread usage in PC level tests for getting to the IceConnected state
2020-07-08 handellm@webrtc.org Logging: reduce locking frequency by fixing TODO.
2020-07-07 handellm@webrtc.org Revert "Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex."
2020-07-07 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision fa6a9612..b5359525 (785707:785857)
2020-07-07 handellm@webrtc.org Migrate pc/ to webrtc::Mutex.
2020-07-07 handellm@webrtc.org Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
2020-07-07 handellm@webrtc.org Migrate modules/remote_bitrate_estimator to webrtc::Mutex.
2020-07-07 handellm@webrtc.org Migrate video/adaptation and video/end_to_end_tests to webrtc::Mutex.
2020-07-07 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 5f083e73..fa6a9612 (785559:785707)
2020-07-07 handellm@webrtc.org Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex.
2020-07-07 handellm@webrtc.org Migrate media/ to webrtc::Mutex.
2020-07-07 handellm@webrtc.org Migrate common_video/ and examples/ to webrtc::Mutex.
2020-07-07 ivoc@webrtc.org Update default max nr of packets to 200.
2020-07-07 handellm@webrtc.org Migrate modules/rtp_rtcp to webrtc::Mutex.
2020-07-07 handellm@webrtc.org Migrate modules/video_coding to webrtc::Mutex.
2020-07-07 titovartem@google.com Implement packets_(sent | received) for RTCTransportStats

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: None
Tbr: webrtc-chromium-sheriffs-robots@google.com
Change-Id: Ida1605b68ce1bb4c568c7fddee238f84aeb98aca
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/2287775Reviewed-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@{#786300}

DEPS

@@ -1453,7 +1453,7 @@ deps = {
   },
 
   'src/third_party/webrtc':
-    Var('webrtc_git') + '/src.git' + '@' + '3444a49d77b094c7fa2aade8873288a55ac0b9e5',
+    Var('webrtc_git') + '/src.git' + '@' + 'e88c95e516b6882be1c61b3a9eb334e55ff70e07',
 
   'src/third_party/libgifcodec':
      Var('skia_git') + '/libgifcodec' + '@'+  Var('libgifcodec_revision'),