All graphs for this bug:
  https://chromeperf.appspot.com/group_report?bug_id=861721

(For debugging:) Original alerts at time of bug-filing:
  https://chromeperf.appspot.com/group_report?sid=00948f466c8182156d161b905865c46b20c99f26115a4528f6e360fd321a055b


Bot(s) for this bug's original alert(s):

webrtc-android-tests-nexus5
webrtc-android-tests-nexus72

Ping, there seems to be a lot of changes related to https://webrtc.googlesource.com/src/+/5e898d612e7f221a259a73b1f4f78a9f2fe8ec5e

This appears to be test cases that are behaving in a non-standard way.
Created https://webrtc-review.googlesource.com/c/src/+/88224

The following revision refers to this bug:
  https://webrtc.googlesource.com/src.git/+/28bb391918e594fd828628b938f97a0a87b220b8

commit 28bb391918e594fd828628b938f97a0a87b220b8
Author: Erik Språng <sprang@webrtc.org>
Date: Wed Jul 11 15:14:19 2018

Fix incorrect screenshare bitrate configs in test cases

A number of full stack tests had screenshare configured at max 2000kbps,
which differs from the 1000kbps we use in production. This CL fixes that
plus a few incorrectly set simulcast bitrates.

It also removes obsolete screenshare ALR tests (ALR is not default-on
for screenshare).

Additionally, it updates the loopback tests to not set a max cap for the
bandwidth estiamte.


Bug:  chromium:861721 
Change-Id: I87b9dd40f12bfc0358ccb8f8f044a6b5e51e1af5
Reviewed-on: https://webrtc-review.googlesource.com/88224
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23935}
[modify] https://crrev.com/28bb391918e594fd828628b938f97a0a87b220b8/video/full_stack_tests.cc
[modify] https://crrev.com/28bb391918e594fd828628b938f97a0a87b220b8/video/screenshare_loopback.cc
[modify] https://crrev.com/28bb391918e594fd828628b938f97a0a87b220b8/video/video_loopback.cc

The following revision refers to this bug:
  https://chromium.googlesource.com/chromium/src.git/+/2b0529974fae81606d19de5722d21320de4678e8

commit 2b0529974fae81606d19de5722d21320de4678e8
Author: webrtc-chromium-autoroll <webrtc-chromium-autoroll@skia-buildbots.google.com.iam.gserviceaccount.com>
Date: Thu Jul 12 02:51:41 2018

Roll src/third_party/webrtc ef43aafcf595..d78323faba86 (47 commits)

https://webrtc.googlesource.com/src.git/+log/ef43aafcf595..d78323faba86


git log ef43aafcf595..d78323faba86 --date=short --no-merges --format='%ad %ae %s'
2018-07-11 steveanton@webrtc.org Remove AddTrack override with MediaStreams
2018-07-11 buildbot@webrtc.org Roll chromium_revision c797a4c559..2e3b05bf93 (574101:574261)
2018-07-11 danilchap@webrtc.org Reland "Remove rtc::Optional alias and api:optional target"
2018-07-11 tkchin@webrtc.org Clear interrupted flag on CallKit audio activation.
2018-07-11 yujo@chromium.org Remove not-updating-stats log message.
2018-07-11 qingsi@google.com Revert "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-11 fbarchard@google.com Include libyuv/planar_functions.h for CopyPlane
2018-07-11 stefan@webrtc.org Moved PayloadRouter to call/.
2018-07-11 sprang@webrtc.org Fix incorrect screenshare bitrate configs in test cases
2018-07-11 magjed@webrtc.org Android: Remove use of EGLContexts in PeerConnectionFactory
2018-07-11 magjed@webrtc.org VCMPacket: Set VCMNaluCompleteness for generic codecs
2018-07-11 mbonadei@webrtc.org Removing unneeded warning suppression flags from audio_device.
2018-07-11 yura.yaroshevich@gmail.com Fixed crash when PCF is destroyed before RTCRtpReceiver in ObjC
2018-07-11 srte@webrtc.org Removes unnecessary destructor checks in tests.
2018-07-11 hbos@webrtc.org RtpReceiverInterface::stream_ids() added.
2018-07-11 mbonadei@webrtc.org Remove -Wno-reorder from examples/BUILD.gn.
2018-07-11 andersc@webrtc.org Fix null check in CricketToWebRtcVideoDecoderFactory
2018-07-11 mbonadei@webrtc.org Add missing dependency on //third_party/abseil-cpp/absl/memory.
2018-07-11 buildbot@webrtc.org Roll chromium_revision 916138ba7e..c797a4c559 (573665:574101)
2018-07-11 mbonadei@webrtc.org Removing -Wno-sentinel from examples/BUILD.gn.
2018-07-11 mbonadei@webrtc.org Add missing dependencies on //third_party/abseil-cpp/absl/memory.
2018-07-11 asapersson@webrtc.org Add unit tests for simulcast layer configurations.
2018-07-11 ilnik@webrtc.org Revert "Remove rtc::Optional alias and api:optional target"
2018-07-11 qingsi@webrtc.org Revert "Add Profile 2 configuration to VP9 Encoder and Decoder"
2018-07-11 qingsi@google.com Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
2018-07-11 emircan@webrtc.org Add Profile 2 to default supported profiles
2018-07-10 emircan@webrtc.org Add Profile 2 configuration to VP9 Encoder and Decoder
2018-07-10 qingsi@google.com Revert "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
2018-07-10 ouj@fb.com Add unittest for default severity level of RTCCallbackLogger
2018-07-10 qingsi@google.com Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
2018-07-10 danilchap@webrtc.org Remove rtc::Optional alias and api:optional target
2018-07-10 magjed@webrtc.org Android: Fix QueryVideoEncoder() for H264 High Profile
2018-07-10 philipel@webrtc.org Actually input packets into the PacketBuffer.
2018-07-10 alessiob@webrtc.org Reland "IWYU: Add <cmath> for fabsf() and powf(), switch to C++ versions"
2018-07-10 magjed@webrtc.org Android: Honor disabling legacy video HW codec acceleration
2018-07-10 saza@webrtc.org Fix fuzzer-found flow-over in AGC1
2018-07-10 mbonadei@webrtc.org Removing -Wno-microsoft-extra-qualification.
2018-07-10 mbonadei@webrtc.org Removing rtc_unittests warning suppression flags.
2018-07-10 mbonadei@webrtc.org Remove -Wno-sign-compare, -Wno-unused-variable and -Wno-format.
2018-07-10 philipel@webrtc.org Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file.
2018-07-10 srte@webrtc.org Moves VideoAnalyzer class to a separate file.
2018-07-10 srte@webrtc.org Removes unnecessary destructor checks in tests.
2018-07-10 buildbot@webrtc.org Roll chromium_revision ef3275516a..916138ba7e (573561:573665)
2018-07-10 ouj@fb.com Regenerate mock peer connection to add missing mock methods.
2018-07-10 sakal@webrtc.org Update examples to use the new VideoCapturer interface.
2018-07-10 ouj@fb.com Make the default severity level of RTCCallbackLogger match the comment on its header.
2018-07-10 andersc@webrtc.org Migrate legacy Android video codec factories.


Created with:
  gclient setdep -r src/third_party/webrtc@d78323faba86

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:861721,chromium:None,chromium:None,chromium:None,chromium:none,chromium:819294,chromium:860638,chromium:none,chromium:None,chromium:None
TBR=webrtc-chromium-sheriffs-robots@google.com

Change-Id: I76bc9222cb5ac91beb290da2d1740e35a029679c
Reviewed-on: https://chromium-review.googlesource.com/1134448
Reviewed-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@{#574470}
[modify] https://crrev.com/2b0529974fae81606d19de5722d21320de4678e8/DEPS

Is this finished?