All graphs for this bug:
  https://chromeperf.appspot.com/group_report?bug_id=827080

(For debugging:) Original alerts at time of bug-filing:
  https://chromeperf.appspot.com/group_report?sid=cf448e8b75ebb598b512743259d144672803a8b9d76ca41cdd0b581bc5f9c164


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

webrtc-android-tests-nexus4-lollipop

This shows a large change in WebRTCPerf/webrtc-android-tests-nexus4-lollipop/webrtc_perf_tests / ramp-up-rtx-media-sent / TOffsetSimulcastRedRtx, which seems correlated with my cl #22595 'Reland "Move rtp-specific config out of EncoderSettings."'

I'm having some difficulty seeing how that could cause an android-only change, though. Other nearby cls are #22594 'Reduce implementations of CallStatsObserver by 2.' and #22596 'Disable FieldTrials/CallOperationEndToEndTest.TransmitsFirstFrame/2 due to flakiness.'

Stefan, do you have any guidance?

Not really, but try to reproduce on a device perhaps? Probably the fastest way forward. Otherwise speculative revert...

BTW, cc the authors of all cls and get their input.

The CallStatsObserver change removes a couple of vtables and reduces the size of an observer vector down from ~4 to ~2 (+fewer calls to register/unregister).

Is the same change perhaps to blame/thank for this improvement? (any chance that being at 0 is a bug?)

https://chromeperf.appspot.com/report?sid=b49ab6f38aefc61d32ec982fd09350008fe671c3c9ff6b8f910f2f4bbdd244d0&rev=22596

Is CallStatsObserver involved in anyway in the measurements for this perf test?

Ah, and it seems that the dashboard pages initially showed me only one of several graphs. When expanding the others, it looks like ramp-up-media-sent goes down and ramp-up-rtx-media-sent goes up by roughly the same amount. And it affects all platforms.

So there's definitely some change in behavior (or possibly in measurements). There's also one graph showing reduced ramp up time.

I don't know to be sure. It does affect all platforms (but it should really be a no-op change as far as this goes). Have you tried running this locally to see if there's a change with/without the CallStatsObserver change or the EncoderSettings change?

To save folks from looking up the two CLs, here they are:
EncoderSettings: https://webrtc-review.googlesource.com/c/src/+/63721
CallStatsObserver: https://webrtc-review.googlesource.com/c/src/+/64524

The CSO change landed slightly before and there's one datapoint we have for that CL that indicates no change.  Do we have a datapoint in isolation for the EncoderSettings change? I see it's a re-land, so maybe we do?

Looks like we do here:
https://webrtc.googlesource.com/src/+log/08006d4133d50c3bc7f8fe70b78ea693b5a97e9c..08126349f5cc5d682a3398014a45088b4bb310f5

In the graph, I don't see any change there, so then we have one data point for each CL in isolation and they both indicate no change :)

I'll attempt to revert my change (and then I'll leave for Easter and be back on Thursday). I got a few merge conflicts with 

  https://webrtc-review.googlesource.com/55264
  https://webrtc-review.googlesource.com/60340

but hopefully not so hard to get right.

Revert cl: https://webrtc-review.googlesource.com/c/src/+/65520/2

I've looked at what happened with the graph when I originally landed my cl, #22532, reverted as #22537, and there are corresponding spikes. So it seems quitelikely that it's my cl.

The following revision refers to this bug:
  https://webrtc.googlesource.com/src.git/+/6c2c13af06b32778b86950681758a7970d1c5d9e

commit 6c2c13af06b32778b86950681758a7970d1c5d9e
Author: Niels Möller <nisse@webrtc.org>
Date: Thu Mar 29 11:45:18 2018

Revert "Reland "Move rtp-specific config out of EncoderSettings.""

This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.

Reason for revert: Regression in ramp up perf tests.

Original change's description:
> Reland "Move rtp-specific config out of EncoderSettings."
>
> This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
>
> Original change's description:
> > Move rtp-specific config out of EncoderSettings.
> >
> > In VideoSendStream::Config, move payload_name and payload_type from
> > EncoderSettings to Rtp.
> >
> > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > and should perhaps be renamed in a follow up cl. It's no longer
> > passed as an argument to VideoCodecInitializer::SetupCodec.
> >
> > The latter then needs a different way to know the codec type,
> > which is provided by a new codec_type member in VideoEncoderConfig.
> >
> > Bug:  webrtc:8830 
> > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22532}
>
> Bug:  webrtc:8830 
> Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> Reviewed-on: https://webrtc-review.googlesource.com/63721
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22595}

TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org

Bug:  webrtc:8830 , chromium:827080 
Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
Reviewed-on: https://webrtc-review.googlesource.com/65520
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22677}
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/call/bitrate_estimator_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/call/call.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/call/rampup_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/call/video_config.h
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/call/video_send_stream.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/call/video_send_stream.h
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/media/engine/fakewebrtccall.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/media/engine/webrtcvideoengine.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/media/engine/webrtcvideoengine_unittest.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/modules/video_coding/include/video_codec_initializer.h
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/modules/video_coding/video_codec_initializer.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/modules/video_coding/video_codec_initializer_unittest.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/test/call_test.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/test/encoder_settings.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/test/encoder_settings.h
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/encoder_rtcp_feedback_unittest.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/end_to_end_tests/codec_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/end_to_end_tests/extended_reports_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/end_to_end_tests/fec_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/end_to_end_tests/histogram_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/end_to_end_tests/log_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/end_to_end_tests/multi_stream_tester.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/end_to_end_tests/retransmission_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/end_to_end_tests/rtp_rtcp_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/picture_id_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/replay.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/send_statistics_proxy.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/video_quality_test.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/video_send_stream.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/video_send_stream_tests.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/video_stream_encoder.cc
[modify] https://crrev.com/6c2c13af06b32778b86950681758a7970d1c5d9e/video/video_stream_encoder_unittest.cc

Graphs look recovering after revert.

The following revision refers to this bug:
  https://chromium.googlesource.com/chromium/src.git/+/99fe96fc2e9a19d26e85756f203e5cb09e8ac0b4

commit 99fe96fc2e9a19d26e85756f203e5cb09e8ac0b4
Author: webrtc-chromium-autoroll@skia-buildbots.google.com.iam.gserviceaccount.com <webrtc-chromium-autoroll@skia-buildbots.google.com.iam.gserviceaccount.com>
Date: Thu Mar 29 18:35:19 2018

Roll src/third_party/webrtc/ 7abc9a07d..a21090b77 (15 commits)

https://webrtc.googlesource.com/src.git/+log/7abc9a07d774..a21090b7706e

$ git log 7abc9a07d..a21090b77 --date=short --no-merges --format='%ad %ae %s'

Created with:
  roll-dep src/third_party/webrtc
BUG= chromium:827080 , chromium:827101 ,chromium:None


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=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng;master.tryserver.chromium.win:win-msvc-dbg
TBR=webrtc-chromium-sheriffs-robots@google.com

Change-Id: I26226844a15fd3ed25ded98c0563fbeb5bd2ca3e
Reviewed-on: https://chromium-review.googlesource.com/986413
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@{#546870}
[modify] https://crrev.com/99fe96fc2e9a19d26e85756f203e5cb09e8ac0b4/DEPS

I've found one difference, starting from the logs from 

./out/Release/webrtc_perf_tests --gtest_filter='RampUpTest.TransportSequenceNumberSimulcastRedRtx' --logs

before and after my change. My change results in one new log line:

(video_codec_initializer.cc:187): No temporal layers and nack enabled -> resilience off

If we look at CallTest::CreateVideoSendConfig, it uses the payload name "fake". Before my change, this trickles down to VideoCodecInitializer::SetupCodec, which translates that to kVideoCodecGeneric.

After my change, CallTest::CreateVideoSendConfig, via a new argument to FillEncoderConfiguration, sets VideoEncoderConfig::codec_type more correctly, to kVideoCodecVP8. This enables the VP8-specific logic in VideoCodecInitializer::SetupCodec and its helpers. I'm not sure where VP8 settings were selected before my change, but I guess it has the effect of changing |resilience| from kResilientStream to kResilienceOff.

Unclear if that explains all the perf alerts, though. Why did we use "FAKE" as payload name in the first place?

FAKE should have been used when we used a fake encoder and not a real VP8 encoder.

Ok, I'm changing kVideoCodecVP8 to kVideoCodecGeneric in ps#3 of the reland cl https://webrtc-review.googlesource.com/c/src/+/66862/2..3

Seems to restore the old behavior in local testing of RampUpTest.TransportSequenceNumberSimulcastRedRtx.

Master:

RESULT ramp-up-media-sent: TransportSequenceNumberSimulcastRedRtx= 17870 bytes
RESULT ramp-up-padding-sent: TransportSequenceNumberSimulcastRedRtx= 0 bytes
RESULT ramp-up-rtx-media-sent: TransportSequenceNumberSimulcastRedRtx= 3804 bytes
RESULT ramp-up-rtx-padding-sent: TransportSequenceNumberSimulcastRedRtx= 672 bytes
RESULT ramp-up-time: TransportSequenceNumberSimulcastRedRtx= 201 milliseconds
RESULT ramp-up-average-network-latency: TransportSequenceNumberSimulcastRedRtx= 0 milliseconds
[       OK ] RampUpTest.TransportSequenceNumberSimulcastRedRtx (211 ms)

Reland cl ps#2 fix: 

RESULT ramp-up-media-sent: TransportSequenceNumberSimulcastRedRtx= 14090 bytes
RESULT ramp-up-padding-sent: TransportSequenceNumberSimulcastRedRtx= 0 bytes
RESULT ramp-up-rtx-media-sent: TransportSequenceNumberSimulcastRedRtx= 1902 bytes
RESULT ramp-up-rtx-padding-sent: TransportSequenceNumberSimulcastRedRtx= 0 bytes
RESULT ramp-up-time: TransportSequenceNumberSimulcastRedRtx= 161 milliseconds
RESULT ramp-up-average-network-latency: TransportSequenceNumberSimulcastRedRtx= 0 milliseconds
[       OK ] RampUpTest.TransportSequenceNumberSimulcastRedRtx (172 ms)

Reland cl ps #3:

RESULT ramp-up-media-sent: TransportSequenceNumberSimulcastRedRtx= 17870 bytes
RESULT ramp-up-padding-sent: TransportSequenceNumberSimulcastRedRtx= 0 bytes
RESULT ramp-up-rtx-media-sent: TransportSequenceNumberSimulcastRedRtx= 3804 bytes
RESULT ramp-up-rtx-padding-sent: TransportSequenceNumberSimulcastRedRtx= 672 bytes
RESULT ramp-up-time: TransportSequenceNumberSimulcastRedRtx= 201 milliseconds
RESULT ramp-up-average-network-latency: TransportSequenceNumberSimulcastRedRtx= 0 milliseconds
[       OK ] RampUpTest.TransportSequenceNumberSimulcastRedRtx (211 ms)

The following revision refers to this bug:
  https://webrtc.googlesource.com/src.git/+/259a497632eabad68d5d9aeaea31d5a7a2e12897

commit 259a497632eabad68d5d9aeaea31d5a7a2e12897
Author: Niels Möller <nisse@webrtc.org>
Date: Thu Apr 05 14:30:09 2018

Reland "Reland "Move rtp-specific config out of EncoderSettings.""

This reverts commit 6c2c13af06b32778b86950681758a7970d1c5d9e.

Reason for revert: Intend to investigate and fix perf problems.

Original change's description:
> Revert "Reland "Move rtp-specific config out of EncoderSettings.""
> 
> This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.
> 
> Reason for revert: Regression in ramp up perf tests.
> 
> Original change's description:
> > Reland "Move rtp-specific config out of EncoderSettings."
> >
> > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
> >
> > Original change's description:
> > > Move rtp-specific config out of EncoderSettings.
> > >
> > > In VideoSendStream::Config, move payload_name and payload_type from
> > > EncoderSettings to Rtp.
> > >
> > > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > > and should perhaps be renamed in a follow up cl. It's no longer
> > > passed as an argument to VideoCodecInitializer::SetupCodec.
> > >
> > > The latter then needs a different way to know the codec type,
> > > which is provided by a new codec_type member in VideoEncoderConfig.
> > >
> > > Bug:  webrtc:8830 
> > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22532}
> >
> > Bug:  webrtc:8830 
> > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> > Reviewed-on: https://webrtc-review.googlesource.com/63721
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22595}
> 
> TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
> 
> Bug:  webrtc:8830 , chromium:827080 
> Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
> Reviewed-on: https://webrtc-review.googlesource.com/65520
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22677}

TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug:  webrtc:8830 ,  chromium:827080 
Change-Id: I9b62987bf5daced90dfeb3ebb6739c80117c487f
Reviewed-on: https://webrtc-review.googlesource.com/66862
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22751}
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/call/bitrate_estimator_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/call/call.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/call/rampup_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/call/video_config.h
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/call/video_send_stream.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/call/video_send_stream.h
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/media/engine/fakewebrtccall.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/media/engine/webrtcvideoengine.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/media/engine/webrtcvideoengine_unittest.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/modules/video_coding/include/video_codec_initializer.h
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/modules/video_coding/video_codec_initializer.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/modules/video_coding/video_codec_initializer_unittest.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/test/call_test.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/test/encoder_settings.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/test/encoder_settings.h
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/encoder_rtcp_feedback_unittest.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/end_to_end_tests/codec_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/end_to_end_tests/extended_reports_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/end_to_end_tests/fec_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/end_to_end_tests/histogram_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/end_to_end_tests/log_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/end_to_end_tests/multi_stream_tester.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/end_to_end_tests/retransmission_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/end_to_end_tests/rtp_rtcp_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/picture_id_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/replay.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/send_statistics_proxy.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/video_quality_test.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/video_send_stream.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/video_send_stream_tests.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/video_stream_encoder.cc
[modify] https://crrev.com/259a497632eabad68d5d9aeaea31d5a7a2e12897/video/video_stream_encoder_unittest.cc

The following revision refers to this bug:
  https://chromium.googlesource.com/chromium/src.git/+/4123a5e9892f0353f62fe5b80a00f634ca891839

commit 4123a5e9892f0353f62fe5b80a00f634ca891839
Author: webrtc-chromium-autoroll@skia-buildbots.google.com.iam.gserviceaccount.com <webrtc-chromium-autoroll@skia-buildbots.google.com.iam.gserviceaccount.com>
Date: Fri Apr 06 06:32:58 2018

Roll src/third_party/webrtc/ 4c9b3c840..19aab2ee7 (16 commits)

https://webrtc.googlesource.com/src.git/+log/4c9b3c840df3..19aab2ee7c6b

$ git log 4c9b3c840..19aab2ee7 --date=short --no-merges --format='%ad %ae %s'

Created with:
  roll-dep src/third_party/webrtc
BUG=chromium:None,chromium:812974,chromium:827080,chromium:None,chromium:794210,chromium:825524


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=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng;master.tryserver.chromium.win:win-msvc-dbg
TBR=webrtc-chromium-sheriffs-robots@google.com

Change-Id: If6b26fb8305b2e0d8c796358684d52a63d181a77
Reviewed-on: https://chromium-review.googlesource.com/999352
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@{#548696}
[modify] https://crrev.com/4123a5e9892f0353f62fe5b80a00f634ca891839/DEPS

Relanded as #22751, and graphs look ok. Closing.