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}
This is part of a series of CLs adding a network controller based on
the BBR congestion control method. The code is based on the QUIC BBR
implementation in Chromium.
Bug: webrtc:8415
Change-Id: I4478e8d5e2abd361b0d22de00ebe04bde9ef18f6
Reviewed-on: https://webrtc-review.googlesource.com/63681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22592}
This class is completely new and tracks data transfer in a slightly
different way compared to the BBR implementation in QUIC. The
fundamental change is that receive time is used rather than packet
index to identify the packets over which data rates should be
calculated.
This is part of a series of CLs adding a network controller based on
the BBR congestion control method. The code is based on the QUIC BBR
implementation in Chromium.
Bug: webrtc:8415
Change-Id: I9d1f12634073ac89c4d542f965e3677a89a1526c
Reviewed-on: https://webrtc-review.googlesource.com/63680
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22589}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
NOPRESUBMIT=true
Change-Id: Ie2879aca5fc1667e4222499d2a8fc2bba9ae2425
Reviewed-on: https://webrtc-review.googlesource.com/21328
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22587}
The neteq_rtpplay tool can crash when the replacement audio file is too short. The desired behavior is that the audio file is looped as much as necessary.
Bug: webrtc:9061
Change-Id: Iefba4c47271584845662a415598bf2197dba0fae
Reviewed-on: https://webrtc-review.googlesource.com/64460
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22585}
This also has the benefit of deleting one unneeded call to
RTPPayloadRegistry::last_received_media_payload_type.
To make this work, also extend NackModule with a OnReceivedPacket
method taking only the sequence number and the is_keyframe flag,
rather than a complete VCMPacket.
Bug: webrtc:8995
Change-Id: Ice379581166e7b1609ec719e944a5a543d69acc1
Reviewed-on: https://webrtc-review.googlesource.com/64120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22584}
In this CL the OnFrame function is implemented.
Bug: webrtc:8909
Change-Id: I68488a033e86eadd0b16d091faad14e9cda7cc36
Reviewed-on: https://webrtc-review.googlesource.com/64121
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22583}
This is part of a series of CLs adding a network controller based on
the BBR congestion control method. The code is based on the QUIC BBR
implementation in Chromium.
Bug: webrtc:8415
Change-Id: Ib952b9c3f1cdd0741330a5ae95aebcf2488f7f1d
Reviewed-on: https://webrtc-review.googlesource.com/63644
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22581}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I440974da4d347b09ff042478720d7983056b62b9
Reviewed-on: https://webrtc-review.googlesource.com/21226
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22579}
Extend TestAudioDeviceModule API to be able to create WavReader/WavWriter from rtc::PlatformFile
Bug: webrtc:8946
Change-Id: Ieea16be91c40a5928689cdeaa8a17d75fee0cf82
Reviewed-on: https://webrtc-review.googlesource.com/63266
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22576}
This CL adds the ability to configure RTPSender to include the
MID header extension when sending packets. The MID will be
included on every packet at the start of the stream until an RTCP
acknoledgment is received for that SSRC at which point it will
stop being included. The MID will be included on regular RTP
streams as well as RTX streams.
Bug: webrtc:4050
Change-Id: Ie27ebee1cd00a67f2b931f5363788f523e3e684f
Reviewed-on: https://webrtc-review.googlesource.com/60582
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22574}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
Change-Id: I51dfe8879c28c91bd1c667fc47b4892373671e0f
Reviewed-on: https://webrtc-review.googlesource.com/21540
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22569}
The estimation on how well the linear filter in the AEC3 is performing
is done through an estimation of the ERLE. That estimation is then
used for knowing how much the suppressor needs to react in order to
cancel all the echoes.
In the current code, the ERLE is quite conservative during farend
inactivity and it is common that it goes to a minimum value during
those periods. Under highly varying conditions, that is probably the
right approach. However, in other scenarios where conditions does not
change that fast there is a loss in transparency that could be avoided
by means of a different ERLE estimation.
In the current CL, the ERLE estimation has been changed in the
following way:
- During farend activity the ERLE is estimated through a 1st order AR
smoother. This smoother goes faster toward lower ERLE values than to
larger ones in order to avoid overestimation of this
value. Furthermore, during the beginning of the farend burst, an
estimation of the ERLE is done that aim to represent the performance
of the linear filter during onsets. Under highly variant environments,
those quantities, the ERLE during onsets and the one computed during
the whole farend duration, would differ a lot. If the environment is
more stationary, those quantities would be much more similar.
- During nearend activity the ERLE estimation is decreased toward a
value of the ERLE during onsets.
Bug: webrtc:9040
Change-Id: Ieab86370a4333d2d0cd7041047d29651de4f6827
Reviewed-on: https://webrtc-review.googlesource.com/62342
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22568}
This CL adds a function to the SendSideCongestionController interface
for reporting per packet feedback availability.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: Ifcb6837bb80c5fcfc1f12f4f93ec38cc2903118f
Reviewed-on: https://webrtc-review.googlesource.com/63205
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22565}
This CL adds robustifications for avoiding that the headset mode
is triggered for reverberant or weak echo paths.
Bug: webrtc:9047,chromium:824111,webrtc:8314,webrtc:8671,webrtc:5201,webrtc:5919
Change-Id: Ib111e617f765377c021a5b633cf13a7917fe62a6
Reviewed-on: https://webrtc-review.googlesource.com/64002
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22557}
First step of the transition needed to reland cl
https://webrtc-review.googlesource.com/62062, and move payload_name
and payload_type out of VideoSendStream::Config::EncoderSettings.
If the new field is set to something different than kVideoCodecUnkown,
payload_name from EncoderSettings is ignored.
Bug: webrtc:8830
Change-Id: I515a91f8291cda79017332102cc6a10736d8a648
Reviewed-on: https://webrtc-review.googlesource.com/64001
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22555}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
Bug: webrtc:8445
NOPRESUBMIT=true
Change-Id: I30d01fcb9cbe1427a7703a3cdd7befae751066b5
Reviewed-on: https://webrtc-review.googlesource.com/21982
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22550}
This CL robustifies the echo removal behavior when headsets are used.
In particular it:
-Introduces a secondary, more refined alignment when no alignment can
be found using the delay estimator.
-Changes decision logic for when to use the linear filter output.
-Changes the decision logic for when to be transparent.
-Changes the way that the transparent mode works.
-Makes the nonlinear mode less aggressive.
-Removes the detector for non-audible echoes.
-Makes the attenuation when there are signals with strong narrowband
characteristics more mild in scenarios with low render.
Furthermore the CL:
-Removes the input of external echo leakage information.
Bug: webrtc:9047,chromium:824111,webrtc:8314,webrtc:8671,webrtc:5201,webrtc:5919
Change-Id: Ied1fe0c0a35d3c31b47606ed2db319a73644d406
Reviewed-on: https://webrtc-review.googlesource.com/60866
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22548}
This reverts commit bc900cb1d1810fcf678fe41cf1e3966daa39c88c.
Reason for revert: Broke downstream projects.
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}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Change-Id: I01f06c1fcf21eb2cd40dca7d4f268614200ee490
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/63720
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22537}
Time triggered tasks in the SendSideCongestionController caused
flakyness in long running unit tests of SendSideCongestionController.
This CL lets the unit test code disable the periodic tasks so they are
only triggered on demand.
Bug: webrtc:9039
Change-Id: I934045d7e6eeaa765dd221cef87389f1d98b58a5
Reviewed-on: https://webrtc-review.googlesource.com/63265
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22536}
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}
As the rate allocation has been moved into entirely into
SimulcastRateAllocator, and the listeners are thus no longer needed,
this class doesn't fill any other purpose than to determine if
ScreenshareLayers or TemporalLayers should be created for a given
simulcast stream. This can however be done just from looking at the
VideoCodec instance, so changing this into a static factory method.
Due to dependencies from upstream projects, keep the class name and
field in VideoCodec around for now.
Bug: webrtc:9012
Change-Id: I028fe6b2a19e0d16b35956cc2df01dcf5bfa7979
Reviewed-on: https://webrtc-review.googlesource.com/63264
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22529}
This CL resolves some minor issues related to running ADM unittests on Windows.
It is rather common on Windows that devices can't be opened up in mono mode and
some tests have been hardcoded to use mono and that leads to crashes and/or error
logs. Now, all tests runs in stereo as well.
NOTRY=TRUE
Bug: None
Change-Id: Iebf11a6ff63c19ff1be45575a8e0a3df4e112bd4
Reviewed-on: https://webrtc-review.googlesource.com/62940
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22510}
This CL moves all temporal layer rate allocation from
DefaultTemporalLayers and ScreenshareLayers into SimulcastRateAllocator.
This means we don't need an extra call-out to the TemporalLayers
interface to get the last allocation, which simplifies the code path a
lot.
It also paves the wave for removing the TemporalLayersFactory interface
(in a separate cl), which will further simplify the ownership model.
Bug: webrtc:9012
Change-Id: I6540b1848efa1a136dce449f13902ad479d5ee37
Reviewed-on: https://webrtc-review.googlesource.com/62420
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22502}
Added for the structs VideoCodecVP8, VideoCodecVP9, VideoCodecH264,
and SpatialLayer.
New operators are used to replace memcmp in VCMEncoderDataBase. Using
memcmp to compare structs is generally unreliable, since the struct
may contain random padding bytes due to alignment requirements
(affects at least VideoCodecH264). And in the case of VideoCodecVP8,
we need to exclude the tl_factory pointers from the comparison.
Bug: webrtc:8830
Change-Id: I40432ea7834e288f8c89ce0a28a630ae1800dff8
Reviewed-on: https://webrtc-review.googlesource.com/62761
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22500}
Since we want the VideoStreamDecoder to callback with the last
continuous frame we need to move the FrameKey into the public API.
Bug: webrtc:8909
Change-Id: I39634145d848b8163778e31a1e0d04d91f9bbeb8
Reviewed-on: https://webrtc-review.googlesource.com/60864
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22495}
And fix typo in UMA metric.
We have this pattern in the FrameCombiner component of the AudioMixer:
if (number_of_streams <= 1) {
// Copy or fill with zeros.
return;
}
// Mix and limit
LogMixingStats(/* args */);
When there is only one remote stream, info about active streams and
sample rate is not logged. This CL moves the call to log stats before
the 'return'.
Bug: webrtc:8925
Change-Id: I7b54f61f628273631909dafbfafa21e155e18d4a
Reviewed-on: https://webrtc-review.googlesource.com/62860
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22493}
Replacing the unique pointer used for access checks with a raw pointer
pointing to the object owned by the unique pointer. This is to stop
tsan from detecting a race between .get() done on the task queue and
.reset() done in the destructor.
Bug: webrtc:8415
Change-Id: Iae2ea9a2d38f319e73146e6b1e360b11b1708c76
Reviewed-on: https://webrtc-review.googlesource.com/62560
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22492}
The task queue based SendSideCongestionController (SSCC) was accessing
a unique pointer to the task queue from the task queue itself. This
triggered a tsan check failure when resetting the same unique pointer.
Also move declaration of SSCC member in RtpTransportControllerSend last,
to ensure that it, and its TaskQueue, are destroyed before other members.
Bug: webrtc:8415
Change-Id: I75c93f41deab637f7e4766ac4b61713c86f866e9
Reviewed-on: https://webrtc-review.googlesource.com/62143
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22478}
This CL significantly improves the response time
of the AEC3 delay estimator to audio buffer issues.
The CL adds ensures that the delay estimator
correlators reacts to buffer issues from the
zero state which is much faster than if it has already
achieved a state matching a previous alignment.
The CL has been extensively tested on offline
recordings.
Bug: webrtc:9023, chromium:822245
Change-Id: Ic149b9429e592d4c3535eb8432582f435a1b4745
Reviewed-on: https://webrtc-review.googlesource.com/62081
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22461}
This CL prepares for adding the BBR network controller and
unit tests for GoogCC network controller.
The changes include:
* Adding pad_rate helper method on PacerConfig.
* Adding ostream operators for controller feedback structs.
* Adding increment operator to Timestamp class.
* Adding kEpoch to Timestamp class to represent 0.
* Rounding when multiplying with double.
Bug: webrtc:8415
Change-Id: I58289f37a6f9f2eee0a88bb06fb24dc295942862
Reviewed-on: https://webrtc-review.googlesource.com/61503
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22458}
We currently have one build target containing everything for audio_device: the interfaces,
the "fine" audio buffer, and the actual implementations for each platform.
Since we are planning to move the Android implementation to the sdk/android folder,
we only want to depend on the interfaces and the "fine" audio buffer, not the other platform
specific implementations. This CL splits the audio_device target into three different targets:
the interfaces, the fine audio buffer, and the platform specific implementations. The default
audio_device target now points to the interfaces instead.
Bug: webrtc:7452
Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd
Reviewed-on: https://webrtc-review.googlesource.com/61321
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22452}
This CL ensures a smooth transition from the parameters used during
the startup phase in the call to the parameters used in the rest of the
call. This is achieved by slowly transitioning between the parameter
sets via interpolation.
Bug: chromium:819240,webrtc:8983
Change-Id: Ifbac4b93fc6ad6efc441f41fb88ef09e8ee3d669
Reviewed-on: https://webrtc-review.googlesource.com/60360
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22451}
The interface of the audioproc_f tool should be located in the api/ directory, so it becomes visible to the outside world.
Bug: webrtc:8732
Change-Id: Ia7475883aeb0e1f7a6afa5e791204b38dc53a8b8
Reviewed-on: https://webrtc-review.googlesource.com/61801
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22449}
