This removes use of the MockPeerConnection and replaces it with the
FakePeerConnectionForStats testing class.
Bug: webrtc:8764
Change-Id: I78553c5a4e4d68cb6666a83f443f72f7c25488dc
Reviewed-on: https://webrtc-review.googlesource.com/46940
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21882}
This is a reland of eed5aa8904d09179971d3f4e7e10c109d7c62bfc
Original change's description:
> Structured ICE logging via RtcEventLog.
>
> This change list contains the structured logging module for ICE using
> the RtcEventLog infrastructure, and also extension to the log parser
> and analyzer.
>
> Bug: None
> Change-Id: I6539cf282155c2cde4d3161c53500c0746671a02
> Reviewed-on: https://webrtc-review.googlesource.com/34622
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21816}
TBR=pthatcher@webrtc.org,terelius@webrtc.org,deadbeef@webrtc.org
Bug: None
Change-Id: I3df585bf636315ceb0273967146111346a83be86
Reviewed-on: https://webrtc-review.googlesource.com/47545
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21881}
When a simulcast stream is enabled or disabled, we want this state
change to be reflected properly in the RtpRtcp modules. Each video send
stream can contain multiple rtp_rtcp_modules pertaining to different
simulcast streams. These modules are currently all turned on/off when
the send stream is started and stopped. This change allows for
individual modules to be turned on/off. This means if a module stops
sending it will send a bye message, so the receiving side will not
expect more frames to be sent when the stream is inactive and the
encoder is no longer encoding/sending images.
Bug: webrtc:8653
Change-Id: Ib6d00240f627b4ff1714646e847026f24c7c3aa4
Reviewed-on: https://webrtc-review.googlesource.com/42841
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21880}
This reverts commit 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9.
Reason for revert: Major regressions on perf bots.
Original change's description:
> Moved congestion controller to task queue.
>
> The goal of this work is to make it easier to experiment with the
> bandwidth estimation implementation. For this reason network control
> functionality is moved from SendSideCongestionController(SSCC),
> PacedSender and BitrateController to the newly created
> GoogCcNetworkController which implements the newly created
> NetworkControllerInterface. This allows the implementation to be
> replaced at runtime in the future.
>
> This is the first part of a split of a larger CL, see:
> https://webrtc-review.googlesource.com/c/src/+/39788/8
> For further explanations.
>
> Bug: webrtc:8415
> Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
> Reviewed-on: https://webrtc-review.googlesource.com/43840
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21868}
TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: Ia8a273eb9e92b7d0d960c49658c228208170962d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/47560
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21877}
end_to_end_tests.cc was over 5000 lines and covered many different
areas in it's testing. In this change it is separated into multiple
smaller test sets separated by the functionality they are testing. The
reasoning behind this is that the fact that a test is working end to end
should be secondary to what functionality the test is actually testing.
A slight functional change is that for some of the tests the
parametrization over round robin pacing being controlled with a field
trial is removed since they are simple enough that they should not be
affected by the pacing method.
Bug: None
Change-Id: I4b7eba80fc142ecfc8fa642dab9b6f587d914048
Reviewed-on: https://webrtc-review.googlesource.com/46143
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21876}
WebRTC internal code should always use include paths that start
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I8a7ab64dfecdb3da4099fdec61e5fc27af4f8ccc
Reviewed-on: https://webrtc-review.googlesource.com/47380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21874}
So we can report perf results using JSON and not parsing stdout.
I reordered the way the arguments are parsed, so that options go
at the end, and not at the middle, which is an awkward place to put them.
Regular usage specifying [-I], bottleneck_value, infile and outfile
shouldn't be affected.
Bug: chromium:807737
Change-Id: Ida863846400326c33e443d723f384971b891b6e5
Reviewed-on: https://webrtc-review.googlesource.com/47161
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21873}
In MediaCodecVideoEncoder, VideoFrame timestamp was used as a
presentation timestamp. With this change timestamp maintained in C++
code is used instead. This matches the behaviour with old frame
callbacks.
Bug: b/72832862
Change-Id: I1f0543ebe837ccac22c83a81a81f3ea128e2a866
Reviewed-on: https://webrtc-review.googlesource.com/47381
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21872}
WebRTC internal code should always use include paths that start
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I36f01784fa5f5b77eefc02db479b1f7f6ee1a8c3
Reviewed-on: https://webrtc-review.googlesource.com/46263
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21871}
The goal of this work is to make it easier to experiment with the
bandwidth estimation implementation. For this reason network control
functionality is moved from SendSideCongestionController(SSCC),
PacedSender and BitrateController to the newly created
GoogCcNetworkController which implements the newly created
NetworkControllerInterface. This allows the implementation to be
replaced at runtime in the future.
This is the first part of a split of a larger CL, see:
https://webrtc-review.googlesource.com/c/src/+/39788/8
For further explanations.
Bug: webrtc:8415
Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3
Reviewed-on: https://webrtc-review.googlesource.com/43840
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21868}
The VideoCodecTest class is a fixture base class for the
libvpx-VP8, libvpx-VP9, and OpenH264 unit tests. It is unrelated
to the VideoProcessor tests, which we colloquially refer to as
the "codec test".
This rename is thus to reduce this confusion. It should have no
functional impact.
Bug: webrtc:8448
Change-Id: If73443bda5df0f29a71ce6ce069ac128795ff0ad
Reviewed-on: https://webrtc-review.googlesource.com/47160
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21867}
Previously, the code which reported cipher stats to UMA for all
transports would classify the media type based on the transport name,
which is brittle and misleading with BUNDLE. This corrects the code to
track all media types (audio, video, data) which use the transport and
report once for each.
Bug: None
Change-Id: I8506f64f0011788b744b8386ac58518a21914b52
Reviewed-on: https://webrtc-review.googlesource.com/47247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21863}
This is a reland of 001546da953275c7a39eb220592b440c9b47d756
Original change's description:
> Break up rtc_event_log_api to solve circular dependencies.
>
> The original rtc_event_log_api is refactored to a pure API target plus
> multiple targets coupled with WebRTC implementations.
>
> Bug: None
> Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
> Reviewed-on: https://webrtc-review.googlesource.com/43247
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#21811}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org
Bug: None
Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647
Reviewed-on: https://webrtc-review.googlesource.com/46900
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21862}
https://chromium-review.googlesource.com/c/external/webrtc/+/575910 pretty much made it a mandate to have OpenSSL 1.1.0 to compile webrtc.
So, let's be explicit about it and cleanup old code for older version support.
Also, generate a compiler error for older OpenSSL versions.
Bug: webrtc:8817
Change-Id: I28590348137b6a04503eabdcc6328297ecf5213e
Reviewed-on: https://webrtc-review.googlesource.com/46502
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#21861}
Original change's description:
> Parameterize PeerConnection signaling tests for Unified Plan
>
> This also changes the behavior of CreateAnswer to fail unless
> the signaling state is kHaveRemoteOffer or kHaveLocalPranswer,
> as per the WebRTC specification.
>
> Bug: webrtc:8765
> Change-Id: I60ac67cd92b17fcbff964afc14d049481e816a28
> Reviewed-on: https://webrtc-review.googlesource.com/41042
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21779}
Bug: webrtc:8813
Change-Id: I9f608fcd0b7aca00b4c1092e271dbd9cd710c38a
Reviewed-on: https://webrtc-review.googlesource.com/46861
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21860}
This is intended to ensure compatibility between Plan B and
Unified Plan endpoints for the single audio - single video case.
If Unified Plan is the offerer, it will add a=msid and a=ssrc MSID
entries to its offer.
If Unified Plan is the answerer, it will use whatever MSID
signaling mechanism was used in the offer (either a=msid or
a=ssrc).
Bug: webrtc:7600
Change-Id: I6192dec19123fbb56f5d04540d2175c7fb30b9b6
Reviewed-on: https://webrtc-review.googlesource.com/44162
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21859}
Instead of chartjson-result-file, since that's the flag passed by the recipe.
TBR=phoglund@webrtc.org
Bug: chromium:807737
Change-Id: I3a679ab7e5c0a446e675d0f4647344cc4194b357
Reviewed-on: https://webrtc-review.googlesource.com/46541
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21858}
This changes the behavior of CreateOffer/CreateAnswer when Unified
Plan is enabled to be in line with that specified in JSEP.
In particular, MSID information is now only included if the
RtpTransceiver is not stopped and either is sending or has ever
sent.
Bug: webrtc:7600
Change-Id: I6400f0583525c7776331eeb0e1bb53973bc02dfb
Reviewed-on: https://webrtc-review.googlesource.com/46400
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21857}
This is done to solve a problem where a string literal is implicitly cast
to a temporary std::string when calling webrtc::test::InitFieldTrialsFromString
which passes a pointer to the internal representation to
webrtc::field_trial::InitFieldTrialFromString(char*). This pointer is
stored for later use, but the temporary std::string is destroyed as soon
as the function returns.
Using webrtc::field_trial::InitFieldTrialFromString(char*) instead,
avoids the implicit casts (but the caller still needs to ensure that
the char* outlives the program). The validation previously done by
webrtc::test::InitFieldTrialsFromString can now be done by manually
calling webrtc::test::ValidateFieldTrialsStringOrDie(const std::string&).
Add system_wrappers:field_trial_default as a direct dependency to
various targets to allow including the field_trials_default.h header.
Bug: webrtc:8812
Change-Id: Ib5a641ea255b1c16a8f7f35e1fe67f6c38a61da6
Reviewed-on: https://webrtc-review.googlesource.com/46141
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21856}
The network preference is added to RTCConfiguration and passed to ICE.
ICE considers now the preference set by applications over network
interface types when making decisions in candidate pair switching.
Bug: webrtc:8816
Change-Id: I40d2612705b54c83dd45772ac855808e0a76b1e1
Reviewed-on: https://webrtc-review.googlesource.com/44020
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21855}
It is broken by very long file names in Chromium.
TBR=phoglund@webrtc.org
Bug: chromium:808111
Change-Id: If3dd556be506b90f8efaa01c50e3d8608ba9be20
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/46104
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21854}
This avoids a data race in which the lifetime TimeInterval is accessed
by the owning Call objects concurrently with SendRtp calls on the
underlying Channel object.
Bug: webrtc:8794
Change-Id: If53d5680095c0177656b659162457287cb8e45dd
Reviewed-on: https://webrtc-review.googlesource.com/46525
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21853}
Merge OveruseDetector into the TrendlineEstimator (send side BWE only) and remove the OveruseDetector from DelayBasedBwe. The behavior should be the same as before. One expection is that if no packets were received for kStreamTimeOutMs (2 seconds), it would previously reset the trendline estimator but not the detector. Since they have been merged, it now resets both.
Create an interface that the estimators will implement to facilitate experimentation with different estimators/detectors.
Bug: webrtc:8729
Change-Id: I5c3d2161a0d0dcb2e8a140c0fd887f0286d70fd4
Reviewed-on: https://webrtc-review.googlesource.com/38781
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21852}
This can be used to wrap Objective-C components in C++ classes, so users
can use the WebRTC C++ API directly together with the iOS specific
components provided by our SDK.
Bug: webrtc:8832
Change-Id: I6d34f7ec62d51df8d3a5340a2e17d30ae73e13e8
Reviewed-on: https://webrtc-review.googlesource.com/46162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21850}
WebRTC internal code should always use include paths that start
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I1fc4cb50d81522a486888a626d4a95df7847d591
Reviewed-on: https://webrtc-review.googlesource.com/46743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21849}
WebRTC internal code should always use include paths that start
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I6c345c38fd990f66bc1a8129e7f7cee7d161e926
Reviewed-on: https://webrtc-review.googlesource.com/47120
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21848}
WebRTC internal code should always used include paths that starts
from the root of the project and that clearly identify the header file.
This allows 'gn check' to actually keep dependencies under control
because 'gn check' cannot enforce anything if the include path
is not fully qualified (starting from the root of the project).
Bug: webrtc:8815
Change-Id: I23fb4fed0c27a4d98bea360315b959af843587bc
Reviewed-on: https://webrtc-review.googlesource.com/46101
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21847}
Building with the newly published cocoapod generated a few warnings,
which looked a bit bad.
Bug: webrtc:8831
Change-Id: I70c06930603b328e4d11c599a5b5dd77b45150c6
Reviewed-on: https://webrtc-review.googlesource.com/46163
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21846}
Test of packet loss used a simplified calculation of lost packets and
loss ratio. Changed the calculation to be more accurate. This protects
against triggering for future implementations with more precise
calculations.
Bug: webrtc:8415
Change-Id: I721dc83954e8738fdf8ea729dee4cc8b8c8fa091
Reviewed-on: https://webrtc-review.googlesource.com/46740
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21845}
Encoded frames are preserved and decoded after all layers are
encoded.
Each spatial layer is decoded with separate decoder.
For quality evaluation of lowres layers original input frame is
downscaled with bilinear interpolation.
Encoded and decoded frames are dumped into separate files.
For async codecs encoded frames are passed to decoder in encode
callback, as before.
Bug: webrtc:8524
Change-Id: Idb0c92c7274c1915cff9a011a2794f1cf4bc8cb1
Reviewed-on: https://webrtc-review.googlesource.com/43381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21844}
When executed on swarming, the script is run from //out/<android build dir>,
so it's better to keep that convention.
Given that all paths are given, cwd doesn't seem to be needed.
TBR=phoglund@webrtc.org
Bug: chromium:755660
Change-Id: Iabf6603983ff88b1281e8113da1aad3320967b72
Reviewed-on: https://webrtc-review.googlesource.com/46142
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21843}
Reorganizes methods in java_types.h to logical groups. The order in
the source file matches the order in the header file.
Bug: webrtc:8769
Change-Id: Id3e1e80276a747a3d9952598207ac55493ac46b6
Reviewed-on: https://webrtc-review.googlesource.com/46146
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21842}
Variable `LAST_SYSTEM_ERROR` was introduced in https://webrtc-review.googlesource.com/c/src/+/32780.
It seems to be the same codeblock in `physicalsocketserver.cc`, only difference is it did not
include the header <errno.h>.
Also, probably a good idea to make the include conditional.
Bug: None
Change-Id: I3241dd83be4a248c6c1db2fab8f924a185e354cb
Reviewed-on: https://webrtc-review.googlesource.com/45864
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21841}
This CL adds #error to spot where rtc_base/win32.h is unconditionally
included and fixes all the places where it happens.
Bug: webrtc:8814
Change-Id: I3c005acf2cdb58a51f1bcaa4acaeebd272c56660
Reviewed-on: https://webrtc-review.googlesource.com/46060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21840}
This CL fixes an issue where the aecdump file handle gets garbage
collected and closed early in the call.
Bug: webrtc:8822
Change-Id: I959908da164b0ec61ccd976fc52f3d919da11b52
Reviewed-on: https://webrtc-review.googlesource.com/46103
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21839}
