This reverts commit 6c0c55c31817ecfa32409424495eb68b31828c40.
Reason for revert:
Fixed the flake.
Original change's description:
> Revert "Added PeerConnectionObserver::OnRemoveTrack."
>
> This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc.
>
> Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway.
>
> Original change's description:
> > Added PeerConnectionObserver::OnRemoveTrack.
> >
> > This corresponds to processing the removal of a remote track step of
> > the spec, with processing the addition of a remote track already
> > covered by OnAddTrack.
> > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
> >
> > Bug: webrtc:8260, webrtc:8315
> > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
> > Reviewed-on: https://webrtc-review.googlesource.com/4722
> > Commit-Queue: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20214}
>
> TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
>
> Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8260, webrtc:8315
> Reviewed-on: https://webrtc-review.googlesource.com/7940
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20218}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,aleloi@webrtc.org,hbos@webrtc.org
Change-Id: Iab7500bebf98535754b102874259de43831fff6b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8260, webrtc:8315
Reviewed-on: https://webrtc-review.googlesource.com/8180
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20227}
Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
Store sent frame info in map to solve potential issue where sent framerate statistics could be
incorrect.
Bug: webrtc:8375
Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
Reviewed-on: https://webrtc-review.googlesource.com/7880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20225}
This CL is the same CL we had at https://codereview.webrtc.org/3014543002/.
Since we cannot land it with Rietveld anymore let's move the discussion
to Gerrit.
BUG=webrtc:7641
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
Change-Id: I5662bec318544b07f476c12ecada997d726e7361
Reviewed-on: https://webrtc-review.googlesource.com/7981
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20224}
This adds and extra param to SquareGenerator's constructor that sets the number of squares used. By default, it uses the same value that was previously hard-coded.
Bug: webrtc:8326
Change-Id: Ie7cff94e4a54fd5bb91f981930cad5e624e0e132
Reviewed-on: https://webrtc-review.googlesource.com/6020
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20223}
This will make it easier for users to specify that they want iSAC in
their codec factories, since they'll no longer have to worry about
choosing either the fix or the float implementation.
BUG=webrtc:8343
Change-Id: I5fb713710a8dd86162b5de73a2f0a851947f1411
Reviewed-on: https://webrtc-review.googlesource.com/6540
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20222}
A '--store-test-artifacts' flag is introduced in
gtest-parallel-wrapper.py to make it possible for test running on
swarming to save test artifacts to the swarming output dir.
Bug: chromium:755660
Change-Id: I6bc1fbf210c1f224f3a7186c5444ae063a6af222
Reviewed-on: https://webrtc-review.googlesource.com/7840
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20221}
In order to unblock the chromium roll into WebRTC we have to
uniform the WebRTC DEPS file to the change done in:
https://chromium-review.googlesource.com/c/chromium/src/+/706280.
third_party/instrumented_libraries/scripts/download_binaries.py is
not needed anymore and msan instrumented libraries are downloaded
by default.
Bug: None
Change-Id: I46e7442866be38b366a6d53efa2b81bcd276c919
Reviewed-on: https://webrtc-review.googlesource.com/7863
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Cr-Commit-Position: refs/heads/master@{#20220}
This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc.
Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway.
Original change's description:
> Added PeerConnectionObserver::OnRemoveTrack.
>
> This corresponds to processing the removal of a remote track step of
> the spec, with processing the addition of a remote track already
> covered by OnAddTrack.
> https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
>
> Bug: webrtc:8260, webrtc:8315
> Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
> Reviewed-on: https://webrtc-review.googlesource.com/4722
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20214}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8260, webrtc:8315
Reviewed-on: https://webrtc-review.googlesource.com/7940
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20218}
The call to |interframe_delay_max_moving_.Add()| below depends on |now|
non decreasing in consequtive calls. However, if two threads are
competing for the lock it may happen that current thread calculates |now|
before the other thread, yet it will get the lock later. This will result
in decreasing local time in consecutive calls and trigger a DCHECK.
The same also applies to |timing_frame_info_counter_|.
Bug: none
Change-Id: I3376d88d4448c2c105e9227a445b11cd6ba8d341
Reviewed-on: https://webrtc-review.googlesource.com/7861
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20217}
Only allow gaps in picture id for key frames.
When a VideoSendStream is destroyed, frames in the queue not yet sent are lost. The recreated stream
should start with a key frame.
Also enable PictureIdIncreasingAfterStreamCountChangeSimulcastEncoderAdapter if forced fallback is
enabled. In this case, the picture id is set in the PayloadRouter and the sequence should be
continuous.
Bug: none
Change-Id: If7987166c86d6a8edbe5e479701f7f04c49cd89c
Reviewed-on: https://webrtc-review.googlesource.com/7363
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20216}
This corresponds to processing the removal of a remote track step of
the spec, with processing the addition of a remote track already
covered by OnAddTrack.
https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks
Bug: webrtc:8260, webrtc:8315
Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700
Reviewed-on: https://webrtc-review.googlesource.com/4722
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20214}
This CL adds a smooth rampup of the NLP gain in AEC3.
Bug: webrtc:8361
Change-Id: I49aa75904751ffe9150db1572271fe7a26232449
Reviewed-on: https://webrtc-review.googlesource.com/7740
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20213}
This CL separates the NLP gain computation for the different variants
of echo estimation. This simplifies the setting of tuning
parameters, with resulting transparency improvements and increased
echo removal performance.
Bug: webrtc:8359
Change-Id: I9b97064396fb6f6e2f418ce534573f68694390a1
Reviewed-on: https://webrtc-review.googlesource.com/7613
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20209}
This runs `gn gen --check` with default args to detect mismatches between
#includes and dependencies in the BUILD.gn files, as well as general build
errors. Run this before uploading a CL for early detection, otherwise such
errors will cause per-platform try jobs to fail.
Bug: webrtc:8279
Change-Id: Ib87e2e3f40b8d1146ea5c1202fb113508a3f05e3
Reviewed-on: https://webrtc-review.googlesource.com/5482
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20208}
Sort alphabetically where it makes sense.
Bug: webrtc:8327
Change-Id: I723f0fda504be0da7fd0d8d6495b16e82b7bebde
Notry: True
Tested: Ran tools_webrtc/autoroller/roll_deps.py --dry-run successfully.
Reviewed-on: https://webrtc-review.googlesource.com/7617
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20207}
This CL adds the ability to set a default echo path delay to use
in AEC3 when there is prior knowledge about the delay in the echo
path.
Bug: webrtc:8358
Change-Id: Ie368f9a6dec9f412e09bf0e095f89d84305045f9
Reviewed-on: https://webrtc-review.googlesource.com/7604
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20204}
Simple rename to reflect that any AEC implementing the EchoControl
interface could be used instead of EchoCanceller3.
Bug: webrtc:8346
Change-Id: Id9abdc15bf3e0b30197077b8c11e20891a7463b3
Reviewed-on: https://webrtc-review.googlesource.com/7611
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20203}
This CL adds some general AEC3 transparency improvements.
Specifically:
-A minimum for how the nearend is masking echo is added.
-A temporal smoothing constant is increased to increase the transparency.
-Parameters are surfaced to the parameter config struct.
Bug: webrtc:8360
Change-Id: I2a4881eb40f4fab53ad740c4001925f0af86bbec
Reviewed-on: https://webrtc-review.googlesource.com/7605
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20200}
This reverts commit b23ed7f1af467a228cbdc63e839cac8856e9df8d.
Reason for revert: Breaks Chromium FYI build
Sample error log:
../../remoting/test/fake_port_allocator.cc:52:7: error: no matching constructor for initialization of 'cricket::BasicPortAllocator'
: BasicPortAllocator(network_manager, socket_factory),
^ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
../../third_party/webrtc/p2p/client/basicportallocator.h:32:12: note: candidate constructor not viable: requires single argument 'network_manager', but 2 arguments were provided
explicit BasicPortAllocator(rtc::NetworkManager* network_manager);
^
../../third_party/webrtc/p2p/client/basicportallocator.h:27:7: note: candidate constructor (the implicit copy constructor) not viable: requires 1 argument, but 2 were provided
class BasicPortAllocator : public PortAllocator {
^
../../third_party/webrtc/p2p/client/basicportallocator.h:29:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
^
../../third_party/webrtc/p2p/client/basicportallocator.h:33:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
^
../../third_party/webrtc/p2p/client/basicportallocator.h:36:3: note: candidate constructor not viable: requires 5 arguments, but 2 were provided
BasicPortAllocator(rtc::NetworkManager* network_manager,
Original change's description:
> TurnCustomizer - an interface for modifying stun messages sent by TurnPort
>
> This patch adds an interface that allows modification of stun messages
> sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
> and the TurnCustomizer will be invoked by TurnPort before sending
> message. This allows user to e.g add custom attributes as described
> in rtf5389.
>
> BUG=webrtc:8313
>
> Change-Id: Ibf5cc10af84c57288f1eb4c578ca064611a769f1
> Reviewed-on: https://webrtc-review.googlesource.com/4781
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20197}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,jonaso@webrtc.org
Change-Id: I624efb22f6e3ceac1b2ff8af1ec47e4cfdde9140
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8313
Reviewed-on: https://webrtc-review.googlesource.com/7680
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20199}
This patch adds an interface that allows modification of stun messages
sent by TurnPort. A user can inject a TurnCustomizer on the RTCConfig
and the TurnCustomizer will be invoked by TurnPort before sending
message. This allows user to e.g add custom attributes as described
in rtf5389.
BUG=webrtc:8313
Change-Id: Ibf5cc10af84c57288f1eb4c578ca064611a769f1
Reviewed-on: https://webrtc-review.googlesource.com/4781
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20197}
Namely:
* Changing destruction_timestamp_ to rtc::Optional, instead of using 0
as a magic value.
* Adding some comments.
* Adding a log statement that would have helped debugging the issue
that hit this DCHECK.
* Getting rid of a 2-line method called in one place, which was not
really helping code readability.
Bug: None
Change-Id: I5fb1ce60edea29cab0c2a8c97e735f26c08aba62
Reviewed-on: https://webrtc-review.googlesource.com/7440
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20196}
The switch to lld caused many new violations.
The ones for MSan are being handled upstream: crbug.com/772559
But memcheck is unlikely to be supported.
BUG=webrtc:8356
TBR=kjellander@webrtc.org
Change-Id: I6b728b7d689a1d883fc6e49eb438336bda08afe1
Reviewed-on: https://webrtc-review.googlesource.com/7080
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20195}
Remove redundant null pointer checks.
move header fields validation when they passed in rather when used.
Validate all used fields from the header.
Bug: webrtc:8335
Change-Id: I20b132c6fb8966e49a5414fe757f74d504f4a61a
Reviewed-on: https://webrtc-review.googlesource.com/7400
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20194}
These versions of StartRtcEventLog() are not used.
Bug: webrtc:8111
Change-Id: I1fb543a908decff203b13f8358598f75d875c111
Reviewed-on: https://webrtc-review.googlesource.com/6782
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20191}
Care should be taken when landing this, because it will affect users of
WebRTC. I'm thinking primarily of Chromium. Chromium will start to
support High profile and Baseline profile using SW codecs with this CL.
Clients who do SDP munging without looking at the H264 profile might
switch from Constrained Baseline to High profile with this change.
Bug: webrtc:8317
Change-Id: Idca3a6b761a66d9e521b913b850c6ae14381f1f4
Reviewed-on: https://webrtc-review.googlesource.com/6341
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20190}
Move RtcEventLogOutput into the API, so that we would be able to change StartRtcEventLog (in PeerConnectionInterface) to use it.
Bug: webrtc:8111
Change-Id: I1d70af792ec584d3f1a8eced1b66c38e4a360642
Reviewed-on: https://webrtc-review.googlesource.com/7220
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20189}
The SW and HW encoder have separate picture id sequences.
Set picture id to not cause sequence discontinuties at encoder changes.
Bug: webrtc:6634
Change-Id: Ie47168791399303d88cbec3ef6ae8ef8c16ced30
Reviewed-on: https://webrtc-review.googlesource.com/5481
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20188}
This is in preparation for deleting the include in rtc_base/refcount.h,
but that change has to wait for some downstream applications to
not rely in the indirect include.
Partial reland of "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
This is a reland of b7239a9dc825ddb25dbc800aed3a065163b9a10e
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I63a42712f6c1ec83823c629d1a954fd1a04d4a6c
Reviewed-on: https://webrtc-review.googlesource.com/7281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20185}
Histogram based percentile counter is added in ReceiveStatisticsProxy.
New 95th percentile metric is reported in the same way as interframe
delay.
Bug: webrtc:8347
Change-Id: I5e476cbb6361dd341cdb97c37d883c3923e5f611
Reviewed-on: https://webrtc-review.googlesource.com/6880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20184}
This is a reland of b7239a9dc825ddb25dbc800aed3a065163b9a10e
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
>
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
>
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}
Bug: webrtc:8270
Change-Id: I9738f6680ab52d0f43639a1a39175fdba5957681
Reviewed-on: https://webrtc-review.googlesource.com/5840
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20180}
There are some internal projects that need to be updated before we remove StartRtcEventLog and StopRtcEventLog. In this CL we take away the pure-virtuality status of the functions. After landing this, we can fix the internal projects, then land https://webrtc-review.googlesource.com/c/src/+/6782.
TBR=stefan@webrtc.org
Bug: webrtc:8111
Change-Id: Ibe495a7e7d6bf8120b1a26f056bd1443031733bf
Reviewed-on: https://webrtc-review.googlesource.com/6980
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20178}
