Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This change introduces a new FrameCadenceAdapter class which takes the
role of being a VideoFrameSinkInterface<> instead of VideoStreamEncoder.
The FrameCadenceAdapter will see its functionality grow in future CLs
and eventually enable screenshare capture sources to have zero hertz as
the minimum capture frequency.
This CL moves logic related to UMA collection and constraints into the
adapter.
The adapter has two major modes. Future functionality is planned to be
added under the WebRTC-ZeroHertzScreenshare field trial. Unit tests are
added that verify passthrough operation when WebRTC-ZeroHertzScreenshare
isn't specified or disabled.
Just specifying the WebRTC-ZeroHertzScreenshare field trial isn't
enough to activate the feature, but the caller has to additionally
configure screen content type, minimum FPS 0, and maximum FPS > 0 for
the new mode.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I1799110ed40843152786ad80df10acfb83a608b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236682
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35315}
According to https://w3c.github.io/webrtc-pc/#datachannel-send it should
return an error, definitely not close the data channel.
While we should probably return an RTCError will better information, this
would break the API and will be done later.
Bug: webrtc:13289
Change-Id: I90baf012440fbe2a38a826cf50b50b2b668fd7ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35306}
Removes the ability to accept nonencrypted answers to encrypted offers.
Fixes some logic around bundled sessions and requirement for
transport parameters.
Bug: webrtc:11066
Change-Id: I56d8628d223614918a1e5260fdb8a117c8c02dbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236344
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35298}
Changes one preexisting enum-to-string function to use the new format.
Also changes the RTC_LOG macros that created collisions with ToString,
for tidiness, and documents the recommended function form.
Bug: webrtc:13272
Change-Id: Ic8bb54ed31402ba32675b142d796cf276ee78df5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235722
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35296}
This reverts commit 3b18208f13e85b356e61a95c0a261e9781403743
and is the third attempt at removing stun origin support
Bug: webrtc:12132
Change-Id: Ic41a6d011fb6239907a257cc4c81ec4d2923dc4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236260
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35294}
The constants are being made private since no new code should use them.
However, the helper functions sill uses "AV1X" internally for backwards
compatibility.
Bug: webrtc:13166
Change-Id: I0a0cd46f31ca70bb7f395c9b1e9cdb202df11f6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35289}
This is part of the removal of support for SDES.
Bug: webrtc:11066
Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35262}
When using VideoEncoderSoftwareFallbackWrapper, releasing and
initialization of encoder_ (H/W) and fallback_encoder_(S/W) happen
repeatedly as reconfiguration procedure is called from higher layer.
Below problems would occur when our encoder_(H/W) fails to initialize
or encode.
Firstly, some encoders' SetFecControllerOverride() functions will fail
during repeated calls since they have checks like
RTC_DCHECK(!fec_controller_override_) to avoid repeated assignment of
fec_controller_override_.
(see : LibvpxVp8Encoder::SetFecControllerOverride())
Secondly, if main_ encoder fails to initialize at first attempt, FEC
setting (fec_controller_override) will not set until reconfiguration
procedure is called again.
This CL comes with two changes to fix above problems.
1. Sets fec_controller_override to both encoders when
SoftwareFallbackWrapper::SetFecController() is called.
2. Removes the current_encoder()->SetFecControllerOverride() in
PrimeEncoder() to avoid redundant calls which may involve fatal error.
Bug: webrtc:13184
Change-Id: I082c93de552bc9ec3141c6490d35acfcee2f8935
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234301
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35231}
Previous limits was only in a comment and users had no way to query it
from the API.
Bug: webrtc:13289
Change-Id: I6187dd9f9482bc3e457909c5e703ef1553d8ef15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235378
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35224}
This is a reland of ba29ce320fe1f9ac69b0ff8eb50fbe402c2912a6
readding the origin to the CreateRelayPortArgs structure to not break
downstream tests yet:
https://webrtc-review.googlesource.com/c/src/+/235300/1..2
Original change's description:
> remove stun origin support
>
> Bug: webrtc:12132
> Change-Id: I0f32e6af77e0c553b0c3b0d047ff03e14c492b31
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234384
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35202}
Bug: webrtc:12132
Change-Id: Ied840b59bb7c9497e98f9b80eb0a54d30008a40f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35220}
This change
- adds new type VideoTrackSourceConstraints expressing min/max FPS
constraints.
- adds new method VideoTrackSourceInterface::ProcessConstraints.
- adds new method VideoSinkInterface<>::OnConstraintsChanged.
- updates AdaptedVideoTrackSource and VideoBroadcaster to forward
the constraints to sinks.
- adds several unit tests for the added functionality.
- and finally, implements OnConstraintsChanged in VideoStreamEncoder.
Chromium will be updated in coming CLs to supply constraints set
through the MediaStream module.
go/rtc-0hz-present
Bug: chromium:1255737
No-Try: true
Change-Id: Iffef239217269c332a1aaa902ddeae2440929e22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235040
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35197}
This change improves echo canceller transparency by enabling the use
of a non-capped ERLE when computing the residual echo spectrum for
dominant nearend detection.
Experimentation has shown that the feature improves echo canceller
transparency and user ratings.
Implementation CL:
https://webrtc-review.googlesource.com/c/src/+/221920
Bug: webrtc:12870
Change-Id: I7dc66810e8300cd35321bcd5b9fae9bc3386836d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234841
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35186}
A preparation for splitting server sockets out into a separate
interface, see https://webrtc-review.googlesource.com/c/src/+/232607.
Transition plan:
1. Land this cl.
2. Update downstream code to use the new name.
3. Attempt landing
https://webrtc-review.googlesource.com/c/src/+/232607. May need
additional steps to not break downstream implementations of
PacketSocketFactory::CreateServerTcpSocket.
Bug: webrtc:13065
Change-Id: Ife448c705222f4c9f66a096e3dc7eb07e0f9c3af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233700
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35155}
This unlocks migration from AsyncResolver to AsyncDnsResolver for
clients that implement PacketSocketFactory.
A default implementation is provided, so that clients that implement
CreateAsyncResolver will still see their name resolution work.
Bug: webrtc:12598
Change-Id: If835cbc753712e9f5b4bd3d5805c7f7d2a561ee5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233500
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35131}
Currently the implementation of FrameTransformers uses distinct,
incompatible types for recevied vs about-to-be-sent frames. This adds a
flag in the interface so we can at least check that we are being given
the correct type. crbug.com/1250638 tracks removing the need for this.
Chrome will be updated after this to check the direction flag and provide
a javascript error if the wrong type of frame is written into the
encoded insertable streams writable stream, rather than crashing.
Bug: chromium:1247260
Change-Id: I9cbb66962ea0718ed47c5e5dba19a8ff9635b0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <toprice@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35100}
* Replace "AV1X" with "AV1";
* Keep mapping of "AV1X" payload name to kVideoCodecAv1 to not break
support of injectable "AV1X".
Bug: webrtc:13166
Change-Id: I9a50481209209f3857bbf28f4ed529ee6972377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34987}
So that applications don't need to construct it from the exposed
network_thread.
The EmulatedNetworkManagerInterface::network_thread() accessor is currently
used as a way to get to emulation's SocketServer, and should be deleted
when applications of the emulation framework have migrated away from
that usage.
Bug: webrtc:13145
Change-Id: I3efa55d117cad8ac601c48a9d2d2aa62a121f9c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231649
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34964}
This reverts commit 2c41cbae37cac548a1133589b9d2c2e8614fa6cb.
Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2.
Original change's description:
> Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
>
> This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.
>
> Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
>
> Original change's description:
> > Wire up non-sender RTT for audio, and implement related standardized stats.
> >
> > The implemented stats are:
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
> >
> > Bug: webrtc:12951, webrtc:12714
> > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#34861}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta,hbos,minyue
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Olga Sharonova <olka@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34897}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12951, webrtc:12714
Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34930}
The explicitly defined constructor suppresses the assignment operator,
which blocks the chromium roll.
Bug: b/198565646
Change-Id: I35917d4b99ad86dcf8b9863e798f5a63d9073824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231123
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34904}
This reverts commit eb89027733c511962120a5f7fd309d1893ad389c.
Reason for revert: We got a successful WebRTC roll into Chromium at last. Relanding, as the issue should be fixed in Chromium by now.
TBR=hta@webrtc.org,philipp.hancke@googlemail.com
Original change's description:
> Revert "frame transformer: make GetPayloadType pure virtual again"
>
> This reverts commit 209ac5fd95594ab3834dad3e3dbd14c8196637bc.
>
> Reason for revert: Breaks WebRTC autoroll presubmit:
> https://chromium-review.googlesource.com/c/chromium/src/+/3134502
> Example failure https://ci.chromium.org/ui/p/chromium/builders/try/mac-rel/775468/overview
>
> ../../buildtools/third_party/libc++/trunk/include/__memory/unique_ptr.h:725:32: error: allocating an object of abstract class type 'testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>'
> return unique_ptr<_Tp>(new _Tp(_VSTD::forward<_Args>(__args)...));
> ^
> ../../third_party/blink/renderer/platform/peerconnection/rtc_encoded_video_stream_transformer_test.cc:69:26: note: in instantiation of function template specialization 'std::make_unique<testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>>' requested here
> auto mock_frame = std::make_unique<NiceMock<MockTransformableVideoFrame>>();
> ^
> ../../third_party/webrtc/api/frame_transformer_interface.h:36:19: note: unimplemented pure virtual method 'GetPayloadType' in 'NiceMock'
> virtual uint8_t GetPayloadType() const = 0;
> ^
>
>
> Original change's description:
> > frame transformer: make GetPayloadType pure virtual again
> >
> > after chrome was updated in
> > https://chromium-review.googlesource.com/c/chromium/src/+/3103323
> >
> > BUG=webrtc:13077
> >
> > Change-Id: I7e5ff6aaae81c5dcfbaa41b09ef01bc95bb7251a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230143
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> > Cr-Commit-Position: refs/heads/main@{#34877}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:13077
> Change-Id: I6b2e4e2804890c857f1f832a6a4faa614ec026c4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230920
> Reviewed-by: Olga Sharonova <olka@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Olga Sharonova <olka@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34891}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:13077
Change-Id: I8414f74be87aad62166a95fac0cd400257fd25a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231120
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34901}
This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.
Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
Original change's description:
> Wire up non-sender RTT for audio, and implement related standardized stats.
>
> The implemented stats are:
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34861}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=hta,hbos,minyue
Bug: webrtc:12951, webrtc:12714
Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34897}
VideoDecoder no longer uses this VideoCodec class,
thus this member is unused.
Bug: webrtc:13045
Change-Id: I6e46a563e90f2538bf288995a3837d95c00ba9cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230941
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34896}
This reverts commit 209ac5fd95594ab3834dad3e3dbd14c8196637bc.
Reason for revert: Breaks WebRTC autoroll presubmit:
https://chromium-review.googlesource.com/c/chromium/src/+/3134502
Example failure https://ci.chromium.org/ui/p/chromium/builders/try/mac-rel/775468/overview
../../buildtools/third_party/libc++/trunk/include/__memory/unique_ptr.h:725:32: error: allocating an object of abstract class type 'testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>'
return unique_ptr<_Tp>(new _Tp(_VSTD::forward<_Args>(__args)...));
^
../../third_party/blink/renderer/platform/peerconnection/rtc_encoded_video_stream_transformer_test.cc:69:26: note: in instantiation of function template specialization 'std::make_unique<testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>>' requested here
auto mock_frame = std::make_unique<NiceMock<MockTransformableVideoFrame>>();
^
../../third_party/webrtc/api/frame_transformer_interface.h:36:19: note: unimplemented pure virtual method 'GetPayloadType' in 'NiceMock'
virtual uint8_t GetPayloadType() const = 0;
^
Original change's description:
> frame transformer: make GetPayloadType pure virtual again
>
> after chrome was updated in
> https://chromium-review.googlesource.com/c/chromium/src/+/3103323
>
> BUG=webrtc:13077
>
> Change-Id: I7e5ff6aaae81c5dcfbaa41b09ef01bc95bb7251a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230143
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#34877}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:13077
Change-Id: I6b2e4e2804890c857f1f832a6a4faa614ec026c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230920
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34891}
