Add media config for using environment monotonic timestamps (i.e. not UTC) in RTCStats constructor, and implemented the usage of the flag.
Bug: chromium:369369568
Change-Id: Ia93d048742c28af201164fe7b2152b791bb6d0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43156}
Setting the standard deviation to 0 is valid and should be interpreted
as directly using the sample value at the coordinates without weighting.
This is made explicit in the documentation for the Corruption Detection
extension:
http://www.webrtc.org/experiments/rtp-hdrext/corruption-detection
Also, change stddev to std_dev in halton_frame_sampler files.
Bug: webrtc:358039777
Change-Id: Id5aa4110194f7f2b2fe9914c94304c90afd64198
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363300
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43070}
The code that restricts the maximum number of simulcast layers based on
resolution is a spec-compliance bug and doesn't make much sense: if the
app asks for 3 layers it should get 3 layers. Since the app knows the
size of the track, it could very easily ask for 1 layer when resolution
is small if that is the behavior it wanted. If the app doesn't ask to
disable layers, WebRTC shouldn't disable layers on its behalf.
This behavior makes even less sense with this "new" API since the app
is explicitly controlling the send resolution in absolute terms.
Removing this behavior in the general case is out of scope since it
would break backwards compatibility, but since `requested_resolution`
has not been exposed to the web yet and existing usage is small, this
is an opportunity to fix the compliance bug for this API.
This CL makes the last web platform test for "scaleResolutionDownTo"
pass.
Bug: chromium:363544347
Change-Id: Ic6fadf3cad69d3beec4ae03d3d031e8062382ad9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43061}
The FrameInstrumentationGenerator is responsible for generating the
instrumentation data that will be used to detect corruption. The data is
then passed to the encoder in the CodecSpecificInfo.
Bug: webrtc:358039777
Change-Id: I79d0534920b4c7fa001e1138371dfd36c13424fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362584
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43060}
This class calculates the corruption score based on the given samples from two frames.
Bug: webrtc:358039777
Change-Id: Ib036f91ec16609e827137cc35d342a2c49764737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362801
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#43043}
For key frames: increase the sequence index until the last 7 bits are
all zeroes. If this results in an overflow, wraparound to 0.
Also:
* Allow setting and getting the sequence index
* Allow getting LayerId
Bug: webrtc:358039777
Change-Id: Ibe16689a3d1eb5706d4fce5c9220770046f26896
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362540
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43042}
The `requested_resolution` API must not change aspect ratio, example:
- Frame is 60x30
- Requested is 30x30
- We expect 30x15 (not 30x30!) as to maintain aspect ratio.
This bug was previously fixed by making VideoAdapter unaware of the
requested resolution behind a flag: this seemed OK since the
VideoStreamEncoder ultimately decides the resolution, whether or not
the incoming frame is adapted.
But this is not desired for some non-Chrome use cases. This CL attempts
to make both Chrome and non-Chrome use cases happy by implementing the
aspect ratio preserving restriction inside VideoAdapter too.
This allows us to get rid of the "use_standard_requested_resolution"
flag and change the "VideoStreamEncoderResolutionTest" TEST_P to
TEST_F.
Bug: webrtc:366067962, webrtc:366284861
Change-Id: I1dfd10963274c5fdfd18d0f4443b2f209d2e9a4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362720
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43037}
The recently added tests resulted in some .log file on some bots being
too long:
video_engine_tests_exe-VideoStreamEncoderStandardOrLegacyRequestedResolutionTest_VideoStreamEncoderStandardOrLegacyRequestedResolutionTest_RequestedResolutionInWrongAspectRatioAndSourceIsAdapting_0-1.log
This CL makes the test names significantly shorter.
# Trivial and believed to fix purple bots, let's land ASAP
NOTRY=True
Bug: webrtc:367066321
Change-Id: I831911947af9d5639d1edb559470f1c9ae702d6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362721
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43030}
When requested_resolution uses a different aspect ratio than the source
the encoder will restrict the frame without changing its aspect ratio,
e.g. a 60x30 input frame that is restricted to 30x30 results in 30x15,
not 30x30.
While this logic works correctly in isolation, if the source also adapts
the frame size based on the sink_wants.requested_resolution that is
signaled back to the source, then the source will produce stretched
30x30 prior to the encoder which happily sends 30x30 not knowing any
wiser.
This is incompatible with the spec[1] and makes this WPT[2] fail. The
correct behavior is to NOT signal the requested_resolution back to the
source, the encoder already configures the correct resolution so this
isn't actually needed and the source shouldn't need to know this API.
In order not to break downstream projects, the new behavior is landed
behind a flag and both behaviors are tested with TEST_P.
This unblocks launching scaleResolutionDownTo API on Web. Migrating
from old to new code path and deleting the flag is a follow-up AI:
webrtc:366284861.
[1] https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-scaleresolutiondownto
[2] https://chromium-review.googlesource.com/c/chromium/src/+/5853944
# Relying on previous green runs for confidence due to purple bots atm,
# see b/367211396
NOTRY=True
NOPRESUBMIT=True
Bug: webrtc:366067962, webrtc:366284861
Change-Id: I7fd1016e9cc6f0b0b9b8c23b0708e521f8e12642
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362541
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43024}
This API should not modify the aspect ratio of the frame, e.g. if the
frame is 1280x720 and requested_resolution is 1280x360, the result
should be 640x360, not a streched out 1280x360 frame. The spec version
of this API calls this "maxWidth" and "maxHeight" which is the right
way to think about it rather than a forced width and height.
VideoAdapter continues to be used to apply adaptation restrictions, but
we now make sure to calculate the correct frame size BEFORE applying
restrictions. Prior to this CL, the VideoAdapter was also used to apply
requested_resolution restrictions. This is actually wrong and would
cause strange scaling factors in some cases, e.g. f=1280x720 + r=720x405
would result in 640x360 instead of 720x405. Now we make f=720x405 first
and only adjust further if restrictions or alignments require us to.
Since this is a change in behavior a WebRtcVideoChannelTest is updated.
Encodings integration test is also added, both for aspect ratio (new
behavior) and orientation agnosticism (old behavior still passing).
Bug: webrtc:366067962
Change-Id: I4e8dc27da5a84d73238b8ab74ef197eb5ee8072a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362101
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43020}
A recent bugfix[1] introduced having to reconfigure the encoder in
response to restrictions updating for the sake of not getting stuck at
the wrong resolution in some cases.
But the newly added ReconfigureEncoder() calls happened too early/often,
and the initial frame dropper got confused thinking resolution changes
were not in response to adaptation (restrictions) when they in fact were.
The CL wrongly disabled the "reset initial frame dropper" logic, when
the correct solution to this problem should have been to delay the
ReconfigurationEncoder() call to the next frame (using
`pending_encoder_reconfiguration_ = true`).
With this delay the initial frame dropper is not confused and we can
restore the old "reset" logic, fixing the regression.
[1] https://webrtc-review.googlesource.com/c/src/+/360200
Bug: b/364252657
Change-Id: I6b93f4cc44eb12b1bbbda0d8d1e9906c29b615a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361740
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42961}
This makes it simpler to use in more contexts.
Bug: b/364184684
Change-Id: I1b08ebd24e51ba1b3f85261eed503a78cd006fd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361480
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42956}
The fieldtrials can be used to override the static QP threshold
that is used in QualityConvergenceMonitor to determine if an
encoded video stream has reached its target quality.
The fieldtrials do not change the dynamic detection.
Bug: chromium:328598314
Change-Id: I5995860eff461f0c712293e34cf75834ce414bed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361201
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42928}
This is to make it clear that this field indicate whether the upper bits
of the sequence number should be communicated. However, the current
implementation only sets the field if it is a key frame.
Bug: webrtc:358039777
Change-Id: Ic2c8b6d91499e4e5cf25b8ce9591d326d7044fb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361402
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42924}
To make it available for constructing ModuleRtpRtcpImpl2
Bug: webrtc:362762208
Change-Id: Ic6ad339170c6aedb6c0bf42419964741d4d32bcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360921
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42888}
Normally (scaleResolutionDownBy) restrictions are applied at the source
which changes the input frame size which triggers reconfiguration with
appropriate scaling factors.
But when requested_resolution is used, encoder settings are by
definition not relative to the input frame size. In order for
restrictions to have an effect, they are applied inside
ReconfigureEncoder(): you get the minimum between the requested
resolution and the restricted resolution.
ReconfigureEncoder() happens when you SetParameters(), but the bug
here is that we don't do it again once the restrictions are updated.
So if restrictions are 540p when you ask for 720p, you get 540p and
after restrictions change to unlimited you're still stuck in 540p.
The fix is to also trigger ReconfigureEncoder() inside
OnVideoSourceRestrictionsUpdated() when the restricted resolution is
changing and a requested_resolution is configured.
To ensure reconfiguring the encoder "on the fly" like this does not
reset initial frame dropping logic, InitialFrameDropper caring about
input frame size changing is made conditional on not using
requested_resolution.
# Slow purple bots failing but they are not affected by this change.
NOTRY=True
Bug: webrtc:361477261
Change-Id: I1389aa16cf408b0d14e0b5b6f68c2442db955be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360200
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42882}
It looks like the map `content_specific_stats_` is just carried over into `aggregated_stats` without doing any important processing. This seems to be redundant and hence, removing it in this CL.
Bug: None
Change-Id: Ia4a5de03999ecd33d387014928cba322bb11ee93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360745
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42870}
The original CL overlooked the possibility that the encoder may be
reconfigured in the middle of a stream.
Restructure the code so that all calls to QP convergence controller
happen on the encoder queue.
A side effect of this CL is that `EncodedImage::SetAtTargetQuality()`
is never called. The information is supplied to the frame cadence
adapter directly without this intermediate step.
`EncodedImage::SetAtTargetQuality()` and
`EncodedImage::IsAtTargetQuality()` are being marked as deprecated
in https://webrtc-review.googlesource.com/c/src/+/359660.
Bug: chromium:359410061
Change-Id: I941b5f60b1a9fd7694dbedf2f3e4ff5253ccf357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359640
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42788}
The functions will be used to map a frame's QP to its optimal standard
deviation for the Gaussian kernel for the filter applied in the
automatic corruption detection algorithm.
Bug: b/358039777
Change-Id: Idb3b8cfdbd4a405151c660df87205e3949f9b085
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359380
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42777}
e.g all files in the api/test folder not including subdirectories
Bug: webrtc:42226242
Change-Id: I18d74a18f8feec41eb252faa9acfffd1d6f45ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42773}
This CL fixes a problem where VSEAM's `queue_` was dereferenced
post destruction. A sequence triggering it is:
0. FrameCadenceAdapter (FCA) is configured with Metronome with >= 34 ms tick delay.
1. A frame is queued for processing on worker queue.
2. The FCA is destroyed. The contained VSyncEncodeAdapterMode instance is scheduled for deletion on worker queue.
3. Encode queue is destroyed.
4. Worker queue is executed, which runs a task that dereferences `queue_`.
Bug: chromium:356423094
Change-Id: Iae8dc070304ef5ec0cfb0b4f27bbb7fe86e7def7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358660
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42745}
This CL only adds variables necessary for the feature, which will be
implemented in later CLs.
Bug: webrtc:350555527
Change-Id: I71e56666e629f56168d316bf693150c0df0e2ecf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356740
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42698}
This requires making CallConfig move-only so it can hold a unique_ptr to
the factory, but as discussed with Danil, that seems fine.
Bug: chromium:355610792
Change-Id: Ie52e33faaa4a2af748daeb25f5327b7a532936e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42679}
Currently this class assumed that if the same RTP sequence number is unwrapped again result would be the same.
That might not be true when several packets were inserted in between these two calls and unwrapper changed its state
This CL propose instead to unwrap once, and save the result in the intermediate struct.
To minimize the change and the risk, only redundant unwrapping is replaced to use unwrapped sequence number
Bug: webrtc:353565743
Change-Id: I8a18c8c206a0e16010951cabcf81dd9cb1588eda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357660
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42662}
This is a reland of commit 09f03be54804e81f626c26e8fde8c86cc952545f
Use max_num_layers instead of encoder_config.number_of_streams when calculation stream resolutions in EncoderStreamFactory::GetStreamResolutions().
Original change's description:
> Pass true stream resolutions to GetSimulcastConfig()
>
> Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig().
>
> Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests:
> * GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow
> * GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow
> * GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
> * GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned
> * GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4
> * GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough).
> * GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow
> * GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544
>
> Bug: webrtc:351644568, b/352504711
> Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42651}
Bug: webrtc:351644568, b/352504711
Change-Id: Ib3fd859257b61c2a5d695b8b8f45c95495117c0e
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42654}
This reverts commit 09f03be54804e81f626c26e8fde8c86cc952545f.
Reason for revert: breaks downstream projects
Original change's description:
> Pass true stream resolutions to GetSimulcastConfig()
>
> Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig().
>
> Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests:
> * GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow
> * GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow
> * GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
> * GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned
> * GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4
> * GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough).
> * GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow
> * GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544
>
> Bug: webrtc:351644568, b/352504711
> Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42651}
Bug: webrtc:351644568, b/352504711
Change-Id: I7aadbe49419b7ac610db4db99284fdcdce9deff5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357500
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42653}
