This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.
Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
This CL removes the deprecated legacy AEC code.
Note that this CL should not be landed before the M80 release has been cut.
Bug: webrtc:11165
Change-Id: I59ee94526e62f702bb9fa9fa2d38c4e48f44753c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161238
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30036}
This CL removes the experimental status of the multi-channel processing
in APM, and accordingly updates the variable naming.
It also splits the activation of multi-channel processing to be separate
for render and capture.
Bug: webrtc:10859
Change-Id: I0e5d04dcb94b6637c33d97146231b8ddddbaea39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160707
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29926}
The new configuration path is via AudioProcessing::ApplyConfig and
AudioProcessing::GetStatistics.
Bug: webrtc:9878
Change-Id: Ic912d67455fcef4895566edb8fef62baf62d7cfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156440
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29454}
The new configuration path is via AudioProcessing::ApplyConfig and
AudioProcessing::GetStatistics.
ApmTest.Process passes with unchanged reference files if
audio_processing_impl would initialize the VAD with
VoiceDetection::kLowLikelihood instead of kVeryLowLikelihood.
This was verified by testing this CL with that modification.
Bug: webrtc:9878
Change-Id: I4d08df37a07e5c72feeec02a07d6b9435f917d72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155445
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29395}
The pointer-to-submodule interfaces are being removed.
This CL:
1) introduces AudioProcessing::Config::GainController1 with most config,
2) adds functions to APM for setting and getting analog gain,
3) creates a temporary GainControlConfigProxy to support the transition
to the new config.
4) Moves the lock references in GainControlForExperimentalAgc and
GainControlImpl into the GainControlConfigProxy, as it becomes the
sole AGC object with functionality exposed to the client.
Bug: webrtc:9947, webrtc:9878
Change-Id: Ic31e15e9bb26d6497a92b77874e0b6cab21ff2b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126485
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27316}
This change reduces the risk of echo due to noise in the headroom
of the linear filter.
Changes:
- Use shorter delay headroom
- Delay headroom is specified in samples (not blocks)
- No hysteresis limit when delay is reduced
Bug: chromium:119942,webrtc:10341
Change-Id: I708e80e26d541dff8ca04b6da2d346a1d59cbfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126420
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27126}
The fuzzer times out on too long inputs.
This CL limits tests to 400 000 bytes, ~ 12 seconds of 8 kHz float audio.
Bug: chromium:940209
Change-Id: I86b772f9d8989a8b129d933d25ece3631a6a365f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126780
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27059}
This change disables the ERLE estimation of onsets and instead assumes
minimum ERLE. This reduces the risk of echo leaks during onsets. The
estimated ERLE was sometimes incorrect due to:
- Not enough data to train on.
- Platform noise suppression can change the echo-path.
Bug: chromium:119942,webrtc:10341
Change-Id: I1dd1c0f160489e76eb784f07e99af02f44f387ec
Reviewed-on: https://webrtc-review.googlesource.com/c/123782
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26794}
This CL removes the legacy, no longer used, render buffering code. It
also removes four unused parameters from the AEC3 config. The change
is tested for bit-exactness.
Bug: webrtc:8671
Change-Id: I2bb6cb7a1097863f228767d757d551c00593bb00
Reviewed-on: https://webrtc-review.googlesource.com/c/119701
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26399}
This Config configuration will eventually replace the AudioProcessing::noise_suppression() interface.
This also introduces a proxy NoiseSuppression, returned by AudioProcessing::noise_suppression.
Without this proxy, ApplyConfig could overwrite NS settings for clients who currently use noise_suppression(). For example, the following code will not preserve the noise suppression level:
apm->noise_suppression()->set_level(NoiseSuppression::kHigh);
auto cfg = apm->GetConfig();
apm->ApplyConfig(cfg);
The NoiseSuppression instance returned by noise_suppression() has no way to update the config inside APM, so GetConfig() will return an out-of-date config which is then re-applied. This CL adds a proxy that makes this update, by forwarding Enable() and set_level() calls to ApplyConfig().
Drive-by change: AudioProcessing::Config substructs are reordered to mirror the capture processing pipeline.
Tested: Ran ToT and this CL builds of audioproc_f and verified identical settings/aecdumps.
Bug: webrtc:9947
Change-Id: I823eade894be115c254d656562564108b2b63b1f
Reviewed-on: https://webrtc-review.googlesource.com/c/116521
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26248}
This updates some tests to use AudioProcesing::Config() and
AudioProcessing::GetStatistics() instead.
Some tests are left with voice_detection() because
a) not all tests make sense to run both APIs in parallel, and
b) we want test coverage of the old VoiceDetection until it is removed.
Bug: webrtc:9947
Change-Id: Ifb21a1e6e931d7ad3c3a4e38f5cc4f146da3c9a3
Reviewed-on: https://webrtc-review.googlesource.com/c/116160
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26134}
This CL makes possible to choose the level estimation for the adaptive
digital GC of AGC2. The options are RMS (default and currently used
estimator) and peak-based (already computed, but not used).
Besides adding the new AGC2 config param for the level estimator, this CL
also refactors the config class by making it more structured.
Bug: webrtc:7494
Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc
Reviewed-on: https://webrtc-review.googlesource.com/c/110144
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25620}
Gain specified by fuzzer in APM config was too high.
Bug: chromium:901661
Change-Id: Id3ea8d23a4284a35c827bb16125902d84e37ca1e
Reviewed-on: https://webrtc-review.googlesource.com/c/110604
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25594}
This CL utilizes the existing, but unused, ability to set
different histogram thresholds for early and late delay
estimation. It does so by tuning the parameters for these.
On top of that, some corrections are added to correctly
handle resets and the use of the hysteresis thresholds.
Bug: webrtc:19886,chromium:896334
Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/106706
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25443}
Rather fail at compile time than at run-time.
Bug: chromium:898373
Bug: webrtc:9855
Change-Id: Iaae81e04e4a8135814c1226f82d3a994de75e9ad
Reviewed-on: https://webrtc-review.googlesource.com/c/107886
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25364}
The fuzzer uses a bitmask to construct the field trials string.
Now that there's 33 relevant field trials it's no longer large enough, so switch to a 64-bit type.
Bug: chromium:898373
Change-Id: I1ea68d451ceadbd9b720079a577b573866293e4b
Reviewed-on: https://webrtc-review.googlesource.com/c/107650
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25330}
Add AecDump to the list of fuzzed stuff. Attaches an AecDump to the
Audio Processing Module in the APM-fuzzer. The AecDump writes to
/dev/null.
Bug: webrtc:7820
Change-Id: I03916ce4d1c69906ca8bb7e6fbe29c11e4ea55e5
Reviewed-on: https://webrtc-review.googlesource.com/c/107622
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25321}
The old render buffering code has been replaced, but can still be
activated by a killswitch. This change enables fuzzer testing of
the old code path.
Bug: webrtc:9726
Change-Id: I6e91cd4b4a95388cc63d1a65dade21b3c44be71b
Reviewed-on: https://webrtc-review.googlesource.com/c/107562
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25303}
"Perfection is achieved, not when there is nothing more to add,
but when there is nothing left to take away."
This CL removes the following kill-switches from AEC3
- WebRTC-Aec3DownSamplingFactor8KillSwitch
- WebRTC-Aec3NewSuppressionKillSwitch
- WebRTC-Aec3ShadowFilterJumpstartKillSwitch
- WebRTC-Aec3SlowFilterAdaptationKillSwitch
- WebRTC-Aec3SuppressorNearendAveragingKillSwitch
It also removes code paths and configuration parameters that are no
longer in use. The list of kill-switches in the audio processing
fuzzer test is updated.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: Ie0af86a14baf853548bf9c00b2b9b3bbc32c1aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/105324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25120}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
The old pointer-to-submodule interface is replaced with
AudioProcessing::Config settings.
Bug: webrtc:9535
Change-Id: I5580d690fdd7664f48fa274b39f12cc41f69da37
Reviewed-on: https://webrtc-review.googlesource.com/102020
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24847}
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).
Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
This CL introduces an adaptive estimation of the early reverb
in the estimation for the room reverberation. The benefits of
this is that for room with long early reflections there is
a lower risk of underestimating the reverberation.
This CL is for a landing the code in
https://webrtc-review.googlesource.com/c/src/+/87420,
and the review of the code was done in that CL. The author of
code is devicentepena@webrtc.org
Bug: webrtc:9479, chromium:865397
Change-Id: Id6f57e2a684664aef96e8c502e66775f37da59da
Reviewed-on: https://webrtc-review.googlesource.com/91162
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24146}
This reverts commit 771b50ca0b92477ce8aabba025e959790dd1a949.
Reason for revert: Introduces error-prone config.
Original change's description:
> Add one-stop-shop for built-in AEC toggling in APM
>
> This does not change what AEC functionality is available.
> However, a client that only uses this interface - and not the submodule
> pointer accessors - gets simpler code, and is guaranteed not to run any
> two AECs in tandem.
>
> The submodule interface EchoControlMobile is being deprecated in
> https://webrtc-review.googlesource.com/c/src/+/89392
>
> Bug: webrtc:9535
> Change-Id: Id9326074e566be6d8768010fc421c457beff402c
> Reviewed-on: https://webrtc-review.googlesource.com/89386
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24066}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: I43283a1b22538a4caa77313499989146b2ce67f1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/90060
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24067}
This does not change what AEC functionality is available.
However, a client that only uses this interface - and not the submodule
pointer accessors - gets simpler code, and is guaranteed not to run any
two AECs in tandem.
The submodule interface EchoControlMobile is being deprecated in
https://webrtc-review.googlesource.com/c/src/+/89392
Bug: webrtc:9535
Change-Id: Id9326074e566be6d8768010fc421c457beff402c
Reviewed-on: https://webrtc-review.googlesource.com/89386
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24066}
This CL re-activates the explicit handling of microphone
gain changes in the AEC3 code. The implementation is done
beneath a kill-switch so that when that switch is active
the changes in this CL are bitexact.
Bug: webrtc:9526,chromium:863826
Change-Id: I58e93d8bc0bce7bec91e102de9891ad48ebc55d8
Reviewed-on: https://webrtc-review.googlesource.com/88620
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23986}
There are many new killswitches for AEC3 code.
By fuzzing them, we ensure the code is more thoroughly tested in case
we need to trigger any switches.
Bug: webrtc:9413
Change-Id: I0bc7609e4d71fc820abfbba7d481b2374c3587cb
Reviewed-on: https://webrtc-review.googlesource.com/88225
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23960}
Add support for stereo in APM fuzzer.
Bug: webrtc:9413
Change-Id: Ie4b19037bd4613c050b03ad0bacf0f44f9feccd3
Reviewed-on: https://webrtc-review.googlesource.com/87221
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23854}
This slightly increases fuzzer coverage of the APM.
(.25 % points more line coverage.)
Bug: webrtc:9413
Change-Id: Ic992423f1dcf34fa0aa9649c8035a8e48b0ccdb2
Reviewed-on: https://webrtc-review.googlesource.com/85342
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23732}
This CL removes the remaining beamformer parts from the APM.
Bug: webrtc:9402
Change-Id: I9ab2795bd2813d17166ed0925125257b82d98a74
Reviewed-on: https://webrtc-review.googlesource.com/83340
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23694}
The fuzzer data is used to configure the field trials of the AEC.
This increases fuzzer coverage of modules/audio_processing/aec3/ by roughly 500 lines of code, ~ 3 % points increase in APM coverage for desktop Chrome.
Bug: webrtc:9413
Change-Id: Iea9059747a8492a7ca2091a359e7883750c45b27
Reviewed-on: https://webrtc-review.googlesource.com/83732
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23650}
We found out that
int16_t x = test::FuzzDataHelper::ReadOrDefaultValue(0)
reads 4 bytes from the fuzzer input instead of 2. That means that
almost half the bits in the input data to audio_processing_fuzzer are
ignored. This change adds template arguments to force reading 2 bytes
when we only need 2.
We also add a small manually generated corpus. During local testing we
let the fuzzer run for a few hours on an empty corpus. Adding the
manually-generated files resulted in an immediate coverage increase by
~3%, and then by another 3% over the next few hours.
The manually generated corpus contains a short segment of speech with
real echo. We suspect that triggering Voice Activity Detection or echo
estimation filter convergence can be difficult for an automatic
fuzzer.
We remove the Level Controller config. We read 20 bytes extra after the
config to guard against future configuration changes.
Bug: webrtc:7820
Change-Id: If60c04f53b27c519c349a40bd13664eef7999368
Reviewed-on: https://webrtc-review.googlesource.com/58744
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22269}
The AEC3 factory is now part of the WebRTC API.
Bug: webrtc:8844
Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e
Reviewed-on: https://webrtc-review.googlesource.com/57141
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22204}
This CL adds the GainCurveApplier (GCA). It owns a
FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve
(IGC). The GCA uses the LE to compute the input signal level, looks up
a gain from IGC and applies it on the signal.
The other IGC and LE submodules were added in previous CLs [1] and
[2].
This CL also turns on AGC2 in the APM fuzzer.
[1] https://webrtc-review.googlesource.com/c/src/+/51920
[2] https://webrtc-review.googlesource.com/c/src/+/52381
Bug: webrtc:7949
Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd
Reviewed-on: https://webrtc-review.googlesource.com/54361
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22103}
Going through the coverage of audio_processing_fuzzer, it was noticed
that it didn't cover AEC-m and AEC-2 code. Therefore this CL adds 2
fuzzer targets that only fuzz the previous generation echo cancellers.
To avoid code duplication, the APM running code was broken out in a
new GN target. We have also changed all fuzzing code to use the
FuzzDataHelper class to avoid manual pointer arithmetic.
Bug: webrtc:7820
Change-Id: Ifea3266e396b487952a736945577fccea15d0e01
Reviewed-on: https://webrtc-review.googlesource.com/36500
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21638}
