Generated using gmock_gen.py with some editing.
This mock doesn't seem to be used by unittest in webrtc, but we need to use it in downstream unittests.
Bug: None
Change-Id: Ia7904ffdd22f3d16fe5fd515fa68833817b44481
Reviewed-on: https://webrtc-review.googlesource.com/85780
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23900}
This test creates a one way audio and video call, allows for bandwidth
estimation to ramp up and then runs the call for 10 seconds. The
average bandwidth estimate over this time is recorded as a perf metric.
This is done at the PeerConnection level with the intention to catch
regressions related to ICE configurations. Stats are taken from
PeerConnection for BWE, and the network simulation is done with a
VirtualSocketServer.
Bug: webrtc:7668
Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db
Reviewed-on: https://webrtc-review.googlesource.com/78361
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23758}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'pc'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12
Reviewed-on: https://webrtc-review.googlesource.com/84181
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23669}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
The proper closing procedure is:
1. Alice resets outgoing stream.
2. Bob receives incoming stream reset, resets his outgoing stream.
3. Alice receives incoming stream reset; channel closed!
4. Bob receives acknowledgement of reset; channel closed!
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7
However, up until now we've been sending both an incoming and outgoing reset
from the side initiating the closing procedure, and doing nothing on the remote
side.
This means that if you call "Close" and the remote endpoint is using an old
version of WebRTC, the channel's state will be stuck at "closing" since the
remote endpoint won't send a reset. Which is already what happens when Firefox
is talking to Chrome.
This CL also fixes an issue where the DataChannel's state prematurely went to
"closed" before the closing procedure was complete. Which could result in a
new DataChannel attempting to re-use the ID and failing.
TBR=magjed@webrtc.org
Bug: chromium:449934, webrtc:4453
Change-Id: Ic1ba813e46538c6c65868961aae6a9780d68a5e2
Reviewed-on: https://webrtc-review.googlesource.com/79061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23478}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
Previous code had a FakePeriodicVideoSource and a
VideoTrackSource, where the latter is reference counted and
outlives the former. That results in potential races when
RemoveSink is called on the VideoTrackSource after the
FakePeriodicVideoSource is destroyed, with a complicated sequence
to do correct shutdown.
The new class, FakePeriodicVideoTrackSource, owns a
FakePeriodicVideoSource, and they get the same lifetime.
Bug: webrtc:6353
Change-Id: Ic33b393e00a31fa28893dce2018948d3f90e0a9e
Reviewed-on: https://webrtc-review.googlesource.com/76961
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23320}
This is a reland of 5faf36ef3c582350fba5ef97a3549e440d81a283
The issue in Chrome has been fixed and this should be safe to reland.
TBR=deadbeef
Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}
Bug: webrtc:7580
Change-Id: Iabd41fb21afdf452c039d5513824ae334f8d1d3f
Reviewed-on: https://webrtc-review.googlesource.com/76980
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23247}
Replaces both BitrateConstraintsMask and
PeerConnectionInterface::BitrateParameters. The latter is kept
temporarily for backwards compatibility.
Bug: None
Change-Id: Ibe1d043f2a76e56ff67809774e9c0f5e0ec9e00f
Reviewed-on: https://webrtc-review.googlesource.com/74020
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23148}
The transaction_id field should be refreshed for every getParameters()
call and checked at each setParameters() call.
This also checks that getParameters() was ever called to return a proper
error code.
Bug: webrtc:7580
Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
Reviewed-on: https://webrtc-review.googlesource.com/70820
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23120}
To prepare for making the software codecs optional and injectable, these
codec factories provide a way to pass in identical factories as were the
default old behaviour.
Bug: webrtc:7925
Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921
Reviewed-on: https://webrtc-review.googlesource.com/71162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23096}
This profile will now not be used unless the application explicitly
sets the flag in CryptoOptions to true. As a result, an 80-bit
authentication tag will be used instead of a 32-bit one. See bug for
more details.
Bug: webrtc:7670
Change-Id: I7c0a118fd7b1e7aac23b9eb8717099f055de0441
Reviewed-on: https://webrtc-review.googlesource.com/66600
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22757}
This CL removes PushCaptureData(), which is unused.
The reason I'm removing it is since this method is cauing chromium-style
violations for all files that includes
modules/audio_device/include/audio_device_defines.h, and it's annoying
to suppress it everywhere.
Bug: webrtc:8659
Change-Id: I9133d05259075d8e8ec89b764be934f37b5fa77e
Reviewed-on: https://webrtc-review.googlesource.com/66404
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22717}
In previous implementation, the SctpTransport always assumes the
DtlsTransport underneath is non-null, which is not true after switching
to new JsepTransportController model.
This CL adds nullptr when connecting/disconnecting the SctpTransport with
the DtlsTransport.
The "channel" related methods and variables are also renamed.
Bug: chromium:827917, chromium:828220
Change-Id: I95aa2900d23b0885f45500e2c53def771abdccad
Reviewed-on: https://webrtc-review.googlesource.com/66160
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22700}
This is a reland of f351c3408a0c7f695447a2a9f4e6a1719a0d6a26
Reland history:
The original CL broke tests in chromium which were manually tested in
the first reland. Another small fix was added to the reland to fix a
downstream bug, which caused separate tests to fail in chromium.
These were not caught because the chromium trybot was down. These
are temporarily disabled in chrome to allow this change to roll in.
Original change's description:
> Reland "Adds support for multiple or no media stream ids."
>
> This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb
>
> Original change's description:
> > Adds support for multiple or no media stream ids.
> >
> > With Unified Plan SDP semantics, this adds support for specifying
> > either no media stream ids or multiple media stream ids for a
> > transceiver/sender/receiver. This includes serializing/deserializing
> > SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> > <appdata>" line to indicate the no stream case. Note that this does
> > not synchronize between multiple streams, this is still just supported
> > based upon the first media stream id.
> >
> > Bug: webrtc:7932, webrtc:7933
> > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> > Reviewed-on: https://webrtc-review.googlesource.com/61341
> > Commit-Queue: Seth Hampson <shampson@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22611}
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
> Reviewed-on: https://webrtc-review.googlesource.com/65560
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22687}
TBR=deadbeef@webrtc.org
Bug: webrtc:7932, webrtc:7933
Change-Id: Ideb30219b2f952dd51428cd4e8bd43ef49df5b17
Reviewed-on: https://webrtc-review.googlesource.com/66280
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22699}
The TransportController will be replaced by the JsepTransportController
and JsepTransport will be replace be JsepTransport2.
The JsepTransportController will take the entire SessionDescription
and handle the RtcpMux, Sdes and BUNDLE internally.
The ownership model is also changed. The P2P layer transports are not
ref-counted and will be owned by the JsepTransport2.
In ORTC aspect, RtpTransportAdapter is now a wrapper over RtpTransport
or SrtpTransport and it implements the public and internal interface
by calling the transport underneath.
Bug: webrtc:8587
Change-Id: Ia7fa61288a566f211f8560072ea0eecaf19e48df
Reviewed-on: https://webrtc-review.googlesource.com/59586
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22693}
This reverts commit f351c3408a0c7f695447a2a9f4e6a1719a0d6a26.
Reason for revert: Breaks chromium import
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_rel_ng/58012
Failin tests:
WebRtcRtpBrowserTest.TrackAddedToSecondStream
WebRtcRtpBrowserTest.TrackSwitchingStream
Original change's description:
> Reland "Adds support for multiple or no media stream ids."
>
> This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb
>
> Original change's description:
> > Adds support for multiple or no media stream ids.
> >
> > With Unified Plan SDP semantics, this adds support for specifying
> > either no media stream ids or multiple media stream ids for a
> > transceiver/sender/receiver. This includes serializing/deserializing
> > SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> > <appdata>" line to indicate the no stream case. Note that this does
> > not synchronize between multiple streams, this is still just supported
> > based upon the first media stream id.
> >
> > Bug: webrtc:7932, webrtc:7933
> > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> > Reviewed-on: https://webrtc-review.googlesource.com/61341
> > Commit-Queue: Seth Hampson <shampson@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22611}
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
> Reviewed-on: https://webrtc-review.googlesource.com/65560
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22687}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,shampson@webrtc.org
Change-Id: I1835419f963762bc308a91d81c423d8e7bf65026
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7932, webrtc:7933
Reviewed-on: https://webrtc-review.googlesource.com/65700
Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22690}
This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb
Original change's description:
> Adds support for multiple or no media stream ids.
>
> With Unified Plan SDP semantics, this adds support for specifying
> either no media stream ids or multiple media stream ids for a
> transceiver/sender/receiver. This includes serializing/deserializing
> SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> <appdata>" line to indicate the no stream case. Note that this does
> not synchronize between multiple streams, this is still just supported
> based upon the first media stream id.
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> Reviewed-on: https://webrtc-review.googlesource.com/61341
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22611}
Bug: webrtc:7932, webrtc:7933
Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
Reviewed-on: https://webrtc-review.googlesource.com/65560
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22687}
With Unified Plan SDP semantics, this adds support for specifying
either no media stream ids or multiple media stream ids for a
transceiver/sender/receiver. This includes serializing/deserializing
SDPs with multiple a=msid lines in a m section, or an "a=msid:-
<appdata>" line to indicate the no stream case. Note that this does
not synchronize between multiple streams, this is still just supported
based upon the first media stream id.
Bug: webrtc:7932, webrtc:7933
Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
Reviewed-on: https://webrtc-review.googlesource.com/61341
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22611}
This exposes the stats selection algorithm[1] on the PeerConnection.
Per-spec, there are four flavors of getStats():
1. RTCPeerConnection.getStats().
2. RTCPeerConnection.getStats(MediaStreamTrack selector).
3. RTCRtpSender.getStats().
4. RTCRtpReceiver.getStats().
1) is the parameterless getStats() which is already shipped.
2) is the same as 3) and 4) except the track is used to look up the
corresponding sender/receiver to use as the selector.
3) and 4) perform stats collection with a filter, which is implemented
in RTCStatsCollector.GetStatsReport(selector).
For technical reasons, it is easier to place GetStats() on the
PeerConnection where the RTCStatsCollector lives than to place it on the
sender/receiver. Passing the selector as an argument or as a "this"
makes little difference other than style. Wiring Chrome up such that the
JavaScript APIs is like the spec is trivial after GetStats() is added to
PeerConnectionInterface.
This CL also adds comments documenting our intent to deprecate and
remove the legacy GetStats() APIs some time in the future.
[1] https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
Bug: chromium:680172
Change-Id: I09316ba6f20b25d4f9c11785d0a1a1262d6062a1
Reviewed-on: https://webrtc-review.googlesource.com/62900
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22602}
Downstreams have been updated, and this now updates all uses of label()
to id() within WebRTC code. This change also makes id() pure virtual and
removes label().
Bug: webrtc:8977
Change-Id: Ib045ea4fabba6f14447c64875c7aeba87dc2be24
Reviewed-on: https://webrtc-review.googlesource.com/60382
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22431}
Needed in order to return error codes to Chromium.
Bug: chromium:819629, chromium:589455
Change-Id: Iab22250db62a348eee21c6d8bfc44020a7380586
Reviewed-on: https://webrtc-review.googlesource.com/60522
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22367}
There was an implementation, but it relied on SSLCertificate::GetChain,
which was never implemented. Except in the fake certificate classes
used by the stats collector tests, hence the tests were passing.
Instead of implementing GetChain, we decided (in
https://webrtc-review.googlesource.com/c/src/+/6500) to add
methods that return a SSLCertChain directly, since it results in a
somewhat cleaner object model.
So this CL switches everything to use the "chain" methods, and gets
rid of the obsolete methods and member variables.
Bug: webrtc:8920
Change-Id: Ie9d7d53654ba859535462521b54c788adec7badf
Reviewed-on: https://webrtc-review.googlesource.com/56961
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22177}
STUN candidates use STUN binding requests to keep NAT bindings open.
Related stats including packet loss and RTT can be now collected via the
legacy GetStats in PeerConnection.
Bug: None
Change-Id: I7b0eee1ccb07eb670a32ee303c9590047b25f31c
Reviewed-on: https://webrtc-review.googlesource.com/54100
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22113}
These methods no longer work with Unified Plan and have been
replaced by iterating over RtpTransceivers to get all the
VoiceChannels and VideoChannels.
Bug: webrtc:8587
Change-Id: I66ec282ee9f7eb987c32e30957733c13c6cf45b8
Reviewed-on: https://webrtc-review.googlesource.com/55760
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22110}
This adds a callback corresponding to the ontrack event as defined
in the WebRTC specification.
Bug: webrtc:7600
Change-Id: Ied8c55e11dcea864428fb194623c1595c21657c7
Reviewed-on: https://webrtc-review.googlesource.com/52660
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22066}
This class adds a convenience method that allows *sending* a task
to the queue (as opposed to posting). Sending is essentially
Post+Wait, a pattern that we don't want to encourage use of
in production code, but is convenient to have from a testing
perspective and there are already several places in the
source code where we use it.
Change-Id: I6efd1b2257e6c641294bb6e4eb53b0021d9553ca
Bug: webrtc:8848
Reviewed-on: https://webrtc-review.googlesource.com/50441
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22022}
This changes the StatsCollector to handle stats from multiple
MediaChannels of the same type (e.g., audio or video).
Bug: webrtc:8764
Change-Id: I91ba50d10cf469420189a311acdafbf6f78579b2
Reviewed-on: https://webrtc-review.googlesource.com/49560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22009}
