Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
This adds WebRTC-Vp9InterLayerPred field trial that allows to control
inter-layer prediction mode in VP9 encoder.
Bug: chromium:949536
Change-Id: Iea03db07fd21f28ab58382c5fdaac68acacc701c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131322
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27521}
This change introduces new logic to allow the injection of the FrameDecryptor
into an arbitrary already running VideoReceiveStream without resetting it. It
does this by taking advantage of the BufferedFrameDecryptor which will
forcefully be created regardless of whether a FrameDecryptor is passed in
during construction of the VideoReceiver if the
crypto_option.require_frame_encryption is true. By allowing the
BufferedFrameDecryptor to swap out which FrameDecryptor it uses this allows the
Receiver to switch decryptors without resetting the stream.
This is intended to mostly be used when you set your FrameDecryptor at a point
post creation for the first time.
Bug: webrtc:10416
Change-Id: If656b2acc447e2e77537cfa394729e5c3a8b660a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130361
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27458}
This is a better solution than https://webrtc-review.googlesource.com/c/src/+/129929 (which got reverted).
This CL instead filters out unused SSRCs from RtpConfig for RtpVideoSender.
Bug: webrtc:10485
Change-Id: Iaa8d07681419a2387c8253eb38e08a0828e9e688
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130505
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27433}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
This reverts commit ab65d8aab5fe63619033371fca1ce2711c2c2137.
Reason for revert: Fails video_engine_tests ExtendedReportsEndToEndTest.TestExtendedReportsCanSignalZeroTargetBitrate
https://ci.chromium.org/p/webrtc/builders/ci/Linux%20MSan/18366
Original change's description:
> Fix target bitrate RTCP messages behavior for SVC streams
>
> Before this CL for SVC streams (e.g VP9) still 3 separate RTP_RTCP senders
> were created. The RTCP target bitrate messages were treated as simulcast
> and were split and send for each separate spatial layer in a separate SSRC.
>
> To fix that an svc flag is now wired to VideoSendStream config
> and filled based on the encoder config in WebrtcVideoEngine. This flag is
> used to differentiate between simulcast and SVC mode in RtpVideoSender.
>
> Bug: webrtc:10485
> Change-Id: Ifa01d12a7d4f01fcbe448ad11e0cc39ab2d1df55
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129929
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27345}
TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org
Change-Id: I184f87289d9dccc67de165038d76a5690158a3b5
No-Tree-Checks: True
No-Try: True
Bug: webrtc:10485
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130467
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27355}
in production code keep using GlobalTaskQueueFactory()
in tests switch to use DefaultTaskQueueFactory directly.
This reverts commit e27ccf9a1681e0e4ff9281f9a18fea357d2bc890.
Reason for revert: addressed the failure with patchset#2
Original change's description:
> Revert "in WebrtcVoiceEngine allow to set TaskQueueFactory"
>
> This reverts commit a39254da593bbdb0b1e072a44827229680afe3ee.
>
> Reason for revert: Tests are failing due to ThreadChecker's called on valid thread.
>
> Original change's description:
> > in WebrtcVoiceEngine allow to set TaskQueueFactory
> >
> > in production code keep using GlobalTaskQueueFactory()
> > in tests switch to use DefaultTaskQueueFactory directly.
> >
> > Bug: webrtc:10284
> > Change-Id: I170274a98324796623089a965a39f0cbb7e281d9
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128878
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27296}
>
> TBR=danilchap@webrtc.org,steveanton@webrtc.org
>
> Change-Id: I9742e5d0171a94f3840e197c40fdb44523e4963b
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10284
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129780
> Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27297}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10284
Change-Id: I55fd5811c68d04c3e8cf537974496460b38c1d4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129933
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27348}
Before this CL for SVC streams (e.g VP9) still 3 separate RTP_RTCP senders
were created. The RTCP target bitrate messages were treated as simulcast
and were split and send for each separate spatial layer in a separate SSRC.
To fix that an svc flag is now wired to VideoSendStream config
and filled based on the encoder config in WebrtcVideoEngine. This flag is
used to differentiate between simulcast and SVC mode in RtpVideoSender.
Bug: webrtc:10485
Change-Id: Ifa01d12a7d4f01fcbe448ad11e0cc39ab2d1df55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129929
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27345}
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 reverts commit a39254da593bbdb0b1e072a44827229680afe3ee.
Reason for revert: Tests are failing due to ThreadChecker's called on valid thread.
Original change's description:
> in WebrtcVoiceEngine allow to set TaskQueueFactory
>
> in production code keep using GlobalTaskQueueFactory()
> in tests switch to use DefaultTaskQueueFactory directly.
>
> Bug: webrtc:10284
> Change-Id: I170274a98324796623089a965a39f0cbb7e281d9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128878
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27296}
TBR=danilchap@webrtc.org,steveanton@webrtc.org
Change-Id: I9742e5d0171a94f3840e197c40fdb44523e4963b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129780
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27297}
in production code keep using GlobalTaskQueueFactory()
in tests switch to use DefaultTaskQueueFactory directly.
Bug: webrtc:10284
Change-Id: I170274a98324796623089a965a39f0cbb7e281d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128878
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27296}
(so far SetBitrate did not do anything for media transport)
Bug: webrtc:9719
Change-Id: I48e669341ffe6c9e4697ff9146c314be7796a209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27169}
WebRTC video engine now configures bitrate on media transport
correctly.
Bug: webrtc:9719
Change-Id: I85884cd76644b7eca3763cec8ce9e31b5b64db27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127941
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27167}
Add a feature (gated by field trial) that stores
packets with unknown ssrc in a circular buffer
and replays them once a receive stream with matching
ssrc is created.
This improves situation where media is incoming
but signaling or SetFrameDecryptor is slow.
BUG=webrtc:10405
Change-Id: I7c7b2f4bd96c942c09e96db0cdae4ce5efef2541
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127543
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27159}
MediaChannel accepted the RtpPacket buffers through non-const pointer.
This is both unclear and introduces questions regarding if the buffer is
actually copied or not.
This change modifies the method to accept by value to reduce ambiguity.
Usage of the non-const data() method which could potentially copy the
buffer contents is also reduced in favor of cdata() which never copies.
Bug: None
Change-Id: I3b2daef0d31cb6aacceb46c86da3a40ce836242b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127340
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27090}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
WebRtcVideoEngine is only ever accessed from one thread, so remove
the lock and replace it with ThreadChecker assertions.
Bug: None
Change-Id: I8c34eb6473f0ebaaaafe8a163c3f5d6f19074021
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125240
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26957}
CodecSpecificInfo has a default constructor, so initializing by memset is not necessary and is in the way of adding non-trivial members.
Related chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/1495533
Bug: webrtc:10342
Change-Id: I36046f919f5fc34ea51de7288ff5c9cc0f2950b8
Reviewed-on: https://webrtc-review.googlesource.com/c/125093
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26924}
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.
Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
RIDs will now appear in the sent packets when they are supplied.
This is relevant for the Simulcast scenario which uses RIDs to
identify the different layers.
Bug: webrtc:10074
Change-Id: I2f281abc144f467e151a30ec13b8c375be4ac3e6
Reviewed-on: https://webrtc-review.googlesource.com/c/124140
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26843}
The discardability flag denotes whether the frame may be dropped by
the decoder with no effect on the decodability of subsequent frames.
Bug: webrtc:10214
Change-Id: I3654951d8863b50effe9670b8d1d7eb051240039
Reviewed-on: https://webrtc-review.googlesource.com/c/122241
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26763}
Reland with fixes for failing chromium tests.
Propagate VideoFrame::UpdateRect to encoder
Accumulate it in all places where frames can be dropped before they reach the encoder.
Reset UpdateRect in VideoBroadcaster if frame the previous frame is dropped.
No accumulation is done here since it's supposed to be a brief occasion then configuration have changed.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/123102
Bug: webrtc:10310
Change-Id: I18be73f47f227d6392bf9cb220b549ced225714f
Reviewed-on: https://webrtc-review.googlesource.com/c/123230
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26738}
Latency corresponds to base minimum delay on NetEq.
Bug: webrtc:10287
Change-Id: I538d202e3e4fe07b779c46bf560e2fde38e0468e
Reviewed-on: https://webrtc-review.googlesource.com/c/121704
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26724}
Accumulate it in all places where frames can be dropped before they reach
the encoder.
Reset UpdateRect in VideoBroadcaster if frame the previous frame is dropped.
No accumulation is done here since it's supposed to be a brief occusion then
configuration have changed.
Bug: webrtc:10310
Change-Id: I2813ecd009eb730bd99ffa0a02f979091b56bf80
Reviewed-on: https://webrtc-review.googlesource.com/c/123102
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26711}
(reverted in https://webrtc-review.googlesource.com/c/src/+/123182/1)
Original cl description:
Always offer transport sequence number header extension for audio
If the extension is negotiated, it will only be used if
the field trial WebRTC-Audio-SendSideBwe is enabled.
This allows simpler experimentation if it should be used or not.
Patchset 3 contain the only change:
Add the field trial WebRTC-Audio-SendSideBwe to call/rampup_tests.cc
TBR: srte@webrtc.org,ossu@webrtc.org
Bug: webrtc:10309 webrtc:10286
Change-Id: I2c1224e8a9fab52c1030369c1364686322e88a0f
Reviewed-on: https://webrtc-review.googlesource.com/c/123183
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26706}
This reverts commit fd965c008c7bc395bb276f260262ac11ccd25406.
Reason for revert: Cause test failure.
Original change's description:
> Always offer transport sequence number header extension for audio
>
> If the extension is negotiated, it will only be used if
> the field trial WebRTC-Audio-SendSideBwe is enabled.
> This allows simpler experimentation if it should be used or not.
>
> Bug: webrtc:10309 webrtc:10286
> Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b
> Reviewed-on: https://webrtc-review.googlesource.com/c/122542
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26689}
TBR=ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org
Change-Id: I1b7d3fa5c282a5bf049ca54695ad16c8278a2698
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10309 webrtc:10286
Reviewed-on: https://webrtc-review.googlesource.com/c/123182
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26703}
If the extension is negotiated, it will only be used if
the field trial WebRTC-Audio-SendSideBwe is enabled.
This allows simpler experimentation if it should be used or not.
Bug: webrtc:10309 webrtc:10286
Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b
Reviewed-on: https://webrtc-review.googlesource.com/c/122542
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26689}
This quality boost means that we sometimes drop a _lot_ of frames in the
base layer. It also interacts poorly with the bitrate adjuster since
even if frames are dropped they are often over-sized.
The setting still leaves the current behavior as default, but can be
changed using the WebRTC-VideoRateControl field trial.
Bug: webrtc:10155
Change-Id: I1a92ec69bab61b5148fe9d8bc391ac5ee1019367
Reviewed-on: https://webrtc-review.googlesource.com/c/122840
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26659}
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.
Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).
Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
