This CL removes the use of the rtc::Thread::socketserver() method
in one place.
Bug: webrtc:13145
Change-Id: I1a1b2501450788263d5280c43e4328ade46f4146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263320
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37340}
since the check is done for the more modern "kind" attribute
BUG=None
Change-Id: I520b62f312348b006d8147ce7b2b483c78657b98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266141
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37258}
This is what allowed us to remove "transceiver" stats from the spec.
Bug: webrtc:14191
Change-Id: I687a2dd97de016832005cb4271f6e1a0e0560cd3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266022
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37247}
This should allow standard stats users not to have to rely on the
obsolete "track" stats.
Bug: webrtc:14174
Change-Id: I24e5e1478ee47c73c12fcdecf7314f41fcc76bc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266020
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37246}
This is a reland of commit 626f87d90501fd8d7a4ea071686cd8befd0d430c
Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}
Bug: webrtc:14167
Change-Id: Ib12488fb8510fb3430e92bcd72d88c7879ecb0ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265861
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37226}
This is a reland of commit 626f87d90501fd8d7a4ea071686cd8befd0d430c
Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}
Bug: webrtc:14167
Change-Id: Ifdc37e7a48fea516c727c06d2f510780386cb204
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265805
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37215}
This reverts commit 626f87d90501fd8d7a4ea071686cd8befd0d430c.
Reason for revert: Breaks one downstream project, will re-land after the dependency stops referencing an unimplemented RTT metric
Original change's description:
> [Stats] Cleanup: Remove unimplemented metrics and obsolete TODOs.
>
> In preparation for the spec moving closer to PR, let's not have
> placeholder metrics not implemented.
>
> Bug: webrtc:14167
> Change-Id: If4688ef85b57f88154d490186b306b30414874e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37205}
Bug: webrtc:14167
Change-Id: I7e9ac60eb474b44fab678d4c08ddcae846ce456c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265800
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37206}
In preparation for the spec moving closer to PR, let's not have
placeholder metrics not implemented.
Bug: webrtc:14167
Change-Id: If4688ef85b57f88154d490186b306b30414874e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265383
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37205}
This currently only exists as a goog legacy stat and has no spec
equivalent according to
https://docs.google.com/document/d/1z-D4SngG36WPiMuRvWeTMN7mWQXrf1XKZwVl3Nf1BIE/edit
Yet it is useful to debug issues sometimes. Exposing it as a
nonstandard stat will make it show up in chrome://webrtc-internals,
removing a need to switch to the legacy stats API there.
BUG=webrtc:14118
Change-Id: I506357ad54ff33df3ba46fb81558aa32187ac8e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264420
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37055}
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
media_kind is the old name (that is kept around since we can't deprecate)
BUG=None
Change-Id: I445441a54bb4ff408502d1aba6834cdac874324b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259766
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36625}
This cl/ extends the RTCIceCandidateStats object with
network_adapter_type and vpn, so that it maps the underlying
WebRTC objects completly.
Bug: webrtc:13773
Change-Id: I5cf79972c60ca6bf2a127dc96fa90811263ba6fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36110}
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
In preparation for switching the default from kPlanB to kUnifiedPlan,
which could cause subtle bugs for those not prepared for it, we change
the default to kNotSpecified. The only purpose of kNotSpecified is to
crash, forcing any dependencies to explicitly set their sdp_semantics
value.
Tests are updated to explicitly set sdp_semantics when necessary, and
where the test does not care we update to kUnifiedPlan.
If this change lands without getting reverted we can let it sit for a
few weeks, after which we should change the default to kUnifiedPlan and
delete kNotSpecified.
Bug: webrtc:11121
Change-Id: I19b669b0735d78e269e19eaae86c2d7d95a91141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242968
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35651}
This reverts commit 37ee0f5e594dd772ec6d620b5e5ea8a751b684f0.
Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642
Original change's description:
> Use backticks not vertical bars to denote variables in comments for /pc
>
> Bug: webrtc:12338
> Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34575}
TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34577}
jitterBufferDelay and jitterBufferEmittedCount are defined
in RTCMediaStreamStats for both audio and video.
But for video, they were not populated in RTCInboundRtpStreamStats.
Bug: webrtc:12910
Change-Id: I135d473f055ecfb2c39b078ccf18c1bb9bc4f210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224280
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34398}
Relanding after adding to chromium stats whitelist:
https://chromium-review.googlesource.com/c/chromium/src/+/2983329
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I3633ee428d07b283b0cc503a84d8fa2e79415dfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223761
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34367}
This reverts commit a27cfbffdfa0bf359628d2164db5b9d6321f9c9c.
Reason for revert: WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise failing.
Original change's description:
> Fix echo return loss stats and add to RTCAudioSourceStats.
>
> This solves two problems:
> * Echo return loss stats weren't being gathered in Chrome, because they
> need to be taken from the audio processor attached to the track
> rather than the audio send stream.
> * The standardized location is in RTCAudioSourceStats, not
> RTCMediaStreamTrackStats. For now, will populate the stats in both
> locations.
>
> Bug: webrtc:12770
> Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34344}
TBR=deadbeef@webrtc.org,hbos@webrtc.org,hbos@chromium.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I6b2587d762f005adef67c0d5121f1b58c3b76688
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12770
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223068
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34352}
This solves two problems:
* Echo return loss stats weren't being gathered in Chrome, because they
need to be taken from the audio processor attached to the track
rather than the audio send stream.
* The standardized location is in RTCAudioSourceStats, not
RTCMediaStreamTrackStats. For now, will populate the stats in both
locations.
Bug: webrtc:12770
Change-Id: I47eaf7f2b50b914a1be84156aa831e27497d07e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223182
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34344}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
The RTCReceivedRtpStreamStats hierarchy, which inherit from
RTCRtpStreamStats, already contain members ssrc, kind, codec_id and
transport_id so there's no need to list them inside
RTCRemoteInboundrtpStreamStats.
This CL removes duplicates so that we don't DCHECK-crash on Android,
and adds a unit test ensuring we never accidentally list the same
member twice.
Bug: webrtc:12658
Change-Id: I27925eadddc6224bf6d6a91784ed7cafd7a4cfb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214343
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33649}
Follow up https://webrtc-review.googlesource.com/c/src/+/210340, |RTCReceivedRtpStreamStats| is the new parent of |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats| so the verification structure in test should change accrodingly.
Bug: webrtc:12532
Change-Id: I0e7a832de2bb60ec68fb963a8846f6b15fdc30a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214082
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Di Wu <meetwudi@gmail.com>
Cr-Commit-Position: refs/heads/master@{#33642}
Deleting obsolete stats. Spec: https://www.w3.org/TR/webrtc-stats/
1. RTCInbound/OutboundRtpStats.isRemote: No longer useful with remote stream stats
2. RTCIceCandidateStats.deleted: This field was obsoleted because if the ICE candidate is deleted it no longer appears in getStats()
I also marked as many other obsoleted stats possible according to spec. I am not as confident to delete them but feel free to comment to let me know if anything is off / can be deleted.
Bug: webrtc:12583
Change-Id: I688d0076270f85caa86256349753e5f0e0a44931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211781
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33549}
Changes:
- adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1])
- collection of remote outbound stats (only for audio streams)
- adding `remote_id` to the inbound stats and set with the ID of the
corresponding remote outbound stats only if the latter are available
- unit tests
[1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats
Tested: verified from chrome://webrtc-internals during an appr.tc call
Bug: webrtc:12529
Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33545}
In refactoring CL https://webrtc-review.googlesource.com/c/src/+/210340,
the RTCRemoteInboundRtpStreamStats hierarchy was updated to inherit from
RTCReceivedRtpStreamStats but we forgot to update the
WEBRTC_RTCSTATS_IMPL() macro to say that RTCReceivedRtpStreamStats is
the parent. As a consequence, RTCReceivedRtpStreamStats's members
(jitter and packetsLost) were not included when iterating over all
members of RTCRemoteInboundRtpStreamStats, which means these two merics
stopped being exposed to JavaScript in Chromium.
There is sadly no way to safe-guard against this, but the fix is simple.
TBR=hta@webrtc.org,meetwudi@gmail.com
Bug: webrtc:12532
Change-Id: I0179dad6eaa592ee36cfe48978f2fc22133b8f45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212866
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33543}
Spec: https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict*
According to the spec, |RTCReceivedRtpStreamStats| is the base class for |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. This structure isn't visible in JavaScript but it's important to bring it up to spec for the C++ part. This CL adds the barebone |RTCReceivedRtpStreamStats| with a bunch of TODOs for later migrations.
This commit makes the minimum |RTCReceivedRtpStreamStats| and rebase |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats| to use the new class as the parent class.
This commit also moves |jitter| and |packets_lost| to |RTCReceivedRtpStreamStats|, from |RTCInboundRtpStreamStats| and |RTCRemoteInboundRtpStreamStats|. Moving these two first because they are the two that exist in both subclasses for now.
Bug: webrtc:12532
Change-Id: I0ec74fd241f16c1e1a6498b6baa621ca0489f279
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/210340
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33435}
Spec: https://www.w3.org/TR/webrtc-stats/#dom-rtcvideosourcestats-frames
Wiring up the "frames" stats with the cumulative fps counter on the video source.
Tests:
./out/Default/peerconnection_unittests
./out/Default/video_engine_tests
Bug: webrtc:12499
Change-Id: I4103f56ed04cb464f5f7e70fbf2d77c25a867a68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208782
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33404}
Tests:
./out/Default/peerconnection_unittests
Manually tested with Chromium to see the data populated
Spec: https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
Bug: webrtc:12506
Change-Id: I60ef8061fb31deab06ca5f115246ceb5a8cdc5ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33361}
Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!
Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33325}
