GN recently added support for Apple frameworks to link, rather than
overloading the libs lists. This pulls .frameworks out of the libs
lists, so that GN can stop supporting .frameworks in libs in the
future.
Bug: chromium:1052560
Change-Id: I263230ddd3c468061584423bba9e1f887503bcaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178601
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sylvain Defresne <sdefresne@chromium.org>
Cr-Commit-Position: refs/heads/master@{#31632}
This is a reland of 19df870d924662e3b6efb86078d31a8e086b38b5
Patchset 1 is the original.
Subsequent patchset changes threadchecker that crashed with downstream
code.
Original change's description:
> Reland "Allows FEC generation after pacer step."
>
> This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0
>
> Patchset 2 contains a fix. Old code can in factor call
> RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
> is not supported there - we shouldn't crash.
>
> Original change's description:
> > Allows FEC generation after pacer step.
> >
> > Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> > This CL enables FEC packets to be generated as media packets are sent,
> > rather than generated, i.e. media packets are inserted into the fec
> > generator after the pacing stage rather than at packetization time.
> >
> > This may have some small impact of performance. FEC packets are
> > typically only generated when a new packet with a marker bit is added,
> > which means FEC packets protecting a frame will now be sent after all
> > of the media packets, rather than (potentially) interleaved with them.
> > Therefore this feature is currently behind a flag so we can examine the
> > impact. Once we are comfortable with the behavior we'll make it default
> > and remove the old code.
> >
> > Note that this change does not include the "protect all header
> > extensions" part of the original CL - that will be a follow-up.
> >
> > Bug: webrtc:11340
> > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31558}
>
> Bug: webrtc:11340
> Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31613}
Bug: webrtc:11340
Change-Id: Ib741c8c284f523c959f8aca454088d9eee7b17f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178600
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31619}
This reverts commit 19df870d924662e3b6efb86078d31a8e086b38b5.
Reason for revert: Downstream project failure
Original change's description:
> Reland "Allows FEC generation after pacer step."
>
> This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0
>
> Patchset 2 contains a fix. Old code can in factor call
> RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
> is not supported there - we shouldn't crash.
>
> Original change's description:
> > Allows FEC generation after pacer step.
> >
> > Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> > This CL enables FEC packets to be generated as media packets are sent,
> > rather than generated, i.e. media packets are inserted into the fec
> > generator after the pacing stage rather than at packetization time.
> >
> > This may have some small impact of performance. FEC packets are
> > typically only generated when a new packet with a marker bit is added,
> > which means FEC packets protecting a frame will now be sent after all
> > of the media packets, rather than (potentially) interleaved with them.
> > Therefore this feature is currently behind a flag so we can examine the
> > impact. Once we are comfortable with the behavior we'll make it default
> > and remove the old code.
> >
> > Note that this change does not include the "protect all header
> > extensions" part of the original CL - that will be a follow-up.
> >
> > Bug: webrtc:11340
> > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31558}
>
> Bug: webrtc:11340
> Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31613}
TBR=sprang@webrtc.org,srte@webrtc.org
Change-Id: I3b2b25898ce88b64c2322f68ef83f9f86ac2edb0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178563
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31614}
This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0
Patchset 2 contains a fix. Old code can in factor call
RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec
is not supported there - we shouldn't crash.
Original change's description:
> Allows FEC generation after pacer step.
>
> Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> This CL enables FEC packets to be generated as media packets are sent,
> rather than generated, i.e. media packets are inserted into the fec
> generator after the pacing stage rather than at packetization time.
>
> This may have some small impact of performance. FEC packets are
> typically only generated when a new packet with a marker bit is added,
> which means FEC packets protecting a frame will now be sent after all
> of the media packets, rather than (potentially) interleaved with them.
> Therefore this feature is currently behind a flag so we can examine the
> impact. Once we are comfortable with the behavior we'll make it default
> and remove the old code.
>
> Note that this change does not include the "protect all header
> extensions" part of the original CL - that will be a follow-up.
>
> Bug: webrtc:11340
> Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31558}
Bug: webrtc:11340
Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31613}
Add TimeController to the CreatePeerConnectionE2EQualityTestFixture
method as a first step to make PC level framework compatible with
TimeController abstraction.
Bug: webrtc:11743
Change-Id: I69305abc880059bf9fe1d4f2e3b7c10cf35417db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178485
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31607}
This is a reland of b46df3da44c42f6e5055c69a8247a344887108ea
Test case for issue that caused revert added:
https://webrtc-review.googlesource.com/c/src/+/178203
Fix for issue that caused revert:
https://webrtc-review.googlesource.com/c/src/+/178207
Original change's description:
> Reland "Removes lock release in PacedSender callback."
>
> This is a reland of 6b9c60b06d04bc519195fca1f621b10accfeb46b
>
> Original change's description:
> > Removes lock release in PacedSender callback.
> >
> > The PacedSender currently has logic to temporarily release its internal
> > lock while sending or asking for padding.
> > This creates some tricky situations in the pacing controller where we
> > need to consider if some thread can enter while we the process thread is
> > actually processing, just temporarily busy sending.
> >
> > Since the pacing call stack is no longer cyclic, we can actually remove
> > this lock-release now.
> >
> > Bug: webrtc:10809
> > Change-Id: Ic59c605252bed1f96a03406c908a30cd1012f995
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173592
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31206}
>
> Bug: webrtc:10809
> Change-Id: Id39fc49b0a038e7ae3a0d9818fb0806c33ae0ae0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175656
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31332}
Bug: webrtc:10809
Change-Id: I1dba507220316008c0f3b278df4b732011f257eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178384
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31588}
https://webrtc-review.googlesource.com/c/src/+/178100 reverted a change
that could result in a deadlock if WebRTC-Audio-SendSideBwe was enabled
and WebRTC-Audio-ABWENoTWCC was not while using send-side BWE in a
mixed audio/video setting.
This CL adds an integration test that fails on tsan if above commit is
cherry-picked.
Bug: webrtc:10809
Change-Id: I5028d5794e5c9e970ccd9b7eb25d5b76a7fa4e58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178203
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31574}
This reverts commit b46df3da44c42f6e5055c69a8247a344887108ea.
Reason for revert: May cause deadlock.
Original change's description:
> Reland "Removes lock release in PacedSender callback."
>
> This is a reland of 6b9c60b06d04bc519195fca1f621b10accfeb46b
>
> Original change's description:
> > Removes lock release in PacedSender callback.
> >
> > The PacedSender currently has logic to temporarily release its internal
> > lock while sending or asking for padding.
> > This creates some tricky situations in the pacing controller where we
> > need to consider if some thread can enter while we the process thread is
> > actually processing, just temporarily busy sending.
> >
> > Since the pacing call stack is no longer cyclic, we can actually remove
> > this lock-release now.
> >
> > Bug: webrtc:10809
> > Change-Id: Ic59c605252bed1f96a03406c908a30cd1012f995
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173592
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31206}
>
> Bug: webrtc:10809
> Change-Id: Id39fc49b0a038e7ae3a0d9818fb0806c33ae0ae0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175656
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31332}
TBR=sprang@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10809
Change-Id: I6b06bafad8cd9eeb22107d04b953fd14b8131afa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31564}
This reverts commit 75fd127640bdf1729af6b4a25875e6d01f1570e0.
Reason for revert: Breaks downstream test
Original change's description:
> Allows FEC generation after pacer step.
>
> Split out from https://webrtc-review.googlesource.com/c/src/+/173708
> This CL enables FEC packets to be generated as media packets are sent,
> rather than generated, i.e. media packets are inserted into the fec
> generator after the pacing stage rather than at packetization time.
>
> This may have some small impact of performance. FEC packets are
> typically only generated when a new packet with a marker bit is added,
> which means FEC packets protecting a frame will now be sent after all
> of the media packets, rather than (potentially) interleaved with them.
> Therefore this feature is currently behind a flag so we can examine the
> impact. Once we are comfortable with the behavior we'll make it default
> and remove the old code.
>
> Note that this change does not include the "protect all header
> extensions" part of the original CL - that will be a follow-up.
>
> Bug: webrtc:11340
> Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31558}
TBR=sprang@webrtc.org,srte@webrtc.org
Change-Id: Ie714e5f68580cbd57560e086c9dc7292a052de5f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177983
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31559}
Split out from https://webrtc-review.googlesource.com/c/src/+/173708
This CL enables FEC packets to be generated as media packets are sent,
rather than generated, i.e. media packets are inserted into the fec
generator after the pacing stage rather than at packetization time.
This may have some small impact of performance. FEC packets are
typically only generated when a new packet with a marker bit is added,
which means FEC packets protecting a frame will now be sent after all
of the media packets, rather than (potentially) interleaved with them.
Therefore this feature is currently behind a flag so we can examine the
impact. Once we are comfortable with the behavior we'll make it default
and remove the old code.
Note that this change does not include the "protect all header
extensions" part of the original CL - that will be a follow-up.
Bug: webrtc:11340
Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31558}
For stream sizes that were not multiple of 4, we could end up causing
a size_t wraparound which resulted in an infinite loop.
Bug: webrtc:9510
Change-Id: Ie3fe5345e1477efa6a4ec338bd9f9b00225e688e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177005
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31503}
The //third_party/abseil-cpp:absl target is currently a group that
depends on all the targets needed by WebRTC in Chromium.
It will be switched to a component starting from
https://chromium-review.googlesource.com/c/chromium/src/+/2174434.
Bug: chromium:1046390
Change-Id: I70d450fdbfa895084b481c9884b6361d2fb9580d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176901
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31498}
It may happen that if we have simulcast track with, let's say, 2 streams
A and B, we can receive frame X on A and then receive it again on B
when there is a switch from A to B. TO correctly handle it we need to
skip second receive of X. Later we need to add metric which will show
how many frames were in between when X was received twice.
Bug: webrtc:11557
Change-Id: I8c52a78674b62387f520a587f51e209ed7c0b0bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176853
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31488}
Unit test added to verify root cause is fixed.
Scenario test added to verify high-level behavior.
Bug: webrtc:11654
Change-Id: I1ad6e2750f5272e86b4198749edbbf5dfd8315c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176564
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31462}
Queue with multiple heads is planned to be used in
DefaultVideoQualityAnalyzer to store stream state. Stream state contains
ordered sequence of frame ids that were send for this video stream.
When frame is received by one receiver it should be removed from state
for that receiver and kept for others.
How it is used can be found in this CL:
https://webrtc-review.googlesource.com/c/src/+/176411
Bug: webrtc:11631
Change-Id: Ic7fabf4d77131805a91f08a2ccfffc73c08d3e2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176402
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31444}
This change introduces a new non-reentrant mutex to WebRTC. It
enables eventual migration to Abseil's mutex.
The mutex types supportable by webrtc::Mutex are
- absl::Mutex
- CriticalSection (Windows only)
- pthread_mutex (POSIX only)
In addition to introducing the mutexes, the CL also changes
PacketBuffer to use the new mutex instead of rtc::CriticalSection.
The method of yielding from critical_section.cc was given a
mini-cleanup and YieldCurrentThread() was added to
rtc_base/synchronization/yield.h/cc.
Additionally, google_benchmark benchmarks for the mutexes were added
(test courtesy of danilchap@), and some results from a pthread/Abseil
shootout were added showing Abseil has the advantage in higher
contention.
Bug: webrtc:11567, webrtc:11634
Change-Id: Iaec324ccb32ec3851bf6db3fd290f5ea5dee4c81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176230
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31443}
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.
Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.
The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.
Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
The absl/flags/flag.h header is not #including absl/flags/declare.h
starting from [1] so this transitive #include needs to be removed.
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/2228841
Bug: None
Change-Id: I06e78ed05e0fb570a9ecc8621ec3ae5298fffd1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176444
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31433}
To support multiple participants video quality analyzer may need to know
peer names in advance to simplify internal structures and metrics
reporting.
Bug: webrtc:11631
Change-Id: I4ffb1554ab7f0e015b8e937b7ffddd55aba9826f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176364
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31415}
Add peer name to video quality analyzer interface to make it possible to
add multipeer support.
Change-Id: I2570cd4481503c8634bdd91208b3dd2fa1d62029
Bug: webrtc:11631
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176329
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31395}
Increase test duration to make at least one frame to come through on slow
test bots and remove check in echo emulation for same purposes. Logging
for echo queue should be enough.
Bug: None
Change-Id: I0d2d1c2a87e1a2b4cd035828443f428b0983edad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176300
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31377}
This reduces the number of threads allocated per PeerConnection when
more than one PC is needed.
Bug: webrtc:11598
Change-Id: I3c1fd71705f90c4b4bbb1bc3f0f659c94016e69a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175904
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31347}
This is a reland of 6b9c60b06d04bc519195fca1f621b10accfeb46b
Original change's description:
> Removes lock release in PacedSender callback.
>
> The PacedSender currently has logic to temporarily release its internal
> lock while sending or asking for padding.
> This creates some tricky situations in the pacing controller where we
> need to consider if some thread can enter while we the process thread is
> actually processing, just temporarily busy sending.
>
> Since the pacing call stack is no longer cyclic, we can actually remove
> this lock-release now.
>
> Bug: webrtc:10809
> Change-Id: Ic59c605252bed1f96a03406c908a30cd1012f995
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173592
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31206}
Bug: webrtc:10809
Change-Id: Id39fc49b0a038e7ae3a0d9818fb0806c33ae0ae0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175656
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31332}
