According to the spec, RID values should be constrained to only
alpha-numeric values. This was not enforced in our implementation to
allow for more flexibility.
It has been brought to our attention that some values that we currently
consider legal (such as the '~', '=' ';' characters) might cause confusion
with the simulcast syntax that uses these characters to indicate other
meanings.
What's worse, is that some characters, when used in RIDs (such as
\u{1f937} \u{1f4a9} and \u{1f926}) cause uncontrollable laughter for some
users which might also be a health hazard.
This change resolves these issues by restricting RIDs to alpha-numeric.
Bug: webrtc:10491
Change-Id: I16e262c87525d0289764beacd098e1525a355463
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132061
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27499}
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}
We already support decoding of the x-mt line. This change adds the
a=x-mt line to the SDP offer. This is not a backward compatible change
for media transport (because of the changes in pre-shared key handling)
1) if media transport is enabled, and SDES is enabled, generate the
media transport offer.
2) if media transport generated the offer, add that offer to the x-mt
line.
3) in order to create media transport, require an x-mt line (backward incompatible).
The way it works is that
1) PeerConnection, on the offerer, asks jsep transport for the
configuration of the media transport.
2) Tentative media transport is created in JsepTransportController when
that happens.
3) SessionDescription will include configuration from this tentative
media transport.
4) When the LocalDescription is set on the offerer, the tentative media
transport is promoted to the real media transport.
Caveats:
- now we really only support MaxBundle. In the previous implementations,
two media transports were briefly created in some tests, and the second
one was destroyed shortly after instantiation.
- we, for now, enforce SDES. In the future, whether SDES is used will be
refactored out of the peer connection.
In the future (on the callee) we should ignore 'is_media_transport' setting. If
Offer contains x-mt, media transport should be used (if the factory is
present). However, we need to decide how to negotiate media transport
for data channels vs data transport for media (x-mt line at this point
doesn't differentiate the two, so we still need to use app setting).
This change also removes the negotation of pre-shared key from the
a=crypto line. Instead, media transport will have its own, 256bit key.
Such key should be transported in the x-mt line. This makes the code
much simpler, and simplifies the dependency / a=crypto lines parsing.
Also, adds a proper test for the connection re-offer (on both sides: callee and caller).
Before, it was possible that media transport could get recreated, based on the offer.
The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test.
This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even
when there is a re-offer.
Bug: webrtc:9719
Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01
Reviewed-on: https://webrtc-review.googlesource.com/c/125040
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26933}
This involves inserting an extra layer between jsep_transport_controller
and the cricket::SctpTransportInternal layer. The objects at this layer
are reference counted.
Bug: chromium:818643
Change-Id: Ibed57c4a538de981cee63e0f7f1f319f029cab39
Reviewed-on: https://webrtc-review.googlesource.com/c/123884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26889}
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}
This reverts commit 05d43c6f7fe497fed0f2c8714e2042dd07a86df2.
Reason for revert: It breaks some Chromium trybots:
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_asan_rel_ng/206387https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_tsan_rel_ng/207737https://ci.chromium.org/p/chromium/builders/luci.chromium.try/win10_chromium_x64_rel_ng/202283
Original change's description:
> Fix getStats() freeze bug affecting Chromium but not WebRTC standalone.
>
> PeerConnection::Close() is, per-spec, a blocking operation.
> Unfortunately, PeerConnection is implemented to own resources used by
> the network thread, and Close() - on the signaling thread - destroys
> these resources. As such, tasks run in parallel like getStats() get into
> race conditions with Close() unless synchronized. The mechanism in-place
> is RTCStatsCollector::WaitForPendingRequest(), it waits until the
> network thread is done with the in-parallel stats request.
>
> Prior to this CL, this was implemented by performing
> rtc::Thread::ProcessMessages() in a loop until the network thread had
> posted a task on the signaling thread to say that it was done which
> would then get processed by ProcessMessages(). In WebRTC this works, and
> the test is RTCStatsIntegrationTest.GetsStatsWhileClosingPeerConnection.
>
> But because Chromium's thread wrapper does no support
> ProcessMessages(), calling getStats() followed by close() in Chrome
> resulted in waiting forever (https://crbug.com/850907).
>
> In this CL, the process messages loop is removed. Instead, the shared
> resources are guarded by an rtc::Event. WaitForPendingRequest() still
> blocks the signaling thread, but only while shared resources are in use
> by the network thread. After this CL, calling WaitForPendingRequest() no
> longer has any unexpected side-effects since it no longer processes
> other messages that might have been posted on the thread.
>
> The resource ownership and threading model of WebRTC deserves to be
> revisited, but this fixes a common Chromium crash without redesigning
> PeerConnection, in a way that does not cause more blocking than what
> the other PeerConnection methods are already doing.
>
> Note: An alternative to using rtc::Event is to use resource locks and
> to not perform the stats collection on the network thread if the
> request was cancelled before the start of processing, but this has very
> little benefit in terms of performance: once the network thread starts
> collecting the stats, it would use the lock until collection is
> completed, blocking the signaling thread trying to acquire that lock
> anyway. This defeats the purpose and is a riskier change, since
> cancelling partial collection in this inherently racy edge-case would
> have observable differences from the returned stats, which may cause
> more regressions.
>
> Bug: chromium:850907
> Change-Id: Idceeee0bddc0c9d5518b58a2b263abb2bbf47cff
> Reviewed-on: https://webrtc-review.googlesource.com/c/121567
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26707}
TBR=steveanton@webrtc.org,hbos@webrtc.org
Change-Id: Icd82cdd5bd086a90999f7fd5f8616e1f2d2153bf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:850907
Reviewed-on: https://webrtc-review.googlesource.com/c/123225
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26721}
PeerConnection::Close() is, per-spec, a blocking operation.
Unfortunately, PeerConnection is implemented to own resources used by
the network thread, and Close() - on the signaling thread - destroys
these resources. As such, tasks run in parallel like getStats() get into
race conditions with Close() unless synchronized. The mechanism in-place
is RTCStatsCollector::WaitForPendingRequest(), it waits until the
network thread is done with the in-parallel stats request.
Prior to this CL, this was implemented by performing
rtc::Thread::ProcessMessages() in a loop until the network thread had
posted a task on the signaling thread to say that it was done which
would then get processed by ProcessMessages(). In WebRTC this works, and
the test is RTCStatsIntegrationTest.GetsStatsWhileClosingPeerConnection.
But because Chromium's thread wrapper does no support
ProcessMessages(), calling getStats() followed by close() in Chrome
resulted in waiting forever (https://crbug.com/850907).
In this CL, the process messages loop is removed. Instead, the shared
resources are guarded by an rtc::Event. WaitForPendingRequest() still
blocks the signaling thread, but only while shared resources are in use
by the network thread. After this CL, calling WaitForPendingRequest() no
longer has any unexpected side-effects since it no longer processes
other messages that might have been posted on the thread.
The resource ownership and threading model of WebRTC deserves to be
revisited, but this fixes a common Chromium crash without redesigning
PeerConnection, in a way that does not cause more blocking than what
the other PeerConnection methods are already doing.
Note: An alternative to using rtc::Event is to use resource locks and
to not perform the stats collection on the network thread if the
request was cancelled before the start of processing, but this has very
little benefit in terms of performance: once the network thread starts
collecting the stats, it would use the lock until collection is
completed, blocking the signaling thread trying to acquire that lock
anyway. This defeats the purpose and is a riskier change, since
cancelling partial collection in this inherently racy edge-case would
have observable differences from the returned stats, which may cause
more regressions.
Bug: chromium:850907
Change-Id: Idceeee0bddc0c9d5518b58a2b263abb2bbf47cff
Reviewed-on: https://webrtc-review.googlesource.com/c/121567
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26707}
This is a reland of ce470aab518f067a67aa03aaab1fc61a45afa0ec
Original change's description:
> Enabling Simulcast use via AddTransceiver.
>
> This change removes the restriction on multiple send encodings when
> calling AddTransceiver. If RIDs are not provided in the
> simulcast scenario, they are auto-generated by the platform.
>
> This effectively enables the use of spec-compliant simulcast.
> Tests are also added to verify simulcast functionality.
>
> Bug: webrtc:10075
> Change-Id: I088feba70a26e85abcc7bfbe3ea1fe5103cd47d2
> Reviewed-on: https://webrtc-review.googlesource.com/c/121389
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26590}
Bug: webrtc:10075
Change-Id: Ib4392e36d5d60b966ecff56d4df69bf60d13a73d
Reviewed-on: https://webrtc-review.googlesource.com/c/122962
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26694}
This change removes the restriction on multiple send encodings when
calling AddTransceiver. If RIDs are not provided in the
simulcast scenario, they are auto-generated by the platform.
This effectively enables the use of spec-compliant simulcast.
Tests are also added to verify simulcast functionality.
Bug: webrtc:10075
Change-Id: I088feba70a26e85abcc7bfbe3ea1fe5103cd47d2
Reviewed-on: https://webrtc-review.googlesource.com/c/121389
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26590}
Enables downstream projects to use the existing fake ice transport implementation, without taking dependency on gunit
Bug: None
Change-Id: I78bac9d40aa6e12b55e86f0460bcd98d85c7f214
Reviewed-on: https://webrtc-review.googlesource.com/c/120445
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26456}
UniqueIdGenerator classes are useful outside the pc directory.
This change moves them to the rtc_base directory to enable code
in all directories to reference them.
Bug: None
Change-Id: I1c77da87ea26d9611f37dc1d4d2c16006a6589c6
Reviewed-on: https://webrtc-review.googlesource.com/c/119460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26378}
This CL adds a single class to manage the use case of having a task
that repeats itself by a fixed or variable interval. It replaces the
repeating task previously locally defined for rtp transport controller
send as well as the cancelable periodic task. Furthermore, it is
introduced where one off repeating tasks were created before.
It provides the currently used functionality of the cancelable periodic
task, but not some of the unused features, such as allowing cancellation
of tasks before they are started and cancellation of a task after the
owning task queue has been destroyed.
Bug: webrtc:9883
Change-Id: Ifa7edee836c2a64fce16a7d0f682eb09c879eaca
Reviewed-on: https://webrtc-review.googlesource.com/c/116182
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26313}
This wires up the "state change" event and defines an observer
class that can be used by clients.
Bug: chromium:907849
Change-Id: I3cba2dc051a56280fb958f139f29cbb0022a39c6
Reviewed-on: https://webrtc-review.googlesource.com/c/114884
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26120}
This change adds Simulcast negotiation to media session offers/answers.
Next step is to add negotiation logic to PeerConnection.
Bug: webrtc:10075
Change-Id: Iea3a1084c16058f0efbc974cf623ec05c3c7a74f
Reviewed-on: https://webrtc-review.googlesource.com/c/115790
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26115}
The BBR controller can still be injected, but the trials
will no longer work. This reduces the binary size.
Bug: webrtc:8415
Change-Id: I2c32c414d08ef0cc16bfd72651535a755cde9916
Reviewed-on: https://webrtc-review.googlesource.com/c/114120
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26077}
This includes a refactoring of jseptransport to store a refcounted
object instead of a std::unique_ptr to the cricket::DtlsTransport.
Bug: chromium:907849
Change-Id: Ib557ce72c2e6ce8af297c2b8deb7ec3a103d6d31
Reviewed-on: https://webrtc-review.googlesource.com/c/111920
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25831}
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
This CL moves webrtc::CreatePeerConnectionFactory definitions out of
pc:create_pc_factory and merges it with its declaration in the api/
directory.
In order to avoid circular dependencies a new build target is created:
* api:create_peerconnection_factory
Bug: webrtc:9862
Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af
Reviewed-on: https://webrtc-review.googlesource.com/c/111186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25744}
Fixing bug that allows a channel to be set on a stopped transceiver.
This CL contains the following refactoring:
1. Extracted ChannelInterface from BaseChannel
2. Unified SetXxxMediaChannel (Voice, Video) into SetMediaChannel
Bug: webrtc:9932
Change-Id: I2fbf00c823b7848ad4f2acb6e80b1b58ac45ee38
Reviewed-on: https://webrtc-review.googlesource.com/c/110564
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25641}
This changes PeerConnection to allow sending and receiving data channel
messages over the media transport. If |use_media_transport_for_data_channels|
is set, PeerConnection will use a DCT_MEDIA_TRANSPORT mode for data
channels.
DCT_MEDIA_TRANSPORT acts exactly like DCT_SCTP within the data channel
and peer connection layers. On the transport layer, it uses the media
transport instead of SCTP. It appears as an RTP data channel in SDP
(just as media over media-transport appears as RTP in SDP).
Bug: webrtc:9719
Change-Id: I6a90142bd3f43668479c825ed02689dcd0d58b78
Reviewed-on: https://webrtc-review.googlesource.com/c/109740
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25575}
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782
This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.
Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.
One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.
Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
This reverts commit 61c6e5643e7ea058e653956980a90e033249c055.
Reason for revert: downstream projects prepared for this change
Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
>
> This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
>
> Reason for revert: breaking downstream
>
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> >
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> >
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> >
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
>
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
>
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
Reason for revert: breaking downstream
Original change's description:
> Isolating APM API build target: making :api an actual target.
>
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
>
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
>
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
