Fixes an issue with NullAudioPoller calling the mixer every 10 ms when
no call is ongoing.
Bug: b/142775365
Change-Id: I77eeddadaf08b358cce2b389c70e4f2baf1d5627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157176
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29508}
This was just checked in all places were it was used, moving the check
into RtpRtcp reduces the boiler plate required at the call sites.
Also changing to always register and unregister extensions by URI to
synchronize the code in AudioSendStream with the code in RtpVideoSender.
This prepares for reducing the scope of ChannelSend.
Bug: webrtc:9883
Change-Id: Ia64d79f20eb98f46cbbbe8318770e4fcf9caa1ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29490}
That allows to use SingleThreadedTaskQueueForTesting via TaskQueueBase interface
but still have access to test-only SendTask function.
Bug: webrtc:10933
Change-Id: I3cc397e55ea2f1ed9e5d885d6a2ccda412beb826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29480}
Well, in fact we need to return both. But return codec sample rate
separately and let the SdpAudioFormat contain the RTP clockrate,
otherwise we're essentially lying to our callers.
Bug: webrtc:11028
Change-Id: I40f36cb9db6b9824404ade6b0515a8312ff97009
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156307
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29444}
There's already a DCHECK at construction time ensuring that it's set.
Bug: webrtC:9883
Change-Id: I9f41b77273bb859626546ab3534d483d9172ea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155581
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29393}
The pattern of using a static function rather than a regular function is
not very well motivated and we don't do that in other places.
To maintain consistency over the code base this Cl replaces those static
modifier functions with regular member functions.
Bug: webrtc:9883
Change-Id: I8edd1781d98905de82722458a0d272af90689a2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155522
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29391}
This improves the accuracy of the priority bitrate on IPv6 networks
and when the min frame length is longer than 20 ms. Unless either of
those are true, there's no significant change in behavior.
Bug: webrtc:11001
Change-Id: I29530655cb607a8e7e8186431cd9362ca397910b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155521
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29375}
This is a no-op change that just removes the AudioAllocationSettings
helper class that was previously introduced since the field trials in it
were used in several places. Those other usages has now been removed
and AudioSendStream is now the only user. By moving the trials directly
to AudioSendStream we reduce the reader overhead when trying to follow
what a particular field trial does.
The "WebRTC-Audio-ForceNoTWCC" trial was removed as it is always set
together with "WebRTC-Audio-ABWENoTWCC".
Bug: webrtc:9883
Change-Id: Ib63589255bfe7adb155ea41279bdcd153f1536c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155366
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29371}
This adds the ability to disable legacy overhead calculation so we'll
use the available data on per packet over head and frame length range
to set the min and max total allocatable bitrate.
Bug: webrtc:11001
Change-Id: I2a94499433e15bad11a08f81fe7f1dfc27982cdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155175
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29368}
actual task queue implementation for these tests is intentionally unchanged for now.
while at it, change return type of created transports to unique_ptr to note passing ownership.
Bug: webrtc:10933
Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29335}
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
This class doesn't strictly follow rtc::TaskQueue semantics,
which makes it surprising and hard to use correctly.
Please use TaskQueueForTest instead.
This CL follows usual deprecation process:
1/ Rename.
% for i in `git ls-files` ; sed -i "s:SingleThreadedTaskQueueForTesting:DEPRECATED_SingleThreadedTaskQueueForTesting:" $i
2/ Annotate old name for downstream users and accidental new uses.
Bug: webrtc:10933
Change-Id: I80b4ee5a48df1f63f63a43ed0efdb50eb7fb156a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150788
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29045}
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.
Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
This metric is not used anywhere and is not calculated correctly when the delay manager is in relative arrival delay mode.
Bug: webrtc:10333
Change-Id: Iac79ab40b79b17802ad9d626c130e82f761bae26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150786
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29037}
These methods were defined, and called, but not doing anything.
Bug: None
Change-Id: I9955843a6bd86e4a583b0213ddb6b3b42e2ab815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150792
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29020}
Also annotate a few of the remaining uses, to guide further splits of
that large build target.
Bug: webrtc:8733
Change-Id: I16ac33ab48e6d39a1a8dbc2a3fc671d8db6dbfe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150789
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29001}
The new target does not depend on libjingle_peerconnection_api, and to
do this, the named "audio" and "video" string literals had to be moved from
media_stream_interface.cc to media_types.cc.
In this cl, the dependency on libjingle_peerconnection_api can be
dropped from a few targets.
No-Presubmit: True
Bug: webrtc:8733
Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28998}
This interface/config field is now unused, let's remove it.
Bug: webrtc:10633
Change-Id: I56ff3d47ba784d973de411ada52ec9485bad9864
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150531
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28978}
The methods are no longer in use, this CL cleans away references and
updates any tests using them.
Bug: webrtc:10633
Change-Id: I2db301e0a021a2f85a8b9a74e409303baba407da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28956}
And a corresponding struct RtpReceiveStats. This is intended
to hold the information exposed via GetStats, which is quite
different from the stats reported to the peer via RTCP.
This is a preparation for moving ReceiveStatistics out of the
individual receive stream objects, and instead have a shared instance
owned by RtpStreamReceiverController or maybe Call.
Bug: webrtc:10679,chromium:677543
Change-Id: Ibb52ee769516ddc51da109b7f2319405693be5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148982
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28943}
It's propagated from ReceiveStatistics up to VoiceReceiverInfo,
and then not used. It's not part of the standard stats.
Bug: None
Change-Id: I90ce6a72e3ca846adbbba5d3023fef18a2169018
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149164
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28933}
The SetSSRC() method is slated for removal, make sure we set the local
SSRC at construction time.
Bug: webrtc:10774
Change-Id: I431e828caf60c5e0134adbe82d1d3345745cc6ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149827
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28926}
The name media_send_ssrc makes less sense when used mostly for the
RtcpReceiver functionality.
The old member is still there and used as a fallback. That will be
cleaned away after downstream code is fixed.
Bug: webrtc:10774
Change-Id: I4ec18db76910f31dfe76bc9b137ffe89220d3fa8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149836
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28923}
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.
The new stat will not be reported unless it is GT 0.
Reporting of decoding_codec_plc events
Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
Some of the macros in format_macros.h follow the C standard and try to fill holes in it (on Windows). But this one has no direct equivalent in the standard and is just mimicking the naming convention. That's not nice.
References:
https://devblogs.microsoft.com/cppblog/c99-library-support-in-visual-studio-2013/https://stackoverflow.com/a/2524673
Change-Id: I53f3faca2976a5b5d4b04a67ffb56ae0f4e930b2
Bug: webrtc:10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147862
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28794}
This reverts commit 67008dfb366237469fe088a61b62c0cad852c024.
Reason for revert: Tests in the Chromium repo have been changed to accomodate this CL: https://chromium-review.googlesource.com/c/chromium/src/+/1728565
Original change's description:
> Revert "Replace the implementation of `GetContributingSources()` on the audio side."
>
> This reverts commit 8fa7151e4bbad40fec1f964fe0c003b8787bb78a.
>
> Reason for revert: Speculative revert to fix roll of webrtc into chrome. Right now tests related to RTCRtpReceiver failing and looks like it is main candidate, who can affect that behavior.
>
> Original change's description:
> > Replace the implementation of `GetContributingSources()` on the audio side.
> >
> > This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the audio side with the spec-compliant `SourceTracker`-implementation.
> >
> > The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network.
> >
> > This change is almost identical to the previous video side change at: https://webrtc-review.googlesource.com/c/src/+/143177
> >
> > Bug: webrtc:10545
> > Change-Id: Ife7f08ee8ca1346099b7466837a3756947085fc5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144422
> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> > Commit-Queue: Chen Xing <chxg@google.com>
> > Cr-Commit-Position: refs/heads/master@{#28459}
>
> TBR=ossu@webrtc.org,chxg@google.com
>
> Change-Id: I5c631d4dcfb39601055ffce9b104f45eea871fd3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10545
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144562
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28478}
TBR=ossu@webrtc.org,titovartem@webrtc.org,chxg@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10545
Change-Id: I609cca4f0ca4e1d31a156ba9eb44407518409f57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147865
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28746}
This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358
Original change's description:
> Only include payload in bytes sent/received.
>
> According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and
> https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent
> statistic should not include headers or padding.
>
> Similarly, according to
> https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes
> received are calculated the same way as bytes sent (eg. not including
> padding or headers).
>
> This change stops adding padding and headers to these statistics.
>
> Bug: webrtc:8516,webrtc:10525
> Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28647}
Bug: webrtc:8516, webrtc:10525
Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28731}
This reverts commit 74a1b4b1321b426392d4c32e4a02361226ad5358.
Reason for revert: requested by chromium
Original change's description:
> Only include payload in bytes sent/received.
>
> According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and
> https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent
> statistic should not include headers or padding.
>
> Similarly, according to
> https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes
> received are calculated the same way as bytes sent (eg. not including
> padding or headers).
>
> This change stops adding padding and headers to these statistics.
>
> Bug: webrtc:8516,webrtc:10525
> Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28647}
TBR=steveanton@webrtc.org,ilnik@webrtc.org,hbos@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,mellem@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8516, webrtc:10525
Change-Id: Ibd31a8264c19f0c6f57d8deb3974593d198046ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147400
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28701}
By design:
* OnPacketAdded() is meant to be called on pacer thread.
* Reconfigure() is meant to be called on worker thread.
Thus we guard against race condition on config_ member.
Possible downside: packet filtering based on ssrc might be slowed down.
Bug: webrtc:9849
Change-Id: I734bb9b34b01db160705897adb1b58e866e12639
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146980
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28691}
This reverts commit fab3460a821abe336ab610c6d6dfc0d392dac263.
Reason for revert: fix downstream instead
Original change's description:
> Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
>
> This reverts commit 9973933d2e606d64fcdc753acb9ba3afd6e30569.
>
> Reason for revert: breaking downstream projects and not reviewed by direct owners
>
> Original change's description:
> > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
> >
> > This reverts commit 24192c267a40eb7d6b1850489ccdbf7a84f8ff0f.
> >
> > Reason for revert: Analyzed the performance regression in more detail.
> >
> > Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f60cb9429bf4bcf13a40a41794ac8fb0 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac.
> >
> > There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall.
> >
> > Original change's description:
> > > Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
> > >
> > > This reverts commit 3e8ef940fe86cf6285afb80e68d2a0bedc631b9f.
> > >
> > > Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260.
> > >
> > > Original change's description:
> > > > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
> > > >
> > > > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
> > > >
> > > > Bug: webrtc:10668
> > > > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > > Reviewed-by: Minyue Li <minyue@webrtc.org>
> > > > Commit-Queue: Chen Xing <chxg@google.com>
> > > > Cr-Commit-Position: refs/heads/master@{#28434}
> > >
> > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com
> > >
> > > Bug: webrtc:10668, chromium:982260
> > > Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339
> > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#28561}
> >
> > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: webrtc:10668, chromium:982260
> > Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Chen Xing <chxg@google.com>
> > Cr-Commit-Position: refs/heads/master@{#28664}
>
> TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
>
> Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10668, chromium:982260
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28671}
TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10668, chromium:982260
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28672}
