This reverts commit a3df0f2d05c7b0973c31fe171507e97e588671a5.
Reason for revert: We decided to keep a killswitch in M70 just in case.
Original reviewed at: https://webrtc-review.googlesource.com/c/src/+/90251
Bug: chromium:690537
Change-Id: Ieb0eb8d5487e03fc55a221f10366ed9768a6eb16
Reviewed-on: https://webrtc-review.googlesource.com/95061
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24385}
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.
This is reland after fixes for broken perf tests.
Original Reviewed-on: https://webrtc-review.googlesource.com/91861
Bug: none
Change-Id: I6999a499408787be43a74a26a16b7826a0814a7b
Reviewed-on: https://webrtc-review.googlesource.com/95182
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24383}
In particular, FullStackTest.ForemanCif30kbpsWithoutPacketLoss and
FullStackTest.VP9KSVC_3SL_Medium_Network_Restricted, part of
webrtc_perf_tests.
Bug: None
Change-Id: If11bbcc1d897f048d7ab36b44cf16e67e0f6bacc
Reviewed-on: https://webrtc-review.googlesource.com/95147
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24380}
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.
Bug: none
Change-Id: I5a34c52e66bbd6c05859729ed14ae87ac26b5969
Reviewed-on: https://webrtc-review.googlesource.com/91861
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24359}
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
The was another definition of the --logs flag in full_stack_tests.cc.
The effect was that --logs set an unused bool, and the value of
FLAG_logs in test_main.cc was always false, regardless of actual
command line.
Tbr: sprang@webrtc.org
Bug: None
Change-Id: I073f8025dd897909c7e2b8d7c0ee080cb4b456ca
Reviewed-on: https://webrtc-review.googlesource.com/94900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24347}
Because DirectTransport is switched on SimulatedPacketReceiverInterface
we can't create it from some specific config in ctor, so all ctors,
that accept specific config are deprecated and you should pass concrete
implementation of underlying implememntation instead.
Bug: webrtc:9630
Change-Id: I7f241f310c993d8136b40898e55a6915924d61bd
Reviewed-on: https://webrtc-review.googlesource.com/94841
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24344}
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport
on this interface. Also switch part of related users on
DefaultNetworkSimulationConfig.
This two changes united into single CL to prevent work duplication.
Most changes were done because of stop including fake_network_pipe.h
into direct_transport.h, so splitting this into 2 CLs will require
first fix all imports of fake_network_pipe.h and then replace them
on new API imports again.
Bug: webrtc:9630
Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6
Reviewed-on: https://webrtc-review.googlesource.com/94762
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24336}
Keep a small window in order to distinguish old and new timestamp.
Bug: chromium:816819
Change-Id: Iefa694c744e8e4b19d3857c567162cdc9410b4da
Reviewed-on: https://webrtc-review.googlesource.com/94141
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24285}
This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.
Reason for revert: Breaks downstream project.
Original change's description:
> Add initial support for RtpEncodingParameters max_framerate.
>
> Add support to set the framerate to the maximum of |max_framerate|.
> Different framerates are currently not supported per stream for video.
>
> Bug: webrtc:9597
> Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> Reviewed-on: https://webrtc-review.googlesource.com/92392
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24270}
TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9597
Reviewed-on: https://webrtc-review.googlesource.com/94060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24277}
Add support to set the framerate to the maximum of |max_framerate|.
Different framerates are currently not supported per stream for video.
Bug: webrtc:9597
Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
Reviewed-on: https://webrtc-review.googlesource.com/92392
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24270}
* Make shorter 4-frame pattern default if 2 temporal layers are used.
* Make DefaultTemporalLayers usable by upper simulcast stream with 2tl.
* If experimental settings are enable, bump the max bitrate for the top
stream. Since we're now using probing everywhere the rampup should be
less of an issue.
* Additionally, fixes an issue in full stack tests, where
ScopedFieldTrials in an experiment would override the
--force_fieldtrials specified at command line. Some trials added by
the test bots caused timeouts without this.
Bug: webrtc:9477
Change-Id: I42410605d416b51c4fbfe5b6b850997484af583c
Reviewed-on: https://webrtc-review.googlesource.com/92883
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24252}
Follow-up to 92800 that inadvertedly excluded capturer startup from the CPU timing interval. Also a few style fixes.
Bug: b/112299470
Change-Id: Ida9100ffd8e125fa9a893a4470a0c934c518767b
Reviewed-on: https://webrtc-review.googlesource.com/92882
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24222}
This is a quick fix to ensure that we don't wrap the value. A proper
solution would be to ensure that the packets_lost field is signed and
handled as signed at all places it's used.
Bug: webrtc:9598
Change-Id: I3622f2a61aa3af57db6292ef4c0a8e97c4833aa4
Reviewed-on: https://webrtc-review.googlesource.com/92881
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24212}
Allows enabling audio for RunWithAnalyzer method, and prints out audio jitterbuffer performance stats. Also fixes for RunWithRenderer when enabling audio (seg-faulted).
Bug: b/112299470
Change-Id: Ic7c0de1c455891f38cca317001c6c216e82f6ec3
Reviewed-on: https://webrtc-review.googlesource.com/92800
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24208}
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
Since the linux_memcheck trybot is no more, this CL removes all the
code needed to make it work.
Bug: webrtc:7737, webrtc:8356, webrtc:9570
Change-Id: I09a9467b8bf895146a3384c2c915b54662721af6
Reviewed-on: https://webrtc-review.googlesource.com/90863
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24201}
Downstream projects have been updated, so this can now be relanded.
This is a revert (and rebase) of: https://webrtc-review.googlesource.com/c/src/+/88820
Bug: none
Change-Id: I424664ddef7aeebd3c6c94ae67c7f70a342dc9a4
Reviewed-on: https://webrtc-review.googlesource.com/92082
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24181}
rtc::LogMessage::SetLogToStderr should only be invoked by the main
function in order to enable or disable logging in a consistent way [1].
Usage of rtc::LogMessage::SetLogToStderr in other parts of the codebase
creates complex behaviors and confusion.
[1] - https://cs.chromium.org/chromium/src/third_party/webrtc/test/test_main.cc?l=88&rcl=665174fdbb4e0540eccb27cf7412348f1b65534c
Bug: None
Change-Id: Iae86fb14d7ca40af6d78d0f0cd81c5a39f65068d
Reviewed-on: https://webrtc-review.googlesource.com/91442
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24154}
In order to have public video bitrate allocator factory, the video bitrate allocator has be part of
the api.
Bug: webrtc:9513
Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2
Reviewed-on: https://webrtc-review.googlesource.com/88083
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#24073}
This change removes the constructors in favor of naming the fields
of the struct.
TBR=kwiberg@webrtc.org
Bug: None
Change-Id: I23ae1165c20994d2efef10184570065957b279af
Reviewed-on: https://webrtc-review.googlesource.com/90081
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24071}
This is a reland of d43c692ba7f53b5576a494c0343bc7a4bb36831b after fixes
to failing chromium tests. No change to the original CL were done.
Original CL reviewed on: https://webrtc-review.googlesource.com/87560TBR=stefan@webrtc.org
Bug: chromium:690537
Change-Id: I6b59ffc90d789aff21c7e52b118d3dfbe756c8a9
Reviewed-on: https://webrtc-review.googlesource.com/89081
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24013}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
