This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I71350feb45c23d5a0e17e4828c514ce1c353f6a7
Reviewed-on: https://webrtc-review.googlesource.com/96700
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24524}
Primarily, this is intended to reduce flakyness of
RampUpTest.AudioTransportSequenceNumber. We shouldn't expect audio
send rate >= 300 kbps at all time in these tests. And in general, if
it's at all relevant to test that bitrate doesn't drop below the start
bitrate, a perf test isn't the right place for that.
A run of
./third_party/gtest-parallel/gtest-parallel -r 1000 -w 1000 \
--gtest_filter=RampUpTest.AudioTransportSequenceNumber \
out/Release/webrtc_perf_tests
passes when I ran it locally after this change, but fails around 4 out
of 1000 times before the change.
Bug: webrtc:8878
Change-Id: I08614ce5683c9ba6fe4b72bfde83e6a81445a59b
Reviewed-on: https://webrtc-review.googlesource.com/96900
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24523}
Thanks to the CL https://webrtc-review.googlesource.com/c/src/+/83822
there is now no need to destroy the stream asynchronously.
But the CL above introduced a leak, the streams were stopped but
not destroyed. This CL essentially fixes the leak, it is now safe
to destroy the stream right after being stopped as everything happen
in the same capture thread.
Bug: chromium:851883
Change-Id: I4bf7409246f3957d90040d0d8cf09e98f28d6559
Reviewed-on: https://webrtc-review.googlesource.com/96621
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24519}
Sine a starting rate field was added to the constraints struct. Having
it in the initial config separately is reduntant. To simplify the code,
the extra field is removed. This is a follow up on:
https://webrtc-review.googlesource.com/c/src/+/92624
Bug: webrtc:9586
Change-Id: I9b01b16b2fc4b8479e83b7e998308be2295e0325
Reviewed-on: https://webrtc-review.googlesource.com/96801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24514}
The build toolchain now requires xcode 10, so we have to roll.
Bug: None
Change-Id: Iafec62e7927ca8a81117710d09e2c42bcf18c0d1
Reviewed-on: https://webrtc-review.googlesource.com/97060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24510}
LappedTransform is only used in BandwidthAdaptationTest and therefore it
should not be anymore a visible target under common_audio.
This CL moves LappedTransform and other two classes it depends on (and which
are not used elsewhere) to modules/audio_coding/codecs/opus/test.
Bug: webrtc:9577, webrtc:5298
Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a
Reviewed-on: https://webrtc-review.googlesource.com/96440
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24509}
Remove partition support for test helper and from tests.
Merge Init function into constructor
Replace extra macroses in favor of Bit helper function
Replace extra members in favor of local variables
Remove fixture
Bug: None
Change-Id: Ibf1600dda9f59abe5afd2bbe40c3e232a2d269ea
Reviewed-on: https://webrtc-review.googlesource.com/96940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24508}
This is a reland of da0898dfae3b0a013ca8ad3828e9adfdc749748d
Original change's description:
> Add spatial index to EncodedImage.
>
> Replaces the VP8 simulcast index and VP9 spatial index formely part of
> CodecSpecificInfo.
>
> Bug: webrtc:9378
> Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
> Reviewed-on: https://webrtc-review.googlesource.com/83161
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24485}
Tbr: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: Iff20b656581ef63317e073833d1a326f7118fdfd
Reviewed-on: https://webrtc-review.googlesource.com/96780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24507}
This CL introduces the ability to control the usage of the shadow filter
output in the echo canceller output.
Bug: webrtc:9694,chromium:879451
Change-Id: I01f90de60de1799b32892051c176bda5e1a8d33e
Reviewed-on: https://webrtc-review.googlesource.com/97020
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24506}
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).
Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
This reverts commit 7bcd2a98be3fa8c246866d6b343c7f94752977b3.
Reason for revert: peerconnection_unittests fails on downstream test runner.
Original change's description:
> Reland "Optimize execution time of RTPSender::UpdateDelayStatistics"
>
> The reland has a lot of additional DCHECKS for easier debugging,
> so in debug builds it will actually be a ~2x slowdown compared to the old code.
> The excessive DCHECKS should be removed in a followup CL.
>
> Bug: webrtc:9439
> Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
> Reviewed-on: https://webrtc-review.googlesource.com/96641
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24501}
TBR=terelius@webrtc.org,asapersson@webrtc.org,philipel@webrtc.org
Change-Id: Ia48444d2a7647cf826ef93b4720f6d7ff9a712c3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9439
Reviewed-on: https://webrtc-review.googlesource.com/96960
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24502}
The reland has a lot of additional DCHECKS for easier debugging,
so in debug builds it will actually be a ~2x slowdown compared to the old code.
The excessive DCHECKS should be removed in a followup CL.
Bug: webrtc:9439
Change-Id: I493de337bf20c998aa32c2532212cac85c5517fb
Reviewed-on: https://webrtc-review.googlesource.com/96641
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24501}
This CL fixes a bug in the feedback based GoogCC where packets lost
was swapped with expected packets received. Since this version of
GoogCC isn't yet used this wasn't discovered. There was also a lack
of unit test coverage. To ensure reasonable behavior, unit tests was
added.
Unit tests was also converted from relevant unit tests on send side
congestion controller for the regular GoogCC controller.
Bug: webrtc:9586
Change-Id: I83c40ff4766104820cb72ec1e8b95c5782def19a
Reviewed-on: https://webrtc-review.googlesource.com/59401
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24498}
This reverts commit 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a.
Reason for revert: Makes the perf test RampUpTest.AudioTransportSequenceNumber fail on windows, almost every time.
Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
>
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
>
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}
TBR=henrika@webrtc.org,nisse@webrtc.org,titovartem@webrtc.org
Change-Id: I8867a22d695494bd5abfda6a97f0719cb3ff3d66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3380
Reviewed-on: https://webrtc-review.googlesource.com/96840
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24496}
Return value always passed as the |retransmitted| argument to
ReceiveStatistics::IncomingPacket. The implementation of this method,
StreamStatisticianImpl::IncomingPacket, can call its own
IsRetransmitOfOldPacket, which is demoted to a private method.
Bug: webrtc:7135
Change-Id: I904db676738689c7a1db4caa588f70e64e3c357d
Reviewed-on: https://webrtc-review.googlesource.com/95649
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24494}
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.
A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.
The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.
The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.
Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
In addition, let the processing thread loop explicitly, and not use
the deprecated builtin looping in PlatformThread.
Bug: webrtc:3380
Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
Reviewed-on: https://webrtc-review.googlesource.com/96544
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24492}
Merge SetPayloadData into constructor,
Remove payload size member because now used only during construction.
Remove member that should be constant
Bug: None
Change-Id: Ib2083439f466ad9151ce8e54fceede6cef51d955
Reviewed-on: https://webrtc-review.googlesource.com/96740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24491}
This change injects the FrameEncryptorInterface and the FrameDecryptorInterface
into the RtpSenderInterface and RtpReceiverInterface respectively. This is the
second stage of the injection. In a follow up CL non owning pointers to these
values will be passed down into the media channel.
This change also updates the corresponding mock files.
Bug: webrtc:9681
Change-Id: I964084fc270e10af9d1127979e713493e6fbba7d
Reviewed-on: https://webrtc-review.googlesource.com/96625
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24489}
FrameDecryptorInterface into the public WebRTC API surface.
This change just addresses the headers and not the internal changes.
Bug: webrtc:9681
Change-Id: I1db0172fe55ba378f62e7781c2b7dcdb93d63239
Reviewed-on: https://webrtc-review.googlesource.com/96622
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24488}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
CustomAudioAnalyzer is an interface of a component into APM that
reads AudioBuffer without changing it.
The APM sub-module is optional. It operates in full band.
As described in the comments, it is an experimental interface which
may be changed in the nearest future.
Change-Id: I21edf729d97947529256407b10fa4b5219bb2bf5
Bug: webrtc:9678
Reviewed-on: https://webrtc-review.googlesource.com/96560
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Valeriia Nemychnikova <valeriian@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24481}
It's reasonable to allow clients implementing their own VideoCodecTests
to decide wether they should run in real-time.
Removes the IsAsyncCodec helper, as the assumptions it made are outdated,
and it is no longer useful.
Bug: None
Change-Id: If766935d4947555af54f499a30cfe553bde4c1ab
Reviewed-on: https://webrtc-review.googlesource.com/95722
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24478}
Add ability to provide custom implementation of
NetworkSimulatedInterface for sender and receiver network in
VideoQualityTestFixtureInterface, passing them to the factory method.
Also unite this mechanism with FecControllerFactoryInterface injection.
Bug: webrtc:9630
Change-Id: I79259113e0fc00d933b73ca299afa836a4cd19d2
Reviewed-on: https://webrtc-review.googlesource.com/96280
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24476}
