This CL sets all visibility to ":*" (this buildfile) where no users
outside this directory are known, and marks up publicly exported
targets and Chrome dependencies explicitly.
Bug: webrtc:13661
Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36004}
This macro is not used in WebRTC code.
Bug: None
Change-Id: I5af1299594e8644ce2a9c772e29507367fd7440d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250140
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35872}
This will be used in the frame buffer 3 scheduler.
Bug: webrtc:13343
Change-Id: Ib699072021da30022a34aabe24e36a37e89ddf41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/245642
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35658}
Important: This change does not in any way affect echo cancellation or standardized stats. The user audio experience is unchanged. Only non-standard stats are affected. Echo return loss metrics are unchanged. Residual echo likelihood {recent max} will no longer be computed by default.
Important: The echo detector is no longer enabled by default.
API change, PSA: https://groups.google.com/g/discuss-webrtc/c/mJV5cDysBDI/m/7PTPBjVHCgAJ
This CL removes the default usage of the residual echo detector in APM.
It can now only be used via injection and the helper function webrtc::CreateEchoDetector. See how the function audio_processing_unittest.cc:CreateApm() changed, for an example.
The echo detector implementation is marked poisonous, to avoid accidental dependencies.
Some cleanup is done:
- EchoDetector::PackRenderAudioBuffer is declared in one target but is defined in another target. It is not necessary to keep in the API. It is made an implementation detail, and the echo detector input is documented in the API.
- The internal state of APM is large and difficult to track. Submodule pointers that are set permanently on construction are now appropriately marked const.
Tested:
- existing + new unit tests
- audioproc_f is bitexact on a large number of aecdumps
Bug: webrtc:11539
Change-Id: I00cc2ee112fedb06451a533409311605220064d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239652
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35550}
In logging.cc, use the pointer of the static variable so that
it doesn't need a global constructor/exit time destructor.
In RTCFieldTrials.mm, store the field trial string as a char pointer
instead of a std::unique_ptr to ensure that it is never freed.
LSAN will be unhappy with this fix, but WebRTC itself hasn't been
tested with LSAN enabled, and any code changed in this CL does not
build with build_with_chromium=true, so it shouldn't be a problem.
Bug: webrtc:9693, webrtc:11665
Change-Id: Ia28e3534170e0817b815717f6efe862f7b51ef62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237320
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35391}
Long term, ideally, these would be fixed and this flag can be removed.
For now, this is an expedient way to allow enabling -Wshadow in
Chromium.
Bug: chromium:794619
Change-Id: I7aba1c7bb7dfe0598cdb077cb97def752b8bac79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232660
Commit-Queue: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35110}
This is useful when building the .framework which doesn't need to
export C++ symbols.
Bug: webrtc:12408
Change-Id: Ied775811a72a06b9ad678c9fb549bca286dd7f37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227089
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34613}
-Winconsistent-missing-override is part of -Wall so there is no need
to explicitly set it.
-Wthread-safety and -Wimplicit-fallthrough are set by default by
Chromium's toolchain and there is no need to duplicated it in WebRTC
GN files:
gn desc out/Debug/ //rtc_base cflags | grep "Wthread-safety"
-Wthread-safety
-Wthread-safety
gn desc out/Debug/ //rtc_base cflags | grep "implicit"
-Wimplicit-fallthrough
-Wimplicit-fallthrough
Bug: None
Change-Id: Ie5104f7c6d508c7b45788420bf111a17b8b10939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226868
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#34549}
This is a reland of 9f2a20f4342a3e86e1f9fdfe6f3d76fb539d41c2
See https://webrtc-review.googlesource.com/c/src/+/226563/1..2
for the fix. RTC_DCHECK_ALWAYS_ON needs to be in public_configs
in order to be propagated together with header #includes and
avoid ODR violations.
Original change's description:
> Add WebRTC specific dcheck_always_on.
>
> Inspired by V8 CL: crrev.com/c/3038528.
>
> This makes the WebRTC's dcheck control independent of Chromium's and
> prepares switching Chromium's default behavior without affecting
> WebRTC developers or builders.
>
> Preparation for: https://crrev.com/c/2893204
>
> Bug: chromium:1225701, webrtc:12988
> Change-Id: Ia0d21f9fb8e9d7704fd1beca16504c301a263b3b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226465
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Dirk Pranke <dpranke@google.com>
> Cr-Commit-Position: refs/heads/master@{#34512}
Bug: chromium:1225701, webrtc:12988
Change-Id: I1f78587487ee7b1a4a07b8c91b737a9e797b2323
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226563
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34519}
Inspired by V8 CL: crrev.com/c/3038528.
This makes the WebRTC's dcheck control independent of Chromium's and
prepares switching Chromium's default behavior without affecting
WebRTC developers or builders.
Preparation for: https://crrev.com/c/2893204
Bug: chromium:1225701, webrtc:12988
Change-Id: Ia0d21f9fb8e9d7704fd1beca16504c301a263b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226465
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Dirk Pranke <dpranke@google.com>
Cr-Commit-Position: refs/heads/master@{#34512}
It was recommened to me to move this define to the top level BUILD.gn
file to avoid potential issues with the define not being available
where we need it.
Bug: webrtc:9273
Change-Id: Id0e939a51d1e381f684a3ae970569a255f52a5bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214101
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33661}
Packets, chunks, parameters and error causes - the SCTP entities
that are sent on the wire - are buffers with fields that are stored
in big endian and that generally consist of a fixed header size, and
a variable sized part, that can e.g. be encoded sub-fields or
serialized strings.
The BoundedByteReader and BoundedByteWriter utilities make it easy
to read those fields with as much aid from the compiler as possible,
by having compile-time assertions that fields are not accessed
outside the buffer's span.
There are some byte reading functionality already in modules/rtp_rtcp,
but that module would be a bit unfortunate to depend on, and doesn't
have the compile time bounds checking that is the biggest feature of
this abstraction of an rtc::ArrayView.
Bug: webrtc:12614
Change-Id: I9fc641aff22221018dda9add4e2c44853c0f64f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212967
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33597}
We need to be able build chromium with rtc_include_tests = true. It
reveals a lot of targets that are not compatible with chromium but
aren't marked so.
`rtc_include_tests=true` has been considered a way to disable targets for the Chromium build, causing an overload on rtc_include_tests while the meaning of the two GN args (rtc_include_tests and build_with_chromium) should be kept separated.
Bug: webrtc:12404
Change-Id: I2f72825445916eae7c20ef9338672d6a07a9b9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203890
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33124}
Some targets depends on targets under enable_google_benchmarks. But they
are not under such if statement themeself.
Bug: webrtc:12404
Change-Id: I7c0b9a75bd3fa18090ef6a44fda22ed5f33d79b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204063
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33104}
This CL adds a string to the resulting WebRTC library (trying to make
sure the version string will be there no matter how WebRTC is packaged).
This CL should be followed by some process to regularly and
automatically update the version string.
No-Try: True
No-Presubmit: True
Bug: webrtc:12159
Change-Id: I9143aeae2cd54d0d4048c138772888100d7873cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191223
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32825}
As discussed on a design review, the name RoboCaller is not clear
enough and switching to CallbackList will provide readability benefits.
Bug: webrtc:11943
Change-Id: I010cf0a91b5323e4e9c96b83703be7af1e67439c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190142
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32478}
The name was chosen because just like a real-world robocaller
[https://en.wikipedia.org/wiki/Robocall], webrtc::RoboCaller will
call multiple recipients and give all of them the same message,
without giving them the chance to reply.
Change-Id: Ia95f4543b15b48fa6388a50706e489dfccc19f71
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184621
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32152}
This will make it easier to find stuff...
Bug: webrtc:11943
Change-Id: I4f1ae80b40b4966cb2d8db36701bbc02ac148df6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184512
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32137}
- Fix the minor issues with the initial library implementation.
- Add unit tests to cover basic scenarios.
Bug: none
Change-Id: Ibf28b4e20f74792fce2fe11d4780fd375a4ad3a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183343
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32122}
Currently is_linux is set to true on Chrome OS build,
but it is planned to be set false. This CL is the preparation
to keep the compatibility.
Bug: chromium:1110266
Test: Build locally.
Change-Id: Ic79a202b0b3baeff157955cd03a07556bfb958a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Hidehiko Abe <hidehiko@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32073}
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.
Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
This is a reland of 1ca8d87239f1209031bbc77a6443bc7ac2dcee8c
Original change's description:
> Support AVX2/FMA intrinsics in Audio Resampler module
>
> From the test result, using AVX2/FMA is 1.60x faster than SSE on atlas.
>
> Bug: webrtc:11663
> Test: common_audio_unittests on atlas and octopus.
> Change-Id: Ibd45ea46aa97d5790a24e5116f741592b95f6416
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176382
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31810}
Bug: webrtc:11663
Change-Id: I92f5832a42c0314853c9fead46425c08e2040dc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181800
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31945}
From the test result, using AVX2/FMA is 1.60x faster than SSE on atlas.
Bug: webrtc:11663
Test: common_audio_unittests on atlas and octopus.
Change-Id: Ibd45ea46aa97d5790a24e5116f741592b95f6416
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176382
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31810}
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 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}
This is one last CL that includes the rest of VoIP API implementation.
Bug: webrtc:11251
Change-Id: I3f1b0bf2fd48be864ffc73482105f9514f75f9e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173860
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31168}
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.
Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
This is based on channel_receive.cc implementation where non-audio
related logics are trimmed off for smaller footprint in size.
Bug: webrtc:11251
Change-Id: I743c9f93f509fa6fcc12981fa73a6f01ce38348c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172821
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31117}
https://webrtc-review.googlesource.com/c/src/+/172847
------------ original description --------------
Preparation for ReceiveStatisticsProxy lock reduction.
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).
Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.
One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.
Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.
Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
Bug: webrtc:11489
Change-Id: I491e13344b9fa714de0741dd927d907de7e39e83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173583
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31077}
This reverts commit 24eed2735b2135227bcfefbabf34a89f9a5fec99.
Reason for revert: Speculative revert: breaks downstream project
Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
>
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
>
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
>
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
>
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
>
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
>
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}
TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
