-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}
- 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 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}
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}
This allows clients to exclude the transient suppression submodule from WebRTC builds, by defining WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR.
The changes have been shown to be bitexact for a test dataset (when the flag is _not_ defined.)
No-Try: True
Bug: webrtc:11226, webrtc:11292
Change-Id: I6931c82a280a9b40a53ee1c2a9820ed9e674a9a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171421
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30978}
For simplicity and flexibility on audio only API, it deemed
to be better to trim off all audio unrelated logic to serve
the purpose.
Bug: webrtc:11251
Change-Id: I40e3eba2714c171f7c98b158303a7b3f744ceb78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169462
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30922}
This CL simply moves the VideoSourceRestrictor from being an inner class
of OveruseFrameDetectorResourceAdaptationModule to a new class,
VideoStreamAdapter.
In follow-up CLs, the responsibility of determining what the next step
for adapting up or down should also be moved to the VideoStreamAdapter.
The end-goal is that the VideoStreamAdapter takes care of "can adapt?"
and "do adapt!" type of logic so that a multi-stream aware adaptation
module can decide which stream (adapter) to adapt, and the adapter can
take care of the nitty gritty details of doing so.
In this CL the "can?"/"do!" part is realized but not the logic for
determining what the next step up or down is, and the class interface
needs improvement.
This CL also sets up the video/adaptation/ subdirectory and moves the
AdaptationCounters class here. Other adaptation-related classes (e.g.
the module and its resources) should move into this directory as well
in the future.
Bug: webrtc:11393
Change-Id: I2c12c1281eca854c62791abb65f0aca47a119726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169542
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30705}
I had to pivot and make tests output protos instead of JSON.
I basically move the proto -> JSON conversion into this script instead
of doing it in the test binary.
This is a temporary state. Later it will be enough to just read up
the file and pass it straight to the Catapult implementation, once
it learns to de-serialize the proto directly.
Bug: chromium:1029452
Change-Id: I7ce992eeeb1a5ae0f20eed54174b08b496e74dfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166920
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30419}
Add it to default build target, so it won't get broken accidentally
again. Fix configuration issue with field trials (new parameter was
added recently, but wasn't set by video_replay)
Bug: webrtc:11287
Change-Id: I9c18746d899acd7ac68c1b9b3a646b862c41897a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166900
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30345}
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This is the first step to move //:android_junit_tests to the righ
package (the target is triggering presubmit errors every time //BUILD.gn
gets updated).
Next steps:
* Update recipes
* Remove //:android_junit_tests
Issues with GN formatting, introduced by [1] will be addressed
separately in a "format all" CL.
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11289
No-Presubmit: True
Change-Id: I70c0927d722911f82dd971c30c7ffb581aed69c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166603
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30328}
Disable the C5041 warning which makes the build fail. This is a
C++17-only change and WebRTC doesn't support C++17 yet, so the code is
technically correct, but fails to build on MSVC 2019 and
warning-as-error active.
Also fix another warning-as-error build error with MSVC 2019 due to
ignoring the result of a [[nodiscard]] function.
No-Presubmit: True
Bug: webrtc:11275,webrtc:11276
Change-Id: I891a894ee87252f96e84fd8d282576f46907256f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30244}
Replace all usages of java_files with sources in gn files, and
automatically format.
This is in preparation for java_files being completely removed upstream
in favor of sources.
NOPRESUBMIT=true
Bug: chromium:1035074
Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Natalie Chouinard <chouinard@google.com>
Cr-Commit-Position: refs/heads/master@{#30135}
This CL adds Resource, ResourceConsumer, ResourceConsumerConfiguration
and ResourceAdaptationProcessor and implements the algorithm outlined
in
https://docs.google.com/presentation/d/13jyqCWNpIa873iKT6yDuB5Q5ma-c0CvxBpX--0tCclY/edit?usp=sharing.
Simply put, if any resource (such as "CPU") is overusing, the most
expensive consumer (e.g. encoded stream) is adapted one step down.
If all resources are underusing, the least expensive consumer is
adapted one step up.
The current resources, consumers and configurations are all fakes;
this CL has no effect on the current adaptation algorithms used in
practise, but it lays down the foundation for future work in this
area.
Bug: webrtc:11167, webrtc:11168, webrtc:11169
Change-Id: I4054ec7728a52a49e137eee6fa67fa27debd9254
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161237
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30053}
