Currently, when users want to use the screen sharing and are using the
Wayland display server (the default on Fedora distribution), then it
doesn't work, because the WebRTC only includes the X11 implementation.
This change adds the support by using the PipeWire multimedia server.
The PipeWire implementation in WebRTC stays in
screen-capturer-pipewire.c and is guarded by the rtc_use_pipewire build
flag that is automatically enabled on Linux.
More information are included in the relevant commit messages.
Tested on the current Chromium master and Firefox.
The sysroot changes are requested in:
https://chromium-review.googlesource.com/c/chromium/src/+/1258174
Co-authored-by: Jan Grulich <grulja@gmail.com>
Co-authored-by: Eike Rathke <erathke@redhat.com>
Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a
BUG=chromium:682122
Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a
Reviewed-on: https://webrtc-review.googlesource.com/c/103504
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25461}
This drops the locks and annotations in EchoCancellationImpl,
now that the interface is no longer externally accessible.
Bug: webrtc:9929
Change-Id: I401256f523340cbabce23a5914ab28ce44179935
Reviewed-on: https://webrtc-review.googlesource.com/c/108602
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25460}
This CL consistently use:
* relative paths for WebRTC dependent targets (test_support)
* absolute paths for shared dependent targets (abseil)
This is a necessary (but insufficient) step to build WebRTC tests
from Chromium tree (rtc_include_tests=true), since test/ doesn't
sit anymore in the top level directory.
We also make sure that target declarations and uses are
consistent in regard to build_with_chromium flag.
Bug: webrtc:9943
Bug: webrtc:9855
Change-Id: I21dea98894df2fd4bfe2fd7ee7b71ba971e0ab5b
Reviewed-on: https://webrtc-review.googlesource.com/c/108720
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25445}
When FlexFEC is enabled, sometimes media packet will be recovered by FEC before the actual media packet's arrival. In current implementation this will be considered as packet out of order and nack will be sent, thus cause large increase in retransmit bitrate.
This fix:
1. Avoid sending nack for packet out of order caused by "early" recovered media packets.
2. Save recovered media packet in a set, and do not send nack for these packets.
Bug: None
Change-Id: I008ef4e33668bce6d2cb9ff52b4b5c8e3f349965
Reviewed-on: https://webrtc-review.googlesource.com/c/108090
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25444}
This CL utilizes the existing, but unused, ability to set
different histogram thresholds for early and late delay
estimation. It does so by tuning the parameters for these.
On top of that, some corrections are added to correctly
handle resets and the use of the hysteresis thresholds.
Bug: webrtc:19886,chromium:896334
Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/106706
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25443}
This class exposes Wait()-Set() logic to synchronize events.
- There is a bug in checking EventWrapper::Wait() as it returns [1,2]. Negating
these values cause us to always pass timeout checks.
- There is a general problem in this class with waiter. There are 2 scenarios:
1) Lock()-Unlock()-DisplaysReconfigured()
In this scenario, Wait() in DisplaysReconfigured() immediately passes as event
is already signaled. Next Lock() call won't continue until Set() is called in
DisplaysReconfigured(). This blocks capture thread from accessing display until
reconfiguration completes.
2) Lock()-DisplaysReconfigured()-Unlock()
In this scenario, Wait() in DisplaysReconfigured() passes when Unlock() called.
Capture thread accesses display while reconfiguration happens. Note that we are
only delaying the OS delegate thread here. As an experiment, adding Sleep() in
DisplaysReconfigured() results in no change, because it looks like OS uses this
thread for only delegates but not for the actual display switch.
Overall, (1) doesnt seem necessary as (2) already accesses display while
reconfiguration happens. (2) doesn't seem necessary as blocking system delegate
thread doesn't help. Therefore, I changed the class to only protect from race
condition on |desktop_configuration_|.
Bug: chromium:796889
Change-Id: I37263305e5ac629e21ff9e977952cf4a21bae19f
Reviewed-on: https://webrtc-review.googlesource.com/c/108560
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25437}
This reverts commit 9a0662ac7e4a3bc6b3a316397a7fdf25f0025d35.
Reason for revert: breaks some av sync perf tests
Original change's description:
> Use only first payload timestamp for RTCP SR generation for audio
>
> Since now RTP rate is set correctly for audio, there's no need to
> use the very last data packet rtp/capture timestamps for generating
> RTCP SR packets.
>
> Using only one (first) packet timestamp eliminates the jitter between
> rtp and capture timestamps for audio. This jitter comes from the fact
> that capture timestamp for audio is unknown and we generate bogus
> timestamp at arbitrary, non-constant offset from the real capture time.
>
> Bug: webrtc:9905
> Change-Id: I855556184cfe994be39ab7780836a050f5a38c35
> Reviewed-on: https://webrtc-review.googlesource.com/c/108580
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25430}
TBR=danilchap@webrtc.org,ilnik@webrtc.org,ossu@webrtc.org
Change-Id: I208a659379b1075258ee94613e42afd9aebe4754
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9905
Reviewed-on: https://webrtc-review.googlesource.com/c/108623
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25435}
Since now RTP rate is set correctly for audio, there's no need to
use the very last data packet rtp/capture timestamps for generating
RTCP SR packets.
Using only one (first) packet timestamp eliminates the jitter between
rtp and capture timestamps for audio. This jitter comes from the fact
that capture timestamp for audio is unknown and we generate bogus
timestamp at arbitrary, non-constant offset from the real capture time.
Bug: webrtc:9905
Change-Id: I855556184cfe994be39ab7780836a050f5a38c35
Reviewed-on: https://webrtc-review.googlesource.com/c/108580
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25430}
Source height may be negative, causing libyuv to invert the image.
However the height of the destination buffer specified by crop_height
should be positive. Remaining calls in common_video_unittests are valid.
Bug: webrtc:9447
Change-Id: I6d398909ae80a99d228ccbbd8c1d7ae804e5bf8d
Reviewed-on: https://webrtc-review.googlesource.com/c/86540
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25427}
Since they rely on a real time simulation, a new build target is
introduced that is intended to be used for real time tests.
Bug: webrtc:9518
Change-Id: Iea58f6a2b687f026e9ab1f37b4aabf8261ed7d23
Reviewed-on: https://webrtc-review.googlesource.com/c/107345
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25410}
Use helper TimeMicrosToNtp() on clock TimeInMicroseconds()
instead of CurrentNtpTime() and CurrentNtpTimeMillis()
Also update TimeMicrosToNtp() to not introduce fractional in
milliseconds offset. Expose that offset in time_utils.h
Add test showing indended behavior.
Bug: webrtc:9919
Change-Id: I8b019e11ae5b79d0b8ba113a84066b0369cd2575
Reviewed-on: https://webrtc-review.googlesource.com/c/107889
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25391}
This means that the PacedSender::Process function becomes slightly
larger, however, it makes it much more obvious to the reader where
the locks are held or not. Confusion over this has previously caused
bugs.
Bug: webrtc:9870
Change-Id: I63257eae59ecf5e7dd28ea24f63157cefe9f81bd
Reviewed-on: https://webrtc-review.googlesource.com/c/105460
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25389}
This log is triggering many times a second for Chrome Remote Desktop on some
browsers. This CL just turns it off for release builds to avoid log files
filling up users' disks until we figure out what's going on.
Bug: chromium:888038
Change-Id: Ibbe9d47295b3633314feb28e155e3f59b878dbdb
Reviewed-on: https://webrtc-review.googlesource.com/c/107688
Commit-Queue: Jamie Walch <jamiewalch@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25372}
This CL creates a new API for the parser of APM json config that
that provides an explicit way for the user to know when there has
been an issue in the parsing of the json config data.
Bug: webrtc:9921
Change-Id: Idd8f40529f40ab6871efb5b356c0fd2cea21b7d9
Reviewed-on: https://webrtc-review.googlesource.com/c/107841
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25355}
This CL modifies the internal data logging and the audioproc_f tool
to allow controlling that via the command line, rather than solely via a
build flag. The logging of internal data is by default off.
Bug: webrtc:5298
Change-Id: I96d1b4f990582938527b9039d6c2ecbb6f76e9ca
Reviewed-on: https://webrtc-review.googlesource.com/c/107713
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25352}
This deprecates the following methods in VideoEncoder:
virtual ScalingSettings GetScalingSettings() const;
virtual bool SupportsNativeHandle() const;
virtual const char* ImplementationName() const;
Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.
Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().
This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.
Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
This CL allows control over the dominant nearend functionality so that
it is not active during the initial phase, when estimates are less
certain.
Bug: webrtc:9906,chromium:898273
Change-Id: I5f61dac806ec3b1ebc1a3ec72f0a16d07a67f14a
Reviewed-on: https://webrtc-review.googlesource.com/c/107632
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25326}
Limiter has been renamed to LimiterDbGainCurve, which is a more correct name
and will allow in a follow-up CL to reuse the Limiter name for GainCurveApplier.
This is done to allow to use the limiter without instancing the fixed digital
gain controller and then to fix an AGC2 issue (namely, fixed gain applied after
the adaptive one).
Bug: webrtc:7494
Change-Id: Icd7050e3e51b832bfbf35e5cc61109215c5b1ca6
Reviewed-on: https://webrtc-review.googlesource.com/c/106901
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25322}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
deprecated version guarantees (using PostTaskAndReply) callback task will run on the task queue,
and thus doesn't guarantee to run it if task queue is destroyed,
new callback versions instead guarantee callback will always run,
but may run off the task queue if task queue is destroyed.
Both keep guarantee observer callbacks will not run after on_destroyed/on_removed is called.
Bug: None
Change-Id: I61bf52127f3084c0186aa8bc89037bf9296801d8
Reviewed-on: https://webrtc-review.googlesource.com/c/107305
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25301}
This change makes the dominant nearend detection more accurate.
- The hangover is increased not leave nearend state between words.
- The SNR requirement is increased to not enter nearend state without
speech activity.
- An early exit mechanism has been added to leave nearend state quickly
when the echo is strong.
Bug: chromium:897701,webrtc:9897
Change-Id: I9e0f3e6ecb80eee1c0c917d4835f110555f74acf
Reviewed-on: https://webrtc-review.googlesource.com/c/107347
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25299}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: I67a4d016a11deca5ac5459826741dd2d3f7931d5
Reviewed-on: https://webrtc-review.googlesource.com/c/107400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25298}
This will allow downstream projects to use it to construct their own
injected codecs without pulling in dependencies on the software codecs.
Bug: webrtc:7925
Change-Id: If8628fedd18e57a51a8b6e5baf4f63a686bf52e8
Reviewed-on: https://webrtc-review.googlesource.com/c/107027
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25297}
This change prevents decoding corruption by not allowing keyframes with a
newer frame id but an older timestamp to be decoded. This does not handle
reordering well.
Bug: none
Change-Id: I4a67ca84ee86a782da74a10530c531d893d3bd3c
Reviewed-on: https://webrtc-review.googlesource.com/c/107304
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25292}
In this CL we change the signal that controls the updates of the ERLE estimator. Until now, the render signal was used which is not optimum for reverberant signals. In this CL, a reverberation has been added to the the render signal and this new signal has been used for controlling when to update the ERLE estimator.
Bug: webrtc:9873
Change-Id: I0ebea3fc208f97aa237af015ba543015d49ed978
Reviewed-on: https://webrtc-review.googlesource.com/c/105660
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25285}
This is a reland of 5ccdc1331fcc3cd78eaa14408fe0c38d37a5a51d
Original change's description:
> Prefix flag macros with WEBRTC_.
>
> Macros defined in rtc_base/flags.h are intended to be used to define
> flags in WebRTC's binaries (e.g. tests).
>
> They are currently not prefixed and this could cause problems with
> downstream clients since these names are quite common.
>
> This CL adds the 'WEBRTC_' prefix to them.
>
> Generated with:
>
> for x in DECLARE DEFINE; do
> for y in bool int float string FLAG; do
> git grep -l "\b$x\_$y\b" | \
> xargs sed -i "s/\b$x\_$y\b/WEBRTC_$x\_$y/g"
> done
> done
> git cl format
>
> Bug: webrtc:9884
> Change-Id: I7b524762b6a3e5aa5b2fc2395edd3e1a0fe72591
> Reviewed-on: https://webrtc-review.googlesource.com/c/106682
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25270}
TBR=kwiberg@webrtc.org
Bug: webrtc:9884
Change-Id: I5ba5368a231a334d135ed5e6fd7a279629ced8a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/107161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25277}
