Use a fake task queue to test the RepeatingTask rather than a real
task queue, which removes the need for Sleep(). This fixes the flakiness
issues as the class is no deterministic.
BUG=webrtc:12808
Change-Id: I8c6a8535165b076f5fe6ec3e65ebcf7f07008737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237803
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35349}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This makes relay candidates identifiable in scenarios
where the TURN server is behind another entity and the peer
sees a different ip address:
client -> turn -> relay address -> third party relay + address -> peer
In those cases, the relay candidate will become peer-reflexive
since the peer sends the third party relay's address in the xor-mapped
address and it is currently not easily possible to determine this is a
relay candidate anymore.
BUG=webrtc:13392
Change-Id: I6787339d0abdc735f8a43f636a676cccd8cadcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237561
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35346}
this could happen if setCodecPreferences is used to prefer
red over opus as it is done for red+opus.
BUG=webrtc:13287
Change-Id: I3d61cd8f1a364572bc531a75dcc239c3919138cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237800
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35344}
This makes it not an error to bind a SequenceChecker before the
global TaskQueueBase::Current() is set.
Unbreaks the SDP integration fuzzer.
Bug: webrtc:13374
Change-Id: Ic4c23fa29f4598290cf9196550e5133ba753f44f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237620
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35341}
This CL set the spatial id in LibaomAv1Encoder and set correct number
of spatial layers for AV1 in FrameEncodeMetadataWriter.
Bug: None
Change-Id: I40092e45be88ec9ab75f228d9ca84c44e3cad326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237662
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/main@{#35339}
- Reusing RTP stack may have contributed to some flakiness as
the previous state could have persisted to new test being performed.
Bug: webrtc:13241
Change-Id: Idf70b56bd3377bc99321fddf7191d7a72c37b085
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237540
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35336}
Change search for next separator to be linear in length of the string
(instead of potentially quadratic)
Reduce copying of std::string by switch to string_view
Throttle logging about unknown key.
Bug: b/204541739
Change-Id: I81d5cd4432966a0a5808077f9001bc62960e5e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237500
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35330}
This adds the Main 3.1 profile to the list of supported H264 codecs. This unifies the output of WebRTC codecs among macOS/Windows (which both have Main 3.1 codecs) and headless Linux browsers.
Bug: None
Change-Id: Ife2fe8c1827be9368fabccc5f24ba316671b1b8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236600
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35328}
It's put in the public folder since the intention is to expose it in
SendOptions.
Additionally, use TimeMs::InfiniteFuture() to represent sending a
message with no limited lifetime (i.e. to send it reliably).
One benefit for these two is avoiding using absl::optional more than
necessary, as it results in larger struct sizes for the outstanding
data chunks.
Bug: webrtc:12943
Change-Id: I87a340f0e0905342878fe9d2a74869bfcd6b0076
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235984
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35323}
VideoStreamEncoder receives frames on an undefined threading
context with the only requirement being that frames are serially
arriving. This CL changes this to post all frames arriving at the
FrameCadenceAdapter to the worker thread before further
processing, transitively leading to frame entry into the
VideoStreamEncoder on the worker thread.
Bug: chromium:1255737
Change-Id: I04d69cb4a5048d671d2dcd3bd6d669fbcda52b3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35320}
The test has a ModuleRtpRtcpImpl2 which is destroyed long
after test transports are removed, leading to a UAF.
Fix by overriding OnStreamsStopped which is called before
transports are removed.
TESTED=Asan now passes 1000/1000, failed 4/1000 before.
Fixed: chromium:1235251, b:192567426
Change-Id: Ie9135685e81712e38c4b00355debfc67c1f603bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237345
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35318}
Also switch to "LT(meaured, limit)" format as this is easier to read.
There have been bot runs that exceeded the old limit.
Bug: webrtc:13354
Change-Id: I1c19c98e1c1777177e7dc14ae4679765c1c40550
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237342
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35317}
This change introduces a new FrameCadenceAdapter class which takes the
role of being a VideoFrameSinkInterface<> instead of VideoStreamEncoder.
The FrameCadenceAdapter will see its functionality grow in future CLs
and eventually enable screenshare capture sources to have zero hertz as
the minimum capture frequency.
This CL moves logic related to UMA collection and constraints into the
adapter.
The adapter has two major modes. Future functionality is planned to be
added under the WebRTC-ZeroHertzScreenshare field trial. Unit tests are
added that verify passthrough operation when WebRTC-ZeroHertzScreenshare
isn't specified or disabled.
Just specifying the WebRTC-ZeroHertzScreenshare field trial isn't
enough to activate the feature, but the caller has to additionally
configure screen content type, minimum FPS 0, and maximum FPS > 0 for
the new mode.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I1799110ed40843152786ad80df10acfb83a608b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236682
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35315}
* Clearing while waiting for a frame should return a new frame
entering the buffer.
* Stopping while waiting for a frame should cancel the wait.
Bug: webrtc:13343
Change-Id: Ife9abfa8b6ea56141c9f32ff37d3b2a2e62a44f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236849
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35314}
close to the codec they relate to as it is done by video_engine.
This does not change functionality but improves readability of the SDP
BUG=webrtc:13287
Change-Id: I9b03cd3131eaa932ffed3fb4e66cbf55faedcdd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235665
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35311}
UpdateMonitors() crops the selected RANDR monitor to the root window,
in case X returns a monitor that lies outside it. But it wasn't enough.
SelectSource() alters the selection directly and doesn't call
UpdateMonitors(), so it also needs to crop. This fixes the case
where a virtual monitor is added, the screen resolution is reduced,
then the new monitor is selected (which now extends outside the reduced
screen size).
This CL also fixes an issue where the ScreenCapturerHelper would
sometimes expand a damage-region outside the DesktopFrame boundary.
This occurred because the helper's size was set to the full
pixel-buffer, so it didn't crop correctly to the monitor's rect.
This CL sets the helper's correct size, and removes some unnecessary
cropping now that the helper will do it correctly.
Bug: chromium:1266179
Change-Id: I8eb8f3302701be4f393934c0899f41def3512853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237120
Commit-Queue: Joe Downing <joedow@chromium.org>
Reviewed-by: Joe Downing <joedow@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35304}
The AsyncListenSocket::SetOption method then gets unused, and can be
deleted.
Bug: webrtc:13065
Change-Id: Idcf70a75b96036290fdceff6e0f96a8d5617f87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236580
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35302}
