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}
Implement AnalogGainStatsReporter and add it in AudioProcessingImpl.
This class computes statistics for analog gain updates and
periodically reports them into a histogram.
The added histograms for analog gain update statistics:
- WebRTC.Audio.ApmAnalogGainDecreaseRate
- WebRTC.Audio.ApmAnalogGainIncreaseRate
- WebRTC.Audio.ApmAnalogGainUpdateRate
- WebRTC.Audio.ApmAnalogGainDecreaseAverage
- WebRTC.Audio.ApmAnalogGainIncreaseAverage
- WebRTC.Audio.ApmAnalogGainUpdateAverage
The histograms are defined in
https://chromium-review.googlesource.com/c/chromium/src/+/3207987
Bug: webrtc:12774
Change-Id: I3c58d4bb3eb034a11c3f39ab8edb2bc67c5fd5e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234140
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35301}
The main change is to remove sockets from epoll if there are no
requested events, which happens when a socket is considered closed
(due to an error or otherwise). This prevents a busy loop when a socket
is an error condition where it will constantly be signaled, but not
deleted by higher level code.
Other related changes:
* Set DE_CLOSE on errors regardless of whether the socket is readable or
writable.
* Don't set DE_ACCEPT on errors.
* Handle getsockopt(SO_ERROR) errors.
* In IsDescriptorClosed:
* Retry recv on EINTR.
* Treat ECONNABORTED and EPIPE as errors.
Original patch contributed by andrey.semashev@gmail.com.
Bug: webrtc:11124
Change-Id: I67f33213efc1418b1ffc8f4867f606b7f8dc4ece
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235863
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35300}
Instead use `parameters_.config.rtp.ssrcs.size()` directly to make decisions about the number of temporal and spatial layer used.
Bug: none
Change-Id: Icba553178ae7fea281c2c67654c510228d9ab5b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237080
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35299}
Removes the ability to accept nonencrypted answers to encrypted offers.
Fixes some logic around bundled sessions and requirement for
transport parameters.
Bug: webrtc:11066
Change-Id: I56d8628d223614918a1e5260fdb8a117c8c02dbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236344
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35298}
Remove analog_level_minimum and analog_level_maximum from
AudioProcessing GainController1 and replace their use with fixed
values 0 and 255, respectively.
Bug: webrtc:12774
Change-Id: Ia4bfe5ed43a65f1587ed67f36bfbb2966b6fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235822
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35297}
Changes one preexisting enum-to-string function to use the new format.
Also changes the RTC_LOG macros that created collisions with ToString,
for tidiness, and documents the recommended function form.
Bug: webrtc:13272
Change-Id: Ic8bb54ed31402ba32675b142d796cf276ee78df5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235722
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35296}
This reverts commit 3b18208f13e85b356e61a95c0a261e9781403743
and is the third attempt at removing stun origin support
Bug: webrtc:12132
Change-Id: Ic41a6d011fb6239907a257cc4c81ec4d2923dc4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236260
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35294}
