By calling release() the ref counter is not decremented and when the
observer is later wrapped in a scoped_refprt its counter will be 2
instead of 1, leading to a memory leak.
Bug: b/231912631
Change-Id: I466e5c29ba4f4aee43e8cf23f2e04fa4b6351b4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261954
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36862}
HandleXEvent() returns true to indicate the event is consumed and
should not be passed to other registered handlers of the same
event-type.
In ScreenCapturerX11, this makes sense for XDamage events because they
are scoped to the object's |damage_handle_|. But RRScreenChangeNotify
and ConfigureNotify events are scoped to the root window, so this CL
changes the return value to false for these events. This allows other
handlers (including other screen-capturer instances) to see these
events.
Bug: webrtc:14060
Change-Id: Id18917b0b62d125da08578e08df9648062500cad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262142
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Lambros Lambrou <lambroslambrou@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36858}
Apple renamed CGColorSpaceCopyICCProfile to CGColorSpaceCopyICCData in
10.13. If you compile code with a minimum OS required of 10.13 or
newer, [-Werror,-Wdeprecated-declarations] will cause an error to
occur on use of the function with the old name.
Add a compile switch so that no error is emitted regardless of the
deployment configuration.
Bug: chromium:1322548
Change-Id: Ie969aa9e5c4fc9bee2ec88b126d4c07701c3e9e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261953
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Avi Drissman <avi@chromium.org>
Cr-Commit-Position: refs/heads/main@{#36855}
Define a local function in the only place where it is used, for calling
SSL_set1_curves_list.
Bug: webrtc:6424
Change-Id: I7b9c372aaed15dbc88ced55652f5afd93db55e49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261313
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36853}
Since the lifetime of an SctpDataChannel is not strictly controlled
by its controller, the controller might go away before the channel
does. This CL guards against this.
Bug: webrtc:13931
Change-Id: I07046fe896d1a66bf89287429beb0587382a13a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261940
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36852}
This was found during code review. This code is essentially dead code
until interleaved messaging is implemented, which is disabled both in
configuration and due to missing code.
Bug: None
Change-Id: Idea87dfe2be204361774d8964140fd9947a66410
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261944
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36850}
- Filter out very old packets (to ensure that the estimate doesn't drop to zero if sending is paused and later resumed).
- Discard packets older than previously discarded packets (to avoid the estimate dropping after deep reordering.)
- Add tests cases for high loss, deep reordering and paused/resumed streams to unittest.
- Remove some field trial settings that have very minor effect and rename some of the others.
- Change analyzer.cc to only draw data points if the estimators have valid estimates.
Bug: webrtc:13402
Change-Id: I47ead8aa4454cced5134d10895ca061d2c3e32f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236347
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36849}
This is in preparation for adding a portal / pipewire backend.
This just renames one class and moved the code to different files.
There are no changes to the implementation.
Bug: webrtc:13177
Change-Id: Iae101fcabafdb6cddd4d82adbb26219e4b37557f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261680
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36848}
The change bounds the estimate increment by MaxIncreaseFactor in DelayedIncreaseWindow after seeing loss. MaxIncreaseFactor is set to 1000 to disable the change by default.
Improve trendline integration: always allow to decrease the estimate, and only allow to increase the estimate if overusing and underusing are not in the state window.
Other improvement: bound candidates by delay based estimate, instance upper bound, and bandwidth limit in the current window.
Clean: remove the flag BackoffWhenOverusing since it has negative impacts when experimenting.
Bug: webrtc:12707
Change-Id: Ia4c1e58d692071967e8807a8b9d64b8ae4caf837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261240
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36847}
Tracing can be disabled by setting the build flag
rtc_disable_trace_events = true
This causes the variable to be unused.
Bug: webrtc:12787
Change-Id: Iebbb8cbb5ede5453ad24ce7710de3b1dd68ad83f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261683
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36843}
GetRtpExtensions() is still used in one corner case for audio receive
streams, so GetRtpExtensions has migrated to AudioReceiveStream.
Updated FlexfecReceiveStream config management (incl. pass by value) and
now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl.
Call GetRtpExtensionMap() from call.cc instead of constructing one on
the fly for each rtp packet (for video packets at least).
Bug: webrtc:11993
Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36839}
This is in pursuit of an issue with another CL, but large enough
to be worth submitting separately.
Bug: webrtc:13931
Change-Id: If470488f092f8640d3a773922f6f0d22765b9e97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261728
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36833}
The killswitch is no longer needed, because the googScreencastMinBitrate
has been successfully removed from the web platform.
The native RTCConfiguration::screencast_min_bitrate is still available
though because there are other downstream users than Chrome.
Bug: chromium:1315155
Change-Id: I2145f9014dbe57bb50e61f1faeacd533d76acb29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261725
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36831}
Instead offer accessors for the specific config values from the struct
that are needed at different times. The remote_ssrc and rtx_ssrc
properties maybe accessed from any thread, other properties have
stricter requiremets.
Bug: webrtc:11993
Change-Id: I3ff8527b13452c773fae1b2574f1e3fd2583b481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261319
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36823}
This is to avoid accessing the array via the config struct.
Moving forward we might want to consider using the RtpHeaderExtensionMap
instead of a std::vector of RtpExtension.
Bug: webrtc:11993
Change-Id: I8469dbbd9bb95a69f87b5912bfc4bf8b8f603beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261317
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36820}
`remote_ssrc` can be considered const while some other state represented
by rtp_config() can not and also is tied to a specific thread.
Separating access to these variables, makes moving things around easier.
Bug: webrtc:11993
Change-Id: I70aa000daab6174a401e01dca163213174e8f284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261316
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36818}
Intended to let Vp8TemporalLayersFactory (an api/ target) reuse
this function, without depending on the codec implementation, and
without introducing a dependency cycle with the webrtc_vp8 build
target.
Bug: webrtc:11607
Change-Id: I671422e994e1005da8c7d768e8dd8ff795553e51
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261308
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36816}
This reduces the surface of externally accessible state that belongs
to the class, which makes it easier to control what state belongs to
what thread. In this CL enforcing remote_ssrc() to be conceptually const
and sync_group to conceptually belong to the packet delivery thread.
Bug: webrtc:11993
Change-Id: I7de9366dc0c2bf451b5c58595c2d073b4016f2e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261450
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36813}
