This CL prevents dereferencing potentially null pointer by
setting the pointer in client code.
We can now safely call PeerConnection::Close(), which happens
to trigger OnIceConnectionChange() on the observer.
This is a followup to: https://webrtc-review.googlesource.com/c/src/+/107706
Bug: webrtc:9855
Change-Id: Ieebf8415f0a12fe87d8cd80d1eb06797926005df
Reviewed-on: https://webrtc-review.googlesource.com/c/108785
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25462}
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}
to Mdns.*.
MdnsResponderInterface now explicitly requires the reference counting
of created names to allow the coexistence of multiple users of the same
responder where one user would not remove identical names created by
others.
MDns.* is also renamed to Mdns.* per the style guide.
TBR=aleloi@webrtc.org
Bug: webrtc:9605
Change-Id: I047fc41f34de8d4e97c980409a7f373769c4c252
Reviewed-on: https://webrtc-review.googlesource.com/c/101921
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25458}
This change deletes the default implementations of state and data
channel methods (SetMediaTransportStateCallback, SendData, CloseChannel,
and SetDataSink). It adds stub implementations to LoopbackMediaTransport
and FakeMediaTransport.
Bug: webrtc:9719
Change-Id: I49b7780c055b552330546b460c2e79ce8df81833
Reviewed-on: https://webrtc-review.googlesource.com/c/108940
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25457}
Adds the types and methods required for sending and receiving data
channel messages over the media transport. These are:
- A DataMessageType to distinguish between text, binary, and control
messages
- A parameters struct for sending data messages, which specifies the
channel id, type, and ordering/reliability parameters
- A sink for data-channel related callbacks (receive data, begin
closing procedure, and end closing procedure)
- A method to set the sink for data channels
- Methods to open, close, and send on data channels
These methods, combined with the state sink, allow PeerConnection to
implement the DataChannelProviderInterface using MediaTransport as the
underlying transport.
Change-Id: Iccb2ba374594762a5b4f995564e2a1ff7d8805f5
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/108541
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25454}
This allows to use secure, end to end communication if SDES cryptos are
passed. MediaTransport can use a derived key to secure its own
communication.
Bug: webrtc:9719
Change-Id: If1a20b136b3b4af0cb24f10b52fc5ce1eb31daa2
Reviewed-on: https://webrtc-review.googlesource.com/c/108504
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25452}
This makes the calculation more similar to the one in WebRTCVoiceEngine.
Bug: webrtc:9510
Change-Id: Ibca69842726e51c07b9cc9550ff9f15a24161e28
Reviewed-on: https://webrtc-review.googlesource.com/c/107653
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25448}
Moved methods: GetReadData, ConsumeReadData, GetWriteBuffer,
ConsumeWriteBuffer, GetWriteRemaining.
These methods represented an optional interface for reading and
writing streams, intended to optimize certain use cases. However,
it was implemented only in the FifoBuffer subclass, and the few
users of that class all have a concrete FifoBuffer, and hence
don't need the methods on the abstract StreamInterface.
Bug: webrtc:6424
Change-Id: I6de74d1a9205fcb7037ad84e24679d4a27c1d219
Reviewed-on: https://webrtc-review.googlesource.com/c/108621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25446}
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 change just wraps the openssl key derivation functions in a simple
interface in a similar way to how we do it for messagedigest.h so we aren't
coupled to openssl in the core implementation.
Bug: webrtc:9917
Change-Id: I8556bd6e38b7da34d93abbe29415c3366f6532ba
Reviewed-on: https://webrtc-review.googlesource.com/c/107981
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25440}
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}
Refactor only remaining user, IsDefaultRoute (helper function
called from BasicNetworkManager::IsIgnoredNetwork) to use a
FILE* and fgets instead.
Bug: webrtc:6424
Change-Id: I57652f664b9a6965c19575c1b5d7f7de24f2ed44
Reviewed-on: https://webrtc-review.googlesource.com/c/108089
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25433}
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}
Clients of media_transport_interface need the ability to monitor BWE
estimates, and this change adds a TargetBitrate observer to the media
transport interface.
Bug: webrtc:9719
Change-Id: I90ebbf684c6f269e0c3cd58428010cfa511cc970
Reviewed-on: https://webrtc-review.googlesource.com/c/108106
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25415}
The struct is more generic and easier to extend than parameters to the
Factory. In addition, the list of parameters to the factory might grow,
making invocations awkward if not difficult to read.
Bug: webrtc:9719
Change-Id: I4b98e26f1f4c0d5ea840f9c28e7ed7abee072b74
Reviewed-on: https://webrtc-review.googlesource.com/c/107984
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25413}
