Drive-by: Add empty dummy targets for all the things left in
the peerconnection target. They should all move out.
Bug: webrtc:13634
Change-Id: I93b193804668decf5feee2a8847403466330e128
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35870}
The DataChannelIntegrationTest.SomeQueuedPacketsGetDroppedInMaxRetransmitsMode
test is flaky on Android.
Bug: None
Change-Id: Ia72081905368e405441d5518b53d03e60fac233b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250120
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35868}
This CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
This is a reland of 9d230d54c7eef31ac1100f0aeef1374dd1ac62fa
Original change's description:
> (Un/)Subscribe RtpVideoSender for feedback on the transport queue.
>
> * RtpVideoSender now registers/unregisters for feedback callback
> inside of SetActive(), which runs on the transport queue.
> * Transport feedback is given on the transport queue
> * Registration/unregistration for feedback is done on the same
> * Removed the last mutex from TransportFeedbackDemuxer.
>
> Ultimately, this work is related to moving state from the Call
> class, that's related to network configuration, but due to the code
> is currently written is attached to the worker thread, over to the
> Transport, where it's used (e.g. suspended_video_send_ssrcs_).
>
> Bug: webrtc:13517, webrtc:11993
> Change-Id: I057d0e2597e6cb746b335e0308599cd547350e56
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248165
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35777}
Bug: webrtc:13517, webrtc:11993
Change-Id: I766e569abea8bae96d32267a951fcdc195ced8a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249782
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35863}
This substantially reduces the amount of text logged in release
builds when running:
rtc_unittests.exe --gtest_filter=*RtcEventLog*
Bug: none
Change-Id: I7b7c7e66fa467924e4414f1d9bfc1590ff01e0c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249981
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35859}
This will enable Chrome to inject its metronome for use in WebRTC for
tasks like synchronized decoding.
Bug: webrtc:13560, chromium:1253787
Change-Id: I2488d746f57152a32d3adf92a3cdfdfdb8000c06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249381
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35853}
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will
be used to store audio timestaps in future changes.
This is a part of the A/V sync metric metric feature for mobile. The metric
have already launched for web clients.
Bug: webrtc:13609
Change-Id: I0031843476ff1b573b262308fca52d587fae30b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35851}
Following [1], add many more checks for safe access to member variables.
This change is effectively a no-op, but landed separately from the
earlier change that's smaller but contains a fundamental assumption
gleaned from the implementation (and its use).
[1]: https://webrtc-review.googlesource.com/c/src/+/249942
Bug: webrtc:11988
Change-Id: I1568e2160c9faa6993c5b68044312f83d00e4815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249943
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35850}
This emulates behaviour from frame buffer 2, but does not handle stats.
In contrast to frame buffer 2, all work happens on the same task queue.
FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind
a field trial WebRTC-FrameBuffer3.
This separates frame scheduling behaviour into a few components,
VideoReceiveStreamTimeoutTracker
* Handles the stream timeouts.
FrameDecodeScheduler
* Manages the scheduling and cancelling of frames being sent to the
decoder.
FrameDecodeTiming
* Handles the timing and ordering of frames to be decoded.
Other changes
* Adds CurrentSize() method to FrameBuffer3
* Move timing to a separate library
* Does a thread check for Receive statistics as this is now
on the worker thread.
* Adds `FlushImmediate` method to RunLoop so that
video_receive_stream2_unittest can pass when scheduling is happening
on the worker thread.
Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721
Bug: webrtc:13343
Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35847}
This removes a couple of methods from the PeerConnectionSdpMethods
interface.
Bug: webrtc:11995
Change-Id: I0a68178b1f0a99e779e6d7f94d03b493d811f500
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249794
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35841}
This is part of a project to make sdp_offer_answer be a separate
compile target from peerconnection.
This CL affects sctp_data_channel and data_channel_utils.
Bug: webrtc:11995
Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35840}
Any Mac builder should be able to compile ARM64 builds when using target_cpu=arm64
Bug: chromium:1238267
Change-Id: I72dac3b6f170f09d5c158ec11650e0cff7b9e638
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249790
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35838}
This reverts commit 3babb8af238a531cbff27951604b09bb78b762cd.
Reason for revert:
- Causes regressions to transceivers, see https://crbug.com/1291956 for more information, including tests to reproduce the issue.
This CL is not a pure revert. While it reverts everything else, it does
keep the new enum value (kProfilePredictiveHigh444). This is as to not
break Chromium which already depend on it. It is not listed in the
kProfilePatterns though so the enum value should never be applicable.
Original change's description:
> Added support for H264 YUV444 (I444) decoding.
>
> Added Nutanix Inc. to the AUTHORS file.
>
> PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
>
> Bug: chromium:1251096
> Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35684}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1251096, chromium:1291956
Change-Id: Ib4d8ea4898f9832914d88e7076e6b39da0c804ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249791
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35835}
1) Fixes crash on dlclose when using NVidia driver
2) Closes EGLDisplay and EGLContext on destruction
3) Prints correct errors for EGL calls
Bug: chromium:1290566
Change-Id: Icfb3cad2e7c054030821479be7e48d77a4e0d5e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249795
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35833}
All uses of the RTC_DISALLOW_COPY_AND_ASSIGN macro has replaced,
so it is safe to delete this file.
Bug: webrtc:13555, webrtc:13082
Change-Id: I2db1f53d7056d1c31d3ae9daab6e705a7e6a9526
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249261
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35831}
Running bitexactness tests only on Linux makes it significantly easier to
update them, while still giving many of the same benefits.
Bug: webrtc:12518, b/216736217
Change-Id: I7f3c9a27c0fc14b7ee0e83aede2e7702cfa79141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249787
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35829}
This CL changes the SdpOfferAnswerHandler class to depend on a new class
PerConnectionInternalMethods, which is implemented by PeerConnection.
This means that SdpOfferAnswerHandler no longer depends on
PeerConnectionInterface.
This opens the way for refactoring PeerConnection so that
PeerConnectionInternalMethods is a member object (encapsulation not
inheritance), which will make it possible to break some of the
dependency cycles that make the "peerconnection" target in the BUILD
file so huge.
Bug: webrtc:11995
Change-Id: Ib8413a31c0148b8d8602764b7367dfd3068da58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249785
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35828}
This reverts the async operation introduced here:
https://webrtc-review.googlesource.com/c/src/+/248170
The race that happened was that the "flush" operation in the dtor
of ChannelManager, could run _after_ PeerConnection::Close() which
is where the Call object gets deleted. Inside the dtor of Call, there
are DCHECKs that could hit when the pending deletions hadn't run.
In most cases the Invoke() that is used to delete the Call object
would run after the pending tasks, but there's still one code path
that I'm looking for that could trigger the deletion of a channel
after Call is destructed.
Bug: webrtc:11992, webrtc:13540, chromium:1291383
Change-Id: I160742907cc0c097a4b2bb1b7c3da03b4e8cd8d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35822}
Thanks to machenbach@ for the heads up.
WebRTC's MB was generating a Swarming command that was using vpython
(defaulting on python2 on some platforms). This CL switches that to
vpython3 (fixing gtest-parallel-wrapper.py to be python3 compliant).
No-Presubmit: True
Bug: webrtc:13607
Change-Id: Icfa7d23b81e30cebfe8243d4ba65284955593465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249604
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35821}
