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Merged
merged 10 commits into from
Aug 26, 2020
Merged

Enable remote audio recording #3

merged 10 commits into from
Aug 26, 2020

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@mininny
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@mininny mininny commented Aug 5, 2020 •
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This PR adds an RTCAudioSink interface for recording remote/local audio stream for iOS Devices.
It hijacks RTCPeerConnectionFactory, and adds AudioSourceSink c++ interface into AudioDeviceModuleIOS and AudioDeviceIOS.

When AudioDeviceIOS::OnGetPlayoutData is called, respective method in RTCAudioSink for remote audio stream is called, and AudioDeviceIOS::OnDeliverRecordedData calls the respective method in RTCAudioSink for local audio stream.

By overriding RTCAudioSink, you can add a sink to local/remote streams of ongoing call, and use them to record into audio files.

@mininny mininny changed the title [WIP] Enable remote audio recording Enable remote audio recording Aug 5, 2020
@sf-nathan-park
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Local audio recording쪽도 interface가 없었나요?

@mininny
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mininny commented Aug 6, 2020

Local audio recording쪽도 interface가 없었나요?

WebRTC내에서는 인터페이스가 따로 없었습니당. iOS에서 제공하는 시스템오디오를 녹음하는 거는 있었는데, remote랑 통일성있게 WebRTC stream으로 녹화하는게 좋을것같아서, local쪽도 interface를 추가하였습니다.

@sf-nathan-park
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엇 그랬군요 Android에는 local쪽 interface만 있어서 remote쪽 비슷하게 추가했습니다. JavaAudioDevice 쪽에 추가하고 PeerConnectionFactory builder에 set한 모양이라서 거의 비슷한 모양새일 것 같습니다. LGTM! GOGO

sf-nathan-park
sf-nathan-park approved these changes Aug 6, 2020
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LGTM! 🚀

@mininny mininny requested a review from x-0o0 August 11, 2020 07:34
x-0o0
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LG™️

ipagong
ipagong approved these changes Aug 26, 2020
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👍 👍 👍

@mininny mininny merged commit eedf0c4 into develop Aug 26, 2020
mininny pushed a commit that referenced this pull request Aug 26, 2020
Merge to M84: Fixes excessive stats updating in TaskQueuePacedSender.
f9e2d03 
Merge note: Production code is a plain merge, unit tests needed some
back-porting due to refactoring of test harness in tot.

TaskQueuePacedSender::MaybeUpdateStats() is intended to be called when
packets are sent or by a sequence of "scheduled" calls. There should
only be one scheduled call in flight at a time - and that one
reschedules itself if needed when it runs.

A bug however caused the "schedules task in flight" flag to
incorrectly be set to false, leading to more and more schedules tasks
being alive - eating CPU cycles.

This CL fixes that and also makes sure the queue time properly goes
down to zero before the next idle interval check, even if there are no
more packets to send.

(cherry picked from commit 998524a)

Bug: webrtc:10809
Change-Id: I4e13fcf95619a43dcaf0ed38bce9684a5b0d8d5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176330
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#31390}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176409
Cr-Commit-Position: refs/branch-heads/4147@{#3}
Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
mininny pushed a commit that referenced this pull request Feb 24, 2021
Revert "NV12 support for VP8 simulcast"
7ef8093 
This reverts commit 76d3e7a.

Reason for revert: Causes multiple Chromium WPT tests to crash, preventing rolls.

Sample failed run:
https://ci.chromium.org/p/chromium/builders/try/win10_chromium_x64_rel_ng/685757?

Sample stack trace:
#0 0x7ff8623fbde9 base::debug::CollectStackTrace()
STDERR: #1 0x7ff862311ca3 [2665012:17:1009/162250.249660:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652370324, interval (us) = 834
STDERR: base::debug::StackTrace::StackTrace()
STDERR: #2 0x7ff8623fb93b base::debug::(anonymous namespace)::StackDumpSignalHandler()
STDERR: #3 0x7ff857a70140 [2665012:17:1009/162250.249947:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652370634, interval (us) = 742
STDERR: (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f)
STDERR: #4 0x7ff85778edb1 gsignal
STDERR: #5 0x7ff857778537 abort
STDERR: #6 0x7ff855d5eee2 [2665012:17:1009/162250.250342:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652371030, interval (us) = 706
STDERR: [2665012:17:1009/162250.250514:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652371204, interval (us) = 963
STDERR: rtc::webrtc_checks_impl::FatalLog()
STDERR: #7 0x7ff855f14e62 webrtc::LibvpxVp8Encoder::PrepareRawImagesForEncoding()
STDERR: #8 0x7ff855f14412 webrtc::LibvpxVp8Encoder::Encode()
STDERR: #9 0x7ff855bae765 webrtc::SimulcastEncoderAdapter::Encode()
STDERR: #10 0x7ff85607d598 webrtc::VideoStreamEncoder::EncodeVideoFrame()
STDERR: #11 0x7ff85607c60d webrtc::VideoStreamEncoder::MaybeEncodeVideoFrame()
STDERR: #12 0x7ff8560816f5 webrtc::webrtc_new_closure_impl::ClosureTask<>::Run()
STDERR: #13 0x7ff855b352b5 (anonymous namespace)::WebrtcTaskQueue::RunTask()
STDERR: #14 0x7ff855b3531e base::internal::Invoker<>::RunOnce()
STDERR: #15 0x7ff86239785b base::TaskAnnotator::RunTask()
STDERR: #16 0x7ff8623c8557 base::internal::TaskTracker::RunSkipOnShutdown()
STDERR: #17 0x7ff8623c7d92 base::internal::TaskTracker::RunTask()
STDERR: #18 0x7ff862415a06 base::internal::TaskTrackerPosix::RunTask()
STDERR: #19 0x7ff8623c75e6 base::internal::TaskTracker::RunAndPopNextTask()
STDERR: #20 0x7ff8623d3a8d base::internal::WorkerThread::RunWorker()
STDERR: #21 0x7ff8623d368d base::internal::WorkerThread::RunPooledWorker()
STDERR: #22 0x7ff862416509 base::(anonymous namespace)::ThreadFunc()
STDERR: #23 0x7ff857a64ea7 start_thread 

Original change's description:
> NV12 support for VP8 simulcast
>
> Tested using video_loopback with generated NV12 frames.
>
> Bug: webrtc:11635, webrtc:11975
> Change-Id: I14b2d663c55a83d80e48e226fcf706cb18903193
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186722
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32325}

TBR=ilnik@webrtc.org,eshr@google.com

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11635
Bug: webrtc:11975
Change-Id: I61c8aed1270bc9c2f7f0577fa5ca717a325f548a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187484
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32369}
mininny pushed a commit that referenced this pull request Feb 24, 2021
Revert "Reland "Replace sigslot usages with robocaller library.""
ff79132 
This reverts commit c5f7108.

Reason for revert: Causes Chromium WPT Tests to fail, preventing rolls.

Sample failed run:
https://ci.chromium.org/p/chromium/builders/try/linux-rel/511995?

Sample logs:

STDERR: # Fatal error in: ../../third_party/webrtc/pc/peer_connection.cc, line 575
STDERR: # last system error: 0
STDERR: # Check failed: (signaling_thread())->IsCurrent()
STDERR: # Received signal 6
STDERR: #0 0x7f81d39e3de9 base::debug::CollectStackTrace()
STDERR: #1 0x7f81d38f9ca3 base::debug::StackTrace::StackTrace()
STDERR: #2 0x7f81d39e393b base::debug::(anonymous namespace)::StackDumpSignalHandler()
STDERR: #3 0x7f81c9054140 (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f)
STDERR: #4 0x7f81c8d72db1 gsignal
STDERR: #5 0x7f81c8d5c537 abort
STDERR: #6 0x7f81c7344032 rtc::webrtc_checks_impl::FatalLog()
STDERR: #7 0x7f81c722e5c0 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>()
STDERR: #8 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach()
STDERR: #9 0x7f81c72d6e8e webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>()
STDERR: #10 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach()
STDERR: #11 0x7f81c71c6df3 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>()
STDERR: #12 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach()
STDERR: #13 0x7f81c73135bc rtc::OpenSSLStreamAdapter::ContinueSSL()
STDERR: #14 0x7f81c7312fd4 rtc::OpenSSLStreamAdapter::OnEvent()
STDERR: #15 0x7f81c71c30d9 cricket::StreamInterfaceChannel::OnPacketReceived()
STDERR: #16 0x7f81c71c640a cricket::DtlsTransport::OnReadPacket()
STDERR: #17 0x7f81c71cad61 cricket::P2PTransportChannel::OnReadPacket()
STDERR: #18 0x7f81c71bc90f cricket::Connection::OnReadPacket()
STDERR: #19 0x7f81c71e6255 cricket::UDPPort::HandleIncomingPacket()
STDERR: #20 0x7f81cd1f1bff blink::(anonymous namespace)::IpcPacketSocket::OnDataReceived()
STDERR: #21 0x7f81cd1f645d blink::P2PSocketClientImpl::DataReceived()
STDERR: #22 0x7f81cd50a16b network::mojom::blink::P2PSocketClientStubDispatch::Accept()
STDERR: #23 0x7f81d2b4f642 mojo::InterfaceEndpointClient::HandleValidatedMessage()
STDERR: #24 0x7f81d2b5304b mojo::MessageDispatcher::Accept()
STDERR: #25 0x7f81d2b50bb1 mojo::InterfaceEndpointClient::HandleIncomingMessage()
STDERR: #26 0x7f81d2b58a3a mojo::internal::MultiplexRouter::ProcessIncomingMessage()
STDERR: #27 0x7f81d2b57f7f mojo::internal::MultiplexRouter::Accept()
STDERR: #28 0x7f81d2b5304b mojo::MessageDispatcher::Accept()
STDERR: #29 0x7f81d2b48851 mojo::Connector::DispatchMessage()
STDERR: #30 0x7f81d2b492e7 mojo::Connector::ReadAllAvailableMessages()
STDERR: #31 0x7f81d2b490a3 mojo::Connector::OnHandleReadyInternal()
STDERR: #32 0x7f81d2b498f0 mojo::SimpleWatcher::DiscardReadyState()
STDERR: #33 0x7f81d2d0e67a mojo::SimpleWatcher::OnHandleReady()
STDERR: #34 0x7f81d2d0eb2b base::internal::Invoker<>::RunOnce()
STDERR: #35 0x7f81d397f85b base::TaskAnnotator::RunTask()
STDERR: #36 0x7f81d399a04c base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl()
STDERR: #37 0x7f81d3999c78 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork()
STDERR: #38 0x7f81d391fe64 base::MessagePumpDefault::Run()
STDERR: #39 0x7f81d399a8dc base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run()
STDERR: #40 0x7f81d395ae55 base::RunLoop::Run()
STDERR: #41 0x7f81d39c87f2 base::Thread::Run()




Original change's description:
> Reland "Replace sigslot usages with robocaller library."
>
> This is a reland of 40261c3
>
> Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError
> added a new member with a different name and used it in webrtc code.
> After this change do two more follow up CLs to completely remove the old code
> from google3.
>
> Original change's description:
> > Replace sigslot usages with robocaller library.
> >
> > - Replace all the top level signals from jsep_transport_controller.
> > - There are still sigslot usages in this file so keep the inheritance
> >   and that is the reason for not having a binary size gain in this CL.
> >
> > Bug: webrtc:11943
> > Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32321}
>
> Bug: webrtc:11943
> Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32359}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org

Change-Id: I6bce1775d10758ac4a9d05b855f473dd70bd9815
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187487
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32372}
mininny pushed a commit that referenced this pull request Feb 24, 2021
Fixes issue triggered by WebRTC-VP9-PerformanceFlags trial.
54d4910 
Using WebRTC-VP9-PerformanceFlags and settings a multi-layer config,
and then configuring the codec in non-svc mode would cause us to not
set the cpu speed in libvpx. For some reason, that could trigger a
crash in the encoder.

This CL fixes that, and adds new test coverage for the code affected
byt the trial.

(cherry picked from commit 03eed7c)

Bug: chromium:1167353, webrtc:11551
Change-Id: Iddb92fe03fc12bac37717908a8b5df4f3d411bf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202761
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#33051}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203521
Cr-Commit-Position: refs/branch-heads/4389@{#3}
Cr-Branched-From: 7acc2d9-refs/heads/master@{#32986}
mininny pushed a commit that referenced this pull request Sep 3, 2021
Slight code clarification in RemoveStoppedTransceivers.
7fa8d46 
There's no change in functionality, which was verified by adding
an 'else' catch-all clause in the loop with an RTC_NOTREACHED()
statement. See patchset #3.

This is mostly a cosmetic change that modifies the loop such that
it's guaranteed that Remove() is always called for transceivers
whose state is "stopped" and there's just one place where Remove()
is called.

Bug: none
Change-Id: Iffe237bb2f08e5e6ef316a6b76c4b183df671f3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215232
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33765}
mininny pushed a commit that referenced this pull request Sep 3, 2021
Merge M93: Calculate pacing delay based on decode start time
caf19bc 
Schedule the frames to be decoded based on the pacing delay from the
last decode scheduled time. In the current implementation, multiple
threads and different functions in same thread can call
MaxWaitingTime(), thereby increasing the wait time each time the
function is called. Instead of returning the wait time for a future
frame based on the number of times the function is called, return the
wait time only for the next frame to be decoded. Threads can call the
function repeatedly to check the waiting time for next frame and wake
up and then go back to waiting if an encoded frame is not available.

(cherry picked from commit 82c2248)

Change-Id: I00886c1619599f94bde5d5eb87405572e435bd73
Bug: chromium:1237402
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226502
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#34660}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228532
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/branch-heads/4577@{#3}
Cr-Branched-From: 5196931-refs/heads/master@{#34463}
mininny pushed a commit that referenced this pull request Jul 28, 2022
@boivie
dcsctp: Abandon chunks marked for fast retransmit
784f4cb 
The current code assumed that chunks that were scheduled for fast
retransmission would never be abandoned, as chunks marked for fast
retransmission would be immediately sent after the SACK has been
processed, giving no time for them to be abandoned.

But fuzzers keep on fuzzing, and can craft a sequence of chunks that
result in a SACK that both marks the chunks for fast retransmission and
later (while processing the same SACK) abandons them.

(cherry picked from commit 7726b7d)

Bug: chromium:1331087
Change-Id: Id218607e18a6f3a9d6d51044dccb920e1e77372a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264960
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#37108}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265163
Cr-Commit-Position: refs/branch-heads/5060@{#3}
Cr-Branched-From: 1389c76-refs/heads/main@{#36868}
mininny pushed a commit that referenced this pull request Nov 17, 2023
Implement support for Chrome task origin tracing. #3.5/4
a1ceae2 
This CL migrates unit tests to the new TaskQueueBase interface.

Bug: chromium:1416199
Change-Id: Ic15c694b28eb67450ac99fdd56754de1246a4d95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295621
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39434}
mininny pushed a commit that referenced this pull request Nov 17, 2023
Implement support for Chrome task origin tracing. #3.6/4
c0f8813 
This CL migrates the task queue paced sender unit test
to the new TaskQueueBase interface.

Bug: chromium:1416199
Change-Id: Id0568bb9a08bf43b92e33fdf45fe75a57e5a7a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39436}
mininny pushed a commit that referenced this pull request Nov 17, 2023
Implement support for Chrome task origin tracing. #3.7/4
ae61aca 
This CL completes migration to the new TaskQueueBase interface
permitting location tracing in Chrome.

Bug: chromium:1416199
Change-Id: Iff7ff5796752a1520384a3db0135a1d4b9438988
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294540
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39439}
mininny pushed a commit that referenced this pull request Nov 17, 2023
Implement support for Chrome task origin tracing. #3.9/4
fb727f3 
This CL forwards repeating task client locations to the passed
task queue.

Bug: chromium:1416199
Change-Id: I437d596f8d327d13498b47dfc0a03812af870331
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295623
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39443}
mininny pushed a commit that referenced this pull request Nov 17, 2023
Implement support for Chrome task origin tracing. #3.8/4
2fa39bd 
This CL forwards TaskQueue locations to the contained
task queue.

Bug: chromium:1416199
Change-Id: I989ae445a67991bf5a857407135dbe8bacbd3c55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295622
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39446}
mininny pushed a commit that referenced this pull request Nov 17, 2023
[M118] Fire MaybeSignalReadyToSend in a PostTask when recursive
d8f2b03 
Speculative fix. Writing the test for it is more complex.

(cherry picked from commit 83894d3)

Bug: chromium:1483874
Change-Id: Icfaf1524b0499c609010753e0b6f3cadbd0e98f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321480
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#40820}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322124
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5993@{#3}
Cr-Branched-From: 5afcec0-refs/heads/main@{#40703}
sf-jed-kyung pushed a commit that referenced this pull request Jul 7, 2025
Fix license name in README.chromium for fft4g
cec4dae 
(cherry picked from commit f844699)

Bug: chromium:414606466
Change-Id: I627dd5b01e157125c9811ec36086efd7e16855b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389920
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Auto-Submit: Andrew Grieve <agrieve@google.com>
Cr-Original-Commit-Position: refs/heads/main@{#44522}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/391000
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/branch-heads/7151@{#3}
Cr-Branched-From: dc428bd-refs/heads/main@{#44472}
sf-jed-kyung pushed a commit that referenced this pull request Aug 25, 2025
Use FieldTrialsView::IsEnabled for DTLS 1.3
23d8e44 
No behavior changes.

(cherry picked from commit 5ff715d)

Bug: webrtc:383141571, chromium:433885045, chromium:434133034
Change-Id: Ice5f3e5cbd245ddea407248a6f29c61c646e6a72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/401740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#45206}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/402200
Cr-Commit-Position: refs/branch-heads/7258@{#3}
Cr-Branched-From: 74fa937-refs/heads/main@{#44974}
sf-jed-kyung pushed a commit that referenced this pull request Oct 16, 2025
@palak8669
[M141] Add logging while creating {Window|Screen}Capturer
d2eaa55 
(cherry picked from commit 6908505ae0eba5d530ad0bb4b37d4654a4f36c95)

Fixed: chromium:450184498
Bug: chromium:448881311
Change-Id: I0f043f58bf831e6822451cac99a8972a054ffdf7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/414800
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Original-Commit-Position: refs/heads/main@{#45856}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/415420
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/branch-heads/7390@{#3}
Cr-Branched-From: 2f553bf-refs/heads/main@{#45520}
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@danney-chun danney-chun Awaiting requested review from danney-chun

@famous90 famous90 Awaiting requested review from famous90

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@ipagong ipagong ipagong approved these changes

@sf-nathan-park sf-nathan-park sf-nathan-park approved these changes

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@mininny @sf-nathan-park @ipagong @x-0o0