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@mininny mininny commented Aug 3, 2020

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@mininny mininny closed this Aug 5, 2020
@mininny mininny deleted the feature/mininny/add-remote-audio-sink branch August 5, 2020 14:44
mininny pushed a commit that referenced this pull request Aug 26, 2020
TBR=sprang@webrtc.org
(cherry picked from commit 43c108b)

Bug: chromium:1084963
Change-Id: I2c6b6a2a62bbcd058b8ed336e6e0f36b8b0d0844
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175220
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#31321}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175903
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/branch-heads/4147@{#2}
Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
mininny pushed a commit that referenced this pull request Feb 24, 2021
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
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
DependencyDescriptor and vp9 wrapper understand key frame differently
when it comes to the first layer frame with spatial_id>0
This CL adds and use DD's interpretation of the key frame when deciding
if DD should be supported going forward.

(cherry picked from commit 0be1846)

Bug: webrtc:11999, chromium:1169060
Change-Id: I11a809a315e18bd856bb391576c6ea1f427e33be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202760
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#33046}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203263
Cr-Commit-Position: refs/branch-heads/4389@{#2}
Cr-Branched-From: 7acc2d9-refs/heads/master@{#32986}
mininny pushed a commit that referenced this pull request Sep 3, 2021
…transport at add-track"

This reverts commit 7a2db8a.

After this commit, the PeerConnection would assume that any new m=
sections will be added to the first existing BUNDLE group. This is true
of JSEP endpoints (if they don't do SDP munging), but is not necessarily
true for non-JSEP endpoints. This breaks the following scenarios:

* Remote offer adding a new m= section that's not part of any BUNDLE group.
* Remote offer adding a m= section to the second BUNDLE group.

The latter is specifically problematic for any application that wants
to bundle all audio streams in one group and all video streams in
another group when using Unified Plan SDP, to replicate the behavior of
using now-deprecated Plan B without bundling.

TBR=hta@webrtc.org

Bug: webrtc:12837, webrtc:12906, chromium:1236202
Change-Id: I97a348c96443dee95e2b42792b73ab7b101fd64c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227681
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/branch-heads/4577@{#2}
Cr-Branched-From: 5196931-refs/heads/master@{#34463}
mininny pushed a commit that referenced this pull request Apr 18, 2022
…eBuffer3

R=​philipel@webrtc.org

(cherry picked from commit d4ff12f)

Change-Id: I9a961417f86fd1f968a42730249d479e3ebd4784
Bug: webrtc:13343, webrtc:13755, chromium:1302207
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252583
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#36100}
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253940
Cr-Commit-Position: refs/branch-heads/4896@{#2}
Cr-Branched-From: 962bf18-refs/heads/main@{#36026}
mininny pushed a commit that referenced this pull request Jul 28, 2022
This allows for its use in test targets in Chromium Windows, which fixes
the compiled errors found in https://chromium-review.googlesource.com/c/chromium/src/+/3649679

(cherry picked from commit 5b8dc1d)

Change-Id: I738b2eaab8eca73c40e847ede67ff5e7757ec512
Bug: chromium:1331333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262811
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#36939}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264981
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5060@{#2}
Cr-Branched-From: 1389c76-refs/heads/main@{#36868}
mininny pushed a commit that referenced this pull request May 9, 2023
This reverts commit be03c09.
Causes regression in web projects that
1/ add a stopped-by-default extension in SRD
2/ call createAnswer
3/ munge the stopped-by-default extension back in SLD
4/ create a subsequent offer and expect the extension to be present

BUG=chromium:1427442,chromium:1051821

Change-Id: I2e48831e92384963a254d873398f54eaee96739a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299143
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/branch-heads/5615@{#2}
Cr-Branched-From: cdfeb4f-refs/heads/main@{#39376}
mininny pushed a commit that referenced this pull request Nov 17, 2023
This prepares TaskQueueBase sub classes to be able to migrate to
the location and traits-based API. It re-introduces a Location class
into the webrtc namespace, which is meant to be overridden by Chromium.

Bug: chromium:1416199
Change-Id: I712c7806a71b3b99b2a2bf95e555b357c21c15ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294381
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
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@{#39400}
mininny pushed a commit that referenced this pull request Nov 17, 2023
BUG=chromium:1478690

(cherry picked from commit a8e3111)

Change-Id: I7a1caad7bbd2fc82507b61b59be71546494a304c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319580
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Original-Commit-Position: refs/heads/main@{#40724}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320580
Cr-Commit-Position: refs/branch-heads/5993@{#2}
Cr-Branched-From: 5afcec0-refs/heads/main@{#40703}
sf-jed-kyung pushed a commit that referenced this pull request Jul 7, 2025
This adds two UMA metrics for the type of SDP munging that occurred and
the port of the candidate that was restricted. The metrics descriptions
are being added here: crrev.com/c/6521706.

(cherry picked from commit d5b3b1e)

Bug: b/409713509
Fixed: b/417142969
Change-Id: I3232cb0cee6848074ab103f4d4edb2e080e09568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Tom Van Goethem <tov@google.com>
Cr-Original-Commit-Position: refs/heads/main@{#44559}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/390920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/branch-heads/7151@{#2}
Cr-Branched-From: dc428bd-refs/heads/main@{#44472}
sf-jed-kyung pushed a commit that referenced this pull request Aug 25, 2025
Original change's description:
> Add chrome-cherry-picker account to bot allowlist
> 
> chrome-cherry-picker@chops-service-accounts.iam.gserviceaccount.com is
> being by the Chrome Cherry Picker (go/chromecherrypicker) and needs to
> be able to skip the author check for presubmits.
> 
> Bug: chromium:414375466
> Change-Id: Ib9f15dd67a4efe5346e6631135e1bcd7196b992c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/400480
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Gennady Tsitovich <gtsitovich@google.com>
> Cr-Commit-Position: refs/heads/main@{#45148}

Bug: chromium:431157710,chromium:414375466
Change-Id: Ib9f15dd67a4efe5346e6631135e1bcd7196b992c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/400700
Commit-Queue: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Chrome Cherry Picker <chrome-cherry-picker@chops-service-accounts.iam.gserviceaccount.com>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/branch-heads/7258@{#2}
Cr-Branched-From: 74fa937-refs/heads/main@{#44974}
sf-jed-kyung pushed a commit that referenced this pull request Sep 17, 2025
DTLS 1.3 considers itself connected earlier than DTLS 1.2 did - when
second flight reaches the client. This CL fixes a bug that when
client is connected (state_ != SSL_CONNECTING), it would not
continue retransmitting. Continuous retransmission is needed
when the third flight is lost multiple times. Or really anytime that DTLS request it :)

This fixes the TODO in dtls_ice_integrationtest.cc in which dtls1.3
spuriously failed with certain (packet loss intensive) configurations.

CREDITS: sergeysu@ that found and fixed the problem!

(cherry picked from commit 4221e1b)

No-Try: true
Bug: chromium:441245658, chromium:441486101
Change-Id: I3302f6f384d7e4cda090184094a6fadaf7e4f129
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406320
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#45439}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407780
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/branch-heads/7339@{#2}
Cr-Branched-From: 9bd6475-refs/heads/main@{#45270}
sf-jed-kyung pushed a commit that referenced this pull request Oct 16, 2025
This condition neglected the fact that if Insertable Streams API is used
we can have frames or samples despite packets never being received.

This CL fixes this unintended regression.

# Ignore unrelated compile issues on ios webrtc bots
NOTRY=True

(cherry picked from commit c15949eda5a00122e2f3b5a643e15781049b9927)

Bug: chromium:444048024, chromium:444384230
Change-Id: Ie6e17a3bc96701476787f5898446f3f706715d15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/408884
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#45616}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/409020
Cr-Commit-Position: refs/branch-heads/7390@{#2}
Cr-Branched-From: 2f553bf-refs/heads/main@{#45520}
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