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Cherry-picked commit for codec blacklist. #6

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Cherry-picked commit for codec blacklist. #6

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@jianjunz jianjunz commented Mar 28, 2019 •
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This change is originally contributed by Hank.

@jianjunz
Cherry-picked commit for codec blacklist.
445672d 
Change-Id: I454f7800cae883b4623ec80d50e7c1a74b853a9f
Reviewed-on: https://git-ccr-1.devtools.intel.com/gerrit/98200
Tested-by: webrtctest <webrtctest@intel.com>
Reviewed-by: Zhu, Jianjun <jianjun.zhu@intel.com>
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Close this one because it is included in PR7.

@jianjunz jianjunz closed this Mar 30, 2019
taste1981 pushed a commit that referenced this pull request Dec 30, 2020
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}
taste1981 pushed a commit that referenced this pull request Dec 30, 2020
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}
JamesTerm pushed a commit to JamesTerm/owt-deps-webrtc that referenced this pull request Apr 15, 2021
For simulcast screenshare, make 2 tl default for high stream.
22c8fc5 
(cherry picked from commit 7461eff)

Bug: webrtc:9734, chromium:883434
Change-Id: I00400782686296b191f0f7a10a65f99253bea929
Reviewed-on: https://webrtc-review.googlesource.com/99101
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#24642}
Reviewed-on: https://webrtc-review.googlesource.com/100101
Cr-Commit-Position: refs/branch-heads/70@{open-webrtc-toolkit#6}
Cr-Branched-From: f18b352-refs/heads/master@{#24472}
@jianjunz jianjunz deleted the encoder branch June 4, 2021 07:17
Dragon-S pushed a commit to Dragon-S/owt-deps-webrtc that referenced this pull request Sep 22, 2021
Merge M89: Fix low-latency renderer with unset playout delays
742109a 
The low-latency renderer is activated by the RTP header extension
playout-delay if the min value is set to 0 and the max value is
set to something greater than 0.

According to the specification of the playout-delay header
extension it doesn't have to be set for every frame but only if
it is changed. The bug that this CL fixes occured if a playout
delay had been set previously but some frames without any specified
playout-delay were received. In this case max composition delay
would not be set and the low-latency renderer algorithm would be
disabled for the rest of the session.

(cherry picked from commit 0093a38)

Bug: chromium:1138888
Change-Id: I12d10715fd5ec29f6ee78296ddfe975d7edab8a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208581
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#33330}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211000
Cr-Commit-Position: refs/branch-heads/4389@{open-webrtc-toolkit#6}
Cr-Branched-From: 7acc2d9-refs/heads/master@{#32986}
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