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WIP: [Do Not Merge]Enable partial-output packetization for slice-based encoding #15
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taste1981
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This patch is placed here allowing sending h.264 bitstream chunk-by-chunk without requiring it to be an entire AU before packetization; a chunk is allowed to be either a group of NALUs, or even splitted within of of the NALU. It however, assumes end of each chunk, except last chunk, is not a complete NAL. So to apply this patch, pre-condition is the 'slice-based-encoding' here is partial-output of one slice instead of real multi-slice encoding. SDK customized encoder needs to be updated as well for this patch to work: |
taste1981
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Dec 30, 2020
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
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Dec 30, 2020
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
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Apr 15, 2021
Bug: chromium:892040,webrtc:9816 Change-Id: I46e8b2de61eedf67e235fcea8f3b9e85f690e64f Reviewed-on: https://webrtc-review.googlesource.com/c/103661 Reviewed-by: Per Åhgren <peah@webrtc.org> TBR: peah@webrtc.org Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#24982} Reviewed-on: https://webrtc-review.googlesource.com/c/104742 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/branch-heads/70@{open-webrtc-toolkit#15} Cr-Branched-From: f18b352-refs/heads/master@{#24472}
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webrtc stack change for slice-based encoding enabling. Now allow slice to be not started with NAL header, and also allow last NAL in the slice to be in-complete.