[Temporary] Rebase webrtc #4
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…ors on key-frames [email protected] (cherry picked from commit 35fc153) Bug: webrtc:11575, chromium:1084963 Change-Id: I09be17ab5155e9f610c8f7c451ca52d7d65e24d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175222 Commit-Queue: Ilya Nikolaevskiy <[email protected]> Reviewed-by: Rasmus Brandt <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31295} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175902 Reviewed-by: Ilya Nikolaevskiy <[email protected]> Cr-Commit-Position: refs/branch-heads/4147@{#1} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
[email protected] (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 <[email protected]> Commit-Queue: Ilya Nikolaevskiy <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31321} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175903 Reviewed-by: Ilya Nikolaevskiy <[email protected]> Cr-Commit-Position: refs/branch-heads/4147@{#2} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
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 <[email protected]> Commit-Queue: Erik Språng <[email protected]> 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}
This fixes a crash that could happen if substreams exist but there is no kMedia substream yet. There was an assumption that we either had no substreams or at least one kMedia substream, but this was not true. The correct thing to do is to ignore substream stats that are not associated with any kMedia substream, because we only produce "outbound-rtp" stats objects for existing kMedia substreams. A test is added to make sure no stats are returned. Prior to the fix, this test would crash. (cherry picked from commit e917379) Bug: chromium:1090712 Change-Id: Ib1f8494a162542ae56bdd2df7618775a3473419b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176446 Commit-Queue: Henrik Boström <[email protected]> Reviewed-by: Erik Språng <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31442} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176751 Reviewed-by: Henrik Boström <[email protected]> Cr-Commit-Position: refs/branch-heads/4147@{#4} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
Unit test added to verify root cause is fixed. Scenario test added to verify high-level behavior. (cherry picked from commit 576db1b) Bug: webrtc:11654, chromium:1092476 Change-Id: I1ad6e2750f5272e86b4198749edbbf5dfd8315c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176564 Commit-Queue: Erik Språng <[email protected]> Reviewed-by: Erik Språng <[email protected]> Reviewed-by: Sebastian Jansson <[email protected]> Reviewed-by: Ilya Nikolaevskiy <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31462} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176840 Cr-Commit-Position: refs/branch-heads/4147@{#5} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
When video frame encoding is done on an external thread (for example in the case of hardware encoders), the WebRTC TaskQueueBase::Current() is null; in this case use the worker queue instead to send transformed frames. (cherry picked from commit 2e69660) Bug: chromium:1086373 Change-Id: I903ddc52ad6832557fc5b5f76396fe26cf5a88f3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176303 Reviewed-by: Magnus Flodman <[email protected]> Reviewed-by: Danil Chapovalov <[email protected]> Commit-Queue: Marina Ciocea <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31388} TBR: [email protected] Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176844 Commit-Queue: Guido Urdaneta <[email protected]> Reviewed-by: Guido Urdaneta <[email protected]> Cr-Commit-Position: refs/branch-heads/4147@{#6} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
…eger scaling factors are used // Skipping CQ, because bots are broken. The same bot passed yesterday. // Now two windows bots fail with "no executable found" or gn errors. [email protected] (cherry picked from commit 09eb6e2) No-Try: True Bug: webrtc:11652 Change-Id: Id3642d607f62b72a567d521d9874b8588c2ce429 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176517 Reviewed-by: Erik Språng <[email protected]> Commit-Queue: Ilya Nikolaevskiy <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31465} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176848 Reviewed-by: Ilya Nikolaevskiy <[email protected]> Cr-Commit-Position: refs/branch-heads/4147@{#7} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
…excessive rounding (cherry picked from commit 2899b3b) Bug: webrtc:11652 Change-Id: I8bfa91c3115d6ebb17beefbb2a5e51efbbd599e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177000 Commit-Queue: Ilya Nikolaevskiy <[email protected]> Reviewed-by: Erik Språng <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31502} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177105 Cr-Commit-Position: refs/branch-heads/4147@{#8} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
We were using the address of the SctpTransport object as the sconn_addr field in usrsctp, which is used to get access to the SctpTransport object in various callbacks. However, this address is sent in the clear in the SCTP cookie, which is undesirable. This change uses a monotonically increasing id instead, which is mapped back to a SctpTransport using a SctpTransportMap helper class. [email protected] Bug: chromium:1076703 Change-Id: I5c6a44801293e3b0aacd032f16f41802f4fecf6d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176422 Reviewed-by: Harald Alvestrand <[email protected]> Reviewed-by: Tommi <[email protected]> Commit-Queue: Taylor <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31449} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177330 Reviewed-by: Taylor <[email protected]> Cr-Commit-Position: refs/branch-heads/4147@{#9} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
…or vp9 1) Fix several typos and small mistakes which could lead to crashes 2) Adjust bitrates if leading layers are disabled 3) Wire up webrtc quality scaler (cherry picked from commit 7a82467) Bug: webrtc:11319 Change-Id: I16e52bdb1c315d64906288e4f2be55fe698d5ceb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177525 Commit-Queue: Ilya Nikolaevskiy <[email protected]> Reviewed-by: Erik Språng <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31546} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177660 Cr-Commit-Position: refs/branch-heads/4147@{#10} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
(cherry picked from commit c7c412a) [email protected] Bug: chromium:1104061 Change-Id: I52d44ff1603341777a873e747c625665bc11bfa5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179161 Commit-Queue: Taylor <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#31720} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179900 Reviewed-by: Taylor <[email protected]> Cr-Commit-Position: refs/branch-heads/4147@{#11} Cr-Branched-From: 2b7d969-refs/heads/master@{#31262}
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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 <[email protected]> > Reviewed-by: Ilya Nikolaevskiy <[email protected]> > Cr-Commit-Position: refs/heads/master@{#32325} [email protected],[email protected] # 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 <[email protected]> Reviewed-by: Ilya Nikolaevskiy <[email protected]> Commit-Queue: Guido Urdaneta <[email protected]> Cr-Commit-Position: refs/heads/master@{#32369}
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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 <[email protected]> > > Reviewed-by: Mirko Bonadei <[email protected]> > > Reviewed-by: Karl Wiberg <[email protected]> > > 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 <[email protected]> > Reviewed-by: Karl Wiberg <[email protected]> > Cr-Commit-Position: refs/heads/master@{#32359} [email protected],[email protected],[email protected] 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 <[email protected]> Commit-Queue: Guido Urdaneta <[email protected]> Cr-Commit-Position: refs/heads/master@{#32372}
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TEST: tested on chromium. (cherry picked from commit 49dbad0) No-Try: True Bug: webrtc:12397, chromium:1171206 Change-Id: I1e15605f90e253a6ef61ab7ead8c576a80e8f01b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203886 Reviewed-by: Jakob Ivarsson <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Reviewed-by: Sam Zackrisson <[email protected]> Commit-Queue: Minyue Li <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#33080} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204302 Reviewed-by: Mirko Bonadei <[email protected]> Reviewed-by: Minyue Li <[email protected]> Cr-Commit-Position: refs/branch-heads/4389@{#4} Cr-Branched-From: 7acc2d9-refs/heads/master@{#32986}
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When pacing is enabled for the low latency rendering path, frames are sent to the decoder in regular intervals. In case of a jitter, these frames intervals could add up to create a large latency. Hence, disable frame pacing if the pre-decode queue grows beyond the threshold. The threshold for when to disable frame pacing is set through a field trial. The default value is high enough so that the behavior is not changed unless the field trial is specified. (cherry picked from commit 2ddc39e) Bug: chromium:1237402 Change-Id: I901fd579f68da286eca3d654118f60d3c55e21ce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228241 Reviewed-by: Ilya Nikolaevskiy <[email protected]> Commit-Queue: Johannes Kron <[email protected]> Cr-Original-Commit-Position: refs/heads/master@{#34705} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228540 Cr-Commit-Position: refs/branch-heads/4577@{#4} Cr-Branched-From: 5196931-refs/heads/master@{#34463}
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(cherry picked from commit f4a6928) Bug: chromium:1330019 Change-Id: I1a22967dff3231c1522fb94de38b309f441d468e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265442 Reviewed-by: Frank Barchard <[email protected]> Reviewed-by: Alexander Cooper <[email protected]> Commit-Queue: Joe Downing <[email protected]> Cr-Original-Commit-Position: refs/heads/main@{#37158} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265482 Commit-Queue: Joe Downing <[email protected]> Cr-Commit-Position: refs/branch-heads/5060@{#4} Cr-Branched-From: 1389c76-refs/heads/main@{#36868}
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This CL forwards BlockingCall client locations to BlockingCallImpl. Fixed: chromium:1416199 Change-Id: Ic5865280ec5c72a09a64419940b8343e8a54986b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295624 Commit-Queue: Markus Handell <[email protected]> Reviewed-by: Harald Alvestrand <[email protected]> Cr-Commit-Position: refs/heads/main@{#39442}
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