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FAILED: obj/third_party/webrtc/rtc_base/logging/logging.o #9

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Billshuai opened this issue Apr 3, 2019 · 0 comments
Open

FAILED: obj/third_party/webrtc/rtc_base/logging/logging.o #9

Billshuai opened this issue Apr 3, 2019 · 0 comments

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@Billshuai
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Billshuai commented Apr 3, 2019 •
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when " build libwebrtc for OWT Android SDK with scripts/build_android.py " error,
无法生存文件obj/third_party/webrtc/rtc_base/logging/logging.o

building libjingle_peerconnection_so for arm release ninja: Entering directory/home/hou/chromium/src/out/releasearm'
[6/929] CXX obj/third_party/webrtc/rtc_base/logging/logging.o
FAILED: obj/third_party/webrtc/rtc_base/logging/logging.o 错误原因是：../../third_party/webrtc/rtc_base/logging.cc:53:36: error: use of undeclared identifier 'LS_NONE'
static LoggingSeverity g_min_sev = LS_NONE;
^
../../third_party/webrtc/rtc_base/logging.cc:54:36: error: use of undeclared identifier 'LS_NONE'
static LoggingSeverity g_dbg_sev = LS_NONE;
^
../../third_party/webrtc/rtc_base/logging.cc:82:6: error: use of undeclared identifier 'LogSink'
void LogSink::OnLogMessage(const std::string& msg,

......
`

fatal error: too many errors emitted, stopping now [-ferror-limit=] 20 errors generated. [11/929] CXX obj/third_party/webrtc/pc/rtc_pc_base/srtpsession.o ninja: build stopped: subcommand failed.

taste1981 pushed a commit that referenced this issue 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 issue 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 issue Apr 15, 2021
Add hysteresis to enabling of simulcast streams.
c80264f 
If the bandwidth is just on the edge of being able to enable a new
stream, the keyframe generated when it is enabled might be large enough
to trigger an overuse and force the stream off again.

To avoid toggling, this CL adds hysteresis so that the available
bandwidth needs to be above X% to start bitrate in order to enable the
stream. It will be shut down once available bitrate falls below the
original enabling bitrate.

For screen content, X defaults to 35.
For realtime content, X defaults to 0.

Both can be individually modified via field trials.

(cherry picked from commit 3064f31)

Bug: webrtc:9734, chromium:884630
Change-Id: I941332d7be7f2a801d13d9202b2076d330e7df32
Reviewed-on: https://webrtc-review.googlesource.com/100308
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#24745}
Reviewed-on: https://webrtc-review.googlesource.com/100761
Cr-Commit-Position: refs/branch-heads/70@{open-webrtc-toolkit#9}
Cr-Branched-From: f18b352-refs/heads/master@{#24472}
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