Revert "Reland "Replace sigslot usages with robocaller library.""

This reverts commit c5f71087589b18bb4df1b78f2c452c4083edf2d9.

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 40261c3663fe316cfe40262c59cee993165ccf63
>
> 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}
11 files changed
tree: 39ab8cdd8ca39b21eba6cd275852c8a46d89ca8a
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. DEPS
  36. ENG_REVIEW_OWNERS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. webrtc_lib_link_test.cc
  52. whitespace.txt
README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info