P2PTransportChannel::OnCandidateResolved: fix resolver leak.

This change fixes a problem where the eventual destruction of a
completed resolver sometimes doesn't happen. This is because the
destruction is posted to the network thread, and if it's destroyed
before the closure is executed, the resolver is leaked.

The fix is in three parts:
1. The resolver->Destroy call is performed on closure destruction
   to make sure it will always run.
2. The closure is executed with task queue. This because the
   RTC_DCHECK on thread:140 fires with the invoker_.
3. It's not possible to guarantee the context Destroy is called on due
   to TaskQueue semantics. Therefore SignalThread::Destroy was changed
   to accept any calling context and only requiring it's the last
   public call to the object.

For unknown reasons, this leak doesn't trigger the leak checker, see
referred bugs for further investigation.

Bug: webrtc:7723, webrtc:11605, chromium:905542
Change-Id: I2681ff1d2416ccbc564974a65ac84781a9ed7aee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176125
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31359}
5 files changed
tree: 984a4339f72105031b62b3323c50cf03688bd637
  1. .clang-format
  2. .git-blame-ignore-revs
  3. .gitignore
  4. .gn
  5. .vpython
  6. AUTHORS
  7. BUILD.gn
  8. CODE_OF_CONDUCT.md
  9. DEPS
  10. ENG_REVIEW_OWNERS
  11. LICENSE
  12. OWNERS
  13. PATENTS
  14. PRESUBMIT.py
  15. README.chromium
  16. README.md
  17. WATCHLISTS
  18. abseil-in-webrtc.md
  19. api/
  20. audio/
  21. build_overrides/
  22. call/
  23. codereview.settings
  24. common_audio/
  25. common_types.h
  26. common_video/
  27. data/
  28. docs/
  29. examples/
  30. license_template.txt
  31. logging/
  32. media/
  33. modules/
  34. native-api.md
  35. p2p/
  36. pc/
  37. presubmit_test.py
  38. presubmit_test_mocks.py
  39. pylintrc
  40. resources/
  41. rtc_base/
  42. rtc_tools/
  43. sdk/
  44. stats/
  45. style-guide.md
  46. style-guide/
  47. system_wrappers/
  48. test/
  49. tools_webrtc/
  50. video/
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. 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