Fix potential deadlock in RTCStatsCollector during shutdown.

When stats gathering on the network thread is cancelled (e.g., during PC
closure), the `SafeTask` evaluating the partial results on the network
thread may be dropped without executing. This causes its closures
(including `RtpTransceiverStatsInfo` objects) to be destructed on the
network thread.

Because `RtpTransceiverStatsInfo` holds `scoped_refptr` references to
`RtpTransceiver` and `RtpReceiverInternal` (which may respectively hold
`VideoTrackProxyWithInternal`), their destruction on the network thread
can trigger a blocking call back to the signaling thread (via the
proxies), potentially leading to a deadlock if the signaling thread is
issuing a blocking call to the network thread.

This change prevents clears the object references (`transceiver` and
`receivers`) on the signaling thread before posting to the network
thread. When the task is dropped on the network thread, the subsequent
destruction of `RtpTransceiverStatsInfo` will be a no-op that avoids
cross-thread blocking.

Bug: b/483026914
Change-Id: Ifc5c317e6a26cc0c97c9ca67101ef8b0cc90ba1e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/448940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46897}
2 files changed
tree: 0a70614a31a67bb0736ffb61a5668a49501920cb
  1. agents/
  2. api/
  3. audio/
  4. build_overrides/
  5. call/
  6. common_audio/
  7. common_video/
  8. data/
  9. docs/
  10. examples/
  11. experiments/
  12. g3doc/
  13. infra/
  14. logging/
  15. media/
  16. modules/
  17. net/
  18. p2p/
  19. pc/
  20. resources/
  21. rtc_base/
  22. rtc_tools/
  23. sdk/
  24. stats/
  25. system_wrappers/
  26. test/
  27. tools_webrtc/
  28. video/
  29. .clang-format
  30. .clang-tidy
  31. .git-blame-ignore-revs
  32. .gitignore
  33. .gn
  34. .mailmap
  35. .rustfmt.toml
  36. .style.yapf
  37. .vpython3
  38. AUTHORS
  39. BUILD.gn
  40. CODE_OF_CONDUCT.md
  41. codereview.settings
  42. DEPS
  43. DIR_METADATA
  44. ENG_REVIEW_OWNERS
  45. GEMINI.md
  46. LICENSE
  47. license_template.txt
  48. native-api.md
  49. OWNERS
  50. OWNERS_INFRA
  51. PATENTS
  52. PRESUBMIT.py
  53. presubmit_test.py
  54. presubmit_test_mocks.py
  55. pylintrc
  56. pylintrc_old_style
  57. README.chromium
  58. README.md
  59. unsafe_buffers_paths.txt
  60. WATCHLISTS
  61. webrtc.gni
  62. webrtc_lib_link_test.cc
  63. 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