[M148] Fix TOCTOU race in RtpVideoSender::OnVideoLayersAllocationUpdated

RtpVideoSender::OnVideoLayersAllocationUpdated runs on the encoder thread,
checks if it's active, and then posts a task to the transport queue to
update module status.

If SetSending(false) is called on the transport queue before the posted
task runs, the task would previously re-register modules in PacketRouter
even though active_ was false. This lead to dangling pointers in
PacketRouter when the RtpVideoSender was destroyed.

This CL adds an IsActive() check inside the posted task to ensure we don't
re-register modules if the sender has been deactivated.

(cherry picked from commit 77d265670fa7d7f358c86c54704a6e97c27a6005)

Bug: chromium:500767595
Test: RtpVideoSenderTest.PostTaskRaceDoesNotLeadToDanglingPointer
Change-Id: I18827c34bece7f5ebc41d159d9e19e356afbc404
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/462880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#47384}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/465961
Cr-Commit-Position: refs/branch-heads/7778@{#5}
Cr-Branched-From: ca896b7ffef011bbf6957c99d413c5aac602c99f-refs/heads/main@{#47319}
3 files changed
tree: 6d02ff90f67dea371e73c724be9b2a615cec4594
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. sdk/
  25. stats/
  26. system_wrappers/
  27. test/
  28. tools_webrtc/
  29. video/
  30. .clang-format
  31. .clang-tidy
  32. .git-blame-ignore-revs
  33. .gitignore
  34. .gn
  35. .mailmap
  36. .rustfmt.toml
  37. .style.yapf
  38. .vpython3
  39. AUTHORS
  40. BUILD.gn
  41. CODE_OF_CONDUCT.md
  42. codereview.settings
  43. DEPS
  44. DIR_METADATA
  45. ENG_REVIEW_OWNERS
  46. GEMINI.md
  47. LICENSE
  48. license_template.txt
  49. native-api.md
  50. OWNERS
  51. OWNERS_INFRA
  52. PATENTS
  53. PRESUBMIT.py
  54. presubmit_test.py
  55. presubmit_test_mocks.py
  56. pylintrc
  57. pylintrc_old_style
  58. README.chromium
  59. README.md
  60. unsafe_buffers_paths.txt
  61. WATCHLISTS
  62. webrtc.gni
  63. webrtc_lib_link_test.cc
  64. 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