Enforce consistent network thread usage in Call::OnSentPacket

Remove the `sent_packet_sequence_checker_` checker in Call and use the
less ambiguous network_thread_ instead. Update OnSentPacket with a
strict requirement to run on the network thread. Previously, a detached
sequence checker allowed inconsistent threading behavior between
production and test environments, which obscured the actual threading
model and complicated maintenance of outgoing traffic logic.

Update tests to consistently follow this for better alignment with the
production code.

Move the responsibility of doing a thread hop for `DeliverRtcpPacket`
out of PeerConnection and into Call. This thread hop is a current
implementation detail for the Call class that we'd like to remove.

Key modifications:
* Update transport wrappers to ensure sent packet notifications are
  dispatched on the correct thread, hopping if necessary.
* Add internal thread-hopping to DeliverRtcpPacket to bridge from the
  network thread to the worker thread.
* Standardize the use of Thread in test configurations to
  simulate realistic network thread behavior.

Bug: webrtc:42222117
Change-Id: Ibcdfe0c3b5d0b20ff47b536929e8100fb70fd2f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/469080
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47596}
21 files changed
tree: 4eb0d0ba6379244df3b4a516cc23964893611042
  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.mdformat
  38. .style.yapf
  39. .vpython3
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. WATCHLISTS
  64. webrtc.gni
  65. webrtc_lib_link_test.cc
  66. 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