Use asynchronous posting for PeerConnection signaling tasks

Update RunOnSignalingThread to use PostTask with thread safety flags
instead of BlockingCall. The places where RunOnSignalingThread is used,
bypass the proxy but used BlockingCall to remain functionally compatible
with the proxies. The methods are however asynchronous, so the
BlockingCall can be avoided. That was not done in the first step when
RunOnSignalingThread was introduced in order to not bring too many
changes (the main focus was on allowing non thread-hopping getters).
This change is a follow up step that transitions the initial step of the
task execution from a synchronous, blocking operation to an asynchronous
when called from a different thread.

As a btw change, SetLocalDescription and SetRemoteDescription now
consistently manage the lifetime of their arguments using
std::unique_ptr and scoped_refptr to ensure that
SessionDescriptionInterface and SetSessionDescriptionObserver objects
remain valid until the asynchronous task completes on the signaling
thread, preventing potential use-after-free or memory leak issues.

Bug: webrtc:442220720
Change-Id: I71cda2643c8ddd89622436df08bf889c0a35c56d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/454980
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47095}
3 files changed
tree: c9c0aea813f9c8960154e83030c9a7bde1689c9d
  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