Change who gets the clone when creating SDPs in SdpOfferAnswerHandler

Previously, `CreateDescriptionObserverWrapperWithCreationCallback` and
associated callbacks, created a clone of the new sdp for internal
storage (last_created_offer_ and last_created_answer_) passed the
original session description to the wrapped observer.

This patch inverts the logic and gives the cloned object to the wrapped
observer instead. The reason for that is that in order to create a
clone, the state of the original object needs to be accessed and doing
so will (once thread checkers are in place) pin that state to the
calling thread. The calling thread is the signaling thread, which is
suitable for SdpOfferAnswerHandler but the wrapped observer should get
the pristine object which is less bound.

A side effect of this CL may be that we'll get less flake on tsan bots
since a test observer implementation will now get a newly constructed
object rather than one who's state may be accessed on different threads
in parallel. However, the local_description() and remote_description()
accessors still need to be fixed (wip).

Along the way use AnyInvocable&& instead of std::function.

Bug: webrtc:442220720
Change-Id: Idfef673b59e1f50bc2a6e82fb142d5f1aa528d8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/409280
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45646}
1 file changed
tree: ab3cbdcb821346d6522961218b71e5d3d4d6e850
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .clang-tidy
  30. .git-blame-ignore-revs
  31. .gitignore
  32. .gn
  33. .mailmap
  34. .rustfmt.toml
  35. .style.yapf
  36. .vpython3
  37. AUTHORS
  38. BUILD.gn
  39. CODE_OF_CONDUCT.md
  40. codereview.settings
  41. DEPS
  42. DIR_METADATA
  43. ENG_REVIEW_OWNERS
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. OWNERS_INFRA
  49. PATENTS
  50. PRESUBMIT.py
  51. presubmit_test.py
  52. presubmit_test_mocks.py
  53. pylintrc
  54. pylintrc_old_style
  55. README.chromium
  56. README.md
  57. WATCHLISTS
  58. webrtc.gni
  59. webrtc_lib_link_test.cc
  60. 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