Remove deprecated Port setters and make Port state const

Now that downstream callers have migrated to the rvalue
RelayPortFactoryInterface::Create overloads and constructor-time
Port::Callbacks:

- Remove legacy const-ref RelayPortFactoryInterface::Create overloads
  and make Port::Callbacks a move-only aggregate.
- Pass component, generation, and send_retransmit_count_attribute via
  Port::PortParametersRef and CreateRelayPortArgs instead of post-
  construction setters.
- Remove Port::set_component, Port::set_generation,
  Port::set_send_retransmit_count_attribute,
  SubscribeCandidateReadyCallback, SubscribeCandidateError,
  SubscribePortComplete, SubscribePortError, and
  TurnPort::SetTurnLoggingId.
- Replace Port's single-subscriber CallbackLists with absl::AnyInvocable
  members and make immutable members const across Port, UDPPort,
  TurnPort, TCPPort, BasicPortAllocator, BasicPortAllocatorSession,
  AllocationSequence, and FakePortAllocatorSession.

Bug: webrtc:42224914, webrtc:42222117
Change-Id: I1949215efeedaa4261a2cf6f66f20639d24924c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/509620
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48976}
17 files changed
tree: 1fc0f4e39191fd38af5cb22f1e464aa52b6b1dba
  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. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.mdformat
  39. .style.yapf
  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. vpython.toml
  64. vpython.toml.uv.lock
  65. WATCHLISTS
  66. webrtc.gni
  67. webrtc_lib_link_test.cc
  68. 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