Remove SetOptions and other unused methods from channel interfaces

Delete SetOptions from VoiceMediaSendChannelInterface,
VoiceMediaReceiveChannelInterface, and VideoMediaSendChannelInterface,
as they lack any external or polymorphic callers.

* Remove unused implementations and associated internal logic from
  WebRtcVideoSendChannel, WebRtcVideoSendStream, and
  WebRtcVoiceReceiveChannel.
* Retain WebRtcVoiceSendChannel::SetOptions as a private non-virtual
  helper function.
* Mark internal tracking fields (default_send_options_ and options_)
  as const and drop their RTC_GUARDED_BY thread annotations, as they
  no longer undergo mutations or require explicit synchronization.
* Remove an unused VideoOptions parameter from the constructor of
  WebRtcVideoReceiveChannel and its creation call site.
* Drop now-obsolete virtual overrides and mock definitions across
  fake and mock implementations.

Bug: none
Change-Id: I508393646b4968c808d0c163fc036be969b807e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/477040
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47886}
14 files changed
tree: 7525b03f38e5180e309dc82a472fff3c3f7ce503
  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.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