Use StrongAlias for RTP header extension identification.

This change introduces RtpHeaderExtensionId as a StrongAlias to
improve type safety for RTP header extension IDs. The type is used
throughout the RTP stack.

The code previous to this change has many different places where it
defines what a valid ID is (between 1 and 255) by checking it - having
a separate type allows us to centralize that check for validity.

IDs need to be allocated at an RTP transport level. Manufacturing your
own ID is a footgun - sooner or later the ID you manufacture will be
used by the allocator, leading to an unexpected error situation. Having
a separate type allows the reader to easily figure out where IDs are
being created, and verify that it's either in an unit test or in the
allocator.

Key changes:
- RtpHeaderExtensionId defined in api/rtp_header_extension_id.h
- Backwards compatible overloads (taking int) added to public APIs in
  api/ and modules/rtp_rtcp/ to ensure no breaking changes for
  downstream projects.
- RtpPacket, RtpHeaderExtensionPicker, and other internal components
  updated to use the new type.
- Updated SRTP components to handle RtpHeaderExtensionId.
- Fixed naming conflict in fake_rtp.h by renaming kRtcpReport to
  kFakeRtcpReport.
- Updated numerous tests to comply with the new explicit type.

Bug: webrtc:514817938
Change-Id: I79eb728f2ebd5c1fe5a1711eae2162c742fd029c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/473880
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47832}
89 files changed
tree: 2d1e9d989dc5091d0122c5f1f50ce2b3b7f57100
  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