Consolidate RTP header extension validation in BaseChannel

Refactor WebRtcVideoEngine and WebRtcVoiceEngine to eliminate redundant
RTP header extension mapping and validation. This change shifts the
responsibility for verifying extension IDs and URI consistency from the
media engine level to BaseChannel.

Key changes:
* Remove ValidateRtpExtensions and its manual overlapping checks from
  WebRtcMediaEngine, as these are now centralized.
* Eliminate recv_rtp_extension_map_ and IdentifyExtensions calls from
  video and voice receive channels.
* Update media channels to filter extensions dynamically when
  requesting RTP parameters instead of maintaining a local filtered
  cache.
* Add CheckRtpExtensionValidity to BaseChannel to verify ID ranges,
  duplicates, and URI reassignment.
* Add historical_rtp_header_extensions_ to BaseChannel to track
  previously seen mappings and prevent illegal ID-to-URI reassignments
  during session renegotiation.

Bug: webrtc:42220456
Change-Id: I165e7f9af21b73d6e2ba66ee9367d140f00826eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/453661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47086}
11 files changed
tree: e5e12ce1416f9b52f23c91951020b122675ec5ce
  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