Add AES-256-CTR-HMAC-SHA512 Sframe cipher suites

Expose the three AES-256-CTR-HMAC-SHA512 cipher suites from
draft-barnes-sframe-iana-256 (registry values 6, 7 and 8), already
implemented by third_party/sframe, through SframeCipherSuite and map
them in ToSframeCipherSuite. This brings the WebRTC enum in line with
the eight suites listed in the WebRTC Encoded Transform
SFrameCipherSuite IDL, which gained these suites in
https://github.com/w3c/webrtc-encoded-transform/pull/303.

Bug: webrtc:479862368
Change-Id: Ie5f0bbcccfeee9ea7dd69b253c75c8c47082fbc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507780
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Kacper Wasniowski <kwasniow@cisco.com>
Cr-Commit-Position: refs/heads/main@{#48822}
3 files changed
tree: 00f425f943eac5bd32b0144524c5bab468ad5d3e
  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