Sframe API support for sender and receiver

This CL introduces the public API surface for Sframe (Secure Frame)
encryption and decryption on RTP senders and receivers. The actual
pipeline wiring is deferred to a follow-up CL; all factory methods
currently return nullptr.

New API headers (api/sframe/):

- sframe_types.h: shared enums SframeMode (per-frame / per-packet)
  and SframeCipherSuite (AES-CTR-HMAC variants, AES-GCM-128/256).
- sframe_encrypter_interface.h: SframeEncrypterInit config struct
  and SframeEncrypterInterface for sender-side key management
  (SetEncryptionKey).
- sframe_decrypter_interface.h: SframeDecrypterInit config struct
  and SframeDecrypterInterface for receiver-side key management
  (AddDecryptionKey, RemoveDecryptionKey).

Interface changes:

- RtpSenderInterface gains CreateSframeEncrypter(SframeEncrypterInit)
- RtpReceiverInterface gains CreateSframeDecrypter(SframeDecrypterInit)
- Both proxy files updated with corresponding PROXY_METHOD entries.

New RtpReceiverBase class introduced inheriting RtpReceiverInternal,
providing CreateSframeDecrypter. This mirrors the existing RtpSenderBase
pattern. AudioRtpReceiver and VideoRtpReceiver now inherit
RtpReceiverBase instead of RtpReceiverInternal, eliminating future code
duplication.

Bug: webrtc:479862368
Change-Id: I2d62b61f2d10325d8dca9f15f08a0eb0509eed01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/452260
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47224}
17 files changed
tree: 34fa2823385fd5e48ea4b862a69d61037861db06
  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