Reland "Sframe encryptor implementation"

This reverts commit 3590bc3fc94418ecddf2a2cf392e19f8a1a17a05.

Reason for revert: Dependency added downstream

Original change's description:
> Revert "Sframe encryptor implementation"
>
> This reverts commit 4820cf11c56af592ea4ed02cbce7c0b1a7656fbe.
>
> Reason for revert: Breaks downstream projects due to new third party dependency.
>
> Original change's description:
> > Sframe encryptor implementation
> >
> > This CL introduces the concrete Sframe encryptor that sits between the
> > public key-management API and the media pipeline, implementing the
> > sender-side wiring for Sframe end-to-end encryption.
> >
> > # High-level design
> > The implementation follows a three-layer split that matches the existing
> > WebRTC layering and respects the api/ visibility rules: 1. Public API
> > layer (api/sframe/) - A narrow, stable interface that exposes only key
> > management to applications: setting an encryption key on the sender
> > side.
> >
> > 2. Internal media interface layer (modules/sframe/) - Extends the public
> > key-management interface with the Encrypt and "max ciphertext size"
> > entry points used by the media pipeline. This is the surface the media
> > channel speaks to; application code never sees it.
> >
> > 3. Concrete impl layer (modules/sframe/) - Implements the media
> > interface on top of the cisco/sframe library.
> >
> > Splitting key management from frame protection lets the public handle
> > expose only the small surface applications need, while the media
> > pipeline talks to a richer interface internally.
> >
> > Bug: webrtc:479862368
> > Change-Id: Id93dba5874b6f5b1f32723755dd404af202f6ecf
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/481301
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#48065}
>
> Bug: webrtc:479862368
> Change-Id: Icef4fe1ce6e391b045f7b94087ee419b7899a96c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/485200
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#48077}

Bug: webrtc:479862368
Change-Id: Ie9dae7c1e52f459b5b5b762937febadab9a33588
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/488000
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#48170}
7 files changed
tree: a92fa5c33fc6989aa680bb27281320d793d1b67a
  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. .vpython3
  41. .yapfignore
  42. AUTHORS
  43. BUILD.gn
  44. CODE_OF_CONDUCT.md
  45. codereview.settings
  46. DEPS
  47. DIR_METADATA
  48. ENG_REVIEW_OWNERS
  49. GEMINI.md
  50. LICENSE
  51. license_template.txt
  52. native-api.md
  53. OWNERS
  54. OWNERS_INFRA
  55. PATENTS
  56. PRESUBMIT.py
  57. presubmit_test.py
  58. presubmit_test_mocks.py
  59. pylintrc
  60. pylintrc_old_style
  61. README.chromium
  62. README.md
  63. unsafe_buffers_paths.txt
  64. WATCHLISTS
  65. webrtc.gni
  66. webrtc_lib_link_test.cc
  67. 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