Move Sframe mode from encryptor to SframeEncryptionConfig

Sframe mode determines how the send pipeline applies encryption, not how
the encryptor performs cryptographic operations. Introduce
SframeEncryptionConfig to carry the mode and encryptor together through
the send pipeline.

Keep mode in SframeEncryptorInit as the application-facing input, but
construct SframeEncryptor using only the cipher suite. Remove its stored
mode and the mode() accessor from the media encryptor interface.

Store an optional SframeEncryptionConfig in RtpSenderBase and
SframeSendOptions. Rename the media-channel setter to
SetSframeEncryptionConfig and propagate the configuration during
installation, SSRC binding, and video send-stream creation or
recreation. RtpVideoSender forwards it to each RTPSenderVideo, including
simulcast streams.

Keep the requirement for Sframe protection independent of the encryption
configuration. Rename the RTP-level flag to
sframe_required_by_negotiation to clarify that it reflects applied SDP.
Preserve the existing frame-dropping behavior while Sframe encryption is
not yet wired into the RTP sender; audio propagation remains a TODO.

Bug: webrtc:479862368
Change-Id: I7e0ffbf1843dd0231354460a25ee052f26552ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507741
Commit-Queue: Kacper Wasniowski <kwasniow@cisco.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48819}
25 files changed
tree: 4d0e15d4f6b8f23c3a2df6bed991524f51a33def
  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