Propagate Sframe options from media channels to the Video RTP send path

WebRtcVideoSendChannel::EnableSframe() and
WebRtcVideoReceiveChannel::EnableSframe() previously only flipped a
local flag with a TODO; nothing below the media channel could observe
it. This CL plumbs the Sframe options from the media channels down to
the video RTP senders.

Changes:

- VideoSendStream::Config gains an SframeSendOptions field and
  VideoReceiveStreamInterface::Config gains an SframeReceiveOptions field.
  Both are populated from the media channel when a stream is created, and
  the receive config copies the field in Config::Copy().
- RtpTransportControllerSendInterface::CreateRtpVideoSender() takes the send
  options and forwards them through RtpVideoSender into
  RTPSenderVideo::Config::sframe_required.
- RTPSenderVideo::SendEncodedImage() fails closed: when Sframe is required
  the frame is dropped instead of being sent in the clear. The encryptor is
  not wired up yet, so today this drops every frame on an Sframe-enabled
  stream.
- WebRtcVideoSendChannel::EnableSframe() pushes the new options to already
  created send streams via WebRtcVideoSendStream::UpdateSframeOptions(),
  which recreates the underlying VideoSendStream. Receive streams still only
  pick the options up at construction; pushing to live receive streams needs
  a setter on VideoReceiveStreamInterface and is left as a TODO.

The audio path is intentionally untouched and will be handled
separately.

Tests: new unit tests cover that Sframe is off by default, that enabling
it before AddSendStream()/AddRecvStream() reaches the stream config, and
that RTPSenderVideo drops encoded images when Sframe is required but no
encryptor is present.

Bug: webrtc:479862368
Change-Id: I40d8298f7a4e4054ef3306dc05d13867a6bf3772
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/506321
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Kacper Wasniowski <kwasniow@cisco.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48782}
10 files changed
tree: d824472db6c2502f4cf14e78dda72d9661d43525
  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