Wire Sframe enable callback through sender and receiver constructors.

This CL implements the C++ plumbing for the SFrame enable signal
described in the WebRTC Encoded Transform spec
(https://w3c.github.io/webrtc-encoded-transform/#attribute, Section
2.2).

Per the spec, when an SFrame transform is set on an RTCRtpSender or
RTCRtpReceiver, the setter must propagate [[useSFrame]] = true to the
owning RTCRtpTransceiver and trigger SDP renegotiation. If the
transceiver's [[useSFrame]] has already been negotiated to false, it
must throw InvalidModificationError.

This is modeled by wiring an on_sframe_enable_requested callback from
each sender/receiver to RtpTransceiver::TryToEnableSframe(). When the
application calls CreateSframeEncrypterOrError() (sender) or
CreateSframeDecrypterOrError() (receiver), the callback fires, which
sets sframe_enabled_ = true on the transceiver and triggers
on_negotiation_needed_(). After negotiation completes,
ApplySframeEnabled() locks the state — matching the spec's behavior
where [[useSFrame]] becomes immutable once associated with a media
description.

Bug: webrtc:479862368
Change-Id: Ie5edd365aa4896bfda44cf34a2ab52dfe32bb441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459460
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47536}
16 files changed
tree: 2ea5a5fa0073581bc490033f18616524895e411d
  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. sdk/
  25. stats/
  26. system_wrappers/
  27. test/
  28. tools_webrtc/
  29. video/
  30. .clang-format
  31. .clang-tidy
  32. .git-blame-ignore-revs
  33. .gitignore
  34. .gn
  35. .mailmap
  36. .rustfmt.toml
  37. .style.mdformat
  38. .style.yapf
  39. .vpython3
  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. WATCHLISTS
  64. webrtc.gni
  65. webrtc_lib_link_test.cc
  66. 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