Revert "RtpSender: Create the Sframe encryptor and hand out a worker proxy"

This reverts commit 8354b7c19d32a3dbe7268319af6085e6ec9ec099.

Reason for revert: breaks webrtc-internal bot. Compile error in sframe

Failure Link: https://fusion2.corp.google.com/invocations/e9a2a344-3234-4d69-a6e9-b95f8700b396/targets/%2F%2Fthird_party%2Fwebrtc:webrtc_all_cc/log?q=label:session_id%3Dbtbs-224212c0-8627-4ddc-9137-60e85c0d9b6c

Original change's description:
> RtpSender: Create the Sframe encryptor and hand out a worker proxy
>
> Implements `RtpSenderBase::CreateSframeEncryptorOrError`
>
> When called, it creates a `SframeEncryptor` on the worker thread and —
> if an SSRC is already assigned — immediately pushes it to the media
> channel via `SetSframeEncryptor`. If the SSRC is not yet known,
> `SetSsrc` is updated to push the encryptor to the channel once an SSRC
> is assigned.
>
> The returned `SframeEncryptorInterface` is wrapped in a new
> `SframeEncryptorProxy` that marshals `SetEncryptionKey` calls to the
> worker thread, so callers can safely invoke it from any other thread.
>
> Bug: webrtc:479862368
> Change-Id: Ic647c4658fa672c820d09a13c34ee0b158f3ae89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/493780
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#48542}

Bug: webrtc:479862368
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I7e24587bf67f8f4a5f85a29c672fbccc22a566b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/501520
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48544}
13 files changed
tree: 4a2933e7ee5f37a78b4102c284e857928661a6ae
  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