Revert "Pipe CSRCs down through the audio and video send streams"

This reverts commit bf33699a2d8eeff670bc1150f7e05eec4818c398.

Reason for revert: Part of CL chain that breaks downstream projects

Bug: b/410811496
Original change's description:
> Pipe CSRCs down through the audio and video send streams
>
> This CL is part of a chain. It exposes methods on the audio and video
> send streams to set the CSRCs on the underlying senders (support for
> this is added in https://webrtc-review.googlesource.com/c/src/+/392940).
> These methods are used in
> https://webrtc-review.googlesource.com/c/src/+/392980.
>
> Bug: b/410811496
> Change-Id: I5e2445c70152724a9837634112e148e71d180ef5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392961
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Helmer Nylén <helmern@google.com>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#44825}

Bug: b/410811496
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I640031b3f42f660968818ad1f04507458b57bbdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395180
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Helmer Nylén <helmern@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44837}
13 files changed
tree: f6f2ceafdf83762b739125ca5b823020e5b60f98
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .rustfmt.toml
  34. .style.yapf
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. OWNERS_INFRA
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  53. pylintrc_old_style
  54. README.chromium
  55. README.md
  56. WATCHLISTS
  57. webrtc.gni
  58. webrtc_lib_link_test.cc
  59. 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