Reland "Add support for setting CSRCs on audio and video senders"

This reverts commit 84f48e824a68e7dc72b7ed229726341e12271157.

Reason for revert: revert breaks ToT

Bug: b/410811496
Original change's description:
> Revert "Add support for setting CSRCs on audio and video senders"
>
> This reverts commit dd3768ef7266e0e4840e883a0f652e3c75887cad.
>
> Reason for revert: breaks downstream projects
>
> Bug: b/410811496
> Original change's description:
> > Add support for setting CSRCs on audio and video senders
> >
> > With this change, CSRCs can be added to video packets sent via
> > RTPSenderVideo::SendEncodedImage. This is implemented by keeping a list
> > of CSRCs in the calling class RtpVideoSender, which is included in all
> > calls to SendEncodedImage.
> >
> > This CL is part of a chain, with the next being
> > https://webrtc-review.googlesource.com/c/src/+/392961. Ultimately, the
> > point is to support setting the CSRC list via RtpEncodingParameters.
> > This is done in https://webrtc-review.googlesource.com/c/src/+/392980.
> >
> > Bug: b/410811496
> > Change-Id: I2b9c430c6b19b423f2f29cf8e81b04ad04c2b915
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392940
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> > Commit-Queue: Helmer Nylén <helmern@google.com>
> > Cr-Commit-Position: refs/heads/main@{#44824}
>
> Bug: b/410811496
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Change-Id: I0c7730b89468740e2692dd52eb72fcd67cf0f040
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395160
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#44833}

Bug: b/410811496
Change-Id: I9fab185d8ab6e9dfe7583f409f7d6dd14fc4e429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395002
Owners-Override: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#44835}
19 files changed
tree: 67cdafda79387e06b44da3173ef98c60441b33f7
  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