Reland "Migrate WebRTC documentation to new renderer"

This reverts commit 0f2ce5cc1c779f9bf33f51f29bfffbcbe105d1b1.

Reason for revert: Downstream infrastructure should be ready now

Original change's description:
> Revert "Migrate WebRTC documentation to new renderer"
>
> This reverts commit 3eceaf46695518f25bef43f155f82ed174827197.
>
> Reason for revert:
>
> Original change's description:
> > Migrate WebRTC documentation to new renderer
> >
> > Bug: b/258408932
> > Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39205}
>
> Bug: b/258408932
> Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Artem Titov <titovartem@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39209}

Bug: b/258408932
Change-Id: Ia172e4a6ad1cc7953b48eed08776e9d1e44eb074
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291660
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39231}
40 files changed
tree: feac96f8e9b494085d8f1e80e26f47b8a7faffab
  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. .style.yapf
  34. .vpython
  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. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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