Reland "RtpEncodingParameters::request_resolution patch 1"

This reverts commit b625101da8d798c936cfd695505a5514644158b0.

Reason for revert: Found problem that was specific how
configuration is handled for VP9. A 1-line change in webrtc_video_engine.cc line 3715.
Thanks Rasmus and great that this was tested!

Original change's description:
> Revert "RtpEncodingParameters::request_resolution patch 1"
>
> This reverts commit ef7359e679e579ccb79afacf5c42e8c6020124e2.
>
> Reason for revert: Breaks downstream test
>
> Original change's description:
> > RtpEncodingParameters::request_resolution patch 1
> >
> > This patch adds RtpEncodingParameters::request_resolution
> > with documentation and plumming. No behaviour is changed yet.
> >
> > Bug: webrtc:14451
> > Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38172}
>
> Bug: webrtc:14451
> Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38176}

Bug: webrtc:14451
Change-Id: Ica9b74180bce22d09bf289126bb5ac137bf9eb70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276543
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38178}
18 files changed
tree: 6bce174495e886836b1fd8a8f969ae1103bebdb6
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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