Reland "Add initial support for RtpEncodingParameters max_framerate."

This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad.

Reason for revert: downstream project fixed.

Original change's description:
> Revert "Add initial support for RtpEncodingParameters max_framerate."
>
> This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > Add initial support for RtpEncodingParameters max_framerate.
> >
> > Add support to set the framerate to the maximum of |max_framerate|.
> > Different framerates are currently not supported per stream for video.
> >
> > Bug: webrtc:9597
> > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> > Reviewed-on: https://webrtc-review.googlesource.com/92392
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24270}
>
> TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
>
> Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9597
> Reviewed-on: https://webrtc-review.googlesource.com/94060
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24277}

TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Bug: webrtc:9597
Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e
Reviewed-on: https://webrtc-review.googlesource.com/100080
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24720}
14 files changed
tree: 87558adcb47fd27f04692c85e688ad5c36016626
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. style-guide/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. abseil-in-webrtc.md
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. WATCHLISTS
  51. webrtc.gni
  52. 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 http://www.webrtc.org/native-code/development 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