[Adaptation] Move EffectiveDegradationPreference to RA-Processor.

This CL is part of the Call-Level Adaptation Processing design doc:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing

The VideoStreamAdapter is responsible for generating adaptation
suggestions according to its DegradationPreference. Today there is one
DegradationPreference that you set, but internally the preference it
uses to make decisions is EffectiveDegradationPreference() which
reinterprets “balanced” as “maintain-resolution” if screenshare is used.

By moving the “effective” logic to the ResourceAdaptationProcessor, the
VideoStreamAdapter will not need to know about the type of track, and
the responsibility of the adapter is minimized. The “effective” logic
is non-standard and something we want to get rid of - until then, it
should be the responsibility of the processor to configure the adapter
to use the appropriate strategy, rather than for the adapter to know
about more states of the system than it needs to.

Future CLs will further minimize what the adapter needs to know, moving
"decision-making" logic to the Processor and "is adapt up allowed?"
logic to the Resources.

By removing the VideoInputMode enum the VideoStreamAdapter does not
know if we have input which has to be checked externally. Input
handling is followed-up on in the next CL.

Bug: webrtc:11172
Change-Id: I37ec9e7392f835cf8fef9829a2c945183f0e9b65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172927
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#31096}
5 files changed
tree: 62bc85b094c4da296b5e77f70f01fe3df8299340
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  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. webrtc_lib_link_test.cc
  53. 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