[Adaptation] Introduce VideoStreamInputState and its Provider.

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

The "input state" of a VideoStream, needed for adaptation and
decision-making, are: source resolution and frame rate, codec type and
min pixels per frame (based on encoder scaling settings). These values
are modified on the encoder queue of the VideoStreamEncoder.

But in order to unblock call-level adaptation processing, where
adaptation and decision making happens off the encoder queue, a snapshot
of the input states need to be available at point of processing:
introducing the VideoStreamInputState.

In this CL, the VideoStreamInputStateProvider is added to provide input
state snapshots across threads based on input from VideoStreamEncoder
and VideoStreamEncoderObserver.

The input state's HasInputFrameSizeAndFramesPerSecond() can now be
DCHECKed inside the VideoStreamAdapter in favor of having less
Adaptation::Status codes. Whether input is "sufficient" for adaptation
is now the responsibility of the Processor. (Goal: adapter is purely a
Adaptation generator and apply-er.)

Somewhat tangental, this CL also deletes VideoStreamEncoder-specific
methods from ResourceAdaptationProcessorInterface making them an
implementation detail of ResourceAdaptationProcessor. In a future CL,
the "processor" will be split up into a "processor" part and a "video
stream encoder resource manager" part - more on that later.

Bug: webrtc:11172
Change-Id: Id9b158f569db0140b75360aaf0f7e2e28fb924f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172928
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@{#31098}
13 files changed
tree: 3c16ca3eda3ee183fe556621a29a053efe3f72fb
  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