[Adaptation] Make Manager's Resources not depend on encoder queue.

This CL makes the VideoStreamEncoderResourceManager's inner Resources
(PreventAdaptUpDueToActiveCounts,
PreventIncreaseResolutionDueToBitrateResource and
PreventAdaptUpInBalancedResource) not directly depend on any of the
manager's states that will continue to live on the encoder task queue
when the adaptation task queue is introduced in the next CL.

PreventAdaptUpDueToActiveCounts depends on effective degradation
preference, which it can get from the Processor, and the active counts,
which will move to the adaptation queue and is safe to use.

PreventIncreaseResolutionDueToBitrateResource depends on encoder
settings and target bitrate. This Resource now listens to these states
being updated, which may be implemented with a PostTask when the
adaptation queue is added.

PreventAdaptUpInBalancedResource depends on the effective degradation
preference, which it can get from the Processor; balanced settings,
which is a const readonly struct (thread-safe); and encoder target
bitrate, which it listens for being updated (to be PostTask'ed).

All resources depends on GetReasonFromResource() which will be callable
from the adaptation queue.

Bug: webrtc:11542, webrtc:11520
Change-Id: Ifa7bd87d9d8729988073f78f6a37c6f3b8aa4db1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174807
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@{#31220}
2 files changed
tree: 056095e2c062339a4cc318723bf584ed89d16835
  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