Introduce ResourceAdaptationModuleListener and VideoSourceRestrictions.

The VideoSourceRestrictions describe the maximum pixels per frame and
max frame rate of a video source.

This CL makes the ResourceAdaptationModuleInterface responsible for the
reconfiguration of video sources. The VideoSourceRestrictions is the
output of an adaptation module, and the ResourceAdaptationModuleListener
handles the callback for when the source restrictions change.

The OveruseFrameDetectorResourceAdaptationModule is updated to output
its changes using these interfaces, and VideoStreamEncoder - now a
listener - is made responsible for triggering the reconfiguring the
video source.

Performing the reconfiguration still requires interacting with the
VideoSourceProxy - it is still partially responsible for keeping track
of rtc::VideoSinkWants settings and performing AddOrUpdateSink(). For
now this may look a bit weird: the VideoSourceProxy tells the
VideoStreamEncoder about the new restrictions, and then the
VideoStreamEncoder tells the VideoSourceProxy to apply these
restrictions on the source/sink. This exercises the listener though, and
unblocks the next CL.

The next CL should move all "configuring the source" logic to the
VideoStreamEncoder instead, so that the only information that is tracked
by OveruseFrameDetectorResourceAdaptationModule is what it actually
outputs to the listener. See the next CL
(https://webrtc-review.googlesource.com/c/src/+/162802) where a
VideoSourceController is introduced, to be owned by the
VideoStreamEncoder rather than the adaptation module.

Bug: webrtc:11222
Change-Id: I450ce74f51d96c4b98009a06134db671893d8fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162522
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30227}
6 files changed
tree: fe8c2494bfdafa4e1880317712ced88dff655b0b
  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 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