Google Git
Sign in
webrtc / src / a3d4252a019b808bb7c55b93ecc19adc755ff2ab
commita3d4252a019b808bb7c55b93ecc19adc755ff2ab[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Thu Jan 16 12:55:29 2020
committerCommit Bot <commit-bot@chromium.org>Fri Jan 17 10:43:02 2020
tree452b185c8472b3888107af88c0a748dfdbc119d5
parent0695df1a5990ffd02cff4f7b49a865d7085f9d0b [diff]
ResourceAdaptationModule HasInputVideo+DegradationPreference setters.

The "has_input_video_ logic" is simplified to abort AdaptUp() and
AdaptDown() directly rather than in each calling place of the
VideoSourceRestrictor. The intent is no change in behavior.
The degradation_preference_ is removed from the VideoSourceRestrictor
as its only usage was DCHECKing (not worth it).

ResourceAdaptationModuleInterface gets SetHasInputVideo() and
SetDegradationPreference(), making these things controllable without
knowing implementation details.

StartCheckForOveruse() and StopCheckForOveruse() are renamed to
StartResourceAdaptation() and StopResourceAdaptation().

Bug: webrtc:11222
Change-Id: Id2d7f34d427dfb3ecd4831b1a245d07becae6520
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166173
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@{#30296}
5 files changed
tree: 452b185c8472b3888107af88c0a748dfdbc119d5
  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