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webrtc / src.git / 4c07605f8d55a8ea96937e3f775226200692e923
commit4c07605f8d55a8ea96937e3f775226200692e923[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Wed Mar 18 09:09:59 2020
committerCommit Bot <commit-bot@chromium.org>Wed Mar 18 11:04:32 2020
tree09f9207eeeb4330b08d071e60d9327d5031946b6
parente5c9f6d106896e870141c3e391e09cbfb593d49f [diff]
[Adaptation] VideoStreamAdapter unit tests added.

This code was previously only exercised by
video_stream_adapter_unittest.cc and other tests, acting more like
integration tests than unit tests. Now that the VideoStreamAdapter is
in a good state, more extensive test coverage is added.

Testing includes:
- Default restrictions.
- Adapting up or down in "maintain-framerate", "maintain-resolution"
  and "balanced", including...
- expecting how frame rate and/or resolution is affected,
- reaching kLimitReached,
- and reaching unrestricted.
- That "disabled" does not adapt.
- When adaptation is not possible, including...
- kInsufficientInput
- kAwaitingPreviousAdaptation
- kIsBitrateConstrained
- PeekNextRestrictions()
- "balanced" + "screenshare" = "maintain-resolution"
- Change degradation preference to/from "balanced" clears restrictions.
- That using invalidated adaptations triggers DCHECKs.

Bug: webrtc:11393
Change-Id: I28e2cf227bc1fd8871ee0d18d9570d4063449160
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170625
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@{#30816}
4 files changed
tree: 09f9207eeeb4330b08d071e60d9327d5031946b6
  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