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webrtc / src / 9297a16140aca937c60501583c321821c99f9790
commit9297a16140aca937c60501583c321821c99f9790[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Fri Oct 10 12:03:24 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Fri Oct 10 13:41:32 2025
tree2a8a362b7091e92de7a3b811f24a9cf57b85983d
parentc17ba2ed3c9e42f5c1fbc92d85b4061d906cd527 [diff]
PixelLimitResource: Add parameter for adaptation reason.

The "reason" for adapting, which can be "cpu" or "quality" depending on
what the resource is reportedly monitoring, impacts which
RTCQualityAdaptationReason[1] we report in getStats() where these are
mapped to "cpu" and "bandwidth".

For testing purposes, it's useful to be able to control if
PixelLimitResource results in being CPU or BW limited. This CL adds the
new field trial enum parameter, "reason".

In addition to unit tests in this CL I have manually verified this gets
correctly plumbed to getStats() by going to
https://henbos.github.io/codec-quality/src/index.html using
--force-fieldtrials=WebRTC-PixelLimitResource/target_pixels:64800,interval:1s,reason:quality,toggle:10s/

[1] https://w3c.github.io/webrtc-stats/#rtcqualitylimitationreason-enum

Bug: webrtc:447037084
Change-Id: I575e1a984a6e31a997ba43f3565b709465fafea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/413160
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45896}
4 files changed
tree: 2a8a362b7091e92de7a3b811f24a9cf57b85983d
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .clang-tidy
  30. .git-blame-ignore-revs
  31. .gitignore
  32. .gn
  33. .mailmap
  34. .rustfmt.toml
  35. .style.yapf
  36. .vpython3
  37. AUTHORS
  38. BUILD.gn
  39. CODE_OF_CONDUCT.md
  40. codereview.settings
  41. DEPS
  42. DIR_METADATA
  43. ENG_REVIEW_OWNERS
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. OWNERS_INFRA
  49. PATENTS
  50. PRESUBMIT.py
  51. presubmit_test.py
  52. presubmit_test_mocks.py
  53. pylintrc
  54. pylintrc_old_style
  55. README.chromium
  56. README.md
  57. WATCHLISTS
  58. webrtc.gni
  59. webrtc_lib_link_test.cc
  60. 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