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webrtc / src / d9f1208de7077d3c195544fbe54a205200c52e0a
commitd9f1208de7077d3c195544fbe54a205200c52e0a[log] [tgz]
authorAlessio Bazzica <alessiob@webrtc.org>Wed Jul 20 14:31:56 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Jul 20 15:44:13 2022
tree37b2367f1191a4ec38c2cd5e575c34ea19cd0c53
parentfb6cbd9201989991cacee2ce8b9aa1b7ba992556 [diff]
`AgcManagerDirect` parametrized unit tests

It is now easier to fully test `AgcManagerDirect` with different values
for the used field trials. In particular, this CL adds tests for the
field trial named `WebRTC-Audio-2ndAgcMinMicLevelExperiment`.

1. `UnmutingRaisesTooLowVolume` and `MicVolumeIsLimited`
The expectations for the lowest input volume are not hard-coded anymore
since the parametrized tests use different values for the enforced
minimum.

2. `RecoveryAfterManualLevelChangeBelowMin`
The recovery behavior after manual input volume change depends on
whether the minimum input volume is overridden. When that's the case,
the minimum volume is applied immediately after the manual adjustment.
Hence, the existing test is left and a parametrized version of it has been added to test the "instant recovery" behavior. The latter test is
skipped when the minimum input volume is not overridden since that case
is covered by the existing test.

Bug: chromium:1275566
Change-Id: Ib0d4427b32b88f33138d4062b365916a3c47a406
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37577}
3 files changed
tree: 37b2367f1191a4ec38c2cd5e575c34ea19cd0c53
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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