Reland "APM: log both applied and recommended input volume stats"

This is a reland of commit 8d7273357d92fab881561d886ce8dfe94e6e2238

Root cause:
audioproc_f doesn't call `metrics::Enable()` and therefore the stats
reporter crashed when `metrics::HistogramFactoryGetCountsLinear()`
returned a nullptr.

Bug fix:
Added `InputVolumeStatsReporter::cannot_log_stats_`, a const flag
that is set to true if any histogram factory returns a nullptr.
When true, the class does nothing.

This CL also includes other code readability improvements that were
not part of the original CL.

Original change's description:
> APM: log both applied and recommended input volume stats
>
> This CL replaces the existing `WebRTC.Audio.ApmAnalogGain.*` stats
> with `WebRTC.Audio.Apm.AppliedInputVolume.*` and adds the
> `WebRTC.Audio.Apm.RecommendedInputVolume.*` stats.
>
> Bug: webrtc:7494
> Change-Id: I70be710d20b1589fc814cbce3d3329ac1500686f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280220
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38468}

Bug: webrtc:7494
Change-Id: I8373d16beb06b84f439d2c2274ededea7c5e95b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280661
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38484}
6 files changed
tree: 19fc68164fcaafe89be508f8609029585e0cce16
  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. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  34. .vpython
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. g3doc.lua
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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