Reland "Switch to use 32 kHz processing inside APM"

This reverts commit ddb627e555615c39f877a280092a15062ad037c5.

Reason for revert: The downstream tests causing the breaking of the Chromium roll are now addressed. 

Original change's description:
> Revert "Switch to use 32 kHz processing inside APM"
>
> This reverts commit 179be29133c50285dcbb4e273a0b6a42d265886c.
>
> Reason for revert: Broke Chromium roll.
>
> Error: https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/2423056/overview
>
> Original change's description:
> > Switch to use 32 kHz processing inside APM
> >
> > This CL switches to using 32 kHz processing internally inside WebRTC APM
> > to avoid using the current 3-band split filter which has been shown to
> > have issues with aliasing impacting the speech quality.
> >
> > The intention is to revert this change once the issues in the 3-band
> > split filter have been addressed.
> >
> > Bug: webrtc:454695115
> > Change-Id: Id87e7f8d2ba37a915b3640f7eeb5c996037c59aa
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/419860
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#46026}
>
> Bug: webrtc:454695115
> Change-Id: I599adf846217f0cb81588e84f541277465ef856a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/420580
> Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46037}

Bug: webrtc:454695115
Change-Id: Ic0bb60b8c7103c7eac9563bae2d06935c152764e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/420463
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46046}
4 files changed
tree: 5d93d55795f8693932f3388e83e02b25b63c7213
  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