Refactor AudioProcessingTest to match current audio processing code

The unittest suite AudioProcessingTest was written long ago at a time
when the audio processing code was very different from what it is now
and has only received minor updates during 10+ years.

It has since long reached a state where its core testing has become
irrelevant. E.g., its assumptions that sample-wise differences
should be low in the output for cases where downmixing and internal
resampling is used are no longer accurate enough for the test to pass.

To get the test to pass, a common workaround has over the years been
to reduce the accuracy in the comparison of the output against the
desired reference (i.e., reduce the tabularized threshold
expected_snr).
The reason for that workaround is that there is essentially no way to
tell what magnitude of difference is correct or incorrect.
That has resulted in that in several cases, that threshold has been
set to 0, effectively deactivating the test checks.

This CL refactors the test to remove the code that evaluates the
accuracy of the output (which is no longer relevant) and keep its
iteration over different channels counts and sample rates.
This reduces the test from doing soft-bitexactness to be a crash-test
that makes sure that the code is verified to run for relevant
configurations.


Bug: webrtc:42230301
Change-Id: I52a7753d61db627cbf6f10dbc1a1d2bfe87deddf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/446920
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#46841}
1 file changed
tree: 8fe664b895b31205019466aab379130573fc77d8
  1. agents/
  2. api/
  3. audio/
  4. build_overrides/
  5. call/
  6. common_audio/
  7. common_video/
  8. data/
  9. docs/
  10. examples/
  11. experiments/
  12. g3doc/
  13. infra/
  14. logging/
  15. media/
  16. modules/
  17. net/
  18. p2p/
  19. pc/
  20. resources/
  21. rtc_base/
  22. rtc_tools/
  23. sdk/
  24. stats/
  25. system_wrappers/
  26. test/
  27. tools_webrtc/
  28. video/
  29. .clang-format
  30. .clang-tidy
  31. .git-blame-ignore-revs
  32. .gitignore
  33. .gn
  34. .mailmap
  35. .rustfmt.toml
  36. .style.yapf
  37. .vpython3
  38. AUTHORS
  39. BUILD.gn
  40. CODE_OF_CONDUCT.md
  41. codereview.settings
  42. DEPS
  43. DIR_METADATA
  44. ENG_REVIEW_OWNERS
  45. GEMINI.md
  46. LICENSE
  47. license_template.txt
  48. native-api.md
  49. OWNERS
  50. OWNERS_INFRA
  51. PATENTS
  52. PRESUBMIT.py
  53. presubmit_test.py
  54. presubmit_test_mocks.py
  55. pylintrc
  56. pylintrc_old_style
  57. README.chromium
  58. README.md
  59. unsafe_buffers_paths.txt
  60. WATCHLISTS
  61. webrtc.gni
  62. webrtc_lib_link_test.cc
  63. 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