Reland "Improve structuring of test for audio glitches."

This reverts commit 3ae09f541900f18a8b680e70372f7f1d5e06bd0a.

Reason for revert: Revert didn't actually stabilize test. Decreasing # of rounds instead.

Original change's description:
> Revert "Improve structuring of test for audio glitches."
>
> This reverts commit fdbaeda00362a385de85b4c08aa0b536062a8415.
>
> Reason for revert: Breaks downstream project, see https://bugs.chromium.org/p/webrtc/issues/detail?id=12371
>
> Original change's description:
> > Improve structuring of test for audio glitches.
> >
> > Bug: webrtc:12361
> > Change-Id: Ieddc3dafbb638b3bd73dd79bcafa499290fa4340
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201723
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32973}
>
> TBR=hbos@webrtc.org,hta@webrtc.org
>
> Change-Id: Ie337de79a80113958607a7508d136c05fe6d9167
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12361
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202024
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32993}

TBR=aleloi@webrtc.org,hbos@webrtc.org,hta@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:12361
Change-Id: Ice79f2144d76bd7576cb415538afdd210625cc4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202247
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33019}
1 file changed
tree: 43cf7a3138782ebdce87e9b476ff05aed82c2a05
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. DEPS
  36. ENG_REVIEW_OWNERS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. webrtc_lib_link_test.cc
  52. 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