Revert "Ensure AV1 is always available in PeerConnectionSimulcastTests."

This reverts commit 2d3b294e49027607c80766c50f1c3c8d7d4b38b9.

Reason for revert: The CL was believed to make AV1 always available
but it turned out that the import bots still failed due to not
having AV1, so it is better to use the built in factories than
to make custom test-only ones.

Original change's description:
> Ensure AV1 is always available in PeerConnectionSimulcastTests.
>
> Unblocks a WebRTC import where a bot without AV1 support would
> otherwise have been running and failing during setting codec
> preferences.
>
> # Non-chromium bots passed, no need to wait for chromium to land.
> # Want to unblock importer.
> NOTRY=True
>
> Bug: webrtc:15005
> Change-Id: I93c6a0ce5591a057c3a0ee49f6dbaef3676c0e1d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298021
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#39592}

Bug: webrtc:15005
Change-Id: I8f0850852edb0d0234000b2d956e2648a9adf904
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298120
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39596}
2 files changed
tree: d9e4cb51c35d4bf339861ea577f482a22ea51081
  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. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. OWNERS_INFRA
  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