Revert "Propagate known Encoder SinkWants when configured instead of after first frame."

This reverts commit 1ee24a650c116509d855e2ed23b8d28a0bb37384.

Reason for revert: Suspected upstream test breakage. 

Original change's description:
> Propagate known Encoder SinkWants when configured instead of after first frame.
>
> Propagate requested resolution and max frame rate to the source when
> configured rather than after the first frame.
> This is so that the source can be configured immediately. There is no
> reason why source should be updated until after first frame since it may lead
> to unnecessary reconfigurations and thread jumps. Wants that depend on actual frame size is not moved.
>
> Cl also change default behaviour in VideoStreamEncoderTest to not
> set restriction on max frame rate. This aligns with how its used.
>
> Bug: webrtc:14451
> Change-Id: I96a3675d3ccabb1d2ecb4354b6932bc6563b1760
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342801
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41906}

Bug: webrtc:14451
Change-Id: I3aa669f8cc61a43b0820a06edf1497f3c86e3958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343220
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41911}
3 files changed
tree: 3ec906c599b957ba8238473b96ae82f5328f09fc
  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. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. OWNERS_INFRA
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. pylintrc_old_style
  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