IWYU modules/video_coding

using
  find modules/video_coding/ -name "*.h" -o -name "*.cc" | grep -v android | xargs tools_webrtc/iwyu/apply-include-cleaner
followed by
  tools_webrtc/gn_check_autofix.py -C out/Default/
and git cl format

H264 changes led to link failures so were reverted.

Manual changes required:
* added I420Buffer to test/testsupport/frame_reader.h
* modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc -- added VP8 constants
* modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc -- added uncompressed header parser
* modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc -- added vpx_encoder for VPX_DL_REALTIME
* modules/video_coding/video_receiver2_unittest.cc -- added video_coding_defines.h
* modules/video_coding/generic_decoder.cc -- same
* modules/video_coding/generic_decoder_unittest.cc -- same
* modules/video_coding/deprecated/jitter_buffer_unittest.cc -- added video_coding/encoded_frame.h
* media/engine/simulcast_encoder_adapter_unittest.cc -- added video_error_codes
* modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc -- same
* modules/video_coding/video_receiver2_unittest.cc -- same
* rtc_tools/video_replay.cc -- same
* test/video_codec_tester_unittest.cc -- same
* rtc_tools/video_encoder/video_encoder.cc -- same
* modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h -- moved around includes
* rtc_tools/DEPS: allowed include from video_coding:video_codec_interface
* modules/video_coding/codecs/av1/libaom_av1_encoder.cc -- expand full aom path

Additional changes resulting from moving VP9 defines and feedback.

BUG=webrtc:42226242

Change-Id: If7040e1cab93cf587f25ee8492604a7f7af9a573
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/381860
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44345}
213 files changed
tree: 2018eef6f466f37a4ff94b58b4f5f287cf6a5afa
  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. .rustfmt.toml
  34. .style.yapf
  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. pylintrc_old_style
  54. README.chromium
  55. README.md
  56. WATCHLISTS
  57. webrtc.gni
  58. webrtc_lib_link_test.cc
  59. 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