Google Git
Sign in
webrtc / src / / ecde63eab5b643e9d84220b8a50208b4cd8ad866
commitecde63eab5b643e9d84220b8a50208b4cd8ad866[log] [tgz]
authorSergey Silkin <ssilkin@webrtc.org>Wed Mar 26 08:25:28 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Mar 26 11:31:51 2025
tree672b0616cf0c8522db8579ae5d3f776bed8aec0b
parent0802e02aa246865a95d2fcb13f614a5740bb1d4b [diff]
Do not crop to render resolution by default

Disable cropping to render resolution by default. This functionality was added in https://webrtc-review.googlesource.com/c/src/+/381340 and https://webrtc-review.googlesource.com/c/src/+/382241 and was enabled by default to make things work with HW encoders which use render resolution to signal cropping. After discussions with AV1 folks the consensus is that encoders should not use render resolution to signal crop. Currently there is no way to signal crop in AV1 bitstream. There is a proposal for adding this info to AV1 metadata: https://github.com/AOMediaCodec/av1-spec/pull/346.

Bug: webrtc:405341160, b:402019802, b:405299912
Change-Id: I9cff9f2f8e7c9b8933fd0fbb100ce37221eaf084
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/382640
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44228}
3 files changed
tree: 672b0616cf0c8522db8579ae5d3f776bed8aec0b
  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