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webrtc / src / 7976b77345ee91b44c06850ddee17cd71a681493
commit7976b77345ee91b44c06850ddee17cd71a681493[log] [tgz]
authorMirko Bonadei <mbonadei@webrtc.org>Wed Dec 18 15:41:08 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Dec 18 17:16:09 2024
tree5990d8906a4599203907fe95564637debfb571df
parent29af9f0c87394dd6288922900e2d27c5d5bb719e [diff]
Revert "Follow codec preference order for sending codec fallback."

This reverts commit 1ad3e14e9981772554a848c5034c7c555680aef7.

Reason for revert: Breaks downstream project. We are investigating into a potential problem when running on mobile platforms. We will get back with info or reland.

Original change's description:
> Follow codec preference order for sending codec fallback.
>
> When encoder selector is not enabled, currently we always fallback to
> VP8 no matter how the codec preference is setup. Update to follow codec
> preference order for the fallback.
>
> Bug: chromium:378566918
> Change-Id: Ia3fbfc9d407683ef7b3d6246af7e9ec58535dc89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370707
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43566}

Bug: chromium:378566918
Change-Id: I09086b5ad100a8f66e87df167e903d0b5fe5b589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372080
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43600}
6 files changed
tree: 5990d8906a4599203907fe95564637debfb571df
  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