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webrtc / src / / 64e8e64e80b590bd7e90bd1bbcbfd628df9cb1c5
commit64e8e64e80b590bd7e90bd1bbcbfd628df9cb1c5[log] [tgz]
authorTomas Lundqvist <tomasl@google.com>Thu Sep 19 13:53:17 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Fri Sep 20 10:38:19 2024
tree08c31ae28706c00b842e44051576e4d8f40a5ce0
parentf566dee902440e06d0e3f251d9cf8f40b5b3fc3d [diff]
Revert "Reland "Return audio stats regarless if we have a codec.""

This reverts commit 4334cdfc5c0619a5f06125ea1f039cb123ccf21e.

Reason for revert: Breaks downstream project.

Original change's description:
> Reland "Return audio stats regarless if we have a codec."
>
> This is a reland of commit 7fff587a096c6ef40f5601f47ef50b221b3a4abf
>
> Original change's description:
> > Return audio stats regarless if we have a codec.
> >
> > Bug: b/331602608
> > Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743
> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> > Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42964}
>
> Bug: b/331602608
> Change-Id: I95c89e7059005bc8dd8569ef41bfe9e863b4082f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361762
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42969}

Bug: b/331602608
Change-Id: Ifbe332a8749d024f603b75b6e787551dd6762dd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363001
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43059}
1 file changed
tree: 08c31ae28706c00b842e44051576e4d8f40a5ce0
  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