Google Git
Sign in
webrtc / src / 153c9e51074b3004ffeaa6fdb9749e4269b71ba8
commit153c9e51074b3004ffeaa6fdb9749e4269b71ba8[log] [tgz]
authorSergey Silkin <ssilkin@webrtc.org>Mon Feb 21 11:05:25 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Mon Feb 21 13:31:24 2022
tree0430131e5970431a1feefafe5f54bc9ab0e6d4bd
parentba2677061aa0872f742a8f4b7327e2878f461df6 [diff]
Revert "Use internal() in VideoTrack when invoking the source."

This reverts commit 962bf1896185c0d84232f2bfae492eeb04e1236d.

Reason for revert: Regressions in PC tests https://crbug.com/webrtc/13697

Original change's description:
> Use internal() in VideoTrack when invoking the source.
>
> This skips going through the proxy and potentially hide a thread hop
> should a regression occur.
>
> This CL contains a part of a previously reviewed, landed, reverted,
> relanded and re-reverted CL:
> https://webrtc-review.googlesource.com/c/src/+/250180
>
> Bug: webrtc:13540
> Change-Id: If098f5c04a263547fb53f44e9f9738b8e941a294
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251861
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36026}

Bug: webrtc:13540
Change-Id: Iea76094aedda91271154f89c356b140c95717976
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251981
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36035}
3 files changed
tree: 0430131e5970431a1feefafe5f54bc9ab0e6d4bd
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .mailmap
  31. .style.yapf
  32. .vpython
  33. .vpython3
  34. AUTHORS
  35. BUILD.gn
  36. CODE_OF_CONDUCT.md
  37. codereview.settings
  38. DEPS
  39. DIR_METADATA
  40. ENG_REVIEW_OWNERS
  41. g3doc.lua
  42. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. PATENTS
  47. PRESUBMIT.py
  48. presubmit_test.py
  49. presubmit_test_mocks.py
  50. pylintrc
  51. README.chromium
  52. README.md
  53. WATCHLISTS
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. 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