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webrtc / src / 369111bf30e5cb22c1ded18be493618c99680a6b
commit369111bf30e5cb22c1ded18be493618c99680a6b[log] [tgz]
authorMarkus Handell <handellm@webrtc.org>Sat Nov 11 09:08:05 2023
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Sat Nov 11 10:52:57 2023
tree4bccd50e16d7549b13f68589d6c87651b050cd8c
parent93ec8048b2817b64ebfb2aea1fa86b88a72c5827 [diff]
Revert "VideoStreamEncoder: exclude screencast from capture time measurement."

This reverts commit 03bc3a0fa67e274efb4518da005f4c5b77c607e9.

Reason for revert: measurement complete

Original change's description:
> VideoStreamEncoder: exclude screencast from capture time measurement.
>
> This CL avoids measurement for screencast encoding work. The reason is
> screencast can cling on to and re-encode old video frames for which
> webrtc::VideoFrame::reference_time() is unchanged.
>
> Bug: chromium:1498378
> Change-Id: I5bf79d29ef7f57ddff2622cbb6c3436480bd16ba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326103
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41097}

Bug: chromium:1498378
Change-Id: I42c1a86123eb1d6c7ad7c8981769f5560884a2f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327025
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41133}
1 file changed
tree: 4bccd50e16d7549b13f68589d6c87651b050cd8c
  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. .vpython
  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