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gerrit-public.fairphone.software / platform / external / webrtc / 189849fa0feca530781aa20f95de93539bbe12f0
commit189849fa0feca530781aa20f95de93539bbe12f0[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Fri Feb 07 16:12:16 2020 +0100
committerCommit Bot <commit-bot@chromium.org>Fri Feb 07 15:14:38 2020 +0000
tree9be81b0905b139dfd46d7a815ff24025eb373b8d
parent8e8b36a94a7a7a1fd0f8093979a406afa56e18c1 [diff]
[Stats] Remove jitterBufferDelay TODO; it's already implemented.

This TODO says this metric is only available for audio and should also
be implemented for video, but ever since M76 this has been implemented
for both audio and video (https://crbug.com/webrtc/10450).

TBR=guido@webrtc.org, hta@webrtc.org
NOTRY=True

Bug: webrtc:10450
Change-Id: Icf2b60fdacae606c66f9d03492f107df9e32ba33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168343
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30485}
1 file changed
tree: 9be81b0905b139dfd46d7a815ff24025eb373b8d
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. WATCHLISTS
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. 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 http://www.webrtc.org/native-code/development 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