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webrtc / src.git / dda828f707b1505f75a62a66c7bdb59f13e183fd
commitdda828f707b1505f75a62a66c7bdb59f13e183fd[log] [tgz]
authorJonas Oreland <jonaso@webrtc.org>Fri Jan 28 13:49:34 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Mon Jan 31 11:51:31 2022
tree35d21ba82315064d6c325e682869adacf9ce0b7f
parent9a99905301ffa8a08a717ddaea3e4879440c97f9 [diff]
Field trials for DSCP

Add a field trial "WebRTC-DscpFieldTrial"
that allows user to set any int value to be
used as tagging. This tag value will be used
for all packets on the PeerConnection,
whether they are audio, video, data or ICE

e.g WebRTC-DscpFieldTrial/override_dscp:40/

see https://webrtc.googlesource.com/src/+/b477fc73cfd2f4c09bb9c416b170ba4b566cecaf/rtc_base/dscp.h
for names of popular ints.

Bug: webrtc:13622
Change-Id: Iedbedd0f918100259678eb5bc083c9bf89b343b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249786
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35848}
2 files changed
tree: 35d21ba82315064d6c325e682869adacf9ce0b7f
  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. .style.yapf
  31. .vpython
  32. .vpython3
  33. AUTHORS
  34. BUILD.gn
  35. CODE_OF_CONDUCT.md
  36. codereview.settings
  37. DEPS
  38. DIR_METADATA
  39. ENG_REVIEW_OWNERS
  40. g3doc.lua
  41. LICENSE
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  45. PATENTS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. WATCHLISTS
  53. webrtc.gni
  54. webrtc_lib_link_test.cc
  55. 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