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webrtc / src / 8dfc90f94752a31db0bfde9674591996d98e0a90
commit8dfc90f94752a31db0bfde9674591996d98e0a90[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Fri Sep 02 07:39:29 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Fri Sep 02 10:52:49 2022
tree55ac70a945633d1198035384434569e3b3991f5d
parenta5d80a764693ea67724a8d0132377e1a9318d2ef [diff]
Make RTCStats IDs more concise.

Ultimately, IDs should be random according to spec[1], so we shouldn't
rely on the ID to convey easily readable information. By making the IDs
shorter we reduce the overhead of string copies and make report dumps a
little bit smaller.

Drive-by: Add "DEPRECATED_" prefic to the RTCMediaStreamStats ID.

[1] https://w3c.github.io/webrtc-pc/#dom-rtcstats-id

# Examples of IDs before and after this CL #

RTCDataChannel_3
-> D3

RTCPeerConnection
-> P

RTCTransport_0_1
-> T01

RTCCodec_RTCTransport_0_1_100_minptime=10;useinbandfec=1
-> CIT01_100_minptime=10;useinbandfec=1

RTCInboundRTPAudioStream_6666
-> IA6666

RTCAudioSource_1
-> SA1

RTCOutboundRTPAudioStream_2943129392
-> OA2943129392

RTCRemoteInboundRtpAudioStream_3541280085
-> RIA3541280085

RTCIceCandidate_6cWRqicY
-> I6cWRqicY

RTCIceCandidatePair_6cWRqicY_haEcM2xD
-> CP6cWRqicY_haEcM2xD

RTCCertificate_FD1:BC:58:90:DF:E8:40:58:8D:04:91:44:93:4E:6C:52:9E:F0:14:98:AA:67:7B:8B:C8:30:C8:31:D0:84:1B:BF
-> CFD1:BC:58:90:DF:E8:40:58:8D:04:91:44:93:4E:6C:52:9E:F0:14:98:AA:67:7B:8B:C8:30:C8:31:D0:84:1B:BF

DEPRECATED_RTCMediaStreamTrack_receiver_3
-> DEPRECATED_TI3

RTCMediaStream_45a6e766-5d1a-40f9-a55c-ea8fdefcde49
-> DEPRECATED_S45a6e766-5d1a-40f9-a55c-ea8fdefcde49

Bug: webrtc:14416, webrtc:14419
Change-Id: I11f0a8b8354203fea1df1093d8864a6d47ee71e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273709
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37992}
2 files changed
tree: 55ac70a945633d1198035384434569e3b3991f5d
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  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. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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