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webrtc / src / c7b964cd71fb3fc937d98f695b5960eb23580059
commitc7b964cd71fb3fc937d98f695b5960eb23580059[log] [tgz]
authorSteve Anton <steveanton@webrtc.org>Thu Feb 01 22:39:45 2018
committerCommit Bot <commit-bot@chromium.org>Fri Feb 02 00:56:44 2018
tree3f1f2e30b09871fd11e338032e75738725e61122
parent970b088878bd7e23c3f5fecb3010f96a36a9dd27 [diff]
Report cipher usage to UMA for all media types on a transport

Previously, the code which reported cipher stats to UMA for all
transports would classify the media type based on the transport name,
which is brittle and misleading with BUNDLE. This corrects the code to
track all media types (audio, video, data) which use the transport and
report once for each.

Bug: None
Change-Id: I8506f64f0011788b744b8386ac58518a21914b52
Reviewed-on: https://webrtc-review.googlesource.com/47247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21863}
2 files changed
tree: 3f1f2e30b09871fd11e338032e75738725e61122
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  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. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.cc
  35. common_types.h
  36. DEPS
  37. LICENSE
  38. license_template.txt
  39. LICENSE_THIRD_PARTY
  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. typedefs.h
  51. WATCHLISTS
  52. webrtc.gni
  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