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gerrit-public.fairphone.software / platform / external / webrtc / 0a5fdbb455b7c93c3585dcbf86bddf682522cade
commit0a5fdbb455b7c93c3585dcbf86bddf682522cade[log] [tgz]
authorZhi Huang <zhihuang@webrtc.org>Thu Jun 14 10:40:19 2018 -0700
committerCommit Bot <commit-bot@chromium.org>Thu Jun 14 18:35:11 2018 +0000
tree7b4143f97836b93ea93eeba4222ef43da29a6ff5
parent9eb38866cd65bd19a7ee89fdb99d3a1c2d7a1e4f [diff]
Use RTC_HISTOGRAM_ENUMERATION to report SRTP/SRTCP unprotect error.

Besides using the MetricsObserverInterface, using RTC_HISTOGRAM_ENUMERATION
directly using RTC_HISTOGRAM_ENUMERATION to report the error which is
needed by internal projects.

Bug: b/110121202, webrtc:9409
Change-Id: I1aaece91200905ea0495229dc2b5e62b1d61279b
Reviewed-on: https://webrtc-review.googlesource.com/83565
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23616}
2 files changed
tree: 7b4143f97836b93ea93eeba4222ef43da29a6ff5
  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. style-guide/
  22. system_wrappers/
  23. test/
  24. third_party/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. THIRD_PARTY_CHROMIUM_DEPS.json
  52. THIRD_PARTY_WEBRTC_DEPS.json
  53. typedefs.h
  54. WATCHLISTS
  55. webrtc.gni
  56. 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