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gerrit.openfyde.cn / webrtc.googlesource.com / src / 22a8f98dd55b2a36146f5d97775464dcc595ded0
commit22a8f98dd55b2a36146f5d97775464dcc595ded0[log] [tgz]
authorBenjamin Wright <benwright@webrtc.org>Sun Oct 28 18:39:00 2018 -0700
committerCommit Bot <commit-bot@chromium.org>Mon Oct 29 02:13:30 2018 +0000
treea340e0afe6f8f20c57ce46e7740858583e7368e9
parent428320c776cd9ce91fa92df17348aa11ce0ccf4d [diff]
Formatted sslidenty.cc and moved non referenced functions into an
anonymous namespace.

There is some really scary code in this function that I did not refactor in
this change. I believe the ASN parsing code should be removed completely
and have attached TODOs to do this once we have a correct test suite to validate
the functionality. I am almost certain openssl has functions that do this
better.

Bug: webrtc:9860
Change-Id: Ice06079eb1e5b10bdb2ee45ae45cbfb2ce8f6f13
Reviewed-on: https://webrtc-review.googlesource.com/c/108206
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25404}
1 file changed
tree: a340e0afe6f8f20c57ce46e7740858583e7368e9
  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. 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. 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