Merge to M77: Reland "Set the usage pattern bits for adding remote ICE candidates from SDP."

This is a reland of 7c6f74ab0344e9c6201de711d54026e9990b8e6c

Compared to the previous commit, new bits are added to log calls of
AddIceCandidate, and the gathering and reception of IPv6 candidates.

Original change's description:
> Set the usage pattern bits for adding remote ICE candidates from SDP.
>
> Currently these bits are only set when a remote ICE candidate is
> successfully added via addIceCandidate. For non-trickled sessions in
> which the remote candidates are added via the remote description, these
> bits are lost. This also happens for trickled sessions, though a rare
> case, when addIceCandidate does not succeed because the peer connection
> is not ready to add any remote candidate.
>
> Bug: webrtc:10868
> Change-Id: Ib2f199f9ffc936060473934d25ba397ef31131a3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148880
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28844}

TBR=hta@webrtc.org

(cherry picked from commit 1ba5dec7694562a095cd08a0008ab3d397345db1)

Bug: webrtc:10868
Change-Id: Ifac0593dcfb64d88619fd24b4ab61c14a0810beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149024
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#28904}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150026
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/branch-heads/m77@{#9}
Cr-Branched-From: 2bac7da1349c75e5cf89612ab9619a1920d5d974-refs/heads/master@{#28685}
3 files changed
tree: b2609c1bd8251dc310599649859fcff0ba9554a8
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  9. examples/
  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