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webrtc / src / 61f74d91f823c9d600490ff8b3856ba2a520ec22
commit61f74d91f823c9d600490ff8b3856ba2a520ec22[log] [tgz]
authorHarald Alvestrand <hta@webrtc.org>Mon Mar 02 10:20:00 2020
committerCommit Bot <commit-bot@chromium.org>Mon Mar 02 11:35:53 2020
treeaa3f92c03d1070f863edb13a437b92c7ef6f6523
parentcb8c40138ca170f841bc45fa6771cdfc4b966e5f [diff]
Reland "Expose can_trickle_ice_candidates on PeerConnection"

This reverts commit cb8c40138ca170f841bc45fa6771cdfc4b966e5f.

Reason for revert: Added missing default.

Original change's description:
> Revert "Expose can_trickle_ice_candidates on PeerConnection"
>
> This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9.
>
> Reason for revert: Breaks downstream due to missing default
>
> Original change's description:
> > Expose can_trickle_ice_candidates on PeerConnection
> >
> > Bug: chromium:708484
> > Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30653}
>
> TBR=deadbeef@webrtc.org,hta@webrtc.org
>
> Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:708484
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30655}

TBR=deadbeef@webrtc.org,hta@webrtc.org

Change-Id: I608da7781f158b4b02dd226d4dcd5615c4935fa8
Bug: chromium:708484
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169541
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30656}
9 files changed
tree: aa3f92c03d1070f863edb13a437b92c7ef6f6523
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  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. webrtc_lib_link_test.cc
  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