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webrtc / src.git / 15845af3cdb27b218b1d3bf7b13e0e1bbb210a37
commit15845af3cdb27b218b1d3bf7b13e0e1bbb210a37[log] [tgz]
authorHarald Alvestrand <hta@webrtc.org>Tue Mar 05 08:50:57 2019
committerCommit Bot <commit-bot@chromium.org>Tue Mar 05 09:21:37 2019
treedd46acb0627f3f01881dcd51e18881fca2b71172
parent9b9344742b186b14d87e827e71a1757f4c94b30e [diff]
Reland "Another mock for GetSctpTransport" (and add test)

This reverts commit 727504cf493f9e03952a6e88348976385a49b9e2.

Reason for revert: Added required INCLUDE to fix compile errors.

Original change's description:
> Revert "Another mock for GetSctpTransport"
>
> This reverts commit b2c4700d39fbedaff9bdbee934e1f3f8032bb35b.
>
> Reason for revert: Breaks Chrome build
>
> Original change's description:
> > Another mock for GetSctpTransport
> >
> > Bug: chromium:818643
> > Change-Id: I4ae7826efa7afa8e7b2ecd8a5928071a1b913ded
> > Reviewed-on: https://webrtc-review.googlesource.com/c/125340
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26941}
>
> TBR=kwiberg@webrtc.org,hta@webrtc.org
>
> Change-Id: I98ddc61ca1e76d69b84138419d91ad9e40b04b1d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:818643
> Reviewed-on: https://webrtc-review.googlesource.com/c/125380
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26943}

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

Change-Id: I3eb410427f6660cd00319b43e7096bd634290e8a
Bug: chromium:818643
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125381
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26964}
2 files changed
tree: dd46acb0627f3f01881dcd51e18881fca2b71172
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. logging/
  10. media/
  11. modules/
  12. p2p/
  13. pc/
  14. resources/
  15. rtc_base/
  16. rtc_tools/
  17. sdk/
  18. stats/
  19. style-guide/
  20. system_wrappers/
  21. test/
  22. tools_webrtc/
  23. video/
  24. .clang-format
  25. .git-blame-ignore-revs
  26. .gitignore
  27. .gn
  28. .vpython
  29. abseil-in-webrtc.md
  30. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.h
  35. DEPS
  36. ENG_REVIEW_OWNERS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. 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