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webrtc / src / 5a25ab298d2305044f2b3c33976a2144446e0d42
commit5a25ab298d2305044f2b3c33976a2144446e0d42[log] [tgz]
authorEdward Lemur <ehmaldonado@webrtc.org>Tue Jan 02 18:26:40 2018
committerCommit Bot <commit-bot@chromium.org>Tue Jan 02 18:26:52 2018
tree3c6c1b838544402637b8b0c82f3584e4d61a24cf
parent49acb1e8ca9f1fd3f604f3ad9cb2b301ac7d2078 [diff]
Revert "Add mock_rtc_event_log.h."

This reverts commit 63aea46a6e424c35f7df1ec6e702c5eeb704996a.

Reason for revert: 
Breaks Chromium build:
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-simulator%2F6721%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files__mb_%2F0%2Fstdout

ERROR at //third_party/webrtc/logging/BUILD.gn:295:5: Can't load input file.
    "../../test:test_support",
    ^------------------------
Unable to load:
  /b/build/slave/mac64/build/src/third_party/test/BUILD.gn
I also checked in the secondary tree for:
  /b/build/slave/mac64/build/src/build/secondary/third_party/test/BUILD.gn

Maybe this should be guarded by an "if (!build_with_chromium)"?

Original change's description:
> Add mock_rtc_event_log.h.
> 
> Bug: webrtc:7642
> Change-Id: Id3aa84d79e5e1a0520a968117cee550c9dd33c16
> Reviewed-on: https://webrtc-review.googlesource.com/37040
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21475}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org

Change-Id: Ib49c7812261c46226ec0b7b3c99af2c3a8b78add
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7642
Reviewed-on: https://webrtc-review.googlesource.com/36981
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21477}
7 files changed
tree: 3c6c1b838544402637b8b0c82f3584e4d61a24cf
  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. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. voice_engine/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  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. typedefs.h
  52. WATCHLISTS
  53. webrtc.gni
  54. 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