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webrtc / src.git / 3bb1194fff251bb63062aa1a807ee7107779f36b
commit3bb1194fff251bb63062aa1a807ee7107779f36b[log] [tgz]
authorPatrik Höglund <phoglund@webrtc.org>Fri Mar 16 12:36:26 2018
committerCommit Bot <commit-bot@chromium.org>Fri Mar 16 14:34:21 2018
treed996d4e71dc1a55c4e808119cbcc7fa01975582c
parent35468356d68f2ef792559b1d73797bf5a25888f6 [diff]
Revert "Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'"

This reverts commit d3070f43b19f503246be4ebad425d87568a71ce0.

Reason for revert: Need to re-enable h264 tests.

Original change's description:
> Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'
> 
> This doesn't change behavior at the moment because Chromium's
> 'proprietary_codecs' is already conditional on 'is_chrome_branded'
> but this guards WebRTC's default from upstream changes like
> https://chromium-review.googlesource.com/c/chromium/src/+/835010/6/build/config/features.gni
> 
> TBR=phoglund@webrtc.org
> 
> Bug: webrtc:8675
> Change-Id: Ic2ae311b5fc70a4d1ac1aefe4cc27574e4fcee40
> Reviewed-on: https://webrtc-review.googlesource.com/36321
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21452}

TBR=phoglund@webrtc.org,oprypin@webrtc.org,hbos@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8675
Change-Id: I97e5b25fb638e9d4731ac9610f9f6009a3789578
Reviewed-on: https://webrtc-review.googlesource.com/62380
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22476}
1 file changed
tree: d996d4e71dc1a55c4e808119cbcc7fa01975582c
  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. style-guide/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  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