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webrtc / src / 58e64bbf3b3f154292c7910d7f5dc3efd9187d5f
commit58e64bbf3b3f154292c7910d7f5dc3efd9187d5f[log] [tgz]
authorMirko Bonadei <mbonadei@webrtc.org>Thu Jul 02 07:47:20 2020
committerCommit Bot <commit-bot@chromium.org>Thu Jul 02 09:10:37 2020
treec6ac6532c08085a6699c4ab3a0e6ee2c14ef6de8
parentf2a4ec19d10dfd8f52173aa105cb05152bce5d5c [diff]
Revert "peerconnection: prefer spec names for signaling state"

This reverts commit f79bfc65e52a35d27cf0db2d212e94043fb44da3.

Reason for revert: Potentially affects Chromium tests, see
failures on https://chromium-review.googlesource.com/c/chromium/src/+/2276338.

Original change's description:
> peerconnection: prefer spec names for signaling state
> 
> Map the internal state names to the spec ones defined in
>   https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum
> instead of exposing them. This only affects the (not specified)
> error strings.
> 
> Bug: None
> Change-Id: Ib0b35bb3106b1688e8386f6fdd0b8c7fdebaf1dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178390
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#31591}

TBR=hbos@webrtc.org,philipp.hancke@googlemail.com

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

Bug: None
Change-Id: I6df20c93f6944b819eb11f22ba30c6221de61d79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31610}
1 file changed
tree: c6ac6532c08085a6699c4ab3a0e6ee2c14ef6de8
  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 here 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