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webrtc / src / / 94d8ccec4cc3b0a0285662346382ec4628cd127a
commit94d8ccec4cc3b0a0285662346382ec4628cd127a[log] [tgz]
authorTaylor Brandstetter <deadbeef@webrtc.org>Fri Jan 26 22:44:21 2018
committerCommit Bot <commit-bot@chromium.org>Fri Jan 26 22:44:30 2018
tree552d1bc589a89799bb2c22e06c77cb3c21531aca
parent4954a77cf81e6793245f52d485834acd3e6eab1c [diff]
Revert "Parameterize PeerConnection signaling tests for Unified Plan"

This reverts commit 65c0a60302202189c37af91fca6abf092f022b1c.

Reason for revert: Breaking downstream test which was calling CreateAnswer in stable state. Will reland after fixing test.

Original change's description:
> Parameterize PeerConnection signaling tests for Unified Plan
> 
> This also changes the behavior of CreateAnswer to fail unless
> the signaling state is kHaveRemoteOffer or kHaveLocalPranswer,
> as per the WebRTC specification.
> 
> Bug: webrtc:8765
> Change-Id: I60ac67cd92b17fcbff964afc14d049481e816a28
> Reviewed-on: https://webrtc-review.googlesource.com/41042
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21779}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,pthatcher@webrtc.org

Change-Id: I90eacadb217353a7e098826563f5aeaaced52452
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8765
Reviewed-on: https://webrtc-review.googlesource.com/44581
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21781}
3 files changed
tree: 552d1bc589a89799bb2c22e06c77cb3c21531aca
  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. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.cc
  35. common_types.h
  36. DEPS
  37. LICENSE
  38. license_template.txt
  39. LICENSE_THIRD_PARTY
  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. typedefs.h
  51. WATCHLISTS
  52. webrtc.gni
  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