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webrtc/src/webrtc/0b9d7ced44cffcae24d881e167c61fe38c5b4985
commit
0b9d7ced44cffcae24d881e167c61fe38c5b4985
[log]
author
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Wed Jan 08 08:59:44 2014
committer
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Wed Jan 08 08:59:44 2014
tree
3d4ca2fff9ef792905c01b02de3c7501724ed4df
parent
9b125e1b0d1bf2cc646ab5f42ba650f62250eb4d [diff]
Add another test case for AST/TOF switching.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5352 4adac7df-926f-26a2-2b94-8c16560cd09d
1 file changed
tree: 3d4ca2fff9ef792905c01b02de3c7501724ed4df
  1. build/
  2. common_audio/
  3. common_video/
  4. examples/
  5. modules/
  6. system_wrappers/
  7. test/
  8. tools/
  9. video/
  10. video_engine/
  11. voice_engine/
  12. .gitignore
  13. call.h
  14. common.h
  15. common_types.h
  16. config.h
  17. engine_configurations.h
  18. experiments.h
  19. frame_callback.h
  20. LICENSE
  21. LICENSE_THIRD_PARTY
  22. PATENTS
  23. PRESUBMIT.py
  24. README.chromium
  25. transport.h
  26. typedefs.h
  27. video_engine_tests.isolate
  28. video_receive_stream.h
  29. video_renderer.h
  30. video_send_stream.h
  31. webrtc.gyp
  32. webrtc_examples.gyp
  33. webrtc_perf_tests.isolate
  34. webrtc_tests.gypi