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webrtc / src / webrtc / 834b475d54959d109c67ce37451d027883dcf604
commit834b475d54959d109c67ce37451d027883dcf604[log] [tgz]
authorpbos@webrtc.org <pbos@webrtc.org>Thu Dec 11 13:26:09 2014
committerpbos@webrtc.org <pbos@webrtc.org>Thu Dec 11 13:26:09 2014
tree26f521109e055233d0096111fee68b49b4df8401
parent28f5f3ba071f2a987dd43538aa729bef16c08571 [diff]
Wire up RTT statistics to webrtc::Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/32249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
5 files changed
tree: 26f521109e055233d0096111fee68b49b4df8401
  1. base/
  2. build/
  3. common_audio/
  4. common_video/
  5. examples/
  6. libjingle/
  7. modules/
  8. overrides/
  9. p2p/
  10. sound/
  11. system_wrappers/
  12. test/
  13. tools/
  14. video/
  15. video_engine/
  16. voice_engine/
  17. .gitignore
  18. BUILD.gn
  19. call.h
  20. codereview.settings
  21. common.gyp
  22. common.h
  23. common_types.h
  24. config.cc
  25. config.h
  26. engine_configurations.h
  27. experiments.h
  28. frame_callback.h
  29. LICENSE
  30. LICENSE_THIRD_PARTY
  31. OWNERS
  32. PATENTS
  33. PRESUBMIT.py
  34. README.chromium
  35. rtc_unittests.isolate
  36. supplement.gypi
  37. transport.h
  38. typedefs.h
  39. video_decoder.h
  40. video_encoder.h
  41. video_engine_tests.isolate
  42. video_frame.h
  43. video_receive_stream.h
  44. video_renderer.h
  45. video_send_stream.h
  46. webrtc.gyp
  47. webrtc_examples.gyp
  48. webrtc_perf_tests.isolate
  49. webrtc_tests.gypi