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webrtc / src / webrtc / 97352846c516ffc407c6faa785f7b37f71c1fc3f
commit97352846c516ffc407c6faa785f7b37f71c1fc3f[log] [tgz]
authorpbos@webrtc.org <pbos@webrtc.org>Fri Nov 14 11:52:04 2014
committerpbos@webrtc.org <pbos@webrtc.org>Fri Nov 14 11:52:04 2014
tree66af05b3918cf27576983341e84462e9dbc62691
parent799bae2c27c4ddd6f76e708f0dd1d563a08b72df [diff]
Report total bitrate for all streams in GetStats.

This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
22 files changed
tree: 66af05b3918cf27576983341e84462e9dbc62691
  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. common.gyp
  21. common.h
  22. common_types.h
  23. config.cc
  24. config.h
  25. engine_configurations.h
  26. experiments.h
  27. frame_callback.h
  28. LICENSE
  29. LICENSE_THIRD_PARTY
  30. OWNERS
  31. PATENTS
  32. PRESUBMIT.py
  33. README.chromium
  34. rtc_unittests.isolate
  35. supplement.gypi
  36. transport.h
  37. typedefs.h
  38. video_decoder.h
  39. video_encoder.h
  40. video_engine_tests.isolate
  41. video_frame.h
  42. video_receive_stream.h
  43. video_renderer.h
  44. video_send_stream.h
  45. webrtc.gyp
  46. webrtc_examples.gyp
  47. webrtc_perf_tests.isolate
  48. webrtc_tests.gypi