)]}'
{
  "commit": "755b04a06ec4ae91ae7fb601c641e683f4e9e87d",
  "tree": "adcb1d0cd3b5524b55eba21f60b9fd8ba2b2a432",
  "parents": [
    "6a85b17a0a8b0c19790b0c01ebafc32eb08b555c"
  ],
  "author": {
    "name": "andrew@webrtc.org",
    "email": "andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d",
    "time": "Tue Nov 15 16:57:56 2011"
  },
  "committer": {
    "name": "andrew@webrtc.org",
    "email": "andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d",
    "time": "Tue Nov 15 16:57:56 2011"
  },
  "message": "Add RMS computation for the RTP level indicator.\n\n- Compute RMS over a packet\u0027s worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.\n- We now use the entire packet rather than the last 10 ms frame.\n- Restore functionality to LevelEstimator.\n- Fix a bug in the splitting filter.\n- Fix a number of bugs in process_test related to a poorly named\n  AudioFrame member.\n- Update the unittest protobuf and float reference output.\n- Add audioproc unittests.\n- Reenable voe_extended_tests, and add a real function test.\n- Use correct minimum level of 127.\n\nTEST\u003daudioproc_unittest, audioproc, voe_extended_test, voe_auto_test\n\nReview URL: http://webrtc-codereview.appspot.com/279003\n\ngit-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d\n",
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