Add RMS computation for the RTP level indicator.

- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
  AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.

TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test

Review URL: http://webrtc-codereview.appspot.com/279003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/audio_processing/audio_processing_impl.h

@@ -81,6 +81,9 @@ class AudioProcessingImpl : public AudioProcessing {
  private:
   int WriteMessageToDebugFile();
   int WriteInitMessage();
+  bool stream_data_changed() const;
+  bool synthesis_needed(bool stream_data_changed) const;
+  bool analysis_needed(bool stream_data_changed) const;
 
   int id_;