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/processing_component.h

@@ -18,16 +18,6 @@
 namespace webrtc {
 class AudioProcessingImpl;
 
-/*template <class T>
-class ComponentHandle {
-  public:
-    ComponentHandle();
-    virtual ~ComponentHandle();
-
-    virtual int Create() = 0;
-    virtual T* ptr() const = 0;
-};*/
-
 class ProcessingComponent {
  public:
   explicit ProcessingComponent(const AudioProcessingImpl* apm);
@@ -37,10 +27,11 @@ class ProcessingComponent {
   virtual int Destroy();
   virtual int get_version(char* version, int version_len_bytes) const = 0;
 
+  bool is_component_enabled() const;
+
  protected:
   virtual int Configure();
   int EnableComponent(bool enable);
-  bool is_component_enabled() const;
   void* handle(int index) const;
   int num_handles() const;