Revert "Pull the Voice Activity Detector out from the AGC"

webrtc/modules/audio_processing/agc/agc_audio_proc.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_processing/vad/common.h"
+#include "webrtc/modules/audio_processing/agc/common.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class AudioFrame;
 class PoleZeroFilter;
 
-class VadAudioProc {
+class AgcAudioProc {
  public:
   // Forward declare iSAC structs.
   struct PitchAnalysisStruct;
   struct PreFiltBankstr;
 
-  VadAudioProc();
-  ~VadAudioProc();
+  AgcAudioProc();
+  ~AgcAudioProc();
 
   int ExtractFeatures(const int16_t* audio_frame,
                       int length,
                       AudioFeatures* audio_features);
 
   static const int kDftSize = 512;
 
  private:
   void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, int length);
   void SubframeCorrelation(double* corr, int length_corr, int subframe_index);
   void GetLpcPolynomials(double* lpc, int length_lpc);
   void FindFirstSpectralPeaks(double* f_peak, int length_f_peak);
   void Rms(double* rms, int length_rms);
   void ResetBuffer();
 
   // To compute spectral peak we perform LPC analysis to get spectral envelope.
   // For every 30 ms we compute 3 spectral peak there for 3 LPC analysis.
   // LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame
   // we need 5 ms of past signal to create the input of LPC analysis.
   static const int kNumPastSignalSamples = kSampleRateHz / 200;
 
   // TODO(turajs): maybe defining this at a higher level (maybe enum) so that
   // all the code recognize it as "no-error."
   static const int kNoError = 0;
 
   static const int kNum10msSubframes = 3;
   static const int kNumSubframeSamples = kSampleRateHz / 100;
-  static const int kNumSamplesToProcess =
-      kNum10msSubframes *
+  static const int kNumSamplesToProcess = kNum10msSubframes *
       kNumSubframeSamples;  // Samples in 30 ms @ given sampling rate.
   static const int kBufferLength = kNumPastSignalSamples + kNumSamplesToProcess;
   static const int kIpLength = kDftSize >> 1;
   static const int kWLength = kDftSize >> 1;
 
   static const int kLpcOrder = 16;
 
   int ip_[kIpLength];
   float w_fft_[kWLength];
 
   // A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ).
   float audio_buffer_[kBufferLength];
   int num_buffer_samples_;
 
   double log_old_gain_;
   double old_lag_;
 
   rtc::scoped_ptr<PitchAnalysisStruct> pitch_analysis_handle_;
   rtc::scoped_ptr<PreFiltBankstr> pre_filter_handle_;
   rtc::scoped_ptr<PoleZeroFilter> high_pass_filter_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_