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_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/agc/common.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class AudioFrame;
-class PoleZeroFilter;
-
-class AgcAudioProc {
- public:
-  // Forward declare iSAC structs.
-  struct PitchAnalysisStruct;
-  struct PreFiltBankstr;
-
-  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 *
-      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_AGC_AGC_AUDIO_PROC_H_