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diff --git a/webrtc/modules/audio_processing/agc/agc_audio_proc.h b/webrtc/modules/audio_processing/agc/agc_audio_proc.h
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+/*
+ *  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_