Reformat existing code. There should be no functional effects.

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.
  */
 
@@ skipping 19 matching lines @@
   ~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 lenght_corr, int subframe_index);
+  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;
@@ skipping 23 matching lines @@
   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_