Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/accelerate.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 27 matching lines @@
   // read from |input|, of length |input_length| elements, and are written to
   // |output|. The number of samples removed through time-stretching is
   // is provided in the output |length_change_samples|. The method returns
   // the outcome of the operation as an enumerator value. If |fast_accelerate|
   // is true, the algorithm will relax the requirements on finding strong
   // correlations, and may remove multiple pitch periods if possible.
   ReturnCodes Process(const int16_t* input,
                       size_t input_length,
                       bool fast_accelerate,
                       AudioMultiVector* output,
-                      int16_t* length_change_samples);
+                      size_t* length_change_samples);
 
  protected:
   // Sets the parameters |best_correlation| and |peak_index| to suitable
   // values when the signal contains no active speech.
   void SetParametersForPassiveSpeech(size_t len,
                                      int16_t* best_correlation,
-                                     int* peak_index) const override;
+                                     size_t* peak_index) const override;
 
   // Checks the criteria for performing the time-stretching operation and,
   // if possible, performs the time-stretching.
   ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
                                       size_t input_length,
                                       size_t peak_index,
                                       int16_t best_correlation,
                                       bool active_speech,
                                       bool fast_mode,
                                       AudioMultiVector* output) const override;
 
  private:
   DISALLOW_COPY_AND_ASSIGN(Accelerate);
 };
 
 struct AccelerateFactory {
   AccelerateFactory() {}
   virtual ~AccelerateFactory() {}
 
   virtual Accelerate* Create(int sample_rate_hz,
                              size_t num_channels,
                              const BackgroundNoise& background_noise) const;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_