Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/merge.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 28 matching lines @@
         SyncBuffer* sync_buffer);
   virtual ~Merge() {}
 
   // The main method to produce the audio data. The decoded data is supplied in
   // |input|, having |input_length| samples in total for all channels
   // (interleaved). The result is written to |output|. The number of channels
   // allocated in |output| defines the number of channels that will be used when
   // de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
   // will be used to scale the audio, and is updated in the process. The array
   // must have |num_channels_| elements.
-  virtual int Process(int16_t* input, size_t input_length,
-                      int16_t* external_mute_factor_array,
-                      AudioMultiVector* output);
+  virtual size_t Process(int16_t* input, size_t input_length,
+                         int16_t* external_mute_factor_array,
+                         AudioMultiVector* output);
 
-  virtual int RequiredFutureSamples();
+  virtual size_t RequiredFutureSamples();
 
  protected:
   const int fs_hz_;
   const size_t num_channels_;
 
  private:
   static const int kMaxSampleRate = 48000;
-  static const int kExpandDownsampLength = 100;
-  static const int kInputDownsampLength = 40;
-  static const int kMaxCorrelationLength = 60;
+  static const size_t kExpandDownsampLength = 100;
+  static const size_t kInputDownsampLength = 40;
+  static const size_t kMaxCorrelationLength = 60;
 
   // Calls |expand_| to get more expansion data to merge with. The data is
   // written to |expanded_signal_|. Returns the length of the expanded data,
   // while |expand_period| will be the number of samples in one expansion period
   // (typically one pitch period). The value of |old_length| will be the number
   // of samples that were taken from the |sync_buffer_|.
-  int GetExpandedSignal(int* old_length, int* expand_period);
+  size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
 
   // Analyzes |input| and |expanded_signal| to find maximum values. Returns
   // a muting factor (Q14) to be used on the new data.
-  int16_t SignalScaling(const int16_t* input, int input_length,
+  int16_t SignalScaling(const int16_t* input, size_t input_length,
                         const int16_t* expanded_signal,
                         int16_t* expanded_max, int16_t* input_max) const;
 
   // Downsamples |input| (|input_length| samples) and |expanded_signal| to
   // 4 kHz sample rate. The downsampled signals are written to
   // |input_downsampled_| and |expanded_downsampled_|, respectively.
-  void Downsample(const int16_t* input, int input_length,
-                  const int16_t* expanded_signal, int expanded_length);
+  void Downsample(const int16_t* input, size_t input_length,
+                  const int16_t* expanded_signal, size_t expanded_length);
 
   // Calculates cross-correlation between |input_downsampled_| and
   // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
   // lag is returned.
-  int16_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
-                                 int start_position, int input_length,
-                                 int expand_period) const;
+  size_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
+                                size_t start_position, size_t input_length,
+                                size_t expand_period) const;
 
   const int fs_mult_;  // fs_hz_ / 8000.
-  const int timestamps_per_call_;
+  const size_t timestamps_per_call_;
   Expand* expand_;
   SyncBuffer* sync_buffer_;
   int16_t expanded_downsampled_[kExpandDownsampLength];
   int16_t input_downsampled_[kInputDownsampLength];
   AudioMultiVector expanded_;
 
   DISALLOW_COPY_AND_ASSIGN(Merge);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_