Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_coding/neteq/expand.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 46 matching lines @@
   virtual void SetParametersForMergeAfterExpand();
 
   // Returns the mute factor for |channel|.
   int16_t MuteFactor(size_t channel) {
     assert(channel < num_channels_);
     return channel_parameters_[channel].mute_factor;
   }
 
   // Accessors and mutators.
   virtual size_t overlap_length() const;
-  int16_t max_lag() const { return max_lag_; }
+  size_t max_lag() const { return max_lag_; }
 
  protected:
   static const int kMaxConsecutiveExpands = 200;
   void GenerateRandomVector(int16_t seed_increment,
                             size_t length,
                             int16_t* random_vector);
 
   void GenerateBackgroundNoise(int16_t* random_vector,
                                size_t channel,
                                int mute_slope,
@@ skipping 11 matching lines @@
   void AnalyzeSignal(int16_t* random_vector);
 
   RandomVector* const random_vector_;
   SyncBuffer* const sync_buffer_;
   bool first_expand_;
   const int fs_hz_;
   const size_t num_channels_;
   int consecutive_expands_;
 
  private:
-  static const int kUnvoicedLpcOrder = 6;
-  static const int kNumCorrelationCandidates = 3;
-  static const int kDistortionLength = 20;
-  static const int kLpcAnalysisLength = 160;
-  static const int kMaxSampleRate = 48000;
+  static const size_t kUnvoicedLpcOrder = 6;
+  static const size_t kNumCorrelationCandidates = 3;
+  static const size_t kDistortionLength = 20;
+  static const size_t kLpcAnalysisLength = 160;
+  static const size_t kMaxSampleRate = 48000;
   static const int kNumLags = 3;
 
   struct ChannelParameters {
     ChannelParameters();
     int16_t mute_factor;
     int16_t ar_filter[kUnvoicedLpcOrder + 1];
     int16_t ar_filter_state[kUnvoicedLpcOrder];
     int16_t ar_gain;
     int16_t ar_gain_scale;
     int16_t voice_mix_factor; /* Q14 */
@@ skipping 11 matching lines @@
   void Correlation(const int16_t* input,
                    size_t input_length,
                    int16_t* output,
                    int* output_scale) const;
 
   void UpdateLagIndex();
 
   BackgroundNoise* const background_noise_;
   StatisticsCalculator* const statistics_;
   const size_t overlap_length_;
-  int16_t max_lag_;
+  size_t max_lag_;
   size_t expand_lags_[kNumLags];
   int lag_index_direction_;
   int current_lag_index_;
   bool stop_muting_;
   size_t expand_duration_samples_;
   rtc::scoped_ptr<ChannelParameters[]> channel_parameters_;
 
   DISALLOW_COPY_AND_ASSIGN(Expand);
 };
 
 struct ExpandFactory {
   ExpandFactory() {}
   virtual ~ExpandFactory() {}
 
   virtual Expand* Create(BackgroundNoise* background_noise,
                          SyncBuffer* sync_buffer,
                          RandomVector* random_vector,
                          StatisticsCalculator* statistics,
                          int fs,
                          size_t num_channels) const;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_EXPAND_H_