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

webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h

 /*
  *  Copyright (c) 2014 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 42 matching lines @@
   // target signal es present and to false otherwise. This methods can be called
   // to know if the data is target signal or interference and process it
   // accordingly.
   bool is_target_present() override { return is_target_present_; }
 
  protected:
   // Process one frequency-domain block of audio. This is where the fun
   // happens. Implements LappedTransform::Callback.
   void ProcessAudioBlock(const complex<float>* const* input,
                          int num_input_channels,
-                         int num_freq_bins,
+                         size_t num_freq_bins,
                          int num_output_channels,
                          complex<float>* const* output) override;
 
  private:
   typedef Matrix<float> MatrixF;
   typedef ComplexMatrix<float> ComplexMatrixF;
   typedef complex<float> complex_f;
 
   void InitDelaySumMasks();
   void InitTargetCovMats();  // TODO(aluebs): Make this depend on target angle.
@@ skipping 19 matching lines @@
   void ApplyLowFrequencyCorrection();
 
   // Postfilter masks are also unreliable at high frequencies. Average mid-high
   // frequency masks to calculate a single mask per block which can be applied
   // in the time-domain. Further, we average these block-masks over a chunk,
   // resulting in one postfilter mask per audio chunk. This allows us to skip
   // both transforming and blocking the high-frequency signal.
   void ApplyHighFrequencyCorrection();
 
   // Compute the means needed for the above frequency correction.
-  float MaskRangeMean(int start_bin, int end_bin);
+  float MaskRangeMean(size_t start_bin, size_t end_bin);
 
   // Applies both sets of masks to |input| and store in |output|.
   void ApplyMasks(const complex_f* const* input, complex_f* const* output);
 
   void EstimateTargetPresence();
 
-  static const int kFftSize = 256;
-  static const int kNumFreqBins = kFftSize / 2 + 1;
+  static const size_t kFftSize = 256;
+  static const size_t kNumFreqBins = kFftSize / 2 + 1;
 
   // Deals with the fft transform and blocking.
-  int chunk_length_;
+  size_t chunk_length_;
   rtc::scoped_ptr<LappedTransform> lapped_transform_;
   float window_[kFftSize];
 
   // Parameters exposed to the user.
   const int num_input_channels_;
   int sample_rate_hz_;
 
   const std::vector<Point> array_geometry_;
 
   // Calculated based on user-input and constants in the .cc file.
-  int low_mean_start_bin_;
-  int low_mean_end_bin_;
-  int high_mean_start_bin_;
-  int high_mean_end_bin_;
+  size_t low_mean_start_bin_;
+  size_t low_mean_end_bin_;
+  size_t high_mean_start_bin_;
+  size_t high_mean_end_bin_;
 
   // Quickly varying mask updated every block.
   float new_mask_[kNumFreqBins];
   // Time smoothed mask.
   float time_smooth_mask_[kNumFreqBins];
   // Time and frequency smoothed mask.
   float final_mask_[kNumFreqBins];
 
   // Array of length |kNumFreqBins|, Matrix of size |1| x |num_channels_|.
   ComplexMatrixF delay_sum_masks_[kNumFreqBins];
@@ skipping 21 matching lines @@
   // The microphone normalization factor.
   ComplexMatrixF eig_m_;
 
   // For processing the high-frequency input signal.
   float high_pass_postfilter_mask_;
 
   // True when the target signal is present.
   bool is_target_present_;
   // Number of blocks after which the data is considered interference if the
   // mask does not pass |kMaskSignalThreshold|.
-  int hold_target_blocks_;
+  size_t hold_target_blocks_;
   // Number of blocks since the last mask that passed |kMaskSignalThreshold|.
-  int interference_blocks_count_;
+  size_t interference_blocks_count_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_BEAMFORMER_NONLINEAR_BEAMFORMER_H_