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

webrtc/modules/audio_processing/ns/ns_core.h

 /*
  *  Copyright (c) 2011 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 33 matching lines @@
   float maxSpecDiff;
   float minSpecDiff;
   // Criteria of weight of histogram peak to accept/reject feature.
   int thresWeightSpecFlat;
   int thresWeightSpecDiff;
 
 } NSParaExtract;
 
 typedef struct NoiseSuppressionC_ {
   uint32_t fs;
-  int blockLen;
-  int windShift;
-  int anaLen;
-  int magnLen;
+  size_t blockLen;
+  size_t windShift;
+  size_t anaLen;
+  size_t magnLen;
   int aggrMode;
   const float* window;
   float analyzeBuf[ANAL_BLOCKL_MAX];
   float dataBuf[ANAL_BLOCKL_MAX];
   float syntBuf[ANAL_BLOCKL_MAX];
 
   int initFlag;
   // Parameters for quantile noise estimation.
   float density[SIMULT * HALF_ANAL_BLOCKL];
   float lquantile[SIMULT * HALF_ANAL_BLOCKL];
   float quantile[HALF_ANAL_BLOCKL];
   int counter[SIMULT];
   int updates;
   // Parameters for Wiener filter.
   float smooth[HALF_ANAL_BLOCKL];
   float overdrive;
   float denoiseBound;
   int gainmap;
   // FFT work arrays.
-  int ip[IP_LENGTH];
+  size_t ip[IP_LENGTH];
   float wfft[W_LENGTH];
 
   // Parameters for new method: some not needed, will reduce/cleanup later.
   int32_t blockInd;  // Frame index counter.
   int modelUpdatePars[4];  // Parameters for updating or estimating.
   // Thresholds/weights for prior model.
   float priorModelPars[7];  // Parameters for prior model.
   float noise[HALF_ANAL_BLOCKL];  // Noise spectrum from current frame.
   float noisePrev[HALF_ANAL_BLOCKL];  // Noise spectrum from previous frame.
   // Magnitude spectrum of previous analyze frame.
@@ skipping 86 matching lines @@
  *      - self          : Instance that should be initialized
  *      - inFrame       : Input speech frame for each band
  *      - num_bands     : Number of bands
  *
  * Output:
  *      - self          : Updated instance
  *      - outFrame      : Output speech frame for each band
  */
 void WebRtcNs_ProcessCore(NoiseSuppressionC* self,
                           const float* const* inFrame,
-                          int num_bands,
+                          size_t num_bands,
                           float* const* outFrame);
 
 #ifdef __cplusplus
 }
 #endif
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_NS_CORE_H_