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

webrtc/common_audio/signal_processing/get_scaling_square.c

 /*
  *  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.
  */
 
 
 /*
  * This file contains the function WebRtcSpl_GetScalingSquare().
  * The description header can be found in signal_processing_library.h
  *
  */
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
-                                   int in_vector_length,
-                                   int times)
+                                   size_t in_vector_length,
+                                   size_t times)
 {
     int16_t nbits = WebRtcSpl_GetSizeInBits((uint32_t)times);
-    int i;
+    size_t i;
     int16_t smax = -1;
     int16_t sabs;
     int16_t *sptr = in_vector;
     int16_t t;
-    int looptimes = in_vector_length;
+    size_t looptimes = in_vector_length;
 
     for (i = looptimes; i > 0; i--)
     {
         sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
         smax = (sabs > smax ? sabs : smax);
     }
     t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
 
     if (smax == 0)
     {
         return 0; // Since norm(0) returns 0
     } else
     {
         return (t > nbits) ? 0 : nbits - t;
     }
 }