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

webrtc/common_audio/signal_processing/auto_correlation.c

 /*
  *  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.
  */
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 int WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
-                              int in_vector_length,
-                              int order,
+                              size_t in_vector_length,
+                              size_t order,
                               int32_t* result,
                               int* scale) {
   int32_t sum = 0;
-  int i = 0, j = 0;
+  size_t i = 0, j = 0;
   int16_t smax = 0;
   int scaling = 0;
 
   if (order > in_vector_length) {
     /* Undefined */
     return -1;
-  } else if (order < 0) {
-    order = in_vector_length;
   }
 
   // Find the maximum absolute value of the samples.
   smax = WebRtcSpl_MaxAbsValueW16(in_vector, in_vector_length);
 
   // In order to avoid overflow when computing the sum we should scale the
   // samples so that (in_vector_length * smax * smax) will not overflow.
   if (smax == 0) {
     scaling = 0;
   } else {
@@ skipping 19 matching lines @@
       sum += (in_vector[j + 2] * in_vector[i + j + 2]) >> scaling;
       sum += (in_vector[j + 3] * in_vector[i + j + 3]) >> scaling;
     }
     for (; j < in_vector_length - i; j++) {
       sum += (in_vector[j] * in_vector[i + j]) >> scaling;
     }
     *result++ = sum;
   }
 
   *scale = scaling;
-  return order + 1;
+  return (int)(order + 1);
 }