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

webrtc/common_audio/signal_processing/levinson_durbin.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_LevinsonDurbin().
  * The description header can be found in signal_processing_library.h
  *
  */
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 #define SPL_LEVINSON_MAXORDER 20
 
 int16_t WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
-                                 int16_t order)
+                                 size_t order)
 {
-    int16_t i, j;
+    size_t i, j;
     // Auto-correlation coefficients in high precision
     int16_t R_hi[SPL_LEVINSON_MAXORDER + 1], R_low[SPL_LEVINSON_MAXORDER + 1];
     // LPC coefficients in high precision
     int16_t A_hi[SPL_LEVINSON_MAXORDER + 1], A_low[SPL_LEVINSON_MAXORDER + 1];
     // LPC coefficients for next iteration
     int16_t A_upd_hi[SPL_LEVINSON_MAXORDER + 1], A_upd_low[SPL_LEVINSON_MAXORDER + 1];
     // Reflection coefficient in high precision
     int16_t K_hi, K_low;
     // Prediction gain Alpha in high precision and with scale factor
     int16_t Alpha_hi, Alpha_low, Alpha_exp;
@@ skipping 201 matching lines @@
     for (i = 1; i <= order; i++)
     {
         // temp1W32 in Q27
         temp1W32 = WEBRTC_SPL_LSHIFT_W32((int32_t)A_hi[i], 16)
                 + WEBRTC_SPL_LSHIFT_W32((int32_t)A_low[i], 1);
         // Round and store upper word
         A[i] = (int16_t)(((temp1W32 << 1) + 32768) >> 16);
     }
     return 1; // Stable filters
 }