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

webrtc/common_audio/signal_processing/filter_ma_fast_q12.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_FilterMAFastQ12().
  * The description header can be found in signal_processing_library.h
  *
  */
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 void WebRtcSpl_FilterMAFastQ12(const int16_t* in_ptr,
                                int16_t* out_ptr,
                                const int16_t* B,
-                               int16_t B_length,
-                               int16_t length)
+                               size_t B_length,
+                               size_t length)
 {
-    int i, j;
+    size_t i, j;
     for (i = 0; i < length; i++)
     {
         int32_t o = 0;
 
         for (j = 0; j < B_length; j++)
         {
           o += B[j] * in_ptr[i - j];
         }
 
         // If output is higher than 32768, saturate it. Same with negative side
         // 2^27 = 134217728, which corresponds to 32768 in Q12
 
         // Saturate the output
         o = WEBRTC_SPL_SAT((int32_t)134215679, o, (int32_t)-134217728);
 
         *out_ptr++ = (int16_t)((o + (int32_t)2048) >> 12);
     }
     return;
 }