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

webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.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.
  */
 
@@ skipping 24 matching lines @@
     int16_t *lsf,  /* (i) the unquantized lsf coefficients Q13 */
     int16_t *lsfdeq,  /* (i) the dequantized lsf coefficients Q13 */
     int16_t *lsfold,  /* (i) the unquantized lsf coefficients of
                                            the previous signal frame Q13 */
     int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the
                                    previous signal frame Q13 */
     int16_t length,  /* (i) should equate FILTERORDER */
     IlbcEncoder *iLBCenc_inst
     /* (i/o) the encoder state structure */
                                         ) {
-  int i, pos, lp_length;
+  size_t i;
+  int pos, lp_length;
 
   int16_t *lsf2, *lsfdeq2;
   /* Stack based */
   int16_t lp[LPC_FILTERORDER + 1];
 
   lsf2 = lsf + length;
   lsfdeq2 = lsfdeq + length;
   lp_length = length + 1;
 
   if (iLBCenc_inst->mode==30) {
@@ skipping 65 matching lines @@
 
     /* update memory */
 
     WEBRTC_SPL_MEMCPY_W16(lsfold, lsf, length);
     WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq, length);
 
   }
 
   return;
 }