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

webrtc/modules/audio_coding/codecs/ilbc/decoder_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 16 matching lines @@
 
 void WebRtcIlbcfix_DecoderInterpolateLsp(
     int16_t *syntdenum,  /* (o) synthesis filter coefficients */
     int16_t *weightdenum, /* (o) weighting denumerator
                                    coefficients */
     int16_t *lsfdeq,   /* (i) dequantized lsf coefficients */
     int16_t length,   /* (i) length of lsf coefficient vector */
     IlbcDecoder *iLBCdec_inst
     /* (i) the decoder state structure */
                                           ){
-  int  i, pos, lp_length;
+  size_t i;
+  int pos, lp_length;
   int16_t  lp[LPC_FILTERORDER + 1], *lsfdeq2;
 
   lsfdeq2 = lsfdeq + length;
   lp_length = length + 1;
 
   if (iLBCdec_inst->mode==30) {
     /* subframe 1: Interpolation between old and first LSF */
 
     WebRtcIlbcfix_LspInterpolate2PolyDec(lp, (*iLBCdec_inst).lsfdeqold, lsfdeq,
                                          WebRtcIlbcfix_kLsfWeight30ms[0], length);
@@ skipping 25 matching lines @@
   }
 
   /* update memory */
 
   if (iLBCdec_inst->mode==30) {
     WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq2, length);
   } else {
     WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq, length);
   }
 }