Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.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"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
 
 /* Filter coefficicients in Q15. */
 static const int16_t kDampFilter[PITCH_DAMPORDER] = {
   -2294, 8192, 20972, 8192, -2294
 };
 
 void WebRtcIsacfix_PitchFilterCore(int loopNumber,
                                    int16_t gain,
-                                   int index,
+                                   size_t index,
                                    int16_t sign,
                                    int16_t* inputState,
                                    int16_t* outputBuf2,
                                    const int16_t* coefficient,
                                    int16_t* inputBuf,
                                    int16_t* outputBuf,
                                    int* index2) {
   int i = 0, j = 0;  /* Loop counters. */
   int16_t* ubufQQpos2 = &outputBuf2[PITCH_BUFFSIZE - (index + 2)];
   int16_t tmpW16 = 0;
@@ skipping 32 matching lines @@
 
     /* Subtract from input and update buffer. */
     tmpW32 = inputBuf[*index2] - sign * tmpW16;
     outputBuf[*index2] = WebRtcSpl_SatW32ToW16(tmpW32);
     tmpW32 = inputBuf[*index2] + outputBuf[*index2];
     outputBuf2[*index2 + PITCH_BUFFSIZE] = WebRtcSpl_SatW32ToW16(tmpW32);
 
     (*index2)++;
   }
 }