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

webrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c

 /*
  *  Copyright (c) 2013 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 <assert.h>
 
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
 void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
                                int16_t* data_out,
                                const int16_t* __restrict coefficients,
-                               int coefficients_length,
-                               int data_length) {
+                               size_t coefficients_length,
+                               size_t data_length) {
   int r0, r1, r2, r3;
   int coef0, offset;
   int i, j, k;
   int coefptr, outptr, tmpout, inptr;
 #if !defined(MIPS_DSP_R1_LE)
   int max16 = 0x7FFF;
   int min16 = 0xFFFF8000;
 #endif  // #if !defined(MIPS_DSP_R1_LE)
 
   assert(data_length > 0);
@@ skipping 102 matching lines @@
     : [coefficients] "r" (coefficients), [data_length] "r" (data_length),
       [coefficients_length] "r" (coefficients_length),
 #if !defined(MIPS_DSP_R1_LE)
       [max16] "r" (max16), [min16] "r" (min16),
 #endif
       [data_out] "r" (data_out), [data_in] "r" (data_in)
     : "hi", "lo", "memory"
   );
 }