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_armv7.S

 @
 @ 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.
 @
 
@@ skipping 137 matching lines @@
 
 END:
   pop {r4-r11}
   bx  lr
 
 @Reference C code:
 @
 @void WebRtcSpl_FilterARFastQ12(int16_t* data_in,
 @                               int16_t* data_out,
 @                               int16_t* __restrict coefficients,
-@                               int coefficients_length,
-@                               int data_length) {
-@  int i = 0;
-@  int j = 0;
+@                               size_t coefficients_length,
+@                               size_t data_length) {
+@  size_t i = 0;
+@  size_t j = 0;
 @
 @  assert(data_length > 0);
 @  assert(coefficients_length > 1);
 @
 @  for (i = 0; i < data_length - 1; i += 2) {
 @    int32_t output1 = 0;
 @    int32_t sum1 = 0;
 @    int32_t output2 = 0;
 @    int32_t sum2 = 0;
 @
@@ skipping 37 matching lines @@
 @      sum1 += coefficients[1] * data_out[i - 1];
 @    }
 @
 @    output1 = coefficients[0] * data_in[i];
 @    output1 -= sum1;
 @    // Saturate and store the output.
 @    output1 = WEBRTC_SPL_SAT(134215679, output1, -134217728);
 @    data_out[i] = (int16_t)((output1 + 2048) >> 12);
 @  }
 @}