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

webrtc/common_audio/signal_processing/downsample_fast.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"
 
 // TODO(Bjornv): Change the function parameter order to WebRTC code style.
 // C version of WebRtcSpl_DownsampleFast() for generic platforms.
 int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
-                              int data_in_length,
+                              size_t data_in_length,
                               int16_t* data_out,
-                              int data_out_length,
+                              size_t data_out_length,
                               const int16_t* __restrict coefficients,
-                              int coefficients_length,
+                              size_t coefficients_length,
                               int factor,
-                              int delay) {
-  int i = 0;
-  int j = 0;
+                              size_t delay) {
+  size_t i = 0;
+  size_t j = 0;
   int32_t out_s32 = 0;
-  int endpos = delay + factor * (data_out_length - 1) + 1;
+  size_t endpos = delay + factor * (data_out_length - 1) + 1;
 
   // Return error if any of the running conditions doesn't meet.
-  if (data_out_length <= 0 || coefficients_length <= 0
+  if (data_out_length == 0 || coefficients_length == 0
                            || data_in_length < endpos) {
     return -1;
   }
 
   for (i = delay; i < endpos; i += factor) {
     out_s32 = 2048;  // Round value, 0.5 in Q12.
 
     for (j = 0; j < coefficients_length; j++) {
       out_s32 += coefficients[j] * data_in[i - j];  // Q12.
     }
 
     out_s32 >>= 12;  // Q0.
 
     // Saturate and store the output.
     *data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
   }
 
   return 0;
 }