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

webrtc/modules/audio_processing/high_pass_filter_impl.cc

 /*
  *  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 29 matching lines @@
   } else {
     hpf->ba = kFilterCoefficients;
   }
 
   WebRtcSpl_MemSetW16(hpf->x, 0, 2);
   WebRtcSpl_MemSetW16(hpf->y, 0, 4);
 
   return AudioProcessing::kNoError;
 }
 
-int Filter(FilterState* hpf, int16_t* data, int length) {
+int Filter(FilterState* hpf, int16_t* data, size_t length) {
   assert(hpf != NULL);
 
   int32_t tmp_int32 = 0;
   int16_t* y = hpf->y;
   int16_t* x = hpf->x;
   const int16_t* ba = hpf->ba;
 
-  for (int i = 0; i < length; i++) {
+  for (size_t i = 0; i < length; i++) {
     //  y[i] = b[0] * x[i] + b[1] * x[i-1] + b[2] * x[i-2]
     //         + -a[1] * y[i-1] + -a[2] * y[i-2];
 
     tmp_int32 = y[1] * ba[3];  // -a[1] * y[i-1] (low part)
     tmp_int32 += y[3] * ba[4];  // -a[2] * y[i-2] (low part)
     tmp_int32 = (tmp_int32 >> 15);
     tmp_int32 += y[0] * ba[3];  // -a[1] * y[i-1] (high part)
     tmp_int32 += y[2] * ba[4];  // -a[2] * y[i-2] (high part)
     tmp_int32 = (tmp_int32 << 1);
 
@@ skipping 90 matching lines @@
 int HighPassFilterImpl::num_handles_required() const {
   return apm_->num_output_channels();
 }
 
 int HighPassFilterImpl::GetHandleError(void* handle) const {
   // The component has no detailed errors.
   assert(handle != NULL);
   return apm_->kUnspecifiedError;
 }
 }  // namespace webrtc