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

webrtc/modules/audio_coding/main/acm2/acm_resampler.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 11 matching lines @@
 ACMResampler::ACMResampler() {
 }
 
 ACMResampler::~ACMResampler() {
 }
 
 int ACMResampler::Resample10Msec(const int16_t* in_audio,
                                  int in_freq_hz,
                                  int out_freq_hz,
                                  int num_audio_channels,
-                                 int out_capacity_samples,
+                                 size_t out_capacity_samples,
                                  int16_t* out_audio) {
-  int in_length = in_freq_hz * num_audio_channels / 100;
+  size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
   int out_length = out_freq_hz * num_audio_channels / 100;
   if (in_freq_hz == out_freq_hz) {
     if (out_capacity_samples < in_length) {
       assert(false);
       return -1;
     }
     memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
-    return in_length / num_audio_channels;
+    return static_cast<int>(in_length / num_audio_channels);
   }
 
   if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
                                     num_audio_channels) != 0) {
     LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
               num_audio_channels);
     return -1;
   }
 
   out_length =
       resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
   if (out_length == -1) {
     LOG_FERR4(LS_ERROR,
               Resample,
               in_audio,
               in_length,
               out_audio,
               out_capacity_samples);
     return -1;
   }
 
   return out_length / num_audio_channels;
 }
 
 }  // namespace acm2
 }  // namespace webrtc