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

webrtc/modules/audio_processing/intelligibility/test/intelligibility_proc.cc

 /*
  *  Copyright (c) 2014 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 93 matching lines @@
 
   WavReader noise_file(FLAGS_noise_file);
   std::vector<float> noise_fpcm(samples);
   noise_file.ReadSamples(samples, &noise_fpcm[0]);
 
   // Run intelligibility enhancement.
   IntelligibilityEnhancer::Config config;
   config.sample_rate_hz = FLAGS_sample_rate;
   config.var_type = static_cast<VarianceArray::StepType>(FLAGS_clear_type);
   config.var_decay_rate = static_cast<float>(FLAGS_clear_alpha);
-  config.var_window_size = FLAGS_clear_window;
+  config.var_window_size = static_cast<size_t>(FLAGS_clear_window);
   config.analysis_rate = FLAGS_ana_rate;
   config.gain_change_limit = FLAGS_gain_limit;
   IntelligibilityEnhancer enh(config);
 
   // Slice the input into smaller chunks, as the APM would do, and feed them
   // through the enhancer.
   float* clear_cursor = &in_fpcm[0];
   float* noise_cursor = &noise_fpcm[0];
 
   for (size_t i = 0; i < samples; i += fragment_size) {
@@ skipping 18 matching lines @@
   }
 }
 
 }  // namespace
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   webrtc::void_main(argc, argv);
   return 0;
 }