Update audio code to use size_t more correctly, webrtc/modules/audio_processing/

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 50 matching lines @@
 DEFINE_double(gain_limit, 1000.0, "Maximum gain change in one block.");
 
 DEFINE_string(clear_file, "speech.wav", "Input file with clear speech.");
 DEFINE_string(noise_file, "noise.wav", "Input file with noise data.");
 DEFINE_string(out_file,
               "proc_enhanced.wav",
               "Enhanced output. Use '-' to "
               "play through aplay immediately.");
 
 // Constant IntelligibilityEnhancer constructor parameters.
-const int kErbResolution = 2;
+const size_t kErbResolution = 2;
 const int kNumChannels = 1;
 
 // void function for gtest
 void void_main(int argc, char* argv[]) {
   google::SetUsageMessage(
       "\n\nVariance algorithm types are:\n"
       "  0 - infinite/normal,\n"
       "  1 - exponentially decaying,\n"
       "  2 - rolling window.\n"
       "\nInput files must be little-endian 16-bit signed raw PCM.\n");
@@ skipping 22 matching lines @@
   in_file.ReadSamples(samples, &in_fpcm[0]);
 
   WavReader noise_file(FLAGS_noise_file);
   std::vector<float> noise_fpcm(samples);
   noise_file.ReadSamples(samples, &noise_fpcm[0]);
 
   // Run intelligibility enhancement.
 
   IntelligibilityEnhancer enh(
       kErbResolution, FLAGS_sample_rate, kNumChannels, FLAGS_clear_type,
-      static_cast<float>(FLAGS_clear_alpha), FLAGS_clear_window, FLAGS_ana_rate,
-      FLAGS_var_rate, FLAGS_gain_limit);
+      static_cast<float>(FLAGS_clear_alpha),
+      static_cast<size_t>(FLAGS_clear_window), FLAGS_ana_rate, FLAGS_var_rate,
+      FLAGS_gain_limit);
 
   // 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) {
     enh.ProcessCaptureAudio(&noise_cursor);
     enh.ProcessRenderAudio(&clear_cursor);
     clear_cursor += fragment_size;
@@ skipping 15 matching lines @@
   }
 }
 
 }  // namespace
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   webrtc::void_main(argc, argv);
   return 0;
 }