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

webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.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 33 matching lines @@
   int16_t data[kDataLength] = {0};
   AudioFeatures features;
   double sp[kMaxNumFrames];
   while (fread(data, sizeof(int16_t), kDataLength, pcm_file) == kDataLength) {
     audioproc.ExtractFeatures(data, kDataLength, &features);
     if (features.num_frames > 0) {
       ASSERT_LT(features.num_frames, kMaxNumFrames);
       // Read reference values.
       const size_t num_frames = features.num_frames;
       ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file));
-      for (int n = 0; n < features.num_frames; n++)
+      for (size_t n = 0; n < features.num_frames; n++)
         EXPECT_NEAR(features.spectral_peak[n], sp[n], 3);
     }
   }
 
   fclose(peak_file);
   fclose(pcm_file);
 }
 
 }  // namespace webrtc