Prepare to convert various types to size_t.

webrtc/common_audio/audio_converter_unittest.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 31 matching lines @@
 // |test_frame|. It searches around |expected_delay| in samples between the
 // signals to compensate for the resampling delay.
 float ComputeSNR(const ChannelBuffer<float>& ref,
                  const ChannelBuffer<float>& test,
                  int expected_delay) {
   VerifyParams(ref, test);
   float best_snr = 0;
   int best_delay = 0;
 
   // Search within one sample of the expected delay.
-  for (int delay = std::max(expected_delay - 1, 0);
+  for (int delay = std::max(expected_delay, 1) - 1;
        delay <= std::min(expected_delay + 1, ref.num_frames());
        ++delay) {
     float mse = 0;
     float variance = 0;
     float mean = 0;
     for (int i = 0; i < ref.num_channels(); ++i) {
       for (int j = 0; j < ref.num_frames() - delay; ++j) {
         float error = ref.channels()[i][j] - test.channels()[i][j + delay];
         mse += error * error;
         variance += ref.channels()[i][j] * ref.channels()[i][j];
@@ skipping 85 matching lines @@
         for (int dst_channel = 0; dst_channel < kChannelsSize; ++dst_channel) {
           RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate],
                                 kChannels[dst_channel], kSampleRates[dst_rate]);
         }
       }
     }
   }
 }
 
 }  // namespace webrtc