[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_processing/test/test_utils.cc

 /*
  *  Copyright (c) 2015 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 26 matching lines @@
                   RawFile* raw_file) {
   if (wav_file) {
     wav_file->WriteSamples(data, length);
   }
   if (raw_file) {
     raw_file->WriteSamples(data, length);
   }
 }
 
 void WriteFloatData(const float* const* data,
-                    int samples_per_channel,
-                    int num_channels,
+                    size_t samples_per_channel,
+                    size_t num_channels,
                     WavWriter* wav_file,
                     RawFile* raw_file) {
   size_t length = num_channels * samples_per_channel;
   rtc::scoped_ptr<float[]> buffer(new float[length]);
   Interleave(data, samples_per_channel, num_channels, buffer.get());
   if (raw_file) {
     raw_file->WriteSamples(buffer.get(), length);
   }
   // TODO(aluebs): Use ScaleToInt16Range() from audio_util
   for (size_t i = 0; i < length; ++i) {
@@ skipping 19 matching lines @@
   return AudioProcessing::kChunkSizeMs * rate / 1000;
 }
 
 void SetFrameSampleRate(AudioFrame* frame,
                         int sample_rate_hz) {
   frame->sample_rate_hz_ = sample_rate_hz;
   frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
       sample_rate_hz / 1000;
 }
 
-AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels) {
+AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
   switch (num_channels) {
     case 1:
       return AudioProcessing::kMono;
     case 2:
       return AudioProcessing::kStereo;
     default:
       assert(false);
       return AudioProcessing::kMono;
   }
 }
@@ skipping 11 matching lines @@
     double y = values[i + 1];
     double z = values[i + 2];
     result.push_back(Point(x, y, z));
   }
 
   return result;
 }
 
 
 }  // namespace webrtc