Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 namespace test {
 
 bool ResampleInputAudioFile::Read(size_t samples,
                                   int output_rate_hz,
                                   int16_t* destination) {
   const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
   CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
       << "Frame size and sample rates don't add up to an integer.";
   rtc::scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
   if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
     return false;
   resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
-  int output_length = 0;
-  CHECK_EQ(resampler_.Push(temp_destination.get(),
-                           static_cast<int>(samples_to_read),
-                           destination,
-                           static_cast<int>(samples),
-                           output_length),
+  size_t output_length = 0;
+  CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read, destination,
+                           samples, output_length),
            0);
-  CHECK_EQ(static_cast<int>(samples), output_length);
+  CHECK_EQ(samples, output_length);
   return true;
 }
 
 bool ResampleInputAudioFile::Read(size_t samples, int16_t* destination) {
   CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
   return Read(samples, output_rate_hz_, destination);
 }
 
 void ResampleInputAudioFile::set_output_rate_hz(int rate_hz) {
   output_rate_hz_ = rate_hz;
 }
 
 }  // namespace test
 }  // namespace webrtc