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

webrtc/modules/utility/source/file_player_impl.h

 /*
  *  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 13 matching lines @@
 
 namespace webrtc {
 class FilePlayerImpl : public FilePlayer
 {
 public:
     FilePlayerImpl(uint32_t instanceID, FileFormats fileFormat);
     ~FilePlayerImpl();
 
     virtual int Get10msAudioFromFile(
         int16_t* outBuffer,
-        int& lengthInSamples,
+        size_t& lengthInSamples,
         int frequencyInHz);
     virtual int32_t RegisterModuleFileCallback(FileCallback* callback);
     virtual int32_t StartPlayingFile(
         const char* fileName,
         bool loop,
         uint32_t startPosition,
         float volumeScaling,
         uint32_t notification,
         uint32_t stopPosition = 0,
         const CodecInst* codecInst = NULL);
@@ skipping 25 matching lines @@
 
     CodecInst _codec;
     int32_t _numberOf10MsPerFrame;
     int32_t _numberOf10MsInDecoder;
 
     Resampler _resampler;
     float _scaling;
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_