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

webrtc/modules/audio_device/dummy/file_audio_device.h

 /*
  *  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 156 matching lines @@
   int32_t _playout_index;
   int32_t _record_index;
   AudioDeviceModule::BufferType _playBufType;
   AudioDeviceBuffer* _ptrAudioBuffer;
   int8_t* _recordingBuffer;  // In bytes.
   int8_t* _playoutBuffer;  // In bytes.
   uint32_t _recordingFramesLeft;
   uint32_t _playoutFramesLeft;
   CriticalSectionWrapper& _critSect;
 
-  uint32_t _recordingBufferSizeIn10MS;
-  uint32_t _recordingFramesIn10MS;
-  uint32_t _playoutFramesIn10MS;
+  size_t _recordingBufferSizeIn10MS;
+  size_t _recordingFramesIn10MS;
+  size_t _playoutFramesIn10MS;
 
   rtc::scoped_ptr<ThreadWrapper> _ptrThreadRec;
   rtc::scoped_ptr<ThreadWrapper> _ptrThreadPlay;
 
   bool _playing;
   bool _recording;
   uint64_t _lastCallPlayoutMillis;
   uint64_t _lastCallRecordMillis;
 
   FileWrapper& _outputFile;
   FileWrapper& _inputFile;
   std::string _outputFilename;
   std::string _inputFilename;
 
   Clock* _clock;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H