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

webrtc/modules/utility/interface/file_player.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 20 matching lines @@
     static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
                                         const FileFormats fileFormat);
 
     static void DestroyFilePlayer(FilePlayer* player);
 
     // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
     // will be set to the number of samples read (not the number of samples per
     // channel).
     virtual int Get10msAudioFromFile(
         int16_t* outBuffer,
-        int& lengthInSamples,
+        size_t& lengthInSamples,
         int frequencyInHz) = 0;
 
     // Register callback for receiving file playing notifications.
     virtual int32_t RegisterModuleFileCallback(
         FileCallback* callback) = 0;
 
     // API for playing audio from fileName to channel.
     // Note: codecInst is used for pre-encoded files.
     virtual int32_t StartPlayingFile(
         const char* fileName,
@@ skipping 50 matching lines @@
                                      const uint32_t /*outHeight*/) {
       return -1;
     }
 
 protected:
     virtual ~FilePlayer() {}
 
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_