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

webrtc/modules/audio_coding/main/test/PCMFile.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 18 matching lines @@
     if (pcm_file_ != NULL) {
       fclose(pcm_file_);
     }
   }
 
   void Open(const std::string& filename, uint16_t frequency, const char* mode,
             bool auto_rewind = false);
 
   int32_t Read10MsData(AudioFrame& audio_frame);
 
-  void Write10MsData(int16_t *playout_buffer, uint16_t length_smpls);
+  void Write10MsData(int16_t *playout_buffer, size_t length_smpls);
   void Write10MsData(AudioFrame& audio_frame);
 
   uint16_t PayloadLength10Ms() const;
   int32_t SamplingFrequency() const;
   void Close();
   bool EndOfFile() const {
     return end_of_file_;
   }
   void Rewind();
   static int16_t ChooseFile(std::string* file_name, int16_t max_len,
                             uint16_t* frequency_hz);
   bool Rewinded();
   void SaveStereo(bool is_stereo = true);
   void ReadStereo(bool is_stereo = true);
  private:
   FILE* pcm_file_;
   uint16_t samples_10ms_;
   int32_t frequency_;
   bool end_of_file_;
   bool auto_rewind_;
   bool rewinded_;
   uint32_t timestamp_;
   bool read_stereo_;
   bool save_stereo_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_