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

webrtc/modules/audio_device/dummy/file_audio_device.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 <iostream>
@@ skipping 154 matching lines @@
   return -1;
 }
 
 int32_t FileAudioDevice::InitRecording() {
   CriticalSectionScoped lock(&_critSect);
 
   if (_recording) {
     return -1;
   }
 
-  _recordingFramesIn10MS = static_cast<uint32_t>(kRecordingFixedSampleRate/100);
+  _recordingFramesIn10MS = static_cast<size_t>(kRecordingFixedSampleRate / 100);
 
   if (_ptrAudioBuffer) {
     _ptrAudioBuffer->SetRecordingSampleRate(kRecordingFixedSampleRate);
     _ptrAudioBuffer->SetRecordingChannels(kRecordingNumChannels);
   }
   return 0;
 }
 
 bool FileAudioDevice::RecordingIsInitialized() const {
   return true;
 }
 
 int32_t FileAudioDevice::StartPlayout() {
   if (_playing) {
       return 0;
   }
 
-  _playoutFramesIn10MS = static_cast<uint32_t>(kPlayoutFixedSampleRate/100);
+  _playoutFramesIn10MS = static_cast<size_t>(kPlayoutFixedSampleRate / 100);
   _playing = true;
   _playoutFramesLeft = 0;
 
   if (!_playoutBuffer) {
       _playoutBuffer = new int8_t[2 *
                                   kPlayoutNumChannels *
                                   kPlayoutFixedSampleRate/100];
   }
   if (!_playoutBuffer) {
     _playing = false;
@@ skipping 338 matching lines @@
         _critSect.Enter();
       }
     }
 
     _critSect.Leave();
     SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
     return true;
 }
 
 }  // namespace webrtc