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

webrtc/test/fake_audio_device.h

 /*
  *  Copyright (c) 2013 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.
  */
 #ifndef WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_
@@ skipping 30 matching lines @@
   bool Recording() const override;
 
   void Start();
   void Stop();
 
  private:
   static bool Run(void* obj);
   void CaptureAudio();
 
   static const uint32_t kFrequencyHz = 16000;
-  static const uint32_t kBufferSizeBytes = 2 * kFrequencyHz;
+  static const size_t kBufferSizeBytes = 2 * kFrequencyHz;
 
   AudioTransport* audio_callback_;
   bool capturing_;
   int8_t captured_audio_[kBufferSizeBytes];
   int8_t playout_buffer_[kBufferSizeBytes];
   int64_t last_playout_ms_;
 
   Clock* clock_;
   rtc::scoped_ptr<EventTimerWrapper> tick_;
   mutable rtc::CriticalSection lock_;
   rtc::scoped_ptr<ThreadWrapper> thread_;
   rtc::scoped_ptr<ModuleFileUtility> file_utility_;
   rtc::scoped_ptr<FileWrapper> input_stream_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_FAKE_AUDIO_DEVICE_H_