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

talk/app/webrtc/test/fakeaudiocapturemodule.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 39 matching lines @@
 }  // namespace rtc
 
 class FakeAudioCaptureModule
     : public webrtc::AudioDeviceModule,
       public rtc::MessageHandler {
  public:
   typedef uint16 Sample;
 
   // The value for the following constants have been derived by running VoE
   // using a real ADM. The constants correspond to 10ms of mono audio at 44kHz.
-  static const int kNumberSamples = 440;
-  static const int kNumberBytesPerSample = sizeof(Sample);
+  static const size_t kNumberSamples = 440;
+  static const size_t kNumberBytesPerSample = sizeof(Sample);
 
   // Creates a FakeAudioCaptureModule or returns NULL on failure.
   static rtc::scoped_refptr<FakeAudioCaptureModule> Create();
 
   // Returns the number of frames that have been successfully pulled by the
   // instance. Note that correctly detecting success can only be done if the
   // pulled frame was generated/pushed from a FakeAudioCaptureModule.
   int frames_received() const;
 
   // Following functions are inherited from webrtc::AudioDeviceModule.
@@ skipping 202 matching lines @@
 
   // Protects variables that are accessed from process_thread_ and
   // the main thread.
   mutable rtc::CriticalSection crit_;
   // Protects |audio_callback_| that is accessed from process_thread_ and
   // the main thread.
   rtc::CriticalSection crit_callback_;
 };
 
 #endif  // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_