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

talk/app/webrtc/test/fakeaudiocapturemodule.h

Index: talk/app/webrtc/test/fakeaudiocapturemodule.h
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.h b/talk/app/webrtc/test/fakeaudiocapturemodule.h
index be1df9a7bd7e48c249b8011256994d9757bb7cc7..9f36ed80af45e87ba029d6da170bbc6dc018ae41 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.h
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule.h
@@ -57,8 +57,8 @@ class FakeAudioCaptureModule
 
   // 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();