Update audio code to use size_t more correctly, talk/ portion.

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 8ff4aa19e7552eea7a0520bf6f4907ce865a081c..2cf316e9fd181fee9842255785c906bb4f5e44c4 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.h
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule.h
@@ -58,8 +58,8 @@ 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.
   // |process_thread| is used to push and pull audio frames to and from the