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

webrtc/common_types.h

Index: webrtc/common_types.h
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 1156bbcd77b8a7a34cc5203cdcfe439b6a5dd92d..8d930e996e7b9b2993b1362fd1d635a486b64c9b 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -381,7 +381,7 @@ struct NetworkStatistics           // NETEQ statistics
     // max packet waiting time in the jitter buffer (ms)
     int maxWaitingTimeMs;
     // added samples in off mode due to packet loss
-    int addedSamples;
+    size_t addedSamples;
 };
 
 // Statistics for calls to AudioCodingModule::PlayoutData10Ms().