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 bda9c7ec50475a1ea4542d3a452823e14655030b..bef8ff42cd6573601cb3163ad08fa5563037534a 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().