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

webrtc/common_types.h

Index: webrtc/common_types.h
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 39ece411979189cafc6d37bbee64044200828e60..693d36db7fc302e96dbfb07f512bb58b2d0626f4 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().