Move RTP stats histograms from VieChannel to SendStatisticsProxy.

webrtc/common_types.h

Index: webrtc/common_types.h
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 2999bc95bc82a353fd820e87506059c29d552d8c..d3fc204f55dc68a600b851cea29a8ee4c65f234b 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -791,6 +791,13 @@ struct RtpPacketCounter {
     packets += other.packets;
   }
 
+  void Subtract(const RtpPacketCounter& other) {
+    header_bytes -= other.header_bytes;

[mflodman, 2016/02/29 09:41:17]

There's no >= 0 check, do we count on the user to handle that?

[sprang_webrtc, 2016/02/29 13:00:33]

Yes. But I'll add a DCHECK to be sure.

+    payload_bytes -= other.payload_bytes;
+    padding_bytes -= other.padding_bytes;
+    packets -= other.packets;
+  }
+
   void AddPacket(size_t packet_length, const RTPHeader& header) {
     ++packets;
     header_bytes += header.headerLength;
@@ -825,6 +832,18 @@ struct StreamDataCounters {
     }
   }
 
+  void Subtract(const StreamDataCounters& other) {
+    transmitted.Subtract(other.transmitted);
+    retransmitted.Subtract(other.retransmitted);
+    fec.Subtract(other.fec);
+    if (other.first_packet_time_ms != -1 &&
+        (other.first_packet_time_ms > first_packet_time_ms ||
+         first_packet_time_ms == -1)) {
+      // Use youngest time.
+      first_packet_time_ms = other.first_packet_time_ms;
+    }
+  }
+
   int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
     return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
   }