Move RTP stats histograms from VieChannel to SendStatisticsProxy.

webrtc/common_types.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 773 matching lines @@
       padding_bytes(0),
       packets(0) {}
 
   void Add(const RtpPacketCounter& other) {
     header_bytes += other.header_bytes;
     payload_bytes += other.payload_bytes;
     padding_bytes += other.padding_bytes;
     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;
     padding_bytes += header.paddingLength;
     payload_bytes +=
         packet_length - (header.headerLength + header.paddingLength);
   }
 
   size_t TotalBytes() const {
     return header_bytes + payload_bytes + padding_bytes;
@@ skipping 14 matching lines @@
     retransmitted.Add(other.retransmitted);
     fec.Add(other.fec);
     if (other.first_packet_time_ms != -1 &&
        (other.first_packet_time_ms < first_packet_time_ms ||
         first_packet_time_ms == -1)) {
       // Use oldest time.
       first_packet_time_ms = other.first_packet_time_ms;
     }
   }
 
+  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);
   }
 
   // Returns the number of bytes corresponding to the actual media payload (i.e.
   // RTP headers, padding, retransmissions and fec packets are excluded).
   // Note this function does not have meaning for an RTX stream.
   size_t MediaPayloadBytes() const {
     return transmitted.payload_bytes - retransmitted.payload_bytes -
            fec.payload_bytes;
@@ skipping 14 matching lines @@
                                    uint32_t ssrc) = 0;
 };
 
 // RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
 // RTCP mode is described by RFC 5506.
 enum class RtcpMode { kOff, kCompound, kReducedSize };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_