Implement googContentType GetStats metric reported on receive side.

webrtc/call/video_receive_stream.h

 /*
  *  Copyright (c) 2013 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 70 matching lines @@
     rtc::Optional<uint64_t> qp_sum;
 
     int current_payload_type = -1;
 
     int total_bitrate_bps = 0;
     int discarded_packets = 0;
 
     int width = 0;
     int height = 0;
 
+    VideoContentType content_type = VideoContentType::UNSPECIFIED;
+
     int sync_offset_ms = std::numeric_limits<int>::max();
 
     uint32_t ssrc = 0;
     std::string c_name;
     StreamDataCounters rtp_stats;
     RtcpPacketTypeCounter rtcp_packet_type_counts;
     RtcpStatistics rtcp_stats;
 
     // Timing frame info: all important timestamps for a full lifetime of a
     // single 'timing frame'.
@@ skipping 148 matching lines @@
   virtual void AddSecondarySink(RtpPacketSinkInterface* sink) = 0;
   virtual void RemoveSecondarySink(const RtpPacketSinkInterface* sink) = 0;
 
  protected:
   virtual ~VideoReceiveStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_