Add histogram stats for send delay for a sent video stream.

webrtc/video/rtp_stream_receiver.cc

 /*
  *  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 35 matching lines @@
   configuration.intra_frame_callback = nullptr;
   configuration.rtt_stats = rtt_stats;
   configuration.rtcp_packet_type_counter_observer =
       rtcp_packet_type_counter_observer;
   configuration.paced_sender = paced_sender;
   configuration.transport_sequence_number_allocator =
       transport_sequence_number_allocator;
   configuration.send_bitrate_observer = nullptr;
   configuration.send_frame_count_observer = nullptr;
   configuration.send_side_delay_observer = nullptr;
+  configuration.send_packet_observer = nullptr;
   configuration.bandwidth_callback = nullptr;
   configuration.transport_feedback_callback = nullptr;
 
   std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration));
   rtp_rtcp->SetSendingStatus(false);
   rtp_rtcp->SetSendingMediaStatus(false);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
 
   return rtp_rtcp;
 }
@@ skipping 381 matching lines @@
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 }  // namespace webrtc