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

webrtc/video/vie_channel.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 45 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 279 matching lines @@
   rtc::CritScope lock(&crit_);
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   rtc::CritScope lock(&crit_);
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc