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

webrtc/modules/rtp_rtcp/include/rtp_rtcp.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 54 matching lines @@
     TransportFeedbackObserver* transport_feedback_callback;
     RtcpRttStats* rtt_stats;
     RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
     RtpAudioFeedback* audio_messages;
     RemoteBitrateEstimator* remote_bitrate_estimator;
     RtpPacketSender* paced_sender;
     TransportSequenceNumberAllocator* transport_sequence_number_allocator;
     BitrateStatisticsObserver* send_bitrate_observer;
     FrameCountObserver* send_frame_count_observer;
     SendSideDelayObserver* send_side_delay_observer;
+    SendPacketObserver* send_packet_observer;
   };
 
   /*
    *   Create a RTP/RTCP module object using the system clock.
    *
    *   configuration  - Configuration of the RTP/RTCP module.
    */
   static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);
 
   /**************************************************************************
@@ skipping 547 matching lines @@
 
     /*
     *   send a request for a keyframe
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_