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

webrtc/call/rtc_event_log_unittest.cc

 /*
  *  Copyright (c) 2015 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 301 matching lines @@
 
   RTPSender rtp_sender(false,     // bool audio
                        clock,     // Clock* clock
                        nullptr,   // Transport*
                        nullptr,   // PacedSender*
                        nullptr,   // PacketRouter*
                        nullptr,   // SendTimeObserver*
                        nullptr,   // BitrateStatisticsObserver*
                        nullptr,   // FrameCountObserver*
                        nullptr,   // SendSideDelayObserver*
-                       nullptr);  // RtcEventLog*
+                       nullptr,   // RtcEventLog*
+                       nullptr);  // SendPacketObserver*
 
   std::vector<uint32_t> csrcs;
   for (unsigned i = 0; i < csrcs_count; i++) {
     csrcs.push_back(prng->Rand<uint32_t>());
   }
   rtp_sender.SetCsrcs(csrcs);
   rtp_sender.SetSSRC(prng->Rand<uint32_t>());
   rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true);
   rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>());
 
@@ skipping 365 matching lines @@
                   rtcp_packet.data(), rtcp_packet.size());
 
   VerifyLogEndEvent(parsed_stream.stream(3));
 
   // Clean up temporary file - can be pretty slow.
   remove(temp_filename.c_str());
 }
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG