Don't simulate probing based on rtc event logs since we don't have that info logged.

webrtc/tools/event_log_visualizer/analyzer.cc

 /*
  *  Copyright (c) 2016 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 816 matching lines @@
             *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
       }
       ++rtcp_iterator;
     }
     if (clock.TimeInMicroseconds() >= NextRtpTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
       const LoggedRtpPacket& rtp = *rtp_iterator->second;
       if (rtp.header.extension.hasTransportSequenceNumber) {
         RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
         cc.GetTransportFeedbackObserver()->AddPacket(
-            rtp.header.extension.transportSequenceNumber, rtp.total_length, 0);
+            rtp.header.extension.transportSequenceNumber, rtp.total_length,
+            PacketInfo::kNotAProbe);
         rtc::SentPacket sent_packet(
             rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
         cc.OnSentPacket(sent_packet);
       }
       ++rtp_iterator;
     }
     if (clock.TimeInMicroseconds() >= NextProcessTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime());
       cc.Process();
     }
@@ skipping 96 matching lines @@
     point.y -= estimated_base_delay_ms;
   // Add the data set to the plot.
   plot->series_list_.push_back(std::move(time_series));
 
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
   plot->SetTitle("Network Delay Change.");
 }
 }  // namespace plotting
 }  // namespace webrtc