Don't use LOG_END timestamp to compute RtcEventLog duration.

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 112 matching lines @@
   MediaType media_type;
   uint8_t header[IP_PACKET_SIZE];
   size_t header_length;
   size_t total_length;
 
   for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
     ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
     if (event_type != ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT &&
         event_type != ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT &&
         event_type != ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT &&
-        event_type != ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
+        event_type != ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT &&
+        event_type != ParsedRtcEventLog::LOG_START &&
+        event_type != ParsedRtcEventLog::LOG_END) {
       uint64_t timestamp = parsed_log_.GetTimestamp(i);
       first_timestamp = std::min(first_timestamp, timestamp);
       last_timestamp = std::max(last_timestamp, timestamp);
     }
 
     switch (parsed_log_.GetEventType(i)) {
       case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
         VideoReceiveStream::Config config(nullptr);
         parsed_log_.GetVideoReceiveConfig(i, &config);
         StreamId stream(config.rtp.remote_ssrc, kIncomingPacket,
@@ skipping 506 matching lines @@
   plot->yaxis_label = "Bitrate (kbps)";
   if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
     plot->title = "Incoming bitrate per stream";
   } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
     plot->title = "Outgoing bitrate per stream";
   }
 }
 
 }  // namespace plotting
 }  // namespace webrtc