Use default header extension map in rtc_event_log2text

webrtc/rtc_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 101 matching lines @@
 }
 
 // Return default values for header extensions, to use on streams without stored
 // mapping data. Currently this only applies to audio streams, since the mapping
 // is not stored in the event log.
 // TODO(ivoc): Remove this once this mapping is stored in the event log for
 //             audio streams. Tracking bug: webrtc:6399
 webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
   webrtc::RtpHeaderExtensionMap default_map;
   default_map.Register<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
+  default_map.Register<TransmissionOffset>(
+      webrtc::RtpExtension::kTimestampOffsetDefaultId);
   default_map.Register<AbsoluteSendTime>(
       webrtc::RtpExtension::kAbsSendTimeDefaultId);
+  default_map.Register<VideoOrientation>(
+      webrtc::RtpExtension::kVideoRotationDefaultId);
+  default_map.Register<VideoContentTypeExtension>(
+      webrtc::RtpExtension::kVideoContentTypeDefaultId);
+  default_map.Register<VideoTimingExtension>(
+      webrtc::RtpExtension::kVideoTimingDefaultId);
+  default_map.Register<TransportSequenceNumber>(
+      webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
+  default_map.Register<PlayoutDelayLimits>(
+      webrtc::RtpExtension::kPlayoutDelayDefaultId);
   return default_map;
 }
 
 constexpr float kLeftMargin = 0.01f;
 constexpr float kRightMargin = 0.02f;
 constexpr float kBottomMargin = 0.02f;
 constexpr float kTopMargin = 0.05f;
 
 rtc::Optional<double> NetworkDelayDiff_AbsSendTime(
     const LoggedRtpPacket& old_packet,
@@ skipping 1577 matching lines @@
     plot->AppendTimeSeries(std::move(series.second));
   }
 
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetYAxis(min_y_axis, max_y_axis, "Relative delay (ms)", kBottomMargin,
                  kTopMargin);
   plot->SetTitle("NetEq timing");
 }
 }  // namespace plotting
 }  // namespace webrtc