Use default header extension map in rtc_event_log2text

webrtc/rtc_tools/event_log_visualizer/analyzer.cc

Index: webrtc/rtc_tools/event_log_visualizer/analyzer.cc
diff --git a/webrtc/rtc_tools/event_log_visualizer/analyzer.cc b/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
index b42f0820f48dd878e8263dd03aaf51747769d74b..593a3b8bf72def0728a32feb579d62ab953cd50e 100644
--- a/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
@@ -119,8 +119,20 @@ int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
 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;
 }