diff --git a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
index a3d3456a7b..46ce248ff6 100644
--- a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
+++ b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
@@ -97,9 +97,10 @@ bool ShouldSkipStream(MediaType media_type,
 // Convert a LoggedRtpPacketIncoming to a test::RtpPacket. Header extension IDs
 // are allocated according to the provided extension map. This might not match
 // the extension map used in the actual call.
-void ConvertRtpPacket(const webrtc::LoggedRtpPacketIncoming& incoming,
-                      const webrtc::RtpHeaderExtensionMap default_extension_map,
-                      webrtc::test::RtpPacket* packet) {
+void ConvertRtpPacket(
+    const webrtc::LoggedRtpPacketIncoming& incoming,
+    const webrtc::RtpHeaderExtensionMap& default_extension_map,
+    webrtc::test::RtpPacket* packet) {
   webrtc::RtpPacket reconstructed_packet(&default_extension_map);
 
   reconstructed_packet.SetMarker(incoming.rtp.header.markerBit);