Ignore this CL - here as a baseline only (originally Bjorn's CL)

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index a41e30b6a771efd169528e5d915ea62f18114c87..9b061623704b18a3bae3477ec53d07d7e5f83179 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -1303,7 +1303,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
   }
 
   if (rtcp_delivered)
-    event_log_->LogRtcpPacket(kIncomingPacket, packet, length);
+    event_log_->LogIncomingRtcpPacket(rtc::MakeArrayView(packet, length));
 
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
@@ -1352,7 +1352,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
     if (audio_receiver_controller_.OnRtpPacket(*parsed_packet)) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
-      event_log_->LogRtpHeader(kIncomingPacket, packet, length);
+      event_log_->LogIncomingRtpHeader(*parsed_packet);
       const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
       if (!first_received_rtp_audio_ms_) {
         first_received_rtp_audio_ms_.emplace(arrival_time_ms);
@@ -1364,7 +1364,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
     if (video_receiver_controller_.OnRtpPacket(*parsed_packet)) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
-      event_log_->LogRtpHeader(kIncomingPacket, packet, length);
+      event_log_->LogIncomingRtpHeader(*parsed_packet);
       const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
       if (!first_received_rtp_video_ms_) {
         first_received_rtp_video_ms_.emplace(arrival_time_ms);