Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 1271 matching lines @@
   }
   if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
     ReadLockScoped read_lock(*send_crit_);
     for (auto& kv : audio_send_ssrcs_) {
       if (kv.second->DeliverRtcp(packet, length))
         rtcp_delivered = true;
     }
   }
 
   if (rtcp_delivered)
-    event_log_->LogRtcpPacket(kIncomingPacket, packet, length);

[the sun, 2017/08/22 11:44:31]

Can you remove the PacketDirection enum?

[terelius, 2017/09/04 12:47:51]

Not yet, since there apparently are downstream dependencies.

+    event_log_->LogIncomingRtcpPacket(packet, length);
 
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length,
                                                 const PacketTime& packet_time) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
 
@@ skipping 28 matching lines @@
     return DELIVERY_UNKNOWN_SSRC;
   }
   parsed_packet->IdentifyExtensions(it->second.extensions);
 
   NotifyBweOfReceivedPacket(*parsed_packet, media_type);
 
   if (media_type == MediaType::AUDIO) {
     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(packet, length);
       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);
       }
       last_received_rtp_audio_ms_.emplace(arrival_time_ms);
       return DELIVERY_OK;
     }
   } else if (media_type == MediaType::VIDEO) {
     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(packet, length);
       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);
       }
       last_received_rtp_video_ms_.emplace(arrival_time_ms);
       return DELIVERY_OK;
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
@@ skipping 51 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc