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

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index edb9bf3106c2d008ad9e3976eeed9cf9a5cb6372..e4b57a77bc6dde92b839e0f77e6d0e1bfaded77f 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -1299,7 +1299,8 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
   }
 
   if (rtcp_delivered)
-    event_log_->LogRtcpPacket(kIncomingPacket, packet, length);
+    event_log_->LogIncomingRtcpPacket(
+        rtc::ArrayView<const uint8_t>(packet, length));

[danilchap, 2017/09/05 08:47:16]

for this use case there is 
rtc::MakeArrayView(packet, length)
(i.e. with template paramter deduction)

[terelius, 2017/09/07 12:53:55]

Done.

 
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
@@ -1348,7 +1349,7 @@ 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);
@@ -1360,7 +1361,7 @@ 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);