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); |