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

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 14bbff781c50bfc3fc4ebff9b43da808043298ac..765c9638e85e6c7763e74f28567fe8da6a79171c 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -513,8 +513,7 @@ TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
 TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                kSsrc, kSeqNum, _, _, _));
-  EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
+  EXPECT_CALL(mock_rtc_event_log_, LogOutgoingRtpHeader(_, _));
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@@ -558,8 +557,7 @@ TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
 TEST_P(RtpSenderTest, TrafficSmoothingRetransmits) {
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                kSsrc, kSeqNum, _, _, _));
-  EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
+  EXPECT_CALL(mock_rtc_event_log_, LogOutgoingRtpHeader(_, _));
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@@ -612,9 +610,7 @@ TEST_P(RtpSenderTest, SendPadding) {
   // Make all (non-padding) packets go to send queue.
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                kSsrc, kSeqNum, _, _, _));
-  EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
-      .Times(1 + 4 + 1);
+  EXPECT_CALL(mock_rtc_event_log_, LogOutgoingRtpHeader(_, _)).Times(1 + 4 + 1);
 
   uint16_t seq_num = kSeqNum;
   uint32_t timestamp = kTimestamp;
@@ -813,8 +809,7 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
   EXPECT_CALL(mock_paced_sender_,
               InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _))
       .Times(kNumPayloadSizes);
-  EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
+  EXPECT_CALL(mock_rtc_event_log_, LogOutgoingRtpHeader(_, _))
       .Times(kNumPayloadSizes);
 
   // Send 10 packets of increasing size.
@@ -827,8 +822,7 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
     fake_clock_.AdvanceTimeMilliseconds(33);
   }
 
-  EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
+  EXPECT_CALL(mock_rtc_event_log_, LogOutgoingRtpHeader(_, _))
       .Times(::testing::AtLeast(4));
 
   // The amount of padding to send it too small to send a payload packet.
@@ -924,9 +918,7 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
                                                kFlexfecSsrc, _, _, _, false))
       .WillOnce(testing::SaveArg<2>(&flexfec_seq_num));
   SendGenericPayload();
-  EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
-      .Times(2);
+  EXPECT_CALL(mock_rtc_event_log_, LogOutgoingRtpHeader(_, _)).Times(2);
   EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum,
                                             fake_clock_.TimeInMilliseconds(),
                                             false, PacedPacketInfo()));
@@ -1000,9 +992,7 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
       kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs, kPayloadData,
       sizeof(kPayloadData), nullptr, &video_header, nullptr));
 
-  EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
-      .Times(1);
+  EXPECT_CALL(mock_rtc_event_log_, LogOutgoingRtpHeader(_, _)).Times(1);
   EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum,
                                             fake_clock_.TimeInMilliseconds(),
                                             false, PacedPacketInfo()));
@@ -1025,9 +1015,7 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
       kVideoFrameKey, kPayloadType, kTimestamp + 1, kCaptureTimeMs + 1,
       kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr));
 
-  EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
-      .Times(2);
+  EXPECT_CALL(mock_rtc_event_log_, LogOutgoingRtpHeader(_, _)).Times(2);
   EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum + 1,
                                             fake_clock_.TimeInMilliseconds(),
                                             false, PacedPacketInfo()));
@@ -1073,9 +1061,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
   params.fec_mask_type = kFecMaskRandom;
   rtp_sender_->SetFecParameters(params, params);
 
-  EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
-      .Times(2);
+  EXPECT_CALL(mock_rtc_event_log_, LogOutgoingRtpHeader(_, _)).Times(2);
   SendGenericPayload();
   ASSERT_EQ(2, transport_.packets_sent());
   const RtpPacketReceived& media_packet = transport_.sent_packets_[0];