Removed RtcEventLog deps to call:call_interfaces.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 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 446 matching lines @@
   uint16_t transport_seq_no;
   EXPECT_TRUE(packet.GetExtension<TransportSequenceNumber>(&transport_seq_no));
   EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
   EXPECT_EQ(transport_.last_packet_id_, transport_seq_no);
 }
 
 TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                kSsrc, kSeqNum, _, _, _));
   EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _));
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   auto packet =
@@ skipping 24 matching lines @@
   EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset);
   uint64_t expected_send_time =
       ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
   EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
 }
 
 TEST_P(RtpSenderTest, TrafficSmoothingRetransmits) {
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                kSsrc, kSeqNum, _, _, _));
   EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _));
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   auto packet =
@@ skipping 33 matching lines @@
   EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
 }
 
 // This test sends 1 regular video packet, then 4 padding packets, and then
 // 1 more regular packet.
 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, _, _, _, _))
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(1 + 4 + 1);
 
   uint16_t seq_num = kSeqNum;
   uint32_t timestamp = kTimestamp;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   size_t rtp_header_len = kRtpHeaderSize;
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
@@ skipping 180 matching lines @@
   rtp_sender_->SetRtxSsrc(1234);
 
   const size_t kNumPayloadSizes = 10;
   const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700,
                                                   750, 800, 850, 900, 950};
   // Expect all packets go through the pacer.
   EXPECT_CALL(mock_paced_sender_,
               InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _))
       .Times(kNumPayloadSizes);
   EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _))
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(kNumPayloadSizes);
 
   // Send 10 packets of increasing size.
   for (size_t i = 0; i < kNumPayloadSizes; ++i) {
     int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
     EXPECT_CALL(transport, SendRtp(_, _, _)).WillOnce(testing::Return(true));
     SendPacket(capture_time_ms, kPayloadSizes[i]);
     rtp_sender_->TimeToSendPacket(kSsrc, seq_num++, capture_time_ms, false,
                                   PacedPacketInfo());
     fake_clock_.AdvanceTimeMilliseconds(33);
   }
 
   EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _))
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(::testing::AtLeast(4));
 
   // The amount of padding to send it too small to send a payload packet.
   EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
       .WillOnce(testing::Return(true));
   EXPECT_EQ(kMaxPaddingSize,
             rtp_sender_->TimeToSendPadding(49, PacedPacketInfo()));
 
   EXPECT_CALL(transport,
               SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize, _))
@@ skipping 76 matching lines @@
 
   EXPECT_CALL(mock_paced_sender_,
               InsertPacket(RtpPacketSender::kLowPriority, kMediaSsrc, kSeqNum,
                            _, _, false));
   uint16_t flexfec_seq_num;
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kLowPriority,
                                                kFlexfecSsrc, _, _, _, false))
       .WillOnce(testing::SaveArg<2>(&flexfec_seq_num));
   SendGenericPayload();
   EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _))
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(2);
   EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum,
                                             fake_clock_.TimeInMilliseconds(),
                                             false, PacedPacketInfo()));
   EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kFlexfecSsrc, flexfec_seq_num,
                                             fake_clock_.TimeInMilliseconds(),
                                             false, PacedPacketInfo()));
   ASSERT_EQ(2, transport_.packets_sent());
   const RtpPacketReceived& media_packet = transport_.sent_packets_[0];
   EXPECT_EQ(kMediaPayloadType, media_packet.PayloadType());
@@ skipping 27 matching lines @@
   rtp_sender_->SetSendPayloadType(kMediaPayloadType);
 
   // Parameters selected to generate a single FEC packet per media packet.
   FecProtectionParams params;
   params.fec_rate = 15;
   params.max_fec_frames = 1;
   params.fec_mask_type = kFecMaskRandom;
   rtp_sender_->SetFecParameters(params, params);
 
   EXPECT_CALL(mock_rtc_event_log_,
-              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _, _))
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(2);
   SendGenericPayload();
   ASSERT_EQ(2, transport_.packets_sent());
   const RtpPacketReceived& media_packet = transport_.sent_packets_[0];
   EXPECT_EQ(kMediaPayloadType, media_packet.PayloadType());
   EXPECT_EQ(kMediaSsrc, media_packet.Ssrc());
   const RtpPacketReceived& flexfec_packet = transport_.sent_packets_[1];
   EXPECT_EQ(kFlexfecPayloadType, flexfec_packet.PayloadType());
   EXPECT_EQ(kFlexfecSsrc, flexfec_packet.Ssrc());
 }
@@ skipping 604 matching lines @@
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderTestWithoutPacer,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderVideoTest,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderAudioTest,
                         ::testing::Bool());
 }  // namespace webrtc