Style cleanups in RtpSender.

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 167 matching lines @@
     EXPECT_EQ(payload_, rtp_header.payloadType);
     EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber);
     EXPECT_EQ(kTimestamp, rtp_header.timestamp);
     EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.ssrc);
     EXPECT_EQ(number_of_csrcs, rtp_header.numCSRCs);
     EXPECT_EQ(0U, rtp_header.paddingLength);
   }
 
   void SendPacket(int64_t capture_time_ms, int payload_length) {
     uint32_t timestamp = capture_time_ms * 90;
-    int32_t rtp_length = rtp_sender_->BuildRTPheader(
+    int32_t rtp_length = rtp_sender_->BuildRtpHeader(
         packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
     ASSERT_GE(rtp_length, 0);
 
     // Packet should be stored in a send bucket.
     EXPECT_EQ(0, rtp_sender_->SendToNetwork(
                      packet_, payload_length, rtp_length, capture_time_ms,
                      kAllowRetransmission, RtpPacketSender::kNormalPriority));
   }
 
   void SendGenericPayload() {
     const uint8_t kPayload[] = {47, 11, 32, 93, 89};
     const uint32_t kTimestamp = 1234;
     const uint8_t kPayloadType = 127;
     const int64_t kCaptureTimeMs = fake_clock_.TimeInMilliseconds();
     char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
     EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
                                               0, 1500));
 
     EXPECT_EQ(0, rtp_sender_->SendOutgoingData(
                      kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs,
-                     kPayload, sizeof(kPayload), nullptr));
+                     kPayload, sizeof(kPayload), nullptr, nullptr));
   }
 };
 
 // TODO(pbos): Move tests over from WithoutPacer to RtpSenderTest as this is our
 // default code path.
 class RtpSenderTestWithoutPacer : public RtpSenderTest {
  public:
   void SetUp() override { SetUpRtpSender(false); }
 };
 
 class RtpSenderVideoTest : public RtpSenderTest {
  protected:
   void SetUp() override {
     // TODO(pbos): Set up to use pacer.
     SetUpRtpSender(false);
     rtp_sender_video_.reset(
         new RTPSenderVideo(&fake_clock_, rtp_sender_.get()));
   }
   std::unique_ptr<RTPSenderVideo> rtp_sender_video_;
 
   void VerifyCVOPacket(uint8_t* data,
                        size_t len,
                        bool expect_cvo,
                        RtpHeaderExtensionMap* map,
                        uint16_t seq_num,
                        VideoRotation rotation) {
     webrtc::RtpUtility::RtpHeaderParser rtp_parser(data, len);
 
     webrtc::RTPHeader rtp_header;
-    size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+    size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
         packet_, kPayload, expect_cvo /* marker_bit */, kTimestamp, 0));
     if (expect_cvo) {
       ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(),
                 length);
     } else {
       ASSERT_EQ(kRtpHeaderSize, length);
     }
     ASSERT_TRUE(rtp_parser.Parse(&rtp_header, map));
     ASSERT_FALSE(rtp_parser.RTCP());
     EXPECT_EQ(payload_, rtp_header.payloadType);
@@ skipping 106 matching lines @@
   EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
   EXPECT_EQ(
       RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength),
       rtp_sender_->RtpHeaderExtensionLength());
   EXPECT_EQ(
       0, rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionVideoRotation));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionLength());
 }
 
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) {
-  size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+  size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize, length);
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
   const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, nullptr);
 
   ASSERT_TRUE(valid_rtp_header);
@@ skipping 10 matching lines @@
   EXPECT_EQ(0u, rtp_header.extension.videoRotation);
 }
 
 TEST_F(RtpSenderTestWithoutPacer,
        BuildRTPPacketWithTransmissionOffsetExtension) {
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
 
-  size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+  size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
@@ skipping 18 matching lines @@
 }
 
 TEST_F(RtpSenderTestWithoutPacer,
        BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
   const int kNegTimeOffset = -500;
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kNegTimeOffset));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransmissionTimeOffset,
                    kTransmissionTimeOffsetExtensionId));
 
-  size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+  size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionTransmissionTimeOffset,
                kTransmissionTimeOffsetExtensionId);
   const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
 
   ASSERT_TRUE(valid_rtp_header);
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header);
   EXPECT_EQ(length, rtp_header.headerLength);
   EXPECT_TRUE(rtp_header.extension.hasTransmissionTimeOffset);
   EXPECT_EQ(kNegTimeOffset, rtp_header.extension.transmissionTimeOffset);
 }
 
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
   EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
 
-  size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+  size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId);
   const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map);
@@ skipping 64 matching lines @@
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
   rtp_sender_->SetVideoRotation(kRotation);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
   EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionVideoRotation, kVideoRotationExtensionId);
 
   size_t length = static_cast<size_t>(
-      rtp_sender_->BuildRTPheader(packet_, kPayload, true, kTimestamp, 0));
+      rtp_sender_->BuildRtpHeader(packet_, kPayload, true, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header);
   EXPECT_EQ(length, rtp_header.headerLength);
   EXPECT_TRUE(rtp_header.extension.hasVideoRotation);
   EXPECT_EQ(ConvertVideoRotationToCVOByte(kRotation),
             rtp_header.extension.videoRotation);
 }
 
 // Test CVO header extension is not set when marker bit is false.
 TEST_F(RtpSenderTestWithoutPacer,
        DISABLED_BuildRTPPacketWithVideoRotation_NoMarkerBit) {
   rtp_sender_->SetVideoRotation(kRotation);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
   EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionVideoRotation, kVideoRotationExtensionId);
 
   size_t length = static_cast<size_t>(
-      rtp_sender_->BuildRTPheader(packet_, kPayload, false, kTimestamp, 0));
+      rtp_sender_->BuildRtpHeader(packet_, kPayload, false, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize, length);
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header, &map));
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header, false);
   EXPECT_EQ(length, rtp_header.headerLength);
   EXPECT_FALSE(rtp_header.extension.hasVideoRotation);
 }
 
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
                                                        kAudioLevelExtensionId));
 
-  size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+  size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
   // Updating audio level is done in RTPSenderAudio, so simulate it here.
   rtp_parser.Parse(&rtp_header);
   rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
@@ skipping 22 matching lines @@
   EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
 }
 
 TEST_F(RtpSenderTestWithoutPacer,
        BuildRTPPacketWithCSRCAndAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
                                                        kAudioLevelExtensionId));
   std::vector<uint32_t> csrcs;
   csrcs.push_back(0x23456789);
   rtp_sender_->SetCsrcs(csrcs);
-  size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+  size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
   // Updating audio level is done in RTPSenderAudio, so simulate it here.
   rtp_parser.Parse(&rtp_header);
   EXPECT_TRUE(rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true,
                                             kAudioLevel));
@@ skipping 23 matching lines @@
                    kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(
       0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                                  kAbsoluteSendTimeExtensionId));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
                                                        kAudioLevelExtensionId));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
                    kTransportSequenceNumberExtensionId));
 
-  size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
+  size_t length = static_cast<size_t>(rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionLength(), length);
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
   webrtc::RTPHeader rtp_header;
 
   // Updating audio level is done in RTPSenderAudio, so simulate it here.
   rtp_parser.Parse(&rtp_header);
   rtp_sender_->UpdateAudioLevel(packet_, length, rtp_header, true, kAudioLevel);
@@ skipping 48 matching lines @@
               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();
-  int rtp_length_int = rtp_sender_->BuildRTPheader(
+  int rtp_length_int = rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);
   ASSERT_NE(-1, rtp_length_int);
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
                                           RtpPacketSender::kNormalPriority));
 
   EXPECT_EQ(0, transport_.packets_sent_);
@@ skipping 32 matching lines @@
               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();
-  int rtp_length_int = rtp_sender_->BuildRTPheader(
+  int rtp_length_int = rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);
   ASSERT_NE(-1, rtp_length_int);
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
                                           RtpPacketSender::kNormalPriority));
 
   EXPECT_EQ(0, transport_.packets_sent_);
@@ skipping 60 matching lines @@
   std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
                                          kTransmissionTimeOffsetExtensionId);
   rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
                                          kAbsoluteSendTimeExtensionId);
   webrtc::RTPHeader rtp_header;
 
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
-  int rtp_length_int = rtp_sender_->BuildRTPheader(
+  int rtp_length_int = rtp_sender_->BuildRtpHeader(
       packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);
   const uint32_t media_packet_timestamp = timestamp;
   ASSERT_NE(-1, rtp_length_int);
   size_t rtp_length = static_cast<size_t>(rtp_length_int);
 
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
                                           RtpPacketSender::kNormalPriority));
 
@@ skipping 37 matching lines @@
     EXPECT_EQ(offset, rtp_header.extension.transmissionTimeOffset);
     uint64_t expected_send_time =
         ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
     EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime);
     fake_clock_.AdvanceTimeMilliseconds(kPaddingPeriodMs);
     timestamp += 90 * kPaddingPeriodMs;
   }
 
   // Send a regular video packet again.
   capture_time_ms = fake_clock_.TimeInMilliseconds();
-  rtp_length_int = rtp_sender_->BuildRTPheader(packet_, kPayload, kMarkerBit,
+  rtp_length_int = rtp_sender_->BuildRtpHeader(packet_, kPayload, kMarkerBit,
                                                timestamp, capture_time_ms);
   ASSERT_NE(-1, rtp_length_int);
   rtp_length = static_cast<size_t>(rtp_length_int);
 
   EXPECT_CALL(mock_paced_sender_,
               InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _));
 
   // Packet should be stored in a send bucket.
   EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
                                           capture_time_ms, kAllowRetransmission,
@@ skipping 154 matching lines @@
 }
 
 TEST_F(RtpSenderTestWithoutPacer, SendGenericVideo) {
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
   // Send keyframe
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   const uint8_t* payload_data =
       GetPayloadData(rtp_header, transport_.last_sent_packet_);
   uint8_t generic_header = *payload_data++;
 
   ASSERT_EQ(sizeof(payload) + sizeof(generic_header),
             GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
 
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
 
   EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
 
   // Send delta frame
   payload[0] = 13;
   payload[1] = 42;
   payload[4] = 13;
 
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
-                                             1234, 4321, payload,
-                                             sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+                   kVideoFrameDelta, payload_type, 1234, 4321, payload,
+                   sizeof(payload), nullptr, nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
                                           transport_.last_sent_packet_len_);
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   payload_data = GetPayloadData(rtp_header, transport_.last_sent_packet_);
   generic_header = *payload_data++;
 
   EXPECT_FALSE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
@@ skipping 30 matching lines @@
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
       .Times(::testing::AtLeast(2));
 
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr));
 
   EXPECT_EQ(1U, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   EXPECT_EQ(1, callback.frame_counts_.key_frames);
   EXPECT_EQ(0, callback.frame_counts_.delta_frames);
 
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
-                                             1234, 4321, payload,
-                                             sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+                   kVideoFrameDelta, payload_type, 1234, 4321, payload,
+                   sizeof(payload), nullptr, nullptr));
 
   EXPECT_EQ(2U, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   EXPECT_EQ(1, callback.frame_counts_.key_frames);
   EXPECT_EQ(1, callback.frame_counts_.delta_frames);
 
   rtp_sender_.reset();
 }
 
 TEST_F(RtpSenderTest, BitrateCallbacks) {
@@ skipping 41 matching lines @@
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   // Initial process call so we get a new time window.
   rtp_sender_->ProcessBitrate();
 
   // Send a few frames.
   for (uint32_t i = 0; i < kNumPackets; ++i) {
-    ASSERT_EQ(0,
-              rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
-                                            4321, payload, sizeof(payload), 0));
+    ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+                     kVideoFrameKey, payload_type, 1234, 4321, payload,
+                     sizeof(payload), nullptr, nullptr));
     fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
   }
 
   rtp_sender_->ProcessBitrate();
 
   // We get one call for every stats updated, thus two calls since both the
   // stream stats and the retransmit stats are updated once.
   EXPECT_EQ(2u, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   const uint32_t kTotalPacketSize = kPacketOverhead + sizeof(payload);
@@ skipping 58 matching lines @@
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   rtp_sender_->RegisterRtpStatisticsCallback(&callback);
 
   // Send a frame.
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr));
   StreamDataCounters expected;
   expected.transmitted.payload_bytes = 6;
   expected.transmitted.header_bytes = 12;
   expected.transmitted.padding_bytes = 0;
   expected.transmitted.packets = 1;
   expected.retransmitted.payload_bytes = 0;
   expected.retransmitted.header_bytes = 0;
   expected.retransmitted.padding_bytes = 0;
   expected.retransmitted.packets = 0;
   expected.fec.packets = 0;
@@ skipping 19 matching lines @@
   expected.transmitted.packets = 3;
   callback.Matches(ssrc, expected);
 
   // Send FEC.
   rtp_sender_->SetGenericFECStatus(true, kRedPayloadType, kUlpfecPayloadType);
   FecProtectionParams fec_params;
   fec_params.fec_mask_type = kFecMaskRandom;
   fec_params.fec_rate = 1;
   fec_params.max_fec_frames = 1;
   rtp_sender_->SetFecParameters(&fec_params, &fec_params);
-  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
-                                             1234, 4321, payload,
-                                             sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
+                   kVideoFrameDelta, payload_type, 1234, 4321, payload,
+                   sizeof(payload), nullptr, nullptr));
   expected.transmitted.payload_bytes = 40;
   expected.transmitted.header_bytes = 60;
   expected.transmitted.packets = 5;
   expected.fec.packets = 1;
   callback.Matches(ssrc, expected);
 
   rtp_sender_->RegisterRtpStatisticsCallback(nullptr);
 }
 
 TEST_F(RtpSenderAudioTest, SendAudio) {
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   const uint8_t* payload_data =
       GetPayloadData(rtp_header, transport_.last_sent_packet_);
 
   ASSERT_EQ(sizeof(payload),
             GetPayloadDataLength(rtp_header, transport_.last_sent_packet_len_));
 
   EXPECT_EQ(0, memcmp(payload, payload_data, sizeof(payload)));
 }
 
 TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
                                                        kAudioLevelExtensionId));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
-                                       payload, sizeof(payload), nullptr));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr));
 
   RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
                                          transport_.last_sent_packet_len_);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser.Parse(&rtp_header));
 
   const uint8_t* payload_data =
       GetPayloadData(rtp_header, transport_.last_sent_packet_);
 
   ASSERT_EQ(sizeof(payload),
@@ skipping 30 matching lines @@
   ASSERT_EQ(
       0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   // DTMF event key=9, duration=500 and attenuationdB=10
   rtp_sender_->SendTelephoneEvent(9, 500, 10);
   // During start, it takes the starting timestamp as last sent timestamp.
   // The duration is calculated as the difference of current and last sent
   // timestamp. So for first call it will skip since the duration is zero.
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms, 0, nullptr, 0,
-                                             nullptr));
+                                             nullptr, nullptr));
   // DTMF Sample Length is (Frequency/1000) * Duration.
   // So in this case, it is (8000/1000) * 500 = 4000.
   // Sending it as two packets.
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms + 2000, 0, nullptr,
-                                             0, nullptr));
+                                             0, nullptr, nullptr));
   std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(
       webrtc::RtpHeaderParser::Create());
   ASSERT_TRUE(rtp_parser.get() != nullptr);
   webrtc::RTPHeader rtp_header;
   ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
                                 transport_.last_sent_packet_len_, &rtp_header));
   // Marker Bit should be set to 1 for first packet.
   EXPECT_TRUE(rtp_header.markerBit);
 
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
                                              capture_time_ms + 4000, 0, nullptr,
-                                             0, nullptr));
+                                             0, nullptr, nullptr));
   ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
                                 transport_.last_sent_packet_len_, &rtp_header));
   // Marker Bit should be set to 0 for rest of the packets.
   EXPECT_FALSE(rtp_header.markerBit);
 }
 
 TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
   const char* kPayloadName = "GENERIC";
   const uint8_t kPayloadType = 127;
   rtp_sender_->SetSSRC(1234);
   rtp_sender_->SetRtxSsrc(4321);
   rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
   rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
 
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
-  ASSERT_EQ(
-      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234, 4321,
-                                       payload, sizeof(payload), 0));
+  ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, kPayloadType, 1234,
+                                             4321, payload, sizeof(payload),
+                                             nullptr, nullptr));
 
   // Will send 2 full-size padding packets.
   rtp_sender_->TimeToSendPadding(1, PacketInfo::kNotAProbe);
   rtp_sender_->TimeToSendPadding(1, PacketInfo::kNotAProbe);
 
   StreamDataCounters rtp_stats;
   StreamDataCounters rtx_stats;
   rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
 
   // Payload + 1-byte generic header.
@@ skipping 76 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc