Disable pacer disabling.

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 89 matching lines @@
   std::vector<rtc::Buffer*> sent_packets_;
 };
 
 }  // namespace
 
 class MockRtpPacketSender : public RtpPacketSender {
  public:
   MockRtpPacketSender() {}
   virtual ~MockRtpPacketSender() {}
 
-  MOCK_METHOD6(SendPacket,
-               bool(Priority priority,
+  MOCK_METHOD6(InsertPacket,
+               void(Priority priority,
                     uint32_t ssrc,
                     uint16_t sequence_number,
                     int64_t capture_time_ms,
                     size_t bytes,
                     bool retransmission));
 };
 
 class RtpSenderTest : public ::testing::Test {
  protected:
   RtpSenderTest()
       : fake_clock_(kStartTime),
         mock_paced_sender_(),
         rtp_sender_(),
         payload_(kPayload),
         transport_(),
         kMarkerBit(true) {
-    EXPECT_CALL(mock_paced_sender_,
-        SendPacket(_, _, _, _, _, _)).WillRepeatedly(testing::Return(true));
+    EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
+        .WillRepeatedly(testing::Return());
   }
 
-  void SetUp() override {
+  void SetUp() override { SetUpRtpSender(true); }
+
+  void SetUpRtpSender(bool pacer) {
     rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
-                                    &mock_paced_sender_, nullptr, nullptr,
-                                    nullptr, nullptr, nullptr));
+                                    pacer ? &mock_paced_sender_ : nullptr,
+                                    nullptr, nullptr, nullptr, nullptr,
+                                    nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 
   SimulatedClock fake_clock_;
   MockRtpPacketSender mock_paced_sender_;
   rtc::scoped_ptr<RTPSender> rtp_sender_;
   int payload_;
   LoopbackTransportTest transport_;
   const bool kMarkerBit;
   uint8_t packet_[kMaxPacketLength];
@@ skipping 21 matching lines @@
                                                      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));
   }
 };
 
+// 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:
   virtual void SetUp() override {
-    RtpSenderTest::SetUp();
+    // TODO(pbos): Set up to use pacer.
+    SetUpRtpSender(false);
     rtp_sender_video_.reset(
         new RTPSenderVideo(&fake_clock_, rtp_sender_.get()));
   }
   rtc::scoped_ptr<RTPSenderVideo> rtp_sender_video_;
 
   void VerifyCVOPacket(uint8_t* data,
                        size_t len,
                        bool expect_cvo,
                        RtpHeaderExtensionMap* map,
                        uint16_t seq_num,
@@ skipping 15 matching lines @@
     EXPECT_EQ(seq_num, rtp_header.sequenceNumber);
     EXPECT_EQ(kTimestamp, rtp_header.timestamp);
     EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.ssrc);
     EXPECT_EQ(0, rtp_header.numCSRCs);
     EXPECT_EQ(0U, rtp_header.paddingLength);
     EXPECT_EQ(ConvertVideoRotationToCVOByte(rotation),
               rtp_header.extension.videoRotation);
   }
 };
 
-TEST_F(RtpSenderTest, RegisterRtpTransmissionTimeOffsetHeaderExtension) {
+TEST_F(RtpSenderTestWithoutPacer,
+       RegisterRtpTransmissionTimeOffsetHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(kRtpOneByteHeaderLength + kTransmissionTimeOffsetLength,
             rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
       kRtpExtensionTransmissionTimeOffset));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, RegisterRtpAbsoluteSendTimeHeaderExtension) {
+TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAbsoluteSendTimeHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kAbsoluteSendTimeLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, RegisterRtpAudioLevelHeaderExtension) {
+TEST_F(RtpSenderTestWithoutPacer, RegisterRtpAudioLevelHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAudioLevel, kAudioLevelExtensionId));
   EXPECT_EQ(
       RtpUtility::Word32Align(kRtpOneByteHeaderLength + kAudioLevelLength),
       rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
       kRtpExtensionAudioLevel));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, RegisterRtpHeaderExtensions) {
+TEST_F(RtpSenderTestWithoutPacer, RegisterRtpHeaderExtensions) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kTransmissionTimeOffsetLength),
             rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
   EXPECT_EQ(RtpUtility::Word32Align(kRtpOneByteHeaderLength +
                                     kTransmissionTimeOffsetLength +
@@ skipping 29 matching lines @@
   EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(
       kRtpExtensionAudioLevel));
   EXPECT_EQ(
       RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength),
       rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(
       0, rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionVideoRotation));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, RegisterRtpVideoRotationHeaderExtension) {
+TEST_F(RtpSenderTestWithoutPacer, RegisterRtpVideoRotationHeaderExtension) {
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 
   EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
   EXPECT_EQ(
       RtpUtility::Word32Align(kRtpOneByteHeaderLength + kVideoRotationLength),
       rtp_sender_->RtpHeaderExtensionTotalLength());
   EXPECT_EQ(
       0, rtp_sender_->DeregisterRtpHeaderExtension(kRtpExtensionVideoRotation));
   EXPECT_EQ(0u, rtp_sender_->RtpHeaderExtensionTotalLength());
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacket) {
+TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacket) {
   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);
   ASSERT_FALSE(rtp_parser.RTCP());
   VerifyRTPHeaderCommon(rtp_header);
   EXPECT_EQ(length, rtp_header.headerLength);
   EXPECT_FALSE(rtp_header.extension.hasTransmissionTimeOffset);
   EXPECT_FALSE(rtp_header.extension.hasAbsoluteSendTime);
   EXPECT_FALSE(rtp_header.extension.hasAudioLevel);
   EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
   EXPECT_EQ(0u, rtp_header.extension.absoluteSendTime);
   EXPECT_FALSE(rtp_header.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header.extension.audioLevel);
   EXPECT_EQ(0u, rtp_header.extension.videoRotation);
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
+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(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
             length);
 
   // Verify
@@ skipping 16 matching lines @@
   webrtc::RTPHeader rtp_header2;
   const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
   EXPECT_EQ(length, rtp_header2.headerLength);
   EXPECT_FALSE(rtp_header2.extension.hasTransmissionTimeOffset);
   EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacketWithNegativeTransmissionOffsetExtension) {
+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(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
             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(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
+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(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
             length);
 
   // Verify
@@ skipping 16 matching lines @@
   const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
   EXPECT_EQ(length, rtp_header2.headerLength);
   EXPECT_FALSE(rtp_header2.extension.hasAbsoluteSendTime);
   EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
 }
 
 // Test CVO header extension is only set when marker bit is true.
-TEST_F(RtpSenderTest, BuildRTPPacketWithVideoRotation_MarkerBit) {
+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));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
             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(RtpSenderTest, DISABLED_BuildRTPPacketWithVideoRotation_NoMarkerBit) {
+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));
   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(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) {
+TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAudioLevel, kAudioLevelExtensionId));
 
   size_t length = static_cast<size_t>(rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, 0));
   ASSERT_EQ(kRtpHeaderSize + rtp_sender_->RtpHeaderExtensionTotalLength(),
             length);
 
   // Verify
   webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
@@ skipping 20 matching lines @@
   const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
   EXPECT_EQ(length, rtp_header2.headerLength);
   EXPECT_FALSE(rtp_header2.extension.hasAudioLevel);
   EXPECT_FALSE(rtp_header2.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
 }
 
-TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
+TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithHeaderExtensions) {
   EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
   EXPECT_EQ(0, rtp_sender_->SetAbsoluteSendTime(kAbsoluteSendTime));
   EXPECT_EQ(0,
             rtp_sender_->SetTransportSequenceNumber(kTransportSequenceNumber));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAudioLevel, kAudioLevelExtensionId));
@@ skipping 51 matching lines @@
   EXPECT_FALSE(rtp_header2.extension.hasTransportSequenceNumber);
 
   EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
   EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
   EXPECT_FALSE(rtp_header2.extension.voiceActivity);
   EXPECT_EQ(0u, rtp_header2.extension.audioLevel);
   EXPECT_EQ(0u, rtp_header2.extension.transportSequenceNumber);
 }
 
 TEST_F(RtpSenderTest, TrafficSmoothingWithExtensions) {
-  EXPECT_CALL(mock_paced_sender_,
-              SendPacket(RtpPacketSender::kNormalPriority, _, kSeqNum, _, _, _))
-      .WillOnce(testing::Return(false));
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
+                                               _, kSeqNum, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
   rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int rtp_length_int = rtp_sender_->BuildRTPheader(
       packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);
@@ skipping 27 matching lines @@
   ASSERT_TRUE(valid_rtp_header);
 
   // Verify transmission time offset.
   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_F(RtpSenderTest, TrafficSmoothingRetransmits) {
-  EXPECT_CALL(mock_paced_sender_,
-              SendPacket(RtpPacketSender::kNormalPriority, _, kSeqNum, _, _, _))
-      .WillOnce(testing::Return(false));
+  EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
+                                               _, kSeqNum, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionTransmissionTimeOffset, kTransmissionTimeOffsetExtensionId));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
   rtp_sender_->SetTargetBitrate(300000);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   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_);
 
   EXPECT_CALL(mock_paced_sender_,
-              SendPacket(RtpPacketSender::kHighPriority, _, kSeqNum, _, _, _))
-      .WillOnce(testing::Return(false));
+              InsertPacket(RtpPacketSender::kHighPriority, _, kSeqNum, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   const int kStoredTimeInMs = 100;
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
 
   EXPECT_EQ(rtp_length_int, rtp_sender_->ReSendPacket(kSeqNum));
   EXPECT_EQ(0, transport_.packets_sent_);
 
   rtp_sender_->TimeToSendPacket(kSeqNum, capture_time_ms, false);
 
   // Process send bucket. Packet should now be sent.
@@ skipping 16 matching lines @@
   uint64_t expected_send_time =
       ConvertMsToAbsSendTime(fake_clock_.TimeInMilliseconds());
   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_F(RtpSenderTest, SendPadding) {
   // Make all (non-padding) packets go to send queue.
   EXPECT_CALL(mock_paced_sender_,
-              SendPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
-      .WillRepeatedly(testing::Return(false));
+              InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   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.
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
@@ skipping 98 matching lines @@
 
 TEST_F(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport, nullptr,
                                   &mock_paced_sender_, nullptr, nullptr,
                                   nullptr, nullptr, nullptr));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
   // Make all packets go through the pacer.
   EXPECT_CALL(mock_paced_sender_,
-              SendPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
-      .WillRepeatedly(testing::Return(false));
+              InsertPacket(RtpPacketSender::kNormalPriority, _, _, _, _, _))
+      .WillRepeatedly(testing::Return());
 
   uint16_t seq_num = kSeqNum;
   rtp_sender_->SetStorePacketsStatus(true, 10);
   int32_t rtp_header_len = kRtpHeaderSize;
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime, kAbsoluteSendTimeExtensionId));
   rtp_header_len += 4;  // 4 bytes extension.
   rtp_header_len += 4;  // 4 extra bytes common to all extension headers.
 
   rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
@@ skipping 32 matching lines @@
   EXPECT_CALL(transport, SendRtp(_, kPayloadSizes[kNumPayloadSizes - 1] +
                                         rtp_header_len + kRtxHeaderSize,
                                  _))
       .WillOnce(testing::Return(true));
   EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
       .WillOnce(testing::Return(true));
   EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize,
             rtp_sender_->TimeToSendPadding(999));
 }
 
-TEST_F(RtpSenderTest, SendGenericVideo) {
+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));
@@ skipping 105 matching lines @@
       total_stats_ = total_stats;
       retransmit_stats_ = retransmit_stats;
     }
 
     uint32_t num_calls_;
     uint32_t ssrc_;
     BitrateStatistics total_stats_;
     BitrateStatistics retransmit_stats_;
   } callback;
   rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
-                                  &mock_paced_sender_, nullptr, nullptr,
-                                  &callback, nullptr, nullptr));
+                                  nullptr, nullptr, nullptr, &callback, nullptr,
+                                  nullptr));
 
   // Simulate kNumPackets sent with kPacketInterval ms intervals.
   const uint32_t kNumPackets = 15;
   const uint32_t kPacketInterval = 20;
   // Overhead = 12 bytes RTP header + 1 byte generic header.
   const uint32_t kPacketOverhead = 13;
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   const uint8_t payload_type = 127;
   ASSERT_EQ(
@@ skipping 37 matching lines @@
   rtp_sender_.reset();
 }
 
 class RtpSenderAudioTest : public RtpSenderTest {
  protected:
   RtpSenderAudioTest() {}
 
   void SetUp() override {
     payload_ = kAudioPayload;
     rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
-                                    &mock_paced_sender_, nullptr, nullptr,
-                                    nullptr, nullptr, nullptr));
+                                    nullptr, nullptr, nullptr, nullptr, nullptr,
+                                    nullptr));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 };
 
-TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
+TEST_F(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   class TestCallback : public StreamDataCountersCallback {
    public:
     TestCallback()
       : StreamDataCountersCallback(), ssrc_(0), counters_() {}
     virtual ~TestCallback() {}
 
     void DataCountersUpdated(const StreamDataCounters& counters,
                              uint32_t ssrc) override {
       ssrc_ = ssrc;
       counters_ = counters;
@@ skipping 195 matching lines @@
   ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
                                              capture_time_ms + 4000, 0, nullptr,
                                              0, 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(RtpSenderTest, BytesReportedCorrectly) {
+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));
@@ skipping 32 matching lines @@
   EXPECT_EQ(rtx_stats.transmitted.TotalBytes(),
       rtx_stats.transmitted.payload_bytes +
       rtx_stats.transmitted.header_bytes +
       rtx_stats.transmitted.padding_bytes);
 
   EXPECT_EQ(transport_.total_bytes_sent_,
             rtp_stats.transmitted.TotalBytes() +
             rtx_stats.transmitted.TotalBytes());
 }
 
-TEST_F(RtpSenderTest, RespectsNackBitrateLimit) {
+TEST_F(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
   const int32_t kPacketSize = 1400;
   const int32_t kNumPackets = 30;
 
   rtp_sender_->SetStorePacketsStatus(true, kNumPackets);
   // Set bitrate (in kbps) to fit kNumPackets á kPacketSize bytes in one second.
   rtp_sender_->SetTargetBitrate(kNumPackets * kPacketSize * 8);
   const uint16_t kStartSequenceNumber = rtp_sender_->SequenceNumber();
   std::list<uint16_t> sequence_numbers;
   for (int32_t i = 0; i < kNumPackets; ++i) {
     sequence_numbers.push_back(kStartSequenceNumber + i);
@@ skipping 40 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