1 files changed, 673 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/pacing/packet_router_unittest.cc b/third_party/libwebrtc/modules/pacing/packet_router_unittest.cc
new file mode 100644
index 0000000000..fc26922850
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/packet_router_unittest.cc
@@ -0,0 +1,673 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#include "modules/pacing/packet_router.h"
+
+#include <cstddef>
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fake_clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// TODO(eladalon): Restructure and/or replace the existing monolithic tests
+// (only some of the test are monolithic) according to the new
+// guidelines - small tests for one thing at a time.
+// (I'm not removing any tests during CL, so as to demonstrate no regressions.)
+
+namespace {
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::AtLeast;
+using ::testing::Field;
+using ::testing::Gt;
+using ::testing::Le;
+using ::testing::NiceMock;
+using ::testing::Property;
+using ::testing::Return;
+using ::testing::SaveArg;
+
+constexpr int kProbeMinProbes = 5;
+constexpr int kProbeMinBytes = 1000;
+
+}  // namespace
+
+class PacketRouterTest : public ::testing::Test {
+ public:
+  PacketRouterTest() {
+    extension_manager.Register<TransportSequenceNumber>(/*id=*/1);
+  }
+
+ protected:
+  std::unique_ptr<RtpPacketToSend> BuildRtpPacket(uint32_t ssrc) {
+    std::unique_ptr<RtpPacketToSend> packet =
+        std::make_unique<RtpPacketToSend>(&extension_manager);
+    packet->SetSsrc(ssrc);
+    return packet;
+  }
+
+  PacketRouter packet_router_;
+  RtpHeaderExtensionMap extension_manager;
+};
+
+TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_GeneratePadding) {
+  constexpr DataSize bytes = DataSize::Bytes(300);
+  const PacedPacketInfo paced_info(1, kProbeMinProbes, kProbeMinBytes);
+
+  EXPECT_TRUE(packet_router_.GeneratePadding(bytes).empty());
+}
+
+
+TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_SendRemb) {
+  const std::vector<uint32_t> ssrcs = {1, 2, 3};
+  constexpr uint32_t bitrate_bps = 10000;
+  // Expect not to crash
+  packet_router_.SendRemb(bitrate_bps, ssrcs);
+}
+
+TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_SendTransportFeedback) {
+  std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback;
+  feedback.push_back(std::make_unique<rtcp::TransportFeedback>());
+  // Expect not to crash
+  packet_router_.SendCombinedRtcpPacket(std::move(feedback));
+}
+
+TEST_F(PacketRouterTest, GeneratePaddingPrioritizesRtx) {
+  // Two RTP modules. The first (prioritized due to rtx) isn't sending media so
+  // should not be called.
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 4567;
+
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  ON_CALL(rtp_1, RtxSendStatus()).WillByDefault(Return(kRtxRedundantPayloads));
+  ON_CALL(rtp_1, SSRC()).WillByDefault(Return(kSsrc1));
+  ON_CALL(rtp_1, SupportsPadding).WillByDefault(Return(false));
+
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+  ON_CALL(rtp_2, RtxSendStatus()).WillByDefault(Return(kRtxOff));
+  ON_CALL(rtp_2, SSRC()).WillByDefault(Return(kSsrc2));
+  ON_CALL(rtp_2, SupportsPadding).WillByDefault(Return(true));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddSendRtpModule(&rtp_2, false);
+
+  const size_t kPaddingSize = 123;
+  const size_t kExpectedPaddingPackets = 1;
+  EXPECT_CALL(rtp_1, GeneratePadding(_)).Times(0);
+  EXPECT_CALL(rtp_2, GeneratePadding(kPaddingSize))
+      .WillOnce([&](size_t padding_size) {
+        return std::vector<std::unique_ptr<RtpPacketToSend>>(
+            kExpectedPaddingPackets);
+      });
+  auto generated_padding =
+      packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+  EXPECT_EQ(generated_padding.size(), kExpectedPaddingPackets);
+
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+  packet_router_.RemoveSendRtpModule(&rtp_2);
+}
+
+TEST_F(PacketRouterTest, GeneratePaddingPrioritizesVideo) {
+  // Two RTP modules. Neither support RTX, both support padding,
+  // but the first one is for audio and second for video.
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 4567;
+  const size_t kPaddingSize = 123;
+  const size_t kExpectedPaddingPackets = 1;
+
+  auto generate_padding = [&](size_t padding_size) {
+    return std::vector<std::unique_ptr<RtpPacketToSend>>(
+        kExpectedPaddingPackets);
+  };
+
+  NiceMock<MockRtpRtcpInterface> audio_module;
+  ON_CALL(audio_module, RtxSendStatus()).WillByDefault(Return(kRtxOff));
+  ON_CALL(audio_module, SSRC()).WillByDefault(Return(kSsrc1));
+  ON_CALL(audio_module, SupportsPadding).WillByDefault(Return(true));
+  ON_CALL(audio_module, IsAudioConfigured).WillByDefault(Return(true));
+
+  NiceMock<MockRtpRtcpInterface> video_module;
+  ON_CALL(video_module, RtxSendStatus()).WillByDefault(Return(kRtxOff));
+  ON_CALL(video_module, SSRC()).WillByDefault(Return(kSsrc2));
+  ON_CALL(video_module, SupportsPadding).WillByDefault(Return(true));
+  ON_CALL(video_module, IsAudioConfigured).WillByDefault(Return(false));
+
+  // First add only the audio module. Since this is the only choice we have,
+  // padding should be sent on the audio ssrc.
+  packet_router_.AddSendRtpModule(&audio_module, false);
+  EXPECT_CALL(audio_module, GeneratePadding(kPaddingSize))
+      .WillOnce(generate_padding);
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+
+  // Add the video module, this should now be prioritized since we cannot
+  // guarantee that audio packets will be included in the BWE.
+  packet_router_.AddSendRtpModule(&video_module, false);
+  EXPECT_CALL(audio_module, GeneratePadding).Times(0);
+  EXPECT_CALL(video_module, GeneratePadding(kPaddingSize))
+      .WillOnce(generate_padding);
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+
+  // Remove and the add audio module again. Module order shouldn't matter;
+  // video should still be prioritized.
+  packet_router_.RemoveSendRtpModule(&audio_module);
+  packet_router_.AddSendRtpModule(&audio_module, false);
+  EXPECT_CALL(audio_module, GeneratePadding).Times(0);
+  EXPECT_CALL(video_module, GeneratePadding(kPaddingSize))
+      .WillOnce(generate_padding);
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+
+  // Remove and the video module, we should fall back to padding on the
+  // audio module again.
+  packet_router_.RemoveSendRtpModule(&video_module);
+  EXPECT_CALL(audio_module, GeneratePadding(kPaddingSize))
+      .WillOnce(generate_padding);
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingSize));
+
+  packet_router_.RemoveSendRtpModule(&audio_module);
+}
+
+TEST_F(PacketRouterTest, PadsOnLastActiveMediaStream) {
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 4567;
+  const uint16_t kSsrc3 = 8901;
+
+  // First two rtp modules send media and have rtx.
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
+  EXPECT_CALL(rtp_1, SupportsPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_1, SupportsRtxPayloadPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_1, TrySendPacket).WillRepeatedly(Return(false));
+  EXPECT_CALL(
+      rtp_1,
+      TrySendPacket(
+          ::testing::Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc1)), _))
+      .WillRepeatedly(Return(true));
+
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+  EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
+  EXPECT_CALL(rtp_2, SupportsPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_2, SupportsRtxPayloadPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_2, TrySendPacket).WillRepeatedly(Return(false));
+  EXPECT_CALL(
+      rtp_2,
+      TrySendPacket(
+          ::testing::Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc2)), _))
+      .WillRepeatedly(Return(true));
+
+  // Third module is sending media, but does not support rtx.
+  NiceMock<MockRtpRtcpInterface> rtp_3;
+  EXPECT_CALL(rtp_3, SSRC()).WillRepeatedly(Return(kSsrc3));
+  EXPECT_CALL(rtp_3, SupportsPadding).WillRepeatedly(Return(true));
+  EXPECT_CALL(rtp_3, SupportsRtxPayloadPadding).WillRepeatedly(Return(false));
+  EXPECT_CALL(rtp_3, TrySendPacket).WillRepeatedly(Return(false));
+  EXPECT_CALL(
+      rtp_3,
+      TrySendPacket(
+          ::testing::Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc3)), _))
+      .WillRepeatedly(Return(true));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddSendRtpModule(&rtp_2, false);
+  packet_router_.AddSendRtpModule(&rtp_3, false);
+
+  const size_t kPaddingBytes = 100;
+
+  // Initially, padding will be sent on last added rtp module that sends media
+  // and supports rtx.
+  EXPECT_CALL(rtp_2, GeneratePadding(kPaddingBytes))
+      .Times(1)
+      .WillOnce([&](size_t target_size_bytes) {
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+        packets.push_back(BuildRtpPacket(kSsrc2));
+        return packets;
+      });
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingBytes));
+
+  // Send media on first module. Padding should be sent on that module.
+  packet_router_.SendPacket(BuildRtpPacket(kSsrc1), PacedPacketInfo());
+
+  EXPECT_CALL(rtp_1, GeneratePadding(kPaddingBytes))
+      .Times(1)
+      .WillOnce([&](size_t target_size_bytes) {
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+        packets.push_back(BuildRtpPacket(kSsrc1));
+        return packets;
+      });
+  packet_router_.GeneratePadding(DataSize::Bytes(kPaddingBytes));
+
+  // Send media on second module. Padding should be sent there.
+  packet_router_.SendPacket(BuildRtpPacket(kSsrc2), PacedPacketInfo());
+
+  // If the last active module is removed, and no module sends media before
+  // the next padding request, and arbitrary module will be selected.
+  packet_router_.RemoveSendRtpModule(&rtp_2);
+
+  // Send on and then remove all remaining modules.
+  RtpRtcpInterface* last_send_module;
+  EXPECT_CALL(rtp_1, GeneratePadding(kPaddingBytes))
+      .Times(1)
+      .WillOnce([&](size_t target_size_bytes) {
+        last_send_module = &rtp_1;
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+        packets.push_back(BuildRtpPacket(kSsrc1));
+        return packets;
+      });
+  EXPECT_CALL(rtp_3, GeneratePadding(kPaddingBytes))
+      .Times(1)
+      .WillOnce([&](size_t target_size_bytes) {
+        last_send_module = &rtp_3;
+        std::vector<std::unique_ptr<RtpPacketToSend>> packets;
+        packets.push_back(BuildRtpPacket(kSsrc3));
+        return packets;
+      });
+
+  for (int i = 0; i < 2; ++i) {
+    last_send_module = nullptr;
+    packet_router_.GeneratePadding(DataSize::Bytes(kPaddingBytes));
+    EXPECT_NE(last_send_module, nullptr);
+    packet_router_.RemoveSendRtpModule(last_send_module);
+  }
+}
+
+TEST_F(PacketRouterTest, AllocatesTransportSequenceNumbers) {
+  const uint16_t kStartSeq = 0xFFF0;
+  const size_t kNumPackets = 32;
+  const uint16_t kSsrc1 = 1234;
+
+  PacketRouter packet_router(kStartSeq - 1);
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
+  EXPECT_CALL(rtp_1, TrySendPacket).WillRepeatedly(Return(true));
+  packet_router.AddSendRtpModule(&rtp_1, false);
+
+  for (size_t i = 0; i < kNumPackets; ++i) {
+    auto packet = BuildRtpPacket(kSsrc1);
+    EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+    packet_router.SendPacket(std::move(packet), PacedPacketInfo());
+    uint32_t expected_unwrapped_seq = static_cast<uint32_t>(kStartSeq) + i;
+    EXPECT_EQ(static_cast<uint16_t>(expected_unwrapped_seq & 0xFFFF),
+              packet_router.CurrentTransportSequenceNumber());
+  }
+
+  packet_router.RemoveSendRtpModule(&rtp_1);
+}
+
+TEST_F(PacketRouterTest, SendTransportFeedback) {
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+
+  ON_CALL(rtp_1, RTCP()).WillByDefault(Return(RtcpMode::kCompound));
+  ON_CALL(rtp_2, RTCP()).WillByDefault(Return(RtcpMode::kCompound));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddReceiveRtpModule(&rtp_2, false);
+
+  std::vector<std::unique_ptr<rtcp::RtcpPacket>> feedback;
+  feedback.push_back(std::make_unique<rtcp::TransportFeedback>());
+  EXPECT_CALL(rtp_1, SendCombinedRtcpPacket);
+  packet_router_.SendCombinedRtcpPacket(std::move(feedback));
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+  EXPECT_CALL(rtp_2, SendCombinedRtcpPacket);
+  std::vector<std::unique_ptr<rtcp::RtcpPacket>> new_feedback;
+  new_feedback.push_back(std::make_unique<rtcp::TransportFeedback>());
+  packet_router_.SendCombinedRtcpPacket(std::move(new_feedback));
+  packet_router_.RemoveReceiveRtpModule(&rtp_2);
+}
+
+TEST_F(PacketRouterTest, SendPacketWithoutTransportSequenceNumbers) {
+  const uint16_t kSsrc1 = 1234;
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(true));
+  ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+
+  // Send a packet without TransportSequenceNumber extension registered,
+  // packets sent should not have the extension set.
+  RtpHeaderExtensionMap extension_manager;
+  auto packet = std::make_unique<RtpPacketToSend>(&extension_manager);
+  packet->SetSsrc(kSsrc1);
+  EXPECT_CALL(
+      rtp_1,
+      TrySendPacket(
+          Property(&RtpPacketToSend::HasExtension<TransportSequenceNumber>,
+                   false),
+          _))
+      .WillOnce(Return(true));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+}
+
+TEST_F(PacketRouterTest, SendPacketAssignsTransportSequenceNumbers) {
+  NiceMock<MockRtpRtcpInterface> rtp_1;
+  NiceMock<MockRtpRtcpInterface> rtp_2;
+
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 2345;
+
+  ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
+  ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
+
+  packet_router_.AddSendRtpModule(&rtp_1, false);
+  packet_router_.AddSendRtpModule(&rtp_2, false);
+
+  // Transport sequence numbers start at 1, for historical reasons.
+  uint16_t transport_sequence_number = 1;
+
+  auto packet = BuildRtpPacket(kSsrc1);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+  EXPECT_CALL(
+      rtp_1,
+      TrySendPacket(
+          Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                   transport_sequence_number),
+          _))
+      .WillOnce(Return(true));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  ++transport_sequence_number;
+  packet = BuildRtpPacket(kSsrc2);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+
+  EXPECT_CALL(
+      rtp_2,
+      TrySendPacket(
+          Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                   transport_sequence_number),
+          _))
+      .WillOnce(Return(true));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  packet_router_.RemoveSendRtpModule(&rtp_1);
+  packet_router_.RemoveSendRtpModule(&rtp_2);
+}
+
+TEST_F(PacketRouterTest, DoesNotIncrementTransportSequenceNumberOnSendFailure) {
+  NiceMock<MockRtpRtcpInterface> rtp;
+  constexpr uint32_t kSsrc = 1234;
+  ON_CALL(rtp, SSRC).WillByDefault(Return(kSsrc));
+  packet_router_.AddSendRtpModule(&rtp, false);
+
+  // Transport sequence numbers start at 1, for historical reasons.
+  const uint16_t kStartTransportSequenceNumber = 1;
+
+  // Build and send a packet - it should be assigned the start sequence number.
+  // Return failure status code to make sure sequence number is not incremented.
+  auto packet = BuildRtpPacket(kSsrc);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+  EXPECT_CALL(
+      rtp, TrySendPacket(
+               Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                        kStartTransportSequenceNumber),
+               _))
+      .WillOnce(Return(false));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  // Send another packet, verify transport sequence number is still at the
+  // start state.
+  packet = BuildRtpPacket(kSsrc);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+
+  EXPECT_CALL(
+      rtp, TrySendPacket(
+               Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                        kStartTransportSequenceNumber),
+               _))
+      .WillOnce(Return(true));
+  packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
+
+  packet_router_.RemoveSendRtpModule(&rtp);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+using PacketRouterDeathTest = PacketRouterTest;
+TEST_F(PacketRouterDeathTest, DoubleRegistrationOfSendModuleDisallowed) {
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = false;  // Value irrelevant.
+  packet_router_.AddSendRtpModule(&module, remb_candidate);
+  EXPECT_DEATH(packet_router_.AddSendRtpModule(&module, remb_candidate), "");
+
+  // Test tear-down
+  packet_router_.RemoveSendRtpModule(&module);
+}
+
+TEST_F(PacketRouterDeathTest, DoubleRegistrationOfReceiveModuleDisallowed) {
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = false;  // Value irrelevant.
+  packet_router_.AddReceiveRtpModule(&module, remb_candidate);
+  EXPECT_DEATH(packet_router_.AddReceiveRtpModule(&module, remb_candidate), "");
+
+  // Test tear-down
+  packet_router_.RemoveReceiveRtpModule(&module);
+}
+
+TEST_F(PacketRouterDeathTest, RemovalOfNeverAddedSendModuleDisallowed) {
+  NiceMock<MockRtpRtcpInterface> module;
+
+  EXPECT_DEATH(packet_router_.RemoveSendRtpModule(&module), "");
+}
+
+TEST_F(PacketRouterDeathTest, RemovalOfNeverAddedReceiveModuleDisallowed) {
+  NiceMock<MockRtpRtcpInterface> module;
+
+  EXPECT_DEATH(packet_router_.RemoveReceiveRtpModule(&module), "");
+}
+#endif  // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+TEST(PacketRouterRembTest, ChangeSendRtpModuleChangeRembSender) {
+  rtc::ScopedFakeClock clock;
+  NiceMock<MockRtpRtcpInterface> rtp_send;
+  NiceMock<MockRtpRtcpInterface> rtp_recv;
+  PacketRouter packet_router;
+  packet_router.AddSendRtpModule(&rtp_send, true);
+  packet_router.AddReceiveRtpModule(&rtp_recv, true);
+
+  uint32_t bitrate_estimate = 456;
+  std::vector<uint32_t> ssrcs = {1234, 5678};
+
+  EXPECT_CALL(rtp_send, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Remove the sending module -> should get remb on the second module.
+  packet_router.RemoveSendRtpModule(&rtp_send);
+
+  EXPECT_CALL(rtp_recv, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  packet_router.RemoveReceiveRtpModule(&rtp_recv);
+}
+
+// Only register receiving modules and make sure we fallback to trigger a REMB
+// packet on this one.
+TEST(PacketRouterRembTest, NoSendingRtpModule) {
+  rtc::ScopedFakeClock clock;
+  NiceMock<MockRtpRtcpInterface> rtp;
+  PacketRouter packet_router;
+
+  packet_router.AddReceiveRtpModule(&rtp, true);
+
+  uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+
+  EXPECT_CALL(rtp, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Lower the estimate to trigger a new packet REMB packet.
+  EXPECT_CALL(rtp, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  EXPECT_CALL(rtp, UnsetRemb());
+  packet_router.RemoveReceiveRtpModule(&rtp);
+}
+
+TEST(PacketRouterRembTest, NonCandidateSendRtpModuleNotUsedForRemb) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = false;
+
+  packet_router.AddSendRtpModule(&module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(module, SetRemb(_, _)).Times(0);
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveSendRtpModule(&module);
+}
+
+TEST(PacketRouterRembTest, CandidateSendRtpModuleUsedForRemb) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = true;
+
+  packet_router.AddSendRtpModule(&module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(module, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveSendRtpModule(&module);
+}
+
+TEST(PacketRouterRembTest, NonCandidateReceiveRtpModuleNotUsedForRemb) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = false;
+
+  packet_router.AddReceiveRtpModule(&module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(module, SetRemb(_, _)).Times(0);
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&module);
+}
+
+TEST(PacketRouterRembTest, CandidateReceiveRtpModuleUsedForRemb) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> module;
+
+  constexpr bool remb_candidate = true;
+
+  packet_router.AddReceiveRtpModule(&module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(module, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&module);
+}
+
+TEST(PacketRouterRembTest,
+     SendCandidatePreferredOverReceiveCandidate_SendModuleAddedFirst) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> send_module;
+  NiceMock<MockRtpRtcpInterface> receive_module;
+
+  constexpr bool remb_candidate = true;
+
+  // Send module added - activated.
+  packet_router.AddSendRtpModule(&send_module, remb_candidate);
+
+  // Receive module added - the send module remains the active one.
+  packet_router.AddReceiveRtpModule(&receive_module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(send_module, SetRemb(bitrate_estimate, ssrcs));
+  EXPECT_CALL(receive_module, SetRemb(_, _)).Times(0);
+
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&receive_module);
+  packet_router.RemoveSendRtpModule(&send_module);
+}
+
+TEST(PacketRouterRembTest,
+     SendCandidatePreferredOverReceiveCandidate_ReceiveModuleAddedFirst) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> send_module;
+  NiceMock<MockRtpRtcpInterface> receive_module;
+
+  constexpr bool remb_candidate = true;
+
+  // Receive module added - activated.
+  packet_router.AddReceiveRtpModule(&receive_module, remb_candidate);
+
+  // Send module added - replaces receive module as active.
+  packet_router.AddSendRtpModule(&send_module, remb_candidate);
+
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(send_module, SetRemb(bitrate_estimate, ssrcs));
+  EXPECT_CALL(receive_module, SetRemb(_, _)).Times(0);
+
+  clock.AdvanceTime(TimeDelta::Millis(1000));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&receive_module);
+  packet_router.RemoveSendRtpModule(&send_module);
+}
+
+TEST(PacketRouterRembTest, ReceiveModuleTakesOverWhenLastSendModuleRemoved) {
+  rtc::ScopedFakeClock clock;
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcpInterface> send_module;
+  NiceMock<MockRtpRtcpInterface> receive_module;
+
+  constexpr bool remb_candidate = true;
+
+  // Send module active, receive module inactive.
+  packet_router.AddSendRtpModule(&send_module, remb_candidate);
+  packet_router.AddReceiveRtpModule(&receive_module, remb_candidate);
+
+  // Send module removed - receive module becomes active.
+  packet_router.RemoveSendRtpModule(&send_module);
+  constexpr uint32_t bitrate_estimate = 456;
+  const std::vector<uint32_t> ssrcs = {1234};
+  EXPECT_CALL(send_module, SetRemb(_, _)).Times(0);
+  EXPECT_CALL(receive_module, SetRemb(bitrate_estimate, ssrcs));
+  packet_router.SendRemb(bitrate_estimate, ssrcs);
+
+  // Test tear-down
+  packet_router.RemoveReceiveRtpModule(&receive_module);
+}
+
+}  // namespace webrtc