| /* |
| * 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::ElementsAreArray; |
| 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); |
| } |
| |
| TEST_F(PacketRouterTest, ForwardsAbortedRetransmissions) { |
| NiceMock<MockRtpRtcpInterface> rtp_1; |
| NiceMock<MockRtpRtcpInterface> rtp_2; |
| |
| const uint32_t kSsrc1 = 1234; |
| const uint32_t kSsrc2 = 2345; |
| const uint32_t kInvalidSsrc = 3456; |
| |
| 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); |
| |
| // Sets of retransmission sequence numbers we wish to abort, per ssrc. |
| const uint16_t kAbortedRetransmissionsOnSsrc1[] = {17, 42}; |
| const uint16_t kAbortedRetransmissionsOnSsrc2[] = {1337, 4711}; |
| const uint16_t kAbortedRetransmissionsOnSsrc3[] = {123}; |
| |
| EXPECT_CALL(rtp_1, OnAbortedRetransmissions( |
| ElementsAreArray(kAbortedRetransmissionsOnSsrc1))); |
| EXPECT_CALL(rtp_2, OnAbortedRetransmissions( |
| ElementsAreArray(kAbortedRetransmissionsOnSsrc2))); |
| |
| packet_router_.OnAbortedRetransmissions(kSsrc1, |
| kAbortedRetransmissionsOnSsrc1); |
| packet_router_.OnAbortedRetransmissions(kSsrc2, |
| kAbortedRetransmissionsOnSsrc2); |
| |
| // Should be noop and not cause any issues. |
| packet_router_.OnAbortedRetransmissions(kInvalidSsrc, |
| kAbortedRetransmissionsOnSsrc3); |
| |
| packet_router_.RemoveSendRtpModule(&rtp_1); |
| packet_router_.RemoveSendRtpModule(&rtp_2); |
| } |
| |
| TEST_F(PacketRouterTest, ReportsRtxSsrc) { |
| NiceMock<MockRtpRtcpInterface> rtp_1; |
| NiceMock<MockRtpRtcpInterface> rtp_2; |
| |
| const uint32_t kSsrc1 = 1234; |
| const uint32_t kRtxSsrc1 = 1235; |
| const uint32_t kSsrc2 = 2345; |
| const uint32_t kInvalidSsrc = 3456; |
| |
| ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1)); |
| ON_CALL(rtp_1, RtxSsrc).WillByDefault(Return(kRtxSsrc1)); |
| ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2)); |
| |
| packet_router_.AddSendRtpModule(&rtp_1, false); |
| packet_router_.AddSendRtpModule(&rtp_2, false); |
| |
| EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kSsrc1), kRtxSsrc1); |
| EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kRtxSsrc1), absl::nullopt); |
| EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kSsrc2), absl::nullopt); |
| EXPECT_EQ(packet_router_.GetRtxSsrcForMedia(kInvalidSsrc), absl::nullopt); |
| |
| packet_router_.RemoveSendRtpModule(&rtp_1); |
| packet_router_.RemoveSendRtpModule(&rtp_2); |
| } |
| |
| #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 |