diff --git a/modules/rtp_rtcp/test/testAPI/test_api.cc b/modules/rtp_rtcp/test/testAPI/test_api.cc index 5fa1de07fb..2fd4464b3c 100644 --- a/modules/rtp_rtcp/test/testAPI/test_api.cc +++ b/modules/rtp_rtcp/test/testAPI/test_api.cc @@ -13,7 +13,6 @@ #include #include #include -#include #include "rtc_base/checks.h" #include "rtc_base/rate_limiter.h" @@ -96,10 +95,10 @@ class RtpRtcpAPITest : public ::testing::Test { module_->SetSSRC(kInitialSsrc); } - std::unique_ptr module_; SimulatedClock fake_clock_; test::NullTransport null_transport_; RateLimiter retransmission_rate_limiter_; + std::unique_ptr module_; }; TEST_F(RtpRtcpAPITest, Basic) { diff --git a/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/modules/rtp_rtcp/test/testAPI/test_api_audio.cc index 2618433dd5..fd7b8a7f16 100644 --- a/modules/rtp_rtcp/test/testAPI/test_api_audio.cc +++ b/modules/rtp_rtcp/test/testAPI/test_api_audio.cc @@ -88,78 +88,79 @@ class VerifyingAudioReceiver : public RtpData { class RtpRtcpAudioTest : public ::testing::Test { protected: RtpRtcpAudioTest() - : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {} - ~RtpRtcpAudioTest() override = default; - - void SetUp() override { - receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock_)); - receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock_)); - + : fake_clock_(123456), + retransmission_rate_limiter_(&fake_clock_, 1000), + receive_statistics1_(ReceiveStatistics::Create(&fake_clock_)), + receive_statistics2_(ReceiveStatistics::Create(&fake_clock_)), + rtp_receiver1_( + RtpReceiver::CreateAudioReceiver(&fake_clock_, + &data_receiver1_, + &rtp_payload_registry1_)), + rtp_receiver2_( + RtpReceiver::CreateAudioReceiver(&fake_clock_, + &data_receiver2_, + &rtp_payload_registry2_)) { RtpRtcp::Configuration configuration; configuration.audio = true; configuration.clock = &fake_clock_; configuration.receive_statistics = receive_statistics1_.get(); - configuration.outgoing_transport = &transport1; + configuration.outgoing_transport = &transport1_; configuration.retransmission_rate_limiter = &retransmission_rate_limiter_; - - module1.reset(RtpRtcp::CreateRtpRtcp(configuration)); - rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver( - &fake_clock_, &data_receiver1, &rtp_payload_registry1_)); + module1_.reset(RtpRtcp::CreateRtpRtcp(configuration)); configuration.receive_statistics = receive_statistics2_.get(); - configuration.outgoing_transport = &transport2; + configuration.outgoing_transport = &transport2_; + module2_.reset(RtpRtcp::CreateRtpRtcp(configuration)); - module2.reset(RtpRtcp::CreateRtpRtcp(configuration)); - rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver( - &fake_clock_, &data_receiver2, &rtp_payload_registry2_)); - - transport1.SetSendModule(module2.get(), &rtp_payload_registry2_, - rtp_receiver2_.get(), receive_statistics2_.get()); - transport2.SetSendModule(module1.get(), &rtp_payload_registry1_, - rtp_receiver1_.get(), receive_statistics1_.get()); + transport1_.SetSendModule(module2_.get(), &rtp_payload_registry2_, + rtp_receiver2_.get(), receive_statistics2_.get()); + transport2_.SetSendModule(module1_.get(), &rtp_payload_registry1_, + rtp_receiver1_.get(), receive_statistics1_.get()); } + ~RtpRtcpAudioTest() override = default; + void RegisterPayload(const CodecInst& codec) { - EXPECT_EQ(0, module1->RegisterSendPayload(codec)); + EXPECT_EQ(0, module1_->RegisterSendPayload(codec)); EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(codec.pltype, CodecInstToSdp(codec))); - EXPECT_EQ(0, module2->RegisterSendPayload(codec)); + EXPECT_EQ(0, module2_->RegisterSendPayload(codec)); EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(codec.pltype, CodecInstToSdp(codec))); } - VerifyingAudioReceiver data_receiver1; - VerifyingAudioReceiver data_receiver2; + SimulatedClock fake_clock_; + RateLimiter retransmission_rate_limiter_; + VerifyingAudioReceiver data_receiver1_; + VerifyingAudioReceiver data_receiver2_; std::unique_ptr receive_statistics1_; std::unique_ptr receive_statistics2_; RTPPayloadRegistry rtp_payload_registry1_; RTPPayloadRegistry rtp_payload_registry2_; std::unique_ptr rtp_receiver1_; std::unique_ptr rtp_receiver2_; - std::unique_ptr module1; - std::unique_ptr module2; - LoopBackTransport transport1; - LoopBackTransport transport2; - SimulatedClock fake_clock_; - RateLimiter retransmission_rate_limiter_; + std::unique_ptr module1_; + std::unique_ptr module2_; + LoopBackTransport transport1_; + LoopBackTransport transport2_; }; TEST_F(RtpRtcpAudioTest, Basic) { - module1->SetSSRC(kSsrc); - module1->SetStartTimestamp(kTimestamp); + module1_->SetSSRC(kSsrc); + module1_->SetStartTimestamp(kTimestamp); // Test detection at the end of a DTMF tone. - // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true)); + // EXPECT_EQ(0, module2_->SetTelephoneEventForwardToDecoder(true)); - EXPECT_EQ(0, module1->SetSendingStatus(true)); + EXPECT_EQ(0, module1_->SetSendingStatus(true)); // Start basic RTP test. // Send an empty RTP packet. // Should fail since we have not registered the payload type. - EXPECT_FALSE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, - kPcmuPayloadType, 0, -1, nullptr, 0, - nullptr, nullptr, nullptr)); + EXPECT_FALSE(module1_->SendOutgoingData(webrtc::kAudioFrameSpeech, + kPcmuPayloadType, 0, -1, nullptr, 0, + nullptr, nullptr, nullptr)); CodecInst voice_codec = {}; voice_codec.pltype = kPcmuPayloadType; @@ -168,9 +169,9 @@ TEST_F(RtpRtcpAudioTest, Basic) { memcpy(voice_codec.plname, "PCMU", 5); RegisterPayload(voice_codec); - EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, - kPcmuPayloadType, 0, -1, kTestPayload, - 4, nullptr, nullptr, nullptr)); + EXPECT_TRUE(module1_->SendOutgoingData(webrtc::kAudioFrameSpeech, + kPcmuPayloadType, 0, -1, kTestPayload, + 4, nullptr, nullptr, nullptr)); EXPECT_EQ(kSsrc, rtp_receiver2_->SSRC()); uint32_t timestamp; @@ -189,16 +190,16 @@ TEST_F(RtpRtcpAudioTest, DTMF) { memcpy(voice_codec.plname, "PCMU", 5); RegisterPayload(voice_codec); - module1->SetSSRC(kSsrc); - module1->SetStartTimestamp(kTimestamp); - EXPECT_EQ(0, module1->SetSendingStatus(true)); + module1_->SetSSRC(kSsrc); + module1_->SetStartTimestamp(kTimestamp); + EXPECT_EQ(0, module1_->SetSendingStatus(true)); // Prepare for DTMF. voice_codec.pltype = kDtmfPayloadType; voice_codec.plfreq = 8000; memcpy(voice_codec.plname, "telephone-event", 16); - EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec)); + EXPECT_EQ(0, module1_->RegisterSendPayload(voice_codec)); EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload( voice_codec.pltype, CodecInstToSdp(voice_codec))); @@ -207,37 +208,37 @@ TEST_F(RtpRtcpAudioTest, DTMF) { // Send a DTMF tone using RFC 2833 (4733). for (int i = 0; i < 16; i++) { - EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10)); + EXPECT_EQ(0, module1_->SendTelephoneEventOutband(i, timeStamp, 10)); } timeStamp += 160; // Prepare for next packet. // Send RTP packets for 16 tones a 160 ms 100ms // pause between = 2560ms + 1600ms = 4160ms for (; timeStamp <= 250 * 160; timeStamp += 160) { - EXPECT_TRUE(module1->SendOutgoingData( + EXPECT_TRUE(module1_->SendOutgoingData( webrtc::kAudioFrameSpeech, kPcmuPayloadType, timeStamp, -1, kTestPayload, 4, nullptr, nullptr, nullptr)); fake_clock_.AdvanceTimeMilliseconds(20); - module1->Process(); + module1_->Process(); } - EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10)); + EXPECT_EQ(0, module1_->SendTelephoneEventOutband(32, 9000, 10)); for (; timeStamp <= 740 * 160; timeStamp += 160) { - EXPECT_TRUE(module1->SendOutgoingData( + EXPECT_TRUE(module1_->SendOutgoingData( webrtc::kAudioFrameSpeech, kPcmuPayloadType, timeStamp, -1, kTestPayload, 4, nullptr, nullptr, nullptr)); fake_clock_.AdvanceTimeMilliseconds(20); - module1->Process(); + module1_->Process(); } } TEST_F(RtpRtcpAudioTest, ComfortNoise) { - module1->SetSSRC(kSsrc); - module1->SetStartTimestamp(kTimestamp); + module1_->SetSSRC(kSsrc); + module1_->SetStartTimestamp(kTimestamp); - EXPECT_EQ(0, module1->SetSendingStatus(true)); + EXPECT_EQ(0, module1_->SetSendingStatus(true)); - // Register PCMU and all four comfort noise codecs + // Register PCMU and all four comfort noise codecs. CodecInst voice_codec = {}; voice_codec.pltype = kPcmuPayloadType; voice_codec.plfreq = 8000; @@ -258,7 +259,7 @@ TEST_F(RtpRtcpAudioTest, ComfortNoise) { for (const auto& c : kCngCodecs) { uint32_t timestamp; int64_t receive_time_ms; - EXPECT_TRUE(module1->SendOutgoingData( + EXPECT_TRUE(module1_->SendOutgoingData( webrtc::kAudioFrameSpeech, kPcmuPayloadType, in_timestamp, -1, kTestPayload, 4, nullptr, nullptr, nullptr)); @@ -270,9 +271,9 @@ TEST_F(RtpRtcpAudioTest, ComfortNoise) { in_timestamp += 10; fake_clock_.AdvanceTimeMilliseconds(20); - EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameCN, c.payload_type, - in_timestamp, -1, kTestPayload, 1, - nullptr, nullptr, nullptr)); + EXPECT_TRUE(module1_->SendOutgoingData( + webrtc::kAudioFrameCN, c.payload_type, in_timestamp, -1, kTestPayload, + 1, nullptr, nullptr, nullptr)); EXPECT_EQ(kSsrc, rtp_receiver2_->SSRC()); EXPECT_TRUE( diff --git a/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc index 7551e71c18..7d8628a55a 100644 --- a/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc +++ b/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc @@ -20,7 +20,6 @@ #include "modules/rtp_rtcp/source/rtp_receiver_audio.h" #include "modules/rtp_rtcp/test/testAPI/test_api.h" #include "rtc_base/rate_limiter.h" -#include "test/gmock.h" #include "test/gtest.h" namespace webrtc { @@ -37,57 +36,53 @@ class RtcpCallback : public RtcpIntraFrameObserver { class RtpRtcpRtcpTest : public ::testing::Test { protected: RtpRtcpRtcpTest() - : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {} + : fake_clock_(123456), + retransmission_rate_limiter_(&fake_clock_, 1000), + receive_statistics1_(ReceiveStatistics::Create(&fake_clock_)), + receive_statistics2_(ReceiveStatistics::Create(&fake_clock_)), + rtp_receiver1_( + RtpReceiver::CreateAudioReceiver(&fake_clock_, + &receiver_, + &rtp_payload_registry1_)), + rtp_receiver2_( + RtpReceiver::CreateAudioReceiver(&fake_clock_, + &receiver_, + &rtp_payload_registry2_)) {} ~RtpRtcpRtcpTest() override = default; void SetUp() override { - receiver = new TestRtpReceiver(); - transport1 = new LoopBackTransport(); - transport2 = new LoopBackTransport(); - - receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock_)); - receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock_)); - RtpRtcp::Configuration configuration; configuration.audio = true; configuration.clock = &fake_clock_; configuration.receive_statistics = receive_statistics1_.get(); - configuration.outgoing_transport = transport1; + configuration.outgoing_transport = &transport1_; configuration.intra_frame_callback = &rtcp_callback1_; configuration.retransmission_rate_limiter = &retransmission_rate_limiter_; - - module1 = RtpRtcp::CreateRtpRtcp(configuration); - - rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver( - &fake_clock_, receiver, &rtp_payload_registry1_)); + module1_.reset(RtpRtcp::CreateRtpRtcp(configuration)); configuration.receive_statistics = receive_statistics2_.get(); - configuration.outgoing_transport = transport2; + configuration.outgoing_transport = &transport2_; configuration.intra_frame_callback = &rtcp_callback2_; + module2_.reset(RtpRtcp::CreateRtpRtcp(configuration)); - module2 = RtpRtcp::CreateRtpRtcp(configuration); - - rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver( - &fake_clock_, receiver, &rtp_payload_registry2_)); - - transport1->SetSendModule(module2, &rtp_payload_registry2_, + transport1_.SetSendModule(module2_.get(), &rtp_payload_registry2_, rtp_receiver2_.get(), receive_statistics2_.get()); - transport2->SetSendModule(module1, &rtp_payload_registry1_, + transport2_.SetSendModule(module1_.get(), &rtp_payload_registry1_, rtp_receiver1_.get(), receive_statistics1_.get()); - module1->SetRTCPStatus(RtcpMode::kCompound); - module2->SetRTCPStatus(RtcpMode::kCompound); + module1_->SetRTCPStatus(RtcpMode::kCompound); + module2_->SetRTCPStatus(RtcpMode::kCompound); - module2->SetSSRC(kSsrc + 1); - module2->SetRemoteSSRC(kSsrc); - module1->SetSSRC(kSsrc); - module1->SetSequenceNumber(kSequenceNumber); - module1->SetStartTimestamp(kTimestamp); + module2_->SetSSRC(kSsrc + 1); + module2_->SetRemoteSSRC(kSsrc); + module1_->SetSSRC(kSsrc); + module1_->SetSequenceNumber(kSequenceNumber); + module1_->SetStartTimestamp(kTimestamp); - module1->SetCsrcs(kCsrcs); - EXPECT_EQ(0, module1->SetCNAME("john.doe@test.test")); + module1_->SetCsrcs(kCsrcs); + EXPECT_EQ(0, module1_->SetCNAME("john.doe@test.test")); - EXPECT_EQ(0, module1->SetSendingStatus(true)); + EXPECT_EQ(0, module1_->SetSendingStatus(true)); CodecInst voice_codec; voice_codec.pltype = 96; @@ -95,10 +90,10 @@ class RtpRtcpRtcpTest : public ::testing::Test { voice_codec.rate = 64000; memcpy(voice_codec.plname, "PCMU", 5); - EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec)); + EXPECT_EQ(0, module1_->RegisterSendPayload(voice_codec)); EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload( voice_codec.pltype, CodecInstToSdp(voice_codec))); - EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec)); + EXPECT_EQ(0, module2_->RegisterSendPayload(voice_codec)); EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload( voice_codec.pltype, CodecInstToSdp(voice_codec))); @@ -107,84 +102,74 @@ class RtpRtcpRtcpTest : public ::testing::Test { // Send RTP packet with the data "testtest". const uint8_t test[9] = "testtest"; EXPECT_EQ(true, - module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1, - test, 8, nullptr, nullptr, nullptr)); + module1_->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1, + test, 8, nullptr, nullptr, nullptr)); } - void TearDown() override { - delete module1; - delete module2; - delete transport1; - delete transport2; - delete receiver; - } - - RtcpCallback rtcp_callback1_; - RtcpCallback rtcp_callback2_; - RTPPayloadRegistry rtp_payload_registry1_; - RTPPayloadRegistry rtp_payload_registry2_; - std::unique_ptr receive_statistics1_; - std::unique_ptr receive_statistics2_; - std::unique_ptr rtp_receiver1_; - std::unique_ptr rtp_receiver2_; - RtpRtcp* module1; - RtpRtcp* module2; - TestRtpReceiver* receiver; - LoopBackTransport* transport1; - LoopBackTransport* transport2; - const std::vector kCsrcs = {1234, 2345}; SimulatedClock fake_clock_; RateLimiter retransmission_rate_limiter_; + RtcpCallback rtcp_callback1_; + RtcpCallback rtcp_callback2_; + RTPPayloadRegistry rtp_payload_registry1_; + RTPPayloadRegistry rtp_payload_registry2_; + TestRtpReceiver receiver_; + std::unique_ptr receive_statistics1_; + std::unique_ptr receive_statistics2_; + std::unique_ptr rtp_receiver1_; + std::unique_ptr rtp_receiver2_; + std::unique_ptr module1_; + std::unique_ptr module2_; + LoopBackTransport transport1_; + LoopBackTransport transport2_; }; TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) { // Set cname of mixed. - EXPECT_EQ(0, module1->AddMixedCNAME(kCsrcs[0], "john@192.168.0.1")); - EXPECT_EQ(0, module1->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2")); + EXPECT_EQ(0, module1_->AddMixedCNAME(kCsrcs[0], "john@192.168.0.1")); + EXPECT_EQ(0, module1_->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2")); - EXPECT_EQ(-1, module1->RemoveMixedCNAME(kCsrcs[0] + 1)); - EXPECT_EQ(0, module1->RemoveMixedCNAME(kCsrcs[1])); - EXPECT_EQ(0, module1->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2")); + EXPECT_EQ(-1, module1_->RemoveMixedCNAME(kCsrcs[0] + 1)); + EXPECT_EQ(0, module1_->RemoveMixedCNAME(kCsrcs[1])); + EXPECT_EQ(0, module1_->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2")); // Send RTCP packet, triggered by timer. fake_clock_.AdvanceTimeMilliseconds(7500); - module1->Process(); + module1_->Process(); fake_clock_.AdvanceTimeMilliseconds(100); - module2->Process(); + module2_->Process(); char cName[RTCP_CNAME_SIZE]; - EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC() + 1, cName)); + EXPECT_EQ(-1, module2_->RemoteCNAME(rtp_receiver2_->SSRC() + 1, cName)); // Check multiple CNAME. - EXPECT_EQ(0, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName)); + EXPECT_EQ(0, module2_->RemoteCNAME(rtp_receiver2_->SSRC(), cName)); EXPECT_EQ(0, strncmp(cName, "john.doe@test.test", RTCP_CNAME_SIZE)); - EXPECT_EQ(0, module2->RemoteCNAME(kCsrcs[0], cName)); + EXPECT_EQ(0, module2_->RemoteCNAME(kCsrcs[0], cName)); EXPECT_EQ(0, strncmp(cName, "john@192.168.0.1", RTCP_CNAME_SIZE)); - EXPECT_EQ(0, module2->RemoteCNAME(kCsrcs[1], cName)); + EXPECT_EQ(0, module2_->RemoteCNAME(kCsrcs[1], cName)); EXPECT_EQ(0, strncmp(cName, "jane@192.168.0.2", RTCP_CNAME_SIZE)); - EXPECT_EQ(0, module1->SetSendingStatus(false)); + EXPECT_EQ(0, module1_->SetSendingStatus(false)); // Test that BYE clears the CNAME. - EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName)); + EXPECT_EQ(-1, module2_->RemoteCNAME(rtp_receiver2_->SSRC(), cName)); } TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) { std::vector report_blocks; - - EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks)); + EXPECT_EQ(0, module1_->RemoteRTCPStat(&report_blocks)); EXPECT_EQ(0u, report_blocks.size()); // Send RTCP packet, triggered by timer. fake_clock_.AdvanceTimeMilliseconds(7500); - module1->Process(); + module1_->Process(); fake_clock_.AdvanceTimeMilliseconds(100); - module2->Process(); + module2_->Process(); - EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks)); + EXPECT_EQ(0, module1_->RemoteRTCPStat(&report_blocks)); ASSERT_EQ(1u, report_blocks.size()); // |kSsrc+1| is the SSRC of module2 that send the report. diff --git a/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/modules/rtp_rtcp/test/testAPI/test_api_video.cc index 4d8190bdad..aa524bdb08 100644 --- a/modules/rtp_rtcp/test/testAPI/test_api_video.cc +++ b/modules/rtp_rtcp/test/testAPI/test_api_video.cc @@ -33,30 +33,30 @@ namespace webrtc { class RtpRtcpVideoTest : public ::testing::Test { protected: RtpRtcpVideoTest() - : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {} + : fake_clock_(123456), + retransmission_rate_limiter_(&fake_clock_, 1000), + receive_statistics_(ReceiveStatistics::Create(&fake_clock_)), + rtp_receiver_( + RtpReceiver::CreateVideoReceiver(&fake_clock_, + &receiver_, + &rtp_payload_registry_)) {} ~RtpRtcpVideoTest() override = default; void SetUp() override { - transport_ = new LoopBackTransport(); - receiver_ = new TestRtpReceiver(); - receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock_)); RtpRtcp::Configuration configuration; configuration.audio = false; configuration.clock = &fake_clock_; - configuration.outgoing_transport = transport_; + configuration.outgoing_transport = &transport_; configuration.retransmission_rate_limiter = &retransmission_rate_limiter_; - - video_module_ = RtpRtcp::CreateRtpRtcp(configuration); - rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver( - &fake_clock_, receiver_, &rtp_payload_registry_)); + video_module_.reset(RtpRtcp::CreateRtpRtcp(configuration)); video_module_->SetRTCPStatus(RtcpMode::kCompound); video_module_->SetSSRC(kSsrc); video_module_->SetStorePacketsStatus(true, 600); EXPECT_EQ(0, video_module_->SetSendingStatus(true)); - transport_->SetSendModule(video_module_, &rtp_payload_registry_, - rtp_receiver_.get(), receive_statistics_.get()); + transport_.SetSendModule(video_module_.get(), &rtp_payload_registry_, + rtp_receiver_.get(), receive_statistics_.get()); VideoCodec video_codec; memset(&video_codec, 0, sizeof(video_codec)); @@ -111,22 +111,16 @@ class RtpRtcpVideoTest : public ::testing::Test { return padding_bytes_in_packet + header_length; } - void TearDown() override { - delete video_module_; - delete transport_; - delete receiver_; - } - - std::unique_ptr receive_statistics_; - RTPPayloadRegistry rtp_payload_registry_; - std::unique_ptr rtp_receiver_; - RtpRtcp* video_module_; - LoopBackTransport* transport_; - TestRtpReceiver* receiver_; uint8_t video_frame_[65000]; size_t payload_data_length_; SimulatedClock fake_clock_; RateLimiter retransmission_rate_limiter_; + std::unique_ptr receive_statistics_; + RTPPayloadRegistry rtp_payload_registry_; + TestRtpReceiver receiver_; + std::unique_ptr rtp_receiver_; + std::unique_ptr video_module_; + LoopBackTransport transport_; }; TEST_F(RtpRtcpVideoTest, BasicVideo) { @@ -161,8 +155,8 @@ TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) { const size_t payload_length = packet_size - header.headerLength; EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket( header, payload, payload_length, pl->typeSpecific)); - EXPECT_EQ(0u, receiver_->payload_size()); - EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength); + EXPECT_EQ(0u, receiver_.payload_size()); + EXPECT_EQ(payload_length, receiver_.rtp_header().header.paddingLength); } timestamp += 3000; fake_clock_.AdvanceTimeMilliseconds(33);