| /* |
| * Copyright 2017 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 <memory> |
| #include <utility> // For std::pair, std::move. |
| |
| #include "webrtc/api/ortc/ortcfactoryinterface.h" |
| #include "webrtc/ortc/testrtpparameters.h" |
| #include "webrtc/p2p/base/udptransport.h" |
| #include "webrtc/pc/test/fakeaudiocapturemodule.h" |
| #include "webrtc/pc/test/fakeperiodicvideocapturer.h" |
| #include "webrtc/pc/test/fakevideotrackrenderer.h" |
| #include "webrtc/rtc_base/criticalsection.h" |
| #include "webrtc/rtc_base/fakenetwork.h" |
| #include "webrtc/rtc_base/gunit.h" |
| #include "webrtc/rtc_base/virtualsocketserver.h" |
| |
| namespace { |
| |
| const int kDefaultTimeout = 10000; // 10 seconds. |
| const int kReceivingDuration = 1000; // 1 second. |
| // Default number of audio/video frames to wait for before considering a test a |
| // success. |
| const int kDefaultNumFrames = 3; |
| const rtc::IPAddress kIPv4LocalHostAddress = |
| rtc::IPAddress(0x7F000001); // 127.0.0.1 |
| |
| static const char kTestKeyParams1[] = |
| "inline:WVNfX19zZW1jdGwgKskgewkyMjA7fQp9CnVubGVz"; |
| static const char kTestKeyParams2[] = |
| "inline:PS1uQCVeeCFCanVmcjkpaywjNWhcYD0mXXtxaVBR"; |
| static const char kTestKeyParams3[] = |
| "inline:WVNfX19zZW1jdGwgKskgewkyMjA7fQp9CnVubGVa"; |
| static const char kTestKeyParams4[] = |
| "inline:WVNfX19zZW1jdGwgKskgewkyMjA7fQp9CnVubGVb"; |
| static const cricket::CryptoParams kTestCryptoParams1(1, |
| "AES_CM_128_HMAC_SHA1_80", |
| kTestKeyParams1, |
| ""); |
| static const cricket::CryptoParams kTestCryptoParams2(1, |
| "AES_CM_128_HMAC_SHA1_80", |
| kTestKeyParams2, |
| ""); |
| static const cricket::CryptoParams kTestCryptoParams3(1, |
| "AES_CM_128_HMAC_SHA1_80", |
| kTestKeyParams3, |
| ""); |
| static const cricket::CryptoParams kTestCryptoParams4(1, |
| "AES_CM_128_HMAC_SHA1_80", |
| kTestKeyParams4, |
| ""); |
| } // namespace |
| |
| namespace webrtc { |
| |
| // Used to test that things work end-to-end when using the default |
| // implementations of threads/etc. provided by OrtcFactory, with the exception |
| // of using a virtual network. |
| // |
| // By default, the virtual network manager doesn't enumerate any networks, but |
| // sockets can still be created in this state. |
| class OrtcFactoryIntegrationTest : public testing::Test { |
| public: |
| OrtcFactoryIntegrationTest() |
| : network_thread_(&virtual_socket_server_), |
| fake_audio_capture_module1_(FakeAudioCaptureModule::Create()), |
| fake_audio_capture_module2_(FakeAudioCaptureModule::Create()) { |
| // Sockets are bound to the ANY address, so this is needed to tell the |
| // virtual network which address to use in this case. |
| virtual_socket_server_.SetDefaultRoute(kIPv4LocalHostAddress); |
| network_thread_.Start(); |
| // Need to create after network thread is started. |
| ortc_factory1_ = OrtcFactoryInterface::Create( |
| &network_thread_, nullptr, &fake_network_manager_, |
| nullptr, fake_audio_capture_module1_) |
| .MoveValue(); |
| ortc_factory2_ = OrtcFactoryInterface::Create( |
| &network_thread_, nullptr, &fake_network_manager_, |
| nullptr, fake_audio_capture_module2_) |
| .MoveValue(); |
| } |
| |
| protected: |
| typedef std::pair<std::unique_ptr<UdpTransportInterface>, |
| std::unique_ptr<UdpTransportInterface>> |
| UdpTransportPair; |
| typedef std::pair<std::unique_ptr<RtpTransportInterface>, |
| std::unique_ptr<RtpTransportInterface>> |
| RtpTransportPair; |
| typedef std::pair<std::unique_ptr<SrtpTransportInterface>, |
| std::unique_ptr<SrtpTransportInterface>> |
| SrtpTransportPair; |
| typedef std::pair<std::unique_ptr<RtpTransportControllerInterface>, |
| std::unique_ptr<RtpTransportControllerInterface>> |
| RtpTransportControllerPair; |
| |
| // Helper function that creates one UDP transport each for |ortc_factory1_| |
| // and |ortc_factory2_|, and connects them. |
| UdpTransportPair CreateAndConnectUdpTransportPair() { |
| auto transport1 = ortc_factory1_->CreateUdpTransport(AF_INET).MoveValue(); |
| auto transport2 = ortc_factory2_->CreateUdpTransport(AF_INET).MoveValue(); |
| transport1->SetRemoteAddress( |
| rtc::SocketAddress(virtual_socket_server_.GetDefaultRoute(AF_INET), |
| transport2->GetLocalAddress().port())); |
| transport2->SetRemoteAddress( |
| rtc::SocketAddress(virtual_socket_server_.GetDefaultRoute(AF_INET), |
| transport1->GetLocalAddress().port())); |
| return {std::move(transport1), std::move(transport2)}; |
| } |
| |
| // Creates one transport controller each for |ortc_factory1_| and |
| // |ortc_factory2_|. |
| RtpTransportControllerPair CreateRtpTransportControllerPair() { |
| return {ortc_factory1_->CreateRtpTransportController().MoveValue(), |
| ortc_factory2_->CreateRtpTransportController().MoveValue()}; |
| } |
| |
| // Helper function that creates a pair of RtpTransports between |
| // |ortc_factory1_| and |ortc_factory2_|. Expected to be called with the |
| // result of CreateAndConnectUdpTransportPair. |rtcp_udp_transports| can be |
| // empty if RTCP muxing is used. |transport_controllers| can be empty if |
| // these transports are being created using a default transport controller. |
| RtpTransportPair CreateRtpTransportPair( |
| const RtpTransportParameters& parameters, |
| const UdpTransportPair& rtp_udp_transports, |
| const UdpTransportPair& rtcp_udp_transports, |
| const RtpTransportControllerPair& transport_controllers) { |
| auto transport_result1 = ortc_factory1_->CreateRtpTransport( |
| parameters, rtp_udp_transports.first.get(), |
| rtcp_udp_transports.first.get(), transport_controllers.first.get()); |
| auto transport_result2 = ortc_factory2_->CreateRtpTransport( |
| parameters, rtp_udp_transports.second.get(), |
| rtcp_udp_transports.second.get(), transport_controllers.second.get()); |
| return {transport_result1.MoveValue(), transport_result2.MoveValue()}; |
| } |
| |
| SrtpTransportPair CreateSrtpTransportPair( |
| const RtpTransportParameters& parameters, |
| const UdpTransportPair& rtp_udp_transports, |
| const UdpTransportPair& rtcp_udp_transports, |
| const RtpTransportControllerPair& transport_controllers) { |
| auto transport_result1 = ortc_factory1_->CreateSrtpTransport( |
| parameters, rtp_udp_transports.first.get(), |
| rtcp_udp_transports.first.get(), transport_controllers.first.get()); |
| auto transport_result2 = ortc_factory2_->CreateSrtpTransport( |
| parameters, rtp_udp_transports.second.get(), |
| rtcp_udp_transports.second.get(), transport_controllers.second.get()); |
| return {transport_result1.MoveValue(), transport_result2.MoveValue()}; |
| } |
| |
| // For convenience when |rtcp_udp_transports| and |transport_controllers| |
| // aren't needed. |
| RtpTransportPair CreateRtpTransportPair( |
| const RtpTransportParameters& parameters, |
| const UdpTransportPair& rtp_udp_transports) { |
| return CreateRtpTransportPair(parameters, rtp_udp_transports, |
| UdpTransportPair(), |
| RtpTransportControllerPair()); |
| } |
| |
| SrtpTransportPair CreateSrtpTransportPairAndSetKeys( |
| const RtpTransportParameters& parameters, |
| const UdpTransportPair& rtp_udp_transports) { |
| SrtpTransportPair srtp_transports = CreateSrtpTransportPair( |
| parameters, rtp_udp_transports, UdpTransportPair(), |
| RtpTransportControllerPair()); |
| EXPECT_TRUE(srtp_transports.first->SetSrtpSendKey(kTestCryptoParams1).ok()); |
| EXPECT_TRUE( |
| srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams2).ok()); |
| EXPECT_TRUE( |
| srtp_transports.second->SetSrtpSendKey(kTestCryptoParams2).ok()); |
| EXPECT_TRUE( |
| srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams1).ok()); |
| return srtp_transports; |
| } |
| |
| SrtpTransportPair CreateSrtpTransportPairAndSetMismatchingKeys( |
| const RtpTransportParameters& parameters, |
| const UdpTransportPair& rtp_udp_transports) { |
| SrtpTransportPair srtp_transports = CreateSrtpTransportPair( |
| parameters, rtp_udp_transports, UdpTransportPair(), |
| RtpTransportControllerPair()); |
| EXPECT_TRUE(srtp_transports.first->SetSrtpSendKey(kTestCryptoParams1).ok()); |
| EXPECT_TRUE( |
| srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams2).ok()); |
| EXPECT_TRUE( |
| srtp_transports.second->SetSrtpSendKey(kTestCryptoParams1).ok()); |
| EXPECT_TRUE( |
| srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams2).ok()); |
| return srtp_transports; |
| } |
| |
| // Ends up using fake audio capture module, which was passed into OrtcFactory |
| // on creation. |
| rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack( |
| const std::string& id, |
| OrtcFactoryInterface* ortc_factory) { |
| // Disable echo cancellation to make test more efficient. |
| cricket::AudioOptions options; |
| options.echo_cancellation.emplace(true); |
| rtc::scoped_refptr<webrtc::AudioSourceInterface> source = |
| ortc_factory->CreateAudioSource(options); |
| return ortc_factory->CreateAudioTrack(id, source); |
| } |
| |
| // Stores created capturer in |fake_video_capturers_|. |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> |
| CreateLocalVideoTrackAndFakeCapturer(const std::string& id, |
| OrtcFactoryInterface* ortc_factory) { |
| cricket::FakeVideoCapturer* fake_capturer = |
| new webrtc::FakePeriodicVideoCapturer(); |
| fake_video_capturers_.push_back(fake_capturer); |
| rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source = |
| ortc_factory->CreateVideoSource( |
| std::unique_ptr<cricket::VideoCapturer>(fake_capturer)); |
| return rtc::scoped_refptr<webrtc::VideoTrackInterface>( |
| ortc_factory->CreateVideoTrack(id, source)); |
| } |
| |
| // Helper function used to test two way RTP senders and receivers with basic |
| // configurations. |
| // If |expect_success| is true, waits for kDefaultTimeout for |
| // kDefaultNumFrames frames to be received by all RtpReceivers. |
| // If |expect_success| is false, simply waits for |kReceivingDuration|, and |
| // stores the number of received frames in |received_audio_frame1_| etc. |
| void BasicTwoWayRtpSendersAndReceiversTest(RtpTransportPair srtp_transports, |
| bool expect_success) { |
| received_audio_frames1_ = 0; |
| received_audio_frames2_ = 0; |
| rendered_video_frames1_ = 0; |
| rendered_video_frames2_ = 0; |
| // Create all the senders and receivers (four per endpoint). |
| auto audio_sender_result1 = ortc_factory1_->CreateRtpSender( |
| cricket::MEDIA_TYPE_AUDIO, srtp_transports.first.get()); |
| auto video_sender_result1 = ortc_factory1_->CreateRtpSender( |
| cricket::MEDIA_TYPE_VIDEO, srtp_transports.first.get()); |
| auto audio_receiver_result1 = ortc_factory1_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_AUDIO, srtp_transports.first.get()); |
| auto video_receiver_result1 = ortc_factory1_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_VIDEO, srtp_transports.first.get()); |
| ASSERT_TRUE(audio_sender_result1.ok()); |
| ASSERT_TRUE(video_sender_result1.ok()); |
| ASSERT_TRUE(audio_receiver_result1.ok()); |
| ASSERT_TRUE(video_receiver_result1.ok()); |
| auto audio_sender1 = audio_sender_result1.MoveValue(); |
| auto video_sender1 = video_sender_result1.MoveValue(); |
| auto audio_receiver1 = audio_receiver_result1.MoveValue(); |
| auto video_receiver1 = video_receiver_result1.MoveValue(); |
| |
| auto audio_sender_result2 = ortc_factory2_->CreateRtpSender( |
| cricket::MEDIA_TYPE_AUDIO, srtp_transports.second.get()); |
| auto video_sender_result2 = ortc_factory2_->CreateRtpSender( |
| cricket::MEDIA_TYPE_VIDEO, srtp_transports.second.get()); |
| auto audio_receiver_result2 = ortc_factory2_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_AUDIO, srtp_transports.second.get()); |
| auto video_receiver_result2 = ortc_factory2_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_VIDEO, srtp_transports.second.get()); |
| ASSERT_TRUE(audio_sender_result2.ok()); |
| ASSERT_TRUE(video_sender_result2.ok()); |
| ASSERT_TRUE(audio_receiver_result2.ok()); |
| ASSERT_TRUE(video_receiver_result2.ok()); |
| auto audio_sender2 = audio_sender_result2.MoveValue(); |
| auto video_sender2 = video_sender_result2.MoveValue(); |
| auto audio_receiver2 = audio_receiver_result2.MoveValue(); |
| auto video_receiver2 = video_receiver_result2.MoveValue(); |
| |
| // Add fake tracks. |
| RTCError error = audio_sender1->SetTrack( |
| CreateLocalAudioTrack("audio", ortc_factory1_.get())); |
| EXPECT_TRUE(error.ok()); |
| error = video_sender1->SetTrack( |
| CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get())); |
| EXPECT_TRUE(error.ok()); |
| error = audio_sender2->SetTrack( |
| CreateLocalAudioTrack("audio", ortc_factory2_.get())); |
| EXPECT_TRUE(error.ok()); |
| error = video_sender2->SetTrack( |
| CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory2_.get())); |
| EXPECT_TRUE(error.ok()); |
| |
| // "sent_X_parameters1" are the parameters that endpoint 1 sends with and |
| // endpoint 2 receives with. |
| RtpParameters sent_opus_parameters1 = |
| MakeMinimalOpusParametersWithSsrc(0xdeadbeef); |
| RtpParameters sent_vp8_parameters1 = |
| MakeMinimalVp8ParametersWithSsrc(0xbaadfeed); |
| RtpParameters sent_opus_parameters2 = |
| MakeMinimalOpusParametersWithSsrc(0x13333337); |
| RtpParameters sent_vp8_parameters2 = |
| MakeMinimalVp8ParametersWithSsrc(0x12345678); |
| |
| // Configure the senders' and receivers' parameters. |
| EXPECT_TRUE(audio_receiver1->Receive(sent_opus_parameters2).ok()); |
| EXPECT_TRUE(video_receiver1->Receive(sent_vp8_parameters2).ok()); |
| EXPECT_TRUE(audio_receiver2->Receive(sent_opus_parameters1).ok()); |
| EXPECT_TRUE(video_receiver2->Receive(sent_vp8_parameters1).ok()); |
| EXPECT_TRUE(audio_sender1->Send(sent_opus_parameters1).ok()); |
| EXPECT_TRUE(video_sender1->Send(sent_vp8_parameters1).ok()); |
| EXPECT_TRUE(audio_sender2->Send(sent_opus_parameters2).ok()); |
| EXPECT_TRUE(video_sender2->Send(sent_vp8_parameters2).ok()); |
| |
| FakeVideoTrackRenderer fake_video_renderer1( |
| static_cast<VideoTrackInterface*>(video_receiver1->GetTrack().get())); |
| FakeVideoTrackRenderer fake_video_renderer2( |
| static_cast<VideoTrackInterface*>(video_receiver2->GetTrack().get())); |
| |
| if (expect_success) { |
| EXPECT_TRUE_WAIT( |
| fake_audio_capture_module1_->frames_received() > kDefaultNumFrames && |
| fake_video_renderer1.num_rendered_frames() > kDefaultNumFrames && |
| fake_audio_capture_module2_->frames_received() > |
| kDefaultNumFrames && |
| fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames, |
| kDefaultTimeout) << "Audio capture module 1 received " |
| << fake_audio_capture_module1_->frames_received() |
| << " frames, Video renderer 1 rendered " |
| << fake_video_renderer1.num_rendered_frames() |
| << " frames, Audio capture module 2 received " |
| << fake_audio_capture_module2_->frames_received() |
| << " frames, Video renderer 2 rendered " |
| << fake_video_renderer2.num_rendered_frames() |
| << " frames."; |
| } else { |
| WAIT(false, kReceivingDuration); |
| rendered_video_frames1_ = fake_video_renderer1.num_rendered_frames(); |
| rendered_video_frames2_ = fake_video_renderer2.num_rendered_frames(); |
| received_audio_frames1_ = fake_audio_capture_module1_->frames_received(); |
| received_audio_frames2_ = fake_audio_capture_module2_->frames_received(); |
| } |
| } |
| |
| rtc::VirtualSocketServer virtual_socket_server_; |
| rtc::Thread network_thread_; |
| rtc::FakeNetworkManager fake_network_manager_; |
| rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module1_; |
| rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module2_; |
| std::unique_ptr<OrtcFactoryInterface> ortc_factory1_; |
| std::unique_ptr<OrtcFactoryInterface> ortc_factory2_; |
| // Actually owned by video tracks. |
| std::vector<cricket::FakeVideoCapturer*> fake_video_capturers_; |
| int received_audio_frames1_ = 0; |
| int received_audio_frames2_ = 0; |
| int rendered_video_frames1_ = 0; |
| int rendered_video_frames2_ = 0; |
| }; |
| |
| // Disable for TSan v2, see |
| // https://bugs.chromium.org/p/webrtc/issues/detail?id=7366 for details. |
| #if !defined(THREAD_SANITIZER) |
| |
| // Very basic end-to-end test with a single pair of audio RTP sender and |
| // receiver. |
| // |
| // Uses muxed RTCP, and minimal parameters with a hard-coded config that's |
| // known to work. |
| TEST_F(OrtcFactoryIntegrationTest, BasicOneWayAudioRtpSenderAndReceiver) { |
| auto udp_transports = CreateAndConnectUdpTransportPair(); |
| auto rtp_transports = |
| CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports); |
| |
| auto sender_result = ortc_factory1_->CreateRtpSender( |
| cricket::MEDIA_TYPE_AUDIO, rtp_transports.first.get()); |
| auto receiver_result = ortc_factory2_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_AUDIO, rtp_transports.second.get()); |
| ASSERT_TRUE(sender_result.ok()); |
| ASSERT_TRUE(receiver_result.ok()); |
| auto sender = sender_result.MoveValue(); |
| auto receiver = receiver_result.MoveValue(); |
| |
| RTCError error = |
| sender->SetTrack(CreateLocalAudioTrack("audio", ortc_factory1_.get())); |
| EXPECT_TRUE(error.ok()); |
| |
| RtpParameters opus_parameters = MakeMinimalOpusParameters(); |
| EXPECT_TRUE(receiver->Receive(opus_parameters).ok()); |
| EXPECT_TRUE(sender->Send(opus_parameters).ok()); |
| // Sender and receiver are connected and configured; audio frames should be |
| // able to flow at this point. |
| EXPECT_TRUE_WAIT( |
| fake_audio_capture_module2_->frames_received() > kDefaultNumFrames, |
| kDefaultTimeout); |
| } |
| |
| // Very basic end-to-end test with a single pair of video RTP sender and |
| // receiver. |
| // |
| // Uses muxed RTCP, and minimal parameters with a hard-coded config that's |
| // known to work. |
| TEST_F(OrtcFactoryIntegrationTest, BasicOneWayVideoRtpSenderAndReceiver) { |
| auto udp_transports = CreateAndConnectUdpTransportPair(); |
| auto rtp_transports = |
| CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports); |
| |
| auto sender_result = ortc_factory1_->CreateRtpSender( |
| cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get()); |
| auto receiver_result = ortc_factory2_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get()); |
| ASSERT_TRUE(sender_result.ok()); |
| ASSERT_TRUE(receiver_result.ok()); |
| auto sender = sender_result.MoveValue(); |
| auto receiver = receiver_result.MoveValue(); |
| |
| RTCError error = sender->SetTrack( |
| CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get())); |
| EXPECT_TRUE(error.ok()); |
| |
| RtpParameters vp8_parameters = MakeMinimalVp8Parameters(); |
| EXPECT_TRUE(receiver->Receive(vp8_parameters).ok()); |
| EXPECT_TRUE(sender->Send(vp8_parameters).ok()); |
| FakeVideoTrackRenderer fake_renderer( |
| static_cast<VideoTrackInterface*>(receiver->GetTrack().get())); |
| // Sender and receiver are connected and configured; video frames should be |
| // able to flow at this point. |
| EXPECT_TRUE_WAIT(fake_renderer.num_rendered_frames() > kDefaultNumFrames, |
| kDefaultTimeout); |
| } |
| |
| // Test that if the track is changed while sending, the sender seamlessly |
| // transitions to sending it and frames are received end-to-end. |
| // |
| // Only doing this for video, since given that audio is sourced from a single |
| // fake audio capture module, the audio track is just a dummy object. |
| // TODO(deadbeef): Change this when possible. |
| TEST_F(OrtcFactoryIntegrationTest, SetTrackWhileSending) { |
| auto udp_transports = CreateAndConnectUdpTransportPair(); |
| auto rtp_transports = |
| CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports); |
| |
| auto sender_result = ortc_factory1_->CreateRtpSender( |
| cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get()); |
| auto receiver_result = ortc_factory2_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get()); |
| ASSERT_TRUE(sender_result.ok()); |
| ASSERT_TRUE(receiver_result.ok()); |
| auto sender = sender_result.MoveValue(); |
| auto receiver = receiver_result.MoveValue(); |
| |
| RTCError error = sender->SetTrack( |
| CreateLocalVideoTrackAndFakeCapturer("video_1", ortc_factory1_.get())); |
| EXPECT_TRUE(error.ok()); |
| RtpParameters vp8_parameters = MakeMinimalVp8Parameters(); |
| EXPECT_TRUE(receiver->Receive(vp8_parameters).ok()); |
| EXPECT_TRUE(sender->Send(vp8_parameters).ok()); |
| FakeVideoTrackRenderer fake_renderer( |
| static_cast<VideoTrackInterface*>(receiver->GetTrack().get())); |
| // Expect for some initial number of frames to be received. |
| EXPECT_TRUE_WAIT(fake_renderer.num_rendered_frames() > kDefaultNumFrames, |
| kDefaultTimeout); |
| // Stop the old capturer, set a new track, and verify new frames are received |
| // from the new track. Stopping the old capturer ensures that we aren't |
| // actually still getting frames from it. |
| fake_video_capturers_[0]->Stop(); |
| int prev_num_frames = fake_renderer.num_rendered_frames(); |
| error = sender->SetTrack( |
| CreateLocalVideoTrackAndFakeCapturer("video_2", ortc_factory1_.get())); |
| EXPECT_TRUE(error.ok()); |
| EXPECT_TRUE_WAIT( |
| fake_renderer.num_rendered_frames() > kDefaultNumFrames + prev_num_frames, |
| kDefaultTimeout); |
| } |
| |
| // End-to-end test with two pairs of RTP senders and receivers, for audio and |
| // video. |
| // |
| // Uses muxed RTCP, and minimal parameters with hard-coded configs that are |
| // known to work. |
| TEST_F(OrtcFactoryIntegrationTest, |
| BasicTwoWayAudioVideoRtpSendersAndReceivers) { |
| auto udp_transports = CreateAndConnectUdpTransportPair(); |
| auto rtp_transports = |
| CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports); |
| bool expect_success = true; |
| BasicTwoWayRtpSendersAndReceiversTest(std::move(rtp_transports), |
| expect_success); |
| } |
| |
| #if !(defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_64_BITS) && !defined(NDEBUG)) |
| TEST_F(OrtcFactoryIntegrationTest, |
| BasicTwoWayAudioVideoSrtpSendersAndReceivers) { |
| auto udp_transports = CreateAndConnectUdpTransportPair(); |
| auto srtp_transports = CreateSrtpTransportPairAndSetKeys( |
| MakeRtcpMuxParameters(), udp_transports); |
| bool expect_success = true; |
| BasicTwoWayRtpSendersAndReceiversTest(std::move(srtp_transports), |
| expect_success); |
| } |
| #endif |
| |
| // Tests that the packets cannot be decoded if the keys are mismatched. |
| TEST_F(OrtcFactoryIntegrationTest, SrtpSendersAndReceiversWithMismatchingKeys) { |
| auto udp_transports = CreateAndConnectUdpTransportPair(); |
| auto srtp_transports = CreateSrtpTransportPairAndSetMismatchingKeys( |
| MakeRtcpMuxParameters(), udp_transports); |
| bool expect_success = false; |
| BasicTwoWayRtpSendersAndReceiversTest(std::move(srtp_transports), |
| expect_success); |
| // No frames are expected to be decoded. |
| EXPECT_TRUE(received_audio_frames1_ == 0 && received_audio_frames2_ == 0 && |
| rendered_video_frames1_ == 0 && rendered_video_frames2_ == 0); |
| } |
| |
| // Tests that the frames cannot be decoded if only one side uses SRTP. |
| TEST_F(OrtcFactoryIntegrationTest, OneSideSrtpSenderAndReceiver) { |
| auto rtcp_parameters = MakeRtcpMuxParameters(); |
| auto udp_transports = CreateAndConnectUdpTransportPair(); |
| auto rtcp_udp_transports = UdpTransportPair(); |
| auto transport_controllers = RtpTransportControllerPair(); |
| auto transport_result1 = ortc_factory1_->CreateRtpTransport( |
| rtcp_parameters, udp_transports.first.get(), |
| rtcp_udp_transports.first.get(), transport_controllers.first.get()); |
| auto transport_result2 = ortc_factory2_->CreateSrtpTransport( |
| rtcp_parameters, udp_transports.second.get(), |
| rtcp_udp_transports.second.get(), transport_controllers.second.get()); |
| |
| auto rtp_transport = transport_result1.MoveValue(); |
| auto srtp_transport = transport_result2.MoveValue(); |
| EXPECT_TRUE(srtp_transport->SetSrtpSendKey(kTestCryptoParams1).ok()); |
| EXPECT_TRUE(srtp_transport->SetSrtpReceiveKey(kTestCryptoParams2).ok()); |
| bool expect_success = false; |
| BasicTwoWayRtpSendersAndReceiversTest( |
| {std::move(rtp_transport), std::move(srtp_transport)}, expect_success); |
| |
| // The SRTP side is not expected to decode any audio or video frames. |
| // The RTP side is not expected to decode any video frames while it is |
| // possible that the encrypted audio frames can be accidentally decoded which |
| // is why received_audio_frames1_ is not validated. |
| EXPECT_TRUE(received_audio_frames2_ == 0 && rendered_video_frames1_ == 0 && |
| rendered_video_frames2_ == 0); |
| } |
| |
| // End-to-end test with two pairs of RTP senders and receivers, for audio and |
| // video. Unlike the test above, this attempts to make the parameters as |
| // complex as possible. The senders and receivers use the SRTP transport with |
| // different keys. |
| // |
| // Uses non-muxed RTCP, with separate audio/video transports, and a full set of |
| // parameters, as would normally be used in a PeerConnection. |
| // |
| // TODO(deadbeef): Update this test as more audio/video features become |
| // supported. |
| TEST_F(OrtcFactoryIntegrationTest, |
| FullTwoWayAudioVideoSrtpSendersAndReceivers) { |
| // We want four pairs of UDP transports for this test, for audio/video and |
| // RTP/RTCP. |
| auto audio_rtp_udp_transports = CreateAndConnectUdpTransportPair(); |
| auto audio_rtcp_udp_transports = CreateAndConnectUdpTransportPair(); |
| auto video_rtp_udp_transports = CreateAndConnectUdpTransportPair(); |
| auto video_rtcp_udp_transports = CreateAndConnectUdpTransportPair(); |
| |
| // Since we have multiple RTP transports on each side, we need an RTP |
| // transport controller. |
| auto transport_controllers = CreateRtpTransportControllerPair(); |
| |
| RtpTransportParameters audio_rtp_transport_parameters; |
| audio_rtp_transport_parameters.rtcp.mux = false; |
| auto audio_srtp_transports = CreateSrtpTransportPair( |
| audio_rtp_transport_parameters, audio_rtp_udp_transports, |
| audio_rtcp_udp_transports, transport_controllers); |
| |
| RtpTransportParameters video_rtp_transport_parameters; |
| video_rtp_transport_parameters.rtcp.mux = false; |
| video_rtp_transport_parameters.rtcp.reduced_size = true; |
| auto video_srtp_transports = CreateSrtpTransportPair( |
| video_rtp_transport_parameters, video_rtp_udp_transports, |
| video_rtcp_udp_transports, transport_controllers); |
| |
| // Set keys for SRTP transports. |
| audio_srtp_transports.first->SetSrtpSendKey(kTestCryptoParams1); |
| audio_srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams2); |
| video_srtp_transports.first->SetSrtpSendKey(kTestCryptoParams3); |
| video_srtp_transports.first->SetSrtpReceiveKey(kTestCryptoParams4); |
| |
| audio_srtp_transports.second->SetSrtpSendKey(kTestCryptoParams2); |
| audio_srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams1); |
| video_srtp_transports.second->SetSrtpSendKey(kTestCryptoParams4); |
| video_srtp_transports.second->SetSrtpReceiveKey(kTestCryptoParams3); |
| |
| // Create all the senders and receivers (four per endpoint). |
| auto audio_sender_result1 = ortc_factory1_->CreateRtpSender( |
| cricket::MEDIA_TYPE_AUDIO, audio_srtp_transports.first.get()); |
| auto video_sender_result1 = ortc_factory1_->CreateRtpSender( |
| cricket::MEDIA_TYPE_VIDEO, video_srtp_transports.first.get()); |
| auto audio_receiver_result1 = ortc_factory1_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_AUDIO, audio_srtp_transports.first.get()); |
| auto video_receiver_result1 = ortc_factory1_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_VIDEO, video_srtp_transports.first.get()); |
| ASSERT_TRUE(audio_sender_result1.ok()); |
| ASSERT_TRUE(video_sender_result1.ok()); |
| ASSERT_TRUE(audio_receiver_result1.ok()); |
| ASSERT_TRUE(video_receiver_result1.ok()); |
| auto audio_sender1 = audio_sender_result1.MoveValue(); |
| auto video_sender1 = video_sender_result1.MoveValue(); |
| auto audio_receiver1 = audio_receiver_result1.MoveValue(); |
| auto video_receiver1 = video_receiver_result1.MoveValue(); |
| |
| auto audio_sender_result2 = ortc_factory2_->CreateRtpSender( |
| cricket::MEDIA_TYPE_AUDIO, audio_srtp_transports.second.get()); |
| auto video_sender_result2 = ortc_factory2_->CreateRtpSender( |
| cricket::MEDIA_TYPE_VIDEO, video_srtp_transports.second.get()); |
| auto audio_receiver_result2 = ortc_factory2_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_AUDIO, audio_srtp_transports.second.get()); |
| auto video_receiver_result2 = ortc_factory2_->CreateRtpReceiver( |
| cricket::MEDIA_TYPE_VIDEO, video_srtp_transports.second.get()); |
| ASSERT_TRUE(audio_sender_result2.ok()); |
| ASSERT_TRUE(video_sender_result2.ok()); |
| ASSERT_TRUE(audio_receiver_result2.ok()); |
| ASSERT_TRUE(video_receiver_result2.ok()); |
| auto audio_sender2 = audio_sender_result2.MoveValue(); |
| auto video_sender2 = video_sender_result2.MoveValue(); |
| auto audio_receiver2 = audio_receiver_result2.MoveValue(); |
| auto video_receiver2 = video_receiver_result2.MoveValue(); |
| |
| RTCError error = audio_sender1->SetTrack( |
| CreateLocalAudioTrack("audio", ortc_factory1_.get())); |
| EXPECT_TRUE(error.ok()); |
| error = video_sender1->SetTrack( |
| CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get())); |
| EXPECT_TRUE(error.ok()); |
| error = audio_sender2->SetTrack( |
| CreateLocalAudioTrack("audio", ortc_factory2_.get())); |
| EXPECT_TRUE(error.ok()); |
| error = video_sender2->SetTrack( |
| CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory2_.get())); |
| EXPECT_TRUE(error.ok()); |
| |
| // Use different codecs in different directions for extra challenge. |
| RtpParameters opus_send_parameters = MakeFullOpusParameters(); |
| RtpParameters isac_send_parameters = MakeFullIsacParameters(); |
| RtpParameters vp8_send_parameters = MakeFullVp8Parameters(); |
| RtpParameters vp9_send_parameters = MakeFullVp9Parameters(); |
| |
| // Remove "payload_type" from receive parameters. Receiver will need to |
| // discern the payload type from packets received. |
| RtpParameters opus_receive_parameters = opus_send_parameters; |
| RtpParameters isac_receive_parameters = isac_send_parameters; |
| RtpParameters vp8_receive_parameters = vp8_send_parameters; |
| RtpParameters vp9_receive_parameters = vp9_send_parameters; |
| opus_receive_parameters.encodings[0].codec_payload_type.reset(); |
| isac_receive_parameters.encodings[0].codec_payload_type.reset(); |
| vp8_receive_parameters.encodings[0].codec_payload_type.reset(); |
| vp9_receive_parameters.encodings[0].codec_payload_type.reset(); |
| |
| // Configure the senders' and receivers' parameters. |
| // |
| // Note: Intentionally, the top codec in the receive parameters does not |
| // match the codec sent by the other side. If "Receive" is called with a list |
| // of codecs, the receiver should be prepared to receive any of them, not |
| // just the one on top. |
| EXPECT_TRUE(audio_receiver1->Receive(opus_receive_parameters).ok()); |
| EXPECT_TRUE(video_receiver1->Receive(vp8_receive_parameters).ok()); |
| EXPECT_TRUE(audio_receiver2->Receive(isac_receive_parameters).ok()); |
| EXPECT_TRUE(video_receiver2->Receive(vp9_receive_parameters).ok()); |
| EXPECT_TRUE(audio_sender1->Send(opus_send_parameters).ok()); |
| EXPECT_TRUE(video_sender1->Send(vp8_send_parameters).ok()); |
| EXPECT_TRUE(audio_sender2->Send(isac_send_parameters).ok()); |
| EXPECT_TRUE(video_sender2->Send(vp9_send_parameters).ok()); |
| |
| FakeVideoTrackRenderer fake_video_renderer1( |
| static_cast<VideoTrackInterface*>(video_receiver1->GetTrack().get())); |
| FakeVideoTrackRenderer fake_video_renderer2( |
| static_cast<VideoTrackInterface*>(video_receiver2->GetTrack().get())); |
| |
| // Senders and receivers are connected and configured; audio and video frames |
| // should be able to flow at this point. |
| EXPECT_TRUE_WAIT( |
| fake_audio_capture_module1_->frames_received() > kDefaultNumFrames && |
| fake_video_renderer1.num_rendered_frames() > kDefaultNumFrames && |
| fake_audio_capture_module2_->frames_received() > kDefaultNumFrames && |
| fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames, |
| kDefaultTimeout); |
| } |
| |
| // TODO(deadbeef): End-to-end test for multiple senders/receivers of the same |
| // media type, once that's supported. Currently, it is not because the |
| // BaseChannel model relies on there being a single VoiceChannel and |
| // VideoChannel, and these only support a single set of codecs/etc. per |
| // send/receive direction. |
| |
| // TODO(deadbeef): End-to-end test for simulcast, once that's supported by this |
| // API. |
| |
| #endif // if !defined(THREAD_SANITIZER) |
| |
| } // namespace webrtc |
