| /* |
| * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| // Disable for TSan v2, see |
| // https://code.google.com/p/webrtc/issues/detail?id=1205 for details. |
| #if !defined(THREAD_SANITIZER) |
| |
| #include <stdio.h> |
| |
| #include <algorithm> |
| #include <functional> |
| #include <list> |
| #include <map> |
| #include <memory> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/mediastreaminterface.h" |
| #include "api/peerconnectioninterface.h" |
| #include "api/peerconnectionproxy.h" |
| #include "api/rtpreceiverinterface.h" |
| #include "api/test/loopback_media_transport.h" |
| #include "api/umametrics.h" |
| #include "api/video_codecs/builtin_video_decoder_factory.h" |
| #include "api/video_codecs/builtin_video_encoder_factory.h" |
| #include "api/video_codecs/sdp_video_format.h" |
| #include "call/call.h" |
| #include "logging/rtc_event_log/fake_rtc_event_log_factory.h" |
| #include "logging/rtc_event_log/rtc_event_log_factory_interface.h" |
| #include "media/engine/fakewebrtcvideoengine.h" |
| #include "media/engine/webrtcmediaengine.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "p2p/base/mockasyncresolver.h" |
| #include "p2p/base/p2pconstants.h" |
| #include "p2p/base/portinterface.h" |
| #include "p2p/base/teststunserver.h" |
| #include "p2p/base/testturncustomizer.h" |
| #include "p2p/base/testturnserver.h" |
| #include "p2p/client/basicportallocator.h" |
| #include "pc/dtmfsender.h" |
| #include "pc/localaudiosource.h" |
| #include "pc/mediasession.h" |
| #include "pc/peerconnection.h" |
| #include "pc/peerconnectionfactory.h" |
| #include "pc/rtpmediautils.h" |
| #include "pc/sessiondescription.h" |
| #include "pc/test/fakeaudiocapturemodule.h" |
| #include "pc/test/fakeperiodicvideotracksource.h" |
| #include "pc/test/fakertccertificategenerator.h" |
| #include "pc/test/fakevideotrackrenderer.h" |
| #include "pc/test/mockpeerconnectionobservers.h" |
| #include "rtc_base/fakenetwork.h" |
| #include "rtc_base/firewallsocketserver.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/testcertificateverifier.h" |
| #include "rtc_base/timeutils.h" |
| #include "rtc_base/virtualsocketserver.h" |
| #include "system_wrappers/include/metrics.h" |
| #include "test/gmock.h" |
| |
| using cricket::ContentInfo; |
| using cricket::FakeWebRtcVideoDecoder; |
| using cricket::FakeWebRtcVideoDecoderFactory; |
| using cricket::FakeWebRtcVideoEncoder; |
| using cricket::FakeWebRtcVideoEncoderFactory; |
| using cricket::MediaContentDescription; |
| using cricket::StreamParams; |
| using rtc::SocketAddress; |
| using ::testing::Combine; |
| using ::testing::ElementsAre; |
| using ::testing::Return; |
| using ::testing::SetArgPointee; |
| using ::testing::Values; |
| using ::testing::_; |
| using webrtc::DataBuffer; |
| using webrtc::DataChannelInterface; |
| using webrtc::DtmfSender; |
| using webrtc::DtmfSenderInterface; |
| using webrtc::DtmfSenderObserverInterface; |
| using webrtc::FakeVideoTrackRenderer; |
| using webrtc::MediaStreamInterface; |
| using webrtc::MediaStreamTrackInterface; |
| using webrtc::MockCreateSessionDescriptionObserver; |
| using webrtc::MockDataChannelObserver; |
| using webrtc::MockSetSessionDescriptionObserver; |
| using webrtc::MockStatsObserver; |
| using webrtc::ObserverInterface; |
| using webrtc::PeerConnection; |
| using webrtc::PeerConnectionInterface; |
| using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; |
| using webrtc::PeerConnectionFactory; |
| using webrtc::PeerConnectionProxy; |
| using webrtc::RTCErrorType; |
| using webrtc::RTCTransportStats; |
| using webrtc::RtpSenderInterface; |
| using webrtc::RtpReceiverInterface; |
| using webrtc::RtpSenderInterface; |
| using webrtc::RtpTransceiverDirection; |
| using webrtc::RtpTransceiverInit; |
| using webrtc::RtpTransceiverInterface; |
| using webrtc::SdpSemantics; |
| using webrtc::SdpType; |
| using webrtc::SessionDescriptionInterface; |
| using webrtc::StreamCollectionInterface; |
| using webrtc::VideoTrackInterface; |
| |
| namespace { |
| |
| static const int kDefaultTimeout = 10000; |
| static const int kMaxWaitForStatsMs = 3000; |
| static const int kMaxWaitForActivationMs = 5000; |
| static const int kMaxWaitForFramesMs = 10000; |
| // Default number of audio/video frames to wait for before considering a test |
| // successful. |
| static const int kDefaultExpectedAudioFrameCount = 3; |
| static const int kDefaultExpectedVideoFrameCount = 3; |
| |
| static const char kDataChannelLabel[] = "data_channel"; |
| |
| // SRTP cipher name negotiated by the tests. This must be updated if the |
| // default changes. |
| static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_80; |
| static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM; |
| |
| static const SocketAddress kDefaultLocalAddress("192.168.1.1", 0); |
| |
| // Helper function for constructing offer/answer options to initiate an ICE |
| // restart. |
| PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions() { |
| PeerConnectionInterface::RTCOfferAnswerOptions options; |
| options.ice_restart = true; |
| return options; |
| } |
| |
| // Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic" |
| // attribute from received SDP, simulating a legacy endpoint. |
| void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) { |
| for (ContentInfo& content : desc->contents()) { |
| content.media_description()->mutable_streams().clear(); |
| } |
| desc->set_msid_supported(false); |
| } |
| |
| // Removes all stream information besides the stream ids, simulating an |
| // endpoint that only signals a=msid lines to convey stream_ids. |
| void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) { |
| for (ContentInfo& content : desc->contents()) { |
| std::string track_id; |
| std::vector<std::string> stream_ids; |
| if (!content.media_description()->streams().empty()) { |
| const StreamParams& first_stream = |
| content.media_description()->streams()[0]; |
| track_id = first_stream.id; |
| stream_ids = first_stream.stream_ids(); |
| } |
| content.media_description()->mutable_streams().clear(); |
| StreamParams new_stream; |
| new_stream.id = track_id; |
| new_stream.set_stream_ids(stream_ids); |
| content.media_description()->AddStream(new_stream); |
| } |
| } |
| |
| int FindFirstMediaStatsIndexByKind( |
| const std::string& kind, |
| const std::vector<const webrtc::RTCMediaStreamTrackStats*>& |
| media_stats_vec) { |
| for (size_t i = 0; i < media_stats_vec.size(); i++) { |
| if (media_stats_vec[i]->kind.ValueToString() == kind) { |
| return i; |
| } |
| } |
| return -1; |
| } |
| |
| class SignalingMessageReceiver { |
| public: |
| virtual void ReceiveSdpMessage(SdpType type, const std::string& msg) = 0; |
| virtual void ReceiveIceMessage(const std::string& sdp_mid, |
| int sdp_mline_index, |
| const std::string& msg) = 0; |
| |
| protected: |
| SignalingMessageReceiver() {} |
| virtual ~SignalingMessageReceiver() {} |
| }; |
| |
| class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface { |
| public: |
| explicit MockRtpReceiverObserver(cricket::MediaType media_type) |
| : expected_media_type_(media_type) {} |
| |
| void OnFirstPacketReceived(cricket::MediaType media_type) override { |
| ASSERT_EQ(expected_media_type_, media_type); |
| first_packet_received_ = true; |
| } |
| |
| bool first_packet_received() const { return first_packet_received_; } |
| |
| virtual ~MockRtpReceiverObserver() {} |
| |
| private: |
| bool first_packet_received_ = false; |
| cricket::MediaType expected_media_type_; |
| }; |
| |
| // Used by PeerConnectionWrapper::OnIceCandidate to allow a test to modify an |
| // ICE candidate before it is signaled. |
| class IceCandidateReplacerInterface { |
| public: |
| virtual ~IceCandidateReplacerInterface() = default; |
| // Return nullptr to drop the candidate (it won't be signaled to the other |
| // side). |
| virtual std::unique_ptr<webrtc::IceCandidateInterface> ReplaceCandidate( |
| const webrtc::IceCandidateInterface*) = 0; |
| }; |
| |
| // Helper class that wraps a peer connection, observes it, and can accept |
| // signaling messages from another wrapper. |
| // |
| // Uses a fake network, fake A/V capture, and optionally fake |
| // encoders/decoders, though they aren't used by default since they don't |
| // advertise support of any codecs. |
| // TODO(steveanton): See how this could become a subclass of |
| // PeerConnectionWrapper defined in peerconnectionwrapper.h. |
| class PeerConnectionWrapper : public webrtc::PeerConnectionObserver, |
| public SignalingMessageReceiver { |
| public: |
| // Different factory methods for convenience. |
| // TODO(deadbeef): Could use the pattern of: |
| // |
| // PeerConnectionWrapper = |
| // WrapperBuilder.WithConfig(...).WithOptions(...).build(); |
| // |
| // To reduce some code duplication. |
| static PeerConnectionWrapper* CreateWithDtlsIdentityStore( |
| const std::string& debug_name, |
| std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, |
| rtc::Thread* network_thread, |
| rtc::Thread* worker_thread) { |
| PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name)); |
| webrtc::PeerConnectionDependencies dependencies(nullptr); |
| dependencies.cert_generator = std::move(cert_generator); |
| if (!client->Init(nullptr, nullptr, std::move(dependencies), network_thread, |
| worker_thread, nullptr, |
| /*media_transport_factory=*/nullptr)) { |
| delete client; |
| return nullptr; |
| } |
| return client; |
| } |
| |
| webrtc::PeerConnectionFactoryInterface* pc_factory() const { |
| return peer_connection_factory_.get(); |
| } |
| |
| webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); } |
| |
| // If a signaling message receiver is set (via ConnectFakeSignaling), this |
| // will set the whole offer/answer exchange in motion. Just need to wait for |
| // the signaling state to reach "stable". |
| void CreateAndSetAndSignalOffer() { |
| auto offer = CreateOffer(); |
| ASSERT_NE(nullptr, offer); |
| EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(offer))); |
| } |
| |
| // Sets the options to be used when CreateAndSetAndSignalOffer is called, or |
| // when a remote offer is received (via fake signaling) and an answer is |
| // generated. By default, uses default options. |
| void SetOfferAnswerOptions( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& options) { |
| offer_answer_options_ = options; |
| } |
| |
| // Set a callback to be invoked when SDP is received via the fake signaling |
| // channel, which provides an opportunity to munge (modify) the SDP. This is |
| // used to test SDP being applied that a PeerConnection would normally not |
| // generate, but a non-JSEP endpoint might. |
| void SetReceivedSdpMunger( |
| std::function<void(cricket::SessionDescription*)> munger) { |
| received_sdp_munger_ = std::move(munger); |
| } |
| |
| // Similar to the above, but this is run on SDP immediately after it's |
| // generated. |
| void SetGeneratedSdpMunger( |
| std::function<void(cricket::SessionDescription*)> munger) { |
| generated_sdp_munger_ = std::move(munger); |
| } |
| |
| // Set a callback to be invoked when a remote offer is received via the fake |
| // signaling channel. This provides an opportunity to change the |
| // PeerConnection state before an answer is created and sent to the caller. |
| void SetRemoteOfferHandler(std::function<void()> handler) { |
| remote_offer_handler_ = std::move(handler); |
| } |
| |
| void SetLocalIceCandidateReplacer( |
| std::unique_ptr<IceCandidateReplacerInterface> replacer) { |
| local_ice_candidate_replacer_ = std::move(replacer); |
| } |
| |
| // Every ICE connection state in order that has been seen by the observer. |
| std::vector<PeerConnectionInterface::IceConnectionState> |
| ice_connection_state_history() const { |
| return ice_connection_state_history_; |
| } |
| void clear_ice_connection_state_history() { |
| ice_connection_state_history_.clear(); |
| } |
| |
| // Every PeerConnection state in order that has been seen by the observer. |
| std::vector<PeerConnectionInterface::PeerConnectionState> |
| peer_connection_state_history() const { |
| return peer_connection_state_history_; |
| } |
| |
| // Every ICE gathering state in order that has been seen by the observer. |
| std::vector<PeerConnectionInterface::IceGatheringState> |
| ice_gathering_state_history() const { |
| return ice_gathering_state_history_; |
| } |
| |
| void AddAudioVideoTracks() { |
| AddAudioTrack(); |
| AddVideoTrack(); |
| } |
| |
| rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack() { |
| return AddTrack(CreateLocalAudioTrack()); |
| } |
| |
| rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack() { |
| return AddTrack(CreateLocalVideoTrack()); |
| } |
| |
| rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() { |
| cricket::AudioOptions options; |
| // Disable highpass filter so that we can get all the test audio frames. |
| options.highpass_filter = false; |
| rtc::scoped_refptr<webrtc::AudioSourceInterface> source = |
| peer_connection_factory_->CreateAudioSource(options); |
| // TODO(perkj): Test audio source when it is implemented. Currently audio |
| // always use the default input. |
| return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(), |
| source); |
| } |
| |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() { |
| webrtc::FakePeriodicVideoSource::Config config; |
| config.timestamp_offset_ms = rtc::TimeMillis(); |
| return CreateLocalVideoTrackInternal(config); |
| } |
| |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> |
| CreateLocalVideoTrackWithConfig( |
| webrtc::FakePeriodicVideoSource::Config config) { |
| return CreateLocalVideoTrackInternal(config); |
| } |
| |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> |
| CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) { |
| webrtc::FakePeriodicVideoSource::Config config; |
| config.rotation = rotation; |
| config.timestamp_offset_ms = rtc::TimeMillis(); |
| return CreateLocalVideoTrackInternal(config); |
| } |
| |
| rtc::scoped_refptr<RtpSenderInterface> AddTrack( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| const std::vector<std::string>& stream_ids = {}) { |
| auto result = pc()->AddTrack(track, stream_ids); |
| EXPECT_EQ(RTCErrorType::NONE, result.error().type()); |
| return result.MoveValue(); |
| } |
| |
| std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType( |
| cricket::MediaType media_type) { |
| std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers; |
| for (auto receiver : pc()->GetReceivers()) { |
| if (receiver->media_type() == media_type) { |
| receivers.push_back(receiver); |
| } |
| } |
| return receivers; |
| } |
| |
| rtc::scoped_refptr<RtpTransceiverInterface> GetFirstTransceiverOfType( |
| cricket::MediaType media_type) { |
| for (auto transceiver : pc()->GetTransceivers()) { |
| if (transceiver->receiver()->media_type() == media_type) { |
| return transceiver; |
| } |
| } |
| return nullptr; |
| } |
| |
| bool SignalingStateStable() { |
| return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable; |
| } |
| |
| void CreateDataChannel() { CreateDataChannel(nullptr); } |
| |
| void CreateDataChannel(const webrtc::DataChannelInit* init) { |
| CreateDataChannel(kDataChannelLabel, init); |
| } |
| |
| void CreateDataChannel(const std::string& label, |
| const webrtc::DataChannelInit* init) { |
| data_channel_ = pc()->CreateDataChannel(label, init); |
| ASSERT_TRUE(data_channel_.get() != nullptr); |
| data_observer_.reset(new MockDataChannelObserver(data_channel_)); |
| } |
| |
| DataChannelInterface* data_channel() { return data_channel_; } |
| const MockDataChannelObserver* data_observer() const { |
| return data_observer_.get(); |
| } |
| |
| int audio_frames_received() const { |
| return fake_audio_capture_module_->frames_received(); |
| } |
| |
| // Takes minimum of video frames received for each track. |
| // |
| // Can be used like: |
| // EXPECT_GE(expected_frames, min_video_frames_received_per_track()); |
| // |
| // To ensure that all video tracks received at least a certain number of |
| // frames. |
| int min_video_frames_received_per_track() const { |
| int min_frames = INT_MAX; |
| if (fake_video_renderers_.empty()) { |
| return 0; |
| } |
| |
| for (const auto& pair : fake_video_renderers_) { |
| min_frames = std::min(min_frames, pair.second->num_rendered_frames()); |
| } |
| return min_frames; |
| } |
| |
| // Returns a MockStatsObserver in a state after stats gathering finished, |
| // which can be used to access the gathered stats. |
| rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack( |
| webrtc::MediaStreamTrackInterface* track) { |
| rtc::scoped_refptr<MockStatsObserver> observer( |
| new rtc::RefCountedObject<MockStatsObserver>()); |
| EXPECT_TRUE(peer_connection_->GetStats( |
| observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard)); |
| EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout); |
| return observer; |
| } |
| |
| // Version that doesn't take a track "filter", and gathers all stats. |
| rtc::scoped_refptr<MockStatsObserver> OldGetStats() { |
| return OldGetStatsForTrack(nullptr); |
| } |
| |
| // Synchronously gets stats and returns them. If it times out, fails the test |
| // and returns null. |
| rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() { |
| rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback( |
| new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>()); |
| peer_connection_->GetStats(callback); |
| EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout); |
| return callback->report(); |
| } |
| |
| int rendered_width() { |
| EXPECT_FALSE(fake_video_renderers_.empty()); |
| return fake_video_renderers_.empty() |
| ? 0 |
| : fake_video_renderers_.begin()->second->width(); |
| } |
| |
| int rendered_height() { |
| EXPECT_FALSE(fake_video_renderers_.empty()); |
| return fake_video_renderers_.empty() |
| ? 0 |
| : fake_video_renderers_.begin()->second->height(); |
| } |
| |
| double rendered_aspect_ratio() { |
| if (rendered_height() == 0) { |
| return 0.0; |
| } |
| return static_cast<double>(rendered_width()) / rendered_height(); |
| } |
| |
| webrtc::VideoRotation rendered_rotation() { |
| EXPECT_FALSE(fake_video_renderers_.empty()); |
| return fake_video_renderers_.empty() |
| ? webrtc::kVideoRotation_0 |
| : fake_video_renderers_.begin()->second->rotation(); |
| } |
| |
| int local_rendered_width() { |
| return local_video_renderer_ ? local_video_renderer_->width() : 0; |
| } |
| |
| int local_rendered_height() { |
| return local_video_renderer_ ? local_video_renderer_->height() : 0; |
| } |
| |
| double local_rendered_aspect_ratio() { |
| if (local_rendered_height() == 0) { |
| return 0.0; |
| } |
| return static_cast<double>(local_rendered_width()) / |
| local_rendered_height(); |
| } |
| |
| size_t number_of_remote_streams() { |
| if (!pc()) { |
| return 0; |
| } |
| return pc()->remote_streams()->count(); |
| } |
| |
| StreamCollectionInterface* remote_streams() const { |
| if (!pc()) { |
| ADD_FAILURE(); |
| return nullptr; |
| } |
| return pc()->remote_streams(); |
| } |
| |
| StreamCollectionInterface* local_streams() { |
| if (!pc()) { |
| ADD_FAILURE(); |
| return nullptr; |
| } |
| return pc()->local_streams(); |
| } |
| |
| webrtc::PeerConnectionInterface::SignalingState signaling_state() { |
| return pc()->signaling_state(); |
| } |
| |
| webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() { |
| return pc()->ice_connection_state(); |
| } |
| |
| webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() { |
| return pc()->ice_gathering_state(); |
| } |
| |
| // Returns a MockRtpReceiverObserver for each RtpReceiver returned by |
| // GetReceivers. They're updated automatically when a remote offer/answer |
| // from the fake signaling channel is applied, or when |
| // ResetRtpReceiverObservers below is called. |
| const std::vector<std::unique_ptr<MockRtpReceiverObserver>>& |
| rtp_receiver_observers() { |
| return rtp_receiver_observers_; |
| } |
| |
| void ResetRtpReceiverObservers() { |
| rtp_receiver_observers_.clear(); |
| for (const rtc::scoped_refptr<RtpReceiverInterface>& receiver : |
| pc()->GetReceivers()) { |
| std::unique_ptr<MockRtpReceiverObserver> observer( |
| new MockRtpReceiverObserver(receiver->media_type())); |
| receiver->SetObserver(observer.get()); |
| rtp_receiver_observers_.push_back(std::move(observer)); |
| } |
| } |
| |
| rtc::FakeNetworkManager* network() const { |
| return fake_network_manager_.get(); |
| } |
| cricket::PortAllocator* port_allocator() const { return port_allocator_; } |
| |
| webrtc::FakeRtcEventLogFactory* event_log_factory() const { |
| return event_log_factory_; |
| } |
| |
| private: |
| explicit PeerConnectionWrapper(const std::string& debug_name) |
| : debug_name_(debug_name) {} |
| |
| bool Init( |
| const PeerConnectionFactory::Options* options, |
| const PeerConnectionInterface::RTCConfiguration* config, |
| webrtc::PeerConnectionDependencies dependencies, |
| rtc::Thread* network_thread, |
| rtc::Thread* worker_thread, |
| std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory, |
| std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) { |
| // There's an error in this test code if Init ends up being called twice. |
| RTC_DCHECK(!peer_connection_); |
| RTC_DCHECK(!peer_connection_factory_); |
| |
| fake_network_manager_.reset(new rtc::FakeNetworkManager()); |
| fake_network_manager_->AddInterface(kDefaultLocalAddress); |
| |
| std::unique_ptr<cricket::PortAllocator> port_allocator( |
| new cricket::BasicPortAllocator(fake_network_manager_.get())); |
| port_allocator_ = port_allocator.get(); |
| fake_audio_capture_module_ = FakeAudioCaptureModule::Create(); |
| if (!fake_audio_capture_module_) { |
| return false; |
| } |
| rtc::Thread* const signaling_thread = rtc::Thread::Current(); |
| |
| webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies; |
| pc_factory_dependencies.network_thread = network_thread; |
| pc_factory_dependencies.worker_thread = worker_thread; |
| pc_factory_dependencies.signaling_thread = signaling_thread; |
| pc_factory_dependencies.media_engine = |
| cricket::WebRtcMediaEngineFactory::Create( |
| rtc::scoped_refptr<webrtc::AudioDeviceModule>( |
| fake_audio_capture_module_), |
| webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), |
| webrtc::CreateBuiltinVideoEncoderFactory(), |
| webrtc::CreateBuiltinVideoDecoderFactory(), nullptr, |
| webrtc::AudioProcessingBuilder().Create()); |
| pc_factory_dependencies.call_factory = webrtc::CreateCallFactory(); |
| if (event_log_factory) { |
| event_log_factory_ = event_log_factory.get(); |
| pc_factory_dependencies.event_log_factory = std::move(event_log_factory); |
| } else { |
| pc_factory_dependencies.event_log_factory = |
| webrtc::CreateRtcEventLogFactory(); |
| } |
| if (media_transport_factory) { |
| pc_factory_dependencies.media_transport_factory = |
| std::move(media_transport_factory); |
| } |
| peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory( |
| std::move(pc_factory_dependencies)); |
| |
| if (!peer_connection_factory_) { |
| return false; |
| } |
| if (options) { |
| peer_connection_factory_->SetOptions(*options); |
| } |
| if (config) { |
| sdp_semantics_ = config->sdp_semantics; |
| } |
| |
| dependencies.allocator = std::move(port_allocator); |
| peer_connection_ = CreatePeerConnection(config, std::move(dependencies)); |
| return peer_connection_.get() != nullptr; |
| } |
| |
| rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection( |
| const PeerConnectionInterface::RTCConfiguration* config, |
| webrtc::PeerConnectionDependencies dependencies) { |
| PeerConnectionInterface::RTCConfiguration modified_config; |
| // If |config| is null, this will result in a default configuration being |
| // used. |
| if (config) { |
| modified_config = *config; |
| } |
| // Disable resolution adaptation; we don't want it interfering with the |
| // test results. |
| // TODO(deadbeef): Do something more robust. Since we're testing for aspect |
| // ratios and not specific resolutions, is this even necessary? |
| modified_config.set_cpu_adaptation(false); |
| |
| dependencies.observer = this; |
| return peer_connection_factory_->CreatePeerConnection( |
| modified_config, std::move(dependencies)); |
| } |
| |
| void set_signaling_message_receiver( |
| SignalingMessageReceiver* signaling_message_receiver) { |
| signaling_message_receiver_ = signaling_message_receiver; |
| } |
| |
| void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; } |
| |
| void set_signal_ice_candidates(bool signal) { |
| signal_ice_candidates_ = signal; |
| } |
| |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal( |
| webrtc::FakePeriodicVideoSource::Config config) { |
| // Set max frame rate to 10fps to reduce the risk of test flakiness. |
| // TODO(deadbeef): Do something more robust. |
| config.frame_interval_ms = 100; |
| |
| video_track_sources_.emplace_back( |
| new rtc::RefCountedObject<webrtc::FakePeriodicVideoTrackSource>( |
| config, false /* remote */)); |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> track( |
| peer_connection_factory_->CreateVideoTrack( |
| rtc::CreateRandomUuid(), video_track_sources_.back())); |
| if (!local_video_renderer_) { |
| local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track)); |
| } |
| return track; |
| } |
| |
| void HandleIncomingOffer(const std::string& msg) { |
| RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer"; |
| std::unique_ptr<SessionDescriptionInterface> desc = |
| webrtc::CreateSessionDescription(SdpType::kOffer, msg); |
| if (received_sdp_munger_) { |
| received_sdp_munger_(desc->description()); |
| } |
| |
| EXPECT_TRUE(SetRemoteDescription(std::move(desc))); |
| // Setting a remote description may have changed the number of receivers, |
| // so reset the receiver observers. |
| ResetRtpReceiverObservers(); |
| if (remote_offer_handler_) { |
| remote_offer_handler_(); |
| } |
| auto answer = CreateAnswer(); |
| ASSERT_NE(nullptr, answer); |
| EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(answer))); |
| } |
| |
| void HandleIncomingAnswer(const std::string& msg) { |
| RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer"; |
| std::unique_ptr<SessionDescriptionInterface> desc = |
| webrtc::CreateSessionDescription(SdpType::kAnswer, msg); |
| if (received_sdp_munger_) { |
| received_sdp_munger_(desc->description()); |
| } |
| |
| EXPECT_TRUE(SetRemoteDescription(std::move(desc))); |
| // Set the RtpReceiverObserver after receivers are created. |
| ResetRtpReceiverObservers(); |
| } |
| |
| // Returns null on failure. |
| std::unique_ptr<SessionDescriptionInterface> CreateOffer() { |
| rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer( |
| new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>()); |
| pc()->CreateOffer(observer, offer_answer_options_); |
| return WaitForDescriptionFromObserver(observer); |
| } |
| |
| // Returns null on failure. |
| std::unique_ptr<SessionDescriptionInterface> CreateAnswer() { |
| rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer( |
| new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>()); |
| pc()->CreateAnswer(observer, offer_answer_options_); |
| return WaitForDescriptionFromObserver(observer); |
| } |
| |
| std::unique_ptr<SessionDescriptionInterface> WaitForDescriptionFromObserver( |
| MockCreateSessionDescriptionObserver* observer) { |
| EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout); |
| if (!observer->result()) { |
| return nullptr; |
| } |
| auto description = observer->MoveDescription(); |
| if (generated_sdp_munger_) { |
| generated_sdp_munger_(description->description()); |
| } |
| return description; |
| } |
| |
| // Setting the local description and sending the SDP message over the fake |
| // signaling channel are combined into the same method because the SDP |
| // message needs to be sent as soon as SetLocalDescription finishes, without |
| // waiting for the observer to be called. This ensures that ICE candidates |
| // don't outrace the description. |
| bool SetLocalDescriptionAndSendSdpMessage( |
| std::unique_ptr<SessionDescriptionInterface> desc) { |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer( |
| new rtc::RefCountedObject<MockSetSessionDescriptionObserver>()); |
| RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage"; |
| SdpType type = desc->GetType(); |
| std::string sdp; |
| EXPECT_TRUE(desc->ToString(&sdp)); |
| pc()->SetLocalDescription(observer, desc.release()); |
| if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) { |
| RemoveUnusedVideoRenderers(); |
| } |
| // As mentioned above, we need to send the message immediately after |
| // SetLocalDescription. |
| SendSdpMessage(type, sdp); |
| EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout); |
| return true; |
| } |
| |
| bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc) { |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer( |
| new rtc::RefCountedObject<MockSetSessionDescriptionObserver>()); |
| RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription"; |
| pc()->SetRemoteDescription(observer, desc.release()); |
| if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) { |
| RemoveUnusedVideoRenderers(); |
| } |
| EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout); |
| return observer->result(); |
| } |
| |
| // This is a work around to remove unused fake_video_renderers from |
| // transceivers that have either stopped or are no longer receiving. |
| void RemoveUnusedVideoRenderers() { |
| auto transceivers = pc()->GetTransceivers(); |
| for (auto& transceiver : transceivers) { |
| if (transceiver->receiver()->media_type() != cricket::MEDIA_TYPE_VIDEO) { |
| continue; |
| } |
| // Remove fake video renderers from any stopped transceivers. |
| if (transceiver->stopped()) { |
| auto it = |
| fake_video_renderers_.find(transceiver->receiver()->track()->id()); |
| if (it != fake_video_renderers_.end()) { |
| fake_video_renderers_.erase(it); |
| } |
| } |
| // Remove fake video renderers from any transceivers that are no longer |
| // receiving. |
| if ((transceiver->current_direction() && |
| !webrtc::RtpTransceiverDirectionHasRecv( |
| *transceiver->current_direction()))) { |
| auto it = |
| fake_video_renderers_.find(transceiver->receiver()->track()->id()); |
| if (it != fake_video_renderers_.end()) { |
| fake_video_renderers_.erase(it); |
| } |
| } |
| } |
| } |
| |
| // Simulate sending a blob of SDP with delay |signaling_delay_ms_| (0 by |
| // default). |
| void SendSdpMessage(SdpType type, const std::string& msg) { |
| if (signaling_delay_ms_ == 0) { |
| RelaySdpMessageIfReceiverExists(type, msg); |
| } else { |
| invoker_.AsyncInvokeDelayed<void>( |
| RTC_FROM_HERE, rtc::Thread::Current(), |
| rtc::Bind(&PeerConnectionWrapper::RelaySdpMessageIfReceiverExists, |
| this, type, msg), |
| signaling_delay_ms_); |
| } |
| } |
| |
| void RelaySdpMessageIfReceiverExists(SdpType type, const std::string& msg) { |
| if (signaling_message_receiver_) { |
| signaling_message_receiver_->ReceiveSdpMessage(type, msg); |
| } |
| } |
| |
| // Simulate trickling an ICE candidate with delay |signaling_delay_ms_| (0 by |
| // default). |
| void SendIceMessage(const std::string& sdp_mid, |
| int sdp_mline_index, |
| const std::string& msg) { |
| if (signaling_delay_ms_ == 0) { |
| RelayIceMessageIfReceiverExists(sdp_mid, sdp_mline_index, msg); |
| } else { |
| invoker_.AsyncInvokeDelayed<void>( |
| RTC_FROM_HERE, rtc::Thread::Current(), |
| rtc::Bind(&PeerConnectionWrapper::RelayIceMessageIfReceiverExists, |
| this, sdp_mid, sdp_mline_index, msg), |
| signaling_delay_ms_); |
| } |
| } |
| |
| void RelayIceMessageIfReceiverExists(const std::string& sdp_mid, |
| int sdp_mline_index, |
| const std::string& msg) { |
| if (signaling_message_receiver_) { |
| signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index, |
| msg); |
| } |
| } |
| |
| // SignalingMessageReceiver callbacks. |
| void ReceiveSdpMessage(SdpType type, const std::string& msg) override { |
| if (type == SdpType::kOffer) { |
| HandleIncomingOffer(msg); |
| } else { |
| HandleIncomingAnswer(msg); |
| } |
| } |
| |
| void ReceiveIceMessage(const std::string& sdp_mid, |
| int sdp_mline_index, |
| const std::string& msg) override { |
| RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage"; |
| std::unique_ptr<webrtc::IceCandidateInterface> candidate( |
| webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr)); |
| EXPECT_TRUE(pc()->AddIceCandidate(candidate.get())); |
| } |
| |
| // PeerConnectionObserver callbacks. |
| void OnSignalingChange( |
| webrtc::PeerConnectionInterface::SignalingState new_state) override { |
| EXPECT_EQ(pc()->signaling_state(), new_state); |
| } |
| void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver, |
| const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& |
| streams) override { |
| if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) { |
| rtc::scoped_refptr<VideoTrackInterface> video_track( |
| static_cast<VideoTrackInterface*>(receiver->track().get())); |
| ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) == |
| fake_video_renderers_.end()); |
| fake_video_renderers_[video_track->id()] = |
| absl::make_unique<FakeVideoTrackRenderer>(video_track); |
| } |
| } |
| void OnRemoveTrack( |
| rtc::scoped_refptr<RtpReceiverInterface> receiver) override { |
| if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) { |
| auto it = fake_video_renderers_.find(receiver->track()->id()); |
| RTC_DCHECK(it != fake_video_renderers_.end()); |
| fake_video_renderers_.erase(it); |
| } |
| } |
| void OnRenegotiationNeeded() override {} |
| void OnIceConnectionChange( |
| webrtc::PeerConnectionInterface::IceConnectionState new_state) override { |
| EXPECT_EQ(pc()->ice_connection_state(), new_state); |
| ice_connection_state_history_.push_back(new_state); |
| } |
| void OnConnectionChange( |
| webrtc::PeerConnectionInterface::PeerConnectionState new_state) override { |
| peer_connection_state_history_.push_back(new_state); |
| } |
| |
| void OnIceGatheringChange( |
| webrtc::PeerConnectionInterface::IceGatheringState new_state) override { |
| EXPECT_EQ(pc()->ice_gathering_state(), new_state); |
| ice_gathering_state_history_.push_back(new_state); |
| } |
| std::unique_ptr<webrtc::IceCandidateInterface> ReplaceIceCandidate( |
| const webrtc::IceCandidateInterface* candidate) { |
| std::string candidate_string; |
| candidate->ToString(&candidate_string); |
| |
| auto owned_candidate = |
| local_ice_candidate_replacer_->ReplaceCandidate(candidate); |
| if (!owned_candidate) { |
| RTC_LOG(LS_INFO) << "LocalIceCandidateReplacer dropped \"" |
| << candidate_string << "\""; |
| return nullptr; |
| } |
| std::string owned_candidate_string; |
| owned_candidate->ToString(&owned_candidate_string); |
| RTC_LOG(LS_INFO) << "LocalIceCandidateReplacer changed \"" |
| << candidate_string << "\" to \"" << owned_candidate_string |
| << "\""; |
| return owned_candidate; |
| } |
| void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override { |
| RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate"; |
| |
| const webrtc::IceCandidateInterface* new_candidate = candidate; |
| std::unique_ptr<webrtc::IceCandidateInterface> owned_candidate; |
| if (local_ice_candidate_replacer_) { |
| owned_candidate = ReplaceIceCandidate(candidate); |
| if (!owned_candidate) { |
| return; // The candidate was dropped. |
| } |
| new_candidate = owned_candidate.get(); |
| } |
| |
| std::string ice_sdp; |
| EXPECT_TRUE(new_candidate->ToString(&ice_sdp)); |
| if (signaling_message_receiver_ == nullptr || !signal_ice_candidates_) { |
| // Remote party may be deleted. |
| return; |
| } |
| SendIceMessage(new_candidate->sdp_mid(), new_candidate->sdp_mline_index(), |
| ice_sdp); |
| } |
| void OnDataChannel( |
| rtc::scoped_refptr<DataChannelInterface> data_channel) override { |
| RTC_LOG(LS_INFO) << debug_name_ << ": OnDataChannel"; |
| data_channel_ = data_channel; |
| data_observer_.reset(new MockDataChannelObserver(data_channel)); |
| } |
| |
| std::string debug_name_; |
| |
| std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_; |
| |
| rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
| rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
| peer_connection_factory_; |
| |
| cricket::PortAllocator* port_allocator_; |
| // Needed to keep track of number of frames sent. |
| rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
| // Needed to keep track of number of frames received. |
| std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>> |
| fake_video_renderers_; |
| // Needed to ensure frames aren't received for removed tracks. |
| std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>> |
| removed_fake_video_renderers_; |
| |
| // For remote peer communication. |
| SignalingMessageReceiver* signaling_message_receiver_ = nullptr; |
| int signaling_delay_ms_ = 0; |
| bool signal_ice_candidates_ = true; |
| |
| // Store references to the video sources we've created, so that we can stop |
| // them, if required. |
| std::vector<rtc::scoped_refptr<webrtc::VideoTrackSource>> |
| video_track_sources_; |
| // |local_video_renderer_| attached to the first created local video track. |
| std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_; |
| |
| SdpSemantics sdp_semantics_; |
| PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_; |
| std::function<void(cricket::SessionDescription*)> received_sdp_munger_; |
| std::function<void(cricket::SessionDescription*)> generated_sdp_munger_; |
| std::function<void()> remote_offer_handler_; |
| std::unique_ptr<IceCandidateReplacerInterface> local_ice_candidate_replacer_; |
| rtc::scoped_refptr<DataChannelInterface> data_channel_; |
| std::unique_ptr<MockDataChannelObserver> data_observer_; |
| |
| std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_; |
| |
| std::vector<PeerConnectionInterface::IceConnectionState> |
| ice_connection_state_history_; |
| std::vector<PeerConnectionInterface::PeerConnectionState> |
| peer_connection_state_history_; |
| std::vector<PeerConnectionInterface::IceGatheringState> |
| ice_gathering_state_history_; |
| |
| webrtc::FakeRtcEventLogFactory* event_log_factory_; |
| |
| rtc::AsyncInvoker invoker_; |
| |
| friend class PeerConnectionIntegrationBaseTest; |
| }; |
| |
| class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput { |
| public: |
| virtual ~MockRtcEventLogOutput() = default; |
| MOCK_CONST_METHOD0(IsActive, bool()); |
| MOCK_METHOD1(Write, bool(const std::string&)); |
| }; |
| |
| // This helper object is used for both specifying how many audio/video frames |
| // are expected to be received for a caller/callee. It provides helper functions |
| // to specify these expectations. The object initially starts in a state of no |
| // expectations. |
| class MediaExpectations { |
| public: |
| enum ExpectFrames { |
| kExpectSomeFrames, |
| kExpectNoFrames, |
| kNoExpectation, |
| }; |
| |
| void ExpectBidirectionalAudioAndVideo() { |
| ExpectBidirectionalAudio(); |
| ExpectBidirectionalVideo(); |
| } |
| |
| void ExpectBidirectionalAudio() { |
| CallerExpectsSomeAudio(); |
| CalleeExpectsSomeAudio(); |
| } |
| |
| void ExpectNoAudio() { |
| CallerExpectsNoAudio(); |
| CalleeExpectsNoAudio(); |
| } |
| |
| void ExpectBidirectionalVideo() { |
| CallerExpectsSomeVideo(); |
| CalleeExpectsSomeVideo(); |
| } |
| |
| void ExpectNoVideo() { |
| CallerExpectsNoVideo(); |
| CalleeExpectsNoVideo(); |
| } |
| |
| void CallerExpectsSomeAudioAndVideo() { |
| CallerExpectsSomeAudio(); |
| CallerExpectsSomeVideo(); |
| } |
| |
| void CalleeExpectsSomeAudioAndVideo() { |
| CalleeExpectsSomeAudio(); |
| CalleeExpectsSomeVideo(); |
| } |
| |
| // Caller's audio functions. |
| void CallerExpectsSomeAudio( |
| int expected_audio_frames = kDefaultExpectedAudioFrameCount) { |
| caller_audio_expectation_ = kExpectSomeFrames; |
| caller_audio_frames_expected_ = expected_audio_frames; |
| } |
| |
| void CallerExpectsNoAudio() { |
| caller_audio_expectation_ = kExpectNoFrames; |
| caller_audio_frames_expected_ = 0; |
| } |
| |
| // Caller's video functions. |
| void CallerExpectsSomeVideo( |
| int expected_video_frames = kDefaultExpectedVideoFrameCount) { |
| caller_video_expectation_ = kExpectSomeFrames; |
| caller_video_frames_expected_ = expected_video_frames; |
| } |
| |
| void CallerExpectsNoVideo() { |
| caller_video_expectation_ = kExpectNoFrames; |
| caller_video_frames_expected_ = 0; |
| } |
| |
| // Callee's audio functions. |
| void CalleeExpectsSomeAudio( |
| int expected_audio_frames = kDefaultExpectedAudioFrameCount) { |
| callee_audio_expectation_ = kExpectSomeFrames; |
| callee_audio_frames_expected_ = expected_audio_frames; |
| } |
| |
| void CalleeExpectsNoAudio() { |
| callee_audio_expectation_ = kExpectNoFrames; |
| callee_audio_frames_expected_ = 0; |
| } |
| |
| // Callee's video functions. |
| void CalleeExpectsSomeVideo( |
| int expected_video_frames = kDefaultExpectedVideoFrameCount) { |
| callee_video_expectation_ = kExpectSomeFrames; |
| callee_video_frames_expected_ = expected_video_frames; |
| } |
| |
| void CalleeExpectsNoVideo() { |
| callee_video_expectation_ = kExpectNoFrames; |
| callee_video_frames_expected_ = 0; |
| } |
| |
| ExpectFrames caller_audio_expectation_ = kNoExpectation; |
| ExpectFrames caller_video_expectation_ = kNoExpectation; |
| ExpectFrames callee_audio_expectation_ = kNoExpectation; |
| ExpectFrames callee_video_expectation_ = kNoExpectation; |
| int caller_audio_frames_expected_ = 0; |
| int caller_video_frames_expected_ = 0; |
| int callee_audio_frames_expected_ = 0; |
| int callee_video_frames_expected_ = 0; |
| }; |
| |
| // Tests two PeerConnections connecting to each other end-to-end, using a |
| // virtual network, fake A/V capture and fake encoder/decoders. The |
| // PeerConnections share the threads/socket servers, but use separate versions |
| // of everything else (including "PeerConnectionFactory"s). |
| class PeerConnectionIntegrationBaseTest : public testing::Test { |
| public: |
| explicit PeerConnectionIntegrationBaseTest(SdpSemantics sdp_semantics) |
| : sdp_semantics_(sdp_semantics), |
| ss_(new rtc::VirtualSocketServer()), |
| fss_(new rtc::FirewallSocketServer(ss_.get())), |
| network_thread_(new rtc::Thread(fss_.get())), |
| worker_thread_(rtc::Thread::Create()), |
| loopback_media_transports_(network_thread_.get()) { |
| network_thread_->SetName("PCNetworkThread", this); |
| worker_thread_->SetName("PCWorkerThread", this); |
| RTC_CHECK(network_thread_->Start()); |
| RTC_CHECK(worker_thread_->Start()); |
| webrtc::metrics::Reset(); |
| } |
| |
| ~PeerConnectionIntegrationBaseTest() { |
| // The PeerConnections should deleted before the TurnCustomizers. |
| // A TurnPort is created with a raw pointer to a TurnCustomizer. The |
| // TurnPort has the same lifetime as the PeerConnection, so it's expected |
| // that the TurnCustomizer outlives the life of the PeerConnection or else |
| // when Send() is called it will hit a seg fault. |
| if (caller_) { |
| caller_->set_signaling_message_receiver(nullptr); |
| delete SetCallerPcWrapperAndReturnCurrent(nullptr); |
| } |
| if (callee_) { |
| callee_->set_signaling_message_receiver(nullptr); |
| delete SetCalleePcWrapperAndReturnCurrent(nullptr); |
| } |
| |
| // If turn servers were created for the test they need to be destroyed on |
| // the network thread. |
| network_thread()->Invoke<void>(RTC_FROM_HERE, [this] { |
| turn_servers_.clear(); |
| turn_customizers_.clear(); |
| }); |
| } |
| |
| bool SignalingStateStable() { |
| return caller_->SignalingStateStable() && callee_->SignalingStateStable(); |
| } |
| |
| bool DtlsConnected() { |
| // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS |
| // are connected. This is an important distinction. Once we have separate |
| // ICE and DTLS state, this check needs to use the DTLS state. |
| return (callee()->ice_connection_state() == |
| webrtc::PeerConnectionInterface::kIceConnectionConnected || |
| callee()->ice_connection_state() == |
| webrtc::PeerConnectionInterface::kIceConnectionCompleted) && |
| (caller()->ice_connection_state() == |
| webrtc::PeerConnectionInterface::kIceConnectionConnected || |
| caller()->ice_connection_state() == |
| webrtc::PeerConnectionInterface::kIceConnectionCompleted); |
| } |
| |
| // When |event_log_factory| is null, the default implementation of the event |
| // log factory will be used. |
| std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWrapper( |
| const std::string& debug_name, |
| const PeerConnectionFactory::Options* options, |
| const RTCConfiguration* config, |
| webrtc::PeerConnectionDependencies dependencies, |
| std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory, |
| std::unique_ptr<webrtc::MediaTransportFactory> media_transport_factory) { |
| RTCConfiguration modified_config; |
| if (config) { |
| modified_config = *config; |
| } |
| modified_config.sdp_semantics = sdp_semantics_; |
| if (!dependencies.cert_generator) { |
| dependencies.cert_generator = |
| absl::make_unique<FakeRTCCertificateGenerator>(); |
| } |
| std::unique_ptr<PeerConnectionWrapper> client( |
| new PeerConnectionWrapper(debug_name)); |
| |
| if (!client->Init(options, &modified_config, std::move(dependencies), |
| network_thread_.get(), worker_thread_.get(), |
| std::move(event_log_factory), |
| std::move(media_transport_factory))) { |
| return nullptr; |
| } |
| return client; |
| } |
| |
| std::unique_ptr<PeerConnectionWrapper> |
| CreatePeerConnectionWrapperWithFakeRtcEventLog( |
| const std::string& debug_name, |
| const PeerConnectionFactory::Options* options, |
| const RTCConfiguration* config, |
| webrtc::PeerConnectionDependencies dependencies) { |
| std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory( |
| new webrtc::FakeRtcEventLogFactory(rtc::Thread::Current())); |
| return CreatePeerConnectionWrapper(debug_name, options, config, |
| std::move(dependencies), |
| std::move(event_log_factory), |
| /*media_transport_factory=*/nullptr); |
| } |
| |
| bool CreatePeerConnectionWrappers() { |
| return CreatePeerConnectionWrappersWithConfig( |
| PeerConnectionInterface::RTCConfiguration(), |
| PeerConnectionInterface::RTCConfiguration()); |
| } |
| |
| bool CreatePeerConnectionWrappersWithSdpSemantics( |
| SdpSemantics caller_semantics, |
| SdpSemantics callee_semantics) { |
| // Can't specify the sdp_semantics in the passed-in configuration since it |
| // will be overwritten by CreatePeerConnectionWrapper with whatever is |
| // stored in sdp_semantics_. So get around this by modifying the instance |
| // variable before calling CreatePeerConnectionWrapper for the caller and |
| // callee PeerConnections. |
| SdpSemantics original_semantics = sdp_semantics_; |
| sdp_semantics_ = caller_semantics; |
| caller_ = CreatePeerConnectionWrapper( |
| "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), |
| nullptr, /*media_transport_factory=*/nullptr); |
| sdp_semantics_ = callee_semantics; |
| callee_ = CreatePeerConnectionWrapper( |
| "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), |
| nullptr, /*media_transport_factory=*/nullptr); |
| sdp_semantics_ = original_semantics; |
| return caller_ && callee_; |
| } |
| |
| bool CreatePeerConnectionWrappersWithConfig( |
| const PeerConnectionInterface::RTCConfiguration& caller_config, |
| const PeerConnectionInterface::RTCConfiguration& callee_config) { |
| caller_ = CreatePeerConnectionWrapper( |
| "Caller", nullptr, &caller_config, |
| webrtc::PeerConnectionDependencies(nullptr), nullptr, |
| /*media_transport_factory=*/nullptr); |
| callee_ = CreatePeerConnectionWrapper( |
| "Callee", nullptr, &callee_config, |
| webrtc::PeerConnectionDependencies(nullptr), nullptr, |
| /*media_transport_factory=*/nullptr); |
| return caller_ && callee_; |
| } |
| |
| bool CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory( |
| const PeerConnectionInterface::RTCConfiguration& caller_config, |
| const PeerConnectionInterface::RTCConfiguration& callee_config, |
| std::unique_ptr<webrtc::MediaTransportFactory> caller_factory, |
| std::unique_ptr<webrtc::MediaTransportFactory> callee_factory) { |
| caller_ = |
| CreatePeerConnectionWrapper("Caller", nullptr, &caller_config, |
| webrtc::PeerConnectionDependencies(nullptr), |
| nullptr, std::move(caller_factory)); |
| callee_ = |
| CreatePeerConnectionWrapper("Callee", nullptr, &callee_config, |
| webrtc::PeerConnectionDependencies(nullptr), |
| nullptr, std::move(callee_factory)); |
| return caller_ && callee_; |
| } |
| |
| bool CreatePeerConnectionWrappersWithConfigAndDeps( |
| const PeerConnectionInterface::RTCConfiguration& caller_config, |
| webrtc::PeerConnectionDependencies caller_dependencies, |
| const PeerConnectionInterface::RTCConfiguration& callee_config, |
| webrtc::PeerConnectionDependencies callee_dependencies) { |
| caller_ = |
| CreatePeerConnectionWrapper("Caller", nullptr, &caller_config, |
| std::move(caller_dependencies), nullptr, |
| /*media_transport_factory=*/nullptr); |
| callee_ = |
| CreatePeerConnectionWrapper("Callee", nullptr, &callee_config, |
| std::move(callee_dependencies), nullptr, |
| /*media_transport_factory=*/nullptr); |
| return caller_ && callee_; |
| } |
| |
| bool CreatePeerConnectionWrappersWithOptions( |
| const PeerConnectionFactory::Options& caller_options, |
| const PeerConnectionFactory::Options& callee_options) { |
| caller_ = CreatePeerConnectionWrapper( |
| "Caller", &caller_options, nullptr, |
| webrtc::PeerConnectionDependencies(nullptr), nullptr, |
| /*media_transport_factory=*/nullptr); |
| callee_ = CreatePeerConnectionWrapper( |
| "Callee", &callee_options, nullptr, |
| webrtc::PeerConnectionDependencies(nullptr), nullptr, |
| /*media_transport_factory=*/nullptr); |
| return caller_ && callee_; |
| } |
| |
| bool CreatePeerConnectionWrappersWithFakeRtcEventLog() { |
| PeerConnectionInterface::RTCConfiguration default_config; |
| caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog( |
| "Caller", nullptr, &default_config, |
| webrtc::PeerConnectionDependencies(nullptr)); |
| callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog( |
| "Callee", nullptr, &default_config, |
| webrtc::PeerConnectionDependencies(nullptr)); |
| return caller_ && callee_; |
| } |
| |
| std::unique_ptr<PeerConnectionWrapper> |
| CreatePeerConnectionWrapperWithAlternateKey() { |
| std::unique_ptr<FakeRTCCertificateGenerator> cert_generator( |
| new FakeRTCCertificateGenerator()); |
| cert_generator->use_alternate_key(); |
| |
| webrtc::PeerConnectionDependencies dependencies(nullptr); |
| dependencies.cert_generator = std::move(cert_generator); |
| return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr, |
| std::move(dependencies), nullptr, |
| /*media_transport_factory=*/nullptr); |
| } |
| |
| cricket::TestTurnServer* CreateTurnServer( |
| rtc::SocketAddress internal_address, |
| rtc::SocketAddress external_address, |
| cricket::ProtocolType type = cricket::ProtocolType::PROTO_UDP, |
| const std::string& common_name = "test turn server") { |
| rtc::Thread* thread = network_thread(); |
| std::unique_ptr<cricket::TestTurnServer> turn_server = |
| network_thread()->Invoke<std::unique_ptr<cricket::TestTurnServer>>( |
| RTC_FROM_HERE, |
| [thread, internal_address, external_address, type, common_name] { |
| return absl::make_unique<cricket::TestTurnServer>( |
| thread, internal_address, external_address, type, |
| /*ignore_bad_certs=*/true, common_name); |
| }); |
| turn_servers_.push_back(std::move(turn_server)); |
| // Interactions with the turn server should be done on the network thread. |
| return turn_servers_.back().get(); |
| } |
| |
| cricket::TestTurnCustomizer* CreateTurnCustomizer() { |
| std::unique_ptr<cricket::TestTurnCustomizer> turn_customizer = |
| network_thread()->Invoke<std::unique_ptr<cricket::TestTurnCustomizer>>( |
| RTC_FROM_HERE, |
| [] { return absl::make_unique<cricket::TestTurnCustomizer>(); }); |
| turn_customizers_.push_back(std::move(turn_customizer)); |
| // Interactions with the turn customizer should be done on the network |
| // thread. |
| return turn_customizers_.back().get(); |
| } |
| |
| // Checks that the function counters for a TestTurnCustomizer are greater than |
| // 0. |
| void ExpectTurnCustomizerCountersIncremented( |
| cricket::TestTurnCustomizer* turn_customizer) { |
| unsigned int allow_channel_data_counter = |
| network_thread()->Invoke<unsigned int>( |
| RTC_FROM_HERE, [turn_customizer] { |
| return turn_customizer->allow_channel_data_cnt_; |
| }); |
| EXPECT_GT(allow_channel_data_counter, 0u); |
| unsigned int modify_counter = network_thread()->Invoke<unsigned int>( |
| RTC_FROM_HERE, |
| [turn_customizer] { return turn_customizer->modify_cnt_; }); |
| EXPECT_GT(modify_counter, 0u); |
| } |
| |
| // Once called, SDP blobs and ICE candidates will be automatically signaled |
| // between PeerConnections. |
| void ConnectFakeSignaling() { |
| caller_->set_signaling_message_receiver(callee_.get()); |
| callee_->set_signaling_message_receiver(caller_.get()); |
| } |
| |
| // Once called, SDP blobs will be automatically signaled between |
| // PeerConnections. Note that ICE candidates will not be signaled unless they |
| // are in the exchanged SDP blobs. |
| void ConnectFakeSignalingForSdpOnly() { |
| ConnectFakeSignaling(); |
| SetSignalIceCandidates(false); |
| } |
| |
| void SetSignalingDelayMs(int delay_ms) { |
| caller_->set_signaling_delay_ms(delay_ms); |
| callee_->set_signaling_delay_ms(delay_ms); |
| } |
| |
| void SetSignalIceCandidates(bool signal) { |
| caller_->set_signal_ice_candidates(signal); |
| callee_->set_signal_ice_candidates(signal); |
| } |
| |
| // Messages may get lost on the unreliable DataChannel, so we send multiple |
| // times to avoid test flakiness. |
| void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc, |
| const std::string& data, |
| int retries) { |
| for (int i = 0; i < retries; ++i) { |
| dc->Send(DataBuffer(data)); |
| } |
| } |
| |
| rtc::Thread* network_thread() { return network_thread_.get(); } |
| |
| rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); } |
| |
| webrtc::MediaTransportPair* loopback_media_transports() { |
| return &loopback_media_transports_; |
| } |
| |
| PeerConnectionWrapper* caller() { return caller_.get(); } |
| |
| // Set the |caller_| to the |wrapper| passed in and return the |
| // original |caller_|. |
| PeerConnectionWrapper* SetCallerPcWrapperAndReturnCurrent( |
| PeerConnectionWrapper* wrapper) { |
| PeerConnectionWrapper* old = caller_.release(); |
| caller_.reset(wrapper); |
| return old; |
| } |
| |
| PeerConnectionWrapper* callee() { return callee_.get(); } |
| |
| // Set the |callee_| to the |wrapper| passed in and return the |
| // original |callee_|. |
| PeerConnectionWrapper* SetCalleePcWrapperAndReturnCurrent( |
| PeerConnectionWrapper* wrapper) { |
| PeerConnectionWrapper* old = callee_.release(); |
| callee_.reset(wrapper); |
| return old; |
| } |
| |
| rtc::FirewallSocketServer* firewall() const { return fss_.get(); } |
| |
| // Expects the provided number of new frames to be received within |
| // kMaxWaitForFramesMs. The new expected frames are specified in |
| // |media_expectations|. Returns false if any of the expectations were |
| // not met. |
| bool ExpectNewFrames(const MediaExpectations& media_expectations) { |
| // First initialize the expected frame counts based upon the current |
| // frame count. |
| int total_caller_audio_frames_expected = caller()->audio_frames_received(); |
| if (media_expectations.caller_audio_expectation_ == |
| MediaExpectations::kExpectSomeFrames) { |
| total_caller_audio_frames_expected += |
| media_expectations.caller_audio_frames_expected_; |
| } |
| int total_caller_video_frames_expected = |
| caller()->min_video_frames_received_per_track(); |
| if (media_expectations.caller_video_expectation_ == |
| MediaExpectations::kExpectSomeFrames) { |
| total_caller_video_frames_expected += |
| media_expectations.caller_video_frames_expected_; |
| } |
| int total_callee_audio_frames_expected = callee()->audio_frames_received(); |
| if (media_expectations.callee_audio_expectation_ == |
| MediaExpectations::kExpectSomeFrames) { |
| total_callee_audio_frames_expected += |
| media_expectations.callee_audio_frames_expected_; |
| } |
| int total_callee_video_frames_expected = |
| callee()->min_video_frames_received_per_track(); |
| if (media_expectations.callee_video_expectation_ == |
| MediaExpectations::kExpectSomeFrames) { |
| total_callee_video_frames_expected += |
| media_expectations.callee_video_frames_expected_; |
| } |
| |
| // Wait for the expected frames. |
| EXPECT_TRUE_WAIT(caller()->audio_frames_received() >= |
| total_caller_audio_frames_expected && |
| caller()->min_video_frames_received_per_track() >= |
| total_caller_video_frames_expected && |
| callee()->audio_frames_received() >= |
| total_callee_audio_frames_expected && |
| callee()->min_video_frames_received_per_track() >= |
| total_callee_video_frames_expected, |
| kMaxWaitForFramesMs); |
| bool expectations_correct = |
| caller()->audio_frames_received() >= |
| total_caller_audio_frames_expected && |
| caller()->min_video_frames_received_per_track() >= |
| total_caller_video_frames_expected && |
| callee()->audio_frames_received() >= |
| total_callee_audio_frames_expected && |
| callee()->min_video_frames_received_per_track() >= |
| total_callee_video_frames_expected; |
| |
| // After the combined wait, print out a more detailed message upon |
| // failure. |
| EXPECT_GE(caller()->audio_frames_received(), |
| total_caller_audio_frames_expected); |
| EXPECT_GE(caller()->min_video_frames_received_per_track(), |
| total_caller_video_frames_expected); |
| EXPECT_GE(callee()->audio_frames_received(), |
| total_callee_audio_frames_expected); |
| EXPECT_GE(callee()->min_video_frames_received_per_track(), |
| total_callee_video_frames_expected); |
| |
| // We want to make sure nothing unexpected was received. |
| if (media_expectations.caller_audio_expectation_ == |
| MediaExpectations::kExpectNoFrames) { |
| EXPECT_EQ(caller()->audio_frames_received(), |
| total_caller_audio_frames_expected); |
| if (caller()->audio_frames_received() != |
| total_caller_audio_frames_expected) { |
| expectations_correct = false; |
| } |
| } |
| if (media_expectations.caller_video_expectation_ == |
| MediaExpectations::kExpectNoFrames) { |
| EXPECT_EQ(caller()->min_video_frames_received_per_track(), |
| total_caller_video_frames_expected); |
| if (caller()->min_video_frames_received_per_track() != |
| total_caller_video_frames_expected) { |
| expectations_correct = false; |
| } |
| } |
| if (media_expectations.callee_audio_expectation_ == |
| MediaExpectations::kExpectNoFrames) { |
| EXPECT_EQ(callee()->audio_frames_received(), |
| total_callee_audio_frames_expected); |
| if (callee()->audio_frames_received() != |
| total_callee_audio_frames_expected) { |
| expectations_correct = false; |
| } |
| } |
| if (media_expectations.callee_video_expectation_ == |
| MediaExpectations::kExpectNoFrames) { |
| EXPECT_EQ(callee()->min_video_frames_received_per_track(), |
| total_callee_video_frames_expected); |
| if (callee()->min_video_frames_received_per_track() != |
| total_callee_video_frames_expected) { |
| expectations_correct = false; |
| } |
| } |
| return expectations_correct; |
| } |
| |
| void TestNegotiatedCipherSuite( |
| const PeerConnectionFactory::Options& caller_options, |
| const PeerConnectionFactory::Options& callee_options, |
| int expected_cipher_suite) { |
| ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options, |
| callee_options)); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout); |
| EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite), |
| caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout); |
| // TODO(bugs.webrtc.org/9456): Fix it. |
| EXPECT_EQ(1, webrtc::metrics::NumEvents( |
| "WebRTC.PeerConnection.SrtpCryptoSuite.Audio", |
| expected_cipher_suite)); |
| } |
| |
| void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled, |
| bool remote_gcm_enabled, |
| int expected_cipher_suite) { |
| PeerConnectionFactory::Options caller_options; |
| caller_options.crypto_options.srtp.enable_gcm_crypto_suites = |
| local_gcm_enabled; |
| PeerConnectionFactory::Options callee_options; |
| callee_options.crypto_options.srtp.enable_gcm_crypto_suites = |
| remote_gcm_enabled; |
| TestNegotiatedCipherSuite(caller_options, callee_options, |
| expected_cipher_suite); |
| } |
| |
| protected: |
| SdpSemantics sdp_semantics_; |
| |
| private: |
| // |ss_| is used by |network_thread_| so it must be destroyed later. |
| std::unique_ptr<rtc::VirtualSocketServer> ss_; |
| std::unique_ptr<rtc::FirewallSocketServer> fss_; |
| // |network_thread_| and |worker_thread_| are used by both |
| // |caller_| and |callee_| so they must be destroyed |
| // later. |
| std::unique_ptr<rtc::Thread> network_thread_; |
| std::unique_ptr<rtc::Thread> worker_thread_; |
| // The turn servers and turn customizers should be accessed & deleted on the |
| // network thread to avoid a race with the socket read/write that occurs |
| // on the network thread. |
| std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_; |
| std::vector<std::unique_ptr<cricket::TestTurnCustomizer>> turn_customizers_; |
| webrtc::MediaTransportPair loopback_media_transports_; |
| std::unique_ptr<PeerConnectionWrapper> caller_; |
| std::unique_ptr<PeerConnectionWrapper> callee_; |
| }; |
| |
| class PeerConnectionIntegrationTest |
| : public PeerConnectionIntegrationBaseTest, |
| public ::testing::WithParamInterface<SdpSemantics> { |
| protected: |
| PeerConnectionIntegrationTest() |
| : PeerConnectionIntegrationBaseTest(GetParam()) {} |
| }; |
| |
| class PeerConnectionIntegrationTestPlanB |
| : public PeerConnectionIntegrationBaseTest { |
| protected: |
| PeerConnectionIntegrationTestPlanB() |
| : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB) {} |
| }; |
| |
| class PeerConnectionIntegrationTestUnifiedPlan |
| : public PeerConnectionIntegrationBaseTest { |
| protected: |
| PeerConnectionIntegrationTestUnifiedPlan() |
| : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {} |
| }; |
| |
| // Test the OnFirstPacketReceived callback from audio/video RtpReceivers. This |
| // includes testing that the callback is invoked if an observer is connected |
| // after the first packet has already been received. |
| TEST_P(PeerConnectionIntegrationTest, |
| RtpReceiverObserverOnFirstPacketReceived) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| // Start offer/answer exchange and wait for it to complete. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // Should be one receiver each for audio/video. |
| EXPECT_EQ(2U, caller()->rtp_receiver_observers().size()); |
| EXPECT_EQ(2U, callee()->rtp_receiver_observers().size()); |
| // Wait for all "first packet received" callbacks to be fired. |
| EXPECT_TRUE_WAIT( |
| std::all_of(caller()->rtp_receiver_observers().begin(), |
| caller()->rtp_receiver_observers().end(), |
| [](const std::unique_ptr<MockRtpReceiverObserver>& o) { |
| return o->first_packet_received(); |
| }), |
| kMaxWaitForFramesMs); |
| EXPECT_TRUE_WAIT( |
| std::all_of(callee()->rtp_receiver_observers().begin(), |
| callee()->rtp_receiver_observers().end(), |
| [](const std::unique_ptr<MockRtpReceiverObserver>& o) { |
| return o->first_packet_received(); |
| }), |
| kMaxWaitForFramesMs); |
| // If new observers are set after the first packet was already received, the |
| // callback should still be invoked. |
| caller()->ResetRtpReceiverObservers(); |
| callee()->ResetRtpReceiverObservers(); |
| EXPECT_EQ(2U, caller()->rtp_receiver_observers().size()); |
| EXPECT_EQ(2U, callee()->rtp_receiver_observers().size()); |
| EXPECT_TRUE( |
| std::all_of(caller()->rtp_receiver_observers().begin(), |
| caller()->rtp_receiver_observers().end(), |
| [](const std::unique_ptr<MockRtpReceiverObserver>& o) { |
| return o->first_packet_received(); |
| })); |
| EXPECT_TRUE( |
| std::all_of(callee()->rtp_receiver_observers().begin(), |
| callee()->rtp_receiver_observers().end(), |
| [](const std::unique_ptr<MockRtpReceiverObserver>& o) { |
| return o->first_packet_received(); |
| })); |
| } |
| |
| class DummyDtmfObserver : public DtmfSenderObserverInterface { |
| public: |
| DummyDtmfObserver() : completed_(false) {} |
| |
| // Implements DtmfSenderObserverInterface. |
| void OnToneChange(const std::string& tone) override { |
| tones_.push_back(tone); |
| if (tone.empty()) { |
| completed_ = true; |
| } |
| } |
| |
| const std::vector<std::string>& tones() const { return tones_; } |
| bool completed() const { return completed_; } |
| |
| private: |
| bool completed_; |
| std::vector<std::string> tones_; |
| }; |
| |
| // Assumes |sender| already has an audio track added and the offer/answer |
| // exchange is done. |
| void TestDtmfFromSenderToReceiver(PeerConnectionWrapper* sender, |
| PeerConnectionWrapper* receiver) { |
| // We should be able to get a DTMF sender from the local sender. |
| rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender = |
| sender->pc()->GetSenders().at(0)->GetDtmfSender(); |
| ASSERT_TRUE(dtmf_sender); |
| DummyDtmfObserver observer; |
| dtmf_sender->RegisterObserver(&observer); |
| |
| // Test the DtmfSender object just created. |
| EXPECT_TRUE(dtmf_sender->CanInsertDtmf()); |
| EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50)); |
| |
| EXPECT_TRUE_WAIT(observer.completed(), kDefaultTimeout); |
| std::vector<std::string> tones = {"1", "a", ""}; |
| EXPECT_EQ(tones, observer.tones()); |
| dtmf_sender->UnregisterObserver(); |
| // TODO(deadbeef): Verify the tones were actually received end-to-end. |
| } |
| |
| // Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each |
| // direction). |
| TEST_P(PeerConnectionIntegrationTest, DtmfSenderObserver) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Only need audio for DTMF. |
| caller()->AddAudioTrack(); |
| callee()->AddAudioTrack(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // DTLS must finish before the DTMF sender can be used reliably. |
| ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout); |
| TestDtmfFromSenderToReceiver(caller(), callee()); |
| TestDtmfFromSenderToReceiver(callee(), caller()); |
| } |
| |
| // Basic end-to-end test, verifying media can be encoded/transmitted/decoded |
| // between two connections, using DTLS-SRTP. |
| TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| |
| // Do normal offer/answer and wait for some frames to be received in each |
| // direction. |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol", |
| webrtc::kEnumCounterKeyProtocolDtls)); |
| EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol", |
| webrtc::kEnumCounterKeyProtocolSdes)); |
| } |
| |
| // Uses SDES instead of DTLS for key agreement. |
| TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) { |
| PeerConnectionInterface::RTCConfiguration sdes_config; |
| sdes_config.enable_dtls_srtp.emplace(false); |
| ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config)); |
| ConnectFakeSignaling(); |
| |
| // Do normal offer/answer and wait for some frames to be received in each |
| // direction. |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol", |
| webrtc::kEnumCounterKeyProtocolSdes)); |
| EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol", |
| webrtc::kEnumCounterKeyProtocolDtls)); |
| } |
| |
| // Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS |
| // certificate once the DTLS handshake has finished. |
| TEST_P(PeerConnectionIntegrationTest, |
| GetRemoteAudioSSLCertificateReturnsExchangedCertificate) { |
| auto GetRemoteAudioSSLCertificate = [](PeerConnectionWrapper* wrapper) { |
| auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc()); |
| auto pc = reinterpret_cast<PeerConnection*>(pci->internal()); |
| return pc->GetRemoteAudioSSLCertificate(); |
| }; |
| auto GetRemoteAudioSSLCertChain = [](PeerConnectionWrapper* wrapper) { |
| auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc()); |
| auto pc = reinterpret_cast<PeerConnection*>(pci->internal()); |
| return pc->GetRemoteAudioSSLCertChain(); |
| }; |
| |
| auto caller_cert = rtc::RTCCertificate::FromPEM(kRsaPems[0]); |
| auto callee_cert = rtc::RTCCertificate::FromPEM(kRsaPems[1]); |
| |
| // Configure each side with a known certificate so they can be compared later. |
| PeerConnectionInterface::RTCConfiguration caller_config; |
| caller_config.enable_dtls_srtp.emplace(true); |
| caller_config.certificates.push_back(caller_cert); |
| PeerConnectionInterface::RTCConfiguration callee_config; |
| callee_config.enable_dtls_srtp.emplace(true); |
| callee_config.certificates.push_back(callee_cert); |
| ASSERT_TRUE( |
| CreatePeerConnectionWrappersWithConfig(caller_config, callee_config)); |
| ConnectFakeSignaling(); |
| |
| // When first initialized, there should not be a remote SSL certificate (and |
| // calling this method should not crash). |
| EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(caller())); |
| EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(callee())); |
| EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(caller())); |
| EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(callee())); |
| |
| caller()->AddAudioTrack(); |
| callee()->AddAudioTrack(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout); |
| |
| // Once DTLS has been connected, each side should return the other's SSL |
| // certificate when calling GetRemoteAudioSSLCertificate. |
| |
| auto caller_remote_cert = GetRemoteAudioSSLCertificate(caller()); |
| ASSERT_TRUE(caller_remote_cert); |
| EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(), |
| caller_remote_cert->ToPEMString()); |
| |
| auto callee_remote_cert = GetRemoteAudioSSLCertificate(callee()); |
| ASSERT_TRUE(callee_remote_cert); |
| EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(), |
| callee_remote_cert->ToPEMString()); |
| |
| auto caller_remote_cert_chain = GetRemoteAudioSSLCertChain(caller()); |
| ASSERT_TRUE(caller_remote_cert_chain); |
| ASSERT_EQ(1U, caller_remote_cert_chain->GetSize()); |
| auto remote_cert = &caller_remote_cert_chain->Get(0); |
| EXPECT_EQ(callee_cert->GetSSLCertificate().ToPEMString(), |
| remote_cert->ToPEMString()); |
| |
| auto callee_remote_cert_chain = GetRemoteAudioSSLCertChain(callee()); |
| ASSERT_TRUE(callee_remote_cert_chain); |
| ASSERT_EQ(1U, callee_remote_cert_chain->GetSize()); |
| remote_cert = &callee_remote_cert_chain->Get(0); |
| EXPECT_EQ(caller_cert->GetSSLCertificate().ToPEMString(), |
| remote_cert->ToPEMString()); |
| } |
| |
| // This test sets up a call between two parties with a source resolution of |
| // 1280x720 and verifies that a 16:9 aspect ratio is received. |
| TEST_P(PeerConnectionIntegrationTest, |
| Send1280By720ResolutionAndReceive16To9AspectRatio) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| |
| // Add video tracks with 16:9 aspect ratio, size 1280 x 720. |
| webrtc::FakePeriodicVideoSource::Config config; |
| config.width = 1280; |
| config.height = 720; |
| config.timestamp_offset_ms = rtc::TimeMillis(); |
| caller()->AddTrack(caller()->CreateLocalVideoTrackWithConfig(config)); |
| callee()->AddTrack(callee()->CreateLocalVideoTrackWithConfig(config)); |
| |
| // Do normal offer/answer and wait for at least one frame to be received in |
| // each direction. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 && |
| callee()->min_video_frames_received_per_track() > 0, |
| kMaxWaitForFramesMs); |
| |
| // Check rendered aspect ratio. |
| EXPECT_EQ(16.0 / 9, caller()->local_rendered_aspect_ratio()); |
| EXPECT_EQ(16.0 / 9, caller()->rendered_aspect_ratio()); |
| EXPECT_EQ(16.0 / 9, callee()->local_rendered_aspect_ratio()); |
| EXPECT_EQ(16.0 / 9, callee()->rendered_aspect_ratio()); |
| } |
| |
| // This test sets up an one-way call, with media only from caller to |
| // callee. |
| TEST_P(PeerConnectionIntegrationTest, OneWayMediaCall) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudioAndVideo(); |
| media_expectations.CallerExpectsNoAudio(); |
| media_expectations.CallerExpectsNoVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // This test sets up a audio call initially, with the callee rejecting video |
| // initially. Then later the callee decides to upgrade to audio/video, and |
| // initiates a new offer/answer exchange. |
| TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Initially, offer an audio/video stream from the caller, but refuse to |
| // send/receive video on the callee side. |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioTrack(); |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| PeerConnectionInterface::RTCOfferAnswerOptions options; |
| options.offer_to_receive_video = 0; |
| callee()->SetOfferAnswerOptions(options); |
| } else { |
| callee()->SetRemoteOfferHandler([this] { |
| callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop(); |
| }); |
| } |
| // Do offer/answer and make sure audio is still received end-to-end. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| { |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudio(); |
| media_expectations.ExpectNoVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| // Sanity check that the callee's description has a rejected video section. |
| ASSERT_NE(nullptr, callee()->pc()->local_description()); |
| const ContentInfo* callee_video_content = |
| GetFirstVideoContent(callee()->pc()->local_description()->description()); |
| ASSERT_NE(nullptr, callee_video_content); |
| EXPECT_TRUE(callee_video_content->rejected); |
| |
| // Now negotiate with video and ensure negotiation succeeds, with video |
| // frames and additional audio frames being received. |
| callee()->AddVideoTrack(); |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| PeerConnectionInterface::RTCOfferAnswerOptions options; |
| options.offer_to_receive_video = 1; |
| callee()->SetOfferAnswerOptions(options); |
| } else { |
| callee()->SetRemoteOfferHandler(nullptr); |
| caller()->SetRemoteOfferHandler([this] { |
| // The caller creates a new transceiver to receive video on when receiving |
| // the offer, but by default it is send only. |
| auto transceivers = caller()->pc()->GetTransceivers(); |
| ASSERT_EQ(3U, transceivers.size()); |
| ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO, |
| transceivers[2]->receiver()->media_type()); |
| transceivers[2]->sender()->SetTrack(caller()->CreateLocalVideoTrack()); |
| transceivers[2]->SetDirection(RtpTransceiverDirection::kSendRecv); |
| }); |
| } |
| callee()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| { |
| // Expect additional audio frames to be received after the upgrade. |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| } |
| |
| // Simpler than the above test; just add an audio track to an established |
| // video-only connection. |
| TEST_P(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Do initial offer/answer with just a video track. |
| caller()->AddVideoTrack(); |
| callee()->AddVideoTrack(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // Now add an audio track and do another offer/answer. |
| caller()->AddAudioTrack(); |
| callee()->AddAudioTrack(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // Ensure both audio and video frames are received end-to-end. |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // This test sets up a call that's transferred to a new caller with a different |
| // DTLS fingerprint. |
| TEST_P(PeerConnectionIntegrationTest, CallTransferredForCallee) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Keep the original peer around which will still send packets to the |
| // receiving client. These SRTP packets will be dropped. |
| std::unique_ptr<PeerConnectionWrapper> original_peer( |
| SetCallerPcWrapperAndReturnCurrent( |
| CreatePeerConnectionWrapperWithAlternateKey().release())); |
| // TODO(deadbeef): Why do we call Close here? That goes against the comment |
| // directly above. |
| original_peer->pc()->Close(); |
| |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // Wait for some additional frames to be transmitted end-to-end. |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // This test sets up a call that's transferred to a new callee with a different |
| // DTLS fingerprint. |
| TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Keep the original peer around which will still send packets to the |
| // receiving client. These SRTP packets will be dropped. |
| std::unique_ptr<PeerConnectionWrapper> original_peer( |
| SetCalleePcWrapperAndReturnCurrent( |
| CreatePeerConnectionWrapperWithAlternateKey().release())); |
| // TODO(deadbeef): Why do we call Close here? That goes against the comment |
| // directly above. |
| original_peer->pc()->Close(); |
| |
| ConnectFakeSignaling(); |
| callee()->AddAudioVideoTracks(); |
| caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // Wait for some additional frames to be transmitted end-to-end. |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // This test sets up a non-bundled call and negotiates bundling at the same |
| // time as starting an ICE restart. When bundling is in effect in the restart, |
| // the DTLS-SRTP context should be successfully reset. |
| TEST_P(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| // Remove the bundle group from the SDP received by the callee. |
| callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) { |
| desc->RemoveGroupByName("BUNDLE"); |
| }); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| { |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| // Now stop removing the BUNDLE group, and trigger an ICE restart. |
| callee()->SetReceivedSdpMunger(nullptr); |
| caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions()); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Expect additional frames to be received after the ICE restart. |
| { |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| } |
| |
| // Test CVO (Coordination of Video Orientation). If a video source is rotated |
| // and both peers support the CVO RTP header extension, the actual video frames |
| // don't need to be encoded in different resolutions, since the rotation is |
| // communicated through the RTP header extension. |
| TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Add rotated video tracks. |
| caller()->AddTrack( |
| caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90)); |
| callee()->AddTrack( |
| callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270)); |
| |
| // Wait for video frames to be received by both sides. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 && |
| callee()->min_video_frames_received_per_track() > 0, |
| kMaxWaitForFramesMs); |
| |
| // Ensure that the aspect ratio is unmodified. |
| // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test, |
| // not just assumed. |
| EXPECT_EQ(4.0 / 3, caller()->local_rendered_aspect_ratio()); |
| EXPECT_EQ(4.0 / 3, caller()->rendered_aspect_ratio()); |
| EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio()); |
| EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio()); |
| // Ensure that the CVO bits were surfaced to the renderer. |
| EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation()); |
| EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation()); |
| } |
| |
| // Test that when the CVO extension isn't supported, video is rotated the |
| // old-fashioned way, by encoding rotated frames. |
| TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Add rotated video tracks. |
| caller()->AddTrack( |
| caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90)); |
| callee()->AddTrack( |
| callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270)); |
| |
| // Remove the CVO extension from the offered SDP. |
| callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) { |
| cricket::VideoContentDescription* video = |
| GetFirstVideoContentDescription(desc); |
| video->ClearRtpHeaderExtensions(); |
| }); |
| // Wait for video frames to be received by both sides. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 && |
| callee()->min_video_frames_received_per_track() > 0, |
| kMaxWaitForFramesMs); |
| |
| // Expect that the aspect ratio is inversed to account for the 90/270 degree |
| // rotation. |
| // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test, |
| // not just assumed. |
| EXPECT_EQ(3.0 / 4, caller()->local_rendered_aspect_ratio()); |
| EXPECT_EQ(3.0 / 4, caller()->rendered_aspect_ratio()); |
| EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio()); |
| EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio()); |
| // Expect that each endpoint is unaware of the rotation of the other endpoint. |
| EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation()); |
| EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation()); |
| } |
| |
| // Test that if the answerer rejects the audio m= section, no audio is sent or |
| // received, but video still can be. |
| TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| // Only add video track for callee, and set offer_to_receive_audio to 0, so |
| // it will reject the audio m= section completely. |
| PeerConnectionInterface::RTCOfferAnswerOptions options; |
| options.offer_to_receive_audio = 0; |
| callee()->SetOfferAnswerOptions(options); |
| } else { |
| // Stopping the audio RtpTransceiver will cause the media section to be |
| // rejected in the answer. |
| callee()->SetRemoteOfferHandler([this] { |
| callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)->Stop(); |
| }); |
| } |
| callee()->AddTrack(callee()->CreateLocalVideoTrack()); |
| // Do offer/answer and wait for successful end-to-end video frames. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalVideo(); |
| media_expectations.ExpectNoAudio(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| |
| // Sanity check that the callee's description has a rejected audio section. |
| ASSERT_NE(nullptr, callee()->pc()->local_description()); |
| const ContentInfo* callee_audio_content = |
| GetFirstAudioContent(callee()->pc()->local_description()->description()); |
| ASSERT_NE(nullptr, callee_audio_content); |
| EXPECT_TRUE(callee_audio_content->rejected); |
| if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) { |
| // The caller's transceiver should have stopped after receiving the answer. |
| EXPECT_TRUE(caller() |
| ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO) |
| ->stopped()); |
| } |
| } |
| |
| // Test that if the answerer rejects the video m= section, no video is sent or |
| // received, but audio still can be. |
| TEST_P(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| // Only add audio track for callee, and set offer_to_receive_video to 0, so |
| // it will reject the video m= section completely. |
| PeerConnectionInterface::RTCOfferAnswerOptions options; |
| options.offer_to_receive_video = 0; |
| callee()->SetOfferAnswerOptions(options); |
| } else { |
| // Stopping the video RtpTransceiver will cause the media section to be |
| // rejected in the answer. |
| callee()->SetRemoteOfferHandler([this] { |
| callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop(); |
| }); |
| } |
| callee()->AddTrack(callee()->CreateLocalAudioTrack()); |
| // Do offer/answer and wait for successful end-to-end audio frames. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudio(); |
| media_expectations.ExpectNoVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| |
| // Sanity check that the callee's description has a rejected video section. |
| ASSERT_NE(nullptr, callee()->pc()->local_description()); |
| const ContentInfo* callee_video_content = |
| GetFirstVideoContent(callee()->pc()->local_description()->description()); |
| ASSERT_NE(nullptr, callee_video_content); |
| EXPECT_TRUE(callee_video_content->rejected); |
| if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) { |
| // The caller's transceiver should have stopped after receiving the answer. |
| EXPECT_TRUE(caller() |
| ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO) |
| ->stopped()); |
| } |
| } |
| |
| // Test that if the answerer rejects both audio and video m= sections, nothing |
| // bad happens. |
| // TODO(deadbeef): Test that a data channel still works. Currently this doesn't |
| // test anything but the fact that negotiation succeeds, which doesn't mean |
| // much. |
| TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| // Don't give the callee any tracks, and set offer_to_receive_X to 0, so it |
| // will reject both audio and video m= sections. |
| PeerConnectionInterface::RTCOfferAnswerOptions options; |
| options.offer_to_receive_audio = 0; |
| options.offer_to_receive_video = 0; |
| callee()->SetOfferAnswerOptions(options); |
| } else { |
| callee()->SetRemoteOfferHandler([this] { |
| // Stopping all transceivers will cause all media sections to be rejected. |
| for (auto transceiver : callee()->pc()->GetTransceivers()) { |
| transceiver->Stop(); |
| } |
| }); |
| } |
| // Do offer/answer and wait for stable signaling state. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Sanity check that the callee's description has rejected m= sections. |
| ASSERT_NE(nullptr, callee()->pc()->local_description()); |
| const ContentInfo* callee_audio_content = |
| GetFirstAudioContent(callee()->pc()->local_description()->description()); |
| ASSERT_NE(nullptr, callee_audio_content); |
| EXPECT_TRUE(callee_audio_content->rejected); |
| const ContentInfo* callee_video_content = |
| GetFirstVideoContent(callee()->pc()->local_description()->description()); |
| ASSERT_NE(nullptr, callee_video_content); |
| EXPECT_TRUE(callee_video_content->rejected); |
| } |
| |
| // This test sets up an audio and video call between two parties. After the |
| // call runs for a while, the caller sends an updated offer with video being |
| // rejected. Once the re-negotiation is done, the video flow should stop and |
| // the audio flow should continue. |
| TEST_P(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| { |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| // Renegotiate, rejecting the video m= section. |
| if (sdp_semantics_ == SdpSemantics::kPlanB) { |
| caller()->SetGeneratedSdpMunger( |
| [](cricket::SessionDescription* description) { |
| for (cricket::ContentInfo& content : description->contents()) { |
| if (cricket::IsVideoContent(&content)) { |
| content.rejected = true; |
| } |
| } |
| }); |
| } else { |
| caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop(); |
| } |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs); |
| |
| // Sanity check that the caller's description has a rejected video section. |
| ASSERT_NE(nullptr, caller()->pc()->local_description()); |
| const ContentInfo* caller_video_content = |
| GetFirstVideoContent(caller()->pc()->local_description()->description()); |
| ASSERT_NE(nullptr, caller_video_content); |
| EXPECT_TRUE(caller_video_content->rejected); |
| // Wait for some additional audio frames to be received. |
| { |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudio(); |
| media_expectations.ExpectNoVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| } |
| |
| // Do one offer/answer with audio, another that disables it (rejecting the m= |
| // section), and another that re-enables it. Regression test for: |
| // bugs.webrtc.org/6023 |
| TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| |
| // Add audio track, do normal offer/answer. |
| rtc::scoped_refptr<webrtc::AudioTrackInterface> track = |
| caller()->CreateLocalAudioTrack(); |
| rtc::scoped_refptr<webrtc::RtpSenderInterface> sender = |
| caller()->pc()->AddTrack(track, {"stream"}).MoveValue(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Remove audio track, and set offer_to_receive_audio to false to cause the |
| // m= section to be completely disabled, not just "recvonly". |
| caller()->pc()->RemoveTrack(sender); |
| PeerConnectionInterface::RTCOfferAnswerOptions options; |
| options.offer_to_receive_audio = 0; |
| caller()->SetOfferAnswerOptions(options); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Add the audio track again, expecting negotiation to succeed and frames to |
| // flow. |
| sender = caller()->pc()->AddTrack(track, {"stream"}).MoveValue(); |
| options.offer_to_receive_audio = 1; |
| caller()->SetOfferAnswerOptions(options); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudio(); |
| EXPECT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // Basic end-to-end test, but without SSRC/MSID signaling. This functionality |
| // is needed to support legacy endpoints. |
| // TODO(deadbeef): When we support the MID extension and demuxing on MID, also |
| // add a test for an end-to-end test without MID signaling either (basically, |
| // the minimum acceptable SDP). |
| TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Add audio and video, testing that packets can be demuxed on payload type. |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| // Remove SSRCs and MSIDs from the received offer SDP. |
| callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // Basic end-to-end test, without SSRC signaling. This means that the track |
| // was created properly and frames are delivered when the MSIDs are communicated |
| // with a=msid lines and no a=ssrc lines. |
| TEST_F(PeerConnectionIntegrationTestUnifiedPlan, |
| EndToEndCallWithoutSsrcSignaling) { |
| const char kStreamId[] = "streamId"; |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Add just audio tracks. |
| caller()->AddTrack(caller()->CreateLocalAudioTrack(), {kStreamId}); |
| callee()->AddAudioTrack(); |
| |
| // Remove SSRCs from the received offer SDP. |
| callee()->SetReceivedSdpMunger(RemoveSsrcsAndKeepMsids); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudio(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // Tests that video flows between multiple video tracks when SSRCs are not |
| // signaled. This exercises the MID RTP header extension which is needed to |
| // demux the incoming video tracks. |
| TEST_F(PeerConnectionIntegrationTestUnifiedPlan, |
| EndToEndCallWithTwoVideoTracksAndNoSignaledSsrc) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddVideoTrack(); |
| caller()->AddVideoTrack(); |
| callee()->AddVideoTrack(); |
| callee()->AddVideoTrack(); |
| |
| caller()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids); |
| callee()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| ASSERT_EQ(2u, caller()->pc()->GetReceivers().size()); |
| ASSERT_EQ(2u, callee()->pc()->GetReceivers().size()); |
| |
| // Expect video to be received in both directions on both tracks. |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalVideo(); |
| EXPECT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // Test that if two video tracks are sent (from caller to callee, in this test), |
| // they're transmitted correctly end-to-end. |
| TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Add one audio/video stream, and one video-only stream. |
| caller()->AddAudioVideoTracks(); |
| caller()->AddVideoTrack(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| ASSERT_EQ(3u, callee()->pc()->GetReceivers().size()); |
| |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| static void MakeSpecCompliantMaxBundleOffer(cricket::SessionDescription* desc) { |
| bool first = true; |
| for (cricket::ContentInfo& content : desc->contents()) { |
| if (first) { |
| first = false; |
| continue; |
| } |
| content.bundle_only = true; |
| } |
| first = true; |
| for (cricket::TransportInfo& transport : desc->transport_infos()) { |
| if (first) { |
| first = false; |
| continue; |
| } |
| transport.description.ice_ufrag.clear(); |
| transport.description.ice_pwd.clear(); |
| transport.description.connection_role = cricket::CONNECTIONROLE_NONE; |
| transport.description.identity_fingerprint.reset(nullptr); |
| } |
| } |
| |
| // Test that if applying a true "max bundle" offer, which uses ports of 0, |
| // "a=bundle-only", omitting "a=fingerprint", "a=setup", "a=ice-ufrag" and |
| // "a=ice-pwd" for all but the audio "m=" section, negotiation still completes |
| // successfully and media flows. |
| // TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works. |
| // TODO(deadbeef): Won't need this test once we start generating actual |
| // standards-compliant SDP. |
| TEST_P(PeerConnectionIntegrationTest, |
| EndToEndCallWithSpecCompliantMaxBundleOffer) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| // Do the equivalent of setting the port to 0, adding a=bundle-only, and |
| // removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all |
| // but the first m= section. |
| callee()->SetReceivedSdpMunger(MakeSpecCompliantMaxBundleOffer); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // Test that we can receive the audio output level from a remote audio track. |
| // TODO(deadbeef): Use a fake audio source and verify that the output level is |
| // exactly what the source on the other side was configured with. |
| TEST_P(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Just add an audio track. |
| caller()->AddAudioTrack(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Get the audio output level stats. Note that the level is not available |
| // until an RTCP packet has been received. |
| EXPECT_TRUE_WAIT(callee()->OldGetStats()->AudioOutputLevel() > 0, |
| kMaxWaitForFramesMs); |
| } |
| |
| // Test that an audio input level is reported. |
| // TODO(deadbeef): Use a fake audio source and verify that the input level is |
| // exactly what the source was configured with. |
| TEST_P(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| // Just add an audio track. |
| caller()->AddAudioTrack(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Get the audio input level stats. The level should be available very |
| // soon after the test starts. |
| EXPECT_TRUE_WAIT(caller()->OldGetStats()->AudioInputLevel() > 0, |
| kMaxWaitForStatsMs); |
| } |
| |
| // Test that we can get incoming byte counts from both audio and video tracks. |
| TEST_P(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| // Do offer/answer, wait for the callee to receive some frames. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| |
| // Get a handle to the remote tracks created, so they can be used as GetStats |
| // filters. |
| for (auto receiver : callee()->pc()->GetReceivers()) { |
| // We received frames, so we definitely should have nonzero "received bytes" |
| // stats at this point. |
| EXPECT_GT(callee()->OldGetStatsForTrack(receiver->track())->BytesReceived(), |
| 0); |
| } |
| } |
| |
| // Test that we can get outgoing byte counts from both audio and video tracks. |
| TEST_P(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| auto audio_track = caller()->CreateLocalAudioTrack(); |
| auto video_track = caller()->CreateLocalVideoTrack(); |
| caller()->AddTrack(audio_track); |
| caller()->AddTrack(video_track); |
| // Do offer/answer, wait for the callee to receive some frames. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| |
| // The callee received frames, so we definitely should have nonzero "sent |
| // bytes" stats at this point. |
| EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0); |
| EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0); |
| } |
| |
| // Test that we can get capture start ntp time. |
| TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioTrack(); |
| |
| callee()->AddAudioTrack(); |
| |
| // Do offer/answer, wait for the callee to receive some frames. |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Get the remote audio track created on the receiver, so they can be used as |
| // GetStats filters. |
| auto receivers = callee()->pc()->GetReceivers(); |
| ASSERT_EQ(1u, receivers.size()); |
| auto remote_audio_track = receivers[0]->track(); |
| |
| // Get the audio output level stats. Note that the level is not available |
| // until an RTCP packet has been received. |
| EXPECT_TRUE_WAIT( |
| callee()->OldGetStatsForTrack(remote_audio_track)->CaptureStartNtpTime() > |
| 0, |
| 2 * kMaxWaitForFramesMs); |
| } |
| |
| // Test that we can get stats (using the new stats implemnetation) for |
| // unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in |
| // SDP. |
| TEST_P(PeerConnectionIntegrationTest, |
| GetStatsForUnsignaledStreamWithNewStatsApi) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioTrack(); |
| // Remove SSRCs and MSIDs from the received offer SDP. |
| callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudio(1); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| |
| // We received a frame, so we should have nonzero "bytes received" stats for |
| // the unsignaled stream, if stats are working for it. |
| rtc::scoped_refptr<const webrtc::RTCStatsReport> report = |
| callee()->NewGetStats(); |
| ASSERT_NE(nullptr, report); |
| auto inbound_stream_stats = |
| report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>(); |
| ASSERT_EQ(1U, inbound_stream_stats.size()); |
| ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined()); |
| ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U); |
| ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined()); |
| } |
| |
| // Same as above but for the legacy stats implementation. |
| TEST_P(PeerConnectionIntegrationTest, |
| GetStatsForUnsignaledStreamWithOldStatsApi) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioTrack(); |
| // Remove SSRCs and MSIDs from the received offer SDP. |
| callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| |
| // Note that, since the old stats implementation associates SSRCs with tracks |
| // using SDP, when SSRCs aren't signaled in SDP these stats won't have an |
| // associated track ID. So we can't use the track "selector" argument. |
| // |
| // Also, we use "EXPECT_TRUE_WAIT" because the stats collector may decide to |
| // return cached stats if not enough time has passed since the last update. |
| EXPECT_TRUE_WAIT(callee()->OldGetStats()->BytesReceived() > 0, |
| kDefaultTimeout); |
| } |
| |
| // Test that we can successfully get the media related stats (audio level |
| // etc.) for the unsignaled stream. |
| TEST_P(PeerConnectionIntegrationTest, |
| GetMediaStatsForUnsignaledStreamWithNewStatsApi) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| // Remove SSRCs and MSIDs from the received offer SDP. |
| callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudio(1); |
| media_expectations.CalleeExpectsSomeVideo(1); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| |
| rtc::scoped_refptr<const webrtc::RTCStatsReport> report = |
| callee()->NewGetStats(); |
| ASSERT_NE(nullptr, report); |
| |
| auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>(); |
| auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats); |
| ASSERT_GE(audio_index, 0); |
| EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined()); |
| } |
| |
| // Helper for test below. |
| void ModifySsrcs(cricket::SessionDescription* desc) { |
| for (ContentInfo& content : desc->contents()) { |
| for (StreamParams& stream : |
| content.media_description()->mutable_streams()) { |
| for (uint32_t& ssrc : stream.ssrcs) { |
| ssrc = rtc::CreateRandomId(); |
| } |
| } |
| } |
| } |
| |
| // Test that the "RTCMediaSteamTrackStats" object is updated correctly when |
| // SSRCs are unsignaled, and the SSRC of the received (audio) stream changes. |
| // This should result in two "RTCInboundRTPStreamStats", but only one |
| // "RTCMediaStreamTrackStats", whose counters go up continuously rather than |
| // being reset to 0 once the SSRC change occurs. |
| // |
| // Regression test for this bug: |
| // https://bugs.chromium.org/p/webrtc/issues/detail?id=8158 |
| // |
| // The bug causes the track stats to only represent one of the two streams: |
| // whichever one has the higher SSRC. So with this bug, there was a 50% chance |
| // that the track stat counters would reset to 0 when the new stream is |
| // received, and a 50% chance that they'll stop updating (while |
| // "concealed_samples" continues increasing, due to silence being generated for |
| // the inactive stream). |
| TEST_P(PeerConnectionIntegrationTest, |
| TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) { |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioTrack(); |
| // Remove SSRCs and MSIDs from the received offer SDP, simulating an endpoint |
| // that doesn't signal SSRCs (from the callee's perspective). |
| callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // Wait for 50 audio frames (500ms of audio) to be received by the callee. |
| { |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudio(50); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| // Some audio frames were received, so we should have nonzero "samples |
| // received" for the track. |
| rtc::scoped_refptr<const webrtc::RTCStatsReport> report = |
| callee()->NewGetStats(); |
| ASSERT_NE(nullptr, report); |
| auto track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>(); |
| ASSERT_EQ(1U, track_stats.size()); |
| ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined()); |
| ASSERT_GT(*track_stats[0]->total_samples_received, 0U); |
| // uint64_t prev_samples_received = *track_stats[0]->total_samples_received; |
| |
| // Create a new offer and munge it to cause the caller to use a new SSRC. |
| caller()->SetGeneratedSdpMunger(ModifySsrcs); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // Wait for 25 more audio frames (250ms of audio) to be received, from the new |
| // SSRC. |
| { |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudio(25); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| report = callee()->NewGetStats(); |
| ASSERT_NE(nullptr, report); |
| track_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>(); |
| ASSERT_EQ(1U, track_stats.size()); |
| ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined()); |
| // The "total samples received" stat should only be greater than it was |
| // before. |
| // TODO(deadbeef): Uncomment this assertion once the bug is completely fixed. |
| // Right now, the new SSRC will cause the counters to reset to 0. |
| // EXPECT_GT(*track_stats[0]->total_samples_received, prev_samples_received); |
| |
| // Additionally, the percentage of concealed samples (samples generated to |
| // conceal packet loss) should be less than 50%. If it's greater, that's a |
| // good sign that we're seeing stats from the old stream that's no longer |
| // receiving packets, and is generating concealed samples of silence. |
| constexpr double kAcceptableConcealedSamplesPercentage = 0.50; |
| ASSERT_TRUE(track_stats[0]->concealed_samples.is_defined()); |
| EXPECT_LT(*track_stats[0]->concealed_samples, |
| *track_stats[0]->total_samples_received * |
| kAcceptableConcealedSamplesPercentage); |
| |
| // Also ensure that we have two "RTCInboundRTPStreamStats" as expected, as a |
| // sanity check that the SSRC really changed. |
| // TODO(deadbeef): This isn't working right now, because we're not returning |
| // *any* stats for the inactive stream. Uncomment when the bug is completely |
| // fixed. |
| // auto inbound_stream_stats = |
| // report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>(); |
| // ASSERT_EQ(2U, inbound_stream_stats.size()); |
| } |
| |
| // Test that DTLS 1.0 is used if both sides only support DTLS 1.0. |
| TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) { |
| PeerConnectionFactory::Options dtls_10_options; |
| dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
| ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options, |
| dtls_10_options)); |
| ConnectFakeSignaling(); |
| // Do normal offer/answer and wait for some frames to be received in each |
| // direction. |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // Test getting cipher stats and UMA metrics when DTLS 1.0 is negotiated. |
| TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) { |
| PeerConnectionFactory::Options dtls_10_options; |
| dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
| ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options, |
| dtls_10_options)); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout); |
| EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher( |
| caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT), |
| kDefaultTimeout); |
| EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), |
| caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout); |
| // TODO(bugs.webrtc.org/9456): Fix it. |
| EXPECT_EQ(1, webrtc::metrics::NumEvents( |
| "WebRTC.PeerConnection.SrtpCryptoSuite.Audio", |
| kDefaultSrtpCryptoSuite)); |
| } |
| |
| // Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated. |
| TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) { |
| PeerConnectionFactory::Options dtls_12_options; |
| dtls_12_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
| ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options, |
| dtls_12_options)); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout); |
| EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher( |
| caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT), |
| kDefaultTimeout); |
| EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite), |
| caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout); |
| // TODO(bugs.webrtc.org/9456): Fix it. |
| EXPECT_EQ(1, webrtc::metrics::NumEvents( |
| "WebRTC.PeerConnection.SrtpCryptoSuite.Audio", |
| kDefaultSrtpCryptoSuite)); |
| } |
| |
| // Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the |
| // callee only supports 1.0. |
| TEST_P(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) { |
| PeerConnectionFactory::Options caller_options; |
| caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
| PeerConnectionFactory::Options callee_options; |
| callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
| ASSERT_TRUE( |
| CreatePeerConnectionWrappersWithOptions(caller_options, callee_options)); |
| ConnectFakeSignaling(); |
| // Do normal offer/answer and wait for some frames to be received in each |
| // direction. |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // Test that DTLS 1.0 can be used if the caller only supports DTLS 1.0 and the |
| // callee supports 1.2. |
| TEST_P(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) { |
| PeerConnectionFactory::Options caller_options; |
| caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10; |
| PeerConnectionFactory::Options callee_options; |
| callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12; |
| ASSERT_TRUE( |
| CreatePeerConnectionWrappersWithOptions(caller_options, callee_options)); |
| ConnectFakeSignaling(); |
| // Do normal offer/answer and wait for some frames to be received in each |
| // direction. |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.ExpectBidirectionalAudioAndVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| } |
| |
| // The three tests below verify that "enable_aes128_sha1_32_crypto_cipher" |
| // works as expected; the cipher should only be used if enabled by both sides. |
| TEST_P(PeerConnectionIntegrationTest, |
| Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) { |
| PeerConnectionFactory::Options caller_options; |
| caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true; |
| PeerConnectionFactory::Options callee_options; |
| callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = |
| false; |
| int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80; |
| TestNegotiatedCipherSuite(caller_options, callee_options, |
| expected_cipher_suite); |
| } |
| |
| TEST_P(PeerConnectionIntegrationTest, |
| Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) { |
| PeerConnectionFactory::Options caller_options; |
| caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = |
| false; |
| PeerConnectionFactory::Options callee_options; |
| callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true; |
| int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80; |
| TestNegotiatedCipherSuite(caller_options, callee_options, |
| expected_cipher_suite); |
| } |
| |
| TEST_P(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) { |
| PeerConnectionFactory::Options caller_options; |
| caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true; |
| PeerConnectionFactory::Options callee_options; |
| callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true; |
| int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_32; |
| TestNegotiatedCipherSuite(caller_options, callee_options, |
| expected_cipher_suite); |
| } |
| |
| // Test that a non-GCM cipher is used if both sides only support non-GCM. |
| TEST_P(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) { |
| bool local_gcm_enabled = false; |
| bool remote_gcm_enabled = false; |
| int expected_cipher_suite = kDefaultSrtpCryptoSuite; |
| TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, |
| expected_cipher_suite); |
| } |
| |
| // Test that a GCM cipher is used if both ends support it. |
| TEST_P(PeerConnectionIntegrationTest, GcmCipherUsedWhenGcmSupported) { |
| bool local_gcm_enabled = true; |
| bool remote_gcm_enabled = true; |
| int expected_cipher_suite = kDefaultSrtpCryptoSuiteGcm; |
| TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, |
| expected_cipher_suite); |
| } |
| |
| // Test that GCM isn't used if only the offerer supports it. |
| TEST_P(PeerConnectionIntegrationTest, |
| NonGcmCipherUsedWhenOnlyCallerSupportsGcm) { |
| bool local_gcm_enabled = true; |
| bool remote_gcm_enabled = false; |
| int expected_cipher_suite = kDefaultSrtpCryptoSuite; |
| TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, |
| expected_cipher_suite); |
| } |
| |
| // Test that GCM isn't used if only the answerer supports it. |
| TEST_P(PeerConnectionIntegrationTest, |
| NonGcmCipherUsedWhenOnlyCalleeSupportsGcm) { |
| bool local_gcm_enabled = false; |
| bool remote_gcm_enabled = true; |
| int expected_cipher_suite = kDefaultSrtpCryptoSuite; |
| TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled, |
| expected_cipher_suite); |
| } |
| |
| // Verify that media can be transmitted end-to-end when GCM crypto suites are |
| // enabled. Note that the above tests, such as GcmCipherUsedWhenGcmSupported, |
| // only verify that a GCM cipher is negotiated, and not necessarily that SRTP |
| // works with it. |
| TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) { |
| PeerConnectionFactory::Options gcm_options; |
| gcm_options.crypto_options.srtp.enable_gcm_crypto_suites = true; |
| ASSERT_TRUE( |
| CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options)); |
| ConnectFakeSignaling(); |
| // Do normal offer/answer and wait for some frames to be received in each |
| // direction. |
| caller()->AddAudioVideoTracks(); |
| callee()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| |