| /* |
| * 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. |
| */ |
| |
| #ifndef PC_TEST_INTEGRATION_TEST_HELPERS_H_ |
| #define PC_TEST_INTEGRATION_TEST_HELPERS_H_ |
| |
| #include <limits.h> |
| #include <stdint.h> |
| #include <stdio.h> |
| |
| #include <algorithm> |
| #include <functional> |
| #include <list> |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/types/optional.h" |
| #include "api/audio_options.h" |
| #include "api/call/call_factory_interface.h" |
| #include "api/candidate.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/data_channel_interface.h" |
| #include "api/ice_transport_interface.h" |
| #include "api/jsep.h" |
| #include "api/media_stream_interface.h" |
| #include "api/media_types.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/rtc_error.h" |
| #include "api/rtc_event_log/rtc_event_log_factory.h" |
| #include "api/rtc_event_log/rtc_event_log_factory_interface.h" |
| #include "api/rtc_event_log_output.h" |
| #include "api/rtp_receiver_interface.h" |
| #include "api/rtp_sender_interface.h" |
| #include "api/rtp_transceiver_interface.h" |
| #include "api/scoped_refptr.h" |
| #include "api/stats/rtc_stats.h" |
| #include "api/stats/rtc_stats_report.h" |
| #include "api/stats/rtcstats_objects.h" |
| #include "api/task_queue/default_task_queue_factory.h" |
| #include "api/task_queue/task_queue_factory.h" |
| #include "api/transport/field_trial_based_config.h" |
| #include "api/transport/webrtc_key_value_config.h" |
| #include "api/uma_metrics.h" |
| #include "api/video/video_rotation.h" |
| #include "api/video_codecs/sdp_video_format.h" |
| #include "api/video_codecs/video_decoder_factory.h" |
| #include "api/video_codecs/video_encoder_factory.h" |
| #include "call/call.h" |
| #include "logging/rtc_event_log/fake_rtc_event_log_factory.h" |
| #include "media/base/media_engine.h" |
| #include "media/base/stream_params.h" |
| #include "media/engine/fake_webrtc_video_engine.h" |
| #include "media/engine/webrtc_media_engine.h" |
| #include "media/engine/webrtc_media_engine_defaults.h" |
| #include "modules/audio_device/include/audio_device.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "modules/audio_processing/test/audio_processing_builder_for_testing.h" |
| #include "p2p/base/fake_ice_transport.h" |
| #include "p2p/base/ice_transport_internal.h" |
| #include "p2p/base/mock_async_resolver.h" |
| #include "p2p/base/p2p_constants.h" |
| #include "p2p/base/port.h" |
| #include "p2p/base/port_allocator.h" |
| #include "p2p/base/port_interface.h" |
| #include "p2p/base/test_stun_server.h" |
| #include "p2p/base/test_turn_customizer.h" |
| #include "p2p/base/test_turn_server.h" |
| #include "p2p/client/basic_port_allocator.h" |
| #include "pc/dtmf_sender.h" |
| #include "pc/local_audio_source.h" |
| #include "pc/media_session.h" |
| #include "pc/peer_connection.h" |
| #include "pc/peer_connection_factory.h" |
| #include "pc/peer_connection_proxy.h" |
| #include "pc/rtp_media_utils.h" |
| #include "pc/session_description.h" |
| #include "pc/test/fake_audio_capture_module.h" |
| #include "pc/test/fake_periodic_video_source.h" |
| #include "pc/test/fake_periodic_video_track_source.h" |
| #include "pc/test/fake_rtc_certificate_generator.h" |
| #include "pc/test/fake_video_track_renderer.h" |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "pc/video_track_source.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/fake_clock.h" |
| #include "rtc_base/fake_mdns_responder.h" |
| #include "rtc_base/fake_network.h" |
| #include "rtc_base/firewall_socket_server.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/helpers.h" |
| #include "rtc_base/ip_address.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/mdns_responder_interface.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "rtc_base/rtc_certificate_generator.h" |
| #include "rtc_base/socket_address.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "rtc_base/task_utils/pending_task_safety_flag.h" |
| #include "rtc_base/task_utils/to_queued_task.h" |
| #include "rtc_base/test_certificate_verifier.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/time_utils.h" |
| #include "rtc_base/virtual_socket_server.h" |
| #include "system_wrappers/include/metrics.h" |
| #include "test/field_trial.h" |
| #include "test/gmock.h" |
| |
| namespace webrtc { |
| |
| using ::cricket::ContentInfo; |
| using ::cricket::StreamParams; |
| using ::rtc::SocketAddress; |
| using ::testing::_; |
| using ::testing::Combine; |
| using ::testing::Contains; |
| using ::testing::DoAll; |
| using ::testing::ElementsAre; |
| using ::testing::NiceMock; |
| using ::testing::Return; |
| using ::testing::SetArgPointee; |
| using ::testing::UnorderedElementsAreArray; |
| using ::testing::Values; |
| using RTCConfiguration = PeerConnectionInterface::RTCConfiguration; |
| |
| 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(); |
| |
| // Remove all stream information (SSRCs, track IDs, etc.) and "msid-semantic" |
| // attribute from received SDP, simulating a legacy endpoint. |
| void RemoveSsrcsAndMsids(cricket::SessionDescription* desc); |
| |
| // 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); |
| |
| int FindFirstMediaStatsIndexByKind( |
| const std::string& kind, |
| const std::vector<const webrtc::RTCMediaStreamTrackStats*>& |
| media_stats_vec); |
| |
| 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_; |
| }; |
| |
| // 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 PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver, |
| public SignalingMessageReceiver { |
| public: |
| // Different factory methods for convenience. |
| // TODO(deadbeef): Could use the pattern of: |
| // |
| // PeerConnectionIntegrationWrapper = |
| // WrapperBuilder.WithConfig(...).WithOptions(...).build(); |
| // |
| // To reduce some code duplication. |
| static PeerConnectionIntegrationWrapper* CreateWithDtlsIdentityStore( |
| const std::string& debug_name, |
| std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, |
| rtc::Thread* network_thread, |
| rtc::Thread* worker_thread) { |
| PeerConnectionIntegrationWrapper* client( |
| new PeerConnectionIntegrationWrapper(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, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false)) { |
| 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 = CreateOfferAndWait(); |
| 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 SetRemoteAsyncResolver(rtc::MockAsyncResolver* resolver) { |
| remote_async_resolver_ = resolver; |
| } |
| |
| // 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 standardized ICE connection state in order that has been seen by the |
| // observer. |
| std::vector<PeerConnectionInterface::IceConnectionState> |
| standardized_ice_connection_state_history() const { |
| return standardized_ice_connection_state_history_; |
| } |
| |
| // 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_; |
| } |
| std::vector<cricket::CandidatePairChangeEvent> |
| ice_candidate_pair_change_history() const { |
| return ice_candidate_pair_change_history_; |
| } |
| |
| // Every PeerConnection signaling state in order that has been seen by the |
| // observer. |
| std::vector<PeerConnectionInterface::SignalingState> |
| peer_connection_signaling_state_history() const { |
| return peer_connection_signaling_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 (const 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) { |
| auto observer = rtc::make_ref_counted<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() { |
| auto callback = |
| rtc::make_ref_counted<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::IceConnectionState |
| standardized_ice_connection_state() { |
| return pc()->standardized_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_manager() const { |
| return fake_network_manager_.get(); |
| } |
| cricket::PortAllocator* port_allocator() const { return port_allocator_; } |
| |
| webrtc::FakeRtcEventLogFactory* event_log_factory() const { |
| return event_log_factory_; |
| } |
| |
| const cricket::Candidate& last_candidate_gathered() const { |
| return last_candidate_gathered_; |
| } |
| const cricket::IceCandidateErrorEvent& error_event() const { |
| return error_event_; |
| } |
| |
| // Sets the mDNS responder for the owned fake network manager and keeps a |
| // reference to the responder. |
| void SetMdnsResponder( |
| std::unique_ptr<webrtc::FakeMdnsResponder> mdns_responder) { |
| RTC_DCHECK(mdns_responder != nullptr); |
| mdns_responder_ = mdns_responder.get(); |
| network_manager()->set_mdns_responder(std::move(mdns_responder)); |
| } |
| |
| // Returns null on failure. |
| std::unique_ptr<SessionDescriptionInterface> CreateOfferAndWait() { |
| auto observer = |
| rtc::make_ref_counted<MockCreateSessionDescriptionObserver>(); |
| pc()->CreateOffer(observer, offer_answer_options_); |
| return WaitForDescriptionFromObserver(observer); |
| } |
| bool Rollback() { |
| return SetRemoteDescription( |
| webrtc::CreateSessionDescription(SdpType::kRollback, "")); |
| } |
| |
| // Functions for querying stats. |
| void StartWatchingDelayStats() { |
| // Get the baseline numbers for audio_packets and audio_delay. |
| auto received_stats = NewGetStats(); |
| auto track_stats = |
| received_stats->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>()[0]; |
| ASSERT_TRUE(track_stats->relative_packet_arrival_delay.is_defined()); |
| auto rtp_stats = |
| received_stats->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>()[0]; |
| ASSERT_TRUE(rtp_stats->packets_received.is_defined()); |
| ASSERT_TRUE(rtp_stats->track_id.is_defined()); |
| audio_track_stats_id_ = track_stats->id(); |
| ASSERT_TRUE(received_stats->Get(audio_track_stats_id_)); |
| rtp_stats_id_ = rtp_stats->id(); |
| ASSERT_EQ(audio_track_stats_id_, *rtp_stats->track_id); |
| audio_packets_stat_ = *rtp_stats->packets_received; |
| audio_delay_stat_ = *track_stats->relative_packet_arrival_delay; |
| audio_samples_stat_ = *track_stats->total_samples_received; |
| audio_concealed_stat_ = *track_stats->concealed_samples; |
| } |
| |
| void UpdateDelayStats(std::string tag, int desc_size) { |
| auto report = NewGetStats(); |
| auto track_stats = |
| report->GetAs<webrtc::RTCMediaStreamTrackStats>(audio_track_stats_id_); |
| ASSERT_TRUE(track_stats); |
| auto rtp_stats = |
| report->GetAs<webrtc::RTCInboundRTPStreamStats>(rtp_stats_id_); |
| ASSERT_TRUE(rtp_stats); |
| auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_; |
| auto delta_rpad = |
| *track_stats->relative_packet_arrival_delay - audio_delay_stat_; |
| auto recent_delay = delta_packets > 0 ? delta_rpad / delta_packets : -1; |
| // The purpose of these checks is to sound the alarm early if we introduce |
| // serious regressions. The numbers are not acceptable for production, but |
| // occur on slow bots. |
| // |
| // An average relative packet arrival delay over the renegotiation of |
| // > 100 ms indicates that something is dramatically wrong, and will impact |
| // quality for sure. |
| // Worst bots: |
| // linux_x86_dbg at 0.206 |
| #if !defined(NDEBUG) |
| EXPECT_GT(0.25, recent_delay) << tag << " size " << desc_size; |
| #else |
| EXPECT_GT(0.1, recent_delay) << tag << " size " << desc_size; |
| #endif |
| auto delta_samples = |
| *track_stats->total_samples_received - audio_samples_stat_; |
| auto delta_concealed = |
| *track_stats->concealed_samples - audio_concealed_stat_; |
| // These limits should be adjusted down as we improve: |
| // |
| // Concealing more than 4000 samples during a renegotiation is unacceptable. |
| // But some bots are slow. |
| |
| // Worst bots: |
| // linux_more_configs bot at conceal count 5184 |
| // android_arm_rel at conceal count 9241 |
| // linux_x86_dbg at 15174 |
| #if !defined(NDEBUG) |
| EXPECT_GT(18000U, delta_concealed) << "Concealed " << delta_concealed |
| << " of " << delta_samples << " samples"; |
| #else |
| EXPECT_GT(15000U, delta_concealed) << "Concealed " << delta_concealed |
| << " of " << delta_samples << " samples"; |
| #endif |
| // Concealing more than 20% of samples during a renegotiation is |
| // unacceptable. |
| // Worst bots: |
| // linux_more_configs bot at conceal rate 0.516 |
| // linux_x86_dbg bot at conceal rate 0.854 |
| if (delta_samples > 0) { |
| #if !defined(NDEBUG) |
| EXPECT_GT(0.95, 1.0 * delta_concealed / delta_samples) |
| << "Concealed " << delta_concealed << " of " << delta_samples |
| << " samples"; |
| #else |
| EXPECT_GT(0.6, 1.0 * delta_concealed / delta_samples) |
| << "Concealed " << delta_concealed << " of " << delta_samples |
| << " samples"; |
| #endif |
| } |
| // Increment trailing counters |
| audio_packets_stat_ = *rtp_stats->packets_received; |
| audio_delay_stat_ = *track_stats->relative_packet_arrival_delay; |
| audio_samples_stat_ = *track_stats->total_samples_received; |
| audio_concealed_stat_ = *track_stats->concealed_samples; |
| } |
| |
| private: |
| explicit PeerConnectionIntegrationWrapper(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, |
| bool reset_encoder_factory, |
| bool reset_decoder_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.task_queue_factory = |
| webrtc::CreateDefaultTaskQueueFactory(); |
| pc_factory_dependencies.trials = std::make_unique<FieldTrialBasedConfig>(); |
| cricket::MediaEngineDependencies media_deps; |
| media_deps.task_queue_factory = |
| pc_factory_dependencies.task_queue_factory.get(); |
| media_deps.adm = fake_audio_capture_module_; |
| webrtc::SetMediaEngineDefaults(&media_deps); |
| |
| if (reset_encoder_factory) { |
| media_deps.video_encoder_factory.reset(); |
| } |
| if (reset_decoder_factory) { |
| media_deps.video_decoder_factory.reset(); |
| } |
| |
| if (!media_deps.audio_processing) { |
| // If the standard Creation method for APM returns a null pointer, instead |
| // use the builder for testing to create an APM object. |
| media_deps.audio_processing = AudioProcessingBuilderForTesting().Create(); |
| } |
| |
| media_deps.trials = pc_factory_dependencies.trials.get(); |
| |
| pc_factory_dependencies.media_engine = |
| cricket::CreateMediaEngine(std::move(media_deps)); |
| 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 = |
| std::make_unique<webrtc::RtcEventLogFactory>( |
| pc_factory_dependencies.task_queue_factory.get()); |
| } |
| 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( |
| rtc::make_ref_counted<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> CreateAnswer() { |
| auto observer = |
| rtc::make_ref_counted<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) { |
| auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>(); |
| RTC_LOG(LS_INFO) << debug_name_ << ": SetLocalDescriptionAndSendSdpMessage"; |
| SdpType type = desc->GetType(); |
| std::string sdp; |
| EXPECT_TRUE(desc->ToString(&sdp)); |
| RTC_LOG(LS_INFO) << debug_name_ << ": local SDP contents=\n" << sdp; |
| pc()->SetLocalDescription(observer, desc.release()); |
| 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) { |
| auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>(); |
| RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription"; |
| pc()->SetRemoteDescription(observer, desc.release()); |
| 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() { |
| if (sdp_semantics_ != SdpSemantics::kUnifiedPlan) { |
| return; |
| } |
| auto transceivers = pc()->GetTransceivers(); |
| std::set<std::string> active_renderers; |
| for (auto& transceiver : transceivers) { |
| // Note - we don't check for direction here. This function is called |
| // before direction is set, and in that case, we should not remove |
| // the renderer. |
| if (transceiver->receiver()->media_type() == cricket::MEDIA_TYPE_VIDEO) { |
| active_renderers.insert(transceiver->receiver()->track()->id()); |
| } |
| } |
| for (auto it = fake_video_renderers_.begin(); |
| it != fake_video_renderers_.end();) { |
| // Remove fake video renderers belonging to any non-active transceivers. |
| if (!active_renderers.count(it->first)) { |
| it = fake_video_renderers_.erase(it); |
| } else { |
| 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 { |
| rtc::Thread::Current()->PostDelayedTask( |
| ToQueuedTask(task_safety_.flag(), |
| [this, type, msg] { |
| RelaySdpMessageIfReceiverExists(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 { |
| rtc::Thread::Current()->PostDelayedTask( |
| ToQueuedTask(task_safety_.flag(), |
| [this, sdp_mid, sdp_mline_index, msg] { |
| RelayIceMessageIfReceiverExists(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); |
| peer_connection_signaling_state_history_.push_back(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()] = |
| std::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()); |
| if (it != fake_video_renderers_.end()) { |
| fake_video_renderers_.erase(it); |
| } else { |
| RTC_LOG(LS_ERROR) << "OnRemoveTrack called for non-active renderer"; |
| } |
| } |
| } |
| 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 OnStandardizedIceConnectionChange( |
| webrtc::PeerConnectionInterface::IceConnectionState new_state) override { |
| standardized_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); |
| } |
| |
| void OnIceSelectedCandidatePairChanged( |
| const cricket::CandidatePairChangeEvent& event) { |
| ice_candidate_pair_change_history_.push_back(event); |
| } |
| |
| void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override { |
| RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate"; |
| |
| if (remote_async_resolver_) { |
| const auto& local_candidate = candidate->candidate(); |
| if (local_candidate.address().IsUnresolvedIP()) { |
| RTC_DCHECK(local_candidate.type() == cricket::LOCAL_PORT_TYPE); |
| rtc::SocketAddress resolved_addr(local_candidate.address()); |
| const auto resolved_ip = mdns_responder_->GetMappedAddressForName( |
| local_candidate.address().hostname()); |
| RTC_DCHECK(!resolved_ip.IsNil()); |
| resolved_addr.SetResolvedIP(resolved_ip); |
| EXPECT_CALL(*remote_async_resolver_, GetResolvedAddress(_, _)) |
| .WillOnce(DoAll(SetArgPointee<1>(resolved_addr), Return(true))); |
| EXPECT_CALL(*remote_async_resolver_, Destroy(_)); |
| } |
| } |
| |
| std::string ice_sdp; |
| EXPECT_TRUE(candidate->ToString(&ice_sdp)); |
| if (signaling_message_receiver_ == nullptr || !signal_ice_candidates_) { |
| // Remote party may be deleted. |
| return; |
| } |
| SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp); |
| last_candidate_gathered_ = candidate->candidate(); |
| } |
| void OnIceCandidateError(const std::string& address, |
| int port, |
| const std::string& url, |
| int error_code, |
| const std::string& error_text) override { |
| error_event_ = cricket::IceCandidateErrorEvent(address, port, url, |
| error_code, error_text); |
| } |
| 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_; |
| // Reference to the mDNS responder owned by |fake_network_manager_| after set. |
| webrtc::FakeMdnsResponder* mdns_responder_ = nullptr; |
| |
| 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; |
| cricket::Candidate last_candidate_gathered_; |
| cricket::IceCandidateErrorEvent error_event_; |
| |
| // 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_; |
| rtc::MockAsyncResolver* remote_async_resolver_ = nullptr; |
| 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::IceConnectionState> |
| standardized_ice_connection_state_history_; |
| std::vector<PeerConnectionInterface::PeerConnectionState> |
| peer_connection_state_history_; |
| std::vector<PeerConnectionInterface::IceGatheringState> |
| ice_gathering_state_history_; |
| std::vector<cricket::CandidatePairChangeEvent> |
| ice_candidate_pair_change_history_; |
| std::vector<PeerConnectionInterface::SignalingState> |
| peer_connection_signaling_state_history_; |
| webrtc::FakeRtcEventLogFactory* event_log_factory_; |
| |
| // Variables for tracking delay stats on an audio track |
| int audio_packets_stat_ = 0; |
| double audio_delay_stat_ = 0.0; |
| uint64_t audio_samples_stat_ = 0; |
| uint64_t audio_concealed_stat_ = 0; |
| std::string rtp_stats_id_; |
| std::string audio_track_stats_id_; |
| |
| ScopedTaskSafety task_safety_; |
| |
| friend class PeerConnectionIntegrationBaseTest; |
| }; |
| |
| class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput { |
| public: |
| virtual ~MockRtcEventLogOutput() = default; |
| MOCK_METHOD(bool, IsActive, (), (const, override)); |
| MOCK_METHOD(bool, Write, (const std::string&), (override)); |
| }; |
| |
| // 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; |
| }; |
| |
| class MockIceTransport : public webrtc::IceTransportInterface { |
| public: |
| MockIceTransport(const std::string& name, int component) |
| : internal_(std::make_unique<cricket::FakeIceTransport>( |
| name, |
| component, |
| nullptr /* network_thread */)) {} |
| ~MockIceTransport() = default; |
| cricket::IceTransportInternal* internal() { return internal_.get(); } |
| |
| private: |
| std::unique_ptr<cricket::FakeIceTransport> internal_; |
| }; |
| |
| class MockIceTransportFactory : public IceTransportFactory { |
| public: |
| ~MockIceTransportFactory() override = default; |
| rtc::scoped_refptr<IceTransportInterface> CreateIceTransport( |
| const std::string& transport_name, |
| int component, |
| IceTransportInit init) { |
| RecordIceTransportCreated(); |
| return rtc::make_ref_counted<MockIceTransport>(transport_name, component); |
| } |
| MOCK_METHOD(void, RecordIceTransportCreated, ()); |
| }; |
| |
| // 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: |
| PeerConnectionIntegrationBaseTest( |
| SdpSemantics sdp_semantics, |
| absl::optional<std::string> field_trials = absl::nullopt) |
| : 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()), |
| field_trials_(field_trials.has_value() |
| ? new test::ScopedFieldTrials(*field_trials) |
| : nullptr) { |
| 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 be 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); |
| caller_->pc()->Close(); |
| delete SetCallerPcWrapperAndReturnCurrent(nullptr); |
| } |
| if (callee_) { |
| callee_->set_signaling_message_receiver(nullptr); |
| callee_->pc()->Close(); |
| 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<PeerConnectionIntegrationWrapper> CreatePeerConnectionWrapper( |
| const std::string& debug_name, |
| const PeerConnectionFactory::Options* options, |
| const RTCConfiguration* config, |
| webrtc::PeerConnectionDependencies dependencies, |
| std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory, |
| bool reset_encoder_factory, |
| bool reset_decoder_factory) { |
| RTCConfiguration modified_config; |
| if (config) { |
| modified_config = *config; |
| } |
| modified_config.sdp_semantics = sdp_semantics_; |
| if (!dependencies.cert_generator) { |
| dependencies.cert_generator = |
| std::make_unique<FakeRTCCertificateGenerator>(); |
| } |
| std::unique_ptr<PeerConnectionIntegrationWrapper> client( |
| new PeerConnectionIntegrationWrapper(debug_name)); |
| |
| if (!client->Init(options, &modified_config, std::move(dependencies), |
| network_thread_.get(), worker_thread_.get(), |
| std::move(event_log_factory), reset_encoder_factory, |
| reset_decoder_factory)) { |
| return nullptr; |
| } |
| return client; |
| } |
| |
| std::unique_ptr<PeerConnectionIntegrationWrapper> |
| CreatePeerConnectionWrapperWithFakeRtcEventLog( |
| const std::string& debug_name, |
| const PeerConnectionFactory::Options* options, |
| const RTCConfiguration* config, |
| webrtc::PeerConnectionDependencies dependencies) { |
| return CreatePeerConnectionWrapper( |
| debug_name, options, config, std::move(dependencies), |
| std::make_unique<webrtc::FakeRtcEventLogFactory>(), |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| } |
| |
| 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, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| sdp_semantics_ = callee_semantics; |
| callee_ = CreatePeerConnectionWrapper( |
| "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), |
| nullptr, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| 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, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| callee_ = CreatePeerConnectionWrapper( |
| "Callee", nullptr, &callee_config, |
| webrtc::PeerConnectionDependencies(nullptr), nullptr, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| 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, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| callee_ = |
| CreatePeerConnectionWrapper("Callee", nullptr, &callee_config, |
| std::move(callee_dependencies), nullptr, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| 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, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| callee_ = CreatePeerConnectionWrapper( |
| "Callee", &callee_options, nullptr, |
| webrtc::PeerConnectionDependencies(nullptr), nullptr, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| 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<PeerConnectionIntegrationWrapper> |
| 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, |
| /*reset_encoder_factory=*/false, |
| /*reset_decoder_factory=*/false); |
| } |
| |
| bool CreateOneDirectionalPeerConnectionWrappers(bool caller_to_callee) { |
| caller_ = CreatePeerConnectionWrapper( |
| "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), |
| nullptr, |
| /*reset_encoder_factory=*/!caller_to_callee, |
| /*reset_decoder_factory=*/caller_to_callee); |
| callee_ = CreatePeerConnectionWrapper( |
| "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr), |
| nullptr, |
| /*reset_encoder_factory=*/caller_to_callee, |
| /*reset_decoder_factory=*/!caller_to_callee); |
| return caller_ && callee_; |
| } |
| |
| 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 std::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 std::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(); } |
| |
| PeerConnectionIntegrationWrapper* caller() { return caller_.get(); } |
| |
| // Set the |caller_| to the |wrapper| passed in and return the |
| // original |caller_|. |
| PeerConnectionIntegrationWrapper* SetCallerPcWrapperAndReturnCurrent( |
| PeerConnectionIntegrationWrapper* wrapper) { |
| PeerConnectionIntegrationWrapper* old = caller_.release(); |
| caller_.reset(wrapper); |
| return old; |
| } |
| |
| PeerConnectionIntegrationWrapper* callee() { return callee_.get(); } |
| |
| // Set the |callee_| to the |wrapper| passed in and return the |
| // original |callee_|. |
| PeerConnectionIntegrationWrapper* SetCalleePcWrapperAndReturnCurrent( |
| PeerConnectionIntegrationWrapper* wrapper) { |
| PeerConnectionIntegrationWrapper* old = callee_.release(); |
| callee_.reset(wrapper); |
| return old; |
| } |
| |
| void SetPortAllocatorFlags(uint32_t caller_flags, uint32_t callee_flags) { |
| network_thread()->Invoke<void>(RTC_FROM_HERE, [this, caller_flags] { |
| caller()->port_allocator()->set_flags(caller_flags); |
| }); |
| network_thread()->Invoke<void>(RTC_FROM_HERE, [this, callee_flags] { |
| callee()->port_allocator()->set_flags(callee_flags); |
| }); |
| } |
| |
| 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) { |
| // Make sure there are no bogus tracks confusing the issue. |
| caller()->RemoveUnusedVideoRenderers(); |
| callee()->RemoveUnusedVideoRenderers(); |
| // 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 ClosePeerConnections() { |
| if (caller()) |
| caller()->pc()->Close(); |
| if (callee()) |
| callee()->pc()->Close(); |
| } |
| |
| 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_METRIC_EQ(1, webrtc::metrics::NumEvents( |
| "WebRTC.PeerConnection.SrtpCryptoSuite.Audio", |
| expected_cipher_suite)); |
| } |
| |
| void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled, |
| bool remote_gcm_enabled, |
| bool aes_ctr_enabled, |
| int expected_cipher_suite) { |
| PeerConnectionFactory::Options caller_options; |
| caller_options.crypto_options.srtp.enable_gcm_crypto_suites = |
| local_gcm_enabled; |
| caller_options.crypto_options.srtp.enable_aes128_sha1_80_crypto_cipher = |
| aes_ctr_enabled; |
| PeerConnectionFactory::Options callee_options; |
| callee_options.crypto_options.srtp.enable_gcm_crypto_suites = |
| remote_gcm_enabled; |
| callee_options.crypto_options.srtp.enable_aes128_sha1_80_crypto_cipher = |
| aes_ctr_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_; |
| std::unique_ptr<PeerConnectionIntegrationWrapper> caller_; |
| std::unique_ptr<PeerConnectionIntegrationWrapper> callee_; |
| std::unique_ptr<test::ScopedFieldTrials> field_trials_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_TEST_INTEGRATION_TEST_HELPERS_H_ |