| /* |
| * Copyright 2013 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 WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
| #define WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |
| |
| #include <memory> |
| |
| #include "webrtc/api/peerconnectioninterface.h" |
| #include "webrtc/api/test/fakeconstraints.h" |
| #include "webrtc/pc/test/fakeaudiocapturemodule.h" |
| #include "webrtc/pc/test/fakevideotrackrenderer.h" |
| #include "webrtc/rtc_base/sigslot.h" |
| |
| class PeerConnectionTestWrapper |
| : public webrtc::PeerConnectionObserver, |
| public webrtc::CreateSessionDescriptionObserver, |
| public sigslot::has_slots<> { |
| public: |
| static void Connect(PeerConnectionTestWrapper* caller, |
| PeerConnectionTestWrapper* callee); |
| |
| PeerConnectionTestWrapper(const std::string& name, |
| rtc::Thread* network_thread, |
| rtc::Thread* worker_thread); |
| virtual ~PeerConnectionTestWrapper(); |
| |
| bool CreatePc( |
| const webrtc::MediaConstraintsInterface* constraints, |
| const webrtc::PeerConnectionInterface::RTCConfiguration& config, |
| rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, |
| rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory); |
| |
| webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); } |
| |
| rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel( |
| const std::string& label, |
| const webrtc::DataChannelInit& init); |
| |
| // Implements PeerConnectionObserver. |
| void OnSignalingChange( |
| webrtc::PeerConnectionInterface::SignalingState new_state) override {} |
| void OnAddStream( |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override; |
| void OnRemoveStream( |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override {} |
| void OnDataChannel( |
| rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override ; |
| void OnRenegotiationNeeded() override {} |
| void OnIceConnectionChange( |
| webrtc::PeerConnectionInterface::IceConnectionState new_state) override {} |
| void OnIceGatheringChange( |
| webrtc::PeerConnectionInterface::IceGatheringState new_state) override {} |
| void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; |
| |
| // Implements CreateSessionDescriptionObserver. |
| void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; |
| void OnFailure(const std::string& error) override {} |
| |
| void CreateOffer(const webrtc::MediaConstraintsInterface* constraints); |
| void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints); |
| void ReceiveOfferSdp(const std::string& sdp); |
| void ReceiveAnswerSdp(const std::string& sdp); |
| void AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index, |
| const std::string& candidate); |
| void WaitForCallEstablished(); |
| void WaitForConnection(); |
| void WaitForAudio(); |
| void WaitForVideo(); |
| void GetAndAddUserMedia( |
| bool audio, const webrtc::FakeConstraints& audio_constraints, |
| bool video, const webrtc::FakeConstraints& video_constraints); |
| |
| // sigslots |
| sigslot::signal1<std::string*> SignalOnIceCandidateCreated; |
| sigslot::signal3<const std::string&, |
| int, |
| const std::string&> SignalOnIceCandidateReady; |
| sigslot::signal1<std::string*> SignalOnSdpCreated; |
| sigslot::signal1<const std::string&> SignalOnSdpReady; |
| sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel; |
| |
| private: |
| void SetLocalDescription(const std::string& type, const std::string& sdp); |
| void SetRemoteDescription(const std::string& type, const std::string& sdp); |
| bool CheckForConnection(); |
| bool CheckForAudio(); |
| bool CheckForVideo(); |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( |
| bool audio, const webrtc::FakeConstraints& audio_constraints, |
| bool video, const webrtc::FakeConstraints& video_constraints); |
| |
| std::string name_; |
| rtc::Thread* const network_thread_; |
| rtc::Thread* const worker_thread_; |
| rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; |
| rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> |
| peer_connection_factory_; |
| rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; |
| std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_; |
| }; |
| |
| #endif // WEBRTC_PC_TEST_PEERCONNECTIONTESTWRAPPER_H_ |