1 /* 2 * Copyright 2013 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_ 12 #define PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_ 13 14 #include <memory> 15 #include <string> 16 #include <vector> 17 18 #include "api/audio_codecs/audio_decoder_factory.h" 19 #include "api/audio_codecs/audio_encoder_factory.h" 20 #include "api/audio_options.h" 21 #include "api/data_channel_interface.h" 22 #include "api/jsep.h" 23 #include "api/media_stream_interface.h" 24 #include "api/peer_connection_interface.h" 25 #include "api/rtc_error.h" 26 #include "api/rtp_receiver_interface.h" 27 #include "api/scoped_refptr.h" 28 #include "api/sequence_checker.h" 29 #include "pc/test/fake_audio_capture_module.h" 30 #include "pc/test/fake_video_track_renderer.h" 31 #include "rtc_base/third_party/sigslot/sigslot.h" 32 #include "rtc_base/thread.h" 33 34 class PeerConnectionTestWrapper 35 : public webrtc::PeerConnectionObserver, 36 public webrtc::CreateSessionDescriptionObserver, 37 public sigslot::has_slots<> { 38 public: 39 static void Connect(PeerConnectionTestWrapper* caller, 40 PeerConnectionTestWrapper* callee); 41 42 PeerConnectionTestWrapper(const std::string& name, 43 rtc::SocketServer* socket_server, 44 rtc::Thread* network_thread, 45 rtc::Thread* worker_thread); 46 virtual ~PeerConnectionTestWrapper(); 47 48 bool CreatePc( 49 const webrtc::PeerConnectionInterface::RTCConfiguration& config, 50 rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, 51 rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory); 52 pc_factory()53 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory() 54 const { 55 return peer_connection_factory_; 56 } pc()57 webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); } 58 59 rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel( 60 const std::string& label, 61 const webrtc::DataChannelInit& init); 62 63 void WaitForNegotiation(); 64 65 // Implements PeerConnectionObserver. 66 void OnSignalingChange( 67 webrtc::PeerConnectionInterface::SignalingState new_state) override; 68 void OnAddTrack( 69 rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, 70 const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& 71 streams) override; 72 void OnDataChannel( 73 rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override; OnRenegotiationNeeded()74 void OnRenegotiationNeeded() override {} OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState new_state)75 void OnIceConnectionChange( 76 webrtc::PeerConnectionInterface::IceConnectionState new_state) override {} OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state)77 void OnIceGatheringChange( 78 webrtc::PeerConnectionInterface::IceGatheringState new_state) override {} 79 void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; 80 81 // Implements CreateSessionDescriptionObserver. 82 void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; OnFailure(webrtc::RTCError)83 void OnFailure(webrtc::RTCError) override {} 84 85 void CreateOffer( 86 const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options); 87 void CreateAnswer( 88 const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options); 89 void ReceiveOfferSdp(const std::string& sdp); 90 void ReceiveAnswerSdp(const std::string& sdp); 91 void AddIceCandidate(const std::string& sdp_mid, 92 int sdp_mline_index, 93 const std::string& candidate); 94 void WaitForCallEstablished(); 95 void WaitForConnection(); 96 void WaitForAudio(); 97 void WaitForVideo(); 98 void GetAndAddUserMedia(bool audio, 99 const cricket::AudioOptions& audio_options, 100 bool video); 101 102 // sigslots 103 sigslot::signal3<const std::string&, int, const std::string&> 104 SignalOnIceCandidateReady; 105 sigslot::signal1<const std::string&> SignalOnSdpReady; 106 sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel; 107 108 private: 109 void SetLocalDescription(webrtc::SdpType type, const std::string& sdp); 110 void SetRemoteDescription(webrtc::SdpType type, const std::string& sdp); 111 bool CheckForConnection(); 112 bool CheckForAudio(); 113 bool CheckForVideo(); 114 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( 115 bool audio, 116 const cricket::AudioOptions& audio_options, 117 bool video); 118 119 std::string name_; 120 rtc::SocketServer* const socket_server_; 121 rtc::Thread* const network_thread_; 122 rtc::Thread* const worker_thread_; 123 webrtc::SequenceChecker pc_thread_checker_; 124 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; 125 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> 126 peer_connection_factory_; 127 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; 128 std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_; 129 int num_get_user_media_calls_ = 0; 130 bool pending_negotiation_; 131 }; 132 133 #endif // PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_ 134