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| 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 WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ | |
| 12 #define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ | |
| 13 | |
| 14 #include <memory> | |
| 15 | |
| 16 #include "webrtc/api/peerconnectioninterface.h" | |
| 17 #include "webrtc/api/test/fakeaudiocapturemodule.h" | |
| 18 #include "webrtc/api/test/fakeconstraints.h" | |
| 19 #include "webrtc/api/test/fakevideotrackrenderer.h" | |
| 20 #include "webrtc/base/sigslot.h" | |
| 21 | |
| 22 class PeerConnectionTestWrapper | |
| 23 : public webrtc::PeerConnectionObserver, | |
| 24 public webrtc::CreateSessionDescriptionObserver, | |
| 25 public sigslot::has_slots<> { | |
| 26 public: | |
| 27 // We need these using declarations because there are two versions of each of | |
| 28 // the below methods and we only override one of them. | |
| 29 // TODO(deadbeef): Remove once there's only one version of the methods. | |
| 30 using PeerConnectionObserver::OnAddStream; | |
| 31 using PeerConnectionObserver::OnRemoveStream; | |
| 32 using PeerConnectionObserver::OnDataChannel; | |
| 33 | |
| 34 static void Connect(PeerConnectionTestWrapper* caller, | |
| 35 PeerConnectionTestWrapper* callee); | |
| 36 | |
| 37 PeerConnectionTestWrapper(const std::string& name, | |
| 38 rtc::Thread* network_thread, | |
| 39 rtc::Thread* worker_thread); | |
| 40 virtual ~PeerConnectionTestWrapper(); | |
| 41 | |
| 42 bool CreatePc( | |
| 43 const webrtc::MediaConstraintsInterface* constraints, | |
| 44 const webrtc::PeerConnectionInterface::RTCConfiguration& config); | |
| 45 | |
| 46 webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); } | |
| 47 | |
| 48 rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel( | |
| 49 const std::string& label, | |
| 50 const webrtc::DataChannelInit& init); | |
| 51 | |
| 52 // Implements PeerConnectionObserver. | |
| 53 virtual void OnSignalingChange( | |
| 54 webrtc::PeerConnectionInterface::SignalingState new_state) {} | |
| 55 virtual void OnStateChange( | |
| 56 webrtc::PeerConnectionObserver::StateType state_changed) {} | |
| 57 virtual void OnAddStream( | |
| 58 rtc::scoped_refptr<webrtc::MediaStreamInterface> stream); | |
| 59 virtual void OnRemoveStream( | |
| 60 rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {} | |
| 61 virtual void OnDataChannel( | |
| 62 rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel); | |
| 63 virtual void OnRenegotiationNeeded() {} | |
| 64 virtual void OnIceConnectionChange( | |
| 65 webrtc::PeerConnectionInterface::IceConnectionState new_state) {} | |
| 66 virtual void OnIceGatheringChange( | |
| 67 webrtc::PeerConnectionInterface::IceGatheringState new_state) {} | |
| 68 virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate); | |
| 69 virtual void OnIceComplete() {} | |
| 70 | |
| 71 // Implements CreateSessionDescriptionObserver. | |
| 72 virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc); | |
| 73 virtual void OnFailure(const std::string& error) {} | |
| 74 | |
| 75 void CreateOffer(const webrtc::MediaConstraintsInterface* constraints); | |
| 76 void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints); | |
| 77 void ReceiveOfferSdp(const std::string& sdp); | |
| 78 void ReceiveAnswerSdp(const std::string& sdp); | |
| 79 void AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index, | |
| 80 const std::string& candidate); | |
| 81 void WaitForCallEstablished(); | |
| 82 void WaitForConnection(); | |
| 83 void WaitForAudio(); | |
| 84 void WaitForVideo(); | |
| 85 void GetAndAddUserMedia( | |
| 86 bool audio, const webrtc::FakeConstraints& audio_constraints, | |
| 87 bool video, const webrtc::FakeConstraints& video_constraints); | |
| 88 | |
| 89 // sigslots | |
| 90 sigslot::signal1<std::string*> SignalOnIceCandidateCreated; | |
| 91 sigslot::signal3<const std::string&, | |
| 92 int, | |
| 93 const std::string&> SignalOnIceCandidateReady; | |
| 94 sigslot::signal1<std::string*> SignalOnSdpCreated; | |
| 95 sigslot::signal1<const std::string&> SignalOnSdpReady; | |
| 96 sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel; | |
| 97 | |
| 98 private: | |
| 99 void SetLocalDescription(const std::string& type, const std::string& sdp); | |
| 100 void SetRemoteDescription(const std::string& type, const std::string& sdp); | |
| 101 bool CheckForConnection(); | |
| 102 bool CheckForAudio(); | |
| 103 bool CheckForVideo(); | |
| 104 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( | |
| 105 bool audio, const webrtc::FakeConstraints& audio_constraints, | |
| 106 bool video, const webrtc::FakeConstraints& video_constraints); | |
| 107 | |
| 108 std::string name_; | |
| 109 rtc::Thread* const network_thread_; | |
| 110 rtc::Thread* const worker_thread_; | |
| 111 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | |
| 112 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | |
| 113 peer_connection_factory_; | |
| 114 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | |
| 115 std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_; | |
| 116 }; | |
| 117 | |
| 118 #endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ | |
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Chromium Code Reviews
Issue 2514883002:
Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
