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1 /* | |
2 * libjingle | |
3 * Copyright 2013 Google Inc. | |
4 * | |
5 * Redistribution and use in source and binary forms, with or without | |
6 * modification, are permitted provided that the following conditions are met: | |
7 * | |
8 * 1. Redistributions of source code must retain the above copyright notice, | |
9 * this list of conditions and the following disclaimer. | |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | |
11 * this list of conditions and the following disclaimer in the documentation | |
12 * and/or other materials provided with the distribution. | |
13 * 3. The name of the author may not be used to endorse or promote products | |
14 * derived from this software without specific prior written permission. | |
15 * | |
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED | |
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF | |
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO | |
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, | |
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, | |
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | |
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | |
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | |
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | |
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | |
26 */ | |
27 | |
28 #ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | |
29 #define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | |
30 | |
31 #include "talk/app/webrtc/peerconnectioninterface.h" | |
32 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" | |
33 #include "talk/app/webrtc/test/fakeconstraints.h" | |
34 #include "talk/app/webrtc/test/fakevideotrackrenderer.h" | |
35 #include "webrtc/base/sigslot.h" | |
36 | |
37 class PeerConnectionTestWrapper | |
38 : public webrtc::PeerConnectionObserver, | |
39 public webrtc::CreateSessionDescriptionObserver, | |
40 public sigslot::has_slots<> { | |
41 public: | |
42 static void Connect(PeerConnectionTestWrapper* caller, | |
43 PeerConnectionTestWrapper* callee); | |
44 | |
45 explicit PeerConnectionTestWrapper(const std::string& name); | |
46 virtual ~PeerConnectionTestWrapper(); | |
47 | |
48 bool CreatePc(const webrtc::MediaConstraintsInterface* constraints); | |
49 | |
50 rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel( | |
51 const std::string& label, | |
52 const webrtc::DataChannelInit& init); | |
53 | |
54 // Implements PeerConnectionObserver. | |
55 virtual void OnSignalingChange( | |
56 webrtc::PeerConnectionInterface::SignalingState new_state) {} | |
57 virtual void OnStateChange( | |
58 webrtc::PeerConnectionObserver::StateType state_changed) {} | |
59 virtual void OnAddStream(webrtc::MediaStreamInterface* stream); | |
60 virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream) {} | |
61 virtual void OnDataChannel(webrtc::DataChannelInterface* data_channel); | |
62 virtual void OnRenegotiationNeeded() {} | |
63 virtual void OnIceConnectionChange( | |
64 webrtc::PeerConnectionInterface::IceConnectionState new_state) {} | |
65 virtual void OnIceGatheringChange( | |
66 webrtc::PeerConnectionInterface::IceGatheringState new_state) {} | |
67 virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate); | |
68 virtual void OnIceComplete() {} | |
69 | |
70 // Implements CreateSessionDescriptionObserver. | |
71 virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc); | |
72 virtual void OnFailure(const std::string& error) {} | |
73 | |
74 void CreateOffer(const webrtc::MediaConstraintsInterface* constraints); | |
75 void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints); | |
76 void ReceiveOfferSdp(const std::string& sdp); | |
77 void ReceiveAnswerSdp(const std::string& sdp); | |
78 void AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index, | |
79 const std::string& candidate); | |
80 void WaitForCallEstablished(); | |
81 void WaitForConnection(); | |
82 void WaitForAudio(); | |
83 void WaitForVideo(); | |
84 void GetAndAddUserMedia( | |
85 bool audio, const webrtc::FakeConstraints& audio_constraints, | |
86 bool video, const webrtc::FakeConstraints& video_constraints); | |
87 | |
88 // sigslots | |
89 sigslot::signal1<std::string*> SignalOnIceCandidateCreated; | |
90 sigslot::signal3<const std::string&, | |
91 int, | |
92 const std::string&> SignalOnIceCandidateReady; | |
93 sigslot::signal1<std::string*> SignalOnSdpCreated; | |
94 sigslot::signal1<const std::string&> SignalOnSdpReady; | |
95 sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel; | |
96 | |
97 private: | |
98 void SetLocalDescription(const std::string& type, const std::string& sdp); | |
99 void SetRemoteDescription(const std::string& type, const std::string& sdp); | |
100 bool CheckForConnection(); | |
101 bool CheckForAudio(); | |
102 bool CheckForVideo(); | |
103 rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( | |
104 bool audio, const webrtc::FakeConstraints& audio_constraints, | |
105 bool video, const webrtc::FakeConstraints& video_constraints); | |
106 | |
107 std::string name_; | |
108 rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; | |
109 rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> | |
110 peer_connection_factory_; | |
111 rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; | |
112 rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_; | |
113 }; | |
114 | |
115 #endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ | |
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