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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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34 | 34 |
35 #include "talk/app/webrtc/dtmfsender.h" | 35 #include "talk/app/webrtc/dtmfsender.h" |
36 #include "talk/app/webrtc/fakemetricsobserver.h" | 36 #include "talk/app/webrtc/fakemetricsobserver.h" |
37 #include "talk/app/webrtc/fakeportallocatorfactory.h" | 37 #include "talk/app/webrtc/fakeportallocatorfactory.h" |
38 #include "talk/app/webrtc/localaudiosource.h" | 38 #include "talk/app/webrtc/localaudiosource.h" |
39 #include "talk/app/webrtc/mediastreaminterface.h" | 39 #include "talk/app/webrtc/mediastreaminterface.h" |
40 #include "talk/app/webrtc/peerconnectionfactory.h" | 40 #include "talk/app/webrtc/peerconnectionfactory.h" |
41 #include "talk/app/webrtc/peerconnectioninterface.h" | 41 #include "talk/app/webrtc/peerconnectioninterface.h" |
42 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" | 42 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" |
43 #include "talk/app/webrtc/test/fakeconstraints.h" | 43 #include "talk/app/webrtc/test/fakeconstraints.h" |
44 #include "talk/app/webrtc/test/fakedtlsidentityservice.h" | 44 #include "talk/app/webrtc/test/fakedtlsidentitystore.h" |
45 #include "talk/app/webrtc/test/fakeperiodicvideocapturer.h" | 45 #include "talk/app/webrtc/test/fakeperiodicvideocapturer.h" |
46 #include "talk/app/webrtc/test/fakevideotrackrenderer.h" | 46 #include "talk/app/webrtc/test/fakevideotrackrenderer.h" |
47 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h" | 47 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h" |
48 #include "talk/app/webrtc/videosourceinterface.h" | 48 #include "talk/app/webrtc/videosourceinterface.h" |
49 #include "talk/media/webrtc/fakewebrtcvideoengine.h" | 49 #include "talk/media/webrtc/fakewebrtcvideoengine.h" |
50 #include "talk/session/media/mediasession.h" | 50 #include "talk/session/media/mediasession.h" |
51 #include "webrtc/base/gunit.h" | 51 #include "webrtc/base/gunit.h" |
52 #include "webrtc/base/physicalsocketserver.h" | 52 #include "webrtc/base/physicalsocketserver.h" |
53 #include "webrtc/base/scoped_ptr.h" | 53 #include "webrtc/base/scoped_ptr.h" |
54 #include "webrtc/base/ssladapter.h" | 54 #include "webrtc/base/ssladapter.h" |
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773 } | 773 } |
774 | 774 |
775 protected: | 775 protected: |
776 explicit JsepTestClient(const std::string& id) | 776 explicit JsepTestClient(const std::string& id) |
777 : PeerConnectionTestClientBase<JsepMessageReceiver>(id), | 777 : PeerConnectionTestClientBase<JsepMessageReceiver>(id), |
778 remove_msid_(false), | 778 remove_msid_(false), |
779 remove_bundle_(false), | 779 remove_bundle_(false), |
780 remove_sdes_(false) { | 780 remove_sdes_(false) { |
781 } | 781 } |
782 | 782 |
783 virtual rtc::scoped_refptr<webrtc::PeerConnectionInterface> | 783 rtc::scoped_refptr<webrtc::PeerConnectionInterface> |
784 CreatePeerConnection(webrtc::PortAllocatorFactoryInterface* factory, | 784 CreatePeerConnection( |
785 const MediaConstraintsInterface* constraints) { | 785 webrtc::PortAllocatorFactoryInterface* factory, |
| 786 const MediaConstraintsInterface* constraints) override { |
786 // CreatePeerConnection with IceServers. | 787 // CreatePeerConnection with IceServers. |
787 webrtc::PeerConnectionInterface::IceServers ice_servers; | 788 webrtc::PeerConnectionInterface::IceServers ice_servers; |
788 webrtc::PeerConnectionInterface::IceServer ice_server; | 789 webrtc::PeerConnectionInterface::IceServer ice_server; |
789 ice_server.uri = "stun:stun.l.google.com:19302"; | 790 ice_server.uri = "stun:stun.l.google.com:19302"; |
790 ice_servers.push_back(ice_server); | 791 ice_servers.push_back(ice_server); |
791 | 792 |
792 FakeIdentityService* dtls_service = | 793 rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store( |
793 rtc::SSLStreamAdapter::HaveDtlsSrtp() ? | 794 rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore() |
794 new FakeIdentityService() : NULL; | 795 : nullptr); |
795 return peer_connection_factory()->CreatePeerConnection( | 796 return peer_connection_factory()->CreatePeerConnection( |
796 ice_servers, constraints, factory, dtls_service, this); | 797 ice_servers, constraints, factory, dtls_identity_store.Pass(), this); |
797 } | 798 } |
798 | 799 |
799 void HandleIncomingOffer(const std::string& msg) { | 800 void HandleIncomingOffer(const std::string& msg) { |
800 LOG(INFO) << id() << "HandleIncomingOffer "; | 801 LOG(INFO) << id() << "HandleIncomingOffer "; |
801 if (NumberOfLocalMediaStreams() == 0) { | 802 if (NumberOfLocalMediaStreams() == 0) { |
802 // If we are not sending any streams ourselves it is time to add some. | 803 // If we are not sending any streams ourselves it is time to add some. |
803 AddMediaStream(true, true); | 804 AddMediaStream(true, true); |
804 } | 805 } |
805 rtc::scoped_ptr<SessionDescriptionInterface> desc( | 806 rtc::scoped_ptr<SessionDescriptionInterface> desc( |
806 webrtc::CreateSessionDescription("offer", msg, NULL)); | 807 webrtc::CreateSessionDescription("offer", msg, NULL)); |
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1620 // TODO(holmer): Disabled due to sometimes crashing on buildbots. | 1621 // TODO(holmer): Disabled due to sometimes crashing on buildbots. |
1621 // See issue webrtc/2378. | 1622 // See issue webrtc/2378. |
1622 TEST_F(JsepPeerConnectionP2PTestClient, | 1623 TEST_F(JsepPeerConnectionP2PTestClient, |
1623 DISABLED_LocalP2PTestWithVideoDecoderFactory) { | 1624 DISABLED_LocalP2PTestWithVideoDecoderFactory) { |
1624 ASSERT_TRUE(CreateTestClients()); | 1625 ASSERT_TRUE(CreateTestClients()); |
1625 EnableVideoDecoderFactory(); | 1626 EnableVideoDecoderFactory(); |
1626 LocalP2PTest(); | 1627 LocalP2PTest(); |
1627 } | 1628 } |
1628 | 1629 |
1629 #endif // if !defined(THREAD_SANITIZER) | 1630 #endif // if !defined(THREAD_SANITIZER) |
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