diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc index c250eeab54..0abef38e55 100644 --- a/talk/app/webrtc/peerconnection_unittest.cc +++ b/talk/app/webrtc/peerconnection_unittest.cc @@ -46,14 +46,16 @@ #include "talk/app/webrtc/test/mockpeerconnectionobservers.h" #include "talk/app/webrtc/videosourceinterface.h" #include "talk/media/webrtc/fakewebrtcvideoengine.h" -#include "webrtc/p2p/base/constants.h" -#include "webrtc/p2p/base/sessiondescription.h" #include "talk/session/media/mediasession.h" #include "webrtc/base/gunit.h" +#include "webrtc/base/physicalsocketserver.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/ssladapter.h" #include "webrtc/base/sslstreamadapter.h" #include "webrtc/base/thread.h" +#include "webrtc/base/virtualsocketserver.h" +#include "webrtc/p2p/base/constants.h" +#include "webrtc/p2p/base/sessiondescription.h" #define MAYBE_SKIP_TEST(feature) \ if (!(feature())) { \ @@ -869,10 +871,16 @@ class JsepTestClient template class P2PTestConductor : public testing::Test { public: + P2PTestConductor() + : pss_(new rtc::PhysicalSocketServer), + ss_(new rtc::VirtualSocketServer(pss_.get())), + ss_scope_(ss_.get()) {} + bool SessionActive() { return initiating_client_->SessionActive() && - receiving_client_->SessionActive(); + receiving_client_->SessionActive(); } + // Return true if the number of frames provided have been received or it is // known that that will never occur (e.g. no frames will be sent or // captured). @@ -1058,6 +1066,9 @@ class P2PTestConductor : public testing::Test { SignalingClass* receiving_client() { return receiving_client_.get(); } private: + rtc::scoped_ptr pss_; + rtc::scoped_ptr ss_; + rtc::SocketServerScope ss_scope_; rtc::scoped_ptr initiating_client_; rtc::scoped_ptr receiving_client_; };