364f204d16
TBR=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/4119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d
120 lines
5.2 KiB
C++
120 lines
5.2 KiB
C++
/*
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* libjingle
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* Copyright 2013, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
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#define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
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#include "talk/app/webrtc/peerconnectioninterface.h"
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#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
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#include "talk/app/webrtc/test/fakeconstraints.h"
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#include "talk/app/webrtc/test/fakevideotrackrenderer.h"
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#include "talk/base/sigslot.h"
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#include "talk/base/thread.h"
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namespace webrtc {
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class PortAllocatorFactoryInterface;
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}
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class PeerConnectionTestWrapper
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: public webrtc::PeerConnectionObserver,
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public webrtc::CreateSessionDescriptionObserver,
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public sigslot::has_slots<> {
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public:
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static void Connect(PeerConnectionTestWrapper* caller,
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PeerConnectionTestWrapper* callee);
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explicit PeerConnectionTestWrapper(const std::string& name);
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virtual ~PeerConnectionTestWrapper();
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bool CreatePc(const webrtc::MediaConstraintsInterface* constraints);
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// Implements PeerConnectionObserver.
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virtual void OnError() {}
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virtual void OnSignalingChange(
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webrtc::PeerConnectionInterface::SignalingState new_state) {}
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virtual void OnStateChange(
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webrtc::PeerConnectionObserver::StateType state_changed) {}
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virtual void OnAddStream(webrtc::MediaStreamInterface* stream);
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virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream) {}
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virtual void OnDataChannel(webrtc::DataChannelInterface* data_channel) {}
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virtual void OnRenegotiationNeeded() {}
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virtual void OnIceConnectionChange(
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webrtc::PeerConnectionInterface::IceConnectionState new_state) {}
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virtual void OnIceGatheringChange(
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webrtc::PeerConnectionInterface::IceGatheringState new_state) {}
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virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate);
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virtual void OnIceComplete() {}
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// Implements CreateSessionDescriptionObserver.
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virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc);
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virtual void OnFailure(const std::string& error) {}
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void CreateOffer(const webrtc::MediaConstraintsInterface* constraints);
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void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints);
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void ReceiveOfferSdp(const std::string& sdp);
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void ReceiveAnswerSdp(const std::string& sdp);
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void AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
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const std::string& candidate);
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void WaitForCallEstablished();
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void WaitForConnection();
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void WaitForAudio();
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void WaitForVideo();
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void GetAndAddUserMedia(
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bool audio, const webrtc::FakeConstraints& audio_constraints,
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bool video, const webrtc::FakeConstraints& video_constraints);
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// sigslots
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sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
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sigslot::signal3<const std::string&,
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int,
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const std::string&> SignalOnIceCandidateReady;
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sigslot::signal1<std::string*> SignalOnSdpCreated;
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sigslot::signal1<const std::string&> SignalOnSdpReady;
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private:
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void SetLocalDescription(const std::string& type, const std::string& sdp);
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void SetRemoteDescription(const std::string& type, const std::string& sdp);
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bool CheckForConnection();
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bool CheckForAudio();
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bool CheckForVideo();
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talk_base::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
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bool audio, const webrtc::FakeConstraints& audio_constraints,
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bool video, const webrtc::FakeConstraints& video_constraints);
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std::string name_;
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talk_base::Thread audio_thread_;
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talk_base::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
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allocator_factory_;
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talk_base::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
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talk_base::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
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peer_connection_factory_;
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talk_base::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
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talk_base::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
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};
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#endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
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