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