Move talk/app/webrtc to webrtc/api

talk/app/webrtc/test/peerconnectiontestwrapper.h

-/*
- * 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"
-
-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::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_