Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/api/test/peerconnectiontestwrapper.h

Index: webrtc/api/test/peerconnectiontestwrapper.h
diff --git a/webrtc/api/test/peerconnectiontestwrapper.h b/webrtc/api/test/peerconnectiontestwrapper.h
deleted file mode 100644
index 6433e8fb05546e60f65736ef0ac04bfccb6f5265..0000000000000000000000000000000000000000
--- a/webrtc/api/test/peerconnectiontestwrapper.h
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
-#define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_
-
-#include <memory>
-
-#include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/api/test/fakeaudiocapturemodule.h"
-#include "webrtc/api/test/fakeconstraints.h"
-#include "webrtc/api/test/fakevideotrackrenderer.h"
-#include "webrtc/base/sigslot.h"
-
-class PeerConnectionTestWrapper
-    : public webrtc::PeerConnectionObserver,
-      public webrtc::CreateSessionDescriptionObserver,
-      public sigslot::has_slots<> {
- public:
-  // We need these using declarations because there are two versions of each of
-  // the below methods and we only override one of them.
-  // TODO(deadbeef): Remove once there's only one version of the methods.
-  using PeerConnectionObserver::OnAddStream;
-  using PeerConnectionObserver::OnRemoveStream;
-  using PeerConnectionObserver::OnDataChannel;
-
-  static void Connect(PeerConnectionTestWrapper* caller,
-                      PeerConnectionTestWrapper* callee);
-
-  PeerConnectionTestWrapper(const std::string& name,
-                            rtc::Thread* network_thread,
-                            rtc::Thread* worker_thread);
-  virtual ~PeerConnectionTestWrapper();
-
-  bool CreatePc(
-      const webrtc::MediaConstraintsInterface* constraints,
-      const webrtc::PeerConnectionInterface::RTCConfiguration& config);
-
-  webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); }
-
-  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(
-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream);
-  virtual void OnRemoveStream(
-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {}
-  virtual void OnDataChannel(
-      rtc::scoped_refptr<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::Thread* const network_thread_;
-  rtc::Thread* const worker_thread_;
-  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
-  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
-      peer_connection_factory_;
-  rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
-  std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
-};
-
-#endif  // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_