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source: webrtc/talk/app/webrtc/test/peerconnectiontestwrapper.h @ 0:4bda6873e34c

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Last change on this file since 0:4bda6873e34c was 0:4bda6873e34c, checked in by Michael Prittie <mprittie@…>, 6 years ago

Scrubbed password for publication.

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1/*
2 * libjingle
3 * Copyright 2013, Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 *  1. Redistributions of source code must retain the above copyright notice,
9 *     this list of conditions and the following disclaimer.
10 *  2. Redistributions in binary form must reproduce the above copyright notice,
11 *     this list of conditions and the following disclaimer in the documentation
12 *     and/or other materials provided with the distribution.
13 *  3. The name of the author may not be used to endorse or promote products
14 *     derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
29#define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
30
31#include "talk/app/webrtc/peerconnectioninterface.h"
32#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
33#include "talk/app/webrtc/test/fakeconstraints.h"
34#include "talk/app/webrtc/test/fakevideotrackrenderer.h"
35#include "talk/base/sigslot.h"
36#include "talk/base/thread.h"
37
38namespace webrtc {
39class PortAllocatorFactoryInterface;
40}
41
42class PeerConnectionTestWrapper
43    : public webrtc::PeerConnectionObserver,
44      public webrtc::CreateSessionDescriptionObserver,
45      public sigslot::has_slots<> {
46 public:
47  static void Connect(PeerConnectionTestWrapper* caller,
48                      PeerConnectionTestWrapper* callee);
49
50  explicit PeerConnectionTestWrapper(const std::string& name);
51  virtual ~PeerConnectionTestWrapper();
52
53  bool CreatePc(const webrtc::MediaConstraintsInterface* constraints);
54
55  // Implements PeerConnectionObserver.
56  virtual void OnError() {}
57  virtual void OnSignalingChange(
58     webrtc::PeerConnectionInterface::SignalingState new_state) {}
59  virtual void OnStateChange(
60      webrtc::PeerConnectionObserver::StateType state_changed) {}
61  virtual void OnAddStream(webrtc::MediaStreamInterface* stream);
62  virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream) {}
63  virtual void OnDataChannel(webrtc::DataChannelInterface* data_channel) {}
64  virtual void OnRenegotiationNeeded() {}
65  virtual void OnIceConnectionChange(
66      webrtc::PeerConnectionInterface::IceConnectionState new_state) {}
67  virtual void OnIceGatheringChange(
68      webrtc::PeerConnectionInterface::IceGatheringState new_state) {}
69  virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate);
70  virtual void OnIceComplete() {}
71
72  // Implements CreateSessionDescriptionObserver.
73  virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc);
74  virtual void OnFailure(const std::string& error) {}
75
76  void CreateOffer(const webrtc::MediaConstraintsInterface* constraints);
77  void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints);
78  void ReceiveOfferSdp(const std::string& sdp);
79  void ReceiveAnswerSdp(const std::string& sdp);
80  void AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
81                       const std::string& candidate);
82  void WaitForCallEstablished();
83  void WaitForConnection();
84  void WaitForAudio();
85  void WaitForVideo();
86  void GetAndAddUserMedia(
87    bool audio, const webrtc::FakeConstraints& audio_constraints,
88    bool video, const webrtc::FakeConstraints& video_constraints);
89
90  // sigslots
91  sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
92  sigslot::signal3<const std::string&,
93                   int,
94                   const std::string&> SignalOnIceCandidateReady;
95  sigslot::signal1<std::string*> SignalOnSdpCreated;
96  sigslot::signal1<const std::string&> SignalOnSdpReady;
97
98 private:
99  void SetLocalDescription(const std::string& type, const std::string& sdp);
100  void SetRemoteDescription(const std::string& type, const std::string& sdp);
101  bool CheckForConnection();
102  bool CheckForAudio();
103  bool CheckForVideo();
104  talk_base::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
105      bool audio, const webrtc::FakeConstraints& audio_constraints,
106      bool video, const webrtc::FakeConstraints& video_constraints);
107
108  std::string name_;
109  talk_base::Thread audio_thread_;
110  talk_base::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
111      allocator_factory_;
112  talk_base::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
113  talk_base::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
114      peer_connection_factory_;
115  talk_base::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
116  talk_base::scoped_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
117};
118
119#endif  // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_
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