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
29#ifndef TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
30#define TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
31
32#include "talk/app/webrtc/audiotrack.h"
33#include "talk/app/webrtc/mediastreamsignaling.h"
34#include "talk/app/webrtc/videotrack.h"
35
36static const char kStream1[] = "stream1";
37static const char kVideoTrack1[] = "video1";
38static const char kAudioTrack1[] = "audio1";
39
40static const char kStream2[] = "stream2";
41static const char kVideoTrack2[] = "video2";
42static const char kAudioTrack2[] = "audio2";
43
44class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
45                                 public webrtc::MediaStreamSignalingObserver {
46 public:
47  explicit FakeMediaStreamSignaling(cricket::ChannelManager* channel_manager) :
48    webrtc::MediaStreamSignaling(rtc::Thread::Current(), this,
49                                 channel_manager) {
50  }
51
52  void SendAudioVideoStream1() {
53    ClearLocalStreams();
54    AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
55  }
56
57  void SendAudioVideoStream2() {
58    ClearLocalStreams();
59    AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
60  }
61
62  void SendAudioVideoStream1And2() {
63    ClearLocalStreams();
64    AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
65    AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
66  }
67
68  void SendNothing() {
69    ClearLocalStreams();
70  }
71
72  void UseOptionsAudioOnly() {
73    ClearLocalStreams();
74    AddLocalStream(CreateStream(kStream2, kAudioTrack2, ""));
75  }
76
77  void UseOptionsVideoOnly() {
78    ClearLocalStreams();
79    AddLocalStream(CreateStream(kStream2, "", kVideoTrack2));
80  }
81
82  void ClearLocalStreams() {
83    while (local_streams()->count() != 0) {
84      RemoveLocalStream(local_streams()->at(0));
85    }
86  }
87
88  // Implements MediaStreamSignalingObserver.
89  virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {
90  }
91  virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {
92  }
93  virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {
94  }
95  virtual void OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream,
96                                    webrtc::AudioTrackInterface* audio_track,
97                                    uint32 ssrc) {
98  }
99  virtual void OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream,
100                                    webrtc::VideoTrackInterface* video_track,
101                                    uint32 ssrc) {
102  }
103  virtual void OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream,
104                                     webrtc::AudioTrackInterface* audio_track,
105                                     uint32 ssrc) {
106  }
107
108  virtual void OnAddRemoteVideoTrack(webrtc::MediaStreamInterface* stream,
109                                     webrtc::VideoTrackInterface* video_track,
110                                     uint32 ssrc) {
111  }
112
113  virtual void OnRemoveRemoteAudioTrack(
114      webrtc::MediaStreamInterface* stream,
115      webrtc::AudioTrackInterface* audio_track) {
116  }
117
118  virtual void OnRemoveRemoteVideoTrack(
119      webrtc::MediaStreamInterface* stream,
120      webrtc::VideoTrackInterface* video_track) {
121  }
122
123  virtual void OnRemoveLocalAudioTrack(
124      webrtc::MediaStreamInterface* stream,
125      webrtc::AudioTrackInterface* audio_track,
126      uint32 ssrc) {
127  }
128  virtual void OnRemoveLocalVideoTrack(
129      webrtc::MediaStreamInterface* stream,
130      webrtc::VideoTrackInterface* video_track) {
131  }
132  virtual void OnRemoveLocalStream(webrtc::MediaStreamInterface* stream) {
133  }
134
135 private:
136  rtc::scoped_refptr<webrtc::MediaStreamInterface> CreateStream(
137      const std::string& stream_label,
138      const std::string& audio_track_id,
139      const std::string& video_track_id) {
140    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
141        webrtc::MediaStream::Create(stream_label));
142
143    if (!audio_track_id.empty()) {
144      rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
145          webrtc::AudioTrack::Create(audio_track_id, NULL));
146      stream->AddTrack(audio_track);
147    }
148
149    if (!video_track_id.empty()) {
150      rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
151          webrtc::VideoTrack::Create(video_track_id, NULL));
152      stream->AddTrack(video_track);
153    }
154    return stream;
155  }
156};
157
158#endif  // TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
159