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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 
     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 
     36 static const char kStream1[] = "stream1";
     37 static const char kVideoTrack1[] = "video1";
     38 static const char kAudioTrack1[] = "audio1";
     39 
     40 static const char kStream2[] = "stream2";
     41 static const char kVideoTrack2[] = "video2";
     42 static const char kAudioTrack2[] = "audio2";
     43 
     44 class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
     45                                  public webrtc::MediaStreamSignalingObserver {
     46  public:
     47   explicit FakeMediaStreamSignaling(cricket::ChannelManager* channel_manager) :
     48     webrtc::MediaStreamSignaling(talk_base::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   }
    127   virtual void OnRemoveLocalVideoTrack(
    128       webrtc::MediaStreamInterface* stream,
    129       webrtc::VideoTrackInterface* video_track) {
    130   }
    131   virtual void OnRemoveLocalStream(webrtc::MediaStreamInterface* stream) {
    132   }
    133 
    134  private:
    135   talk_base::scoped_refptr<webrtc::MediaStreamInterface> CreateStream(
    136       const std::string& stream_label,
    137       const std::string& audio_track_id,
    138       const std::string& video_track_id) {
    139     talk_base::scoped_refptr<webrtc::MediaStreamInterface> stream(
    140         webrtc::MediaStream::Create(stream_label));
    141 
    142     if (!audio_track_id.empty()) {
    143       talk_base::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
    144           webrtc::AudioTrack::Create(audio_track_id, NULL));
    145       stream->AddTrack(audio_track);
    146     }
    147 
    148     if (!video_track_id.empty()) {
    149       talk_base::scoped_refptr<webrtc::VideoTrackInterface> video_track(
    150           webrtc::VideoTrack::Create(video_track_id, NULL));
    151       stream->AddTrack(video_track);
    152     }
    153     return stream;
    154   }
    155 };
    156 
    157 #endif  // TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
    158