1 /* 2 * libjingle 3 * Copyright 2015 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_MEDIASTREAMOBSERVER_H_ 29 #define TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_ 30 31 #include "talk/app/webrtc/mediastreaminterface.h" 32 #include "webrtc/base/scoped_ref_ptr.h" 33 #include "webrtc/base/sigslot.h" 34 35 namespace webrtc { 36 37 // Helper class which will listen for changes to a stream and emit the 38 // corresponding signals. 39 class MediaStreamObserver : public ObserverInterface { 40 public: 41 explicit MediaStreamObserver(MediaStreamInterface* stream); 42 ~MediaStreamObserver(); 43 44 const MediaStreamInterface* stream() const { return stream_; } 45 46 void OnChanged() override; 47 48 sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*> 49 SignalAudioTrackAdded; 50 sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*> 51 SignalAudioTrackRemoved; 52 sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*> 53 SignalVideoTrackAdded; 54 sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*> 55 SignalVideoTrackRemoved; 56 57 private: 58 rtc::scoped_refptr<MediaStreamInterface> stream_; 59 AudioTrackVector cached_audio_tracks_; 60 VideoTrackVector cached_video_tracks_; 61 }; 62 63 } // namespace webrtc 64 65 #endif // TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_ 66
