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Side by Side Diff: talk/app/webrtc/mediastreamobserver.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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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_
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