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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 // This file contains classes that implement RtpReceiverInterface. | |
29 // An RtpReceiver associates a MediaStreamTrackInterface with an underlying | |
30 // transport (provided by AudioProviderInterface/VideoProviderInterface) | |
31 | |
32 #ifndef TALK_APP_WEBRTC_RTPRECEIVER_H_ | |
33 #define TALK_APP_WEBRTC_RTPRECEIVER_H_ | |
34 | |
35 #include <string> | |
36 | |
37 #include "talk/app/webrtc/mediastreamprovider.h" | |
38 #include "talk/app/webrtc/rtpreceiverinterface.h" | |
39 #include "webrtc/base/basictypes.h" | |
40 | |
41 namespace webrtc { | |
42 | |
43 class AudioRtpReceiver : public ObserverInterface, | |
44 public AudioSourceInterface::AudioObserver, | |
45 public rtc::RefCountedObject<RtpReceiverInterface> { | |
46 public: | |
47 AudioRtpReceiver(AudioTrackInterface* track, | |
48 uint32_t ssrc, | |
49 AudioProviderInterface* provider); | |
50 | |
51 virtual ~AudioRtpReceiver(); | |
52 | |
53 // ObserverInterface implementation | |
54 void OnChanged() override; | |
55 | |
56 // AudioSourceInterface::AudioObserver implementation | |
57 void OnSetVolume(double volume) override; | |
58 | |
59 // RtpReceiverInterface implementation | |
60 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | |
61 return track_.get(); | |
62 } | |
63 | |
64 std::string id() const override { return id_; } | |
65 | |
66 void Stop() override; | |
67 | |
68 private: | |
69 void Reconfigure(); | |
70 | |
71 const std::string id_; | |
72 const rtc::scoped_refptr<AudioTrackInterface> track_; | |
73 const uint32_t ssrc_; | |
74 AudioProviderInterface* provider_; // Set to null in Stop(). | |
75 bool cached_track_enabled_; | |
76 }; | |
77 | |
78 class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> { | |
79 public: | |
80 VideoRtpReceiver(VideoTrackInterface* track, | |
81 uint32_t ssrc, | |
82 VideoProviderInterface* provider); | |
83 | |
84 virtual ~VideoRtpReceiver(); | |
85 | |
86 // RtpReceiverInterface implementation | |
87 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override { | |
88 return track_.get(); | |
89 } | |
90 | |
91 std::string id() const override { return id_; } | |
92 | |
93 void Stop() override; | |
94 | |
95 private: | |
96 std::string id_; | |
97 rtc::scoped_refptr<VideoTrackInterface> track_; | |
98 uint32_t ssrc_; | |
99 VideoProviderInterface* provider_; | |
100 }; | |
101 | |
102 } // namespace webrtc | |
103 | |
104 #endif // TALK_APP_WEBRTC_RTPRECEIVER_H_ | |
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