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
track()60   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
61     return track_.get();
62   }
63 
id()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
track()87   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
88     return track_.get();
89   }
90 
id()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_
105