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
41namespace webrtc {
42
43class 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
78class 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_
105