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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2012, 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 interfaces for MediaStream, MediaTrack and MediaSource.
29// These interfaces are used for implementing MediaStream and MediaTrack as
30// defined in http://dev.w3.org/2011/webrtc/editor/webrtc.html#stream-api. These
31// interfaces must be used only with PeerConnection. PeerConnectionManager
32// interface provides the factory methods to create MediaStream and MediaTracks.
33
34#ifndef TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_
35#define TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_
36
37#include <string>
38#include <vector>
39
40#include "talk/base/basictypes.h"
41#include "talk/base/refcount.h"
42#include "talk/base/scoped_ref_ptr.h"
43
44namespace cricket {
45
46class AudioRenderer;
47class VideoCapturer;
48class VideoRenderer;
49class VideoFrame;
50
51} // namespace cricket
52
53namespace webrtc {
54
55// Generic observer interface.
56class ObserverInterface {
57 public:
58 virtual void OnChanged() = 0;
59
60 protected:
61 virtual ~ObserverInterface() {}
62};
63
64class NotifierInterface {
65 public:
66 virtual void RegisterObserver(ObserverInterface* observer) = 0;
67 virtual void UnregisterObserver(ObserverInterface* observer) = 0;
68
69 virtual ~NotifierInterface() {}
70};
71
72// Base class for sources. A MediaStreamTrack have an underlying source that
73// provide media. A source can be shared with multiple tracks.
74// TODO(perkj): Implement sources for local and remote audio tracks and
75// remote video tracks.
76class MediaSourceInterface : public talk_base::RefCountInterface,
77 public NotifierInterface {
78 public:
79 enum SourceState {
80 kInitializing,
81 kLive,
82 kEnded,
83 kMuted
84 };
85
86 virtual SourceState state() const = 0;
87
88 protected:
89 virtual ~MediaSourceInterface() {}
90};
91
92// Information about a track.
93class MediaStreamTrackInterface : public talk_base::RefCountInterface,
94 public NotifierInterface {
95 public:
96 enum TrackState {
97 kInitializing, // Track is beeing negotiated.
98 kLive = 1, // Track alive
99 kEnded = 2, // Track have ended
100 kFailed = 3, // Track negotiation failed.
101 };
102
103 virtual std::string kind() const = 0;
104 virtual std::string id() const = 0;
105 virtual bool enabled() const = 0;
106 virtual TrackState state() const = 0;
107 virtual bool set_enabled(bool enable) = 0;
108 // These methods should be called by implementation only.
109 virtual bool set_state(TrackState new_state) = 0;
fischman@webrtc.org32001ef2013-08-12 23:26:21 +0000110
111 protected:
112 virtual ~MediaStreamTrackInterface() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000113};
114
115// Interface for rendering VideoFrames from a VideoTrack
116class VideoRendererInterface {
117 public:
118 virtual void SetSize(int width, int height) = 0;
119 virtual void RenderFrame(const cricket::VideoFrame* frame) = 0;
120
121 protected:
122 // The destructor is protected to prevent deletion via the interface.
123 // This is so that we allow reference counted classes, where the destructor
124 // should never be public, to implement the interface.
125 virtual ~VideoRendererInterface() {}
126};
127
128class VideoSourceInterface;
129
130class VideoTrackInterface : public MediaStreamTrackInterface {
131 public:
132 // Register a renderer that will render all frames received on this track.
133 virtual void AddRenderer(VideoRendererInterface* renderer) = 0;
134 // Deregister a renderer.
135 virtual void RemoveRenderer(VideoRendererInterface* renderer) = 0;
136
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000137 virtual VideoSourceInterface* GetSource() const = 0;
138
139 protected:
140 virtual ~VideoTrackInterface() {}
141};
142
143// AudioSourceInterface is a reference counted source used for AudioTracks.
144// The same source can be used in multiple AudioTracks.
wu@webrtc.org0de29502014-02-13 19:54:28 +0000145// TODO(perkj): Extend this class with necessary methods to allow separate
146// sources for each audio track.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000147class AudioSourceInterface : public MediaSourceInterface {
148};
149
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000150// Interface for receiving audio data from a AudioTrack.
151class AudioTrackSinkInterface {
152 public:
153 virtual void OnData(const void* audio_data,
154 int bits_per_sample,
155 int sample_rate,
156 int number_of_channels,
157 int number_of_frames) = 0;
158 protected:
159 virtual ~AudioTrackSinkInterface() {}
160};
161
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000162class AudioTrackInterface : public MediaStreamTrackInterface {
163 public:
164 // TODO(xians): Figure out if the following interface should be const or not.
165 virtual AudioSourceInterface* GetSource() const = 0;
166
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000167 // Adds/Removes a sink that will receive the audio data from the track.
168 // TODO(xians): Make them pure virtual after Chrome implements these
169 // interfaces.
170 virtual void AddSink(AudioTrackSinkInterface* sink) {}
171 virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
172
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000173 // Gets a pointer to the audio renderer of this AudioTrack.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000174 // The pointer is valid for the lifetime of this AudioTrack.
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000175 // TODO(xians): Remove the following interface after Chrome switches to
176 // AddSink() and RemoveSink() interfaces.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000177 virtual cricket::AudioRenderer* GetRenderer() { return NULL; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000178
179 protected:
180 virtual ~AudioTrackInterface() {}
181};
182
183typedef std::vector<talk_base::scoped_refptr<AudioTrackInterface> >
184 AudioTrackVector;
185typedef std::vector<talk_base::scoped_refptr<VideoTrackInterface> >
186 VideoTrackVector;
187
188class MediaStreamInterface : public talk_base::RefCountInterface,
189 public NotifierInterface {
190 public:
191 virtual std::string label() const = 0;
192
193 virtual AudioTrackVector GetAudioTracks() = 0;
194 virtual VideoTrackVector GetVideoTracks() = 0;
195 virtual talk_base::scoped_refptr<AudioTrackInterface>
196 FindAudioTrack(const std::string& track_id) = 0;
197 virtual talk_base::scoped_refptr<VideoTrackInterface>
198 FindVideoTrack(const std::string& track_id) = 0;
199
200 virtual bool AddTrack(AudioTrackInterface* track) = 0;
201 virtual bool AddTrack(VideoTrackInterface* track) = 0;
202 virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
203 virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
204
205 protected:
206 virtual ~MediaStreamInterface() {}
207};
208
209} // namespace webrtc
210
211#endif // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_