blob: ce5e04536d5156540533791403918a90c7d02765 [file] [log] [blame]
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
jlmiller@webrtc.org5f93d0a2015-01-20 21:36:13 +00003 * Copyright 2012 Google Inc.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00004 *
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
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000040#include "webrtc/base/basictypes.h"
41#include "webrtc/base/refcount.h"
42#include "webrtc/base/scoped_ref_ptr.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000043
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.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000076class MediaSourceInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000077 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.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000093class MediaStreamTrackInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000094 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:
guoweis@webrtc.orgf9a75d92015-03-10 06:37:00 +0000118 virtual void SetSize(int width, int height) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000119 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.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000145class AudioSourceInterface : public MediaSourceInterface {
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000146 public:
147 class AudioObserver {
148 public:
149 virtual void OnSetVolume(double volume) = 0;
150
151 protected:
152 virtual ~AudioObserver() {}
153 };
154
155 // TODO(xians): Makes all the interface pure virtual after Chrome has their
156 // implementations.
157 // Sets the volume to the source. |volume| is in the range of [0, 10].
158 virtual void SetVolume(double volume) {}
159
160 // Registers/unregisters observer to the audio source.
161 virtual void RegisterAudioObserver(AudioObserver* observer) {}
162 virtual void UnregisterAudioObserver(AudioObserver* observer) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000163};
164
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000165// Interface for receiving audio data from a AudioTrack.
166class AudioTrackSinkInterface {
167 public:
168 virtual void OnData(const void* audio_data,
169 int bits_per_sample,
170 int sample_rate,
171 int number_of_channels,
172 int number_of_frames) = 0;
173 protected:
174 virtual ~AudioTrackSinkInterface() {}
175};
176
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000177// Interface of the audio processor used by the audio track to collect
178// statistics.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000179class AudioProcessorInterface : public rtc::RefCountInterface {
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000180 public:
181 struct AudioProcessorStats {
182 AudioProcessorStats() : typing_noise_detected(false),
183 echo_return_loss(0),
184 echo_return_loss_enhancement(0),
185 echo_delay_median_ms(0),
186 aec_quality_min(0.0),
187 echo_delay_std_ms(0) {}
188 ~AudioProcessorStats() {}
189
190 bool typing_noise_detected;
191 int echo_return_loss;
192 int echo_return_loss_enhancement;
193 int echo_delay_median_ms;
194 float aec_quality_min;
195 int echo_delay_std_ms;
196 };
197
198 // Get audio processor statistics.
199 virtual void GetStats(AudioProcessorStats* stats) = 0;
200
201 protected:
202 virtual ~AudioProcessorInterface() {}
203};
204
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000205class AudioTrackInterface : public MediaStreamTrackInterface {
206 public:
207 // TODO(xians): Figure out if the following interface should be const or not.
208 virtual AudioSourceInterface* GetSource() const = 0;
209
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000210 // Add/Remove a sink that will receive the audio data from the track.
211 virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
212 virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0;
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000213
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000214 // Get the signal level from the audio track.
215 // Return true on success, otherwise false.
216 // TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
217 // after Chrome has the correct implementation of the interface.
218 virtual bool GetSignalLevel(int* level) { return false; }
219
220 // Get the audio processor used by the audio track. Return NULL if the track
221 // does not have any processor.
222 // TODO(xians): Make the interface pure virtual.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000223 virtual rtc::scoped_refptr<AudioProcessorInterface>
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000224 GetAudioProcessor() { return NULL; }
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000225
226 // Get a pointer to the audio renderer of this AudioTrack.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000227 // The pointer is valid for the lifetime of this AudioTrack.
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000228 // TODO(xians): Remove the following interface after Chrome switches to
229 // AddSink() and RemoveSink() interfaces.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000230 virtual cricket::AudioRenderer* GetRenderer() { return NULL; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000231
232 protected:
233 virtual ~AudioTrackInterface() {}
234};
235
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000236typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> >
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000237 AudioTrackVector;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000238typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000239 VideoTrackVector;
240
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000241class MediaStreamInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000242 public NotifierInterface {
243 public:
244 virtual std::string label() const = 0;
245
246 virtual AudioTrackVector GetAudioTracks() = 0;
247 virtual VideoTrackVector GetVideoTracks() = 0;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000248 virtual rtc::scoped_refptr<AudioTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000249 FindAudioTrack(const std::string& track_id) = 0;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000250 virtual rtc::scoped_refptr<VideoTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000251 FindVideoTrack(const std::string& track_id) = 0;
252
253 virtual bool AddTrack(AudioTrackInterface* track) = 0;
254 virtual bool AddTrack(VideoTrackInterface* track) = 0;
255 virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
256 virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
257
258 protected:
259 virtual ~MediaStreamInterface() {}
260};
261
262} // namespace webrtc
263
264#endif // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_