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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
kjellanderb24317b2016-02-10 07:54:43 -08002 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00003 *
kjellanderb24317b2016-02-10 07:54:43 -08004 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00009 */
10
11// This file contains interfaces for MediaStream, MediaTrack and MediaSource.
12// These interfaces are used for implementing MediaStream and MediaTrack as
13// defined in http://dev.w3.org/2011/webrtc/editor/webrtc.html#stream-api. These
14// interfaces must be used only with PeerConnection. PeerConnectionManager
15// interface provides the factory methods to create MediaStream and MediaTracks.
16
Henrik Kjellander15583c12016-02-10 10:53:12 +010017#ifndef WEBRTC_API_MEDIASTREAMINTERFACE_H_
18#define WEBRTC_API_MEDIASTREAMINTERFACE_H_
henrike@webrtc.org28e20752013-07-10 00:45:36 +000019
20#include <string>
21#include <vector>
22
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000023#include "webrtc/base/basictypes.h"
24#include "webrtc/base/refcount.h"
25#include "webrtc/base/scoped_ref_ptr.h"
Perc0d31e92016-03-31 17:23:39 +020026#include "webrtc/base/optional.h"
perkja3ede6c2016-03-08 01:27:48 +010027#include "webrtc/media/base/mediachannel.h"
nissee73afba2016-01-28 04:47:08 -080028#include "webrtc/media/base/videosinkinterface.h"
nissedb25d2e2016-02-26 01:24:58 -080029#include "webrtc/media/base/videosourceinterface.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000030
31namespace cricket {
32
33class AudioRenderer;
34class VideoCapturer;
35class VideoRenderer;
36class VideoFrame;
37
38} // namespace cricket
39
40namespace webrtc {
41
42// Generic observer interface.
43class ObserverInterface {
44 public:
45 virtual void OnChanged() = 0;
46
47 protected:
48 virtual ~ObserverInterface() {}
49};
50
51class NotifierInterface {
52 public:
53 virtual void RegisterObserver(ObserverInterface* observer) = 0;
54 virtual void UnregisterObserver(ObserverInterface* observer) = 0;
55
56 virtual ~NotifierInterface() {}
57};
58
59// Base class for sources. A MediaStreamTrack have an underlying source that
60// provide media. A source can be shared with multiple tracks.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000061class MediaSourceInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000062 public NotifierInterface {
63 public:
64 enum SourceState {
65 kInitializing,
66 kLive,
67 kEnded,
68 kMuted
69 };
70
71 virtual SourceState state() const = 0;
72
tommi6eca7e32015-12-15 04:27:11 -080073 virtual bool remote() const = 0;
74
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075 protected:
76 virtual ~MediaSourceInterface() {}
77};
78
79// Information about a track.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000080class MediaStreamTrackInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000081 public NotifierInterface {
82 public:
83 enum TrackState {
perkjc8f952d2016-03-23 00:33:56 -070084 kLive,
85 kEnded,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000086 };
87
deadbeeffac06552015-11-25 11:26:01 -080088 static const char kAudioKind[];
89 static const char kVideoKind[];
90
nissefcc640f2016-04-01 01:10:42 -070091 // The kind() method must return kAudioKind only if the object is a
92 // subclass of AudioTrackInterface, and kVideoKind only if the
93 // object is a subclass of VideoTrackInterface. It is typically used
94 // to protect a static_cast<> to the corresponding subclass.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000095 virtual std::string kind() const = 0;
96 virtual std::string id() const = 0;
97 virtual bool enabled() const = 0;
98 virtual TrackState state() const = 0;
99 virtual bool set_enabled(bool enable) = 0;
fischman@webrtc.org32001ef2013-08-12 23:26:21 +0000100
101 protected:
102 virtual ~MediaStreamTrackInterface() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103};
104
perkja3ede6c2016-03-08 01:27:48 +0100105// VideoTrackSourceInterface is a reference counted source used for VideoTracks.
106// The same source can be used in multiple VideoTracks.
107class VideoTrackSourceInterface
108 : public MediaSourceInterface,
109 public rtc::VideoSourceInterface<cricket::VideoFrame> {
110 public:
nissefcc640f2016-04-01 01:10:42 -0700111 struct Stats {
112 // Original size of captured frame, before video adaptation.
113 int input_width;
114 int input_height;
115 };
perkja3ede6c2016-03-08 01:27:48 +0100116 // Get access to the source implementation of cricket::VideoCapturer.
117 // This can be used for receiving frames and state notifications.
118 // But it should not be used for starting or stopping capturing.
119 // TODO(perkj): We are currently trying to replace all internal use of
120 // cricket::VideoCapturer with rtc::VideoSourceInterface. Once that
121 // refactoring is done,
122 // remove this method.
123 virtual cricket::VideoCapturer* GetVideoCapturer() = 0;
124
125 virtual void Stop() = 0;
126 virtual void Restart() = 0;
127
perkj0d3eef22016-03-09 02:39:17 +0100128 // Indicates that parameters suitable for screencasts should be automatically
129 // applied to RtpSenders.
130 // TODO(perkj): Remove these once all known applications have moved to
131 // explicitly setting suitable parameters for screencasts and dont' need this
132 // implicit behavior.
133 virtual bool is_screencast() const = 0;
134
Perc0d31e92016-03-31 17:23:39 +0200135 // Indicates that the encoder should denoise video before encoding it.
136 // If it is not set, the default configuration is used which is different
137 // depending on video codec.
perkj0d3eef22016-03-09 02:39:17 +0100138 // TODO(perkj): Remove this once denoising is done by the source, and not by
139 // the encoder.
Perc0d31e92016-03-31 17:23:39 +0200140 virtual rtc::Optional<bool> needs_denoising() const = 0;
perkja3ede6c2016-03-08 01:27:48 +0100141
nissefcc640f2016-04-01 01:10:42 -0700142 // Returns false if no stats are available, e.g, for a remote
143 // source, or a source which has not seen its first frame yet.
144 // Should avoid blocking.
145 virtual bool GetStats(Stats* stats) = 0;
146
perkja3ede6c2016-03-08 01:27:48 +0100147 protected:
148 virtual ~VideoTrackSourceInterface() {}
149};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000150
nissedb25d2e2016-02-26 01:24:58 -0800151class VideoTrackInterface
152 : public MediaStreamTrackInterface,
153 public rtc::VideoSourceInterface<cricket::VideoFrame> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000154 public:
nissedb25d2e2016-02-26 01:24:58 -0800155 // Register a video sink for this track.
156 void AddOrUpdateSink(rtc::VideoSinkInterface<cricket::VideoFrame>* sink,
157 const rtc::VideoSinkWants& wants) override{};
158 void RemoveSink(
159 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override{};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000160
perkja3ede6c2016-03-08 01:27:48 +0100161 virtual VideoTrackSourceInterface* GetSource() const = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000162
163 protected:
164 virtual ~VideoTrackInterface() {}
165};
166
tommi6eca7e32015-12-15 04:27:11 -0800167// Interface for receiving audio data from a AudioTrack.
168class AudioTrackSinkInterface {
169 public:
170 virtual void OnData(const void* audio_data,
171 int bits_per_sample,
172 int sample_rate,
Peter Kasting69558702016-01-12 16:26:35 -0800173 size_t number_of_channels,
tommi6eca7e32015-12-15 04:27:11 -0800174 size_t number_of_frames) = 0;
175
176 protected:
177 virtual ~AudioTrackSinkInterface() {}
178};
179
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000180// AudioSourceInterface is a reference counted source used for AudioTracks.
181// The same source can be used in multiple AudioTracks.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000182class AudioSourceInterface : public MediaSourceInterface {
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000183 public:
184 class AudioObserver {
185 public:
186 virtual void OnSetVolume(double volume) = 0;
187
188 protected:
189 virtual ~AudioObserver() {}
190 };
191
192 // TODO(xians): Makes all the interface pure virtual after Chrome has their
193 // implementations.
194 // Sets the volume to the source. |volume| is in the range of [0, 10].
Tommif888bb52015-12-12 01:37:01 +0100195 // TODO(tommi): This method should be on the track and ideally volume should
196 // be applied in the track in a way that does not affect clones of the track.
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000197 virtual void SetVolume(double volume) {}
198
199 // Registers/unregisters observer to the audio source.
200 virtual void RegisterAudioObserver(AudioObserver* observer) {}
201 virtual void UnregisterAudioObserver(AudioObserver* observer) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000202
tommi6eca7e32015-12-15 04:27:11 -0800203 // TODO(tommi): Make pure virtual.
204 virtual void AddSink(AudioTrackSinkInterface* sink) {}
205 virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000206};
207
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000208// Interface of the audio processor used by the audio track to collect
209// statistics.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000210class AudioProcessorInterface : public rtc::RefCountInterface {
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000211 public:
212 struct AudioProcessorStats {
213 AudioProcessorStats() : typing_noise_detected(false),
214 echo_return_loss(0),
215 echo_return_loss_enhancement(0),
216 echo_delay_median_ms(0),
217 aec_quality_min(0.0),
218 echo_delay_std_ms(0) {}
219 ~AudioProcessorStats() {}
220
221 bool typing_noise_detected;
222 int echo_return_loss;
223 int echo_return_loss_enhancement;
224 int echo_delay_median_ms;
225 float aec_quality_min;
226 int echo_delay_std_ms;
227 };
228
229 // Get audio processor statistics.
230 virtual void GetStats(AudioProcessorStats* stats) = 0;
231
232 protected:
233 virtual ~AudioProcessorInterface() {}
234};
235
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000236class AudioTrackInterface : public MediaStreamTrackInterface {
237 public:
238 // TODO(xians): Figure out if the following interface should be const or not.
239 virtual AudioSourceInterface* GetSource() const = 0;
240
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000241 // Add/Remove a sink that will receive the audio data from the track.
242 virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
243 virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0;
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000244
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000245 // Get the signal level from the audio track.
246 // Return true on success, otherwise false.
247 // TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
248 // after Chrome has the correct implementation of the interface.
249 virtual bool GetSignalLevel(int* level) { return false; }
250
251 // Get the audio processor used by the audio track. Return NULL if the track
252 // does not have any processor.
253 // TODO(xians): Make the interface pure virtual.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000254 virtual rtc::scoped_refptr<AudioProcessorInterface>
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000255 GetAudioProcessor() { return NULL; }
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000256
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000257 protected:
258 virtual ~AudioTrackInterface() {}
259};
260
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000261typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> >
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000262 AudioTrackVector;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000263typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000264 VideoTrackVector;
265
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000266class MediaStreamInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000267 public NotifierInterface {
268 public:
269 virtual std::string label() const = 0;
270
271 virtual AudioTrackVector GetAudioTracks() = 0;
272 virtual VideoTrackVector GetVideoTracks() = 0;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000273 virtual rtc::scoped_refptr<AudioTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000274 FindAudioTrack(const std::string& track_id) = 0;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000275 virtual rtc::scoped_refptr<VideoTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000276 FindVideoTrack(const std::string& track_id) = 0;
277
278 virtual bool AddTrack(AudioTrackInterface* track) = 0;
279 virtual bool AddTrack(VideoTrackInterface* track) = 0;
280 virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
281 virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
282
283 protected:
284 virtual ~MediaStreamInterface() {}
285};
286
287} // namespace webrtc
288
Henrik Kjellander15583c12016-02-10 10:53:12 +0100289#endif // WEBRTC_API_MEDIASTREAMINTERFACE_H_