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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
pbos9baddf22017-01-02 06:44:41 -080020#include <stddef.h>
21
henrike@webrtc.org28e20752013-07-10 00:45:36 +000022#include <string>
23#include <vector>
24
nisseaf916892017-01-10 07:44:26 -080025#include "webrtc/api/video/video_frame.h"
26// TODO(nisse): Transition hack, Chrome expects that including this
27// file declares I420Buffer. Delete after users of I420Buffer are
28// fixed to include the new header.
29#include "webrtc/api/video/i420_buffer.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000030#include "webrtc/base/refcount.h"
31#include "webrtc/base/scoped_ref_ptr.h"
Perc0d31e92016-03-31 17:23:39 +020032#include "webrtc/base/optional.h"
perkja3ede6c2016-03-08 01:27:48 +010033#include "webrtc/media/base/mediachannel.h"
nissee73afba2016-01-28 04:47:08 -080034#include "webrtc/media/base/videosinkinterface.h"
nissedb25d2e2016-02-26 01:24:58 -080035#include "webrtc/media/base/videosourceinterface.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037namespace webrtc {
38
39// Generic observer interface.
40class ObserverInterface {
41 public:
42 virtual void OnChanged() = 0;
43
44 protected:
45 virtual ~ObserverInterface() {}
46};
47
48class NotifierInterface {
49 public:
50 virtual void RegisterObserver(ObserverInterface* observer) = 0;
51 virtual void UnregisterObserver(ObserverInterface* observer) = 0;
52
53 virtual ~NotifierInterface() {}
54};
55
56// Base class for sources. A MediaStreamTrack have an underlying source that
57// provide media. A source can be shared with multiple tracks.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000058class MediaSourceInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059 public NotifierInterface {
60 public:
61 enum SourceState {
62 kInitializing,
63 kLive,
64 kEnded,
65 kMuted
66 };
67
68 virtual SourceState state() const = 0;
69
tommi6eca7e32015-12-15 04:27:11 -080070 virtual bool remote() const = 0;
71
henrike@webrtc.org28e20752013-07-10 00:45:36 +000072 protected:
73 virtual ~MediaSourceInterface() {}
74};
75
76// Information about a track.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000077class MediaStreamTrackInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000078 public NotifierInterface {
79 public:
80 enum TrackState {
perkjc8f952d2016-03-23 00:33:56 -070081 kLive,
82 kEnded,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000083 };
84
deadbeeffac06552015-11-25 11:26:01 -080085 static const char kAudioKind[];
86 static const char kVideoKind[];
87
nissefcc640f2016-04-01 01:10:42 -070088 // The kind() method must return kAudioKind only if the object is a
89 // subclass of AudioTrackInterface, and kVideoKind only if the
90 // object is a subclass of VideoTrackInterface. It is typically used
91 // to protect a static_cast<> to the corresponding subclass.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000092 virtual std::string kind() const = 0;
93 virtual std::string id() const = 0;
94 virtual bool enabled() const = 0;
95 virtual TrackState state() const = 0;
96 virtual bool set_enabled(bool enable) = 0;
fischman@webrtc.org32001ef2013-08-12 23:26:21 +000097
98 protected:
99 virtual ~MediaStreamTrackInterface() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000100};
101
perkja3ede6c2016-03-08 01:27:48 +0100102// VideoTrackSourceInterface is a reference counted source used for VideoTracks.
103// The same source can be used in multiple VideoTracks.
104class VideoTrackSourceInterface
105 : public MediaSourceInterface,
nisseacd935b2016-11-11 03:55:13 -0800106 public rtc::VideoSourceInterface<VideoFrame> {
perkja3ede6c2016-03-08 01:27:48 +0100107 public:
nissefcc640f2016-04-01 01:10:42 -0700108 struct Stats {
109 // Original size of captured frame, before video adaptation.
110 int input_width;
111 int input_height;
112 };
perkja3ede6c2016-03-08 01:27:48 +0100113
perkj0d3eef22016-03-09 02:39:17 +0100114 // Indicates that parameters suitable for screencasts should be automatically
115 // applied to RtpSenders.
116 // TODO(perkj): Remove these once all known applications have moved to
117 // explicitly setting suitable parameters for screencasts and dont' need this
118 // implicit behavior.
119 virtual bool is_screencast() const = 0;
120
Perc0d31e92016-03-31 17:23:39 +0200121 // Indicates that the encoder should denoise video before encoding it.
122 // If it is not set, the default configuration is used which is different
123 // depending on video codec.
perkj0d3eef22016-03-09 02:39:17 +0100124 // TODO(perkj): Remove this once denoising is done by the source, and not by
125 // the encoder.
Perc0d31e92016-03-31 17:23:39 +0200126 virtual rtc::Optional<bool> needs_denoising() const = 0;
perkja3ede6c2016-03-08 01:27:48 +0100127
nissefcc640f2016-04-01 01:10:42 -0700128 // Returns false if no stats are available, e.g, for a remote
129 // source, or a source which has not seen its first frame yet.
130 // Should avoid blocking.
131 virtual bool GetStats(Stats* stats) = 0;
132
perkja3ede6c2016-03-08 01:27:48 +0100133 protected:
134 virtual ~VideoTrackSourceInterface() {}
135};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000136
nissedb25d2e2016-02-26 01:24:58 -0800137class VideoTrackInterface
138 : public MediaStreamTrackInterface,
nisseacd935b2016-11-11 03:55:13 -0800139 public rtc::VideoSourceInterface<VideoFrame> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000140 public:
pbos5214a0a2016-12-16 15:39:11 -0800141 // Video track content hint, used to override the source is_screencast
142 // property.
143 // See https://crbug.com/653531 and https://github.com/WICG/mst-content-hint.
144 enum class ContentHint { kNone, kFluid, kDetailed };
145
nissedb25d2e2016-02-26 01:24:58 -0800146 // Register a video sink for this track.
nisseacd935b2016-11-11 03:55:13 -0800147 void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
pbos5214a0a2016-12-16 15:39:11 -0800148 const rtc::VideoSinkWants& wants) override {}
149 void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000150
perkja3ede6c2016-03-08 01:27:48 +0100151 virtual VideoTrackSourceInterface* GetSource() const = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000152
pbos5214a0a2016-12-16 15:39:11 -0800153 virtual ContentHint content_hint() const { return ContentHint::kNone; }
154 virtual void set_content_hint(ContentHint hint) {}
155
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000156 protected:
157 virtual ~VideoTrackInterface() {}
158};
159
tommi6eca7e32015-12-15 04:27:11 -0800160// Interface for receiving audio data from a AudioTrack.
161class AudioTrackSinkInterface {
162 public:
163 virtual void OnData(const void* audio_data,
164 int bits_per_sample,
165 int sample_rate,
Peter Kasting69558702016-01-12 16:26:35 -0800166 size_t number_of_channels,
tommi6eca7e32015-12-15 04:27:11 -0800167 size_t number_of_frames) = 0;
168
169 protected:
170 virtual ~AudioTrackSinkInterface() {}
171};
172
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000173// AudioSourceInterface is a reference counted source used for AudioTracks.
174// The same source can be used in multiple AudioTracks.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000175class AudioSourceInterface : public MediaSourceInterface {
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000176 public:
177 class AudioObserver {
178 public:
179 virtual void OnSetVolume(double volume) = 0;
180
181 protected:
182 virtual ~AudioObserver() {}
183 };
184
185 // TODO(xians): Makes all the interface pure virtual after Chrome has their
186 // implementations.
187 // Sets the volume to the source. |volume| is in the range of [0, 10].
Tommif888bb52015-12-12 01:37:01 +0100188 // TODO(tommi): This method should be on the track and ideally volume should
189 // be applied in the track in a way that does not affect clones of the track.
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000190 virtual void SetVolume(double volume) {}
191
192 // Registers/unregisters observer to the audio source.
193 virtual void RegisterAudioObserver(AudioObserver* observer) {}
194 virtual void UnregisterAudioObserver(AudioObserver* observer) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000195
tommi6eca7e32015-12-15 04:27:11 -0800196 // TODO(tommi): Make pure virtual.
197 virtual void AddSink(AudioTrackSinkInterface* sink) {}
198 virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000199};
200
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000201// Interface of the audio processor used by the audio track to collect
202// statistics.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000203class AudioProcessorInterface : public rtc::RefCountInterface {
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000204 public:
205 struct AudioProcessorStats {
ivoc4e477a12017-01-15 08:29:46 -0800206 AudioProcessorStats()
207 : typing_noise_detected(false),
208 echo_return_loss(0),
209 echo_return_loss_enhancement(0),
210 echo_delay_median_ms(0),
211 echo_delay_std_ms(0),
212 aec_quality_min(0.0),
213 residual_echo_likelihood(0.0f),
214 residual_echo_likelihood_recent_max(0.0f),
215 aec_divergent_filter_fraction(0.0) {}
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000216 ~AudioProcessorStats() {}
217
218 bool typing_noise_detected;
219 int echo_return_loss;
220 int echo_return_loss_enhancement;
221 int echo_delay_median_ms;
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000222 int echo_delay_std_ms;
ivoc8c63a822016-10-21 04:10:03 -0700223 float aec_quality_min;
224 float residual_echo_likelihood;
ivoc4e477a12017-01-15 08:29:46 -0800225 float residual_echo_likelihood_recent_max;
Minyue2a8a78c2016-04-07 16:48:15 +0200226 float aec_divergent_filter_fraction;
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000227 };
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_