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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.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000074class MediaSourceInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075 public NotifierInterface {
76 public:
77 enum SourceState {
78 kInitializing,
79 kLive,
80 kEnded,
81 kMuted
82 };
83
84 virtual SourceState state() const = 0;
85
tommi6eca7e32015-12-15 04:27:11 -080086 virtual bool remote() const = 0;
87
henrike@webrtc.org28e20752013-07-10 00:45:36 +000088 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
deadbeeffac06552015-11-25 11:26:01 -0800103 static const char kAudioKind[];
104 static const char kVideoKind[];
105
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000106 virtual std::string kind() const = 0;
107 virtual std::string id() const = 0;
108 virtual bool enabled() const = 0;
109 virtual TrackState state() const = 0;
110 virtual bool set_enabled(bool enable) = 0;
111 // These methods should be called by implementation only.
112 virtual bool set_state(TrackState new_state) = 0;
fischman@webrtc.org32001ef2013-08-12 23:26:21 +0000113
114 protected:
115 virtual ~MediaStreamTrackInterface() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000116};
117
118// Interface for rendering VideoFrames from a VideoTrack
nisse2098fca2016-01-27 06:12:49 -0800119class VideoRendererInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000120 public:
guoweis@webrtc.org00c509a2015-03-12 21:37:26 +0000121 // |frame| may have pending rotation. For clients which can't apply rotation,
122 // |frame|->GetCopyWithRotationApplied() will return a frame that has the
123 // rotation applied.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000124 virtual void RenderFrame(const cricket::VideoFrame* frame) = 0;
125
126 protected:
127 // The destructor is protected to prevent deletion via the interface.
128 // This is so that we allow reference counted classes, where the destructor
129 // should never be public, to implement the interface.
130 virtual ~VideoRendererInterface() {}
131};
132
133class VideoSourceInterface;
134
135class VideoTrackInterface : public MediaStreamTrackInterface {
136 public:
137 // Register a renderer that will render all frames received on this track.
138 virtual void AddRenderer(VideoRendererInterface* renderer) = 0;
139 // Deregister a renderer.
140 virtual void RemoveRenderer(VideoRendererInterface* renderer) = 0;
141
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000142 virtual VideoSourceInterface* GetSource() const = 0;
143
144 protected:
145 virtual ~VideoTrackInterface() {}
146};
147
tommi6eca7e32015-12-15 04:27:11 -0800148// Interface for receiving audio data from a AudioTrack.
149class AudioTrackSinkInterface {
150 public:
151 virtual void OnData(const void* audio_data,
152 int bits_per_sample,
153 int sample_rate,
Peter Kasting69558702016-01-12 16:26:35 -0800154 size_t number_of_channels,
tommi6eca7e32015-12-15 04:27:11 -0800155 size_t number_of_frames) = 0;
156
157 protected:
158 virtual ~AudioTrackSinkInterface() {}
159};
160
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000161// AudioSourceInterface is a reference counted source used for AudioTracks.
162// The same source can be used in multiple AudioTracks.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000163class AudioSourceInterface : public MediaSourceInterface {
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000164 public:
165 class AudioObserver {
166 public:
167 virtual void OnSetVolume(double volume) = 0;
168
169 protected:
170 virtual ~AudioObserver() {}
171 };
172
173 // TODO(xians): Makes all the interface pure virtual after Chrome has their
174 // implementations.
175 // Sets the volume to the source. |volume| is in the range of [0, 10].
Tommif888bb52015-12-12 01:37:01 +0100176 // TODO(tommi): This method should be on the track and ideally volume should
177 // be applied in the track in a way that does not affect clones of the track.
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000178 virtual void SetVolume(double volume) {}
179
180 // Registers/unregisters observer to the audio source.
181 virtual void RegisterAudioObserver(AudioObserver* observer) {}
182 virtual void UnregisterAudioObserver(AudioObserver* observer) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000183
tommi6eca7e32015-12-15 04:27:11 -0800184 // TODO(tommi): Make pure virtual.
185 virtual void AddSink(AudioTrackSinkInterface* sink) {}
186 virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000187};
188
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000189// Interface of the audio processor used by the audio track to collect
190// statistics.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000191class AudioProcessorInterface : public rtc::RefCountInterface {
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000192 public:
193 struct AudioProcessorStats {
194 AudioProcessorStats() : typing_noise_detected(false),
195 echo_return_loss(0),
196 echo_return_loss_enhancement(0),
197 echo_delay_median_ms(0),
198 aec_quality_min(0.0),
199 echo_delay_std_ms(0) {}
200 ~AudioProcessorStats() {}
201
202 bool typing_noise_detected;
203 int echo_return_loss;
204 int echo_return_loss_enhancement;
205 int echo_delay_median_ms;
206 float aec_quality_min;
207 int echo_delay_std_ms;
208 };
209
210 // Get audio processor statistics.
211 virtual void GetStats(AudioProcessorStats* stats) = 0;
212
213 protected:
214 virtual ~AudioProcessorInterface() {}
215};
216
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000217class AudioTrackInterface : public MediaStreamTrackInterface {
218 public:
219 // TODO(xians): Figure out if the following interface should be const or not.
220 virtual AudioSourceInterface* GetSource() const = 0;
221
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000222 // Add/Remove a sink that will receive the audio data from the track.
223 virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
224 virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0;
mallinath@webrtc.org67ee6b92014-02-03 16:57:16 +0000225
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000226 // Get the signal level from the audio track.
227 // Return true on success, otherwise false.
228 // TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
229 // after Chrome has the correct implementation of the interface.
230 virtual bool GetSignalLevel(int* level) { return false; }
231
232 // Get the audio processor used by the audio track. Return NULL if the track
233 // does not have any processor.
234 // TODO(xians): Make the interface pure virtual.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000235 virtual rtc::scoped_refptr<AudioProcessorInterface>
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000236 GetAudioProcessor() { return NULL; }
henrike@webrtc.org40b3b682014-03-03 18:30:11 +0000237
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000238 protected:
239 virtual ~AudioTrackInterface() {}
240};
241
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000242typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> >
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000243 AudioTrackVector;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000244typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000245 VideoTrackVector;
246
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000247class MediaStreamInterface : public rtc::RefCountInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000248 public NotifierInterface {
249 public:
250 virtual std::string label() const = 0;
251
252 virtual AudioTrackVector GetAudioTracks() = 0;
253 virtual VideoTrackVector GetVideoTracks() = 0;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000254 virtual rtc::scoped_refptr<AudioTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000255 FindAudioTrack(const std::string& track_id) = 0;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000256 virtual rtc::scoped_refptr<VideoTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000257 FindVideoTrack(const std::string& track_id) = 0;
258
259 virtual bool AddTrack(AudioTrackInterface* track) = 0;
260 virtual bool AddTrack(VideoTrackInterface* track) = 0;
261 virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
262 virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
263
264 protected:
265 virtual ~MediaStreamInterface() {}
266};
267
268} // namespace webrtc
269
270#endif // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_