Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(260)

Side by Side Diff: talk/app/webrtc/mediastreaminterface.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updated location for peerconnection_unittests.isolate Created 4 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
(Empty)
1 /*
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 "webrtc/base/basictypes.h"
41 #include "webrtc/base/refcount.h"
42 #include "webrtc/base/scoped_ref_ptr.h"
43
44 namespace cricket {
45
46 class AudioRenderer;
47 class VideoCapturer;
48 class VideoRenderer;
49 class VideoFrame;
50
51 } // namespace cricket
52
53 namespace webrtc {
54
55 // Generic observer interface.
56 class ObserverInterface {
57 public:
58 virtual void OnChanged() = 0;
59
60 protected:
61 virtual ~ObserverInterface() {}
62 };
63
64 class 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 class MediaSourceInterface : public rtc::RefCountInterface,
75 public NotifierInterface {
76 public:
77 enum SourceState {
78 kInitializing,
79 kLive,
80 kEnded,
81 kMuted
82 };
83
84 virtual SourceState state() const = 0;
85
86 virtual bool remote() const = 0;
87
88 protected:
89 virtual ~MediaSourceInterface() {}
90 };
91
92 // Information about a track.
93 class MediaStreamTrackInterface : public rtc::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 static const char kAudioKind[];
104 static const char kVideoKind[];
105
106 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;
113
114 protected:
115 virtual ~MediaStreamTrackInterface() {}
116 };
117
118 // Interface for rendering VideoFrames from a VideoTrack
119 class VideoRendererInterface {
120 public:
121 // |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.
124 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
133 class VideoSourceInterface;
134
135 class 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
142 virtual VideoSourceInterface* GetSource() const = 0;
143
144 protected:
145 virtual ~VideoTrackInterface() {}
146 };
147
148 // Interface for receiving audio data from a AudioTrack.
149 class AudioTrackSinkInterface {
150 public:
151 virtual void OnData(const void* audio_data,
152 int bits_per_sample,
153 int sample_rate,
154 size_t number_of_channels,
155 size_t number_of_frames) = 0;
156
157 protected:
158 virtual ~AudioTrackSinkInterface() {}
159 };
160
161 // AudioSourceInterface is a reference counted source used for AudioTracks.
162 // The same source can be used in multiple AudioTracks.
163 class AudioSourceInterface : public MediaSourceInterface {
164 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].
176 // 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.
178 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) {}
183
184 // TODO(tommi): Make pure virtual.
185 virtual void AddSink(AudioTrackSinkInterface* sink) {}
186 virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
187 };
188
189 // Interface of the audio processor used by the audio track to collect
190 // statistics.
191 class AudioProcessorInterface : public rtc::RefCountInterface {
192 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
217 class 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
222 // 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;
225
226 // 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.
235 virtual rtc::scoped_refptr<AudioProcessorInterface>
236 GetAudioProcessor() { return NULL; }
237
238 // Get a pointer to the audio renderer of this AudioTrack.
239 // The pointer is valid for the lifetime of this AudioTrack.
240 // TODO(xians): Remove the following interface after Chrome switches to
241 // AddSink() and RemoveSink() interfaces.
242 virtual cricket::AudioRenderer* GetRenderer() { return NULL; }
243
244 protected:
245 virtual ~AudioTrackInterface() {}
246 };
247
248 typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> >
249 AudioTrackVector;
250 typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
251 VideoTrackVector;
252
253 class MediaStreamInterface : public rtc::RefCountInterface,
254 public NotifierInterface {
255 public:
256 virtual std::string label() const = 0;
257
258 virtual AudioTrackVector GetAudioTracks() = 0;
259 virtual VideoTrackVector GetVideoTracks() = 0;
260 virtual rtc::scoped_refptr<AudioTrackInterface>
261 FindAudioTrack(const std::string& track_id) = 0;
262 virtual rtc::scoped_refptr<VideoTrackInterface>
263 FindVideoTrack(const std::string& track_id) = 0;
264
265 virtual bool AddTrack(AudioTrackInterface* track) = 0;
266 virtual bool AddTrack(VideoTrackInterface* track) = 0;
267 virtual bool RemoveTrack(AudioTrackInterface* track) = 0;
268 virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
269
270 protected:
271 virtual ~MediaStreamInterface() {}
272 };
273
274 } // namespace webrtc
275
276 #endif // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698