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_ 277