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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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253 // Add/Remove a sink that will receive the audio data from the track. | 253 // Add/Remove a sink that will receive the audio data from the track. |
254 virtual void AddSink(AudioTrackSinkInterface* sink) = 0; | 254 virtual void AddSink(AudioTrackSinkInterface* sink) = 0; |
255 virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0; | 255 virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0; |
256 | 256 |
257 // Get the signal level from the audio track. | 257 // Get the signal level from the audio track. |
258 // Return true on success, otherwise false. | 258 // Return true on success, otherwise false. |
259 // TODO(deadbeef): Change the interface to int GetSignalLevel() and pure | 259 // TODO(deadbeef): Change the interface to int GetSignalLevel() and pure |
260 // virtual after it's implemented in chromium. | 260 // virtual after it's implemented in chromium. |
261 virtual bool GetSignalLevel(int* level) { return false; } | 261 virtual bool GetSignalLevel(int* level) { return false; } |
262 | 262 |
263 // Get the audio processor used by the audio track. Return NULL if the track | 263 // Get the audio processor used by the audio track. Return null if the track |
264 // does not have any processor. | 264 // does not have any processor. |
265 // TODO(deadbeef): Make the interface pure virtual. | 265 // TODO(deadbeef): Make the interface pure virtual. |
266 virtual rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() { | 266 virtual rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() { |
267 return nullptr; | 267 return nullptr; |
268 } | 268 } |
269 | 269 |
270 protected: | 270 protected: |
271 virtual ~AudioTrackInterface() {} | 271 virtual ~AudioTrackInterface() {} |
272 }; | 272 }; |
273 | 273 |
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301 virtual bool RemoveTrack(AudioTrackInterface* track) = 0; | 301 virtual bool RemoveTrack(AudioTrackInterface* track) = 0; |
302 virtual bool RemoveTrack(VideoTrackInterface* track) = 0; | 302 virtual bool RemoveTrack(VideoTrackInterface* track) = 0; |
303 | 303 |
304 protected: | 304 protected: |
305 virtual ~MediaStreamInterface() {} | 305 virtual ~MediaStreamInterface() {} |
306 }; | 306 }; |
307 | 307 |
308 } // namespace webrtc | 308 } // namespace webrtc |
309 | 309 |
310 #endif // WEBRTC_API_MEDIASTREAMINTERFACE_H_ | 310 #endif // WEBRTC_API_MEDIASTREAMINTERFACE_H_ |
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