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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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