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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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105 | 105 |
106 // TODO(pbos): Remove config option once combined A/V BWE is always on. | 106 // TODO(pbos): Remove config option once combined A/V BWE is always on. |
107 bool combined_audio_video_bwe = false; | 107 bool combined_audio_video_bwe = false; |
108 }; | 108 }; |
109 | 109 |
110 virtual Stats GetStats() const = 0; | 110 virtual Stats GetStats() const = 0; |
111 | 111 |
112 // Sets an audio sink that receives unmixed audio from the receive stream. | 112 // Sets an audio sink that receives unmixed audio from the receive stream. |
113 // Ownership of the sink is passed to the stream and can be used by the | 113 // Ownership of the sink is passed to the stream and can be used by the |
114 // caller to do lifetime management (i.e. when the sink's dtor is called). | 114 // caller to do lifetime management (i.e. when the sink's dtor is called). |
115 // Only one sink can be set and passing a null sink, clears an existing one. | 115 // Only one sink can be set and passing a null sink clears an existing one. |
116 // NOTE: Audio must still somehow be pulled through AudioTransport for audio | 116 // NOTE: Audio must still somehow be pulled through AudioTransport for audio |
117 // to stream through this sink. In practice, this happens if mixed audio | 117 // to stream through this sink. In practice, this happens if mixed audio |
118 // is being pulled+rendered and/or if audio is being pulled for the purposes | 118 // is being pulled+rendered and/or if audio is being pulled for the purposes |
119 // of feeding to the AEC. | 119 // of feeding to the AEC. |
120 virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0; | 120 virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0; |
121 }; | 121 }; |
122 } // namespace webrtc | 122 } // namespace webrtc |
123 | 123 |
124 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ | 124 #endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ |
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