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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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| 93 // Starts stream activity. | 93 // Starts stream activity. |
| 94 // When a stream is active, it can receive, process and deliver packets. | 94 // When a stream is active, it can receive, process and deliver packets. |
| 95 virtual void Start() = 0; | 95 virtual void Start() = 0; |
| 96 // Stops stream activity. | 96 // Stops stream activity. |
| 97 // When a stream is stopped, it can't receive, process or deliver packets. | 97 // When a stream is stopped, it can't receive, process or deliver packets. |
| 98 virtual void Stop() = 0; | 98 virtual void Stop() = 0; |
| 99 | 99 |
| 100 // TODO(solenberg): Make payload_type a config property instead. | 100 // TODO(solenberg): Make payload_type a config property instead. |
| 101 virtual bool SendTelephoneEvent(int payload_type, int event, | 101 virtual bool SendTelephoneEvent(int payload_type, int event, |
| 102 int duration_ms) = 0; | 102 int duration_ms) = 0; |
| 103 |
| 104 virtual void SetMuted(bool muted) = 0; |
| 105 |
| 103 virtual Stats GetStats() const = 0; | 106 virtual Stats GetStats() const = 0; |
| 104 | 107 |
| 105 protected: | 108 protected: |
| 106 virtual ~AudioSendStream() {} | 109 virtual ~AudioSendStream() {} |
| 107 }; | 110 }; |
| 108 } // namespace webrtc | 111 } // namespace webrtc |
| 109 | 112 |
| 110 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ | 113 #endif // WEBRTC_AUDIO_SEND_STREAM_H_ |
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