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Side by Side Diff: webrtc/api/call/audio_send_stream.h

Issue 2392883002: Multi frequency DTMF support - sender side (Closed)
Patch Set: rebase Created 4 years, 1 month ago
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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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120 }; 120 };
121 121
122 // Starts stream activity. 122 // Starts stream activity.
123 // When a stream is active, it can receive, process and deliver packets. 123 // When a stream is active, it can receive, process and deliver packets.
124 virtual void Start() = 0; 124 virtual void Start() = 0;
125 // Stops stream activity. 125 // Stops stream activity.
126 // When a stream is stopped, it can't receive, process or deliver packets. 126 // When a stream is stopped, it can't receive, process or deliver packets.
127 virtual void Stop() = 0; 127 virtual void Stop() = 0;
128 128
129 // TODO(solenberg): Make payload_type a config property instead. 129 // TODO(solenberg): Make payload_type a config property instead.
130 virtual bool SendTelephoneEvent(int payload_type, int event, 130 virtual bool SendTelephoneEvent(int payload_type, int payload_frequency,
131 int duration_ms) = 0; 131 int event, int duration_ms) = 0;
132 132
133 virtual void SetMuted(bool muted) = 0; 133 virtual void SetMuted(bool muted) = 0;
134 134
135 virtual Stats GetStats() const = 0; 135 virtual Stats GetStats() const = 0;
136 136
137 protected: 137 protected:
138 virtual ~AudioSendStream() {} 138 virtual ~AudioSendStream() {}
139 }; 139 };
140 } // namespace webrtc 140 } // namespace webrtc
141 141
142 #endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ 142 #endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
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