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Side by Side Diff: webrtc/api/dtmfsenderinterface.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updated location for peerconnection_unittests.isolate Created 4 years, 11 months ago
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1 /* 1 /*
2 * libjingle 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * Copyright 2012 Google Inc.
4 * 3 *
5 * Redistribution and use in source and binary forms, with or without 4 * Use of this source code is governed by a BSD-style license
6 * modification, are permitted provided that the following conditions are met: 5 * that can be found in the LICENSE file in the root of the source
7 * 6 * tree. An additional intellectual property rights grant can be found
8 * 1. Redistributions of source code must retain the above copyright notice, 7 * in the file PATENTS. All contributing project authors may
9 * this list of conditions and the following disclaimer. 8 * be found in the AUTHORS file in the root of the source tree.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 9 */
27 10
28 #ifndef TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_ 11 #ifndef WEBRTC_API_DTMFSENDERINTERFACE_H_
29 #define TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_ 12 #define WEBRTC_API_DTMFSENDERINTERFACE_H_
30 13
31 #include <string> 14 #include <string>
32 15
33 #include "talk/app/webrtc/mediastreaminterface.h" 16 #include "webrtc/api/mediastreaminterface.h"
34 #include "webrtc/base/common.h" 17 #include "webrtc/base/common.h"
35 #include "webrtc/base/refcount.h" 18 #include "webrtc/base/refcount.h"
36 19
37 // This file contains interfaces for DtmfSender. 20 // This file contains interfaces for DtmfSender.
38 21
39 namespace webrtc { 22 namespace webrtc {
40 23
41 // DtmfSender callback interface. Application should implement this interface 24 // DtmfSender callback interface. Application should implement this interface
42 // to get notifications from the DtmfSender. 25 // to get notifications from the DtmfSender.
43 class DtmfSenderObserverInterface { 26 class DtmfSenderObserverInterface {
(...skipping 51 matching lines...) Expand 10 before | Expand all | Expand 10 after
95 // This value will be the value last set via the InsertDtmf() method, or the 78 // This value will be the value last set via the InsertDtmf() method, or the
96 // default value of 50 ms if InsertDtmf() was never called. 79 // default value of 50 ms if InsertDtmf() was never called.
97 virtual int inter_tone_gap() const = 0; 80 virtual int inter_tone_gap() const = 0;
98 81
99 protected: 82 protected:
100 virtual ~DtmfSenderInterface() {} 83 virtual ~DtmfSenderInterface() {}
101 }; 84 };
102 85
103 } // namespace webrtc 86 } // namespace webrtc
104 87
105 #endif // TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_ 88 #endif // WEBRTC_API_DTMFSENDERINTERFACE_H_
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