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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2012, Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
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 */
27
28#ifndef TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_
29#define TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_
30
31#include <string>
32
33#include "talk/app/webrtc/mediastreaminterface.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000034#include "webrtc/base/common.h"
35#include "webrtc/base/refcount.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036
37// This file contains interfaces for DtmfSender.
38
39namespace webrtc {
40
41// DtmfSender callback interface. Application should implement this interface
42// to get notifications from the DtmfSender.
43class DtmfSenderObserverInterface {
44 public:
45 // Triggered when DTMF |tone| is sent.
46 // If |tone| is empty that means the DtmfSender has sent out all the given
47 // tones.
48 virtual void OnToneChange(const std::string& tone) = 0;
49
50 protected:
51 virtual ~DtmfSenderObserverInterface() {}
52};
53
54// The interface of native implementation of the RTCDTMFSender defined by the
55// WebRTC W3C Editor's Draft.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000056class DtmfSenderInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000057 public:
58 virtual void RegisterObserver(DtmfSenderObserverInterface* observer) = 0;
59 virtual void UnregisterObserver() = 0;
60
61 // Returns true if this DtmfSender is capable of sending DTMF.
62 // Otherwise returns false.
63 virtual bool CanInsertDtmf() = 0;
64
65 // Queues a task that sends the DTMF |tones|. The |tones| parameter is treated
66 // as a series of characters. The characters 0 through 9, A through D, #, and
67 // * generate the associated DTMF tones. The characters a to d are equivalent
68 // to A to D. The character ',' indicates a delay of 2 seconds before
69 // processing the next character in the tones parameter.
70 // Unrecognized characters are ignored.
71 // The |duration| parameter indicates the duration in ms to use for each
72 // character passed in the |tones| parameter.
73 // The duration cannot be more than 6000 or less than 70.
74 // The |inter_tone_gap| parameter indicates the gap between tones in ms.
75 // The |inter_tone_gap| must be at least 50 ms but should be as short as
76 // possible.
77 // If InsertDtmf is called on the same object while an existing task for this
78 // object to generate DTMF is still running, the previous task is canceled.
79 // Returns true on success and false on failure.
80 virtual bool InsertDtmf(const std::string& tones, int duration,
81 int inter_tone_gap) = 0;
82
83 // Returns the track given as argument to the constructor.
84 virtual const AudioTrackInterface* track() const = 0;
85
86 // Returns the tones remaining to be played out.
87 virtual std::string tones() const = 0;
88
89 // Returns the current tone duration value in ms.
90 // This value will be the value last set via the InsertDtmf() method, or the
91 // default value of 100 ms if InsertDtmf() was never called.
92 virtual int duration() const = 0;
93
94 // Returns the current value of the between-tone gap in ms.
95 // 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.
97 virtual int inter_tone_gap() const = 0;
98
99 protected:
100 virtual ~DtmfSenderInterface() {}
101};
102
103} // namespace webrtc
104
105#endif // TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_