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1 /* | |
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_DTMFSENDER_H_ | |
29 #define TALK_APP_WEBRTC_DTMFSENDER_H_ | |
30 | |
31 #include <string> | |
32 | |
33 #include "talk/app/webrtc/dtmfsenderinterface.h" | |
34 #include "talk/app/webrtc/mediastreaminterface.h" | |
35 #include "talk/app/webrtc/proxy.h" | |
36 #include "webrtc/base/common.h" | |
37 #include "webrtc/base/messagehandler.h" | |
38 #include "webrtc/base/refcount.h" | |
39 | |
40 // DtmfSender is the native implementation of the RTCDTMFSender defined by | |
41 // the WebRTC W3C Editor's Draft. | |
42 // http://dev.w3.org/2011/webrtc/editor/webrtc.html | |
43 | |
44 namespace rtc { | |
45 class Thread; | |
46 } | |
47 | |
48 namespace webrtc { | |
49 | |
50 // This interface is called by DtmfSender to talk to the actual audio channel | |
51 // to send DTMF. | |
52 class DtmfProviderInterface { | |
53 public: | |
54 // Returns true if the audio track with given id (|track_id|) is capable | |
55 // of sending DTMF. Otherwise returns false. | |
56 virtual bool CanInsertDtmf(const std::string& track_id) = 0; | |
57 // Sends DTMF |code| via the audio track with given id (|track_id|). | |
58 // The |duration| indicates the length of the DTMF tone in ms. | |
59 // Returns true on success and false on failure. | |
60 virtual bool InsertDtmf(const std::string& track_id, | |
61 int code, int duration) = 0; | |
62 // Returns a |sigslot::signal0<>| signal. The signal should fire before | |
63 // the provider is destroyed. | |
64 virtual sigslot::signal0<>* GetOnDestroyedSignal() = 0; | |
65 | |
66 protected: | |
67 virtual ~DtmfProviderInterface() {} | |
68 }; | |
69 | |
70 class DtmfSender | |
71 : public DtmfSenderInterface, | |
72 public sigslot::has_slots<>, | |
73 public rtc::MessageHandler { | |
74 public: | |
75 static rtc::scoped_refptr<DtmfSender> Create( | |
76 AudioTrackInterface* track, | |
77 rtc::Thread* signaling_thread, | |
78 DtmfProviderInterface* provider); | |
79 | |
80 // Implements DtmfSenderInterface. | |
81 void RegisterObserver(DtmfSenderObserverInterface* observer) override; | |
82 void UnregisterObserver() override; | |
83 bool CanInsertDtmf() override; | |
84 bool InsertDtmf(const std::string& tones, | |
85 int duration, | |
86 int inter_tone_gap) override; | |
87 const AudioTrackInterface* track() const override; | |
88 std::string tones() const override; | |
89 int duration() const override; | |
90 int inter_tone_gap() const override; | |
91 | |
92 protected: | |
93 DtmfSender(AudioTrackInterface* track, | |
94 rtc::Thread* signaling_thread, | |
95 DtmfProviderInterface* provider); | |
96 virtual ~DtmfSender(); | |
97 | |
98 private: | |
99 DtmfSender(); | |
100 | |
101 // Implements MessageHandler. | |
102 virtual void OnMessage(rtc::Message* msg); | |
103 | |
104 // The DTMF sending task. | |
105 void DoInsertDtmf(); | |
106 | |
107 void OnProviderDestroyed(); | |
108 | |
109 void StopSending(); | |
110 | |
111 rtc::scoped_refptr<AudioTrackInterface> track_; | |
112 DtmfSenderObserverInterface* observer_; | |
113 rtc::Thread* signaling_thread_; | |
114 DtmfProviderInterface* provider_; | |
115 std::string tones_; | |
116 int duration_; | |
117 int inter_tone_gap_; | |
118 | |
119 RTC_DISALLOW_COPY_AND_ASSIGN(DtmfSender); | |
120 }; | |
121 | |
122 // Define proxy for DtmfSenderInterface. | |
123 BEGIN_PROXY_MAP(DtmfSender) | |
124 PROXY_METHOD1(void, RegisterObserver, DtmfSenderObserverInterface*) | |
125 PROXY_METHOD0(void, UnregisterObserver) | |
126 PROXY_METHOD0(bool, CanInsertDtmf) | |
127 PROXY_METHOD3(bool, InsertDtmf, const std::string&, int, int) | |
128 PROXY_CONSTMETHOD0(const AudioTrackInterface*, track) | |
129 PROXY_CONSTMETHOD0(std::string, tones) | |
130 PROXY_CONSTMETHOD0(int, duration) | |
131 PROXY_CONSTMETHOD0(int, inter_tone_gap) | |
132 END_PROXY() | |
133 | |
134 // Get DTMF code from the DTMF event character. | |
135 bool GetDtmfCode(char tone, int* code); | |
136 | |
137 } // namespace webrtc | |
138 | |
139 #endif // TALK_APP_WEBRTC_DTMFSENDER_H_ | |
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