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: ~DtmfProviderInterface()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_ 140