/* * libjingle * Copyright 2012, Google Inc. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions are met: * * 1. Redistributions of source code must retain the above copyright notice, * this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright notice, * this list of conditions and the following disclaimer in the documentation * and/or other materials provided with the distribution. * 3. The name of the author may not be used to endorse or promote products * derived from this software without specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ #ifndef TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_ #define TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_ #include #include "talk/app/webrtc/mediastreaminterface.h" #include "webrtc/base/common.h" #include "webrtc/base/refcount.h" // This file contains interfaces for DtmfSender. namespace webrtc { // DtmfSender callback interface. Application should implement this interface // to get notifications from the DtmfSender. class DtmfSenderObserverInterface { public: // Triggered when DTMF |tone| is sent. // If |tone| is empty that means the DtmfSender has sent out all the given // tones. virtual void OnToneChange(const std::string& tone) = 0; protected: virtual ~DtmfSenderObserverInterface() {} }; // The interface of native implementation of the RTCDTMFSender defined by the // WebRTC W3C Editor's Draft. class DtmfSenderInterface : public rtc::RefCountInterface { public: virtual void RegisterObserver(DtmfSenderObserverInterface* observer) = 0; virtual void UnregisterObserver() = 0; // Returns true if this DtmfSender is capable of sending DTMF. // Otherwise returns false. virtual bool CanInsertDtmf() = 0; // Queues a task that sends the DTMF |tones|. The |tones| parameter is treated // as a series of characters. The characters 0 through 9, A through D, #, and // * generate the associated DTMF tones. The characters a to d are equivalent // to A to D. The character ',' indicates a delay of 2 seconds before // processing the next character in the tones parameter. // Unrecognized characters are ignored. // The |duration| parameter indicates the duration in ms to use for each // character passed in the |tones| parameter. // The duration cannot be more than 6000 or less than 70. // The |inter_tone_gap| parameter indicates the gap between tones in ms. // The |inter_tone_gap| must be at least 50 ms but should be as short as // possible. // If InsertDtmf is called on the same object while an existing task for this // object to generate DTMF is still running, the previous task is canceled. // Returns true on success and false on failure. virtual bool InsertDtmf(const std::string& tones, int duration, int inter_tone_gap) = 0; // Returns the track given as argument to the constructor. virtual const AudioTrackInterface* track() const = 0; // Returns the tones remaining to be played out. virtual std::string tones() const = 0; // Returns the current tone duration value in ms. // This value will be the value last set via the InsertDtmf() method, or the // default value of 100 ms if InsertDtmf() was never called. virtual int duration() const = 0; // Returns the current value of the between-tone gap in ms. // This value will be the value last set via the InsertDtmf() method, or the // default value of 50 ms if InsertDtmf() was never called. virtual int inter_tone_gap() const = 0; protected: virtual ~DtmfSenderInterface() {} }; } // namespace webrtc #endif // TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_