| /* |
| * 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 <string> |
| |
| #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_ |