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/*
* 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_
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