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/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef API_DTMFSENDERINTERFACE_H_
#define API_DTMFSENDERINTERFACE_H_
#include <string>
#include "api/mediastreaminterface.h"
#include "rtc_base/refcount.h"
namespace webrtc {
// DtmfSender callback interface, used to implement RTCDtmfSender events.
// Applications 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.
// See: https://www.w3.org/TR/webrtc/#peer-to-peer-dtmf
class DtmfSenderInterface : public rtc::RefCountInterface {
public:
// Used to receive events from the DTMF sender. Only one observer can be
// registered at a time. UnregisterObserver should be called before the
// observer object is destroyed.
virtual void RegisterObserver(DtmfSenderObserverInterface* observer) = 0;
virtual void UnregisterObserver() = 0;
// Returns true if this DtmfSender is capable of sending DTMF. Otherwise
// returns false. To be able to send DTMF, the associated RtpSender must be
// able to send packets, and a "telephone-event" codec must be negotiated.
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. Only exists for
// backwards compatibilty; now that DtmfSenders are tied to RtpSenders, it's
// no longer relevant.
// TODO(bugs.webrtc.org/9426): Remove this method.
virtual const AudioTrackInterface* track() const { return nullptr; }
// 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 // API_DTMFSENDERINTERFACE_H_