d4e598d57a
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
139 lines
4.8 KiB
C++
139 lines
4.8 KiB
C++
/*
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* libjingle
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* Copyright 2012, Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_APP_WEBRTC_DTMFSENDER_H_
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#define TALK_APP_WEBRTC_DTMFSENDER_H_
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#include <string>
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#include "talk/app/webrtc/dtmfsenderinterface.h"
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#include "talk/app/webrtc/mediastreaminterface.h"
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#include "talk/app/webrtc/proxy.h"
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#include "webrtc/base/common.h"
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#include "webrtc/base/messagehandler.h"
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#include "webrtc/base/refcount.h"
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// DtmfSender is the native implementation of the RTCDTMFSender defined by
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// the WebRTC W3C Editor's Draft.
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// http://dev.w3.org/2011/webrtc/editor/webrtc.html
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namespace rtc {
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class Thread;
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}
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namespace webrtc {
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// This interface is called by DtmfSender to talk to the actual audio channel
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// to send DTMF.
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class DtmfProviderInterface {
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public:
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// Returns true if the audio track with given id (|track_id|) is capable
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// of sending DTMF. Otherwise returns false.
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virtual bool CanInsertDtmf(const std::string& track_id) = 0;
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// Sends DTMF |code| via the audio track with given id (|track_id|).
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// The |duration| indicates the length of the DTMF tone in ms.
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// Returns true on success and false on failure.
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virtual bool InsertDtmf(const std::string& track_id,
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int code, int duration) = 0;
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// Returns a |sigslot::signal0<>| signal. The signal should fire before
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// the provider is destroyed.
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virtual sigslot::signal0<>* GetOnDestroyedSignal() = 0;
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protected:
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virtual ~DtmfProviderInterface() {}
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};
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class DtmfSender
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: public DtmfSenderInterface,
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public sigslot::has_slots<>,
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public rtc::MessageHandler {
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public:
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static rtc::scoped_refptr<DtmfSender> Create(
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AudioTrackInterface* track,
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rtc::Thread* signaling_thread,
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DtmfProviderInterface* provider);
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// Implements DtmfSenderInterface.
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virtual void RegisterObserver(DtmfSenderObserverInterface* observer) OVERRIDE;
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virtual void UnregisterObserver() OVERRIDE;
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virtual bool CanInsertDtmf() OVERRIDE;
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virtual bool InsertDtmf(const std::string& tones, int duration,
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int inter_tone_gap) OVERRIDE;
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virtual const AudioTrackInterface* track() const OVERRIDE;
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virtual std::string tones() const OVERRIDE;
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virtual int duration() const OVERRIDE;
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virtual int inter_tone_gap() const OVERRIDE;
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protected:
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DtmfSender(AudioTrackInterface* track,
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rtc::Thread* signaling_thread,
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DtmfProviderInterface* provider);
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virtual ~DtmfSender();
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private:
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DtmfSender();
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// Implements MessageHandler.
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virtual void OnMessage(rtc::Message* msg);
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// The DTMF sending task.
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void DoInsertDtmf();
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void OnProviderDestroyed();
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void StopSending();
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rtc::scoped_refptr<AudioTrackInterface> track_;
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DtmfSenderObserverInterface* observer_;
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rtc::Thread* signaling_thread_;
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DtmfProviderInterface* provider_;
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std::string tones_;
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int duration_;
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int inter_tone_gap_;
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DISALLOW_COPY_AND_ASSIGN(DtmfSender);
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};
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// Define proxy for DtmfSenderInterface.
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BEGIN_PROXY_MAP(DtmfSender)
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PROXY_METHOD1(void, RegisterObserver, DtmfSenderObserverInterface*)
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PROXY_METHOD0(void, UnregisterObserver)
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PROXY_METHOD0(bool, CanInsertDtmf)
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PROXY_METHOD3(bool, InsertDtmf, const std::string&, int, int)
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PROXY_CONSTMETHOD0(const AudioTrackInterface*, track)
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PROXY_CONSTMETHOD0(std::string, tones)
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PROXY_CONSTMETHOD0(int, duration)
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PROXY_CONSTMETHOD0(int, inter_tone_gap)
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END_PROXY()
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// Get DTMF code from the DTMF event character.
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bool GetDtmfCode(char tone, int* code);
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} // namespace webrtc
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#endif // TALK_APP_WEBRTC_DTMFSENDER_H_
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