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📄 mediahandler.h

📁 MiniSip Client with DomainKeys Authentication, Sip, Audio communications, Echo Cancel
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/* Copyright (C) 2004-2006 the Minisip Team  This library is free software; you can redistribute it and/or modify it under the terms of the GNU Lesser General Public License as published by the Free Software Foundation; either version 2.1 of the License, or (at your option) any later version.  This library is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for more details.  You should have received a copy of the GNU Lesser General Public License along with this library; if not, write to the Free Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA *//* Copyright (C) 2004  * * Authors: Erik Eliasson <eliasson@it.kth.se> *          Johan Bilien <jobi@via.ecp.fr>*/#ifndef MEDIA_HANDLER_H#define MEDIA_HANDLER_H#include<libminisip/libminisip_config.h>#include<libmutil/MemObject.h>#include<libminisip/mediahandler/Media.h>#include<libminisip/mediahandler/AudioMedia.h>#include<libminisip/mediahandler/MediaStream.h>#include<libminisip/mediahandler/Session.h>#include<libminisip/mediahandler/RtpReceiver.h>#include<libminisip/ipprovider/IpProvider.h>#include<libmutil/CommandString.h>#include<libmutil/MessageRouter.h>class SipSoftPhoneConfiguration;class IpProvider;class LIBMINISIP_API MediaHandler : public virtual MObject, public SessionRegistry, public CommandReceiver {	public:		/**		 * Constructor, created on startup		 * @param config reference to the softphone configuration		 * @param ipProvider reference to the public IP provider, used		 * for NAT traversal mechanisms		 */		MediaHandler( MRef<SipSoftPhoneConfiguration *> config, MRef<IpProvider *> ipProvider );		virtual ~MediaHandler();				/**		 * Creates a new media session, for use in a new VoIP call		 * @param config the call specific configuration		 * @param callId identifier shared with the SIP stack		 * @returns a reference to the session created		 */		MRef<Session *> createSession( SipDialogSecurityConfig &config, SipCommonConfig &common, std::string callId = "" );				/**		 * Registers a new media type (audio or video		 * @param media a reference to the representation of the		 * medium to add		 */		void registerMedia( MRef<Media *> media );		/**		 * Handles a command sent by the user interface		 * @param command the command to handle		 */		void handleCommand(std::string subsystem, const CommandString & command );		CommandString handleCommandResp(std::string subsystem, const CommandString& command);				/**		 * Provides the IP address given as contact to the external		 * peers		 * @returns a string containing the IP address		 */		std::string getExtIP();						virtual std::string getMemObjectType(){return "MediaHandler";}#ifdef DEBUG_OUTPUT			virtual std::string getDebugString();#endif#if 0		/**		True if all but one sender/media sessions are muted.		If turned off, all ongoing sessions receive the audio from our mic.		If turned on, only one source (the active one) will receive it.		*/		bool muteAllButOne;#endif	private:		void init();		std::list< MRef<Media *> > media;		std::string ringtoneFile;		MRef<AudioMedia *> audioMedia;		MRef<IpProvider *> ipProvider;		MRef<SipSoftPhoneConfiguration *> config;				/**		Looks for a Session with callid. If found, set the audio settings		of that Session (side = "receivers" or "senders") to either be on		or off.		@param callid callid of the Session whose settings we want to deal with		@param side either "receivers" (audio received from the network and heard			on the headsets) or "senders" (audio read from the mic/soundcard and			sent over the net).		@param turnOn whether to turn on or off the setting of the specified session.			If turnOn = true -> the receivers are actived (un-silenced) or the			are activated (un-muted).		*/		void setSessionSoundSettings( std::string callid, std::string side, bool turnOn );};#endif

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