callaudio.cpp

KphoneSI (kpsi) is a SIP (Session Initiation Protocol) user agent for Linux, with which you can in

//
// C++ Implementation: callaudio2
//
// Description:
//
//
// Author: root <root@papa>, (C) 2004
//
// Copyright: See COPYING file that comes with this distribution
//
//
#include <stdio.h>
#include <sys/types.h>
#include <unistd.h>
#include <signal.h>
#include <math.h>
#include <sys/ioctl.h>
#include "../config.h"
#include "../dissipate2/sipprotocol.h"
#include "../dissipate2/sipcall.h"
#include "../dissipate2/sdpbuild.h"
#include "kstatics.h"
#include "dspaudioout.h"
#include "dspaudioin.h"
#include "dspoutrtp.h"
#include "dspoutoss.h"
#ifdef ALSA_SUPPORT
#include "dspoutalsa.h"
#endif

#include "callaudio.h"

CallAudio::CallAudio(SessionControl *sessionC, const char *name )
{
	sc = sessionC;
	outrtp = 0;
	inrtp = 0;
	inoss = 0;
	outoss = 0;
	inalsa = 0;
	outalsa = 0;
	output = 0;
	input = 0;
	useoss = true;
	payload = sc->getPayload();
	useStun = false;
	audio_fd = -1;
	charDTMF[0]='0';
	charDTMF[1]='1';
	charDTMF[2]='2';
	charDTMF[3]='3';
	charDTMF[4]='4';
	charDTMF[5]='5';
	charDTMF[6]='6';
	charDTMF[7]='7';
	charDTMF[8]='8';
	charDTMF[9]='9';
	charDTMF[10]='*';
	charDTMF[11]='#';
	dtmfCount = 0;
	dtmfSize  = 0;
	dtmfTimer = new QTimer();
	connect( dtmfTimer, SIGNAL( timeout() ),
		this, SLOT( handleDTMFSeq() ) );
	debug = KStatics::debugLevel;
	audioUser=0;
	printf(":::::: Callaudio start\n");

}


CallAudio::~CallAudio()
{

if( output ) {
		if( output->running() ) {
			output->setCancel();
			output->wait();
		}
		delete output;
	}
	if( input ) {
		if( input->running() ) {
			input->setCancel();
			input->wait();
		}
		delete input;
	};
if(dtmfTimer) {
		delete dtmfTimer;
		dtmfTimer=0;
}
	printf(":::::: Callaudio end\n");

}




void CallAudio::audioIn( SdpBuild *call, int aU )
{
	bool ok;
	audioUser=aU;
	int min, max;
	min = sc->getMinMediaPort();
	max = sc->getMaxMediaPort();
	if(debug >= 2) printf("=====CallAudio audioIn: %d\n",(int) sc->getAudioSys() );
	QString hostname = call->getExtHost();
	unsigned int portnum = call->getAudioPort().toInt( &ok, 10 );;
	if(debug >= 2) printf( "=====CallAudio audioIn: Sending to remote site %s:%d\n", hostname.latin1(), portnum );
	if( input ) {
		stopListeningAudio(aU);
	}
	if( sc->getAudioSys() == SessionControl::isOSS ) {
		if(debug >= 2) printf("=====CallAudio audioIn: OSS used: codec = %d payload size %d\n",call->getRtpCodecNum(),call->getResponsePay());
		if (sc->getSymMediaMode()) {
			outrtp = new DspOutRtp(call->getRtpCodec(), call->getRtpCodecNum(), min, max, hostname, &socket );
		if(debug >= 2) printf( "=====CallAudio  audioIn:  sym media\n" );
		} else {
			outrtp = new DspOutRtp( call->getRtpCodec(), call->getRtpCodecNum(), min, max, hostname );
		}
		outrtp->setPortNum( portnum);
		outrtp->openDevice( DspOut::WriteOnly );
		outrtp->setPayload( call->getResponsePay() );
		if(debug >= 2) printf( "=====CallAudio audioIn: RTP %s:%d %d\n", hostname.latin1(), portnum,outrtp->getPortNum() );

		inoss = new DspOutOss( sc->getOSSFilename2() );

		if( sc->isAudioRW()) {
			if(debug >= 2) printf( "=====CallAudio audioIn: OSS device already open (readwrite)\n" );
			if( !inoss->openDevice( audio_fd ) ) {
				if(debug >= 2) printf( "=====CallAudio audioIn:  openDevice Failed.\n" );
			}
		} else {
			if(debug >= 2) printf( "=====CallAudio audioIn: Opening OSS device %s for Input \n", sc->getOSSFilename2().latin1() );
			if( !inoss->openDevice( DspOut::ReadOnly ) ) {
				if(debug >= 2) printf( "=====CallAudio: audioIn openDevice Failed.\n" );
			}
		}
		inoss->readBuffer(2);
		if(debug >= 2) printf( "=====CallAudio audioIn: Creating OSS->RTP Diverter\n" );
		input = new DspAudioIn( inoss, outrtp );
		output->setCodec(call->getRtpCodec(), call->getRtpCodecNum());
		input->start();
		output->start();

	} else if( sc->getAudioSys() == SessionControl::isALSA ) {
		if(debug >= 2) printf("=====CallAudio audioIn: ALSA used codec = %d\n",call->getRtpCodecNum());
		if (sc->getSymMediaMode()) {
			outrtp = new DspOutRtp( call->getRtpCodec(), call->getRtpCodecNum(), min, max, hostname, &socket );
		if(debug >= 2) printf( "=====CallAudio audioIn:  ALSA sym media\n" );
		} else {
			outrtp = new DspOutRtp( call->getRtpCodec(), call->getRtpCodecNum(), min, max,  hostname );
		}
		outrtp->setPortNum( portnum);
		outrtp->openDevice( DspOut::WriteOnly );
		outrtp->setPayload( call->getResponsePay() );
		inalsa = new DspOutAlsa( sc->getALSAFilename() );//("default");
		if(debug >= 2) printf( "=====CallAudio audioIn:  we open the deviced\n" );

		if( !inalsa->openDevice( DspOut::ReadOnly ) ) {
			if(debug >= 2) printf( "=====CallAudio audioIn: openDevice Failed.\n" );
		}
		inalsa->readBuffer(2);
		if(debug >= 2) printf( "=====CallAudio: audioIn: Creating ALSA->RTP Diverter\n" );
		input = new DspAudioIn( inalsa, outrtp );
		output->setCodec(call->getRtpCodec(), call->getRtpCodecNum());
		input->start();
		output->start();

	}

}

int CallAudio::audioOut( SdpBuild *call,int aU )
{
	audioUser=aU;
	useStun=sc->isStun();
	int min, max;
	min =sc-> getMinMediaPort();
	max = sc->getMaxMediaPort();
	if( output  ) {
		if(debug >= 2) printf( "=====CallAudio audioOut: stop sending audio\n" );
		stopSendingAudio(aU);
		}
	if( sc->getAudioSys() == SessionControl::isOSS )
		{
		if(debug >= 2) printf( "=====CallAudio audioOut: OSS used codec = %d payload size = %d \n",call->getRtpCodecNum(),call->getResponsePay());
		if (sc->getSymMediaMode()) {
			if(debug >= 2) printf( "=====CallAudio audioOut: sym media\n" );
			inrtp = new DspOutRtp( call->getRtpCodec(), call->getRtpCodecNum(), min, max, QString::null, &socket );
		} else {
			inrtp = new DspOutRtp( call->getRtpCodec(), call->getRtpCodecNum() , min, max );
		}
		inrtp->setPayload(call->getResponsePay() );
		if( useStun ) {
		if(debug >= 2) printf( "=====CallAudio audioOut: use stun\n" );
		inrtp->setStunSrv( sc->getStunSrv() );
		}
		inrtp->openDevice( DspOut::ReadOnly );
		int po = 0;
		po=inrtp->getPortNum();
		if(debug >= 2) printf( "=====CallAudio audioOut: RTP  %d\n", po );

		outoss = new DspOutOss( sc->getOSSFilename() );
		if( sc->isAudioRW() ) {
			if( !outoss->openDevice( DspOut::ReadWrite ) ) {
				if(debug >= 2) printf("=====CallAudio audioOut: openDevice Failed.\n" );
				po=0;
			} else {
				audio_fd = outoss->audio_fd;
			}
		} else {
			if( !outoss->openDevice( DspOut::WriteOnly )){
				if(debug >= 2) printf("=====CallAudio audioOut: openDevice Failed.\n" );
				po=0;
			}
		}
		if(debug >= 2) printf( "=====CallAudio audioOut: Creating RTP->OSS Diverter\n" );

		output = new DspAudioOut( inrtp, outoss );
		return po;
	} else 	if( sc->getAudioSys() == SessionControl::isALSA )
		{
		if(debug >= 2) printf( "=====CallAudio audioOut: ALSA used codec = %d\n",call->getRtpCodecNum());
		if (sc->getSymMediaMode()) {
			if(debug >= 2) printf( "=====CallAudio: audioOut sym media\n" );
			inrtp = new DspOutRtp( call->getRtpCodec(), call->getRtpCodecNum(),  min, max,QString::null, &socket );
		} else {
			inrtp = new DspOutRtp(call->getRtpCodec(), call->getRtpCodecNum() , min, max );
		}
		inrtp->setPayload(call->getResponsePay() );
		if( useStun ) {
		if(debug >= 2) printf( "=====CallAudio audioOut: use stun\n" );
		inrtp->setStunSrv( sc->getStunSrv() );
		}

		inrtp->openDevice( DspOut::ReadOnly );

		int po = 0;
		po=inrtp->getPortNum();
		if(debug >= 2) printf( "=====CallAudio audioOut: ALSA\n" );

		outalsa = new DspOutAlsa( sc->getALSAFilename() );//default
		if( !outalsa->openDevice( DspOut::WriteOnly )){
		    if(debug >= 2) printf("=====CallAudio audioOut: openDevice Failed.\n" );
		    po=0;
		}
//		printf ("!!!!!!!!!!!!!!!!!!!---------------> %s \n",socket.getStunnedHost().latin1() );
		if(debug >= 2) printf( "=====CallAudio audioOut: Creating RTP->ALSA Diverter\n" );
		output = new DspAudioOut( inrtp, outalsa );
		return po;
	}

return 0;
}

void CallAudio::stopSendingAudio( int aU ) {

if (aU == audioUser) {

	if(debug >= 2) printf("=====CallAudio: stopSendingAudio %d\n",audioUser);
	if( output ) {
	    if( output->running() ) {
		output->setCancel();
	    	output->wait();
	    }
	    if(debug >= 2) printf("=====CallAudio: stopSendingAudio output deleted %d\n",audioUser);
	    delete output;
	    output = 0;
	}


	if( input ) {
	    if( input->running() ) {
		input->setCancel();
		input->wait();
	    }
    	    if(debug >= 2) printf("=====CallAudio: stopSendingAudio input deleted %d\n",audioUser);
	    delete input;
	    input = 0;
	    
	}
	clearMe(aU);
}

}

void CallAudio::stopListeningAudio( int aU )
{
if (aU == audioUser) {
 
    if(debug >= 2) printf("=====CallAudio: stopListeningAudio %d\n",audioUser);

	if( input ) {
		if( input->running() ) {
			input->setCancel();
			input->wait();
		}
		delete input;
		input = 0;
	}
}
}

bool CallAudio::isAudioOn( void )
{
	return (output || input );
}

void CallAudio::startDTMF(int tone)
{
	if (outrtp) {

	    outrtp->sendDTMF(tone);
	    if(debug >= 2) printf( "tone %d \n",tone);
	}

	if (output) {
		output->startTone(charDTMF[tone]);
	}

	if (input) {
		input->startTone(charDTMF[tone]);
	}
}
void CallAudio::startDTMFSeq(QString string, int size)
{
dtmfCount = 0;
dtmfSize = size;
dtmfString = string;
dtmfTimer->start(500,TRUE);
}
void CallAudio::handleDTMFSeq() {
int tone = 0, id =0;

	if(dtmfCount < dtmfSize) {
		id        = (int)(*dtmfString.mid(dtmfCount,1));
		switch(id) {
			case 0x30: tone = 0; break;
			case 0x31: tone = 1; break;
			case 0x32: tone = 2; break;
			case 0x33: tone = 3; break;
			case 0x34: tone = 4; break;
			case 0x35: tone = 5; break;
			case 0x36: tone = 6; break;
			case 0x37: tone = 7; break;
			case 0x38: tone = 8; break;
			case 0x39: tone = 9;break;
			case 0x2a: tone = 10; break;
			case 0x23: tone = 11;break;
			case 0x2c: dtmfCount ++; dtmfTimer->start(sc->getDTMFWait(),TRUE); return;
			default: tone = 0;
		}
		if (outrtp) outrtp->sendDTMF(tone);
		dtmfCount ++;
	dtmfTimer->start(500,TRUE);
	}

}

void CallAudio::stopDTMF(void)
{
	if (output) {
		output->stopTone();
	}

	if (input) {
		input->stopTone();
	}
}
void CallAudio::ringOnce(int mul) {

	int audio_fd = ::open( sc->getRingtonedevice(), O_WRONLY | O_NONBLOCK );
		if( audio_fd == -1 ) {
			printf( "!!!!!ERROR: %s\n", "Open Failed" );
			return;
		}
		int flags = fcntl( audio_fd, F_GETFL );
		flags &= ~O_NONBLOCK;
		fcntl( audio_fd, F_SETFL, flags );
		int format = AFMT_S16_LE;
		if( ioctl( audio_fd, SNDCTL_DSP_SETFMT, &format ) == -1 ) {
			return;
		}
		if( format != AFMT_S16_LE ) {
			return;
		}
		int channels = 1;
		if( ioctl( audio_fd, SNDCTL_DSP_CHANNELS, &channels ) == -1 ) {
			return;
		}
		if( channels != 1 ) {
			return;
		}
		int rate = 8000;
		if( ioctl( audio_fd, SNDCTL_DSP_SPEED, &rate ) == -1 ) {
			return;
		}
		if( rate != 8000) {
			return;
		}
		int size = 1024*16;
		int samp = 1024*(mul-1)*4;
		int ampl = 1024*16;
		unsigned char devbuf[size];
		int buf[samp];
		int i, p=0;
		double arg1, arg2;
		arg1 = (double)2 * (double)M_PI * (double)941 / (double)samp;
		arg2 = (double)2 * (double)M_PI * (double)1336 / (double)samp;
		for(int i = 0; i < samp; i++) {
			buf[i] = (short)((double)(ampl) * sin(arg1 * i) +
				(double)(ampl) * sin(arg2 * i));
		};
		for (i=0; i<samp; i+=2) {
			devbuf[p++] = (unsigned char)(buf[i] & 0xff);
			devbuf[p++] = (unsigned char)((buf[i] >> 8) & 0xff);
		}
		for(;;) {
			if( write( audio_fd, devbuf, samp ) != -1 ) {
				break;
			}
		}
		::close( audio_fd );
}

void CallAudio::clearMe(int cwuser) {

	if(audioUser==cwuser){
 		audioUser=0;
	}
}

bool CallAudio::audioOwner(int cwuser){
	if(audioUser==cwuser) return true; else return false;
}