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	git-svn-id: svn+ssh://svn.code.sf.net/p/wsjt/wsjt/trunk@109 ab8295b8-cf94-4d9e-aec4-7959e3be5d79
		
			
				
	
	
		
			291 lines
		
	
	
		
			7.8 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			291 lines
		
	
	
		
			7.8 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
#include <stdio.h>
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#include <stdlib.h>
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#include <pthread.h>
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#include <inttypes.h>
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#include <time.h>
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#include <sys/time.h>
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#include <fcntl.h>
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#include <sys/soundcard.h>
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#include "conf.h"
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#include <string.h>
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#define AUDIOBUFSIZE	4096
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#define FRAMESPERBUFFER 1024
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#define TIMEOUT 1000L		/* select time out for audio device */
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char rcv_buf[AUDIOBUFSIZE];	/* XXX grab one from upper app later --db */
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char tx_buf[AUDIOBUFSIZE];	/* XXX grab one from upper app later --db */
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#define DSP "/dev/dsp0.0"
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#define MAXDSPNAME sizeof(DSP)+1	/* quick hack --db */
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char dsp_in[MAXDSPNAME];		/* Both of these must be same length */
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char dsp_out[MAXDSPNAME];
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extern void decode1_(int *iarg);
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void oss_loop(int *iarg);
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/*
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 * local state data referencing some gcom common fortran variables as well
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 */
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struct audio_data {
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  int fd_in;		/* Audio fd in; used only locally in this function  */
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  int fd_out;		/* Audio fd out; used only locally in this function */
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  double *Tsec;		/* Present time                       SoundIn,SoundOut */
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  double *tbuf;		/* Tsec at time of input callback          SoundIn  */
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  int *iwrite;		/* Write pointer to Rx ring buffer         SoundIn  */
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  int *ibuf;		/* Most recent input buffer#               SoundIn  */
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  int *TxOK;		/* OK to transmit?                         SoundIn  */
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  int *ndebug;		/* Write debugging info?                   GUI      */
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  int *ndsec;		/* Dsec in units of 0.1 s                  GUI      */
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  int *Transmitting;	/* Actually transmitting?                  SoundOut */
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  int *nwave;		/* Number of samples in iwave              SoundIn  */
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  int *nmode;		/* Which WSJT mode?                        GUI      */
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  int *trperiod;	/* Tx or Rx period in seconds              GUI      */
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  int nbuflen;
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  int nfs;
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  int16_t *y1;		/* Ring buffer for audio channel 0         SoundIn  */
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  int16_t *y2;		/* Ring buffer for audio channel 1         SoundIn  */
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  short  *iwave;
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}data;
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/*
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 * start_threads()
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 * inputs	- ndevin  device number for input
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 *		- ndevout device number for output
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 *		- y1 short int array for channel 0
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 *		- y2 short int array for channel 1
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 * 		- nmax
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 *		- iwrite
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 *		- iwave
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 *		- nwave
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 *		- rate 
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 *		- NSPB
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 *		- TRPeriod
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 *		- TxOK
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 *		- ndebug debug output or not?
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 *		- Transmitting
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 *		- Tsec
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 *		- ngo
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 *		- nmode
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 *		- tbuf
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 *		- ibuf
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 *		- ndsec
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 * output	- ?
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 * side effects - Called from audio_init.f90 to start audio decode and
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 *		  OSS thread.
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 */
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int
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start_threads_(int *ndevin, int *ndevout, short y1[], short y2[],
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	       int *nbuflen, int *iwrite, short iwave[],
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	       int *nwave, int *nfsample, int *nsamperbuf,
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	       int *TRPeriod, int *TxOK, int *ndebug,
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	       int *Transmitting, double *Tsec, int *ngo, int *nmode,
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	       double tbuf[], int *ibuf, int *ndsec)
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{
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  pthread_t thread1,thread2;
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  int iret1,iret2;
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  int iarg1 = 1,iarg2 = 2;
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  int32_t rate=*nfsample;
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  int samplesize;
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  int format;
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  int channels;
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  double dnfs;
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  /* XXX OSS device is decoded from ndevin and ndevout
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   * This is not strictly speaking the way to do it and is
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   * probably specific to FreeBSD for now. 
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   * i.e. the .0 addition is a vchan; I'll add configure magic later. --db
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   */
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  snprintf(dsp_in, MAXDSPNAME, 
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	   "/dev/dsp%d.0", *ndevin);
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  dsp_in[MAXDSPNAME] = '\0';
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  data.fd_in = open (dsp_in, O_RDWR, 0);
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  if (data.fd_in < 0) { 
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	fprintf(stderr, "Cannot open %s for input.\n", dsp_in);
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	exit(-1);
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  }
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  if (*ndevin == *ndevout) {
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    data.fd_out = data.fd_in;
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    strncpy(dsp_out, dsp_in, sizeof(dsp_out));
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    dsp_out[sizeof(dsp_out)] = '\0';
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    if (ioctl(data.fd_in, SNDCTL_DSP_SETDUPLEX, 0) < 0) {
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	fprintf(stderr, "Cannot use %s for full duplex.\n", dsp_in);
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	exit(-1);
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    }
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  }
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  data.Tsec = Tsec;
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  data.tbuf = tbuf;
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  data.iwrite = iwrite;
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  data.ibuf = ibuf;
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  data.TxOK = TxOK;
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  data.ndebug = ndebug;
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  data.ndsec = ndsec;
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  data.Transmitting = Transmitting;
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  data.y1 = y1;
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  data.y2 = y2;
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  data.nbuflen = *nbuflen;
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  data.nmode = nmode;
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  data.nwave = nwave;
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  data.iwave = iwave;
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  data.nfs = *nfsample;
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  data.trperiod = TRPeriod;
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  dnfs=(double)*nfsample;
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  channels = 2;
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  if (ioctl (data.fd_in, SNDCTL_DSP_CHANNELS, &channels) == -1) {
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	fprintf (stderr, "Unable to set 2 channels for input.\n");
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	exit (-1);
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  }
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  if (channels != 2) {
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    fprintf (stderr, "Unable to set 2 channels.\n");
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    exit (-1);
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  }
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  format = AFMT_S16_NE;
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  if (ioctl (data.fd_in, SNDCTL_DSP_SETFMT, &format) == -1) {
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	fprintf (stderr, "Unable to set format for input.\n");
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	exit (-1);
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  }
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  if (ioctl (data.fd_in, SNDCTL_DSP_SPEED, &rate) == -1) {
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	fprintf (stderr, "Unable to set rate for input\n");
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	exit (-1);
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  }
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  printf("Audio OSS streams running normally.\n");
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  printf("******************************************************************\n");
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  printf("Opened %s for input.\n", dsp_in);
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  printf("Opened %s for output.\n", dsp_out);
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  printf("Rate set = %d\n", rate);
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  //  printf("start_threads: creating thread for oss_loop\n");
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  iret1 = pthread_create(&thread1, NULL,
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			 (void *(*)(void *))oss_loop, &iarg1);
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  // printf("start_threads: creating thread for decode1_\n");
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  iret2 = pthread_create(&thread2, NULL,
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			 (void *(*)(void *))decode1_,&iarg2);
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}
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/*
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 * oss_loop
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 *
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 * inputs	- int pointer NOT USED
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 * output	- none
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 * side effects	-
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 */
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void
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oss_loop(int *iarg)
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{
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  fd_set readfds, writefds;
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  int nfds = 0;
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  struct timeval timeout = {0, 0};
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  struct timeval tv;
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  int nread;
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  unsigned int i;
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  static int n=0;
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  static int n2=0;
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  static int ia=0;
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  static int ib=0;
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  static int ic=0;
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  static int16_t *in;
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  static int16_t *wptr;
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  static int TxOKz=0;
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  static int ncall=0;
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  static int nsec=0;
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  static double stime;
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  for (;;) {
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    FD_ZERO(&readfds );
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    FD_ZERO(&writefds );
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    FD_SET(data.fd_in, &readfds);
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    FD_SET(data.fd_out, &writefds);
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    timeout.tv_usec = TIMEOUT;
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    if (select(FD_SETSIZE, &readfds, &writefds, NULL, &timeout) > 0) {
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      if (FD_ISSET(data.fd_in, &readfds)) {
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	    nread = read (data.fd_in, rcv_buf, AUDIOBUFSIZE);
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	    if (nread <= 0) {
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	      fprintf(stderr, "Read error %d\n", nread);
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	      exit(-1);
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	    }
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	    if (nread == AUDIOBUFSIZE) {
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	      /* Get System time */
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	      gettimeofday(&tv, NULL);
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	      stime = (double) tv.tv_sec + ((double)tv.tv_usec / 1000000.0) +
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		*(data.ndsec) * 0.1;
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	      *(data.Tsec) = stime;
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	      ncall++;
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	      /* increment buffer pointers only if data available */
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	      ia=*(data.iwrite);
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	      ib=*(data.ibuf);
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	      data.tbuf[ib++] = stime;
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	      if(ib>=FRAMESPERBUFFER)
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		ib=0; 
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	      *(data.ibuf) = ib;
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	      in = (int16_t *)rcv_buf;	// XXX
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	      for(i=0; i<FRAMESPERBUFFER; i++) {
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		data.y1[ia] = (*in++);
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		data.y2[ia] = (*in++);
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		ia++;
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	      }
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	      if(ia >= data.nbuflen)
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		ia=0;  //Wrap buffer pointer if necessary
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	      *(data.iwrite) = ia;            /* Save buffer pointer */
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	      fivehz_();                      /* Call fortran routine */
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	    }
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      }
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      if (FD_ISSET(data.fd_in, &writefds)) {
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	/* Get System time */
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	gettimeofday(&tv, NULL);
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	stime = (double) tv.tv_sec + ((double)tv.tv_usec / 1000000.0) +
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	  *(data.ndsec) * 0.1;
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	*(data.Tsec) = stime;
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	if(*(data.TxOK) && (!TxOKz))  {
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	  n=nsec/(*(data.trperiod));
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	  ic = (int)(stime - *(data.trperiod)*n) * data.nfs;
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	  ic = ic % *(data.nwave);
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	}
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	TxOKz = *(data.TxOK);
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	*(data.Transmitting) = *(data.TxOK);
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	wptr = (int16_t *)tx_buf;		/* XXX */
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	if(*(data.TxOK))  {
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	  for(i=0 ; i<FRAMESPERBUFFER; i++ )  {
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	    n2 = data.iwave[ic];
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	    addnoise_(&n2);
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	    *wptr++ = n2;			/* left */
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	    *wptr++ = n2;			/* right */
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	    ic++;
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	    if(ic >= *(data.nwave)) {
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	      ic = ic % *(data.nwave);	/* Wrap buffer pointer if necessary */
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	      if(*(data.nmode) == 2)
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		*(data.TxOK) = 0;
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	    }
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	  }
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	} else {
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	  memset(tx_buf, 0, AUDIOBUFSIZE);
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	}
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	if (write(data.fd_out, tx_buf, AUDIOBUFSIZE) < 0) {
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	  fprintf(stderr, "Can't write to soundcard.\n");
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	  exit(-1);
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	}
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	fivehztx_();                             /* Call fortran routine */
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      }
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    }
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  }
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}
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