resample.c

刻录光盘的程序

		memcpy( newp+(di*4), p+(si*4), 4 );
		idx += (double)(global.playback_rate/100.0);
		si = (long)idx;
		di++;
	}
	return di;
}
#endif

static int guess_endianess(p, p2, SamplesToDo)
	UINT4 *p;
	short *p2;
	unsigned SamplesToDo;
{
    /* analyse samples */
    int vote_for_little = 0;
    int vote_for_big = 0;
    int total_votes;

    while (((UINT4 *)p2 - p) + (unsigned) 1 < SamplesToDo) {
      unsigned char *p3 = (unsigned char *)p2;
#if MY_LITTLE_ENDIAN == 1
      int diff_lowl = *(p2+0) - *(p2+2);
      int diff_lowr = *(p2+1) - *(p2+3);
      int diff_bigl = ((*(p3  ) << 8) + *(p3+1)) - ((*(p3+4) << 8) + *(p3+5));
      int diff_bigr = ((*(p3+2) << 8) + *(p3+3)) - ((*(p3+6) << 8) + *(p3+7));
#else
      int diff_lowl = ((*(p3+1) << 8) + *(p3  )) - ((*(p3+5) << 8) + *(p3+4));
      int diff_lowr = ((*(p3+3) << 8) + *(p3+2)) - ((*(p3+7) << 8) + *(p3+6));
      int diff_bigl = *(p2+0) - *(p2+2);
      int diff_bigr = *(p2+1) - *(p2+3);
#endif

      if ((abs(diff_lowl) + abs(diff_lowr)) <
	  (abs(diff_bigl) + abs(diff_bigr))) {
	vote_for_little++;
      } else {
	if ((abs(diff_lowl) + abs(diff_lowr)) >
	    (abs(diff_bigl) + abs(diff_bigr))) {
	  vote_for_big++;
	}
      }
      p2 += 2;
   }
#ifdef DEBUG_VOTE_ENDIANESS
   if (global.quiet != 1)
     fprintf(stderr, "votes for little: %4d,  votes for big: %4d\n", 
		vote_for_little, vote_for_big);
#endif
   total_votes = vote_for_big + vote_for_little;
   if (total_votes < 3
       || abs(vote_for_big - vote_for_little) < total_votes/3) {
     return -1;
   } else {
	if (vote_for_big > vote_for_little)
		return 1;
	else
		return 0;
   }
}

int jitterShift = 0; 

unsigned char *synchronize(p, SamplesToDo, TotSamplesDone)
	UINT4 *p;
	unsigned SamplesToDo;
	unsigned TotSamplesDone;
{
  static int jitter = 0;
  char *pSrc;                   /* start of cdrom buffer */

  /* if endianess is unknown, guess endianess based on 
     differences between succesive samples. If endianess
     is correct, the differences are smaller than with the
     opposite byte order.
   */
  if ((*in_lendian) < 0) {
    short *p2 = (short *)p;

    /* skip constant samples */
    while ((((UINT4 *)p2 - p) + (unsigned) 1 < SamplesToDo)
           && *p2 == *(p2+2)) p2++;

    if (((UINT4 *)p2 - p) + (unsigned) 1 < SamplesToDo) {
      switch (guess_endianess(p, p2, SamplesToDo)) {
        case -1: break;
        case  1: (*in_lendian) = 0;
#if 0
	         if (global.quiet != 1)
		   fprintf(stderr, "big endian detected\n");
#endif
	break;
        case  0: (*in_lendian) = 1;
#if 0
	         if (global.quiet != 1)
		   fprintf(stderr, "little endian detected\n");
#endif
	break;
      }
    }
  }

  /* ENDIAN ISSUES:
   * the individual endianess of cdrom/cd-writer, cpu, 
   * sound card and audio output format need a careful treatment.
   *
   * For possible sample processing (rate conversion) we need
   * the samples in cpu byte order. This is the first conversion.
   *
   * After processing it depends on the endianness of the output
   * format, whether a second conversion is needed.
   *
   */

  if (global.need_hostorder && (*in_lendian) != MY_LITTLE_ENDIAN) {
    /* change endianess of delivered samples to native cpu order */
    change_endianness(p, SamplesToDo);
  }

  /* synchronisation code */
  if (TotSamplesDone != 0 && global.overlap != 0 && SamplesToDo > CD_FRAMESAMPLES) {

    pSrc = (char *) sync_buffers((unsigned char *)p);
    if (!pSrc ) {
      return NULL;
    }
    if (pSrc) {
      jitter = ((unsigned char *)pSrc - (((unsigned char *)p) + global.overlap*CD_FRAMESIZE_RAW))/4;
      jitterShift += jitter;
      SamplesToDo -= jitter + global.overlap*CD_FRAMESAMPLES;
#if 0
      fprintf(stderr,
	    "Length: pre %d, diff1 %ld, diff2 %ld, min %ld\n", SamplesToDo,
	   (TotSamplesWanted - TotSamplesDone),
	   SamplesNeeded((TotSamplesWanted - TotSamplesDone), undersampling),
	   min(SamplesToDo, SamplesNeeded((TotSamplesWanted - TotSamplesDone), undersampling)));
#endif
    }
  } else {
    pSrc = ( char * ) p;
  }
  return (unsigned char *) pSrc;
}

/* convert cdda data to required output format
 * sync code for unreliable cdroms included
 * 
 */
long 
SaveBuffer (p, SamplesToDo, TotSamplesDone)
	UINT4 *p;
	unsigned long SamplesToDo;
	unsigned long *TotSamplesDone;
{
  UINT4 *pSrc;                   /* start of cdrom buffer */
  UINT4 *pSrcStop;               /* end of cdrom buffer */

  /* in case of different endianness between host and output format,
     copy in a seperate buffer and modify the local copy */
  if ( ((!global.need_hostorder && global.need_big_endian == (*in_lendian)) ||
	 (global.need_hostorder && global.need_big_endian != MY_BIG_ENDIAN))
      && global.OutSampleSize > 1) {
     static UINT4 *localoutputbuffer;
     if (localoutputbuffer == NULL) {
       localoutputbuffer = (UINT4 *) malloc(global.nsectors*CD_FRAMESIZE_RAW);
       if (localoutputbuffer == NULL) {
         perror("cannot allocate local buffer");
         return 1;
       }
     }
     memcpy(localoutputbuffer, p, SamplesToDo*4);
     p = localoutputbuffer;
  }

  pSrc = p;
  pDst = (unsigned char *) p;
  pStart = ( unsigned char * ) pSrc;
  pSrcStop = pSrc + SamplesToDo;

  /* code for subsampling and output stage */

  if (global.ismono && global.findmono) {
    short *pmm;
    for (pmm = (short *)pStart; (UINT4 *) pmm < pSrcStop; pmm += 2) {
      if (*pmm != *(pmm+1)) {
        global.ismono = 0;
        break;
      }
    }
  }
  /* optimize the case of no conversion */
  if (1 && undersampling == 1 && samples_to_do == 1 &&
       global.channels == 2 && global.OutSampleSize == 2 && Halved == 0) {
    /* output format is the original cdda format ->
     * just forward the buffer 
     */
      
    if ( waitforsignal != 0 && any_signal == 0) {
      int *myptr = (int *)pStart;
      while ((UINT4 *)myptr < pSrcStop && *myptr == 0) myptr++;
      pStart = (unsigned char *) myptr;
      /* scan for first signal */
      if ( (UINT4 *)pStart != pSrcStop ) {
	/* first non null amplitude is found in buffer */
	any_signal = 1;
      }
    }
    pDst = (unsigned char *) pSrcStop;		/* set pDst to end */

    if (global.deemphasize) {
      /* this implements an attenuation treble shelving filter 
         to undo the effect of pre-emphasis. The filter is of
         a recursive first order */
      static short lastin[2] = { 0, 0 };
      static double lastout[2] = { 0.0, 0.0 };
      short *pmm;

      /* Here is the gnuplot file for the frequency response
         of the deemphasis. The error is below +-0.1dB

# first define the ideal filter. We use the tenfold sampling frequency.
T=1./441000.
OmegaU=1./15E-6
OmegaL=15./50.*OmegaU
V0=OmegaL/OmegaU
H0=V0-1.
B=V0*tan(OmegaU*T/2.)
# the coefficients follow
a1=(B - 1.)/(B + 1.)
b0=(1.0 + (1.0 - a1) * H0/2.)
b1=(a1 + (a1 - 1.0) * H0/2.)
# helper variables
D=b1/b0
o=2*pi*T
H2(f)=b0*sqrt((1+2*cos(f*o)*D+D*D)/(1+2*cos(f*o)*a1+a1*a1))
# now approximate the ideal curve with a fitted one for sampling frequency
# of 44100 Hz.
T2=1./44100.
V02=0.3365
OmegaU2=1./19E-6
B2=V02*tan(OmegaU2*T2/2.)
# the coefficients follow
a12=(B2 - 1.)/(B2 + 1.)
b02=(1.0 + (1.0 - a12) * (V02-1.)/2.)
b12=(a12 + (a12 - 1.0) * (V02-1.)/2.)
# helper variables
D2=b12/b02
o2=2*pi*T2
H(f)=b02*sqrt((1+2*cos(f*o2)*D2+D2*D2)/(1+2*cos(f*o2)*a12+a12*a12))
# plot best, real, ideal, level with halved attenuation,
#      level at full attentuation, 10fold magnified error
set logscale x
set grid xtics ytics mxtics mytics
plot [f=1000:20000] [-12:2] 20*log10(H(f)),20*log10(H2(f)),  20*log10(OmegaL/(2*pi*f)), 0.5*20*log10(V0), 20*log10(V0), 200*log10(H(f)/H2(f))
pause -1 "Hit return to continue"
       */

#ifdef TEST
#define V0	0.3365
#define OMEGAG	(1./19e-6)
#define T	(1./44100.)
#define H0	(V0-1.)
#define B	(V0*tan((OMEGAG * T)/2.0))
#define a1	((B - 1.)/(B + 1.))
#define b0 	(1.0 + (1.0 - a1) * H0/2.0)
#define b1 	(a1 + (a1 - 1.0) * H0/2.0)
#else
#define a1	-0.62786881719628784282
#define b0 	0.45995451989513153057
#define b1 	-0.08782333709141937339
#endif

      for (pmm = (short *)pStart; pmm < (short *)pDst;) {
        lastout[0] = *pmm * b0 + lastin[0] * b1 - lastout[0] * a1;
        lastin[0] = *pmm;
        *pmm++ = lastout[0] > 0.0 ? lastout[0] + 0.5 : lastout[0] - 0.5;
        lastout[1] = *pmm * b0 + lastin[1] * b1 - lastout[1] * a1;
        lastin[1] = *pmm;
        *pmm++ = lastout[1] > 0.0 ? lastout[1] + 0.5 : lastout[1] - 0.5;
      }
    }

    if (global.swapchannels == 1) {
	swap_channels((UINT4 *)pStart, SamplesToDo);
    }

    if (global.findminmax) {
      short *pmm;
      for (pmm = (short *)pStart; pmm < (short *)pDst; pmm++) {
        if (*pmm < global.minamp[1]) global.minamp[1] = *pmm;
        if (*pmm > global.maxamp[1]) global.maxamp[1] = *pmm;
        pmm++;
        if (*pmm < global.minamp[0]) global.minamp[0] = *pmm;
        if (*pmm > global.maxamp[0]) global.maxamp[0] = *pmm;
      }
    }
  } else {

#define none_missing	0
#define one_missing	1
#define two_missing	2
#define collecting	3

    static int sample_state = collecting;
    static int Toggle_on = 0;

    if (global.channels == 2 && global.swapchannels == 1) {
	swap_channels((UINT4 *)pStart, SamplesToDo);
    }

    /* conversion required */
    while ( pSrc < pSrcStop ) {
	  
	long l,r;

	long iSamples_left = (pSrcStop - pSrc) / sizeof(short) / 2;
	short *myptr = (short *) pSrc;

	/* LSB l, MSB l */
	l = *myptr++;	/* left channel */
	r = *myptr++;	/* right channel */
	pSrc = (UINT4 *) myptr;

	switch (sample_state) {
	case two_missing:
two__missing:
	    ls2 += l; rs2 += r;
	    if (undersampling > 1) {
		ls3 += l; rs3 += r;
	    }
	    sample_state = one_missing;
	    break;
	case one_missing:
	    auxl = l; auxr = r;

	    ls3 += l; rs3 += r;
	    sample_state = none_missing;

	    /* FALLTHROUGH */
none__missing:
	case none_missing:
	    /* Filtered samples are complete. Now interpolate and scale. */

	    if (Halved != 0 && Toggle_on == 0) {
                lsum = interpolate(lsum, ls2, ls3)/(int) undersampling;
	        rsum = interpolate(rsum, rs2, rs3)/(int) undersampling;
            } else {
		lsum /= (int) undersampling;
		rsum /= (int) undersampling;
            }
	    emit_sample(lsum, rsum, global.channels);
	    /* reload counter */
	    samples_to_do = undersampling - 1;
	    lsum = auxl;
	    rsum = auxr;
	    /* reset sample register */
	    auxl = ls2 = ls3 = 0;
	    auxr = rs2 = rs3 = 0;
	    Toggle_on ^= 1;
	    sample_state = collecting;
	    break;
	case collecting:
	    if ( samples_to_do > 0) {
		samples_to_do--;
		if (Halved != 0 && Toggle_on == 0) {
		    /* Divider x.5 : we need data for quadratic interpolation */
		    iSamples_left--;

		    lsum += l; rsum += r;
		    if ( samples_to_do < undersampling - 1) {
			ls2 += l; rs2 += r;
		    }
		    if ( samples_to_do < undersampling - 2) {
			ls3 += l; rs3 += r;
		    }
		} else {
		    /* integral divider */
		    lsum += l;
		    rsum += r;
		    iSamples_left--;
		}
	    } else {
	        if (Halved != 0 && Toggle_on == 0) {
		    sample_state = two_missing;
		    goto two__missing;
		} else {
		    auxl = l;
		    auxr = r;
		    sample_state = none_missing;
		    goto none__missing;
		}
	    }
	    break;
	} /* switch state */

    } /* while */

    /* flush_buffer */
    if ((samples_to_do == 0 && Halved == 0))
    {
	if (Halved != 0 && Toggle_on == 0) {
	    lsum = interpolate(lsum, ls2, ls3)/(int) undersampling;
	    rsum = interpolate(rsum, rs2, rs3)/(int) undersampling;
	} else {
	    lsum /= (int) undersampling;
	    rsum /= (int) undersampling;
	}
	emit_sample(lsum, rsum, global.channels);
	
	/* reload counter */
	samples_to_do = undersampling;
	
	/* reset sample register */
	lsum = auxl = ls2 = ls3 = 0;
	rsum = auxr = rs2 = rs3 = 0;
	Toggle_on ^= 1;
	sample_state = collecting;
    }

  } /* if optimize else */

  if ( waitforsignal == 0 ) pStart = (unsigned char *)p;
  else if (any_signal != 0) global.SkippedSamples += ((UINT4 *)pStart - p);
  else global.SkippedSamples += (pSrcStop - p);

  if ( waitforsignal == 0 || any_signal != 0) {
    int retval = 0;
    unsigned outlen;

    assert(pDst >= pStart);
    outlen = (size_t) (pDst - pStart);

    if (outlen <= 0) return 0;

#ifdef	ECHO_TO_SOUNDCARD

    /* this assumes the soundcard needs samples in native cpu byte order */
    if (global.echo != 0) {
               static unsigned char *newp;
               unsigned    newlen;

               newlen = (100*(outlen/4))/global.playback_rate;
               newlen = (newlen*4);
               if ( (newp != NULL) || (newp = (unsigned char *) malloc( 2*global.nsectors*CD_FRAMESIZE_RAW+32 )) ) {
			newlen = 4*ReSampleBuffer( pStart, newp, outlen/4 );
			write_snd_device((char *)newp, newlen);
               }
    }
#endif

    if ( global.no_file != 0 ) {
        *TotSamplesDone += SamplesToDo;
        return 0;
    }
    if ( (!global.need_hostorder && global.need_big_endian == (*in_lendian)) ||
	 (global.need_hostorder && global.need_big_endian != MY_BIG_ENDIAN)) {
      if ( global.OutSampleSize > 1) {
        /* change endianness from input sample or native cpu order 
           to required output endianness */
        change_endianness((UINT4 *)pStart, outlen/4);
      }
    }
    if ((unsigned)(retval = write ( global.audio, pStart, outlen )) == outlen) {
        *TotSamplesDone += SamplesToDo;
	return 0;
    } else {
        fprintf(stderr, "write(audio, 0x%p, %u) = %d\n",pStart,outlen,retval);
        perror("Probably disk space exhausted");
        return 1;
    }
  } else return 0;
}