resamplesubs.c

音频信号的重采样程序,如44.1K的WAV转换成采样频率为48K的WAV.

    Xread = Xoff;               /* Position in input array to read into */
    Time = (Xoff<<Np);          /* Current-time pointer for converter */
    
    for (i=0; i<Xoff; X1[i++]=0); /* Need Xoff zeros at begining of sample */
    for (i=0; i<Xoff; X2[i++]=0); /* Need Xoff zeros at begining of sample */

    do {
        if (!last)              /* If haven't read last sample yet */
        {
            last = readData(infd, inCount, X1, X2, IBUFFSIZE,
                            nChans, (int)Xread);
            if (last && (last-Xoff<Nx)) { /* If last sample has been read... */
                Nx = last-Xoff; /* ...calc last sample affected by filter */
                if (Nx <= 0)
                  break;
            }
        }

        /* Resample stuff in input buffer */
        Time2 = Time;
        Nout=SrcLinear(X1,Y1,factor,&Time,Nx);
        if (nChans==2)
          Nout=SrcLinear(X2,Y2,factor,&Time2,Nx);

        Time -= (Nx<<Np);       /* Move converter Nx samples back in time */
        Xp += Nx;               /* Advance by number of samples processed */
        Ncreep = (Time>>Np) - Xoff; /* Calc time accumulation in Time */
        if (Ncreep) {
            Time -= (Ncreep<<Np);    /* Remove time accumulation */
            Xp += Ncreep;            /* and add it to read pointer */
        }
        for (i=0; i<IBUFFSIZE-Xp+Xoff; i++) { /* Copy part of input signal */
            X1[i] = X1[i+Xp-Xoff]; /* that must be re-used */
            if (nChans==2)
              X2[i] = X2[i+Xp-Xoff]; /* that must be re-used */
        }
        if (last) {             /* If near end of sample... */
            last -= Xp;         /* ...keep track were it ends */
            if (!last)          /* Lengthen input by 1 sample if... */
              last++;           /* ...needed to keep flag TRUE */
        }
        Xread = i;              /* Pos in input buff to read new data into */
        Xp = Xoff;
        
        Ycount += Nout;
        if (Ycount>outCount) {
            Nout -= (Ycount-outCount);
            Ycount = outCount;
        }

        if (Nout > OBUFFSIZE) /* Check to see if output buff overflowed */
          return err_ret("Output array overflow");
        
        if (nChans==1) {
            for (i = 0; i < Nout; i++)
                obufs[0][i] = HWORD_TO_MUS_SAMPLE_TYPE(Y1[i]);
        } else {
            for (i = 0; i < Nout; i++) {
                obufs[0][i] = HWORD_TO_MUS_SAMPLE_TYPE(Y1[i]);
                obufs[1][i] = HWORD_TO_MUS_SAMPLE_TYPE(Y2[i]);
            }
        }
        /* NB: errors reported within sndlib */
        mus_file_write(outfd, 0, Nout - 1, nChans, obufs);

        printf(".");  fflush(stdout);

    } while (Ycount<outCount); /* Continue until done */

    return(Ycount);             /* Return # of samples in output file */
}


static int resampleWithFilter(  /* number of output samples returned */
    double factor,              /* factor = outSampleRate/inSampleRate */
    int infd,                   /* input and output file descriptors */
    int outfd,
    int inCount,                /* number of input samples to convert */
    int outCount,               /* number of output samples to compute */
    int nChans,                 /* number of sound channels (1 or 2) */
    BOOL interpFilt,            /* TRUE means interpolate filter coeffs */
    HWORD Imp[], HWORD ImpD[],
    UHWORD LpScl, UHWORD Nmult, UHWORD Nwing)
{
    UWORD Time, Time2;          /* Current time/pos in input sample */
    UHWORD Xp, Ncreep, Xoff, Xread;
    int OBUFFSIZE = (int)(((double)IBUFFSIZE)*factor+2.0);
    HWORD X1[IBUFFSIZE], Y1[OBUFFSIZE]; /* I/O buffers */
    HWORD X2[IBUFFSIZE], Y2[OBUFFSIZE]; /* I/O buffers */
    UHWORD Nout, Nx;
    int i, Ycount, last;
    
    mus_sample_t **obufs = sndlib_allocate_buffers(nChans, OBUFFSIZE);
    if (obufs == NULL)
        return err_ret("Can't allocate output buffers");

    /* Account for increased filter gain when using factors less than 1 */
    if (factor < 1)
      LpScl = LpScl*factor + 0.5;

    /* Calc reach of LP filter wing & give some creeping room */
    Xoff = ((Nmult+1)/2.0) * MAX(1.0,1.0/factor) + 10;

    if (IBUFFSIZE < 2*Xoff)      /* Check input buffer size */
      return err_ret("IBUFFSIZE (or factor) is too small");

    Nx = IBUFFSIZE - 2*Xoff;     /* # of samples to process each iteration */
    
    last = 0;                   /* Have not read last input sample yet */
    Ycount = 0;                 /* Current sample and length of output file */
    Xp = Xoff;                  /* Current "now"-sample pointer for input */
    Xread = Xoff;               /* Position in input array to read into */
    Time = (Xoff<<Np);          /* Current-time pointer for converter */
    
    for (i=0; i<Xoff; X1[i++]=0); /* Need Xoff zeros at begining of sample */
    for (i=0; i<Xoff; X2[i++]=0); /* Need Xoff zeros at begining of sample */
        
    do {
        if (!last)              /* If haven't read last sample yet */
        {
            last = readData(infd, inCount, X1, X2, IBUFFSIZE, 
                            nChans, (int)Xread);
            if (last && (last-Xoff<Nx)) { /* If last sample has been read... */
                Nx = last-Xoff; /* ...calc last sample affected by filter */
                if (Nx <= 0)
                  break;
            }
        }
        /* Resample stuff in input buffer */
        Time2 = Time;
        if (factor >= 1) {      /* SrcUp() is faster if we can use it */
            Nout=SrcUp(X1,Y1,factor,&Time,Nx,Nwing,LpScl,Imp,ImpD,interpFilt);
            if (nChans==2)
              Nout=SrcUp(X2,Y2,factor,&Time2,Nx,Nwing,LpScl,Imp,ImpD,
                         interpFilt);
        }
        else {
            Nout=SrcUD(X1,Y1,factor,&Time,Nx,Nwing,LpScl,Imp,ImpD,interpFilt);
            if (nChans==2)
              Nout=SrcUD(X2,Y2,factor,&Time2,Nx,Nwing,LpScl,Imp,ImpD,
                         interpFilt);
        }

        Time -= (Nx<<Np);       /* Move converter Nx samples back in time */
        Xp += Nx;               /* Advance by number of samples processed */
        Ncreep = (Time>>Np) - Xoff; /* Calc time accumulation in Time */
        if (Ncreep) {
            Time -= (Ncreep<<Np);    /* Remove time accumulation */
            Xp += Ncreep;            /* and add it to read pointer */
        }
        for (i=0; i<IBUFFSIZE-Xp+Xoff; i++) { /* Copy part of input signal */
            X1[i] = X1[i+Xp-Xoff]; /* that must be re-used */
            if (nChans==2)
              X2[i] = X2[i+Xp-Xoff]; /* that must be re-used */
        }
        if (last) {             /* If near end of sample... */
            last -= Xp;         /* ...keep track were it ends */
            if (!last)          /* Lengthen input by 1 sample if... */
              last++;           /* ...needed to keep flag TRUE */
        }
        Xread = i;              /* Pos in input buff to read new data into */
        Xp = Xoff;
        
        Ycount += Nout;
        if (Ycount>outCount) {
            Nout -= (Ycount-outCount);
            Ycount = outCount;
        }

        if (Nout > OBUFFSIZE) /* Check to see if output buff overflowed */
          return err_ret("Output array overflow");
        
        if (nChans==1) {
            for (i = 0; i < Nout; i++)
              obufs[0][i] = HWORD_TO_MUS_SAMPLE_TYPE(Y1[i]);
        } else {
            for (i = 0; i < Nout; i++) {
                obufs[0][i] = HWORD_TO_MUS_SAMPLE_TYPE(Y1[i]);
                obufs[1][i] = HWORD_TO_MUS_SAMPLE_TYPE(Y2[i]);
            }
        }
        /* NB: errors reported within sndlib */
        mus_file_write(outfd, 0, Nout - 1, nChans, obufs);

        printf(".");  fflush(stdout);

    } while (Ycount<outCount); /* Continue until done */

    return(Ycount);             /* Return # of samples in output file */
}


int resample(                   /* number of output samples returned */
    double factor,              /* factor = Sndout/Sndin */
    int    infd,                /* input and output file descriptors */
    int    outfd,
    int inCount,                /* number of input samples to convert */
    int outCount,               /* number of output samples to compute */
    int nChans,                 /* number of sound channels (1 or 2) */
    BOOL interpFilt,            /* TRUE means interpolate filter coeffs */
    int fastMode,               /* 0 = highest quality, slowest speed */
    BOOL largeFilter,           /* TRUE means use 65-tap FIR filter */
    char *filterFile)           /* NULL for internal filter, else filename */
{
    UHWORD LpScl;               /* Unity-gain scale factor */
    UHWORD Nwing;               /* Filter table size */
    UHWORD Nmult;               /* Filter length for up-conversions */
    HWORD *Imp=0;               /* Filter coefficients */
    HWORD *ImpD=0;              /* ImpD[n] = Imp[n+1]-Imp[n] */
    
    if (fastMode)
      return resampleFast(factor,infd,outfd,inCount,outCount,nChans);

#ifdef DEBUG
    /* Check for illegal constants */
    if (Np >= 16)
      return err_ret("Error: Np>=16");
    if (Nb+Nhg+NLpScl >= 32)
      return err_ret("Error: Nb+Nhg+NLpScl>=32");
    if (Nh+Nb > 32)
      return err_ret("Error: Nh+Nb>32");
#endif
    
    /* Set defaults */

    if (filterFile != NULL && *filterFile != '\0') {
        if (readFilter(filterFile, &Imp, &ImpD, &LpScl, &Nmult, &Nwing))
          return err_ret("could not find filter file, "
               "or syntax error in contents of filter file");
    } else if (largeFilter) {
        Nmult = LARGE_FILTER_NMULT;
        Imp = LARGE_FILTER_IMP;         /* Impulse response */
        ImpD = LARGE_FILTER_IMPD;       /* Impulse response deltas */
        LpScl = LARGE_FILTER_SCALE;     /* Unity-gain scale factor */
        Nwing = LARGE_FILTER_NWING;     /* Filter table length */
    } else {
        Nmult = SMALL_FILTER_NMULT;
        Imp = SMALL_FILTER_IMP;         /* Impulse response */
        ImpD = SMALL_FILTER_IMPD;       /* Impulse response deltas */
        LpScl = SMALL_FILTER_SCALE;     /* Unity-gain scale factor */
        Nwing = SMALL_FILTER_NWING;     /* Filter table length */
    }
#if DEBUG
    fprintf(stderr,"Attenuating resampler scale factor by 0.95 "
            "to reduce probability of clipping\n");
#endif
    LpScl *= 0.95;
    return resampleWithFilter(factor,infd,outfd,inCount,outCount,nChans, 
                              interpFilt, Imp, ImpD, LpScl, Nmult, Nwing);
}