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diff --git a/libtoolame-dab/subband.c b/libtoolame-dab/subband.c
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+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "common.h"
+#include "encoder.h"
+#include "mem.h"
+#include "bitstream.h"
+#include "encode.h"
+#include "enwindow.h"
+#include "subband.h"
+
+
+#ifdef REFERENCECODE
+/************************************************************************
+*
+* window_subband()
+*
+* PURPOSE: Overlapping window on PCM samples
+*
+* SEMANTICS:
+* 32 16-bit pcm samples are scaled to fractional 2's complement and
+* concatenated to the end of the window buffer #x#. The updated window
+* buffer #x# is then windowed by the analysis window #c# to produce the
+* windowed sample #z#
+*
+************************************************************************/
+
+void window_subband (short **buffer, double z[64], int k)
+{
+ typedef double XX[2][HAN_SIZE];
+ static XX *x;
+ double *xk;
+ int i;
+ static int off[2] = { 0, 0 };
+ static char init = 0;
+ double t;
+ double *ep0, *ep1, *ep2, *ep3, *ep4, *ep5, *ep6, *ep7;
+ if (!init) {
+ x = (XX *) mem_alloc (sizeof (XX), "x");
+ memset (x, 0, 2 * HAN_SIZE * sizeof (double));
+ init = 1;
+ }
+ xk = (*x)[k];
+
+ /* replace 32 oldest samples with 32 new samples */
+ for (i = 0; i < 32; i++)
+ xk[31 - i + off[k]] = (double) *(*buffer)++ / SCALE;
+
+ ep0 = &enwindow[0];
+ ep1 = &enwindow[64];
+ ep2 = &enwindow[128];
+ ep3 = &enwindow[192];
+ ep4 = &enwindow[256];
+ ep5 = &enwindow[320];
+ ep6 = &enwindow[384];
+ ep7 = &enwindow[448];
+
+ /* shift samples into proper window positions */
+ for (i = 0; i < 64; i++) {
+ t = xk[(i + off[k]) & (512 - 1)] * *ep0++;
+ t += xk[(i + 64 + off[k]) & (512 - 1)] * *ep1++;
+ t += xk[(i + 128 + off[k]) & (512 - 1)] * *ep2++;
+ t += xk[(i + 192 + off[k]) & (512 - 1)] * *ep3++;
+ t += xk[(i + 256 + off[k]) & (512 - 1)] * *ep4++;
+ t += xk[(i + 320 + off[k]) & (512 - 1)] * *ep5++;
+ t += xk[(i + 384 + off[k]) & (512 - 1)] * *ep6++;
+ t += xk[(i + 448 + off[k]) & (512 - 1)] * *ep7++;
+ z[i] = t;
+ }
+
+ off[k] += 480; /*offset is modulo (HAN_SIZE-1) */
+ off[k] &= HAN_SIZE - 1;
+
+}
+
+
+/************************************************************************
+*
+* filter_subband()
+*
+* PURPOSE: Calculates the analysis filter bank coefficients
+*
+* SEMANTICS:
+* The windowed samples #z# is filtered by the digital filter matrix #m#
+* to produce the subband samples #s#. This done by first selectively
+* picking out values from the windowed samples, and then multiplying
+* them by the filter matrix, producing 32 subband samples.
+*
+************************************************************************/
+void filter_subband (double z[HAN_SIZE], double s[SBLIMIT])
+{
+ double yprime[32];
+ register int i, j;
+
+ static double m[16][32];
+ static int init = 0;
+
+ if (init == 0) {
+ init++;
+ create_dct_matrix (m);
+ }
+
+ yprime[0] = z[16];
+ for (i = 1; i <= 16; i++)
+ yprime[i] = z[i + 16] + z[16 - i];
+ for (i = 17; i <= 31; i++)
+ yprime[i] = z[i + 16] - z[80 - i];
+
+ for (i = 15; i >= 0; i--) {
+ register double s0 = 0.0, s1 = 0.0;
+ register double *mp = m[i];
+ register double *xinp = yprime;
+ for (j = 0; j < 8; j++) {
+ s0 += *mp++ * *xinp++;
+ s1 += *mp++ * *xinp++;
+ s0 += *mp++ * *xinp++;
+ s1 += *mp++ * *xinp++;
+ }
+ s[i] = s0 + s1;
+ s[31 - i] = s0 - s1;
+ }
+}
+#endif //REFERENCECODE
+
+void create_dct_matrix (double filter[16][32])
+{
+ register int i, k;
+
+ for (i = 0; i < 16; i++)
+ for (k = 0; k < 32; k++) {
+ if ((filter[i][k] = 1e9 * cos ((double) ((2 * i + 1) * k * PI64))) >= 0)
+ modf (filter[i][k] + 0.5, &filter[i][k]);
+ else
+ modf (filter[i][k] - 0.5, &filter[i][k]);
+ filter[i][k] *= 1e-9;
+ }
+}
+
+
+#ifdef NEWWS
+/***********************************************************************
+ An implementation of a modified window subband as seen in Kumar & Zubair's
+"A high performance software implentation of mpeg audio encoder"
+I think from IEEE ASCAP 1996 proceedings
+
+input: shift in 32*12 (384) new samples into a 864 point buffer.
+ch - which channel we're looking at.
+
+This routine basically does 12 calls to window subband all in one go.
+Not yet called in code. here for testing only.
+************************************************************************/
+void window_subband12 (short **buffer, int ch)
+{
+ static double x[2][864]; /* 2 channels, 864 buffer for each */
+ double *xk;
+ double t[12]; /* a temp buffer for summing values */
+ double y[12][64]; /* 12 output arrays of 64 values */
+ int i, j, k, m;
+ static double c[512]; /* enwindow array */
+ static int init = 0;
+ double c0;
+
+ xk = x[ch]; /* an easier way of referencing the array */
+
+ /* shift 384 new samples into the buffer */
+ for (i = 863; i >= 384; i--)
+ xk[i] = xk[i - 384];
+ for (i = 383; i >= 0; i--)
+ xk[i] = (double) *(*buffer)++ / SCALE;
+
+ for (j = 0; j < 64; j++) {
+ for (k = 0; k < 12; k++)
+ t[k] = 0;
+ for (i = 0; i < 8; i++) {
+ m = i * 64 + j;
+ c0 = c[m];
+ t[0] += c0 * xk[m + 352];
+ t[1] += c0 * xk[m + 320];
+ t[2] += c0 * xk[m + 288];
+ t[3] += c0 * xk[m + 256];
+ t[4] += c0 * xk[m + 224];
+ t[5] += c0 * xk[m + 192];
+ t[6] += c0 * xk[m + 160];
+ t[7] += c0 * xk[m + 128];
+ t[8] += c0 * xk[m + 96];
+ t[9] += c0 * xk[m + 64];
+ t[10] += c0 * xk[m + 32];
+ t[11] += c0 * xk[m];
+ }
+ for (i = 0; i < 12; i++) {
+ y[i][j] = t[i];
+ }
+ }
+}
+#endif /* NEWWS */
+
+
+//____________________________________________________________________________
+//____ WindowFilterSubband() _________________________________________
+//____ RS&A - Feb 2003 _______________________________________________________
+void WindowFilterSubband (short *pBuffer, int ch, double s[SBLIMIT])
+{
+ register int i, j;
+ int pa, pb, pc, pd, pe, pf, pg, ph;
+ double t;
+ double *dp, *dp2;
+ double *pEnw;
+ double y[64];
+ double yprime[32];
+
+ static double x[2][512];
+ static double m[16][32];
+ static int init = 0;
+ static int off[2];
+ static int half[2];
+
+ if (init == 0) {
+ init++;
+ off[0] = 0;
+ off[1] = 0;
+ half[0] = 0;
+ half[1] = 0;
+ for (i = 0; i < 2; i++)
+ for (j = 0; j < 512; j++)
+ x[i][j] = 0;
+ create_dct_matrix (m);
+ }
+
+ dp = x[ch] + off[ch] + half[ch] * 256;
+
+ /* replace 32 oldest samples with 32 new samples */
+ for (i = 0; i < 32; i++)
+ dp[(31 - i) * 8] = (double) pBuffer[i] / SCALE;
+
+ // looks like "school example" but does faster ...
+ dp = (x[ch] + half[ch] * 256);
+ pa = off[ch];
+ pb = (pa + 1) % 8;
+ pc = (pa + 2) % 8;
+ pd = (pa + 3) % 8;
+ pe = (pa + 4) % 8;
+ pf = (pa + 5) % 8;
+ pg = (pa + 6) % 8;
+ ph = (pa + 7) % 8;
+
+ for (i = 0; i < 32; i++) {
+ dp2 = dp + i * 8;
+ pEnw = enwindow + i;
+ t = dp2[pa] * pEnw[0];
+ t += dp2[pb] * pEnw[64];
+ t += dp2[pc] * pEnw[128];
+ t += dp2[pd] * pEnw[192];
+ t += dp2[pe] * pEnw[256];
+ t += dp2[pf] * pEnw[320];
+ t += dp2[pg] * pEnw[384];
+ t += dp2[ph] * pEnw[448];
+ y[i] = t;
+ }
+
+ yprime[0] = y[16]; // Michael Chenīs dct filter
+
+ dp = half[ch] ? x[ch] : (x[ch] + 256);
+ pa = half[ch] ? (off[ch] + 1) & 7 : off[ch];
+ pb = (pa + 1) % 8;
+ pc = (pa + 2) % 8;
+ pd = (pa + 3) % 8;
+ pe = (pa + 4) % 8;
+ pf = (pa + 5) % 8;
+ pg = (pa + 6) % 8;
+ ph = (pa + 7) % 8;
+
+ for (i = 0; i < 32; i++) {
+ dp2 = dp + i * 8;
+ pEnw = enwindow + i + 32;
+ t = dp2[pa] * pEnw[0];
+ t += dp2[pb] * pEnw[64];
+ t += dp2[pc] * pEnw[128];
+ t += dp2[pd] * pEnw[192];
+ t += dp2[pe] * pEnw[256];
+ t += dp2[pf] * pEnw[320];
+ t += dp2[pg] * pEnw[384];
+ t += dp2[ph] * pEnw[448];
+ y[i + 32] = t;
+ // 1st pass on Michael Chenīs dct filter
+ if (i > 0 && i < 17)
+ yprime[i] = y[i + 16] + y[16 - i];
+ }
+
+ // 2nd pass on Michael Chenīs dct filter
+ for (i = 17; i < 32; i++)
+ yprime[i] = y[i + 16] - y[80 - i];
+
+ for (i = 15; i >= 0; i--) {
+ register double s0 = 0.0, s1 = 0.0;
+ register double *mp = m[i];
+ register double *xinp = yprime;
+ for (j = 0; j < 8; j++) {
+ s0 += *mp++ * *xinp++;
+ s1 += *mp++ * *xinp++;
+ s0 += *mp++ * *xinp++;
+ s1 += *mp++ * *xinp++;
+ }
+ s[i] = s0 + s1;
+ s[31 - i] = s0 - s1;
+ }
+
+ half[ch] = (half[ch] + 1) & 1;
+ if (half[ch] == 1)
+ off[ch] = (off[ch] + 7) & 7;
+}
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