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changeset 1119:042f189bd225 trunk

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[svn] - rename psycho.c to paranoia.c, as other stuff will be here (such as frequency locking code, to fix warble, etc.)
author nenolod
date Tue, 30 May 2006 22:52:11 -0700
parents 9ebbf4d00d89
children 8e6b7ed7dd88
files Plugins/Input/mpg123/Makefile.in Plugins/Input/mpg123/paranoia.c Plugins/Input/mpg123/psycho.c
diffstat 3 files changed, 448 insertions(+), 448 deletions(-) [+]
line wrap: on
line diff
--- a/Plugins/Input/mpg123/Makefile.in	Tue May 30 22:37:19 2006 -0700
+++ b/Plugins/Input/mpg123/Makefile.in	Tue May 30 22:52:11 2006 -0700
@@ -9,7 +9,7 @@
 	decode_2to1.c decode_4to1.c \
 	layer1.c layer2.c layer3.c \
 	tabinit.c http.c \
-	dxhead.c decode.c dct64.c psycho.c interface.c
+	dxhead.c decode.c dct64.c paranoia.c interface.c
 
 OBJECTS = ${SOURCES:.c=.o} tag_c_hacked.o
 
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/Plugins/Input/mpg123/paranoia.c	Tue May 30 22:52:11 2006 -0700
@@ -0,0 +1,447 @@
+/* libmpgdec: An advanced MPEG layer 1/2/3 decoder.
+ * psycho.c: Psychoaccoustic modeling.
+ *
+ * Copyright (C) 2005-2006 William Pitcock <nenolod@nenolod.net>
+ * Portions copyright (C) 2001 Rafal Bosak <gyver@fanthom.irc.pl>
+ * Portions copyright (C) 1999 Michael Hipp
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ * 
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ * 
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <math.h>
+
+#include "common.h"
+
+int bext_level;
+int stereo_level;
+int filter_level;
+int harmonics_level;
+int bext_sfactor;
+int stereo_sfactor;
+int harmonics_sfactor;
+int enable_plugin = 1;
+int lsine[65536];
+int rsine[65536];
+
+/*
+ * Initialize and configure the psychoaccoustics engine.
+ */
+void psycho_init(void)
+{
+    int i, x;
+    double lsum;
+    double rsum;
+
+    bext_level = 28;
+    stereo_level = stereo_sfactor = 16;
+    filter_level = 3;
+    harmonics_level = harmonics_sfactor = 43;
+
+    bext_sfactor = (float)(((float)16384 * 10) / (float)(bext_level + 1)) + (float)(102 - bext_level) * 128;
+
+#define COND 0
+    /* calculate sinetables */
+    for (i = 0; i < 32768; ++i)
+    {
+	lsum = rsum = 0;
+	if (COND || i < 32768 )lsum+=  ((cos((double)i * 3.141592535 / 32768/2	)) + 0) / 2;
+	if (COND || i < 16384 )rsum-=  ((cos((double)i * 3.141592535 / 16384/2	)) + 0) / 4;
+	rsum = lsum;
+
+	if (COND || i < 8192 ) lsum += ((cos((double)i * 3.141592535 / 8192/2	 )) + 0) /  8;
+	if (COND || i < 5641 ) rsum += ((cos((double)i * 3.141592535 / 5641.333333/2)) + 0) /  8;
+
+	lsine[32768 + i] = (double)(lsum - 0.5) * 32768 * 1.45;
+	lsine[32768 - i] = lsine[32768 + i];
+	rsine[32768 + i] = (double)(rsum - 0.5) * 32768 * 1.45;
+	rsine[32768 - i] = rsine[32768 + i];
+
+	x = i;
+    }
+}
+
+/*
+ * This routine computes a reverb for the track.
+ */
+
+#define DELAY2 21000
+#define DELAY1 35000
+#define DELAY3 14000
+#define DELAY4 5
+
+#define BUF_SIZE DELAY1+DELAY2+DELAY3+DELAY4
+
+void echo3d(gint16 *data, int datasize)
+{
+  int x;
+  int left, right, dif, left0, right0, left1, right1, left2, right2, left3, right3, left4, right4, leftc, rightc, lsf, rsf;
+  static int left0p = 0, right0p = 0;
+  static int rangeErrorsUp = 0;
+  static int rangeErrorsDown = 0;
+  gint16 *dataptr;
+  static int l0, l1, l2, r0, r1, r2, ls, rs, ls1, rs1;
+  static int ll0, ll1, ll2, rr0, rr1, rr2;
+  static int lharmb = 0, rharmb = 0, lhfb = 0, rhfb = 0;
+  int lharm0, rharm0;
+  static gint16 buf[BUF_SIZE];
+  static int bufPos1 = 1 + BUF_SIZE - DELAY1;
+  static int bufPos2 = 1 + BUF_SIZE - DELAY1 - DELAY2;
+  static int bufPos3 = 1 + BUF_SIZE - DELAY1 - DELAY2 - DELAY3;
+  static int bufPos4 = 1 + BUF_SIZE - DELAY1 - DELAY2 - DELAY3 - DELAY4;
+  dataptr = data;
+
+  for (x = 0; x < datasize; x += 4) {
+
+    // ************ load sample **********
+    left0 = dataptr[0];	
+    right0 = dataptr[1];
+
+    // ************ slightly expand stereo for direct input **********
+    ll0=left0;rr0=right0;
+    dif = (ll0+ll1+ll2 - rr0-rr1-rr2) * stereo_sfactor / 256;
+    left0 += dif;
+    right0 -= dif;
+    ll2= ll1; ll1= ll0;
+    rr2= rr1; rr1= rr0;
+    
+    // ************ echo from buffer - first echo **********
+    // ************  **********
+    left1 = buf[bufPos1++];
+    if (bufPos1 == BUF_SIZE)
+      bufPos1 = 0;
+    right1 = buf[bufPos1++];
+    if (bufPos1 == BUF_SIZE)
+      bufPos1 = 0;
+
+    // ************ highly expand stereo for first echo **********
+    dif = (left1 - right1);
+    left1 = left1 + dif;
+    right1 = right1 - dif;
+
+    // ************ second echo  **********
+    left2 = buf[bufPos2++];
+    if (bufPos2 == BUF_SIZE)
+      bufPos2 = 0;	
+    right2 = buf[bufPos2++];
+    if (bufPos2 == BUF_SIZE)
+      bufPos2 = 0;
+
+    // ************ expand stereo for second echo **********
+    dif = (left2 - right2);
+    left2 = left2 - dif;
+    right2 = right2 - dif;
+
+    // ************ third echo  **********
+    left3 = buf[bufPos3++];
+    if (bufPos3 == BUF_SIZE)
+      bufPos3 = 0;	
+    right3 = buf[bufPos3++];
+    if (bufPos3 == BUF_SIZE)
+      bufPos3 = 0;
+
+    // ************ fourth echo  **********
+    left4 = buf[bufPos4++];
+    if (bufPos4 == BUF_SIZE)
+      bufPos4 = 0;	
+    right4 = buf[bufPos4++];
+    if (bufPos4 == BUF_SIZE)
+      bufPos4 = 0;
+
+    left3 = (left4+left3) / 2;
+    right3 = (right4+right3) / 2;
+
+    // ************ expand stereo for second echo **********
+    dif = (left4 - right4);
+    left3 = left3 - dif;
+    right3 = right3 - dif;
+
+    // ************ a weighted sum taken from reverb buffer **********
+    leftc = left1 / 9 + right2 /8  + left3 / 8;
+    rightc = right1 / 11 + left2 / 9 + right3 / 10;
+
+    left = left0p;
+    right = right0p;
+
+    l0 = leftc + left0 / 2;
+    r0 = rightc + right0 / 2;
+
+    ls = l0 + l1 + l2;	// do not reverb high frequencies (filter)
+    rs = r0 + r1 + r2;  //
+
+    // ************ add some extra even harmonics **********
+    // ************ or rather specific nonlinearity
+
+    lhfb = lhfb + (ls * 32768 - lhfb) / 32;
+    rhfb = rhfb + (rs * 32768 - rhfb) / 32;
+
+    lsf = ls - lhfb / 32768;
+    rsf = rs - rhfb / 32768;
+
+    lharm0 = 0
+	+ ((lsf + 10000) * ((((lsine[((lsf/4) + 32768 + 65536) % 65536] * harmonics_sfactor)) / 64))) / 32768
+       	- ((lsine[((lsf/4) + 32768 +65536) % 65536]) * harmonics_sfactor) / 128
+	;
+
+    rharm0 =
+	+ ((rsf + 10000) * ((((lsine[((rsf/4) + 32768 + 65536) % 65536] * harmonics_sfactor)) / 64))) / 32768
+     	- ((rsine[((rsf/4) + 32768 +65536) % 65536]) * harmonics_sfactor) / 128
+	;
+
+    lharmb = lharmb + (lharm0 * 32768 - lharmb) / 16384;
+    rharmb = rharmb + (rharm0 * 32768 - rharmb) / 16384;
+
+    // ************ for convolution filters **********
+    l2= l1; r2= r1;
+    l1 = l0; r1 = r0;
+    ls1 = ls; rs1 = rs;
+
+    left  = 0 + lharm0 - lharmb / 32768 + left;
+    right = 0 + rharm0 - rharmb / 32768 + right;
+
+    left0p = left0;
+    right0p = right0;
+
+
+    // ************ limiter **********
+    if (left < -32768) {
+	left = -32768;	// limit
+	rangeErrorsDown++;
+    }
+    else if (left > 32767) {
+	left = 32767;
+	rangeErrorsUp++;
+    }
+    if (right < -32768) {
+	right = -32768;
+	rangeErrorsDown++;
+    }
+    else if (right > 32767) {
+	right = 32767;
+	rangeErrorsUp++;
+    }
+    // ************ store sample **********
+    dataptr[0] = left;
+    dataptr[1] = right;
+    dataptr += 2;
+
+   }
+}
+
+/*
+ * simple pith shifter, plays short fragments at 1.5 speed
+ */
+void pitchShifter(const int lin, const int rin, int *lout, int *rout)
+{ 
+#define SH_BUF_SIZE 100 * 3
+    static gint16 shBuf[SH_BUF_SIZE];
+    static int shBufPos = SH_BUF_SIZE - 6;
+    static int shBufPos1 = SH_BUF_SIZE - 6;
+    static int cond;
+
+    shBuf[shBufPos++] = lin;
+    shBuf[shBufPos++] = rin;
+
+    if (shBufPos == SH_BUF_SIZE) shBufPos = 0;
+
+    switch (cond){
+	case 1:
+	    *lout = (shBuf[shBufPos1 + 0] * 2 + shBuf[shBufPos1 + 2])/4;
+	    *rout = (shBuf[shBufPos1 + 1] * 2 + shBuf[shBufPos1 + 3])/4;
+	    break;
+	case 0:
+	    *lout = (shBuf[shBufPos1 + 4] * 2 + shBuf[shBufPos1 + 2])/4;
+	    *rout = (shBuf[shBufPos1 + 5] * 2 + shBuf[shBufPos1 + 3])/4;
+	    cond = 2;
+	    shBufPos1 += 6;
+	    if (shBufPos1 == SH_BUF_SIZE) {
+		shBufPos1 = 0;
+	    }
+	    break;
+    }
+    cond--;
+}
+
+
+struct Interpolation{
+    int acount;		// counter
+    int lval, rval;	// value
+    int sal, sar;	// sum
+    int al, ar;		
+    int a1l, a1r;
+};
+
+/*
+ * interpolation routine for ampliude and "energy"
+ */
+static inline void interpolate(struct Interpolation *s, int l, int r)
+{
+#define AMPL_COUNT 64
+    int a0l, a0r, dal = 0, dar = 0;
+
+    if (l < 0) l = -l;
+    if (r < 0) r = -r;
+
+    s->lval += l / 8;
+    s->rval += r / 8;
+
+    s->lval = (s->lval * 120) / 128;
+    s->rval = (s->rval * 120) / 128;
+
+    s->sal += s->lval;
+    s->sar += s->rval;
+
+    s->acount++;
+    if (s->acount == AMPL_COUNT){
+	s->acount = 0;
+	a0l = s->a1l;
+	a0r = s->a1r;
+	s->a1l = s->sal / AMPL_COUNT;
+	s->a1r = s->sar / AMPL_COUNT;
+	s->sal = 0;
+	s->sar = 0;
+	dal = s->a1l - a0l;
+	dar = s->a1r - a0r;
+	s->al = a0l * AMPL_COUNT;
+	s->ar = a0r * AMPL_COUNT;
+    }
+
+    s->al += dal;
+    s->ar += dar;
+}
+
+/*
+ * calculate scalefactor for mixer
+ */
+inline int calc_scalefactor(int a, int e)
+{
+    int x;
+
+    if (a > 8192) a = 8192;
+    else if (a < 0) a = 0;
+
+    if (e > 8192) e = 8192;
+    else if (e < 0) e = 0;
+
+    x = ((e+500) * 4096 )/ (a + 300) + e;
+
+    if (x > 16384) x = 16384;
+    else if (x < 0) x = 0;
+    return x;
+}
+
+static struct Interpolation bandext_energy;
+static struct Interpolation bandext_amplitude;
+
+/*
+ * exact bandwidth extender ("exciter") routine
+ */
+static void bandext(gint16 *data, const int datasize)
+{
+
+    int x;
+    static int saw;		// test stuff
+    int left, right;
+    gint16 *dataptr = data;
+    static int lprev0, rprev0, lprev1, rprev1, lprev2, rprev2;
+    int left0, right0, left1, right1, left2, right2, left3, right3;
+    static int lamplUp, lamplDown;
+    static int ramplUp, ramplDown;
+    int lampl, rampl;
+    int tmp;
+
+    for (x = 0; x < datasize; x += 4) {
+
+	// ************ load sample **********
+       	left0 = dataptr[0];
+       	right0 = dataptr[1];
+
+	// ************ highpass filter part 1 **********
+	left1  = (left0  - lprev0) * 56880 / 65536;
+	right1 = (right0 - rprev0) * 56880 / 65536;
+
+	left2  = (left1  - lprev1) * 56880 / 65536;
+	right2 = (right1 - rprev1) * 56880 / 65536;
+
+	left3  = (left2  - lprev2) * 56880 / 65536;
+	right3 = (right2 - rprev2) * 56880 / 65536;
+
+	switch (filter_level){
+	    case 1:
+	       	pitchShifter(left1, right1, &left, &right);
+		break;
+	    case 2:
+	       	pitchShifter(left2, right2, &left, &right);
+		break;
+	    case 3:
+	       	pitchShifter(left3, right3, &left, &right);
+		break;
+	}
+
+	// ************ amplitude detector ************
+	tmp = left1 + lprev1;
+	if      (tmp * 16 > lamplUp  ) lamplUp   += (tmp - lamplUp  );
+	else if (tmp * 16 < lamplDown) lamplDown += (tmp - lamplDown);
+	lamplUp   = (lamplUp   * 1000) /1024;
+	lamplDown = (lamplDown * 1000) /1024;
+	lampl = lamplUp - lamplDown;
+
+	tmp = right1 + rprev1;
+	if      (tmp * 16 > ramplUp  ) ramplUp   += (tmp - ramplUp  );
+	else if (tmp * 16 < ramplDown) ramplDown += (tmp - ramplDown);
+	ramplUp   = (ramplUp   * 1000) /1024;
+	ramplDown = (ramplDown * 1000) /1024;
+	rampl = ramplUp - ramplDown;
+
+	interpolate(&bandext_amplitude, lampl, rampl);
+
+	// ************ "sound energy" detector (approx. spectrum complexity) ***********
+	interpolate(&bandext_energy, left0  - lprev0, right0 - rprev0);
+
+	// ************ mixer ***********
+	left   = left0 + left  * calc_scalefactor(bandext_amplitude.lval, bandext_energy.lval) / bext_sfactor; 
+	right  = right0 + right * calc_scalefactor(bandext_amplitude.rval, bandext_energy.rval) / bext_sfactor; //16384
+
+	saw = (saw + 2048) & 0x7fff;
+
+	lprev0 = left0;
+	rprev0 = right0;
+	lprev1 = left1;
+	rprev1 = right1;
+	lprev2 = left2;
+	rprev2 = right2;
+
+	if (left < -32768) left = -32768;
+       	else if (left > 32767) left = 32767;
+       	if (right < -32768) right = -32768;
+       	else if (right > 32767) right = 32767;
+
+	dataptr[0] = left;
+       	dataptr[1] = right;
+       	dataptr += 2;
+    }
+}
+
+int psycho_process(void *data, int length, int nch)
+{
+    if (nch != 2)
+        return length;			/* XXX: we cant process mono yet */
+
+    echo3d((gint16 *)data, length);
+    bandext((gint16 *)data, length);
+    return length;
+}
+
--- a/Plugins/Input/mpg123/psycho.c	Tue May 30 22:37:19 2006 -0700
+++ /dev/null	Thu Jan 01 00:00:00 1970 +0000
@@ -1,447 +0,0 @@
-/* libmpgdec: An advanced MPEG layer 1/2/3 decoder.
- * psycho.c: Psychoaccoustic modeling.
- *
- * Copyright (C) 2005-2006 William Pitcock <nenolod@nenolod.net>
- * Portions copyright (C) 2001 Rafal Bosak <gyver@fanthom.irc.pl>
- * Portions copyright (C) 1999 Michael Hipp
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- * 
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * 
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-
-#include "common.h"
-
-int bext_level;
-int stereo_level;
-int filter_level;
-int harmonics_level;
-int bext_sfactor;
-int stereo_sfactor;
-int harmonics_sfactor;
-int enable_plugin = 1;
-int lsine[65536];
-int rsine[65536];
-
-/*
- * Initialize and configure the psychoaccoustics engine.
- */
-void psycho_init(void)
-{
-    int i, x;
-    double lsum;
-    double rsum;
-
-    bext_level = 28;
-    stereo_level = stereo_sfactor = 16;
-    filter_level = 3;
-    harmonics_level = harmonics_sfactor = 43;
-
-    bext_sfactor = (float)(((float)16384 * 10) / (float)(bext_level + 1)) + (float)(102 - bext_level) * 128;
-
-#define COND 0
-    /* calculate sinetables */
-    for (i = 0; i < 32768; ++i)
-    {
-	lsum = rsum = 0;
-	if (COND || i < 32768 )lsum+=  ((cos((double)i * 3.141592535 / 32768/2	)) + 0) / 2;
-	if (COND || i < 16384 )rsum-=  ((cos((double)i * 3.141592535 / 16384/2	)) + 0) / 4;
-	rsum = lsum;
-
-	if (COND || i < 8192 ) lsum += ((cos((double)i * 3.141592535 / 8192/2	 )) + 0) /  8;
-	if (COND || i < 5641 ) rsum += ((cos((double)i * 3.141592535 / 5641.333333/2)) + 0) /  8;
-
-	lsine[32768 + i] = (double)(lsum - 0.5) * 32768 * 1.45;
-	lsine[32768 - i] = lsine[32768 + i];
-	rsine[32768 + i] = (double)(rsum - 0.5) * 32768 * 1.45;
-	rsine[32768 - i] = rsine[32768 + i];
-
-	x = i;
-    }
-}
-
-/*
- * This routine computes a reverb for the track.
- */
-
-#define DELAY2 21000
-#define DELAY1 35000
-#define DELAY3 14000
-#define DELAY4 5
-
-#define BUF_SIZE DELAY1+DELAY2+DELAY3+DELAY4
-
-void echo3d(gint16 *data, int datasize)
-{
-  int x;
-  int left, right, dif, left0, right0, left1, right1, left2, right2, left3, right3, left4, right4, leftc, rightc, lsf, rsf;
-  static int left0p = 0, right0p = 0;
-  static int rangeErrorsUp = 0;
-  static int rangeErrorsDown = 0;
-  gint16 *dataptr;
-  static int l0, l1, l2, r0, r1, r2, ls, rs, ls1, rs1;
-  static int ll0, ll1, ll2, rr0, rr1, rr2;
-  static int lharmb = 0, rharmb = 0, lhfb = 0, rhfb = 0;
-  int lharm0, rharm0;
-  static gint16 buf[BUF_SIZE];
-  static int bufPos1 = 1 + BUF_SIZE - DELAY1;
-  static int bufPos2 = 1 + BUF_SIZE - DELAY1 - DELAY2;
-  static int bufPos3 = 1 + BUF_SIZE - DELAY1 - DELAY2 - DELAY3;
-  static int bufPos4 = 1 + BUF_SIZE - DELAY1 - DELAY2 - DELAY3 - DELAY4;
-  dataptr = data;
-
-  for (x = 0; x < datasize; x += 4) {
-
-    // ************ load sample **********
-    left0 = dataptr[0];	
-    right0 = dataptr[1];
-
-    // ************ slightly expand stereo for direct input **********
-    ll0=left0;rr0=right0;
-    dif = (ll0+ll1+ll2 - rr0-rr1-rr2) * stereo_sfactor / 256;
-    left0 += dif;
-    right0 -= dif;
-    ll2= ll1; ll1= ll0;
-    rr2= rr1; rr1= rr0;
-    
-    // ************ echo from buffer - first echo **********
-    // ************  **********
-    left1 = buf[bufPos1++];
-    if (bufPos1 == BUF_SIZE)
-      bufPos1 = 0;
-    right1 = buf[bufPos1++];
-    if (bufPos1 == BUF_SIZE)
-      bufPos1 = 0;
-
-    // ************ highly expand stereo for first echo **********
-    dif = (left1 - right1);
-    left1 = left1 + dif;
-    right1 = right1 - dif;
-
-    // ************ second echo  **********
-    left2 = buf[bufPos2++];
-    if (bufPos2 == BUF_SIZE)
-      bufPos2 = 0;	
-    right2 = buf[bufPos2++];
-    if (bufPos2 == BUF_SIZE)
-      bufPos2 = 0;
-
-    // ************ expand stereo for second echo **********
-    dif = (left2 - right2);
-    left2 = left2 - dif;
-    right2 = right2 - dif;
-
-    // ************ third echo  **********
-    left3 = buf[bufPos3++];
-    if (bufPos3 == BUF_SIZE)
-      bufPos3 = 0;	
-    right3 = buf[bufPos3++];
-    if (bufPos3 == BUF_SIZE)
-      bufPos3 = 0;
-
-    // ************ fourth echo  **********
-    left4 = buf[bufPos4++];
-    if (bufPos4 == BUF_SIZE)
-      bufPos4 = 0;	
-    right4 = buf[bufPos4++];
-    if (bufPos4 == BUF_SIZE)
-      bufPos4 = 0;
-
-    left3 = (left4+left3) / 2;
-    right3 = (right4+right3) / 2;
-
-    // ************ expand stereo for second echo **********
-    dif = (left4 - right4);
-    left3 = left3 - dif;
-    right3 = right3 - dif;
-
-    // ************ a weighted sum taken from reverb buffer **********
-    leftc = left1 / 9 + right2 /8  + left3 / 8;
-    rightc = right1 / 11 + left2 / 9 + right3 / 10;
-
-    left = left0p;
-    right = right0p;
-
-    l0 = leftc + left0 / 2;
-    r0 = rightc + right0 / 2;
-
-    ls = l0 + l1 + l2;	// do not reverb high frequencies (filter)
-    rs = r0 + r1 + r2;  //
-
-    // ************ add some extra even harmonics **********
-    // ************ or rather specific nonlinearity
-
-    lhfb = lhfb + (ls * 32768 - lhfb) / 32;
-    rhfb = rhfb + (rs * 32768 - rhfb) / 32;
-
-    lsf = ls - lhfb / 32768;
-    rsf = rs - rhfb / 32768;
-
-    lharm0 = 0
-	+ ((lsf + 10000) * ((((lsine[((lsf/4) + 32768 + 65536) % 65536] * harmonics_sfactor)) / 64))) / 32768
-       	- ((lsine[((lsf/4) + 32768 +65536) % 65536]) * harmonics_sfactor) / 128
-	;
-
-    rharm0 =
-	+ ((rsf + 10000) * ((((lsine[((rsf/4) + 32768 + 65536) % 65536] * harmonics_sfactor)) / 64))) / 32768
-     	- ((rsine[((rsf/4) + 32768 +65536) % 65536]) * harmonics_sfactor) / 128
-	;
-
-    lharmb = lharmb + (lharm0 * 32768 - lharmb) / 16384;
-    rharmb = rharmb + (rharm0 * 32768 - rharmb) / 16384;
-
-    // ************ for convolution filters **********
-    l2= l1; r2= r1;
-    l1 = l0; r1 = r0;
-    ls1 = ls; rs1 = rs;
-
-    left  = 0 + lharm0 - lharmb / 32768 + left;
-    right = 0 + rharm0 - rharmb / 32768 + right;
-
-    left0p = left0;
-    right0p = right0;
-
-
-    // ************ limiter **********
-    if (left < -32768) {
-	left = -32768;	// limit
-	rangeErrorsDown++;
-    }
-    else if (left > 32767) {
-	left = 32767;
-	rangeErrorsUp++;
-    }
-    if (right < -32768) {
-	right = -32768;
-	rangeErrorsDown++;
-    }
-    else if (right > 32767) {
-	right = 32767;
-	rangeErrorsUp++;
-    }
-    // ************ store sample **********
-    dataptr[0] = left;
-    dataptr[1] = right;
-    dataptr += 2;
-
-   }
-}
-
-/*
- * simple pith shifter, plays short fragments at 1.5 speed
- */
-void pitchShifter(const int lin, const int rin, int *lout, int *rout)
-{ 
-#define SH_BUF_SIZE 100 * 3
-    static gint16 shBuf[SH_BUF_SIZE];
-    static int shBufPos = SH_BUF_SIZE - 6;
-    static int shBufPos1 = SH_BUF_SIZE - 6;
-    static int cond;
-
-    shBuf[shBufPos++] = lin;
-    shBuf[shBufPos++] = rin;
-
-    if (shBufPos == SH_BUF_SIZE) shBufPos = 0;
-
-    switch (cond){
-	case 1:
-	    *lout = (shBuf[shBufPos1 + 0] * 2 + shBuf[shBufPos1 + 2])/4;
-	    *rout = (shBuf[shBufPos1 + 1] * 2 + shBuf[shBufPos1 + 3])/4;
-	    break;
-	case 0:
-	    *lout = (shBuf[shBufPos1 + 4] * 2 + shBuf[shBufPos1 + 2])/4;
-	    *rout = (shBuf[shBufPos1 + 5] * 2 + shBuf[shBufPos1 + 3])/4;
-	    cond = 2;
-	    shBufPos1 += 6;
-	    if (shBufPos1 == SH_BUF_SIZE) {
-		shBufPos1 = 0;
-	    }
-	    break;
-    }
-    cond--;
-}
-
-
-struct Interpolation{
-    int acount;		// counter
-    int lval, rval;	// value
-    int sal, sar;	// sum
-    int al, ar;		
-    int a1l, a1r;
-};
-
-/*
- * interpolation routine for ampliude and "energy"
- */
-static inline void interpolate(struct Interpolation *s, int l, int r)
-{
-#define AMPL_COUNT 64
-    int a0l, a0r, dal = 0, dar = 0;
-
-    if (l < 0) l = -l;
-    if (r < 0) r = -r;
-
-    s->lval += l / 8;
-    s->rval += r / 8;
-
-    s->lval = (s->lval * 120) / 128;
-    s->rval = (s->rval * 120) / 128;
-
-    s->sal += s->lval;
-    s->sar += s->rval;
-
-    s->acount++;
-    if (s->acount == AMPL_COUNT){
-	s->acount = 0;
-	a0l = s->a1l;
-	a0r = s->a1r;
-	s->a1l = s->sal / AMPL_COUNT;
-	s->a1r = s->sar / AMPL_COUNT;
-	s->sal = 0;
-	s->sar = 0;
-	dal = s->a1l - a0l;
-	dar = s->a1r - a0r;
-	s->al = a0l * AMPL_COUNT;
-	s->ar = a0r * AMPL_COUNT;
-    }
-
-    s->al += dal;
-    s->ar += dar;
-}
-
-/*
- * calculate scalefactor for mixer
- */
-inline int calc_scalefactor(int a, int e)
-{
-    int x;
-
-    if (a > 8192) a = 8192;
-    else if (a < 0) a = 0;
-
-    if (e > 8192) e = 8192;
-    else if (e < 0) e = 0;
-
-    x = ((e+500) * 4096 )/ (a + 300) + e;
-
-    if (x > 16384) x = 16384;
-    else if (x < 0) x = 0;
-    return x;
-}
-
-static struct Interpolation bandext_energy;
-static struct Interpolation bandext_amplitude;
-
-/*
- * exact bandwidth extender ("exciter") routine
- */
-static void bandext(gint16 *data, const int datasize)
-{
-
-    int x;
-    static int saw;		// test stuff
-    int left, right;
-    gint16 *dataptr = data;
-    static int lprev0, rprev0, lprev1, rprev1, lprev2, rprev2;
-    int left0, right0, left1, right1, left2, right2, left3, right3;
-    static int lamplUp, lamplDown;
-    static int ramplUp, ramplDown;
-    int lampl, rampl;
-    int tmp;
-
-    for (x = 0; x < datasize; x += 4) {
-
-	// ************ load sample **********
-       	left0 = dataptr[0];
-       	right0 = dataptr[1];
-
-	// ************ highpass filter part 1 **********
-	left1  = (left0  - lprev0) * 56880 / 65536;
-	right1 = (right0 - rprev0) * 56880 / 65536;
-
-	left2  = (left1  - lprev1) * 56880 / 65536;
-	right2 = (right1 - rprev1) * 56880 / 65536;
-
-	left3  = (left2  - lprev2) * 56880 / 65536;
-	right3 = (right2 - rprev2) * 56880 / 65536;
-
-	switch (filter_level){
-	    case 1:
-	       	pitchShifter(left1, right1, &left, &right);
-		break;
-	    case 2:
-	       	pitchShifter(left2, right2, &left, &right);
-		break;
-	    case 3:
-	       	pitchShifter(left3, right3, &left, &right);
-		break;
-	}
-
-	// ************ amplitude detector ************
-	tmp = left1 + lprev1;
-	if      (tmp * 16 > lamplUp  ) lamplUp   += (tmp - lamplUp  );
-	else if (tmp * 16 < lamplDown) lamplDown += (tmp - lamplDown);
-	lamplUp   = (lamplUp   * 1000) /1024;
-	lamplDown = (lamplDown * 1000) /1024;
-	lampl = lamplUp - lamplDown;
-
-	tmp = right1 + rprev1;
-	if      (tmp * 16 > ramplUp  ) ramplUp   += (tmp - ramplUp  );
-	else if (tmp * 16 < ramplDown) ramplDown += (tmp - ramplDown);
-	ramplUp   = (ramplUp   * 1000) /1024;
-	ramplDown = (ramplDown * 1000) /1024;
-	rampl = ramplUp - ramplDown;
-
-	interpolate(&bandext_amplitude, lampl, rampl);
-
-	// ************ "sound energy" detector (approx. spectrum complexity) ***********
-	interpolate(&bandext_energy, left0  - lprev0, right0 - rprev0);
-
-	// ************ mixer ***********
-	left   = left0 + left  * calc_scalefactor(bandext_amplitude.lval, bandext_energy.lval) / bext_sfactor; 
-	right  = right0 + right * calc_scalefactor(bandext_amplitude.rval, bandext_energy.rval) / bext_sfactor; //16384
-
-	saw = (saw + 2048) & 0x7fff;
-
-	lprev0 = left0;
-	rprev0 = right0;
-	lprev1 = left1;
-	rprev1 = right1;
-	lprev2 = left2;
-	rprev2 = right2;
-
-	if (left < -32768) left = -32768;
-       	else if (left > 32767) left = 32767;
-       	if (right < -32768) right = -32768;
-       	else if (right > 32767) right = 32767;
-
-	dataptr[0] = left;
-       	dataptr[1] = right;
-       	dataptr += 2;
-    }
-}
-
-int psycho_process(void *data, int length, int nch)
-{
-    if (nch != 2)
-        return length;			/* XXX: we cant process mono yet */
-
-    echo3d((gint16 *)data, length);
-    bandext((gint16 *)data, length);
-    return length;
-}
-