shithub: sox

--- a/configure.in

+++ b/configure.in

@@ -155,7 +155,6 @@

 			ac_cv_dev_alsa_dsp=no,

 			ac_cv_dev_alsa_dsp=no)

 	if test "$ac_cv_dev_alsa_dsp" = yes

 	then

 		AC_CHECK_HEADERS(linux/asound.h, alsa_dsp=yes)

@@ -248,7 +247,7 @@

 	if test "$ac_cv_dev_sun_audio" = yes

 	then

-		AC_CHECK_HEADER(sys/audioio.h, sun/audioio.h, sun_audio=yes)

+		AC_CHECK_HEADERS(sys/audioio.h sun/audioio.h, sun_audio=yes)

 		if test "$sun_audio" = auto

 		then

 			AC_WARN([No audioio.h to compile with SUN /dev/audio])

--- a/libst.h

+++ b/libst.h

@@ -51,7 +51,11 @@

 extern int rand();

 extern void srand(/* seed */);

 #endif

-extern void initrand();

+extern void st_initrand();

+extern LONG st_clip24();

+extern void st_sine();

+extern void st_triangle();

 #ifndef HAVE_STRERROR

 char *strerror(/*errorcode*/);

--- a/src/chorus.c

+++ b/src/chorus.c

@@ -69,6 +69,7 @@

 #include <math.h>

 #include <string.h>

 #include "st.h"

+#include "libst.h"

 #define MOD_SINE	0

 #define MOD_TRIANGLE	1

@@ -90,59 +91,6 @@

 	int	maxsamples, fade_out;

 } *chorus_t;

-/* Private data for SKEL file */

-LONG chorus_clip24(l)

-LONG l;

-{

-	if (l >= ((LONG)1 << 24))

-		return ((LONG)1 << 24) - 1;

-	else if (l <= -((LONG)1 << 24))

-		return -((LONG)1 << 24) + 1;

-	else

-		return l;

-}

-/* This was very painful.  We need a sine library. */

-void chorus_sine(buf, len, max, depth)

-int *buf;

-long len;

-int max;

-int depth;

-{

-	long i;

-	int offset;

-	double val;

-	offset = max - depth;

-	for (i = 0; i < len; i++) {

-		val = sin((double)i/(double)len * 2.0 * M_PI);

-		buf[i] = offset + (int) (val * (double)depth);

-	}

-}

-void chorus_triangle(buf, len, max, depth)

-int *buf;

-long len;

-int max;

-int depth;

-{

-	long i;

-	int offset;

-	double val;

-	offset = max - 2 * depth;

-	for (i = 0; i < len / 2; i++) {

-		val = i * 2.0 / len;

-		buf[i] = offset + (int) (val * 2.0 * (double)depth);

-	}

-	for (i = len / 2; i < len ; i++) {

-		val = (len - i) * 2.0 / len;

-		buf[i] = offset + (int) (val * 2.0 * (double)depth);

-	}

-}

/*

  * Process options

*/

@@ -226,10 +174,10 @@

 			fail("chorus: Cannot malloc %d bytes!\n",

 				sizeof(int) * chorus->length[i]);

 		if ( chorus->modulation[i] == MOD_SINE )

-			chorus_sine(chorus->lookup_tab[i], chorus->length[i],

+			st_sine(chorus->lookup_tab[i], chorus->length[i],

 				chorus->samples[i] - 1, chorus->depth_samples[i]);

 		else

-			chorus_triangle(chorus->lookup_tab[i], chorus->length[i],

+			st_triangle(chorus->lookup_tab[i], chorus->length[i],

 				chorus->samples[i] - 1, chorus->depth_samples[i]);

 		chorus->phase[i] = 0;

@@ -285,7 +233,7 @@

 			chorus->maxsamples] * chorus->decay[i];

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_out * chorus->out_gain;

-		out = chorus_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Mix decay of delay and input */

 		chorus->chorusbuf[chorus->counter] = d_in;

@@ -324,7 +272,7 @@

 		chorus->maxsamples] * chorus->decay[i];

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_out * chorus->out_gain;

-		out = chorus_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Mix decay of delay and input */

 		chorus->chorusbuf[chorus->counter] = d_in;

--- a/src/echo.c

+++ b/src/echo.c

@@ -61,6 +61,7 @@

 #endif

 #include <math.h>

 #include "st.h"

+#include "libst.h"

 #define DELAY_BUFSIZ ( 50L * MAXRATE )

 #define MAX_ECHOS 7	/* 24 bit x ( 1 + MAX_ECHOS ) = */

@@ -79,20 +80,6 @@

 /* Private data for SKEL file */

-/* If we are not carefull with the output volume */

-LONG echo_clip24(l)

-LONG l;

-{

-	if (l >= ((LONG)1 << 24))

-		return ((LONG)1 << 24) - 1;

-	else if (l <= -((LONG)1 << 24))

-		return -((LONG)1 << 24) + 1;

-	else

-		return l;

-}

/*

  * Process options

*/

@@ -200,7 +187,7 @@

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_out * echo->out_gain;

-		out = echo_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Store input in delay buffer */

 		echo->delay_buf[echo->counter] = d_in;

@@ -236,7 +223,7 @@

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_out * echo->out_gain;

-		out = echo_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Store input in delay buffer */

 		echo->delay_buf[echo->counter] = d_in;

--- a/src/echos.c

+++ b/src/echos.c

@@ -52,6 +52,7 @@

 #endif

 #include <math.h>

 #include "st.h"

+#include "libst.h"

 #define DELAY_BUFSIZ ( 50L * MAXRATE )

 #define MAX_ECHOS 7	/* 24 bit x ( 1 + MAX_ECHOS ) = */

@@ -69,21 +70,6 @@

 /* Private data for SKEL file */

-/* If we are not carefull with the output volume */

-LONG echos_clip24(l)

-LONG l;

-{

-	if (l >= ((LONG)1 << 24))

-		return ((LONG)1 << 24) - 1;

-	else if (l <= -((LONG)1 << 24))

-		return -((LONG)1 << 24) + 1;

-	else

-		return l;

-}

/*

  * Process options

*/

@@ -188,7 +174,7 @@

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_out * echos->out_gain;

-		out = echos_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Mix decay of delays and input */

 		for ( j = 0; j < echos->num_delays; j++ ) {

@@ -230,7 +216,7 @@

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_out * echos->out_gain;

-		out = echos_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Mix decay of delays and input */

 		for ( j = 0; j < echos->num_delays; j++ ) {

--- a/src/flanger.c

+++ b/src/flanger.c

@@ -59,6 +59,7 @@

 #include <math.h>

 #include <string.h>

 #include "st.h"

+#include "libst.h"

 #define MOD_SINE	0

 #define MOD_TRIANGLE	1

@@ -79,51 +80,6 @@

 /* Private data for SKEL file */

-LONG flanger_clip24(l)

-LONG l;

-{

-	if (l >= ((LONG)1 << 24))

-		return ((LONG)1 << 24) - 1;

-	else if (l <= -((LONG)1 << 24))

-		return -((LONG)1 << 24) + 1;

-	else

-		return l;

-}

-/* This was very painful.  We need a sine library. */

-void flanger_sine(buf, len, depth)

-int *buf;

-long len;

-long depth;

-{

-	long i;

-	double val;

-	for (i = 0; i < len; i++) {

-		val = sin((double)i/(double)len * 2.0 * M_PI);

-		buf[i] = (int) ((1.0 + val) * depth / 2.0);

-	}

-}

-void flanger_triangle(buf, len, depth)

-int *buf;

-long len;

-long depth;

-{

-	long i;

-	double val;

-	for (i = 0; i < len / 2; i++) {

-		val = i * 2.0 / len;

-		buf[i] = (int) (val * depth);

-	}

-	for (i = len / 2; i < len ; i++) {

-		val = (len - i) * 2.0 / len;

-		buf[i] = (int) (val * depth);

-	}

-}

/*

  * Process options

*/

@@ -200,10 +156,10 @@

 			sizeof(int) * flanger->length);

 	if ( flanger->modulation == MOD_SINE )

-		flanger_sine(flanger->lookup_tab, flanger->length,

+		st_sine(flanger->lookup_tab, flanger->length,

 			flanger->maxsamples - 1);

 	else

-		flanger_triangle(flanger->lookup_tab, flanger->length,

+		st_triangle(flanger->lookup_tab, flanger->length,

 			flanger->maxsamples - 1);

 	flanger->counter = 0;

 	flanger->phase = 0;

@@ -237,7 +193,7 @@

 	flanger->maxsamples] * flanger->decay;

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_out * flanger->out_gain;

-		out = flanger_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Mix decay of delay and input */

 		flanger->flangerbuf[flanger->counter] = d_in;

@@ -272,7 +228,7 @@

 	flanger->maxsamples] * flanger->decay;

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_out * flanger->out_gain;

-		out = flanger_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Mix decay of delay and input */

 		flanger->flangerbuf[flanger->counter] = d_in;

--- a/src/mask.c

+++ b/src/mask.c

@@ -38,7 +38,7 @@

 		fail("Mask effect takes no options.");

 	/* should take # of bits */

-	initrand();

+	st_initrand();

/*

--- a/src/misc.c

+++ b/src/misc.c

@@ -11,12 +11,14 @@

  * Sound Tools miscellaneous stuff.

*/

-#include "st.h"

-#include "version.h"

-#include "patchlvl.h"

 #include <stdio.h>

 #include <stdlib.h>

 #include <time.h>

+#include <math.h>

+#include "st.h"

+#include "libst.h"

+#include "version.h"

+#include "patchlvl.h"

 char *sizes[] = {

 	"NONSENSE!",

@@ -256,11 +258,56 @@

 #endif

 /* Util to set initial seed so that we are a little less non-random */

-void initrand() {

+void st_initrand() {

 	time_t t;

 	time(&t);

 	srand(t);

+}

+LONG st_clip24(l)

+LONG l;

+{

+    if (l >= ((LONG)1 << 23))

+	return ((LONG)1 << 23) - 1;

+    else if (l <= -((LONG)1 << 23))

+        return -((LONG)1 << 23) + 1;

+    else

+        return l;

+}

+/* This was very painful.  We need a sine library. */

+void st_sine(buf, len, depth)

+int *buf;

+long len;

+long depth;

+{

+	long i;

+	double val;

+	for (i = 0; i < len; i++) {

+		val = sin((double)i/(double)len * 2.0 * M_PI);

+		buf[i] = (int) ((1.0 + val) * depth / 2.0);

+	}

+}

+void st_triangle(buf, len, depth)

+int *buf;

+long len;

+long depth;

+{

+	long i;

+	double val;

+	for (i = 0; i < len / 2; i++) {

+		val = i * 2.0 / len;

+		buf[i] = (int) (val * depth);

+	}

+	for (i = len / 2; i < len ; i++) {

+		val = (len - i) * 2.0 / len;

+		buf[i] = (int) (val * depth);

+	}

 char *

--- a/src/phaser.c

+++ b/src/phaser.c

@@ -62,6 +62,7 @@

 #include <math.h>

 #include <string.h>

 #include "st.h"

+#include "libst.h"

 #define MOD_SINE	0

 #define MOD_TRIANGLE	1

@@ -80,53 +81,6 @@

 	long	maxsamples, fade_out;

 } *phaser_t;

-/* Private data for SKEL file */

-LONG phaser_clip24(l)

-LONG l;

-{

-	if (l >= ((LONG)1 << 24))

-		return ((LONG)1 << 24) - 1;

-	else if (l <= -((LONG)1 << 24))

-		return -((LONG)1 << 24) + 1;

-	else

-		return l;

-}

-/* This was very painful.  We need a sine library. */

-void phaser_sine(buf, len, depth)

-int *buf;

-long len;

-long depth;

-{

-	long i;

-	double val;

-	for (i = 0; i < len; i++) {

-		val = sin((double)i/(double)len * 2.0 * M_PI);

-		buf[i] = (int) ((1.0 + val) * depth / 2.0);

-	}

-}

-void phaser_triangle(buf, len, depth)

-int *buf;

-long len;

-long depth;

-{

-	long i;

-	double val;

-	for (i = 0; i < len / 2; i++) {

-		val = i * 2.0 / len;

-		buf[i] = (int) (val * depth);

-	}

-	for (i = len / 2; i < len ; i++) {

-		val = (len - i) * 2.0 / len;

-		buf[i] = (int) (val * depth);

-	}

-}

/*

  * Process options

*/

@@ -199,10 +153,10 @@

 			sizeof(int) * phaser->length);

 	if ( phaser->modulation == MOD_SINE )

-		phaser_sine(phaser->lookup_tab, phaser->length,

+		st_sine(phaser->lookup_tab, phaser->length,

 			phaser->maxsamples - 1);

 	else

-		phaser_triangle(phaser->lookup_tab, phaser->length,

+		st_triangle(phaser->lookup_tab, phaser->length,

 			phaser->maxsamples - 1);

 	phaser->counter = 0;

 	phaser->phase = 0;

@@ -236,7 +190,7 @@

 	phaser->maxsamples] * phaser->decay * -1.0;

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_in * phaser->out_gain;

-		out = phaser_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Mix decay of delay and input */

 		phaser->phaserbuf[phaser->counter] = d_in;

@@ -271,7 +225,7 @@

 	phaser->maxsamples] * phaser->decay * -1.0;

 		/* Adjust the output volume and size to 24 bit */

 		d_out = d_in * phaser->out_gain;

-		out = phaser_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		/* Mix decay of delay and input */

 		phaser->phaserbuf[phaser->counter] = d_in;

--- a/src/polyphas.c

+++ b/src/polyphas.c

@@ -385,7 +385,7 @@

 		int total, size, uprate;

     int k;

-    initrand();

+    st_initrand();

     rate->lcmrate = st_lcm((long)effp->ininfo.rate, (long)effp->outinfo.rate);

--- a/src/rate.c

+++ b/src/rate.c

@@ -30,7 +30,7 @@

  * Limited to 16 bit samples and sampling frequency <= 65535 Hz. If

  * the input & output frequencies are equal, a delay of one sample is

- * introduced

+ * introduced.  Limited to processing 32-bit count worth of samples.

  * 1 << FRAC_BITS evaluating to zero in several places.  Changed with

  * an (unsigned long) cast to make it safe.  MarkMLl 2/1/99

@@ -40,13 +40,13 @@

 /* Private data */

 typedef struct ratestuff {

-        u_l opos_frac;  /* fractional position of the output stream in input stream unit */

-        u_l opos;

+        ULONG opos_frac;  /* fractional position of the output stream in input stream unit */

+        ULONG opos;

-        u_l opos_inc_frac;  /* fractional position increment in the output stream */

-        u_l opos_inc;

+        ULONG opos_inc_frac;  /* fractional position increment in the output stream */

+        ULONG opos_inc;

-        u_l ipos;      /* position in the input stream (integer) */

+        ULONG ipos;      /* position in the input stream (integer) */

         LONG ilast; /* last sample in the input stream */

 } *rate_t;

@@ -70,13 +70,18 @@

 eff_t effp;

 	rate_t rate = (rate_t) effp->priv;

-        u_l incr;

+        ULONG incr;

+	if (effp->ininfo.rate >= 65535 || effp->outinfo.rate >= 65535)

+	    fail("rate effect can only handle rates <= 65535");

+	if (effp->ininfo.size == DWORD || effp->ininfo.size == FLOAT)

+	    fail("rate effect does not work on data greater then 16 bits");

         rate->opos_frac=0;

         rate->opos=0;

         /* increment */

-        incr=(u_l)((double)effp->ininfo.rate / (double)effp->outinfo.rate *

+        incr=(ULONG)((double)effp->ininfo.rate / (double)effp->outinfo.rate *

                    (double) ((unsigned long) 1 << FRAC_BITS));

         rate->opos_inc_frac = incr & (((unsigned long) 1 << FRAC_BITS)-1);

@@ -101,7 +106,7 @@

 	LONG *istart,*iend;

 	LONG *ostart,*oend;

 	LONG ilast,icur,out;

-        u_l tmp;

+        ULONG tmp;

         double t;

         ilast=rate->ilast;

@@ -114,13 +119,18 @@

         while (obuf < oend) {

-                /* read as many input samples so that ipos > opos */

+		/* Safety catch to make sure we have input samples.  */

+                if (ibuf >= iend) goto the_end;

+                /* read as many input samples so that ipos > opos */

                 while (rate->ipos <= rate->opos) {

-                        if (ibuf >= iend) goto the_end;

                         ilast = *ibuf++;

                         rate->ipos++;

+			/* See if we finished the input buffer yet */

+                        if (ibuf >= iend) goto the_end;

                 icur = *ibuf;

                 /* interpolate */

@@ -206,10 +216,10 @@

 /* Private data for Lerp via LCM file */

 typedef struct ratestuff {

-	u_l	lcmrate;		/* least common multiple of rates */

-	u_l	inskip, outskip;	/* LCM increments for I & O rates */

-	u_l	total;

-	u_l	intot, outtot;		/* total samples in LCM basis */

+	ULONG	lcmrate;		/* least common multiple of rates */

+	ULONG	inskip, outskip;	/* LCM increments for I & O rates */

+	ULONG	total;

+	ULONG	intot, outtot;		/* total samples in LCM basis */

 	LONG	lastsamp;		/* history */

 } *rate_t;

--- a/src/reverb.c

+++ b/src/reverb.c

@@ -98,6 +98,7 @@

 #endif

 #include <math.h>

 #include "st.h"

+#include "libst.h"

 #define REVERB_FADE_THRESH 10

 #define DELAY_BUFSIZ ( 50L * MAXRATE )

@@ -114,23 +115,6 @@

 	LONG	pl, ppl, pppl;

 } *reverb_t;

-/* Private data for SKEL file */

-#ifndef abs

-#define abs(a) ((a) >= 0 ? (a) : -(a))

-#endif

-LONG reverb_clip24(l)

-LONG l;

-{

-	if (l >= ((LONG)1 << 24))

-		return ((LONG)1 << 24) - 1;

-	else if (l <= -((LONG)1 << 24))

-		return -((LONG)1 << 24) + 1;

-	else

-		return l;

-}

/*

  * Process options

*/

@@ -232,7 +216,7 @@

 			d_in +=

 reverb->reverbbuf[(i + reverb->maxsamples - reverb->samples[j]) % reverb->maxsamples] * reverb->decay[j];

 		d_out = d_in * reverb->out_gain;

-		out = reverb_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		*obuf++ = out * 256;

 		reverb->reverbbuf[i] = d_in;

 		i++;		/* XXX need a % maxsamples here ? */

@@ -265,10 +249,10 @@

 			d_in +=

 reverb->reverbbuf[(i + reverb->maxsamples - reverb->samples[j]) % reverb->maxsamples] * reverb->decay[j];

 		d_out = d_in * reverb->out_gain;

-		out = reverb_clip24((LONG) d_out);

+		out = st_clip24((LONG) d_out);

 		obuf[done++] = out * 256;

 		reverb->reverbbuf[i] = d_in;

-		l = reverb_clip24((LONG) d_in);

+		l = st_clip24((LONG) d_in);

 		reverb->pppl = reverb->ppl;

 		reverb->ppl = reverb->pl;

 		reverb->pl = l;