shithub: sox

Download patch

ref: 8791cbd4a9f89b06f87ea3f85f8c2d2432bf8903
parent: 8abc793d95d141a162b56730d2061c49ca0515b8
author: cbagwell <cbagwell>
date: Sat Aug 26 17:24:57 EDT 2000 

Updated lowp and highp filters to use a different filter algorthim.
highp actually works now.  Adding error checking to sphere handler
and can autodetect it now.  more man page updates.


--- a/Makefile.gcc
+++ b/Makefile.gcc
@@ -133,11 +133,6 @@
 # DOS_DEFINES	= -DDOS
 # RM		= del /q
 
-# Uncomment the following line if compiling under NeXT.
-# defines .snd to mean a NeXT sound format file only.
-#
-# NEXT_DEFINES	= -DNeXT
-
 # MISC DEFINES - The catch all for things that make even less sense
 #  then normal under unix.  If you need more than one of the following
 #  MISC DEFINES remember to include them on one line so it isn't just
--- a/README
+++ b/README
@@ -2,32 +2,41 @@
 
 
 SoX (also known as Sound eXchange) translates sound samples between different
-file formats, and optionally performs various sound effects.
+file formats, and optionally applies various sound effects.
 
+SoX is intended as the Swiss Army knife of sound processing tools.  It 
+doesn't do anything very well, but sooner or later it comes in very handy.
+
 This release understands:
 
   o Raw files in various binary formats
   o Raw textual data
-  o Microsoft .WAV files
+  o Amiga 8svx files
+  o Apple/SGI AIFF files
+  o SUN .au style files
     o PCM, u-law, a-law
-    o MS ADPCM
-    o IMA ADPCM
-    o GSM
-  o MAUD files
-  o Sound Blaster .VOC files
-  o IRCAM SoundFile files
-  o SUN .au files
-    o PCM, u-law, a-law
     o G7xx ADPCM files (read only)
-  o mutant DEC .au files
-  o Apple/SGI AIFF files
+    o mutant DEC .au files
+    o NeXT .snd files
+  o AVR files
   o CD-R (music CD format)
+  o CVS and VMS files (continous variable slope)
+  o GSM files.
   o Macintosh HCOM files
-  o Sounder files
-  o NeXT .snd files
+  o Amiga MAUD files
+  o IRCAM SoundFile files
+  o NIST SPHERE files
+  o Turtle beach SampleVision files.
   o Soundtool (DOS) files
-  o Psion (palmtop) A-law files
-  o AVR files
+  o Yamaha TX-16W sampler files.
+  o Sound Blaster .VOC files
+  o Microsoft .WAV files
+    o PCM, u-law, a-law
+    o MS ADPCM, IMA ADPCM
+    o GSM
+  o Psion (palmtop) A-law WVE files
+  o Pseudo-file fomats that allow direct playing/recording
+    from some audio devices under unix.
 
 The sound effects include:
 
@@ -73,11 +82,6 @@
 
 Technically, SoX is made up of a sound file processing library called libst
 and SoX is a program implemented using this library.
-
-SoX is intended as the Swiss Army knife of sound processing tools.  It 
-doesn't do anything very well, but sooner or later it comes in very handy.
-SoX is really only usable day-to-day if you hide the wacky options with 
-one-line shell scripts.
 
 Installing:
 
--- a/configure
+++ b/configure
@@ -1123,10 +1123,6 @@
 		CFLAGS="$CFLAGS -D_HPUX_SOURCE"
 		;;
 
-	*next*|*rhapsody*|*apple*darwin* )
-		CFLAGS="$CFLAGS -DNeXT"
-		;;
-
 	*cygwin* )
 		CFLAGS="$CFLAGS -mno-cygwin"
 		LDFLAGS="$LDFLAGS -mno-cygwin"
@@ -1145,7 +1141,7 @@
 
 
 echo $ac_n "checking whether byte ordering is bigendian""... $ac_c" 1>&6
-echo "configure:1149: checking whether byte ordering is bigendian" >&5
+echo "configure:1145: checking whether byte ordering is bigendian" >&5
 if eval "test \"`echo '$''{'ac_cv_c_bigendian'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
@@ -1152,7 +1148,7 @@
   ac_cv_c_bigendian=unknown
 # See if sys/param.h defines the BYTE_ORDER macro.
 cat > conftest.$ac_ext <<EOF
-#line 1156 "configure"
+#line 1152 "configure"
 #include "confdefs.h"
 #include <sys/types.h>
 #include <sys/param.h>
@@ -1163,11 +1159,11 @@
 #endif
 ; return 0; }
 EOF
-if { (eval echo configure:1167: \"$ac_compile\") 1>&5; (eval $ac_compile) 2>&5; }; then
+if { (eval echo configure:1163: \"$ac_compile\") 1>&5; (eval $ac_compile) 2>&5; }; then
   rm -rf conftest*
   # It does; now see whether it defined to BIG_ENDIAN or not.
 cat > conftest.$ac_ext <<EOF
-#line 1171 "configure"
+#line 1167 "configure"
 #include "confdefs.h"
 #include <sys/types.h>
 #include <sys/param.h>
@@ -1178,7 +1174,7 @@
 #endif
 ; return 0; }
 EOF
-if { (eval echo configure:1182: \"$ac_compile\") 1>&5; (eval $ac_compile) 2>&5; }; then
+if { (eval echo configure:1178: \"$ac_compile\") 1>&5; (eval $ac_compile) 2>&5; }; then
   rm -rf conftest*
   ac_cv_c_bigendian=yes
 else
@@ -1198,7 +1194,7 @@
     { echo "configure: error: can not run test program while cross compiling" 1>&2; exit 1; }
 else
   cat > conftest.$ac_ext <<EOF
-#line 1202 "configure"
+#line 1198 "configure"
 #include "confdefs.h"
 main () {
   /* Are we little or big endian?  From Harbison&Steele.  */
@@ -1211,7 +1207,7 @@
   exit (u.c[sizeof (long) - 1] == 1);
 }
 EOF
-if { (eval echo configure:1215: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext} && (./conftest; exit) 2>/dev/null
+if { (eval echo configure:1211: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext} && (./conftest; exit) 2>/dev/null
 then
   ac_cv_c_bigendian=no
 else
@@ -1238,14 +1234,14 @@
 
 
 echo "checking if math library is required during link" 1>&6
-echo "configure:1242: checking if math library is required during link" >&5
+echo "configure:1238: checking if math library is required during link" >&5
 echo $ac_n "checking for pow""... $ac_c" 1>&6
-echo "configure:1244: checking for pow" >&5
+echo "configure:1240: checking for pow" >&5
 if eval "test \"`echo '$''{'ac_cv_func_pow'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1249 "configure"
+#line 1245 "configure"
 #include "confdefs.h"
 /* System header to define __stub macros and hopefully few prototypes,
     which can conflict with char pow(); below.  */
@@ -1268,7 +1264,7 @@
 
 ; return 0; }
 EOF
-if { (eval echo configure:1272: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext}; then
+if { (eval echo configure:1268: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext}; then
   rm -rf conftest*
   eval "ac_cv_func_pow=yes"
 else
@@ -1290,7 +1286,7 @@
 if test "$ac_cv_func_pow" = no
 then
   echo $ac_n "checking for pow in -lm""... $ac_c" 1>&6
-echo "configure:1294: checking for pow in -lm" >&5
+echo "configure:1290: checking for pow in -lm" >&5
 ac_lib_var=`echo m'_'pow | sed 'y%./+-%__p_%'`
 if eval "test \"`echo '$''{'ac_cv_lib_$ac_lib_var'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
@@ -1298,7 +1294,7 @@
   ac_save_LIBS="$LIBS"
 LIBS="-lm  $LIBS"
 cat > conftest.$ac_ext <<EOF
-#line 1302 "configure"
+#line 1298 "configure"
 #include "confdefs.h"
 /* Override any gcc2 internal prototype to avoid an error.  */
 /* We use char because int might match the return type of a gcc2
@@ -1309,7 +1305,7 @@
 pow()
 ; return 0; }
 EOF
-if { (eval echo configure:1313: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext}; then
+if { (eval echo configure:1309: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext}; then
   rm -rf conftest*
   eval "ac_cv_lib_$ac_lib_var=yes"
 else
@@ -1357,17 +1353,17 @@
 
 ac_safe=`echo "gsm.h" | sed 'y%./+-%__p_%'`
 echo $ac_n "checking for gsm.h""... $ac_c" 1>&6
-echo "configure:1361: checking for gsm.h" >&5
+echo "configure:1357: checking for gsm.h" >&5
 if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1366 "configure"
+#line 1362 "configure"
 #include "confdefs.h"
 #include <gsm.h>
 EOF
 ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
-{ (eval echo configure:1371: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+{ (eval echo configure:1367: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
 ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"`
 if test -z "$ac_err"; then
   rm -rf conftest*
@@ -1392,7 +1388,7 @@
 if test "$gsminc" = yes
 then
   echo $ac_n "checking for gsm_create in -lgsm""... $ac_c" 1>&6
-echo "configure:1396: checking for gsm_create in -lgsm" >&5
+echo "configure:1392: checking for gsm_create in -lgsm" >&5
 ac_lib_var=`echo gsm'_'gsm_create | sed 'y%./+-%__p_%'`
 if eval "test \"`echo '$''{'ac_cv_lib_$ac_lib_var'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
@@ -1400,7 +1396,7 @@
   ac_save_LIBS="$LIBS"
 LIBS="-lgsm  $LIBS"
 cat > conftest.$ac_ext <<EOF
-#line 1404 "configure"
+#line 1400 "configure"
 #include "confdefs.h"
 /* Override any gcc2 internal prototype to avoid an error.  */
 /* We use char because int might match the return type of a gcc2
@@ -1411,7 +1407,7 @@
 gsm_create()
 ; return 0; }
 EOF
-if { (eval echo configure:1415: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext}; then
+if { (eval echo configure:1411: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext}; then
   rm -rf conftest*
   eval "ac_cv_lib_$ac_lib_var=yes"
 else
@@ -1446,17 +1442,17 @@
 do
 ac_safe=`echo "$ac_hdr" | sed 'y%./+-%__p_%'`
 echo $ac_n "checking for $ac_hdr""... $ac_c" 1>&6
-echo "configure:1450: checking for $ac_hdr" >&5
+echo "configure:1446: checking for $ac_hdr" >&5
 if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1455 "configure"
+#line 1451 "configure"
 #include "confdefs.h"
 #include <$ac_hdr>
 EOF
 ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
-{ (eval echo configure:1460: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+{ (eval echo configure:1456: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
 ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"`
 if test -z "$ac_err"; then
   rm -rf conftest*
@@ -1487,12 +1483,12 @@
 for ac_func in getopt strerror memmove rand
 do
 echo $ac_n "checking for $ac_func""... $ac_c" 1>&6
-echo "configure:1491: checking for $ac_func" >&5
+echo "configure:1487: checking for $ac_func" >&5
 if eval "test \"`echo '$''{'ac_cv_func_$ac_func'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1496 "configure"
+#line 1492 "configure"
 #include "confdefs.h"
 /* System header to define __stub macros and hopefully few prototypes,
     which can conflict with char $ac_func(); below.  */
@@ -1515,7 +1511,7 @@
 
 ; return 0; }
 EOF
-if { (eval echo configure:1519: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext}; then
+if { (eval echo configure:1515: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext}; then
   rm -rf conftest*
   eval "ac_cv_func_$ac_func=yes"
 else
@@ -1544,7 +1540,7 @@
 if test "$alsa_dsp" = auto
 then
 	echo $ac_n "checking whether /proc/asound is functional (ALSA)""... $ac_c" 1>&6
-echo "configure:1548: checking whether /proc/asound is functional (ALSA)" >&5
+echo "configure:1544: checking whether /proc/asound is functional (ALSA)" >&5
 if eval "test \"`echo '$''{'ac_cv_dev_alsa_dsp'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
@@ -1552,7 +1548,7 @@
   ac_cv_dev_alsa_dsp=no
 else
   cat > conftest.$ac_ext <<EOF
-#line 1556 "configure"
+#line 1552 "configure"
 #include "confdefs.h"
 
 			void *opendir(const char *);
@@ -1570,7 +1566,7 @@
 				}
 			
 EOF
-if { (eval echo configure:1574: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext} && (./conftest; exit) 2>/dev/null
+if { (eval echo configure:1570: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext} && (./conftest; exit) 2>/dev/null
 then
   ac_cv_dev_alsa_dsp=yes
 else
@@ -1592,17 +1588,17 @@
 do
 ac_safe=`echo "$ac_hdr" | sed 'y%./+-%__p_%'`
 echo $ac_n "checking for $ac_hdr""... $ac_c" 1>&6
-echo "configure:1596: checking for $ac_hdr" >&5
+echo "configure:1592: checking for $ac_hdr" >&5
 if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1601 "configure"
+#line 1597 "configure"
 #include "confdefs.h"
 #include <$ac_hdr>
 EOF
 ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
-{ (eval echo configure:1606: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+{ (eval echo configure:1602: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
 ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"`
 if test -z "$ac_err"; then
   rm -rf conftest*
@@ -1642,17 +1638,17 @@
 do
 ac_safe=`echo "$ac_hdr" | sed 'y%./+-%__p_%'`
 echo $ac_n "checking for $ac_hdr""... $ac_c" 1>&6
-echo "configure:1646: checking for $ac_hdr" >&5
+echo "configure:1642: checking for $ac_hdr" >&5
 if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1651 "configure"
+#line 1647 "configure"
 #include "confdefs.h"
 #include <$ac_hdr>
 EOF
 ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
-{ (eval echo configure:1656: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+{ (eval echo configure:1652: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
 ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"`
 if test -z "$ac_err"; then
   rm -rf conftest*
@@ -1687,7 +1683,7 @@
 if test "$oss_dsp" = auto
 then
 	echo $ac_n "checking whether /dev/dsp is functional (OSS)""... $ac_c" 1>&6
-echo "configure:1691: checking whether /dev/dsp is functional (OSS)" >&5
+echo "configure:1687: checking whether /dev/dsp is functional (OSS)" >&5
 if eval "test \"`echo '$''{'ac_cv_dev_oss_dsp'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
@@ -1695,7 +1691,7 @@
   ac_cv_dev_oss_dsp=no
 else
   cat > conftest.$ac_ext <<EOF
-#line 1699 "configure"
+#line 1695 "configure"
 #include "confdefs.h"
 
 			int open(const char *, int);
@@ -1713,7 +1709,7 @@
 				}
 			
 EOF
-if { (eval echo configure:1717: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext} && (./conftest; exit) 2>/dev/null
+if { (eval echo configure:1713: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext} && (./conftest; exit) 2>/dev/null
 then
   ac_cv_dev_oss_dsp=yes
 else
@@ -1735,17 +1731,17 @@
 do
 ac_safe=`echo "$ac_hdr" | sed 'y%./+-%__p_%'`
 echo $ac_n "checking for $ac_hdr""... $ac_c" 1>&6
-echo "configure:1739: checking for $ac_hdr" >&5
+echo "configure:1735: checking for $ac_hdr" >&5
 if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1744 "configure"
+#line 1740 "configure"
 #include "confdefs.h"
 #include <$ac_hdr>
 EOF
 ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
-{ (eval echo configure:1749: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+{ (eval echo configure:1745: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
 ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"`
 if test -z "$ac_err"; then
   rm -rf conftest*
@@ -1785,17 +1781,17 @@
 do
 ac_safe=`echo "$ac_hdr" | sed 'y%./+-%__p_%'`
 echo $ac_n "checking for $ac_hdr""... $ac_c" 1>&6
-echo "configure:1789: checking for $ac_hdr" >&5
+echo "configure:1785: checking for $ac_hdr" >&5
 if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1794 "configure"
+#line 1790 "configure"
 #include "confdefs.h"
 #include <$ac_hdr>
 EOF
 ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
-{ (eval echo configure:1799: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+{ (eval echo configure:1795: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
 ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"`
 if test -z "$ac_err"; then
   rm -rf conftest*
@@ -1830,7 +1826,7 @@
 if test "$sun_audio" = auto
 then
 	echo $ac_n "checking whether /dev/audio is functional (SUN style)""... $ac_c" 1>&6
-echo "configure:1834: checking whether /dev/audio is functional (SUN style)" >&5
+echo "configure:1830: checking whether /dev/audio is functional (SUN style)" >&5
 if eval "test \"`echo '$''{'ac_cv_dev_sun_audio'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
@@ -1838,7 +1834,7 @@
   ac_cv_dev_sun_audio=no
 else
   cat > conftest.$ac_ext <<EOF
-#line 1842 "configure"
+#line 1838 "configure"
 #include "confdefs.h"
 
 			int open(const char *, int);
@@ -1856,7 +1852,7 @@
 				}
 			
 EOF
-if { (eval echo configure:1860: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext} && (./conftest; exit) 2>/dev/null
+if { (eval echo configure:1856: \"$ac_link\") 1>&5; (eval $ac_link) 2>&5; } && test -s conftest${ac_exeext} && (./conftest; exit) 2>/dev/null
 then
   ac_cv_dev_sun_audio=yes
 else
@@ -1878,17 +1874,17 @@
 do
 ac_safe=`echo "$ac_hdr" | sed 'y%./+-%__p_%'`
 echo $ac_n "checking for $ac_hdr""... $ac_c" 1>&6
-echo "configure:1882: checking for $ac_hdr" >&5
+echo "configure:1878: checking for $ac_hdr" >&5
 if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1887 "configure"
+#line 1883 "configure"
 #include "confdefs.h"
 #include <$ac_hdr>
 EOF
 ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
-{ (eval echo configure:1892: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+{ (eval echo configure:1888: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
 ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"`
 if test -z "$ac_err"; then
   rm -rf conftest*
@@ -1928,17 +1924,17 @@
 do
 ac_safe=`echo "$ac_hdr" | sed 'y%./+-%__p_%'`
 echo $ac_n "checking for $ac_hdr""... $ac_c" 1>&6
-echo "configure:1932: checking for $ac_hdr" >&5
+echo "configure:1928: checking for $ac_hdr" >&5
 if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
   echo $ac_n "(cached) $ac_c" 1>&6
 else
   cat > conftest.$ac_ext <<EOF
-#line 1937 "configure"
+#line 1933 "configure"
 #include "confdefs.h"
 #include <$ac_hdr>
 EOF
 ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
-{ (eval echo configure:1942: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+{ (eval echo configure:1938: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
 ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"`
 if test -z "$ac_err"; then
   rm -rf conftest*
--- a/configure.in
+++ b/configure.in
@@ -63,10 +63,6 @@
 		CFLAGS="$CFLAGS -D_HPUX_SOURCE"
 		;;
 
-	*next*|*rhapsody*|*apple*darwin* )
-		CFLAGS="$CFLAGS -DNeXT"
-		;;
-
 	*cygwin* )
 		CFLAGS="$CFLAGS -mno-cygwin"
 		LDFLAGS="$LDFLAGS -mno-cygwin"
--- a/pick.c
+++ b/pick.c
@@ -81,12 +81,12 @@
 
 	if (effp->outinfo.channels != 1)  /* must be one output channel */
 	{
-	   st_fail("Can't pick with other than 1 output channel."); 
+	   st_fail("Pick effect requires output to be forced to 1 channel.  Currenty it is set for %d channels.",effp->outinfo.channels); 
 	   return (ST_EOF);
 	}
 	if (effp->ininfo.channels != 2 && effp->ininfo.channels != 4)
 	{
-	        st_fail("Can't pick with other than 2 or 4 input channels.");
+	        st_fail("Pick effect can not work with mono input data.");
 		return (ST_EOF);
 	}
         if (effp->ininfo.channels == 2) {  /* check for valid option */
--- a/sox.1
+++ b/sox.1
@@ -70,11 +70,11 @@
 .br
     \fBflanger\fR \fIgain-in gain-out delay decay speed\fR < -s | -t >
 .br
-    \fBhighp\fR \fIcenter\fR
+    \fBhighp\fR \fIfrequency\fR
 .br
     \fBhighpass\fR \fIfrequency\fR
 .br
-    \fBlowp\fR \fIcenter\fR
+    \fBlowp\fR \fIfrequency\fR
 .br
     \fBlowpass\fR \fIfrequency\fR
 .br
@@ -86,7 +86,7 @@
 .br
     \fBphaser\fR \fIgain-in gain-out delay decay speed\fR < -s | -t >
 .br
-    \fBpick\fR
+    \fBpick\fR [ \fI-1\fR | \fI-2\fR | \fI-3\fR | \fI-4\fR | \fI-l\fR | \fI-r\fR ]
 .br
     \fBpitch\fR \fIshift\fR [ \fIwidth interpole fade\fR ]
 .br
@@ -407,6 +407,17 @@
 Sound Files are used by academic music software 
 such as the CSound package, and the MixView sound sample editor.
 .TP 10
+.B .sph
+.br
+SPHERE (SPeech HEader Resources) is a file format defined by NIST
+(National Institute of Standards and Technology) and is used with
+speech audio.  SoX can read these files when they contain
+ulaw and PCM data.  It will ignore any header information that
+says the data is compressed using \fIshorten\fR compression and
+will treat the data as either ulaw or PCM.  This will allow SoX
+and the command line \fIshorten\fR program to be ran together using
+pipes to uncompress the data and then pass the result to SoX for processing.
+.TP 10
 .B .smp
 Turtle Beach SampleVision files.
 .br
@@ -415,6 +426,16 @@
 sample rates are supported by the package, although not all are supported by
 the samplers themselves. Currently loop points are ignored.
 .TP 10
+.B .snd
+.br
+Under DOS this file format is the same as the \fB.sndt\fR format.  Under all
+other platforms it is the same as the \fB.au\fR format.
+.TP 10
+.B .sndt
+SoundTool files.
+.br
+This is an older DOS file format.
+.TP 10
 .B sunau
 Sun /dev/audio device driver
 .br
@@ -653,11 +674,11 @@
 The modulation is either sinodial (-s) or triangular
 (-t).  Gain-out is the volume of the output.
 .TP 10
-highp \fIcenter
-Apply a high-pass filter.
+highp \fIfrequency\fR
+Apply a single pole recursive high-pass filter.
 The frequency response drops logarithmically with 
-.I center
-frequency in the middle of the drop.
+I frequency 
+in the middle of the drop.
 The slope of the filter is quite gentle.
 See \fBfilter\fR for a highpass effect with sharper cutoff.
 .TP 10
@@ -664,11 +685,11 @@
 highpass \fIfrequency\fB
 Butterworth highpass filter.  Description comming soon!
 .TP 10
-lowp \fIcenter
-Apply a low-pass filter.
+lowp \fIfrequency\fR
+Apply a single pole recursive low-pass filter.
 The frequency response drops logarithmically with 
-.I center
-frequency in the middle of the drop.
+.I frequency 
+in the middle of the drop.
 The slope of the filter is quite gentle.
 See \fBfilter\fR for a lowpass effect with sharper cutoff.
 .TP 10
@@ -710,9 +731,12 @@
 (-t).  The decay should be less than 0.5 to avoid
 feedback.  Gain-out is the volume of the output.
 .TP 10
-pick
+pick [ \fI-1\fR | \fI-2\fR | \fI-3\fR | \fI-4\fR | \fI-l\fR | \fI-r\fR ]
 Select the left or right channel of a stereo sample,
-or one of four channels in a quadrophonic sample.
+or one of four channels in a quadrophonic sample. The \fI-l\fR and \fI-r\fR
+options represent either the left or right channel.  It is required that
+you use the \fB-c 1\fR command line option in order to force the output file to
+contain only 1 channel.
 .TP 10
 pitch \fIshift [ width interpole fade ]\fB
 Change the pitch of file without affecting its duration by cross-fading
--- a/sox.txt
+++ b/sox.txt
@@ -42,15 +42,15 @@
 	   echos  gain-in gain-out delay decay [ delay decay ...]
 	   filter [ low ]-[ high ] [ window-len [ beta ]]
 	   flanger gain-in gain-out delay decay speed < -s | -t >
-	   highp center
+	   highp frequency
 	   highpass frequency
-	   lowp center
+	   lowp frequency
 	   lowpass frequency
 	   map
 	   mask
 	   pan direction
 	   phaser gain-in gain-out delay decay speed < -s | -t >
-	   pick
+	   pick [ -1 | -2 | -3 | -4 | -l | -r ]
 	   pitch shift [ width interpole fade ]
 	   polyphase [ -w < nut / ham > ]
 		     [	-width < long / short / # > ]
@@ -425,35 +425,35 @@
 		 such as the  CSound  package,	and  the  MixView
 		 sound sample editor.
 
+       .sph
+		 SPHERE	 (SPeech HEader Resources) is a file for-
+		 mat defined by NIST (National Institute of Stan-
+		 dards	and  Technology)  and is used with speech
+		 audio.	 SoX can read these files when they  con-
+		 tain  ulaw  and  PCM  data.   It will ignore any
+		 header information that says the  data	 is  com-
+		 pressed using shorten compression and will treat
+		 the data as either ulaw or PCM.  This will allow
+		 SoX  and  the command line shorten program to be
+		 ran together using pipes to uncompress the  data
+		 and  then pass the result to SoX for processing.
+
        .smp	 Turtle Beach SampleVision files.
-		 SMP  files  are  for use with the PC-DOS package
-		 SampleVision by  Turtle  Beach	 Softworks.  This
-		 package  is  for  communication  to several MIDI
-		 samplers. All sample rates are supported by  the
-		 package,  although  not all are supported by the
-		 samplers themselves. Currently loop  points  are
+		 SMP files are for use with  the  PC-DOS  package
+		 SampleVision  by  Turtle  Beach  Softworks. This
+		 package is for	 communication	to  several  MIDI
+		 samplers.  All sample rates are supported by the
+		 package, although not all are supported  by  the
+		 samplers  themselves.	Currently loop points are
 		 ignored.
 
-       sunau	 Sun /dev/audio device driver
-		 This is a pseudo-file type and can be optionally
-		 compiled into Sox.  Run sox -h	 to  see  if  you
-		 have  support	for  this  file	 type.	When this
-		 driver is used it allows you to open  up  a  Sun
-		 /dev/audio file and configure it to use the same
-		 data type as passed in to  Sox.   It  works  for
-		 both  playing and recording sound samples.  When
-		 playing sound files it attempts to  set  up  the
-		 audio driver to use the same format as the input
-		 file.	It is suggested to  always  override  the
-		 output values to use the highest quality samples
-		 your hardware can handle.  Example: -t sunau  -w
-		 -s /dev/audio or -t sunau -U -c 1 /dev/audio for
-		 older sun equipment.
+       .snd
+		 Under DOS this file format is the  same  as  the
+		 .sndt	format.	  Under all other platforms it is
+		 the same as the .au format.
 
-       .txw	 Yamaha TX-16W sampler.
-		 A file format from a  Yamaha  sampling	 keyboard
-		 which	wrote  IBM-PC format 3.5" floppies.  Han-
-		 dles reading of files	which  do  not	have  the
+       .sndt	 SoundTool files.
+		 This is an older DOS file format.
 
 
 
@@ -466,129 +466,129 @@
 SoX(1)							   SoX(1)
 
 
-		 sample	 rate field set to one of the expected by
-		 looking at some other bytes in	 the  attack/loop
-		 length	 fields,  and  defaulting to 33kHz if the
+       sunau	 Sun /dev/audio device driver
+		 This is a pseudo-file type and can be optionally
+		 compiled  into	 Sox.	Run  sox -h to see if you
+		 have support for  this	 file  type.   When  this
+		 driver	 is  used  it allows you to open up a Sun
+		 /dev/audio file and configure it to use the same
+		 data  type  as	 passed	 in to Sox.  It works for
+		 both playing and recording sound samples.   When
+		 playing  sound	 files	it attempts to set up the
+		 audio driver to use the same format as the input
+		 file.	 It  is	 suggested to always override the
+		 output values to use the highest quality samples
+		 your  hardware can handle.  Example: -t sunau -w
+		 -s /dev/audio or -t sunau -U -c 1 /dev/audio for
+		 older sun equipment.
+
+       .txw	 Yamaha TX-16W sampler.
+		 A  file  format  from a Yamaha sampling keyboard
+		 which wrote IBM-PC format 3.5"	 floppies.   Han-
+		 dles reading of files which do not have the sam-
+		 ple rate field set to one  of	the  expected  by
+		 looking  at  some other bytes in the attack/loop
+		 length fields, and defaulting to  33kHz  if  the
 		 sample rate is still unknown.
 
        .vms	 More info to come.
-		 Used to compress speech audio	for  applications
+		 Used  to  compress speech audio for applications
 		 such as voice mail.
 
        .voc	 Sound Blaster VOC files.
-		 VOC  files  are  multi-part  and contain silence
-		 parts, looping, and different sample  rates  for
-		 different  chunks.   On input, the silence parts
-		 are filled out, loops are rejected,  and  sample
-		 data	with  a	 new  sample  rate  is	rejected.
-		 Silence with a different sample rate  is  gener-
-		 ated  appropriately.	On output, silence is not
+		 VOC files are	multi-part  and	 contain  silence
+		 parts,	 looping,  and different sample rates for
+		 different chunks.  On input, the  silence  parts
+		 are  filled  out, loops are rejected, and sample
+		 data  with  a	new  sample  rate  is	rejected.
+		 Silence  with	a different sample rate is gener-
+		 ated appropriately.  On output, silence  is  not
 		 detected, nor are impossible sample rates.
 
        .wav	 Microsoft .WAV RIFF files.
-		 These appear to be very similar  to  IFF  files,
-		 but  not  the	same.	They are the native sound
+		 These	appear	to  be very similar to IFF files,
+		 but not the same.  They  are  the  native  sound
 		 file format of Windows.  (Obviously, Windows was
-		 of  such  incredible  importance to the computer
-		 industry that it just had to have its own  sound
+		 of such incredible importance	to  the	 computer
+		 industry  that it just had to have its own sound
 		 file format.)	Normally .wav files have all for-
-		 matting information in their headers, and so  do
-		 not  need  any	 format	 options specified for an
-		 input file. If any are, they will  override  the
-		 file  header,	and  you  will	be warned to this
+		 matting  information in their headers, and so do
+		 not need any format  options  specified  for  an
+		 input	file.  If any are, they will override the
+		 file header, and you  will  be	 warned	 to  this
 		 effect.  You had better know what you are doing!
-		 Output	 format	 options will cause a format con-
-		 version, and the  .wav	 will  written	appropri-
-		 ately.	  Sox currently can read PCM, ULAW, ALAW,
-		 MS ADPCM, and IMA (or DVI) ADPCM.  It can  write
-		 all of these formats including (NEW!)	the ADPCM
-		 encoding.
+		 Output format options will cause a  format  con-
+		 version,  and	the  .wav  will written appropri-
+		 ately.	 Sox currently can read PCM, ULAW,  ALAW,
 
-       .wve	 Psion 8-bit alaw
-		 These are 8-bit a-law 8khz sound files	 used  on
-		 the Psion palmtop portable computer.
 
-       .raw	 Raw files (no header).
-		 The  sample  rate,  size  (byte, word, etc), and
-		 encoding (signed, unsigned, etc.)  of the sample
-		 file  must  be	 given.	  The  number of channels
-		 defaults to 1.
 
-       .ub, .sb, .uw, .sw, .ul, .sl
-		 These are several  suffices  which  serve  as	a
-		 shorthand  for	 raw  files with a given size and
-		 encoding.  Thus, ub, sb, uw, sw, ul and sl  cor-
-		 respond   to  "unsigned  byte",  "signed  byte",
-		 "unsigned word", "signed word",  "ulaw"  (byte),
+			  July 24, 2000				8
 
 
 
-			  July 24, 2000				8
 
 
+SoX(1)							   SoX(1)
 
 
+		 MS  ADPCM, and IMA (or DVI) ADPCM.  It can write
+		 all of these formats including (NEW!)	the ADPCM
+		 encoding.
 
-SoX(1)							   SoX(1)
+       .wve	 Psion 8-bit alaw
+		 These	are  8-bit a-law 8khz sound files used on
+		 the Psion palmtop portable computer.
 
+       .raw	 Raw files (no header).
+		 The sample rate, size	(byte,	word,  etc),  and
+		 encoding (signed, unsigned, etc.)  of the sample
+		 file must be  given.	The  number  of	 channels
+		 defaults to 1.
 
-		 and  "signed long".  The sample rate defaults to
+       .ub, .sb, .uw, .sw, .ul, .sl
+		 These	are  several  suffices	which  serve as a
+		 shorthand for raw files with a	 given	size  and
+		 encoding.   Thus, ub, sb, uw, sw, ul and sl cor-
+		 respond  to  "unsigned	 byte",	 "signed   byte",
+		 "unsigned  word",  "signed word", "ulaw" (byte),
+		 and "signed long".  The sample rate defaults  to
 		 8000 hz if not explicitly set, and the number of
-		 channels  (as	always) defaults to 1.	There are
-		 lots of Sparc samples floating around	in  u-law
+		 channels (as always) defaults to 1.   There  are
+		 lots  of  Sparc samples floating around in u-law
 		 format with no header and fixed at a sample rate
-		 of 8000 hz.  (Certain sound management	 software
+		 of  8000 hz.  (Certain sound management software
 		 cheerfully  ignores  the  headers.)   Similarly,
 		 most Mac sound files are in unsigned byte format
 		 with a sample rate of 11025 or 22050 hz.
 
-       .auto	 This  is  a  ``meta-type'': specifying this type
-		 for an input file triggers some code that  tries
-		 to  guess  the	 real  type  by looking for magic
-		 words in the  header.	 If  the  type	can't  be
-		 guessed,  the	program	 exits with an error mes-
-		 sage.	The input must be a  plain  file,  not	a
+       .auto	 This is a ``meta-type'':  specifying  this  type
+		 for  an input file triggers some code that tries
+		 to guess the real  type  by  looking  for  magic
+		 words	in  the	 header.   If  the  type can't be
+		 guessed, the program exits with  an  error  mes-
+		 sage.	 The  input  must  be a plain file, not a
 		 pipe.	This type can't be used for output files.
 
 EFFECTS
        Multiple effects may be applied to the audio data by spec-
-       ifying  them  one  after another at the end of the command
+       ifying them one after another at the end	 of  the  command
        line.
 
        avg [ -l | -r ]
-		 Reduce the number of channels by  averaging  the
-		 samples,  or  duplicate channels to increase the
-		 number of channels.  This  effect  is	automati-
-		 cally	used  when  the	 number of input channels
+		 Reduce	 the  number of channels by averaging the
+		 samples, or duplicate channels to  increase  the
+		 number	 of  channels.	 This effect is automati-
+		 cally used when the  number  of  input	 channels
 		 differ from the number of output channels.  When
-		 reducing  the	number of channels it is possible
-		 to manually specify the avg effect and	 use  the
-		 -l  and  -r  options  to select only the left or
-		 right channel for the output instead of  averag-
+		 reducing the number of channels it  is	 possible
+		 to  manually  specify the avg effect and use the
+		 -l and -r options to select  only  the	 left  or
+		 right	channel for the output instead of averag-
 		 ing the two channels.
 
-       band [ -n ] center [ width ]
-		 Apply	 a   band-pass	 filter.   The	frequency
-		 response drops logarithmically around the center
-		 frequency.   The  width  gives	 the slope of the
-		 drop.	The frequencies at  center  +  width  and
-		 center	 -  width  will be half of their original
-		 amplitudes.  Band defaults to a mode oriented to
-		 pitched signals, i.e. voice, singing, or instru-
-		 mental music.	The -n (for  noise)  option  uses
-		 the   alternate  mode	for  un-pitched	 signals.
-		 Warning: -n introduces	 a  power-gain	of  about
-		 11dB  in  the	filter, so beware of output clip-
-		 ping.	Band introduces noise in the shape of the
-		 filter, i.e. peaking at the center frequency and
-		 settling around it.  See filter for  a	 bandpass
-		 effect with steeper shoulders.
 
-       bandpass frequency bandwidth
-		 Butterworth  bandpass filter. Description coming
 
-
-
 			  July 24, 2000				9
 
 
@@ -598,6 +598,25 @@
 SoX(1)							   SoX(1)
 
 
+       band [ -n ] center [ width ]
+		 Apply	a  band-pass   filter.	  The	frequency
+		 response drops logarithmically around the center
+		 frequency.  The width gives  the  slope  of  the
+		 drop.	 The  frequencies  at  center + width and
+		 center - width will be half  of  their	 original
+		 amplitudes.  Band defaults to a mode oriented to
+		 pitched signals, i.e. voice, singing, or instru-
+		 mental	 music.	  The  -n (for noise) option uses
+		 the  alternate	 mode  for  un-pitched	 signals.
+		 Warning:  -n  introduces  a  power-gain of about
+		 11dB in the filter, so beware	of  output  clip-
+		 ping.	Band introduces noise in the shape of the
+		 filter, i.e. peaking at the center frequency and
+		 settling  around  it.	See filter for a bandpass
+		 effect with steeper shoulders.
+
+       bandpass frequency bandwidth
+		 Butterworth bandpass filter. Description  coming
 		 soon!
 
        bandreject frequency bandwidth
@@ -607,10 +626,10 @@
        chorus gain-in gain-out delay decay speed depth
 
 	      -s | -t [ delay decay speed depth -s | -t ... ]
-		 Add  a chorus to a sound sample.  Each quadtuple
-		 delay/decay/speed/depth gives the delay in  mil-
-		 liseconds  and	 the  decay (relative to gain-in)
-		 with a modulation speed in  Hz	 using	depth  in
+		 Add a chorus to a sound sample.  Each	quadtuple
+		 delay/decay/speed/depth  gives the delay in mil-
+		 liseconds and the decay  (relative  to	 gain-in)
+		 with  a  modulation  speed  in Hz using depth in
 		 milliseconds.	The modulation is either sinodial
 		 (-s) or triangular (-t).  Gain-out is the volume
 		 of the output.
@@ -620,41 +639,22 @@
 	       in-dB1,out-dB1[,in-dB2,out-dB2...]
 
 	       [gain] [initial-volume]
-		 Compand  (compress  or expand) the dynamic range
-		 of a sample.  The attack and decay time  specify
-		 the  integration  time	 over  which the absolute
-		 value of  the	input  signal  is  integrated  to
-		 determine  its volume.	 Where more than one pair
-		 of attack/decay parameters are	 specified,  each
-		 channel  is treated separately and the number of
-		 pairs must agree with the number of input  chan-
-		 nels.	 The second parameter is a list of points
-		 on the compander's transfer  function	specified
-		 in  dB	 relative  to the maximum possible signal
+		 Compand (compress or expand) the  dynamic  range
+		 of  a sample.	The attack and decay time specify
+		 the integration time  over  which  the	 absolute
+		 value	of  the	 input	signal	is  integrated to
+		 determine its volume.	Where more than one  pair
+		 of  attack/decay  parameters are specified, each
+		 channel is treated separately and the number  of
+		 pairs	must agree with the number of input chan-
+		 nels.	The second parameter is a list of  points
+		 on  the  compander's transfer function specified
+		 in dB relative to the	maximum	 possible  signal
 		 amplitude.   The  input  values  must	be  in	a
-		 strictly increasing order but the transfer func-
-		 tion does not have to be  monotonically  rising.
-		 The  special  value -inf may be used to indicate
-		 that the input volume should be associated  out-
-		 put  volume.	The  points -inf,-inf and 0,0 are
-		 assumed; the latter may be overridden,	 but  the
-		 former	 may not.  The third (optional) parameter
-		 is a postprocessing gain in dB which is  applied
-		 after	the  compression  has  taken  place;  the
-		 fourth (optional) parameter is an initial volume
-		 to  be	 assumed for each channel when the effect
-		 starts.  This permits the user to supply a nomi-
-		 nal  level  initially,	 so  that, for example, a
-		 very large gain is not applied to initial signal
-		 levels before the companding action has begun to
-		 operate: it is quite probable that  in	 such  an
-		 event,	 the  output  would  be	 severely clipped
-		 while	the  compander	gain   properly	  adjusts
-		 itself.
+		 strictly  increasing  order  but  the	 transfer
 
 
 
-
 			  July 24, 2000			       10
 
 
@@ -664,22 +664,41 @@
 SoX(1)							   SoX(1)
 
 
+		 function  does not have to be monotonically ris-
+		 ing.  The special value  -inf	may  be	 used  to
+		 indicate that the input volume should be associ-
+		 ated output volume.  The  points  -inf,-inf  and
+		 0,0  are  assumed; the latter may be overridden,
+		 but the former may not.   The	third  (optional)
+		 parameter  is	a postprocessing gain in dB which
+		 is  applied  after  the  compression  has  taken
+		 place;	 the  fourth  (optional)  parameter is an
+		 initial volume to be assumed  for  each  channel
+		 when  the  effect starts.  This permits the user
+		 to supply a nominal level  initially,	so  that,
+		 for example, a very large gain is not applied to
+		 initial  signal  levels  before  the  companding
+		 action	 has begun to operate: it is quite proba-
+		 ble that in such an event, the output	would  be
+		 severely  clipped while the compander gain prop-
+		 erly adjusts itself.
+
        copy	 Copy the input file to the output file.  This is
-		 the default effect if both files have	the  same
+		 the  default  effect if both files have the same
 		 sampling rate.
 
        cut loopnumber
 		 Extract loop #N from a sample.
 
-       deemph	 Apply	a  treble  attenuation shelving filter to
+       deemph	 Apply a treble attenuation  shelving  filter  to
 		 samples  in  audio  cd	 format.   The	frequency
-		 response  of pre-emphasized recordings is recti-
-		 fied.	The filtering is defined in the	 standard
+		 response of pre-emphasized recordings is  recti-
+		 fied.	 The filtering is defined in the standard
 		 document ISO 908.
 
        echo gain-in gain-out delay decay [ delay decay ... ]
 		 Add echoing to a sound sample.	 Each delay/decay
-		 part gives the delay  in  milliseconds	 and  the
+		 part  gives  the  delay  in milliseconds and the
 		 decay (relative to gain-in) of that echo.  Gain-
 		 out is the volume of the output.
 
@@ -686,107 +705,88 @@
        echos gain-in gain-out delay decay [ delay decay ... ]
 		 Add a sequence of echos to a sound sample.  Each
 		 delay/decay part gives the delay in milliseconds
-		 and the decay	(relative  to  gain-in)	 of  that
+		 and  the  decay  (relative  to	 gain-in) of that
 		 echo.	Gain-out is the volume of the output.
 
        filter [ low ]-[ high ] [ window-len [ beta ] ]
 		 Apply	a  Sinc-windowed  lowpass,  highpass,  or
-		 bandpass filter of given window  length  to  the
-		 signal.   low	refers	to  the	 frequency of the
-		 lower 6dB corner of the filter.  high refers  to
-		 the  frequency	 of  the  upper 6dB corner of the
+		 bandpass  filter  of  given window length to the
+		 signal.  low refers  to  the  frequency  of  the
+		 lower	6dB corner of the filter.  high refers to
+		 the frequency of the upper  6dB  corner  of  the
 		 filter.
 
-		 A lowpass filter  is  obtained	 by  leaving  low
-		 unspecified,	or   0.	  A  highpass  filter  is
-		 obtained by leaving high unspecified, or  0,  or
-		 greater  than or equal to the Nyquist frequency.
+		 A  lowpass  filter  is	 obtained  by leaving low
+		 unspecified,  or  0.	A  highpass   filter   is
 
-		 The window-len, if unspecified, defaults to 128.
-		 Longer	 windows  give	a sharper cutoff, smaller
-		 windows a more gradual cutoff.
 
-		 The beta, if unspecified, defaults to 16.   This
-		 selects  a Kaiser window.  You can select a Nut-
-		 tall window by specifying anything <= 2.0  here.
-		 For  more  discussion	of  beta,  look under the
-		 resample effect.
 
+			  July 24, 2000			       11
 
-       flanger gain-in gain-out delay decay speed < -s | -t >
-		 Add a flanger to a sound  sample.   Each  triple
-		 delay/decay/speed  gives  the delay in millisec-
-		 onds and the decay (relative to gain-in) with	a
-		 modulation  speed  in	Hz.   The  modulation  is
 
 
 
-			  July 24, 2000			       11
 
+SoX(1)							   SoX(1)
 
 
+		 obtained  by  leaving high unspecified, or 0, or
+		 greater than or equal to the Nyquist  frequency.
 
+		 The window-len, if unspecified, defaults to 128.
+		 Longer windows give a	sharper	 cutoff,  smaller
+		 windows a more gradual cutoff.
 
-SoX(1)							   SoX(1)
+		 The  beta, if unspecified, defaults to 16.  This
+		 selects a Kaiser window.  You can select a  Nut-
+		 tall  window by specifying anything <= 2.0 here.
+		 For more discussion  of  beta,	 look  under  the
+		 resample effect.
 
 
-		 either sinodial (-s) or triangular (-t).   Gain-
+       flanger gain-in gain-out delay decay speed < -s | -t >
+		 Add  a	 flanger  to a sound sample.  Each triple
+		 delay/decay/speed gives the delay  in	millisec-
+		 onds  and the decay (relative to gain-in) with a
+		 modulation  speed  in	Hz.   The  modulation  is
+		 either	 sinodial (-s) or triangular (-t).  Gain-
 		 out is the volume of the output.
 
-       highp center
-		 Apply	 a   high-pass	 filter.   The	frequency
-		 response drops logarithmically with center  fre-
-		 quency	 in the middle of the drop.  The slope of
-		 the filter is quite gentle.  See  filter  for	a
-		 highpass effect with sharper cutoff.
+       highp frequency
+		 Apply a single pole recursive high-pass  filter.
+		 The  frequency	 response  drops  logarithmically
+		 with I frequency in the middle of the drop.  The
+		 slope of the filter is quite gentle.  See filter
+		 for a highpass effect with sharper cutoff.
 
        highpass frequency
-		 Butterworth  highpass	filter.	 Description com-
+		 Butterworth highpass filter.	Description  com-
 		 ming soon!
 
-       lowp center
-		 Apply a low-pass filter.  The frequency response
-		 drops	logarithmically	 with center frequency in
-		 the middle of the drop.  The slope of the filter
-		 is  quite  gentle.   See  filter  for	a lowpass
-		 effect with sharper cutoff.
+       lowp frequency
+		 Apply	a  single pole recursive low-pass filter.
+		 The  frequency	 response  drops  logarithmically
+		 with  frequency  in the middle of the drop.  The
+		 slope of the filter is quite gentle.  See filter
+		 for a lowpass effect with sharper cutoff.
 
        lowpass frequency
-		 Butterworth lowpass filter.  Description  coming
+		 Butterworth  lowpass filter.  Description coming
 		 soon!
 
        map	 Display a list of loops in a sample, and miscel-
 		 laneous loop info.
 
-       mask	 Add "masking  noise"  to  signal.   This  effect
-		 deliberately  adds  white  noise  to  a sound in
-		 order to mask quantization effects,  created  by
-		 the  process  of  playing a sound digitally.  It
-		 tends to mask buzzing voices, for  example.   It
-		 adds  1/2  bit of noise to the sound file at the
+       mask	 Add  "masking	noise"	to  signal.   This effect
+		 deliberately adds white  noise	 to  a	sound  in
+		 order	to  mask quantization effects, created by
+		 the process of playing a  sound  digitally.   It
+		 tends	to  mask buzzing voices, for example.  It
+		 adds 1/2 bit of noise to the sound file  at  the
 		 output bit depth.
 
-       pan direction
-		 Pan the sound of an audio file from one  channel
-		 to another.  This is done by changing the volume
-		 of the input channels so that it  fades  out  on
-		 one  channel  and  fades-in  on another.  If the
-		 number of input channels is different	then  the
-		 number of output channels then this effect tries
-		 to intelligently handle this.	For instance,  if
-		 the input contains 1 channel and the output con-
-		 tains 2 channels, then it will create the  miss-
-		 ing  channel  itself.	 The direction is a value
-		 from -1.0 to 1.0.  -1.0 represents far left  and
-		 1.0  represents  far  right.  Numbers in between
-		 will start the pan effect without totally muting
-		 the opposite channel.
 
-       phaser gain-in gain-out delay decay speed < -s | -t >
-		 Add  a	 phaser	 to  a sound sample.  Each triple
 
-
-
 			  July 24, 2000			       12
 
 
@@ -796,27 +796,50 @@
 SoX(1)							   SoX(1)
 
 
-		 delay/decay/speed gives the delay  in	millisec-
-		 onds  and the decay (relative to gain-in) with a
+       pan direction
+		 Pan  the sound of an audio file from one channel
+		 to another.  This is done by changing the volume
+		 of  the  input	 channels so that it fades out on
+		 one channel and fades-in  on  another.	  If  the
+		 number	 of  input channels is different then the
+		 number of output channels then this effect tries
+		 to  intelligently handle this.	 For instance, if
+		 the input contains 1 channel and the output con-
+		 tains	2 channels, then it will create the miss-
+		 ing channel itself.  The direction  is	 a  value
+		 from  -1.0 to 1.0.  -1.0 represents far left and
+		 1.0 represents far right.   Numbers  in  between
+		 will start the pan effect without totally muting
+		 the opposite channel.
+
+       phaser gain-in gain-out delay decay speed < -s | -t >
+		 Add a phaser to a  sound  sample.   Each  triple
+		 delay/decay/speed  gives  the delay in millisec-
+		 onds and the decay (relative to gain-in) with	a
 		 modulation  speed  in	Hz.   The  modulation  is
-		 either	 sinodial  (-s)	 or triangular (-t).  The
+		 either sinodial (-s) or  triangular  (-t).   The
 		 decay should be less than 0.5 to avoid feedback.
 		 Gain-out is the volume of the output.
 
-       pick	 Select	 the  left  or	right channel of a stereo
-		 sample, or one of four	 channels  in  a  quadro-
-		 phonic sample.
+       pick [ -1 | -2 | -3 | -4 | -l | -r ]
+		 Select the left or right  channel  of	a  stereo
+		 sample,  or  one  of  four channels in a quadro-
+		 phonic sample. The -l and -r  options	represent
+		 either	  the  left  or	 right	channel.   It  is
+		 required that you use	the  -c	 1  command  line
+		 option in order to force the output file to con-
+		 tain only 1 channel.
 
        pitch shift [ width interpole fade ]
-		 Change	 the  pitch of file without affecting its
+		 Change the pitch of file without  affecting  its
 		 duration by cross-fading shifted samples.  shift
 		 is given in cents. Use a positive value to shift
-		 to treble, negative  value  to	 shift	to  bass.
-		 Default  shift	 is 0.	width of window is in ms.
-		 Default width is 20ms. Try 30ms to lower  pitch,
-		 and  10ms to raise pitch.  interpole option, can
+		 to  treble,  negative	value  to  shift to bass.
+		 Default shift is 0.  width of window is  in  ms.
+		 Default  width is 20ms. Try 30ms to lower pitch,
+		 and 10ms to raise pitch.  interpole option,  can
 		 be "cubic" or "linear". Default is "cubic".  The
-		 fade  option,	can be "cos", "hamming", "linear"
+		 fade option, can be "cos",  "hamming",	 "linear"
 		 or "trapezoid".  Default is "cos".
 
        polyphase [ -w < nut / ham > ]
@@ -825,58 +848,59 @@
 
 		 [ -cutoff #  ]
 		 Translate input sampling rate to output sampling
-		 rate  via  polyphase  interpolation, a DSP algo-
-		 rithm.	 This method is slow  and  uses	 lots  of
-		 RAM, but gives much better results than rate.
+		 rate via polyphase interpolation,  a  DSP  algo-
+		 rithm.	  This	method	is  slow and uses lots of
 
-		 -w  <	nut / ham > : select either a Nuttal (~90
-		 dB stopband) or Hamming (~43 dB  stopband)  win-
-		 dow.  Default is nut.
 
-		 -width	 long / short / # : specify the (approxi-
-		 mate) width of the filter.  long  is  1024  sam-
-		 ples;	short  is 128 samples.	Alternatively, an
-		 exact number can be used.  Default is long.  The
-		 short	option is not recommended, as it produces
-		 poor quality results.
 
-		 -cutoff # : specify the filter cutoff	frequency
-		 in  terms  of	fraction  of frequency bandwidth,
-		 also know as the Nyquist frequency.  Please  see
-		 the  resample	effect for further information on
-		 Nyquist frequency.  If upsampling, then this  is
-		 the  fraction of the original signal that should
-		 go through.  If downsampling, this is the  frac-
-		 tion  of  the	signal	left  after downsampling.
-		 Default is 0.95.  Remember that this is a float.
+			  July 24, 2000			       13
 
 
 
 
-			  July 24, 2000			       13
 
+SoX(1)							   SoX(1)
 
 
+		 RAM, but gives much better results than rate.
 
+		 -w < nut / ham > : select either a  Nuttal  (~90
+		 dB  stopband)	or Hamming (~43 dB stopband) win-
+		 dow.  Default is nut.
 
-SoX(1)							   SoX(1)
+		 -width long / short / # : specify the	(approxi-
+		 mate)	width  of  the filter.	long is 1024 sam-
+		 ples; short is 128 samples.   Alternatively,  an
+		 exact number can be used.  Default is long.  The
+		 short option is not recommended, as it	 produces
+		 poor quality results.
 
+		 -cutoff  # : specify the filter cutoff frequency
+		 in terms of  fraction	of  frequency  bandwidth,
+		 also  know as the Nyquist frequency.  Please see
+		 the resample effect for further  information  on
+		 Nyquist  frequency.  If upsampling, then this is
+		 the fraction of the original signal that  should
+		 go  through.  If downsampling, this is the frac-
+		 tion of  the  signal  left  after  downsampling.
+		 Default is 0.95.  Remember that this is a float.
 
+
        rate	 Translate input sampling rate to output sampling
-		 rate via linear interpolation to the Least  Com-
+		 rate  via linear interpolation to the Least Com-
 		 mon Multiple of the two sampling rates.  This is
 		 the default effect if the two files have differ-
-		 ent  sampling	rates and the preview options was
+		 ent sampling rates and the preview  options  was
 		 specified.  This is fast but noisy: the spectrum
-		 of  the  original  sound will be shifted upwards
-		 and duplicated faintly when up-translating by	a
+		 of the original sound will  be	 shifted  upwards
+		 and  duplicated faintly when up-translating by a
 		 multiple.
 
-		 Lerp-ing  is  acceptable  for	cheap 8-bit sound
-		 hardware, but for CD-quality  sound  you  should
-		 instead  use  either  resample or polyphase.  If
+		 Lerp-ing is acceptable	 for  cheap  8-bit  sound
+		 hardware,  but	 for  CD-quality sound you should
+		 instead use either resample  or  polyphase.   If
 		 you are wondering which rate changing effects to
-		 use,  you  will want to read a detailed analysis
+		 use, you will want to read a  detailed	 analysis
 		 of  all  of  them  at	http://eakaw2.et.tu-dres-
 		 den.de/~wilde/resample/resample.html
 
@@ -883,26 +907,37 @@
        resample [ -qs | -q | -ql ] [ rolloff [ beta ] ]
 		 Translate input sampling rate to output sampling
 		 rate  via  simulated  analog  filtration.   This
-		 method	 is slower than rate, but gives much bet-
+		 method is slower than rate, but gives much  bet-
 		 ter results.
 
 		 By default, linear interpolation is used, with a
-		 window	 width	about  45 samples at the lower of
-		 the two rate.	This gives an accuracy	of  about
-		 16  bits, but insufficient stopband rejection in
-		 the case that you want to have	 rolloff  greater
+		 window width about 45 samples at  the	lower  of
+		 the  two  rate.  This gives an accuracy of about
+		 16 bits, but insufficient stopband rejection  in
+		 the  case  that you want to have rolloff greater
 		 than about 0.80 of the Nyquist frequency.
 
-		 The  -q*  options will change the default values
-		 for rolloff and beta as well  as  use	quadratic
-		 interpolation	of filter coefficients, resulting
+
+
+			  July 24, 2000			       14
+
+
+
+
+
+SoX(1)							   SoX(1)
+
+
+		 The -q* options will change the  default  values
+		 for  rolloff  and  beta as well as use quadratic
+		 interpolation of filter coefficients,	resulting
 		 in about 24 bits precision.  The -qs, -q, or -ql
-		 options  specify  increased accuracy at the cost
-		 of lower execution speed.   It	 is  optional  to
-		 specify  rolloff  and beta parameters when using
+		 options specify increased accuracy at	the  cost
+		 of  lower  execution  speed.	It is optional to
+		 specify rolloff and beta parameters  when  using
 		 the -q* options.
 
-		 Following is a table of the reasonable	 defaults
+		 Following  is a table of the reasonable defaults
 		 which are built-in to sox:
 
 		    Option  Window rolloff beta interpolation
@@ -914,77 +949,42 @@
 		    ------  ------ ------- ---- -------------
 
 		 -qs, -q, or -ql use window lengths of 45, 75, or
-		 149 samples, respectively, at the lower  sample-
+		 149  samples, respectively, at the lower sample-
 		 rate of the two files.	 This means progressively
-
-
-
-			  July 24, 2000			       14
-
-
-
-
-
-SoX(1)							   SoX(1)
-
-
-		 sharper stop-band rejection,  at  proportionally
+		 sharper  stop-band  rejection, at proportionally
 		 slower execution times.
 
-		 rolloff  refers  to the cut-off frequency of the
-		 low pass filter and is given  in  terms  of  the
-		 Nyquist  frequency  for  the  lower sample rate.
-		 rolloff therefore should  be  something  between
+		 rolloff refers to the cut-off frequency  of  the
+		 low  pass  filter  and	 is given in terms of the
+		 Nyquist frequency for	the  lower  sample  rate.
+		 rolloff  therefore  should  be something between
 		 0.0 and 1.0, in practice 0.8-0.95.  The defaults
 		 are indicated above.
 
 		 The Nyquist frequency is equal to (sample rate /
-		 2).   Logically,  this	 is  because the A/D con-
-		 verter needs at least	2  samples  to	detect	1
-		 cycle	at  the	 Nyquist  frequency.  Frequencies
-		 higher then the Nyquist will actually appear  as
-		 lower	frequencies  to	 the A/D converter and is
+		 2).  Logically, this is  because  the	A/D  con-
+		 verter	 needs	at  least  2  samples to detect 1
+		 cycle at  the	Nyquist	 frequency.   Frequencies
+		 higher	 then the Nyquist will actually appear as
+		 lower frequencies to the A/D  converter  and  is
 		 called aliasing.  Normally, A/D converts run the
-		 signal	 through a highpass filter first to avoid
+		 signal through a highpass filter first to  avoid
 		 these problems.
 
-		 Similar problems will happen  in  software  when
-		 reducing  the sample rate of an audio file (fre-
-		 quencies above the new Nyquist frequency can  be
-		 aliased  to  lower  frequencies).   Therefore, a
-		 good resample effect will remove  all	frequency
+		 Similar  problems  will  happen in software when
+		 reducing the sample rate of an audio file  (fre-
+		 quencies  above the new Nyquist frequency can be
+		 aliased to  lower  frequencies).   Therefore,	a
+		 good  resample	 effect will remove all frequency
 		 information above the new Nyquist frequency.
 
-		 The  rolloff  refers to how close to the Nyquist
+		 The rolloff refers to how close to  the  Nyquist
 		 frequency this cutoff is, with closer being bet-
-		 ter.	When  increasing  the  sample  rate of an
-		 audio file you would not expect to have any fre-
-		 quencies   exist  that	 are  past  the	 original
-		 Nyquist frequency.  Because of resampling  prop-
-		 erties,  it is common to have alaising data cre-
-		 ated that is above the	 old  Nyquist  frequency.
-		 In  that case the rolloff refers to how close to
-		 the original Nyquist frequency to use a highpass
-		 filter	 to  remove  this false data, with closer
-		 also being better.
+		 ter.  When increasing	the  sample  rate  of  an
+		 audio	file  you  would  not  expect to have any
 
-		 The beta parameter determines the type of filter
-		 window	 used.	Any value greater than 2.0 is the
-		 beta for a Kaiser window.  Beta <= 2.0 selects a
-		 Nuttall  window.  If unspecified, the default is
-		 a Kaiser window with beta 16.
 
-		 In the case of Kaiser window (beta > 2.0), lower
-		 betas	produce a somewhat faster transition from
-		 passband to stopband, at the cost of  noticeable
-		 artifacts.   A	 beta  of 16 is the default, beta
-		 less than 10 is not recommended.  If you want	a
-		 sharper  cutoff,  don't  use  low  beta's, use a
-		 longer	 sample	 window.   A  Nuttall  window  is
-		 selected  by specifying any 'beta' <= 2, and the
 
-
-
 			  July 24, 2000			       15
 
 
@@ -994,22 +994,45 @@
 SoX(1)							   SoX(1)
 
 
-		 Nuttall window has somewhat steeper cutoff  than
-		 the  default  Kaiser  window.	You will probably
-		 not need to  use  the	beta  parameter	 at  all,
-		 unless	 you are just curious about comparing the
+		 frequencies exist that	 are  past  the	 original
+		 Nyquist  frequency.  Because of resampling prop-
+		 erties, it is common to have alaising data  cre-
+		 ated  that  is	 above the old Nyquist frequency.
+		 In that case the rolloff refers to how close  to
+		 the original Nyquist frequency to use a highpass
+		 filter to remove this false  data,  with  closer
+		 also being better.
+
+		 The beta parameter determines the type of filter
+		 window used.  Any value greater than 2.0 is  the
+		 beta for a Kaiser window.  Beta <= 2.0 selects a
+		 Nuttall window.  If unspecified, the default  is
+		 a Kaiser window with beta 16.
+
+		 In the case of Kaiser window (beta > 2.0), lower
+		 betas produce a somewhat faster transition  from
+		 passband  to stopband, at the cost of noticeable
+		 artifacts.  A beta of 16 is  the  default,  beta
+		 less  than 10 is not recommended.  If you want a
+		 sharper cutoff, don't	use  low  beta's,  use	a
+		 longer	 sample	 window.   A  Nuttall  window  is
+		 selected by specifying any 'beta' <= 2, and  the
+		 Nuttall  window has somewhat steeper cutoff than
+		 the default Kaiser window.   You  will	 probably
+		 not  need  to	use  the  beta	parameter at all,
+		 unless you are just curious about comparing  the
 		 effects of Nuttall vs. Kaiser windows.
 
 		 This is the default effect if the two files have
-		 different  sampling  rates.   Default parameters
+		 different sampling  rates.   Default  parameters
 		 are, as indicated above, Kaiser window of length
 		 45, rolloff 0.80, beta 16, linear interpolation.
 
-		 NOTE: -qs is  only  slightly  slower,	but  more
+		 NOTE:	-qs  is	 only  slightly	 slower, but more
 		 accurate for 16-bit or higher precision.
 
-		 NOTE:	In many cases of up-sampling, no interpo-
-		 lation is needed, as exact  filter  coefficients
+		 NOTE: In many cases of up-sampling, no	 interpo-
+		 lation	 is  needed, as exact filter coefficients
 		 can be computed in a reasonable amount of space.
 		 To be precise, this is done when
 
@@ -1019,104 +1042,81 @@
 
        reverb gain-out delay [ delay ... ]
 		 Add reverberation to a sound sample.  Each delay
-		 is given in milliseconds  and	its  feedback  is
-		 depending  on	the  reverb-time in milliseconds.
-		 Each delay should be in the  range  of	 half  to
+		 is  given  in	milliseconds  and its feedback is
+		 depending on the  reverb-time	in  milliseconds.
+		 Each  delay  should  be  in the range of half to
 		 quarter of reverb-time to get a realistic rever-
 		 beration.  Gain-out is the volume of the output.
 
-       reverse	 Reverse  the  sound sample completely.	 Included
+
+
+
+			  July 24, 2000			       16
+
+
+
+
+
+SoX(1)							   SoX(1)
+
+
+       reverse	 Reverse the sound sample  completely.	 Included
 		 for finding Satanic subliminals.
 
        speed factor
-		 Speed up or down the sound, as a  magnetic  tape
+		 Speed	up  or down the sound, as a magnetic tape
 		 with a speed control.	It affects both pitch and
-		 time. A factor of 1.0 means no	 change,  and  is
+		 time.	A  factor  of 1.0 means no change, and is
 		 the  default.	 2.0  doubles  speed,  thus  time
-		 length is cut by a half and pitch is one  octave
-		 higher.   0.5 halves speed thus time length dou-
+		 length	 is cut by a half and pitch is one octave
+		 higher.  0.5 halves speed thus time length  dou-
 		 bles and pitch is one octave lower.
 
        split	 Turn a mono sample into a stereo sample by copy-
-		 ing  the  input  channel  to  the left and right
+		 ing the input channel	to  the	 left  and  right
 		 channels.
 
        stat [ -s n ] [-rms ] [ -v ] [ -d ]
-		 Do a statistical check on the	input  file,  and
+		 Do  a	statistical  check on the input file, and
 		 print results on the standard error file.  Audio
-		 data is passed unmodified from input  to  output
+		 data  is  passed unmodified from input to output
 		 file unless used along with the -e option.
 
 		 The "Volume Adjustment:" field in the statistics
-
-
-
-			  July 24, 2000			       16
-
-
-
-
-
-SoX(1)							   SoX(1)
-
-
-		 gives you the argument to the	-v  number  which
+		 gives	you  the  argument to the -v number which
 		 will make the sample as loud as possible without
 		 clipping.
 
 		 The option -v will print out the "Volume Adjust-
-		 ment:"	 field's  value	 only  and  return.  This
-		 could be of use in scripts to auto  convert  the
+		 ment:" field's	 value	only  and  return.   This
+		 could	be  of use in scripts to auto convert the
 		 volume.
 
-		 The  -s n option is used to scale the input data
-		 by a given factor.  The default value	of  n  is
-		 the   max   value  of	a  signed  long	 variable
+		 The -s n option is used to scale the input  data
+		 by  a	given  factor.	The default value of n is
+		 the  max  value  of  a	 signed	  long	 variable
 		 (0x7fffffff).	Internal effects always work with
-		 signed	 long  PCM  data  and so the value should
+		 signed long PCM data and  so  the  value  should
 		 relate to this fact.
 
-		 The -rms option will convert all output  average
+		 The  -rms option will convert all output average
 		 values to root mean square format.
 
 		 There is also an optional parameter -d that will
-		 print out a hex dump of the sound file from  the
-		 internal  buffer  that	 is  in 32-bit signed PCM
-		 data.	This is mainly only of	use  in	 tracking
-		 down  endian  problems	 that  creep in to sox on
+		 print	out a hex dump of the sound file from the
+		 internal buffer that is  in  32-bit  signed  PCM
+		 data.	 This  is  mainly only of use in tracking
+		 down endian problems that creep  in  to  sox  on
 		 cross-platform versions.
 
 
        stretch factor [window fade shift fading]
-		 Time stretch file  by	a  given  factor.  Change
+		 Time  stretch	file  by  a  given factor. Change
 		 duration without affecting the pitch.	factor of
-		 stretching: >1.0 lengthen,  <1.0  shorten  dura-
-		 tion.	 window	 size  is in ms. Default is 20ms.
-		 The fade option, can be "lin".	 shift ratio,  in
-		 [0.0  1.0].  Default  depends on stretch factor.
-		 1.0 to shorten, 0.8  to  lengthen.   The  fading
-		 ratio,	 in  [0.0  0.5].  The  amount of a fade's
-		 default depends on factor and shift.
+		 stretching:	>1.0   lengthen,   <1.0	  shorten
 
-       swap [ 1 2 | 1 2 3 4 ]
-		 Swap  channels	 in  multi-channel  sound  files.
-		 Optionally,  you  may	specify the channel order
-		 you would like the output in.	This defaults  to
-		 output channel 2 and then 1 for stereo and 2, 1,
-		 4, 3 for quad-channels.  An interesting  feature
-		 is  that  you	may  duplicate a given channel by
-		 overwriting another.  This is done by	repeating
-		 an  output  channel  on  the  command line.  For
-		 example, swap 2 2 will overwrite channel 1  with
-		 channel  2's  data;  creating a stereo file with
-		 both channels containing the same audio data.
 
-       vibro speed  [ depth ]
-		 Add the world-famous  Fender  Vibro-Champ  sound
-		 effect to a sound sample by using a sine wave as
 
-
-
 			  July 24, 2000			       17
 
 
@@ -1126,42 +1126,77 @@
 SoX(1)							   SoX(1)
 
 
+		 duration.  window size	 is  in	 ms.  Default  is
+		 20ms.	The  fade  option,  can	 be "lin".  shift
+		 ratio, in [0.0 1.0]. Default depends on  stretch
+		 factor.  1.0  to  shorten, 0.8 to lengthen.  The
+		 fading ratio, in [0.0	0.5].  The  amount  of	a
+		 fade's default depends on factor and shift.
+
+       swap [ 1 2 | 1 2 3 4 ]
+		 Swap  channels	 in  multi-channel  sound  files.
+		 Optionally, you may specify  the  channel  order
+		 you  would like the output in.	 This defaults to
+		 output channel 2 and then 1 for stereo and 2, 1,
+		 4,  3 for quad-channels.  An interesting feature
+		 is that you may duplicate  a  given  channel  by
+		 overwriting  another.	This is done by repeating
+		 an output channel  on	the  command  line.   For
+		 example,  swap 2 2 will overwrite channel 1 with
+		 channel 2's data; creating a  stereo  file  with
+		 both channels containing the same audio data.
+
+       vibro speed  [ depth ]
+		 Add  the  world-famous	 Fender Vibro-Champ sound
+		 effect to a sound sample by using a sine wave as
 		 the volume knob.  Speed gives the Hertz value of
-		 the  wave.   This must be under 30.  Depth gives
-		 the amount the volume is cut into  by	the  sine
-		 wave,	ranging 0.0 to 1.0 and defaulting to 0.5.
+		 the wave.  This must be under 30.   Depth  gives
+		 the  amount  the  volume is cut into by the sine
+		 wave, ranging 0.0 to 1.0 and defaulting to  0.5.
 
        vol gain	 [ type ]
-		 The vol effect is much	 like  the  command  line
-		 option	 -v.   It allows you to adjust the volume
-		 of an input file and allows you to  specify  the
-		 adjustment  in	 relation to amplitude, power, or
+		 The  vol  effect  is  much like the command line
+		 option -v.  It allows you to adjust  the  volume
+		 of  an	 input file and allows you to specify the
+		 adjustment in relation to amplitude,  power,  or
 		 dB.  When type is amplitude then a linear change
 		 of the amplitude is performed based on the gain.
-		 Therefore, a value of 1.0 will keep  the  volume
-		 the  same, 0.0 to < 1.0 will cause the volume to
+		 Therefore,  a	value of 1.0 will keep the volume
+		 the same, 0.0 to < 1.0 will cause the volume  to
 		 decrease and values of > 1.0 will cause the vol-
-		 ume  to increase.  Beware of clipping audio data
-		 when the gain is greater then 1.0.   A	 negative
-		 value	performs  the  same adjustment while also
+		 ume to increase.  Beware of clipping audio  data
+		 when  the  gain is greater then 1.0.  A negative
+		 value performs the same  adjustment  while  also
 		 changing the phase.
-		 When type is power then  a  value  of	1.0  also
+		 When  type  is	 power	then  a value of 1.0 also
 		 means no change in volume.
-		 When  type  is	 dB the amplitude is change loga-
-		 rithmically.  0.0 is constant while  +6  doubles
+		 When type is dB the amplitude	is  change  loga-
+		 rithmically.	0.0  is constant while +6 doubles
 		 the amplitude.
 
-       Sox  enforces certain effects.  If the two files have dif-
+       Sox enforces certain effects.  If the two files have  dif-
        ferent sampling rates, the requested effect must be one of
-       polyphase,  rate, or resample.  If the two files have dif-
-       ferent numbers of channels, the avg or other channel  mix-
+       polyphase, rate, or resample.  If the two files have  dif-
+       ferent  numbers of channels, the avg or other channel mix-
        ing effect must be requested.
 
+
+
+
+			  July 24, 2000			       18
+
+
+
+
+
+SoX(1)							   SoX(1)
+
+
 BUGS
-       The  syntax  is horrific.  Thats the breaks when trying to
+       The syntax is horrific.	Thats the breaks when  trying  to
        handle all things from the command line.
 
-       Please report any bugs found in this  version  of  sox  to
+       Please  report  any  bugs  found in this version of sox to
        Chris Bagwell (cbagwell@sprynet.com)
 
 FILES
@@ -1169,10 +1204,10 @@
        play(1), rec(1), soxexam(1)
 
 NOTICES
-       The  version  of	 Sox that accompanies this manual page is
-       support by Chris Bagwell	 (cbagwell@sprynet.com).   Please
+       The version of Sox that accompanies this	 manual	 page  is
+       support	by  Chris Bagwell (cbagwell@sprynet.com).  Please
        refer any questions regarding it to this address.  You may
-       obtain  the  latest  version   at   the	 the   web   site
+       obtain	the   latest   version	 at   the  the	web  site
        http://home.sprynet.com/~cbagwell/sox.html
 
 
@@ -1183,6 +1218,37 @@
 
 
 
-			  July 24, 2000			       18
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+			  July 24, 2000			       19
 
 
--- a/src/auto.c
+++ b/src/auto.c
@@ -64,18 +64,17 @@
 	else if (strncmp(header, "2BIT", 4) == 0) {
 		type = "avr";
 	}
-	else if (header[0] == 0 && header[1] == 0) {
-		int rate = (header[2] & 0xff) + ((header[3] & 0xff) << 8);
-		if (rate >= 4000 && rate <= 25000)
-			type = "sndr";
+	else if (strncmp(header, "NIST_1A", 4)) {
+	        type = "sph";
 	}
+
   	if (type == 0) {
   		printf("Type AUTO doesn't recognize this header\n");
-                printf("Trying: -t raw -r 11000 -b -u\n\n");
+                printf("Trying: -t raw -r 44100 -s -w\n\n");
                 type = "raw";
-                ft->info.rate = 11000;
-                ft->info.size = ST_SIZE_BYTE;
-                ft->info.encoding = ST_ENCODING_UNSIGNED;
+                ft->info.rate = 44100;
+                ft->info.size = ST_SIZE_WORD;
+                ft->info.encoding = ST_ENCODING_SIGN2;
                 }
 	st_report("Type AUTO changed to %s", type);
 	ft->filetype = type;
--- a/src/handlers.c
+++ b/src/handlers.c
@@ -46,7 +46,7 @@
 
 static char *aunames[] = {
 	"au",
-#ifdef	NeXT
+#ifndef	DOS
 	"snd",
 #endif
 	(char *) 0
@@ -211,13 +211,6 @@
 extern int  st_smpstartwrite();
 extern int  st_smpstopwrite();
 
-static char *sndrnames[] = {
-	"sndr",
-	(char *) 0
-};
-
-extern int st_sndrstartwrite();
-
 static char *sndtnames[] = {
 	"sndt",
 #ifdef	DOS
@@ -239,6 +232,7 @@
 extern int  st_spherestartread();
 extern LONG st_sphereread();
 extern int  st_spherestartwrite();
+extern LONG st_spherewrite();
 extern int  st_spherestopwrite();
 
 #if	defined(SUNAUDIO_PLAYER)
@@ -413,15 +407,12 @@
 	{smpnames, ST_FILE_STEREO | ST_FILE_LOOPS,/* SampleVision sound */
 		st_smpstartread, st_smpread, st_nothing,
 		st_smpstartwrite, st_smpwrite, st_smpstopwrite},
-	{sndrnames, ST_FILE_STEREO,		/* Sounder Sound File */
-		st_sndtstartread, st_rawread, st_rawstopread,
-		st_sndrstartwrite, st_rawwrite, st_rawstopwrite},
 	{sndtnames, ST_FILE_STEREO,		/* Sndtool Sound File */
 		st_sndtstartread, st_rawread, st_rawstopread, 
 		st_sndtstartwrite, st_sndtwrite, st_sndtstopwrite},
 	{spherenames, ST_FILE_STEREO,		/* NIST Sphere File */
 	        st_spherestartread, st_sphereread, st_rawstopread,
-		st_spherestartwrite, st_rawwrite, st_spherestopwrite},
+		st_spherestartwrite, st_spherewrite, st_spherestopwrite},
 #if	defined(SUNAUDIO_PLAYER)
 	{sunnames, ST_FILE_STEREO,		/* Sun /dev/audio player */
 		st_sunstartread, st_rawread, st_rawstopread,
--- a/src/highp.c
+++ b/src/highp.c
@@ -10,15 +10,30 @@
 /*
  * Sound Tools High-Pass effect file.
  *
- * Algorithm:  1nd order filter.
- * From Fugue source code:
+ * Algorithm:  Recursive single pole high-pass filter
  *
- * 	output[N] = B * (output[N-1] - input[N-1] + input[N])
+ * Reference: The Scientist and Engineer's Guide to Digital Processing
  *
- * 	A = 2.0 * pi * center
- * 	B = exp(-A / frequency)
+ * 	output[N] = A0 * input[N] + A1 * input[N-1] + B1 * output[N-1]
+ *
+ *      X  = exp(-2.0 * pi * Fc)
+ *      A0 = (1 + X) / 2
+ *      A1 = -(1 + X) / 2
+ *      B1 = X
+ *      Fc = cutoff freq / sample rate
+ *
+ * Mimics an RC high-pass filter:
+ *
+ *        || C
+ *    ----||--------->
+ *        ||    |
+ *              <
+ *              > R
+ *              <
+ *              |
+ *              V
+ *
  */
-/* SJB: Note: highp filter is currently broken, see test gnuplot graphs */
 
 #include <math.h>
 #include "st.h"
@@ -25,9 +40,9 @@
 
 /* Private data for Highpass effect */
 typedef struct highpstuff {
-	float	center;
-	double	A, B;
-	double	in1, out1;
+	float	cutoff;
+	double	A0, A1, B1;
+	double	inm1, outm1;
 } *highp_t;
 
 /*
@@ -40,9 +55,9 @@
 {
 	highp_t highp = (highp_t) effp->priv;
 
-	if ((n < 1) || !sscanf(argv[0], "%f", &highp->center))
+	if ((n < 1) || !sscanf(argv[0], "%f", &highp->cutoff))
 	{
-		st_fail("Usage: highp center");
+		st_fail("Usage: highp cutoff");
 		return (ST_EOF);
 	}
 	return (ST_SUCCESS);
@@ -55,16 +70,17 @@
 eff_t effp;
 {
 	highp_t highp = (highp_t) effp->priv;
-	if (highp->center > effp->ininfo.rate*2)
+	if (highp->cutoff > effp->ininfo.rate/2)
 	{
-		st_fail("Highpass: center must be < minimum data rate*2\n");
+		st_fail("Highpass: cutoff must be < sample rate / 2 (Nyquest rate)\n");
 		return (ST_EOF);
 	}
-	
-	highp->A = (M_PI * 2.0 * highp->center) / effp->ininfo.rate;
-	highp->B = exp(-highp->A / effp->ininfo.rate);
-	highp->in1 = 0.0;
-	highp->out1 = 0.0;
+
+	highp->B1 = exp((-2.0 * M_PI * (highp->cutoff / effp->ininfo.rate)));
+	highp->A0 = (1 + highp->B1) / 2;
+	highp->A1 = (-1 * (1 + highp->B1)) / 2;
+	highp->inm1 = 0.0;
+	highp->outm1 = 0.0;
 	return (ST_SUCCESS);
 }
 
@@ -84,21 +100,20 @@
 	LONG l;
 
 	len = ((*isamp > *osamp) ? *osamp : *isamp);
-	d = highp->out1;
 
-	/* yeah yeah yeah registers & integer arithmetic yeah yeah yeah */
 	for(done = 0; done < len; done++) {
 		l = *ibuf++;
-		d = (highp->B * ((d - highp->in1) + (double) l)) / 65536.0;
-		d *= 0.8;
-		if (d > 32767)
-			d = 32767;
-		if (d < - 32767)
-			d = - 32767;
-		highp->in1 = l;
-		*obuf++ = d * 65536L;
+		d = highp->A0 * l + 
+		    highp->A1 * highp->inm1 + 
+		    highp->B1 * highp->outm1;
+                if (d < -2147483647L)
+                    d = -2147483647L;
+                else if (d > 2147483647L)
+                    d = 2147483647L;
+		highp->inm1 = l;
+		highp->outm1 = d;
+		*obuf++ = d;
 	}
-	highp->out1 = d;
 	*isamp = len;
 	*osamp = len;
 	return (ST_SUCCESS);
--- a/src/lowp.c
+++ b/src/lowp.c
@@ -11,13 +11,27 @@
 /*
  * Sound Tools Low-Pass effect file.
  *
- * Algorithm:  1nd order filter.
- * From Fugue source code:
+ * Algorithm:  Recursive single pole lowpass filter
  *
- * 	output[N] = input[N] * A + input[N-1] * B
+ * Reference: The Scientist and Engineer's Guide to Digital Signal Processing
  *
- * 	A = 2.0 * pi * center
- * 	B = exp(-A / frequency)
+ * 	output[N] = input[N] * A + output[N-1] * B
+ *
+ * 	X = exp(-2.0 * pi * Fc)
+ * 	A = 1 - X
+ * 	B = X
+ * 	Fc = cutoff freq / sample rate
+ *
+ * Mimics an RC low-pass filter:   
+ *
+ *     ---/\/\/\/\----------->
+ *                    |
+ *                   --- C
+ *                   ---
+ *                    |
+ *                    |
+ *                    V
+ *
  */
 
 #include <math.h>
@@ -25,9 +39,9 @@
 
 /* Private data for Lowpass effect */
 typedef struct lowpstuff {
-	float	center;
+	float	cutoff;
 	double	A, B;
-	double	in1;
+	double	outm1;
 } *lowp_t;
 
 /*
@@ -40,9 +54,9 @@
 {
 	lowp_t lowp = (lowp_t) effp->priv;
 
-	if ((n < 1) || !sscanf(argv[0], "%f", &lowp->center))
+	if ((n < 1) || !sscanf(argv[0], "%f", &lowp->cutoff))
 	{
-		st_fail("Usage: lowp center");
+		st_fail("Usage: lowp cutoff");
 		return (ST_EOF);
 	}
 	return (ST_SUCCESS);
@@ -55,15 +69,15 @@
 eff_t effp;
 {
 	lowp_t lowp = (lowp_t) effp->priv;
-	if (lowp->center > effp->ininfo.rate*2)
+	if (lowp->cutoff > effp->ininfo.rate / 2)
 	{
-		st_fail("Lowpass: center must be < minimum data rate*2\n");
+		st_fail("Lowpass: cutoff must be < sample rate / 2 (Nyquest rate)\n");
 		return (ST_EOF);
 	}
 
-	lowp->A = (M_PI * 2.0 * lowp->center) / effp->ininfo.rate;
-	lowp->B = exp(-lowp->A / effp->ininfo.rate);
-	lowp->in1 = 0.0;
+	lowp->B = exp((-2.0 * M_PI * (lowp->cutoff / effp->ininfo.rate)));
+	lowp->A = 1 - lowp->B;
+	lowp->outm1 = 0.0;
 	return (ST_SUCCESS);
 }
 
@@ -84,17 +98,15 @@
 
 	len = ((*isamp > *osamp) ? *osamp : *isamp);
 
-	/* yeah yeah yeah registers & integer arithmetic yeah yeah yeah */
 	for(done = 0; done < len; done++) {
 		l = *ibuf++;
-		d = lowp->A * (l / 65536L) + lowp->B * (lowp->in1 / 65536L);
-		d *= 0.8;
-		if (d > 32767)
-			d = 32767;
-		if (d < - 32767)
-			d = - 32767;
-		lowp->in1 = l;
-		*obuf++ = d * 65536L;
+		d = lowp->A * l + lowp->B * lowp->outm1;
+		if (d < -2147483647L)
+		    d = -2147483647L;
+		else if (d > 2147483647L)
+		    d = 2147483647L;
+		lowp->outm1 = l;
+		*obuf++ = d;
 	}
 	*isamp = len;
 	*osamp = len;
--- a/src/sphere.c
+++ b/src/sphere.c
@@ -16,6 +16,7 @@
 /* Private data for SKEL file */
 typedef struct spherestuff {
 	char	shorten_check[4];
+	ULONG   numSamples;
 } *sphere_t;
 
 /*
@@ -68,11 +69,12 @@
 
 	while (strncmp(buf, "end_head", 8) != 0)
 	{
-	    if (strncmp(buf, "sample_n_bytes", 14) == 0)
+	    if (strncmp(buf, "sample_n_bytes", 14) == 0 && ft->info.size == -1)
 	    {
 		sscanf(buf, "%s %s %d", fldname, fldtype, &ft->info.size);
 	    }
-	    if (strncmp(buf, "channel_count", 13) == 0)
+	    if (strncmp(buf, "channel_count", 13) == 0 && 
+		ft->info.channels == -1)
 	    {
 		sscanf(buf, "%s %s %d", fldname, fldtype, &ft->info.channels);
 	    }
@@ -82,12 +84,14 @@
 		/* Only bother looking for ulaw flag.  All others
 		 * should be caught below by default PCM check
 		 */
-		if (strncmp(fldsval,"ulaw",4) == 0)
+		if (ft->info.encoding == -1 && 
+		    strncmp(fldsval,"ulaw",4) == 0)
 		{
 		    ft->info.encoding = ST_ENCODING_ULAW;
 		}
 	    }
-	    if (strncmp(buf, "sample_rate", 11) == 0)
+	    if (strncmp(buf, "sample_rate", 11) == 0 &&
+		ft->info.rate == -1)
 	    {
 #ifdef __alpha__
 		sscanf(buf, "%s %s %d", fldname, fldtype, &ft->info.rate);
@@ -126,7 +130,7 @@
 	}
 
 	if (ft->info.size == -1)
-	    ft->info.size = 1;
+	    ft->info.size = ST_SIZE_BYTE;
 
 	/* sample_coding is optional and is PCM if missing.
 	 * This means encoding is signed if size = word or
@@ -151,10 +155,15 @@
 	}
 
 	sphere->shorten_check[0] = 0;
+
+	/* TODO: Check first four bytes of data to see if its shorten
+	 * compressed or not.  This data will need to be written to
+	 * buffer during first st_sphereread().
+	 */
 #if 0
 	st_reads(ft, sphere->shorten_check, 4);
 
-	if (!strcmp(sphere->shorten_check,"abcd"))
+	if (!strcmp(sphere->shorten_check,"ajkg"))
 	{
 	    st_fail("File uses shorten compression, can not handle this.\n");
 	    return(ST_EOF);
@@ -179,6 +188,9 @@
 
     if (sphere->shorten_check[0])
     {
+	/* TODO: put these 4 bytes into the buffer.  Requires
+	 * knowing how to process ulaw and all version of PCM data size.
+	 */
 	sphere->shorten_check[0] = 0;
     }
     return st_rawread(ft, buf, len);
@@ -189,6 +201,7 @@
 {
     int rc;
     int x;
+    sphere_t sphere = (sphere_t) ft->priv;
 
     if (!ft->seekable)
     {
@@ -196,6 +209,19 @@
 	return (ST_EOF);
     }
 
+    switch (ft->info.encoding)
+    {
+	case ST_ENCODING_ULAW:
+	case ST_ENCODING_SIGN2:
+	case ST_ENCODING_UNSIGNED:
+	    break;
+	default:
+	    st_fail("SPHERE format only supports ulaw and PCM data.");
+	    return(ST_EOF);
+    }
+
+    sphere->numSamples = 0;
+
     /* Needed for rawwrite */
     rc = st_rawstartwrite(ft);
     if (rc)
@@ -210,11 +236,22 @@
 	
 }
 
+LONG st_spherewrite(ft, buf, len) 
+ft_t ft;
+LONG *buf, len;
+{
+    sphere_t sphere = (sphere_t) ft->priv;
+
+    sphere->numSamples += len; /* must later be divided by channels */
+    return st_rawwrite(ft, buf, len);
+}
+
 int st_spherestopwrite(ft) 
 ft_t ft;
 {
     int rc;
     char buf[128];
+    sphere_t sphere = (sphere_t) ft->priv;
 
     rc = st_rawstopwrite(ft);
     if (rc)
@@ -229,7 +266,12 @@
     st_writes(ft, "NIST_1A\n");
     st_writes(ft, "   1024\n");
 
-    /* FIXME: Need sample_count */
+#ifdef __alpha__
+    sprintf(buf, "sample_count -i %d\n", sphere->numSamples/ft->info.channels);
+#else
+    sprintf(buf, "sample_count -i %ld\n", sphere->numSamples/ft->info.channels);
+#endif
+    st_writes(ft, buf);
 
     sprintf(buf, "sample_n_bytes -i %d\n", ft->info.size);
     st_writes(ft, buf);
--- a/src/stat.c
+++ b/src/stat.c
@@ -208,7 +208,7 @@
 		            if (x1 == (stat->fft_size/2) - 1)
 		                magnitude *= 2.0;
 	                }
-	                printf("%f  %f\n",ffa*x1, magnitude);
+	                fprintf(stderr,"%f  %f\n",ffa*x1, magnitude);
 	            }
 		}