diff --git a/Makefile.linux b/Makefile.linux
index ed8b418..d695f92 100644
--- a/Makefile.linux
+++ b/Makefile.linux
@@ -735,7 +735,7 @@ install-rpi :
 # Combine some unit tests into a single regression sanity check.
 
 
-check : dtest ttest tttexttest pftest tlmtest lltest enctest kisstest pad2test xidtest dtmftest check-modem1200 check-modem300 check-modem9600 check-modem19200 check-modem2400 check-modem4800
+check : dtest ttest tttexttest pftest tlmtest lltest enctest kisstest pad2test xidtest dtmftest check-modem1200 check-modem300 check-modem9600 check-modem19200 check-modem2400 check-modem2400-g check-modem4800
 
 # Can we encode and decode at popular data rates?
 
@@ -753,14 +753,14 @@ check-modem300 : gen_packets atest
 
 check-modem9600 : gen_packets atest
 	./gen_packets -B9600 -n 100 -o /tmp/test96.wav
-	./atest -B9600 -F0 -L50 -G54 /tmp/test96.wav
-	./atest -B9600 -F1 -L55 -G59 /tmp/test96.wav
+	./atest -B9600 -F0 -L61 -G65 /tmp/test96.wav
+	./atest -B9600 -F1 -L62 -G66 /tmp/test96.wav
 	rm /tmp/test96.wav
 
 check-modem19200 : gen_packets atest
 	./gen_packets -r 96000 -B19200 -n 100 -o /tmp/test19.wav
-	./atest -B19200 -F0 -L55 -G59 /tmp/test19.wav
-	./atest -B19200 -F1 -L60 -G64 /tmp/test19.wav
+	./atest -B19200 -F0 -L60 -G64 /tmp/test19.wav
+	./atest -B19200 -F1 -L64 -G68 /tmp/test19.wav
 	rm /tmp/test19.wav
 
 check-modem2400 : gen_packets atest
@@ -769,6 +769,11 @@ check-modem2400 : gen_packets atest
 	./atest -B2400 -F1 -L80 -G87 /tmp/test24.wav
 	rm /tmp/test24.wav
 
+check-modem2400-g : gen_packets atest
+	./gen_packets -B2400 -g -n 100 -o /tmp/test24-g.wav
+	./atest -B2400 -g -F0 -L99 -G100 /tmp/test24-g.wav
+	rm /tmp/test24-g.wav
+
 check-modem4800 : gen_packets atest
 	./gen_packets -B2400 -n 100 -o /tmp/test48.wav
 	./atest -B2400 -F0 -L70 -G79 /tmp/test48.wav
diff --git a/Makefile.win b/Makefile.win
index 671d1a0..51f3519 100644
--- a/Makefile.win
+++ b/Makefile.win
@@ -16,7 +16,7 @@
 #
 
 
-all : direwolf decode_aprs text2tt tt2text ll2utm utm2ll aclients log2gpx gen_packets atest ttcalc tnctest kissutil
+all : direwolf decode_aprs text2tt tt2text ll2utm utm2ll aclients log2gpx gen_packets atest ttcalc tnctest tnctest-issue-132 kissutil
 
 
 # People say we need -mthreads option for threads to work properly.
@@ -197,13 +197,13 @@ gen_packets : gen_packets.o  ax25_pad.o hdlc_send.o fcs_calc.o gen_tone.o morse.
 	$(CC) $(CFLAGS) -o $@ $^
 
 
-# Connected mode packet application server.
 
-appserver : appserver.o textcolor.o ax25_pad.o fcs_calc.o misc.a
+# Connected mode sample applications for talking to network TNC with AGW protocol.
+
+appserver : appserver.o agwlib.o dwsock.o textcolor.o dtime_now.o misc.a
 	$(CC) $(CFLAGS) -o $@ $^ -lwinmm -lws2_32
 
 
-
 # -------------------------------------------  Libraries  --------------------------------------------
 
 
@@ -277,7 +277,7 @@ strlcat.o : misc/strlcat.c
 
 # Combine some unit tests into a single regression sanity check.
 
-check : dtest ttest tttexttest pftest tlmtest lltest enctest kisstest pad2test xidtest dtmftest check-modem1200 check-modem300 check-modem9600 check-modem19200 check-modem2400 check-modem4800
+check : dtest ttest tttexttest pftest tlmtest lltest enctest kisstest pad2test xidtest dtmftest check-modem1200 check-modem300 check-modem9600 check-modem19200 check-modem2400 check-modem2400-g check-modem4800
 
 # Can we encode and decode at popular data rates?
 # Verify that single bit fixup increases the count.
@@ -305,8 +305,8 @@ check-modem9600 : gen_packets atest
 	sleep 1
 	gen_packets -B9600 -n 100 -o test96.wav
 	sleep 1
-	atest -B9600 -F0 -L50 -G54 test96.wav
-	atest -B9600 -F1 -L55 -G59 test96.wav
+	atest -B9600 -F0 -L61 -G65 test96.wav
+	atest -B9600 -F1 -L62 -G66 test96.wav
 	sleep 1
 	rm test96.wav
 
@@ -319,8 +319,8 @@ check-modem19200 : gen_packets atest
 	sleep 1
 	gen_packets -r 96000 -B19200 -n 100 -o test19.wav
 	sleep 1
-	atest -B19200 -F0 -L55 -G59 test19.wav
-	atest -B19200 -F1 -L60 -G64 test19.wav
+	atest -B19200 -F0 -L60 -G64 test19.wav
+	atest -B19200 -F1 -L64 -G68 test19.wav
 	sleep 1
 	rm test19.wav
 
@@ -332,6 +332,13 @@ check-modem2400 : gen_packets atest
 	sleep 1
 	rm test24.wav
 
+check-modem2400-g : gen_packets atest
+	gen_packets -B2400 -g -n 100 -o test24-g.wav
+	sleep 1
+	atest -B2400 -g -F0 -L99 -G100 test24-g.wav
+	sleep 1
+	rm test24-g.wav
+
 check-modem4800 : gen_packets atest
 	gen_packets -B4800 -n 100 -o test48.wav
 	sleep 1
@@ -467,6 +474,9 @@ dtmftest : dtmf.c textcolor.o
 tnctest : tnctest.c textcolor.o dtime_now.o serial_port.o misc.a
 	$(CC) $(CFLAGS) -o $@ $^ -lwinmm -lws2_32
 
+tnctest-issue-132 : tnctest-issue-132.c textcolor.o dtime_now.o serial_port.o misc.a
+	$(CC) $(CFLAGS) -o $@ $^ -lwinmm -lws2_32
+
 
 # For tweaking the demodulator.
 
@@ -481,7 +491,7 @@ testagc : atest.c demod.c dsp.c demod_afsk.c demod_psk.c demod_9600.o fsk_demod_
 		dwgpsnmea.o dwgps.o serial_port.o tt_text.o dtime_now.o regex.a misc.a
 	rm -f atest.exe
 	$(CC) $(CFLAGS) -o atest $^
-	./atest -P GGG- -F 0 ../02_Track_2.wav | grep "packets decoded in" >atest.out
+	./atest -P H+ -F 0 ../01_Track_1.wav ../02_Track_2.wav | grep "packets decoded in" >atest.out
 	echo " " > tune.h
 
 
@@ -509,6 +519,8 @@ testagc96 : atest.c fsk_fast_filter.h tune.h demod.c demod_afsk.c demod_psk.c de
 	rm -f atest96.exe
 	$(CC) $(CFLAGS) -o atest96 $^
 	./atest96 -B 9600 ../walkabout9600c.wav | grep "packets decoded in" >atest.out
+	#./atest96 -B 9600 ../walkabout9600c.wav noisy96.wav zzz16.wav zzz16.wav zzz16.wav zzz8.wav zzz8.wav zzz8.wav | grep "packets decoded in" >atest.out
+	#./atest96 -B 9600 zzz16.wav zzz8.wav | grep "packets decoded in" >atest.out
 	#./atest96 -B 9600 noisy96.wav | grep "packets decoded in" >atest.out
 	#./atest96 -B 9600 19990303_0225_9600_8bis_22kHz.wav | grep "packets decoded in" >atest.out
 	#./atest96 -B 9600  19990303_0225_9600_16bit_22kHz.wav | grep "packets decoded in" >atest.out
@@ -561,10 +573,15 @@ aclients : aclients.c ax25_pad.c fcs_calc.c textcolor.c misc.a regex.a
 
 # Note:  kiss_frame.c has conditional compilation on KISSUTIL.
 
-kissutil : kissutil.c kiss_frame.c ax25_pad.o fcs_calc.o textcolor.o serial_port.o sock.o dtime_now.o misc.a regex.a
+kissutil : kissutil.c kiss_frame.c ax25_pad.o fcs_calc.o textcolor.o serial_port.o dwsock.o dtime_now.o misc.a regex.a
 	$(CC) $(CFLAGS) -DKISSUTIL -o $@ $^ -lwinmm -lws2_32
 
 
+mqtest : aprsmsg.c kiss_frame.c encode_aprs.o ax25_pad.o fcs_calc.o textcolor.o serial_port.o dwsock.o dtime_now.o latlong.o misc.a regex.a
+	$(CC) $(CFLAGS) -DMQTEST -DKISSUTIL -o $@ $^ -lwinmm -lws2_32
+
+
+
 # Touch Tone to Speech sample application.
 
 ttcalc : ttcalc.o ax25_pad.o fcs_calc.o textcolor.o misc.a regex.a
@@ -585,6 +602,7 @@ walk96 : walk96.c dwgps.o dwgpsnmea.o kiss_frame.o \
 
 
 
+
 #--------------------------------------------------------------
 
 
diff --git a/atest.c b/atest.c
index 9715712..67b8ed9 100644
--- a/atest.c
+++ b/atest.c
@@ -140,7 +140,10 @@ static int e_o_f;
 static int packets_decoded_one = 0;
 static int packets_decoded_total = 0;
 static int decimate = 0;		/* Reduce that sampling rate if set. */
-					/* 1 = normal, 2 = half, etc. */
+					/* 1 = normal, 2 = half, 3 = 1/3, etc. */
+
+static int upsample = 0;		/* Upsample for G3RUH decoder. */
+					/* Non-zero will override the default. */
 
 static struct audio_s my_audio_config;
 
@@ -175,6 +178,13 @@ static int sample_number = -1;		/* Sample number from the file. */
 					/* Use to print timestamp, relative to beginning */
 					/* of file, when frame was decoded. */
 
+// command line options.
+
+static int B_opt = DEFAULT_BAUD;	// Bits per second.  Need to change all baud references to bps.
+static int g_opt = 0;			// G3RUH modem regardless of speed.
+static int h_opt = 0;			// Hexadecimal display of received packet.
+
+
 int main (int argc, char *argv[])
 {
 
@@ -276,6 +286,7 @@ int main (int argc, char *argv[])
 	  //my_audio_config.achan[channel].passall = 1;				
 	}
 
+
 	while (1) {
           //int this_option_optind = optind ? optind : 1;
           int option_index = 0;
@@ -288,7 +299,7 @@ int main (int argc, char *argv[])
 
 	  /* ':' following option character means arg is required. */
 
-          c = getopt_long(argc, argv, "B:P:D:F:L:G:012",
+          c = getopt_long(argc, argv, "B:P:D:U:gF:L:G:012h",
                         long_options, &option_index);
           if (c == -1)
             break;
@@ -296,63 +307,13 @@ int main (int argc, char *argv[])
           switch (c) {
 
             case 'B':				/* -B for data Bit rate */
-						/*    300 implies 1600/1800 AFSK. */
-						/*    1200 implies 1200/2200 AFSK. */
-						/*    2400 implies V.26 */
-						/*    9600 implies scrambled. */
+						/* Also implies modem type based on speed. */
+	      B_opt = atoi(optarg);
+              break;
 
-              my_audio_config.achan[0].baud = atoi(optarg);
+            case 'g':				/* -G Force G3RUH regardless of speed. */
 
-              dw_printf ("Data rate set to %d bits / second.\n", my_audio_config.achan[0].baud);
-
-              if (my_audio_config.achan[0].baud < MIN_BAUD || my_audio_config.achan[0].baud > MAX_BAUD) {
-		text_color_set(DW_COLOR_ERROR);
-                dw_printf ("Use a more reasonable bit rate in range of %d - %d.\n", MIN_BAUD, MAX_BAUD);
-                exit (EXIT_FAILURE);
-              }
-
-	      /* We have similar logic in direwolf.c, config.c, gen_packets.c, and atest.c, */
-	      /* that need to be kept in sync.  Maybe it could be a common function someday. */
-
-	      if (my_audio_config.achan[0].baud == 100) {
-                my_audio_config.achan[0].modem_type = MODEM_AFSK;
-                my_audio_config.achan[0].mark_freq = 1615;
-                my_audio_config.achan[0].space_freq = 1785;
-	        strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
-	      }
-	      else if (my_audio_config.achan[0].baud < 600) {
-                my_audio_config.achan[0].modem_type = MODEM_AFSK;
-                my_audio_config.achan[0].mark_freq = 1600;
-                my_audio_config.achan[0].space_freq = 1800;
-	        strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
-	      }
-	      else if (my_audio_config.achan[0].baud < 1800) {
-                my_audio_config.achan[0].modem_type = MODEM_AFSK;
-                my_audio_config.achan[0].mark_freq = DEFAULT_MARK_FREQ;
-                my_audio_config.achan[0].space_freq = DEFAULT_SPACE_FREQ;
-		// Should default to E+ or something similar later.
-	      }
-	      else if (my_audio_config.achan[0].baud < 3600) {
-                my_audio_config.achan[0].modem_type = MODEM_QPSK;
-                my_audio_config.achan[0].mark_freq = 0;
-                my_audio_config.achan[0].space_freq = 0;
-	        strlcpy (my_audio_config.achan[0].profiles, "", sizeof(my_audio_config.achan[0].profiles));
-                dw_printf ("Using V.26 QPSK rather than AFSK.\n");
-	      }
-	      else if (my_audio_config.achan[0].baud < 7200) {
-                my_audio_config.achan[0].modem_type = MODEM_8PSK;
-                my_audio_config.achan[0].mark_freq = 0;
-                my_audio_config.achan[0].space_freq = 0;
-	        strlcpy (my_audio_config.achan[0].profiles, "", sizeof(my_audio_config.achan[0].profiles));
-                dw_printf ("Using V.27 8PSK rather than AFSK.\n");
-	      }
-	      else {
-                my_audio_config.achan[0].modem_type = MODEM_SCRAMBLE;
-                my_audio_config.achan[0].mark_freq = 0;
-                my_audio_config.achan[0].space_freq = 0;
-	        strlcpy (my_audio_config.achan[0].profiles, " ", sizeof(my_audio_config.achan[0].profiles));	// avoid getting default later.
-                dw_printf ("Using scrambled baseband signal rather than AFSK.\n");
-	      }
+	      g_opt = 1;
               break;
 
 	    case 'P':				/* -P for modem profile. */
@@ -375,6 +336,23 @@ int main (int argc, char *argv[])
 	      my_audio_config.achan[0].decimate = decimate;
 	      break;	
 
+	    case 'U':				/* -U upsample for G3RUH to improve performance */
+						/* when the sample rate to baud ratio is low. */
+						/* Actually it is set automatically and this will */
+						/* override the normal calculation. */
+
+	      upsample = atoi(optarg);
+
+	      dw_printf ("Multiply audio sample rate by %d\n", upsample);
+	      if (upsample < 1 || upsample > 4) {
+		text_color_set(DW_COLOR_ERROR);
+		dw_printf ("Unreasonable value for -U.\n");
+		exit (EXIT_FAILURE);
+	      }
+	      dw_printf ("Multiply audio sample rate by %d\n", upsample);
+	      my_audio_config.achan[0].upsample = upsample;
+	      break;
+
 	    case 'F':				/* -D set "fix bits" level. */
 
 	      my_audio_config.achan[0].fix_bits = atoi(optarg);
@@ -411,6 +389,11 @@ int main (int argc, char *argv[])
 	       decode_only = 2;
 	       break;
 
+	     case 'h':				/* Hexadecimal display. */
+
+	       h_opt = 1;
+	       break;
+
              case '?':
 
               /* Unknown option message was already printed. */
@@ -426,6 +409,75 @@ int main (int argc, char *argv[])
 	    }
         }
     
+/*
+ * Set modem type based on data rate.
+ * (Could be overridden by -g later.)
+ */
+	/*    300 implies 1600/1800 AFSK. */
+	/*    1200 implies 1200/2200 AFSK. */
+	/*    2400 implies V.26 QPSK. */
+	/*    4800 implies V.27 8PSK. */
+	/*    9600 implies G3RUH baseband scrambled. */
+
+        my_audio_config.achan[0].baud = B_opt;
+
+        if (my_audio_config.achan[0].baud < MIN_BAUD || my_audio_config.achan[0].baud > MAX_BAUD) {
+	  text_color_set(DW_COLOR_ERROR);
+          dw_printf ("Use a more reasonable bit rate in range of %d - %d.\n", MIN_BAUD, MAX_BAUD);
+          exit (EXIT_FAILURE);
+        }
+
+	/* We have similar logic in direwolf.c, config.c, gen_packets.c, and atest.c, */
+	/* that need to be kept in sync.  Maybe it could be a common function someday. */
+
+	if (my_audio_config.achan[0].baud == 100) {
+	  my_audio_config.achan[0].modem_type = MODEM_AFSK;
+	  my_audio_config.achan[0].mark_freq = 1615;
+	  my_audio_config.achan[0].space_freq = 1785;
+	  strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
+	}
+	else if (my_audio_config.achan[0].baud < 600) {
+	  my_audio_config.achan[0].modem_type = MODEM_AFSK;
+	  my_audio_config.achan[0].mark_freq = 1600;
+	  my_audio_config.achan[0].space_freq = 1800;
+	  strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
+	}
+	else if (my_audio_config.achan[0].baud < 1800) {
+	  my_audio_config.achan[0].modem_type = MODEM_AFSK;
+	  my_audio_config.achan[0].mark_freq = DEFAULT_MARK_FREQ;
+	  my_audio_config.achan[0].space_freq = DEFAULT_SPACE_FREQ;
+	// Should default to E+ or something similar later.
+	}
+	else if (my_audio_config.achan[0].baud < 3600) {
+	  my_audio_config.achan[0].modem_type = MODEM_QPSK;
+	  my_audio_config.achan[0].mark_freq = 0;
+	  my_audio_config.achan[0].space_freq = 0;
+	  strlcpy (my_audio_config.achan[0].profiles, "", sizeof(my_audio_config.achan[0].profiles));
+	}
+	else if (my_audio_config.achan[0].baud < 7200) {
+	  my_audio_config.achan[0].modem_type = MODEM_8PSK;
+	  my_audio_config.achan[0].mark_freq = 0;
+	  my_audio_config.achan[0].space_freq = 0;
+	  strlcpy (my_audio_config.achan[0].profiles, "", sizeof(my_audio_config.achan[0].profiles));
+	}
+	else {
+	  my_audio_config.achan[0].modem_type = MODEM_SCRAMBLE;
+	  my_audio_config.achan[0].mark_freq = 0;
+	  my_audio_config.achan[0].space_freq = 0;
+	  strlcpy (my_audio_config.achan[0].profiles, " ", sizeof(my_audio_config.achan[0].profiles));	// avoid getting default later.
+	}
+
+/*
+ * -g option means force g3RUH regardless of speed.
+ */
+
+	if (g_opt) {
+          my_audio_config.achan[0].modem_type = MODEM_SCRAMBLE;
+          my_audio_config.achan[0].mark_freq = 0;
+          my_audio_config.achan[0].space_freq = 0;
+	  strlcpy (my_audio_config.achan[0].profiles, " ", sizeof(my_audio_config.achan[0].profiles));	// avoid getting default later.
+	}
+
 	memcpy (&my_audio_config.achan[1], &my_audio_config.achan[0], sizeof(my_audio_config.achan[0]));
 
 
@@ -763,6 +815,21 @@ void dlq_rec_frame (int chan, int subchan, int slice, packet_t pp, alevel_t alev
 	ax25_safe_print ((char *)pinfo, info_len, 0);
 	dw_printf ("\n");
 
+/*
+ * -h option for hexadecimal display.  (new in 1.6)
+ */
+
+	if (h_opt) {
+
+	  text_color_set(DW_COLOR_DEBUG);
+	  dw_printf ("------\n");
+	  ax25_hex_dump (pp);
+	  dw_printf ("------\n");
+	}
+
+
+
+
 #if 1		// temp experiment  	TODO: remove this.
 
 #include "decode_aprs.h"
@@ -814,8 +881,12 @@ static void usage (void) {
 	dw_printf ("               1200 (default) baud uses 1200/2200 Hz AFSK.\n");
 	dw_printf ("               9600 baud uses K9NG/G2RUH standard.\n");
 	dw_printf ("\n");
+	dw_printf ("        -g     Force G3RUH modem rather rather than default for data rate.\n");
+	dw_printf ("\n");
 	dw_printf ("        -D n   Divide audio sample rate by n.\n");
 	dw_printf ("\n");
+	dw_printf ("        -h     Print frame contents as hexadecimal bytes.\n");
+	dw_printf ("\n");
 	dw_printf ("        -F n   Amount of effort to try fixing frames with an invalid CRC.  \n");
 	dw_printf ("               0 (default) = consider only correct frames.  \n");
 	dw_printf ("               1 = Try to fix only a single bit.  \n");
diff --git a/audio.h b/audio.h
index 9ff86c5..7bdf374 100644
--- a/audio.h
+++ b/audio.h
@@ -138,6 +138,8 @@ struct audio_s {
 	    int decimate;		/* Reduce AFSK sample rate by this factor to */
 					/* decrease computational requirements. */
 
+	    int upsample;		/* Upsample by this factor for G3RUH. */
+
 	    int interleave;		/* If > 1, interleave samples among multiple decoders. */
 					/* Quick hack for experiment. */
 
diff --git a/ax25_pad.c b/ax25_pad.c
index 23e061b..e584d0b 100644
--- a/ax25_pad.c
+++ b/ax25_pad.c
@@ -2308,16 +2308,30 @@ void ax25_hex_dump (packet_t this_p)
 	  dw_printf ("%s\n", cp_text);
 	}
 
+	// Address fields must be only upper case letters and digits.
+	// If less than 6 characters, trailing positions are filled with ASCII space.
+	// Using all zero bits in one of these 6 positions is wrong.
+	// Any non printable characters will be printed as "." here.
 
-	dw_printf (" dest    %c%c%c%c%c%c %2d c/r=%d res=%d last=%d\n", 
-				fptr[0]>>1, fptr[1]>>1, fptr[2]>>1, fptr[3]>>1, fptr[4]>>1, fptr[5]>>1,
+	dw_printf (" dest    %c%c%c%c%c%c %2d c/r=%d res=%d last=%d\n",
+				isprint(fptr[0]>>1) ? fptr[0]>>1 : '.',
+				isprint(fptr[1]>>1) ? fptr[1]>>1 : '.',
+				isprint(fptr[2]>>1) ? fptr[2]>>1 : '.',
+				isprint(fptr[3]>>1) ? fptr[3]>>1 : '.',
+				isprint(fptr[4]>>1) ? fptr[4]>>1 : '.',
+				isprint(fptr[5]>>1) ? fptr[5]>>1 : '.',
 				(fptr[6]&SSID_SSID_MASK)>>SSID_SSID_SHIFT,
 				(fptr[6]&SSID_H_MASK)>>SSID_H_SHIFT, 
 				(fptr[6]&SSID_RR_MASK)>>SSID_RR_SHIFT,
 				fptr[6]&SSID_LAST_MASK);
 
 	dw_printf (" source  %c%c%c%c%c%c %2d c/r=%d res=%d last=%d\n", 
-				fptr[7]>>1, fptr[8]>>1, fptr[9]>>1, fptr[10]>>1, fptr[11]>>1, fptr[12]>>1,
+				isprint(fptr[7]>>1) ? fptr[7]>>1 : '.',
+				isprint(fptr[8]>>1) ? fptr[8]>>1 : '.',
+				isprint(fptr[9]>>1) ? fptr[9]>>1 : '.',
+				isprint(fptr[10]>>1) ? fptr[10]>>1 : '.',
+				isprint(fptr[11]>>1) ? fptr[11]>>1 : '.',
+				isprint(fptr[12]>>1) ? fptr[12]>>1 : '.',
 				(fptr[13]&SSID_SSID_MASK)>>SSID_SSID_SHIFT,
 				(fptr[13]&SSID_H_MASK)>>SSID_H_SHIFT, 
 				(fptr[13]&SSID_RR_MASK)>>SSID_RR_SHIFT,
@@ -2327,7 +2341,12 @@ void ax25_hex_dump (packet_t this_p)
 
 	  dw_printf (" digi %d  %c%c%c%c%c%c %2d   h=%d res=%d last=%d\n", 
 				n - 1,
-				fptr[n*7+0]>>1, fptr[n*7+1]>>1, fptr[n*7+2]>>1, fptr[n*7+3]>>1, fptr[n*7+4]>>1, fptr[n*7+5]>>1,
+				isprint(fptr[n*7+0]>>1) ? fptr[n*7+0]>>1 : '.',
+				isprint(fptr[n*7+1]>>1) ? fptr[n*7+1]>>1 : '.',
+				isprint(fptr[n*7+2]>>1) ? fptr[n*7+2]>>1 : '.',
+				isprint(fptr[n*7+3]>>1) ? fptr[n*7+3]>>1 : '.',
+				isprint(fptr[n*7+4]>>1) ? fptr[n*7+4]>>1 : '.',
+				isprint(fptr[n*7+5]>>1) ? fptr[n*7+5]>>1 : '.',
 				(fptr[n*7+6]&SSID_SSID_MASK)>>SSID_SSID_SHIFT,
 				(fptr[n*7+6]&SSID_H_MASK)>>SSID_H_SHIFT, 
 				(fptr[n*7+6]&SSID_RR_MASK)>>SSID_RR_SHIFT,
diff --git a/config.c b/config.c
index b5dab4e..12013da 100644
--- a/config.c
+++ b/config.c
@@ -1261,7 +1261,10 @@ void config_init (char *fname, struct audio_s *p_audio_config,
  * Options:
  *	mark:space	- AFSK tones.  Defaults based on speed.
  *	num@offset	- Multiple decoders on different frequencies.
- *	
+ *	/9		- Divide sample rate by specified number.
+ *	*9		- Upsample ratio for G3RUH.
+ *	[A-Z+-]+	- Letters, plus, minus for the demodulator "profile."
+ *	g3ruh		- This modem type regardless of default for speed.
  */
 
 	  else if (strcasecmp(t, "MODEM") == 0) {
@@ -1320,7 +1323,7 @@ void config_init (char *fname, struct audio_s *p_audio_config,
               p_audio_config->achan[channel].space_freq = 0;
 	    }
 
-	    /* Get mark frequency. */
+	    /* Get any options. */
 
 	    t = split(NULL,0);
 	    if (t == NULL) {
@@ -1332,6 +1335,9 @@ void config_init (char *fname, struct audio_s *p_audio_config,
 
 /* old style */
 
+	      text_color_set(DW_COLOR_ERROR);
+	      dw_printf ("Line %d: Old style (pre version 1.2) format will no longer be supported in next version.\n", line);
+
 	      n = atoi(t);
 	      /* Originally the upper limit was 3000. */
 	      /* Version 1.0 increased to 5000 because someone */
@@ -1516,6 +1522,25 @@ void config_init (char *fname, struct audio_s *p_audio_config,
 		  }
 		}
 
+		else if (*t == '*') {		/* *upsample */
+		  int n = atoi(t+1);
+
+                  if (n >= 1 && n <= 4) {
+	            p_audio_config->achan[channel].upsample = n;
+		  }
+	    	  else {
+	            text_color_set(DW_COLOR_ERROR);
+                    dw_printf ("Line %d: Ignoring unreasonable upsample ratio of %d.\n", line, n);
+		  }
+		}
+
+		else if (strcasecmp(t, "G3RUH") == 0) {		/* Force G3RUH modem regardless of default for speed. New in 1.6. */
+
+                  p_audio_config->achan[channel].modem_type = MODEM_SCRAMBLE;
+                  p_audio_config->achan[channel].mark_freq = 0;
+                  p_audio_config->achan[channel].space_freq = 0;
+		}
+
 		else {
 	          text_color_set(DW_COLOR_ERROR);
                   dw_printf ("Line %d: Unrecognized option for MODEM: %s\n", line, t);
diff --git a/decode_aprs.c b/decode_aprs.c
index 3b7f6c8..7a8831c 100644
--- a/decode_aprs.c
+++ b/decode_aprs.c
@@ -219,8 +219,8 @@ void decode_aprs (decode_aprs_t *A, packet_t pp, int quiet)
 	if ( ( ! A->g_quiet ) && ( (int)strlen((char*)pinfo) != info_len) ) {
 
 	  text_color_set(DW_COLOR_ERROR);
-	  dw_printf("'nul' character found in Information part.  This should never happen.\n");
-	  dw_printf("It seems that %s is transmitting with defective software.\n", A->g_src);
+	  dw_printf("'nul' character found in Information part.  This should never happen with APRS.\n");
+	  dw_printf("If this is meant to be APRS, %s is transmitting with defective software.\n", A->g_src);
 
 	  if (strcmp((char*)pinfo, "4P") == 0) {
 	    dw_printf("The TM-D710 will do this intermittently.  A firmware upgrade is needed to fix it.\n");
diff --git a/demod.c b/demod.c
index 3953622..c945b49 100644
--- a/demod.c
+++ b/demod.c
@@ -63,7 +63,7 @@ static struct audio_s          *save_audio_config_p;
 
 // TODO: temp experiment.
 
-static int upsample = 2;	// temp experiment.
+
 static int zerostuff = 1;	// temp experiment.
 
 // Current state of all the decoders.
@@ -648,20 +648,70 @@ int demod_init (struct audio_s *pa)
 //#endif
 	      }
 
-#ifdef TUNE_UPSAMPLE
-	      upsample = TUNE_UPSAMPLE;
-#endif
 
 
 #ifdef TUNE_ZEROSTUFF
 	      zerostuff = TUNE_ZEROSTUFF;
 #endif
 
+
+/*
+ * We need a minimum number of audio samples per bit time for good performance.
+ * Easier to check here because demod_9600_init might have an adjusted sample rate.
+ */
+
+	      float ratio = (float)(save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec)
+							/ (float)(save_audio_config_p->achan[chan].baud);
+
+/*
+ * Set reasonable upsample ratio if user did not override.
+ */
+
+	      if (save_audio_config_p->achan[chan].upsample == 0) {
+
+	        if (ratio < 5) {
+
+	          // example: 44100 / 9600 is 4.59
+	          // Big improvement with x2.
+	          // x4 seems to work the best.
+	          // The other parameters are not as touchy.
+	          // Might reduce on ARM if it takes too much CPU power.
+
+	          save_audio_config_p->achan[chan].upsample = 4;
+	        }
+	        else if (ratio < 10) {
+
+	          // 48000 / 9600 is 5.00
+	          // Need more reasearch.  Treat like above for now.
+
+	          save_audio_config_p->achan[chan].upsample = 4;
+	        }
+	        else if (ratio < 15) {
+
+	          // ...
+
+	          save_audio_config_p->achan[chan].upsample = 2;
+	        }
+	        else {	// >= 15
+	          //
+	          // An example of this might be .....
+	          // Probably no benefit.
+
+	          save_audio_config_p->achan[chan].upsample = 1;
+	        }
+	      }
+
+#ifdef TUNE_UPSAMPLE
+	      save_audio_config_p->achan[chan].upsample = TUNE_UPSAMPLE;
+#endif
+
 	      text_color_set(DW_COLOR_DEBUG);
 	      dw_printf ("Channel %d: %d baud, K9NG/G3RUH, %s, %d sample rate x %d",
-		    chan, save_audio_config_p->achan[chan].baud, 
+		    chan,
+	            save_audio_config_p->achan[chan].baud,
 		    save_audio_config_p->achan[chan].profiles,
-		    save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec, upsample);
+		    save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec,
+	            save_audio_config_p->achan[chan].upsample);
 	      if (save_audio_config_p->achan[chan].dtmf_decode != DTMF_DECODE_OFF) 
 	        dw_printf (", DTMF decoder enabled");
 	      dw_printf (".\n");
@@ -669,6 +719,7 @@ int demod_init (struct audio_s *pa)
 	      struct demodulator_state_s *D;
 	      D = &demodulator_state[chan][0];	// first subchannel
 
+
 	      save_audio_config_p->achan[chan].num_subchan = 1;
               save_audio_config_p->achan[chan].num_slicers = 1;
 
@@ -681,12 +732,6 @@ int demod_init (struct audio_s *pa)
      	      }
 	        
 
-	      /* We need a minimum number of audio samples per bit time for good performance. */
-	      /* Easier to check here because demod_9600_init might have an adjusted sample rate. */
-
-	      float ratio = (float)(save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec)
-							/ (float)(save_audio_config_p->achan[chan].baud);
-
 	      text_color_set(DW_COLOR_INFO);
 	      dw_printf ("The ratio of audio samples per sec (%d) to data rate in baud (%d) is %.1f\n",
 				save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec,
@@ -698,6 +743,9 @@ int demod_init (struct audio_s *pa)
 	      }
 	      else if (ratio < 5) {
 	        dw_printf ("This is on the low side for best performance.  Can you use a higher sample rate?\n");
+	        if (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec == 44100) {
+	          dw_printf ("For example, can you use 48000 rather than 44100?\n");
+	        }
 	      }
 	      else if (ratio < 6) {
 	        dw_printf ("Increasing the sample rate should improve decoder performance.\n");
@@ -709,7 +757,7 @@ int demod_init (struct audio_s *pa)
 	        dw_printf ("This is a suitable ratio for good performance.\n");
 	      }
 
-	      demod_9600_init (upsample * save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec, save_audio_config_p->achan[chan].baud, D);
+	      demod_9600_init (save_audio_config_p->achan[chan].upsample * save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec, save_audio_config_p->achan[chan].baud, D);
 
 	      if (strchr(save_audio_config_p->achan[chan].profiles, '+') != NULL) {
 
@@ -942,17 +990,17 @@ void demod_process_sample (int chan, int subchan, int sam)
 	      /* So far, both are same in tests with different */
 	      /* optimal low pass filter parameters. */
 
-	      for (k=1; k<upsample; k++) {
+	      for (k=1; k<save_audio_config_p->achan[chan].upsample; k++) {
 	        demod_9600_process_sample (chan, 0, D);
 	      }
-	      demod_9600_process_sample (chan, sam * upsample, D);
+	      demod_9600_process_sample (chan, sam * save_audio_config_p->achan[chan].upsample, D);
 	    }
 	    else {
 
 	      /* Linear interpolation. */
 	      static int prev_sam;
 
-	      switch (upsample) {
+	      switch (save_audio_config_p->achan[chan].upsample) {
 	        case 1:
 	          demod_9600_process_sample (chan, sam, D);
 	          break;
diff --git a/demod_9600.c b/demod_9600.c
index e94f8aa..e6e97cd 100644
--- a/demod_9600.c
+++ b/demod_9600.c
@@ -1,7 +1,7 @@
 //
 //    This file is part of Dire Wolf, an amateur radio packet TNC.
 // 
-//    Copyright (C) 2011, 2012, 2013, 2015  John Langner, WB2OSZ
+//    Copyright (C) 2011, 2012, 2013, 2015, 2019  John Langner, WB2OSZ
 //
 //    This program is free software: you can redistribute it and/or modify
 //    it under the terms of the GNU General Public License as published by
@@ -141,23 +141,43 @@ void demod_9600_init (int samples_per_sec, int baud, struct demodulator_state_s
 //	  case 'K':			// upsample x3 with filtering.
 //	  case 'L':			// upsample x4 with filtering.
 
-	    D->lp_filter_len_bits =  76 * 9600.0 / (44100.0 * 2.0);
+
+	    D->lp_filter_len_bits =  1.0;
 
 	    // Works best with odd number in some tests.  Even is better in others.
 	    //D->lp_filter_size = ((int) (0.5f * ( D->lp_filter_len_bits * (float)samples_per_sec / (float)baud ))) * 2 + 1;
+
 	    D->lp_filter_size = (int) (( D->lp_filter_len_bits * (float)samples_per_sec / baud) + 0.5f);
 
-	    D->lp_window = BP_WINDOW_HAMMING;
-	    D->lpf_baud = 0.62;
+	    D->lp_window = BP_WINDOW_COSINE;
+
+	    D->lpf_baud = 1.00;
 
 	    D->agc_fast_attack = 0.080;
 	    D->agc_slow_decay =  0.00012;
 
 	    D->pll_locked_inertia = 0.89;
 	    D->pll_searching_inertia = 0.67;
+
+	    D->play_it_again_sample = 0;		// TODO: 1.6 experiment.
+							// assuming lp_filter_size > lp2_filter_size
+
+	    D->lp2_filter_size =  samples_per_sec / baud;	// samples for 1 bit
+
+
 //	    break;
 //	}
 
+#if 0
+	text_color_set(DW_COLOR_DEBUG);
+	dw_printf ("----------  %s  (%d, %d)  -----------\n", __func__, samples_per_sec, baud);
+	dw_printf ("filter_len_bits = %.2f\n", D->lp_filter_len_bits);
+	dw_printf ("lp_filter_size = %d\n", D->lp_filter_size);
+	dw_printf ("lp_window = %d\n", D->lp_window);
+	dw_printf ("lpf_baud = %.2f\n", D->lpf_baud);
+	dw_printf ("samples per bit = %.1f\n", (double)samples_per_sec / baud);
+#endif
+
         D->pll_step_per_sample = 
 		(int) round(TICKS_PER_PLL_CYCLE * (double) baud / (double)samples_per_sec);
 
@@ -194,7 +214,13 @@ void demod_9600_init (int samples_per_sec, int baud, struct demodulator_state_s
 
 	//dw_printf ("demod_9600_init: call gen_lowpass(fc=%.2f, , size=%d, )\n", fc, D->lp_filter_size);
 
-	gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window);
+	(void)gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window, 0);
+
+// Go back and resample where bit is expected.
+
+	fc = (float)baud * 1 / (float)samples_per_sec;
+
+	(void)gen_lowpass (fc, D->lp2_filter, D->lp2_filter_size, D->lp_window, 0);
 
 	/* Version 1.2: Experiment with different slicing levels. */
 
@@ -481,15 +507,14 @@ void demod_9600_process_sample (int chan, int sam, struct demodulator_state_s *D
  *
  *		Results???  TBD
  *
+ * Version 1.6:	New experiment where filter size to extract clock is not the same
+ *		as filter to extract the data bit value.
+ *
  *--------------------------------------------------------------------*/
 
 __attribute__((hot))
 inline static void nudge_pll (int chan, int subchan, int slice, float demod_out_f, struct demodulator_state_s *D)
 {
-
-/*
- */
-
 	D->slicer[slice].prev_d_c_pll = D->slicer[slice].data_clock_pll;
 
 	// Perform the add as unsigned to avoid signed overflow error.
@@ -499,7 +524,31 @@ inline static void nudge_pll (int chan, int subchan, int slice, float demod_out_
 
 	  /* Overflow.  Was large positive, wrapped around, now large negative. */
 
-	  hdlc_rec_bit (chan, subchan, slice, demod_out_f > 0, 1, D->slicer[slice].lfsr);
+
+	  if (D->play_it_again_sample) {	// New experiment in 1.6.
+
+// FIXME: double check position and draw picture.
+
+	    int offset = ( D->lp_filter_size - D->lp2_filter_size ) / 2;
+
+	    float amp = convolve (D->raw_cb + offset, D->lp2_filter, D->lp2_filter_size);
+
+	    int resampled;
+
+	    if (D->num_slicers > 1) {
+	      resampled = amp - slice_point[slice] > 0;;
+	    }
+	    else {
+	      resampled = amp > 0;
+	    }
+
+	    hdlc_rec_bit (chan, subchan, slice, resampled, 1, D->slicer[slice].lfsr);
+	  }
+	  else {
+
+// traditional
+	    hdlc_rec_bit (chan, subchan, slice, demod_out_f > 0, 1, D->slicer[slice].lfsr);
+	  }
 	}
 
 /*
diff --git a/demod_afsk.c b/demod_afsk.c
index 80bd40e..59afbfa 100644
--- a/demod_afsk.c
+++ b/demod_afsk.c
@@ -382,6 +382,45 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	    D->pll_searching_inertia = 0.64;
 	    break;
 
+	  case 'H':
+
+		/* Experiment in Version 1.6 			*/
+		/* 1200 baud - Started out as a copy of E but	*/
+		/* will probably have little tweaks after the	*/
+		/* major experiment.				*/
+		/* Enhancements: 				*/
+		/*  + Look back and sample the bit position.	*/
+		/*  + Avoid smearing by long filter and low pass. */
+
+	    D->use_prefilter = 1;		/* first, a bandpass filter. */
+	    D->prefilter_baud = 0.21;
+	    D->pre_filter_len_bits = 184 * 1200. / 44100.;
+	    D->pre_filter_len_bits = 235 * 1200. / 44100.;
+	    D->pre_window = BP_WINDOW_TRUNCATED;
+
+	    D->ms_filter_len_bits = 65 * 1200. / 44100.;	// Just over 2 bit times.
+	    D->ms_window = BP_WINDOW_COSINE;
+
+	    /* New for synchronous re-demod in 1.6. */
+	    D->play_it_again_sample = 1;
+	    D->m2_filter_len_bits = 44 * 1200. / 44100.;	// a little more than 1 bit time.
+	    D->s2_filter_len_bits = 48 * 1200. / 44100.;	// a little more than 1 bit time.
+	    D->ms2_window = BP_WINDOW_TRUNCATED;
+	    D->lp_delay_fract = 0.53;		// FIXME:  This is backwards.
+
+	    D->lpf_use_fir = 1;
+	    D->lpf_baud = 1.10;
+	    D->lp_filter_len_bits = 64 * 1200. / 44100.;
+	    D->lp_window = BP_WINDOW_TRUNCATED;
+
+	    D->agc_fast_attack = 0.820;
+	    D->agc_slow_decay = 0.000214;
+	    D->hysteresis = 0.001;
+
+	    D->pll_locked_inertia = 0.765;
+	    D->pll_searching_inertia = 0.44;
+	    break;
+
 	  default:
 
 	    text_color_set(DW_COLOR_ERROR);
@@ -395,6 +434,9 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 #ifdef TUNE_MS_WINDOW
 	D->ms_window = TUNE_MS_WINDOW;
 #endif
+#ifdef TUNE_MS2_WINDOW
+	D->ms2_window = TUNE_MS2_WINDOW;
+#endif
 #ifdef TUNE_LP_WINDOW
 	D->lp_window = TUNE_LP_WINDOW;
 #endif
@@ -417,7 +459,9 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 #ifdef TUNE_PRE_BAUD
 	D->prefilter_baud = TUNE_PRE_BAUD;
 #endif
-
+#ifdef TUNE_LP_DELAY_FRACT
+	D->lp_delay_fract = TUNE_LP_DELAY_FRACT;
+#endif
 
 /*
  * Calculate constants used for timing.
@@ -432,6 +476,8 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 
 	D->pre_filter_size = (int) round( D->pre_filter_len_bits * (float)samples_per_sec / (float)baud );
 	D->ms_filter_size = (int) round( D->ms_filter_len_bits * (float)samples_per_sec / (float)baud );
+	D->m2_filter_size = (int) round( D->m2_filter_len_bits * (float)samples_per_sec / (float)baud );
+	D->s2_filter_size = (int) round( D->s2_filter_len_bits * (float)samples_per_sec / (float)baud );
 	D->lp_filter_size = (int) round( D->lp_filter_len_bits * (float)samples_per_sec / (float)baud );
 	  	 
 /* Experiment with other sizes. */
@@ -442,6 +488,12 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 #ifdef TUNE_MS_FILTER_SIZE
 	D->ms_filter_size = TUNE_MS_FILTER_SIZE;
 #endif
+#ifdef TUNE_M2_FILTER_SIZE
+	D->m2_filter_size = TUNE_M2_FILTER_SIZE;
+#endif
+#ifdef TUNE_S2_FILTER_SIZE
+	D->s2_filter_size = TUNE_S2_FILTER_SIZE;
+#endif
 #ifdef TUNE_LP_FILTER_SIZE
 	D->lp_filter_size = TUNE_LP_FILTER_SIZE;
 #endif
@@ -450,7 +502,7 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	assert (D->ms_filter_size >= 4);
 	//assert (D->lp_filter_size >= 4);
 
-	if (D->pre_filter_size > MAX_FILTER_SIZE) 
+	if (D->pre_filter_size > MAX_FILTER_SIZE)
 	{
 	  text_color_set (DW_COLOR_ERROR);
 	  dw_printf ("Calculated filter size of %d is too large.\n", D->pre_filter_size);
@@ -460,7 +512,7 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	  exit (1);
 	}
 
-	if (D->ms_filter_size > MAX_FILTER_SIZE) 
+	if (D->ms_filter_size > MAX_FILTER_SIZE)
 	{
 	  text_color_set (DW_COLOR_ERROR);
 	  dw_printf ("Calculated filter size of %d is too large.\n", D->ms_filter_size);
@@ -470,6 +522,25 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	  exit (1);
 	}
 
+	if (D->m2_filter_size * 2 > MAX_FILTER_SIZE)
+	{
+	  text_color_set (DW_COLOR_ERROR);
+	  dw_printf ("Calculated filter size of %d is too large.\n", D->m2_filter_size);
+	  dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
+	  dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
+							MAX_FILTER_SIZE);
+	  exit (1);
+	}
+	if (D->s2_filter_size * 2 > MAX_FILTER_SIZE)
+	{
+	  text_color_set (DW_COLOR_ERROR);
+	  dw_printf ("Calculated filter size of %d is too large.\n", D->s2_filter_size);
+	  dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
+	  dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
+							MAX_FILTER_SIZE);
+	  exit (1);
+	}
+
 	if (D->lp_filter_size > MAX_FILTER_SIZE) 
 	{
 	  text_color_set (DW_COLOR_ERROR);
@@ -502,10 +573,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	  f1 = f1 / (float)samples_per_sec;
 	  f2 = f2 / (float)samples_per_sec;
 	  
-	  //gen_bandpass (f1, f2, D->pre_filter, D->pre_filter_size, BP_WINDOW_HAMMING);
-	  //gen_bandpass (f1, f2, D->pre_filter, D->pre_filter_size, BP_WINDOW_BLACKMAN);
-	  //gen_bandpass (f1, f2, D->pre_filter, D->pre_filter_size, BP_WINDOW_COSINE);
-	  //gen_bandpass (f1, f2, D->pre_filter, D->pre_filter_size, D->bp_window);
 	  gen_bandpass (f1, f2, D->pre_filter, D->pre_filter_size, D->pre_window);
 	}
 
@@ -525,46 +592,8 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	  dw_printf ("   j     shape   M sin   M cos \n");
 #endif
 
-	  float Gs = 0, Gc = 0;
-
-          for (j=0; j<D->ms_filter_size; j++) {
-	    float am;
-	    float center;
-	    float shape = 1.0f;		/* Shape is an attempt to smooth out the */
-					/* abrupt edges in hopes of reducing */
-					/* overshoot and ringing. */
-					/* My first thought was to use a cosine shape. */
-					/* Should investigate Hamming and Blackman */
-					/* windows mentioned in the literature. */
-					/* http://en.wikipedia.org/wiki/Window_function */
-
-	    center = 0.5f * (D->ms_filter_size - 1);
-	    am = ((float)(j - center) / (float)samples_per_sec) * ((float)mark_freq) * (2.0f * (float)M_PI);
-
-	    shape = window (D->ms_window, D->ms_filter_size, j);
-
-	    D->m_sin_table[j] = sinf(am) * shape;
-	    D->m_cos_table[j] = cosf(am) * shape;
-
-	    Gs += D->m_sin_table[j] * sinf(am);
-	    Gc += D->m_cos_table[j] * cosf(am);
-
-#if DEBUG1
-	    dw_printf ("%6d  %6.2f  %6.2f  %6.2f\n", j, shape, D->m_sin_table[j], D->m_cos_table[j]) ;
-#endif
-          }
-
-
-/* Normalize for unity gain */
-
-#if DEBUG1
-	  dw_printf ("Before normalizing, Gs = %.2f, Gc = %.2f\n", Gs, Gc) ;
-#endif
-          for (j=0; j<D->ms_filter_size; j++) {
-	    D->m_sin_table[j] = D->m_sin_table[j] / Gs;
-	    D->m_cos_table[j] = D->m_cos_table[j] / Gc;
-	  }
 
+	  gen_ms (mark_freq, samples_per_sec, D->m_sin_table, D->m_cos_table, D->ms_filter_size, D->ms_window);
 
 #if DEBUG1
 	  text_color_set(DW_COLOR_DEBUG);
@@ -572,39 +601,14 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	  dw_printf ("Space\n");
 	  dw_printf ("   j     shape   S sin   S cos\n");
 #endif
-	  Gs = 0;
-	  Gc = 0;
 
-          for (j=0; j<D->ms_filter_size; j++) {
-	    float as;
-	    float center;
-	    float shape = 1.0f;
+	  gen_ms (space_freq, samples_per_sec, D->s_sin_table, D->s_cos_table, D->ms_filter_size, D->ms_window);
 
-	    center = 0.5 * (D->ms_filter_size - 1);
-	    as = ((float)(j - center) / (float)samples_per_sec) * ((float)space_freq) * (2.0f * (float)M_PI);
+	  // Note that these are twice as long so we can try matching different phases.
 
-	    shape = window (D->ms_window, D->ms_filter_size, j);
-
-	    D->s_sin_table[j] = sinf(as) * shape;
-	    D->s_cos_table[j] = cosf(as) * shape;
-
-	    Gs += D->s_sin_table[j] * sinf(as);
-	    Gc += D->s_cos_table[j] * cosf(as);
-
-#if DEBUG1
-	    dw_printf ("%6d  %6.2f  %6.2f  %6.2f\n", j, shape, D->s_sin_table[j], D->s_cos_table[j] ) ;
-#endif
-          }
-
-
-/* Normalize for unity gain */
-
-#if DEBUG1
-	  dw_printf ("Before normalizing, Gs = %.2f, Gc = %.2f\n", Gs, Gc) ;
-#endif
-          for (j=0; j<D->ms_filter_size; j++) {
-	    D->s_sin_table[j] = D->s_sin_table[j] / Gs;
-	    D->s_cos_table[j] = D->s_cos_table[j] / Gc;
+	  if (D->play_it_again_sample) {
+	    gen_ms (mark_freq, samples_per_sec, D->m2_sin_table, D->m2_cos_table, D->m2_filter_size * 2, D->ms2_window);
+	    gen_ms (space_freq, samples_per_sec, D->s2_sin_table, D->s2_cos_table, D->s2_filter_size * 2, D->ms2_window);
 	  }
 
 /*
@@ -616,7 +620,11 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	if (D->lpf_use_fir) {
 	  float fc;
 	  fc = baud * D->lpf_baud / (float)samples_per_sec;
-	  gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window);
+	  D->lp_filter_delay = gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window, D->lp_delay_fract);
+	}
+	else {
+	  // D->lp_filter_delay = 
+	  // Only needed for looking back and I don't expect to use IIR in that case.
 	}
 
 /*
@@ -853,15 +861,22 @@ void demod_afsk_process_sample (int chan, int subchan, int sam, struct demodulat
  * Optional bandpass filter before the mark/space discriminator.
  */
 
+// FIXME:  calculate how much we really need.
+
+	int extra = 0;
+	if (D->play_it_again_sample) {
+	  extra = D->lp_filter_size;
+	}
+
 	if (D->use_prefilter) {
 	  float cleaner;
 
 	  push_sample (fsam, D->raw_cb, D->pre_filter_size);
 	  cleaner = convolve (D->raw_cb, D->pre_filter, D->pre_filter_size);
-	  push_sample (cleaner, D->ms_in_cb, D->ms_filter_size);
+	  push_sample (cleaner, D->ms_in_cb, D->ms_filter_size + extra);
 	}
 	else {
-	  push_sample (fsam, D->ms_in_cb, D->ms_filter_size);
+	  push_sample (fsam, D->ms_in_cb, D->ms_filter_size + extra);
 	}
 
 /*
@@ -978,7 +993,7 @@ void demod_afsk_process_sample (int chan, int subchan, int sam, struct demodulat
  * Here is an excellent explanation:
  * http://www.febo.com/packet/layer-one/transmit.html
  *
- * Under real conditions, we find that the higher tone has a
+ * Under real conditions, we find that the higher tone usually has a
  * considerably smaller amplitude due to the passband characteristics
  * of the transmitter and receiver.  To make matters worse, it
  * varies considerably from one station to another.
@@ -1133,9 +1148,80 @@ inline static void nudge_pll (int chan, int subchan, int slice, int demod_data,
 
 	  /* Overflow. */
 
-	  hdlc_rec_bit (chan, subchan, slice, demod_data, 0, -1);
+// In version 1.6 we will try a new experiment.
+// The tone filters are about 2 bit times wide so this smears the signal.
+// I originally expected this to be about 1 bit time but 2 turned out
+// to give the best results after extensive experimentation.
+// We are looking at half of each adjacent bit, not just the one we want.
+// The low pass filter also smears the signal.
+//
+// We will try to look back in time and re-demodulate only the time period of the bit.
+//
+
+	  if (D->play_it_again_sample) {	// New in 1.6.  Currently 'H' demod profile.
+
+// FIXME: double check position and draw picture.
+
+#if 0
+	    // This provided a slight benefit.
+
+	    int offset = ( D->ms_filter_size - D->m2_filter_size ) / 2 + D->lp_filter_delay;
+
+	    float m_sum1 = convolve (D->ms_in_cb + offset, D->m2_sin_table, D->m2_filter_size);
+	    float m_sum2 = convolve (D->ms_in_cb + offset, D->m2_cos_table, D->m2_filter_size);
+	    float m_amp = sqrtf(m_sum1 * m_sum1 + m_sum2 * m_sum2);
+
+	    offset = ( D->ms_filter_size - D->s2_filter_size ) / 2 + D->lp_filter_delay;
+
+	    float s_sum1 = convolve (D->ms_in_cb + offset, D->s2_sin_table, D->s2_filter_size);
+	    float s_sum2 = convolve (D->ms_in_cb + offset, D->s2_cos_table, D->s2_filter_size);
+	    float s_amp = sqrtf(s_sum1 * s_sum1 + s_sum2 * s_sum2);
+#else
+	    // Here we try something completely new.
+	    // Rather than taking the vector magnitude of the I+Q components,
+	    // correlate it with only a sine wave.  We don't know the phase so
+	    // we will try matching with a bunch of different phases and take the best match.
+
+	    // Trying match with cosine as well could be beneficial for lower sample rates.
+
+	    int j;
+	    float m_amp = 0;
+	    float s_amp = 0;
+
+	    int offset = ( D->ms_filter_size - D->m2_filter_size ) / 2 + D->lp_filter_delay;
+
+	    for (j = 0; j <= D->m2_filter_size; j++) {
+	      float match = fabsf(convolve (D->ms_in_cb + offset, D->m2_sin_table + j, D->m2_filter_size));
+	      if (match > m_amp) m_amp = match;
+	    }
+
+	    offset = ( D->ms_filter_size - D->s2_filter_size ) / 2 + D->lp_filter_delay;
+
+	    for (j = 0; j <= D->s2_filter_size; j++) {
+	      float match = fabsf(convolve (D->ms_in_cb + offset, D->s2_sin_table + j, D->s2_filter_size));
+	      if (match > s_amp) s_amp = match;
+	    }
+#endif
+
+	    int resampled;
+	    if (D->num_slicers > 1) {
+	      resampled = m_amp > s_amp * space_gain[slice];
+	    }
+	    else {
+	      resampled = m_amp > s_amp;
+	    }
+
+	    hdlc_rec_bit (chan, subchan, slice, resampled, 0, -1);
+	  }
+	  else {
+	    // Traditional way, after the low pass filter.
+	    hdlc_rec_bit (chan, subchan, slice, demod_data, 0, -1);
+	  }
 	}
 
+	// Even if we used alternative method to extract the data bit,
+	//  we still use the low pass output for the PLL.
+
         if (demod_data != D->slicer[slice].prev_demod_data) {
 
 	  if (hdlc_rec_gathering (chan, subchan, slice)) {
diff --git a/demod_psk.c b/demod_psk.c
index ee660ef..3b2a487 100644
--- a/demod_psk.c
+++ b/demod_psk.c
@@ -510,7 +510,7 @@ void demod_psk_init (enum modem_t modem_type, int samples_per_sec, int bps, char
  */
 
 	float fc = correct_baud * D->lpf_baud / (float)samples_per_sec;
-	gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window);
+	(void)gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window, 0);
 
 /*
  * No point in having multiple numbers for signal level.
diff --git a/direwolf.c b/direwolf.c
index d920957..f02f624 100644
--- a/direwolf.c
+++ b/direwolf.c
@@ -190,7 +190,7 @@ int main (int argc, char *argv[])
 	struct digi_config_s digi_config;
 	struct cdigi_config_s cdigi_config;
 	struct igate_config_s igate_config;
-	int r_opt = 0, n_opt = 0, b_opt = 0, B_opt = 0, D_opt = 0;	/* Command line options. */
+	int r_opt = 0, n_opt = 0, b_opt = 0, B_opt = 0, D_opt = 0, U_opt = 0;	/* Command line options. */
 	char P_opt[16];
 	char l_opt_logdir[80];
 	char L_opt_logfile[80];
@@ -199,6 +199,7 @@ int main (int argc, char *argv[])
 	
 	int t_opt = 1;		/* Text color option. */				
 	int a_opt = 0;		/* "-a n" interval, in seconds, for audio statistics report.  0 for none. */
+	int g_opt = 0;		/* G3RUH mode, ignoring default for speed. */				
 
 	int d_k_opt = 0;	/* "-d k" option for serial port KISS.  Can be repeated for more detail. */					
 	int d_n_opt = 0;	/* "-d n" option for Network KISS.  Can be repeated for more detail. */	
@@ -263,7 +264,7 @@ int main (int argc, char *argv[])
 	text_color_init(t_opt);
 	text_color_set(DW_COLOR_INFO);
 	//dw_printf ("Dire Wolf version %d.%d (%s) Beta Test 4\n", MAJOR_VERSION, MINOR_VERSION, __DATE__);
-	dw_printf ("Dire Wolf DEVELOPMENT version %d.%d %s (%s)\n", MAJOR_VERSION, MINOR_VERSION, "A", __DATE__);
+	dw_printf ("Dire Wolf DEVELOPMENT version %d.%d %s (%s)\n", MAJOR_VERSION, MINOR_VERSION, "B", __DATE__);
 	//dw_printf ("Dire Wolf version %d.%d\n", MAJOR_VERSION, MINOR_VERSION);
 
 
@@ -352,7 +353,7 @@ int main (int argc, char *argv[])
 
 	  /* ':' following option character means arg is required. */
 
-          c = getopt_long(argc, argv, "P:B:D:c:pxr:b:n:d:q:t:Ul:L:Sa:E:T:",
+          c = getopt_long(argc, argv, "P:B:gD:U:c:pxr:b:n:d:q:t:ul:L:Sa:E:T:",
                         long_options, &option_index);
           if (c == -1)
             break;
@@ -404,13 +405,18 @@ int main (int argc, char *argv[])
             }
             break;
 
+          case 'g':				/* -g G3RUH modem, overriding default mode for speed. */
+	 
+	    g_opt = 1;
+            break;
+
 	  case 'P':				/* -P for modem profile. */
 
 	    //debug: dw_printf ("Demodulator profile set to \"%s\"\n", optarg);
 	    strlcpy (P_opt, optarg, sizeof(P_opt)); 
 	    break;	
 
-          case 'D':				/* -D decrease AFSK demodulator sample rate */
+          case 'D':				/* -D divide AFSK demodulator sample rate */
 	 
 	    D_opt = atoi(optarg);
             if (D_opt < 1 || D_opt > 8) {
@@ -420,6 +426,16 @@ int main (int argc, char *argv[])
             }
             break;
 
+          case 'U':				/* -U multiply G3RUH demodulator sample rate (upsample) */
+	 
+	    U_opt = atoi(optarg);
+            if (U_opt < 1 || U_opt > 4) {
+	      text_color_set(DW_COLOR_ERROR);
+              dw_printf ("Crazy value for -U. \n");
+              exit (EXIT_FAILURE);
+            }
+            break;
+
           case 'x':				/* -x for transmit calibration tones. */
 
 	    xmit_calibrate_option = 1;
@@ -518,7 +534,7 @@ int main (int argc, char *argv[])
 	    break;
 
 
-	  case 'U':				/* Print UTF-8 test and exit. */
+	  case 'u':				/* Print UTF-8 test and exit. */
 
 	    dw_printf ("\n  UTF-8 test string: ma%c%cana %c%c F%c%c%c%ce\n\n", 
 			0xc3, 0xb1,
@@ -610,15 +626,18 @@ int main (int argc, char *argv[])
 	if (r_opt != 0) {
 	  audio_config.adev[0].samples_per_sec = r_opt;
 	}
+
 	if (n_opt != 0) {
 	  audio_config.adev[0].num_channels = n_opt;
 	  if (n_opt == 2) {
 	    audio_config.achan[1].valid = 1;
 	  }
 	}
+
 	if (b_opt != 0) {
 	  audio_config.adev[0].bits_per_sample = b_opt;
 	}
+
 	if (B_opt != 0) {
 	  audio_config.achan[0].baud = B_opt;
 
@@ -661,6 +680,16 @@ int main (int argc, char *argv[])
 	  }
 	}
 
+	if (g_opt) {
+
+	  // Force G3RUH mode, overriding default for speed.
+	  //   Example:   -B 2400 -g  
+
+	  audio_config.achan[0].modem_type = MODEM_SCRAMBLE;
+          audio_config.achan[0].mark_freq = 0;
+          audio_config.achan[0].space_freq = 0;
+	}
+
 	audio_config.statistics_interval = a_opt;
 
 	if (strlen(P_opt) > 0) { 
@@ -674,6 +703,13 @@ int main (int argc, char *argv[])
 	    audio_config.achan[0].decimate = D_opt;
 	}
 
+	if (U_opt != 0) {
+	    // Increase G3RUH audio sampling rate to improve performance.
+	    // The value is normally determined automatically based on audio
+	    // sample rate and baud.  This allows override for experimentation.
+	    audio_config.achan[0].upsample = U_opt;
+	}
+
 	strlcpy(audio_config.timestamp_format, T_opt_timestamp, sizeof(audio_config.timestamp_format));
 
 	// temp - only xmit errors.
@@ -1145,9 +1181,10 @@ void app_process_rec_packet (int chan, int subchan, int slice, packet_t pp, alev
  * If it came from DTMF decoder, send it to APRStt gateway.
  * Otherwise, it is a candidate for IGate and digipeater.
  *
- * TODO: It might be useful to have some way to simulate touch tone
- * sequences with BEACON sendto=R... for testing.
+ * TODO: It would be useful to have some way to simulate touch tone
+ * sequences with BEACON sendto=R0 for testing.
  */
+
 	if (subchan == -1) {
 	  if (tt_config.gateway_enabled && info_len >= 2) {
 	    aprs_tt_sequence (chan, (char*)(pinfo+1));
@@ -1254,6 +1291,7 @@ static void usage (char **argv)
 	dw_printf ("                     2400 bps uses QPSK based on V.26 standard.\n");
 	dw_printf ("                     4800 bps uses 8PSK based on V.27 standard.\n");
 	dw_printf ("                     9600 bps and up uses K9NG/G3RUH standard.\n");
+	dw_printf ("    -g             Force G3RUH modem regardless of speed.\n");
 	dw_printf ("    -D n           Divide audio sample rate by n for channel 0.\n");
 	dw_printf ("    -d             Debug options:\n");
 	dw_printf ("       a             a = AGWPE network protocol client.\n");
@@ -1281,7 +1319,7 @@ static void usage (char **argv)
 	dw_printf ("    -p             Enable pseudo terminal for KISS protocol.\n");
 #endif
 	dw_printf ("    -x             Send Xmit level calibration tones.\n");
-	dw_printf ("    -U             Print UTF-8 test string and exit.\n");
+	dw_printf ("    -u             Print UTF-8 test string and exit.\n");
 	dw_printf ("    -S             Print symbol tables and exit.\n");
 	dw_printf ("    -T fmt         Time stamp format for sent and received frames.\n");
 	dw_printf ("\n");
diff --git a/dsp.c b/dsp.c
index fa16305..6ba7094 100644
--- a/dsp.c
+++ b/dsp.c
@@ -1,7 +1,7 @@
 //
 //    This file is part of Dire Wolf, an amateur radio packet TNC.
 //
-//    Copyright (C) 2011, 2012, 2013, 2015  John Langner, WB2OSZ
+//    Copyright (C) 2011, 2012, 2013, 2015, 2019  John Langner, WB2OSZ
 //
 //    This program is free software: you can redistribute it and/or modify
 //    it under the terms of the GNU General Public License as published by
@@ -51,7 +51,7 @@
 
 // Don't remove this.  It serves as a reminder that an experiment is underway.
 
-#if defined(TUNE_MS_FILTER_SIZE) || defined(TUNE_AGC_FAST) || defined(TUNE_LPF_BAUD) || defined(TUNE_PLL_LOCKED) || defined(TUNE_PROFILE)
+#if defined(TUNE_MS_FILTER_SIZE) || defined(TUNE_MS2_FILTER_SIZE) || defined(TUNE_AGC_FAST) || defined(TUNE_LPF_BAUD) || defined(TUNE_PLL_LOCKED) || defined(TUNE_PROFILE)
 #define DEBUG1 1		// Don't remove this.
 #endif
 
@@ -118,13 +118,16 @@ float window (bp_window_t type, int size, int j)
  * Inputs:   	fc		- Cutoff frequency as fraction of sampling frequency.
  *		filter_size	- Number of filter taps.
  *		wtype		- Window type, BP_WINDOW_HAMMING, etc.
+ *		lp_delay_fract	- Fudge factor for the delay value.
  *
  * Outputs:     lp_filter
  *		
+ * Returns:	Signal delay thru the filter in number of audio samples.
+ *
  *----------------------------------------------------------------*/
 
  
-void gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype)
+int gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype, float lp_delay_fract)
 {
 	int j;
 	float G;
@@ -171,14 +174,69 @@ void gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype
         for (j=0; j<filter_size; j++) {
 	  lp_filter[j] = lp_filter[j] / G;
 	}
-}
+
+
+// Calculate the signal delay.
+// If a signal at level 0 steps to level 1, this is the time that it would
+// take for the output to reach 0.5.
+//
+// Examples:
+//
+// Filter has one tap with value of 1.0.
+//	Output is immediate so I would call this delay of 0.
+//
+// Filter coefficients:	0.2, 0.2, 0.2, 0.2, 0.2
+//	"1" inputs	Out
+//	1		0.2
+//	2		0.4
+//	3		0.6
+//
+// In this case, the output does not change immediately.
+// It takes two more samples to reach the half way point
+// so it has a delay of 2.
+
+	float sum = 0;
+	int delay = 0;
+
+	if (lp_delay_fract == 0) lp_delay_fract = 0.5;
+
+        for (j=0; j<filter_size; j++) {
+	  sum += lp_filter[j];
+#if DEBUG1
+	dw_printf ("lp_filter[%d] = %.3f   sum = %.3f   lp_delay_fract = %.3f\n", j, lp_filter[j], sum, lp_delay_fract);
+#endif
+	  if (sum > lp_delay_fract) {
+	    delay = j;
+	    break;
+	  }
+	}
+
+#if DEBUG1
+	  dw_printf ("Low Pass Delay = %d samples\n", delay) ;
+#endif
+
+// Hmmm.  This might have been wasted effort.  The result is always half the number of taps.
+
+	if (delay < 2 || delay > filter_size - 2) {
+	  text_color_set(DW_COLOR_ERROR);
+	  dw_printf ("Internal error, %s %d, delay %d for size %d\n", __func__, __LINE__, delay, filter_size);
+	}
+
+	return (delay);
+
+}  /* end gen_lowpass */
+
+
+#undef DEBUG1
+
 
 
 /*------------------------------------------------------------------
  *
  * Name:        gen_bandpass
  *
- * Purpose:     Generate band pass filter kernel.
+ * Purpose:     Generate band pass filter kernel for the prefilter.
+ *		This is NOT for the mark/space filters.
  *
  * Inputs:   	f1		- Lower cutoff frequency as fraction of sampling frequency.
  *		f2		- Upper cutoff frequency...
@@ -187,10 +245,10 @@ void gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype
  *
  * Outputs:     bp_filter
  *
- * Reference:	http://www.labbookpages.co.uk/audio/firWindowing.html	
+ * Reference:	http://www.labbookpages.co.uk/audio/firWindowing.html
  *
  *		Does it need to be an odd length?
- *	
+ *
  *----------------------------------------------------------------*/
 
 
@@ -201,7 +259,6 @@ void gen_bandpass (float f1, float f2, float *bp_filter, int filter_size, bp_win
 	float G;
 	float center = 0.5 * (filter_size - 1);
 
-
 #if DEBUG1
 	text_color_set(DW_COLOR_DEBUG);
 
@@ -229,7 +286,8 @@ void gen_bandpass (float f1, float f2, float *bp_filter, int filter_size, bp_win
 #if DEBUG1
 	  dw_printf ("%6d  %6.2f  %6.3f  %6.3f\n", j, shape, sinc, bp_filter[j] ) ;
 #endif
-        }
+	}
+
 
 /*
  * Normalize bandpass for unity gain in middle of passband.
@@ -249,6 +307,66 @@ void gen_bandpass (float f1, float f2, float *bp_filter, int filter_size, bp_win
         for (j=0; j<filter_size; j++) {
 	  bp_filter[j] = bp_filter[j] / G;
 	}
-}
 
-/* end dsp.c */
\ No newline at end of file
+} /* end gen_bandpass */
+
+
+
+/*------------------------------------------------------------------
+ *
+ * Name:        gen_ms
+ *
+ * Purpose:     Generate mark and space filters.
+ *
+ * Inputs:   	fc		- Tone frequency, i.e. mark or space.
+ *		sps		- Samples per second.
+ *		filter_size	- Number of filter taps.
+ *		wtype		- Window type, BP_WINDOW_HAMMING, etc.
+ *
+ * Outputs:     bp_filter
+ *
+ * Reference:	http://www.labbookpages.co.uk/audio/firWindowing.html
+ *
+ *		Does it need to be an odd length?
+ *
+ *----------------------------------------------------------------*/
+
+
+void gen_ms (int fc, int sps, float *sin_table, float *cos_table, int filter_size, int wtype)
+{
+	int j;
+	float Gs = 0, Gc = 0;;
+
+        for (j=0; j<filter_size; j++) {
+
+	  float center = 0.5f * (filter_size - 1);
+	  float am = ((float)(j - center) / (float)sps) * ((float)fc) * (2.0f * (float)M_PI);
+
+	  float shape = window (wtype, filter_size, j);
+
+	  sin_table[j] = sinf(am) * shape;
+	  cos_table[j] = cosf(am) * shape;
+
+	  Gs += sin_table[j] * sinf(am);
+	  Gc += cos_table[j] * cosf(am);
+
+#if DEBUG1
+	  dw_printf ("%6d  %6.2f  %6.2f  %6.2f\n", j, shape, sin_table[j], cos_table[j]) ;
+#endif
+        }
+
+
+/* Normalize for unity gain */
+
+#if DEBUG1
+	dw_printf ("Before normalizing, Gs = %.2f, Gc = %.2f\n", Gs, Gc) ;
+#endif
+        for (j=0; j<filter_size; j++) {
+	  sin_table[j] = sin_table[j] / Gs;
+	  cos_table[j] = cos_table[j] / Gc;
+	}
+
+} /* end gen_ms */
+
+
+/* end dsp.c */
diff --git a/dsp.h b/dsp.h
index 1f5aaa5..5d5b882 100644
--- a/dsp.h
+++ b/dsp.h
@@ -5,6 +5,9 @@
 
 float window (bp_window_t type, int size, int j);
 
-void gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype);
+int gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype, float lp_delay_fract);
+
+void gen_bandpass (float f1, float f2, float *bp_filter, int filter_size, bp_window_t wtype);
+
+void gen_ms (int fc, int samples_per_sec, float *sin_table, float *cos_table, int filter_size, int wtype);
 
-void gen_bandpass (float f1, float f2, float *bp_filter, int filter_size, bp_window_t wtype);
\ No newline at end of file
diff --git a/fsk_demod_state.h b/fsk_demod_state.h
index a37b24d..cb791dd 100644
--- a/fsk_demod_state.h
+++ b/fsk_demod_state.h
@@ -31,6 +31,8 @@ struct demodulator_state_s
 	char profile;			// 'A', 'B', etc.	Upper case.
 					// Only needed to see if we are using 'F' to take fast path.
 
+	int play_it_again_sample;	// Enable new synchronous demod in version 1.6.
+
 #define TICKS_PER_PLL_CYCLE ( 256.0 * 256.0 * 256.0 * 256.0 )
 
 	int pll_step_per_sample;	// PLL is advanced by this much each audio sample.
@@ -39,9 +41,17 @@ struct demodulator_state_s
 
 	int ms_filter_size;		/* Size of mark & space filters, in audio samples. */
 					/* Started off as a guess of one bit length */
-					/* but somewhat longer turned out to be better. */
+					/* but about 2 bit times turned out to be better. */
 					/* Currently using same size for any prefilter. */
 
+	int m2_filter_size;
+	int s2_filter_size;		/* Size of mark & space filters, in audio samples */
+					/* for the synchronous demodulator.  I'm expecting */
+					/* smaller, perhaps just over 1 bit time here. */
+
+	int lp2_filter_size;		/* FSK resampling - Size of Low Pass filter, in audio samples. */
+
+
 #define MAX_FILTER_SIZE 320		/* 304 is needed for profile C, 300 baud & 44100. */
 
 /*
@@ -49,6 +59,9 @@ struct demodulator_state_s
  * e.g.  1 means 1/1200 second for 1200 baud.
  */
 	float ms_filter_len_bits;
+	float m2_filter_len_bits;
+	float s2_filter_len_bits;
+	float lp_delay_fract;
 
 /* 
  * Window type for the various filters.
@@ -57,6 +70,7 @@ struct demodulator_state_s
 	bp_window_t pre_window;
 	bp_window_t ms_window;
 	bp_window_t lp_window;
+	bp_window_t ms2_window;		/* New in 1.6. */
 
 
 /*
@@ -80,6 +94,12 @@ struct demodulator_state_s
 					/* Previously it was always the same as the M/S */
 					/* filters but in version 1.2 it's now independent. */
 
+	int lp_filter_delay;		/* Number of samples that the low pass filter */
+					/* delays the signal. */
+	
+					/* New in 1.6. */
+
+
 /*
  * Automatic gain control.  Fast attack and slow decay factors.
  */
@@ -135,6 +155,18 @@ struct demodulator_state_s
 	float s_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
 	float s_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
 
+/*
+ * Same for the synchronous re-demodulator.
+ */
+
+	float m2_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+	float m2_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+
+	float s2_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+	float s2_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+
+	float lp2_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+
 /*
  * These are for PSK only.
  * They are number of delay line taps into previous symbol.
diff --git a/gen_packets.c b/gen_packets.c
index 62133a9..1f4543e 100644
--- a/gen_packets.c
+++ b/gen_packets.c
@@ -250,7 +250,9 @@ int main(int argc, char **argv)
 						/*    9600 implies scrambled. */
 
 						/* If you want something else, specify -B first */
-						/* then anything to override these defaults. */
+						/* then anything to override these defaults with -m, -s, or -g. */
+
+						// FIXME: options should not be order dependent.
 
               modem.achan[0].baud = atoi(optarg);
               text_color_set(DW_COLOR_INFO); 
@@ -308,6 +310,8 @@ int main(int argc, char **argv)
 
             case 'g':				/* -g for g3ruh scrambling */
 
+	      // FIXME:  order dependent.  -g must come after -B.
+
               modem.achan[0].modem_type = MODEM_SCRAMBLE;
               text_color_set(DW_COLOR_INFO); 
               dw_printf ("Using scrambled baseband signal rather than AFSK.\n");
diff --git a/gen_tone.c b/gen_tone.c
index 0d7c255..26120f1 100644
--- a/gen_tone.c
+++ b/gen_tone.c
@@ -1,10 +1,7 @@
-//#define DEBUG 1
-//#define DEBUG2 1
-
 //
 //    This file is part of Dire Wolf, an amateur radio packet TNC.
 //
-//    Copyright (C) 2011, 2014, 2015, 2016  John Langner, WB2OSZ
+//    Copyright (C) 2011, 2014, 2015, 2016, 2019  John Langner, WB2OSZ
 //
 //    This program is free software: you can redistribute it and/or modify
 //    it under the terms of the GNU General Public License as published by
@@ -106,59 +103,9 @@ static int bit_count[MAX_CHANS];	// Counter incremented for each bit transmitted
 static int save_bit[MAX_CHANS];
 
 
-/*
- * The K9NG/G3RUH output originally took a very simple and lazy approach.
- * We simply generated a square wave with + or - the desired amplitude.
- * This has a couple undesirable properties.
- *
- *	- Transmitting a square wave would splatter into adjacent
- *	   channels of the transmitter doesn't limit the bandwidth.
- *
- *	- The usual sample rate of 44100 is not a multiple of the 
- *	   baud rate so jitter would be added to the zero crossings.
- *
- * Starting in version 1.2, we try to overcome these issues by using
- * a higher sample rate, low pass filtering, and down sampling.
- *
- * What sort of low pass filter would be appropriate?  Intuitively,
- * we would expect a cutoff frequency somewhere between baud/2 and baud.
- * The current values were found with a small amount of trial and 
- * error for best results.  Future improvement is certainly possible.
- */
-
-/* 
- * For low pass filtering of 9600 baud data. 
- */
-
-/* Add sample to buffer and shift the rest down. */
-// TODO:  Can we have one copy of these in dsp.h?
-
-static inline void push_sample (float val, float *buff, int size)
-{
-	memmove(buff+1,buff,(size-1)*sizeof(float));
-	buff[0] = val; 
-}
+static int prev_dat[MAX_CHANS];		// Previous data bit.  Used for G3RUH style.
 
 
-/* FIR filter kernel. */
-
-static inline float convolve (const float *data, const float *filter, int filter_size)
-{
-	  float sum = 0;
-	  int j;
-
-	  for (j=0; j<filter_size; j++) {
-	    sum += filter[j] * data[j];
-	  }
-	  return (sum);
-}
-
-static int lp_filter_size[MAX_CHANS];
-static float raw[MAX_CHANS][MAX_FILTER_SIZE] __attribute__((aligned(16)));
-static float lp_filter[MAX_CHANS][MAX_FILTER_SIZE] __attribute__((aligned(16)));
-static int resample[MAX_CHANS];
-
-#define UPSAMPLE 2
 
 
 /*------------------------------------------------------------------
@@ -180,6 +127,8 @@ static int resample[MAX_CHANS];
  *
  *		amp		- Signal amplitude on scale of 0 .. 100.
  *
+ *				  100% uses the full 16 bit sample range of +-32k.
+ *
  *		gen_packets	- True if being called from "gen_packets" utility
  *				  rather than the "direwolf" application.
  *
@@ -211,9 +160,7 @@ int gen_tone_init (struct audio_s *audio_config_p, int amp, int gen_packets)
 
 	save_audio_config_p = audio_config_p;
 	
-
-	amp16bit = (32767 * amp) / 100;
-
+	amp16bit = (int)((32767 * amp) / 100);
 
 	for (chan = 0; chan < MAX_CHANS; chan++) {
 
@@ -266,7 +213,15 @@ int gen_tone_init (struct audio_s *audio_config_p, int amp, int gen_packets)
 	        f2_change_per_sample[chan] = f1_change_per_sample[chan];	// Not used.
 	        break;
 
-	      default:
+	      case MODEM_BASEBAND:
+	      case MODEM_SCRAMBLE:
+
+		// Tone is half baud.
+	        ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / (double)audio_config_p->achan[chan].baud ) + 0.5);
+	        f1_change_per_sample[chan] = (int) (((double)audio_config_p->achan[chan].baud * 0.5 * TICKS_PER_CYCLE / (double)audio_config_p->adev[a].samples_per_sec ) + 0.5);
+	        break;
+
+	      default:		// AFSK
 
 	        ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / (double)audio_config_p->achan[chan].baud ) + 0.5);
 	        f1_change_per_sample[chan] = (int) (((double)audio_config_p->achan[chan].mark_freq * TICKS_PER_CYCLE / (double)audio_config_p->adev[a].samples_per_sec ) + 0.5);
@@ -298,75 +253,6 @@ int gen_tone_init (struct audio_s *audio_config_p, int amp, int gen_packets)
 	  sine_table[j] = s;
         }
 
-
-/*
- * Low pass filter for 9600 baud. 
- */
-
-	for (chan = 0; chan < MAX_CHANS; chan++) {
-
-	  if (audio_config_p->achan[chan].valid && 
-		(audio_config_p->achan[chan].modem_type == MODEM_SCRAMBLE 
-		  ||  audio_config_p->achan[chan].modem_type == MODEM_BASEBAND)) {
-
-	    int a = ACHAN2ADEV(chan);
-	    int samples_per_sec;		/* Might be scaled up! */
-	    int baud;
-
-	    /* These numbers were by trial and error.  Need more investigation here. */
-
-	    float filter_len_bits =  88 * 9600.0 / (44100.0 * 2.0);
-						/* Filter length in number of data bits. */
-						/* Currently 9.58 */
-	
-	    float lpf_baud = 0.8;		/* Lowpass cutoff freq as fraction of baud rate */
-
-	    float fc;				/* Cutoff frequency as fraction of sampling frequency. */
-
-/*
- * Normally, we want to generate the same thing whether sending over the air
- * or putting it into a file for other testing.
- * (There is an important exception.  gen_packets can introduce random noise.)
- * In this case, we want more aggressive low pass filtering so it looks more like
- * what we see coming out of a receiver.
- * Specifically, single bits of the same state have considerably reduced amplitude
- * below several same values in a row.
- */
-
-	    if (gen_packets) {
-	      filter_len_bits = 4;
-	      lpf_baud = 0.55;		/* Lowpass cutoff freq as fraction of baud rate */
-	    }
-
-	    samples_per_sec = audio_config_p->adev[a].samples_per_sec * UPSAMPLE;		
-	    baud = audio_config_p->achan[chan].baud;
-
-	    ticks_per_sample[chan] = (int) ((TICKS_PER_CYCLE / (double)samples_per_sec ) + 0.5);
-	    ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / (double)baud ) + 0.5);
-
-	    lp_filter_size[chan] = (int) (( filter_len_bits * (float)samples_per_sec / baud) + 0.5);
-
-	    if (lp_filter_size[chan] < 10) {
-	      text_color_set(DW_COLOR_DEBUG);
-	      dw_printf ("gen_tone_init: unexpected, chan %d, lp_filter_size %d < 10\n", chan, lp_filter_size[chan]);
-	      lp_filter_size[chan] = 10;
-	    }
-	    else if (lp_filter_size[chan] > MAX_FILTER_SIZE) {
-	      text_color_set(DW_COLOR_DEBUG);
-	      dw_printf ("gen_tone_init: unexpected, chan %d, lp_filter_size %d > %d\n", chan, lp_filter_size[chan], MAX_FILTER_SIZE);
-	      lp_filter_size[chan] = MAX_FILTER_SIZE;
-	    }
-
-	    fc = (float)baud * lpf_baud / (float)samples_per_sec;
-
-	    //text_color_set(DW_COLOR_DEBUG);
-	    //dw_printf ("gen_tone_init: chan %d, call gen_lowpass(fc=%.2f, , size=%d, )\n", chan, fc, lp_filter_size[chan]);
-
-	    gen_lowpass (fc, lp_filter[chan], lp_filter_size[chan], BP_WINDOW_HAMMING);
-
-	  }
-	}
-
 	return (0);
 
  } /* end gen_tone_init */
@@ -389,6 +275,11 @@ int gen_tone_init (struct audio_s *audio_config_p, int amp, int gen_packets)
  *
  * Version 1.4:	Attempt to implement 2400 and 4800 bps PSK modes.
  *
+ * Version 1.6: For G3RUH, rather than generating square wave and low
+ *		pass filtering, generate the waveform directly.
+ *		This avoids overshoot, ringing, and adding more jitter.
+ *		Alternating bits come out has sine wave of baud/2 Hz.
+ *
  *--------------------------------------------------------------------*/
 
 static const int gray2phase_v26[4] = {0, 1, 3, 2};
@@ -469,7 +360,6 @@ void tone_gen_put_bit (int chan, int dat)
 	do {		/* until enough audio samples for this symbol. */
 
 	  int sam;
-	  float fsam;
 
 	  switch (save_audio_config_p->achan[chan].modem_type) {
 
@@ -499,23 +389,17 @@ void tone_gen_put_bit (int chan, int dat)
 	    case MODEM_BASEBAND:
 	    case MODEM_SCRAMBLE:
 
-#if DEBUG2
-	      text_color_set(DW_COLOR_DEBUG);
-	      dw_printf ("tone_gen_put_bit %d SCR\n", __LINE__);
-#endif
-	      fsam = dat ? amp16bit : (-amp16bit);
-
-	      /* version 1.2 - added a low pass filter instead of square wave out. */
-
-	      push_sample (fsam, raw[chan], lp_filter_size[chan]);
-
-	      resample[chan]++;
-	      if (resample[chan] >= UPSAMPLE) {
-
-	        sam = (int) convolve (raw[chan], lp_filter[chan], lp_filter_size[chan]);
-	        resample[chan] = 0;
-	        gen_tone_put_sample (chan, a, sam);
+	      if (dat != prev_dat[chan]) {
+	        tone_phase[chan] += f1_change_per_sample[chan];
 	      }
+	      else {
+	        if (tone_phase[chan] & 0x80000000)
+	          tone_phase[chan] = 0xc0000000;	// 270 degrees.
+	        else
+	          tone_phase[chan] = 0x40000000;	// 90 degrees.
+	      }
+              sam = sine_table[(tone_phase[chan] >> 24) & 0xff];
+	      gen_tone_put_sample (chan, a, sam);
 	      break;
 
 	    default:
@@ -532,7 +416,10 @@ void tone_gen_put_bit (int chan, int dat)
         } while (bit_len_acc[chan] < ticks_per_bit[chan]);
 
 	bit_len_acc[chan] -= ticks_per_bit[chan];
-}
+
+	prev_dat[chan] = dat;		// Only needed for G3RUH baseband/scrambled.
+
+}  /* end tone_gen_put_bit */
 
 
 void gen_tone_put_sample (int chan, int a, int sam) {
@@ -547,10 +434,21 @@ void gen_tone_put_sample (int chan, int a, int sam) {
 
 	assert (save_audio_config_p->adev[a].bits_per_sample == 16 || save_audio_config_p->adev[a].bits_per_sample == 8);
 
-	// TODO: Should print message telling user to reduce output level.
+	// Bad news if we are clipping and distorting the signal.
+	// We are using the full range.
+	// Too late to change now because everyone would need to recalibrate their
+	// transmit audio level.
 
-	if (sam < -32767) sam = -32767;
-	else if (sam > 32767) sam = 32767;
+	if (sam < -32767) {
+	  text_color_set(DW_COLOR_ERROR);
+	  dw_printf ("Warning: Audio sample %d clipped to -32767.\n", sam);
+	  sam = -32767;
+	}
+	else if (sam > 32767) {
+	  text_color_set(DW_COLOR_ERROR);
+	  dw_printf ("Warning: Audio sample %d clipped to +32767.\n", sam);
+	  sam = 32767;
+	}
 
 	if (save_audio_config_p->adev[a].num_channels == 1) {