Add EAS to gen_packets.

man/gen_packets.1

@@ -46,6 +46,8 @@ Data rate in bits/sec for first channel.  Standard values are 300, 1200, 2400, 4
 4800 bps uses 8PSK based on V.27 standard.
 .P
 9600 bps and up uses K9NG/G3RUH standard.
+.P
+EAS for Emergency Alert System (EAS) Specific Area Message Encoding (SAME).
 .RE
 .RE
 .PD

src/gen_packets.c

@@ -1,7 +1,7 @@
 //
 //    This file is part of Dire Wolf, an amateur radio packet TNC.
 //
-//    Copyright (C) 2011, 2013, 2014, 2015, 2016, 2019, 2021  John Langner, WB2OSZ
+//    Copyright (C) 2011, 2013, 2014, 2015, 2016, 2019, 2021, 2023  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
@@ -118,13 +118,48 @@ static void send_packet (char *str)
     	packet_t pp;
     	unsigned char fbuf[AX25_MAX_PACKET_LEN+2];
     	int flen;
-	int c;
+	int c = 0;	// channel number.
 
 	if (g_morse_wpm > 0) {
 
-	  // TODO: Why not use the destination field instead of command line option?
+	  // Why not use the destination field instead of command line option?
+	  // For one thing, this is not in TNC-2 monitor format.
 
-	  morse_send (0, str, g_morse_wpm, 100, 100);
+	  morse_send (c, str, g_morse_wpm, 100, 100);
+	}
+	else if (modem.achan[0].modem_type == MODEM_EAS) {
+
+// Generate EAS SAME signal FOR RESEARCH AND TESTING ONLY!!!
+// There could be legal consequences for sending unauhorized SAME
+// over the radio so don't do it!
+
+	  // I'm expecting to see TNC 2 monitor format.
+	  // The source and destination are ignored.
+	  // The optional destination SSID is the number of times to repeat.
+	  // The user defined data type indicator can optionally be used
+	  // for compatibility with how it is received and presented to client apps.
+	  // Examples:
+	  //	X>X-3:{DEZCZC-WXR-RWT-033019-033017-033015-033013-033011-025011-025017-033007-033005-033003-033001-025009-025027-033009+0015-1691525-KGYX/NWS-
+	  //	X>X:NNNN
+
+	  pp = ax25_from_text (str, 1);
+	  if (pp == NULL) {
+            text_color_set(DW_COLOR_ERROR);
+            dw_printf ("\"%s\" is not valid TNC2 monitoring format.\n", str);
+	    return;
+	  }
+	  unsigned char *pinfo;
+	  int info_len = ax25_get_info (pp, &pinfo);
+	  if (info_len >= 3 && strncmp((char*)pinfo, "{DE", 3) == 0) {
+	    pinfo += 3;
+	    info_len -= 3;
+	  }
+
+	  int repeat = ax25_get_ssid (pp, AX25_DESTINATION);
+	  if (repeat == 0) repeat = 1;
+
+	  eas_send (c, pinfo, repeat, 500, 500);
+	  ax25_delete (pp);
 	}
 	else {
 	  pp = ax25_from_text (str, 1);
@@ -135,6 +170,9 @@ static void send_packet (char *str)
 	  }
 	  flen = ax25_pack (pp, fbuf);
 	  (void)flen;
+
+	  // If stereo, put same thing in each channel.
+
 	  for (c=0; c<modem.adev[0].num_channels; c++)
 	  {
 
@@ -282,23 +320,31 @@ int main(int argc, char **argv)
 
 						// FIXME: options should not be order dependent.
 
+              if (strcasecmp(optarg, "EAS") == 0) {
+	        modem.achan[0].baud = 0xEA5EA5;	// See special case below.
+	      }
+	      else {
 	        modem.achan[0].baud = atoi(optarg);
+	      }
+
               text_color_set(DW_COLOR_INFO); 
               dw_printf ("Data rate set to %d bits / second.\n", modem.achan[0].baud);
-              if (modem.achan[0].baud != 100 && (modem.achan[0].baud < MIN_BAUD || modem.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 (modem.achan[0].baud == 100) {
+	      if (modem.achan[0].baud == 100) {			// What was this for?
                   modem.achan[0].modem_type = MODEM_AFSK;
                   modem.achan[0].mark_freq = 1615;
                   modem.achan[0].space_freq = 1785;
 	      }
+	      else if (modem.achan[0].baud == 0xEA5EA5) {
+		  modem.achan[0].baud = 521;			// Fine tuned later. 520.83333
+								// Proper fix is to make this float.
+                  modem.achan[0].modem_type = MODEM_EAS;
+                  modem.achan[0].mark_freq = 2083.3333;		// Ideally these should be floating point.
+                  modem.achan[0].space_freq = 1562.5000 ;
+	      }
 	      else if (modem.achan[0].baud < 600) {
                   modem.achan[0].modem_type = MODEM_AFSK;
                   modem.achan[0].mark_freq = 1600;		// Typical for HF SSB
@@ -334,6 +380,11 @@ int main(int argc, char **argv)
                   text_color_set(DW_COLOR_INFO); 
                   dw_printf ("Using scrambled baseband signal rather than AFSK.\n");
 	      }
+              if (modem.achan[0].baud != 100 && (modem.achan[0].baud < MIN_BAUD || modem.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);
+              }
               break;
 
             case 'g':				/* -g for g3ruh scrambling */
@@ -740,15 +791,24 @@ int main(int argc, char **argv)
 	}
 	else {
 
+	  // This should send a total of 6.
+	  // Note that sticking in the user defined type {DE is optional.
+
+	  if (modem.achan[0].modem_type == MODEM_EAS) {
+	    send_packet ("X>X-3:{DEZCZC-WXR-RWT-033019-033017-033015-033013-033011-025011-025017-033007-033005-033003-033001-025009-025027-033009+0015-1691525-KGYX/NWS-");
+	    send_packet ("X>X-2:{DENNNN");
+	    send_packet ("X>X:NNNN");
+	  }
+	  else {
 /*
  * Builtin default 4 packets.
  */
-
 	    send_packet ("WB2OSZ-15>TEST:,The quick brown fox jumps over the lazy dog!  1 of 4");
 	    send_packet ("WB2OSZ-15>TEST:,The quick brown fox jumps over the lazy dog!  2 of 4");
 	    send_packet ("WB2OSZ-15>TEST:,The quick brown fox jumps over the lazy dog!  3 of 4");
 	    send_packet ("WB2OSZ-15>TEST:,The quick brown fox jumps over the lazy dog!  4 of 4");
 	  }
+	}
 
 	audio_file_close();
 
@@ -765,7 +825,7 @@ static void usage (char **argv)
 	dw_printf ("Options:\n");
 	dw_printf ("  -a <number>   Signal amplitude in range of 0 - 200%%.  Default 50.\n");
 	dw_printf ("  -b <number>   Bits / second for data.  Default is %d.\n", DEFAULT_BAUD);
-	dw_printf ("  -B <number>   Bits / second for data.  Proper modem selected for 300, 1200, 2400, 4800, 9600.\n");
+	dw_printf ("  -B <number>   Bits / second for data.  Proper modem selected for 300, 1200, 2400, 4800, 9600, EAS.\n");
 	dw_printf ("  -g            Scrambled baseband rather than AFSK.\n");
 	dw_printf ("  -j            2400 bps QPSK compatible with direwolf <= 1.5.\n");
 	dw_printf ("  -J            2400 bps QPSK compatible with MFJ-2400.\n");
@@ -788,6 +848,7 @@ static void usage (char **argv)
 	dw_printf ("the default built-in message. The format should correspond to\n");
 	dw_printf ("the standard packet monitoring representation such as,\n\n");
 	dw_printf ("    WB2OSZ-1>APDW12,WIDE2-2:!4237.14NS07120.83W#\n");
+	dw_printf ("User defined content can't be used with -n option.\n");
 	dw_printf ("\n");
 	dw_printf ("Example:  gen_packets -o x.wav \n");
 	dw_printf ("\n");

src/gen_tone.c

@@ -1,7 +1,7 @@
 //
 //    This file is part of Dire Wolf, an amateur radio packet TNC.
 //
-//    Copyright (C) 2011, 2014, 2015, 2016, 2019  John Langner, WB2OSZ
+//    Copyright (C) 2011, 2014, 2015, 2016, 2019, 2023  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
@@ -70,6 +70,7 @@ static int ticks_per_sample[MAX_CHANS];	/* Same for both channels of same soundc
 static int ticks_per_bit[MAX_CHANS];
 static int f1_change_per_sample[MAX_CHANS];
 static int f2_change_per_sample[MAX_CHANS];
+static float samples_per_symbol[MAX_CHANS];
 
 
 static short sine_table[256];
@@ -198,8 +199,11 @@ int gen_tone_init (struct audio_s *audio_config_p, int amp, int gen_packets)
 	        ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / ((double)audio_config_p->achan[chan].baud * 0.5)) + 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);
 	        f2_change_per_sample[chan] = f1_change_per_sample[chan];	// Not used.
+	        samples_per_symbol[chan] = 2. * (float)audio_config_p->adev[a].samples_per_sec / (float)audio_config_p->achan[chan].baud;
 
 	        tone_phase[chan] = PHASE_SHIFT_45;	// Just to mimic first attempt.
+							// ??? Why?  We are only concerned with the difference
+							// from one symbol to the next.
 	        break;
 
 	      case MODEM_8PSK:
@@ -211,6 +215,7 @@ int gen_tone_init (struct audio_s *audio_config_p, int amp, int gen_packets)
 	        ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / ((double)audio_config_p->achan[chan].baud / 3.)) + 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);
 	        f2_change_per_sample[chan] = f1_change_per_sample[chan];	// Not used.
+	        samples_per_symbol[chan] = 3. * (float)audio_config_p->adev[a].samples_per_sec / (float)audio_config_p->achan[chan].baud;
 	        break;
 
 	      case MODEM_BASEBAND:
@@ -220,11 +225,23 @@ int gen_tone_init (struct audio_s *audio_config_p, int amp, int gen_packets)
 		// 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);
+	        samples_per_symbol[chan] = (float)audio_config_p->adev[a].samples_per_sec / (float)audio_config_p->achan[chan].baud;
+	        break;
+
+	      case MODEM_EAS:		//  EAS.
+
+		// TODO: Proper fix would be to use float for baud, mark, space.
+
+	        ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / 520.833333333333 ) + 0.5);
+	        samples_per_symbol[chan] = (int)((audio_config_p->adev[a].samples_per_sec / 520.83333) + 0.5);
+	        f1_change_per_sample[chan] = (int) ((2083.33333333333 * TICKS_PER_CYCLE / (double)audio_config_p->adev[a].samples_per_sec ) + 0.5);
+	        f2_change_per_sample[chan] = (int) ((1562.5000000 * 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);
+	        samples_per_symbol[chan] = (float)audio_config_p->adev[a].samples_per_sec / (float)audio_config_p->achan[chan].baud;
 	        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);
 	        f2_change_per_sample[chan] = (int) (((double)audio_config_p->achan[chan].space_freq * TICKS_PER_CYCLE / (double)audio_config_p->adev[a].samples_per_sec ) + 0.5);
 	        break;
@@ -285,9 +302,64 @@ int gen_tone_init (struct audio_s *audio_config_p, int amp, int gen_packets)
  *
  *--------------------------------------------------------------------*/
 
+// Interpolate between two values.
+// My original approximation simply jumped between phases, producing a discontinuity,
+// and increasing bandwidth.
+// According to multiple sources, we should transition more gently.
+// Below see see a rough approximation of:
+//  * A step function, immediately going to new value.
+//  * Linear interpoation.
+//  * Raised cosine.  Square root of cosine is also mentioned.
+//
+//	new	      -		    /		   --
+//		      |		   /		  /
+//		      |		  /		  |
+//		      |		 /		  /
+//	old	-------		/		--
+//		step		linear		raised cosine
+//
+// Inputs are the old (previous value), new value, and a blending control
+// 0 -> take old value
+// 1 -> take new value.
+// inbetween some sort of weighted average.
+
+static inline float interpol8 (float oldv, float newv, float bc)
+{
+	// Step function.
+	//return (newv);				// 78 on 11/7
+
+	assert (bc >= 0);
+	assert (bc <= 1.1);
+
+	if (bc < 0) return (oldv);
+	if (bc > 1) return (newv);
+
+	// Linear interpolation, just for comparison.
+	//return (bc * newv + (1.0f - bc) * oldv);	// 39 on 11/7
+
+	float rc = 0.5f * (cosf(bc * M_PI - M_PI) + 1.0f);
+	float rrc = bc >= 0.5f
+				? 0.5f * (sqrtf(fabsf(cosf(bc * M_PI - M_PI))) + 1.0f)
+				: 0.5f * (-sqrtf(fabsf(cosf(bc * M_PI - M_PI))) + 1.0f);
+
+	(void)rrc;
+	return (rc * newv + (1.0f - bc) * oldv);	// 49 on 11/7
+	//return (rrc * newv + (1.0f - bc) * oldv);	// 55 on 11/7
+}
+
 static const int gray2phase_v26[4] = {0, 1, 3, 2};
 static const int gray2phase_v27[8] = {1, 0, 2, 3, 6, 7, 5, 4};
 
+// #define PSKIQ 1  // not ready for prime time yet.
+#if PSKIQ
+static int xmit_octant[MAX_CHANS];	// absolute phase in 45 degree units.
+static int xmit_prev_octant[MAX_CHANS];	// from previous symbol.
+
+// For PSK, we generate the final signal by combining fixed frequency cosine and
+// sine by the following weights.
+static const float ci[8] = { 1,	.7071,	0,	-.7071,	-1,	-.7071,	0,	.7071	};
+static const float sq[8] = { 0,	.7071,	1,	.7071,	0,	-.7071,	-1,	-.7071	};
+#endif
 
 void tone_gen_put_bit (int chan, int dat)
 {
@@ -324,14 +396,28 @@ void tone_gen_put_bit (int chan, int dat)
 
 	  // All zero bits should give us steady 1800 Hz.
 	  // All one bits should flip phase by 180 degrees each time.
+	  // For V.26B, add another 45 degrees.
+	  // This seems to work a little better.
 
 	  dibit = (save_bit[chan] << 1) | dat;
 
-	  symbol = gray2phase_v26[dibit];
+	  symbol = gray2phase_v26[dibit];	// 0 .. 3 for QPSK.
+#if PSKIQ
+	  // One phase shift unit is 45 degrees.
+	  // Remember what it was last time and calculate new.
+	  // values 0 .. 7.
+	  xmit_prev_octant[chan] = xmit_octant[chan];
+	  xmit_octant[chan] += symbol * 2;
+	  if (save_audio_config_p->achan[chan].v26_alternative == V26_B) {
+	    xmit_octant[chan] += 1;
+	  }
+	  xmit_octant[chan] &= 0x7;
+#else
 	  tone_phase[chan] += symbol * PHASE_SHIFT_90;
 	  if (save_audio_config_p->achan[chan].v26_alternative == V26_B) {
 	    tone_phase[chan] += PHASE_SHIFT_45;
 	  }
+#endif
 	  bit_count[chan]++;
 	}
 
@@ -370,7 +456,9 @@ void tone_gen_put_bit (int chan, int dat)
 	  lfsr[chan] = (lfsr[chan] << 1) | (x & 1);
 	  dat = x;
 	}
-	  
+#if PSKIQ
+	int blend = 1;
+#endif
 	do {		/* until enough audio samples for this symbol. */
 
 	  int sam;
@@ -395,9 +483,58 @@ void tone_gen_put_bit (int chan, int dat)
 	      gen_tone_put_sample (chan, a, sam);
 	      break;
 
-	    case MODEM_QPSK:
+	    case MODEM_EAS:
-	    case MODEM_8PSK:
 
+	      tone_phase[chan] += dat ? f1_change_per_sample[chan] : f2_change_per_sample[chan];
+              sam = sine_table[(tone_phase[chan] >> 24) & 0xff];
+	      gen_tone_put_sample (chan, a, sam);
+	      break;
+
+	    case MODEM_QPSK:
+
+#if DEBUG2
+	      text_color_set(DW_COLOR_DEBUG);
+	      dw_printf ("tone_gen_put_bit %d PSK\n", __LINE__);
+#endif
+	      tone_phase[chan] += f1_change_per_sample[chan];
+#if PSKIQ
+#if 1  // blend JWL
+	      // remove loop invariant
+	      float old_i = ci[xmit_prev_octant[chan]];
+	      float old_q = sq[xmit_prev_octant[chan]];
+
+	      float new_i = ci[xmit_octant[chan]];
+	      float new_q = sq[xmit_octant[chan]];
+
+	      float b = blend / samples_per_symbol[chan];	// roughly 0 to 1
+	      blend++;
+	     // b = (b - 0.5) * 20 + 0.5;
+	     // if (b < 0) b = 0;
+	     // if (b > 1) b = 1;
+		// b = b > 0.5;
+		//b = 1;		// 78 decoded with this.
+					// only 39 without.
+
+
+	      //float blended_i = new_i * b + old_i * (1.0f - b);
+	      //float blended_q = new_q * b + old_q * (1.0f - b);
+
+	      float blended_i = interpol8 (old_i, new_i, b);
+	      float blended_q = interpol8 (old_q, new_q, b);
+
+	      sam = blended_i * sine_table[((tone_phase[chan] - PHASE_SHIFT_90) >> 24) & 0xff] +
+	            blended_q * sine_table[(tone_phase[chan] >> 24) & 0xff];
+#else  // jump
+	      sam = ci[xmit_octant[chan]] * sine_table[((tone_phase[chan] - PHASE_SHIFT_90) >> 24) & 0xff] +
+	            sq[xmit_octant[chan]] * sine_table[(tone_phase[chan] >> 24) & 0xff];
+#endif
+#else
+              sam = sine_table[(tone_phase[chan] >> 24) & 0xff];
+#endif
+	      gen_tone_put_sample (chan, a, sam);
+	      break;
+
+	    case MODEM_8PSK:
 #if DEBUG2
 	      text_color_set(DW_COLOR_DEBUG);
 	      dw_printf ("tone_gen_put_bit %d PSK\n", __LINE__);
@@ -521,6 +658,20 @@ void gen_tone_put_sample (int chan, int a, int sam) {
 	}
 }
 
+void gen_tone_put_quiet_ms (int chan, int time_ms) {
+
+	int a = ACHAN2ADEV(chan);	/* device for channel. */
+	int sam = 0;
+
+	int nsamples = (int) ((time_ms * (float)save_audio_config_p->adev[a].samples_per_sec / 1000.) + 0.5);
+
+	for (int j=0; j<nsamples; j++)  {
+	  gen_tone_put_sample (chan, a, sam);
+        };
+
+	// Avoid abrupt change when it starts up again.
+	tone_phase[chan] = 0;
+}
 
 
 /*-------------------------------------------------------------------

src/gen_tone.h

@@ -15,3 +15,5 @@ int gen_tone_init (struct audio_s *pp, int amp, int gen_packets);
 void tone_gen_put_bit (int chan, int dat);
 
 void gen_tone_put_sample (int chan, int a, int sam);
+
+void gen_tone_put_quiet_ms (int chan, int time_ms);

src/hdlc_send.c

@@ -295,4 +295,82 @@ static void send_bit_nrzi (int chan, int b)
 	number_of_bits_sent[chan]++;
 }
 
+
+//  The rest of this is for EAS SAME.
+//  This is sort of a logical place because it serializes a frame, but not in HDLC.
+//  We have a parallel where SAME deserialization is in hdlc_rec.
+//  Maybe both should be pulled out and moved to a same.c.
+
+
+/*-------------------------------------------------------------------
+ *
+ * Name:        eas_send
+ *
+ * Purpose:    	Serialize EAS SAME for transmission.
+ *
+ * Inputs:	chan	- Radio channel number.
+ *		str	- Character string to send.
+ *		repeat	- Number of times to repeat with 1 sec quiet between.
+ *		txdelay	- Delay (ms) from PTT to first preamble bit.
+ *		txtail	- Delay (ms) from last data bit to PTT off.	
+ *		
+ *
+ * Returns:	Total number of milliseconds to activate PTT.
+ *		This includes delays before the first character
+ *		and after the last to avoid chopping off part of it.
+ *
+ * Description:	xmit_thread calls this instead of the usual hdlc_send
+ *		when we have a special packet that means send EAS SAME
+ *		code.
+ *
+ *--------------------------------------------------------------------*/
+
+static inline void eas_put_byte (int chan, unsigned char b)
+{
+	for (int n=0; n<8; n++) {
+	  tone_gen_put_bit (chan, (b & 1));
+	  b >>= 1;
+	}
+}
+
+int eas_send (int chan, unsigned char *str, int repeat, int txdelay, int txtail)
+{
+	int bytes_sent = 0;
+	const int gap = 1000;
+	int gaps_sent = 0;
+
+	gen_tone_put_quiet_ms (chan, txdelay);
+
+	for (int r=0; r<repeat; r++ ) {
+	  for (int j=0; j<16; j++) {
+	    eas_put_byte (chan, 0xAB);
+	    bytes_sent++;
+	  }
+
+	  for (unsigned char *p = str; *p != '\0'; p++) {
+	    eas_put_byte (chan, *p);
+	    bytes_sent++;
+	  }
+
+	  if (r < repeat-1) {
+	    gen_tone_put_quiet_ms (chan, gap);
+	    gaps_sent++;
+	  }
+	}
+
+	gen_tone_put_quiet_ms (chan, txtail);
+
+	audio_flush(ACHAN2ADEV(chan));
+
+	int elapsed = txdelay + (int) (bytes_sent * 8 * 1.92) + (gaps_sent * gap) + txtail;
+
+// dw_printf ("DEBUG:  EAS total time = %d ms\n", elapsed);
+
+	return (elapsed);
+
+}  /* end eas_send */
+
+
+
+
 /* end hdlc_send.c */

src/hdlc_send.h

@@ -3,7 +3,10 @@
 
 // In version 1.7 an extra layer of abstraction was added here.
 // Rather than calling hdlc_send_frame, we now use another function
-// which sends AX.25, FX.25, or IL2P depending on
+// which sends AX.25, FX.25, or IL2P depending on mode.
+
+// eas_send fits here logically because it also serializes a packet.
+
 
 #include "ax25_pad.h"
 #include "audio.h"
@@ -12,6 +15,8 @@ int layer2_send_frame (int chan, packet_t pp, int bad_fcs, struct audio_s *audio
 
 int layer2_preamble_postamble (int chan, int flags, int finish, struct audio_s *audio_config_p);
 
+int eas_send (int chan, unsigned char *str, int repeat, int txdelay, int txtail);
+
 /* end hdlc_send.h */