aboutsummaryrefslogtreecommitdiffstats
path: root/src/amps/dsp.c
blob: ae71432c302815cd7781205f5b134bd9a82e520c (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
/* AMPS audio processing
 *
 * (C) 2016 by Andreas Eversberg <jolly@eversberg.eu>
 * All Rights Reserved
 *
 * This program is free software: you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation, either version 3 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program.  If not, see <http://www.gnu.org/licenses/>.
 */

/* How does FSK decoding work:
 * ---------------------------
 *
 * AMPS modulates the carrier frequency. If it is 8 kHz above, it is high level,
 * if it is 8 kHz below, it is low level.  The bits are coded using Manchester
 * code. A 1 is coded by low level, followed by a hight level. A 0 is coded by
 * a high level, followed by a low level. This will cause at least one level
 * change within each bit.  Also the level changes between equal bits, see
 * Manchester coding.  The bit rate is 10 KHz.
 *
 * In order to detect and demodulate a frame, the dotting sequnce is searched.
 * The dotting sequnece are alternate bits: 101010101...  The duration of a
 * level change within the dotting sequnene ist 100uS.  If all offsets of 8
 * level changes lay within +-50% of the expected time, the dotting sequence is
 * valid.  Now the next 12 bits will be searched for sync sequnece.  If better
 * dotting-offsets are found, the counter for searching the sync sequence is
 * reset, so the next 12 bits will be searched for sync too.  If no sync was
 * detected, the state changes to search for next dotting sequence.
 *
 * The average level change offsets of the dotting sequence is used to set the
 * window for the first bit.  When all samples for the window are received, a
 * raise in level is detected as 1, fall in level is detected as 0. This is done
 * by substracting the average sample value of the left side of the window by
 * the average sample value of the right side.  After the bit has been detected,
 * the samples for the next window will be received and detected.
 *
 * +-----+-----+-----+-----+
 * |     |     |   __|__   |
 * |     |     |  /  |  \  |
 * |     |     | /   |   \ |
 * |     |     |/    |    \|
 * +-----+-----+-----+-----+
 * |\    |    /|     |     |
 * | \   |   / |     |     |
 * |  \__|__/  |     |     |
 * |     |     |     |     |
 * +-----+-----+-----+-----+
 *       End   Half  Begin
 *
 * The Rx window is depiced above. In this example there is a raising edge.
 * The window is analyzed in backward direction. The average level between
 * 'Half' position and 'Begin' position is calculated, also the average level
 * between 'End' position and 'Half' position. Because the right (second)
 * side of the average level is higher than the left (first) side, a raising
 * edge is detected.
 *
 * Tests showed that comparing half of the regions of the window will cause
 * more errors than only quarter regions of the regions. Especially this is
 * true with NBFM receivers that are normally not sufficient for AMPS signals.
 *
 * As soon as a sync pattern is detected, the polarity of the pattern is used
 * to decode the following frame bits with correct polarity.  During reception
 * of the frame bits, no sync and no dotting sequnece is searched or detected.
 *
 * After reception of the bit, the bits are re-assembled, parity checked and
 * decoded. Then the process hunts for next dotting sequence.  
 */

#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>
#include <math.h>
#include "../libsample/sample.h"
#include "../common/debug.h"
#include "../common/call.h"
#include "amps.h"
#include "frame.h"
#include "dsp.h"
#include "main.h"

#define CHAN amps->sender.kanal

/* uncomment this to debug the encoding process */
//#define DEBUG_ENCODER

/* uncomment this to debug the decoding process */
//#define DEBUG_DECODER

#define PI			M_PI

#define COMPANDOR_0DB		1.0     /* A level of 0dBm (1.0) shall be unaccected */
#define AMPS_MAX_DEVIATION	8000.0
#define AMPS_MAX_MODULATION	10000.0
#define AMPS_DBM0_DEVIATION	2900.0  /* deviation of dBm0 at 1 kHz */
#define AMPS_FSK_DEVIATION	(8000.0 / AMPS_DBM0_DEVIATION)	/* no emphasis */
#define AMPS_SAT_DEVIATION	(2000.0 / AMPS_DBM0_DEVIATION)	/* no emphasis */
#define AMPS_MAX_DISPLAY	(10000.0 / AMPS_DBM0_DEVIATION)	/* no emphasis */
#define AMPS_BITRATE		10000
/* for some reason, 4000 Hz deviation works better */
#define TACS_DBM0_DEVIATION	4000.0  /* 2300 Hz deviation at 1 kHz (according to panasonic manual) */
#define TACS_MAX_DEVIATION	6400.0	/* (according to texas intruments and other sources) */
#define TACS_MAX_MODULATION	9500.0	/* (according to panasonic manual) */
#define TACS_FSK_DEVIATION	(6400.0 / TACS_DBM0_DEVIATION)	/* no emphasis */
#define TACS_SAT_DEVIATION	(1700.0 / TACS_DBM0_DEVIATION)	/* no emphasis (panasonic / TI) */
#define TACS_MAX_DISPLAY	(8000.0 / TACS_DBM0_DEVIATION)	/* no emphasis */
#define TACS_BITRATE		8000
#define SAT_DURATION		0.05	/* duration of SAT signal measurement */
#define SAT_QUALITY		0.85	/* quality needed to detect SAT signal */
#define DTX_LEVEL		0.50	/* SAT level needed to mute/unmute */
#define SIG_QUALITY		0.80	/* quality needed to detect Signaling Tone */
#define SAT_DETECT_COUNT	5	/* number of measures to detect SAT signal (specs say 250ms) */
#define SAT_LOST_COUNT		5	/* number of measures to loose SAT signal (specs say 250ms) */
#define SIG_DETECT_COUNT	6	/* number of measures to detect Signaling Tone */
#define SIG_LOST_COUNT		4	/* number of measures to loose Signaling Tone */
#define CUT_OFF_HIGHPASS	300.0   /* cut off frequency for high pass filter to remove dc level from sound card / sample */
#define BEST_QUALITY		0.68	/* Best possible RX quality */
#define COMFORT_NOISE		0.02	/* audio level of comfort noise (relative to ISDN level) */

static sample_t ramp_up[256], ramp_down[256];

static double sat_freq[4] = {
	5970.0,
	6000.0,
	6030.0,
	5800.0, /* noise level to check against */
};

static sample_t dsp_sine_sat[65536];

static uint8_t dsp_sync_check[0x800];

/* global init for FSK */
void dsp_init(void)
{
	int i;
	double s;

	PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for SAT signal.\n");
	for (i = 0; i < 65536; i++) {
		s = sin((double)i / 65536.0 * 2.0 * PI);
		dsp_sine_sat[i] = s * ((!tacs) ? AMPS_SAT_DEVIATION : TACS_SAT_DEVIATION);
	}

	/* sync checker */
	for (i = 0; i < 0x800; i++) {
		dsp_sync_check[i] = 0xff; /* no sync */
	}
	for (i = 0; i < 11; i++) {
		dsp_sync_check[0x712 ^ (1 << i)] = 0x01; /* one bit error */
		dsp_sync_check[0x0ed ^ (1 << i)] = 0x81; /* one bit error */
	}
	dsp_sync_check[0x712] = 0x00; /* no bit error */
	dsp_sync_check[0x0ed] = 0x80; /* no bit error */
}

static void dsp_init_ramp(amps_t *amps)
{
	double c;
        int i;

	PDEBUG(DDSP, DEBUG_DEBUG, "Generating smooth ramp table.\n");
	for (i = 0; i < 256; i++) {
		c = cos((double)i / 256.0 * PI);
#if 0
		if (c < 0)
			c = -sqrt(-c);
		else
			c = sqrt(c);
#endif
		ramp_down[i] = c * (double)amps->fsk_deviation;
		ramp_up[i] = -ramp_down[i];
	}
}

static void sat_reset(amps_t *amps, const char *reason);

/* Init FSK of transceiver */
int dsp_init_sender(amps_t *amps, int tolerant)
{
	sample_t *spl;
	int i;
	int rc;
	int half;

	/* attack (3ms) and recovery time (13.5ms) according to amps specs */
	init_compandor(&amps->cstate, 8000, 3.0, 13.5, COMPANDOR_0DB);

	PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for transceiver.\n");

	/* set modulation parameters */
	sender_set_fm(&amps->sender,
		(!tacs) ? AMPS_MAX_DEVIATION : TACS_MAX_DEVIATION,
		(!tacs) ? AMPS_MAX_MODULATION : TACS_MAX_MODULATION,
		(!tacs) ? AMPS_DBM0_DEVIATION : TACS_DBM0_DEVIATION,
		(!tacs) ? AMPS_MAX_DISPLAY : TACS_MAX_DISPLAY);

	if (amps->sender.samplerate < 96000) {
		PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 96000 Hz to process FSK and SAT signals.\n");
		return -EINVAL;
	}

	amps->fsk_bitduration = (double)amps->sender.samplerate / (double)((!tacs) ? AMPS_BITRATE : TACS_BITRATE);
	amps->fsk_bitstep = 1.0 / amps->fsk_bitduration;
	PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", amps->fsk_bitduration, amps->sender.samplerate);

	amps->fsk_tx_buffer_size = amps->fsk_bitduration + 10; /* 10 extra to avoid overflow due to rounding */
	spl = calloc(sizeof(*spl), amps->fsk_tx_buffer_size);
	if (!spl) {
		PDEBUG(DDSP, DEBUG_DEBUG, "No memory!\n");
		rc = -ENOMEM;
		goto error;
	}
	amps->fsk_tx_buffer = spl;

	amps->fsk_rx_window_length = ceil(amps->fsk_bitduration); /* buffer holds one bit (rounded up) */
	half = amps->fsk_rx_window_length >> 1;
	amps->fsk_rx_window_begin = half >> 1;
	amps->fsk_rx_window_half = half;
	amps->fsk_rx_window_end = amps->fsk_rx_window_length - (half >> 1);
	PDEBUG(DDSP, DEBUG_DEBUG, "Bit window length: %d\n", amps->fsk_rx_window_length);
	PDEBUG(DDSP, DEBUG_DEBUG, " -> Samples in window to analyse level left of edge: %d..%d\n", amps->fsk_rx_window_begin, amps->fsk_rx_window_half - 1);
	PDEBUG(DDSP, DEBUG_DEBUG, " -> Samples in window to analyse level right of edge: %d..%d\n", amps->fsk_rx_window_half, amps->fsk_rx_window_end - 1);
	spl = calloc(sizeof(*amps->fsk_rx_window), amps->fsk_rx_window_length);
	if (!spl) {
		PDEBUG(DDSP, DEBUG_DEBUG, "No memory!\n");
		rc = -ENOMEM;
		goto error;
	}
	amps->fsk_rx_window = spl;

	/* create devation and ramp */
	amps->fsk_deviation = (!tacs) ? AMPS_FSK_DEVIATION : TACS_FSK_DEVIATION;
	dsp_init_ramp(amps);

	/* allocate ring buffer for SAT signal detection */
	amps->sat_samples = (int)((double)amps->sender.samplerate * SAT_DURATION + 0.5);
	spl = calloc(sizeof(*spl), amps->sat_samples);
	if (!spl) {
		PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
		return -ENOMEM;
	}
	amps->sat_filter_spl = spl;

	/* count SAT tones */
	for (i = 0; i < 4; i++) {
		audio_goertzel_init(&amps->sat_goertzel[i], sat_freq[i], amps->sender.samplerate);
		if (i < 3) {
			amps->sat_phaseshift65536[i] = 65536.0 / ((double)amps->sender.samplerate / sat_freq[i]);
			PDEBUG(DDSP, DEBUG_DEBUG, "sat_phaseshift65536[%d] = %.4f\n", i, amps->sat_phaseshift65536[i]);
		}
	}
	/* signaling tone */
	audio_goertzel_init(&amps->sat_goertzel[4], (!tacs) ? 10000.0 : 8000.0, amps->sender.samplerate);
	sat_reset(amps, "Initial state");

	/* be more tolerant when syncing */
	amps->fsk_rx_sync_tolerant = tolerant;

	amps->dmp_frame_level = display_measurements_add(&amps->sender, "Frame Level", "%.1f %% (last)", DISPLAY_MEAS_LAST, DISPLAY_MEAS_LEFT, 0.0, 150.0, 100.0);
	amps->dmp_frame_quality = display_measurements_add(&amps->sender, "Frame Quality", "%.1f %% (last)", DISPLAY_MEAS_LAST, DISPLAY_MEAS_LEFT, 0.0, 100.0, 100.0);

	return 0;

error:
	dsp_cleanup_sender(amps);

	return rc;
}

/* Cleanup transceiver instance. */
void dsp_cleanup_sender(amps_t *amps)
{
	PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for treansceiver.\n");

	if (amps->fsk_tx_buffer)
		free(amps->fsk_tx_buffer);
	if (amps->fsk_rx_window)
		free(amps->fsk_rx_window);
	if (amps->sat_filter_spl) {
		free(amps->sat_filter_spl);
		amps->sat_filter_spl = NULL;
	}
#if 0
	if (amps->frame_spl) {
		free(amps->frame_spl);
		amps->frame_spl = NULL;
	}
#endif
}

static int fsk_encode(amps_t *amps, char bit)
{
	sample_t *spl;
	double phase, bitstep, deviation;
	int count;
	char last;

	deviation = amps->fsk_deviation;
	spl = amps->fsk_tx_buffer;
	phase = amps->fsk_tx_phase;
	last = amps->fsk_tx_last_bit;
	bitstep = amps->fsk_bitstep * 256.0 * 2.0; /* half bit ramp */

//printf("%d %d\n", (bit) & 1, last & 1);
	if ((bit & 1)) {
		if ((last & 1)) {
			/* last bit was 1, this bit is 1, so we ramp down first */
			do {
				*spl++ = ramp_down[(uint8_t)phase];
				phase += bitstep;
			} while (phase < 256.0);
			phase -= 256.0;
		} else {
			/* last bit was 0, this bit is 1, so we stay down first */
			do {
				*spl++ = -deviation;
				phase += bitstep;
			} while (phase < 256.0);
			phase -= 256.0;
		}
		/* ramp up */
		do {
			*spl++ = ramp_up[(uint8_t)phase];
			phase += bitstep;
		} while (phase < 256.0);
		phase -= 256.0;
	} else {
		if ((last & 1)) {
			/* last bit was 1, this bit is 0, so we stay up first */
			do {
				*spl++ = deviation;
				phase += bitstep;
			} while (phase < 256.0);
			phase -= 256.0;
		} else {
			/* last bit was 0, this bit is 0, so we ramp up first */
			do {
				*spl++ = ramp_up[(uint8_t)phase];
				phase += bitstep;
			} while (phase < 256.0);
			phase -= 256.0;
		}
		/* ramp down */
		do {
			*spl++ = ramp_down[(uint8_t)phase];
			phase += bitstep;
		} while (phase < 256.0);
		phase -= 256.0;
	}
	last = bit;
	/* depending on the number of samples, return the number */
	count = ((uintptr_t)spl - (uintptr_t)amps->fsk_tx_buffer) / sizeof(*spl);

	amps->fsk_tx_last_bit = last;
	amps->fsk_tx_phase = phase;
	amps->fsk_tx_buffer_length = count;

	return count;
}

static int fsk_frame(amps_t *amps, sample_t *samples, int length)
{
	int count = 0, len, pos, copy, i;
	sample_t *spl;
	int rc;
	char c;

	len = amps->fsk_tx_buffer_length;
	pos = amps->fsk_tx_buffer_pos;
	spl = amps->fsk_tx_buffer;

again:
	/* there must be length, otherwise we would skip blocks */
	if (count == length)
		goto done;

	/* start of new bit, so generate buffer for one bit */
	if (pos == 0) {
		c = amps->fsk_tx_frame[amps->fsk_tx_frame_pos];
		/* start new frame, so we generate one */
		if (c == '\0') {
			if (amps->dsp_mode == DSP_MODE_AUDIO_RX_FRAME_TX)
				rc = amps_encode_frame_fvc(amps, amps->fsk_tx_frame);
			else
				rc = amps_encode_frame_focc(amps, amps->fsk_tx_frame);
			/* check if we have not bit string (change to tx audio)
			 * we may not store fsk_tx_buffer_pos, because is was reset on a mode achange */
			if (rc)
				return count;
			amps->fsk_tx_frame_pos = 0;
			c = amps->fsk_tx_frame[0];
		}
		if (c == 'i')
			c = (amps->channel_busy) ? '0' : '1';
		/* invert, if polarity of the cell is negative */
		if (amps->flip_polarity)
			c ^= 1;
		len = fsk_encode(amps, c);
		amps->fsk_tx_frame_pos++;
	}

	copy = len - pos;
	if (length - count < copy)
		copy = length - count;
//printf("pos=%d length=%d copy=%d\n", pos, length, copy);
	for (i = 0; i < copy; i++) {
#ifdef DEBUG_ENCODER
		puts(debug_amplitude((double)spl[pos]));
#endif
		*samples++ = spl[pos++];
	}
	count += copy;
	if (pos == len) {
		pos = 0;
		goto again;
	}

done:
	amps->fsk_tx_buffer_length = len;
	amps->fsk_tx_buffer_pos = pos;

	return count;
}

/* send comfort noise */
static void comfort_noise(sample_t *samples, int length)
{
	int i;
	int16_t r;

	for (i = 0; i < length; i++) {
		r = random();
		samples[i] = (double)r / 32768.0 * COMFORT_NOISE;
	}
}

/* Generate audio stream with SAT signal. Keep phase for next call of function. */
static void sat_encode(amps_t *amps, sample_t *samples, int length)
{
        double phaseshift, phase;
	int i;

	phaseshift = amps->sat_phaseshift65536[amps->sat];
	phase = amps->sat_phase65536;

	for (i = 0; i < length; i++) {
		*samples++ += dsp_sine_sat[(uint16_t)phase];
		phase += phaseshift;
		if (phase >= 65536)
			phase -= 65536;
	}

	amps->sat_phase65536 = phase;
}

/* Provide stream of audio toward radio unit */
void sender_send(sender_t *sender, sample_t *samples, uint8_t *power, int length)
{
	amps_t *amps = (amps_t *) sender;
	int count;

again:
	switch (amps->dsp_mode) {
	case DSP_MODE_OFF:
		memset(power, 0, length);
		memset(samples, 0, sizeof(*samples) * length);
		break;
	case DSP_MODE_AUDIO_RX_AUDIO_TX:
		memset(power, 1, length);
		jitter_load(&amps->sender.dejitter, samples, length);
		/* pre-emphasis */
		if (amps->pre_emphasis)
			pre_emphasis(&amps->estate, samples, length);
		/* encode sat */
		sat_encode(amps, samples, length);
		break;
	case DSP_MODE_AUDIO_RX_FRAME_TX:
	case DSP_MODE_FRAME_RX_FRAME_TX:
		/* Encode frame into audio stream. If frames have
		 * stopped, process again for rest of stream. */
		count = fsk_frame(amps, samples, length);
		memset(power, 1, count);
		samples += count;
		power += count;
		length -= count;
		if (length)
			goto again;
	}
}

static void fsk_rx_bit(amps_t *amps, sample_t *spl, int len, int pos, int begin, int half, int end)
{
	int i;
	double first, second;
	int bit;
	double max = 0, min = 0;

	/* decode one bit. substact the first half from the second half.
	 * the result shows the direction of the bit change: 1 == positive.
	 */
	pos -= begin; /* possible wrap is handled below */
	second = first = 0;
	for (i = begin; i < half; i++) {
		if (--pos < 0)
			pos += len;
//printf("second %d: %d\n", pos, spl[pos]);
		second += spl[pos];
		if (i == 0 || spl[pos] > max)
			max = spl[pos];
		if (i == 0 || spl[pos] < min)
			min = spl[pos];
	}
	second /= (half - begin);
	for (i = half; i < end; i++) {
		if (--pos < 0)
			pos += len;
//printf("first %d: %d\n", pos, spl[pos]);
		first += spl[pos];
		if (spl[pos] > max)
			max = spl[pos];
		if (spl[pos] < min)
			min = spl[pos];
	}
	first /= (end - half);
//printf("first = %d second = %d\n", first, second);
	/* get bit */
	if (second > first)
		bit = 1;
	else
		bit = 0;
#ifdef DEBUG_DECODER
	if (amps->fsk_rx_sync != FSK_SYNC_POSITIVE && amps->fsk_rx_sync != FSK_SYNC_NEGATIVE)
		printf("Decoded bit as %d (dotting life = %d)\n", bit, amps->fsk_rx_dotting_life);
	else
		printf("Decoded bit as %d\n", bit);
#endif

	if (amps->fsk_rx_sync != FSK_SYNC_POSITIVE && amps->fsk_rx_sync != FSK_SYNC_NEGATIVE) {
		amps->fsk_rx_sync_register = (amps->fsk_rx_sync_register << 1) | bit;
		/* check if we received a sync */
		switch (dsp_sync_check[amps->fsk_rx_sync_register & 0x7ff]) {
		case 0x01:
			if (!amps->fsk_rx_sync_tolerant)
				break;
		case 0x00:
#ifdef DEBUG_DECODER
			printf("Sync word detected (positive)\n");
#endif
			amps->fsk_rx_sync = FSK_SYNC_POSITIVE;
prepare_frame:
			amps->fsk_rx_frame_count = 0;
			amps->fsk_rx_frame_quality = 0.0;
			amps->fsk_rx_frame_level = 0.0;
			amps->fsk_rx_sync_register = 0x555;
			amps->when_received = get_time() - (21.0 / (double)((!tacs) ? AMPS_BITRATE : TACS_BITRATE));
			return;
		case 0x81:
			if (!amps->fsk_rx_sync_tolerant)
				break;
		case 0x80:
#ifdef DEBUG_DECODER
			printf("Sync word detected (negative)\n");
#endif
			amps->fsk_rx_sync = FSK_SYNC_NEGATIVE;
			goto prepare_frame;
			return;
		}
		/* if no sync, count down the dotting life counter */
		if (--amps->fsk_rx_dotting_life == 0) {
#ifdef DEBUG_DECODER
			printf("No Sync detected after dotting\n");
#endif
			amps->fsk_rx_sync = FSK_SYNC_NONE;
			amps->channel_busy = 0;
			return;
		}
		return;
	}

	/* count level and quality */
	amps->fsk_rx_frame_level += (double)(max - min) / (double)((!tacs) ? AMPS_FSK_DEVIATION : TACS_FSK_DEVIATION) / 2.0;
	if (bit)
		amps->fsk_rx_frame_quality += (double)(second - first) / (double)((!tacs) ? AMPS_FSK_DEVIATION : TACS_FSK_DEVIATION) / 2.0 / BEST_QUALITY;
	else
		amps->fsk_rx_frame_quality += (double)(first - second) / (double)((!tacs) ? AMPS_FSK_DEVIATION : TACS_FSK_DEVIATION) / 2.0 / BEST_QUALITY;

	/* invert bit if negative sync was detected */
	if (amps->fsk_rx_sync == FSK_SYNC_NEGATIVE)
		bit = 1 - bit;

	/* read next bit. after all bits, we reset to FSK_SYNC_NONE */
	amps->fsk_rx_frame[amps->fsk_rx_frame_count++] = bit + '0';
	if (amps->fsk_rx_frame_count > FSK_MAX_BITS) {
		fprintf(stderr, "our fsk_tx_count (%d) is larger than our max bits we can handle, please fix!\n", amps->fsk_rx_frame_count);
		abort();
	}
	if (amps->fsk_rx_frame_count == amps->fsk_rx_frame_length) {
		int more;

		/* update measurements */
		display_measurements_update(amps->dmp_frame_level, amps->fsk_rx_frame_level / (double)amps->fsk_rx_frame_count * 100.0, 0.0);
		display_measurements_update(amps->dmp_frame_quality, amps->fsk_rx_frame_quality / (double)amps->fsk_rx_frame_count * 100.0, 0.0);

		/* a complete frame was received, so we process it */
		amps->fsk_rx_frame[amps->fsk_rx_frame_count] = '\0';
		more = amps_decode_frame(amps, amps->fsk_rx_frame, amps->fsk_rx_frame_count, amps->fsk_rx_frame_level / (double)amps->fsk_rx_frame_count, amps->fsk_rx_frame_quality / amps->fsk_rx_frame_level, (amps->fsk_rx_sync == FSK_SYNC_NEGATIVE));
		if (more) {
			/* switch to next word length without DCC included */
			amps->fsk_rx_frame_length = 240;
			goto prepare_frame;
		} else {
			/* switch back to first word length with DCC included */
			if (amps->fsk_rx_frame_length == 240)
				amps->fsk_rx_frame_length = 247;
			amps->fsk_rx_sync = FSK_SYNC_NONE;
			amps->channel_busy = 0;
		}
	}
}

static void fsk_rx_dotting(amps_t *amps, double _elapsed)
{
	uint8_t pos = amps->fsk_rx_dotting_pos++;
	double average, elapsed, offset;
	int i;

#ifdef DEBUG_DECODER
	printf("Level change detected\n");
#endif
	/* store into dotting list */
	amps->fsk_rx_dotting_elapsed[pos++] = _elapsed;

	/* check quality of dotting sequence.
	 * in case this is not a dotting sequence, noise or speech, the quality
	 * should be bad.
	 * count (only) 7 'elapsed' values between 8 zero-crossings.
	 * calculate the average relative to the current position.
	 */
	average = 0.0;
	elapsed = 0.0;
	for (i = 1; i < 8; i++) {
		elapsed += amps->fsk_rx_dotting_elapsed[--pos];
		offset = elapsed - (double)i;
		if (offset >= 0.5 || offset <= -0.5) {
#ifdef DEBUG_DECODER
//			printf("offset %.3f (last but %d) not within -0.5 .. 0.5 bit position, detecting no dotting.\n", offset, i - 1);
#endif
			return;
		}
		average += offset;
	}
	average /= (double)i;

	amps->fsk_rx_dotting_life = 12;

	/* if we are already found dotting, we detect better dotting.
	 * this happens, if dotting was falsely detected due to noise.
	 * then the real dotting causes a reastart of hunting for sync sequence.
	 */
	if (amps->fsk_rx_sync == FSK_SYNC_NONE || fabs(average) < amps->fsk_rx_dotting_average) {
#ifdef DEBUG_DECODER
		printf("Found (better) dotting sequence (average = %.3f)\n", average);
#endif
		amps->fsk_rx_sync = FSK_SYNC_DOTTING;
		amps->fsk_rx_dotting_average = fabs(average);
		amps->fsk_rx_bitcount = 0.5 + average;
		if (amps->si.acc_type.bis)
			amps->channel_busy = 1;
	}
}

/* decode frame */
static void sender_receive_frame(amps_t *amps, sample_t *samples, int length)
{
	int i;

	for (i = 0; i < length; i++) {
#ifdef DEBUG_DECODER
		puts(debug_amplitude(samples[i] / (double)FSK_DEVIATION));
#endif
		/* push sample to detection window and shift */
		amps->fsk_rx_window[amps->fsk_rx_window_pos++] = samples[i];
		if (amps->fsk_rx_window_pos == amps->fsk_rx_window_length)
			amps->fsk_rx_window_pos = 0;
		if (amps->fsk_rx_sync != FSK_SYNC_POSITIVE && amps->fsk_rx_sync != FSK_SYNC_NEGATIVE) {
			/* check for change in polarity */
			if (amps->fsk_rx_last_sample <= 0) {
				if (samples[i] > 0) {
					fsk_rx_dotting(amps, amps->fsk_rx_elapsed);
					amps->fsk_rx_elapsed = 0.0;
				}
			} else {
				if (samples[i] <= 0) {
					fsk_rx_dotting(amps, amps->fsk_rx_elapsed);
					amps->fsk_rx_elapsed = 0.0;
				}
			}
		}
		amps->fsk_rx_last_sample = samples[i];
		amps->fsk_rx_elapsed += amps->fsk_bitstep;
//		printf("%.4f\n", bitcount);
		if (amps->fsk_rx_sync != FSK_SYNC_NONE) {
			amps->fsk_rx_bitcount += amps->fsk_bitstep;
			if (amps->fsk_rx_bitcount >= 1.0) {
				amps->fsk_rx_bitcount -= 1.0;
				fsk_rx_bit(amps,
					amps->fsk_rx_window,
					amps->fsk_rx_window_length,
					amps->fsk_rx_window_pos,
					amps->fsk_rx_window_begin,
					amps->fsk_rx_window_half,
					amps->fsk_rx_window_end);
			}
		}
	}
}


/* decode SAT and signaling tone */
/* compare supervisory signal against noise floor on 5800 Hz */
static void sat_decode(amps_t *amps, sample_t *samples, int length)
{
	double result[3], quality[2];

	audio_goertzel(&amps->sat_goertzel[amps->sat], samples, length, 0, &result[0], 1);
	audio_goertzel(&amps->sat_goertzel[3], samples, length, 0, &result[1], 1);
	audio_goertzel(&amps->sat_goertzel[4], samples, length, 0, &result[2], 1);

	quality[0] = (result[0] - result[1]) / result[0];
	if (quality[0] < 0)
		quality[0] = 0;
	quality[1] = (result[2] - result[1]) / result[2];
	if (quality[1] < 0)
		quality[1] = 0;

	PDEBUG_CHAN(DDSP, DEBUG_NOTICE, "SAT level %.2f%% quality %.0f%%\n", result[0] / ((!tacs) ? AMPS_SAT_DEVIATION : TACS_SAT_DEVIATION) / 0.63662 * 100.0, quality[0] * 100.0);
	if (amps->sender.loopback || debuglevel == DEBUG_DEBUG) {
		PDEBUG_CHAN(DDSP, debuglevel, "Signaling Tone level %.2f%% quality %.0f%%\n", result[2] / ((!tacs) ? AMPS_FSK_DEVIATION : TACS_FSK_DEVIATION) / 0.63662 * 100.0, quality[1] * 100.0);
	}
	if (quality[0] > SAT_QUALITY && result[0] / ((!tacs) ? AMPS_SAT_DEVIATION : TACS_SAT_DEVIATION) / 0.63662 > DTX_LEVEL)
		amps->dtx_state = 1;
	else
		amps->dtx_state = 0;
	if (quality[0] > SAT_QUALITY) {
		if (amps->sat_detected == 0) {
			amps->sat_detect_count++;
			if (amps->sat_detect_count == SAT_DETECT_COUNT) {
				amps->sat_detected = 1;
				amps->sat_detect_count = 0;
				PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "SAT signal detected with level=%.0f%%, quality=%.0f%%.\n", result[0] / 0.63662 * 100.0, quality[0] * 100.0);
				amps_rx_sat(amps, 1, quality[0]);
			}
		} else
			amps->sat_detect_count = 0;
	} else {
		if (amps->sat_detected == 1) {
			amps->sat_detect_count++;
			if (amps->sat_detect_count == SAT_LOST_COUNT) {
				amps->sat_detected = 0;
				amps->sat_detect_count = 0;
				PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "SAT signal lost.\n");
				amps_rx_sat(amps, 0, 0.0);
			}
		} else
			amps->sat_detect_count = 0;
	}
	if (quality[1] > SIG_QUALITY) {
		if (amps->sig_detected == 0) {
			amps->sig_detect_count++;
			if (amps->sig_detect_count == SIG_DETECT_COUNT) {
				amps->sig_detected = 1;
				amps->sig_detect_count = 0;
				PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Signaling Tone detected with level=%.0f%%, quality=%.0f%%.\n", result[2] / 0.63662 * 100.0, quality[1] * 100.0);
				amps_rx_signaling_tone(amps, 1, quality[1]);
			}
		} else
			amps->sig_detect_count = 0;
	} else {
		if (amps->sig_detected == 1) {
			amps->sig_detect_count++;
			if (amps->sig_detect_count == SIG_LOST_COUNT) {
				amps->sig_detected = 0;
				amps->sig_detect_count = 0;
				PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Signaling Tone lost.\n");
				amps_rx_signaling_tone(amps, 0, 0.0);
			}
		} else
			amps->sig_detect_count = 0;
	}
}

static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
{
	transaction_t *trans = amps->trans_list;
	sample_t *spl;
	int max, pos;
	int i;

	/* SAT / signalling tone detection */
	max = amps->sat_samples;
	spl = amps->sat_filter_spl;
	pos = amps->sat_filter_pos;
	for (i = 0; i < length; i++) {
		spl[pos++] = samples[i];
		if (pos == max) {
			pos = 0;
			sat_decode(amps, spl, max);
		}
	}
	amps->sat_filter_pos = pos;

	/* receive audio, but only if call established and SAT detected */

	if ((amps->dsp_mode == DSP_MODE_AUDIO_RX_AUDIO_TX || amps->dsp_mode == DSP_MODE_AUDIO_RX_FRAME_TX)
	 && trans && trans->callref) {
		int pos, count;
		int i;

		/* de-emphasis */
		if (amps->de_emphasis)
			de_emphasis(&amps->estate, samples, length);
		/* downsample */
		count = samplerate_downsample(&amps->sender.srstate, samples, length);
		expand_audio(&amps->cstate, samples, count);
		spl = amps->sender.rxbuf;
		pos = amps->sender.rxbuf_pos;
		for (i = 0; i < count; i++) {
			spl[pos++] = samples[i];
			if (pos == 160) {
				if (amps->dtx_state == 0)
					comfort_noise(spl, 160);
				call_up_audio(trans->callref, spl, 160);
				pos = 0;
			}
		}
		amps->sender.rxbuf_pos = pos;
	} else
		amps->sender.rxbuf_pos = 0;
}

/* Process received audio stream from radio unit. */
void sender_receive(sender_t *sender, sample_t *samples, int length, double __attribute__((unused)) rf_level_db)
{
	amps_t *amps = (amps_t *) sender;

	/* dc filter required for FSK decoding and tone detection */
	if (amps->de_emphasis)
		dc_filter(&amps->estate, samples, length);

	switch (amps->dsp_mode) {
	case DSP_MODE_OFF:
		break;
	case DSP_MODE_FRAME_RX_FRAME_TX:
		sender_receive_frame(amps, samples, length);
		break;
	case DSP_MODE_AUDIO_RX_AUDIO_TX:
	case DSP_MODE_AUDIO_RX_FRAME_TX:
		sender_receive_audio(amps, samples, length);
		break;
	}
}

/* Reset SAT detection states, so ongoing tone will be detected again. */
static void sat_reset(amps_t *amps, const char *reason)
{
	PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "SAT detector reset: %s.\n", reason);
	amps->sat_detected = 0;
	amps->sat_detect_count = 0;
	amps->sig_detected = 0;
	amps->sig_detect_count = 0;
}

void amps_set_dsp_mode(amps_t *amps, enum dsp_mode mode, int frame_length)
{
#if 0
	/* reset telegramm */
	if (mode == DSP_MODE_FRAME && amps->dsp_mode != mode)
		amps->frame = 0;
#endif
	if (mode == DSP_MODE_FRAME_RX_FRAME_TX) {
		/* reset SAT detection */
		sat_reset(amps, "Change to FOCC");
		PDEBUG_CHAN(DDSP, DEBUG_INFO, "Change mode to FOCC\n");
	}
	if (amps->dsp_mode == DSP_MODE_FRAME_RX_FRAME_TX
	 && (mode == DSP_MODE_AUDIO_RX_AUDIO_TX || mode == DSP_MODE_AUDIO_RX_FRAME_TX)) {
		/* reset SAT detection */
		sat_reset(amps, "Change from FOCC to FVC");
		PDEBUG_CHAN(DDSP, DEBUG_INFO, "Change mode from FOCC to FVC\n");
	}
	if (amps->dsp_mode == DSP_MODE_OFF
	 && (mode == DSP_MODE_AUDIO_RX_AUDIO_TX || mode == DSP_MODE_AUDIO_RX_FRAME_TX)) {
		/* reset SAT detection */
		sat_reset(amps, "Enable FVC");
		PDEBUG_CHAN(DDSP, DEBUG_INFO, "Change mode from OFF to FVC\n");
	}
	if (mode == DSP_MODE_OFF) {
		/* reset SAT detection */
		sat_reset(amps, "Disable FVC");
		PDEBUG_CHAN(DDSP, DEBUG_INFO, "Change mode from FVC to OFF\n");
	}

	amps->dsp_mode = mode;
	if (frame_length)
		amps->fsk_rx_frame_length = frame_length;

	/* reset detection process */
	amps->fsk_rx_sync = FSK_SYNC_NONE;
	amps->channel_busy = 0;
	amps->fsk_rx_sync_register = 0x555;

	/* reset transmitter */
	amps->fsk_tx_buffer_pos = 0;
	amps->fsk_tx_frame[0] = '\0';
	amps->fsk_tx_frame_pos = 0;
}