1 files changed, 0 insertions, 256 deletions
diff --git a/src/common/ffsk.c b/src/common/ffsk.c
deleted file mode 100644
index fdbf255..0000000
--- a/src/common/ffsk.c
+++ /dev/null
@@ -1,256 +0,0 @@
-/* FFSK audio processing (NMT / Radiocom 2000)
- *
- * (C) 2017 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/>.
- */
-
-#define CHAN ffsk->channel
-
-#include <stdio.h>
-#include <stdint.h>
-#include <stdlib.h>
-#include <string.h>
-#include <errno.h>
-#include <math.h>
-#include "../common/sample.h"
-#include "../common/debug.h"
-#include "ffsk.h"
-
-#define PI			M_PI
-
-#define BIT_RATE		1200	/* baud rate */
-#define FILTER_STEPS		0.1	/* step every 1/12000 sec */
-
-/* two signaling tones */
-static double ffsk_freq[2] = {
-	1800.0,
-	1200.0,
-};
-
-static sample_t dsp_tone_bit[2][2][65536]; /* polarity, bit, phase */
-
-/* global init for FFSK */
-void ffsk_global_init(double peak_fsk)
-{
-	int i;
-	double s;
-
-	PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for FFSK tones.\n");
-	for (i = 0; i < 65536; i++) {
-		s = sin((double)i / 65536.0 * 2.0 * PI);
-		/* bit(1) 1 cycle */
-		dsp_tone_bit[0][1][i] = s * peak_fsk;
-		dsp_tone_bit[1][1][i] = -s * peak_fsk;
-		/* bit(0) 1.5 cycles */
-		s = sin((double)i / 65536.0 * 3.0 * PI);
-		dsp_tone_bit[0][0][i] = s * peak_fsk;
-		dsp_tone_bit[1][0][i] = -s * peak_fsk;
-	}
-}
-
-/* Init FFSK */
-int ffsk_init(ffsk_t *ffsk, void *inst, void (*receive_bit)(void *inst, int bit, double quality, double level), int channel, int samplerate)
-{
-	sample_t *spl;
-	int i;
-
-	/* a symbol rate of 1200 Hz, times check interval of FILTER_STEPS */
-	if (samplerate < (double)BIT_RATE / (double)FILTER_STEPS) {
-		PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process FSK+supervisory signal.\n");
-		return -EINVAL;
-	}
-
-	memset(ffsk, 0, sizeof(*ffsk));
-	ffsk->inst = inst;
-	ffsk->receive_bit = receive_bit;
-	ffsk->channel = channel;
-	ffsk->samplerate = samplerate;
-
-	ffsk->samples_per_bit = (double)ffsk->samplerate / (double)BIT_RATE;
-	ffsk->bits_per_sample = 1.0 / ffsk->samples_per_bit;
-	PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", ffsk->samples_per_bit, ffsk->samplerate);
-
-	/* allocate ring buffers, one bit duration */
-	ffsk->filter_size = floor(ffsk->samples_per_bit); /* buffer holds one bit (rounded down) */
-	spl = calloc(1, ffsk->filter_size * sizeof(*spl));
-	if (!spl) {
-		PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
-		ffsk_cleanup(ffsk);
-		return -ENOMEM;
-	}
-	ffsk->filter_spl = spl;
-	ffsk->filter_bit = -1;
-
-	/* count symbols */
-	for (i = 0; i < 2; i++)
-		audio_goertzel_init(&ffsk->goertzel[i], ffsk_freq[i], ffsk->samplerate);
-	ffsk->phaseshift65536 = 65536.0 / ffsk->samples_per_bit;
-	PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", ffsk->phaseshift65536);
-
-	return 0;
-}
-
-/* Cleanup transceiver instance. */
-void ffsk_cleanup(ffsk_t *ffsk)
-{
-	PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for Transceiver.\n");
-
-	if (ffsk->filter_spl) {
-		free(ffsk->filter_spl);
-		ffsk->filter_spl = NULL;
-	}
-}
-
-//#define DEBUG_MODULATOR
-//#define DEBUG_FILTER
-//#define DEBUG_QUALITY
-
-/* Filter one chunk of audio an detect tone, quality and loss of signal.
- * The chunk is a window of 1/1200s. This window slides over audio stream
- * and is processed every 1/12000s. (one step) */
-static inline void ffsk_decode_step(ffsk_t *ffsk, int pos)
-{
-	double level, result[2], softbit, quality;
-	int max;
-	sample_t *spl;
-	int bit;
-	
-	max = ffsk->filter_size;
-	spl = ffsk->filter_spl;
-
-	level = audio_level(spl, max);
-	/* limit level to prevent division by zero */
-	if (level < 0.001)
-		level = 0.001;
-
-	audio_goertzel(ffsk->goertzel, spl, max, pos, result, 2);
-
-	/* calculate soft bit from both frequencies */
-	softbit = (result[1] / level - result[0] / level + 1.0) / 2.0;
-//printf("%.3f: %.3f\n", level, softbit);
-	/* scale it, since both filters overlap by some percent */
-#define MIN_QUALITY 0.33
-	softbit = (softbit - MIN_QUALITY) / (1.0 - MIN_QUALITY - MIN_QUALITY);
-#ifdef DEBUG_FILTER
-//	printf("|%s", debug_amplitude(result[0]/level));
-//	printf("|%s| low=%.3f high=%.3f level=%d\n", debug_amplitude(result[1]/level), result[0]/level, result[1]/level, (int)level);
-	printf("|%s| softbit=%.3f\n", debug_amplitude(softbit), softbit);
-#endif
-	if (softbit > 1)
-		softbit = 1;
-	if (softbit < 0)
-		softbit = 0;
-	if (softbit > 0.5)
-		bit = 1;
-	else
-		bit = 0;
-
-	if (ffsk->filter_bit != bit) {
-		/* If we have a bit change, move sample counter towards one half bit duration.
-		 * We may have noise, so the bit change may be wrong or not at the correct place.
-		 * This can cause bit slips.
-		 * Therefore we change the sample counter only slightly, so bit slips may not
-		 * happen so quickly.
-		 * */
-#ifdef DEBUG_FILTER
-		puts("bit change");
-#endif
-		ffsk->filter_bit = bit;
-		if (ffsk->filter_sample < 5)
-			ffsk->filter_sample++;
-		if (ffsk->filter_sample > 5)
-			ffsk->filter_sample--;
-	} else if (--ffsk->filter_sample == 0) {
-		/* if sample counter bit reaches 0, we reset sample counter to one bit duration */
-#ifdef DEBUG_FILTER
-		puts("sample");
-#endif
-//		quality = result[bit] / level;
-		if (softbit > 0.5)
-			quality = softbit * 2.0 - 1.0;
-		else
-			quality = 1.0 - softbit * 2.0;
-#ifdef DEBUG_QUALITY
-		printf("|%s| quality=%.2f ", debug_amplitude(softbit), quality);
-		printf("|%s|\n", debug_amplitude(quality));
-#endif
-		/* adjust level, so a peak level becomes 100% */
-		ffsk->receive_bit(ffsk->inst, bit, quality, level / 0.63662);
-		ffsk->filter_sample = 10;
-	}
-}
-
-void ffsk_receive(ffsk_t *ffsk, sample_t *sample, int length)
-{
-	sample_t *spl;
-	int max, pos;
-	double step, bps;
-	int i;
-
-	/* write received samples to decode buffer */
-	max = ffsk->filter_size;
-	pos = ffsk->filter_pos;
-	step = ffsk->filter_step;
-	bps = ffsk->bits_per_sample;
-	spl = ffsk->filter_spl;
-	for (i = 0; i < length; i++) {
-#ifdef DEBUG_MODULATOR
-		printf("|%s|\n", debug_amplitude((double)samples[i] / 2333.0 /*fsk peak*/ / 2.0));
-#endif
-		/* write into ring buffer */
-		spl[pos++] = sample[i];
-		if (pos == max)
-			pos = 0;
-		/* if 1/10th of a bit duration is reached, decode buffer */
-		step += bps;
-		if (step >= FILTER_STEPS) {
-			step -= FILTER_STEPS;
-			ffsk_decode_step(ffsk, pos);
-		}
-	}
-	ffsk->filter_step = step;
-	ffsk->filter_pos = pos;
-}
-
-/* render frame */
-int ffsk_render_frame(ffsk_t *ffsk, const char *frame, int length, sample_t *sample)
-{
-	int bit, polarity;
-        double phaseshift, phase;
-	int count = 0, i;
-
-	polarity = ffsk->polarity;
-	phaseshift = ffsk->phaseshift65536;
-	phase = ffsk->phase65536;
-	for (i = 0; i < length; i++) {
-		bit = (frame[i] == '1');
-		do {
-			*sample++ = dsp_tone_bit[polarity][bit][(uint16_t)phase];
-			count++;
-			phase += phaseshift;
-		} while (phase < 65536.0);
-		phase -= 65536.0;
-		/* flip polarity when we have 1.5 sine waves */
-		if (bit == 0)
-			polarity = 1 - polarity;
-	}
-	ffsk->phase65536 = phase;
-	ffsk->polarity = polarity;
-
-	/* return number of samples created for frame */
-	return count;
-}
-