aboutsummaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
authorAndreas Eversberg <jolly@eversberg.eu>2017-07-24 16:18:10 +0200
committerAndreas Eversberg <jolly@eversberg.eu>2017-08-08 12:53:34 +0200
commit6c6402571758340c640bd2350599ff6a9e5ffeb6 (patch)
tree3492e67b9e9ec10d478aace75385c65003d96523
parent92ce6d4a428bb72692800ef32b5b80e69fef032b (diff)
Move FFSK modem from NMT to common code, so it can be used by other networks
-rw-r--r--src/common/Makefile.am1
-rw-r--r--src/common/ffsk.c256
-rw-r--r--src/common/ffsk.h27
-rw-r--r--src/nmt/dms.c9
-rw-r--r--src/nmt/dms.h2
-rw-r--r--src/nmt/dsp.c260
-rw-r--r--src/nmt/dsp.h1
-rw-r--r--src/nmt/main.c3
-rw-r--r--src/nmt/nmt.c4
-rw-r--r--src/nmt/nmt.h37
-rw-r--r--src/test/test_dms.c10
11 files changed, 371 insertions, 239 deletions
diff --git a/src/common/Makefile.am b/src/common/Makefile.am
index abdb4b5..f719c6e 100644
--- a/src/common/Makefile.am
+++ b/src/common/Makefile.am
@@ -24,6 +24,7 @@ libcommon_a_SOURCES = \
compandor.c \
fft.c \
fm_modulation.c \
+ ffsk.c \
sender.c \
display_wave.c \
display_status.c \
diff --git a/src/common/ffsk.c b/src/common/ffsk.c
new file mode 100644
index 0000000..fdbf255
--- /dev/null
+++ b/src/common/ffsk.c
@@ -0,0 +1,256 @@
+/* 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;
+}
+
diff --git a/src/common/ffsk.h b/src/common/ffsk.h
new file mode 100644
index 0000000..84fc52a
--- /dev/null
+++ b/src/common/ffsk.h
@@ -0,0 +1,27 @@
+#include "../common/goertzel.h"
+
+typedef struct ffsk {
+ void *inst;
+ void (*receive_bit)(void *inst, int bit, double quality, double level);
+ int channel; /* channel number */
+ int samplerate; /* current sample rate */
+ double samples_per_bit; /* number of samples for one bit (1200 Baud) */
+ double bits_per_sample; /* fraction of a bit per sample */
+ goertzel_t goertzel[2]; /* filter for fsk decoding */
+ int polarity; /* current polarity state of bit */
+ sample_t *filter_spl; /* array to hold ring buffer for bit decoding */
+ int filter_size; /* size of ring buffer */
+ int filter_pos; /* position to write next sample */
+ double filter_step; /* counts bit duration, to trigger decoding every 10th bit */
+ int filter_bit; /* last bit state, so we detect a bit change */
+ int filter_sample; /* count until it is time to sample bit */
+ double phaseshift65536; /* how much the phase of fsk synbol changes per sample */
+ double phase65536; /* current phase */
+} ffsk_t;
+
+void ffsk_global_init(double peak_fsk);
+int ffsk_init(ffsk_t *ffsk, void *inst, void (*receive_bit)(void *inst, int bit, double quality, double level), int channel, int samplerate);
+void ffsk_cleanup(ffsk_t *ffsk);
+void ffsk_receive(ffsk_t *ffsk, sample_t *sample, int lenght);
+int ffsk_render_frame(ffsk_t *ffsk, const char *frame, int length, sample_t *sample);
+
diff --git a/src/nmt/dms.c b/src/nmt/dms.c
index 2fcaf3c..8a27be0 100644
--- a/src/nmt/dms.c
+++ b/src/nmt/dms.c
@@ -25,7 +25,6 @@
#include "../common/debug.h"
#include "../common/timer.h"
#include "nmt.h"
-#include "dsp.h"
#define MUTE_DURATION 0.300 /* 200ms, and about 95ms for the frame itself */
@@ -288,7 +287,8 @@ static void dms_encode_dt(nmt_t *nmt, uint8_t d, uint8_t s, uint8_t n, uint8_t *
#endif
/* render wave form */
- dms->frame_length = fsk_render_frame(nmt, frame, 127, dms->frame_spl);
+ test_dms_frame(frame, 127); // used by test program
+ dms->frame_length = ffsk_render_frame(&nmt->ffsk, frame, 127, dms->frame_spl);
dms->frame_valid = 1;
dms->frame_pos = 0;
if (dms->frame_length > dms->frame_size) {
@@ -335,7 +335,8 @@ static void dms_encode_rr(nmt_t *nmt, uint8_t d, uint8_t s, uint8_t n)
#endif
/* render wave form */
- dms->frame_length = fsk_render_frame(nmt, frame, 77, dms->frame_spl);
+ test_dms_frame(frame, 77); // used by test program
+ dms->frame_length = ffsk_render_frame(&nmt->ffsk, frame, 77, dms->frame_spl);
dms->frame_valid = 1;
dms->frame_pos = 0;
if (dms->frame_length > dms->frame_size) {
@@ -662,7 +663,7 @@ void fsk_receive_bit_dms(nmt_t *nmt, int bit, double quality, double level)
memset(dms->rx_frame_quality, 0, sizeof(dms->rx_frame_quality));
/* set muting of receive path */
- nmt->fsk_filter_mute = (int)((double)nmt->sender.samplerate * MUTE_DURATION);
+ nmt->rx_mute = (int)((double)nmt->sender.samplerate * MUTE_DURATION);
return;
}
diff --git a/src/nmt/dms.h b/src/nmt/dms.h
index e4bfd37..1810e00 100644
--- a/src/nmt/dms.h
+++ b/src/nmt/dms.h
@@ -60,3 +60,5 @@ void dms_send(nmt_t *nmt, const uint8_t *data, int length, int eight_bits);
void dms_all_sent(nmt_t *nmt);
void dms_receive(nmt_t *nmt, const uint8_t *data, int length, int eight_bits);
+void test_dms_frame(const char *frame, int length);
+
diff --git a/src/nmt/dsp.c b/src/nmt/dsp.c
index adc588b..0a1ba2d 100644
--- a/src/nmt/dsp.c
+++ b/src/nmt/dsp.c
@@ -59,9 +59,8 @@
#define COMPANDOR_0DB 1.0 /* A level of 0dBm (1.0) shall be unaccected */
#define TX_PEAK_FSK (4200.0 / 1800.0 * 1000.0 / DBM0_DEVIATION)
#define TX_PEAK_SUPER (300.0 / 4015.0 * 1000.0 / DBM0_DEVIATION)
+#define BIT_RATE 1200
#define MAX_DISPLAY 1.4 /* something above dBm0 */
-#define BIT_RATE 1200 /* baud rate */
-#define FILTER_STEPS 0.1 /* step every 1/12000 sec */
#define DIALTONE_HZ 425.0 /* dial tone frequency */
#define TX_PEAK_DIALTONE 0.5 /* dial tone peak FIXME */
#define SUPER_DURATION 0.25 /* duration of supervisory signal measurement */
@@ -69,12 +68,6 @@
#define SUPER_DETECT_COUNT 6 /* number of measures to detect supervisory signal */
#define MUTE_DURATION 0.280 /* a tiny bit more than two frames */
-/* two signaling tones */
-static double fsk_freq[2] = {
- 1800.0,
- 1200.0,
-};
-
/* two supervisory tones */
static double super_freq[5] = {
3955.0, /* 0-Signal 1 */
@@ -85,48 +78,39 @@ static double super_freq[5] = {
};
/* table for fast sine generation */
-static sample_t dsp_tone_bit[2][2][65536]; /* polarity, bit, phase */
static sample_t dsp_sine_super[65536];
static sample_t dsp_sine_dialtone[65536];
-/* global init for FSK */
+/* global init for FFSK */
void dsp_init(void)
{
int i;
double s;
- PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for supervisory signal.\n");
+ PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for supervisory signal and dial tone.\n");
for (i = 0; i < 65536; i++) {
s = sin((double)i / 65536.0 * 2.0 * PI);
/* supervisor sine */
dsp_sine_super[i] = s * TX_PEAK_SUPER;
/* dialtone sine */
dsp_sine_dialtone[i] = s * TX_PEAK_DIALTONE;
- /* bit(1) 1 cycle */
- dsp_tone_bit[0][1][i] = s * TX_PEAK_FSK;
- dsp_tone_bit[1][1][i] = -s * TX_PEAK_FSK;
- /* bit(0) 1.5 cycles */
- s = sin((double)i / 65536.0 * 3.0 * PI);
- dsp_tone_bit[0][0][i] = s * TX_PEAK_FSK;
- dsp_tone_bit[1][0][i] = -s * TX_PEAK_FSK;
}
+
+ ffsk_global_init(TX_PEAK_FSK);
}
+static void fsk_receive_bit(void *inst, int bit, double quality, double level);
+
/* Init FSK of transceiver */
int dsp_init_sender(nmt_t *nmt, double deviation_factor)
{
sample_t *spl;
+ double samples_per_bit;
int i;
/* attack (3ms) and recovery time (13.5ms) according to NMT specs */
init_compandor(&nmt->cstate, 8000, 3.0, 13.5, COMPANDOR_0DB);
- /* a symbol rate of 1200 Hz, times check interval of FILTER_STEPS */
- if (nmt->sender.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;
- }
-
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for Transceiver.\n");
/* set modulation parameters */
@@ -135,22 +119,16 @@ int dsp_init_sender(nmt_t *nmt, double deviation_factor)
PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.3f (%.3f KHz deviation @ 1500 Hz)\n", TX_PEAK_FSK * deviation_factor, 3.5 * deviation_factor);
PDEBUG(DDSP, DEBUG_DEBUG, "Using Supervisory level of %.3f (%.3f KHz deviation @ 4015 Hz)\n", TX_PEAK_SUPER * deviation_factor, 0.3 * deviation_factor);
- nmt->fsk_samples_per_bit = (double)nmt->sender.samplerate / (double)BIT_RATE;
- nmt->fsk_bits_per_sample = 1.0 / nmt->fsk_samples_per_bit;
- PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", nmt->fsk_samples_per_bit, nmt->sender.samplerate);
-
- /* allocate ring buffers, one bit duration */
- nmt->fsk_filter_size = floor(nmt->fsk_samples_per_bit); /* buffer holds one bit (rounded down) */
- spl = calloc(1, nmt->fsk_filter_size * sizeof(*spl));
- if (!spl) {
- PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
- return -ENOMEM;
+ /* init ffsk */
+ if (ffsk_init(&nmt->ffsk, nmt, fsk_receive_bit, nmt->sender.kanal, nmt->sender.samplerate) < 0) {
+ PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "FFSK init failed!\n");
+ return -EINVAL;
}
- nmt->fsk_filter_spl = spl;
- nmt->fsk_filter_bit = -1;
/* allocate transmit buffer for a complete frame, add 10 to be safe */
- nmt->frame_size = 166.0 * (double)nmt->fsk_samples_per_bit + 10;
+
+ samples_per_bit = (double)nmt->sender.samplerate / (double)BIT_RATE;
+ nmt->frame_size = 166.0 * samples_per_bit + 10;
spl = calloc(nmt->frame_size, sizeof(*spl));
if (!spl) {
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
@@ -159,7 +137,7 @@ int dsp_init_sender(nmt_t *nmt, double deviation_factor)
nmt->frame_spl = spl;
/* allocate DMS transmit buffer for a complete frame, add 10 to be safe */
- nmt->dms.frame_size = 127.0 * (double)nmt->fsk_samples_per_bit + 10;
+ nmt->dms.frame_size = 127.0 * samples_per_bit + 10;
spl = calloc(nmt->dms.frame_size, sizeof(*spl));
if (!spl) {
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
@@ -176,12 +154,6 @@ int dsp_init_sender(nmt_t *nmt, double deviation_factor)
}
nmt->super_filter_spl = spl;
- /* count symbols */
- for (i = 0; i < 2; i++)
- audio_goertzel_init(&nmt->fsk_goertzel[i], fsk_freq[i], nmt->sender.samplerate);
- nmt->fsk_phaseshift65536 = 65536.0 / nmt->fsk_samples_per_bit;
- PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", nmt->fsk_phaseshift65536);
-
/* count supervidory tones */
for (i = 0; i < 5; i++) {
audio_goertzel_init(&nmt->super_goertzel[i], super_freq[i], nmt->sender.samplerate);
@@ -207,6 +179,8 @@ void dsp_cleanup_sender(nmt_t *nmt)
{
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for Transceiver.\n");
+ ffsk_cleanup(&nmt->ffsk);
+
if (nmt->frame_spl) {
free(nmt->frame_spl);
nmt->frame_spl = NULL;
@@ -215,10 +189,6 @@ void dsp_cleanup_sender(nmt_t *nmt)
free(nmt->dms.frame_spl);
nmt->dms.frame_spl = NULL;
}
- if (nmt->fsk_filter_spl) {
- free(nmt->fsk_filter_spl);
- nmt->fsk_filter_spl = NULL;
- }
if (nmt->super_filter_spl) {
free(nmt->super_filter_spl);
nmt->super_filter_spl = NULL;
@@ -226,29 +196,38 @@ void dsp_cleanup_sender(nmt_t *nmt)
}
/* Check for SYNC bits, then collect data bits */
-static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
+static void fsk_receive_bit(void *inst, int bit, double quality, double level)
{
- double frames_elapsed;
+ nmt_t *nmt = (nmt_t *)inst;
+ uint64_t frames_elapsed;
int i;
+ /* normalize FSK level */
+ level /= TX_PEAK_FSK;
+
+ nmt->rx_bits_count++;
+
+ if (nmt->dms_call)
+ fsk_receive_bit_dms(nmt, bit, quality, level);
+
// printf("bit=%d quality=%.4f\n", bit, quality);
- if (!nmt->fsk_filter_in_sync) {
- nmt->fsk_filter_sync = (nmt->fsk_filter_sync << 1) | bit;
+ if (!nmt->rx_in_sync) {
+ nmt->rx_sync = (nmt->rx_sync << 1) | bit;
/* level and quality */
- nmt->fsk_filter_level[nmt->fsk_filter_count & 0xff] = level;
- nmt->fsk_filter_quality[nmt->fsk_filter_count & 0xff] = quality;
- nmt->fsk_filter_count++;
+ nmt->rx_level[nmt->rx_count & 0xff] = level;
+ nmt->rx_quality[nmt->rx_count & 0xff] = quality;
+ nmt->rx_count++;
/* check if pattern 1010111100010010 matches */
- if (nmt->fsk_filter_sync != 0xaf12)
+ if (nmt->rx_sync != 0xaf12)
return;
/* average level and quality */
level = quality = 0;
for (i = 0; i < 16; i++) {
- level += nmt->fsk_filter_level[(nmt->fsk_filter_count - 1 - i) & 0xff];
- quality += nmt->fsk_filter_quality[(nmt->fsk_filter_count - 1 - i) & 0xff];
+ level += nmt->rx_level[(nmt->rx_count - 1 - i) & 0xff];
+ quality += nmt->rx_quality[(nmt->rx_count - 1 - i) & 0xff];
}
level /= 16.0; quality /= 16.0;
// printf("sync (level = %.2f, quality = %.2f\n", level, quality);
@@ -262,114 +241,38 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
nmt->rx_bits_count_current = nmt->rx_bits_count - 26.0;
/* rest sync register */
- nmt->fsk_filter_sync = 0;
- nmt->fsk_filter_in_sync = 1;
- nmt->fsk_filter_count = 0;
+ nmt->rx_sync = 0;
+ nmt->rx_in_sync = 1;
+ nmt->rx_count = 0;
/* set muting of receive path */
- nmt->fsk_filter_mute = (int)((double)nmt->sender.samplerate * MUTE_DURATION);
+ nmt->rx_mute = (int)((double)nmt->sender.samplerate * MUTE_DURATION);
return;
}
/* read bits */
- nmt->fsk_filter_frame[nmt->fsk_filter_count] = bit + '0';
- nmt->fsk_filter_level[nmt->fsk_filter_count] = level;
- nmt->fsk_filter_quality[nmt->fsk_filter_count] = quality;
- if (++nmt->fsk_filter_count != 140)
+ nmt->rx_frame[nmt->rx_count] = bit + '0';
+ nmt->rx_level[nmt->rx_count] = level;
+ nmt->rx_quality[nmt->rx_count] = quality;
+ if (++nmt->rx_count != 140)
return;
/* end of frame */
- nmt->fsk_filter_frame[140] = '\0';
- nmt->fsk_filter_in_sync = 0;
+ nmt->rx_frame[140] = '\0';
+ nmt->rx_in_sync = 0;
/* average level and quality */
level = quality = 0;
for (i = 0; i < 140; i++) {
- level += nmt->fsk_filter_level[i];
- quality += nmt->fsk_filter_quality[i];
+ level += nmt->rx_level[i];
+ quality += nmt->rx_quality[i];
}
level /= 140.0; quality /= 140.0;
/* send telegramm */
- frames_elapsed = (nmt->rx_bits_count_current - nmt->rx_bits_count_last) / 166.0;
+ frames_elapsed = (nmt->rx_bits_count_current - nmt->rx_bits_count_last + 83) / 166; /* round to nearest frame */
/* convert level so that received level at TX_PEAK_FSK results in 1.0 (100%) */
- nmt_receive_frame(nmt, nmt->fsk_filter_frame, quality, level, frames_elapsed);
-}
-
-//#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 fsk_decode_step(nmt_t *nmt, int pos)
-{
- double level, result[2], softbit, quality;
- int max;
- sample_t *spl;
- int bit;
-
- max = nmt->fsk_filter_size;
- spl = nmt->fsk_filter_spl;
-
- /* count time in bits */
- nmt->rx_bits_count += FILTER_STEPS;
-
- level = audio_level(spl, max);
- /* limit level to prevent division by zero */
- if (level < 0.001)
- level = 0.001;
-
- audio_goertzel(nmt->fsk_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 (nmt->fsk_filter_bit != bit) {
- /* if we have a bit change, reset sample counter to one half bit duration */
-#ifdef DEBUG_FILTER
- puts("bit change");
-#endif
- nmt->fsk_filter_bit = bit;
- nmt->fsk_filter_sample = 5;
- } else if (--nmt->fsk_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% */
- fsk_receive_bit(nmt, bit, quality, level / 0.63662 / TX_PEAK_FSK);
- if (nmt->dms_call)
- fsk_receive_bit_dms(nmt, bit, quality, level / 0.63662 / TX_PEAK_FSK);
- nmt->fsk_filter_sample = 10;
- }
+ nmt_receive_frame(nmt, nmt->rx_frame, quality, level, frames_elapsed);
}
/* compare supervisory signal against noise floor on 3900 Hz */
@@ -425,7 +328,6 @@ void sender_receive(sender_t *sender, sample_t *samples, int length)
nmt_t *nmt = (nmt_t *) sender;
sample_t *spl;
int max, pos;
- double step, bps;
int i;
/* write received samples to decode buffer */
@@ -442,34 +344,15 @@ void sender_receive(sender_t *sender, sample_t *samples, int length)
}
nmt->super_filter_pos = pos;
- /* write received samples to decode buffer */
- max = nmt->fsk_filter_size;
- pos = nmt->fsk_filter_pos;
- step = nmt->fsk_filter_step;
- bps = nmt->fsk_bits_per_sample;
- spl = nmt->fsk_filter_spl;
+ ffsk_receive(&nmt->ffsk, samples, length);
+
+ /* muting audio while receiving frame */
for (i = 0; i < length; i++) {
-#ifdef DEBUG_MODULATOR
- printf("|%s|\n", debug_amplitude((double)samples[i] / TX_PEAK_FSK / 2.0));
-#endif
- /* write into ring buffer */
- spl[pos++] = samples[i];
- if (pos == max)
- pos = 0;
- /* muting audio while receiving frame */
- if (nmt->fsk_filter_mute && !nmt->sender.loopback) {
+ if (nmt->rx_mute && !nmt->sender.loopback) {
samples[i] = 0;
- nmt->fsk_filter_mute--;
- }
- /* if 1/10th of a bit duration is reached, decode buffer */
- step += bps;
- if (step >= FILTER_STEPS) {
- step -= FILTER_STEPS;
- fsk_decode_step(nmt, pos);
+ nmt->rx_mute--;
}
}
- nmt->fsk_filter_step = step;
- nmt->fsk_filter_pos = pos;
if ((nmt->dsp_mode == DSP_MODE_AUDIO || nmt->dsp_mode == DSP_MODE_DTMF)
&& nmt->trans && nmt->trans->callref) {
@@ -494,35 +377,6 @@ void sender_receive(sender_t *sender, sample_t *samples, int length)
nmt->sender.rxbuf_pos = 0;
}
-/* render frame */
-int fsk_render_frame(nmt_t *nmt, const char *frame, int length, sample_t *sample)
-{
- int bit, polarity;
- double phaseshift, phase;
- int count = 0, i;
-
- polarity = nmt->fsk_polarity;
- phaseshift = nmt->fsk_phaseshift65536;
- phase = nmt->fsk_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;
- }
- nmt->fsk_phase65536 = phase;
- nmt->fsk_polarity = polarity;
-
- /* return number of samples created for frame */
- return count;
-}
-
static int fsk_frame(nmt_t *nmt, sample_t *samples, int length)
{
const char *frame;
@@ -539,7 +393,7 @@ next_frame:
return length;
}
/* render frame */
- nmt->frame_length = fsk_render_frame(nmt, frame, 166, nmt->frame_spl);
+ nmt->frame_length = ffsk_render_frame(&nmt->ffsk, frame, 166, nmt->frame_spl);
nmt->frame_pos = 0;
if (nmt->frame_length > nmt->frame_size) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n");
diff --git a/src/nmt/dsp.h b/src/nmt/dsp.h
index 34f04e6..fff2dba 100644
--- a/src/nmt/dsp.h
+++ b/src/nmt/dsp.h
@@ -2,7 +2,6 @@
void dsp_init(void);
int dsp_init_sender(nmt_t *nmt, double deviation_factor);
void dsp_cleanup_sender(nmt_t *nmt);
-int fsk_render_frame(nmt_t *nmt, const char *frame, int length, sample_t *sample);
void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode);
void super_reset(nmt_t *nmt);
diff --git a/src/nmt/main.c b/src/nmt/main.c
index 396f7d0..d29ee62 100644
--- a/src/nmt/main.c
+++ b/src/nmt/main.c
@@ -427,3 +427,6 @@ fail:
return 0;
}
+// dummy, will be replaced by DMS test program
+void test_dms_frame(const char __attribute__((unused)) *frame, int __attribute__((unused)) length) {}
+
diff --git a/src/nmt/nmt.c b/src/nmt/nmt.c
index 01a7b3c..847be72 100644
--- a/src/nmt/nmt.c
+++ b/src/nmt/nmt.c
@@ -1527,7 +1527,7 @@ static void tx_active(nmt_t *nmt, frame_t *frame)
* general handlers to call sub handling
*/
-void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double level, double frames_elapsed)
+void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double level, int frames_elapsed)
{
frame_t frame;
int rc;
@@ -1541,7 +1541,7 @@ void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double leve
}
/* frame counter */
- nmt->rx_frame_count += (int)(frames_elapsed + 0.5);
+ nmt->rx_frame_count += frames_elapsed;
PDEBUG_CHAN(DNMT, (nmt->sender.loopback) ? DEBUG_NOTICE : DEBUG_DEBUG, "Received frame %s\n", nmt_frame_name(frame.mt));
diff --git a/src/nmt/nmt.h b/src/nmt/nmt.h
index 1ed438e..3f9577c 100644
--- a/src/nmt/nmt.h
+++ b/src/nmt/nmt.h
@@ -1,8 +1,8 @@
-#include "../common/goertzel.h"
#include "../common/sender.h"
#include "../common/compandor.h"
#include "../common/dtmf.h"
#include "../common/call.h"
+#include "../common/ffsk.h"
#include "dms.h"
#include "sms.h"
@@ -96,40 +96,29 @@ typedef struct nmt {
/* dsp states */
enum dsp_mode dsp_mode; /* current mode: audio, durable tone 0 or 1, paging */
- double fsk_samples_per_bit; /* number of samples for one bit (1200 Baud) */
- double fsk_bits_per_sample; /* fraction of a bit per sample */
+ ffsk_t ffsk; /* ffsk processing */
int super_samples; /* number of samples in buffer for supervisory detection */
- goertzel_t fsk_goertzel[2]; /* filter for fsk decoding */
goertzel_t super_goertzel[5]; /* filter for supervisory decoding */
- int fsk_polarity; /* current polarity state of bit */
- sample_t *fsk_filter_spl; /* array to hold ring buffer for bit decoding */
- int fsk_filter_size; /* size of ring buffer */
- int fsk_filter_pos; /* position to write next sample */
- double fsk_filter_step; /* counts bit duration, to trigger decoding every 10th bit */
- int fsk_filter_bit; /* last bit state, so we detect a bit change */
- int fsk_filter_sample; /* count until it is time to sample bit */
- uint16_t fsk_filter_sync; /* shift register to detect sync */
- int fsk_filter_in_sync; /* if we are in sync and receive bits */
- int fsk_filter_mute; /* mute count down after sync */
- char fsk_filter_frame[141]; /* receive frame (one extra byte to terminate string) */
- int fsk_filter_count; /* next bit to receive */
- double fsk_filter_level[256]; /* level infos */
- double fsk_filter_quality[256];/* quality infos */
sample_t *super_filter_spl; /* array with sample buffer for supervisory detection */
int super_filter_pos; /* current sample position in filter_spl */
double super_phaseshift65536[4];/* how much the phase of sine wave changes per sample */
double super_phase65536; /* current phase */
double dial_phaseshift65536; /* how much the phase of sine wave changes per sample */
double dial_phase65536; /* current phase */
- double fsk_phaseshift65536; /* how much the phase of fsk synbol changes per sample */
- double fsk_phase65536; /* current phase */
+ uint16_t rx_sync; /* shift register to detect sync */
+ int rx_in_sync; /* if we are in sync and receive bits */
+ int rx_mute; /* mute count down after sync */
+ char rx_frame[141]; /* receive frame (one extra byte to terminate string) */
+ int rx_count; /* next bit to receive */
+ double rx_level[256]; /* level infos */
+ double rx_quality[256]; /* quality infos */
sample_t *frame_spl; /* samples to store a complete rendered frame */
int frame_size; /* total size of sample buffer */
int frame_length; /* current length of data in sample buffer */
int frame_pos; /* current sample position in frame_spl */
- double rx_bits_count; /* sample counter */
- double rx_bits_count_current; /* sample counter of current frame */
- double rx_bits_count_last; /* sample counter of last frame */
+ uint64_t rx_bits_count; /* sample counter */
+ uint64_t rx_bits_count_current; /* sample counter of current frame */
+ uint64_t rx_bits_count_last; /* sample counter of last frame */
int super_detected; /* current detection state flag */
int super_detect_count; /* current number of consecutive detections/losses */
@@ -149,7 +138,7 @@ int nmt_create(int nmt_system, const char *country, int channel, enum nmt_chan_t
void nmt_check_channels(int nmt_system);
void nmt_destroy(sender_t *sender);
void nmt_go_idle(nmt_t *nmt);
-void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double level, double frames_elapsed);
+void nmt_receive_frame(nmt_t *nmt, const char *bits, double quality, double level, int frames_elapsed);
const char *nmt_get_frame(nmt_t *nmt);
void nmt_rx_super(nmt_t *nmt, int tone, double quality);
void timeout_mt_paging(struct transaction *trans);
diff --git a/src/test/test_dms.c b/src/test/test_dms.c
index a35a6f1..c71f87c 100644
--- a/src/test/test_dms.c
+++ b/src/test/test_dms.c
@@ -56,14 +56,12 @@ void dms_all_sent(nmt_t *nmt)
}
/* receive bits from DMS */
-int fsk_render_frame(nmt_t *nmt, const char *frame, int length, sample_t *sample)
+void test_dms_frame(const char *frame, int length)
{
printf("(getting %d bits from DMS layer)\n", length);
memcpy(current_bits, frame, length);
current_bit_count = length;
-
- return nmt->fsk_samples_per_bit * length;
}
nmt_t *alloc_nmt(void)
@@ -71,9 +69,11 @@ nmt_t *alloc_nmt(void)
nmt_t *nmt;
nmt = calloc(sizeof(*nmt), 1);
+ nmt->sender.samplerate = 40 * 1200;
dms_init_sender(nmt);
- nmt->fsk_samples_per_bit = 40;
- nmt->dms.frame_size = nmt->fsk_samples_per_bit * 127 + 10;
+ ffsk_global_init(1.0);
+ ffsk_init(&nmt->ffsk, nmt, NULL, 1, nmt->sender.samplerate);
+ nmt->dms.frame_size = nmt->ffsk.samples_per_bit * 127 + 10;
nmt->dms.frame_spl = calloc(nmt->dms.frame_size, sizeof(nmt->dms.frame_spl[0]));
dms_reset(nmt);