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-rw-r--r--src/nmt/dsp.c260
1 files changed, 57 insertions, 203 deletions
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");
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-27 03:08:15 +0000