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NMT: Rework on FSK process, no more requirement for special sample rates

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This commit is contained in:
Andreas Eversberg
2017-01-07 06:13:31 +01:00
parent 2d0bc92cde
commit d447be5e26
6 changed files with 112 additions and 85 deletions
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14
src/nmt/dms.c
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@@ -287,10 +287,13 @@ static void dms_encode_dt(nmt_t *nmt, uint8_t d, uint8_t s, uint8_t n, uint8_t *
#endif
/* render wave form */
fsk_render_frame(nmt, frame, 127, dms->frame_spl);
dms->frame_length = fsk_render_frame(nmt, frame, 127, dms->frame_spl);
dms->frame_valid = 1;
dms->frame_pos = 0;
dms->frame_length = nmt->samples_per_bit * 127;
if (dms->frame_length > dms->frame_size) {
PDEBUG(DDMS, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n");
abort();
}
}
/* encode RR frame and schedule for next transmission */
@@ -331,10 +334,13 @@ static void dms_encode_rr(nmt_t *nmt, uint8_t d, uint8_t s, uint8_t n)
#endif
/* render wave form */
fsk_render_frame(nmt, frame, 77, dms->frame_spl);
dms->frame_length = fsk_render_frame(nmt, frame, 77, dms->frame_spl);
dms->frame_valid = 1;
dms->frame_pos = 0;
dms->frame_length = nmt->samples_per_bit * 77;
if (dms->frame_length > dms->frame_size) {
PDEBUG(DDMS, DEBUG_ERROR, "Frame exceeds buffer, please fix!\n");
abort();
}
}
/* check if we have to transmit a frame and render it

3
src/nmt/dms.h
View File

@@ -26,8 +26,9 @@ typedef struct dms {
/* DMS transmission */
int frame_valid; /* set, if there is a valid frame in sample buffer */
int16_t *frame_spl; /* 127 * fsk_bit_length */
int frame_size; /* total size of buffer */
int frame_pos; /* current sample position in frame_spl */
int frame_length; /* number of samples in frame_spl */
int frame_length; /* number of samples currently in frame_spl */
uint16_t rx_sync; /* shift register to detect sync */
double rx_sync_level[256]; /* level infos */
double rx_sync_quality[256]; /* quality infos */

142
src/nmt/dsp.c
View File

@@ -55,7 +55,7 @@
#define MUTE_DURATION 0.280 /* a tiny bit more than two frames */
/* two signaling tones */
static double fsk_bits[2] = {
static double fsk_freq[2] = {
1800.0,
1200.0,
};
@@ -70,8 +70,9 @@ static double super_freq[5] = {
};
/* table for fast sine generation */
int dsp_sine_super[256];
int dsp_sine_dialtone[256];
uint16_t dsp_tone_bit[2][2][256]; /* polarity, bit, phase */
uint16_t dsp_sine_super[256];
uint16_t dsp_sine_dialtone[256];
/* global init for FSK */
void dsp_init(void)
@@ -82,8 +83,17 @@ void dsp_init(void)
PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for supervisory signal.\n");
for (i = 0; i < 256; i++) {
s = sin((double)i / 256.0 * 2.0 * PI);
/* supervisor sine */
dsp_sine_super[i] = (int)(s * TX_PEAK_SUPER);
/* dialtone sine */
dsp_sine_dialtone[i] = (int)(s * TX_PEAK_DIALTONE);
/* bit(1) 1 cycle */
dsp_tone_bit[0][1][i] = (int)(s * TX_PEAK_FSK);
dsp_tone_bit[1][1][i] = (int)(-s * TX_PEAK_FSK);
/* bit(0) 1.5 cycles */
s = sin((double)i / 256.0 * 3.0 * PI);
dsp_tone_bit[0][0][i] = (int)(s * TX_PEAK_FSK);
dsp_tone_bit[1][0][i] = (int)(-s * TX_PEAK_FSK);
}
}
@@ -97,46 +107,24 @@ int dsp_init_sender(nmt_t *nmt)
/* attack (3ms) and recovery time (13.5ms) according to NMT specs */
init_compandor(&nmt->cstate, 8000, 3.0, 13.5, COMPANDOR_0DB);
if ((nmt->sender.samplerate % (BIT_RATE * STEPS_PER_BIT))) {
PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be a multiple of %d bits per second.\n", BIT_RATE * STEPS_PER_BIT);
return -EINVAL;
}
/* this should not happen. it is implied by previous check */
if (nmt->supervisory && nmt->sender.samplerate < 12000) {
PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 12000 Hz to process supervisory signal.\n");
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");
/* allocate sample for 2 bits with 2 polarities */
nmt->samples_per_bit = nmt->sender.samplerate / BIT_RATE;
PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per bit duration.\n", nmt->samples_per_bit);
nmt->fsk_filter_step = nmt->samples_per_bit / STEPS_PER_BIT;
PDEBUG(DDSP, DEBUG_DEBUG, "Using %d samples per filter step.\n", nmt->fsk_filter_step);
nmt->fsk_sine[0][0] = calloc(4, nmt->samples_per_bit * sizeof(int16_t));
nmt->fsk_sine[0][1] = nmt->fsk_sine[0][0] + nmt->samples_per_bit;
nmt->fsk_sine[1][0] = nmt->fsk_sine[0][1] + nmt->samples_per_bit;
nmt->fsk_sine[1][1] = nmt->fsk_sine[1][0] + nmt->samples_per_bit;
if (!nmt->fsk_sine[0][0]) {
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
return -ENOMEM;
}
PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.0f (3.5 KHz deviation @ 1500 Hz)\n", TX_PEAK_FSK);
PDEBUG(DDSP, DEBUG_DEBUG, "Using Supervisory level of %.0f (0.3 KHz deviation @ 4015 Hz)\n", TX_PEAK_SUPER);
/* generate sines */
for (i = 0; i < nmt->samples_per_bit; i++) {
nmt->fsk_sine[0][0][i] = TX_PEAK_FSK * sin(3.0 * PI * (double)i / (double)nmt->samples_per_bit); /* 1.5 waves */
nmt->fsk_sine[0][1][i] = TX_PEAK_FSK * sin(2.0 * PI * (double)i / (double)nmt->samples_per_bit); /* 1 wave */
nmt->fsk_sine[1][0][i] = -nmt->fsk_sine[0][0][i];
nmt->fsk_sine[1][1][i] = -nmt->fsk_sine[0][1][i];
}
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 */
spl = calloc(1, nmt->samples_per_bit * sizeof(*spl));
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;
@@ -144,16 +132,18 @@ int dsp_init_sender(nmt_t *nmt)
nmt->fsk_filter_spl = spl;
nmt->fsk_filter_bit = -1;
/* allocate transmit buffer for a complete frame */
spl = calloc(166, nmt->samples_per_bit * sizeof(*spl));
/* allocate transmit buffer for a complete frame, add 10 to be safe */
nmt->frame_size = 166.0 * (double)nmt->fsk_samples_per_bit + 10;
spl = calloc(nmt->frame_size, sizeof(*spl));
if (!spl) {
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
return -ENOMEM;
}
nmt->frame_spl = spl;
/* allocate DMS transmit buffer for a complete frame */
spl = calloc(127, nmt->samples_per_bit * sizeof(*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;
spl = calloc(nmt->dms.frame_size, sizeof(*spl));
if (!spl) {
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
return -ENOMEM;
@@ -171,10 +161,12 @@ int dsp_init_sender(nmt_t *nmt)
/* count symbols */
for (i = 0; i < 2; i++) {
coeff = 2.0 * cos(2.0 * PI * fsk_bits[i] / (double)nmt->sender.samplerate);
coeff = 2.0 * cos(2.0 * PI * fsk_freq[i] / (double)nmt->sender.samplerate);
nmt->fsk_coeff[i] = coeff * 32768.0;
PDEBUG(DDSP, DEBUG_DEBUG, "coeff[%d] = %d\n", i, (int)nmt->fsk_coeff[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "fsk_coeff[%d] = %d\n", i, (int)nmt->fsk_coeff[i]);
}
nmt->fsk_phaseshift256 = 256.0 / nmt->fsk_samples_per_bit;
PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", nmt->fsk_phaseshift256);
/* count supervidory tones */
for (i = 0; i < 5; i++) {
@@ -184,13 +176,14 @@ int dsp_init_sender(nmt_t *nmt)
if (i < 4) {
nmt->super_phaseshift256[i] = 256.0 / ((double)nmt->sender.samplerate / super_freq[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift_super[%d] = %.4f\n", i, nmt->super_phaseshift256[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "super_phaseshift[%d] = %.4f\n", i, nmt->super_phaseshift256[i]);
}
}
super_reset(nmt);
/* dial tone */
nmt->dial_phaseshift256 = 256.0 / ((double)nmt->sender.samplerate / DIALTONE_HZ);
PDEBUG(DDSP, DEBUG_DEBUG, "dial_phaseshift = %.4f\n", nmt->dial_phaseshift256);
/* dtmf */
dtmf_init(&nmt->dtmf, 8000);
@@ -254,8 +247,8 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
return;
/* sync time */
nmt->rx_sample_count_last = nmt->rx_sample_count_current;
nmt->rx_sample_count_current = nmt->rx_sample_count - nmt->samples_per_bit * 26;
nmt->rx_bits_count_last = nmt->rx_bits_count_current;
nmt->rx_bits_count_current = nmt->rx_bits_count - 26.0;
/* rest sync register */
nmt->fsk_filter_sync = 0;
@@ -287,7 +280,7 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
level /= 140.0; quality /= 140.0;
/* send telegramm */
frames_elapsed = (double)(nmt->rx_sample_count_current - nmt->rx_sample_count_last) / (double)(nmt->samples_per_bit * 166);
frames_elapsed = (nmt->rx_bits_count_current - nmt->rx_bits_count_last) / 166.0;
/* 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);
}
@@ -297,8 +290,8 @@ static void fsk_receive_bit(nmt_t *nmt, int bit, double quality, double level)
//#define DEBUG_QUALITY
/* Filter one chunk of audio an detect tone, quality and loss of signal.
* The chunk is a window of 10ms. This window slides over audio stream
* and is processed every 1ms. (one step) */
* 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;
@@ -306,11 +299,11 @@ static inline void fsk_decode_step(nmt_t *nmt, int pos)
int16_t *spl;
int bit;
max = nmt->samples_per_bit;
max = nmt->fsk_filter_size;
spl = nmt->fsk_filter_spl;
/* count time in samples*/
nmt->rx_sample_count += nmt->fsk_filter_step;
/* count time in bits */
nmt->rx_bits_count += 0.1;
level = audio_level(spl, max);
/* limit level to prevent division by zero */
@@ -430,7 +423,8 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
{
nmt_t *nmt = (nmt_t *) sender;
int16_t *spl;
int max, pos, step;
int max, pos;
double step, bps;
int i;
/* write received samples to decode buffer */
@@ -448,26 +442,32 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
nmt->super_filter_pos = pos;
/* write received samples to decode buffer */
max = nmt->samples_per_bit;
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;
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 (nmt->fsk_filter_mute) {
if (pos == max)
pos = 0;
/* muting audio while receiving frame */
if (nmt->fsk_filter_mute && !nmt->sender.loopback) {
samples[i] = 0;
nmt->fsk_filter_mute--;
}
if (pos == max)
pos = 0;
/* if filter step has been reched */
if (!(pos % step)) {
/* if 1/10th of a bit duration is reached, decode buffer */
step += bps;
if (step >= 0.1) {
step -= 0.1;
fsk_decode_step(nmt, pos);
}
}
nmt->fsk_filter_step = step;
nmt->fsk_filter_pos = pos;
if ((nmt->dsp_mode == DSP_MODE_AUDIO || nmt->dsp_mode == DSP_MODE_DTMF)
@@ -495,21 +495,32 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
}
/* render frame */
void fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
int fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
{
int bit, polarity;
int i;
double phaseshift, phase;
int count = 0, i;
polarity = nmt->fsk_polarity;
phaseshift = nmt->fsk_phaseshift256;
phase = nmt->fsk_phase256;
for (i = 0; i < length; i++) {
bit = (frame[i] == '1');
memcpy(sample, nmt->fsk_sine[polarity][bit], nmt->samples_per_bit * sizeof(*sample));
sample += nmt->samples_per_bit;
do {
*sample++ = dsp_tone_bit[polarity][bit][((uint8_t)phase) & 0xff];
count++;
phase += phaseshift;
} while (phase < 256.0);
phase -= 256.0;
/* flip polarity when we have 1.5 sine waves */
if (bit == 0)
polarity = 1 - polarity;
}
nmt->fsk_phase256 = phase;
nmt->fsk_polarity = polarity;
/* return number of samples created for frame */
return count;
}
static int fsk_frame(nmt_t *nmt, int16_t *samples, int length)
@@ -520,21 +531,24 @@ static int fsk_frame(nmt_t *nmt, int16_t *samples, int length)
int count, max;
next_frame:
if (!nmt->frame) {
if (!nmt->frame_length) {
/* request frame */
frame = nmt_get_frame(nmt);
if (!frame) {
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Stop sending frames.\n");
return length;
}
nmt->frame = 1;
nmt->frame_pos = 0;
/* render frame */
fsk_render_frame(nmt, frame, 166, nmt->frame_spl);
nmt->frame_length = fsk_render_frame(nmt, 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");
abort();
}
}
/* send audio from frame */
max = nmt->samples_per_bit * 166;
max = nmt->frame_length;
count = max - nmt->frame_pos;
if (count > length)
count = length;
@@ -546,7 +560,7 @@ next_frame:
nmt->frame_pos += count;
/* check for end of telegramm */
if (nmt->frame_pos == max) {
nmt->frame = 0;
nmt->frame_length = 0;
/* we need more ? */
if (length)
goto next_frame;
@@ -666,7 +680,7 @@ void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode)
{
/* reset telegramm */
if (mode == DSP_MODE_FRAME && nmt->dsp_mode != mode)
nmt->frame = 0;
nmt->frame_length = 0;
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "DSP mode %s -> %s\n", nmt_dsp_mode_name(nmt->dsp_mode), nmt_dsp_mode_name(mode));
nmt->dsp_mode = mode;

2
src/nmt/dsp.h
View File

@@ -2,7 +2,7 @@
void dsp_init(void);
int dsp_init_sender(nmt_t *nmt);
void dsp_cleanup_sender(nmt_t *nmt);
void fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample);
int fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample);
void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode);
void super_reset(nmt_t *nmt);

27
src/nmt/nmt.h
View File

@@ -91,17 +91,17 @@ typedef struct nmt {
/* dsp states */
enum dsp_mode dsp_mode; /* current mode: audio, durable tone 0 or 1, paging */
int samples_per_bit; /* number of samples for one bit (1200 Baud) */
double fsk_samples_per_bit; /* number of samples for one bit (1200 Baud) */
double fsk_bits_per_sample; /* fraction of a bit per sample */
int super_samples; /* number of samples in buffer for supervisory detection */
int fsk_coeff[2]; /* coefficient k = 2*cos(2*PI*f/samplerate), k << 15 */
int super_coeff[5]; /* coefficient for supervisory signal */
int16_t *fsk_sine[2][2]; /* 4 pointers to 4 precalc. sine curves */
int fsk_polarity; /* current polarity state of bit */
int samples_per_chunk; /* how many samples lasts one chunk */
int16_t *fsk_filter_spl; /* array with samples_per_chunk */
int fsk_filter_pos; /* current sample position in filter_spl */
int fsk_filter_step; /* number of samples for each analyzation */
int fsk_filter_bit; /* last bit, so we detect a bit change */
int16_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 */
@@ -116,12 +116,15 @@ typedef struct nmt {
double super_phase256; /* current phase */
double dial_phaseshift256; /* how much the phase of sine wave changes per sample */
double dial_phase256; /* current phase */
int frame; /* set, if there is a valid frame */
int16_t *frame_spl; /* 166 * fsk_bit_length */
double fsk_phaseshift256; /* how much the phase of fsk synbol changes per sample */
double fsk_phase256; /* current phase */
int16_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 */
uint64_t rx_sample_count; /* sample counter */
uint64_t rx_sample_count_current;/* sample counter of current frame */
uint64_t rx_sample_count_last; /* sample counter of last frame */
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 */
int super_detected; /* current detection state flag */
int super_detect_count; /* current number of consecutive detections/losses */

9
src/test/test_dms.c
View File

@@ -55,12 +55,14 @@ void dms_all_sent(nmt_t *nmt)
}
/* receive bits from DMS */
void fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
int fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
{
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)
@@ -69,8 +71,9 @@ nmt_t *alloc_nmt(void)
nmt = calloc(sizeof(*nmt), 1);
dms_init_sender(nmt);
nmt->dms.frame_spl = calloc(1000000, 1);
nmt->samples_per_bit = 40;
nmt->fsk_samples_per_bit = 40;
nmt->dms.frame_size = nmt->fsk_samples_per_bit * 127 + 10;
nmt->dms.frame_spl = calloc(nmt->dms.frame_size, sizeof(nmt->dms.frame_spl[0]));
dms_reset(nmt);