Split FSK modem code into separate modulator and demodulator

This commit is contained in:
Andreas Eversberg 2019-09-01 15:38:16 +02:00
parent 360729d270
commit 6dcc8baad4
9 changed files with 213 additions and 137 deletions

View File

@ -87,7 +87,8 @@ typedef struct bnetz {
/* dsp states */
enum dsp_mode dsp_mode; /* current mode: audio, durable tone 0 or 1, "Telegramm" */
fsk_t fsk; /* fsk modem instance */
fsk_mod_t fsk_mod; /* fsk modem instance */
fsk_demod_t fsk_demod;
uint16_t rx_telegramm; /* rx shift register for receiveing telegramm */
double rx_telegramm_quality[16];/* quality of each bit in telegramm */
double rx_telegramm_level[16]; /* level of each bit in telegramm */

View File

@ -61,7 +61,7 @@ int tx_telegramm_pos = 0;
int latspl;
/* instances */
fsk_t fsk;
fsk_mod_t fsk_mod;
#ifdef HAVE_ALSA
void *audio = NULL;
#endif
@ -226,7 +226,7 @@ again:
break;
case TX_MODE_FSK:
/* send FSK until it stops, then fill with silence */
count = fsk_send(&fsk, samples, length, 0);
count = fsk_mod_send(&fsk_mod, samples, length, 0);
samples += count;
length -= count;
if (length)
@ -286,7 +286,7 @@ int main(int argc, char *argv[])
/* init */
bnetz_init_telegramm();
memset(&fsk, 0, sizeof(fsk));
memset(&fsk_mod, 0, sizeof(fsk_mod));
/* latency of send buffer in samples */
latspl = samplerate * latency / 1000;
@ -340,7 +340,7 @@ int main(int argc, char *argv[])
sprintf(funkwahl, "wwww%c%s%se%c%s%se", start_digit, station_id, dialing + 1, start_digit, station_id, dialing + 1);
/* init fsk */
if (fsk_init(&fsk, NULL, fsk_send_bit, NULL, samplerate, BIT_RATE, F0, F1, 1.0, 0, 0) < 0) {
if (fsk_mod_init(&fsk_mod, NULL, fsk_send_bit, samplerate, BIT_RATE, F0, F1, 1.0, 0) < 0) {
PDEBUG(DDSP, DEBUG_ERROR, "FSK init failed!\n");
goto exit;
}
@ -389,7 +389,7 @@ exit:
#endif
/* exit fsk */
fsk_cleanup(&fsk);
fsk_mod_cleanup(&fsk_mod);
return 0;
}

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@ -96,7 +96,11 @@ int dsp_init_sender(bnetz_t *bnetz, double squelch_db)
PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.3f (%.3f KHz deviation @ 2000 Hz)\n", TX_PEAK_FSK, 4.0);
/* init fsk */
if (fsk_init(&bnetz->fsk, bnetz, fsk_send_bit, fsk_receive_bit, bnetz->sender.samplerate, BIT_RATE, F0, F1, TX_PEAK_FSK, 0, BIT_ADJUST) < 0) {
if (fsk_mod_init(&bnetz->fsk_mod, bnetz, fsk_send_bit, bnetz->sender.samplerate, BIT_RATE, F0, F1, TX_PEAK_FSK, 0) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
if (fsk_demod_init(&bnetz->fsk_demod, bnetz, fsk_receive_bit, bnetz->sender.samplerate, BIT_RATE, F0, F1, BIT_ADJUST) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
@ -121,7 +125,8 @@ void dsp_cleanup_sender(bnetz_t *bnetz)
{
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for 'Sender'.\n");
fsk_cleanup(&bnetz->fsk);
fsk_mod_cleanup(&bnetz->fsk_mod);
fsk_demod_cleanup(&bnetz->fsk_demod);
}
/* If good tone is received, we just set this tone, if not already and reset counters.
@ -270,7 +275,7 @@ void sender_receive(sender_t *sender, sample_t *samples, int length, double rf_l
int i;
/* fsk/tone signal */
fsk_receive(&bnetz->fsk, samples, length);
fsk_demod_receive(&bnetz->fsk_demod, samples, length);
/* process signal mute/loss, without signalling tone / FSK frames */
switch (squelch(&bnetz->squelch, rf_level_db, (double)length / (double)bnetz->sender.samplerate)) {
@ -375,7 +380,7 @@ again:
case DSP_MODE_TELEGRAMM:
/* Encode tone/frame into audio stream. If frames have
* stopped, process again for rest of stream. */
count = fsk_send(&bnetz->fsk, samples, length, 0);
count = fsk_mod_send(&bnetz->fsk_mod, samples, length, 0);
samples += count;
length -= count;
if (length)
@ -412,7 +417,7 @@ void bnetz_set_dsp_mode(bnetz_t *bnetz, enum dsp_mode mode)
/* reset telegramm */
if (mode == DSP_MODE_TELEGRAMM && bnetz->dsp_mode != mode) {
bnetz->tx_telegramm = 0;
fsk_tx_reset(&bnetz->fsk);
fsk_mod_tx_reset(&bnetz->fsk_mod);
}
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "DSP mode %s -> %s\n", bnetz_dsp_mode_name(bnetz->dsp_mode), bnetz_dsp_mode_name(mode));

View File

@ -33,21 +33,18 @@
* fsk = instance of fsk modem
* inst = instance of user
* send_bit() = function to be called whenever a new bit has to be sent
* receive_bit() = function to be called whenever a new bit was received
* samplerate = samplerate
* bitrate = bits per second
* f0, f1 = two frequencies for bit 0 and bit 1
* level = level to modulate the frequencies
* coherent = use coherent modulation (FFSK)
* bitadjust = how much to adjust the sample clock when a bitchange was detected. (0 = nothing, don't use this, 0.5 full adjustment)
*/
int fsk_init(fsk_t *fsk, void *inst, int (*send_bit)(void *inst), void (*receive_bit)(void *inst, int bit, double quality, double level), int samplerate, double bitrate, double f0, double f1, double level, int coherent, double bitadjust)
int fsk_mod_init(fsk_mod_t *fsk, void *inst, int (*send_bit)(void *inst), int samplerate, double bitrate, double f0, double f1, double level, int coherent)
{
double bandwidth;
int i;
int rc;
PDEBUG(DDSP, DEBUG_DEBUG, "Setup FSK for Transceiver. (F0 = %.1f, F1 = %.1f, peak = %.1f)\n", f0, f1, level);
PDEBUG(DDSP, DEBUG_DEBUG, "Setup FSK for Transmitter. (F0 = %.1f, F1 = %.1f, peak = %.1f)\n", f0, f1, level);
memset(fsk, 0, sizeof(*fsk));
@ -62,9 +59,6 @@ int fsk_init(fsk_t *fsk, void *inst, int (*send_bit)(void *inst), void (*receive
fsk->sin_tab[i] = sin((double)i / 65536.0 * 2.0 * PI) * level;
fsk->inst = inst;
fsk->rx_bit = -1;
fsk->rx_bitadjust = bitadjust;
fsk->receive_bit = receive_bit;
fsk->tx_bit = -1;
fsk->level = level;
fsk->send_bit = send_bit;
@ -78,14 +72,6 @@ int fsk_init(fsk_t *fsk, void *inst, int (*send_bit)(void *inst), void (*receive
fsk->high_bit = 0;
}
/* calculate bandwidth */
bandwidth = fabs(f0 - f1) * 2.0;
/* init fm demodulator */
rc = fm_demod_init(&fsk->demod, (double)samplerate, (f0 + f1) / 2.0, bandwidth);
if (rc < 0)
goto error;
fsk->bits_per_sample = (double)bitrate / (double)samplerate;
PDEBUG(DDSP, DEBUG_DEBUG, "Bitduration of %.4f bits per sample @ %d.\n", fsk->bits_per_sample, samplerate);
@ -118,19 +104,150 @@ int fsk_init(fsk_t *fsk, void *inst, int (*send_bit)(void *inst), void (*receive
return 0;
error:
fsk_cleanup(fsk);
fsk_mod_cleanup(fsk);
return rc;
}
/* Cleanup transceiver instance. */
void fsk_cleanup(fsk_t *fsk)
void fsk_mod_cleanup(fsk_mod_t *fsk)
{
PDEBUG(DDSP, DEBUG_DEBUG, "Cleanup FSK for Transceiver.\n");
PDEBUG(DDSP, DEBUG_DEBUG, "Cleanup FSK for Transmitter.\n");
if (fsk->sin_tab) {
free(fsk->sin_tab);
fsk->sin_tab = NULL;
}
}
/* modulate bits
*
* If first/next bit is required, callback function send_bit() is called.
* If there is no (more) data to be transmitted, the callback functions shall
* return -1. In this case, this function stops and returns the number of
* samples that have been rendered so far, if any.
*
* For coherent mode (FSK), we round the phase on every bit change to the
* next zero crossing. This prevents phase shifts due to rounding errors.
*/
int fsk_mod_send(fsk_mod_t *fsk, sample_t *sample, int length, int add)
{
int count = 0;
double phase, phaseshift;
phase = fsk->tx_phase65536;
/* get next bit */
if (fsk->tx_bit < 0) {
next_bit:
fsk->tx_bit = fsk->send_bit(fsk->inst);
#ifdef DEBUG_MODULATOR
printf("bit change to %d\n", fsk->tx_bit);
#endif
if (fsk->tx_bit < 0)
goto done;
/* correct phase when changing bit */
if (fsk->coherent) {
/* round phase to nearest zero crossing */
if (phase > 16384.0 && phase < 49152.0)
phase = 32768.0;
else
phase = 0;
/* set phase according to current position in bit */
phase += fsk->tx_bitpos * fsk->cycles_per_bit65536[fsk->tx_bit & 1];
#ifdef DEBUG_MODULATOR
printf("phase %.3f bitpos=%.6f\n", phase, fsk->tx_bitpos);
#endif
}
}
/* modulate bit */
phaseshift = fsk->phaseshift65536[fsk->tx_bit & 1];
while (count < length && fsk->tx_bitpos < 1.0) {
if (add)
sample[count++] += fsk->sin_tab[(uint16_t)phase];
else
sample[count++] = fsk->sin_tab[(uint16_t)phase];
#ifdef DEBUG_MODULATOR
printf("|%s|\n", debug_amplitude(fsk->sin_tab[(uint16_t)phase] / fsk->level));
#endif
phase += phaseshift;
if (phase >= 65536.0)
phase -= 65536.0;
fsk->tx_bitpos += fsk->bits_per_sample;
}
if (fsk->tx_bitpos >= 1.0) {
fsk->tx_bitpos -= 1.0;
goto next_bit;
}
done:
fsk->tx_phase65536 = phase;
return count;
}
/* reset transmitter state, so we get a clean start */
void fsk_mod_tx_reset(fsk_mod_t *fsk)
{
fsk->tx_phase65536 = 0;
fsk->tx_bitpos = 0;
fsk->tx_bit = -1;
}
/*
* fsk = instance of fsk modem
* inst = instance of user
* receive_bit() = function to be called whenever a new bit was received
* samplerate = samplerate
* bitrate = bits per second
* f0, f1 = two frequencies for bit 0 and bit 1
* bitadjust = how much to adjust the sample clock when a bitchange was detected. (0 = nothing, don't use this, 0.5 full adjustment)
*/
int fsk_demod_init(fsk_demod_t *fsk, void *inst, void (*receive_bit)(void *inst, int bit, double quality, double level), int samplerate, double bitrate, double f0, double f1, double bitadjust)
{
double bandwidth;
int rc;
PDEBUG(DDSP, DEBUG_DEBUG, "Setup FSK for Receiver. (F0 = %.1f, F1 = %.1f)\n", f0, f1);
memset(fsk, 0, sizeof(*fsk));
fsk->inst = inst;
fsk->rx_bit = -1;
fsk->rx_bitadjust = bitadjust;
fsk->receive_bit = receive_bit;
fsk->f0_deviation = (f0 - f1) / 2.0;
fsk->f1_deviation = (f1 - f0) / 2.0;
if (f0 < f1) {
fsk->low_bit = 0;
fsk->high_bit = 1;
} else {
fsk->low_bit = 1;
fsk->high_bit = 0;
}
/* calculate bandwidth */
bandwidth = fabs(f0 - f1) * 2.0;
/* init fm demodulator */
rc = fm_demod_init(&fsk->demod, (double)samplerate, (f0 + f1) / 2.0, bandwidth);
if (rc < 0)
goto error;
fsk->bits_per_sample = (double)bitrate / (double)samplerate;
PDEBUG(DDSP, DEBUG_DEBUG, "Bitduration of %.4f bits per sample @ %d.\n", fsk->bits_per_sample, samplerate);
return 0;
error:
fsk_demod_cleanup(fsk);
return rc;
}
/* Cleanup transceiver instance. */
void fsk_demod_cleanup(fsk_demod_t *fsk)
{
PDEBUG(DDSP, DEBUG_DEBUG, "Cleanup FSK for Receiver.\n");
fm_demod_exit(&fsk->demod);
}
@ -150,7 +267,7 @@ void fsk_cleanup(fsk_t *fsk)
* Therefore we change the sample counter only slightly, so bit slips may not
* happen so quickly.
*/
void fsk_receive(fsk_t *fsk, sample_t *sample, int length)
void fsk_demod_receive(fsk_demod_t *fsk, sample_t *sample, int length)
{
sample_t I[length], Q[length], frequency[length], f;
int i;
@ -211,78 +328,3 @@ void fsk_receive(fsk_t *fsk, sample_t *sample, int length)
}
}
/* modulate bits
*
* If first/next bit is required, callback function send_bit() is called.
* If there is no (more) data to be transmitted, the callback functions shall
* return -1. In this case, this function stops and returns the number of
* samples that have been rendered so far, if any.
*
* For coherent mode (FSK), we round the phase on every bit change to the
* next zero crossing. This prevents phase shifts due to rounding errors.
*/
int fsk_send(fsk_t *fsk, sample_t *sample, int length, int add)
{
int count = 0;
double phase, phaseshift;
phase = fsk->tx_phase65536;
/* get next bit */
if (fsk->tx_bit < 0) {
next_bit:
fsk->tx_bit = fsk->send_bit(fsk->inst);
#ifdef DEBUG_MODULATOR
printf("bit change to %d\n", fsk->tx_bit);
#endif
if (fsk->tx_bit < 0)
goto done;
/* correct phase when changing bit */
if (fsk->coherent) {
/* round phase to nearest zero crossing */
if (phase > 16384.0 && phase < 49152.0)
phase = 32768.0;
else
phase = 0;
/* set phase according to current position in bit */
phase += fsk->tx_bitpos * fsk->cycles_per_bit65536[fsk->tx_bit & 1];
#ifdef DEBUG_MODULATOR
printf("phase %.3f bitpos=%.6f\n", phase, fsk->tx_bitpos);
#endif
}
}
/* modulate bit */
phaseshift = fsk->phaseshift65536[fsk->tx_bit & 1];
while (count < length && fsk->tx_bitpos < 1.0) {
if (add)
sample[count++] += fsk->sin_tab[(uint16_t)phase];
else
sample[count++] = fsk->sin_tab[(uint16_t)phase];
#ifdef DEBUG_MODULATOR
printf("|%s|\n", debug_amplitude(fsk->sin_tab[(uint16_t)phase] / fsk->level));
#endif
phase += phaseshift;
if (phase >= 65536.0)
phase -= 65536.0;
fsk->tx_bitpos += fsk->bits_per_sample;
}
if (fsk->tx_bitpos >= 1.0) {
fsk->tx_bitpos -= 1.0;
goto next_bit;
}
done:
fsk->tx_phase65536 = phase;
return count;
}
/* reset transmitter state, so we get a clean start */
void fsk_tx_reset(fsk_t *fsk)
{
fsk->tx_phase65536 = 0;
fsk->tx_bitpos = 0;
fsk->tx_bit = -1;
}

View File

@ -1,14 +1,12 @@
#include "../libfm/fm.h"
typedef struct ffsk {
typedef struct fsk_mod {
void *inst;
int (*send_bit)(void *inst);
void (*receive_bit)(void *inst, int bit, double quality, double level);
fm_demod_t demod;
double bits_per_sample; /* fraction of a bit per sample */
double *sin_tab; /* sine table with correct peak level */
double phaseshift65536[2]; /* how much the phase of fsk synbol changes per sample */
double cycles_per_bit65536[2]; /* cacles of one bit */
double cycles_per_bit65536[2]; /* cycles of one bit */
double tx_phase65536; /* current transmit phase */
double level; /* level (amplitude) of signal */
int coherent; /* set, if coherent TX mode */
@ -16,15 +14,27 @@ typedef struct ffsk {
double f1_deviation;
int low_bit, high_bit; /* a low or high deviation means which bit? */
int tx_bit; /* current transmitting bit (-1 if not set) */
int rx_bit; /* current receiving bit (-1 if not yet measured) */
double tx_bitpos; /* current transmit position in bit */
} fsk_mod_t;
typedef struct fsk_demod {
void *inst;
void (*receive_bit)(void *inst, int bit, double quality, double level);
fm_demod_t demod;
double bits_per_sample; /* fraction of a bit per sample */
double f0_deviation; /* deviation of frequencies, relative to center */
double f1_deviation;
int low_bit, high_bit; /* a low or high deviation means which bit? */
int rx_bit; /* current receiving bit (-1 if not yet measured) */
double rx_bitpos; /* current receive position in bit (sampleclock) */
double rx_bitadjust; /* how much does a bit change cause the sample clock to be adjusted in phase */
} fsk_t;
} fsk_demod_t;
int fsk_init(fsk_t *fsk, void *inst, int (*send_bit)(void *inst), void (*receive_bit)(void *inst, int bit, double quality, double level), int samplerate, double bitrate, double f0, double f1, double level, int coherent, double bitadjust);
void fsk_cleanup(fsk_t *fsk);
void fsk_receive(fsk_t *fsk, sample_t *sample, int length);
int fsk_send(fsk_t *fsk, sample_t *sample, int length, int add);
void fsk_tx_reset(fsk_t *fsk);
int fsk_mod_init(fsk_mod_t *fsk, void *inst, int (*send_bit)(void *inst), int samplerate, double bitrate, double f0, double f1, double level, int coherent);
void fsk_mod_cleanup(fsk_mod_t *fsk);
int fsk_mod_send(fsk_mod_t *fsk, sample_t *sample, int length, int add);
void fsk_mod_tx_reset(fsk_mod_t *fsk);
int fsk_demod_init(fsk_demod_t *fsk, void *inst, void (*receive_bit)(void *inst, int bit, double quality, double level), int samplerate, double bitrate, double f0, double f1, double bitadjust);
void fsk_demod_cleanup(fsk_demod_t *fsk);
void fsk_demod_receive(fsk_demod_t *fsk, sample_t *sample, int length);

View File

@ -120,7 +120,11 @@ int dsp_init_sender(nmt_t *nmt, double 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);
/* init fsk */
if (fsk_init(&nmt->fsk, nmt, fsk_send_bit, fsk_receive_bit, nmt->sender.samplerate, BIT_RATE, F0, F1, TX_PEAK_FSK, 1, BIT_ADJUST) < 0) {
if (fsk_mod_init(&nmt->fsk_mod, nmt, fsk_send_bit, nmt->sender.samplerate, BIT_RATE, F0, F1, TX_PEAK_FSK, 1) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
if (fsk_demod_init(&nmt->fsk_demod, nmt, fsk_receive_bit, nmt->sender.samplerate, BIT_RATE, F0, F1, BIT_ADJUST) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
@ -163,7 +167,8 @@ void dsp_cleanup_sender(nmt_t *nmt)
{
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for Transceiver.\n");
fsk_cleanup(&nmt->fsk);
fsk_mod_cleanup(&nmt->fsk_mod);
fsk_demod_cleanup(&nmt->fsk_demod);
if (nmt->super_filter_spl) {
free(nmt->super_filter_spl);
@ -336,7 +341,7 @@ void sender_receive(sender_t *sender, sample_t *samples, int length, double __at
nmt->super_filter_pos = pos;
/* fsk signal */
fsk_receive(&nmt->fsk, samples, length);
fsk_demod_receive(&nmt->fsk_demod, samples, length);
/* muting audio while receiving frame */
for (i = 0; i < length; i++) {
@ -449,7 +454,7 @@ again:
jitter_load(&nmt->sender.dejitter, samples, length);
/* send after dejitter, so audio is flushed */
if (nmt->dms.tx_frame_valid) {
fsk_send(&nmt->fsk, samples, length, 0);
fsk_mod_send(&nmt->fsk_mod, samples, length, 0);
break;
}
if (nmt->supervisory)
@ -464,7 +469,7 @@ again:
case DSP_MODE_FRAME:
/* Encode frame into audio stream. If frames have
* stopped, process again for rest of stream. */
count = fsk_send(&nmt->fsk, samples, length, 0);
count = fsk_mod_send(&nmt->fsk_mod, samples, length, 0);
/* special case: add supervisory signal to frame at loop test */
if (nmt->sender.loopback && nmt->supervisory)
super_encode(nmt, samples, count);
@ -501,7 +506,7 @@ void nmt_set_dsp_mode(nmt_t *nmt, enum dsp_mode mode)
{
/* reset frame */
if (mode == DSP_MODE_FRAME && nmt->dsp_mode != mode) {
fsk_tx_reset(&nmt->fsk);
fsk_mod_tx_reset(&nmt->fsk_mod);
nmt->tx_frame_length = 0;
}

View File

@ -104,7 +104,8 @@ typedef struct nmt {
/* dsp states */
enum dsp_mode dsp_mode; /* current mode: audio, durable tone 0 or 1, paging */
fsk_t fsk; /* fsk processing */
fsk_mod_t fsk_mod; /* fsk processing */
fsk_demod_t fsk_demod;
int super_samples; /* number of samples in buffer for supervisory detection */
goertzel_t super_goertzel[5]; /* filter for supervisory decoding */
sample_t *super_filter_spl; /* array with sample buffer for supervisory detection */

View File

@ -85,7 +85,11 @@ int dsp_init_sender(r2000_t *r2000)
PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.3f\n", TX_PEAK_FSK);
/* init fsk */
if (fsk_init(&r2000->fsk, r2000, fsk_send_bit, fsk_receive_bit, r2000->sender.samplerate, FSK_BIT_RATE, FSK_F0, FSK_F1, TX_PEAK_FSK, 1, FSK_BIT_ADJUST) < 0) {
if (fsk_mod_init(&r2000->fsk_mod, r2000, fsk_send_bit, r2000->sender.samplerate, FSK_BIT_RATE, FSK_F0, FSK_F1, TX_PEAK_FSK, 1) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
if (fsk_demod_init(&r2000->fsk_demod, r2000, fsk_receive_bit, r2000->sender.samplerate, FSK_BIT_RATE, FSK_F0, FSK_F1, FSK_BIT_ADJUST) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
@ -95,7 +99,11 @@ int dsp_init_sender(r2000_t *r2000)
r2000->rx_max = 144;
/* init supervisorty fsk */
if (fsk_init(&r2000->super_fsk, r2000, super_send_bit, super_receive_bit, r2000->sender.samplerate, SUPER_BIT_RATE, SUPER_F0, SUPER_F1, TX_PEAK_SUPER, 0, SUPER_BIT_ADJUST) < 0) {
if (fsk_mod_init(&r2000->super_fsk_mod, r2000, super_send_bit, r2000->sender.samplerate, SUPER_BIT_RATE, SUPER_F0, SUPER_F1, TX_PEAK_SUPER, 0) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
if (fsk_demod_init(&r2000->super_fsk_demod, r2000, super_receive_bit, r2000->sender.samplerate, SUPER_BIT_RATE, SUPER_F0, SUPER_F1, SUPER_BIT_ADJUST) < 0) {
PDEBUG_CHAN(DDSP, DEBUG_ERROR, "FSK init failed!\n");
return -EINVAL;
}
@ -122,8 +130,10 @@ void dsp_cleanup_sender(r2000_t *r2000)
{
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Cleanup DSP for Transceiver.\n");
fsk_cleanup(&r2000->fsk);
fsk_cleanup(&r2000->super_fsk);
fsk_mod_cleanup(&r2000->fsk_mod);
fsk_demod_cleanup(&r2000->fsk_demod);
fsk_mod_cleanup(&r2000->super_fsk_mod);
fsk_demod_cleanup(&r2000->super_fsk_demod);
}
/* Check for SYNC bits, then collect data bits */
@ -254,14 +264,14 @@ void sender_receive(sender_t *sender, sample_t *samples, int length, double __at
if (r2000->dsp_mode == DSP_MODE_AUDIO_TX
|| r2000->dsp_mode == DSP_MODE_AUDIO_TX_RX
|| r2000->sender.loopback)
fsk_receive(&r2000->super_fsk, samples, length);
fsk_demod_receive(&r2000->super_fsk_demod, samples, length);
/* do de-emphasis */
if (r2000->de_emphasis)
de_emphasis(&r2000->estate, samples, length);
/* fsk signal */
fsk_receive(&r2000->fsk, samples, length);
fsk_demod_receive(&r2000->fsk_demod, samples, length);
/* we must have audio mode for both ways and a call */
if (r2000->dsp_mode == DSP_MODE_AUDIO_TX_RX
@ -342,19 +352,19 @@ again:
if (r2000->pre_emphasis)
pre_emphasis(&r2000->estate, samples, length);
/* add supervisory to sample buffer */
fsk_send(&r2000->super_fsk, samples, length, 1);
fsk_mod_send(&r2000->super_fsk_mod, samples, length, 1);
break;
case DSP_MODE_FRAME:
/* Encode frame into audio stream. If frames have
* stopped, process again for rest of stream. */
count = fsk_send(&r2000->fsk, samples, length, 0);
count = fsk_mod_send(&r2000->fsk_mod, samples, length, 0);
/* do pre-emphasis */
if (r2000->pre_emphasis)
pre_emphasis(&r2000->estate, samples, count);
/* special case: add supervisory signal to frame at loop test */
if (r2000->sender.loopback) {
/* add supervisory to sample buffer */
fsk_send(&r2000->super_fsk, samples, count, 1);
fsk_mod_send(&r2000->super_fsk_mod, samples, count, 1);
}
memset(power, 1, count);
samples += count;
@ -390,12 +400,12 @@ void r2000_set_dsp_mode(r2000_t *r2000, enum dsp_mode mode, int super)
/* reset telegramm */
if (mode == DSP_MODE_FRAME && r2000->dsp_mode != mode) {
r2000->tx_frame_length = 0;
fsk_tx_reset(&r2000->fsk);
fsk_mod_tx_reset(&r2000->fsk_mod);
}
if ((mode == DSP_MODE_AUDIO_TX || mode == DSP_MODE_AUDIO_TX_RX)
&& (r2000->dsp_mode != DSP_MODE_AUDIO_TX && r2000->dsp_mode != DSP_MODE_AUDIO_TX_RX)) {
r2000->super_tx_word_length = 0;
fsk_tx_reset(&r2000->super_fsk);
fsk_mod_tx_reset(&r2000->super_fsk_mod);
}
if (super >= 0) {

View File

@ -85,7 +85,8 @@ typedef struct r2000 {
/* dsp states */
enum dsp_mode dsp_mode; /* current mode: audio, durable tone 0 or 1, paging */
fsk_t fsk; /* fsk processing */
fsk_mod_t fsk_mod; /* fsk processing */
fsk_demod_t fsk_demod;
char tx_frame[208]; /* carries bits of one frame to transmit */
int tx_frame_length;
int tx_frame_pos;
@ -102,7 +103,8 @@ typedef struct r2000 {
uint64_t rx_bits_count_last; /* sample counter of last frame */
/* supervisory dsp states */
fsk_t super_fsk; /* fsk processing */
fsk_mod_t super_fsk_mod; /* fsk processing */
fsk_demod_t super_fsk_demod;
uint32_t super_tx_word; /* supervisory info to transmit */
int super_tx_word_length;
int super_tx_word_pos;