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Correcting all levels and move all remaining integer samples to sample_t

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The leves are based on the standards of each mobile network. They
are adjusted to the specified frequency deviation now.
This commit is contained in:
Andreas Eversberg
2017-01-29 07:25:12 +01:00
parent bd7ccc5fa0
commit 7e45f556ce
38 changed files with 484 additions and 397 deletions
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64
src/bnetz/dsp.c
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@@ -34,9 +34,19 @@
#define PI 3.1415927
/* Notes on TX_PEAK_TONE level:
*
* At 2000 Hz the deviation shall be 4 kHz, so with emphasis the deviation
* at 1000 Hz would be theoretically 2 kHz. This is factor 0.714 below
* 2.8 kHz deviation we want at dBm0.
*/
/* signaling */
#define BANDWIDTH 5000.0 /* maximum bandwidth */
#define TX_PEAK_TONE 5000.0 /* peak amplitude for all tones */
#define MAX_DEVIATION 4000.0
#define MAX_MODULATION 3000.0
#define DBM0_DEVIATION 2800.0 /* deviation of dBm0 at 1 kHz */
#define TX_PEAK_TONE (4000.0 / 2000.0 * 1000.0 / DBM0_DEVIATION)
#define MAX_DISPLAY 1.4 /* something above dBm0 */
#define BIT_DURATION 0.010 /* bit length: 10 ms */
#define FILTER_STEP 0.001 /* step every 1 ms */
#define METERING_HZ 2900 /* metering pulse frequency */
@@ -54,7 +64,7 @@ static double fsk_bits[2] = {
};
/* table for fast sine generation */
static sample_t dsp_sine[256];
static sample_t dsp_sine[65536];
/* global init for FSK */
void dsp_init(void)
@@ -62,13 +72,8 @@ void dsp_init(void)
int i;
PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table.\n");
for (i = 0; i < 256; i++) {
dsp_sine[i] = (int)(sin((double)i / 256.0 * 2.0 * PI) * TX_PEAK_TONE);
}
if (TX_PEAK_TONE > 32767.0) {
fprintf(stderr, "TX_PEAK_TONE definition too high, please fix!\n");
abort();
for (i = 0; i < 65536; i++) {
dsp_sine[i] = sin((double)i / 65536.0 * 2.0 * PI) * TX_PEAK_TONE;
}
}
@@ -85,9 +90,8 @@ int dsp_init_sender(bnetz_t *bnetz)
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for 'Sender'.\n");
/* set deviation and modulation parameters */
bnetz->sender.bandwidth = BANDWIDTH;
bnetz->sender.sample_deviation = 1250.0 / (double)TX_PEAK_TONE; // FIXME: calc real value
/* set modulation parameters */
sender_set_fm(&bnetz->sender, MAX_DEVIATION, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
audio_init_loss(&bnetz->sender.loss, LOSS_INTERVAL, bnetz->sender.loss_volume, LOSS_TIME);
@@ -114,8 +118,8 @@ int dsp_init_sender(bnetz_t *bnetz)
/* count symbols */
for (i = 0; i < 2; i++) {
audio_goertzel_init(&bnetz->fsk_goertzel[i], fsk_bits[i], bnetz->sender.samplerate);
bnetz->phaseshift256[i] = 256.0 / ((double)bnetz->sender.samplerate / fsk_bits[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift[%d] = %.4f (must be arround 64 at 8000hz)\n", i, bnetz->phaseshift256[i]);
bnetz->phaseshift65536[i] = 65536.0 / ((double)bnetz->sender.samplerate / fsk_bits[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift[%d] = %.4f (must be arround 64 at 8000hz)\n", i, bnetz->phaseshift65536[i]);
}
return 0;
@@ -241,9 +245,9 @@ static inline void fsk_decode_step(bnetz_t *bnetz, int pos)
// FIXME: better threshold
/* adjust level, so we get peak of sine curve */
if (level / 0.63 > 0.05 && (softbit > 0.75 || softbit < 0.25)) {
fsk_receive_tone(bnetz, bit, 1, level / 0.63662 * 32768.0 / TX_PEAK_TONE, quality);
fsk_receive_tone(bnetz, bit, 1, level / 0.63662 / TX_PEAK_TONE, quality);
} else
fsk_receive_tone(bnetz, bit, 0, level / 0.63662 * 32768.0 / TX_PEAK_TONE, quality);
fsk_receive_tone(bnetz, bit, 0, level / 0.63662 / TX_PEAK_TONE, quality);
if (bnetz->fsk_filter_bit != bit) {
/* if we have a bit change, reset sample counter to one half bit duration */
@@ -256,7 +260,7 @@ static inline void fsk_decode_step(bnetz_t *bnetz, int pos)
printf("|%s|\n", debug_amplitude(quality);
#endif
/* adjust level, so we get peak of sine curve */
fsk_receive_bit(bnetz, bit, level / 0.63662 * 32768.0 / TX_PEAK_TONE, quality);
fsk_receive_bit(bnetz, bit, level / 0.63662 / TX_PEAK_TONE, quality);
bnetz->fsk_filter_sample = 10;
}
}
@@ -308,17 +312,17 @@ static void fsk_tone(bnetz_t *bnetz, sample_t *samples, int length, int tone)
double phaseshift, phase;
int i;
phase = bnetz->phase256;
phaseshift = bnetz->phaseshift256[tone];
phase = bnetz->phase65536;
phaseshift = bnetz->phaseshift65536[tone];
for (i = 0; i < length; i++) {
*samples++ = dsp_sine[(uint8_t)phase];
*samples++ = dsp_sine[(uint16_t)phase];
phase += phaseshift;
if (phase >= 256)
phase -= 256;
if (phase >= 65536)
phase -= 65536;
}
bnetz->phase256 = phase;
bnetz->phase65536 = phase;
}
static int fsk_telegramm(bnetz_t *bnetz, sample_t *samples, int length)
@@ -342,17 +346,17 @@ next_telegramm:
bnetz->telegramm_pos = 0;
spl = bnetz->telegramm_spl;
/* render telegramm */
phase = bnetz->phase256;
phase = bnetz->phase65536;
for (i = 0; i < 16; i++) {
phaseshift = bnetz->phaseshift256[telegramm[i] == '1'];
phaseshift = bnetz->phaseshift65536[telegramm[i] == '1'];
for (j = 0; j < bnetz->samples_per_bit; j++) {
*spl++ = dsp_sine[(uint8_t)phase];
*spl++ = dsp_sine[(uint16_t)phase];
phase += phaseshift;
if (phase >= 256)
phase -= 256;
if (phase >= 65536)
phase -= 65536;
}
}
bnetz->phase256 = phase;
bnetz->phase65536 = phase;
}
/* send audio from telegramm */