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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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120
src/nmt/dsp.c
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@@ -34,23 +34,36 @@
#define PI M_PI
/* Notes on frequency deviation of supervidory signal:
/* Notes on TX_PEAK_FSK level:
*
* The FSK deviation at 1500 Hz is 3.5 KHz. If we use a level of 10000
* The supervisory deviation shall be 0.3 KHz: 10000 / 3.5 * 0.3 = 857
* Supervisory is raised by pre-emphasis by factor 2.68 (4015 / 1500),
* so we need to lower it: 857 / 2.68 = 320
* This deviation is -2.2db below the dBm0 deviation.
*
* At 1800 Hz the deviation shall be 4.2 kHz, so with emphasis the deviation
* at 1000 Hz would be theoretically 2.333 kHz. This is factor 0.777 below
* 3 kHz deviation we want at dBm0.
*/
/* Notes on TX_PEAK_SUPER (supervisory signal) level:
*
* This level has 0.3 kHz deviation at 4015 Hz.
*
* Same calculation as above, but now we want 0.3 kHz deviation after emphasis,
* so we calculate what we would need at 1000 Hz in relation to 3 kHz
* deviation.
*/
/* signaling */
#define BANDWIDTH 6000.0 /* maximum bandwidth FIXME */
#define COMPANDOR_0DB 32767 /* works quite well */
#define TX_PEAK_FSK 10000.0 /* peak amplitude of signaling FSK +-3.5 KHz @ 1500 Hz */
#define TX_PEAK_SUPER (TX_PEAK_FSK / 3.5 * 0.3 / 2.68) /* peak amplitude of supervisory signal +-0.3 KHz @ 4015 Hz */
#define MAX_DEVIATION 4700.0
#define MAX_MODULATION 4055.0
#define DBM0_DEVIATION 3000.0 /* deviation of dBm0 at 1 kHz */
#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 MAX_DISPLAY 1.4 /* something above dBm0 */
#define BIT_RATE 1200 /* baud rate */
#define STEPS_PER_BIT 10 /* step every 1/12000 sec */
#define FILTER_STEPS 0.1 /* step every 1/12000 sec */
#define DIALTONE_HZ 425.0 /* dial tone frequency */
#define TX_PEAK_DIALTONE 16000.0 /* dial tone peak */
#define TX_PEAK_DIALTONE 0.5 /* dial tone peak FIXME */
#define SUPER_DURATION 0.25 /* duration of supervisory signal measurement */
#define SUPER_DETECT_COUNT 4 /* number of measures to detect supervisory signal */
#define MUTE_DURATION 0.280 /* a tiny bit more than two frames */
@@ -71,9 +84,9 @@ static double super_freq[5] = {
};
/* table for fast sine generation */
static double dsp_tone_bit[2][2][256]; /* polarity, bit, phase */
static double dsp_sine_super[256];
static double dsp_sine_dialtone[256];
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 */
void dsp_init(void)
@@ -82,8 +95,8 @@ void dsp_init(void)
double s;
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);
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 */
@@ -92,7 +105,7 @@ void dsp_init(void)
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 / 256.0 * 3.0 * PI);
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;
}
@@ -115,9 +128,8 @@ int dsp_init_sender(nmt_t *nmt)
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for Transceiver.\n");
/* set deviation and modulation parameters */
nmt->sender.bandwidth = BANDWIDTH;
nmt->sender.sample_deviation = 2500.0 / (double)TX_PEAK_FSK; // FIXME: calc real value
/* set modulation parameters */
sender_set_fm(&nmt->sender, MAX_DEVIATION, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
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);
@@ -166,22 +178,22 @@ int dsp_init_sender(nmt_t *nmt)
/* count symbols */
for (i = 0; i < 2; i++)
audio_goertzel_init(&nmt->fsk_goertzel[i], fsk_freq[i], nmt->sender.samplerate);
nmt->fsk_phaseshift256 = 256.0 / nmt->fsk_samples_per_bit;
PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", nmt->fsk_phaseshift256);
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);
if (i < 4) {
nmt->super_phaseshift256[i] = 256.0 / ((double)nmt->sender.samplerate / super_freq[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "super_phaseshift[%d] = %.4f\n", i, nmt->super_phaseshift256[i]);
nmt->super_phaseshift65536[i] = 65536.0 / ((double)nmt->sender.samplerate / super_freq[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "super_phaseshift[%d] = %.4f\n", i, nmt->super_phaseshift65536[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);
nmt->dial_phaseshift65536 = 65536.0 / ((double)nmt->sender.samplerate / DIALTONE_HZ);
PDEBUG(DDSP, DEBUG_DEBUG, "dial_phaseshift = %.4f\n", nmt->dial_phaseshift65536);
/* dtmf */
dtmf_init(&nmt->dtmf, 8000);
@@ -305,9 +317,8 @@ static inline void fsk_decode_step(nmt_t *nmt, int pos)
level = audio_level(spl, max);
/* limit level to prevent division by zero */
if (level < 0.01)
level = 0.01;
// level = 0.63662 / 2.0;
if (level < 0.001)
level = 0.001;
audio_goertzel(nmt->fsk_goertzel, spl, max, pos, result, 2);
@@ -353,9 +364,9 @@ static inline void fsk_decode_step(nmt_t *nmt, int pos)
printf("|%s|\n", debug_amplitude(quality));
#endif
/* adjust level, so a peak level becomes 100% */
fsk_receive_bit(nmt, bit, quality, level / 0.63662 * 32768.0 / TX_PEAK_FSK);
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 * 32768.0 / TX_PEAK_FSK);
fsk_receive_bit_dms(nmt, bit, quality, level / 0.63662 / TX_PEAK_FSK);
nmt->fsk_filter_sample = 10;
}
}
@@ -368,26 +379,19 @@ static void super_decode(nmt_t *nmt, sample_t *samples, int length)
audio_goertzel(&nmt->super_goertzel[nmt->supervisory - 1], samples, length, 0, &result[0], 1);
audio_goertzel(&nmt->super_goertzel[4], samples, length, 0, &result[1], 1); /* noise floor detection */
#if 0
/* normalize levels */
result[0] *= 32768.0 / TX_PEAK_SUPER / 0.63662;
result[1] *= 32768.0 / TX_PEAK_SUPER / 0.63662;
printf("signal=%.4f noise=%.4f\n", result[0], result[1]);
#endif
quality = (result[0] - result[1]) / result[0];
if (quality < 0)
quality = 0;
if (nmt->state == STATE_ACTIVE)
PDEBUG_CHAN(DDSP, DEBUG_NOTICE, "Supervisory level %.0f%% quality %.0f%%\n", result[0] / 0.63662 * 32768.0 / TX_PEAK_SUPER * 100.0, quality * 100.0);
PDEBUG_CHAN(DDSP, DEBUG_NOTICE, "Supervisory level %.0f%% quality %.0f%%\n", result[0] / 0.63662 / TX_PEAK_SUPER * 100.0, quality * 100.0);
if (quality > 0.5) {
if (nmt->super_detected == 0) {
nmt->super_detect_count++;
if (nmt->super_detect_count == SUPER_DETECT_COUNT) {
nmt->super_detected = 1;
nmt->super_detect_count = 0;
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Supervisory signal detected with level=%.0f%%, quality=%.0f%%.\n", result[0] / 0.63662 * 32768.0 / TX_PEAK_SUPER * 100.0, quality * 100.0);
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Supervisory signal detected with level=%.0f%%, quality=%.0f%%.\n", result[0] / 0.63662 / TX_PEAK_SUPER * 100.0, quality * 100.0);
nmt_rx_super(nmt, 1, quality);
}
} else
@@ -497,21 +501,21 @@ int fsk_render_frame(nmt_t *nmt, const char *frame, int length, sample_t *sample
int count = 0, i;
polarity = nmt->fsk_polarity;
phaseshift = nmt->fsk_phaseshift256;
phase = nmt->fsk_phase256;
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][(uint8_t)phase];
*sample++ = dsp_tone_bit[polarity][bit][(uint16_t)phase];
count++;
phase += phaseshift;
} while (phase < 256.0);
phase -= 256.0;
} while (phase < 65536.0);
phase -= 65536.0;
/* flip polarity when we have 1.5 sine waves */
if (bit == 0)
polarity = 1 - polarity;
}
nmt->fsk_phase256 = phase;
nmt->fsk_phase65536 = phase;
nmt->fsk_polarity = polarity;
/* return number of samples created for frame */
@@ -570,17 +574,17 @@ static void super_encode(nmt_t *nmt, sample_t *samples, int length)
double phaseshift, phase;
int i;
phaseshift = nmt->super_phaseshift256[nmt->supervisory - 1];
phase = nmt->super_phase256;
phaseshift = nmt->super_phaseshift65536[nmt->supervisory - 1];
phase = nmt->super_phase65536;
for (i = 0; i < length; i++) {
*samples++ += dsp_sine_super[(uint8_t)phase];
*samples++ += dsp_sine_super[(uint16_t)phase];
phase += phaseshift;
if (phase >= 256)
phase -= 256;
if (phase >= 65536)
phase -= 65536;
}
nmt->super_phase256 = phase;
nmt->super_phase65536 = phase;
}
/* Generate audio stream from dial tone. Keep phase for next call of function. */
@@ -589,17 +593,17 @@ static void dial_tone(nmt_t *nmt, sample_t *samples, int length)
double phaseshift, phase;
int i;
phaseshift = nmt->dial_phaseshift256;
phase = nmt->dial_phase256;
phaseshift = nmt->dial_phaseshift65536;
phase = nmt->dial_phase65536;
for (i = 0; i < length; i++) {
*samples++ = dsp_sine_dialtone[(uint8_t)phase];
*samples++ = dsp_sine_dialtone[(uint16_t)phase];
phase += phaseshift;
if (phase >= 256)
phase -= 256;
if (phase >= 65536)
phase -= 65536;
}
nmt->dial_phase256 = phase;
nmt->dial_phase65536 = phase;
}
/* Provide stream of audio toward radio unit */