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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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79
src/amps/dsp.c
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@@ -99,10 +99,13 @@
#define PI M_PI
#define BANDWIDTH 20000.0 /* maximum bandwidth */
#define FSK_DEVIATION 32767.0 /* +-8 KHz */
#define SAT_DEVIATION 8192.0 /* +-2 KHz */
#define COMPANDOR_0DB 45000 /* works quite well */
#define MAX_DEVIATION 8000.0
#define MAX_MODULATION 10000.0
#define DBM0_DEVIATION 2900.0 /* deviation of dBm0 at 1 kHz */
#define COMPANDOR_0DB 1.0 /* A level of 0dBm (1.0) shall be unaccected */
#define FSK_DEVIATION (8000.0 / DBM0_DEVIATION) /* no emphasis */
#define SAT_DEVIATION (2000.0 / DBM0_DEVIATION) /* no emphasis */
#define MAX_DISPLAY (8000.0 / DBM0_DEVIATION) /* no emphasis */
#define BITRATE 10000
#define SIG_TONE_CROSSINGS 2000 /* 2000 crossings are 100ms @ 10 KHz */
#define SIG_TONE_MINBITS 950 /* minimum bit durations to detect signaling tone (1000 is perfect for 100 ms) */
@@ -116,7 +119,7 @@
#define CUT_OFF_HIGHPASS 300.0 /* cut off frequency for high pass filter to remove dc level from sound card / sample */
#define BEST_QUALITY 0.68 /* Best possible RX quality */
static int16_t ramp_up[256], ramp_down[256];
static sample_t ramp_up[256], ramp_down[256];
static double sat_freq[5] = {
5970.0,
@@ -126,8 +129,8 @@ static double sat_freq[5] = {
10000.0, /* signaling tone */
};
static int dsp_sine_sat[256];
static int dsp_sine_test[256];
static sample_t dsp_sine_sat[65536];
static sample_t dsp_sine_test[65536];
static uint8_t dsp_sync_check[0x800];
@@ -138,10 +141,10 @@ void dsp_init(void)
double s;
PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table for SAT signal.\n");
for (i = 0; i < 256; i++) {
s = sin((double)i / 256.0 * 2.0 * PI);
dsp_sine_sat[i] = (int)(s * SAT_DEVIATION);
dsp_sine_test[i] = (int)(s * FSK_DEVIATION);
for (i = 0; i < 65536; i++) {
s = sin((double)i / 65536.0 * 2.0 * PI);
dsp_sine_sat[i] = s * SAT_DEVIATION;
dsp_sine_test[i] = s * FSK_DEVIATION;
}
/* sync checker */
@@ -170,7 +173,7 @@ static void dsp_init_ramp(amps_t *amps)
else
c = sqrt(c);
#endif
ramp_down[i] = (int)(c * (double)amps->fsk_deviation);
ramp_down[i] = c * (double)amps->fsk_deviation;
ramp_up[i] = -ramp_down[i];
}
}
@@ -190,9 +193,8 @@ int dsp_init_sender(amps_t *amps, int tolerant)
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for transceiver.\n");
/* set deviation and modulation parameters */
amps->sender.bandwidth = BANDWIDTH;
amps->sender.sample_deviation = 8000.0 / (double)FSK_DEVIATION;
/* set modulation parameters */
sender_set_fm(&amps->sender, MAX_DEVIATION, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
if (amps->sender.samplerate < 96000) {
PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 96000 Hz to process FSK and SAT signals.\n");
@@ -229,7 +231,7 @@ int dsp_init_sender(amps_t *amps, int tolerant)
amps->fsk_rx_window = spl;
/* create devation and ramp */
amps->fsk_deviation = FSK_DEVIATION; /* be sure not to overflow 32767 */
amps->fsk_deviation = FSK_DEVIATION;
dsp_init_ramp(amps);
/* allocate ring buffer for SAT signal detection */
@@ -245,15 +247,15 @@ int dsp_init_sender(amps_t *amps, int tolerant)
for (i = 0; i < 5; i++) {
audio_goertzel_init(&amps->sat_goertzel[i], sat_freq[i], amps->sender.samplerate);
if (i < 3) {
amps->sat_phaseshift256[i] = 256.0 / ((double)amps->sender.samplerate / sat_freq[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "sat_phaseshift256[%d] = %.4f\n", i, amps->sat_phaseshift256[i]);
amps->sat_phaseshift65536[i] = 65536.0 / ((double)amps->sender.samplerate / sat_freq[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "sat_phaseshift65536[%d] = %.4f\n", i, amps->sat_phaseshift65536[i]);
}
}
sat_reset(amps, "Initial state");
/* test tone */
amps->test_phaseshift256 = 256.0 / ((double)amps->sender.samplerate / 1000.0);
PDEBUG(DDSP, DEBUG_DEBUG, "test_phaseshift256 = %.4f\n", amps->test_phaseshift256);
amps->test_phaseshift65536 = 65536.0 / ((double)amps->sender.samplerate / 1000.0);
PDEBUG(DDSP, DEBUG_DEBUG, "test_phaseshift65536 = %.4f\n", amps->test_phaseshift65536);
/* be more tolerant when syncing */
amps->fsk_rx_sync_tolerant = tolerant;
@@ -401,7 +403,7 @@ again:
//printf("pos=%d length=%d copy=%d\n", pos, length, copy);
for (i = 0; i < copy; i++) {
#ifdef DEBUG_ENCODER
puts(debug_amplitude((double)spl[pos] / 32767.0));
puts(debug_amplitude((double)spl[pos]));
#endif
*samples++ = spl[pos++];
}
@@ -422,26 +424,19 @@ done:
static void sat_encode(amps_t *amps, sample_t *samples, int length)
{
double phaseshift, phase;
int32_t sample;
int i;
phaseshift = amps->sat_phaseshift256[amps->sat];
phase = amps->sat_phase256;
phaseshift = amps->sat_phaseshift65536[amps->sat];
phase = amps->sat_phase65536;
for (i = 0; i < length; i++) {
sample = *samples;
sample += dsp_sine_sat[(uint8_t)phase];
if (sample > 32767)
sample = 32767;
else if (sample < -32767)
sample = -32767;
*samples++ = sample;
*samples++ += dsp_sine_sat[(uint16_t)phase];
phase += phaseshift;
if (phase >= 256)
phase -= 256;
if (phase >= 65536)
phase -= 65536;
}
amps->sat_phase256 = phase;
amps->sat_phase65536 = phase;
}
static void test_tone_encode(amps_t *amps, sample_t *samples, int length)
@@ -449,17 +444,17 @@ static void test_tone_encode(amps_t *amps, sample_t *samples, int length)
double phaseshift, phase;
int i;
phaseshift = amps->test_phaseshift256;
phase = amps->test_phase256;
phaseshift = amps->test_phaseshift65536;
phase = amps->test_phase65536;
for (i = 0; i < length; i++) {
*samples++ = dsp_sine_test[(uint8_t)phase];
*samples++ = dsp_sine_test[(uint16_t)phase];
phase += phaseshift;
if (phase >= 256)
phase -= 256;
if (phase >= 65536)
phase -= 65536;
}
amps->test_phase256 = phase;
amps->test_phase65536 = phase;
}
/* Provide stream of audio toward radio unit */
@@ -735,9 +730,9 @@ static void sat_decode(amps_t *amps, sample_t *samples, int length)
if (quality[1] < 0)
quality[1] = 0;
PDEBUG_CHAN(DDSP, DEBUG_NOTICE, "SAT level %.2f%% quality %.0f%%\n", result[0] * 32767.0 / SAT_DEVIATION / 0.63662 * 100.0, quality[0] * 100.0);
PDEBUG_CHAN(DDSP, DEBUG_NOTICE, "SAT level %.2f%% quality %.0f%%\n", result[0] / SAT_DEVIATION / 0.63662 * 100.0, quality[0] * 100.0);
if (amps->sender.loopback || debuglevel == DEBUG_DEBUG) {
PDEBUG_CHAN(DDSP, debuglevel, "Signaling Tone level %.2f%% quality %.0f%%\n", result[2] * 32767.0 / FSK_DEVIATION / 0.63662 * 100.0, quality[1] * 100.0);
PDEBUG_CHAN(DDSP, debuglevel, "Signaling Tone level %.2f%% quality %.0f%%\n", result[2] / FSK_DEVIATION / 0.63662 * 100.0, quality[1] * 100.0);
}
if (quality[0] > SAT_QUALITY) {
if (amps->sat_detected == 0) {