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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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8
src/anetz/anetz.h
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@@ -33,12 +33,12 @@ typedef struct anetz {
int fsk_filter_pos; /* current sample position in filter_spl */
int tone_detected; /* what tone has been detected */
int tone_count; /* how long has that tone been detected */
double tone_phaseshift256; /* how much the phase of sine wave changes per sample */
double tone_phase256; /* current phase */
double tone_phaseshift65536; /* how much the phase of sine wave changes per sample */
double tone_phase65536; /* current phase */
double page_gain; /* factor to raise the paging tones */
int page_sequence; /* if set, use paging tones in sequence rather than parallel */
double paging_phaseshift256[4];/* how much the phase of sine wave changes per sample */
double paging_phase256[4]; /* current phase */
double paging_phaseshift65536[4];/* how much the phase of sine wave changes per sample */
double paging_phase65536[4]; /* current phase */
int paging_tone; /* current tone (0..3) in sequenced mode */
int paging_count; /* current sample count of tone in seq. mode */
int paging_transition; /* set to number of samples during transition */

118
src/anetz/dsp.c
View File

@@ -35,8 +35,12 @@
#define PI 3.1415927
/* signaling */
#define BANDWIDTH 15000.0 /* maximum bandwidth */
#define TX_PEAK_TONE 8192.0 /* peak amplitude for all tones */
#define MAX_DEVIATION 15000.0
#define MAX_MODULATION 4000.0
#define DBM0_DEVIATION 10500.0 /* deviation of dBm0 at 1 kHz */
#define TX_PEAK_TONE (10500.0 / DBM0_DEVIATION) /* 10.5 kHz, no emphasis */
#define TX_PEAK_PAGE (15000.0 / DBM0_DEVIATION) /* 15 kHz, no emphasis */
#define MAX_DISPLAY (15000.0 / DBM0_DEVIATION) /* 15 kHz, no emphasis */
#define CHUNK_DURATION 0.010 /* 10 ms */
// FIXME: how long until we detect a tone?
@@ -53,7 +57,8 @@ static double fsk_tones[2] = {
};
/* table for fast sine generation */
sample_t dsp_sine_tone[256];
static sample_t dsp_sine_tone[65536];
static sample_t dsp_sine_page[65536];
/* global init for audio processing */
void dsp_init(void)
@@ -62,14 +67,10 @@ void dsp_init(void)
double s;
PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine tables.\n");
for (i = 0; i < 256; i++) {
s = sin((double)i / 256.0 * 2.0 * PI);
dsp_sine_tone[i] = (int)(s * 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++) {
s = sin((double)i / 65536.0 * 2.0 * PI);
dsp_sine_tone[i] = s * TX_PEAK_TONE;
dsp_sine_page[i] = s * TX_PEAK_PAGE;
}
}
@@ -82,15 +83,10 @@ int dsp_init_sender(anetz_t *anetz, double page_gain, int page_sequence)
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for 'Sender'.\n");
/* set deviation and modulation parameters */
anetz->sender.bandwidth = BANDWIDTH;
anetz->sender.sample_deviation = 11000.0 / (double)TX_PEAK_TONE;
/* set modulation parameters */
sender_set_fm(&anetz->sender, MAX_DEVIATION * page_gain, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
anetz->page_gain = page_gain;
if (page_gain * TX_PEAK_TONE > 32767.0) {
page_gain = 32767.0 / TX_PEAK_TONE;
PDEBUG_CHAN(DDSP, DEBUG_NOTICE, "Highest possible gain of paging tones is %.1f dB.\n", log10(page_gain) * 20);
}
anetz->page_sequence = page_sequence;
audio_init_loss(&anetz->sender.loss, LOSS_INTERVAL, anetz->sender.loss_volume, LOSS_TIME);
@@ -109,8 +105,8 @@ int dsp_init_sender(anetz_t *anetz, double page_gain, int page_sequence)
for (i = 0; i < 2; i++)
audio_goertzel_init(&anetz->fsk_tone_goertzel[i], fsk_tones[i], anetz->sender.samplerate);
tone = fsk_tones[(anetz->sender.loopback == 0) ? 0 : 1];
anetz->tone_phaseshift256 = 256.0 / ((double)anetz->sender.samplerate / tone);
PDEBUG(DDSP, DEBUG_DEBUG, "TX %.0f Hz phaseshift = %.4f\n", tone, anetz->tone_phaseshift256);
anetz->tone_phaseshift65536 = 65536.0 / ((double)anetz->sender.samplerate / tone);
PDEBUG(DDSP, DEBUG_DEBUG, "TX %.0f Hz phaseshift = %.4f\n", tone, anetz->tone_phaseshift65536);
return 0;
}
@@ -169,19 +165,19 @@ static void fsk_decode_chunk(anetz_t *anetz, sample_t *spl, int max)
/* show quality of tone */
if (anetz->sender.loopback) {
/* adjust level, so we get peak of sine curve */
PDEBUG_CHAN(DDSP, DEBUG_NOTICE, "Tone %.0f: Level=%3.0f%% Quality=%3.0f%%\n", fsk_tones[1], level / 0.63662 * 100.0 * 32768.0 / TX_PEAK_TONE, result[1] / level * 100.0);
PDEBUG_CHAN(DDSP, DEBUG_NOTICE, "Tone %.0f: Level=%3.0f%% Quality=%3.0f%%\n", fsk_tones[1], level / 0.63662 * 100.0 / TX_PEAK_TONE, result[1] / level * 100.0);
}
if (level / 0.63 > 0.05 && result[0] / level > 0.5)
PDEBUG_CHAN(DDSP, DEBUG_INFO, "Tone %.0f: Level=%3.0f%% Quality=%3.0f%%\n", fsk_tones[0], level / 0.63662 * 100.0 * 32768.0 / TX_PEAK_TONE, result[0] / level * 100.0);
PDEBUG_CHAN(DDSP, DEBUG_INFO, "Tone %.0f: Level=%3.0f%% Quality=%3.0f%%\n", fsk_tones[0], level / 0.63662 * 100.0 / TX_PEAK_TONE, result[0] / level * 100.0);
/* adjust level, so we get peak of sine curve */
/* indicate detected tone */
if (level / 0.63 > 0.05 && result[0] / level > 0.5)
fsk_receive_tone(anetz, 0, 1, level / 0.63662 * 32768.0 / TX_PEAK_TONE);
fsk_receive_tone(anetz, 0, 1, level / 0.63662 / TX_PEAK_TONE);
else if (level / 0.63 > 0.05 && result[1] / level > 0.5)
fsk_receive_tone(anetz, 1, 1, level / 0.63662 * 32768.0 / TX_PEAK_TONE);
fsk_receive_tone(anetz, 1, 1, level / 0.63662 / TX_PEAK_TONE);
else
fsk_receive_tone(anetz, -1, 0, level / 0.63662 * 32768.0 / TX_PEAK_TONE);
fsk_receive_tone(anetz, -1, 0, level / 0.63662 / TX_PEAK_TONE);
}
/* Process received audio stream from radio unit. */
@@ -230,42 +226,36 @@ void dsp_set_paging(anetz_t *anetz, double *freq)
int i;
for (i = 0; i < 4; i++) {
anetz->paging_phaseshift256[i] = 256.0 / ((double)anetz->sender.samplerate / freq[i]);
anetz->paging_phase256[i] = 0;
anetz->paging_phaseshift65536[i] = 65536.0 / ((double)anetz->sender.samplerate / freq[i]);
anetz->paging_phase65536[i] = 0;
}
}
/* Generate audio stream of 4 simultanious paging tones. Keep phase for next call of function.
* Use TX_PEAK_TONE*page_gain for all tones, which gives peak of 1/4th for each individual tone. */
* Use TX_PEAK_PAGE*page_gain for all tones, which gives peak of 1/4th for each individual tone. */
static void fsk_paging_tone(anetz_t *anetz, sample_t *samples, int length)
{
double phaseshift[4], phase[4];
double *phaseshift, *phase;
int i;
double sample;
for (i = 0; i < 4; i++) {
phaseshift[i] = anetz->paging_phaseshift256[i];
phase[i] = anetz->paging_phase256[i];
}
phaseshift = anetz->paging_phaseshift65536;
phase = anetz->paging_phase65536;
for (i = 0; i < length; i++) {
sample = (int32_t)dsp_sine_tone[(uint8_t)phase[0]]
+ (int32_t)dsp_sine_tone[(uint8_t)phase[1]]
+ (int32_t)dsp_sine_tone[(uint8_t)phase[2]]
+ (int32_t)dsp_sine_tone[(uint8_t)phase[3]];
sample = dsp_sine_page[(uint16_t)phase[0]]
+ dsp_sine_page[(uint16_t)phase[1]]
+ dsp_sine_page[(uint16_t)phase[2]]
+ dsp_sine_page[(uint16_t)phase[3]];
*samples++ = sample / 4.0 * anetz->page_gain;
phase[0] += phaseshift[0];
phase[1] += phaseshift[1];
phase[2] += phaseshift[2];
phase[3] += phaseshift[3];
if (phase[0] >= 256) phase[0] -= 256;
if (phase[1] >= 256) phase[1] -= 256;
if (phase[2] >= 256) phase[2] -= 256;
if (phase[3] >= 256) phase[3] -= 256;
}
for (i = 0; i < 4; i++) {
anetz->paging_phase256[i] = phase[i];
if (phase[0] >= 65536) phase[0] -= 65536;
if (phase[1] >= 65536) phase[1] -= 65536;
if (phase[2] >= 65536) phase[2] -= 65536;
if (phase[3] >= 65536) phase[3] -= 65536;
}
}
@@ -280,14 +270,11 @@ static void fsk_paging_tone(anetz_t *anetz, sample_t *samples, int length)
*/
static void fsk_paging_tone_sequence(anetz_t *anetz, sample_t *samples, int length, int numspl)
{
double phaseshift[4], phase[4];
int i;
double *phaseshift, *phase;
int tone, count, transition;
for (i = 0; i < 4; i++) {
phaseshift[i] = anetz->paging_phaseshift256[i];
phase[i] = anetz->paging_phase256[i];
}
phaseshift = anetz->paging_phaseshift65536;
phase = anetz->paging_phase65536;
tone = anetz->paging_tone;
count = anetz->paging_count;
transition = anetz->paging_transition;
@@ -295,21 +282,21 @@ static void fsk_paging_tone_sequence(anetz_t *anetz, sample_t *samples, int leng
while (length) {
/* use tone, but during transition of tones, keep phase 0 degrees (high level) until next tone reaches 0 degrees (high level) */
if (!transition)
*samples++ = dsp_sine_tone[(uint8_t)phase[tone]] * anetz->page_gain;
*samples++ = dsp_sine_page[(uint16_t)phase[tone]] * anetz->page_gain;
else {
/* fade between old an new tone */
*samples++
= (double)dsp_sine_tone[(uint8_t)phase[(tone - 1) & 3]] * (double)(transition - count) / (double)transition / 2.0 * anetz->page_gain
+ (double)dsp_sine_tone[(uint8_t)phase[tone]] * (double)count / (double)transition / 2.0 * anetz->page_gain;
= (double)dsp_sine_page[(uint16_t)phase[(tone - 1) & 3]] * (double)(transition - count) / (double)transition / 2.0 * anetz->page_gain
+ (double)dsp_sine_page[(uint16_t)phase[tone]] * (double)count / (double)transition / 2.0 * anetz->page_gain;
}
phase[0] += phaseshift[0];
phase[1] += phaseshift[1];
phase[2] += phaseshift[2];
phase[3] += phaseshift[3];
if (phase[0] >= 256) phase[0] -= 256;
if (phase[1] >= 256) phase[1] -= 256;
if (phase[2] >= 256) phase[2] -= 256;
if (phase[3] >= 256) phase[3] -= 256;
if (phase[0] >= 65536) phase[0] -= 65536;
if (phase[1] >= 65536) phase[1] -= 65536;
if (phase[2] >= 65536) phase[2] -= 65536;
if (phase[3] >= 65536) phase[3] -= 65536;
count++;
if (transition && count == transition) {
transition = 0;
@@ -327,9 +314,6 @@ static void fsk_paging_tone_sequence(anetz_t *anetz, sample_t *samples, int leng
length--;
}
for (i = 0; i < 4; i++) {
anetz->paging_phase256[i] = phase[i];
}
anetz->paging_tone = tone;
anetz->paging_count = count;
anetz->paging_transition = transition;
@@ -341,17 +325,17 @@ static void fsk_tone(anetz_t *anetz, sample_t *samples, int length)
double phaseshift, phase;
int i;
phaseshift = anetz->tone_phaseshift256;
phase = anetz->tone_phase256;
phaseshift = anetz->tone_phaseshift65536;
phase = anetz->tone_phase65536;
for (i = 0; i < length; i++) {
*samples++ = dsp_sine_tone[(uint8_t)phase];
*samples++ = dsp_sine_tone[(uint16_t)phase];
phase += phaseshift;
if (phase >= 256)
phase -= 256;
if (phase >= 65536)
phase -= 65536;
}
anetz->tone_phase256 = phase;
anetz->tone_phase65536 = phase;
}
/* Provide stream of audio toward radio unit */