A-Netz: Calibration of tone levels on TX and RX side

pull/1/head
Andreas Eversberg 7 years ago
parent b73894ac49
commit aae58d5a68
  1. 8
      src/anetz/anetz.c
  2. 2
      src/anetz/anetz.h
  3. 80
      src/anetz/dsp.c

@ -248,9 +248,13 @@ void anetz_receive_tone(anetz_t *anetz, int tone)
else
PDEBUG(DANETZ, DEBUG_DEBUG, "Continuous tone is gone.\n");
if (anetz->sender.loopback) {
/* skip any handling in loopback mode */
if (anetz->sender.loopback)
return;
/* skip tone 2280 Hz, because it is not relevant for base station */
if (tone == 0)
return;
}
switch (anetz->state) {
case ANETZ_FREI:

@ -24,7 +24,7 @@ typedef struct anetz {
/* dsp states */
enum dsp_mode dsp_mode; /* current mode: audio, durable tone 0 or 1, paging */
int fsk_tone_coeff; /* coefficient k = 2*cos(2*PI*f/samplerate), k << 15 */
int fsk_tone_coeff[2]; /* coefficient k = 2*cos(2*PI*f/samplerate), k << 15 */
int samples_per_chunk; /* how many samples lasts one chunk */
int16_t *fsk_filter_spl; /* array with samples_per_chunk */
int fsk_filter_pos; /* current sample position in filter_spl */

@ -33,7 +33,11 @@
#define PI 3.1415927
/* signalling */
#define TX_PEAK 8190.0 /* peak amplitude of sine wave (must be less than 32768/4) */
/* NOTE: The peak deviation is similar for paging tone and signalling tone,
* so both tones should be equal after pre-emphasis. This is why the paging
* tones is so much louder.*/
#define TX_PEAK_TONE 8192.0 /* peak amplitude for all tones */
#define TX_PEAK_PAGE 32767.0 /* peak amplitude paging tone */
// FIXME: what is the allowed deviation of tone?
#define CHUNK_DURATION 0.010 /* 10 ms */
@ -53,20 +57,28 @@ static double fsk_tones[2] = {
};
/* table for fast sine generation */
int dsp_sine[256];
int dsp_sine_tone[256];
int dsp_sine_page[256];
/* global init for audio processing */
void dsp_init(void)
{
int i;
double s;
PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine table.\n");
PDEBUG(DDSP, DEBUG_DEBUG, "Generating sine tables.\n");
for (i = 0; i < 256; i++) {
dsp_sine[i] = (int)(sin((double)i / 256.0 * 2.0 * PI) * TX_PEAK);
s = sin((double)i / 256.0 * 2.0 * PI);
dsp_sine_tone[i] = (int)(s * TX_PEAK_TONE);
dsp_sine_page[i] = (int)(s * TX_PEAK_PAGE);
}
if (TX_PEAK > 8191.0) {
fprintf(stderr, "TX_PEAK definition too high, please fix!\n");
if (TX_PEAK_TONE > 32767.0) {
fprintf(stderr, "TX_PEAK_TONE definition too high, please fix!\n");
abort();
}
if (TX_PEAK_PAGE > 32767.0) {
fprintf(stderr, "TX_PEAK_PAGE definition too high, please fix!\n");
abort();
}
}
@ -76,7 +88,8 @@ int dsp_init_sender(anetz_t *anetz)
{
int16_t *spl;
double coeff;
int detect_tone = (anetz->sender.loopback) ? 0 : 1;
int i;
double tone;
PDEBUG(DDSP, DEBUG_DEBUG, "Init DSP for 'Sender'.\n");
@ -93,11 +106,14 @@ int dsp_init_sender(anetz_t *anetz)
anetz->tone_detected = -1;
coeff = 2.0 * cos(2.0 * PI * fsk_tones[detect_tone] / (double)anetz->sender.samplerate);
anetz->fsk_tone_coeff = coeff * 32768.0;
PDEBUG(DDSP, DEBUG_DEBUG, "RX %.0f Hz coeff = %d\n", fsk_tones[detect_tone], (int)anetz->fsk_tone_coeff);
anetz->tone_phaseshift256 = 256.0 / ((double)anetz->sender.samplerate / fsk_tones[0]);
PDEBUG(DDSP, DEBUG_DEBUG, "TX %.0f Hz phaseshift = %.4f\n", fsk_tones[0], anetz->tone_phaseshift256);
for (i = 0; i < 2; i++) {
coeff = 2.0 * cos(2.0 * PI * fsk_tones[i] / (double)anetz->sender.samplerate);
anetz->fsk_tone_coeff[i] = coeff * 32768.0;
PDEBUG(DDSP, DEBUG_DEBUG, "RX %.0f Hz coeff = %d\n", fsk_tones[i], (int)anetz->fsk_tone_coeff[i]);
}
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);
return 0;
}
@ -119,7 +135,7 @@ static void fsk_receive_tone(anetz_t *anetz, int tone, int goodtone, double leve
/* lost tone because it is not good anymore or has changed */
if (!goodtone || tone != anetz->tone_detected) {
if (anetz->tone_count >= TONE_DETECT_TH) {
PDEBUG(DDSP, DEBUG_DEBUG, "Lost %.0f Hz tone after %d ms.\n", fsk_tones[anetz->tone_detected], 1000.0 * CHUNK_DURATION * anetz->tone_count);
PDEBUG(DDSP, DEBUG_INFO, "Lost %.0f Hz tone after %d ms.\n", fsk_tones[anetz->tone_detected], 1000.0 * CHUNK_DURATION * anetz->tone_count);
anetz_receive_tone(anetz, -1);
}
if (goodtone)
@ -136,7 +152,7 @@ static void fsk_receive_tone(anetz_t *anetz, int tone, int goodtone, double leve
if (anetz->tone_count >= TONE_DETECT_TH)
audio_reset_loss(&anetz->sender.loss);
if (anetz->tone_count == TONE_DETECT_TH) {
PDEBUG(DDSP, DEBUG_DEBUG, "Detecting continous %.0f Hz tone. (level = %d%%)\n", fsk_tones[anetz->tone_detected], (int)(level * 100));
PDEBUG(DDSP, DEBUG_INFO, "Detecting continous %.0f Hz tone. (level = %d%%)\n", fsk_tones[anetz->tone_detected], (int)(level * 100.0 + 0.5));
anetz_receive_tone(anetz, anetz->tone_detected);
}
}
@ -144,27 +160,31 @@ static void fsk_receive_tone(anetz_t *anetz, int tone, int goodtone, double leve
/* Filter one chunk of audio an detect tone, quality and loss of signal. */
static void fsk_decode_chunk(anetz_t *anetz, int16_t *spl, int max)
{
double level, result;
double level, result[2];
level = audio_level(spl, max);
if (audio_detect_loss(&anetz->sender.loss, level))
anetz_loss_indication(anetz);
audio_goertzel(spl, max, 0, &anetz->fsk_tone_coeff, &result, 1);
audio_goertzel(spl, max, 0, anetz->fsk_tone_coeff, result, 2);
/* show quality of tone */
if (anetz->sender.loopback) {
/* adjust level, so we get peak of sine curve */
PDEBUG(DDSP, DEBUG_NOTICE, "Quality Tone:%3.0f%% Level:%3.0f%%\n", result / level * 100.0, level / 0.63662 * 100.0);
PDEBUG(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);
}
if (level / 0.63 > 0.05 && result[0] / level > 0.5)
PDEBUG(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);
/* adjust level, so we get peak of sine curve */
/* indicate detected tone 1 (1750 Hz) or tone 0 (2280 Hz) at loopback */
if (level / 0.63 > 0.05 && result / level > 0.5)
fsk_receive_tone(anetz, (anetz->sender.loopback) ? 0 : 1, 1, level / 0.63662);
/* 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);
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);
else
fsk_receive_tone(anetz, (anetz->sender.loopback) ? 0 : 1, 0, level / 0.63662);
fsk_receive_tone(anetz, -1, 0, level / 0.63662 * 32768.0 / TX_PEAK_TONE);
}
/* Process received audio stream from radio unit. */
@ -220,11 +240,12 @@ void dsp_set_paging(anetz_t *anetz, double *freq)
}
/* Generate audio stream of 4 simultanious paging tones. Keep phase for next call of function.
* Use TX_PEAK for one tone, which gives peak of TX_PEAK * 4 for all tones. */
* Use TX_PEAK_PAGE for all tones, which gives peak of (TX_PEAK_PAGE / 4) for each individual tone. */
static void fsk_paging_tone(anetz_t *anetz, int16_t *samples, int length)
{
double phaseshift[5], phase[5];
int i;
int32_t sample;
for (i = 0; i < 4; i++) {
phaseshift[i] = anetz->paging_phaseshift256[i];
@ -232,10 +253,11 @@ static void fsk_paging_tone(anetz_t *anetz, int16_t *samples, int length)
}
for (i = 0; i < length; i++) {
*samples++ = dsp_sine[((uint8_t)phase[0]) & 0xff]
+ dsp_sine[((uint8_t)phase[1]) & 0xff]
+ dsp_sine[((uint8_t)phase[2]) & 0xff]
+ dsp_sine[((uint8_t)phase[3]) & 0xff];
sample = (int32_t)dsp_sine_page[((uint8_t)phase[0]) & 0xff]
+ (int32_t)dsp_sine_page[((uint8_t)phase[1]) & 0xff]
+ (int32_t)dsp_sine_page[((uint8_t)phase[2]) & 0xff]
+ (int32_t)dsp_sine_page[((uint8_t)phase[3]) & 0xff];
*samples++ = sample >> 2;
phase[0] += phaseshift[0];
phase[1] += phaseshift[1];
phase[2] += phaseshift[2];
@ -252,7 +274,7 @@ static void fsk_paging_tone(anetz_t *anetz, int16_t *samples, int length)
}
/* Generate audio stream of 4 sequenced paging tones. Keep phase for next call of function.
* Use TX_PEAK * 2 for each tone. We will use a lower peak, because the radio might not TX it. */
* Use TX_PEAK_PAGE / 2 for each tone, that is twice as much peak per tone. */
static void fsk_paging_tone_sequence(anetz_t *anetz, int16_t *samples, int length, int numspl)
{
double phaseshift, phase;
@ -266,7 +288,7 @@ next_tone:
phaseshift = anetz->paging_phaseshift256[tone];
while (length) {
*samples++ = dsp_sine[((uint8_t)phase) & 0xff] << 2;
*samples++ = dsp_sine_page[((uint8_t)phase) & 0xff] >> 1;
phase += phaseshift;
if (phase >= 256)
phase -= 256;
@ -294,7 +316,7 @@ static void fsk_tone(anetz_t *anetz, int16_t *samples, int length)
phase = anetz->tone_phase256;
for (i = 0; i < length; i++) {
*samples++ = dsp_sine[((uint8_t)phase) & 0xff];
*samples++ = dsp_sine_tone[((uint8_t)phase) & 0xff];
phase += phaseshift;
if (phase >= 256)
phase -= 256;

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