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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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41
src/common/call.c
View File

@@ -35,6 +35,23 @@
#define DISC_TIMEOUT 30
//#define DEBUG_LEVEL
#ifdef DEBUG_LEVEL
static double level_of(double *samples, int count)
{
double level = 0;
int i;
for (i = 0; i < count; i++) {
if (samples[i] > level)
level = samples[i];
}
return level;
}
#endif
/* stream patterns/announcements */
int16_t *test_spl = NULL;
int16_t *ringback_spl = NULL;
@@ -221,7 +238,7 @@ static void get_test_patterns(int16_t *samples, int length)
if (pos >= size)
*samples++ = 0;
else
*samples++ = spl[pos] >> 1;
*samples++ = spl[pos] >> 2;
if (++pos == max)
pos = 0;
}
@@ -477,7 +494,8 @@ int call_init(const char *station_id, const char *audiodev, int samplerate, int
return 0;
/* open sound device for call control */
call.sound = sound_open(audiodev, NULL, NULL, 1, 0.0, samplerate, 3700.0, 0.0);
/* use +3.17 dBm0 (factor 1.44) for complete range of sound card */
call.sound = sound_open(audiodev, NULL, NULL, 1, 0.0, samplerate, 1.44, 4000.0);
if (!call.sound) {
PDEBUG(DSENDER, DEBUG_ERROR, "No sound device!\n");
@@ -677,7 +695,6 @@ void process_call(int c)
default:
jitter_load(&call.dejitter, samples, count);
}
samples_to_int16(spl, samples, count);
samples_list[0] = samples;
rc = sound_write(call.sound, samples_list, count, NULL, NULL, 1);
if (rc < 0) {
@@ -904,8 +921,6 @@ void call_in_release(int callref, int cause)
/* forward audio to MNCC or call instance */
void call_tx_audio(int callref, sample_t *samples, int count)
{
int16_t spl[count];
if (!callref)
return;
@@ -920,6 +935,10 @@ void call_tx_audio(int callref, sample_t *samples, int count)
/* forward audio */
data->msg_type = ANALOG_8000HZ;
data->callref = callref;
#ifdef DEBUG_LEVEL
double lev = level_of(samples, count);
printf(" mobil-level: %s%.4f\n", debug_db(lev), (20 * log10(lev)));
#endif
samples_to_int16((int16_t *)data->data, samples, count);
mncc_write(buf, sizeof(buf));
@@ -934,6 +953,7 @@ void call_tx_audio(int callref, sample_t *samples, int count)
} else
/* else, if no sound is used, send test tone to mobile */
if (call.state == CALL_CONNECT) {
int16_t spl[count];
get_test_patterns(spl, count);
int16_to_samples(samples, spl, count);
call_rx_audio(callref, samples, count);
@@ -953,6 +973,13 @@ void call_mncc_clock(void)
data->callref = process->callref;
/* try to get patterns, else copy the samples we got */
get_process_patterns(process, (int16_t *)data->data, 160);
#ifdef DEBUG_LEVEL
sample_t samples[160];
int16_to_samples(samples, (int16_t *)data->data, 160);
double lev = level_of(samples, 160);
printf(" mobil-level: %s%.4f\n", debug_db(lev), (20 * log10(lev)));
samples_to_int16((int16_t *)data->data, samples, 160);
#endif
mncc_write(buf, sizeof(buf));
}
process = process->next;
@@ -978,6 +1005,10 @@ void call_mncc_recv(uint8_t *buf, int length)
if (is_process_pattern(data->callref))
return;
int16_to_samples(samples, (int16_t *)data->data, count);
#ifdef DEBUG_LEVEL
double lev = level_of(samples, count);
printf("festnetz-level: %s %.4f\n", debug_db(lev), (20 * log10(lev)));
#endif
call_rx_audio(data->callref, samples, count);
return;
}

7
src/common/compandor.c
View File

@@ -35,6 +35,9 @@
/* Minimum level value to keep state */
#define ENVELOPE_MIN 0.001
/* Maximum level, to prevent sqrt_tab to overflow */
#define ENVELOPE_MAX 9.990
static double sqrt_tab[10000];
/*
@@ -43,7 +46,7 @@ static double sqrt_tab[10000];
* Hopefully this is correct
*
*/
void init_compandor(compandor_t *state, int samplerate, double attack_ms, double recovery_ms, int unaffected_level)
void init_compandor(compandor_t *state, int samplerate, double attack_ms, double recovery_ms, double unaffected_level)
{
int i;
@@ -95,6 +98,8 @@ void compress_audio(compandor_t *state, sample_t *samples, int num)
envelope = peak;
if (envelope < ENVELOPE_MIN)
envelope = ENVELOPE_MIN;
if (envelope > ENVELOPE_MAX)
envelope = ENVELOPE_MAX;
value = value / sqrt_tab[(int)(envelope / 0.001)];
//if (i > 47000.0 && i < 48144)

2
src/common/compandor.h
View File

@@ -15,7 +15,7 @@ typedef struct compandor {
} e;
} compandor_t;
void init_compandor(compandor_t *state, int samplerate, double attack_ms, double recovery_ms, int unaffected_level);
void init_compandor(compandor_t *state, int samplerate, double attack_ms, double recovery_ms, double unaffected_level);
void compress_audio(compandor_t *state, sample_t *samples, int num);
void expand_audio(compandor_t *state, sample_t *samples, int num);

2
src/common/display.h
View File

@@ -23,7 +23,7 @@ void get_win_size(int *w, int *h);
void display_wave_init(sender_t *sender, int samplerate);
void display_wave_on(int on);
void display_wave_limit_scroll(int on);
void display_wave(sender_t *sender, sample_t *samples, int length);
void display_wave(sender_t *sender, sample_t *samples, int length, double range);
void display_iq_init(int samplerate);
void display_iq_on(int on);

16
src/common/display_wave.c
View File

@@ -20,6 +20,7 @@
#include <stdio.h>
#include <stdint.h>
#include <string.h>
#include <math.h>
#include <sys/ioctl.h>
#include "sample.h"
#include "sender.h"
@@ -110,10 +111,10 @@ void display_wave_limit_scroll(int on)
* HEIGHT is odd, so the center line's char is '-' (otherwise '_')
* (HEIGHT - 1) / 2 = 1, so the center line is drawn in line 1
*
* y is in range of 0..5, so these are 5 steps, where 2 to 2.999 is the
* y is in range of 0..4, so these are 5 steps, where 2 is the
* center line. this is calculated by (HEIGHT * 2 - 1)
*/
void display_wave(sender_t *sender, sample_t *samples, int length)
void display_wave(sender_t *sender, sample_t *samples, int length, double range)
{
dispwav_t *disp = &sender->dispwav;
int pos, max;
@@ -147,9 +148,16 @@ void display_wave(sender_t *sender, sample_t *samples, int length)
if (pos == width) {
memset(&screen, ' ', sizeof(screen));
for (j = 0; j < width; j++) {
y = (32767 - (int32_t)buffer[j]) * (HEIGHT * 2 - 1) / 65536;
/* Input value is scaled to range -1 .. 1 and then substracted from 1,
* so the result ranges from 0 .. 2.
* HEIGHT-1 is multiplied with the range, so a HEIGHT of 3 would allow
* 0..4 (5 steps) and a HEIGHT of 11 would allow 0..20 (21 steps).
* We always use odd number of steps, so there will be a center between
* values.
*/
y = round((1.0 - buffer[j] / range) * (double)(HEIGHT - 1));
/* only display level, if it is in range */
if (y >= 0 && y < HEIGHT * 2)
if (y >= 0 && y < HEIGHT * 2 - 1)
screen[y >> 1][j] = (y & 1) ? '_' : '-';
}
sprintf(screen[0], "(chan %d", sender->kanal);

34
src/common/dtmf.c
View File

@@ -25,10 +25,12 @@
#define PI M_PI
#define TX_PEAK_DTMF 7000 /* single dtmf tone peak (note this is half to total peak) */
#define DTMF_DURATION 0.100 /* duration in seconds */
static double tx_peak_dtmf_low = 0.2818 / SPEECH_LEVEL; /* -11 dBm, relative to speech level */
static double tx_peak_dtmf_high = 0.3548 / SPEECH_LEVEL;/* -9 dBm, relative to speech level */
#define DTMF_DURATION 0.100 /* duration in seconds */
static double dsp_sine_dtmf[256];
static sample_t dsp_sine_dtmf_low[65536];
static sample_t dsp_sine_dtmf_high[65536];
void dtmf_init(dtmf_t *dtmf, int samplerate)
{
@@ -39,8 +41,10 @@ void dtmf_init(dtmf_t *dtmf, int samplerate)
dtmf->max = (int)((double)samplerate * DTMF_DURATION + 0.5);
// FIXME: do this globally and not per instance */
for (i = 0; i < 256; i++)
dsp_sine_dtmf[i] = (int)(sin((double)i / 256.0 * 2.0 * PI) * TX_PEAK_DTMF);
for (i = 0; i < 65536; i++) {
dsp_sine_dtmf_low[i] = sin((double)i / 65536.0 * 2.0 * PI) * tx_peak_dtmf_low;
dsp_sine_dtmf_high[i] = sin((double)i / 65536.0 * 2.0 * PI) * tx_peak_dtmf_high;
}
}
/* set dtmf tone */
@@ -71,8 +75,8 @@ void dtmf_set_tone(dtmf_t *dtmf, char tone)
}
dtmf->tone = tone;
dtmf->pos = 0;
dtmf->phaseshift256[0] = 256.0 / ((double)dtmf->samplerate / f1);
dtmf->phaseshift256[1] = 256.0 / ((double)dtmf->samplerate / f2);
dtmf->phaseshift65536[0] = 65536.0 / ((double)dtmf->samplerate / f1);
dtmf->phaseshift65536[1] = 65536.0 / ((double)dtmf->samplerate / f2);
}
/* Generate audio stream from DTMF tone. Keep phase for next call of function. */
@@ -87,20 +91,20 @@ void dtmf_tone(dtmf_t *dtmf, sample_t *samples, int length)
return;
}
phaseshift = dtmf->phaseshift256;
phase = dtmf->phase256;
phaseshift = dtmf->phaseshift65536;
phase = dtmf->phase65536;
pos = dtmf->pos;
max = dtmf->max;
for (i = 0; i < length; i++) {
*samples++ = dsp_sine_dtmf[((uint8_t)phase[0]) & 0xff]
+ dsp_sine_dtmf[((uint8_t)phase[1]) & 0xff];
*samples++ = dsp_sine_dtmf_low[(uint16_t)phase[0]]
+ dsp_sine_dtmf_high[(uint16_t)phase[1]];
phase[0] += phaseshift[0];
if (phase[0] >= 256)
phase[0] -= 256;
if (phase[0] >= 65536)
phase[0] -= 65536;
phase[1] += phaseshift[1];
if (phase[1] >= 256)
phase[1] -= 256;
if (phase[1] >= 65536)
phase[1] -= 65536;
/* tone ends */
if (++pos == max) {

4
src/common/dtmf.h
View File

@@ -4,8 +4,8 @@ typedef struct dtmf {
char tone; /* current tone to be played */
int pos; /* sample counter for tone */
int max; /* max number of samples for tone duration */
double phaseshift256[2]; /* how much the phase of sine wave changes per sample */
double phase256[2]; /* current phase */
double phaseshift65536[2]; /* how much the phase of sine wave changes per sample */
double phase65536[2]; /* current phase */
} dtmf_t;
void dtmf_init(dtmf_t *dtmf, int samplerate);

13
src/common/emphasis.c
View File

@@ -21,6 +21,7 @@
#include <stdint.h>
#include <string.h>
#include <math.h>
#include "sample.h"
#include "filter.h"
#include "emphasis.h"
#include "debug.h"
@@ -29,7 +30,7 @@
#define CUT_OFF_H 100.0 /* cut-off frequency for high-pass filter */
static void gen_sine(double *samples, int num, int samplerate, double freq)
static void gen_sine(sample_t *samples, int num, int samplerate, double freq)
{
int i;
@@ -37,7 +38,7 @@ static void gen_sine(double *samples, int num, int samplerate, double freq)
samples[i] = cos(2.0 * M_PI * freq / (double)samplerate * (double)i);
}
static double get_level(double *samples, int num)
static double get_level(sample_t *samples, int num)
{
int i;
double envelope = 0;
@@ -52,7 +53,7 @@ static double get_level(double *samples, int num)
int init_emphasis(emphasis_t *state, int samplerate, double cut_off)
{
double factor;
double test_samples[samplerate / 10];
sample_t test_samples[samplerate / 10];
memset(state, 0, sizeof(*state));
@@ -78,7 +79,7 @@ int init_emphasis(emphasis_t *state, int samplerate, double cut_off)
return 0;
}
void pre_emphasis(emphasis_t *state, double *samples, int num)
void pre_emphasis(emphasis_t *state, sample_t *samples, int num)
{
double x, y, x_last, factor, amp;
int i;
@@ -101,7 +102,7 @@ void pre_emphasis(emphasis_t *state, double *samples, int num)
state->p.x_last = x_last;
}
void de_emphasis(emphasis_t *state, double *samples, int num)
void de_emphasis(emphasis_t *state, sample_t *samples, int num)
{
double x, y, y_last, factor, amp;
int i;
@@ -125,7 +126,7 @@ void de_emphasis(emphasis_t *state, double *samples, int num)
}
/* high pass filter to remove DC and low frequencies */
void dc_filter(emphasis_t *state, double *samples, int num)
void dc_filter(emphasis_t *state, sample_t *samples, int num)
{
filter_process(&state->d.hp, samples, num);
}

6
src/common/emphasis.h
View File

@@ -15,7 +15,7 @@ typedef struct emphasis {
#define CUT_OFF_EMPHASIS_DEFAULT 300.0
int init_emphasis(emphasis_t *state, int samplerate, double cut_off);
void pre_emphasis(emphasis_t *state, double *samples, int num);
void de_emphasis(emphasis_t *state, double *samples, int num);
void dc_filter(emphasis_t *state, double *samples, int num);
void pre_emphasis(emphasis_t *state, sample_t *samples, int num);
void de_emphasis(emphasis_t *state, sample_t *samples, int num);
void dc_filter(emphasis_t *state, sample_t *samples, int num);

3
src/common/filter.c
View File

@@ -22,6 +22,7 @@
#include <string.h>
#include <stdlib.h>
#include <math.h>
#include "sample.h"
#include "filter.h"
#define PI M_PI
@@ -65,7 +66,7 @@ void filter_highpass_init(filter_t *bq, double frequency, int samplerate, int it
bq->b2 = (1 - K / Q + K * K) * norm;
}
void filter_process(filter_t *bq, double *samples, int length)
void filter_process(filter_t *bq, sample_t *samples, int length)
{
double a0, a1, a2, b1, b2;
double *z1, *z2;

2
src/common/filter.h
View File

@@ -9,6 +9,6 @@ typedef struct filter {
void filter_lowpass_init(filter_t *bq, double frequency, int samplerate, int iterations);
void filter_highpass_init(filter_t *bq, double frequency, int samplerate, int iterations);
void filter_process(filter_t *bq, double *samples, int length);
void filter_process(filter_t *bq, sample_t *samples, int length);
#endif /* _FILTER_H */

59
src/common/sample.c
View File

@@ -1,27 +1,64 @@
/* Sample definition
*
* (C) 2017 by Andreas Eversberg <jolly@eversberg.eu>
* All Rights Reserved
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdint.h>
#include "sample.h"
#include "sample.h"
/*
* A regular voice conversation takes place at this factor below the full range
* of 16 bits signed value:
*/
static double int_16_speech_level = SPEECH_LEVEL * 0.7079; /* 16 dBm below dBm0, which is about 3dBm below full 16 bit range */
/* A sample_t is a value that has virtually infinite precision but will also
* support high numbers. 'double' or 'float' types are sufficient.
*
* When using sample_t inside signal processing of each base station, the
* level of +- 1 is relative to the normal speach evenlope.
*
* When converting sample_t to int16_t, the level of +- 1 is reduced by factor.
* This way the speech may be louder before clipping happens.
*
* When using sample_t to modulate (SDR or sound card), the level is changed,
* so it represents the frequency deviation in Hz. The deviation of speech
* envelope is network dependant.
*/
void samples_to_int16(int16_t *spl, sample_t *samples, int length)
{
int32_t value;
while (length--) {
if (*samples > 32767.0)
*spl = 32767;
else if (*samples < -32767.0)
*spl = -32767;
value = *samples++ * int_16_speech_level * 32768.0;
if (value > 32767.0)
*spl++ = 32767;
else if (value < -32767.0)
*spl++ = -32767;
else
*spl = (uint16_t)(*samples);
samples++;
spl++;
*spl++ = (uint16_t)value;
}
}
void int16_to_samples(sample_t *samples, int16_t *spl, int length)
{
while (length--) {
*samples = (double)(*spl);
samples++;
spl++;
*samples++ = (double)(*spl++) / 32767.0 / int_16_speech_level;
}
}

2
src/common/sample.h
View File

@@ -1,6 +1,8 @@
typedef double sample_t;
#define SPEECH_LEVEL 0.1585
void samples_to_int16(int16_t *spl, sample_t *samples, int length);
void int16_to_samples(sample_t *samples, int16_t *spl, int length);

2
src/common/samplerate.c
View File

@@ -124,7 +124,7 @@ int samplerate_upsample(samplerate_t *state, sample_t *input, int input_num, sam
if (input == output) {
/* copy samples */
for (i = 0; i < input_num; i++)
for (i = 0; i < output_num; i++)
*output++ = samples[i];
}

60
src/common/sdr.c
View File

@@ -47,7 +47,6 @@ typedef struct sdr {
sdr_chan_t *chan; /* settings for all channels */
int paging_channel; /* if set, points to paging channel */
sdr_chan_t paging_chan; /* settings for extra paging channel */
double spl_deviation; /* how to convert a sample step into deviation (Hz) */
int channels; /* number of frequencies */
double samplerate; /* IQ rate */
double amplitude; /* amplitude of each carrier */
@@ -84,15 +83,19 @@ int sdr_init(const char *device_args, double rx_gain, double tx_gain, const char
return 0;
}
void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_frequency, double *rx_frequency, int channels, double paging_frequency, int samplerate, double bandwidth, double sample_deviation)
void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_frequency, double *rx_frequency, int channels, double paging_frequency, int samplerate, double max_deviation, double max_modulation)
{
sdr_t *sdr;
double bandwidth;
double tx_center_frequency, rx_center_frequency;
int rc;
int c;
display_iq_init(samplerate);
bandwidth = 2.0 * (max_deviation + max_modulation);
PDEBUG(DSDR, DEBUG_INFO, "Using Bandwidth of 2 * (%.1f + %.1f) = %.1f\n", max_deviation / 1000, max_modulation / 1000, bandwidth / 1000);
if (channels < 1) {
PDEBUG(DSDR, DEBUG_ERROR, "No channel given, please fix!\n");
abort();
@@ -105,7 +108,6 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
}
sdr->channels = channels;
sdr->samplerate = samplerate;
sdr->spl_deviation = sample_deviation;
sdr->amplitude = 0.4 / (double)channels; // FIXME: actual amplitude 0.1?
/* special case where we use a paging frequency */
@@ -124,8 +126,8 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
PDEBUG(DSDR, DEBUG_INFO, "Frequency #%d: TX = %.6f MHz, RX = %.6f MHz\n", c, tx_frequency[c] / 1e6, rx_frequency[c] / 1e6);
sdr->chan[c].tx_frequency = tx_frequency[c];
sdr->chan[c].rx_frequency = rx_frequency[c];
filter_lowpass_init(&sdr->chan[c].rx_lp[0], bandwidth, samplerate, 1);
filter_lowpass_init(&sdr->chan[c].rx_lp[1], bandwidth, samplerate, 1);
filter_lowpass_init(&sdr->chan[c].rx_lp[0], bandwidth / 2.0, samplerate, 1);
filter_lowpass_init(&sdr->chan[c].rx_lp[1], bandwidth / 2.0, samplerate, 1);
}
if (sdr->paging_channel) {
PDEBUG(DSDR, DEBUG_INFO, "Paging Frequency: TX = %.6f MHz\n", paging_frequency / 1e6);
@@ -190,21 +192,21 @@ void *sdr_open(const char __attribute__((__unused__)) *audiodev, double *tx_freq
PDEBUG(DSDR, DEBUG_INFO, "Using gain: TX %.1f dB, RX %.1f dB\n", sdr_tx_gain, sdr_rx_gain);
if (sdr_write_iq_rx_wave) {
rc = wave_create_record(&sdr->wave_rx_rec, sdr_write_iq_rx_wave, sdr->samplerate, 2);
rc = wave_create_record(&sdr->wave_rx_rec, sdr_write_iq_rx_wave, sdr->samplerate, 2, 1.0);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
goto error;
}
}
if (sdr_write_iq_tx_wave) {
rc = wave_create_record(&sdr->wave_tx_rec, sdr_write_iq_tx_wave, sdr->samplerate, 2);
rc = wave_create_record(&sdr->wave_tx_rec, sdr_write_iq_tx_wave, sdr->samplerate, 2, 1.0);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
goto error;
}
}
if (sdr_read_iq_rx_wave) {
rc = wave_create_playback(&sdr->wave_rx_play, sdr_read_iq_rx_wave, sdr->samplerate, 2);
rc = wave_create_playback(&sdr->wave_rx_play, sdr_read_iq_rx_wave, sdr->samplerate, 2, 1.0);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE playback instance!\n");
goto error;
@@ -269,7 +271,7 @@ int sdr_write(void *inst, sample_t **samples, int num, enum paging_signal __attr
/* modulate */
for (s = 0, ss = 0; s < num; s++) {
/* deviation is defined by the sample value and the offset */
dev = offset + (double)samples[c][s] * sdr->spl_deviation;
dev = offset + samples[c][s];
#ifdef FAST_SINE
phase += 256.0 * dev / rate;
if (phase < 0.0)
@@ -294,20 +296,8 @@ int sdr_write(void *inst, sample_t **samples, int num, enum paging_signal __attr
if (sdr->wave_tx_rec.fp) {
sample_t spl[2][num], *spl_list[2] = { spl[0], spl[1] };
for (s = 0, ss = 0; s < num; s++) {
if (buff[ss] >= 1.0)
spl[0][s] = 32767.0;
else if (buff[ss] <= -1.0)
spl[0][s] = -32767.0;
else
spl[0][s] = 32767.0 * buff[ss];
ss++;
if (buff[ss] >= 1.0)
spl[1][s] = 32767.0;
else if (buff[ss] <= -1.0)
spl[1][s] = -32767.0;
else
spl[1][s] = 32767.0 * buff[ss];
ss++;
spl[0][s] = buff[ss++];
spl[1][s] = buff[ss++];
}
wave_write(&sdr->wave_tx_rec, spl_list, num);
}
@@ -325,7 +315,7 @@ int sdr_read(void *inst, sample_t **samples, int num, int channels)
{
sdr_t *sdr = (sdr_t *)inst;
float buff[num * 2];
double I[num], Q[num], i, q;
sample_t I[num], Q[num], i, q;
int count;
int c, s, ss;
double phase, rot, last_phase, dev, rate;
@@ -341,20 +331,8 @@ int sdr_read(void *inst, sample_t **samples, int num, int channels)
if (sdr->wave_rx_rec.fp) {
sample_t spl[2][count], *spl_list[2] = { spl[0], spl[1] };
for (s = 0, ss = 0; s < count; s++) {
if (buff[ss] >= 1.0)
spl[0][s] = 32767.0;
else if (buff[ss] <= -1.0)
spl[0][s] = -32767.0;
else
spl[0][s] = 32767.0 * buff[ss];
ss++;
if (buff[ss] >= 1.0)
spl[1][s] = 32767.0;
else if (buff[ss] <= -1.0)
spl[1][s] = -32767.0;
else
spl[1][s] = 32767.0 * buff[ss];
ss++;
spl[0][s] = buff[ss++];
spl[1][s] = buff[ss++];
}
wave_write(&sdr->wave_rx_rec, spl_list, count);
}
@@ -362,8 +340,8 @@ int sdr_read(void *inst, sample_t **samples, int num, int channels)
sample_t spl[2][count], *spl_list[2] = { spl[0], spl[1] };
wave_read(&sdr->wave_rx_play, spl_list, count);
for (s = 0, ss = 0; s < count; s++) {
buff[ss++] = spl[0][s] / 32767.0;
buff[ss++] = spl[1][s] / 32767.0;
buff[ss++] = spl[0][s];
buff[ss++] = spl[1][s];
}
}
display_iq(buff, count);
@@ -391,7 +369,7 @@ int sdr_read(void *inst, sample_t **samples, int num, int channels)
else if (dev > 0.49)
dev -= 1.0;
dev *= rate;
samples[c][s] = dev / sdr->spl_deviation;
samples[c][s] = dev;
}
sdr->chan[c].rx_last_phase = last_phase;
}

29
src/common/sender.c
View File

@@ -41,8 +41,6 @@ int sender_create(sender_t *sender, int kanal, double sendefrequenz, double empf
sender->kanal = kanal;
sender->sendefrequenz = sendefrequenz;
sender->empfangsfrequenz = empfangsfrequenz;
sender->bandwidth = 4000; /* default is overwritten by dsp.c */
sender->sample_deviation = 0.2; /* default is overwritten by dsp.c */
strncpy(sender->audiodev, audiodev, sizeof(sender->audiodev) - 1);
sender->samplerate = samplerate;
sender->rx_gain = rx_gain;
@@ -168,21 +166,21 @@ int sender_open_audio(void)
}
if (master->write_rx_wave) {
rc = wave_create_record(&master->wave_rx_rec, master->write_rx_wave, master->samplerate, channels);
rc = wave_create_record(&master->wave_rx_rec, master->write_rx_wave, master->samplerate, channels, master->max_deviation);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
return rc;
}
}
if (master->write_tx_wave) {
rc = wave_create_record(&master->wave_tx_rec, master->write_tx_wave, master->samplerate, channels);
rc = wave_create_record(&master->wave_tx_rec, master->write_tx_wave, master->samplerate, channels, master->max_deviation);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE recoding instance!\n");
return rc;
}
}
if (master->read_rx_wave) {
rc = wave_create_playback(&master->wave_rx_play, master->read_rx_wave, master->samplerate, channels);
rc = wave_create_playback(&master->wave_rx_play, master->read_rx_wave, master->samplerate, channels, master->max_deviation);
if (rc < 0) {
PDEBUG(DSENDER, DEBUG_ERROR, "Failed to create WAVE playback instance!\n");
return rc;
@@ -190,7 +188,7 @@ int sender_open_audio(void)
}
/* open device */
master->audio = master->audio_open(master->audiodev, tx_f, rx_f, channels, paging_frequency, master->samplerate, master->bandwidth, master->sample_deviation);
master->audio = master->audio_open(master->audiodev, tx_f, rx_f, channels, paging_frequency, master->samplerate, master->max_deviation, master->max_modulation);
if (!master->audio) {
PDEBUG(DSENDER, DEBUG_ERROR, "No audio device!\n");
return -EIO;
@@ -225,6 +223,17 @@ void sender_destroy(sender_t *sender)
jitter_destroy(&sender->dejitter);
}
void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation, double dBm0_deviation, double max_display)
{
sender->max_deviation = max_deviation;
sender->max_modulation = max_modulation;
sender->dBm0_deviation = dBm0_deviation;
sender->max_display = max_display;
PDEBUG_CHAN(DSENDER, DEBUG_DEBUG, "Maxium deviation: %.1f kHz, Maximum modulation: %.1f kHz\n", max_deviation / 1000.0, max_modulation / 1000.0);
PDEBUG_CHAN(DSENDER, DEBUG_DEBUG, "Deviation at dBm0 (audio level): %.1f kHz\n", dBm0_deviation / 1000.0);
}
static void gain_samples(sample_t *samples, int length, double gain)
{
int i;
@@ -278,12 +287,14 @@ cant_recover:
sender_send(inst, samples[i], count);
/* internal loopback: loop back TX audio to RX */
if (inst->loopback == 1) {
display_wave(inst, samples[i], count);
display_wave(inst, samples[i], count, inst->max_display);
sender_receive(inst, samples[i], count);
}
/* do pre emphasis towards radio */
if (inst->pre_emphasis)
pre_emphasis(&inst->estate, samples[i], count);
/* normal level to frequency deviation of dBm0 */
gain_samples(samples[i], count, inst->dBm0_deviation);
/* set paging signal */
paging_signal[i] = inst->paging_signal;
on[i] = inst->paging_on;
@@ -328,6 +339,8 @@ transmit_later:
/* loop through all channels */
for (i = 0, inst = sender; inst; i++, inst = inst->slave) {
/* frequency deviation of dBm0 to normal level */
gain_samples(samples[i], count, 1.0 / inst->dBm0_deviation);
/* rx gain */
if (inst->rx_gain != 1.0)
gain_samples(samples[i], count, inst->rx_gain);
@@ -337,7 +350,7 @@ transmit_later:
de_emphasis(&inst->estate, samples[i], count);
}
if (inst->loopback != 1) {
display_wave(inst, samples[i], count);
display_wave(inst, samples[i], count, inst->max_display);
sender_receive(inst, samples[i], count);
}
if (inst->loopback == 3)

9
src/common/sender.h
View File

@@ -31,8 +31,12 @@ typedef struct sender {
double sendefrequenz; /* transmitter frequency */
double empfangsfrequenz; /* receiver frequency */
double ruffrequenz; /* special paging frequency used for B-Netz */
double bandwidth; /* max NF frequency to be transmitted unaffected by filtering */
double sample_deviation; /* frequency deviation of one sample step (after pre-emphasis) */
/* fm levels */
double max_deviation; /* max frequency deviation */
double max_modulation; /* max frequency modulated */
double dBm0_deviation; /* deviation of 1000 Hz reference tone at dBm0 */
double max_display; /* level of displaying wave form */
/* audio */
void *audio;
@@ -85,6 +89,7 @@ extern int cant_recover;
int sender_create(sender_t *sender, int kanal, double sendefrequenz, double empfangsfrequenz, const char *audiodev, int samplerate, double rx_gain, int pre_emphasis, int de_emphasis, const char *write_rx_wave, const char *write_tx_wave, const char *read_rx_wave, int loopback, double loss_volume, enum paging_signal paging_signal);
void sender_destroy(sender_t *sender);
void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation, double dBm0_deviation, double max_display);
int sender_open_audio(void);
void process_sender_audio(sender_t *sender, int *quit, int latspl);
void sender_send(sender_t *sender, sample_t *samples, int count);

24
src/common/sound_alsa.c
View File

@@ -27,6 +27,7 @@
typedef struct sound {
snd_pcm_t *phandle, *chandle;
int pchannels, cchannels;
double spl_deviation; /* how much deviation is one sample step */
double paging_phaseshift; /* phase to shift every sample */
double paging_phase; /* current phase */
} sound_t;
@@ -131,7 +132,7 @@ static int sound_prepare(sound_t *sound)
return 0;
}
void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int channels, double __attribute__((unused)) paging_frequency, int samplerate, double __attribute__((unused)) bandwidth, double __attribute__((unused)) sample_deviation)
void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int channels, double __attribute__((unused)) paging_frequency, int samplerate, double max_deviation, double __attribute__((unused)) max_modulation)
{
sound_t *sound;
int rc;
@@ -147,6 +148,7 @@ void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_freque
return NULL;
}
sound->spl_deviation = max_deviation / 32767.0;
sound->paging_phaseshift = 1.0 / ((double)samplerate / 1000.0);
rc = snd_pcm_open(&sound->phandle, audiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
@@ -252,6 +254,7 @@ static void gen_paging_tone(sound_t *sound, int16_t *samples, int length, enum p
int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *paging_signal, int *on, int channels)
{
sound_t *sound = (sound_t *)inst;
double spl_deviation = sound->spl_deviation;
int32_t value;
int16_t buff[num << 1];
int rc;
@@ -263,7 +266,7 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
int16_t paging[num << 1];
gen_paging_tone(sound, paging, num, paging_signal[0], on[0]);
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i];
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
@@ -273,13 +276,13 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
}
} else if (channels == 2) {
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i];
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
value = samples[1][i];
value = samples[1][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
@@ -288,7 +291,7 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
}
} else {
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i];
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
@@ -300,7 +303,7 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
} else {
/* one channel */
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i];
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
@@ -328,6 +331,7 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
int sound_read(void *inst, sample_t **samples, int num, int channels)
{
sound_t *sound = (sound_t *)inst;
double spl_deviation = sound->spl_deviation;
int16_t buff[num << 1];
int32_t spl;
int in, rc;
@@ -361,17 +365,17 @@ int sound_read(void *inst, sample_t **samples, int num, int channels)
for (i = 0, ii = 0; i < rc; i++) {
spl = buff[ii++];
spl += buff[ii++];
samples[0][i] = (sample_t)spl;
samples[0][i] = (double)spl * spl_deviation;
}
} else {
for (i = 0, ii = 0; i < rc; i++) {
samples[0][i] = (sample_t)buff[ii++];
samples[1][i] = (sample_t)buff[ii++];
samples[0][i] = (double)buff[ii++] * spl_deviation;
samples[1][i] = (double)buff[ii++] * spl_deviation;
}
}
} else {
for (i = 0, ii = 0; i < rc; i++) {
samples[0][i] = (sample_t)buff[ii++];
samples[0][i] = (double)buff[ii++] * spl_deviation;
}
}

18
src/common/wave.c
View File

@@ -34,7 +34,7 @@ struct fmt {
uint16_t bits_sample; /* bits per sample (one channel) */
};
int wave_create_record(wave_rec_t *rec, const char *filename, int samplerate, int channels)
int wave_create_record(wave_rec_t *rec, const char *filename, int samplerate, int channels, double max_deviation)
{
/* RIFFxxxxWAVEfmt xxxx(fmt size)dataxxxx... */
char dummyheader[4 + 4 + 4 + 4 + 4 + sizeof(struct fmt) + 4 + 4];
@@ -43,6 +43,7 @@ int wave_create_record(wave_rec_t *rec, const char *filename, int samplerate, in
memset(rec, 0, sizeof(*rec));
rec->samplerate = samplerate;
rec->channels = channels;
rec->max_deviation = max_deviation;
rec->fp = fopen(filename, "w");
if (!rec->fp) {
@@ -58,7 +59,7 @@ int wave_create_record(wave_rec_t *rec, const char *filename, int samplerate, in
return 0;
}
int wave_create_playback(wave_play_t *play, const char *filename, int samplerate, int channels)
int wave_create_playback(wave_play_t *play, const char *filename, int samplerate, int channels, double max_deviation)
{
uint8_t buffer[256];
struct fmt fmt;
@@ -68,6 +69,7 @@ int wave_create_playback(wave_play_t *play, const char *filename, int samplerate
memset(play, 0, sizeof(*play));
play->channels = channels;
play->max_deviation = max_deviation;
play->fp = fopen(filename, "r");
if (!play->fp) {
@@ -196,6 +198,8 @@ error:
int wave_read(wave_play_t *play, sample_t **samples, int length)
{
double max_deviation = play->max_deviation;
int16_t value; /* must be int16, so assembling bytes work */
uint8_t buff[2 * length * play->channels];
int __attribute__((__unused__)) len;
int i, j, c;
@@ -212,11 +216,12 @@ int wave_read(wave_play_t *play, sample_t **samples, int length)
if (!play->left)
printf("*** Finished reading WAVE file.\n");
/* read and correct endiness */
/* read and correct endianness */
len = fread(buff, 1, 2 * length * play->channels, play->fp);
for (i = 0, j = 0; i < length; i++) {
for (c = 0; c < play->channels; c++) {
samples[c][i] = (double)(buff[j] + (buff[j + 1] << 8));
value = buff[j] + (buff[j + 1] << 8);
samples[c][i] = (double)value / 32767.0 * max_deviation;
j += 2;
}
}
@@ -226,15 +231,16 @@ int wave_read(wave_play_t *play, sample_t **samples, int length)
int wave_write(wave_rec_t *rec, sample_t **samples, int length)
{
double max_deviation = rec->max_deviation;
int32_t value;
uint8_t buff[2 * length * rec->channels];
int __attribute__((__unused__)) len;
int i, j, c;
/* write and correct endiness */
/* write and correct endianness */
for (i = 0, j = 0; i < length; i++) {
for (c = 0; c < rec->channels; c++) {
value = samples[c][i];
value = samples[c][i] / max_deviation * 32767.0;
if (value > 32767)
value = 32767;
else if (value < -32767)

6
src/common/wave.h
View File

@@ -2,6 +2,7 @@
typedef struct wave_rec {
FILE *fp;
int channels;
double max_deviation;
int samplerate;
uint32_t written; /* how much samples written */
} wave_rec_t;
@@ -9,11 +10,12 @@ typedef struct wave_rec {
typedef struct wave_play {
FILE *fp;
int channels;
double max_deviation;
uint32_t left; /* how much samples left */
} wave_play_t;
int wave_create_record(wave_rec_t *rec, const char *filename, int samplerate, int channels);
int wave_create_playback(wave_play_t *play, const char *filename, int samplerate, int channels);
int wave_create_record(wave_rec_t *rec, const char *filename, int samplerate, int channels, double max_deviation);
int wave_create_playback(wave_play_t *play, const char *filename, int samplerate, int channels, double max_deviation);
int wave_read(wave_play_t *play, sample_t **samples, int length);
int wave_write(wave_rec_t *rec, sample_t **samples, int length);
void wave_destroy_record(wave_rec_t *rec);