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-rw-r--r--src/amps/dsp.c5
-rw-r--r--src/anetz/dsp.c10
-rw-r--r--src/bnetz/dsp.c5
-rw-r--r--src/cnetz/dsp.c5
-rw-r--r--src/common/call.c2
-rw-r--r--src/common/sender.c4
-rw-r--r--src/common/sender.h4
-rw-r--r--src/common/sound.h2
-rw-r--r--src/common/sound_alsa.c2
-rw-r--r--src/nmt/dsp.c5
10 files changed, 35 insertions, 9 deletions
diff --git a/src/amps/dsp.c b/src/amps/dsp.c
index 776ec3b..4b7e08b 100644
--- a/src/amps/dsp.c
+++ b/src/amps/dsp.c
@@ -99,6 +99,7 @@
#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 */
@@ -191,6 +192,10 @@ int dsp_init_sender(amps_t *amps, int high_pass, 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;
+
if (amps->sender.samplerate < 96000) {
PDEBUG(DDSP, DEBUG_ERROR, "Sample rate must be at least 96000 Hz to process FSK and SAT signals.\n");
return -EINVAL;
diff --git a/src/anetz/dsp.c b/src/anetz/dsp.c
index 41888cd..e47f7f2 100644
--- a/src/anetz/dsp.c
+++ b/src/anetz/dsp.c
@@ -35,12 +35,10 @@
#define PI 3.1415927
/* signaling */
-/* NOTE: The peak deviation is similar for paging tone and signaling tone,
- * so both tones should be equal after pre-emphasis. This is why the paging
- * tones is so much louder.*/
+#define BANDWIDTH 15000.0 /* maximum bandwidth */
#define TX_PEAK_TONE 8192.0 /* peak amplitude for all tones */
+#warning FIXME: only with emphasis, use seperate option for volume, override by sdr
#define TX_PEAK_PAGE 32766.0 /* peak amplitude paging tone */
-// FIXME: what is the allowed deviation of tone?
#define CHUNK_DURATION 0.010 /* 10 ms */
// FIXME: how long until we detect a tone?
@@ -94,6 +92,10 @@ int dsp_init_sender(anetz_t *anetz, 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;
+
anetz->page_sequence = page_sequence;
audio_init_loss(&anetz->sender.loss, LOSS_INTERVAL, anetz->sender.loss_volume, LOSS_TIME);
diff --git a/src/bnetz/dsp.c b/src/bnetz/dsp.c
index 2e2a7f1..90a1821 100644
--- a/src/bnetz/dsp.c
+++ b/src/bnetz/dsp.c
@@ -35,6 +35,7 @@
#define PI 3.1415927
/* signaling */
+#define BANDWIDTH 5000.0 /* maximum bandwidth */
#define TX_PEAK_TONE 5000.0 /* peak amplitude for all tones */
#define BIT_DURATION 0.010 /* bit length: 10 ms */
#define FILTER_STEP 0.001 /* step every 1 ms */
@@ -85,6 +86,10 @@ int dsp_init_sender(bnetz_t *bnetz)
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for 'Sender'.\n");
+ /* set deviation and modulation parameters */
+ bnetz->sender.bandwidth = BANDWIDTH;
+ bnetz->sender.sample_deviation = 1250.0 / (double)TX_PEAK_TONE; // FIXME: calc real value
+
audio_init_loss(&bnetz->sender.loss, LOSS_INTERVAL, bnetz->sender.loss_volume, LOSS_TIME);
bnetz->samples_per_bit = bnetz->sender.samplerate * BIT_DURATION;
diff --git a/src/cnetz/dsp.c b/src/cnetz/dsp.c
index a35e8c8..ae453c6 100644
--- a/src/cnetz/dsp.c
+++ b/src/cnetz/dsp.c
@@ -42,6 +42,7 @@ extern int voice_deviation;
#define PI M_PI
+#define BANDWIDTH 5500.0 /* maximum bandwidth */
#define FSK_DEVIATION 10000
#define COMPANDOR_0DB 30000
#define BITRATE 5280.0 /* bits per second */
@@ -88,6 +89,10 @@ int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], do
PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init FSK for 'Sender'.\n");
+ /* set deviation and modulation parameters */
+ cnetz->sender.bandwidth = BANDWIDTH;
+ cnetz->sender.sample_deviation = 2500.0 / (double)FSK_DEVIATION;
+
if (measure_speed) {
cnetz->measure_speed = measure_speed;
cant_recover = 1;
diff --git a/src/common/call.c b/src/common/call.c
index 7f8fae6..ab57390 100644
--- a/src/common/call.c
+++ b/src/common/call.c
@@ -473,7 +473,7 @@ 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, samplerate);
+ call.sound = sound_open(audiodev, NULL, NULL, 1, samplerate, 3700.0, 0.0);
if (!call.sound) {
PDEBUG(DSENDER, DEBUG_ERROR, "No sound device!\n");
diff --git a/src/common/sender.c b/src/common/sender.c
index 45760e1..3438554 100644
--- a/src/common/sender.c
+++ b/src/common/sender.c
@@ -40,6 +40,8 @@ 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;
@@ -172,7 +174,7 @@ int sender_open_audio(void)
}
/* open device */
- master->audio = master->audio_open(master->audiodev, tx_f, rx_f, channels, master->samplerate);
+ master->audio = master->audio_open(master->audiodev, tx_f, rx_f, channels, master->samplerate, master->bandwidth, master->sample_deviation);
if (!master->audio) {
PDEBUG(DSENDER, DEBUG_ERROR, "No audio device!\n");
return -EIO;
diff --git a/src/common/sender.h b/src/common/sender.h
index c89b405..94bdee8 100644
--- a/src/common/sender.h
+++ b/src/common/sender.h
@@ -27,11 +27,13 @@ typedef struct sender {
int kanal; /* channel number */
double sendefrequenz; /* transmitter frequency */
double empfangsfrequenz; /* receiver frequency */
+ double bandwidth; /* max NF frequency to be transmitted unaffected by filtering */
+ double sample_deviation; /* frequency deviation of one sample step (after pre-emphasis) */
/* audio */
void *audio;
char audiodev[64]; /* audio device name (alsa or sdr) */
- void *(*audio_open)(const char *, double *, double *, int, int);
+ void *(*audio_open)(const char *, double *, double *, int, int, double, double);
void (*audio_close)(void *);
int (*audio_write)(void *, int16_t **, int, int);
int (*audio_read)(void *, int16_t **, int, int);
diff --git a/src/common/sound.h b/src/common/sound.h
index 45eb2c0..313a559 100644
--- a/src/common/sound.h
+++ b/src/common/sound.h
@@ -1,5 +1,5 @@
-void *sound_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int channels, int samplerate);
+void *sound_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int channels, int samplerate, double bandwidth, double sample_deviation);
void sound_close(void *inst);
int sound_write(void *inst, int16_t **samples, int num, int channels);
int sound_read(void *inst, int16_t **samples, int num, int channels);
diff --git a/src/common/sound_alsa.c b/src/common/sound_alsa.c
index 15fe1a9..c8d0668 100644
--- a/src/common/sound_alsa.c
+++ b/src/common/sound_alsa.c
@@ -128,7 +128,7 @@ 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, int samplerate)
+void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int channels, int samplerate, double __attribute__((unused)) bandwidth, double __attribute__((unused)) sample_deviation)
{
sound_t *sound;
int rc;
diff --git a/src/nmt/dsp.c b/src/nmt/dsp.c
index 49f7fb4..e3ff165 100644
--- a/src/nmt/dsp.c
+++ b/src/nmt/dsp.c
@@ -43,6 +43,7 @@
*/
/* 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 */
@@ -115,6 +116,10 @@ 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
+
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);
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-26 19:54:52 +0000