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AMPS: Add option to use the BUSY/IDLE bit on FOCC

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This commit is contained in:
Andreas Eversberg
2016-06-20 17:16:46 +02:00
parent 76f6285846
commit 86daa5a64a
7 changed files with 133 additions and 129 deletions
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172
src/amps/dsp.c
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@@ -158,7 +158,7 @@ int dsp_init_sender(amps_t *amps, int high_pass)
amps->fsk_bitstep = 1.0 / amps->fsk_bitduration;
PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", amps->fsk_bitduration, amps->sender.samplerate);
amps->fsk_tx_buffer_size = amps->fsk_bitduration * (double)FSK_MAX_BITS + 10; /* 10 extra to avoid overflow due to routing */
amps->fsk_tx_buffer_size = amps->fsk_bitduration + 10; /* 10 extra to avoid overflow due to rounding */
amps->fsk_tx_buffer = calloc(sizeof(int16_t), amps->fsk_tx_buffer_size);
if (!amps->fsk_tx_buffer) {
PDEBUG(DDSP, DEBUG_DEBUG, "No memory!\n");
@@ -237,76 +237,66 @@ void dsp_cleanup_sender(amps_t *amps)
#endif
}
static int fsk_encode(amps_t *amps, const char *bits)
static int fsk_encode(amps_t *amps, char bit)
{
int16_t *spl;
double phase, bitstep, deviation;
int count;
char last;
if (strlen(bits) > FSK_MAX_BITS) {
fprintf(stderr, "FSK buffer too small\n");
abort();
}
deviation = amps->fsk_deviation;
spl = amps->fsk_tx_buffer;
phase = amps->fsk_tx_phase;
last = amps->fsk_tx_last_bit;
bitstep = amps->fsk_bitstep * 256.0 * 2.0; /* half bit ramp */
//printf("%s\n", bits);
while (*bits) {
//printf("%d %d\n", (*bits) & 1, last & 1);
if (((*bits) & 1)) {
if ((last & 1)) {
/* last bit was 1, this bit is 1, so we ramp down first */
do {
*spl++ = ramp_down[(int)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* last bit was 0, this bit is 1, so we stay down first */
do {
*spl++ = -deviation;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
/* ramp up */
do {
*spl++ = ramp_up[(int)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
if ((last & 1)) {
/* last bit was 1, this bit is 0, so we stay up first */
do {
*spl++ = deviation;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* last bit was 0, this bit is 0, so we ramp up first */
do {
*spl++ = ramp_up[(int)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
/* ramp up */
//printf("%d %d\n", (bit) & 1, last & 1);
if ((bit & 1)) {
if ((last & 1)) {
/* last bit was 1, this bit is 1, so we ramp down first */
do {
*spl++ = ramp_down[(int)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* last bit was 0, this bit is 1, so we stay down first */
do {
*spl++ = -deviation;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
last = *bits;
bits++;
/* ramp up */
do {
*spl++ = ramp_up[(int)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
if ((last & 1)) {
/* last bit was 1, this bit is 0, so we stay up first */
do {
*spl++ = deviation;
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
} else {
/* last bit was 0, this bit is 0, so we ramp up first */
do {
*spl++ = ramp_up[(int)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
/* ramp up */
do {
*spl++ = ramp_down[(int)phase];
phase += bitstep;
} while (phase < 256.0);
phase -= 256.0;
}
last = bit;
/* depending on the number of samples, return the number */
count = ((uintptr_t)spl - (uintptr_t)amps->fsk_tx_buffer) / sizeof(*spl);
@@ -319,46 +309,60 @@ static int fsk_encode(amps_t *amps, const char *bits)
int fsk_frame(amps_t *amps, int16_t *samples, int length)
{
int count = 0, pos, copy, i;
int count = 0, len, pos, copy, i;
int16_t *spl;
const char *bits;
int rc;
char c;
len = amps->fsk_tx_buffer_length;
pos = amps->fsk_tx_buffer_pos;
spl = amps->fsk_tx_buffer;
again:
/* there must be length, otherwise we would skip blocks */
if (count == length)
return count;
goto done;
pos = amps->fsk_tx_buffer_pos;
spl = amps->fsk_tx_buffer + pos;
/* start new frame, so we generate one */
/* start of new bit, so generate buffer for one bit */
if (pos == 0) {
if (amps->dsp_mode == DSP_MODE_AUDIO_RX_FRAME_TX)
bits = amps_encode_frame_fvc(amps);
else
bits = amps_encode_frame_focc(amps);
if (!bits)
return 0;
fsk_encode(amps, bits);
c = amps->fsk_tx_frame[amps->fsk_tx_frame_pos];
/* start new frame, so we generate one */
if (c == '\0') {
if (amps->dsp_mode == DSP_MODE_AUDIO_RX_FRAME_TX)
rc = amps_encode_frame_fvc(amps, amps->fsk_tx_frame);
else
rc = amps_encode_frame_focc(amps, amps->fsk_tx_frame);
/* check if we have not bit string (change to tx audio)
* we may not store fsk_tx_buffer_pos, because is was reset on a mode achange */
if (rc)
return count;
amps->fsk_tx_frame_pos = 0;
c = amps->fsk_tx_frame[0];
}
if (c == 'i')
c = (amps->channel_busy) ? '0' : '1';
len = fsk_encode(amps, c);
amps->fsk_tx_frame_pos++;
}
copy = amps->fsk_tx_buffer_length - pos;
copy = len - pos;
if (length - count < copy)
copy = length - count;
//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) / 32767.0));
puts(debug_amplitude((double)spl[pos] / 32767.0));
#endif
*samples++ = *spl++;
*samples++ = spl[pos++];
}
pos += copy;
count += copy;
if (pos ==amps->fsk_tx_buffer_length) {
amps->fsk_tx_buffer_pos = 0;
if (pos == len) {
pos = 0;
goto again;
}
done:
amps->fsk_tx_buffer_length = len;
amps->fsk_tx_buffer_pos = pos;
return count;
@@ -399,12 +403,10 @@ void sender_send(sender_t *sender, int16_t *samples, int length)
again:
switch (amps->dsp_mode) {
case DSP_MODE_OFF:
off:
/* silence, if transmitter is off */
memset(samples, 0, length * sizeof(*samples));
break;
case DSP_MODE_AUDIO_RX_AUDIO_TX:
audio:
jitter_load(&amps->sender.audio, samples, length);
/* pre-emphasis */
if (amps->pre_emphasis)
@@ -417,21 +419,10 @@ audio:
/* Encode frame into audio stream. If frames have
* stopped, process again for rest of stream. */
count = fsk_frame(amps, samples, length);
#if 0
/* special case: add SAT signal to frame at loop test */
if (amps->sender.loopback)
sat_encode(amps, samples, length);
#endif
/* count == 0: no frame, this should not happen */
if (count == 0)
goto off;
/* * also if the mode changed to audio during processing */
if (amps->dsp_mode == DSP_MODE_AUDIO_RX_AUDIO_TX)
goto audio;
samples += count;
length -= count;
goto again;
break;
if (length)
goto again;
}
}
@@ -512,6 +503,7 @@ prepare_frame:
printf("No Sync detected after dotting\n");
#endif
amps->fsk_rx_sync = FSK_SYNC_NONE;
amps->channel_busy = 0;
return;
}
return;
@@ -549,6 +541,7 @@ prepare_frame:
if (amps->fsk_rx_frame_length == 240)
amps->fsk_rx_frame_length = 247;
amps->fsk_rx_sync = FSK_SYNC_NONE;
amps->channel_busy = 0;
}
}
}
@@ -599,6 +592,8 @@ void fsk_rx_dotting(amps_t *amps, double _elapsed, int dir)
amps->fsk_rx_sync = FSK_SYNC_DOTTING;
amps->fsk_rx_dotting_average = fabs(average);
amps->fsk_rx_bitcount = 0.5 + average;
if (amps->si.acc_type.bis)
amps->channel_busy = 1;
}
}
@@ -859,9 +854,12 @@ void amps_set_dsp_mode(amps_t *amps, enum dsp_mode mode, int frame_length)
/* reset detection process */
amps->fsk_rx_sync = FSK_SYNC_NONE;
amps->channel_busy = 0;
amps->fsk_rx_sync_register = 0x555;
/* reset transmitter */
amps->fsk_tx_buffer_pos = 0;
amps->fsk_tx_frame[0] = '\0';
amps->fsk_tx_frame_pos = 0;
}