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Add global DC-Filter and remove all individual DC-Filters

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This commit is contained in:
Andreas Eversberg
2017-01-28 18:18:44 +01:00
parent 71e556e7ff
commit bd7ccc5fa0
9 changed files with 32 additions and 58 deletions
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35
src/amps/dsp.c
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@@ -178,12 +178,11 @@ static void dsp_init_ramp(amps_t *amps)
static void sat_reset(amps_t *amps, const char *reason);
/* Init FSK of transceiver */
int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
int dsp_init_sender(amps_t *amps, int tolerant)
{
sample_t *spl;
int i;
int rc;
double RC, dt;
int half;
/* attack (3ms) and recovery time (13.5ms) according to amps specs */
@@ -256,14 +255,6 @@ int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
amps->test_phaseshift256 = 256.0 / ((double)amps->sender.samplerate / 1000.0);
PDEBUG(DDSP, DEBUG_DEBUG, "test_phaseshift256 = %.4f\n", amps->test_phaseshift256);
/* use this filter to remove dc level for 0-crossing detection
* if we have de-emphasis, we don't need it, so high_pass is not set. */
if (high_pass) {
RC = 1.0 / (CUT_OFF_HIGHPASS * 2.0 *3.14);
dt = 1.0 / amps->sender.samplerate;
amps->highpass_factor = RC / (RC + dt);
}
/* be more tolerant when syncing */
amps->fsk_rx_sync_tolerant = tolerant;
@@ -808,8 +799,7 @@ static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
int max, pos;
int i;
/* SAT detection */
/* SAT / signalling tone detection */
max = amps->sat_samples;
spl = amps->sat_filter_spl;
pos = amps->sat_filter_pos;
@@ -853,25 +843,10 @@ static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
void sender_receive(sender_t *sender, sample_t *samples, int length)
{
amps_t *amps = (amps_t *) sender;
double x, y, x_last, y_last, factor;
int i;
/* high pass filter to remove 0-level
* if factor is not set, we should already have 0-level. */
factor = amps->highpass_factor;
if (factor) {
x_last = amps->highpass_x_last;
y_last = amps->highpass_y_last;
for (i = 0; i < length; i++) {
x = (double)samples[i];
y = factor * (y_last + x - x_last);
x_last = x;
y_last = y;
samples[i] = y;
}
amps->highpass_x_last = x_last;
amps->highpass_y_last = y_last;
}
/* dc filter required for FSK decoding and tone detection */
if (amps->de_emphasis)
dc_filter(&amps->estate, samples, length);
switch (amps->dsp_mode) {
case DSP_MODE_OFF: