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Quisk version 4.2.45

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This commit is contained in:
James C. Ahlstrom
2025-10-08 16:29:31 -04:00
parent 458e43252d
commit 0528e8bbbe
17 changed files with 97 additions and 1331 deletions
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microphone.c
+4 -150
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@@ -8,7 +8,6 @@
#include <sys/types.h>
#include "microphone.h"
#include "filter.h"
#include "freedv.h"
#ifdef MS_WINDOWS
#include <winsock2.h>
@@ -245,11 +244,9 @@ static void init_alc(struct alc * pt, int size)
case DGT_U:
case DGT_L:
case DGT_IQ:
pt->gain_now[i] = 1.4;
break;
case FDV_U:
case FDV_L:
pt->gain_now[i] = 2.0;
pt->gain_now[i] = 1.4;
break;
default:
pt->gain_now[i] = 1.0;
@@ -617,7 +614,7 @@ static int tx_filter_digital(complex double * filtered, int count)
quisk_filt_dInit(&filter1, quiskDgtFilt48Coefs, sizeof(quiskDgtFilt48Coefs)/sizeof(double)); // pass 1350, stop 1650
}
if ( ! filtered) { // Change to rxMode
quisk_filt_tune(&filter1, 1650.0 / 48000, rxMode != DGT_L && rxMode != LSB);
quisk_filt_tune(&filter1, 1650.0 / 48000, rxMode != DGT_L && rxMode != LSB && rxMode != FDV_L);
return 0;
}
for (i = 0; i < count; i++) // FIR bandpass filter; separate into I and Q
@@ -626,140 +623,6 @@ static int tx_filter_digital(complex double * filtered, int count)
return count;
}
static int tx_filter_freedv(complex double * filtered, int count, int encode)
{ // Input samples are creal(filtered), output is filtered.
// This filter is used for digital voice.
// Changes by Dave Roberts, G8KBB, for additional modes June, 2020.
// Modified by N2ADR January, 2024.
int i;
// The input sample rate is mic_sample_rate, either 8000 or 48000 sps.
// The FreeDV codec requires an input speech_sample_rate of either 8000 or 16000 sps depending on the mode. Input is always real.
// The FreeDV codec will output samples at modem_sample_rate of 8000 or 48000 sps depending on the mode, and output may be real or complex.
double dtmp, magn, dsample;
int modem_sample_rate, speech_sample_rate, use_filter, is_real;
static int samples_size = 0;
static int do_init = 1;
static double aaa, bbb, ccc, Xmin, Xmax, Ymax;
static double time_long, time_short;
static double inMax=0.3;
static double * dsamples = NULL;
static struct quisk_cFilter filter1, filter2;
static struct quisk_dFilter filtDecim48to16, filtDecim16to8;;
static struct quisk_cFilter cfiltInterp8to48;
if (do_init) { // initialization
//QuiskWavWriteOpen(&hWav, "jim.wav", 3, 1, 4, 8000, 1.0 / CLIP16);
do_init = 0;
quisk_filt_cInit(&filter1, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double)); // at 48000 sps pass 3000 stop 4000 times two plus 3000
quisk_filt_cInit(&filter2, quiskFilt53D2Coefs, sizeof(quiskFilt53D2Coefs)/sizeof(double)); // at 8000 sps pass 1500 stop 1800 times two plus 1500
quisk_filt_cInit(&cfiltInterp8to48, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));
quisk_filt_dInit(&filtDecim48to16, quiskAudio24p3Coefs, sizeof(quiskAudio24p3Coefs)/sizeof(double)); // pass 6000 stop 8000
quisk_filt_dInit(&filtDecim16to8, quiskFilt16dec8Coefs, sizeof(quiskFilt16dec8Coefs)/sizeof(double)); // pass 3000 stop 4000
Ymax = pow(10.0, - 1 / 20.0); // maximum y
Xmax = pow(10.0, 3 / 20.0); // x where slope is zero; for x > Xmax, y == Ymax
Xmin = Ymax - fabs(Ymax - Xmax); // x where slope is 1 and y = x; start of compression
//printf ("Xmin %f\n", Xmin);
aaa = 1.0 / (2.0 * (Xmin - Xmax)); // quadratic
bbb = -2.0 * aaa * Xmax;
ccc = Ymax - aaa * Xmax * Xmax - bbb * Xmax;
}
switch (freedv_current_mode) { // n_modem_sample_rate and n_speech_sample_rate may not be valid
case FREEDV_MODE_2400A:
case FREEDV_MODE_2400B:
use_filter = 1; // center frequency 3000
is_real = 1;
modem_sample_rate = 48000;
speech_sample_rate = 8000;
break;
case FREEDV_MODE_2020:
use_filter = 0;
is_real = 0;
modem_sample_rate = 8000;
speech_sample_rate = 16000;
break;
case FREEDV_MODE_800XA:
use_filter = 2; // center frequency 1500
is_real = 1;
modem_sample_rate = 8000;
speech_sample_rate = 8000;
break;
default:
use_filter = 0;
is_real = 0;
modem_sample_rate = 8000;
speech_sample_rate = 8000;
break;
}
if ( ! filtered) { // rxMode changed
quisk_filt_tune((struct quisk_dFilter *)&filter1, 3000.0 / 48000.0, rxMode != FDV_L);
quisk_filt_tune((struct quisk_dFilter *)&filter2, 1500.0 / 8000.0, rxMode != FDV_L);
return 0;
}
// check size of dsamples[] buffer
if (count > samples_size) {
samples_size = count * 2;
if (dsamples)
free(dsamples);
dsamples = (double *)malloc(samples_size * sizeof(double));
}
// copy to dsamples[]
for (i = 0; i < count; i++)
dsamples[i] = creal(filtered[i]) / CLIP16; // normalize to 1.0000
// Decimate to 8000 Hz or to 16000 depending on mode (and hence setting of sample rate)
if (quisk_sound_state.mic_sample_rate == 48000) {
switch (speech_sample_rate) {
case 16000:
count = quisk_dDecimate(dsamples, count, &filtDecim48to16, 3);
break;
case 8000:
count = quisk_dDecimate(dsamples, count, &filtDecim48to16, 3);
count = quisk_dDecimate(dsamples, count, &filtDecim16to8, 2);
break;
default:
QuiskPrintf("Failure to convert speech input rate in tx_filter_freedv\n");
break;
}
}
else if (quisk_sound_state.mic_sample_rate == 8000 && speech_sample_rate != 8000) {
QuiskPrintf("Failure to convert input rate in tx_filter_freedv\n");
}
// Measure average peak input audio level and limit
dtmp = 1.0 / speech_sample_rate; // sample time
time_long = 1.0 - exp(- dtmp / 3.000);
time_short = 1.0 - exp(- dtmp / 0.005);
for (i = 0; i < count; i++) {
dsample = dsamples[i];
magn = fabs(dsample);
if (magn > inMax)
inMax = inMax * (1 - time_short) + time_short * magn;
else if(magn > mic_agc_level)
inMax = inMax * (1 - time_long) + time_long * magn;
else
inMax = inMax * (1 - time_long) + time_long * mic_agc_level;
dsample = dsample / inMax * Xmin * 0.7;
magn = fabs(dsample);
if (magn < Xmin)
dsamples[i] = dsample;
else if (magn > Xmax)
dsamples[i] = copysign(Ymax, dsample);
else
dsamples[i] = copysign(aaa * magn * magn + bbb * magn + ccc, dsample);
dsamples[i] = dsamples[i] * CLIP16;
}
//QuiskWavWriteD(&hWav, dsamples, count);
if (encode && pt_quisk_freedv_tx) // Encode audio into digital modulation
count = (* pt_quisk_freedv_tx)(filtered, dsamples, count, is_real);
//quisk_calc_audio_graph(CLIP16, filtered, NULL, count, 0);
if (use_filter == 1) // convert float samples to complex with an analytic filter
count = quisk_cCDecimate(filtered, count, &filter1, 1);
else if (use_filter == 2)
count = quisk_cCDecimate(filtered, count, &filter2, 1);
// Interpolate up to 48000
if (MIC_OUT_RATE != modem_sample_rate )
count = quisk_cInterpolate(filtered, count, &cfiltInterp8to48, MIC_OUT_RATE / modem_sample_rate);
return count;
}
PyObject * quisk_get_tx_filter(PyObject * self, PyObject * args)
{ // return the TX filter response to display on the graph
// This is for debugging. Change quisk.py to call QS.get_tx_filter() instead
@@ -1397,6 +1260,8 @@ int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int
case DGT_L:
case DGT_IQ:
case DGT_FM:
case FDV_U: // FreeDV
case FDV_L:
count = tx_filter_digital(cSamples, count); // filter samples, minimal processing
process_alc(cSamples, count, &tx_alc, rxMode);
break;
@@ -1404,11 +1269,6 @@ int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int
count *= interp;
transmit_mic_imd(cSamples, count, quiskImdLevel / 1000.0);
break;
case FDV_U: // FDV
case FDV_L:
count = tx_filter_freedv(cSamples, count, 1);
process_alc(cSamples, count, &tx_alc, rxMode);
break;
default:
break;
}
@@ -1499,11 +1359,6 @@ int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int
count = tx_filter(cSamples, count);
process_alc(cSamples, count, &tx_alc, rxMode);
break;
case FDV_U: // FDV
case FDV_L:
count = tx_filter_freedv(cSamples, count, 0);
process_alc(cSamples, count, &tx_alc, rxMode);
break;
default:
for (i = 0; i < count; i++)
cSamples[i] = 0;
@@ -1673,7 +1528,6 @@ void quisk_set_tx_mode(void) // called when the mode rxMode is changed
tx_filter(NULL, 0);
tx_filter_digital(NULL, 0);
transmit_udp(NULL, 0);
tx_filter_freedv(NULL, 0, 0);
serial_key_samples(NULL, 0);
}