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Quisk version 4.2.29

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This commit is contained in:
James C. Ahlstrom
2024-01-11 12:31:13 -05:00
parent 15375aa4ea
commit 26773d5d09
20 changed files with 361 additions and 316 deletions
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microphone.c
+63 -46
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@@ -626,32 +626,32 @@ static int tx_filter_digital(complex double * filtered, int 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.
// 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 n_speech_sample_rate of either 8000 or 16000 sps depending on the mode. Input is always real.
// The FreeDV codec will output samples at n_modem_sample_rate of 8000, 16000 or 48000 sps depending on the mode, and output may be real or complex.
// 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 last_rx_mode;
static int last_freedv_mode;
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 filter2;
static struct quisk_dFilter filtDecim;
static struct quisk_cFilter cfiltInterp;
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;
last_rx_mode = -1;
last_freedv_mode = -1;
memset(&filter2, 0, sizeof(filter2));
quisk_filt_dInit(&filtDecim, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double)); // pass 3000, stop 4000
quisk_filt_cInit(&cfiltInterp, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));
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
@@ -660,36 +660,38 @@ static int tx_filter_freedv(complex double * filtered, int count, int encode)
bbb = -2.0 * aaa * Xmax;
ccc = Ymax - aaa * Xmax * Xmax - bbb * Xmax;
}
if (last_freedv_mode != freedv_current_mode) { // correct filter2 depends on freedv mode
last_freedv_mode = freedv_current_mode;
last_rx_mode = -1; // re-tune new filter
if (filter2.cSamples)
free(filter2.cSamples);
filter2.cSamples = NULL;
if (filter2.cpxCoefs)
free(filter2.cpxCoefs);
filter2.cpxCoefs = NULL;
if (filter2.cBuf)
free(filter2.cBuf);
filter2.cBuf = NULL;
switch(freedv_current_mode) {
case FREEDV_MODE_700D: // filter2 is not used
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;
// TODO: add modes that return real samples that must be converted to complex
default:
quisk_filt_cInit(&filter2, quiskFilt53D2Coefs, sizeof(quiskFilt53D2Coefs)/sizeof(double)); // pass 1500, stop 1800
use_filter = 0;
is_real = 0;
modem_sample_rate = 8000;
speech_sample_rate = 8000;
break;
}
}
if (last_rx_mode != (int)rxMode) { // change to sideband
last_rx_mode = rxMode;
if (rxMode == FDV_U) // upper sideband
quisk_filt_tune((struct quisk_dFilter *)&filter2, 1600.0 / n_modem_sample_rate, 1);
else if (rxMode == FDV_L) // lower sideband
quisk_filt_tune((struct quisk_dFilter *)&filter2, 1600.0 / n_modem_sample_rate, 0);
}
if ( ! filtered)
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;
@@ -701,12 +703,25 @@ static int tx_filter_freedv(complex double * filtered, int count, int encode)
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)
count = quisk_dDecimate(dsamples, count, &filtDecim, quisk_sound_state.mic_sample_rate / n_speech_sample_rate);
else if (quisk_sound_state.mic_sample_rate == 8000 && n_speech_sample_rate != 8000)
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 / n_speech_sample_rate; // sample time
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++) {
@@ -730,13 +745,15 @@ static int tx_filter_freedv(complex double * filtered, int count, int encode)
}
//QuiskWavWriteD(&hWav, dsamples, count);
if (encode && pt_quisk_freedv_tx) // Encode audio into digital modulation
count = (* pt_quisk_freedv_tx)(filtered, dsamples, count);
count = (* pt_quisk_freedv_tx)(filtered, dsamples, count, is_real);
//quisk_calc_audio_graph(CLIP16, filtered, NULL, count, 0);
if (filter2.cSamples) // convert float samples to complex with an analytic filter
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 != n_modem_sample_rate )
count = quisk_cInterpolate(filtered, count, &cfiltInterp, MIC_OUT_RATE / n_modem_sample_rate);
if (MIC_OUT_RATE != modem_sample_rate )
count = quisk_cInterpolate(filtered, count, &cfiltInterp8to48, MIC_OUT_RATE / modem_sample_rate);
return count;
}