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quisk-kc4upr/microphone.c

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#include <Python.h>
#include <stdlib.h>
#include <math.h>
#include <sys/time.h>
#include <complex.h>
#include <fftw3.h>
#include "quisk.h"
#include <sys/types.h>
#include "microphone.h"
#include "filter.h"
#include "freedv.h"
#ifdef MS_WINDOWS
#include <Winsock2.h>
static int mic_cleanup = 0; // must clean up winsock
#else
#include <sys/socket.h>
#include <arpa/inet.h>
#include <netinet/in.h>
#endif
#define DEBUG 0
#define DEBUG_LEVEL 0
#if DEBUG_LEVEL || DEBUG_IO || DEBUG
static int debug_timer = 1; // count up number of samples
#endif
// The microphone samples must be 48000 sps or 8000 sps. The output sample
// rate is always MIC_OUT_RATE samples per second
// FM needs pre-emphasis and de-emphasis. See vk1od.net/FM/FM.htm for details.
// For IIR design, see http://www.abvolt.com/research/publications2.htm.
// Microhone preemphasis: boost high frequencies 0.00 to 1.00
double quisk_mic_preemphasis;
// Microphone clipping; try 3.0 or 4.0
double quisk_mic_clip;
// If true, decimate 48000 sps mic samples to 8000 sps for processing
#define DECIM_8000 1
struct alc {
complex double * buffer;
int buf_size;
int index;
int block_index;
int counter;
int fault;
double max_magn;
double gain_now[20];
double gain_max;
double gain_min;
double gain_change;
double next_change;
double final_gain;
} ;
// These are external:
int mic_max_display; // display value of maximum microphone signal level 0 to 2**15 - 1
int quiskSpotLevel = -1; // level is -1 for Spot button Off; else the Spot level 0 to 1000.
int quiskImdLevel = 500; // level for rxMode IMD, 0 to 1000
static SOCKET mic_socket = INVALID_SOCKET; // send microphone samples to a socket
static double mic_agc_level = 0.10; // Mic levels below this are noise and are ignored
static int mic_level; // maximum microphone signal level for display
static int mic_timer; // time to display maximum mic level
static int align4; // add two bytes to start of audio samples to align to 4 bytes
static double modulation_index = 1.6; // For FM transmit, the modulation index
static int is_vox = 0; // Is the VOX level exceeded?
static int vox_level = CLIP16; // VOX trigger level as a number 0 to CLIP16
static int timeVOX = 2000; // VOX hang time in milliseconds
static int tx_sample_rate = 48000; // Used for SoapySDR
static int reverse_tx_sideband;
static int doTxCorrect = 0; // Corrections for UDP sample transmit
static double TxCorrectLevel;
static complex TxCorrectDc;
// Used for the Hermes protocol
#define HERMES_TX_BUF_SAMPLES 4800 // buffer size in I/Q samples (two shorts)
#define HERMES_TX_BUF_SHORTS (HERMES_TX_BUF_SAMPLES * 2)
static int hermes_read_index; // index to read from buffer
static int hermes_write_index; // index to write to buffer
static int hermes_num_samples; // number of samples in the buffer
static short hermes_buf[HERMES_TX_BUF_SHORTS]; // buffer to store Tx I/Q samples waiting to be sent at 48 ksps
static int hermes_filter_rx; // hermes filter to use for Rx
static int hermes_filter_tx; // hermes filter to use for Tx
static int alex_hpf_rx; // Alex HPF to use for Rx
static int alex_hpf_tx; // Alex HPF to use for Tx
static int alex_lpf_rx; // Alex LPF to use for Rx
static int alex_lpf_tx; // Alex LPF to use for Tx
#define TX_BLOCK_SHORTS 600 // transmit UDP packet with this many shorts (two bytes) (perhaps + 1)
#define MIC_MAX_HOLD_TIME 400 // Time to hold the maximum mic level on the Status screen in milliseconds
// If USE_GET_SIN is not zero, replace mic samples with a sin wave at a
// frequency determined by the sidetone slider and an amplitude determined
// by the Spot button level. LEVEL FAILS
// If USE_GET_SIN is 1, pass these samples through the transmit filters.
// If USE_GET_SIN is 2, transmit these samples directly.
#define USE_GET_SIN 0
// If USE_2TONE is not zero, replace samples with a 2-tone test signal.
#define USE_2TONE 0
#if USE_GET_SIN
static void get_sin(complex double * cSamples, int count)
{ // replace mic samples with a sin wave
int i;
double freq;
complex double phase1; // Phase increment
static complex double vector1 = CLIP32 / 2;
// Use the sidetone slider 0 to 1000 to set frequency
//freq = (quisk_sidetoneCtrl - 500) / 1000.0 * MIC_OUT_RATE;
freq = quisk_sidetoneCtrl * 5;
freq = ((int)freq / 50) * 50;
#if USE_GET_SIN == 2
phase1 = cexp(I * 2.0 * M_PI * freq / MIC_OUT_RATE);
count *= MIC_OUT_RATE / quisk_sound_state.mic_sample_rate;
#else
phase1 = cexp(I * 2.0 * M_PI * freq / quisk_sound_state.mic_sample_rate);
#endif
for (i = 0; i < count; i++) {
vector1 *= phase1;
cSamples[i] = vector1;
}
#if DEBUG_IO || DEBUG
if (debug_timer == 0)
printf ("get_sin freq %.0lf\n", freq);
#endif
}
#endif
#if USE_2TONE
static void get_2tone(complex double * cSamples, int count)
{ // replace mic samples
int i;
static complex double phase1=0, phase2; // Phase increment
static complex double vector1;
static complex double vector2;
if (phase1 == 0) { // initialize
phase1 = cexp((I * 2.0 * M_PI * IMD_TONE_1) / quisk_sound_state.mic_sample_rate);
phase2 = cexp((I * 2.0 * M_PI * IMD_TONE_2) / quisk_sound_state.mic_sample_rate);
vector1 = CLIP32 / 2.0;
vector2 = CLIP32 / 2.0;
}
for (i = 0; i < count; i++) {
vector1 *= phase1;
vector2 *= phase2;
cSamples[i] = (vector1 + vector2);
}
}
#endif
static double CcmPeak(double * dsamples, complex double * csamples, int count)
{
int i, j;
complex double csample;
double dtmp, dsample, newlevel, oldlevel;
static double out_short, out_long;
static struct Ccmpr {
int buf_size;
int index_read;
double themax;
double level;
double * d_samp;
complex double * c_samp;
double * levl;
} dat = {0};
if ( ! dat.buf_size) { // initialize; the sample rate is 8000
dat.buf_size = 8000 * 30 / 1000; // total delay in samples
dat.index_read = 0; // index to output; and then write a new sample here
dat.themax = 1.0; // maximum level in the buffer
dat.level = 1.0; // current output level
dat.d_samp = (double *) malloc(dat.buf_size * sizeof(double)); // buffer for double samples
dat.c_samp = (complex double *) malloc(dat.buf_size * sizeof(complex double)); // buffer for complex samples
dat.levl = (double *) malloc(dat.buf_size * sizeof(double)); // magnitude of the samples
for (i = 0; i < dat.buf_size; i++) {
dat.d_samp[i] = 0;
dat.c_samp[i] = 0;
dat.levl[i] = 1.0;
}
dtmp = 1.0 / 8000; // sample time
out_short = 1.0 - exp(- dtmp / 0.010); // short time constant
out_long = 1.0 - exp(- dtmp / 3.000); // long time constant
return 1.0;
}
for (i = 0; i < count; i++) {
if (dsamples) {
dsample = dsamples[i];
dsamples[i] = dat.d_samp[dat.index_read] / dat.level; // FIFO output
dat.d_samp[dat.index_read] = dsample; // write new sample at read index
newlevel = fabs(dsample);
}
else {
csample = csamples[i];
csamples[i] = dat.c_samp[dat.index_read] / dat.level; // FIFO output
dat.c_samp[dat.index_read] = csample; // write new sample at read index
newlevel = cabs(csample);
}
oldlevel = dat.levl[dat.index_read];
dat.levl[dat.index_read] = newlevel;
if (newlevel < dat.themax && oldlevel < dat.themax) { // some other sample is the maximum
// no change to dat.themax
}
else if (newlevel > dat.themax && newlevel > oldlevel) { // newlevel is the maximum
dat.themax = newlevel;
}
else { // search for the maximum level
dat.themax = 0; // Find the maximim level in the buffer
for (j = 0; j < dat.buf_size; j++) {
if (dat.levl[j] > dat.themax)
dat.themax = dat.levl[j];
}
}
// Increase dat.level if the maximum level is greater than 1.0;
// decrease it slowly back to 1.0 if it is lower. Output is modulated by dat.level.
if (dat.themax > 1.0) // increase rapidly to the peak level
dat.level = dat.level * (1.0 - out_short) + dat.themax * out_short;
else // decrease slowly back to 1.0
dat.level = dat.level * (1.0 - out_long) + 1.0 * out_long;
if (++dat.index_read >= dat.buf_size)
dat.index_read = 0;
}
return dat.level;
}
static void init_alc(struct alc * pt, int size)
{ // Call first to set the buffer size. Then call to initialize the structure on each key down.
int i;
if (pt->buffer == NULL) {
pt->buf_size = size; // do not change the size
pt->buffer = (complex double *)malloc(size * sizeof(complex double));
for (i = 0; i < 20; i++) // initial gain by rx_mode
switch(i) {
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;
break;
default:
pt->gain_now[i] = 1.0;
break;
}
}
pt->index = 0;
pt->block_index = 0;
pt->counter = 0;
pt->fault = 0;
pt->max_magn = 0;
pt->gain_max = 3.0;
pt->gain_min = 0.1;
pt->gain_change = 0;
pt->next_change = 0;
pt->final_gain = 0;
for (i = 0; i < pt->buf_size; i++)
pt->buffer[i] = 0;
}
static void process_alc(complex double * cSamples, int count, struct alc * pt, rx_mode_type rx_mode)
{ // Automatic Level Control (ALC)
int i;
double d, magn;
complex double csamp;
for (i = 0; i < count; i++) {
csamp = cSamples[i]; // new sample to add to buffer
cSamples[i] = pt->buffer[pt->index] * pt->gain_now[rx_mode]; // remove sample from buffer and apply gain
#if DEBUG_LEVEL || DEBUG_IO || DEBUG
magn = cabs(cSamples[i]);
if (magn >= CLIP16)
printf("ALC clip gain %9.6f level %9.6f\n", pt->gain_now[rx_mode], magn / CLIP16);
#endif
pt->buffer[pt->index] = csamp; // add new sample to buffer
magn = cabs(csamp); // measure new sample
if (magn * (pt->gain_now[rx_mode] + pt->gain_change * pt->buf_size) > (CLIP16 - 10)) {
pt->gain_change = ((CLIP16 - 10) / magn - pt->gain_now[rx_mode]) / pt->buf_size;
pt->final_gain = pt->gain_now[rx_mode] + pt->gain_change * pt->buf_size;
if (pt->final_gain > pt->gain_max) {
pt->final_gain = pt->gain_max;
pt->gain_change = (pt->final_gain - pt->gain_now[rx_mode]) / pt->buf_size;
}
else if (pt->final_gain < pt->gain_min) {
pt->final_gain = pt->gain_min;
pt->gain_change = (pt->final_gain - pt->gain_now[rx_mode]) / pt->buf_size;
}
pt->block_index = pt->index;
pt->counter = 0;
pt->fault = 0;
pt->next_change = 1E10;
//printf("ALC DEC: gain %9.6f change %9.6f final gain %9.6f\n", pt->gain_now[rx_mode], pt->gain_change, pt->final_gain);
}
else if (pt->index == pt->block_index) {
//d = cabs(cSamples[i]) / (CLIP16 - 10);
//printf("ALC Fin: gain %9.6f out level %9.6f\n", pt->gain_now[rx_mode], d);
#if 0
alc_start_gain = alc_gain;
k = alc_block_index;
for (j = 1; j < pt->buf_size; j++) {
if (++k >= pt->buf_size)
k = 0;
magn = cabs(alc_buffer[k]);
if (magn < 100) {
alc_fault++;
continue;
}
else {
d = ((CLIP16 - 10) / magn - alc_start_gain) / j;
if ( alc_next_change > d)
alc_next_change = d;
}
}
#endif
d = 5.0; // number of seconds to double gain
d = 1.0 / (48000.0 * d);
if (pt->next_change > d)
pt->next_change = d;
if (pt->next_change != 1E10 && pt->fault < pt->buf_size - 10) {
pt->gain_change = pt->next_change;
}
pt->final_gain = pt->gain_now[rx_mode] + pt->gain_change * pt->buf_size;
if (pt->final_gain > pt->gain_max) {
pt->final_gain = pt->gain_max;
pt->gain_change = (pt->final_gain - pt->gain_now[rx_mode]) / pt->buf_size;
}
else if (pt->final_gain < pt->gain_min) {
pt->final_gain = pt->gain_min;
pt->gain_change = (pt->final_gain - pt->gain_now[rx_mode]) / pt->buf_size;
}
//printf("ALC New: gain %9.6f change %9.6f final gain %9.6f\n", pt->gain_now[rx_mode], pt->gain_change, pt->final_gain);
pt->fault = 0;
pt->counter = 0;
pt->next_change = 1E10;
}
else {
if (magn < 100) {
pt->fault++;
}
else {
d = ((CLIP16 - 10) / magn - pt->final_gain) / ++pt->counter;
if ( pt->next_change > d)
pt->next_change = d;
}
}
pt->gain_now[rx_mode] += pt->gain_change;
if (++pt->index >= pt->buf_size)
pt->index = 0;
}
#if DEBUG_LEVEL || DEBUG_IO || DEBUG
for (i = 0; i < count; i++) {
magn = cabs(cSamples[i]);
if (pt->max_magn < magn)
pt->max_magn = magn;
}
if (debug_timer == 0) {
printf("ALC Out: gain %9.6f max lvl%9.6f final gain %9.6f\n", pt->gain_now[rx_mode], pt->max_magn / CLIP16, pt->final_gain);
pt->max_magn = 0;
}
#endif
}
static int tx_filter(complex double * filtered, int count)
{ // Input samples are creal(filtered), output is filtered. The input rate must be 8000 or 48000 sps.
int i, is_ssb;
int sample_rate = 8000;
double dsample, dtmp, magn;
complex double csample;
static double inMax=0.3;
static double x_1=0;
static double aaa, bbb, ccc, Xmin, Xmax, Ymax;
static int samples_size = 0;
static double * dsamples = NULL;
static complex double * csamples = NULL;
static double time_long, time_short;
static struct quisk_dFilter filtDecim, dfiltInterp;
static struct quisk_dFilter filtAudio1, filtAudio2, dfiltAudio3;
static struct quisk_cFilter cfiltAudio3, cfiltInterp;
static struct quisk_dFilter filter1={NULL}, filter2;
#if DEBUG_IO || DEBUG
char * clip;
static double dbOut = 0, Level0 = 0, Level1 = 0, Level2 = 0, Level3 = 0, Level4 = 0;
#endif
is_ssb = (rxMode == LSB || rxMode == USB);
if (!filtered) { // initialization
if (! filter1.dCoefs) {
quisk_filt_dInit(&filter1, quiskMicFilt8Coefs, sizeof(quiskMicFilt8Coefs)/sizeof(double));
quisk_filt_dInit(&filter2, quiskMicFilt8Coefs, sizeof(quiskMicFilt8Coefs)/sizeof(double));
quisk_filt_dInit(&filtDecim, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));
quisk_filt_dInit(&dfiltInterp, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));
quisk_filt_cInit(&cfiltInterp, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));
quisk_filt_dInit(&filtAudio1, quiskFiltTx8kAudioB, sizeof(quiskFiltTx8kAudioB)/sizeof(double));
quisk_filt_dInit(&filtAudio2, quiskFiltTx8kAudioB, sizeof(quiskFiltTx8kAudioB)/sizeof(double));
quisk_filt_dInit(&dfiltAudio3, quiskFiltTx8kAudioB, sizeof(quiskFiltTx8kAudioB)/sizeof(double));
quisk_filt_cInit(&cfiltAudio3, quiskFiltTx8kAudioB, sizeof(quiskFiltTx8kAudioB)/sizeof(double));
dtmp = 1.0 / sample_rate; // sample time
time_long = 1.0 - exp(- dtmp / 3.000);
time_short = 1.0 - exp(- dtmp / 0.005);
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
aaa = 1.0 / (2.0 * (Xmin - Xmax)); // quadratic
bbb = -2.0 * aaa * Xmax;
ccc = Ymax - aaa * Xmax * Xmax - bbb * Xmax;
#if DEBUG_IO || DEBUG
printf("Compress to %.2lf dB from %.2lf to %.2lf dB\n",
20 * log10(Ymax), 20 * log10(Xmin), 20 * log10(Xmax));
#endif
}
if (is_ssb) {
quisk_filt_tune(&filter1, 1650.0 / sample_rate, rxMode != LSB);
quisk_filt_tune(&filter2, 1650.0 / sample_rate, rxMode != LSB);
}
return 0;
}
// check size of dsamples[] and csamples[] buffer
if (count > samples_size) {
samples_size = count * 2;
if (dsamples)
free(dsamples);
if (csamples)
free(csamples);
dsamples = (double *)malloc(samples_size * sizeof(double));
csamples = (complex double *)malloc(samples_size * sizeof(complex double));
}
// copy to dsamples[], normalize to +/- 1.0
for (i = 0; i < count; i++)
dsamples[i] = creal(filtered[i]) / CLIP16;
// Decimate to 8000 Hz
if (quisk_sound_state.mic_sample_rate != sample_rate)
count = quisk_dDecimate(dsamples, count, &filtDecim, quisk_sound_state.mic_sample_rate / sample_rate);
// restrict bandwidth 300 to 2700 Hz
count = quisk_dFilter(dsamples, count, &filtAudio1);
#if DEBUG_IO || DEBUG
// Measure peak input audio level
for (i = 0; i < count; i++) {
magn = fabs(dsamples[i]);
if (magn > Level0)
Level0 = magn;
}
#endif
// high pass filter for preemphasis: See Radcom, January 2010, page 76.
// quisk_mic_preemphasis == 1 was measured as 6 dB / octave.
// gain at 800 Hz was measured as 0.104672.
for (i = 0; i < count; i++) {
dtmp = dsamples[i];
dsamples[i] = dtmp - quisk_mic_preemphasis * x_1;
x_1 = dtmp; // delayed sample
dsamples[i] *= 2; // compensate for loss
#if DEBUG_IO || DEBUG
magn = fabs(dsamples[i]);
if (magn > Level1)
Level1 = magn;
#endif
}
if (is_ssb) { // SSB
// FIR bandpass filter; separate into I and Q
for (i = 0; i < count; i++) {
csample = quisk_dC_out(dsamples[i], &filter1) * 2.0; // filter loss 0.5
// Measure average peak input audio level and normalize
magn = cabs(csample);
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;
csample /= inMax;
magn /= inMax;
#if DEBUG_IO || DEBUG
if (magn > Level2)
Level2 = magn;
#endif
// Audio compression.
csample *= quisk_mic_clip;
magn *= quisk_mic_clip;
if (magn > 1.0)
csample = csample / magn;
dsamples[i] = creal(csample);
}
}
else { // AM and FM
// Measure average peak input audio level and normalize
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 /= inMax;
magn /= inMax;
#if DEBUG_IO || DEBUG
if (magn > Level2)
Level2 = magn;
#endif
// Audio compression.
dsample *= quisk_mic_clip;
magn *= quisk_mic_clip;
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);
}
}
// remove clipping distortion; restrict bandwidth 300 to 2700 Hz
count = quisk_dFilter(dsamples, count, &filtAudio2);
if (is_ssb) { // SSB
// FIR bandpass filter; separate into I and Q
for (i = 0; i < count; i++) {
csamples[i] = quisk_dC_out(dsamples[i], &filter2) * 2.0; // filter loss 0.5
#if DEBUG_IO || DEBUG
magn = cabs(csamples[i]);
if (magn > Level3)
Level3 = magn;
#endif
}
// round off peaks
CcmPeak(NULL, csamples, count);
#if DEBUG_IO || DEBUG
for (i = 0; i < count; i++) {
magn = cabs(csamples[i]);
if (magn > Level4)
Level4 = magn;
}
#endif
// remove clipping distortion
count = quisk_cDecimate(csamples, count, &cfiltAudio3, 1);
// Interpolate up to 48000
if (MIC_OUT_RATE != sample_rate)
count = quisk_cInterpolate(csamples, count, &cfiltInterp, MIC_OUT_RATE / sample_rate);
// convert back to 16 bits
for (i = 0; i < count; i++) {
filtered[i] = csamples[i] * CLIP16;
#if DEBUG_IO || DEBUG
magn = cabs(csamples[i]);
if (magn > dbOut)
dbOut = magn;
#endif
}
}
else { // AM and FM
#if DEBUG_IO || DEBUG
for (i = 0; i < count; i++) {
magn = fabs(dsamples[i]);
if (magn > Level3)
Level3 = magn;
}
#endif
// round off peaks
CcmPeak(dsamples, NULL, count);
#if DEBUG_IO || DEBUG
for (i = 0; i < count; i++) {
magn = fabs(dsamples[i]);
if (magn > Level4)
Level4 = magn;
}
#endif
// remove clipping distortion
count = quisk_dFilter(dsamples, count, &dfiltAudio3);
// Interpolate up to 48000
if (MIC_OUT_RATE != sample_rate)
count = quisk_dInterpolate(dsamples, count, &dfiltInterp, MIC_OUT_RATE / sample_rate);
// convert back to 16 bits
for (i = 0; i < count; i++) {
filtered[i] = dsamples[i] * CLIP16;
#if DEBUG_IO || DEBUG
magn = fabs(dsamples[i]);
if (magn > dbOut)
dbOut = magn;
#endif
}
}
#if DEBUG_IO || DEBUG
if (debug_timer == 0) {
if (dbOut > 1.0)
clip = "Clip";
else
clip = "";
dbOut = 20 * log10(dbOut);
printf ("pre %3.1lf dB clip %2.0lf InMax %6.2lf Level0 %6.2lf Level1 %6.2lf Level2 %6.2lf Level3 %6.2lf Level4 %6.2lf dbOut %6.2lf %s\n",
quisk_mic_preemphasis, 20 * log10(quisk_mic_clip), 20 * log10(inMax), 20 * log10(Level0),
20 * log10(Level1), 20 * log10(Level2), 20 * log10(Level3), 20 * log10(Level4), dbOut, clip);
Level0 = Level1 = Level2 = Level3 = Level4 = dbOut = 0;
}
//QuiskPrintTime(" tx_filter", 2);
#endif
return count;
}
static int tx_filter_digital(complex double * filtered, int count)
{ // Input samples are creal(filtered), output is filtered.
// This filter has minimal processing and is used for digital modes.
int i;
static int do_init = 1;
static struct quisk_dFilter filter1;
if (do_init) { // initialization
do_init = 0;
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);
return 0;
}
for (i = 0; i < count; i++) // FIR bandpass filter; separate into I and Q
filtered[i] = quisk_dC_out(creal(filtered[i]), &filter1) * 2.00; // tuned filter loss 0.5
//quisk_calc_audio_graph(CLIP16, filtered, NULL, count, 0);
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.
int i;
int sample_rate = 8000;
double dtmp, magn, dsample;
static int samples_size = 0;
static int last_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;
if (do_init) { // initialization
//QuiskWavWriteOpen(&hWav, "jim.wav", 3, 1, 4, 8000, 1.0 / CLIP16);
do_init = 0;
quisk_filt_cInit(&filter2, quiskFilt53D2Coefs, sizeof(quiskFilt53D2Coefs)/sizeof(double)); // pass 1500, stop 1800
quisk_filt_tune((struct quisk_dFilter *)&filter2, 1600.0 / 8000, 1); // upper sideband
last_mode = 11;
quisk_filt_dInit(&filtDecim, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double)); // pass 3000, stop 4000
quisk_filt_cInit(&cfiltInterp, quiskLpFilt48Coefs, sizeof(quiskLpFilt48Coefs)/sizeof(double));
dtmp = 1.0 / sample_rate; // sample time
time_long = 1.0 - exp(- dtmp / 3.000);
time_short = 1.0 - exp(- dtmp / 0.005);
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;
}
if (last_mode != rxMode) { // change to sideband
if (rxMode == FDV_U) { // upper sideband
last_mode = rxMode;
quisk_filt_tune((struct quisk_dFilter *)&filter2, 1600.0 / 8000, 1);
}
else if (rxMode == FDV_L) { // lower sideband
last_mode = rxMode;
quisk_filt_tune((struct quisk_dFilter *)&filter2, 1600.0 / 8000, 0);
}
}
if ( ! filtered)
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
if (quisk_sound_state.mic_sample_rate != sample_rate)
count = quisk_dDecimate(dsamples, count, &filtDecim, quisk_sound_state.mic_sample_rate / sample_rate);
// Measure average peak input audio level and limit
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);
//quisk_calc_audio_graph(CLIP16, filtered, NULL, count, 0);
if (freedv_current_mode != FREEDV_MODE_700D) // 700D has its own filter
count = quisk_cCDecimate(filtered, count, &filter2, 1);
// Interpolate up to 48000
if (MIC_OUT_RATE != sample_rate)
count = quisk_cInterpolate(filtered, count, &cfiltInterp, MIC_OUT_RATE / 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
// of QS.get_filter().
int i, j, k;
int freq, time;
PyObject * tuple2;
complex double cx;
double scale;
double * average, * fft_window, * bufI, * bufQ;
fftw_complex * samples, * pt; // complex data for fft
fftw_plan plan; // fft plan
double phase, delta;
int nTaps = 325;
if (!PyArg_ParseTuple (args, ""))
return NULL;
// Create space for the fft of size data_width
pt = samples = (fftw_complex *) fftw_malloc(sizeof(fftw_complex) * data_width);
plan = fftw_plan_dft_1d(data_width, pt, pt, FFTW_FORWARD, FFTW_MEASURE);
average = (double *) malloc(sizeof(double) * (data_width + nTaps));
fft_window = (double *) malloc(sizeof(double) * data_width);
bufI = (double *) malloc(sizeof(double) * nTaps);
bufQ = (double *) malloc(sizeof(double) * nTaps);
for (i = 0, j = -data_width / 2; i < data_width; i++, j++) // Hanning
fft_window[i] = 0.5 + 0.5 * cos(2. * M_PI * j / data_width);
for (i = 0; i < data_width + nTaps; i++)
average[i] = 0.5; // Value for freq == 0
for (freq = 1; freq < data_width / 2.0 - 10.0; freq++) {
//freq = data_width * 0.2 / 48.0;
delta = 2 * M_PI / data_width * freq;
phase = 0;
// generate some initial samples to fill the filter pipeline
for (time = 0; time < data_width + nTaps; time++) {
average[time] += cos(phase); // current sample
phase += delta;
if (phase > 2 * M_PI)
phase -= 2 * M_PI;
}
}
// now filter the signal using the transmit filter
tx_filter(NULL, 0); // initialize
scale = 1.0;
for (i = 0; i < data_width; i++)
if (fabs(average[i + nTaps]) > scale)
scale = fabs(average[i + nTaps]);
scale = CLIP16 / scale; // limit to CLIP16
for (i = 0; i < nTaps; i++)
samples[i] = average[i] * scale;
tx_filter(samples, nTaps); // process initial samples
for (i = 0; i < data_width; i++)
samples[i] = average[i + nTaps] * scale;
tx_filter(samples, data_width); // process the samples
for (i = 0; i < data_width; i++) // multiply by window
samples[i] *= fft_window[i];
fftw_execute(plan); // Calculate FFT
// Normalize and convert to log10
scale = 0.3 / data_width / scale;
for (k = 0; k < data_width; k++) {
cx = samples[k];
average[k] = cabs(cx) * scale;
if (average[k] <= 1e-7) // limit to -140 dB
average[k] = -7;
else
average[k] = log10(average[k]);
}
// Return the graph data
tuple2 = PyTuple_New(data_width);
i = 0;
// Negative frequencies:
for (k = data_width / 2; k < data_width; k++, i++)
PyTuple_SetItem(tuple2, i, PyFloat_FromDouble(20.0 * average[k]));
// Positive frequencies:
for (k = 0; k < data_width / 2; k++, i++)
PyTuple_SetItem(tuple2, i, PyFloat_FromDouble(20.0 * average[k]));
free(bufQ);
free(bufI);
free(average);
free(fft_window);
fftw_destroy_plan(plan);
fftw_free(samples);
return tuple2;
}
// Send samples using the Metis-Hermes protocol. A frame is 8 bytes: L/R audio and I/Q mic samples.
// All samples are 2 bytes. The 1032 byte UDP packet contains 63*2 radio sound samples, and 63*2 mic I/Q samples.
// Samples are sent synchronously with the input samples.
static void quisk_hermes_tx_reset(void)
{ // Reset the buffer to half full of zero samples
int i;
hermes_num_samples = HERMES_TX_BUF_SAMPLES / 2;
hermes_read_index = 0;
hermes_write_index = hermes_num_samples * 2;
for (i = 0; i < HERMES_TX_BUF_SHORTS; i++) // Put zero mic samples into the buffer
hermes_buf[i] = 0;
//printf("quisk_hermes_tx_reset: hermes_num_samples %d\n", hermes_num_samples);
}
static void quisk_hermes_tx_add(complex double * cSamples, int tx_count, int key_down)
{ // Add samples to the Tx buffer.
int i;
static int hermes_buf_has_samples=1;
if (key_down) { // add non-zero samples to buffer
hermes_buf_has_samples = 1;
}
else if (hermes_buf_has_samples) { // key is not down; reset buffer to zero
quisk_hermes_tx_reset();
hermes_buf_has_samples = 0;
return;
}
else { // key is not down but buffer is zeroed; just reset pointers
hermes_num_samples = HERMES_TX_BUF_SAMPLES / 2;
hermes_read_index = 0;
hermes_write_index = hermes_num_samples * 2;
return;
}
//printf("hermes_tx_add start: hermes_num_samples %d, tx_count %d\n", hermes_num_samples, tx_count);
if (hermes_num_samples + tx_count >= HERMES_TX_BUF_SAMPLES) { // no more space; throw away half the samples
quisk_udp_mic_error("Tx hermes buffer overflow");
//printf("Tx hermes buffer overflow: hermes_num_samples %d tx_count %d\n", hermes_num_samples, tx_count);
i = hermes_num_samples - HERMES_TX_BUF_SAMPLES / 2; // number of samples to remove
hermes_num_samples -= i;
hermes_read_index += i * 2;
if (hermes_read_index >= HERMES_TX_BUF_SHORTS)
hermes_read_index -= HERMES_TX_BUF_SHORTS;
}
hermes_num_samples += tx_count;
for (i = 0; i < tx_count; i++) { // Put transmit mic samples into the buffer
hermes_buf[hermes_write_index++] = (short)cimag(cSamples[i]);
hermes_buf[hermes_write_index++] = (short)creal(cSamples[i]);
if (hermes_write_index >= HERMES_TX_BUF_SHORTS)
hermes_write_index = 0;
}
//printf ("Buffer usage %.1f %%, hermes_num_samples %d, tx_count %d\n", 100.0 * hermes_num_samples / HERMES_TX_BUF_SAMPLES, hermes_num_samples, tx_count);
//printf("hermes_tx_add end: hermes_num_samples %d\n", hermes_num_samples);
}
void quisk_hermes_tx_send(int tx_socket, int * tx_records)
{ // Send one UDP block of mic samples using the Metis-Hermes protocol. Timing is from blocks received, rate is 48k.
// If the key is up we send samples anyway, but the samples are zero.
int i, offset, sent, ratio;
short s;
unsigned char sendbuf[1032];
unsigned char * pt_buf;
static unsigned int seq = 0;
static unsigned char C0_index = 0;
static int mox_bit=0; // On key up, send some zero samples before releasing the T/R relay to Rx.
static int mox_counter; // Timer for key-up delay.
unsigned int hlwp = 0;
//printf("hermes_tx_send start 1: hermes_num_samples %d\n", hermes_num_samples);
if (tx_records == NULL) {
seq = 0;
C0_index = 0;
quisk_hermes_tx_reset();
return;
}
if (quisk_is_key_down()) {
mox_bit = 1;
mox_counter = (int)((quiskKeyupDelay + 30) / 2.625 + 0.5); // 126 samples per block / 48000 == 2.625 msec
}
else { // Delay un-setting the MOX bit and continue sending zero-valued samples; when Tx signal is zero, unset MOX.
if (mox_bit) { // This allows the Tx signal to go to zero before switching T/R relay.
if (mox_counter > 0)
mox_counter--;
else
mox_bit = 0;
}
}
//printf("hermes_tx_send start 2: hermes_num_samples %d\n", hermes_num_samples);
ratio = quisk_sound_state.sample_rate / 48000; // send rate is 48 ksps
//printf ("quisk_hermes_tx_send ratio %d count %d\n", ratio, *tx_records);
if (*tx_records / ratio < 63 * 2) // tx_records is the number of samples received for each receiver
return;
// Send 63*2 Tx samples with control bytes
//printf ("Buffer usage %.1f %%\n", 100.0 * hermes_num_samples / HERMES_TX_BUF_SAMPLES);
//printf ("Tx quisk_hermes_tx_send ratio %d, count %d, samples %d\n", ratio, *tx_records, hermes_num_samples);
*tx_records -= 63 * 2 * ratio;
if (hermes_num_samples < 63 * 2) { // Not enough samples to send
//printf("Tx hermes buffer underflow: hermes_num_samples %d\n", hermes_num_samples);
quisk_udp_mic_error("Tx hermes buffer underflow");
quisk_hermes_tx_reset();
}
hermes_num_samples -= 63 * 2;
sendbuf[0] = 0xEF;
sendbuf[1] = 0xFE;
sendbuf[2] = 0x01;
sendbuf[3] = 0x02;
sendbuf[4] = seq >> 24 & 0xFF;
sendbuf[5] = seq >> 16 & 0xFF;
sendbuf[6] = seq >> 8 & 0xFF;
sendbuf[7] = seq & 0xFF;
seq++;
sendbuf[8] = 0x7F;
sendbuf[9] = 0x7F;
sendbuf[10] = 0x7F;
offset = C0_index * 4; // offset into quisk_pc_to_hermes is C0[7:1] * 4
sendbuf[11] = C0_index << 1 | mox_bit; // C0
sendbuf[12] = quisk_pc_to_hermes[offset++]; // C1
sendbuf[13] = quisk_pc_to_hermes[offset++]; // C2
sendbuf[14] = quisk_pc_to_hermes[offset++]; // C3
sendbuf[15] = quisk_pc_to_hermes[offset++]; // C4
if (C0_index == 0) { // Do not change receiver count without stopping Hermes and restarting
sendbuf[15] = quisk_multirx_count << 3 | 0x04; // Send the old count, not the changed count
if (mox_bit) // send filter selection on J16
sendbuf[13] = hermes_filter_tx << 1;
else
sendbuf[13] = hermes_filter_rx << 1;
}
else if ( ! quisk_is_vna && C0_index == 9) {
if (mox_bit) { // send Alex filter selection
sendbuf[14] = alex_hpf_tx;
sendbuf[15] = alex_lpf_tx;
}
else {
sendbuf[14] = alex_hpf_rx;
sendbuf[15] = alex_lpf_rx;
}
}
if (++C0_index > 16)
C0_index = 0;
pt_buf = sendbuf + 16;
for (i = 0; i < 63; i++) { // add 63 samples
*pt_buf++ = 0x00; // Left/Right audio sample
*pt_buf++ = 0x00;
*pt_buf++ = 0x00;
*pt_buf++ = 0x00;
s = hermes_buf[hermes_read_index++];
*pt_buf++ = (s >> 8) & 0xFF; // Two bytes of I
*pt_buf++ = s & 0xFF;
s = hermes_buf[hermes_read_index++];
*pt_buf++ = (s >> 8) & 0xFF; // Two bytes of Q
*pt_buf++ = s & 0xFF;
if (hermes_read_index >= HERMES_TX_BUF_SHORTS)
hermes_read_index = 0;
}
sendbuf[520] = 0x7F;
sendbuf[521] = 0x7F;
sendbuf[522] = 0x7F;
// Changes for HermesLite v2 thanks to Steve, KF7O
if ((quisk_hermeslite_writepointer > 0) && (quisk_hermeslite_writeattempts++ % 8 == 0)) {
// Only send periodic hermeslite writes in second part of frame
hlwp = 5*(quisk_hermeslite_writepointer-1);
sendbuf[523] = quisk_hermeslite_writequeue[hlwp++] << 1 | mox_bit;
sendbuf[524] = quisk_hermeslite_writequeue[hlwp++];
sendbuf[525] = quisk_hermeslite_writequeue[hlwp++];
sendbuf[526] = quisk_hermeslite_writequeue[hlwp++];
sendbuf[527] = quisk_hermeslite_writequeue[hlwp++];
if ((sendbuf[523] & 0x80) == 0) {
// No acknowledge requested so fire and forget
quisk_hermeslite_writepointer--;
quisk_hermeslite_writeattempts = 0;
}
} else {
offset = C0_index * 4; // offset into quisk_pc_to_hermes is C0[7:1] * 4
sendbuf[523] = C0_index << 1 | mox_bit; // C0
sendbuf[524] = quisk_pc_to_hermes[offset++]; // C1
sendbuf[525] = quisk_pc_to_hermes[offset++]; // C2
sendbuf[526] = quisk_pc_to_hermes[offset++]; // C3
sendbuf[527] = quisk_pc_to_hermes[offset++]; // C4
if (C0_index == 0) {
sendbuf[527] = quisk_multirx_count << 3 | 0x04; // Send the old count, not the changed count
if (mox_bit) // send filter selection on J16
sendbuf[525] = hermes_filter_tx << 1;
else
sendbuf[525] = hermes_filter_rx << 1;
}
else if ( ! quisk_is_vna && C0_index == 9) {
if (mox_bit) { // send Alex filter selection
sendbuf[526] = alex_hpf_tx;
sendbuf[527] = alex_lpf_tx;
}
else {
sendbuf[526] = alex_hpf_rx;
sendbuf[527] = alex_lpf_rx;
}
}
if (++C0_index > 16)
C0_index = 0;
if (quisk_hermeslite_writepointer > 0) quisk_hermeslite_writeattempts++;
}
// Abort after 53/8 ~= 5 retries
if ((quisk_hermeslite_writepointer > 0) && (quisk_hermeslite_writeattempts > 53)) {
printf("ERROR: Maximum Hermes-Lite write attempts\n");
// Cancel entire write sequence
quisk_hermeslite_writepointer = 0;
quisk_hermeslite_writeattempts = 0;
}
pt_buf = sendbuf + 528;
for (i = 0; i < 63; i++) { // add 63 samples
*pt_buf++ = 0x00; // Left/Right audio sample
*pt_buf++ = 0x00;
*pt_buf++ = 0x00;
*pt_buf++ = 0x00;
s = hermes_buf[hermes_read_index++];
*pt_buf++ = (s >> 8) & 0xFF; // Two bytes of I
*pt_buf++ = s & 0xFF;
s = hermes_buf[hermes_read_index++];
*pt_buf++ = (s >> 8) & 0xFF; // Two bytes of Q
*pt_buf++ = s & 0xFF;
if (hermes_read_index >= HERMES_TX_BUF_SHORTS)
hermes_read_index = 0;
}
sent = send(tx_socket, (char *)sendbuf, 1032, 0);
if (sent != 1032)
quisk_udp_mic_error("Tx UDP socket error in Hermes");
//printf("hermes_tx_send end: hermes_num_samples %d\n", hermes_num_samples);
}
// udp_iq has an initial zero followed by the I/Q samples.
// The initial zero is sent iff align4 == 1.
static void transmit_udp(complex double * cSamples, int count)
{ // Send count samples using the HiQSDR protocol. Each sample is sent as two shorts (4 bytes) of I/Q data.
// Transmission is delayed until a whole block of data is available.
int i, sent;
static short udp_iq[TX_BLOCK_SHORTS + 1] = {0};
static int udp_size = 1;
if (mic_socket == INVALID_SOCKET)
return;
if ( ! cSamples) { // initialization
udp_size = 1;
udp_iq[0] = 0; // should not be necessary
return;
}
if (doTxCorrect) {
for (i = 0; i < count; i++)
cSamples[i] = cSamples[i] * TxCorrectLevel + TxCorrectDc;
}
for (i = 0; i < count; i++) { // transmit samples
udp_iq[udp_size++] = (short)creal(cSamples[i]);
udp_iq[udp_size++] = (short)cimag(cSamples[i]);
if (udp_size >= TX_BLOCK_SHORTS) { // check count
if (align4)
sent = send(mic_socket, (char *)udp_iq, udp_size * 2, 0);
else
sent = send(mic_socket, (char *)udp_iq + 1, --udp_size * 2, 0);
if (sent != udp_size * 2)
printf("Send socket returned %d\n", sent);
udp_size = 1;
}
}
}
static void transmit_mic_carrier(complex double * cSamples, int count, double level)
{ // send a CW carrier instead of mic samples
#if 1
// transmit a carrier equal to the number of samples
int i;
for (i = 0; i < count; i++)
cSamples[i] = level * CLIP16;
#else
// replace mic samples with a sin wave
int i;
double freq;
complex double phase1; // Phase increment
static complex double vector1 = CLIP16 / 2;
// Use the sidetone slider 0 to 1000 to set frequency
freq = quisk_sidetoneCtrl * 5;
freq = ((int)freq / 50) * 50;
phase1 = cexp(I * 2.0 * M_PI * freq / 48000.0);
//phase1 = conj(phase1);
for (i = 0; i < count; i++) {
vector1 *= phase1;
cSamples[i] = vector1 * level;
}
#endif
}
static void transmit_mic_imd(complex double * cSamples, int count, double level)
{ // send a 2-tone test signal instead of mic samples
int i;
complex double v;
static complex double phase1=0, phase2; // Phase increment
static complex double vector1;
static complex double vector2;
if (phase1 == 0) { // initialize
phase1 = cexp((I * 2.0 * M_PI * IMD_TONE_1) / MIC_OUT_RATE);
phase2 = cexp((I * 2.0 * M_PI * IMD_TONE_2) / MIC_OUT_RATE);
vector1 = CLIP16 / 2.0;
vector2 = CLIP16 / 2.0;
}
for (i = 0; i < count; i++) { // transmit a carrier equal to the number of samples
vector1 *= phase1;
vector2 *= phase2;
v = level * (vector1 + vector2);
cSamples[i] = v;
}
}
int quisk_process_microphone(int mic_sample_rate, complex double * cSamples, int count)
{
int i, sample, maximum, interp, mic_interp, key_down;
double d, ctcss_delta, audio_scale, ctcss_scale;
static int key_was_down=0, key_down_counter=0;
static struct quisk_cFilter filtInterp={NULL};
static struct quisk_cFilter filtInterp2={NULL};
static struct quisk_cFilter filt240D4 = {NULL};
static struct quisk_cFilter filt300D6 = {NULL};
static struct quisk_cHB45Filter HalfBand = {NULL, 0, 0};
static double ctcss_angle=0;
static struct alc tx_alc = {NULL};
#if 0
// Measure soundcard actual sample rate
static time_t seconds = 0;
static int total = 0;
struct timeval tb;
static double dtime;
gettimeofday(&tb);
total += count;
if (seconds == 0) {
seconds = tb.tv_sec;
dtime = tb.tv_sec + 0.000001 * tb.tv_usec;
}
else if (tb.tv_sec - seconds > 4) {
printf("Mic soundcard rate %.3f\n", total / (tb.tv_sec + .000001 * tb.tv_usec - dtime));
seconds = tb.tv_sec;
printf("backlog %d, count %d\n", backlog, count);
}
#endif
#if DEBUG_IO || DEBUG
//QuiskPrintTime("", -1);
#endif
#if DEBUG_IO || DEBUG
double magn;
static double out_max=0;
#endif
#if DEBUG_LEVEL || DEBUG_IO || DEBUG
debug_timer += count;
if (debug_timer >= mic_sample_rate) // one second
debug_timer = 0;
#endif
// Microphone sample are input at mic_sample_rate. But after processing,
// the output rate is MIC_OUT_RATE.
interp = MIC_OUT_RATE / mic_sample_rate;
#if USE_GET_SIN
get_sin(cSamples, count); // Replace mic samples with a sin wave
#endif
#if USE_2TONE
get_2tone(cSamples, count); // Replace mic samples with a 2-tone test signal
#endif
// measure maximum microphone level
maximum = 1;
for (i = 0; i < count; i++) {
cSamples[i] *= (double)CLIP16 / CLIP32; // convert 32-bit samples to 16 bits
d = creal(cSamples[i]);
sample = (int)fabs(d);
if (sample > maximum)
maximum = sample;
}
// VOX processing
if (maximum > vox_level) {
is_vox = mic_sample_rate / 1000 * timeVOX; // reset timer to maximum
}
else if(is_vox) {
is_vox -= count; // decrement timer
if (is_vox < 0)
is_vox = 0;
}
// mic display level
if (maximum > mic_level)
mic_level = maximum;
mic_timer -= count; // time out the max microphone level to display
if (mic_timer <= 0) {
mic_timer = mic_sample_rate / 1000 * MIC_MAX_HOLD_TIME;
mic_max_display = mic_level;
mic_level = 1;
}
if ( ! tx_alc.buffer)
init_alc(&tx_alc, 960); // at 48000 sps, 960 is 20 msec
// quiskTxHoldState is a state machine to implement a pause for a repeater frequency shift for FM
key_down = quisk_is_key_down();
if (rxMode == FM || rxMode == DGT_FM) {
switch (quiskTxHoldState) {
case 0: // Never implement any hold
break;
case 1: // Start hold when key goes down
if (key_down)
quiskTxHoldState = 2;
break;
case 2: // Key down hold is in progress; wait until state changes to 3
break;
case 3: // Hold is released; when key goes up, hold starts again
if ( ! key_down)
quiskTxHoldState = 4;
break;
case 4: // Key up hold is in progress; wait until state changes to 1
break;
}
}
if (quiskTxHoldState == 2 || quiskTxHoldState == 4) { // don't transmit until the hold is cleared
key_down = 0;
for (i = 0; i < count; i++)
cSamples[i] = 0;
}
if (key_down != key_was_down) {
key_down_counter = 4800; // Key was pressed. Zero the first few samples to clear the buffers.
init_alc(&tx_alc, 0); // init ALC for Tx and RECORD_MIC
key_was_down = key_down;
}
if (key_down || DEBUG_MIC) { // create transmit I/Q samples
#if USE_GET_SIN == 2
transmit_udp(cSamples, count * interp);
#else
if (quiskSpotLevel >= 0) { // Spot is in use
count *= interp;
transmit_mic_carrier(cSamples, count, quiskSpotLevel / 1000.0);
}
else switch (rxMode) {
case LSB: // LSB
case USB: // USB
if (quisk_record_state == PLAYBACK)
count = tx_filter_digital(cSamples, count); // filter samples, minimal processing
else
count = tx_filter(cSamples, count); // filter samples
process_alc(cSamples, count, &tx_alc, rxMode);
break;
case AM: // AM
if (quisk_record_state != PLAYBACK) // no audio processing for recorded sound
count = tx_filter(cSamples, count);
for (i = 0; i < count; i++) // transmit (0.5 + ampl/2, 0)
cSamples[i] = (creal(cSamples[i]) + CLIP16) * 0.5;
process_alc(cSamples, count, &tx_alc, rxMode);
break;
case FM: // FM
if (quisk_record_state != PLAYBACK) // no audio processing for recorded sound
count = tx_filter(cSamples, count);
if (quisk_ctcss_freq > 9) {
ctcss_delta = 2.0 * M_PI / MIC_OUT_RATE * quisk_ctcss_freq;
ctcss_scale = 450.0 * modulation_index / quisk_ctcss_freq; // for 15% of total deviation
audio_scale = 0.85 * modulation_index / CLIP16;
for (i = 0; i < count; i++) {
ctcss_angle += ctcss_delta;
if (ctcss_angle >= 2.0 * M_PI)
ctcss_angle -= 2.0 * M_PI;
cSamples[i] = CLIP16 * cexp(I * (audio_scale * creal(cSamples[i]) + ctcss_scale * sin (ctcss_angle)));
}
}
else {
audio_scale = modulation_index / CLIP16;
for (i = 0; i < count; i++) // this is phase modulation == FM and 6 dB /octave preemphasis
cSamples[i] = CLIP16 * cexp(I * audio_scale * creal(cSamples[i]));
}
process_alc(cSamples, count, &tx_alc, rxMode);
break;
case DGT_U: // external digital modes
case DGT_L:
case DGT_IQ:
case DGT_FM:
count = tx_filter_digital(cSamples, count); // filter samples, minimal processing
process_alc(cSamples, count, &tx_alc, rxMode);
break;
case IMD: // transmit IMD 2-tone test
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;
}
#if DEBUG_IO || DEBUG
for (i = 0; i < count; i++) {
magn = cabs(cSamples[i]);
if (out_max < magn)
out_max = magn;
}
if (debug_timer == 0) {
printf("Max cSamples[] Output Level %9.6f\n", out_max / CLIP16);
out_max = 0;
}
#endif
if (key_down_counter > 0) { // zero the first few samples to clear the buffers.
for (i = 0; i < count && key_down_counter > 0; i++, key_down_counter--) {
if (key_down_counter < 480)
cSamples[i] *= (1.0 - key_down_counter / 480.0); // slow increase at end
else
cSamples[i] = 0; // initial samples are zero
}
}
#endif
}
if (reverse_tx_sideband)
for (i = 0; i < count; i++)
cSamples[i] = conj(cSamples[i]);
if(quisk_pt_sample_write) { // Used for SoapySDR
// Interpolate the mic samples to the Tx sample rate
//printf("Tx sample rate %i\n", tx_sample_rate);
switch (tx_sample_rate) {
case 100000:
count = quisk_cInterp2HB45(cSamples, count, &HalfBand);
// Fall through
case 50000:
if (! filt240D4.dCoefs)
quisk_filt_cInit(&filt240D4, quiskFilt240D4Coefs, sizeof(quiskFilt240D4Coefs)/sizeof(double));
if (! filt300D6.dCoefs)
quisk_filt_cInit(&filt300D6, quiskFilt300D6Coefs, sizeof(quiskFilt300D6Coefs)/sizeof(double));
count = quisk_cInterpDecim(cSamples, count, &filt240D4, 5, 4); // 60 kSps
count = quisk_cInterpDecim(cSamples, count, &filt300D6, 5, 6); // 50 kSps
break;
case 96000:
if (! filtInterp2.dCoefs)
quisk_filt_cInit(&filtInterp2, quiskFilt48dec24Coefs, sizeof(quiskFilt48dec24Coefs)/sizeof(double));
count = quisk_cInterpolate(cSamples, count, &filtInterp2, 2);
break;
case 192000:
if (! filtInterp2.dCoefs)
quisk_filt_cInit(&filtInterp2, quiskFilt48dec24Coefs, sizeof(quiskFilt48dec24Coefs)/sizeof(double));
count = quisk_cInterp2HB45(cSamples, count, &HalfBand);
count = quisk_cInterpolate(cSamples, count, &filtInterp2, 2);
break;
default: // 48000
break;
}
(*quisk_pt_sample_write)(cSamples, count);
}
else if (quisk_use_rx_udp == 10) { // Send Hermes mic samples when key is up or down
if ( ! quisk_rx_udp_started)
;
else {
if ((rxMode == CWL || rxMode == CWU) && quiskSpotLevel < 0) // CW and no Spot
for (i = 0; i < count; i++)
cSamples[i] = 0;
quisk_hermes_tx_add(cSamples, count, key_down);
}
}
else if (quisk_use_rx_udp && key_down) { // Send mic samples to UDP when key is down
transmit_udp(cSamples, count);
}
if (quisk_record_state == RECORD_MIC) {
switch (rxMode) {
case LSB: // LSB
case USB: // USB
count = tx_filter(cSamples, count); // filter samples
process_alc(cSamples, count, &tx_alc, rxMode);
break;
case AM: // AM
count = tx_filter(cSamples, count);
process_alc(cSamples, count, &tx_alc, rxMode);
break;
case FM: // FM
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;
break;
}
// Perhaps interpolate the mic samples back to the sound play rate
mic_interp = quisk_sound_state.playback_rate / MIC_OUT_RATE;
if (mic_interp > 1) {
if (! filtInterp.dCoefs)
quisk_filt_cInit(&filtInterp, quiskFilt12_19Coefs, sizeof(quiskFilt12_19Coefs)/sizeof(double));
count = quisk_cInterpolate(cSamples, count, &filtInterp, mic_interp);
}
quisk_tmp_record(cSamples, count, (double)CLIP32 / CLIP16); // convert 16 to 32 bits
}
#if DEBUG_IO || DEBUG
//QuiskPrintTime(" process_mic", 1);
#endif
return count;
}
PyObject * quisk_set_tx_audio(PyObject * self, PyObject * args, PyObject * keywds)
{ /* Call with keyword arguments ONLY; change Tx audio parameters */
static char * kwlist[] = {"vox_level", "vox_time", "mic_clip", "mic_preemphasis", "tx_sample_rate",
"reverse_tx_sideband", NULL} ;
int vlevel = -9999, clevel = -9999;
if (!PyArg_ParseTupleAndKeywords (args, keywds, "|iiidii", kwlist,
&vlevel, &timeVOX, &clevel, &quisk_mic_preemphasis, &tx_sample_rate, &reverse_tx_sideband))
return NULL;
if (vlevel != -9999)
vox_level = (int)(pow(10.0, vlevel / 20.0) * CLIP16); // Convert dB to 16-bit sample
if (clevel != -9999)
quisk_mic_clip = pow(10.0, clevel / 20.0); // Convert dB to factor
Py_INCREF (Py_None);
return Py_None;
}
PyObject * quisk_set_udp_tx_correct(PyObject * self, PyObject * args) // Called from GUI thread
{
double DcI, DcQ, level;
if (!PyArg_ParseTuple (args, "ddd", &DcI, &DcQ, &level))
return NULL;
if (DcI == 0 && DcQ == 0 && level == 1.0){
doTxCorrect = 0;
}
else {
doTxCorrect = 1;
TxCorrectDc = (DcI + I * DcQ) * CLIP16;
DcI = fabs(DcI);
DcQ = fabs(DcQ);
if (DcI > DcQ)
TxCorrectLevel = 1.0 - DcI;
else
TxCorrectLevel = 1.0 - DcQ;
TxCorrectLevel *= level;
}
Py_INCREF (Py_None);
return Py_None;
}
PyObject * quisk_is_vox(PyObject * self, PyObject * args)
{ /* return the VOX state */
if (!PyArg_ParseTuple (args, ""))
return NULL;
return PyInt_FromLong(is_vox);
}
PyObject * quisk_set_hermes_filter(PyObject * self, PyObject * args)
{
if (!PyArg_ParseTuple (args, "ii", &hermes_filter_rx, &hermes_filter_tx))
return NULL;
Py_INCREF (Py_None);
return Py_None;
}
PyObject * quisk_set_alex_hpf(PyObject * self, PyObject * args)
{
if (!PyArg_ParseTuple (args, "ii", &alex_hpf_rx, &alex_hpf_tx))
return NULL;
Py_INCREF (Py_None);
return Py_None;
}
PyObject * quisk_set_alex_lpf(PyObject * self, PyObject * args)
{
if (!PyArg_ParseTuple (args, "ii", &alex_lpf_rx, &alex_lpf_tx))
return NULL;
Py_INCREF (Py_None);
return Py_None;
}
void quisk_close_mic(void)
{
if (mic_socket != INVALID_SOCKET) {
close(mic_socket);
mic_socket = INVALID_SOCKET;
}
#ifdef MS_WINDOWS
if (mic_cleanup)
WSACleanup();
#endif
}
void quisk_open_mic(void)
{
struct sockaddr_in Addr;
int sndsize = 48000;
#if DEBUG_IO || DEBUG
int intbuf;
#ifdef MS_WINDOWS
int bufsize = sizeof(int);
#else
socklen_t bufsize = sizeof(int);
#endif
#endif
#ifdef MS_WINDOWS
WORD wVersionRequested;
WSADATA wsaData;
#endif
modulation_index = QuiskGetConfigDouble("modulation_index", 1.6);
mic_agc_level = QuiskGetConfigDouble("mic_agc_level", 0.1);
if (quisk_sound_state.tx_audio_port == 0x553B)
align4 = 0; // Using old port: data starts at byte 42.
else
align4 = 1; // Start data at byte 44; align to dword
if (quisk_sound_state.mic_ip[0]) {
#ifdef MS_WINDOWS
wVersionRequested = MAKEWORD(2, 2);
if (WSAStartup(wVersionRequested, &wsaData) != 0)
return; // failure to start winsock
mic_cleanup = 1;
#endif
mic_socket = socket(PF_INET, SOCK_DGRAM, 0);
if (mic_socket != INVALID_SOCKET) {
setsockopt(mic_socket, SOL_SOCKET, SO_SNDBUF, (char *)&sndsize, sizeof(sndsize));
Addr.sin_family = AF_INET;
// This is the UDP port for TX microphone samples, and must agree with the microcontroller.
Addr.sin_port = htons(quisk_sound_state.tx_audio_port);
#ifdef MS_WINDOWS
Addr.sin_addr.S_un.S_addr = inet_addr(quisk_sound_state.mic_ip);
#else
inet_aton(quisk_sound_state.mic_ip, &Addr.sin_addr);
#endif
if (connect(mic_socket, (const struct sockaddr *)&Addr, sizeof(Addr)) != 0) {
close(mic_socket);
mic_socket = INVALID_SOCKET;
}
else {
#if DEBUG_IO || DEBUG
if (getsockopt(mic_socket, SOL_SOCKET, SO_SNDBUF, (char *)&intbuf, &bufsize) == 0)
printf("UDP mic socket send buffer size %d\n", intbuf);
else
printf ("Failure SO_SNDBUF\n");
#endif
}
}
}
}
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);
}
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