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sbitx.c
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#include <stdlib.h>
#include <string.h>
#include <stdio.h>
#include <fcntl.h>
#include <math.h>
#include <complex.h>
#include <fftw3.h>
#include <unistd.h>
#include <wiringPi.h>
#include <wiringSerial.h>
#include <linux/types.h>
#include <linux/limits.h>
#include <stdint.h>
#include <time.h>
#include <signal.h>
#include <pthread.h>
#include <errno.h>
#include "sdr.h"
#include "sdr_ui.h"
#include "sound.h"
#include "i2cbb.h"
#include "si5351.h"
#include "ini.h"
int set_field(char *, char *); // This should be moved to a .h file
#define DEBUG 0
char audio_card[32];
static int tx_shift = 512;
FILE *pf_debug = NULL;
//this is for processing FT8 decodes
//unsigned int wallclock = 0;
#define TX_LINE 4
#define TX_POWER 27
#define BAND_SELECT 5
#define LPF_A 5
#define LPF_B 6
#define LPF_C 10
#define LPF_D 11
#define SBITX_DE (0)
#define SBITX_V2 (1)
int sbitx_version = SBITX_V2;
int fwdpower, vswr;
float fft_bins[MAX_BINS]; // spectrum ampltiudes
int spectrum_plot[MAX_BINS];
fftw_complex *fft_spectrum;
fftw_plan plan_spectrum;
float spectrum_window[MAX_BINS];
void set_rx1(int frequency);
void tr_switch(int tx_on);
float min_fft_level;
int rx_gain_slow_count = 0;
int rx_gain_fast_attack = 0; // Flag to enable Fast IF Gain Changes
int rx_gain_changed = 0; // Flag to indicate a change in rx_gain has been called for
// Wisdom Defines for the FFTW and FFTWF libraries
// Options for WISDOM_MODE from least to most rigorous are FFTW_ESTIMATE, FFTW_MEASURE, FFTW_PATIENT, and FFTW_EXHAUSTIVE
// The FFTW_ESTIMATE mode seems to make completely incorrect Wisdom plan choices sometimes, and is not recommended.
// Wisdom plans found in an existing Wisdom file will negate the need for time consuming Wisdom plan calculations
// if the Wisdom plans in the file were generated at the same or more rigorous level.
#define WISDOM_MODE FFTW_MEASURE
#define PLANTIME -1 // spend no more than plantime seconds finding the best FFT algorithm. -1 turns the platime cap off.
char wisdom_file[] = "/home/pi/sbitx/data/sbitx_wisdom.wis"; // Moved to default data directory - N3SB
fftw_complex *fft_out; // holds the incoming samples in freq domain (for rx as well as tx)
fftw_complex *fft_in; // holds the incoming samples in time domain (for rx as well as tx)
fftw_complex *fft_m; // holds previous samples for overlap and discard convolution
fftw_plan plan_fwd, plan_tx;
int bfo_freq = 40035000;
int freq_hdr = -1;
static double volume = 100.0;
static int tx_drive = 40;
static int rx_gain = 100;
static int rx_vol = 100;
static int tx_gain = 100;
static int tx_compress = 0;
static double spectrum_speed = 0.3;
static int in_tx = 0;
static int rx_tx_ramp = 0;
static int sidetone = 2000000000;
struct vfo tone_a, tone_b; //these are audio tone generators
static int tx_use_line = 0;
struct rx *rx_list = NULL;
struct rx *tx_list = NULL;
struct filter *tx_filter; //convolution filter
static double tx_amp = 0.0;
static double alc_level = 1.0;
static int tr_relay = 0;
static int rx_pitch = 700; //used only to offset the lo for CW,CWR
static int bridge_compensation = 100;
static double voice_clip_level = 0.022;
static int in_calibration = 1; // this turns off alc, clipping et al
#define MUTE_MAX 6
static int mute_count = 50;
FILE *pf_record;
int16_t record_buffer[1024];
int32_t modulation_buff[MAX_BINS];
/* the power gain of the tx varies widely from
band to band. these data structures help in flattening
the gain */
struct power_settings {
int f_start;
int f_stop;
int max_watts;
double scale;
};
struct power_settings band_power[] ={
{ 3500000, 4000000, 37, 0.002},
{ 7000000, 7300009, 40, 0.0015},
{10000000, 10200000, 35, 0.0019},
{14000000, 14300000, 35, 0.0025},
{18000000, 18200000, 20, 0.0023},
{21000000, 21450000, 20, 0.003},
{24800000, 25000000, 20, 0.0034},
{28000000, 29700000, 20, 0.0037}
};
#define CMD_TX (2)
#define CMD_RX (3)
#define TUNING_SHIFT (0)
#define MDS_LEVEL (-135)
struct Queue qremote;
void radio_tune_to(u_int32_t f){
if (rx_list->mode == MODE_CW)
si5351bx_setfreq(2, f + bfo_freq - 24000 + TUNING_SHIFT - rx_pitch);
else if (rx_list->mode == MODE_CWR)
si5351bx_setfreq(2, f + bfo_freq - 24000 + TUNING_SHIFT + rx_pitch);
else
si5351bx_setfreq(2, f + bfo_freq - 24000 + TUNING_SHIFT);
// printf("Setting radio rx_pitch %d\n", rx_pitch);
}
void fft_init(){
// int mem_needed;
//printf("initializing the fft\n");
fflush(stdout);
// mem_needed = sizeof(fftw_complex) * MAX_BINS;
fft_m = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * MAX_BINS/2);
fft_in = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * MAX_BINS);
fft_out = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * MAX_BINS);
fft_spectrum = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * MAX_BINS);
memset(fft_spectrum, 0, sizeof(fftw_complex) * MAX_BINS);
memset(fft_in, 0, sizeof(fftw_complex) * MAX_BINS);
memset(fft_out, 0, sizeof(fftw_complex) * MAX_BINS);
memset(fft_m, 0, sizeof(fftw_complex) * MAX_BINS/2);
fftw_set_timelimit(PLANTIME);
fftwf_set_timelimit(PLANTIME);
int e = fftw_import_wisdom_from_filename(wisdom_file);
if (e == 0)
{
printf("Generating Wisdom File...\n");
}
plan_fwd = fftw_plan_dft_1d(MAX_BINS, fft_in, fft_out, FFTW_FORWARD, WISDOM_MODE); // Was FFTW_ESTIMATE N3SB
plan_spectrum = fftw_plan_dft_1d(MAX_BINS, fft_in, fft_spectrum, FFTW_FORWARD, WISDOM_MODE); // Was FFTW_ESTIMATE N3SB
fftw_export_wisdom_to_filename(wisdom_file);
//zero up the previous 'M' bins
for (int i= 0; i < MAX_BINS/2; i++){
__real__ fft_m[i] = 0.0;
__imag__ fft_m[i] = 0.0;
}
make_hann_window(spectrum_window, MAX_BINS);
}
void fft_reset_m_bins(){
//zero up the previous 'M' bins
memset(fft_in, 0, sizeof(fftw_complex) * MAX_BINS);
memset(fft_out, 0, sizeof(fftw_complex) * MAX_BINS);
memset(fft_m, 0, sizeof(fftw_complex) * MAX_BINS/2);
memset(fft_spectrum, 0, sizeof(fftw_complex) * MAX_BINS);
memset(tx_list->fft_time, 0, sizeof(fftw_complex) * MAX_BINS);
memset(tx_list->fft_freq, 0, sizeof(fftw_complex) * MAX_BINS);
/* for (int i= 0; i < MAX_BINS/2; i++){
__real__ fft_m[i] = 0.0;
__imag__ fft_m[i] = 0.0;
}
*/
}
int mag2db(double mag){
int m = abs(mag) * 10000000;
int c = 31;
int p = 0x80000000;
while(c > 0){
if (p & m)
break;
c--;
p = p >> 1;
}
return c;
}
void set_spectrum_speed(int speed){
spectrum_speed = speed;
for (int i = 0; i < MAX_BINS; i++)
fft_bins[i] = 0;
}
void spectrum_reset(){
for (int i = 0; i < MAX_BINS; i++)
fft_bins[i] = 0;
}
void spectrum_update(){
//we are only using the lower half of the bins,
//so this copies twice as many bins,
//it can be optimized. leaving it here just in case
//someone wants to try I Q channels
//in hardware
// this has been hand optimized to lower
//the inordinate cpu usage
for (int i = 1269; i < 1803; i++){
fft_bins[i] = ((1.0 - spectrum_speed) * fft_bins[i]) +
(spectrum_speed * cabs(fft_spectrum[i]));
int y = power2dB(cnrmf(fft_bins[i]));
spectrum_plot[i] = y;
}
}
int remote_audio_output(int16_t *samples){
int length = q_length(&qremote);
for (int i = 0; i < length; i++){
samples[i] = q_read(&qremote) / 32786;
}
return length;
}
static int prev_lpf = -1;
void set_lpf_40mhz(int frequency){
int lpf = 0;
if (frequency < 5500000)
lpf = LPF_D;
else if (frequency < 10500000)
lpf = LPF_C;
else if (frequency < 18500000)
lpf = LPF_B;
else if (frequency < 30000000)
lpf = LPF_A;
if (lpf == prev_lpf){
#if DEBUG > 0
puts("LPF not changed");
#endif
return;
}
#if DEBUG > 0
printf("##################Setting LPF to %d\n", lpf);
#endif
digitalWrite(LPF_A, LOW);
digitalWrite(LPF_B, LOW);
digitalWrite(LPF_C, LOW);
digitalWrite(LPF_D, LOW);
#if DEBUG > 0
printf("################ setting %d high\n", lpf);
#endif
digitalWrite(lpf, HIGH);
prev_lpf = lpf;
}
void set_rx1(int frequency){
static int last_frequency; // Holds the last frequency set - used by the Auto IF Gain algorithm
if ((abs(frequency - last_frequency) > 1000000) && in_tx == 0)
{
rx_gain_fast_attack = 1; // Kick the Auto IF Gain algorithm into fast attack mode if the frequency change is > 1 MHz
}
last_frequency = frequency;
if (frequency == freq_hdr)
return;
radio_tune_to(frequency);
freq_hdr = frequency;
set_lpf_40mhz(frequency);
}
void set_volume(double v){
volume = v;
}
FILE *wav_start_writing(const char* path)
{
char subChunk1ID[4] = { 'f', 'm', 't', ' ' };
uint32_t subChunk1Size = 16; // 16 for PCM
uint16_t audioFormat = 1; // PCM = 1
uint16_t numChannels = 1;
uint16_t bitsPerSample = 16;
uint32_t sampleRate = 12000;
uint16_t blockAlign = numChannels * bitsPerSample / 8;
uint32_t byteRate = sampleRate * blockAlign;
char subChunk2ID[4] = { 'd', 'a', 't', 'a' };
uint32_t subChunk2Size = 0Xffffffff; //num_samples * blockAlign;
char chunkID[4] = { 'R', 'I', 'F', 'F' };
uint32_t chunkSize = 4 + (8 + subChunk1Size) + (8 + subChunk2Size);
char format[4] = { 'W', 'A', 'V', 'E' };
FILE* f = fopen(path, "w");
// NOTE: works only on little-endian architecture
fwrite(chunkID, sizeof(chunkID), 1, f);
fwrite(&chunkSize, sizeof(chunkSize), 1, f);
fwrite(format, sizeof(format), 1, f);
fwrite(subChunk1ID, sizeof(subChunk1ID), 1, f);
fwrite(&subChunk1Size, sizeof(subChunk1Size), 1, f);
fwrite(&audioFormat, sizeof(audioFormat), 1, f);
fwrite(&numChannels, sizeof(numChannels), 1, f);
fwrite(&sampleRate, sizeof(sampleRate), 1, f);
fwrite(&byteRate, sizeof(byteRate), 1, f);
fwrite(&blockAlign, sizeof(blockAlign), 1, f);
fwrite(&bitsPerSample, sizeof(bitsPerSample), 1, f);
fwrite(subChunk2ID, sizeof(subChunk2ID), 1, f);
fwrite(&subChunk2Size, sizeof(subChunk2Size), 1, f);
return f;
}
void wav_record(int32_t *samples, int count){
int16_t *w;
int32_t *s;
int i = 0, j = 0;
int decimation_factor = 96000 / 12000;
if (!pf_record)
return;
w = record_buffer;
while(i < count){
*w++ = *samples / 32786;
samples += decimation_factor;
i += decimation_factor;
j++;
}
fwrite(record_buffer, j, sizeof(int16_t), pf_record);
}
/*
The sound process is called by the duplex sound system for each block of samples
In this demo, we read and equivalent block from the file instead of processing from
the input I and Q signals.
*/
int32_t in_i[MAX_BINS];
int32_t in_q[MAX_BINS];
int32_t out_i[MAX_BINS];
int32_t out_q[MAX_BINS];
short is_ready = 0;
void tx_init(int frequency, short mode, int bpf_low, int bpf_high){
//we assume that there are 96000 samples / sec, giving us a 48khz slice
//the tuning can go up and down only by 22 KHz from the center_freq
tx_filter = filter_new(1024, 1025);
filter_tune(tx_filter, (1.0 * bpf_low)/96000.0, (1.0 * bpf_high)/96000.0 , 5);
}
struct rx *add_tx(int frequency, short mode, int bpf_low, int bpf_high){
//we assume that there are 96000 samples / sec, giving us a 48khz slice
//the tuning can go up and down only by 22 KHz from the center_freq
struct rx *r = malloc(sizeof(struct rx));
r->low_hz = bpf_low;
r->high_hz = bpf_high;
r->tuned_bin = 512;
//create fft complex arrays to convert the frequency back to time
r->fft_time = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * MAX_BINS);
r->fft_freq = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * MAX_BINS);
int e = fftw_import_wisdom_from_filename(wisdom_file);
if (e == 0)
{
printf("Generating Wisdom File...\n");
}
r->plan_rev = fftw_plan_dft_1d(MAX_BINS, r->fft_freq, r->fft_time, FFTW_BACKWARD, WISDOM_MODE); // Was FFTW_ESTIMATE N3SB
fftw_export_wisdom_to_filename(wisdom_file);
r->output = 0;
r->next = NULL;
r->mode = mode;
r->filter = filter_new(1024, 1025);
filter_tune(r->filter, (1.0 * bpf_low)/96000.0, (1.0 * bpf_high)/96000.0 , 5);
if (abs(bpf_high - bpf_low) < 1000){
r->agc_speed = 10;
r->agc_threshold = -60;
r->agc_loop = 0;
}
else {
r->agc_speed = 10;
r->agc_threshold = -60;
r->agc_loop = 0;
}
//the modems drive the tx at 12000 Hz, this has to be upconverted
//to the radio's sampling rate
r->next = tx_list;
tx_list = r;
}
struct rx *add_rx(int frequency, short mode, int bpf_low, int bpf_high){
//we assume that there are 96000 samples / sec, giving us a 48khz slice
//the tuning can go up and down only by 22 KHz from the center_freq
struct rx *r = malloc(sizeof(struct rx));
r->low_hz = bpf_low;
r->high_hz = bpf_high;
r->tuned_bin = 512;
r->agc_gain = 0.0;
//create fft complex arrays to convert the frequency back to time
r->fft_time = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * MAX_BINS);
r->fft_freq = (fftw_complex*) fftw_malloc(sizeof(fftw_complex) * MAX_BINS);
int e = fftw_import_wisdom_from_filename(wisdom_file);
if (e == 0)
{
printf("Generating Wisdom File...\n");
}
r->plan_rev = fftw_plan_dft_1d(MAX_BINS, r->fft_freq, r->fft_time, FFTW_BACKWARD, WISDOM_MODE); // Was FFTW_ESTIMATE N3SB
fftw_export_wisdom_to_filename(wisdom_file);
r->output = 0;
r->next = NULL;
r->mode = mode;
r->filter = filter_new(1024, 1025);
filter_tune(r->filter, (1.0 * bpf_low)/96000.0, (1.0 * bpf_high)/96000.0 , 5);
if (abs(bpf_high - bpf_low) < 1000){
r->agc_speed = 300;
r->agc_threshold = -60;
r->agc_loop = 0;
r->signal_avg = 0;
}
else {
r->agc_speed = 300;
r->agc_threshold = -60;
r->agc_loop = 0;
r->signal_avg = 0;
}
// the modems are driven by 12000 samples/sec
// the queue is for 20 seconds, 5 more than 15 sec needed for the FT8
r->next = rx_list;
rx_list = r;
}
int count = 0;
double agc2(struct rx *r){
int i;
double signal_strength, agc_gain_should_be;
//do nothing if agc is off
if (r->agc_speed == -1){
for (i=0; i < MAX_BINS/2; i++)
__imag__ (r->fft_time[i+(MAX_BINS/2)]) *=10000000;
return 10000000;
}
//find the peak signal amplitude
signal_strength = 0.0;
for (i=0; i < MAX_BINS/2; i++){
double s = cimag(r->fft_time[i+(MAX_BINS/2)]) * 1000;
if (signal_strength < s)
signal_strength = s;
}
//also calculate the moving average of the signal strength
r->signal_avg = (r->signal_avg * 0.93) + (signal_strength * 0.07);
if (signal_strength == 0)
agc_gain_should_be = 10000000;
else
agc_gain_should_be = 100000000000/signal_strength;
r->signal_strength = signal_strength;
// printf("Agc temp, g:%g, s:%g, f:%g ", r->agc_gain, signal_strength, agc_gain_should_be);
double agc_ramp = 0.0;
// climb up the agc quickly if the signal is louder than before
if (agc_gain_should_be < r->agc_gain){
r->agc_gain = agc_gain_should_be;
//reset the agc to hang count down
r->agc_loop = r->agc_speed;
// printf("attack %g %d ", r->agc_gain, r->agc_loop);
}
else if (r->agc_loop <= 0){
agc_ramp = (agc_gain_should_be - r->agc_gain) / (MAX_BINS/2);
// printf("release %g %d ", r->agc_gain, r->agc_loop);
}
// else if (r->agc_loop > 0)
// printf("hanging %g %d ", r->agc_gain, r->agc_loop);
if (agc_ramp != 0){
// printf("Ramping from %g ", r->agc_gain);
for (i = 0; i < MAX_BINS/2; i++){
__imag__ (r->fft_time[i+(MAX_BINS/2)]) *= r->agc_gain;
}
r->agc_gain += agc_ramp;
// printf("by %g to %g ", agc_ramp, r->agc_gain);
}
else
for (i = 0; i < MAX_BINS/2; i++)
__imag__ (r->fft_time[i+(MAX_BINS/2)]) *= r->agc_gain;
// printf("\n");
r->agc_loop--;
//printf("%d:s meter: %d %d %d \n", count++, (int)r->agc_gain, (int)r->signal_strength, r->agc_loop);
return 100000000000 / r->agc_gain;
}
void my_fftw_execute(fftw_plan f){
fftw_execute(f);
}
//TODO : optimize the memory copy and moves to use the memcpy
void rx_process(int32_t *input_rx, int32_t *input_mic,
int32_t *output_speaker, int32_t *output_tx, int n_samples)
{
int i, j = 0;
double i_sample, q_sample;
//STEP 1: first add the previous M samples to
for (i = 0; i < MAX_BINS/2; i++)
fft_in[i] = fft_m[i];
//STEP 2: then add the new set of samples
// m is the index into incoming samples, starting at zero
// i is the index into the time samples, picking from
// the samples added in the previous step
int m = 0;
//gather the samples into a time domain array
for (i= MAX_BINS/2; i < MAX_BINS; i++){
i_sample = (1.0 *input_rx[j])/200000000.0;
q_sample = 0;
j++;
__real__ fft_m[m] = i_sample;
__imag__ fft_m[m] = q_sample;
__real__ fft_in[i] = i_sample;
__imag__ fft_in[i] = q_sample;
m++;
}
// STEP 3: convert the time domain samples to frequency domain
my_fftw_execute(plan_fwd);
//STEP 3B: this is a side line, we use these frequency domain
// values to paint the spectrum in the user interface
// I discovered that the raw time samples give horrible spectrum
// and they need to be multiplied wiht a window function
// they use a separate fft plan
// NOTE: the spectrum update has nothing to do with the actual
// signal processing. If you are not showing the spectrum or the
// waterfall, you can skip these steps
for (i = 0; i < MAX_BINS; i++)
__real__ fft_in[i] *= spectrum_window[i];
my_fftw_execute(plan_spectrum);
// the spectrum display is updated
spectrum_update();
// Make adjustments to IF Gain
min_fft_level = 10000; // Set high before starting to find the lowest level in the fft_bins array
for(i=1269; i<1800; i++)
{
if (fft_bins[i] < min_fft_level)
{
min_fft_level = fft_bins[i]; // new lowest level
}
}
#define TARGET_FFT_LEVEL 0.015
#define FAST_MIN_FFT_LEVEL 0.008
#define FAST_MAX_FFT_LEVEL 0.025
// Allow for fast IF Gain changes if the rx_gain_fast_attack flag is set (see set_rx1 function in this file)
if (((min_fft_level < FAST_MIN_FFT_LEVEL) || (min_fft_level > FAST_MAX_FFT_LEVEL)) && (rx_gain_fast_attack == 1))
{
rx_gain_slow_count = 0; // Reset slow IF Gain Loop counter
if (min_fft_level < FAST_MIN_FFT_LEVEL)
{
rx_gain += 5; // Plan to increase up RX Gain
rx_gain_changed = 1; // Flag to request a big RX Gain change
}
else if (min_fft_level > FAST_MAX_FFT_LEVEL)
{
rx_gain -= 5; // Plan to decrease RX Gain
rx_gain_changed = 1; // Flag to request a big RX Gain change
}
}
else
{
rx_gain_slow_count++; // Only make fine adjustments to rx_gain every second.
if (rx_gain_slow_count > 106) // Approx 106 blocks of samples per second
{
rx_gain_fast_attack = 0; // If the IF gain is close to correct for a second, stop fast IF Gain changes.
rx_gain_slow_count = 0;
if (min_fft_level < TARGET_FFT_LEVEL)
{
rx_gain += 1;
rx_gain_changed = 1; // Flag to request a small RX Gain change
}
else
{
rx_gain -= 1;
rx_gain_changed = 1; // Flag to request a small RX Gain change
}
}
}
if((!in_tx) && rx_gain_changed == 1)
{
// sound_mixer(audio_card, "Capture", rx_gain); // This function call is not needed
char rx_gain_buff[8];
(void) sprintf(rx_gain_buff, "%d", rx_gain);
set_field("r1:gain", rx_gain_buff);
rx_gain_changed = 0;
}
// ... back to the actual processing, after spectrum update
// we may add another sub receiver within the pass band later,
// hence, the linkced list of receivers here
// at present, we handle just the first receiver
struct rx *r = rx_list;
//STEP 4: we rotate the bins around by r-tuned_bin
for (i = 0; i < MAX_BINS; i++){
int b = i + r->tuned_bin;
if (b >= MAX_BINS)
b = b - MAX_BINS;
if (b < 0)
b = b + MAX_BINS;
r->fft_freq[i] = fft_out[b];
}
// STEP 5:zero out the other sideband
if (r->mode == MODE_LSB || r->mode == MODE_CWR)
for (i = 0; i < MAX_BINS/2; i++){
__real__ r->fft_freq[i] = 0;
__imag__ r->fft_freq[i] = 0;
}
else
for (i = MAX_BINS/2; i < MAX_BINS; i++){
__real__ r->fft_freq[i] = 0;
__imag__ r->fft_freq[i] = 0;
}
// STEP 6: apply the filter to the signal,
// in frequency domain we just multiply the filter
// coefficients with the frequency domain samples
for (i = 0; i < MAX_BINS; i++)
r->fft_freq[i] *= r->filter->fir_coeff[i];
//STEP 7: convert back to time domain
my_fftw_execute(r->plan_rev);
//STEP 8 : AGC
agc2(r);
//STEP 9: send the output back to where it needs to go
int is_digital = 0;
if (rx_list->output == 0){
for (i= 0; i < MAX_BINS/2; i++){
int32_t sample;
sample = cimag(r->fft_time[i+(MAX_BINS/2)]);
//keep transmit buffer empty
output_speaker[i] = sample;
output_tx[i] = 0;
}
//push the samples to the remote audio queue, decimated to 16000 samples/sec
for (i = 0; i < MAX_BINS/2; i += 6)
q_write(&qremote, output_speaker[i]);
}
if (mute_count){
memset(output_speaker, 0, MAX_BINS/2 * sizeof(int32_t));
mute_count--;
}
//push the data to any potential modem
modem_rx(rx_list->mode, output_speaker, MAX_BINS/2);
}
void read_power(){
uint8_t response[4];
int16_t vfwd, vref;
char buff[20];
if (!in_tx)
return;
if(i2cbb_read_i2c_block_data(0x8, 0, 4, response) == -1)
return;
vfwd = vref = 0;
memcpy(&vfwd, response, 2);
memcpy(&vref, response+2, 2);
// printf("%d:%d\n", vfwd, vref);
if (vref >= vfwd)
vswr = 100;
else
vswr = (10*(vfwd + vref))/(vfwd-vref);
//here '400' is the scaling factor as our ref power output is 40 watts
//this calculates the power as 1/10th of a watt, 400 = 40 watts
int fwdvoltage = (vfwd * 40)/bridge_compensation;
fwdpower = (fwdvoltage * fwdvoltage)/400;
int rf_v_p2p = (fwdvoltage * 126)/400;
// printf("rf volts: %d, alc %g, %d watts ", rf_v_p2p, alc_level, fwdpower/10);
if (rf_v_p2p > 135 && !in_calibration){
alc_level *= 135.0 / (1.0 * rf_v_p2p);
printf("ALC tripped, to %d percent\n", (int)(100 * alc_level));
}
/* else if (alc_level < 0.95){
printf("alc releasing to ");
alc_level *= 1.02;
}
*/
// printf("alc: %g\n", alc_level);
}
static int tx_process_restart = 0;
void tx_process(
int32_t *input_rx, int32_t *input_mic,
int32_t *output_speaker, int32_t *output_tx,
int n_samples)
{
int i;
double i_sample, q_sample;
struct rx *r = tx_list;
//fix the burst at the start of transmission
if (tx_process_restart){
fft_reset_m_bins();
tx_process_restart = 0;
}
if (mute_count && (r->mode == MODE_USB || r->mode == MODE_LSB)){
memset(input_mic, 0, n_samples * sizeof(int32_t));
mute_count--;
}
//first add the previous M samples
for (i = 0; i < MAX_BINS/2; i++)
fft_in[i] = fft_m[i];
int m = 0;
int j = 0;
//double max = -10.0, min = 10.0;
//gather the samples into a time domain array
for (i= MAX_BINS/2; i < MAX_BINS; i++){
if (r->mode == MODE_2TONE)
i_sample = (1.0 * (vfo_read(&tone_a)
+ vfo_read(&tone_b))) / 50000000000.0;
else if (r->mode == MODE_CALIBRATE)
i_sample = (1.0 * (vfo_read(&tone_a))) / 25000000000.0;
else if (r->mode == MODE_CW || r->mode == MODE_CWR || r->mode == MODE_FT8)
i_sample = modem_next_sample(r->mode) / 3;
else {
i_sample = (1.0 * input_mic[j]) / 2000000000.0;
}
//clip the overdrive to prevent damage up the processing chain, PA
if (r->mode == MODE_USB || r->mode == MODE_LSB){
if (i_sample < (-1.0 * voice_clip_level))
i_sample = -1.0 * voice_clip_level;
else if (i_sample > voice_clip_level)
i_sample = voice_clip_level;
}
/*
//to measure the voice peaks, used to measure voice_clip_level
if (max < i_sample)
max = i_sample;
if (min > i_sample)
min = i_sample;
*/
//don't echo the voice modes
if (r->mode == MODE_USB || r->mode == MODE_LSB || r->mode == MODE_AM
|| r->mode == MODE_NBFM)
output_speaker[j] = 0;
else
output_speaker[j] = i_sample * sidetone;
q_sample = 0;
j++;
__real__ fft_m[m] = i_sample;
__imag__ fft_m[m] = q_sample;
__real__ fft_in[i] = i_sample;
__imag__ fft_in[i] = q_sample;
m++;
}
//push the samples to the remote audio queue, decimated to 16000 samples/sec
for (i = 0; i < MAX_BINS/2; i += 6)
q_write(&qremote, output_speaker[i]);
//convert to frequency
fftw_execute(plan_fwd);
// NOTE: fft_out holds the fft output (in freq domain) of the
// incoming mic samples
// the naming is unfortunate
// apply the filter
for (i = 0; i < MAX_BINS; i++)
fft_out[i] *= tx_filter->fir_coeff[i];
// the usb extends from 0 to MAX_BINS/2 - 1,
// the lsb extends from MAX_BINS - 1 to MAX_BINS/2 (reverse direction)
// zero out the other sideband
// TBD: Something strange is going on, this should have been the otherway
if (r->mode == MODE_LSB || r->mode == MODE_CWR)
// zero out the LSB
for (i = 0; i < MAX_BINS/2; i++){
__real__ fft_out[i] = 0;
__imag__ fft_out[i] = 0;
}
else
// zero out the USB
for (i = MAX_BINS/2; i < MAX_BINS; i++){
__real__ fft_out[i] = 0;
__imag__ fft_out[i] = 0;
}
//now rotate to the tx_bin
for (i = 0; i < MAX_BINS; i++){
int b = i + tx_shift;
if (b >= MAX_BINS)
b = b - MAX_BINS;
if (b < 0)
b = b + MAX_BINS;
r->fft_freq[b] = fft_out[i];
}
// the spectrum display is updated
//spectrum_update();
//convert back to time domain
fftw_execute(r->plan_rev);
int min = 10000000;
int max = -10000000;
float scale = volume;
for (i= 0; i < MAX_BINS/2; i++){
double s = creal(r->fft_time[i+(MAX_BINS/2)]);
output_tx[i] = s * scale * tx_amp * alc_level;
if (min > output_tx[i])
min = output_tx[i];
if (max < output_tx[i])
max = output_tx[i];
//output_tx[i] = 0;
}
// printf("min %d, max %d\n", min, max);
read_power();
sdr_modulation_update(output_tx, MAX_BINS/2, tx_amp);
}
/*
This is called each time there is a block of signal samples ready
either from the mic or from the rx IF
*/
void sound_process(
int32_t *input_rx, int32_t *input_mic,
int32_t *output_speaker, int32_t *output_tx,
int n_samples)
{
if (in_tx)
tx_process(input_rx, input_mic, output_speaker, output_tx, n_samples);
else
rx_process(input_rx, input_mic, output_speaker, output_tx, n_samples);
if (pf_record)
wav_record(in_tx == 0 ? output_speaker : input_mic, n_samples);
}
void set_rx_filter(){
if(rx_list->mode == MODE_LSB || rx_list->mode == MODE_CWR)
filter_tune(rx_list->filter,
(1.0 * -rx_list->high_hz)/96000.0,
(1.0 * -rx_list->low_hz)/96000.0 ,
5);
else
filter_tune(rx_list->filter,
(1.0 * rx_list->low_hz)/96000.0,
(1.0 * rx_list->high_hz)/96000.0 ,
5);
}
/*
Write code that mus repeatedly so things, it is called during the idle time
of the event loop
*/
void loop(){
delay(10);
}
void signal_handler(int signum){
digitalWrite(TX_LINE, LOW);
}
void setup_audio_codec(){
strcpy(audio_card, "hw:0");
//configure all the channels of the mixer
sound_mixer(audio_card, "Input Mux", UNUSED_PARAMETER, 0);
sound_mixer(audio_card, "Line", UNUSED_PARAMETER, 1);
sound_mixer(audio_card, "Mic", UNUSED_PARAMETER, 0);
sound_mixer(audio_card, "Mic Boost", RX_VOLUME_CONTROL, 0); // One of these is probably not required - N3SB
sound_mixer(audio_card, "Mic Boost", TX_GAIN_CONTROL, 0); // One of these is probably not required - N3SB
sound_mixer(audio_card, "Playback Deemphasis", UNUSED_PARAMETER, 0);
// Set a starting sound level
sound_mixer(audio_card, "Master", RX_VOLUME_CONTROL, 10); // Need to change - N3SB
sound_mixer(audio_card, "Master", TX_GAIN_CONTROL, 0); // Now initializing TX Gain to 0 - N3SB
sound_mixer(audio_card, "Output Mixer HiFi", UNUSED_PARAMETER, 1);
sound_mixer(audio_card, "Output Mixer Mic Sidetone", UNUSED_PARAMETER, 0);
}
void setup_oscillators(){
//initialize the SI5351
delay(200);
si5351bx_init();
delay(200);
si5351bx_setfreq(1, bfo_freq);
delay(200);
si5351bx_setfreq(1, bfo_freq);
si5351_reset();
}
static int hw_init_index = 0;
static int hw_settings_handler(void* user, const char* section,
const char* name, const char* value)
{
char cmd[1000];
char new_value[200];
if (!strcmp(name, "f_start"))