kgoba-ft8_lib/gen_ft8.cpp

184 wiersze
5.1 KiB
C++

#include <cstdlib>
#include <cstring>
#include <cstdio>
#include <cmath>
#include "common/wave.h"
#include "common/debug.h"
//#include "ft8/v1/pack.h"
//#include "ft8/v1/encode.h"
#include "ft8/pack.h"
#include "ft8/encode.h"
#include "ft8/constants.h"
#define LOG_LEVEL LOG_INFO
void gfsk_pulse(int n_spsym, float b, float *pulse)
{
const float c = M_PI * sqrtf(2 / logf(2));
for (int i = 0; i < 3 * n_spsym; ++i)
{
float t = i / (float)n_spsym - 1.5f;
pulse[i] = (erff(c * b * (t + 0.5f)) - erff(c * b * (t - 0.5f))) / 2;
}
}
// Same as synth_fsk, but uses GFSK phase shaping
void synth_gfsk(const uint8_t *symbols, int n_sym, float f0, int n_spsym, int signal_rate, float *signal)
{
LOG(LOG_DEBUG, "n_spsym = %d\n", n_spsym);
int n_wave = n_sym * n_spsym;
float hmod = 1.0f;
// Compute the smoothed frequency waveform.
// Length = (nsym+2)*nsps samples, first and last symbols extended
float dphi_peak = 2 * M_PI * hmod / n_spsym;
float dphi[n_wave + 2 * n_spsym];
// Shift frequency up by f0
for (int i = 0; i < n_wave + 2 * n_spsym; ++i)
{
dphi[i] = 2 * M_PI * f0 / signal_rate;
}
float pulse[3 * n_spsym];
gfsk_pulse(n_spsym, 2.0f, pulse);
for (int i = 0; i < n_sym; ++i)
{
int ib = i * n_spsym;
for (int j = 0; j < 3 * n_spsym; ++j)
{
dphi[j + ib] += dphi_peak * symbols[i] * pulse[j];
}
}
// Add dummy symbols at beginning and end with tone values equal to 1st and last symbol, respectively
for (int j = 0; j < 2 * n_spsym; ++j)
{
dphi[j] += dphi_peak * pulse[j + n_spsym] * symbols[0];
dphi[j + n_sym * n_spsym] += dphi_peak * pulse[j] * symbols[n_sym - 1];
}
// Calculate and insert the audio waveform
float phi = 0;
for (int k = 0; k < n_wave; ++k)
{ // Don't include dummy symbols
signal[k] = sinf(phi);
phi = fmodf(phi + dphi[k + n_spsym], 2 * M_PI);
}
// Apply envelope shaping to the first and last symbols
int n_ramp = n_spsym / 8;
for (int i = 0; i < n_ramp; ++i)
{
float env = (1 - cosf(2 * M_PI * i / (2 * n_ramp))) / 2;
signal[i] *= env;
signal[n_wave - 1 - i] *= env;
}
}
// Convert a sequence of symbols (tones) into a sinewave of continuous phase (FSK).
// Symbol 0 gets encoded as a sine of frequency f0, the others are spaced in increasing
// fashion.
void synth_fsk(const uint8_t *symbols, int num_symbols, float f0, float spacing,
float symbol_rate, float signal_rate, float *signal)
{
float phase = 0;
float dt = 1 / signal_rate;
float dt_sym = 1 / symbol_rate;
float t = 0;
int j = 0;
int i = 0;
while (j < num_symbols)
{
float f = f0 + symbols[j] * spacing;
phase = fmodf(phase + 2 * M_PI * f / signal_rate, 2 * M_PI);
signal[i] = sinf(phase);
t += dt;
if (t >= dt_sym)
{
// Move to the next symbol
t -= dt_sym;
++j;
}
++i;
}
}
void usage()
{
printf("Generate a 15-second WAV file encoding a given message.\n");
printf("Usage:\n");
printf("\n");
printf("gen_ft8 MESSAGE WAV_FILE [FREQUENCY]\n");
printf("\n");
printf("(Note that you might have to enclose your message in quote marks if it contains spaces)\n");
}
int main(int argc, char **argv)
{
// Expect two command-line arguments
if (argc < 3)
{
usage();
return -1;
}
const char *message = argv[1];
const char *wav_path = argv[2];
float frequency = 1000.0;
if (argc > 3)
{
frequency = atof(argv[3]);
}
// First, pack the text data into binary message
uint8_t packed[FT8_K_BYTES];
//int rc = packmsg(message, packed);
int rc = pack77(message, packed);
if (rc < 0)
{
printf("Cannot parse message!\n");
printf("RC = %d\n", rc);
return -2;
}
printf("Packed data: ");
for (int j = 0; j < 10; ++j)
{
printf("%02x ", packed[j]);
}
printf("\n");
// Second, encode the binary message as a sequence of FSK tones
uint8_t tones[FT8_NN]; // FT8_NN = 79, lack of better name at the moment
//genft8(packed, 0, tones);
genft8(packed, tones);
printf("FSK tones: ");
for (int j = 0; j < FT8_NN; ++j)
{
printf("%d", tones[j]);
}
printf("\n");
// Third, convert the FSK tones into an audio signal
const int sample_rate = 12000;
const float symbol_rate = 6.25f;
const int num_samples = (int)(0.5f + FT8_NN / symbol_rate * sample_rate);
const int num_silence = (15 * sample_rate - num_samples) / 2;
float signal[num_silence + num_samples + num_silence];
for (int i = 0; i < num_silence + num_samples + num_silence; i++)
{
signal[i] = 0;
}
// synth_fsk(tones, FT8_NN, frequency, symbol_rate, symbol_rate, sample_rate, signal + num_silence);
synth_gfsk(tones, FT8_NN, frequency, sample_rate / symbol_rate, sample_rate, signal + num_silence);
save_wav(signal, num_silence + num_samples + num_silence, sample_rate, wav_path);
return 0;
}