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Fixed FT8 waveform generation for high frequencies

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This commit is contained in:
Karlis Goba 2019-11-15 10:22:45 +02:00
parent 0b8f6f8b48
commit 2d1337d47e
1 changed files with 66 additions and 3 deletions
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69
gen_ft8.cpp
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@ -4,12 +4,74 @@
#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.
@ -23,8 +85,8 @@ void synth_fsk(const uint8_t *symbols, int num_symbols, float f0, float spacing,
int i = 0;
while (j < num_symbols) {
float f = f0 + symbols[j] * spacing;
phase += 2 * M_PI * f / signal_rate;
signal[i] = sin(phase);
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
@ -97,7 +159,8 @@ int main(int argc, char **argv) {
signal[i] = 0;
}
synth_fsk(tones, ft8::NN, frequency, symbol_rate, symbol_rate, sample_rate, signal + num_silence);
// 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;