radio95/fm95
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bug fixes

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This commit is contained in:
KubaPro010
2025-05-31 11:14:07 +02:00
parent ffcbc4b0c0
commit 5fd2383b03
3 changed files with 13 additions and 21 deletions
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4
dsp/filters.c
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@@ -12,7 +12,3 @@ inline float apply_preemphasis(ResistorCapacitor *filter, float sample) {
filter->prev_sample = sample;
return out;
}
inline float hard_clip(float sample, float threshold) {
return fmaxf(-threshold, fminf(threshold, sample));
}

2
dsp/filters.h
View File

@@ -12,5 +12,3 @@ typedef struct
void init_preemphasis(ResistorCapacitor *filter, float tau, float sample_rate);
float apply_preemphasis(ResistorCapacitor *filter, float sample);
float hard_clip(float sample, float threshold);

24
src/fm95.c
View File

@@ -78,14 +78,13 @@ void uninterleave(const float *input, float *left, float *right, size_t num_samp
}
float compute_darc_amplitude(float stereo_injection_percent) {
if (stereo_injection_percent <= 0.025f) {
return 0.04f;
} else if (stereo_injection_percent <= 0.05f) {
float slope = (0.1f - 0.04f) / (0.05f - 0.025f);
return 0.04f + slope * (stereo_injection_percent - 0.025f);
} else {
return 0.1f;
if (stereo_injection_percent <= 0.025f) return 0.04f;
else if (stereo_injection_percent <= 0.05f) return 0.04f + ((0.1f - 0.04f) / (0.05f - 0.025f)) * (stereo_injection_percent - 0.025f);
else return 0.1f;
}
inline float hard_clip(float sample, float threshold) {
return fmaxf(-threshold, fminf(threshold, sample));
}
static void stop(int signum) {
@@ -530,18 +529,17 @@ int main(int argc, char **argv) {
float ready_l = apply_preemphasis(&preemp_l, left[i]);
float ready_r = apply_preemphasis(&preemp_r, right[i]);
iirfilt_rrrf_execute(lpf_l, ready_l, &ready_l);
iirfilt_rrrf_execute(lpf_r, ready_r, &ready_r);
ready_l = process_agc_stereo(&agc, ready_l, ready_r, &ready_r);
ready_l = hard_clip(ready_l*audio_volume, clipper_threshold);
ready_r = hard_clip(ready_r*audio_volume, clipper_threshold);
float mid = (ready_l + ready_r) / 2.0f;
float side = (ready_l - ready_r) / 2.0f;
audio = mid*MONO_VOLUME;
if(stereo) {
float side = (ready_l - ready_r) / 2.0f;
float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, polar_stereo ? 1 : 8); // 31.25 or 38 KHz
if(polar_stereo) audio += ((side+0.2)*stereo_carrier)*STEREO_VOLUME; // 0.2 in polar stereo because it also includes a carrier wave, so we add a carrier wave via DC
@@ -554,7 +552,7 @@ int main(int argc, char **argv) {
if(rds_on && polar_stereo == 0) {
float rds_carrier = get_oscillator_cos_multiplier_ni(&osc, 12); // 57 KHz
mpx += (rds1_in[i]*rds_carrier)*RDS_VOLUME;
if(!darc_on) { // DARC is hardcoded into 76 khz, according to a screenshot of a fm mpx with darc in it, it takes like 65 to 85 khz
if(!darc_on) { // DARC is hardcoded into 76 khz, according to a screenshot of a fm mpx with darc in it, it takes like 65 to 85 khz, so all of RDS2
float rds2_carrier_66 = get_oscillator_cos_multiplier_ni(&osc, 14); // 66.5 KHz
mpx += (rds2_in[i]*rds2_carrier_66)*RDS2_VOLUME;
if(rds2_on) {
@@ -565,9 +563,9 @@ int main(int argc, char **argv) {
}
}
}
if(mpx_on) mpx += hard_clip(mpx_in[i], 1.0f)*MPX_VOLUME;
if(mpx_on) mpx += mpx_in[i]*MPX_VOLUME;
if(sca_on) mpx += modulate_fm(&sca_mod, hard_clip(sca_in[i], sca_clipper_threshold))*SCA_VOLUME;
if(darc_on && polar_stereo == 0) mpx += hard_clip(refrenced_modulate_fm(&darc_modulator, last_darc_data, 16.0f)*compute_darc_amplitude(stereo*STEREO_VOLUME), 0.1f); // should never be over 10%, the docs say so
if(darc_on && polar_stereo == 0) mpx += refrenced_modulate_fm(&darc_modulator, last_darc_data, 16.0f)*compute_darc_amplitude(side*STEREO_VOLUME);
float mpx_only = measure_mpx(&mpx_only_power, mpx * mpx_deviation);
float mpower = measure_mpx(&power, (audio+mpx) * mpx_deviation); // Standard requires that the output is measured specifically