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mirror of https://github.com/radio95-rnt/fm95.git synced 2026-02-26 11:22:00 +01:00

tilt correction? (yes, ai wrote it, you hate me because of that? find a tilt filter yourself then that you can copy the code of)

This commit is contained in:
2025-07-08 19:54:15 +02:00
parent 8932a9a24d
commit 52f2f61a9a
3 changed files with 64 additions and 1 deletions

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@@ -13,4 +13,49 @@ inline float apply_preemphasis(ResistorCapacitor *filter, float sample) {
float out = (sample - filter->alpha * filter->prev_sample) * filter->gain;
filter->prev_sample = sample;
return out;
}
void tilt_init(TiltCorrectionFilter* filter, float correction_strength) {
// This filter is a first-order IIR low-shelf filter.
// The difference equation is: y[n] = b0*x[n] + b1*x[n-1] - a1*y[n-1]
// We simplify it to y[n] = x[n] - a1*y[n-1] which acts as a leaky integrator.
// The "correction_strength" is our leaky factor. It is the pole of the filter.
// A value close to 1.0 places the pole very close to the unit circle,
// providing a large boost to low frequencies (and DC).
if (correction_strength >= 1.0f) {
correction_strength = 0.99999f; // Prevent instability
}
filter->b0 = 1.0f;
filter->b1 = 0.0f;
filter->a1 = -correction_strength; // The feedback coefficient
// Reset filter state
filter->x_prev = 0.0f;
filter->y_prev = 0.0f;
}
float tilt(TiltCorrectionFilter* filter, float input_sample) {
// Apply the difference equation: y[n] = b0*x[n] + b1*x[n-1] - a1*y[n-1]
float output_sample = filter->b0 * input_sample + filter->b1 * filter->x_prev - filter->a1 * filter->y_prev;
// Important: Prevent output from running away due to DC offset accumulation
// This is a simple guard. If the filter becomes unstable or the output
// grows too large, it gets reset. For square waves, the absolute value of the
// output should not significantly exceed the absolute value of the input.
if (fabsf(output_sample) > 2.0f * fabsf(input_sample) && fabsf(input_sample) > 0.001f) {
// This condition indicates the filter state might be diverging. Resetting it.
// You may need to adjust the '2.0f' factor based on your signal.
filter->y_prev = 0;
output_sample = input_sample;
}
// Update the state for the next iteration
filter->x_prev = input_sample;
filter->y_prev = output_sample;
return output_sample;
}

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@@ -12,3 +12,14 @@ typedef struct
void init_preemphasis(ResistorCapacitor *filter, float tau, float sample_rate, float ref_freq);
float apply_preemphasis(ResistorCapacitor *filter, float sample);
typedef struct {
float b0, b1;
float a1;
float x_prev;
float y_prev;
} TiltCorrectionFilter;
void tilt_init(TiltCorrectionFilter *filter, float correction_strength);
float tilt(TiltCorrectionFilter *filter, float input_sample);

View File

@@ -65,6 +65,7 @@ typedef struct
float clipper_threshold;
float preemphasis;
float tilt;
uint8_t calibration;
float mpx_power;
float mpx_deviation;
@@ -168,6 +169,9 @@ int run_fm95(const FM95_Config config, FM95_Runtime* runtime) {
MPXPowerMeasurement power;
init_modulation_power_measure(&power, config.sample_rate);
TiltCorrectionFilter tilter;
tilt_init(&tilter, config.tilt);
StereoEncoder stencode;
init_stereo_encoder(&stencode, 4.0f, &osc, (config.stereo == 2), config.volumes.mono, config.volumes.pilot, config.volumes.stereo);
@@ -248,7 +252,7 @@ int run_fm95(const FM95_Config config, FM95_Runtime* runtime) {
mpx *= bs412_audio_gain;
output[i] = hard_clip((mpx_in[i]+mpx)*config.master_volume, 1.0); // Ensure peak deviation of 75 khz, assuming we're calibrated correctly (lower)
output[i] = hard_clip(tilt(&tilter, (mpx_in[i]+mpx))*config.master_volume, 1.0); // Ensure peak deviation of 75 khz, assuming we're calibrated correctly (lower)
advance_oscillator(&osc);
}
@@ -365,6 +369,8 @@ static int config_handler(void* user, const char* section, const char* name, con
pconfig->volumes.rds = strtof(value, NULL);
} else if(MATCH("volumes", "rds_step")) {
pconfig->volumes.rds_step = strtof(value, NULL);
} else if(MATCH("fm95", "tilt")) {
pconfig->tilt = strtof(value, NULL);
} else {
return 0; // Unknown section/name
}
@@ -452,6 +458,7 @@ int main(int argc, char **argv) {
.clipper_threshold = DEFAULT_CLIPPER_THRESHOLD,
.preemphasis = DEFAULT_PREEMPHASIS_TAU,
.tilt = 0,
.calibration = 0,
.mpx_power = DEFAULT_MPX_POWER,
.mpx_deviation = DEFAULT_MPX_DEVIATION,