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mirror of https://github.com/radio95-rnt/fm95.git synced 2026-02-27 03:23:54 +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

View File

@@ -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;
}