mirror of
https://github.com/radio95-rnt/fm95.git
synced 2026-02-26 19:23:51 +01:00
damn it
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2
.vscode/.server-controller-port.log
vendored
2
.vscode/.server-controller-port.log
vendored
@@ -1,5 +1,5 @@
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{
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"port": 13452,
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"time": 1742842256267,
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"time": 1742920067458,
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"version": "0.0.3"
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}
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@@ -13,32 +13,3 @@ float apply_preemphasis(ResistorCapacitor *filter, float sample) {
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float hard_clip(float sample, float threshold) {
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return fmaxf(-threshold, fminf(threshold, sample));
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}
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float sincf(float x) {
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return (x == 0.0f) ? 1.0f : sinf(M_PI * x) / (M_PI * x);
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}
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void init_lpf(FIRFilter *lpf, float freq) {
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int m = FILTER_LEN - 1;
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float sum = 0.0f;
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for(int n = 0; n < FILTER_LEN; n++) {
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float x = n-m/2.0f;
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lpf->filter[n] = sincf(2.0f*freq*x) * (0.54 - 0.46 * cosf(M_2PI * n / m));
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sum += lpf->filter[n];
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}
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for(int n = 0; n < FILTER_LEN; n++) {
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lpf->filter[n] /= sum;
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}
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}
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float fir_filter(FIRFilter *fir, float sample) {
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float out = 0.0f;
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for(int i = 0; i < FILTER_LEN; i++) {
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out += fir->filter[i] * sample;
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}
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fir->filter_idx++;
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if(fir->filter_idx == FILTER_LEN) fir->filter_idx = 0;
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return out;
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}
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@@ -17,11 +17,3 @@ void init_preemphasis(ResistorCapacitor *filter, float tau, float sample_rate);
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float apply_preemphasis(ResistorCapacitor *filter, float sample);
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float hard_clip(float sample, float threshold);
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typedef struct filters
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{
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float filter[FILTER_LEN];
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int filter_idx;
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} FIRFilter;
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void init_lpf(FIRFilter *lpf, float freq);
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float fir_filter(FIRFilter *fir, float sample);
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@@ -414,10 +414,6 @@ int main(int argc, char **argv) {
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ResistorCapacitor preemp_l, preemp_r;
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init_preemphasis(&preemp_l, preemphasis_tau, sample_rate);
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init_preemphasis(&preemp_r, preemphasis_tau, sample_rate);
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FIRFilter lpf_l, lpf_r;
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init_lpf(&lpf_l, 15000);
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init_lpf(&lpf_r, 15000);
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// #endregion
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signal(SIGINT, stop);
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@@ -475,8 +471,6 @@ int main(int argc, char **argv) {
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float ready_l = apply_preemphasis(&preemp_l, l_in)*2;
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float ready_r = apply_preemphasis(&preemp_r, r_in)*2;
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ready_l = fir_filter(&lpf_l, ready_l);
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ready_r = fir_filter(&lpf_l, ready_r);
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ready_l = hard_clip(ready_l*audio_volume, clipper_threshold);
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ready_r = hard_clip(ready_r*audio_volume, clipper_threshold);
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