mirror of
https://github.com/radio95-rnt/fm95.git
synced 2026-02-26 19:23:51 +01:00
95 lines
3.3 KiB
C
95 lines
3.3 KiB
C
#include "bs412.h"
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#define BS412_TIME 60
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#define CLAMP(x, lo, hi) (((x) < (lo)) ? (lo) : ((x) > (hi) ? (hi) : (x)))
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#define SQRT19000 180499999.99999997f // (19000 / sqrt(2)) * 19000 / sqrt(2)
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// inline float dbr_to_deviation(float dbr) {
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// return 19000.0f * sqrtf(pow(10.0, dbr / 10.0));
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// }
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inline float deviation_to_dbr(float deviation) {
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if (deviation < 1e-6f) return -100.0f;
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return 10*log10f((deviation*deviation)/SQRT19000);
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}
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void init_bs412(BS412Compressor* comp, uint32_t mpx_deviation, float target_power, float attack, float release, float max_gain, uint32_t sample_rate) {
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comp->mpx_deviation = mpx_deviation;
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comp->avg_power = 0.0f;
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comp->alpha = 1.0f / (BS412_TIME * sample_rate);
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comp->sample_rate = sample_rate;
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comp->attack = expf(-1.0f / (attack * sample_rate));
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comp->release = expf(-1.0f / (release * sample_rate));
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comp->target = deviation_to_dbr(19000.0f * pow(10.0, target_power / 10.0)); // target is expected to not be our rms format
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comp->gain = 0.0f;
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comp->can_compress = 0;
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comp->second_counter = 0;
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comp->last_output = 0.0f;
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comp->max_gain = max_gain;
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#ifdef BS412_DEBUG
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debug_printf("Initialized MPX power measurement with sample rate: %d\n", sample_rate);
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#endif
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}
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float bs412_compress(BS412Compressor* comp, float audio, float sample_mpx) {
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float combined = audio + sample_mpx;
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comp->avg_power += comp->alpha * ((comp->last_output * comp->last_output * comp->mpx_deviation * comp->mpx_deviation) - comp->avg_power);
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float avg_deviation = sqrtf(comp->avg_power);
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float modulation_power = deviation_to_dbr(avg_deviation);
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if(comp->target <= -100.0f) {
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#ifdef BS412_DEBUG
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if(comp->sample_counter > comp->sample_rate) {
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debug_printf("MPX power: %.2f dBr (%.1f Hz)\n", modulation_power, avg_deviation);
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comp->sample_counter = 0;
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}
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#endif
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comp->sample_counter++;
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return combined;
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}
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if(comp->sample_counter > comp->sample_rate) {
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#ifdef BS412_DEBUG
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debug_printf("MPX power: %.2f dBr with gain %.2fx (%.2f dBr)\n", modulation_power, mpx->gain, deviation_to_dbr(avg_deviation * mpx->gain));
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#endif
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comp->sample_counter = 0;
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if(comp->can_compress == 0) comp->second_counter++;
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}
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if(comp->can_compress == 0 && comp->second_counter > BS412_TIME) {
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#ifdef BS412_DEBUG
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debug_printf("Can compress.\n");
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#endif
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comp->gain = powf(10.0f, (comp->target - modulation_power) / 10.0f);
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comp->can_compress = 1;
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}
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if(comp->can_compress == 0) {
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comp->sample_counter++;
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return combined;
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}
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float target_gain = expf((comp->target - modulation_power) * 0.2302585093f); // 1/10 * ln(10)
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if (modulation_power > comp->target) comp->gain = comp->attack * comp->gain + (1.0f - comp->attack) * target_gain;
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else comp->gain = comp->release * comp->gain + (1.0f - comp->release) * target_gain;
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comp->gain = CLAMP(comp->gain, 0.0f, comp->max_gain);
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float output_sample = (audio * comp->gain) + sample_mpx;
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if(deviation_to_dbr(avg_deviation * comp->gain) > comp->target && modulation_power < comp->target) {
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// Gain is too much, reduce
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float overshoot_dbr = deviation_to_dbr(avg_deviation * comp->gain) - comp->target;
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float reduction_factor = powf(10.0f, -overshoot_dbr / 10.0f);
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comp->gain *= reduction_factor;
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comp->gain = fmaxf(0.01f, fminf(comp->max_gain, comp->gain));
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}
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comp->sample_counter++;
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comp->last_output = output_sample;
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return output_sample;
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}
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