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mirror of https://github.com/radio95-rnt/fm95.git synced 2026-02-26 19:23:51 +01:00

change bs412 gain reduction logic, still issue #2 should be unchanged

This commit is contained in:
2025-06-09 16:10:27 +02:00
parent 4764e5768e
commit f11f25c8f0
2 changed files with 24 additions and 7 deletions

View File

@@ -1,5 +1,7 @@
#include "bs412.h"
#define BS412_RMS
#define LOG2_19000 log2f(19000.0f)
inline float dbr_to_deviation(float dbr) {
@@ -21,11 +23,19 @@ void init_modulation_power_measure(MPXPowerMeasurement* mpx, int sample_rate) {
}
float measure_mpx(MPXPowerMeasurement* mpx, float deviation) {
#ifdef BS412_RMS
mpx->sample += deviation * deviation; // rmS
#else
mpx->sample += fabsf(deviation);
#endif
mpx->sample_counter++;
float inv_counter = 1.0f / mpx->sample_counter;
#ifdef BS412_RMS
float avg_deviation = sqrtf(mpx->sample * inv_counter); // RMs
#else
float avg_deviation = mpx->sample * inv_counter; // RMs
#endif
float modulation_power = deviation_to_dbr(avg_deviation);
#ifdef BS412_DEBUG
@@ -38,7 +48,7 @@ float measure_mpx(MPXPowerMeasurement* mpx, float deviation) {
#ifdef BS412_DEBUG
debug_printf("Resetting MPX power measurement\n");
#endif
mpx->sample = avg_deviation * avg_deviation;
mpx->sample = deviation * deviation;
mpx->sample_counter = 1;
}

View File

@@ -3,7 +3,7 @@
#include <getopt.h>
#include <liquid/liquid.h>
#define LPF_ORDER 18
#define LPF_ORDER 17
#define buffer_maxlength 12288
#define buffer_tlength_fragsize 12288
@@ -518,11 +518,18 @@ int main(int argc, char **argv) {
float mpower = measure_mpx(&power, (audio+mpx) * mpx_deviation); // Standard requires that the output is measured specifically
if (mpower > mpx_power) {
float excess_power = mpower - mpx_power;
excess_power = deviation_to_dbr(dbr_to_deviation(excess_power) - dbr_to_deviation(mpx_only)); // make sure mpx is not included in the power to attenuate, because we'd be attuating the mpx signal for audio
float target_gain = dbr_to_deviation(-excess_power)/mpx_deviation;
bs412_audio_gain = 0.9f * bs412_audio_gain + 0.1f * target_gain;
audio *= bs412_audio_gain;
excess_power = deviation_to_dbr(dbr_to_deviation(excess_power) - dbr_to_deviation(mpx_only));
if (excess_power > 0.0f && excess_power < 10.0f) {
float target_gain = dbr_to_deviation(-excess_power) / mpx_deviation;
target_gain = fmaxf(target_gain, 0.1f);
target_gain = fminf(target_gain, 1.0f);
bs412_audio_gain = 0.9f * bs412_audio_gain + 0.1f * target_gain;
}
} else {
bs412_audio_gain = fminf(1.0f, bs412_audio_gain + 0.001f);
}
iirfilt_rrrf_execute(mpx_lpf, audio, &audio); // Should have no effect, as audio should be 0-15, and 23-53, this is a filter for 53, assuming the filter is good, this is precaution and recomendation