radio95/fm95
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start modularization

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This commit is contained in:
KubaPro010
2025-06-21 15:06:27 +02:00
parent 33ffcd9682
commit 16f0a53c05
18 changed files with 79 additions and 43 deletions
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2
src/chimer95.c
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@@ -8,7 +8,7 @@
#define buffer_tlength_fragsize 1024
#define buffer_prebuf 0
#include "../lib/oscillator.h"
#include "../dsp/oscillator.h"
#include "../lib/optimization.h"
#define DEFAULT_FREQ 1000.0f

2
src/dcf95.c
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@@ -10,7 +10,7 @@
// #define DEBUG
#include "../lib/oscillator.h"
#include "../dsp/oscillator.h"
#define DEFAULT_FREQ 77500.0f
#define DEFAULT_SAMPLE_RATE 192000

43
src/fm95.c
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@@ -19,12 +19,13 @@
#define DEFAULT_MPX_DEVIATION 75000.0f // for BS412
#define DEFAULT_DEVIATION 75000.0f // another way to set the volume
#include "../lib/oscillator.h"
#include "../dsp/filters.h"
#include "../dsp/fm_modulator.h"
#include "../dsp/oscillator.h"
#include "../filter/iir.h"
#include "../modulation/fm_modulator.h"
#include "../modulation/stereo_encoder.h"
#include "../lib/optimization.h"
#include "../dsp/bs412.h"
#include "../dsp/gain_control.h"
#include "../filter/bs412.h"
#include "../filter/gain_control.h"
#define DEFAULT_SAMPLE_RATE 192000
@@ -343,18 +344,19 @@ int main(int argc, char **argv) {
FMModulator sca_mod;
init_fm_modulator(&sca_mod, sca_frequency, sca_deviation, sample_rate);
iirfilt_rrrf lpf_l, lpf_r, mpx_lpf;
iirfilt_rrrf lpf_l, lpf_r;
lpf_l = iirfilt_rrrf_create_prototype(LIQUID_IIRDES_CHEBY2, LIQUID_IIRDES_LOWPASS, LIQUID_IIRDES_SOS, LPF_ORDER, (15000.0f/sample_rate), 0.0f, 1.0f, 40.0f);
lpf_r = iirfilt_rrrf_create_prototype(LIQUID_IIRDES_CHEBY2, LIQUID_IIRDES_LOWPASS, LIQUID_IIRDES_SOS, LPF_ORDER, (15000.0f/sample_rate), 0.0f, 1.0f, 40.0f);
mpx_lpf = iirfilt_rrrf_create_prototype(LIQUID_IIRDES_BUTTER, LIQUID_IIRDES_LOWPASS, LIQUID_IIRDES_SOS, 1, (polar_stereo ? (46250.0f/sample_rate) : (53000.0f/sample_rate)), 0.0f, 1.0f, 1.0f);
ResistorCapacitor preemp_l, preemp_r;
init_preemphasis(&preemp_l, preemphasis_tau, sample_rate, 15250.0f);
init_preemphasis(&preemp_r, preemphasis_tau, sample_rate, 15250.0f);
MPXPowerMeasurement power, mpx_only_power;
MPXPowerMeasurement power;
init_modulation_power_measure(&power, sample_rate);
init_modulation_power_measure(&mpx_only_power, sample_rate);
StereoEncoder stencode;
init_stereo_encoder(&stencode, 4.0f, &osc, polar_stereo, MONO_VOLUME, PILOT_VOLUME, STEREO_VOLUME);
float bs412_audio_gain = 1.0f;
@@ -410,7 +412,6 @@ int main(int argc, char **argv) {
for (uint16_t i = 0; i < BUFFER_SIZE; i++) {
float mpx = 0.0f;
float audio = 0.0f;
float ready_l = apply_preemphasis(&preemp_l, audio_stereo_input[2*i+0]);
float ready_r = apply_preemphasis(&preemp_r, audio_stereo_input[2*i+1]);
@@ -424,20 +425,8 @@ int main(int argc, char **argv) {
ready_l = hard_clip(ready_l*audio_volume, clipper_threshold);
ready_r = hard_clip(ready_r*audio_volume, clipper_threshold);
float mid = (ready_l + ready_r) * 0.5f;
float side = (ready_l - ready_r) * 0.5f;
mpx = stereo_encode(&stencode, stereo, ready_l, ready_r);
audio = mid*MONO_VOLUME;
if(stereo) {
float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, 8); // 31.25 or 38 KHz
if(polar_stereo) audio += ((side+0.2)*stereo_carrier)*STEREO_VOLUME; // 0.2 in polar stereo because it also includes a carrier wave, so we add a carrier wave via DC
else {
float pilot = get_oscillator_sin_multiplier_ni(&osc, 4); // 19 KHz
mpx += pilot*PILOT_VOLUME;
audio += (side*stereo_carrier)*STEREO_VOLUME;
}
}
if(rds_on && !polar_stereo) {
for(uint8_t stream = 0; stream < rds_streams; stream++) {
uint8_t osc_stream = 12+stream;
@@ -448,11 +437,9 @@ int main(int argc, char **argv) {
if(mpx_on) mpx += mpx_in[i];
if(sca_on) mpx += modulate_fm(&sca_mod, hard_clip(sca_in[i], sca_clipper_threshold))*SCA_VOLUME;
float mpxonly_power = measure_mpx(&mpx_only_power, mpx * mpx_deviation);
float mpx_power = measure_mpx(&power, (audio+mpx) * mpx_deviation); // Standard requires that the output is measured specifically
float mpx_power = measure_mpx(&power, mpx * mpx_deviation);
if (mpx_power > max_mpx_power) {
float excess_power = mpx_power - max_mpx_power;
excess_power = deviation_to_dbr(dbr_to_deviation(excess_power) - dbr_to_deviation(mpxonly_power));
if (excess_power > 0.0f && excess_power < 10.0f) {
float target_gain = dbr_to_deviation(-excess_power) / mpx_deviation;
@@ -466,9 +453,8 @@ int main(int argc, char **argv) {
bs412_audio_gain = fminf(1.0f, bs412_audio_gain + 0.001f);
}
audio *= bs412_audio_gain;
mpx *= bs412_audio_gain;
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
audio = hard_clip(audio, 1.0f-mpx); // Prevent clipping, via clipping the audio signal with relation to the mpx signal
output[i] = hard_clip((audio+mpx), 1.0f)*master_volume; // Ensure peak deviation of 75 khz, assuming we're calibrated correctly
@@ -485,7 +471,6 @@ int main(int argc, char **argv) {
printf("Cleaning up...\n");
iirfilt_rrrf_destroy(lpf_l);
iirfilt_rrrf_destroy(lpf_r);
iirfilt_rrrf_destroy(mpx_lpf);
free_PulseInputDevice(&input_device);
if(mpx_on) free_PulseInputDevice(&mpx_device);