radio95/fm95
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Code Issues Activity

change vban max packet size, add a rds2 device to fm95 and others

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This commit is contained in:
KubaPro010
2025-05-30 17:28:19 +02:00
parent 14630e5c7f
commit 7213c3b13d
7 changed files with 85 additions and 65 deletions
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1
README.md
View File

@@ -14,6 +14,7 @@ Supports these inputs:
- Audio (via Pulse)
- MPX (via Pulse, basically passthrough, i don't recommend this unless you have something else than rds or sca to modulate, you could run chimer95 via here)
- RDS (via Pulse, expects unmodulated RDS, stereo, left channel on 57 KHz, right on 66.5, rds95 is recommended here, in modulation this is inphase to the pilot)
- RDS2 (via Pulse, expects unmodulated RDS, stereo, left channel on 71.25 KHz, rigth on 76 KHz)
- SCA (via Pulse, by default on 67 khz with a 7 khz deviation)
- DARC (via Pulse, stereo, with one left clock channel [change when data is changed] and other one is data which is expected to be nrzi with bit 1 being 1 here, and bit 0 being -1)

4
dsp/bs412.c
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@@ -1,10 +1,10 @@
#include "bs412.h"
float dbr_to_deviation(float dbr) {
inline float dbr_to_deviation(float dbr) {
return 19000.0f * powf(10.0f, dbr / 10.0f);
}
float deviation_to_dbr(float deviation) {
inline float deviation_to_dbr(float deviation) {
if(deviation == 0.0f) return -100.0f;
return 10 * log10f(deviation / 19000.0f);
}

10
dsp/filters.c
View File

@@ -1,16 +1,18 @@
#include "filters.h"
void init_preemphasis(ResistorCapacitor *filter, float tau, float sample_rate) {
filter->prev_sample = 0.0f;
filter->alpha = expf(-1 / (tau*sample_rate));
float dt = 1.0f / sample_rate;
filter->alpha = tau / (tau + dt);
filter->gain = 1.0f / sqrtf(1.0f - filter->alpha);
filter->prev_sample = 0.0f;
}
float apply_preemphasis(ResistorCapacitor *filter, float sample) {
inline float apply_preemphasis(ResistorCapacitor *filter, float sample) {
float out = (sample - filter->alpha * filter->prev_sample) * filter->gain;
filter->prev_sample = sample;
return out;
}
float hard_clip(float sample, float threshold) {
inline float hard_clip(float sample, float threshold) {
return fmaxf(-threshold, fminf(threshold, sample));
}

2
dsp/oscillator.c
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@@ -6,7 +6,7 @@ void init_oscillator(Oscillator *osc, float frequency, float sample_rate) {
osc->sample_rate = sample_rate;
}
void change_oscillator_frequency(Oscillator *osc, float frequency) {
inline void change_oscillator_frequency(Oscillator *osc, float frequency) {
osc->phase_increment = (M_2PI * frequency) / osc->sample_rate;
}

2
src/chimer95.c
View File

@@ -17,7 +17,7 @@
#define OUTPUT_DEVICE "FM_MPX"
#define BUFFER_SIZE 256
#define BUFFER_SIZE 128
#include "../io/audio.h"

121
src/fm95.c
View File

@@ -32,6 +32,7 @@
#define INPUT_DEVICE "FM_Audio.monitor"
#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
#define RDS_DEVICE "RDS.monitor"
#define RDS2_DEVICE "\0" // Disabled, this is for the additional two RDS channels, 71.25 and 76 khz
#define MPX_DEVICE "FM_MPX.monitor"
#define SCA_DEVICE "\0" // Disabled
#define DARC_DEVICE "\0" // Disabled
@@ -46,8 +47,10 @@
#define MONO_VOLUME 0.45f // 45%
#define PILOT_VOLUME 0.09f // 9%
#define STEREO_VOLUME 0.3f // 30%
#define RDS_VOLUME 0.06f // 6%
#define RDS2_VOLUME 0.045f // 4.5%
#define RDS_VOLUME 0.0475f // 4.75%
#define RDS2_VOLUME 0.04f // 4%
#define RDS3_VOLUME 0.0375f // 3.75%
#define RDS4_VOLUME 0.035f // 3.5%
#define SCA_VOLUME 0.1f // 10%, needs to be high because this is analog
#define MPX_VOLUME 1.0f
@@ -92,7 +95,7 @@ static void stop(int signum) {
}
void show_version() {
printf("fm95 (an FM Processor by radio95) version 1.6\n");
printf("fm95 (an FM Processor by radio95) version 1.7\n");
}
void show_help(char *name) {
printf(
@@ -102,14 +105,15 @@ void show_help(char *name) {
"\t-o,--output\tOverride output device [default: %s]\n"
"\t-M,--mpx\tOverride MPX input device [default: %s]\n"
"\t-r,--rds\tOverride RDS95 input device [default: %s]\n"
"\t-R,--rds2\tOverride the RDS2 additional stream device [default: %s]\n"
"\t-S,--sca\tOverride the SCA input device [default: %s]\n"
"\t-d,--darc\tOverride the DARC input device [default: %s]\n"
"\t-f,--sca_freq\tOverride the SCA frequency [default: %.1f]\n"
"\t-F,--sca_dev\tOverride the SCA deviation [default: %.2f]\n"
"\t-C,--sca_clip\tOverride the SCA clipper threshold [default: %.2f]\n"
"\t-c,--clipper\tOverride the clipper threshold [default: %.2f]\n"
"\t-O,--polar\tForce Polar Stereo (does not take effect with -m%s)\n"
"\t-R,--preemp\tOverride preemphasis [default: %.2f µs]\n"
"\t-O,--polar\tForce Polar Stereo (does not take effect with -s0%s)\n"
"\t-e,--preemp\tOverride preemphasis [default: %.2f µs]\n"
"\t-V,--calibrate\tEnable Calibration mode [default: off, option 2 enables a 60 hz square wave instead of the 400 hz sine wave]\n"
"\t-p,--power\tSet the MPX power [default: %.1f]\n"
"\t-P,--mpx_dev\tSet the MPX deviation [default: %.1f]\n"
@@ -120,26 +124,11 @@ void show_help(char *name) {
,DEFAULT_STEREO
,INPUT_DEVICE
,OUTPUT_DEVICE
#ifdef MPX_DEVICE
,MPX_DEVICE
#else
,"not set"
#endif
#ifdef RDS_DEVICE
,RDS_DEVICE
#else
,"not set"
#endif
#ifdef SCA_DEVICE
,RDS2_DEVICE
,SCA_DEVICE
#else
,"not set"
#endif
#ifdef DARC_DEVICE
,DARC_DEVICE
#else
,"not set"
#endif
,DEFAULT_SCA_FREQUENCY
,DEFAULT_SCA_DEVIATION
,DEFAULT_SCA_CLIPPER_THRESHOLD
@@ -157,7 +146,7 @@ void show_help(char *name) {
int main(int argc, char **argv) {
show_version();
PulseInputDevice mpx_device, rds_device, sca_device, darc_device;
PulseInputDevice mpx_device, rds_device, rds2_device, sca_device, darc_device;
PulseInputDevice input_device;
PulseOutputDevice output_device;
@@ -174,6 +163,7 @@ int main(int argc, char **argv) {
char audio_output_device[64] = OUTPUT_DEVICE;
char audio_mpx_device[64] = MPX_DEVICE;
char audio_rds_device[64] = RDS_DEVICE;
char audio_rds2_device[64] = RDS2_DEVICE;
char audio_sca_device[64] = SCA_DEVICE;
char audio_darc_device[64] = DARC_DEVICE;
float preemphasis_tau = DEFAULT_PREEMPHASIS_TAU;
@@ -188,7 +178,7 @@ int main(int argc, char **argv) {
// #region Parse Arguments
int opt;
const char *short_opt = "s::i:o:M:r:S:d:f:F:C:c:O::R:V::p:P:A:v:D:h";
const char *short_opt = "s::i:o:M:r:R:S:d:f:F:C:c:O::e:V::p:P:A:v:D:h";
struct option long_opt[] =
{
{"stereo", optional_argument, NULL, 's'},
@@ -196,6 +186,7 @@ int main(int argc, char **argv) {
{"output", required_argument, NULL, 'o'},
{"mpx", required_argument, NULL, 'M'},
{"rds", required_argument, NULL, 'r'},
{"rds2", required_argument, NULL, 'R'},
{"sca", required_argument, NULL, 'S'},
{"darc", required_argument, NULL, 'd'},
{"sca_freq", required_argument, NULL, 'f'},
@@ -203,7 +194,7 @@ int main(int argc, char **argv) {
{"sca_clip", required_argument, NULL, 'C'},
{"clipper", required_argument, NULL, 'c'},
{"polar", optional_argument, NULL, 'O'},
{"preemp", required_argument, NULL, 'R'},
{"preemp", required_argument, NULL, 'e'},
{"calibrate", optional_argument, NULL, 'V'},
{"power", required_argument, NULL, 'p'},
{"mpx_dev", required_argument, NULL, 'P'},
@@ -233,6 +224,9 @@ int main(int argc, char **argv) {
case 'r': // RDS in
memcpy(audio_rds_device, optarg, 63);
break;
case 'R': // RDS2 in
memcpy(audio_rds2_device, optarg, 63);
break;
case 'S': //SCA in
memcpy(audio_sca_device, optarg, 63);
break;
@@ -255,7 +249,7 @@ int main(int argc, char **argv) {
if(optarg) polar_stereo = atoi(optarg);
else polar_stereo = 1;
break;
case 'R': // Preemp
case 'e': // Preemp
preemphasis_tau = strtof(optarg, NULL)*1.0e-6f;
break;
case 'V': // Calibration
@@ -286,6 +280,7 @@ int main(int argc, char **argv) {
int mpx_on = (strlen(audio_mpx_device) != 0);
int rds_on = (strlen(audio_rds_device) != 0);
int rds2_on = (strlen(audio_rds2_device) != 0);
int sca_on = (strlen(audio_sca_device) != 0);
int darc_on = (strlen(audio_darc_device) != 0);
@@ -332,6 +327,18 @@ int main(int argc, char **argv) {
return 1;
}
}
if(rds2_on) {
printf("Connecting to RDS2 device... (%s)\n", audio_rds2_device);
opentime_pulse_error = init_PulseInputDevice(&rds2_device, sample_rate, 1, "fm95", "RDS2 Input", audio_rds2_device, &input_buffer_atr);
if (opentime_pulse_error) {
fprintf(stderr, "Error: cannot open RDS2 device: %s\n", pa_strerror(opentime_pulse_error));
free_PulseInputDevice(&input_device);
if(mpx_on) free_PulseInputDevice(&mpx_device);
if(rds_on) free_PulseInputDevice(&rds_device);
return 1;
}
}
if(sca_on) {
printf("Connecting to SCA device... (%s)\n", audio_sca_device);
@@ -342,6 +349,7 @@ int main(int argc, char **argv) {
free_PulseInputDevice(&input_device);
if(mpx_on) free_PulseInputDevice(&mpx_device);
if(rds_on) free_PulseInputDevice(&rds_device);
if(rds2_on) free_PulseInputDevice(&rds2_device);
return 1;
}
}
@@ -355,6 +363,7 @@ int main(int argc, char **argv) {
free_PulseInputDevice(&input_device);
if(mpx_on) free_PulseInputDevice(&mpx_device);
if(rds_on) free_PulseInputDevice(&rds_device);
if(rds2_on) free_PulseInputDevice(&rds2_device);
if(sca_on) free_PulseInputDevice(&sca_device);
return 1;
}
@@ -368,6 +377,7 @@ int main(int argc, char **argv) {
free_PulseInputDevice(&input_device);
if(mpx_on) free_PulseInputDevice(&mpx_device);
if(rds_on) free_PulseInputDevice(&rds_device);
if(rds2_on) free_PulseInputDevice(&rds2_device);
if(sca_on) free_PulseInputDevice(&sca_device);
if(darc_on) free_PulseInputDevice(&darc_device);
return 1;
@@ -399,6 +409,7 @@ int main(int argc, char **argv) {
free_PulseInputDevice(&input_device);
if(mpx_on) free_PulseInputDevice(&mpx_device);
if(rds_on) free_PulseInputDevice(&rds_device);
if(rds2_on) free_PulseInputDevice(&rds2_device);
if(sca_on) free_PulseInputDevice(&sca_device);
if(darc_on) free_PulseInputDevice(&darc_device);
free_PulseOutputDevice(&output_device);
@@ -415,7 +426,7 @@ int main(int argc, char **argv) {
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);
iirfilt_rrrf mpx_lpf = iirfilt_rrrf_create_prototype(LIQUID_IIRDES_BUTTER, LIQUID_IIRDES_LOWPASS, LIQUID_IIRDES_SOS, 2, (polar_stereo ? (46250.0f/sample_rate) : (53000.0f/sample_rate)), 0.0f, 1.0f, 1.0f);
iirfilt_rrrf 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);
@@ -432,8 +443,8 @@ int main(int argc, char **argv) {
float bs412_audio_gain = 1.0f;
AGC agc;
// fs target min max attack relese
initAGC(&agc, sample_rate, 0.625f, 0.0f, 1.25f, 0.025f, 0.25f);
// fs target min max attack release
initAGC(&agc, sample_rate, 0.7071f, 0.0f, 1.5f, 0.05f, 0.25f);
signal(SIGINT, stop);
signal(SIGTERM, stop);
@@ -446,6 +457,10 @@ int main(int argc, char **argv) {
float rds1_in[BUFFER_SIZE] = {0};
float rds2_in[BUFFER_SIZE] = {0};
float rds3_rds4_in[BUFFER_SIZE*2] = {0};
float rds3_in[BUFFER_SIZE] = {0};
float rds4_in[BUFFER_SIZE] = {0};
float darc_data[BUFFER_SIZE*2] = {0}; // DARC data and clock
float darc_clock[BUFFER_SIZE] = {0};
float darc_data_out[BUFFER_SIZE] = {0};
@@ -480,6 +495,14 @@ int main(int argc, char **argv) {
}
uninterleave(rds1_rds2_in, rds1_in, rds2_in, BUFFER_SIZE*2);
}
if(rds2_on) {
if((pulse_error = read_PulseInputDevice(&rds2_device, rds3_rds4_in, sizeof(rds3_rds4_in)))) {
fprintf(stderr, "Error reading from RDS2 device: %s\n", pa_strerror(pulse_error));
to_run = 0;
break;
}
uninterleave(rds3_rds4_in, rds3_in, rds4_in, BUFFER_SIZE*2);
}
if(sca_on) {
if((pulse_error = read_PulseInputDevice(&sca_device, sca_in, sizeof(sca_in)))) {
fprintf(stderr, "Error reading from SCA device: %s\n", pa_strerror(pulse_error));
@@ -500,20 +523,13 @@ int main(int argc, char **argv) {
float mpx = 0.0f;
float audio = 0.0f;
float l_in = left[i];
float r_in = right[i];
float current_mpx_in = mpx_in[i];
float current_rds_in = rds1_in[i];
float current_rds2_in = rds2_in[i];
float current_sca_in = sca_in[i];
if(darc_clock[i] != last_darc_clock) {
last_darc_clock = darc_clock[i];
last_darc_data = darc_data_out[i];
}
float ready_l = apply_preemphasis(&preemp_l, l_in);
float ready_r = apply_preemphasis(&preemp_r, r_in);
float ready_l = apply_preemphasis(&preemp_l, left[i]);
float ready_r = apply_preemphasis(&preemp_r, right[i]);
iirfilt_rrrf_execute(lpf_l, ready_l, &ready_l);
iirfilt_rrrf_execute(lpf_r, ready_r, &ready_r);
@@ -522,29 +538,35 @@ 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 mono = (ready_l + ready_r) / 2.0f;
audio = mono*MONO_VOLUME;
float mid = (ready_l + ready_r) / 2.0f;
audio = mid*MONO_VOLUME;
if(stereo) {
float stereo_signal = (ready_l - ready_r) / 2.0f;
float side = (ready_l - ready_r) / 2.0f;
float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, polar_stereo ? 1 : 8); // 31.25 or 38 KHz
if(polar_stereo) audio += ((stereo_signal+0.2)*stereo_carrier)*STEREO_VOLUME;
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 += (stereo_signal*stereo_carrier)*STEREO_VOLUME;
audio += (side*stereo_carrier)*STEREO_VOLUME;
}
}
if(rds_on && polar_stereo == 0) {
float rds_carrier = get_oscillator_cos_multiplier_ni(&osc, 12); // 57 KHz
mpx += (current_rds_in*rds_carrier)*RDS_VOLUME;
mpx += (rds1_in[i]*rds_carrier)*RDS_VOLUME;
if(!darc_on) { // DARC is hardcoded into 76 khz, according to a screenshot of a fm mpx with darc in it, it takes like 65 to 85 khz
float rds2_carrier_66 = get_oscillator_cos_multiplier_ni(&osc, 14); // 66.5 KHz
mpx += (current_rds2_in*rds2_carrier_66)*RDS2_VOLUME;
mpx += (rds2_in[i]*rds2_carrier_66)*RDS2_VOLUME;
if(rds2_on) {
float rds2_carrier_71 = get_oscillator_cos_multiplier_ni(&osc, 15); // 71.25 KHz
float rds2_carrier_76 = get_oscillator_cos_multiplier_ni(&osc, 16); // 76 KHz
mpx += (rds3_in[i]*rds2_carrier_71)*RDS3_VOLUME;
mpx += (rds4_in[i]*rds2_carrier_76)*RDS4_VOLUME;
}
}
}
if(mpx_on) mpx += hard_clip(current_mpx_in, 1.0f)*MPX_VOLUME;
if(sca_on) mpx += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
if(mpx_on) mpx += hard_clip(mpx_in[i], 1.0f)*MPX_VOLUME;
if(sca_on) mpx += modulate_fm(&sca_mod, hard_clip(sca_in[i], sca_clipper_threshold))*SCA_VOLUME;
if(darc_on && polar_stereo == 0) mpx += hard_clip(refrenced_modulate_fm(&darc_modulator, last_darc_data, 16.0f)*compute_darc_amplitude(stereo*STEREO_VOLUME), 0.1f); // should never be over 10%, the docs say so
float mpx_only = measure_mpx(&mpx_only_power, mpx * mpx_deviation);
@@ -558,11 +580,10 @@ int main(int argc, char **argv) {
audio *= 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
iirfilt_rrrf_execute(mpx_lpf, audio, &audio);
output[i] = (audio+mpx)*master_volume;
output[i] = hard_clip((audio+mpx), 1.0f)*master_volume; // Ensure peak deviation of 75 khz, assuming we're calibrated correctly
if(rds_on || stereo || darc_on) advance_oscillator(&osc);
}

10
src/vban95.c
View File

@@ -20,7 +20,7 @@
#define BUF_SIZE 1500
#define MAX_AUDIO_DATA_SIZE (BUF_SIZE - sizeof(VBANHeader))
#define MAX_BUFFER_PACKETS 128
#define MAX_BUFFER_PACKETS 16
#define POLL_TIMEOUT_MS 75
@@ -251,14 +251,10 @@ int main(int argc, char *argv[]) {
signal(SIGTERM, stop);
while (to_run) {
ssize_t recv_len = recvfrom(sockfd, buffer, BUF_SIZE, 0,
(struct sockaddr *)&sender_addr, &sender_len);
ssize_t recv_len = recvfrom(sockfd, buffer, BUF_SIZE, 0, (struct sockaddr *)&sender_addr, &sender_len);
if (recv_len < 0) {
if (errno == EAGAIN || errno == EWOULDBLOCK) {
// No data available, just continue with the loop
// Add a small sleep to avoid consuming too much CPU
usleep(POLL_TIMEOUT_MS * 1000); // Convert ms to microseconds
usleep(POLL_TIMEOUT_MS * 1000);
continue;
} else {
perror("recvfrom error");