radio95/fm95
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6a5d1994833b6e8ace13e9aa40c6bc0192d987ae
fm95/src/fm95.c

457 lines
16 KiB
C
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#include <stdio.h>
#include <stdlib.h>
#include <getopt.h>
#define buffer_maxlength 12288
#define buffer_tlength_fragsize 12288
#define buffer_prebuf 32
#define DEFAULT_STEREO 1
#define DEFAULT_STEREO_POLAR 0
#define DEFAULT_CLIPPER_THRESHOLD 1.0f
#define DEFAULT_SCA_FREQUENCY 67000.0f
#define DEFAULT_SCA_DEVIATION 7000.0f
#define DEFAULT_SCA_CLIPPER_THRESHOLD 1.0f // Full deviation, if you set this to 0.5 then you may as well set the deviation to 3.5k
#define DEFAULT_PREEMPHASIS_TAU 50e-6 // Europe, the freedomers use 75µs
//#define USB
#include "../lib/constants.h"
#include "../lib/oscillator.h"
#include "../lib/filters.h"
#include "../lib/fm_modulator.h"
#define SAMPLE_RATE 192000
#define INPUT_DEVICE "FM_Audio.monitor"
#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
#define MPX_DEVICE "FM_MPX.monitor"
// #define SCA_DEVICE ""
#define BUFFER_SIZE 2048
#include <pulse/simple.h>
#include <pulse/error.h>
#define MASTER_VOLUME 1.0f // Volume of everything combined, for calibration
#define AUDIO_VOLUME 1.0f // Audio volume, before clipper
#define MONO_VOLUME 0.45f // L+R Signal
#define PILOT_VOLUME 0.09f // 19 KHz Pilot
#define STEREO_VOLUME 0.45f // L-R signal, should be same as MONO
#define SCA_VOLUME 0.1f // FM SCA signal, 10%
#define MPX_VOLUME 1.0f // Passtrough
#define MPX_CLIPPER_THRESHOLD 1.0f
#define LPF_CUTOFF 15000 // Should't need to be changed
volatile sig_atomic_t to_run = 1;
void uninterleave(const float *input, float *left, float *right, size_t num_samples) {
// For stereo, usually it is like this: LEFT RIGHT LEFT RIGHT LEFT RIGHT so this is used to get LEFT LEFT LEFT and RIGHT RIGHT RIGHT
for (size_t i = 0; i < num_samples/2; i++) {
left[i] = input[i * 2];
right[i] = input[i * 2 + 1];
}
}
static void stop(int signum) {
(void)signum;
printf("\nReceived stop signal.\n");
to_run = 0;
}
void show_version() {
printf("fm95 (an FM Processor by radio95) version 1.3\n");
}
void show_help(char *name) {
printf(
"Usage: %s\n"
" -m,--mono Force Mono [default: %d]\n"
" -s,--stereo Force Stereo [default: %d]\n"
" -i,--input Override input device [default: %s]\n"
" -o,--output Override output device [default: %s]\n"
" -M,--mpx Override MPX input device [default: %s]\n"
" -C,--sca Override the SCA input device [default: %s]\n"
" -f,--sca_freq Override the SCA frequency [default: %.1f]\n"
" -F,--sca_dev Override the SCA deviation [default: %.2f]\n"
" -L,--sca_clip Override the SCA clipper threshold [default: %.2f]\n"
" -c,--clipper Override the clipper threshold [default: %.2f]\n"
" -P,--polar Force Polar Stereo (does not take effect with -m%s)\n"
" -g,--ge Force Zenith/GE stereo (does not take effect with -m%s)\n"
" -R,--preemp Override preemphasis [default: %.2f µs]\n"
" -V,--calibrate Enable Calibration mode [default: off]\n"
" -A,--master_vol Set master volume [default: %.3f]\n"
" -v,--audio_vol Set audio volume [default: %.3f]\n"
,name
,DEFAULT_STEREO^1
,DEFAULT_STEREO
,INPUT_DEVICE
,OUTPUT_DEVICE
#ifdef MPX_DEVICE
,MPX_DEVICE
#else
,"not set"
#endif
#ifdef SCA_DEVICE
,SCA_DEVICE
#else
,"not set"
#endif
,DEFAULT_SCA_FREQUENCY
,DEFAULT_SCA_DEVIATION
,DEFAULT_SCA_CLIPPER_THRESHOLD
,DEFAULT_CLIPPER_THRESHOLD
,(DEFAULT_STEREO_POLAR == 1) ? ", default" : ""
,(DEFAULT_STEREO_POLAR == 1) ? "" : ", default"
,DEFAULT_PREEMPHASIS_TAU/0.000001
,MASTER_VOLUME
,AUDIO_VOLUME
);
}
int main(int argc, char **argv) {
show_version();
pa_simple *mpx_device;
pa_simple *sca_device;
pa_simple *output_device;
float clipper_threshold = DEFAULT_CLIPPER_THRESHOLD;
int stereo = DEFAULT_STEREO;
int polar_stereo = DEFAULT_STEREO_POLAR;
float sca_frequency = DEFAULT_SCA_FREQUENCY;
float sca_deviation = DEFAULT_SCA_DEVIATION;
float sca_clipper_threshold = DEFAULT_SCA_CLIPPER_THRESHOLD;
char audio_input_device[64] = INPUT_DEVICE;
char audio_output_device[64] = OUTPUT_DEVICE;
#ifndef MPX_DEVICE
char audio_mpx_device[64] = "\0";
#else
char audio_mpx_device[64] = MPX_DEVICE;
#endif
#ifndef SCA_DEVICE
char audio_sca_device[64] = "\0";
#else
char audio_sca_device[64] = SCA_DEVICE;
#endif
float preemphasis_tau = DEFAULT_PREEMPHASIS_TAU;
int calibration_mode = 0;
float master_volume = MASTER_VOLUME;
float audio_volume = AUDIO_VOLUME;
// #region Parse Arguments
int opt;
const char *short_opt = "msi:o:apM:C:f:F:L:c:l:PgSDR:VA:v:h";
struct option long_opt[] =
{
{"mono", no_argument, NULL, 'm'},
{"stereo", no_argument, NULL, 's'},
{"input", required_argument, NULL, 'i'},
{"output", required_argument, NULL, 'o'},
{"mpx", required_argument, NULL, 'M'},
{"sca", required_argument, NULL, 'C'},
{"sca_freq", required_argument, NULL, 'f'},
{"sca_dev", required_argument, NULL, 'F'},
{"sca_clip", required_argument, NULL, 'L'},
{"clipper", required_argument, NULL, 'c'},
{"polar", no_argument, NULL, 'P'},
{"ge", no_argument, NULL, 'g'},
{"preemp", required_argument, NULL, 'R'},
{"calibrate", no_argument, NULL, 'V'},
{"master_vol", required_argument, NULL, 'A'},
{"audio_vol", required_argument, NULL, 'v'},
{"help", no_argument, NULL, 'h'},
{0, 0, 0, 0}
};
while((opt = getopt_long(argc, argv, short_opt, long_opt, NULL)) != -1) {
switch(opt) {
case 'm': // Mono
stereo = 0;
break;
case 's': // Stereo
stereo = 1;
break;
case 'i': // Input Device
memcpy(audio_input_device, optarg, 63);
break;
case 'o': // Output Device
memcpy(audio_output_device, optarg, 63);
break;;
case 'M': //MPX in
memcpy(audio_mpx_device, optarg, 63);
break;
case 'C': //SCA in
memcpy(audio_sca_device, optarg, 63);
break;
case 'f': //SCA freq
sca_frequency = strtof(optarg, NULL);
break;
case 'F': //SCA deviation
sca_deviation = strtof(optarg, NULL);
break;
case 'L': //SCA clip
sca_clipper_threshold = strtof(optarg, NULL);
break;
case 'c': //Clipper
clipper_threshold = strtof(optarg, NULL);
break;
case 'P': //Polar
polar_stereo = 1;
break;
case 'g': //GE
polar_stereo = 0;
break;
case 'R': // Preemp
preemphasis_tau = strtof(optarg, NULL)*0.000001;
break;
case 'V': // Calibration
calibration_mode = 1;
break;
case 'A': // Master vol
master_volume = strtof(optarg, NULL);
break;
case 'v': // Audio Volume
audio_volume = strtof(optarg, NULL);
break;
case 'h':
show_help(argv[0]);
return 1;
}
}
// #endregion
// #region Setup devices
// Define formats and buffer atributes
pa_sample_spec stereo_format = {
.format = PA_SAMPLE_FLOAT32NE, //Float32 NE, or Float32 Native Endian, the float in c uses the endianess of your pc, or native endian, and float is float32, and double is float64
.channels = 2,
.rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
};
pa_sample_spec mono_format = {
.format = PA_SAMPLE_FLOAT32NE,
.channels = 1,
.rate = SAMPLE_RATE
};
pa_buffer_attr input_buffer_atr = {
.maxlength = buffer_maxlength,
.fragsize = buffer_tlength_fragsize
};
pa_buffer_attr output_buffer_atr = {
.maxlength = buffer_maxlength,
.tlength = buffer_tlength_fragsize,
.prebuf = buffer_prebuf
};
int opentime_pulse_error;
printf("Connecting to input device... (%s)\n", audio_input_device);
pa_simple *input_device = pa_simple_new(
NULL,
"fm95",
PA_STREAM_RECORD,
audio_input_device,
"Main Audio Input",
&stereo_format,
NULL,
&input_buffer_atr,
&opentime_pulse_error
);
if (!input_device) {
fprintf(stderr, "Error: cannot open input device: %s\n", pa_strerror(opentime_pulse_error));
return 1;
}
if(strlen(audio_mpx_device) != 0) {
printf("Connecting to MPX device... (%s)\n", audio_mpx_device);
mpx_device = pa_simple_new(
NULL,
"fm95",
PA_STREAM_RECORD,
audio_mpx_device,
"MPX Input",
&mono_format,
NULL,
&input_buffer_atr,
&opentime_pulse_error
);
if (!mpx_device) {
fprintf(stderr, "Error: cannot open MPX device: %s\n", pa_strerror(opentime_pulse_error));
pa_simple_free(input_device);
return 1;
}
}
if(strlen(audio_sca_device) != 0) {
printf("Connecting to SCA device... (%s)\n", audio_sca_device);
sca_device = pa_simple_new(
NULL,
"fm95",
PA_STREAM_RECORD,
audio_sca_device,
"SCA Input",
&mono_format,
NULL,
&input_buffer_atr,
&opentime_pulse_error
);
if (!sca_device) {
fprintf(stderr, "Error: cannot open SCA device: %s\n", pa_strerror(opentime_pulse_error));
pa_simple_free(input_device);
if(strlen(audio_mpx_device) != 0) pa_simple_free(mpx_device);
return 1;
}
}
printf("Connecting to output device... (%s)\n", audio_output_device);
output_device = pa_simple_new(
NULL,
"StereoEncoder",
PA_STREAM_PLAYBACK,
audio_output_device,
"MPX Output",
&mono_format,
NULL,
&output_buffer_atr,
&opentime_pulse_error
);
if (!output_device) {
fprintf(stderr, "Error: cannot open output device: %s\n", pa_strerror(opentime_pulse_error));
pa_simple_free(input_device);
if(strlen(audio_mpx_device) != 0) pa_simple_free(mpx_device);
if(strlen(audio_sca_device) != 0) pa_simple_free(sca_device);
return 1;
}
// #endregion
if(calibration_mode) {
Oscillator osc;
init_oscillator(&osc, 400, SAMPLE_RATE);
signal(SIGINT, stop);
signal(SIGTERM, stop);
int pulse_error;
float output[BUFFER_SIZE]; // MPX, this goes to the output
while(to_run) {
for (int i = 0; i < BUFFER_SIZE; i++) {
output[i] = get_oscillator_sin_sample(&osc)*master_volume;
}
if (pa_simple_write(output_device, output, sizeof(output), &pulse_error) < 0) {
fprintf(stderr, "Error writing to output device: %s\n", pa_strerror(pulse_error));
to_run = 0;
break;
}
}
printf("Cleaning up...\n");
pa_simple_free(input_device);
if(strlen(audio_mpx_device) != 0) pa_simple_free(mpx_device);
if(strlen(audio_sca_device) != 0) pa_simple_free(sca_device);
pa_simple_free(output_device);
return 0;
}
// #region Setup Filters/Modulaltors/Oscillators
Oscillator osc;
init_oscillator(&osc, polar_stereo ? 31250.0 : 19000, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
FMModulator sca_mod;
init_fm_modulator(&sca_mod, sca_frequency, sca_deviation, SAMPLE_RATE);
ResistorCapacitor preemp_l, preemp_r;
init_preemphasis(&preemp_l, preemphasis_tau, SAMPLE_RATE);
init_preemphasis(&preemp_r, preemphasis_tau, SAMPLE_RATE);
// Use https://www.earlevel.com/main/2021/09/02/biquad-calculator-v3/
BiquadFilter lpf_l, lpf_r;
init_lpf(&lpf_l, LPF_CUTOFF, 0.17675f, SAMPLE_RATE);
init_lpf(&lpf_r, LPF_CUTOFF, 0.17675f, SAMPLE_RATE);
// #endregion
signal(SIGINT, stop);
signal(SIGTERM, stop);
int pulse_error;
float audio_stereo_input[BUFFER_SIZE*2]; // Input from device, interleaved stereo
float mpx_in[BUFFER_SIZE] = {0}; // Input from MPX device
float sca_in[BUFFER_SIZE] = {0}; // Input from SCA device
float left[BUFFER_SIZE+64], right[BUFFER_SIZE+64]; // Audio, same thing as in input but uninterleaved, ai told be there could be a buffer overflow here
float output[BUFFER_SIZE]; // MPX, this goes to the output
while (to_run) {
if (pa_simple_read(input_device, audio_stereo_input, sizeof(audio_stereo_input), &pulse_error) < 0) {
fprintf(stderr, "Error reading from input device: %s\n", pa_strerror(pulse_error));
to_run = 0;
break;
}
uninterleave(audio_stereo_input, left, right, BUFFER_SIZE*2);
if(strlen(audio_mpx_device) != 0) {
if (pa_simple_read(mpx_device, mpx_in, sizeof(mpx_in), &pulse_error) < 0) {
fprintf(stderr, "Error reading from MPX device: %s\n", pa_strerror(pulse_error));
to_run = 0;
break;
}
}
if(strlen(audio_sca_device) != 0) {
if (pa_simple_read(sca_device, sca_in, sizeof(sca_in), &pulse_error) < 0) {
fprintf(stderr, "Error reading from SCA device: %s\n", pa_strerror(pulse_error));
to_run = 0;
break;
}
}
for (int i = 0; i < BUFFER_SIZE; i++) {
float l_in = left[i];
float r_in = right[i];
float current_mpx_in = mpx_in[i];
float current_sca_in = sca_in[i];
float ready_l = apply_biquad(&lpf_l, l_in);
float ready_r = apply_biquad(&lpf_r, r_in);
ready_l = apply_preemphasis(&preemp_l, ready_l)*2;
ready_r = apply_preemphasis(&preemp_r, ready_r)*2;
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; // Stereo to Mono
output[i] = mono*MONO_VOLUME;
if(stereo == 1) {
float stereo = (ready_l - ready_r) / 2.0f; // Also Stereo to Mono but a bit diffrent
float stereo_carrier = get_oscillator_sin_multiplier_ni(&osc, polar_stereo ? 1 : 2);
if(polar_stereo) {
output[i] += ((stereo+0.2)*stereo_carrier)*STEREO_VOLUME;
} else {
float pilot = get_oscillator_sin_multiplier_ni(&osc, 1);
output[i] += pilot*PILOT_VOLUME +
(stereo*stereo_carrier)*STEREO_VOLUME;
}
advance_oscillator(&osc);
}
if(strlen(audio_mpx_device) != 0) output[i] += hard_clip(current_mpx_in, MPX_CLIPPER_THRESHOLD)*MPX_VOLUME;
if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
output[i] *= master_volume;
}
if (pa_simple_write(output_device, output, sizeof(output), &pulse_error) < 0) {
fprintf(stderr, "Error writing to output device: %s\n", pa_strerror(pulse_error));
to_run = 0;
break;
}
}
printf("Cleaning up...\n");
pa_simple_free(input_device);
if(strlen(audio_mpx_device) != 0) pa_simple_free(mpx_device);
if(strlen(audio_sca_device) != 0) pa_simple_free(sca_device);
pa_simple_free(output_device);
return 0;
}
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