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mirror of https://github.com/radio95-rnt/fm95.git synced 2026-02-27 03:23:54 +01:00

modify read me

This commit is contained in:
2025-01-25 23:49:31 +01:00
parent a1feb6c0a9
commit d4c7334a96
3 changed files with 5 additions and 3 deletions

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src/fm95.c Normal file
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#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <stdint.h>
#include <signal.h>
#include <string.h>
#include <getopt.h>
#include "options.h"
#define DEFAULT_STEREO 1
#define DEFAULT_STEREO_POLAR 0
#define DEFAULT_STEREO_SSB 0
#define DEFAULT_CLIPPER_THRESHOLD 1.0f
#define DEFAULT_ALSA_OUTPUT 0
//#define USB
#include "../lib/constants.h"
#include "../lib/oscillator.h"
#include "../lib/filters.h"
#include "../lib/hilbert.h"
#define SAMPLE_RATE 192000 // Don't go lower than 108 KHz, becuase it (53000*2) and (38000+15000)
#define INPUT_DEVICE "real_real_tx_audio_input.monitor"
#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
// #define MPX_DEVICE ""
#define BUFFER_SIZE 512
#include <pulse/simple.h>
#include <pulse/error.h>
#include <alsa/asoundlib.h>
#define MONO_VOLUME 0.45f // L+R Signal
#define PILOT_VOLUME 0.09f // 19 KHz Pilot
#define STEREO_VOLUME 0.45f // L-R signal
#define MPX_VOLUME 1.0f
#ifdef PREEMPHASIS
#define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america
#endif
volatile sig_atomic_t to_run = 1;
float hard_clip(float sample, float threshold) {
if (sample > threshold) {
return threshold; // Clip to the upper threshold
} else if (sample < -threshold) {
return -threshold; // Clip to the lower threshold
} else {
return sample; // No clipping
}
}
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. Cleaning up...\n");
to_run = 0;
}
void show_version() {
printf("fm95 (an FM Processor by radio95) version 1.0\n");
}
void show_help(char *name) {
printf(
"fm95 (an FM Processor by radio95)\n"
"Usage: %s\n\n"
" -m,--mono Force Mono\n"
" -s,--stereo Force Stereo\n"
" -i,--input Override input device\n"
" -o,--output Override output device\n"
" -M,--mpx Override MPX input device\n"
" -c,--clipper Override the clipper threshold\n"
" -P,--polar Force Polar Stereo (does not take effect with -m)\n"
" -g,--ge Force Zenith/GE stereo (does not take effect with -m, default)\n"
" -S,--ssb Force SSB\n"
" -D,--dsb Force DSB\n"
,name
);
}
int main(int argc, char **argv) {
show_version();
int stereo = DEFAULT_STEREO;
#ifndef MPX_DEVICE
char audio_mpx_device[64] = "\0";
#else
char audio_mpx_device[64] = MPX_DEVICE;
#endif
pa_simple *mpx_device;
pa_simple *output_device;
snd_pcm_hw_params_t *output_params;
snd_pcm_t *output_handle;
float clipper_threshold = DEFAULT_CLIPPER_THRESHOLD;
int polar_stereo = DEFAULT_STEREO_POLAR;
int ssb = DEFAULT_STEREO_SSB;
char audio_input_device[64] = INPUT_DEVICE;
char audio_output_device[64] = OUTPUT_DEVICE;
int alsa_output = DEFAULT_ALSA_OUTPUT;
int opt;
const char *short_opt = "msi:o:apM:c:PgSDhv";
struct option long_opt[] =
{
{"mono", no_argument, NULL, 'm'},
{"stereo", no_argument, NULL, 's'},
{"input", optional_argument, NULL, 'i'},
{"output", optional_argument, NULL, 'o'},
{"alsa_out", no_argument, NULL, 'a'},
{"pulse_put", no_argument, NULL, 'p'},
{"mpx", optional_argument, NULL, 'M'},
{"clipper", optional_argument, NULL, 'c'},
{"polar", no_argument, NULL, 'P'},
{"ge", no_argument, NULL, 'g'},
{"ssb", no_argument, NULL, 'S'},
{"dsb", no_argument, NULL, 'D'},
{"help", no_argument, NULL, 'h'},
{"version", no_argument, NULL, 'v'},
{ 0, 0, 0, 0 }
};
while((opt = getopt_long(argc, argv, short_opt, long_opt, NULL)) != -1) {
switch(opt) {
case 'm': // Mono
stereo = 0;
printf("Running in Mono\n");
break;
case 's': // Stereo
stereo = 1;
printf("Running in Stereo\n");
break;
case 'i': // Input Device
memcpy(audio_input_device, optarg, 63);
break;
case 'o': // Output Device
memcpy(audio_output_device, optarg, 63);
break;
case 'a': // Alsa output
alsa_output = 1;
printf("Outputting via alsa\n");
break;
case 'p': // Pulse output
alsa_output = 0;
printf("Outputting via pulse\n");
break;
case 'M': //MPX in
memcpy(audio_mpx_device, optarg, 63);
break;
case 'c': //Clipper
clipper_threshold = strtof(optarg, NULL);
printf("Running with a clipper threshold of %f\n", clipper_threshold);
break;
case 'P': //Polar
polar_stereo = 1;
printf("Using polar stereo\n");
break;
case 'g': //GE
polar_stereo = 0;
printf("Using Zenith/GE stereo\n");
break;
case 'S': //SSB
ssb = 1;
printf("Using SSB\n");
break;
case 'D': //DSB
ssb = 0;
printf("Using DSB\n");
break;
case 'v': // Version
show_version();
return 0;
case 'h':
show_help(argv[0]);
return 1;
}
}
// 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,
"StereoEncoder",
PA_STREAM_RECORD,
audio_input_device,
"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,
"StereoEncoder",
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));
return 1;
}
}
printf("Connecting to output device... (%s)\n", audio_output_device);
if(alsa_output == 0) {
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);
return 1;
}
} else {
int output_error = snd_pcm_open(&output_handle, audio_output_device, SND_PCM_STREAM_PLAYBACK, 0);
if(output_error < 0) {
fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error));
pa_simple_free(input_device);
return 1;
}
snd_pcm_hw_params_malloc(&output_params);
snd_pcm_hw_params_any(output_handle, output_params);
snd_pcm_hw_params_set_access(output_handle, output_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(output_handle, output_params, SND_PCM_FORMAT_FLOAT); // Same as pulse's Float32NE
snd_pcm_hw_params_set_channels(output_handle, output_params, 1);
unsigned int rate = SAMPLE_RATE;
int dir;
snd_pcm_hw_params_set_rate_near(output_handle, output_params, &rate, &dir);
snd_pcm_uframes_t frames = BUFFER_SIZE;
snd_pcm_hw_params_set_period_size_near(output_handle, output_params, &frames, &dir); // i don't have a clue why like this
output_error = snd_pcm_hw_params(output_handle, output_params);
if(output_error < 0) {
fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error));
snd_pcm_close(output_handle);
pa_simple_free(input_device);
snd_pcm_hw_params_free(output_params);
return 1;
}
}
Oscillator pilot_osc;
if(polar_stereo == 1) {
init_oscillator(&pilot_osc, 31250.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
} else {
init_oscillator(&pilot_osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
}
HilbertTransformer hilbert; // An Hilbert shifts a signal in quadrature, generating the I/Q data
init_hilbert(&hilbert);
DelayLine monoDelay; // Hilbert introduces a delay of 99 samples, this should be here to sync the mono with stereo to a sample
init_delay_line(&monoDelay, 99);
#ifdef PREEMPHASIS
ResistorCapacitor preemp_l, preemp_r;
init_rc_tau(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
init_rc_tau(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
#endif
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]; // Input from MPX 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;
}
}
for (int i = 0; i < BUFFER_SIZE; i++) {
float l_in = left[i];
float r_in = right[i];
float multiplex_in = mpx_in[i];
#ifdef PREEMPHASIS
float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in)*2;
float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in)*2;
float current_left_input = hard_clip(preemphasized_left, clipper_threshold);
float current_right_input = hard_clip(preemphasized_right, clipper_threshold);
#else
float current_left_input = hard_clip(l_in, clipper_threshold);
float current_right_input = hard_clip(r_in, clipper_threshold);
#endif
float mono = (current_left_input + current_right_input) / 2.0f; // Stereo to Mono
if(stereo == 1) {
float stereo = (current_left_input - current_right_input) / 2.0f; // Also Stereo to Mono but a bit diffrent
if(polar_stereo == 1) {
if(ssb) {
float stereo_carrier = get_oscillator_sin_multiplier_ni(&pilot_osc, 1); // Get stereo carrier via multiplication
float stereo_carrier_cos = get_oscillator_cos_sample(&pilot_osc); // Get Carrier Q of I/Q
float stereo_i, stereo_q;
stereo += 0.2;
apply_hilbert(&hilbert, stereo, &stereo_i, &stereo_q); // Compute I/Q
#ifdef USB
float signal = (stereo_i*stereo_carrier_cos+stereo_q*(stereo_carrier*0.775f));
#else
float signal = (stereo_i*stereo_carrier_cos-stereo_q*(stereo_carrier*0.775f));
#endif
output[i] = delay_line(&monoDelay, mono)*MONO_VOLUME +
signal*STEREO_VOLUME;
if(strlen(audio_mpx_device) != 0) output[i] += multiplex_in*MPX_VOLUME;
} else {
float stereo_carrier = get_oscillator_sin_sample(&pilot_osc);
output[i] = mono*MONO_VOLUME +
((stereo+0.2)*stereo_carrier)*STEREO_VOLUME;
if(strlen(audio_mpx_device) != 0) output[i] += multiplex_in*MPX_VOLUME;
}
} else {
if(ssb) {
float stereo_carrier = get_oscillator_sin_multiplier_ni(&pilot_osc, 2); // Get stereo carrier via multiplication
float stereo_carrier_cos = get_oscillator_cos_multiplier_ni(&pilot_osc, 2); // Get Carrier Q of I/Q
float pilot = get_oscillator_sin_sample(&pilot_osc);
float stereo_i, stereo_q;
apply_hilbert(&hilbert, stereo, &stereo_i, &stereo_q); // Compute I/Q
#ifdef USB
float signal = (stereo_i*stereo_carrier_cos+stereo_q*(stereo_carrier*0.775f));
#else
float signal = (stereo_i*stereo_carrier_cos-stereo_q*(stereo_carrier*0.775f));
#endif
output[i] = delay_line(&monoDelay, mono)*MONO_VOLUME +
pilot*PILOT_VOLUME +
signal*STEREO_VOLUME;
if(strlen(audio_mpx_device) != 0) output[i] += multiplex_in*MPX_VOLUME;
} else {
float stereo_carrier = get_oscillator_sin_multiplier_ni(&pilot_osc,2);
float pilot = get_oscillator_sin_sample(&pilot_osc);
output[i] = mono*MONO_VOLUME +
pilot*PILOT_VOLUME +
(stereo*stereo_carrier)*STEREO_VOLUME;
if(strlen(audio_mpx_device) != 0) output[i] += multiplex_in*MPX_VOLUME;
}
}
} else {
output[i] = mono*MONO_VOLUME; // Only Mono
}
}
if(alsa_output == 0) {
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;
}
} else {
snd_pcm_writei(output_handle, output, sizeof(output));
}
}
printf("Cleaning up...\n");
pa_simple_free(input_device);
if(strlen(audio_mpx_device) != 0) pa_simple_free(mpx_device);
if(alsa_output == 0) {
pa_simple_free(output_device);
} else {
snd_pcm_drain(output_handle);
snd_pcm_close(output_handle);
snd_pcm_hw_params_free(output_params);
}
exit_hilbert(&hilbert);
exit_delay_line(&monoDelay);
return 0;
}