add sca

.vscode/.server-controller-port.log

@@ -1,5 +1,5 @@
 {
   "port": 13452,
-  "time": 1737845288156,
+  "time": 1737886041050,
   "version": "0.0.3"
 }

README.md

@@ -1,6 +1,3 @@
-# FMTools
-FMTools is a repository of apps you can use to make your FM broadcast better, pirate or not this will help you if you don't have something, maybe you want a better stereo encoder? SCA? We have what you need, for RDS just use MiniRDS
-
 # fm95
 FM95 is a audio processor for FM, it does:
 - Pre-Emphasis
@@ -9,7 +6,7 @@ FM95 is a audio processor for FM, it does:
 - SSB Stereo
 - Polar Stereo
 - Polar SSB Stereo (huh)
-- TODO: SCA
+- SCA
 
 Supports 2 inputs:
 - Audio (via Pulse)
@@ -18,13 +15,6 @@ Supports 2 inputs:
 and one output:
 - MPX (via Pulse or ALSA)
 
-Note that i haven't tested it, but i will on monday (29-01-25)
-
-# SCAMod
-SCAMod is a simple FM modulator which can be used to modulate a secondary audio stream, has similiar cpu usage and latency as STCode
-
-Has a fine quality, but as it goes for 12 khz fm signals
-
 # How to compile?
 To compile you need `cmake`, `libasound2-dev` and `libpulse-dev`, if you have those then do these commands:
 ```

src/fm95.c

@@ -1,18 +1,22 @@
 #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 PREEMPHASIS
+
+#define buffer_maxlength 12288
+#define buffer_tlength_fragsize 8192
+#define buffer_prebuf 16
 
 #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 DEFAULT_SCA_FREQUENCY 67000.0f
+#define DEFAULT_SCA_DEVIATION 7000.0f
+#define DEFAULT_SCA_CLIPPER_THRESHOLD 1.0f
+#define DEFAULT_PREEMPHASIS_TAU 50e-6
 
 //#define USB
 
@@ -20,12 +24,14 @@
 #include "../lib/oscillator.h"
 #include "../lib/filters.h"
 #include "../lib/hilbert.h"
+#include "../lib/fm_modulator.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 SCA_DEVICE ""
 
 #define BUFFER_SIZE 512
 
@@ -36,11 +42,9 @@
 #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 SCA_VOLUME 0.1f
 #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;
 
@@ -71,64 +75,115 @@ static void stop(int signum) {
 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"
+        "   -m,--mono       Force Mono [default: %d]\n"
-        "   -s,--stereo     Force Stereo\n"
+        "   -s,--stereo     Force Stereo [default: %d]\n"
-        "   -i,--input      Override input device\n"
+        "   -i,--input      Override input device [default: %s]\n"
-        "   -o,--output     Override output device\n"
+        "   -o,--output     Override output device [default: %s]\n"
-        "   -M,--mpx        Override MPX input device\n"
+        "   -a,--alsa_out   Force alsa output [default: %d]\n"
-        "   -c,--clipper    Override the clipper threshold\n"
+        "   -p,--pulse_out  Force pulse output [default: %d]\n"
-        "   -P,--polar      Force Polar Stereo (does not take effect with -m)\n"
+        "   -M,--mpx        Override MPX input device [default: %s]\n"
-        "   -g,--ge         Force Zenith/GE stereo (does not take effect with -m, default)\n"
+        "   -C,--sca        Override the SCA input device [default: %s]\n"
-        "   -S,--ssb        Force SSB\n"
+        "   -f,--sca_freq   Override the SCA frequency [default: %f]\n"
-        "   -D,--dsb        Force DSB\n"
+        "   -F,--sca_dev    Override the SCA deviation [default: %f]\n"
+        "   -L,--sca_clip   Override the SCA clipper threshold [default: %f]\n"
+        "   -c,--clipper    Override the clipper threshold [default: %f]\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"
+        "   -S,--ssb        Force SSB [default: %d]\n"
+        "   -D,--dsb        Force DSB [default: %d]\n"
+        "   -R,--preemp     Override preemphasis [default: %f]\n"
         ,name
+        ,DEFAULT_STEREO^1
+        ,DEFAULT_STEREO
+        ,INPUT_DEVICE
+        ,OUTPUT_DEVICE
+        ,DEFAULT_ALSA_OUTPUT
+        ,DEFAULT_ALSA_OUTPUT^1
+        #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_STEREO_SSB
+        ,DEFAULT_STEREO_SSB^1
+        ,DEFAULT_PREEMPHASIS_TAU
     );
 }
 
 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 *sca_device;
     pa_simple *output_device;
     snd_pcm_hw_params_t *output_params;
     snd_pcm_t *output_handle;
+
     float clipper_threshold = DEFAULT_CLIPPER_THRESHOLD;
+    int stereo = DEFAULT_STEREO;
     int polar_stereo = DEFAULT_STEREO_POLAR;
     int ssb = DEFAULT_STEREO_SSB;
+
+    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
     int alsa_output = DEFAULT_ALSA_OUTPUT;
+    float preemphasis_tau = DEFAULT_PREEMPHASIS_TAU;
 
+    // #region Parse Arguments
     int opt;
-    const char	*short_opt = "msi:o:apM:c:PgSDhv";
+    const char	*short_opt = "msi:o:apM:C:f:F:L:c:PgSDR:hv";
     struct option	long_opt[] =
 	{
-		{"mono",		no_argument, NULL, 'm'},
+        {"mono",        no_argument,       NULL, 'm'},
-		{"stereo",		no_argument, NULL, 's'},
+        {"stereo",      no_argument,       NULL, 's'},
-		{"input",		optional_argument, NULL, 'i'},
+        {"input",       required_argument, NULL, 'i'},
-		{"output",		optional_argument, NULL, 'o'},
+        {"output",      required_argument, NULL, 'o'},
-		{"alsa_out",	no_argument, NULL, 'a'},
+        {"alsa_out",    no_argument,       NULL, 'a'},
-		{"pulse_put",	no_argument, NULL, 'p'},
+        {"pulse_out",   no_argument,       NULL, 'p'},
-		{"mpx",	     	optional_argument, NULL, 'M'},
+        {"mpx",         required_argument, NULL, 'M'},
-		{"clipper",		optional_argument, NULL, 'c'},
+        {"sca",         required_argument, NULL, 'C'},
-		{"polar",	no_argument, NULL, 'P'},
+        {"sca_freq",    required_argument, NULL, 'f'},
-		{"ge",	no_argument, NULL, 'g'},
+        {"sca_dev",     required_argument, NULL, 'F'},
-		{"ssb",	no_argument, NULL, 'S'},
+        {"sca_clip",    required_argument, NULL, 'L'},
-		{"dsb",	no_argument, NULL, 'D'},
+        {"clipper",     required_argument, NULL, 'c'},
-
+        {"polar",       no_argument,       NULL, 'P'},
-		{"help",	no_argument, NULL, 'h'},
+        {"ge",          no_argument,       NULL, 'g'},
-		{"version",	no_argument, NULL, 'v'},
+        {"ssb",         no_argument,       NULL, 'S'},
-		{ 0,		0,		0,	0 }
+        {"dsb",         no_argument,       NULL, 'D'},
+        {"preemp",      no_argument,       NULL, 'R'},
+        
+        {"help",        no_argument,       NULL, 'h'},
+        {"version",     no_argument,       NULL, 'v'},
+        {0,             0,                 0,    0}  // No trailing comma here
 	};
 
     while((opt = getopt_long(argc, argv, short_opt, long_opt, NULL)) != -1) {
@@ -158,6 +213,21 @@ int main(int argc, char **argv) {
             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);
+                printf("Running with a SCA frequency of %f\n", sca_frequency);
+                break;
+            case 'F': //SCA deviation
+                sca_deviation = strtof(optarg, NULL);
+                printf("Running with a SCA deviation of %f\n", sca_deviation);
+                break;
+            case 'L': //SCA clip
+                sca_clipper_threshold = strtof(optarg, NULL);
+                printf("Running with a SCA clipper threshold of %f\n", sca_clipper_threshold);
+                break;
             case 'c': //Clipper
                 clipper_threshold = strtof(optarg, NULL);
                 printf("Running with a clipper threshold of %f\n", clipper_threshold);
@@ -178,6 +248,10 @@ int main(int argc, char **argv) {
                 ssb = 0;
                 printf("Using DSB\n");
                 break;
+            case 'R': // Preemp
+                preemphasis_tau = strtof(optarg, NULL)*0.000001;
+                printf("Running with a premp of %f\n", preemphasis_tau);
+                break;
             case 'v': // Version
                 show_version();
                 return 0;
@@ -186,6 +260,9 @@ int main(int argc, char **argv) {
                 return 1;
         }
     }
+    // #endregion
+
+    // #region Setup devices
 
     // Define formats and buffer atributes
     pa_sample_spec stereo_format = {
@@ -215,10 +292,10 @@ int main(int argc, char **argv) {
 
     pa_simple *input_device = pa_simple_new(
         NULL,
-        "StereoEncoder",
+        "fm95",
         PA_STREAM_RECORD,
         audio_input_device,
-        "Audio Input",
+        "Main Audio Input",
         &stereo_format,
         NULL,
         &input_buffer_atr,
@@ -234,7 +311,7 @@ int main(int argc, char **argv) {
 
         mpx_device = pa_simple_new(
             NULL,
-            "StereoEncoder",
+            "fm95",
             PA_STREAM_RECORD,
             audio_mpx_device,
             "MPX Input",
@@ -245,6 +322,28 @@ int main(int argc, char **argv) {
         );
         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;
         }
     }
@@ -266,6 +365,8 @@ int main(int argc, char **argv) {
         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;
         }
     } else {
@@ -273,6 +374,8 @@ int main(int argc, char **argv) {
         if(output_error < 0) {
             fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_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;
         }
         snd_pcm_hw_params_malloc(&output_params);
@@ -294,29 +397,35 @@ int main(int argc, char **argv) {
             return 1;
         }
     }
+    // #endregion
 
     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
+        init_oscillator(&pilot_osc, 31250.0, SAMPLE_RATE); // The stereo carrier itself, the stereo carrier in polar is modulated directly on 31.25 khz with a bit of a 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
     }
+
+    FMModulator sca_mod;
+    init_fm_modulator(&sca_mod, sca_frequency, sca_deviation, SAMPLE_RATE);
+
     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_l, preemphasis_tau, SAMPLE_RATE);
-    init_rc_tau(&preemp_r, 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 sca_in[BUFFER_SIZE]; // 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) {
@@ -333,25 +442,30 @@ int main(int argc, char **argv) {
                 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 multiplex_in = mpx_in[i];
+            float current_mpx_in = mpx_in[i];
+            float current_sca_in = sca_in[i];
 
-#ifdef PREEMPHASIS
+            float ready_l = l_in;
-            float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in)*2;
+            float ready_r = r_in;
-            float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in)*2;
+            ready_l = apply_pre_emphasis(&preemp_l, ready_l)*2;
-            float current_left_input = hard_clip(preemphasized_left, clipper_threshold);
+            ready_r = apply_pre_emphasis(&preemp_r, ready_r)*2;
-            float current_right_input = hard_clip(preemphasized_right, clipper_threshold);
+            ready_l = hard_clip(ready_l, clipper_threshold);
-#else
+            ready_r = hard_clip(ready_r, clipper_threshold);
-            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
+            float mono = (ready_l + ready_r) / 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
+                float stereo = (ready_l - ready_r) / 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
@@ -367,12 +481,14 @@ int main(int argc, char **argv) {
                         #endif
                         output[i] = delay_line(&monoDelay, mono)*MONO_VOLUME +
                             signal*STEREO_VOLUME;
-                        if(strlen(audio_mpx_device) != 0) output[i] += multiplex_in*MPX_VOLUME;
+                        if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
+                        if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_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;
+                        if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
+                        if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
                     }
                 } else {
                     if(ssb) {
@@ -390,18 +506,22 @@ int main(int argc, char **argv) {
                         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;
+                        if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
+                        if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_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;
+                        if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
+                        if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
                     }
                 }
             } else {
                 output[i] = mono*MONO_VOLUME; // Only Mono
+                if(strlen(audio_mpx_device) != 0) output[i] += current_mpx_in*MPX_VOLUME;
+                if(strlen(audio_sca_device) != 0) output[i] += modulate_fm(&sca_mod, hard_clip(current_sca_in, sca_clipper_threshold))*SCA_VOLUME;
             }
         }
 
@@ -418,6 +538,7 @@ int main(int argc, char **argv) {
     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);
     if(alsa_output == 0) {
         pa_simple_free(output_device);
     } else {

src/options.h

@@ -1,5 +0,0 @@
-#define PREEMPHASIS
-
-#define buffer_maxlength 12288
-#define buffer_tlength_fragsize 8192
-#define buffer_prebuf 16

src/sca_mod.c

@@ -1,195 +0,0 @@
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <stdint.h>
-#include <signal.h>
-#include <string.h>
-
-#include "../lib/constants.h"
-#include "../lib/oscillator.h"
-#include "../lib/filters.h"
-#include "../lib/fm_modulator.h"
-
-#include "options.h"
-
-#define SAMPLE_RATE 192000
-
-#define INPUT_DEVICE "SCA.monitor"
-#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
-// #define ALSA_OUTPUT // Output, not input or both
-#define BUFFER_SIZE 512
-#define CLIPPER_THRESHOLD 1 // Adjust this as needed, this also limits deviation, so if you set this to 0.5 then the deviation will be limited to half
-
-#include <pulse/simple.h>
-#include <pulse/error.h>
-#ifdef ALSA_OUTPUT
-#include <alsa/asoundlib.h>
-#endif
-
-#define VOLUME 0.1f // SCA Volume
-#define VOLUME_AUDIO 1.0f // SCA Audio volume
-#define FREQUENCY 67000 // SCA Frequency
-#define DEVIATION 7000 // SCA Deviation
-
-#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) {
-    if (sample > CLIPPER_THRESHOLD) {
-        return CLIPPER_THRESHOLD;  // Clip to the upper threshold
-    } else if (sample < -CLIPPER_THRESHOLD) {
-        return -CLIPPER_THRESHOLD;  // Clip to the lower threshold
-    } else {
-        return sample;  // No clipping
-    }
-}
-
-static void stop(int signum) {
-    (void)signum;
-    printf("\nReceived stop signal. Cleaning up...\n");
-    to_run = 0;
-}
-
-int main() {
-    printf("SCAMod : SCA Modulator (based on the Stereo encoder STCode) made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
-
-    // Define formats and buffer atributes
-    pa_sample_spec audio_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 = 1,
-        .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
-    };
-
-    pa_buffer_attr input_buffer_atr = {
-        .maxlength = buffer_maxlength,
-	    .fragsize = buffer_tlength_fragsize
-    };
-#ifndef ALSA_OUTPUT
-    pa_buffer_attr output_buffer_atr = {
-        .maxlength = buffer_maxlength,
-        .tlength = buffer_tlength_fragsize,
-	    .prebuf = buffer_prebuf
-    };
-#endif
-
-    printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
-
-    pa_simple *input_device = pa_simple_new(
-        NULL,
-        "SCAMod",
-        PA_STREAM_RECORD,
-        INPUT_DEVICE,
-        "Audio Input",
-        &audio_format,
-        NULL,
-        &input_buffer_atr,
-        NULL
-    );
-    if (!input_device) {
-        fprintf(stderr, "Error: cannot open input device.\n");
-        return 1;
-    }
-
-    printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE);
-    #ifndef ALSA_OUTPUT
-    pa_simple *output_device = pa_simple_new(
-        NULL,
-        "SCAMod",
-        PA_STREAM_PLAYBACK,
-        OUTPUT_DEVICE,
-        "Signal",
-        &audio_format,
-        NULL,
-        &output_buffer_atr,
-        NULL
-    );
-    if (!output_device) {
-        fprintf(stderr, "Error: cannot open output device.\n");
-        pa_simple_free(input_device);
-        return 1;
-    }
-    #else
-    snd_pcm_hw_params_t *output_params;
-    snd_pcm_t *output_handle;
-    int output_error = snd_pcm_open(&output_handle, 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);
-        return 1;
-    }
-    #endif
-
-    FMModulator mod;
-    init_fm_modulator(&mod, FREQUENCY, DEVIATION, SAMPLE_RATE);
-#ifdef PREEMPHASIS
-    ResistorCapacitor preemp;
-    init_rc_tau(&preemp, PREEMPHASIS_TAU, SAMPLE_RATE);
-#endif
-
-    signal(SIGINT, stop);
-    signal(SIGTERM, stop);
-    
-    int pulse_error;
-    float input[BUFFER_SIZE]; // Input from device
-    float signal[BUFFER_SIZE]; // this goes to the output
-    while (to_run) {
-        if (pa_simple_read(input_device, input, sizeof(input), &pulse_error) < 0) {
-            fprintf(stderr, "Error reading from input device: %s\n", pa_strerror(pulse_error));
-            to_run = 0;
-            break;
-        }
-
-        for (int i = 0; i < BUFFER_SIZE; i++) {
-            float in = input[i];
-
-#ifdef PREEMPHASIS
-            float preemphasized = apply_pre_emphasis(&preemp, in)*2;
-            float current_input = hard_clip(preemphasized);
-#else
-            float current_input = hard_clip(in);
-#endif
-
-            signal[i] = modulate_fm(&mod, current_input)*VOLUME;
-        }
-
-#ifndef ALSA_OUTPUT
-        if (pa_simple_write(output_device, signal, sizeof(signal), &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, signal, sizeof(signal));
-#endif
-    }
-    printf("Cleaning up...\n");
-    pa_simple_free(input_device);
-    #ifndef ALSA_OUTPUT
-    pa_simple_free(output_device);
-    #else
-    snd_pcm_drain(output_handle);
-    snd_pcm_close(output_handle);
-    snd_pcm_hw_params_free(&output_params);
-    #endif
-    return 0;
-}