diff --git a/README.md b/README.md
index 4d55ef4..981a939 100644
--- a/README.md
+++ b/README.md
@@ -23,21 +23,4 @@ This also has a cpu usage of 20% with lpf, but goes to 13-15% without the lpf
 # 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
-
-# QDCode
-QD code is a FM quadrophonic encoder, following the Dorren standard
-
-I haven't tested this, but i'm scared, i don't have a decoder anyway
-
-# CSTCode
-This is a stereo encoder, but using the crosby system, as we all know, stereo is made of these things:
-0-15: mono
-19 khz: pilot
-38 khz: stereo
-but the crosby system is:
-0-15: mono (seems normal, right?)
-50 khz: fm modulated l-r
-
-
-yeah (https://en.wikipedia.org/wiki/Crosby_system)
\ No newline at end of file
+Has a fine quality, but as it goes for 12 khz fm signals
\ No newline at end of file
diff --git a/src/crosby_stereo_encoder.c b/src/crosby_stereo_encoder.c
deleted file mode 100644
index 0833253..0000000
--- a/src/crosby_stereo_encoder.c
+++ /dev/null
@@ -1,198 +0,0 @@
-// This is a stereo encoder using the crosby system (https://en.wikipedia.org/wiki/Crosby_system)
-
-#include <stdio.h>
-#include <pulse/simple.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"
-
-// Features
-#include "features.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 BUFFER_SIZE 512
-#define CLIPPER_THRESHOLD 1.0 // Adjust this as needed
-
-#define MONO_VOLUME 0.5f // L+R Signal
-#define STEREO_VOLUME_AUDIO 1.0f // L-R signal (once demodulated)
-#define STEREO_VOLUME_MODULATION 0.5f // L-R signal (on MPX)
-
-#ifdef PREEMPHASIS
-#define PREEMPHASIS_TAU 0.00005  // 50 microseconds, use 0.000075 if in america
-#endif
-
-#ifdef LPF
-#define LPF_CUTOFF 15000
-#endif
-
-volatile sig_atomic_t to_run = 1;
-
-float 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
-    }
-}
-
-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;
-}
-
-int main() {
-    printf("CSTCode : Stereo encoder (Using the crosby system) made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
-    // 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 = 4096, // You can lower this to 512, but this is fine, it's sub-second delay, you're probably not gonna notice unless you're looking for it
-	    .fragsize = 2048
-    };
-    pa_buffer_attr output_buffer_atr = {
-        .maxlength = 4096,
-        .tlength = 2048,
-	    .prebuf = 0
-    };
-
-    printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
-
-    pa_simple *input_device = pa_simple_new(
-        NULL,
-        "CrosbyStereoEncoder",
-        PA_STREAM_RECORD,
-        INPUT_DEVICE,
-        "Audio Input",
-        &stereo_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);
-
-    pa_simple *output_device = pa_simple_new(
-        NULL,
-        "CrosbyStereoEncoder",
-        PA_STREAM_PLAYBACK,
-        OUTPUT_DEVICE,
-        "MPX",
-        &mono_format,
-        NULL,
-        &output_buffer_atr,
-        NULL
-    );
-    if (!output_device) {
-        fprintf(stderr, "Error: cannot open output device.\n");
-        pa_simple_free(input_device);
-        return 1;
-    }
-
-    Oscillator osc;
-    init_oscillator(&osc, 50000.0, SAMPLE_RATE);
-#ifdef PREEMPHASIS
-    Emphasis preemp_l, preemp_r;
-    init_emphasis(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
-    init_emphasis(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
-#endif
-#ifdef LPF
-    LowPassFilter lpf_l, lpf_r;
-    init_low_pass_filter(&lpf_l, LPF_CUTOFF, SAMPLE_RATE);
-    init_low_pass_filter(&lpf_r, LPF_CUTOFF, SAMPLE_RATE);
-#endif
-
-    signal(SIGINT, stop);
-    signal(SIGTERM, stop);
-    
-    float input[BUFFER_SIZE*2]; // Input from device, interleaved stereo
-    float left[BUFFER_SIZE+64], right[BUFFER_SIZE+64]; // Audio, same thing as in input but ininterleaved, ai told be there could be a buffer overflow here
-    float mpx[BUFFER_SIZE]; // MPX, this goes to the output
-    while (to_run) {
-        if (pa_simple_read(input_device, input, sizeof(input), NULL) < 0) {
-            fprintf(stderr, "Error reading from input device.\n");
-            break;
-        }
-        uninterleave(input, left, right, BUFFER_SIZE*2);
-
-        for (int i = 0; i < BUFFER_SIZE; i++) {
-            float l_in = left[i];
-            float r_in = right[i];
-
-#ifdef PREEMPHASIS
-#ifdef LPF
-            float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in);
-            float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
-            float preemphasized_left = apply_pre_emphasis(&preemp_l, lowpassed_left);
-            float preemphasized_right = apply_pre_emphasis(&preemp_r, lowpassed_right);
-            float current_left_input = clip(preemphasized_left);
-            float current_right_input = clip(preemphasized_right);
-#else
-            float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in);
-            float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in);
-            float current_left_input = clip(preemphasized_left);
-            float current_right_input = clip(preemphasized_right);
-#endif
-#else
-#ifdef LPF
-            float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in);
-            float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
-            float current_left_input = clip(lowpassed_left);
-            float current_right_input = clip(lowpassed_right);
-#else
-            float current_left_input = clip(l_in);
-            float current_right_input = clip(r_in);
-#endif
-#endif
-
-            float mono = (current_left_input + current_right_input) / 2.0f; // Stereo to Mono
-            float stereo = (current_left_input - current_right_input) / 2.0f; // Also Stereo to Mono but a bit diffrent
-
-            change_oscillator_frequency(&osc, (50000+((stereo*STEREO_VOLUME_AUDIO)*15000)));
-
-            mpx[i] = mono * MONO_VOLUME +
-                get_oscillator_sin_sample(&osc)*STEREO_VOLUME_MODULATION;
-        }
-
-        if (pa_simple_write(output_device, mpx, sizeof(mpx), NULL) < 0) {
-            fprintf(stderr, "Error writing to output device.\n");
-            break;
-        }
-    }
-    printf("Cleaning up...\n");
-    pa_simple_free(input_device);
-    pa_simple_free(output_device);
-    return 0;
-}
diff --git a/src/quadro_encoder.c b/src/quadro_encoder.c
deleted file mode 100644
index 297435e..0000000
--- a/src/quadro_encoder.c
+++ /dev/null
@@ -1,232 +0,0 @@
-// what am i doing with my life, writing some quadro encoders? (https://en.wikipedia.org/wiki/FM_broadcasting#Quadraphonic_FM)
-
-#include <stdio.h>
-#include <pulse/simple.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"
-
-// Features
-#include "features.h"
-
-#define SAMPLE_RATE 192000 // Don't go lower than 182 KHz (91*2)
-
-#define INPUT_DEVICE "real_real_tx_audio_input.monitor"
-#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
-#define BUFFER_SIZE 512
-#define CLIPPER_THRESHOLD 1.0f // Adjust this as needed
-
-#define MONO_VOLUME 0.45f // L+R Signal
-#define PILOT_VOLUME 0.0175f // 19 KHz Pilot
-#define SIN38_VOLUME 0.15f
-#define COS38_VOLUME 0.15f
-#define SIN76_VOLUME 0.15f
-
-#ifdef PREEMPHASIS
-#define PREEMPHASIS_TAU 0.00005  // 50 microseconds, use 0.000075 if in america
-#endif
-
-#ifdef LPF
-#define LPF_CUTOFF 15000
-#endif
-
-volatile sig_atomic_t to_run = 1;
-
-float 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
-    }
-}
-
-void uninterleave(const float *input, float *front_left, float *front_right, float *rear_left, float *rear_right, size_t num_samples) {
-    for (size_t i = 0; i < num_samples / 4; i++) {
-        front_left[i] = input[i * 4];
-        front_right[i] = input[i * 4 + 1];
-        rear_left[i] = input[i * 4 + 2];
-        rear_right[i] = input[i * 4 + 3];
-    }
-}
-
-static void stop(int signum) {
-    (void)signum;
-    printf("\nReceived stop signal. Cleaning up...\n");
-    to_run = 0;
-}
-
-int main() {
-    printf("QDCode : Quad encoder made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
-    // 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 = 4,
-        .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 = 4096, // You can lower this to 512, but this is fine, it's sub-second delay, you're probably not gonna notice unless you're looking for it
-	    .fragsize = 2048
-    };
-    pa_buffer_attr output_buffer_atr = {
-        .maxlength = 4096,
-        .tlength = 2048,
-	    .prebuf = 0
-    };
-
-    printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
-
-    pa_simple *input_device = pa_simple_new(
-        NULL,
-        "QuadCoder",
-        PA_STREAM_RECORD,
-        INPUT_DEVICE,
-        "Audio Input",
-        &stereo_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);
-
-    pa_simple *output_device = pa_simple_new(
-        NULL,
-        "QuadCoder",
-        PA_STREAM_PLAYBACK,
-        OUTPUT_DEVICE,
-        "MPX",
-        &mono_format,
-        NULL,
-        &output_buffer_atr,
-        NULL
-    );
-    if (!output_device) {
-        fprintf(stderr, "Error: cannot open output device.\n");
-        pa_simple_free(input_device);
-        return 1;
-    }
-
-    Oscillator pilot_osc;
-    init_oscillator(&pilot_osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
-#ifdef PREEMPHASIS
-    Emphasis preemp_lf, preemp_lr, preemp_rf, preemp_rr;
-    init_emphasis(&preemp_lf, PREEMPHASIS_TAU, SAMPLE_RATE);
-    init_emphasis(&preemp_lr, PREEMPHASIS_TAU, SAMPLE_RATE);
-    init_emphasis(&preemp_rf, PREEMPHASIS_TAU, SAMPLE_RATE);
-    init_emphasis(&preemp_rr, PREEMPHASIS_TAU, SAMPLE_RATE);
-#endif
-#ifdef LPF
-    LowPassFilter lpf_lf, lpf_lr, lpf_rf, lpf_rr;
-    init_low_pass_filter(&lpf_lf, LPF_CUTOFF, SAMPLE_RATE);
-    init_low_pass_filter(&lpf_lr, LPF_CUTOFF, SAMPLE_RATE);
-    init_low_pass_filter(&lpf_rf, LPF_CUTOFF, SAMPLE_RATE);
-    init_low_pass_filter(&lpf_rr, LPF_CUTOFF, SAMPLE_RATE);
-#endif
-
-    signal(SIGINT, stop);
-    signal(SIGTERM, stop);
-    
-    float input[BUFFER_SIZE*4]; // Input from device, interleaved
-    float left_front[BUFFER_SIZE+64], left_rear[BUFFER_SIZE+64]; // Audio, same thing as in input but ininterleaved, ai told be there could be a buffer overflow here
-    float right_front[BUFFER_SIZE+64], right_rear[BUFFER_SIZE+64]; // Audio, same thing as in input but ininterleaved, ai told be there could be a buffer overflow here
-    float mpx[BUFFER_SIZE]; // MPX, this goes to the output
-    while (to_run) {
-        if (pa_simple_read(input_device, input, sizeof(input), NULL) < 0) {
-            fprintf(stderr, "Error reading from input device.\n");
-            break;
-        }
-        uninterleave(input, left_front, right_front, left_rear, right_rear, BUFFER_SIZE*4);
-
-        for (int i = 0; i < BUFFER_SIZE; i++) {
-            float sin38 = sinf((pilot_osc.phase+(0.5*PI))*2);
-            float cos38 = cosf((pilot_osc.phase+(0.5*PI))*2);
-            float sin76 = sinf((pilot_osc.phase+(0.5*PI))*4);
-            float pilot = get_oscillator_sin_sample(&pilot_osc);
-            float lf_in = left_front[i];
-            float lr_in = left_rear[i];
-            float rf_in = right_front[i];
-            float rr_in = right_rear[i];
-
-#ifdef PREEMPHASIS
-#ifdef LPF
-            float lowpassed_frontleft = apply_low_pass_filter(&lpf_lf, lf_in);
-            float lowpassed_frontright = apply_low_pass_filter(&lpf_rf, rf_in);
-            float lowpassed_rearleft = apply_low_pass_filter(&lpf_lr, lr_in);
-            float lowpassed_rearright = apply_low_pass_filter(&lpf_rr, rr_in);
-            float preemphasized_frontleft = apply_pre_emphasis(&preemp_lf, lowpassed_frontleft);
-            float preemphasized_frontright = apply_pre_emphasis(&preemp_rf, lowpassed_frontright);
-            float preemphasized_rearleft = apply_pre_emphasis(&preemp_lr, lowpassed_rearleft);
-            float preemphasized_rearright = apply_pre_emphasis(&preemp_rr, lowpassed_rearright);
-            float current_lf_input = clip(preemphasized_frontleft);
-            float current_rf_input = clip(preemphasized_frontright);
-            float current_lr_input = clip(preemphasized_rearleft);
-            float current_rr_input = clip(preemphasized_rearright);
-#else
-            float preemphasized_frontleft = apply_pre_emphasis(&preemp_lf, lf_in);
-            float preemphasized_frontright = apply_pre_emphasis(&preemp_rf, rf_in);
-            float preemphasized_rearleft = apply_pre_emphasis(&preemp_lr, lr_in);
-            float preemphasized_rearright = apply_pre_emphasis(&preemp_rr, rr_in);
-            float current_lf_input = clip(preemphasized_frontleft);
-            float current_rf_input = clip(preemphasized_frontright);
-            float current_lr_input = clip(preemphasized_rearleft);
-            float current_rr_input = clip(preemphasized_rearright);
-#endif
-#else
-#ifdef LPF
-            float lowpassed_frontleft = apply_low_pass_filter(&lpf_lf, lf_in);
-            float lowpassed_frontright = apply_low_pass_filter(&lpf_rf, rf_in);
-            float lowpassed_rearleft = apply_low_pass_filter(&lpf_lr, lr_in);
-            float lowpassed_rearright = apply_low_pass_filter(&lpf_rr, rr_in);
-            float current_lf_input = clip(lowpassed_frontleft);
-            float current_rf_input = clip(lowpassed_frontright);
-            float current_lr_input = clip(lowpassed_rearleft);
-            float current_rr_input = clip(lowpassed_rearright);
-#else
-            float current_lf_input = clip(lf_in);
-            float current_rf_input = clip(rf_in);
-            float current_lr_input = clip(lr_in);
-            float current_rr_input = clip(rr_in);
-#endif
-#endif
-
-            float mono = (current_lf_input+current_rf_input+current_lr_input+current_rr_input)/4;
-            float signal_sin38 = ((current_lf_input+current_lr_input)-(current_rf_input+current_rr_input))/4;
-            float signal_cos38 = ((current_lf_input+current_rr_input)-(current_lr_input+current_rf_input))/4;
-            float signal_sin76 = ((current_lf_input+current_rf_input)-(current_lr_input+current_rr_input))/4;
-
-            mpx[i] = mono * MONO_VOLUME +
-                pilot * PILOT_VOLUME +
-                (sin38*signal_sin38)*SIN38_VOLUME +
-                (cos38*signal_cos38)*COS38_VOLUME +
-                (sin76*signal_sin76)*SIN76_VOLUME;
-
-        }
-
-        if (pa_simple_write(output_device, mpx, sizeof(mpx), NULL) < 0) {
-            fprintf(stderr, "Error writing to output device.\n");
-            break;
-        }
-    }
-    printf("Cleaning up...\n");
-    pa_simple_free(input_device);
-    pa_simple_free(output_device);
-    return 0;
-}