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remove some code

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KubaPro010
2025-01-01 02:20:22 +01:00
parent b871d05c95
commit 4f6083a79b
3 changed files with 1 additions and 448 deletions
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19
README.md
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@@ -23,21 +23,4 @@ This also has a cpu usage of 20% with lpf, but goes to 13-15% without the lpf
# SCAMod # 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 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 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)

198
src/crosby_stereo_encoder.c
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@@ -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;
}

232
src/quadro_encoder.c
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@@ -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;
}