mirror of
https://github.com/radio95-rnt/fm95.git
synced 2026-02-27 03:23:54 +01:00
remove some code
This commit is contained in:
@@ -1,198 +0,0 @@
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// This is a stereo encoder using the crosby system (https://en.wikipedia.org/wiki/Crosby_system)
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#include <stdio.h>
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#include <pulse/simple.h>
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#include <stdlib.h>
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#include <math.h>
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#include <stdint.h>
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#include <signal.h>
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#include <string.h>
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#include "../lib/constants.h"
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#include "../lib/oscillator.h"
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#include "../lib/filters.h"
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// Features
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#include "features.h"
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#define SAMPLE_RATE 192000 // Don't go lower than 108 KHz, becuase it (53000*2) and (38000+15000)
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#define INPUT_DEVICE "real_real_tx_audio_input.monitor"
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#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
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#define BUFFER_SIZE 512
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#define CLIPPER_THRESHOLD 1.0 // Adjust this as needed
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#define MONO_VOLUME 0.5f // L+R Signal
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#define STEREO_VOLUME_AUDIO 1.0f // L-R signal (once demodulated)
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#define STEREO_VOLUME_MODULATION 0.5f // L-R signal (on MPX)
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#ifdef PREEMPHASIS
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#define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america
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#endif
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#ifdef LPF
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#define LPF_CUTOFF 15000
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#endif
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volatile sig_atomic_t to_run = 1;
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float clip(float sample) {
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if (sample > CLIPPER_THRESHOLD) {
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return CLIPPER_THRESHOLD; // Clip to the upper threshold
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} else if (sample < -CLIPPER_THRESHOLD) {
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return -CLIPPER_THRESHOLD; // Clip to the lower threshold
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} else {
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return sample; // No clipping
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}
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}
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void uninterleave(const float *input, float *left, float *right, size_t num_samples) {
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// 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
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for (size_t i = 0; i < num_samples/2; i++) {
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left[i] = input[i * 2];
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right[i] = input[i * 2 + 1];
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}
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}
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static void stop(int signum) {
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(void)signum;
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printf("\nReceived stop signal. Cleaning up...\n");
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to_run = 0;
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}
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int main() {
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printf("CSTCode : Stereo encoder (Using the crosby system) made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
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// Define formats and buffer atributes
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pa_sample_spec stereo_format = {
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.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
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.channels = 2,
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.rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
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};
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pa_sample_spec mono_format = {
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.format = PA_SAMPLE_FLOAT32NE,
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.channels = 1,
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.rate = SAMPLE_RATE
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};
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pa_buffer_attr input_buffer_atr = {
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.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
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.fragsize = 2048
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};
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pa_buffer_attr output_buffer_atr = {
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.maxlength = 4096,
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.tlength = 2048,
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.prebuf = 0
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};
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printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
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pa_simple *input_device = pa_simple_new(
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NULL,
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"CrosbyStereoEncoder",
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PA_STREAM_RECORD,
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INPUT_DEVICE,
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"Audio Input",
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&stereo_format,
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NULL,
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&input_buffer_atr,
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NULL
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);
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if (!input_device) {
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fprintf(stderr, "Error: cannot open input device.\n");
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return 1;
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}
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printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE);
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pa_simple *output_device = pa_simple_new(
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NULL,
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"CrosbyStereoEncoder",
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PA_STREAM_PLAYBACK,
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OUTPUT_DEVICE,
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"MPX",
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&mono_format,
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NULL,
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&output_buffer_atr,
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NULL
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);
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if (!output_device) {
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fprintf(stderr, "Error: cannot open output device.\n");
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pa_simple_free(input_device);
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return 1;
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}
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Oscillator osc;
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init_oscillator(&osc, 50000.0, SAMPLE_RATE);
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#ifdef PREEMPHASIS
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Emphasis preemp_l, preemp_r;
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init_emphasis(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
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init_emphasis(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
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#endif
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#ifdef LPF
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LowPassFilter lpf_l, lpf_r;
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init_low_pass_filter(&lpf_l, LPF_CUTOFF, SAMPLE_RATE);
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init_low_pass_filter(&lpf_r, LPF_CUTOFF, SAMPLE_RATE);
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#endif
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signal(SIGINT, stop);
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signal(SIGTERM, stop);
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float input[BUFFER_SIZE*2]; // Input from device, interleaved stereo
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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
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float mpx[BUFFER_SIZE]; // MPX, this goes to the output
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while (to_run) {
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if (pa_simple_read(input_device, input, sizeof(input), NULL) < 0) {
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fprintf(stderr, "Error reading from input device.\n");
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break;
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}
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uninterleave(input, left, right, BUFFER_SIZE*2);
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for (int i = 0; i < BUFFER_SIZE; i++) {
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float l_in = left[i];
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float r_in = right[i];
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#ifdef PREEMPHASIS
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#ifdef LPF
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float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in);
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float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
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float preemphasized_left = apply_pre_emphasis(&preemp_l, lowpassed_left);
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float preemphasized_right = apply_pre_emphasis(&preemp_r, lowpassed_right);
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float current_left_input = clip(preemphasized_left);
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float current_right_input = clip(preemphasized_right);
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#else
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float preemphasized_left = apply_pre_emphasis(&preemp_l, l_in);
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float preemphasized_right = apply_pre_emphasis(&preemp_r, r_in);
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float current_left_input = clip(preemphasized_left);
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float current_right_input = clip(preemphasized_right);
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#endif
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#else
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#ifdef LPF
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float lowpassed_left = apply_low_pass_filter(&lpf_l, l_in);
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float lowpassed_right = apply_low_pass_filter(&lpf_r, r_in);
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float current_left_input = clip(lowpassed_left);
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float current_right_input = clip(lowpassed_right);
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#else
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float current_left_input = clip(l_in);
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float current_right_input = clip(r_in);
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#endif
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#endif
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float mono = (current_left_input + current_right_input) / 2.0f; // Stereo to Mono
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float stereo = (current_left_input - current_right_input) / 2.0f; // Also Stereo to Mono but a bit diffrent
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change_oscillator_frequency(&osc, (50000+((stereo*STEREO_VOLUME_AUDIO)*15000)));
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mpx[i] = mono * MONO_VOLUME +
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get_oscillator_sin_sample(&osc)*STEREO_VOLUME_MODULATION;
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}
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if (pa_simple_write(output_device, mpx, sizeof(mpx), NULL) < 0) {
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fprintf(stderr, "Error writing to output device.\n");
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break;
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}
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}
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printf("Cleaning up...\n");
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pa_simple_free(input_device);
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pa_simple_free(output_device);
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return 0;
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}
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@@ -1,232 +0,0 @@
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// what am i doing with my life, writing some quadro encoders? (https://en.wikipedia.org/wiki/FM_broadcasting#Quadraphonic_FM)
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#include <stdio.h>
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#include <pulse/simple.h>
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#include <stdlib.h>
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#include <math.h>
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#include <stdint.h>
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#include <signal.h>
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#include <string.h>
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#include "../lib/constants.h"
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#include "../lib/oscillator.h"
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#include "../lib/filters.h"
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// Features
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#include "features.h"
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#define SAMPLE_RATE 192000 // Don't go lower than 182 KHz (91*2)
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#define INPUT_DEVICE "real_real_tx_audio_input.monitor"
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#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
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#define BUFFER_SIZE 512
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#define CLIPPER_THRESHOLD 1.0f // Adjust this as needed
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#define MONO_VOLUME 0.45f // L+R Signal
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#define PILOT_VOLUME 0.0175f // 19 KHz Pilot
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#define SIN38_VOLUME 0.15f
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#define COS38_VOLUME 0.15f
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#define SIN76_VOLUME 0.15f
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#ifdef PREEMPHASIS
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#define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america
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#endif
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#ifdef LPF
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#define LPF_CUTOFF 15000
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#endif
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volatile sig_atomic_t to_run = 1;
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float clip(float sample) {
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if (sample > CLIPPER_THRESHOLD) {
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return CLIPPER_THRESHOLD; // Clip to the upper threshold
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} else if (sample < -CLIPPER_THRESHOLD) {
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return -CLIPPER_THRESHOLD; // Clip to the lower threshold
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} else {
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return sample; // No clipping
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}
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}
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void uninterleave(const float *input, float *front_left, float *front_right, float *rear_left, float *rear_right, size_t num_samples) {
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for (size_t i = 0; i < num_samples / 4; i++) {
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front_left[i] = input[i * 4];
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front_right[i] = input[i * 4 + 1];
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rear_left[i] = input[i * 4 + 2];
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rear_right[i] = input[i * 4 + 3];
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}
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}
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static void stop(int signum) {
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(void)signum;
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printf("\nReceived stop signal. Cleaning up...\n");
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to_run = 0;
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}
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int main() {
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printf("QDCode : Quad encoder made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
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// Define formats and buffer atributes
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pa_sample_spec stereo_format = {
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.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
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.channels = 4,
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.rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
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};
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pa_sample_spec mono_format = {
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.format = PA_SAMPLE_FLOAT32NE,
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.channels = 1,
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.rate = SAMPLE_RATE
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};
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pa_buffer_attr input_buffer_atr = {
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.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
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.fragsize = 2048
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};
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pa_buffer_attr output_buffer_atr = {
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.maxlength = 4096,
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.tlength = 2048,
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.prebuf = 0
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};
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printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
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pa_simple *input_device = pa_simple_new(
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NULL,
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"QuadCoder",
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PA_STREAM_RECORD,
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INPUT_DEVICE,
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"Audio Input",
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&stereo_format,
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NULL,
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&input_buffer_atr,
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NULL
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);
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if (!input_device) {
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fprintf(stderr, "Error: cannot open input device.\n");
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return 1;
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}
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printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE);
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pa_simple *output_device = pa_simple_new(
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NULL,
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"QuadCoder",
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PA_STREAM_PLAYBACK,
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OUTPUT_DEVICE,
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"MPX",
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&mono_format,
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NULL,
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&output_buffer_atr,
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NULL
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);
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if (!output_device) {
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fprintf(stderr, "Error: cannot open output device.\n");
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pa_simple_free(input_device);
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return 1;
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}
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Oscillator pilot_osc;
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init_oscillator(&pilot_osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
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#ifdef PREEMPHASIS
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Emphasis preemp_lf, preemp_lr, preemp_rf, preemp_rr;
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init_emphasis(&preemp_lf, PREEMPHASIS_TAU, SAMPLE_RATE);
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init_emphasis(&preemp_lr, PREEMPHASIS_TAU, SAMPLE_RATE);
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init_emphasis(&preemp_rf, PREEMPHASIS_TAU, SAMPLE_RATE);
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init_emphasis(&preemp_rr, PREEMPHASIS_TAU, SAMPLE_RATE);
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#endif
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#ifdef LPF
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LowPassFilter lpf_lf, lpf_lr, lpf_rf, lpf_rr;
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init_low_pass_filter(&lpf_lf, LPF_CUTOFF, SAMPLE_RATE);
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init_low_pass_filter(&lpf_lr, LPF_CUTOFF, SAMPLE_RATE);
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init_low_pass_filter(&lpf_rf, LPF_CUTOFF, SAMPLE_RATE);
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init_low_pass_filter(&lpf_rr, LPF_CUTOFF, SAMPLE_RATE);
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#endif
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signal(SIGINT, stop);
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signal(SIGTERM, stop);
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float input[BUFFER_SIZE*4]; // Input from device, interleaved
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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
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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
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float mpx[BUFFER_SIZE]; // MPX, this goes to the output
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while (to_run) {
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if (pa_simple_read(input_device, input, sizeof(input), NULL) < 0) {
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fprintf(stderr, "Error reading from input device.\n");
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break;
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}
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uninterleave(input, left_front, right_front, left_rear, right_rear, BUFFER_SIZE*4);
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for (int i = 0; i < BUFFER_SIZE; i++) {
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float sin38 = sinf((pilot_osc.phase+(0.5*PI))*2);
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float cos38 = cosf((pilot_osc.phase+(0.5*PI))*2);
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float sin76 = sinf((pilot_osc.phase+(0.5*PI))*4);
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float pilot = get_oscillator_sin_sample(&pilot_osc);
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float lf_in = left_front[i];
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float lr_in = left_rear[i];
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float rf_in = right_front[i];
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float rr_in = right_rear[i];
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#ifdef PREEMPHASIS
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#ifdef LPF
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float lowpassed_frontleft = apply_low_pass_filter(&lpf_lf, lf_in);
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float lowpassed_frontright = apply_low_pass_filter(&lpf_rf, rf_in);
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float lowpassed_rearleft = apply_low_pass_filter(&lpf_lr, lr_in);
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float lowpassed_rearright = apply_low_pass_filter(&lpf_rr, rr_in);
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float preemphasized_frontleft = apply_pre_emphasis(&preemp_lf, lowpassed_frontleft);
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float preemphasized_frontright = apply_pre_emphasis(&preemp_rf, lowpassed_frontright);
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float preemphasized_rearleft = apply_pre_emphasis(&preemp_lr, lowpassed_rearleft);
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float preemphasized_rearright = apply_pre_emphasis(&preemp_rr, lowpassed_rearright);
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float current_lf_input = clip(preemphasized_frontleft);
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float current_rf_input = clip(preemphasized_frontright);
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float current_lr_input = clip(preemphasized_rearleft);
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float current_rr_input = clip(preemphasized_rearright);
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#else
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float preemphasized_frontleft = apply_pre_emphasis(&preemp_lf, lf_in);
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float preemphasized_frontright = apply_pre_emphasis(&preemp_rf, rf_in);
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float preemphasized_rearleft = apply_pre_emphasis(&preemp_lr, lr_in);
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float preemphasized_rearright = apply_pre_emphasis(&preemp_rr, rr_in);
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float current_lf_input = clip(preemphasized_frontleft);
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float current_rf_input = clip(preemphasized_frontright);
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float current_lr_input = clip(preemphasized_rearleft);
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float current_rr_input = clip(preemphasized_rearright);
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#endif
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#else
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||||
#ifdef LPF
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float lowpassed_frontleft = apply_low_pass_filter(&lpf_lf, lf_in);
|
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float lowpassed_frontright = apply_low_pass_filter(&lpf_rf, rf_in);
|
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float lowpassed_rearleft = apply_low_pass_filter(&lpf_lr, lr_in);
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float lowpassed_rearright = apply_low_pass_filter(&lpf_rr, rr_in);
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float current_lf_input = clip(lowpassed_frontleft);
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float current_rf_input = clip(lowpassed_frontright);
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float current_lr_input = clip(lowpassed_rearleft);
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float current_rr_input = clip(lowpassed_rearright);
|
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#else
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float current_lf_input = clip(lf_in);
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float current_rf_input = clip(rf_in);
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float current_lr_input = clip(lr_in);
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float current_rr_input = clip(rr_in);
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||||
#endif
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||||
#endif
|
||||
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||||
float mono = (current_lf_input+current_rf_input+current_lr_input+current_rr_input)/4;
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float signal_sin38 = ((current_lf_input+current_lr_input)-(current_rf_input+current_rr_input))/4;
|
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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;
|
||||
}
|
||||
Reference in New Issue
Block a user