sap

.vscode/.server-controller-port.log

@@ -1,5 +1,5 @@
 {
   "port": 13452,
-  "time": 1735749319488,
+  "time": 1735839820567,
   "version": "0.0.3"
 }

README.md

@@ -37,3 +37,6 @@ Also it doesnsn't sound bad, how may ask where did i find a decoder for it? Made
 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
+
+# SSAPCoder
+This "standard" was made by me, it encoder stereo am-quality audio into the FM carrier

src/stereo_sap_coder.c

@@ -0,0 +1,200 @@
+#include <stdio.h>
+#include <pulse/simple.h>
+#include <pulse/error.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
+
+#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 0.75 // Adjust this as needed, this also limits deviation, so if you set this to 0.5 then the deviation will be limited to half
+
+#define MONO_VOLUME 0.03f // Mono Volume
+#define STEREO_VOLUME 0.01f // Stereo Volume
+
+#ifdef PREEMPHASIS
+#define PREEMPHASIS_TAU 0.00005  // 50 microseconds, use 0.000075 if in america
+#endif
+
+#ifdef LPF
+#define LPF_CUTOFF 8000
+#endif
+
+volatile sig_atomic_t to_run = 1;
+
+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];
+    }
+}
+
+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
+    }
+}
+
+static void stop(int signum) {
+    (void)signum;
+    printf("\nReceived stop signal. Cleaning up...\n");
+    to_run = 0;
+}
+
+int main() {
+    printf("SSCAPMod : Stereo SAP Modulator (based on the SCA encoder SCAMod) made by radio95 (with help of ChatGPT and Claude, thanks!)\n");
+
+    // Define formats and buffer atributes
+    pa_sample_spec mono_audio_format = {
+        .format = PA_SAMPLE_FLOAT32LE,
+        .channels = 1,
+        .rate = SAMPLE_RATE // Same sample rate makes it easy, leave the resampling to pipewire, it should know better
+    };
+    pa_sample_spec stereo_audio_format = {
+        .format = PA_SAMPLE_FLOAT32LE,
+        .channels = 2,
+        .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 = 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,
+        "SSAPMod",
+        PA_STREAM_RECORD,
+        INPUT_DEVICE,
+        "Audio Input",
+        &stereo_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);
+
+    pa_simple *output_device = pa_simple_new(
+        NULL,
+        "SSAPMod",
+        PA_STREAM_PLAYBACK,
+        OUTPUT_DEVICE,
+        "Signal",
+        &mono_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;
+    }
+
+    Oscillator osc_mono, osc_stereo;
+    init_oscillator(&osc_mono, 68000, SAMPLE_RATE);
+    init_oscillator(&osc_mono, 77000, SAMPLE_RATE);
+#ifdef PREEMPHASIS
+    Emphasis preemp_l, premp_r;
+    init_emphasis(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
+    init_emphasis(&premp_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);
+    
+    int pulse_error;
+    float input[BUFFER_SIZE*2]; // Input from device
+    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 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;
+        }
+        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;
+            float stereo = (current_left_input-current_right_input)/2.0f;
+
+            change_oscillator_frequency(&osc_mono, (68000+(mono*8000)));
+            change_oscillator_frequency(&osc_stereo, (77000+(stereo*8000)));
+            signal[i] = get_oscillator_sin_sample(&osc_mono)*MONO_VOLUME+
+                get_oscillator_sin_sample(&osc_stereo)*STEREO_VOLUME;
+        }
+
+        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;
+        }
+    }
+    printf("Cleaning up...\n");
+    pa_simple_free(input_device);
+    pa_simple_free(output_device);
+    return 0;
+}