add alsa output to pstcode, scamod

.vscode/.server-controller-port.log

@@ -1,5 +1,5 @@
 {
   "port": 13452,
-  "time": 1737738374932,
+  "time": 1737831115102,
   "version": "0.0.3"
 }

satire/crosby_stereo_coder.c

@@ -1,197 +0,0 @@
-#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"
-
-#include "options.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 0.525 // Adjust this as needed
-
-#define MONO_VOLUME 0.5f // L+R Signal
-#define STEREO_VOLUME_AUDIO 1.0f // L-R signal
-#define STEREO_VOLUME_MODULATION 0.5f // L-R signal
-
-#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("CrosbySTCode : 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 = buffer_maxlength,
-	    .fragsize = buffer_tlength_fragsize
-    };
-    pa_buffer_attr output_buffer_atr = {
-        .maxlength = buffer_maxlength,
-        .tlength = buffer_tlength_fragsize,
-	    .prebuf = buffer_prebuf
-    };
-
-    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
-    ResistorCapacitor preemp_l, preemp_r;
-    init_rc_tau(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
-    init_rc_tau(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
-#endif
-#ifdef LPF
-    ResistorCapacitor 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, 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), &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; // 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), &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;
-}

satire/stereo_sca_mod.c

@@ -1,198 +0,0 @@
-#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"
-#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 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 0.075f // Mono Volume
-#define STEREO_VOLUME 0.025f // 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("StereoSCAMod : Stereo SCA 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 = buffer_maxlength,
-	    .fragsize = buffer_tlength_fragsize
-    };
-    pa_buffer_attr output_buffer_atr = {
-        .maxlength = buffer_maxlength,
-        .tlength = buffer_tlength_fragsize,
-	    .prebuf = buffer_prebuf
-    };
-
-    printf("Connecting to input device... (%s)\n", INPUT_DEVICE);
-
-    pa_simple *input_device = pa_simple_new(
-        NULL,
-        "StereoSCAMod",
-        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,
-        "StereoSCAMod",
-        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;
-    }
-
-    FMModulator mod_mono, mod_stereo;
-    init_fm_modulator(&mod_mono, 67000, 6000, SAMPLE_RATE);
-    init_fm_modulator(&mod_stereo, 80000, 6000, SAMPLE_RATE);
-#ifdef PREEMPHASIS
-    ResistorCapacitor preemp_l, preemp_r;
-    init_rc_tau(&preemp_l, PREEMPHASIS_TAU, SAMPLE_RATE);
-    init_rc_tau(&preemp_r, PREEMPHASIS_TAU, SAMPLE_RATE);
-#endif
-#ifdef LPF
-    ResistorCapacitor 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;
-
-            signal[i] = modulate_fm(&mod_mono, mono)*MONO_VOLUME+
-                modulate_fm(&mod_stereo, 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;
-}

src/polar_stereo_coder.c

@@ -17,6 +17,7 @@
 
 #define INPUT_DEVICE "real_real_tx_audio_input.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 0.525 // Adjust this as needed
 
@@ -75,11 +76,13 @@ int main() {
         .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);
 
@@ -101,6 +104,7 @@ int main() {
 
     printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE);
 
+#ifndef ALSA_OUTPUT
     pa_simple *output_device = pa_simple_new(
         NULL,
         "PolarStereoEncoder",
@@ -117,6 +121,33 @@ int main() {
         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
 
     Oscillator stereo_osc;
     init_oscillator(&stereo_osc, 31250.0, SAMPLE_RATE);
@@ -186,14 +217,22 @@ int main() {
                 ((stereo+0.2) * stereo_carrier)*STEREO_VOLUME; // the 0.2 add DC, you know what happens then? Carrier wave
         }
 
+#ifndef ALSA_OUTPUT
         if (pa_simple_write(output_device, mpx, sizeof(mpx), &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, mpx, sizeof(mpx));
+#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_free(output_handle);
     return 0;
 }

src/sca_mod.c

@@ -1,6 +1,4 @@
 #include <stdio.h>
-#include <pulse/simple.h>
-#include <pulse/error.h>
 #include <stdlib.h>
 #include <math.h>
 #include <stdint.h>
@@ -18,9 +16,16 @@
 
 #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
@@ -66,11 +71,13 @@ int main() {
         .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);
 
@@ -91,7 +98,7 @@ int main() {
     }
 
     printf("Connecting to output device... (%s)\n", OUTPUT_DEVICE);
-
+    #ifndef ALSA_OUTPUT
     pa_simple *output_device = pa_simple_new(
         NULL,
         "SCAMod",
@@ -108,6 +115,33 @@ int main() {
         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);
@@ -157,14 +191,23 @@ int main() {
             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_free(output_handle);
+    #endif
     return 0;
 }

src/stereo_coder.c

@@ -23,7 +23,7 @@
 
 #define INPUT_DEVICE "real_real_tx_audio_input.monitor"
 #define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
-#define ALSA_OUTPUT // Output, not input or both
+// #define ALSA_OUTPUT // Output, not input or both
 #define BUFFER_SIZE 512
 #define CLIPPER_THRESHOLD 0.525 // Adjust this as needed
 
@@ -89,11 +89,13 @@ int main() {
         .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
 
     int open_pulse_error;
 

wip/tv_encoder.c

@@ -1,48 +0,0 @@
-// This will encode a black and white TV signal using a luminance value, how does it work?
-/*
-    It encodes the luminance into negative values, so totally white pixel should output -1, a black one should be 0
-    
-    Every new line it sends a 0.5, every frame it is a 1.0
-*/
-
-#include "../lib/fm_modulator.h"
-
-unsigned int rgb_to_luminance(unsigned int r, unsigned int g, unsigned int b) {
-    return (unsigned int)(0.299 * r + 0.587 * g + 0.114 * b);
-}
-
-typedef struct {
-    int line;
-    int pixel;
-    int lines;
-    int pixels;
-} TVEncoder;
-
-void init_tv_modulator(TVEncoder* tv, int lines, int pixels) {
-    tv->pixels = pixels;
-    tv->lines = lines;
-    tv->line = 0;
-    tv->pixel = 0;
-}
-
-float tv_encode(TVEncoder* tv, float luminance) {
-    float normalized_luminance = luminance / 255.0f;  // Normalize luminance to [0, 1]
-
-    if (tv->line < tv->lines) {
-        if (tv->pixel < tv->pixels) {
-            // Process pixel within the current line
-            tv->pixel++;
-            return -normalized_luminance;
-        } else {
-            // End of line: reset pixel counter and move to the next line
-            tv->pixel = 0;
-            tv->line++;
-            return 0.5f;
-        }
-    } else {
-        // End of frame: reset frame counters
-        tv->line = 0;
-        tv->pixel = 0;
-        return 1.0f;
-    }
-}