#include #include #include #include #include #include #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 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 #include #ifdef ALSA_OUTPUT #include #endif #define VOLUME 0.1f // SCA Volume #define VOLUME_AUDIO 1.0f // SCA Audio volume #define FREQUENCY 67000 // SCA Frequency #define DEVIATION 7000 // SCA Deviation #ifdef PREEMPHASIS #define PREEMPHASIS_TAU 0.00005 // 50 microseconds, use 0.000075 if in america #endif volatile sig_atomic_t to_run = 1; float hard_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("SCAMod : SCA Modulator (based on the Stereo encoder STCode) made by radio95 (with help of ChatGPT and Claude, thanks!)\n"); // Define formats and buffer atributes pa_sample_spec audio_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 = 1, .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 }; #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); pa_simple *input_device = pa_simple_new( NULL, "SCAMod", PA_STREAM_RECORD, INPUT_DEVICE, "Audio Input", &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); #ifndef ALSA_OUTPUT pa_simple *output_device = pa_simple_new( NULL, "SCAMod", PA_STREAM_PLAYBACK, OUTPUT_DEVICE, "Signal", &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; } #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); #ifdef PREEMPHASIS ResistorCapacitor preemp; init_rc_tau(&preemp, PREEMPHASIS_TAU, SAMPLE_RATE); #endif signal(SIGINT, stop); signal(SIGTERM, stop); int pulse_error; float input[BUFFER_SIZE]; // Input from device 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; } for (int i = 0; i < BUFFER_SIZE; i++) { float in = input[i]; #ifdef PREEMPHASIS float preemphasized = apply_pre_emphasis(&preemp, in)*2; float current_input = hard_clip(preemphasized); #else float current_input = hard_clip(in); #endif 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_close(output_handle); snd_pcm_hw_params_free(&output_params); #endif return 0; }