#include #include #include #include #include #include #define INPUT_DEVICE "real_real_tx_audio_input.monitor" #define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback" #define BUFFER_SIZE 512 #define MONO_VOLUME 0.5f // L+R Signal #define PILOT_VOLUME 0.025f // 19 KHz Pilot #define STEREO_VOLUME 0.275f // L-R signal #define TWO_BUFFER_SIZE (BUFFER_SIZE*2) // Don't touch this volatile sig_atomic_t to_run = 1; const float format_scale = 1.0f / 32768.0f; void stereo_s16le_to_float(const int16_t *input, float *left, float *right, size_t num_samples) { for (size_t i = 0; i < num_samples/2; i++) { left[i] = input[i * 2] * format_scale; right[i] = input[i * 2 + 1] * format_scale; } } void float_array_to_s16le(const float *input, int16_t *output, size_t num_samples) { for (size_t i = 0; i < num_samples; i++) { output[i] = (int16_t)((fminf(fmaxf(input[i], -1.0f), 1.0f)) * 32767.0f); } } #define M_2PI (3.14159265358979323846 * 2.0) // Track phase continuously to maintain frequency accuracy typedef struct { float phase; float phase_increment; } Oscillator; void init_oscillator(Oscillator *osc, float frequency, float sample_rate) { osc->phase = 0.0f; osc->phase_increment = (M_2PI * frequency) / sample_rate; } float get_next_sample(Oscillator *osc) { float sample = sinf(osc->phase); osc->phase += osc->phase_increment; if (osc->phase >= M_2PI) { osc->phase -= M_2PI; } return sample; } static void stop(int signum) { (void)signum; printf("\nReceived stop signal. Cleaning up...\n"); to_run = 0; } int main() { printf("STCode : Stereo encoder made by radio95 (with help of ChatGPT and Claude, thanks!)\n"); const float SAMPLE_RATE = 192000.0f; // Don't go lower than 108 KHz, becuase it (53000*2) and (38000+15000) const float PILOT_FREQ = 19000.0f; const float STEREO_FREQ = 38000.0f; // Define formats and buffer atributes pa_sample_spec stereo_format = { .format = PA_SAMPLE_S16LE, .channels = 2, .rate = SAMPLE_RATE }; pa_sample_spec mono_format = { .format = PA_SAMPLE_S16LE, .channels = 1, .rate = SAMPLE_RATE }; pa_buffer_attr input_buffer_atr = { .maxlength = 4096, .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, "StereoEncoder", 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, "StereoEncoder", 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, stereo_osc; init_oscillator(&pilot_osc, PILOT_FREQ, SAMPLE_RATE); init_oscillator(&stereo_osc, STEREO_FREQ, SAMPLE_RATE); signal(SIGINT, stop); signal(SIGTERM, stop); int16_t input[TWO_BUFFER_SIZE]; float left[BUFFER_SIZE], right[BUFFER_SIZE]; float mpx[BUFFER_SIZE]; int16_t output[BUFFER_SIZE]; while (to_run) { if (pa_simple_read(input_device, input, sizeof(input), NULL) < 0) { fprintf(stderr, "Error reading from input device.\n"); break; } stereo_s16le_to_float(input, left, right, TWO_BUFFER_SIZE); for (int i = 0; i < BUFFER_SIZE; i++) { float pilot = get_next_sample(&pilot_osc); float stereo_carrier = get_next_sample(&stereo_osc); float mono = (left[i] + right[i]) / 2.0f; float stereo = (left[i] - right[i]) / 2.0f; mpx[i] = mono*MONO_VOLUME + (stereo * stereo_carrier)*STEREO_VOLUME + (pilot * PILOT_VOLUME); } float_array_to_s16le(mpx, output, BUFFER_SIZE); if (pa_simple_write(output_device, output, sizeof(output), 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; }