mirror of
https://github.com/radio95-rnt/fm95.git
synced 2026-02-27 11:33:54 +01:00
196 lines
6.1 KiB
C
196 lines
6.1 KiB
C
#include <stdio.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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#include "../lib/fm_modulator.h"
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#include "options.h"
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#define SAMPLE_RATE 192000
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#define INPUT_DEVICE "SCA.monitor"
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#define OUTPUT_DEVICE "alsa_output.platform-soc_sound.stereo-fallback"
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// #define ALSA_OUTPUT // Output, not input or both
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#define BUFFER_SIZE 512
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#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
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#include <pulse/simple.h>
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#include <pulse/error.h>
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#ifdef ALSA_OUTPUT
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#include <alsa/asoundlib.h>
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#endif
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#define VOLUME 0.1f // SCA Volume
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#define VOLUME_AUDIO 1.0f // SCA Audio volume
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#define FREQUENCY 67000 // SCA Frequency
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#define DEVIATION 7000 // SCA Deviation
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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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volatile sig_atomic_t to_run = 1;
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float hard_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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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("SCAMod : SCA Modulator (based on the Stereo encoder STCode) 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 audio_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 = 1,
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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_buffer_attr input_buffer_atr = {
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.maxlength = buffer_maxlength,
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.fragsize = buffer_tlength_fragsize
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};
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#ifndef ALSA_OUTPUT
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pa_buffer_attr output_buffer_atr = {
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.maxlength = buffer_maxlength,
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.tlength = buffer_tlength_fragsize,
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.prebuf = buffer_prebuf
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};
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#endif
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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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"SCAMod",
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PA_STREAM_RECORD,
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INPUT_DEVICE,
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"Audio Input",
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&audio_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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#ifndef ALSA_OUTPUT
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pa_simple *output_device = pa_simple_new(
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NULL,
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"SCAMod",
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PA_STREAM_PLAYBACK,
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OUTPUT_DEVICE,
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"Signal",
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&audio_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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#else
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snd_pcm_hw_params_t *output_params;
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snd_pcm_t *output_handle;
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int output_error = snd_pcm_open(&output_handle, OUTPUT_DEVICE, SND_PCM_STREAM_PLAYBACK, 0);
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if(output_error < 0) {
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fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error));
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pa_simple_free(input_device);
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return 1;
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}
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snd_pcm_hw_params_malloc(&output_params);
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snd_pcm_hw_params_any(output_handle, output_params);
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snd_pcm_hw_params_set_access(output_handle, output_params, SND_PCM_ACCESS_RW_INTERLEAVED);
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snd_pcm_hw_params_set_format(output_handle, output_params, SND_PCM_FORMAT_FLOAT); // Same as pulse's Float32NE
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snd_pcm_hw_params_set_channels(output_handle, output_params, 1);
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unsigned int rate = SAMPLE_RATE;
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int dir;
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snd_pcm_hw_params_set_rate_near(output_handle, output_params, &rate, &dir);
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snd_pcm_uframes_t frames = BUFFER_SIZE;
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snd_pcm_hw_params_set_period_size_near(output_handle, output_params, &frames, &dir); // i don't have a clue why like this
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output_error = snd_pcm_hw_params(output_handle, output_params);
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if(output_error < 0) {
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fprintf(stderr, "Error: cannot open output device: %s\n", snd_strerror(output_error));
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snd_pcm_close(output_handle);
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pa_simple_free(input_device);
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return 1;
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}
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#endif
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FMModulator mod;
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init_fm_modulator(&mod, FREQUENCY, DEVIATION, SAMPLE_RATE);
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#ifdef PREEMPHASIS
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ResistorCapacitor preemp;
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init_rc_tau(&preemp, PREEMPHASIS_TAU, 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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int pulse_error;
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float input[BUFFER_SIZE]; // Input from device
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float signal[BUFFER_SIZE]; // 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), &pulse_error) < 0) {
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fprintf(stderr, "Error reading from input device: %s\n", pa_strerror(pulse_error));
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to_run = 0;
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break;
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}
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for (int i = 0; i < BUFFER_SIZE; i++) {
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float in = input[i];
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#ifdef PREEMPHASIS
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float preemphasized = apply_pre_emphasis(&preemp, in)*2;
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float current_input = hard_clip(preemphasized);
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#else
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float current_input = hard_clip(in);
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#endif
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signal[i] = modulate_fm(&mod, current_input)*VOLUME;
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}
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#ifndef ALSA_OUTPUT
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if (pa_simple_write(output_device, signal, sizeof(signal), &pulse_error) < 0) {
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fprintf(stderr, "Error writing to output device: %s\n", pa_strerror(pulse_error));
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to_run = 0;
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break;
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}
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#else
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snd_pcm_writei(output_handle, signal, sizeof(signal));
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#endif
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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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#ifndef ALSA_OUTPUT
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pa_simple_free(output_device);
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#else
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snd_pcm_drain(output_handle);
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snd_pcm_close(output_handle);
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snd_pcm_hw_params_free(&output_params);
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#endif
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return 0;
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}
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