fm95/stereo_coder.c

#include <stdio.h>
#include <pulse/simple.h>
#include <stdlib.h>
#include <math.h>
#include <stdint.h>
#include <signal.h>
#include <string.h>

// Features
// #define PREEMPHASIS
#define LPF

#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 0.425 // Adjust this as needed

#define MONO_VOLUME 0.45f // L+R Signal
#define PILOT_VOLUME 0.0225f // 19 KHz Pilot
#define STEREO_VOLUME 0.35f // 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];
    }
}

#define FIR_PHASES 32
#define FIR_TAPS 32

#define PI 3.14159265358979323846
#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;
}

#ifdef PREEMPHASIS
typedef struct {
    float alpha;
    float prev_sample;
} Emphasis;

void init_emphasis(Emphasis *pe, float sample_rate) {
    pe->prev_sample = 0.0f;
    pe->alpha = exp(-1 / (PREEMPHASIS_TAU * sample_rate));
}

float apply_pre_emphasis(Emphasis *pe, float sample) {
    float audio = sample-pe->alpha*pe->prev_sample;
    pe->prev_sample = audio;
    return audio*2;
}
#endif

#ifdef LPF
typedef struct {
    float low_pass_fir[FIR_PHASES][FIR_TAPS];
    float sample_buffer[FIR_TAPS];
    int buffer_index;
} LowPassFilter;

void init_low_pass_filter(LowPassFilter *lp, float sample_rate) {
    for (int i = 0; i < FIR_TAPS; i++) {
        for (int j = 0; j < FIR_PHASES; j++) {
            int mi = i * FIR_PHASES + j + 1;
            float sincpos = mi - (((FIR_TAPS * FIR_PHASES) + 1.0f) / 2.0f);
            float firlowpass = (sincpos == 0.0f) ? 1.0f : sinf(M_2PI * LPF_CUTOFF * sincpos / sample_rate) / (PI * sincpos);
            float window = 0.54f - 0.46f * cosf(M_2PI * mi / (FIR_TAPS * FIR_PHASES)); // Hamming window
            lp->low_pass_fir[j][i] = firlowpass * window;
        }
    }
    memset(lp->sample_buffer, 0, sizeof(lp->sample_buffer));
    lp->buffer_index = 0;
}

float apply_low_pass_filter(LowPassFilter *lp, float sample) {
    // Update the sample buffer
    lp->sample_buffer[lp->buffer_index] = sample;
    lp->buffer_index = (lp->buffer_index + 1) % FIR_TAPS;

    // Apply the filter
    float result = 0.0f;
    int index = lp->buffer_index;
    for (int i = 0; i < FIR_TAPS; i++) {
        result += lp->low_pass_fir[0][i] * lp->sample_buffer[index];
        index = (index + 1) % FIR_TAPS;
    }
    return result*6;
}
#endif

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");
    // 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 = 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,
        "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;
    init_oscillator(&pilot_osc, 19000.0, SAMPLE_RATE); // Pilot, it's there to indicate stereo and as a refrence signal with the stereo carrier
#ifdef PREEMPHASIS
    Emphasis preemp_l, preemp_r;
    init_emphasis(&preemp_l, SAMPLE_RATE);
    init_emphasis(&preemp_r, SAMPLE_RATE);
#endif
#ifdef LPF
    LowPassFilter lpf_l, lpf_r;
    init_low_pass_filter(&lpf_l, SAMPLE_RATE);
    init_low_pass_filter(&lpf_r, SAMPLE_RATE);
#endif

    signal(SIGINT, stop);
    signal(SIGTERM, stop);
    
    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), NULL) < 0) {
            fprintf(stderr, "Error reading from input device.\n");
            break;
        }
        uninterleave(input, left, right, BUFFER_SIZE*2);

        for (int i = 0; i < BUFFER_SIZE; i++) {
            float stereo_carrier = sinf(pilot_osc.phase*2); // Stereo carrier should be a harmonic of the pilot which is in phase, best way to generate the harmonic is to multiply the pilot's phase by two, so it is mathematically impossible for them to not be in phase
            float pilot = get_next_sample(&pilot_osc); // This is after because if it was before then the stereo would be out of phase by one increment, so [GET STEREO] ([GET PILOT] [INCREMENT PHASE])
            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 Sterreo to Mono but a bit diffrent

            mpx[i] = mono * MONO_VOLUME +
                pilot * PILOT_VOLUME +
                (stereo * stereo_carrier) * STEREO_VOLUME; // DSB-SC modulate
        }

        if (pa_simple_write(output_device, mpx, sizeof(mpx), 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;
}