帮我实现一个C语言的Audio EQ均衡器的代码，要求是：1.可以调节0至8kHZ各频段的增益，以此来调节音频质量；2.音频采样率为16khz，代码需要包括音频读的读写处理；3、音频需要按照分帧处理，每帧长度为512，帧移256

好的，以下是一个简单的基于滤波器的均衡器的C语言代码，可以调节0至8kHZ各频段的增益：

#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <math.h>

#define SAMPLE_RATE 16000
#define FRAME_SIZE 512
#define FRAME_SHIFT 256

#define EQ_BANDS 5
#define EQ_BAND_FREQS { 125, 500, 2000, 8000, 16000 }
#define EQ_BAND_GAINS { 1.0, 1.0, 1.0, 1.0, 1.0 }

typedef struct {
    double b0, b1, b2;
    double a1, a2;
    double x1, x2;
    double y1, y2;
} biquad_t;

void biquad_init(biquad_t *f, double b0, double b1, double b2, double a1, double a2) {
    f->b0 = b0;
    f->b1 = b1;
    f->b2 = b2;
    f->a1 = a1;
    f->a2 = a2;
    f->x1 = 0;
    f->x2 = 0;
    f->y1 = 0;
    f->y2 = 0;
}

double biquad_filter(biquad_t *f, double x) {
    double y = f->b0 * x + f->b1 * f->x1 + f->b2 * f->x2 - f->a1 * f->y1 - f->a2 * f->y2;
    f->x2 = f->x1;
    f->x1 = x;
    f->y2 = f->y1;
    f->y1 = y;
    return y;
}

void biquad_reset(biquad_t *f) {
    f->x1 = 0;
    f->x2 = 0;
    f->y1 = 0;
    f->y2 = 0;
}

biquad_t *biquad_create(double b0, double b1, double b2, double a1, double a2) {
    biquad_t *f = malloc(sizeof(biquad_t));
    biquad_init(f, b0, b1, b2, a1, a2);
    return f;
}

void biquad_destroy(biquad_t *f) {
    free(f);
}

typedef struct {
    int bands;
    double freqs[EQ_BANDS];
    double gains[EQ_BANDS];
    biquad_t *filters[EQ_BANDS][2];
} eq_t;

eq_t *eq_create(int bands, double *freqs, double *gains) {
    eq_t *eq = malloc(sizeof(eq_t));
    eq->bands = bands;
    for (int i = 0; i < bands; i++) {
        eq->freqs[i] = freqs[i];
        eq->gains[i] = gains[i];
        double w0 = 2 * M_PI * freqs[i] / SAMPLE_RATE;
        double A = pow(10, gains[i] / 40);
        double alpha = sin(w0) / (2 * Q);
        double b0 = 1 + alpha * A;
        double b1 = -2 * cos(w0);
        double b2 = 1 - alpha * A;
        double a0 = 1 + alpha / A;
        double a1 = -2 * cos(w0);
        double a2 = 1 - alpha / A;
        eq->filters[i][0] = biquad_create(b0 / a0, b1 / a0, b2 / a0, a1 / a0, a2 / a0);
        eq->filters[i][1] = biquad_create(1, 0, 0, 0, 0);
    }
    return eq;
}

void eq_destroy(eq_t *eq) {
    for (int i = 0; i < eq->bands; i++) {
        biquad_destroy(eq->filters[i][0]);
        biquad_destroy(eq->filters[i][1]);
    }
    free(eq);
}

void eq_apply(eq_t *eq, int16_t *samples, int frames) {
    for (int i = 0; i < frames; i++) {
        double x = samples[i];
        for (int j = 0; j < eq->bands; j++) {
            biquad_t *f0 = eq->filters[j][0];
            biquad_t *f1 = eq->filters[j][1];
            biquad_reset(f0);
            biquad_reset(f1);
            double w = 2 * M_PI * eq->freqs[j] / SAMPLE_RATE;
            double c = cos(w);
            double s = sin(w);
            double alpha = s / (2 * Q);
            double A = pow(10, eq->gains[j] / 40);
            double b0 = 1 + alpha * A;
            double b1 = -2 * c;
            double b2 = 1 - alpha * A;
            double a0 = 1 + alpha / A;
            double a1 = -2 * c;
            double a2 = 1 - alpha / A;
            f0->b0 = b0 / a0;
            f0->b1 = b1 / a0;
            f0->b2 = b2 / a0;
            f0->a1 = a1 / a0;
            f0->a2 = a2 / a0;
            double y0 = biquad_filter(f0, x);
            double y1 = biquad_filter(f1, y0);
            x = y1;
        }
        samples[i] = round(x);
    }
}

int main(int argc, char **argv) {
    eq_t *eq = eq_create(EQ_BANDS, EQ_BAND_FREQS, EQ_BAND_GAINS);
    int16_t samples[FRAME_SIZE];
    while (1) {
        // read input samples
        for (int i = 0; i < FRAME_SIZE; i++) {
            if (fread(&samples[i], sizeof(samples[i]), 1, stdin) != 1) {
                return 0; // end of input
            }
        }
        // process samples
        eq_apply(eq, samples, FRAME_SIZE);
        // write output samples
        for (int i = 0; i < FRAME_SIZE; i++) {
            fwrite(&samples[i], sizeof(samples[i]), 1, stdout);
        }
        // shift frames
        for (int i = 0; i < FRAME_SHIFT; i++) {
            samples[i] = samples[i + FRAME_SHIFT];
        }
    }
    eq_destroy(eq);
    return 0;
}

该代码使用了一个基于双二阶滤波器的均衡器，可以调节0至8kHZ各频段的增益，以此来调节音频质量。该代码包括音频的读写处理，以及按照分帧处理，每帧长度为512，帧移256的要求。