使用fopen_s怎么求wav的时域信号均值、均方值、方差，计算信号的自协方差矩阵和功率谱

以下是使用fopen_s计算wav文件的时域信号均值、均方值、方差，自协方差矩阵和功率谱的示例代码：

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

#define BUFFER_SIZE 1024

int main()
{
    FILE *fp;
    errno_t err;
    char filename[] = "audio.wav";
    char buffer[BUFFER_SIZE];
    int num_samples, sample_rate, bits_per_sample, num_channels;
    double mean, rms, variance, *signal, **covariance, *power_spectrum;
    int i, j, k;

    err = fopen_s(&fp, filename, "rb");
    if (err != 0)
    {
        printf("Failed to open file %s\n", filename);
        return 1;
    }

    // Read WAV header
    fseek(fp, 22, SEEK_SET);
    fread(&num_channels, sizeof(short), 1, fp);
    fseek(fp, 24, SEEK_SET);
    fread(&sample_rate, sizeof(int), 1, fp);
    fseek(fp, 34, SEEK_SET);
    fread(&bits_per_sample, sizeof(short), 1, fp);
    fseek(fp, 40, SEEK_SET);
    fread(&num_samples, sizeof(int), 1, fp);

    signal = (double*)malloc(num_samples * sizeof(double));
    power_spectrum = (double*)malloc(num_samples / 2 * sizeof(double));
    covariance = (double**)malloc(num_channels * sizeof(double*));
    for (i = 0; i < num_channels; i++)
    {
        covariance[i] = (double*)malloc(num_samples * num_samples * sizeof(double));
    }

    // Read WAV data
    fseek(fp, 44, SEEK_SET);
    for (i = 0; i < num_samples; i++)
    {
        fread(buffer, sizeof(char), bits_per_sample / 8 * num_channels, fp);
        signal[i] = 0;
        for (j = 0; j < num_channels; j++)
        {
            k = j * bits_per_sample / 8;
            signal[i] += (double)(((short)(buffer[k + 1] << 8) | buffer[k]) / 32768.0);
        }
        signal[i] /= num_channels;
    }

    // Calculate signal statistics
    mean = 0;
    for (i = 0; i < num_samples; i++)
    {
        mean += signal[i];
    }
    mean /= num_samples;

    rms = 0;
    for (i = 0; i < num_samples; i++)
    {
        rms += pow(signal[i] - mean, 2);
    }
    rms = sqrt(rms / num_samples);

    variance = 0;
    for (i = 0; i < num_samples; i++)
    {
        variance += pow(signal[i] - mean, 2);
    }
    variance /= num_samples;

    printf("Mean: %lf\n", mean);
    printf("RMS: %lf\n", rms);
    printf("Variance: %lf\n", variance);

    // Calculate covariance matrix
    for (i = 0; i < num_channels; i++)
    {
        for (j = 0; j < num_samples; j++)
        {
            for (k = 0; k < num_samples; k++)
            {
                covariance[i][j * num_samples + k] = (signal[j] - mean) * (signal[k] - mean);
            }
        }
    }

    // Calculate power spectrum
    for (i = 0; i < num_samples / 2; i++)
    {
        power_spectrum[i] = 0;
        for (j = 0; j < num_samples; j++)
        {
            power_spectrum[i] += signal[j] * cos(2 * M_PI * i * j / num_samples);
        }
        power_spectrum[i] /= num_samples;
        power_spectrum[i] = pow(power_spectrum[i], 2);
    }

    // Free memory
    free(signal);
    free(power_spectrum);
    for (i = 0; i < num_channels; i++)
    {
        free(covariance[i]);
    }
    free(covariance);

    // Close file
    fclose(fp);

    return 0;
}

该代码使用了以下步骤来计算wav文件的时域信号均值、均方值、方差，自协方差矩阵和功率谱：

1. 打开wav文件并读取其头部信息，包括采样率、样本数、每个样本的位数和通道数等。
2. 根据采样率和样本数，分配一个double类型的数组来存储音频信号。
3. 从wav文件中读取音频数据，并将每个样本的值存储在上一步分配的数组中。如果wav文件是多通道的，则将每个通道的样本值求平均后再存储在数组中。
4. 计算音频信号的均值、均方值和方差，并输出结果。
5. 分配一个二维double类型的数组来存储音频信号的自协方差矩阵。对于每个通道，将音频信号的每个样本值与所有其他样本值的乘积存储在数组中。
6. 分配一个double类型的数组来存储音频信号的功率谱。对于每个频率，将音频信号的所有样本值与相应的余弦函数的乘积求和，然后除以样本数，然后再平方。
7. 释放分配的内存并关闭wav文件。

注意：该代码只适用于采样率、样本数和位数相同的音频文件。如果需要处理其他格式的音频文件，可能需要进行一些修改。