vs2019C语言fopen_s通过文件名读WAV文件内的数据,统计时域信号的均值、均方值、方差,计算信号的自协方差矩阵
时间: 2024-05-11 19:14:48 浏览: 104
下面是一个简单的示例代码,可以通过文件名读取WAV文件内的数据,并计算信号的时域统计量和自协方差矩阵:
```c
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
#include <math.h>
#define BUFFER_SIZE 1024
struct WaveHeader {
char chunkId[4];
uint32_t chunkSize;
char format[4];
char subchunk1Id[4];
uint32_t subchunk1Size;
uint16_t audioFormat;
uint16_t numChannels;
uint32_t sampleRate;
uint32_t byteRate;
uint16_t blockAlign;
uint16_t bitsPerSample;
char subchunk2Id[4];
uint32_t subchunk2Size;
};
int main(int argc, char *argv[]) {
if (argc < 2) {
printf("Usage: %s <filename>\n", argv[0]);
return 1;
}
FILE *fp = NULL;
fopen_s(&fp, argv[1], "rb");
if (fp == NULL) {
printf("Error opening file %s\n", argv[1]);
return 1;
}
// Read WAV header
struct WaveHeader header;
fread(&header, sizeof(header), 1, fp);
if (strncmp(header.chunkId, "RIFF", 4) != 0 ||
strncmp(header.format, "WAVE", 4) != 0 ||
strncmp(header.subchunk1Id, "fmt ", 4) != 0 ||
header.audioFormat != 1) {
printf("Invalid WAV file format\n");
fclose(fp);
return 1;
}
// Read audio data
uint32_t numSamples = header.subchunk2Size / (header.bitsPerSample / 8);
float *data = (float *) malloc(numSamples * sizeof(float));
int16_t buffer[BUFFER_SIZE];
int readSize = 0;
uint32_t i = 0;
while ((readSize = fread(buffer, sizeof(int16_t), BUFFER_SIZE, fp)) > 0) {
for (int j = 0; j < readSize; j++) {
data[i++] = buffer[j] / 32768.0f;
}
}
fclose(fp);
// Calculate time domain statistics
float mean = 0.0f, variance = 0.0f, rms = 0.0f;
for (i = 0; i < numSamples; i++) {
mean += data[i];
rms += data[i] * data[i];
}
mean /= numSamples;
rms = sqrtf(rms / numSamples);
for (i = 0; i < numSamples; i++) {
variance += (data[i] - mean) * (data[i] - mean);
}
variance /= numSamples;
printf("Mean: %f\n", mean);
printf("RMS: %f\n", rms);
printf("Variance: %f\n", variance);
// Calculate autocovariance matrix
uint32_t maxLag = 100;
float *autoCov = (float *) malloc((maxLag + 1) * (maxLag + 1) * sizeof(float));
for (uint32_t m = 0; m <= maxLag; m++) {
for (uint32_t n = 0; n <= maxLag; n++) {
float sum = 0.0f;
for (i = 0; i < numSamples - m; i++) {
sum += (data[i] - mean) * (data[i + m] - mean);
}
autoCov[m * (maxLag + 1) + n] = sum / (numSamples - m);
if (m != n) {
autoCov[n * (maxLag + 1) + m] = autoCov[m * (maxLag + 1) + n];
}
}
}
// Print autocovariance matrix
for (uint32_t m = 0; m <= maxLag; m++) {
for (uint32_t n = 0; n <= maxLag; n++) {
printf("%f ", autoCov[m * (maxLag + 1) + n]);
}
printf("\n");
}
free(data);
free(autoCov);
return 0;
}
```
该代码首先通过 `fopen_s` 函数打开指定的 WAV 文件,并读取 WAV 文件头部信息。然后,它将 WAV 文件中的音频数据读入到一个浮点数数组中,将每个采样值除以 32768.0,以将其归一化到 [-1, 1] 的范围内。接下来,它计算了时域统计量(均值、均方值和方差),并使用这些统计量计算了自协方差矩阵。最后,它打印了自协方差矩阵的值。需要注意的是,该代码的自协方差矩阵是一个对称矩阵,并且仅计算了最多 100 个滞后样本的值。如果需要更高的分辨率和/或更大的滞后样本数量,请相应地更改 `maxLag` 变量的值。
阅读全文