C实现读取音频然后对音频进行FFt后输出

以下是一个简单的C程序，可以读取一个WAV文件，对其进行FFT变换，并输出变换结果。

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

#define PI 3.14159265358979323846

typedef struct WAVHeader {
    char chunkID[4];
    int chunkSize;
    char format[4];
    char subchunk1ID[4];
    int subchunk1Size;
    short audioFormat;
    short numChannels;
    int sampleRate;
    int byteRate;
    short blockAlign;
    short bitsPerSample;
    char subchunk2ID[4];
    int subchunk2Size;
} WAVHeader;

void readWAVHeader(FILE* file, WAVHeader* header) {
    fread(header, sizeof(WAVHeader), 1, file);
}

void printWAVHeader(WAVHeader* header) {
    printf("Chunk ID: %c%c%c%c\n", header->chunkID[0], header->chunkID[1], header->chunkID[2], header->chunkID[3]);
    printf("Chunk Size: %d\n", header->chunkSize);
    printf("Format: %c%c%c%c\n", header->format[0], header->format[1], header->format[2], header->format[3]);
    printf("Subchunk1 ID: %c%c%c%c\n", header->subchunk1ID[0], header->subchunk1ID[1], header->subchunk1ID[2], header->subchunk1ID[3]);
    printf("Subchunk1 Size: %d\n", header->subchunk1Size);
    printf("Audio Format: %d\n", header->audioFormat);
    printf("Num Channels: %d\n", header->numChannels);
    printf("Sample Rate: %d\n", header->sampleRate);
    printf("Byte Rate: %d\n", header->byteRate);
    printf("Block Align: %d\n", header->blockAlign);
    printf("Bits Per Sample: %d\n", header->bitsPerSample);
    printf("Subchunk2 ID: %c%c%c%c\n", header->subchunk2ID[0], header->subchunk2ID[1], header->subchunk2ID[2], header->subchunk2ID[3]);
    printf("Subchunk2 Size: %d\n", header->subchunk2Size);
}

void readWAVData(FILE* file, short* data, int numSamples) {
    fread(data, sizeof(short), numSamples, file);
}

void printWAVData(short* data, int numSamples) {
    for (int i = 0; i < numSamples; i++) {
        printf("%d\n", data[i]);
    }
}

void fft(double* data, int n) {
    if (n <= 1) return;
    double* even = (double*) malloc(n/2 * sizeof(double));
    double* odd = (double*) malloc(n/2 * sizeof(double));
    for (int i = 0; i < n/2; i++) {
        even[i] = data[2*i];
        odd[i] = data[2*i+1];
    }
    fft(even, n/2);
    fft(odd, n/2);
    for (int i = 0; i < n/2; i++) {
        double re = even[i];
        double im = odd[i] * exp(-2 * PI * i / n);
        data[i] = re + im;
        data[i+n/2] = re - im;
    }
    free(even);
    free(odd);
}

int main(int argc, char* argv[]) {
    if (argc != 2) {
        printf("Usage: %s <filename>\n", argv[0]);
        return 1;
    }
    FILE* file = fopen(argv[1], "rb");
    if (file == NULL) {
        printf("Error opening file\n");
        return 1;
    }
    WAVHeader header;
    readWAVHeader(file, &header);
    printWAVHeader(&header);
    int numSamples = header.subchunk2Size / 2;
    short* data = (short*) malloc(numSamples * sizeof(short));
    readWAVData(file, data, numSamples);
    fclose(file);
    double* fftData = (double*) malloc(numSamples * sizeof(double));
    for (int i = 0; i < numSamples; i++) {
        fftData[i] = (double) data[i];
    }
    fft(fftData, numSamples);
    for (int i = 0; i < numSamples; i++) {
        printf("%f\n", fftData[i]);
    }
    free(data);
    free(fftData);
    return 0;
}

在上面的程序中，我们首先定义了一个`WAVHeader`结构体，用于存储WAV文件的头信息。然后定义了一些函数用于读取WAV文件的头信息和数据，以及进行FFT变换。在`main`函数中，我们首先读取WAV头信息，然后读取WAV数据，并将其转换为一个`double`类型的数组。接着，我们对这个数组进行FFT变换，最后将变换结果输出到控制台上。

注意，上面的程序中没有做任何错误处理，比如读取文件失败或者WAV文件格式不正确等情况。在实际使用中，需要做相应的错误处理。另外，在进行FFT变换时，需要注意数组长度必须是2的整数次幂，否则需要进行补零操作。