kiss_fftr的KissFFT输出 [英] KissFFT output of kiss_fftr

问题描述

我收到含有320样本分组PCM数据槽承插连接。声音的采样速率是每秒8000个样品。我用它做这样的事情：

I'm receiving PCM data trough socket connection in packets containing 320 samples. Sample rate of sound is 8000 samples per second. I am doing with it something like this:

int size = 160 * 2;//160;
int isinverse = 1;
kiss_fft_scalar zero;
memset(&zero,0,sizeof(zero));
kiss_fft_cpx fft_in[size];
kiss_fft_cpx fft_out[size];
kiss_fft_cpx fft_reconstructed[size];

kiss_fftr_cfg fft = kiss_fftr_alloc(size*2 ,0 ,0,0);
kiss_fftr_cfg ifft = kiss_fftr_alloc(size*2,isinverse,0,0);

for (int i = 0; i < size; i++) {
    fft_in[i].r = zero;
    fft_in[i].i = zero;
    fft_out[i].r = zero;
    fft_out[i].i = zero;
    fft_reconstructed[i].r = zero;
    fft_reconstructed[i].i = zero;
}

// got my data through socket connection

for (int i = 0; i < size; i++) {
     // samples are type of short
     fft_in[i].r = samples[i];
     fft_in[i].i = zero;
     fft_out[i].r = zero;
     fft_out[i].i = zero;
 }

 kiss_fftr(fft, (kiss_fft_scalar*) fft_in, fft_out);
 kiss_fftri(ifft, fft_out, (kiss_fft_scalar*)fft_reconstructed);

 // lets normalize samples
 for (int i = 0; i < size; i++) {
     short* samples = (short*) bufTmp1;
     samples[i] = rint(fft_reconstructed[i].r/(size*2));
 }

在我填写的OpenAL缓冲区并播放它们。一切工作得很好，但我想这样做 kiss_fftr 和 kiss_fftri 之间的音频的一些过滤。出发点，我认为这是声音从时域转换到频域，但我真的不明白我从 kiss_fftr 函数接收什么样的数据。什么样的信息存储在每个那些复数，是什么它真正的部和虚部能告诉我有关的频率。我不知道哪个频率覆盖（什么频率范围）的 fft_out - 哪些索引对应于频率

After that I fill OpenAL buffers and play them. Everything works just fine but I would like to do some filtering of audio between kiss_fftr and kiss_fftri. Starting point as I think for this is to convert sound from time domain to frequency domain, but I don't really understand what kind of data I'm receiving from kiss_fftr function. What information is stored in each of those complex number, what its real and imaginary part can tell me about frequency. And I don't know which frequencies are covered (what frequency span) in fft_out - which indexes corresponds to which frequencies.

我总新手信号处理和傅立叶变换主题。

I am total newbie in signal processing and Fourier transform topics.

任何帮助吗？

推荐答案

在用双脚跳成C实现之前，熟悉数字滤波器，ESP的 FIR过滤器。

Before you jump in with both feet into a C implementation, get familiar with digital filters, esp FIR filters.

您可以设计使用类似 GNU八度的信号工具箱中的FIR滤波器。看看命令FIR1（简单），firls或雷米兹。或者，你也许可以通过网页来设计FIR滤波器。快速的Web搜索在线FIR滤波器设计发现这（我没有用它，但它似乎使用了雷米兹或firpm命令中使用的等波纹设计）

You can design the FIR filter using something like GNU Octave's signal toolbox. Look at the command fir1(the simplest), firls, or remez. Alternately, you might be able to design a FIR filter through a web page. A quick web search for "online fir filter design" found this (I have not used it, but it appears to use the equiripple design used in the remez or firpm command )

尝试用直接卷积（不含FFT）算法实现第一次过滤器，看看速度是可以接受的 - 这是一个更容易的路径。如果你需要一个基于FFT的方法，还有在kissfft /工具/ kiss_fastfir.c文件的一个样本实现重叠保留。

Try implementing your filter first with a direct convolution (without FFTs) and see if the speed is acceptable -- that is an easier path. If you need an FFT-based approach, there is a sample implementation of overlap-save in the kissfft/tools/kiss_fastfir.c file.

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