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filter
- filter about FFT and microsoft wave audio
highpassfilter
- 问题是高切滤波或低通滤波，如果采用硬件实现，速度最快，但是现在计算机速度这么快，硬件实现起来相对麻烦，成本高，且灵活性性差， 软件实现相对方便多了，具体体的操作步骤如下（频率域） 先对数据作正FFT,再对变换后的频域数据乘以滤波因子，滤波因子的选择有理想高通、巴特沃斯高通、指数高通、梯形高通，一般选择梯形高通滤波因子，具体原因你可从数字信号处理的书上找到，，接下来将数据再作逆FFT就是了，还有就是在空间域作滤波，这里有一个带同滤波的程序的程序，你可参考一下，（将低切参数设为零就是
FIRIIR
- 本文档里含有诸多c++文件，其功能实现有FIR滤波，IIR滤波，，还有正弦曲线，还实现了128点FFT，DFT-This document contains a lot of c++ file, its function to achieve a FIR filter, IIR filtering, as well as the sine curve, but also to achieve a 128-point FFT, DFT
FFTlvbo
- 对信号进行FFT变换，再将特定的频率段滤波滤除，最后描绘出滤波前后的振幅谱以及滤波后的时间域信号。-FFT transformation on the signal, then the specific frequency segment filter to filter out, and finally depict a time domain signal after the amplitude spectrum before and after filtering and filterin
pmffft.pdf
- 采用匹配滤波器和FFT结合的伪码捕获方案-Matched filter and FFT using a combination of pseudo-code acquisition scheme
fft_fir_filter
- FFT-based FIR Filter is a unit to perform the finite impulse responce filter based on the Fast Fourier Transform (FFT). It performs the convolution of the unlimited signal sequence with the synthesized impulse responce of the length of Ni=N/2 samples
rectangular
- 信号经fft变换后通过一个矩形窗滤波器，再通过ifft变换到时域-After the signal is converted by a fft rectangular window filter, and then transformed into the time domain by ifft
Filter-by-using-DSP-Lib
- stm32 cortex-m4的dsp库，包括各种滤波算法，例如FIR、IIR以及FFT等-dsp library of stm32 with cortex-m4 core which includes varies of algorithm like fir/IIR and fft etc
SPPP3.0PUserPGuide
- SP++3.0手册，C++库的信号处理库，包括信号产生，FFT，滤波器设计，频谱分析，小波变换-SP++ 3.0 Manual, C++ library signal processing including signal generation, FFT, filter design, spectrum analysis, wavelet transform