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[Software Engineering] adaptivefilter DL : 1: 该文件含有自适应滤波的四篇国外经典文章,其中有连续自适应滤波的结构,无迟延子带自适应滤波器构造,FIR滤波器的时频实现等域.是自适应滤波器构造的经典文献-the adaptive filter paper containing the four foreign classic articles, which are continuous adaptive filter structure, without delay subband adaptive filter structure, FIR filters achieve such time-frequency domain. adaptive filter is constructed of classic literature
Date : 2008-10-13 Size : 2.57mb User : 单昊
[Software Engineering] adaptivefilter DL : 1: 该文件含有自适应滤波的四篇国外经典文章,其中有连续自适应滤波的结构,无迟延子带自适应滤波器构造,FIR滤波器的时频实现等域.是自适应滤波器构造的经典文献-the adaptive filter paper containing the four foreign classic articles, which are continuous adaptive filter structure, without delay subband adaptive filter structure, FIR filters achieve such time-frequency domain. adaptive filter is constructed of classic literature
Date : 2025-12-13 Size : 2.57mb User : 单昊
[Software Engineering] xiaoxingshuzixitong DL : 0: 本课题研究的主要内容是在空间域直接处理图像的基础上，几种典型的变换域快速算法分析与研究及这些快速算法的程序设计实现。还研究了图像增强中的变换域处理方法，主要针对几种典型变换域滤波器及其程序实现；最后对小波变换的算法分析及在图像压缩中的应用进行了理论研究。-This task includes a series of speedy arithmetic analysis and their programme implement in frequency domain after spatial domain.It also includes image enchancement in the frequency domain that is several frequency-domain filter.At last the task includes the arithmetic of wavelet transformation and its application in image copression.
Date : 2025-12-13 Size : 1.08mb User : waa
[Software Engineering] highpassfilter DL : 0: 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你的高切滤波）-The problem is the high cut filter or low pass filter, if the use of hardware, the fastest, but now the computer speed so fast, hardware problems are relatively high cost and poor flexibility, Software is relatively easy to achieve more concrete steps are as follows body (frequency domain) Is the first of the data for FFT, then the transformed frequency domain data multiplied by the filter factor, choose the ideal filter factor of the high-pass, Butterworth high-pass, index of high-pass, high pass ladder, generally choose high-pass ladder filter factor the specific reason you can book on digital signal processing found, then the data further inverse FFT is it, there is in the spatial domain for filtering, there is a filtering process with the same program, you can refer to , (the low-cut parameters set to zero is your high-cut filter)
Date : 2025-12-13 Size : 7kb User : 夏燕苹
[Software Engineering] 111 DL : 0: 录制一段个人的语音信号，并对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；给定滤波器的性能指标，采用双线性变换法和窗函数法设计滤波器，并画出滤波器的频率响应；然后用自己设计的滤波器对采集的信号进行滤波，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号；最后，设计一个信号处理系统界面。-Record a personal voice signal, and recording the signal is sampled draw sampled speech signal time-domain waveform and frequency spectrum for a given filter performance, the bilinear transformation and design of filter window functions and draw the frequency response filter then filter of their own design collection of the signal filtering, to draw the filtered signal in time domain waveforms and frequency spectrum, and filtering the signal before and after comparison, analysis of signal changes playback audio signals Finally, a signal processing system interface design.
Date : 2025-12-13 Size : 6kb User : liudan
[Software Engineering] gabor DL : 0: The real and imaginary components of a complex Gabor filter are phase sensitive, i.e., as a consequence their response to a sinusoid is another sinusoid (see Figure 1.2). By getting the magnitude of the output (square root of the sum of squared real and imaginary outputs) we can get a response that phase insensitive and thus unmodulated positive response to a target sinusoid input (see Figure 1.2). In some cases it is useful to compute the overall output of the two out of phase filters. One common way of doing so is to add the squared output (the energy) of each filter, equivalently we can get the magnitude. This corresponds to the magnitude (more precisely the squared magnitude) of the complex Gabor filter output. In the frequency domain, the magnitude of the response to a particular frequency is simply the magnitude of the complex Fourier transform, i.e.
Date : 2025-12-13 Size : 156kb User : cestrada
[Software Engineering] 54681686 DL : 0: 主要研究了利用重叠滤波理论进行全相位数字滤波器设计。首先介绍了设计 FIR 滤波器的常规方法及其优缺点，然后借助重叠理论提出全相位的概念，继而给出频域全相位数字滤波器(简记为 APFF)结构图并理论推导得到与其等效的其他三种系统。-Research the use of overlapping full-phase digital filter theory, filter design. First introduced the conventional FIR filter design methods and their advantages and disadvantages, and then overlap with the theory put The whole concept of phase, frequency domain and then give the whole phase digital filter (abbreviated as APFF) Structure and Theoretical analysis to be equivalent with the other three systems.
Date : 2025-12-13 Size : 3.13mb User :
[Software Engineering] THE-EKF-IN-THE-FREQUENCY-DOMAIN DL : 0: 频域的卡尔曼滤波器，这个比较有意思，有兴趣的朋友可以看看，是一篇论文-THE EXTENDED KALMAN FILTER IN THE FREQUENCY DOMAIN FOR THE IDENTIFICATION OF MECHANICAL STRUCTURES EXCITED BY SINUSOIDAL MULTIPLE INPUTS
Date : 2025-12-13 Size : 171kb User : ywauto
[Software Engineering] FFTlvbo DL : 0: 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 filtering.
Date : 2025-12-13 Size : 7kb User : pingzi
[Software Engineering] HPF DL : 0: A High Pass filter matlab code for image processing in frequency domain
Date : 2025-12-13 Size : 1kb User : Ahmed Ragib Kafi
[Software Engineering] qun_v19 DL : 0: 最终的权值矩阵就是滤波器的系数，分析了该信号的时域、频域、倒谱，循环谱等,最大信噪比的独立分量分析算法。- The final weight matrix is ??the filter coefficient, Analysis of the signal time domain, frequency domain, cepstrum, cyclic spectrum, etc. SNR largest independent component analysis algorithm.
Date : 2025-12-13 Size : 4kb User : 郭崇
[Software Engineering] Guass-noise DL : 0: 产生一个高斯白噪声利用MATLAB自带的fir1函数产生一个低通滤波器，限制高斯白噪声的带宽，由此产生了视频噪声。利用产生的视频噪声，分别代入噪声调幅干扰的时域表达式，并且进行100次的积累后求平均值，对其进行快速傅里叶变换后，由此画出噪声调幅干扰的频域波形。重复上述步骤，分别代入噪声调频干扰与噪声调相干扰的时域表达式，分别画出其频域波形与时域波形。(Generate a Gauss white noise The fir1 function of MATLAB is used to generate a low-pass filter, which limits the bandwidth of Gauss white noise, and thus produces video noise. Using the generated video noise, the time domain expressions of noise amplitude modulation interference are substituted, and the average value is calculated after 100 times accumulation. After fast Fourier transform, the frequency domain waveform of noise amplitude modulation interference is drawn. Repeat the steps mentioned above, and substitute the time-domain expressions of noise FM interference and noise coherent interference, and draw the frequency-domain waveform and time-domain waveform respectively.)
Date : 2025-12-13 Size : 3kb User : 大丶哥