Description: 生成一个正弦信号，并加入白噪声，得到正弦信号以及白噪声的混合信号，通过低通滤波器对白噪声进行处理-generate a sinusoidal signal, and add white noise, sinusoidal signal to be white noise and mixed-signal, through low-pass filter white noise processing Platform: |
Size: 66560 |
Author:谢晓丹 |
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Description: IIR-butter低通滤波器设计，数字信号处理专业学生的课程设计。-IIR-butter low-pass filter design, digital signal processing professional students in curriculum design. Platform: |
Size: 30720 |
Author:qjyong |
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Description: 这是用三角窗设计的FIR低通滤波器，来消除音乐信号的噪声-This is the triangular window design FIR low-pass filter to eliminate the noise signal Music Platform: |
Size: 3072 |
Author:xiaochong |
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Description: DFT进行频谱分析时的三种现象
1、混叠：对连续信号采样，要求连续信号是带限的，采样频率要足够高。Fs应满足Nyquist采样定理才不产生混叠。
采样前加低通滤波器防混叠
2、频谱泄漏：DFT对时域信号进行了截断（相当于采样信号乘了一个窗函数），时域的乘积相当于频域卷积，改变了原信号频谱。
改用其他窗函数（非矩形）改善泄漏。
3、栅栏现象：DFT 只给出了频谱在采样点上的取值，采样点间的频谱内容丢失。-DFT spectrum analysis of a three phenomena, aliasing : continuous signal sampling, requiring continuous signal is band-limited, the sampling frequency should be high enough. Fs meet the Nyquist Sampling Theorem would not have aliasing. Sampling former low-pass filter plus anti-aliasing 2, spectrum leakage : DFT right time domain signal a disconnect (equivalent to a sampling signal by a window function). time domain is equivalent to the product of frequency-domain convolution, changes in the spectrum of the original signal. Window functions to switch to other (non-rectangular) to improve leakage. 3, fenced phenomenon : DFT spectrum is given only at the sampling point value, sampling points of the spectrum as lost. Platform: |
Size: 148480 |
Author:qinyang |
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Description: 这片是写的数据信号处理可查那个设计的报告，报告包括卷积演示程序、演示采样定理、模拟滤波器设计演示—从模拟低通滤波器到模拟高通、带通、带阻的幅度
特性对比演示、切比雪夫I型低通滤波器、利用凯塞窗设计高通滤波器、使用双线性变换法设计巴特沃斯低通数字滤波器 这几个部分..........用matlab实现
-this piece is written data signal processing design can be found that the report, including the convolution Demonstration Program, Presentations sampling theorem, Analog Filter Design demo-from analog low-pass filter to simulate high-pass, band-pass, with the rate of resistance properties compared demo, I Chebyshev low-pass filter, the use of Kaiser window design high-pass filter, use bilinear transform design Butterworth low-pass digital filter these parts .......... used mat Implementation lab Platform: |
Size: 325632 |
Author:风 |
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Description: 实现一个数字信号处理的仿真系统 。要求具有界面并实现以下功能： 1）能产生并选择各种数字信号（sin、方波、三角波）；2）用滤波器实现低通、高通、带通和带阻滤波；3）得到输出信号的频域特性和时间序列。
-The realization of a digital signal processing simulation system. Require an interface and realize the following functions: 1) to produce and select a variety of digital signal (sin, square wave, triangular wave) 2) low-pass filter, high pass, band pass and band-stop filter 3) to be output frequency domain characteristics of signals and time series. Platform: |
Size: 11264 |
Author:袁峰 |
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Description: 1．产生一个连续信号，包含低频，中频，高频分量，对其进行采样，进行频谱分析，分别设计三种高通，
低通，带通滤波器对信号进行滤波处理，观察滤波后信号的频谱。
2.采集一段含有噪音的语音信号（可以录制含有噪音的信号，或者录制语音后再加进噪音信号），对其进行
采样和频谱分析，根据分析结果设计出一合适的滤波器滤除噪音信号-1. Produce a continuous signal, including low-frequency, medium frequency, high-frequency components, its sampling, spectral analysis, are designed for three high-pass, low pass, band-pass filter the signal filtering treatment to observe the signal spectrum after filtering. 2. Acquisition section of the speech signal contains noise (can record a signal containing noise, or voice recording, additional noise after the signal), its sampling and spectral analysis, based on an analysis of the results to design a suitable filter to filter out one noise signal Platform: |
Size: 1024 |
Author: |
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Description: 本程序界面实现的是在MATLAB下的语音信号处理,采用是巴特沃斯低通滤波器-The program interface is in the MATLAB realize under the voice signal processing, is the use of Butterworth low-pass filter Platform: |
Size: 4096 |
Author:chengdu |
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Description: 小波变换的一级分解过程是，原始信号分别进行低通、高通滤波，再分别进行二元下抽样，就得到低频、高频（也称为平均、细节）两部分系数；而多级分解则是对上一级分解得到的低频系数再进行小波分解，是一个递归过程。-Wavelet decomposition level is, the original signal were low-pass, high pass filter, and then carried out under the binary sample, we obtained low-frequency, high-frequency (also known as the average, details) coefficient of two parts and multi-level decomposition it is the decomposition level to be low-frequency coefficients further wavelet decomposition is a recursive process. Platform: |
Size: 7420928 |
Author:王正 |
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Description: 这个是我数字信号处理这门课的课程设计,文件包括用matlab编写的代码和设计报告.这个设计包括卷积演示程序、采样定理演示程序、模拟滤波器设计演示程序、设计切比雪夫I型低通滤波器、切比雪夫I型低通滤波器、双线性变换法设计巴特沃斯低通数字滤波器、用凯塞窗设计高通滤波器.-This is my digital signal processing the course of the curriculum design, preparation of documents including the use of matlab code and design report. Convolution demonstration of this design include the procedures, sampling theorem demo program, analog filter design demo program, design Chebyshev-I low-pass filter, Chebyshev type I low-pass filter, bilinear transform design Butterworth low-pass digital filter with Kaiser window design high-pass filter. Platform: |
Size: 264192 |
Author:蔡泽鹏 |
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Description: 输入自己事先录制好的语音信号，该程序实现将语音信号通过FIR低通、高通、带通滤波器后还原出来的语音质量-Enter their own pre-recorded speech signal, the program will realize voice signal through the FIR low-pass, high pass, band-pass filter to restore them after the voice quality Platform: |
Size: 150528 |
Author:shelly |
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Description: ％ y=filter(sig,fmin,fmax)
％ 对信号进行理想滤波，可完成低通，带通
％ sig为被滤波信号
％ fmin为下限频率，当fmin为0时为低通滤波器，当fmin>0时，为带通滤波器
％ fmax为上限频率，fmax>fmin
％ fs为信号的采样频率
％ y为经过滤波后的信号
-％ Y = filter (sig, fmin, fmax)％ of the ideal signal filtering, to be completed by low-pass, band-pass filter％ sig for signal％ fmin is the minimum frequency, when fmin for 0:00 for the low-pass filter, when fmin > 0, for the bandpass filter％ fmax is the upper limit frequency, fmax> fmin％ fs for signal sampling frequency％ y for the filtered signal Platform: |
Size: 1024 |
Author:liu |
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Description: 这个是对一个一维信号的Excel数据加载进matlab程序，然后进行巴特沃斯的低通滤波。-This is a one-dimensional signal of Excel data is loaded into the matlab program, and then Butterworth low-pass filtering. Platform: |
Size: 5120 |
Author:廖勇 |
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Description: 设计一个IIR数字低通滤波器， 逼近一组模拟滤波器的指标参数（通带截止频率Wp=2*pi*2000rad/s,阻带边界频率Ws=2*pi*3000rad/s,通带波纹 Rp=3db, 阻带衰减Rs=15db, 采样频率f=10000Hz）； 分别用脉冲响应不变法和双线性变换法实现设计，列出传递函数并描绘模拟和数字滤波器的幅频和相频响应曲线。用上述设计滤波器完成几组给定信号的滤波，证明滤波器的有效性和滤波范围限制.-Design a IIR digital low-pass filter, approaching a group of analog filters indicator parameters (pass-band cut-off frequency Wp = 2* pi* 2000rad/s, stop-band edge frequency Ws = 2* pi* 3000rad/s, pass-band ripple Rp = 3db, stop-band attenuation Rs = 15db, the sampling frequency f = 10000Hz) were used to change the impulse response method and bilinear transformation method for design, list and describe the transfer function of analog and digital filter amplitude-frequency and phase-frequency response curve. A filter was designed with the completion of the above-mentioned groups of a given signal filtering, to prove the effectiveness of filters and filter limit Platform: |
Size: 71680 |
Author:刘文珍 |
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Description: 基于MATLAB的语音滤波器，通过MATLAB使原来的语音信号分别进行带通，高通，低通滤波，得到更好的效果-MATLAB-based voice filter through the MATLAB so that the original speech signal, respectively, for band-pass, high pass, low pass filter, to get better results Platform: |
Size: 2048 |
Author:柳沐璇 |
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Description: 使用matlab设计一个巴特沃斯低通数字滤波器，使用它对一个混合信号进行滤波，加深对低通滤波器的理解-Using matlab to design a Butterworth low-pass digital filter, use it to a mixed-signal filtering, to deepen the understanding of the low-pass filter Platform: |
Size: 40960 |
Author:jihangui |
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Description: It's a pass filter that based on matlab platform can remove noise from orignal signal Platform: |
Size: 8677376 |
Author:LESLIEWEI
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Search - low pass filter signal matlab