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二维小波变换
- 关于二维小波变换的程序 [精华] 说明:此算法重在概念,速度并不是很快。因为FOR循环的缘故。此程序从循环矩阵的观点出发,把圆周卷积和快速幅里叶变换建立了联系。实现了分解和无失真重构。它只做了一层分解,即将256x256图形分解成为64x64的四个图形,避免了使用WKEEP()的困惑。主要思想为用小波滤波器族构造正交阵W,变换写为B=W*A*W ,反变换为:A=W *A*W,这与所有正交变换无异。W为循环正交矩阵,因此可用FFT实现快速运算,难点就在重构矩阵上。若用矩阵概念明确,一个
srfft
- 我的一个分裂基快速FFT变换程序.不知道有没有错-I split a fast FFT-based procedures. Do not know if it is right or wrong
CXIMAGE_SRC.ZIP
- Windows下图象处理。可读取/保存常见的多种图象格式,如bmp,jpg,jpeg,png,pcx,tif,tga,gif,wmf,emf,wbmp,可做常规的图象处理,如缩放,裁减,翻转,镜像,旋转,柔化,锐化,亮度,对比度,模糊,伪彩,浮雕,边缘检测,中值滤波,不同图片的颜色通道分离,合并,gamma,fft变换处理等,有多文档窗口的demo程序,demo本身就能算是个功能完善的图象处理工具,能够学到很多图象处理知识。-Windows image processing. Reads / p
DSPHOMEWORK
- 设信号 ,用 对x(t)采样得x(n),是否会发生频谱混叠?现利用FFT分析其频谱。 1.编程绘制该信号的波形。 2.若令N=16,编程对x(n)做FFT运算,并绘制其幅频特性曲线。 3.令N=1024,编程对x(n)做FFT运算,并绘制其幅频特性曲线。 4.分析2、3的运算结果。 设计调试报告要求: 1.工作原理简述; 2.设计思路; 3.难点及解决方法; 4.设计、调试结果及分析; 5.程序文本及操作步骤。-based signal
FFTfrequencychartisanalytical
- 1.通过实验加深对快速傅立叶变换(FFT)基本原理的理解。 2.了解FFT点数与频谱分辨率的关系,以及两种加长序列FFT与原序列FFT的关系。 离散傅里叶变换(DFT)和卷积是信号处理中两个最基本也是最常用的运算,它们涉及到信号与系统的分析与综合这一广泛的信号处理领域。实际上卷积与DFT之间有着互通的联系:卷积可化为DFT来实现,其它的许多算法,如相关、滤波和谱估计等都可化为DFT来实现,DFT也可化为卷积来实现。-1. Deepen the experimental fast Fou
multiratefliter
- 设模拟带通信号频率范围1KHz—1.1KHz,设计一个采样频率8KHz数字频谱分析系统,要求频率分辨率0.1Hz。若对信号直接用FFT进行频谱分析-simulation based communications, with the frequency range 1KHz - 1.1KHz. Design of an 8 KHz sampling frequency digital spectrum analysis system requirements frequency resolutio
2FFT
- DIT-FFT算法子程序 FFT长度为2m,m为正整数。FFT输出结果放在输入复数数组中。其中xr存放信号序列实部,xi存放信号序列虚部。如果信号长度小于N,需要给xr和xi后面补足0,计算结果x(k)的实部和虚部仍然分别存放在数组xr和xi中。-DIT - FFT subroutine FFT length of 2m, m is a positive integer. FFT output on the importation of complex array. Xr which ke
DIT-FFTsuanfazichengxu
- DIT-FFT算法子程序 FFT长度为2m,m为正整数。FFT输出结果放在输入复数数组中。其中xr存放信号序列实部,xi存放信号序列虚部。如果信号长度小于N,需要给xr和xi后面补足0,计算结果x(k)的实部和虚部仍然分别存放在数组xr和xi中。-DIT - FFT subroutine FFT length of 2m, m is a positive integer. FFT output on the importation of complex array. Xr which ke
myfft_ifft
- FFT and IFFT for 16-bits wave, you can use this instead of convolution calculate, input samples must be 2^n. If its not 2^n samples just stuff zero in last array. I already make it to C++ class you can new a calss and do it easy
cdaudioxsrc
- 这是一个数字信号处理的函数库,包括FFT变换,各种滤波算法, 如果你要编写数字信号处理相关的程序,这是个不错的选择-This a digital signal processing functions, including FFT, various filtering algorithm, if you want to prepare digital signal processing related to the procedure, which is not a bad choice
siginalprosessing.zip
- 这是一个数字信号处理的函数库,包括FFT变换各种滤波算法如果你要编写数字信号处理相关的程序这是个不错的选择,This is a digital signal processing library, including all kinds of filtering algorithm FFT transform you to be prepared if the digital signal processing-related procedures This is a good choice
multichanneltest.zip
- 完成飛行情報區交叉可以高保真的夢想,它的大幅過濾器和線性信號階段。 but current FFT type of FIR crossover on PC loses time information of audio signal.但目前的FFT類型的FIR交叉在PC失去時間信息的音頻信號。 If you use 10ms frame, time information in 10ms are lost.如果您使用10ms幀,在10ms時間信息都將丟失。 Now nVidia CUDA G
15883865Wojjj
- FFT算法研究,看看对要使用VC++进行数字信号处理的朋友还是很实用的-FFT algorithm to see if want to use VC++ For digital signal processing of a friend or a very practical
933174vanessa
- FFT算法研究,看看对要使用VC++进行数字信号处理的朋友还是很实用的-FFT algorithm to see if want to use VC++ For digital signal processing of a friend or a very practical
fftRound2
- fft的东西,自己要是用的自己看看,希望对你有用-fft things that they could see for himself if used, in the hope that useful to you
soundstudiocs
- very nice fft h. System will automatically delete the directory of debug and release, so please do not put files on these two directory. If you have not registered, Please [regist first].You should upload at least five so
VCS
- code FFT for matlab I do project about voice classification system. But, I can t continuous coding. If you understan my project, you will contect to me. Thank.-code FFT for matlab I do project about voice classification system. But, I can
FM_phase_noise_
- The file calculates and plots FM noise sidebands for a carrier. It also does sinusoidal modulation. This simple way of adding noise to a carrier is useful for simulation of PLLs. It turns out, though, that the mean of the randn function is not as clo
digital-signal-processing-application-in-21xx
- 这是一本关于在ADI ADSP 21XX系列上开发数字信号算法应用的讲解,涉及浮点、定点,一维,二维FFT/DCT等算法!即使对于其他DSP开发平台的工程师我想也有不少帮助。-This is a book about the ADI ADSP 21XX series of digital signal algorithm is applied to develop explanations, involving floating-point, fixed point, one-dimension
fft_ana
- 给定了一个初始函数,利用FFT进行频谱分析,如果有自己的采样数据,只需设置采样频率即可实现频谱分析。(Given an initial function, spectrum analysis is carried out by using FFT. If there is one's own sampling data, only setting sampling frequency can realize spectrum analysis.)