Search Results

Search found 61 results on 3 pages for 'fourier'.

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

Page 1/3 | 1 2 3  | Next Page >

  • Displaying Fourier transforms in OpenCV

    - by Simonw
    Hi, I'm just learning to use OpenCV and am having a problem with using DFT. I've done a signal processing class which used MatLab, so I'm trying to go through some of the exercises we did in that class. I'm trying to get and display the FT of an image, so I can mask some of the frequencies. I'd like to be able to see the FT, so I know how big to make the mask, but when I tried, I got an image like this: rather than like one of these Am I forgetting a step somewhere? I'm loading the image, converting its type to CV_32FC1, getting the matrix of it, getting the DFT, and then getting turning the resulting matrix back into an image. I'll post the code I'm using if it will be of any help? Or if someone has a link to an example of displaying the FT? I could only find ones which used it for the convolution. EDIT: Did I get the Phase of the image?

    Read the article

  • Simple in-place discrete fourier transform ( DFT )

    - by Adam
    I'm writing a very simple in-place DFT. I am using the formula shown here: http://en.wikipedia.org/wiki/Discrete_Fourier_transform#Definition along with Euler's formula to avoid having to use a complex number class just for this. So far I have this: private void fft(double[] data) { double[] real = new double[256]; double[] imag = new double[256]; double pi_div_128 = -1 * Math.PI / 128; for (int k = 0; k < 256; k++) { for (int n = 0; n < 256; n++) { real[k] += data[k] * Math.Cos(pi_div_128 * k * n); imag[k] += data[k] * Math.Sin(pi_div_128 * k * n); } data[k] = Math.Sqrt(real[k] * real[k] + imag[k] * imag[k]); } } But the Math.Cos and Math.Sin terms eventually go both positive and negative, so as I'm adding those terms multiplied with data[k], they cancel out and I just get some obscenely small value. I see how it is happening, but I can't make sense of how my code is perhaps mis-representing the mathematics. Any help is appreciated. FYI, I do have to write my own, I realize I can get off-the shelf FFT's.

    Read the article

  • spike in my inverse fourier transform

    - by Jon
    I am trying to compare two data sets in MATLAB. To do this I need to filter the data sets by Fourier transforming the data, filtering it and then inverse Fourier transforming it. When I inverse Fourier transform the data however I get a spike at either end of the red data set (picture shows the first spike), it should be close to zero at the start, like the blue line. I am comparing many data sets and this only happens occasionally. I have three questions about this phenomenon. First, what may be causing it, secondly, how can I remedy it, and third, will it affect the data further along the time series or just at the beginning and end of the time series as it appears to from the picture. Any help would be great thanks.

    Read the article

  • DSP - Problems using the inverse Fast Fourier Transform

    - by Trap
    I've been playing around a little with the Exocortex implementation of the FFT, but I'm having some problems. First, after calculating the inverse FFT of an unchanged frequency spectrum obtained by a previous forward FFT, one would expect to get the original signal back, but this is not the case. I had to figure out that I needed to scale the FFT output to about 1 / fftLength to get the amplitudes ok. Why is this? Second, whenever I modify the amplitudes of the frequency bins before calling the iFFT the signal gets distorted at low frequencies. However, this does not happen if I attenuate all the bins by the same factor. Let me put a very simplified example of the output buffer of a 4-sample FFT: // Bin 0 (DC) FFTOut[0] = 0.0000610351563 FFTOut[1] = 0.0 // Bin 1 FFTOut[2] = 0.000331878662 FFTOut[3] = 0.000629425049 // Central bin FFTOut[4] = -0.0000381469727 FFTOut[5] = 0.0 // Bin 3, this is a negative frequency bin. FFTOut[6] = 0.000331878662 FFTOut[7] = -0.000629425049 The output is composed of pairs of floats, each representing the real and imaginay parts of a single bin. So, bin 0 (array indexes 0, 1) would represent the real and imaginary parts of the DC frequency. As you can see, bins 1 and 3 both have the same values, (except for the sign of the Im part), so I guess these are the negative frequency values, and finally indexes (4, 5) would be the central frequency bin. To attenuate the frequency bin 1 this is what I do: // Attenuate the 'positive' bin FFTOut[2] *= 0.5; FFTOut[3] *= 0.5; // Attenuate its corresponding negative bin. FFTOut[6] *= 0.5; FFTOut[7] *= 0.5; For the actual tests I'm using a 1024-length FFT and I always provide all the samples so no 0-padding is needed. // Attenuate var halfSize = fftWindowLength / 2; float leftFreq = 0f; float rightFreq = 22050f; for( var c = 1; c < halfSize; c++ ) { var freq = c * (44100d / halfSize); // Calc. positive and negative frequency locations. var k = c * 2; var nk = (fftWindowLength - c) * 2; // This kind of attenuation corresponds to a high-pass filter. // The attenuation at the transition band is linearly applied, could // this be the cause of the distortion of low frequencies? var attn = (freq < leftFreq) ? 0 : (freq < rightFreq) ? ((freq - leftFreq) / (rightFreq - leftFreq)) : 1; mFFTOut[ k ] *= (float)attn; mFFTOut[ k + 1 ] *= (float)attn; mFFTOut[ nk ] *= (float)attn; mFFTOut[ nk + 1 ] *= (float)attn; } Obviously I'm doing something wrong but can't figure out what or where.

    Read the article

  • Fourier transform software

    - by CFP
    Hello everyone! After spending a lot of time searching for this, I thought that some SuperUser gurus might know the answer :) I'm searching for an open source application to compute an FFT, that could: * Import a list of points from a text file (in any format, I could write conversion scripts if needed), for example 0,1; 1,2; 4,5 * Compute the associated discrete transform, and output the list of coefficients Ideally, it would also display the plot and the associated fourier decomposition on the same graph, to allow comparison, but this is not absolutely needed. It can be either on Windows or on Linux/Unix. Can you think of a solution? Thanks, CFP.

    Read the article

  • help with matlab and Discrete Fourier transform

    - by user504363
    Hi all I have previous experience with Matlab, but the problem that I face some problems in apply a problem in (DSP : Digital signal processing) which is not my study field, but I must finish that problems in days to complete my project. all i want is help me with method and steps of solving this problem in matlab and then I can write the code with myself. the problem is about the signal x(t) = exp(-a*t); 1) what's the Discrete Fourier transform of the sampled signal with sample rate fs 2) if a=1 and fs =1 , plot the amplitude spectrum of sampled signal 3) fix the sampling frequency at fs = 1(hz) [what's it mean ?] and plot the magnitude of the Fourier Transform of the sampled signal at various values of a thanks

    Read the article

  • fourier transform to transpose key of a wav file

    - by tbischel
    I want to write an app to transpose the key a wav file plays in (for fun, I know there are apps that already do this)... my main understanding of how this might be accomplished is to 1) chop the audio file into very small blocks (say 1/10 a second) 2) run an FFT on each block 3) phase shift the frequency space up or down depending on what key I want 4) use an inverse FFT to return each block to the time domain 5) glue all the blocks together But now I'm wondering if the transformed blocks would no longer be continuous when I try to glue them back together. Are there ideas how I should do this to guarantee continuity, or am I just worrying about nothing?

    Read the article

  • How to extract semi-precise frequencies from a WAV file using Fourier Transforms

    - by Seisatsu
    Let us say that I have a WAV file. In this file, is a series of sine tones at precise 1 second intervals. I want to use the FFTW library to extract these tones in sequence. Is this particularly hard to do? How would I go about this? Also, what is the bast way to write tones of this kind into a WAV file? I assume I would only need a simple audio library for the output. My language of choice is C

    Read the article

  • Using GNU Octave FFT functions

    - by CFP
    Hello world! I'm playing with octave's fft functions, and I can't really figure out how to scale their output: I use the following (very short) code to approximate a function: function y = f(x) y = x .^ 2; endfunction; X=[-4096:4095]/64; Y = f(X); # plot(X, Y); F = fft(Y); S = [0:2047]/2048; function points = approximate(input, count) size = size(input)(2); fourier = [fft(input)(1:count) zeros(1, size-count)]; points = ifft(fourier); endfunction; Y = f(X); plot(X, Y, X, approximate(Y, 10)); Basically, what it does is take a function, compute the image of an interval, fft-it, then keep a few harmonics, and ifft the result. Yet I get a plot that is vertically compressed (the vertical scale of the output is wrong). Any ideas? Thanks!

    Read the article

  • Pitch detection and change java

    - by omegas27
    Hello, I'm french so I'm sorry if you have trouble to understand some of my sentences. Aniways, I saw in some topics that the pitch could be fetected thanks to the Fourier transform but I didn't really understand how to implement it. Moreover, I didn't find how to change the pitch of a wav file and if possibl ,a mp3 file I am listening to music using javaSound for the wav and JLayer for the mp3. Thanks

    Read the article

  • convolution in R

    - by user236215
    I tried to do convolution in R directly and using FFTs then taking inverse. But it seems from simple observation it is not correct. Look at this example: # DIRECTLY > x2$xt [1] 24.610 24.605 24.610 24.605 24.610 > h2$xt [1] 0.003891051 0.003875910 0.003860829 0.003845806 0.003830842 > convolve(h2$xt,x2$xt) [1] 0.4750436 0.4750438 0.4750435 0.4750437 0.4750435 # USING INVERSE FOURIER TRANSFORM > f=fft(fft(h2$xt)*fft(x2$xt), inv=TRUE) > Re(f)/length(f) [1] 0.4750438 0.4750435 0.4750437 0.4750435 0.4750436 > Lets take the index 0. At 0, the convolution should simply be the last value of x2$xt (24.610) multiplied by first value of h2$xt (0.003891051) which should give convolution at index 0 = 24.610*0.003891051 = 0.09575877 which is way off from 0.4750436. Am I doing something wrong? Why is the values so different from expected?

    Read the article

  • How to generate a lower frequency version of a signal in Matlab?

    - by estourodepilha.com
    With a sine input, I tried to modify it's frequency cutting some lower frequencies in the spectrum, shifting the main frequency towards zero. As the signal is not fftshifted I tried to do that by eliminating some samples at the begin and at the end of the fft vector: interval = 1; samplingFrequency = 44100; signalFrequency = 440; sampleDuration = 1 / samplingFrequency; timespan = 1 : sampleDuration : (1 + interval); original = sin(2 * pi * signalFrequency * timespan); fourierTransform = fft(original); frequencyCut = 10; %% Hertz frequencyCut = floor(frequencyCut * (length(pattern) / samplingFrequency) / 4); %% Samples maxFrequency = length(fourierTransform) - (2 * frequencyCut); signal = ifft(fourierTransform(frequencyCut + 1:maxFrequency), 'symmetric'); But it didn't work as expected. I also tried to remove the center part of the spectrum, but it wielded a higher frequency sine wave too. How to make it right?

    Read the article

  • DSP - Filtering in the frequency domain via FFT

    - by Trap
    I've been playing around a little with the Exocortex implementation of the FFT, but I'm having some problems. Whenever I modify the amplitudes of the frequency bins before calling the iFFT the resulting signal contains some clicks and pops, especially when low frequencies are present in the signal (like drums or basses). However, this does not happen if I attenuate all the bins by the same factor. Let me put an example of the output buffer of a 4-sample FFT: // Bin 0 (DC) FFTOut[0] = 0.0000610351563 FFTOut[1] = 0.0 // Bin 1 FFTOut[2] = 0.000331878662 FFTOut[3] = 0.000629425049 // Bin 2 FFTOut[4] = -0.0000381469727 FFTOut[5] = 0.0 // Bin 3, this is the first and only negative frequency bin. FFTOut[6] = 0.000331878662 FFTOut[7] = -0.000629425049 The output is composed of pairs of floats, each representing the real and imaginay parts of a single bin. So, bin 0 (array indexes 0, 1) would represent the real and imaginary parts of the DC frequency. As you can see, bins 1 and 3 both have the same values, (except for the sign of the Im part), so I guess bin 3 is the first negative frequency, and finally indexes (4, 5) would be the last positive frequency bin. Then to attenuate the frequency bin 1 this is what I do: // Attenuate the 'positive' bin FFTOut[2] *= 0.5; FFTOut[3] *= 0.5; // Attenuate its corresponding negative bin. FFTOut[6] *= 0.5; FFTOut[7] *= 0.5; For the actual tests I'm using a 1024-length FFT and I always provide all the samples so no 0-padding is needed. // Attenuate var halfSize = fftWindowLength / 2; float leftFreq = 0f; float rightFreq = 22050f; for( var c = 1; c < halfSize; c++ ) { var freq = c * (44100d / halfSize); // Calc. positive and negative frequency indexes. var k = c * 2; var nk = (fftWindowLength - c) * 2; // This kind of attenuation corresponds to a high-pass filter. // The attenuation at the transition band is linearly applied, could // this be the cause of the distortion of low frequencies? var attn = (freq < leftFreq) ? 0 : (freq < rightFreq) ? ((freq - leftFreq) / (rightFreq - leftFreq)) : 1; // Attenuate positive and negative bins. mFFTOut[ k ] *= (float)attn; mFFTOut[ k + 1 ] *= (float)attn; mFFTOut[ nk ] *= (float)attn; mFFTOut[ nk + 1 ] *= (float)attn; } Obviously I'm doing something wrong but can't figure out what. I don't want to use the FFT output as a means to generate a set of FIR coefficients since I'm trying to implement a very basic dynamic equalizer. What's the correct way to filter in the frequency domain? what I'm missing? Thanks in advance.

    Read the article

  • Difficulty creating classes and arrays of those classes C#

    - by Lucifer Fayte
    I'm trying to implement a Discrete Fourier Transformation algorithm for a project I'm doing in school. But creating a class is seeming to be difficult(which it shouldn't be). I'm using Visual Studio 2012. Basically I need a class called Complex to store the two values I get from a DFT; The real portion and the imaginary portion. This is what I have so far for that: using System; using System.Collections.Generic; using System.Linq; using System.Text; using System.Threading.Tasks; namespace SoundEditor_V3 { public class Complex { public double real; public double im; public Complex() { real = 0; im = 0; } } } The problem is that it doesn't recognize the constructor as a constructor, now I'm just learning C#, but I looked it up online and this is how it's supposed to look apparently. It recognizes my constructor as a method. Why is that? Am I creating the class wrong? It's doing the same thing for my Fourier class as well. So each time I try to create a Fourier object and then use it's method...there is no such thing. example, I do this: Fourier fou = new Fourier(); fou.DFT(s, N, amp, 0); and it tells me fou is a 'field' but is used like a 'type' why is it saying that? Here is the code for my Fourier class as well: using System; using System.Collections.Generic; using System.Linq; using System.Text; using System.Threading.Tasks; namespace SoundEditor_V3 { public class Fourier { //FOURIER //N = number of samples //s is the array of samples(data) //amp is the array where the complex result will be written to //start is the where in the array to start public void DFT(byte[] s, int N, ref Complex[] amp, int start) { Complex tem = new Complex(); int f; int t; for (f = 0; f < N; f++) { tem.real = 0; tem.im = 0; for (t = 0; t < N; t++) { tem.real += s[t + start] * Math.Cos(2 * Math.PI * t * f / N); tem.im -= s[t + start] * Math.Sin(2 * Math.PI * t * f / N); } amp[f].real = tem.real; amp[f].im = tem.im; } } //INVERSE FOURIER public void IDFT(Complex[] A, ref int[] s) { int N = A.Length; int t, f; double result; for (t = 0; t < N; t++) { result = 0; for (f = 0; f < N; f++) { result += A[f].real * Math.Cos(2 * Math.PI * t * f / N) - A[f].im * Math.Sin(2 * Math.PI * t * f / N); } s[t] = (int)Math.Round(result); } } } } I'm very much stuck at the moment, any and all help would be appreciated. Thank you.

    Read the article

  • iPhone: CPU power to do DSP/Fourier transform/frequency domain?

    - by mahboudz
    I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and other music recognition apps, and guitar tuner apps out there. However, I don't know what limitations I'll have to deal with. Anyone play around with this area?

    Read the article

  • C socket programming: connect() hangs

    - by Fantastic Fourier
    Hey all, I'm about to rip my hair out. I have this client that tries to connect to a server, everything seems to be fine, using gethostbyname(), socket(), bind(), but when trying toconnect()` it just hangs there and the server doesn't see anything from the client. I know that the server works because another client (also in C) can connect just fine. What causes the server to not see this incoming connection? I'm at the end of my wits here. The two different clients are pretty similar too so I'm even more lost. if (argc == 2) { host = argv[1]; // server address } else { printf("plz read the manual\n"); exit(1); } hserver = gethostbyname(host); if (hserver) { printf("host found: %p\n", hserver); printf("host found: %s\n", hserver->h_name ); } else { printf("host not found\n"); exit(1); } bzero((char * ) &server_address, sizeof(server_address)); // copy zeroes into string server_address.sin_family = AF_INET; server_address.sin_addr.s_addr = htonl(hserver->h_addr); server_address.sin_port = htons(SERVER_PORT); bzero((char * ) &client_address, sizeof(client_address)); // copy zeroes into string client_address.sin_family = AF_INET; client_address.sin_addr.s_addr = htonl(INADDR_ANY); client_address.sin_port = htons(SERVER_PORT); sockfd = socket(AF_INET, SOCK_STREAM, 0); if (sockfd < 0) exit(1); else { printf("socket is opened: %i \n", sockfd); info.sock_fd = sockfd; rv = fcntl(sockfd, F_SETFL, O_NONBLOCK); // socket set to NONBLOCK if(rv < 0) printf("nonblock failed: %i %s\n", errno, strerror(errno)); else printf("socket is set nonblock\n"); } timeout.tv_sec = 0; // seconds timeout.tv_usec = 500000; // micro seconds ( 0.5 seconds) setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, &timeout, sizeof(struct timeval)); rv = bind(sockfd, (struct sockaddr *) &client_address, sizeof(client_address)); if (rv < 0) { printf("MAIN: ERROR bind() %i: %s\n", errno, strerror(errno)); exit(1); } else printf("socket is bound\n"); rv = connect(sockfd, (struct sockaddr *) &server_address, sizeof(server_address)); printf("rv = %i\n", rv); if (rv < 0) { printf("MAIN: ERROR connect() %i: %s\n", errno, strerror(errno)); exit(1); } else printf("connected\n"); Any thoughts or insights are deeply greatly humongously appreciated. -Fourier

    Read the article

  • Finding text orientation in image (angle for rotation)

    - by maximus
    There is an image captured by camera, and I need to find the angle of the text in order to rotate it to make the image better for OCR results. So I know that the fourier transform can be used for that purpose, My question is, does it really gives good results or may be it is better to use something different than that? Can you tell me if there is a good method for this purpose? I am afraid that not every image containing the text will give me a good result after using fourier transform method. Actually, if I make like it is written here: link text (see the part related with an example of text image) calculating the logarithm of the magnitude of the Fourier transform of image with text and then thresholding it, I get that points and I can calculate the line approximately passing through them, and after getting the line calculate the angle, and then make an affine transform, But, what if I do not get a good result every time using this method , and make a false transform? Any ideas please to judge wether the result is correct or not, or may be another method is better? The binary image can contain noise, even if there are not so much of them, the angle found as a result can be not accurate.

    Read the article

  • pdfLaTeX + memoir class compile error

    - by Sebastien
    Hi, I'm in the middle of writing my thesis, and was using KOMA-Script. The document compiles just fine. I stumbled upon the memoir class yesterday, and was thinking of giving it a try, so here I am trying to compile with this class instead of KOMA-Script. First compilation is OK On second compilation, the document would not build (./fourier/fourier.tex [98] ! Undefined control sequence. <argument> ... C\protect \noexpand \protect \bond \protect \noexpand \protec... l.1 \chapter {Homogénéisation numérique par transformée de Fourier rapide} ? It has apparently not connected to hyperlink (btw, I'm using memhfixc), since I've commented this one out. Here is the preamble of my document, any thoughts ? Thanks in advance, S %\documentclass[draft, 11pt, a4paper, chapterprefix]{scrreprt} \documentclass[draft, 11pt, a4paper]{memoir} \usepackage[authoryear, round]{natbib} \usepackage[french]{babel} \usepackage[latin1]{inputenc} \usepackage{pdfsync} \usepackage[version=3]{mhchem} \usepackage{pgf} \usepackage{microtype} \usepackage{txfonts} % Polices times \usepackage[notref, notcite]{showkeys} \usepackage{amsmath} \usepackage{amssymb} \usepackage[bvec]{sbmacros} \usepackage{micromechanics} \usepackage{pgfcad} %\usepackage[breaklinks=true]{hyperref} %\usepackage{memhfixc} % Pour assurer la compatibilité entre memoir et hyperref %\newcommand{\url}[1]{\texttt{#1}} % Options KOMA-Script % \addtokomafont{caption}{\small} % \pagestyle{headings}

    Read the article

  • TCP multicast and multithreading

    - by Fantastic Fourier
    I need to come up with clients that can multicast to other clients reliably. That implies I'll be using TCP to connect reliably between clients within a multicast group. Doesn't that come up to n^2 number of connections? That seems a little silly to me. Wouldn't/shouldn't there be a way to more easily multicast with reliability?

    Read the article

  • C socket programming: calling recv() changes my socket file descriptor?

    - by fourier
    Hey all, I have this strange problem with recv(). I'm programming client/server where client send() a message (a structure to be exact) and server recv() it. I am also working with multiple sockets and select(). while(1) { readset = info->read_set; info->copy_set = info->read_set; timeout.tv_sec = 1; timeout.tv_usec = 0; // 0.5 seconds ready = select(info->max_fd+1, &readset, NULL, NULL, &timeout); if (ready == -1) { printf("S: ERROR: select(): %s\nEXITING...", strerror(errno)); exit(1); } else if (ready == 0) { continue; } else { printf("S: oh finally you have contacted me!\n"); for(i = 0; i < (info->max_fd+1); i++) { if(FD_ISSET(i, &readset)) //this is where problem begins { printf("S: %i is set\n", i); printf("S: we talking about socket %i son\n", i); // i = 4 num_bytes = recv(i, &msg, MAX_MSG_BYTE, 0); printf("S: number of bytes recieved in socket %i is %i\n", i, num_bytes); // prints out i = 0 what?? if (num_bytes == 0) { printf("S: socket has been closed\n"); break; } else if (num_bytes == -1) { printf("S: ERROR recv: %d %s \n", i, strerror(errno)); continue; } else { handle_request(arg, &msg); printf("S: msg says %s\n", msg->_payload); } } // if (FD_ISSET(i, &readset) else printf("S: %i is not set\n", i); } // for (i = 0; i < maxfd+1; i++) to check sockets for msg } // if (ready == -1) info->read_set = info->copy_set; printf("S: copied\n"); } the problem I have is that in read_set, 0~3 aren't set and 4 is. That is fine. But when i call recv(), i suddently becomes 0. Why is that? It doesn't make sense to me why recv() would take an socket file descriptor number and modify to another number. Is that normal? Am I missing something? S: 0 is not set S: 1 is not set S: 2 is not set S: 3 is not set S: 4 is set S: we talking about socket 4 son S: i is strangely or unstrangely 0 S: number of bytes recieved in socket 0 is 40 That's what it prints out.

    Read the article

  • NET Math Libraries

    - by JoshReuben
    NET Mathematical Libraries   .NET Builder for Matlab The MathWorks Inc. - http://www.mathworks.com/products/netbuilder/ MATLAB Builder NE generates MATLAB based .NET and COM components royalty-free deployment creates the components by encrypting MATLAB functions and generating either a .NET or COM wrapper around them. .NET/Link for Mathematica www.wolfram.com a product that 2-way integrates Mathematica and Microsoft's .NET platform call .NET from Mathematica - use arbitrary .NET types directly from the Mathematica language. use and control the Mathematica kernel from a .NET program. turns Mathematica into a scripting shell to leverage the computational services of Mathematica. write custom front ends for Mathematica or use Mathematica as a computational engine for another program comes with full source code. Leverages MathLink - a Wolfram Research's protocol for sending data and commands back and forth between Mathematica and other programs. .NET/Link abstracts the low-level details of the MathLink C API. Extreme Optimization http://www.extremeoptimization.com/ a collection of general-purpose mathematical and statistical classes built for the.NET framework. It combines a math library, a vector and matrix library, and a statistics library in one package. download the trial of version 4.0 to try it out. Multi-core ready - Full support for Task Parallel Library features including cancellation. Broad base of algorithms covering a wide range of numerical techniques, including: linear algebra (BLAS and LAPACK routines), numerical analysis (integration and differentiation), equation solvers. Mathematics leverages parallelism using .NET 4.0's Task Parallel Library. Basic math: Complex numbers, 'special functions' like Gamma and Bessel functions, numerical differentiation. Solving equations: Solve equations in one variable, or solve systems of linear or nonlinear equations. Curve fitting: Linear and nonlinear curve fitting, cubic splines, polynomials, orthogonal polynomials. Optimization: find the minimum or maximum of a function in one or more variables, linear programming and mixed integer programming. Numerical integration: Compute integrals over finite or infinite intervals, over 2D and higher dimensional regions. Integrate systems of ordinary differential equations (ODE's). Fast Fourier Transforms: 1D and 2D FFT's using managed or fast native code (32 and 64 bit) BigInteger, BigRational, and BigFloat: Perform operations with arbitrary precision. Vector and Matrix Library Real and complex vectors and matrices. Single and double precision for elements. Structured matrix types: including triangular, symmetrical and band matrices. Sparse matrices. Matrix factorizations: LU decomposition, QR decomposition, singular value decomposition, Cholesky decomposition, eigenvalue decomposition. Portability and performance: Calculations can be done in 100% managed code, or in hand-optimized processor-specific native code (32 and 64 bit). Statistics Data manipulation: Sort and filter data, process missing values, remove outliers, etc. Supports .NET data binding. Statistical Models: Simple, multiple, nonlinear, logistic, Poisson regression. Generalized Linear Models. One and two-way ANOVA. Hypothesis Tests: 12 14 hypothesis tests, including the z-test, t-test, F-test, runs test, and more advanced tests, such as the Anderson-Darling test for normality, one and two-sample Kolmogorov-Smirnov test, and Levene's test for homogeneity of variances. Multivariate Statistics: K-means cluster analysis, hierarchical cluster analysis, principal component analysis (PCA), multivariate probability distributions. Statistical Distributions: 25 29 continuous and discrete statistical distributions, including uniform, Poisson, normal, lognormal, Weibull and Gumbel (extreme value) distributions. Random numbers: Random variates from any distribution, 4 high-quality random number generators, low discrepancy sequences, shufflers. New in version 4.0 (November, 2010) Support for .NET Framework Version 4.0 and Visual Studio 2010 TPL Parallellized – multicore ready sparse linear program solver - can solve problems with more than 1 million variables. Mixed integer linear programming using a branch and bound algorithm. special functions: hypergeometric, Riemann zeta, elliptic integrals, Frensel functions, Dawson's integral. Full set of window functions for FFT's. Product  Price Update subscription Single Developer License $999  $399  Team License (3 developers) $1999  $799  Department License (8 developers) $3999  $1599  Site License (Unlimited developers in one physical location) $7999  $3199    NMath http://www.centerspace.net .NET math and statistics libraries matrix and vector classes random number generators Fast Fourier Transforms (FFTs) numerical integration linear programming linear regression curve and surface fitting optimization hypothesis tests analysis of variance (ANOVA) probability distributions principal component analysis cluster analysis built on the Intel Math Kernel Library (MKL), which contains highly-optimized, extensively-threaded versions of BLAS (Basic Linear Algebra Subroutines) and LAPACK (Linear Algebra PACKage). Product  Price Update subscription Single Developer License $1295 $388 Team License (5 developers) $5180 $1554   DotNumerics http://www.dotnumerics.com/NumericalLibraries/Default.aspx free DotNumerics is a website dedicated to numerical computing for .NET that includes a C# Numerical Library for .NET containing algorithms for Linear Algebra, Differential Equations and Optimization problems. The Linear Algebra library includes CSLapack, CSBlas and CSEispack, ports from Fortran to C# of LAPACK, BLAS and EISPACK, respectively. Linear Algebra (CSLapack, CSBlas and CSEispack). Systems of linear equations, eigenvalue problems, least-squares solutions of linear systems and singular value problems. Differential Equations. Initial-value problem for nonstiff and stiff ordinary differential equations ODEs (explicit Runge-Kutta, implicit Runge-Kutta, Gear's BDF and Adams-Moulton). Optimization. Unconstrained and bounded constrained optimization of multivariate functions (L-BFGS-B, Truncated Newton and Simplex methods).   Math.NET Numerics http://numerics.mathdotnet.com/ free an open source numerical library - includes special functions, linear algebra, probability models, random numbers, interpolation, integral transforms. A merger of dnAnalytics with Math.NET Iridium in addition to a purely managed implementation will also support native hardware optimization. constants & special functions complex type support real and complex, dense and sparse linear algebra (with LU, QR, eigenvalues, ... decompositions) non-uniform probability distributions, multivariate distributions, sample generation alternative uniform random number generators descriptive statistics, including order statistics various interpolation methods, including barycentric approaches and splines numerical function integration (quadrature) routines integral transforms, like fourier transform (FFT) with arbitrary lengths support, and hartley spectral-space aware sequence manipulation (signal processing) combinatorics, polynomials, quaternions, basic number theory. parallelized where appropriate, to leverage multi-core and multi-processor systems fully managed or (if available) using native libraries (Intel MKL, ACMS, CUDA, FFTW) provides a native facade for F# developers

    Read the article

  • HTML text input and using the input as a variable in a script(tcl)/sql(sqlite)

    - by Fantastic Fourier
    Hello all, I'm very VERY new at this whole web thing. And I'm just very confused in general. Basically, what I want to do is take an input via text using HTML and adding that input to database, table trans. Should be simple but I am lost. <li>Transaction Number</li> <li><input type=|text| name=|tnumber| </li> // do i need to use value? <li>Employee Name</li> <li><input type=|text| name=|ename| </li> <li><input type=|SUBMIT| value=|Add|></li> ...... ...... sqlite3 db $::env(ROOT)/database.db mb eval {INSERT INTO trans VALUES ($tnumber, $ename} mb close They are both in a same file and there are only two fields to the database to keep things simple. What I can see here is that tnumber and ename aren't declared as variables. So how do I do that so that the text input is assigned to respective variables?

    Read the article

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

1 2 3  | Next Page >