I\'m baffled by the results I\'m getting from FFT and would appreciate any help. I\'m using FFTW 3.2.2 but have gotten similar results with other FFT implementations (in Java). When I take the FFT o
There is a Table of pairs , which defines pieces bounds. And we are using straightforward algorithm: y = f开发者_开发百科(x)
I\'m trying to understand the effect of the Block Size and best strategy of choosing the Coefficients in DCT compression.
I\'m trying to implement this extenstion of the Karplus-Strong plucked string algorithm, but I don\'t understand the notation there used. Maybe it will take years of study, but maybe it won\'t - maybe
I have performed an fft (fast fourier transform) on a time series waveform in Matlab, but I seem to have a weird wave actually in the fourier transform plot, although there are spikes this开发者_如何学
I have two waveforms which are linked by a numerical facto开发者_运维问答r. I need to use optimal scaling (least squares) between the two waveforms to calculate this factor in Matlab. Unfortunately I
I\'m trying to interpolate some data for the purpose of plotting. For instance, given N data points, I\'d like to be able to generate a \"smooth\" plot, made up of 10*N or so interpolated data points.
What is the best method for selecting design properties for a digital filter in Matlab with the GUI sptool?More specifically, if I have a signal, how do I go about determining which filter values will
My problem is this: I\'m developing a reasonably small application (which needs to be able to grow in the future, but for now, limited functionality will suffice) which recieves audio (16bit mono @ 4
I need to do some basic and non basic DSP programming in C#. At its core, it includes the generation of a sin wave deciding its frequency i开发者_高级运维n Hertz. Then I\'d like to Frequency Modulate
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