By default, butter produces a discrete-time filter design. Therefore the transfer function is defined in terms of z (z-transform), not s (Laplace transform). A discrete-time Butterworth filter of order n has an n-order zero at z=-1 and n poles within the unit circle. This is in accordance with your results....

matlab,signals,system,lowpass-filter

You should use z = filter(b,1,x);, Fs = 6000; T = 1/Fs; N=512; t = (0:N-1)*T; x1 = 4*sin(2*pi*1000*t); x2 = sin(2*pi*2500*t); x = x1 + x2; NFFT = 2^nextpow2(N); y = fft(x,NFFT)/N; fx = Fs/2*linspace(0,1,NFFT/2+1); subplot(211) plot(fx,2*abs(y(1:NFFT/2+1))) fc = 1500; Wn = (2/Fs)*fc; b = fir1(20,Wn,'low',kaiser(21,3)); z = filter(b,1,x);...

image-processing,lowpass-filter

There are many methods to do this, since convolution to Fourier analysis among others. If you wants to do a convolution you have to run over the image I(mn) with a mask M(AB). considering that you only run over internal parts of the image you will do (N-A)(M-B)(A*B*2) operation approximately....

matlab,filter,fft,frequency,lowpass-filter

Short Answer: Using freqz: you only get the single-sided frequency response of the filter and hence you can only apply the filter on the single-sided spectrum of the signal, x to get the single-sided frequency spectrum of the output (filtered) signal o1: z = freqz( b, a, N/2+1, Fs); o1...

You need to pass your filter into waveOut.Init rather than audioFileReader, so that the audio from the file is pulled through the filter. waveOut.Init(myFilter); ...

c#,signal-processing,lowpass-filter

IIR filters are notorious for being sensitive to numerical precision, and especially so as the number of terms in the recurrence equation increases. Fortunately, in this case it is possible to obtain a different filter topology which is a bit less sensitive by implementing the filter as a cascade of...

android,smoothing,lowpass-filter

A lowpass filter is supposed to let in low frequencies and elimiate high frequencies. So the wikipedia version looks right to me- for a low frequency, dT should be big, for a high frequency it should be small. However, I'd really suggest either hiring someone who knows DSP or doing...

matlab,filtering,fft,lowpass-filter

Just to remind ourselves of how MATLAB stores frequency content for Y = fft(y,N): Y(1) is the constant offset Y(2:N/2 + 1) is the set of positive frequencies Y(N/2 + 2:end) is the set of negative frequencies... (normally we would plot this left of the vertical axis) In order to...

javascript,c#,unityscript,lowpass-filter

The example code may be wrong. Use Vector3.Lerp() instead. A full list of Lerps is here.