Specify an equiripple bandpass FIR filter. In this example, you Open a default filter in Filter Designer. For more details, see the Algorithms section on the dsp.SineWave object page. You can import an existing filter into SPTool, or you can design and edit a new filter using Filter Designer. In other words, the ratio F s / f i must be an integer. The value of T s equals 1 / F s, the variable f i is the frequency of the sinusoidal signal, and F s is the sample rate of the signal. That is, 1 / f i T s = k i must be an integer value for every channel i = 1, 2. This method of generating the sinusoidal signal requires that the period of every sinusoid in the output be evenly divisible by the sample period. To generate a fixed-point signal, set the Method property of the dsp.SineWave object to 'Table lookup'. The fraction length is computed based on the values of the generated sinusoidal signal to give the best possible precision. For the purpose of this example, set the value to numerictype(,16). To generate a fixed-point signal, set the CustomOutputDataType property to a numerictype object. It instantly designs high-order IIR filters based on the Butterworth, Chebyshev, or Elliptic analog prototypes. Both approaches give similar results.ДЄnswer to question# 2: You could, if you wish, call an integrator a 'filter.' But it's unstable, and unstable filters are of no value in trying to filter real-world signals.Create a dsp.SineWave object with SampleRate set to 22.05 kHz, SamplesPerFrame set to 32, and OutputDataType set to 'Custom'. Design ScopeIIR provides a comprehensive IIR filter design capability. The two approaches result in freq magnitude responses that are zero-valued at all frequencies except at zero Hz. The DFT of an impulse response that's a finite-length sequence of ones, evaluated at zero Hz, is merely the sum of the impulse response samples. So freqz() gives a freq mag response of infinity at zero Hz. And that ratio has a zero-valued denominator at zero Hz. But I'll bet a pint of beer that the region of convergence is outside the pole (and therefore outside the unit circle) because the integrator's impulse response is causal.ДЄnswer to question# 1: Matlab's freqz() command computes the freq response at zero Hz by evaluating a ratio of polynomials at various frequencies. Those are good questions! Here's my 'two cents.' Hilmar is correct, the integrator is not stable because there's a pole on the z-plane's unit circle. Is a leaky integrator the same thing as a low pass filter? However it is a leaky integrator not a rectangular (i.e. I forgot to mention that this is the closest to this question I could find on the site. This will give the block an additional in port. Change the Initial Condition Source property of the integrator to external. Why does the freqz method does not match the impulse response + FFT approach?ДЄm I wrong in trying to look at an integrator as a digital filter? 1 Answer Sorted by: 1 Using a workspace variable as you are doing is the wrong approach. I guess this kind of makes sense mathematically, as the output of the integrator is a step function, which, after FFT, is the addition of as many samples of the step output as the FFT length. However, when finding the freq response by sending an delta pulse through the integrator via filter Matlab function, and performing the FFT on its output, I just get a pulse (with amplitude equal to number of FFT points) followed of zeroes. Plotting the frequency response with Matlab using freqz shows what it looks like a kind of low pass filter response. I have a simple discrete time rectangular integrator with Z transform H(z) = 1/(1-Z^(-1)) Complementary Filter Design on the Special Euclidean Group SE. A very simple question which can't really figure out:

Matlab filter designer integrator