# Questions tagged [finite-impulse-response]

A finite impulse response (FIR) filter is a type of a signal processing filter whose impulse response is of finite duration.

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39 views

### Does a FIR filter always perform a convolution operation?

I've come across this question in textbook. I guess it's basically asking are there any input or impulse response conditions that a FIR filter won't be able to compute. I can't think of any.
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### How to find the impulse response from the following input/output relation

Suppose we have a system defined with input $x(t)$, and output $y(t)$, related by $$y(t) = \sum_i \alpha_i(t)x(t-\tau_i(t)) \tag 1$$ This is a linear system, (according to the textbook), so we can ...
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### Frequency Sampling Filters (FSFs) NOT the Frequency Sampling Method

A class of digital filters called Frequency Sampling Filters (FSF) has been mentioned in a side note by Richard Lyons in another related question comparing IIR digital filter design using mapping ...
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### Minimum Phase - All Pass Decomposition For Large Linear Phase Filters

UPDATE: I am looking for a robust approach to decompose linear phase FIR filters with 100s of coefficients into its minimum phase and all pass components. I originally thought determining all the ...
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### Mapping of Classic Filters for Digital Filter Design

Of the four classic analog filter types: Butterworth, Chebyshev, Elliptic and Bessel- are any of these relegated to obsolescence for purposes of digital filter design in comparison to optimized ...
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### How to calculate magnitude of frequency response?

I'm trying to write a function in Python that calculates the magnitude of an FIR filters frequency response. I tried doing it by first calculating the Fourier transform with np.fft.fft and then ...
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### Causal inverse of $h[n]=\delta[n]-\alpha\delta[n-1]$

Find the causal inverse of $$h[n]=\delta[n]-\alpha\delta[n-1]$$ we have $h[0]=1$ and $h[1]=-\alpha$ also $h[n]=0$ for $n>1$ From the formula $$h_i[n]=\sum_{i=1}^n\frac{h[n]h_i[n-i]}{h[0]}$$ we ...
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### strange waveform after highness FIR filter

I was trying to apply FIR filter on the ECG signal: a bandstop for 45-55 Hz for noise removal and a highpass filter for moving baseline wander. But after that, the ECG signal show strange waveform at ...
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### How to get the difference equation from impulse response?

I'm trying to solve a problem about filter design. We want to design a equalizer filter which have the following frequency response : How can i get the difference equation from the impulse responce ...
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### Is a second order low pass or high pass fir filter possible?

I'm learning about filters and obviously the first kind of filters I came across were low pass and high pass filters: $$\frac{1}{2}(1+z^{-1})$$ is a low pass filter and $$\frac{1}{2}(1-z^{-1})$$ is a ...
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### How to estimate multidimensional linear response function from measured input and output signals?

Problem definition: I measure N sources of noise $x_i$ and one output signal $y$ which contains real signal $y_0$ plus noise from all these sources $x_i$ transformed by linear response functions $h_i$ ...
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### Where can I find a solved numerical example of FIR digital filter design using Kaiser window?

I am studying for digital signal processing exam. And I need to learn about Kaiser window numerical for it. It is pretty important. I searched the whole internet but could not find 1 example of Kaiser ...
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### How would I approach implementing a windowed sinc filter with changing cutoff in real time?

I moved this out of another question, where I'd put it to give more context, and answered it here for the next reader. ELI5 how to implement a windowed sinc FIR filter in real time for an audio ...
190 views

### What is a simple method, if any, to calculate a variable IIR filter for audio effects in real time?

Question edited based on first answer which raised a lot of good points about what problem I'm trying to solve. Also reordered some things to stick context of how I did this with FIR so far at the ...
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### FIR filtering returns "static" signal

//PROBLEM I'm trying to filter EEG trials from an array in which each trail has dimensions [8x125] = [number of channels x samples@250Hz], but the resulting signal looks static and wrong. Why is this ...
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### Notch Filter via Bernstein Polynomial

Context: In the review article [1], the authors explain that we can model the frequency response of an FIR notch filter using Bernstein polynomials. The paper shows the details, but at the end of it, ...
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### NLMS algorithm greatly attenuating signal

Im writing an NLMS MATLAB program to remove powerline noise from ecg signals. I sweep through tap widths and learning rates which get the best SNR values. Some of the combinations produce great SNRs ...
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### Determine the lattice coefficients corresponding to FIR filter with system function

And sketch lattice structure $H(z)=1+ \frac {13} {24} Z^{-1}+ \frac {5} {8} Z^{-2}+ \frac {1} {8} Z^{-3}$ This is my engineering assignment question from Digital Signal Processing subject on the topic ...
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### How do large tap counts effect FIR Filters for block data?

I am using the ARM CMSIS DSP library for a signal processing application. I find good filter performance when using tap counts that are in excess of 10% of the length of my data block. I understand ...
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### Accuracy of time-domain filters and frequency domain filters

I am trying to calculate the Hilbert transform using 3 different method: scipy.signal.hilbert function, time domain filter, and frequency domain filter. The test ...
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### FM demodulation using complex differentiation

I'm trying to implement the complex differentiation discriminator which is taken from Software-Defined Radio Using MATLAB, Simulink, and the RTL-SDR. The final result is s(t) = \frac{s_q'(t) s_i(t) -...
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### Find coefficients of optimal Wiener filter of length 2

I need to find the coefficients (impulse response) of a FIR Wiener filter with length equal to 2. I have a gaussian white noise signal that is generated using the Standard Normal Distribution (mean = ...
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### Half-band FIR Filter desired output

I am a trying to implement the Receiver Signal Path of the ADRV9009 in Verilog, starting with the 2 half-band filters. These are the coefficient of the two filters: This is the result that I got. My ...
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### Can simple FIR/IIR filtering recover/estimate a signal plagued with low SNR white noise?

I'm trying to design a filter to restore the original signal from strong noise (low SNR). I have tried low pass FIR and it could work well as the amplitude of original signal is much higher than the ...
282 views

### Implementing FIR Hilbert Transformer in Python

I want to implement FIR Hilbert Transformer in Python using window method. I'm new to digital signal processing so please point out to me all mistakes you notice. Ideal Hilbert transformer is ...
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### Frequency domain filtering: must filter coefficients always be conjugate symetric?

While trying to do frequency domain filtering of audio signals using windowed overlap-add methodology, I came across some pre-calculated frequency domain coefficients. I noticed that the coefficients, ...