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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Impulse Response and Frequency Response of a FBLMS filter

Suppose I work with sampling frequency FS and block length of L. I implemented the Frequency-domain Block LMS (FBLMS) algorithm ...
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Multi-thread Audio Processing With C and JACK

I'm currently trying to create a NUPOLS (Non-Uniformly Partitioned OverLap Save) convolution reverb processor using C and JACK. According to the literature I've consulted; this is equivalent to ...
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145 views

How to solve a time delay from an FIR filter?

I have designed a FIR high-pass filter $\{h_n\}$ with length 257 to filter a signal $\{f_n\}$ with length 512. To convolve signal with filter in time, I performed an FFT with length 768, padding ...
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51 views

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, ...
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347 views

What is the slope of a transfer function?

Lets say I have a Transfer function (H) plot as below. I need to find the SLOPE of the transfer function in dB/MHz. That is... i need a plot (db/MHz) vs frequency. How can i go about it ? ...
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FIR filter attenuation in the passband

OK, so I am designing a 2 kHz bandpass FIR filter using the following parameters: Fa = 1900 Hz Fb = 2100 Hz Fs = 50 kHz M = 33 pts Att = 40 dB I calculated the coefficients using this online tool (...
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Amplitude and phase compensation of analog filter

My analog filter perform much needed antialiasing and input protection. Unfortunately, the filter adds undesired changes in the gain/phase across frequency. For example, here is an example of ...
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Impacted of a conjugated filter

What is the effect of using a conjugated FIR filter? I have a use case where a weight vector, x, can be estimated using the least squares approximation: $$Ax=b$$ $$ A^HAx = A^Hb$$ $$ x = (A^HA)^{-1} ...
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65 views

Do we guess the cut off frequency when passband edge frequency and stop band edge frequency are given? FIR Filter design-LOW PASS FILTER

Say passband edge frequency=0.25 $\pi$ Stopband edge frequency=0.35 $\pi$ What is the cut off frequency? Do we average these 2 frequencies and write the answer or what is the procedure to find cut off ...
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Writing the specifications of digital filter, am I correct?

Design the FIR filter using suitable window for the specifications-: $0.899\leq |H (e^{j\omega}\;)|\leq1\;for\; |\omega| \leq0.2\pi$ $\; |H(e^{j\omega}\;)| \leq 0.01, \text{ for } \; 0.4\pi\leq \...
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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 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 ...
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323 views

How can a RLS algorithm utilise Wiener filter as FIR (M-tap) block?

I'm currently working with a dataset of $5000$ pulses of $N=15000$ samples each. I managed to implement the RLS algorithms with a FIR M-Tap filter such that $M\leq 15000$ ($150$ seems to achieve the ...
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Minimum phase conversion with Cepstrum method, how to scale the result?

I am trying to convert a zero phase spectrum (magnidue response curve with zero phases) to a minimum phase spectrum, because I need a totally causal impulse response for FFT spectral filtering, and ...
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how to get N point FIR filter( convolution) output when input and impulse response are both N points ? ideally convolution results in length N+N-1

x[n] -> Input Signal of length N h[n] -> Impulse response of length N y[n] = x[n]*h[n] (convolution to get the output of this FIR filter) Ideally length of y[n] should be N+N-1 but is 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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FIR filter format - Brutefir - Volumio

I am trying to compensate for the high frequency droop of a TDA1543 based nos DAC. One straightforward method is described here. Having played the resulting .wav, ...
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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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144 views

Amplitude problem by puting 5 FIR in a row

I'm trying to use 5 FIR in a row to improve the S/N ratio on my signal, but I have some difficulties doing it. In fact I have 2 signals, let's call them sig_I and sig_Q. I want to apply my 5 filter ...
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How to compare many bandpass filters (for their signal propogation delay)?

Let's say we have an audio signal sampled at 96 kHz, and we want to compare several bandpass filters to find the one with the lowest rectangular-envelope-propagation-delay (see graph below). The band ...
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How to properly match bit growth of a FIR filter with a reference value

I am trying to implement a chain of CIC/FIR filters on an ZYNQ FPGA. Using the Xilinx FIR compiler works fine so far but I am unable to properly get all the math. At the moment I have 2 chains of ...
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463 views

Side Lobes in the Magnitude Response of a Low Pass FIR Filter

I have the impulse response (from the filter coefficients) of an FIR filter obtained from MATLAB's "interp" function using the command: [y, b] = interp(x, 5); % where b contains the interpolation ...
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340 views

Implementing an analog approximation of an FIR filter given its impulse response

Problem: I have the impulse response of a matched filter(therefore its phase and magnitude response. See figure below) of a filter, and I need to implement its response using only off-the-shelf ...
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315 views

How to obtain a meaningful group delay graph using matlab

Problem: I'm trying to analyze the behavior of an FIR filter with the following impulse response/kernel: Using Matlab's function grpdelay(myKernel,length(myKernel)), I obtained the following figure:...
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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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Digtial FIR Impulse response & transfer function

I am currently working the figure through below. as it is an FIR Filter i have worked out using convolution that the output is 4,2,4,6,0,0. i am trying to obtain the 'z' domain transfer function of ...
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388 views

Generating FIR filter in Web Audio API

I'm creating an HTML that applies a low-pass Windowed Sinc filter with a window size of 8192 to white, pink, or Brownian noise. There are nine radio buttons, eight of which set a cutoff frequency and ...
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45 views

Lower bound of weighted average of sequence

Can anyone prove that $$\mathrm{avg}\left(\frac {a_i}{\left(1+a_i\right)^2}\right) \ge \frac{\mathrm {avg}({a_i})}{\left(1+\mathrm{avg}{(a_i)}\right)^2}$$ for a sequence of positive valued elements $...
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How Is the MATLAB `filter()` Function So Fast?

How is the filter() implemented in MATLAB? How is it so fast? What's the fastest implementation of a FIR filter?
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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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103 views

FIR Filter implementation vs Linear convolution implementation

I came across a C code for the FIR filter on one of the websites. It is as follows ...
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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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Decimating Polyphase Filter in Simulink

I have read many of the articles on the website regarding a polyphase decimating filter structure. I'm sure we are all familiar with the typical image shown in Fred Harris Multirate book and many ...
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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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88 views

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 ...
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75 views

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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103 views

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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Nyquist Frequency and Window Length

The Nyquist theorem states that the sampling rate must be twice the highest frequency to be observed. How does the length of the interval play into this relationship? The main resource that I found ...
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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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44 views

Arbitrary phase FIR filter design

I would like to design an FIR filter using the frequency sampling method with nonlinear phase specifications. For example, between $0$ and $\frac{f_s}{4}$ a phase of $180^\circ$ and between $\frac{f_s}...
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41 views

Low Pass Filtering, Welch's Method, ENBW

I am working on using Welch's method (https://www.osti.gov/servlets/purl/5688766/) to estimate a Power Spectral Density (PSD). This algorithm involves scaling a set of averaged FFTs by the equivalent ...
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715 views

Applying multiple IIR and FIR filters at once

I would like to appy multiple filters to a signal at once. Let's assume I got two frequency bands I want to extract. I compute a Butterworth Filter for 300-500Hz and one for 700-800Hz. I would like to ...
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Why is z (and not ω) the variable of interest for discrete time systems?

A continuous time domain system is well described by the Laplace transform. It allows to express any continuous signal x(t) as the integral sum of weighted complex and exponentially growing/decaying ...
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Is parallel structure applicable in realizing FIR filters?

This may sound weird but can FIR filters be realized by implementing it in parallel form like in IIR? I know that FIR can be realized in cascade, but I can't seem to find any resources that realize ...
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Who first coined "Direct Form I" and "Direct Form II"?

I see "Direct Form I" and "Direct Form II" commonly in literature to refer to FIR / IIR implementation from the filter transfer function, and reference it myself, but where did ...
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Convolving a LP FIR filter with a HP FIR filter

I currently feed a sampled audio input signal into a 511 tap highpass FIR filter, the output of which is then fed to another 511 tap lowpass FIR filter, resulting in a very narrow bandpass filter ...
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Amplitude / gain of difference of 2 known FIR low-pass filters

Given a discretely sampled sinusoidal signal (frequency $1/40$), and 2 Simple Moving Averages (SMA), better known as FIR low-pass filters of order e.g. $10$ and $21$, it can easily be calculated what ...

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