Questions tagged [digital-filters]

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1answer
60 views

How to design a filter that can filter out noise accurately, after setting the parameters of the filter using standard signal?

I meet a problem with designing a filter. I have two different instruments that could measure the same AC signal (usually ~200hz, always <1kHz), A and B. A can carry out signal measurement during ...
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1answer
346 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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18 views

Subband Tuning in Matlab

Suppose you have an i/q stream with a sample rate of 20 MHz. You want to bandpass filter a 1 MHz bandwidth (ie passband from 5 to 6 MHz). You then want to heterodyne that 1 MHz chunk. How would you do ...
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1answer
61 views

How to identify low pass, bandpass, band reject, high pass filter etc from given $|H(e^{j\omega})|$?

1) How is this a low pass filter? $0.99 \leq |H(e^{j\omega}\;)|\leq1.01 for \; 0\leq|\omega|\leq0.19\pi$ $|H(e^{j\omega}\;)|\leq0.01\;for\; 0.21\pi\leq|\omega|\leq\pi$ 2) What kind of filter is this? ...
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1answer
43 views

What is the transfer function of the filtfilt?

I designed a Parks McClellan filter to design a lowpass filter. First, I created the order of the filter with firpmord(). Then I designed the filter with firpm() and created the impulse response. I ...
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1answer
130 views

How do I know what is causing the noise in my signal

I have a noisy audio file, available here (It is safe to download if you wish to) https://ufile.io/pstrt Now, I inspect it in matlab: And then i plot it on normalised frequency: Now I need to ...
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1answer
157 views

How to implement a series of second-order, digital state-variable filters in MATLAB?

How to implement a series of second-order, digital state-variable filters, using the high-pass output from each section to effect the low-pass to high-pass transformation in MATLAB? Hello, I am trying ...
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1answer
39 views

Why an does an integrate-and-dump stage in a CIC filter provide the same functionality as the integrator and comb in series?

I have a hard time understanding why the combination of integrator and comb (integrate-and-dump) is the same as a separate integrator and comb in series like in this picture: Please explain why these ...
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2answers
175 views

Can digital filters actually separate signals?

I've come to realize what appears to be a hard limit in convolution theorem: to avoid time-domain aliasing, we must pad the signal/filter, but the padding distorts the spectrum. Consider a ...
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1answer
28 views

Fourier transform and Zero Order Hold

I was asked to explain why a zero order hold (ZOH) filter is not a good choice for signal reconstruction. To answer this question, I was thinking about studying the Fourier transform of the ...
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1answer
524 views

How can I get the continuous-time transfer function coefficients (or poles and zeros) from the corresponding discrete-time TF and vice versa?

Let's say I have a continuous time transfer function which I know its numerator coefficients $(B^c = [b^c_m, ..., b^c_1, b^c_0])$ and denominator coefficients $(A^c = [a^c_m, ..., a^c_1, a^c_0])$. ...
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1answer
37 views

Filter model of passive attenuation of headphones

It is possible to model passive attenuation of headphones as a Matlab/Python filter ? For example I would like to find digital filter which will be simulating the way of the passive attentuation of ...
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2answers
40 views

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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1answer
60 views

Moving average filter implementation in the floating point [closed]

I have following C++ implementation of the moving average digital filter. ...
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1answer
47 views

Prove the following property of All-Pass Filters

If A(z) is an all-pass filter given by $$A(z)=\frac{z^{-1}-d_1}{1-d_1 z^{-1}}$$ where $d_1$ is a real coefficient Then prove that $$ |A(z)|\left\{ \begin{aligned} &<1 &|z|>1 \\ &=1 &...
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1answer
80 views

How can I find the transform function, amplitude and frequency response of a digital filter in python?

I have applied a laplace filter mask to an image and now I want to find the amplitude and freqency response of a laplacian filter: [[1,1,1], [1,-8,1], [1,1,1] ]. I know I need to first find the ...
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2answers
39 views

Digital IIR LPF Difference Equation from Transfer Function

I want a digital IIR filter with f0=225kHz and fs=53.125GHz. I can come up with the transfer function and plot it using Matlab. ...
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0answers
49 views

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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109 views

Design an FIR linear phase lowpass filter using windows to meet the following specifications. What is given in the question? How to start solving it?

What is given in this question? I have not seen a single solution of these types of filter design problems in the internet. So I am confused, how do I proceed solving these problems. Can you guide me ...
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1answer
27 views

Estimating reverberant time (T60) of an IIR filter

I'm learning a method to genenrate pink noise by filtering a white noise in this book, and the author mentioned an estimation of T60 given the coefficients of an IIR filter in the code that ...
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1answer
65 views

Phase lag changes cross spectrum

I have a same noise passing through two different ARMA filter. The autospectrum of first filter is given by S11=G1*G1*S_in where S_in is input spectrum created by set of random numbers. Similarly the ...
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2answers
3k views

highpass filter design in Matlab

I hope to design 1st order highpass filter from transfer function. In the example of 1st order lowpass filter, I first get the coefficients of numerator and denominator in the variable 'b' and 'a'. In ...
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1answer
49 views

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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0answers
5 views

Active noise control in LabVIEW using cRIO (FPGA and real-time module)

Has anyone tried to implement the example code in LabVIEW for active noise control usin the my CRIO controller? I have an experimental setup of two microphones (primary and error microphone) and two ...
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1answer
171 views

Does it make sense for my filter to depend on the discrete sample variable n?

I feel a little nervous asking this question as I am not sure if know what exactly I am trying to ask. But please see if you can help. Ok, I have a filter whose output grows as long as the input is ...
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0answers
30 views

How to find lattice coefficients when k2=1?

$H(z)=1+2z^{-1}+z^{-2}$ We know the formula $a_{m-1}\;(k)=\frac{a_m\;(k)-k_m\;a_m\;(m-k)}{1-k_m\;^2}$ Here $k_2=1$ That makes $a_{m-1}\;(k)=infinity$ How to find lattice coefficients of all pole IIR ...
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0answers
96 views

How to read the filter IIR design numericals specifications?

First of all I want to let you know about notations that I am using-: $\alpha_p$=attenuation factor of passband(in dB) $\alpha_s$=attenuation factor of stopband (in dB) $A_p$=Gain at passband edge ...
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1answer
26 views

Design of digital filter with desired phase response

I want to design a digital filter with the following phase response in MATLAB. i.e. at 1kHz, the phase response should be 9 degrees, at 2khz phase response should be 18 degrees ,at 3kHz phase response ...
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1answer
68 views

How is CIC interpolation gain calculated?

For a programmable CIC interpolation IP core configured as follows: Input bits : 17 Output bits : 49 (Full precision output) Stages, N : 5 Diff Delay, M : 1 According to E. B. Hogenauer in An ...
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1answer
234 views

Adaptive Band pass filter for extracting respiratory waveform from accelerometer data

I am recording data from an accelerometer attached to the chest (1000Hz). I need to extract the respiratory waveform. I tried an adaptive bandpass filter based on a dominant frequency in my signal ...
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1answer
58 views

Lattice structure-Lattice coefficients-Sign positive or negative?

This is the question $$ H(z) = \frac{1}{2 + 1.4z^{-1} + 1.8z^{-2}} $$ My professor has told that we don't need to take those 2 values as negative and instead we can take them as positives(with the ...
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4answers
37k views

How many taps does an FIR filter need?

I am looking to design a set of FIR filters to implement a low pass filter. I am also trying to reduce the latency of the signal through the filter so I am wondering what the minimum number of taps I ...
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1answer
112 views

Why is $y(n)=x(n)-x(n−1)+x(n-2)$ a low-pass filter?

I know that it's a low-pass FIR filter with impulse response of length equal to $3$, but I don't know how to explain exactly why it's low-pass.
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1answer
186 views

Help in designing filter for DUC (Digital Up-Converter) and DDC (Digital Down-Converter) using the built-in DUC and DDC blocks in MATLAB Simulink

Can anyone please help me in simulating a simple BPSK transceiver system that correctly receives a signal through a DUC and DDC? I have attached the screenshot of my Simulink system along with ...
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1answer
84 views

Can you explain remez exchange algorithm?

What I don't understand about this algorithm-: What is the diffeence between w and w cap? What is the meaning of extremals? What does it means to reject p-r superfluous potential...
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0answers
27 views

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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1answer
70 views

How is this given impulse response of infinite duration? Isn't it just from -π to +π?

How is hd(n) infinite duration when it is from -π to +π. Book says as it is infinite duration, we in the next step take-: h[n]=hd[n] from n=-(N-1)/2 to (N-1)/2 and 0 otherwise. I can't see how this ...
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1answer
65 views

Varying phase shift between two signals

I have two microphones separated by a distance d placed in end-fire configuration. Audio input data was played from the front side (i.e corresponding to 0 degrees.) The data picked by microphone1 and ...
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1answer
44 views

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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1answer
387 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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1answer
38 views

GNU Radio filter normalization

I have noticed that all filter taps from the firdes class are normalized by dividing them over the sum of all taps magnitude, as shown in the example below for Gaussian Filter. Is this normalization ...
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1answer
53 views

Aritifacts of time-varying IIR filters

I'm doing something like spatial audio algorithm using IIR modelled HRTFs. At each frame I get a direction information with azimuth and elevation, and then I have to find or interpolate the ...
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1answer
25 views

Load cell noise filteration

I'm currently working in a project as apart of a student team. Setup A platform is attached (placed over and screwed) two 1 Kg load cells in order to measure the weight of the system. A vibration ...
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1answer
57 views

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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2answers
48 views

When applying analog and digital effects on an audio signal is it better to start with a clean digital signal or use the analog effects before ADC?

If I am digitally capturing an audio signal from an analog source and I know I will want some analog effects applied, is it better to use them on the analog signal before doing the ADC, or should I ...
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0answers
35 views

How to calculate analog cutoff frequency to design a digital filter?

$\Omega_c$=cutoff analog frequency In this example, what is going to be value of $\Omega_c$? Some teachers in youtube are teaching this method-: Is this method correct? If I use this formula, then ...
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1answer
48 views

Phase Response of $N$-th order Digital All-Pass Filter

I am having trouble reconciling my derivation of the phase response of an N-th order all-pass filter with those I am finding in the literature, and I figured someone here could help me. Real Version: ...
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0answers
7 views

Processing accelerometer and gyroscope data

I'm working on an instrumented headform to measure linear and rotation acceleration of the on impact, for which I'm using ADXL377 accelerometer and ICM-20649 angular rate sensor. What kind of ...
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0answers
21 views

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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1answer
78 views

Is the sampling period arbitrary for a bilinear transform, and why?

I have been given the specifications for a digital highpass filter (stopband, passband, stopband attenuation and maximum passband ripple). I am expected to design a prototype lowpass filter in the ...

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