Signal Processing

Questions tagged [digital-filters]

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38 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
43 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
46 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
42 views

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

1) How is this a low pass filter? $0.99 \leq |H(e^{jw}\;)|\leq1.01 for \; 0\leq|w|\leq0.19\pi$ $|H(e^{jw}\;)|\leq0.01\;for\; 0.21\pi\leq|w|\leq\pi$ 2) What kind of filter is this? Low pass, high pass, ...
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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
48 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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108 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
46 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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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
95 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
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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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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1answer
166 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 ...
2
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1answer
65 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 ...
3
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1answer
109 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
80 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
63 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
35 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
52 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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2answers
47 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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1answer
56 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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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
72 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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2answers
51 views

IIR Filter Stability

I have designed an inverse IIR high pass filter by inverting its transfer function B(z)/A(z) to A(z)/B(z). However, the inverse filter has a pole at z=-1. Is this filter considered stable (marginally ...
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1answer
41 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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1answer
63 views

How to calculate the SNR from the data collected by ADC?

I want to design a filter to deal with periodic signals according to the actual collected data. In practice, I would like to use the SNR as the criterion to judge the performance of the filter. ...
0
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1answer
106 views

Having trouble interpreting z-transform description of a predictor from a codec paper

I've been looking at the opus paper (https://arxiv.org/pdf/1602.04845.pdf); in particular, in section 4.1, they describe the predictor for the current band energy based on energy from both the current ...
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2answers
243 views

How to remove power line interference?

I have a couple of datasets that show a peak around the 50-60 Hz range (mostly around 55Hz and in some cases at their harmonics i.e. 100-120Hz.) In some datasets the spike is significant in comparison ...
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1answer
59 views

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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1answer
39 views

Ideal Allpass filter

Is it possible to implement an ideal allpass filter in Matlab? The filter response should be like this: Note: the photo is imaginary (I did it using paint).
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0answers
24 views

Can digital filters be arranged as a hierarchy?

I am developing a software framework for image processing. So, I need to arrange the digital filters as a hierarchy. Digital Filters The above link lists the following digital filters: Mean Filter -...
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1answer
54 views

Definition of frequency response

I studied various signal processing materials for a long time, and I have a question. Considering an LTI filter, one can define its frequency response by evaluating its transfer function $H(z)$ on the ...
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1answer
47 views

Zeros and poles from transfer function

I have a transfer function $$ H(z) = \frac{Y(z)}{X(z)} = 1 - 0.5z^{-1} \text{.}$$ I'm interested in zeros and poles. I know I need to adjust the function to $$ H(z) = \frac{\prod_i(z-n_i)}{\prod_i(z-...
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0answers
61 views

Problems discretizing a notch filter - Chebyshev Type I

I'm trying to implement and discretize a notch filter (Chebyshev type I) these specifications: Ws = 2pi[2000 2500]; Wp = 2pi[1000 7000]; Rp = 0.5; Rs = 40; But when I try to use discretization methods,...
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1answer
50 views

80-tap Decimation Filter: divide into stages? / Cost of having a number of taps

I did a FIR filter hardware implementation with 80 taps (FPGA/VHDL). I will use it later as the decimation filter. My questions: Does the number of taps affect a cost of implementation of FIR? FIR ...
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1answer
22 views

Phase in FIR filters [closed]

Does anyone have any idea what is Clear Phase technology ? How is it implemented ? And which audio channels whether left ,right center , sub woofer should have this technology implemented and why ?
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3answers
306 views

Filter odd or even harmonics with notch or inverse notch filter

Hi i had the following question. I have a signal containing a 200Hz sine wave and it's odd and even harmonics (no other frequencys or disturbing signals are contained). What i'm looking for is a kind ...
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1answer
53 views

When is a LTI IIR $H(z)$ system minimum phase

EDIT: I failed to mention that the system's inverse also needs to be causal and stable. I cannot wrap my mind around on how when a system and its inverse are both causal and stable and LTI IIR it is ...
2
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2answers
51 views

How to check if a signal is mainly composed of a few impulse peaks?

I am working on a problem that produces different types signals, and I need to detect the peaks after low-pass filtering, then find_peaks from ...
2
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3answers
115 views

Why must the order of a Band-Pass and Notch filter always be even?

My professor mentioned that the order of a band-pass and a notch filter must always be even, when showing an example of designing a digital filter using the bilinear transformation. Then he also ...
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1answer
40 views

Power Spectral Density of a Filter

I need to calculate the output power spectral density of the following digital filter My calculations are as follows: $y\left(n\right)\:=\:x\left(n-1\right)+d\left[x\left(n-1\right)+x\left(n\right)\...
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0answers
30 views

Upmixing after Equalization [duplicate]

Can anyone tell me why in a stereo mode many channels are derived from the 2 channel input . It looks like upmixing.. My questions are summarized below : If it is Stereo mode why upmix it to many ...

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