Questions tagged [digital-filters]

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Design of Parks-McClellan Lowpass Filter

Can we design Parks-McClellan Lowpass Filter with only sampling frequency of 12KHz and cutoff frequency of 4KHz and order of 30? I went through Matlab help, where we need parameter like passband ...
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1answer
56 views

Advice on designing a digital filter that doesn't have phase-sensitive edge artifacts?

I'm pretty well versed in statistics, but not really digital signal filtering. I have a data scenario where I expected to be able to pretty easily filter out some noise (human pulse) that's at a known ...
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52 views

Question about ideal filters

Following shows the discrete time Fourier transform of an ideal low pass filter with cutoff frequency $\omega_c$: $$H\left(e^{j\omega}\right) = \begin{cases} 1, & \text{if $|\omega| \le \omega_c$...
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Understanding How to Create a Goertzel Implementation

I understand how to use the Goertzel filter, and now I'm trying to understand the inner workings of it. According to wiki, the Goertzel filter can be thought of as an IIR followed by FIR. I have ...
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1answer
38 views

Which one can be accompanied by linear filters?

I have a matrix : \begin{bmatrix} 1 & 2 & 3\\ 1 & 4 & 5\\ 2 & 6 & 7 \end{bmatrix} After doing operation 1 , I get \begin{bmatrix} 0 & 0 & 0\\ 1 & 2 & 3\\ 1 &...
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1answer
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How to find minimum length of a FIR symmetric filter if I am given DTFT

I am practising for upcoming exams and came across this question. Let $h[n]$ be an FIR filter such that $h[n] = 0$ when $|n| > M$ and $h[n] = h[−n]$. A plot of $H(e^{j\omega})$ (DTFT of $h[n]$) is ...
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1answer
40 views

Resampling FIR coefficients

I have received from a client a 32-coefficient FIR filter running at 1 kHz. I would like to adapt the filter to a 24-coefficient FIR filter running at 750 Hz while preserving the 0-375 Hz frequency ...
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1answer
109 views

Digital filter coefficients from low-pass to high-pass

Given I have coefficients a0, a1, a2, b1, and b2, defining the difference equation for a digital filter as: ...
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2answers
89 views

Complex Samples (IQ) - Baseband Filtering

We are currently analyzing a large set of IQ samples in a desktop application and we are interested in implementing many different bandpass filters dynamically. We realized, that working with scipy ...
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1answer
40 views

How to calculate DTFT of cosine function divided by n

I'm having a hard time to calculate the next function, and I don't really know Matlab good enough to calculate it there. Help would be appreciated: $$h[n]=\frac{A_1 \cos⁡[\theta_1(n-N/2)]}{n-N/2}$$
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cascade realisation from direct realisation

I am trying to convert filter realisation from Direct form 1 to cascade using function provided from Digital signal processing by Proakis. I designed lowpass IIR filter and extracted numerator and ...
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1answer
57 views

Frequency conversion in IIR lowpass-to-lowpass transformation

We use $\ z^{-1} \xrightarrow{} \frac{z^{-1}-\alpha}{1-{\alpha}z^{-1}}$ and $ \alpha = \frac{\sin(\omega_c^{'}-\omega_c)/2}{\sin(\omega_c^{'}+\omega_c)/2}$ for a lowpass-to-lowpass frequency ...
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0answers
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Effect of time duration in bandpass filtering

I am new in DSP, but I am trying to process EEG data to decompose into the bands, theta, alpha, and others. The raw data is processed: Remove the DC offset by subtracting the average value from the ...
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1answer
40 views

How to quantize filter coefficients to 10 bits

I am trying to learn digital filter design. One of the concept is quantization of filter coefficients. I used Designfilt to design an lowpass IIR filter and obtained coefficients. Now I want to ...
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0answers
33 views

The size of an FIR filter for PDM-PCM conversion

I have been trying to understand how the conversion from PDM to PCM happens so that I could interface with a microphone. I have been doing some research online and I found that the most common way to ...
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1answer
36 views

3 dB bandwidth of a filter

I am solving one of the question from signal processing about calculating the 3dB bandwidth of the filter. But I am stuck once I got the magnitude response of the filter. Can someone help me with this?...
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1answer
32 views

Stop band ripple vs stop band attenuation

What is difference between stop band ripple and stop band attenuation in context of filters?
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Fixed Point, Time Domain and Freq Domain Filter Difference

I am trying to compare my IIR filter in time domain and freq domain, but i am getting different results which i dont understand why ?. So please help me out in understanding the issue. My code is : <...
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1answer
40 views

Finding number of alternations in parks mcclellan

I'm given this question and asked to find number of alternations for filter 1 and 2, using parks mcclellan algorithm. I search for solution and find this answer : My question is why the number of ...
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1answer
42 views

Impulse Invariant method for digital filter design

One of the known methods for discretizing analog filters is impulse response invariant. We get the impulse response in time domain, discretize it and then get the Z transform. What I am trying to ...
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24 views

Remove noise from distance calculated from accelerometer data

I got distance calculated from accelerometer raw datas using double integral accelerometer.I attached the image below.But it shows noise with it.Please help me to remove this noise.
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2answers
229 views

Resources for learning about filters from scratch?

Topic of filters is very important in signal processing So i want to learn about filters from scratch and then eventually reaching FIR and IIR filters I have some idea about low pass,high pass,band ...
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2answers
100 views

Ideal Lowpass Filters [closed]

I am a beginner to Digital Signal Processing. I took this course of DSP from Coursera in which I am stuck with this question. Since, I am new, please be descriptive. Any help will be appreciated. ...
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28 views

Increase signal to noise ratio

I work with the K-LC6 microwave module. Previously, when the microwave module was assembled on discrete elements, the signal-to-noise ratio was 5. Now, with the transition to specialized microcircuits,...
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2answers
109 views

Converting cascaded biquads to parallel biquads

I need to convert a fourth-order IIR filter made using two cascaded second-order sections with the transfer function: $$ H(z) = \frac{b2_{1}z^{-2} + b1_{1}z^{-1} + b0_{1}}{a2_{1}z^{-2} + a1_{1}z^{-1} ...
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2answers
81 views

3 dB cut-off frequency of first-order IIR high-pass filter

As descipbed in this great answer (thank you Matt L.), the suggested exact formula was very precise. I need a first order high-pass with similar precision. EDIT: (clarification) The low-pass filter ...
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0answers
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Threshold Estimation from a Linear FM Pulse Position Modulated signal

I have a signal chain to detect multi channel LFM (Linear Fm - Hann window being applied as shaping filter for the pulses) pulses as shown in picture below. (Numbers in the picture is in Hz., for ...
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2answers
66 views

Coefficient Scaling of IIR filter to obtain unity gain response

I am designing a 2nd order IIR digital filter : My tf equation with coefficients is ...
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2answers
81 views

Calculating the 3dB frequency of an nth order low pass filter and nth order high pass filter

I am using the 6th order high pass filter in the C code. And the cut-off frequency selected for it is 150 Hz and the sampling frequency being used is 32000Hz. Now, I want to calculate the 3 dB ...
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1answer
75 views

How should we read FIR filter phase response graph?

What should we see from the FIR phase response graph, as an example for a linear phase? And what does this straight line at $0.5\pi$ mean?
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Why when you use any filter, it results first a steepness at the beginning to a certain level?

Is there any tool to avoid that? For instance, I wish after filterin my signal, the filtered signal start from a certain level, in other words, at the offset level. In this case, I've use Butterworth,...
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2answers
78 views

Why FIR uses often direct-form instead of transposed-form?

Even though tranposed-form has less computational burden and memory requirements, why FIR uses mostly direct-form?
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What is the intuitive idea behind FIR filter order and filter parameters? [duplicate]

I know that if we decrease stop-band and pass-band attenuation, and also decreasing the transition band, the filter order will increase as well as filter performance. But what is the intuitive idea ...
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2answers
47 views

Understanding zero-phase filters

I'm reading the book "Digital Image Processing" by Gonzalez and Woods and I'm wondering how their definition of zero-phase-shift filter is equivalent to the one given here. "Digital Image Processing" ...
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1answer
38 views

How to remove the bump from the signal using filters?

I want to have only the slope of the signal. How can I do? I have tried to apply some filters but so far I have not been able to hit the nail on the head. I use Matlab.
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27 points median filter

I trying to build 27 points long median filter on FPGA using Verilog. The filter is 1-D (used to clean noise from a spectrum - I want to clean some random peaks). I already built an 9 points median ...
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40 views

IIR filter relaxation time computation

Let's define the relaxation time of a filter as the delay needed for the impulse response of a filter to reach and remain under an amplitude threshold. Is there a closed-form solution to the ...
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1answer
39 views

why there is no 0 on the right side? (matlab, low pass filter)

I took even number of sample (in this case 6) from sinc, hamming window, and multiple of those two. then after fourier transform, frequency-domain variables have zero only left side, in this case its ...
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1answer
53 views

Applications for the Maximum Phase Filter

A maximum phase digital filter has all zeros outside the unit circle, and has the maximum phase and therefore longest delay for a given magnitude response. Besides the possibility of needing a ...
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1answer
27 views

Remove a signal that is known to repeat every 4 samples at a known phase i.e. exactly 500 Hz signal in 2000 Hz data

I have a situation that the measurement device is generating a known signal in the data at 1/4 of the rate. In this case that is a 500 Hz signal when sampling at 2000 Hz. Now, I can already measure ...
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0answers
24 views

Filter design with arbitrary response [duplicate]

I would like to design a feedforward filter for a current controller. I could of course run MATLAB to brute force an arbitrary response for me, but I have some ranges "don't cares" in the response ...
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1answer
45 views

Extract a signal that is known to repeat every 4 samples at a known phase i.e. exactly 500 Hz signal in 2000 Hz data

I have a situation that the measurement device is generating a known signal in the data at 1/4 of the rate. In this case that is a 500 Hz signal when sampling at 2000 Hz. Now I can extract the 500 Hz ...
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1answer
33 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
44 views

Numerical differentiation on circular data

I have multiple time series of position data (x, y, z) and orientation data expressed as Euler/Tait-Bryan angles (yaw, pitch, roll) obtained from a head tracking device. I need to obtain velocity ...
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2answers
31 views

IIR Cascaded Filter Response, Unexpected Results

I am attempting to understand how to manipulate portions of a cascaded series of biquad notch filters in a control system I am working with. As part of determining the composite response of all the ...
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1answer
42 views

Calculating the cutoff frequency of a highpass digital filter

I am new to DSP and I am trying to find the cutoff frequency of a HP digital filter. I know the equation that describes the system, its frequency response: $H(e^{jω})= 1 - \frac{e^{jω}}{2} - \frac{1}{...
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chromatic and polarization mode dispersion compensation in electrical dmain

I am currently working on adaptive compensation of chromatic and polarization dispersions in the electric domain using LMS (least squares method) and CMA (constant module algorithm) algorithms. I am ...
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1answer
55 views

Polynomial division to separate FIR and IIR in cascade

I have a FIR filter $L(z)$ and an IIR filter $H(z) = \frac{B(z)}{A(z)}$ which are cascaded together as $L(z)H(z) = L(z)\frac{B(z)}{A(z)}$. For specific reasons I want to calculate a parallel topology ...
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2answers
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Digital filtering based on frequency component amplitude… why not?

I've got a very general question. I've just implemented a filter working in frequency domain that simply cuts every frequency component whose modulus amplitude falls below a certain threshold value. ...
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28 views

How to use background noise information to de-noise a (non-stationary) signal

I performed some measurements, which unfortunately include some noise. The measured signals are non-stationary, because the phenomena I am capturing is a transient. Before the transient, I recorded a ...

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