Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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3answers
72 views

Create filter without regard to phase response (amplitude only)

Why I only care about amplitude response I'm working on equalizing a DAC where I only care about the flatness of an AWGN spectrum. Since AWGN has random phase for any frequency bin, I do not care ...
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1answer
44 views

FIR filter | how can I change the axis to unnormalized?

I don't understand why my output graphs and not showing the real frequencies, tried to change it in a number of ways with no luck. so far I'm stuck. This is what I have done so far while implementing ...
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1answer
198 views

Confused about filter's cutoff frequency in rational sampling rate converter

A interpolator which increase sampling rate by L times is followed by an anti-imaging filter, the filter is said to have a cutoff frequency of Fs/2L. Is Fs here is the frequency before or after ...
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1answer
44 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
282 views

Where to learn about "analog prototype filters"?

Where to learn about "analog prototype filters"? I've heard about them, but I'm unsure about what they really are and how they're constructed.
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1answer
115 views

What is the best way of filtering EEG signal for the specific frequency band?

I have an EEG data with a sampling rate is 500 Hz. Actually, there are too many ways to filter EEG signals. However, the abundance of options is not valid for me. I have to use a causal, linear-phase ...
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1answer
84 views

EEG Sampling, Which Order should cover my result?

I have EEG data sampling rate is 250 Hz and I want to band pass this data in ( 0.5-3 Hz & 3-8 Hz & 8-13 Hz & 13-30 Hz & 30-40 Hz). Currently, the data is in the frequency range of ...
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1answer
285 views
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2answers
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Designing high-pass filters using impulse invariance

One of the main disadvantages of realizing digital filters using impulse invariance is aliasing. According to the Nyquist sampling criterion, in order for the frequency response of the digital filter ...
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2answers
66 views

Minimal group delay filter to remove harmonics but preserve fundamental

I have a signal that contains energy at a fundamental frequency (60 Hz) plus significant energy at harmonics (2nd, 3rd etc up to 15th). The signal is sampled by an ADC at 1920 Hz. I want to construct ...
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1answer
842 views

Recursive Implementation of the Gaussian Filter (1D & 2D)

I'm trying to implement an IIR form to approximate the Gaussian Blur Filter. I'm working with the article "Recursive Implementation of the Gaussian Filter" by Ian T. Young and Lucas J. van Vliet. ...
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1answer
30 views

Filter for ISI Channel

I know that OFDM has many advantages in removing ISI. But I think there is one more way of removing ISI(equalizing or inverting technique). Syppose there is a channel $H(\omega)$ and we have our ...
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2answers
208 views

Filter design to realize Cauchy product

I come from Computer Science so please pardon for my possibly wrong terminology. I need to design a filter which has coefficients $$h_0, h_1, \ldots, h_n, \ldots \quad\text{such that}\quad h_0 > ...
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2answers
293 views

High Dynamic Range FIR Filters

Related to this question on using the Windowing method for FIR filter design versus the optimized algorithms such as least squares (firls in MATLAB, Octave and ...
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4answers
7k views

FIR Filter Design: Window vs Parks McClellan and Least Squares

Are there any advantages to use a window approach over Parks-McClellan (further abbreviated here as PMcC) or Least Squares algorithms for FIR filter design of a low pass filter? Assume with today's ...
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1answer
25 views

Estimating change in RMS of noise given FIR filter coefficients

Filtering noise leads to a change in the overall RMS of the noise as demonstrated by the following code-block. which uses scipy's firwin2 to calculate the taps. The ...
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1answer
276 views

Estimate smartphone accelerometer bias

I was planning to develop Android / iOS applications that enable users to measure 3D length using their smartphones. According to this question, you need to know at least the time-varying bias that ...
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1answer
38 views

Implementing HPF using frequency sampling method

I'm on it for a few hours trying to tweak it in all sort of ways but the output comes out scrambled and I can't understand why. I am trying to implement an HPF with a stopband frequency of 500Hz and ...
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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
37 views

how would gradient descent work in optimizing a practical frequency response of a sinc function?

The gradient descent algorithm should work by minimizing the ripples in the stop and pass band, then increase the slope of the transition band. So basically the parameters of interest will be passband,...
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1answer
50 views

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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1answer
594 views

Sampling the impulse response

Given two IIR filters (one Low-Pass and one High-Pass) designed using Butterworth, how can I know if they can be obtained using sampling the impulse response?
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0answers
30 views

How to derive a 2d jinc highpass filter from a lowpass filter?

The following code define a jinc lowpass filter, how to derive a corresponding highpass filter? Is it possible? ...
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3answers
3k views

How to implement Polyphase filter?

I am trying to implement the following 4-Tap Polyphase window shown in figure below and used in : polyphase filters- radio astronomy: lecture #8 by Prof. Dale E. Gary on Matlab. I have tried splitting ...
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1answer
20 views

Find the output signal in a Low Pass Filter for a given cutoff frequency range

I have a scenario where 𝑥(𝑡)=2⋅sin150𝜋t +sin250𝜋t and g(t)=𝑥(𝑡)sin250𝜋 . The signal g(t) is passed as input through an ideal lowpass filter with cutoff frequency(fc)= 300𝜋 and passband gain= 3....
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0answers
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Signal filtering with physical limitations and estimated STD

I have a prediction mechanism that predicts some signal value. Generally the prediction look like this: The are some ranges of systematic error and I cannot do anything with this. But in many points ...
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3answers
367 views

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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1answer
92 views

Why is there a dip in this Chebyshev filter response?

I want to build and simulate a 6 GHz, 4th order lumped-element Chebyshev bandpass filter with a maximum passband ripple of 0.1 dB and a bandwidth of 1 GHz, and as I understand it, there should be 3 ...
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2answers
546 views

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 ...
0
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1answer
548 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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1answer
179 views

Wavelets and bank filters

The wavelet transform has a problem as it gives poor time resolution for low frequencies and poor frequency resolution for high frequencies according to uncertainty conditions. This appears well ...
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2answers
69 views

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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0answers
76 views

implement bandpass filter on complex valued signal? (C++)

I want to filter out complex valued samples - Bandpass filter. I have 100 complex samples and I wish to filter out a range of frequencies from it, as you can see in the result of the FFT, there are 3 ...
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1answer
382 views

Biquad coefficients using Magnitude (or Phase) Invariance Mapping Method

I'm trying MIM (Magnitude Invariance Method) and PIM (Phase Invariance Method) for to improve biquad LPF response at low sampling rates. I'm looking some help and examples of usage if available. ...
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1answer
101 views

How to design a digital Butterworth bandpass filter?

I am looking into designing a Bandpass Butterworth filter in python, but, I was not sure I am designing my filter correctly. What I have are the following: High cutoff frequency = 200Hz Low cutoff ...
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1answer
441 views

python rewrite lfilter (iir) with for

It's a beginner question, but useful to users from python - signal.lfilter, I was using lfilter from Find reverse one pole ...
0
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1answer
66 views

Measure 3-D length using smart-phone's accelerometer

I'm wondering if we can measure 3-D length only by our smart-phone's accelerometer. And we all know these low cost IMUs are not accurate. You can model accelerometer's error this way: $$ a = f*a' + g ...
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5answers
3k views

What's the Q factor of a digital filter's pole?

For the analog S-plane, the Q factor depends on the angle of the complex pole $p$ from the horizontal axis, $\theta = \arg p$: damping factor $\zeta = \frac{1}{2Q}$, and $\zeta = \cos(\theta)$, and $\...
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3answers
164 views

How can the location of the resonant peak in RBJ's shelving filters be calculated?

I'm attempting to use the low- and high-shelf filters from robert bristow-johnson's Audio EQ Cookbook1 for a filter that will have a pronounced peak - high $Q$, with a range of up to +24dB. The goal ...
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1answer
1k views

Common Filter Types for Audio Applications

I'm beginning to write a basic "virtual analog" synthesizer and want to implement a lowpass with variable cutoff frequency and resonance. Does anyone have insight into filter types commonly used to ...
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2answers
3k views

Making an EQ from cascading filters

I hope this isn't a dublicate . I checked but maybe didn't see everything. Here goes : I've been scratching my head over some EQ related questions. I read a lot of papers but my limited math and ...
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1answer
211 views

Filter: IIR Filter

How IIR Filter coefficients formula derived? for LPF, HPF, BPF, BSF there are coefficients formula for a0, a1, a2, b0, b1. How it came? I would like to know the derivation for all those coeffs w.r.t ...
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4answers
5k views

Audio EQ Cookbook without frequency warping

The famous https://www.w3.org/TR/audio-eq-cookbook/ offers a set of [real] biquad filter calculation formulas that generally work fine. However, when filter's frequency approaches Nyquist frequency ...
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1answer
1k views

Prewarping both resonant frequency $f_0$ and bandwidth (or $Q$) when using bilinear transform

RBJ's Audio EQ cookbook takes into account only frequency prewarping when case Q is used for bandwidth. Why not $Q$ prewarping as well with some of those filter ...
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2answers
282 views

Where do RBJ's analog shelving filters come from?

In RBJ's Audio EQ Cookbook, the analog prototype for a low shelf filter is given as such: $$H(s)=A\cdot\frac{s^2+\left(\frac{\sqrt{A}}{Q}\right)\cdot s + A}{A\cdot s^2 + \left(\frac{\sqrt{A}}{Q}\right)...
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2answers
939 views

Help with audio EQ cookbook BPF filters and Q

I am fairly new DSP, and am using maths that I have not used in 20-30 years. I have been working with Cookbook formulae for audio EQ biquad filter coefficients by Robert Bristow-Johnson and am having ...
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1answer
169 views

Question regarding filter implementation - Audio EQ Cookbook

Some of you may know the infamous Audio EQ Cookbook. You can take a look here. We tried to implement the functions there to achieve a implementation for a Lowpassfilter. However, when we try to lower ...
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1answer
361 views

Parameters of Butterworth filter that approximates the amplitude response of a Gaussian filter?

For a given narrowband Gaussian filter with a specific central frequency and filter width, I need corners of a bandpass Butterworth filter whose amplitude response is close enough to the Gaussian ...
2
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2answers
401 views

filter design for ECG

I would like to ask how to design a digital filter to remove $50$ Hz interference on an ECG signal. Do I need to use notch filter? Thanks.

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