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Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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Given a filter (coefficients) how can I make it minimum phase?

Consider we have a low pass FIR filter (or other type of filter) that means the filter coefficients are given. Now we want to have the same (or almost) frequency response but a minimum phase filter. ...
4
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1answer
133 views

Efficient implementation of 2-d circularly symmetric low-pass filter

How can a two-dimensional ideal circularly symmetric low-pass filter or its approximation be efficiently implemented on data sampled on a square grid? I'm referring to an ideal filter with a spatial ...
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1answer
861 views

Minimum phase FIR method

I am trying to make a minimum phase filter (in wxMaxima) according to these steps: first create a "normal" FIR (a simple sinc, wc=0.4, random example, but in the pictures a remez with Octave) ...
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0answers
41 views

Understanding the magnitude of frequency response filterbank based on elliptic filters

I've implemented a 10-channels filterbank with octave-scaled second-order elliptic filters using the Python's library scipy.signal. Here is the magnitude of the frequency response: Can someone ...
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0answers
10 views

remove \ minimize first side lobe ringing artifacts due to data truncation

I have a truncated data set that I need to transform, for example: ...
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1answer
71 views

How to prove a filter has linear phase response?

I have designed an FIR filter to have linear phase response using odd-symmetry design. The coefficients of this filter are {2,1,3,1,0,-1,-3,-1,-2}. I am now being asked to prove it has linear phase ...
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0answers
26 views

How to design a bandpass FIR filter?

I want to design a bandpass filter (BPF) using a FIR filter by the window method. I have $f_{c1} = 3\, \mathrm{kHz}$, $f_{c2} = 6\, \mathrm{kHz}$ and slope is $-80\, \mathrm{dB/dec}$. How can I find ...
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1answer
34 views

Bandpass filter with very high sampling rate

Consider a bandpass filter with Low Cut 17Hz, High Cut 22 Hz, Fs = 45000 Hz and Order = 6. When I pass a mixture of multiple sinusoidal waves through this filter (with a sine wave of frequency 20 Hz), ...
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1answer
95 views

Filtering on time or frequency domain?

I'm applying an ideal filter on the frequency domain. My code below: ...
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1answer
46 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
51 views

What is the proper way to filter out unwanted noise from accelerometers?

I'm a beginner in signal processing and have a hard time to filter out environmental noise from the floor vibration data collected using accelerometers. I have tried using butterworth filter (IIR) ...
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2answers
5k views

What is the difference between a lag filter and “PI” control?

A lag filter/compensator has the form $$G_c(s) = \frac{s+z}{s+p}$$ with $-z < -p < 0$. In practice, the effect of lag compensation in feedback control is to increase the DC gain of the open-...
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2answers
162 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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1answer
35 views

How to design a a filter and how to determine the poles and zeros on the basis of following informations?

A sinusoidal signal of 300Hz frequency,3V amplitude is contaminated with line frequency (50Hz,1V). Design a digital filter including poles and zeros to remove the interference component. Consider the ...
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0answers
38 views

Where to start in the design of my filter to remove 50 Hz

pick any 3 random files from this database, : https://physionet.org/pn3/ecgiddb/ They are subject to 50 Hz powerline interference. We wish to convert from time domain to frequency domain and remove ...
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1answer
36 views

FIR filter design with time and frequency domain constraints

I deal with the problem of the synthesis of the filter with the requirements of both the frequency and time characteristics. Now there is a filter that meets the requirements. Roughly speaking the ...
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1answer
75 views

Change durration/tempo of a .Wav file without introducing pops or pitch shift?

I am attempting to develop a similar function as Audacity's Effects>Change Tempo effect. Or more specifically: I am in need of a way to create a funciton in my ...
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0answers
11 views

How to choose the order of the filter - Capon's method

I am studying power-spectra and I want to find the best estimation method for my problem. I stumbled upon Capon's method (sometimes also called minimum variance), which on paper looks very nice. On ...
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2answers
1k 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
176 views
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2answers
1k views

What is the difference between the sampling frequency of signal and sampling frequency of filter

I believe that there is no connection between the sampling frequency used for converting an analogue filter to digital filter and the one used to sample a signal that the filter will be used on. But I ...
2
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1answer
137 views

How does a digital filter work?

To introduce my situation: I'm developing a digital synthesizer in a form of a C++ library, working with low level APIs like WASAPI, ASIO, ALSA etc. It's probably not very practical and I'm mostly "...
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2answers
4k views

Filter Order Rule of Thumb

Is there a rule of thumb for calculating the approximate filter order of a FIR/IIR based on passband vs sample rate? I seem to remember reading about it once in Undergrad. I'd think you could make a ...
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0answers
24 views

Seperation of multi-carrier signal using Square Root Nyquist filter

My goal is to separate multi-carrier to single carrier outputs. I intend to do it using Polyphase FBMC. Each of my carrier is 10 MHz in bandwidth. I have 256 such carrier in the channel. Thus I ...
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1answer
245 views

Weighting function for output error IIR filter design

I noticed, when I try to fit filter coefficients to a given complex transfer function with the output error method, implemented for example in the MATLAB function ...
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0answers
14 views

Concept of scales in Gabor filter

I've been reading about Gabor filters, and there is a concept of scale. While I understand that Gabor features help in analyzing both the frequency and orientations of an image, what is the idea of ...
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1answer
170 views

Implementation of halfband pass filter

I need to interpolate a complex valued bandlimited periodic function using local interpolation. I can have the signal sampled at any frequency I want over at equispace intervals. I am aware that for ...
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0answers
19 views

Is it possible to create a causal filter with a positively-sloped phase response?

I am currently prototyping a set of noise-cancelling headphones that utilize active EQ to modify the noise-cancelling signal, allowing the user to modify the frequencies that they hear or don't hear. ...
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1answer
257 views

Is there a $z$-transform like for variable sampling rate signals?

I'm working with signals with variable sampling rate (the time space between samples is not constant). I know the delay between samples but I don't wont to interpolate the signal. Is there a way to ...
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2answers
160 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 ...
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2answers
4k views

Low pass vs band pass filter

I've been reading some EEG papers where the authors use a bandpass filter at, say, 1Hz and 30Hz. I've also seen some accelerometer papers where they bandpass at 0.5Hz and 10Hz What is the difference ...
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1answer
105 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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1answer
313 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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0answers
30 views

Python scipy firwin2 has a low frequency error when designing arbitrary amplitude filter

I've been trying to design an arbitrary amplitude FIR filter which has a specific attenuation for each frequency. However I get an error at low frequency when comparing the results to my expected ...
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1answer
361 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 ...
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1answer
272 views

Filtering White Noise to Generate Harmonic Signal

I have studied convolutions and filters a long time ago. Today, I am trying to review the basics using some notes of mine, but I am finding difficult to solve easy problems. Since I don't have ...
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1answer
235 views

Least Squares with Non Zero Mean Noise

What happens if the noise has no zero mean? I mean, if the exercise is something like: $$y(k) = A x + \eta(k)$$ When I have zero mean, I start from: $$y = A x$$ $$\Rightarrow \hat{y} = A \hat{x}$$ ...
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1answer
44 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 ...
2
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1answer
128 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
201 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
104 views

Stable digital filter with leading zeroes

I'm looking construct a stable pole-only filter where the feedback coefficients start with a block of zeroes, i.e. \begin{align} a_0 &= 1\\ a_i &= 0, \textrm{ for}\ 1 \le i \lt k\\ a_i &\...
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0answers
24 views

FIR filter window method [duplicate]

In the window method for filter design it is explained that we do an ideal filter and then we pass it to time domain, but it will be time limited and then... Why don't we filter directly in the ...
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0answers
17 views

Signal ECG , collecte & visualize data

I am currently working on a sensor that visualizes an ECG signal and visualizes its data. Its principle is that it has 3 electrodes that must be placed in specific parts of the body .... I would like ...
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0answers
12 views

Frequency warping while creating filter coefficients

If I have an audio signal at 16kHz with frequencies band-limited to 8kHz. I want to low-pass filter this signal such that fc=4000Hz. Meaning when I reconstruct the signal I do not want any content ...
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0answers
15 views

Need help with detecting onset of audio in wav file

I'm new to DSP and I'm learning as I go. I have an audio file with a chirp signal(2kHz). I played and recorded this chirp signal and the recording has a lot of environmental sounds. Here is my code:
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1answer
37 views

Electret microphone RC filter design contradiction among information sources

I'm building a circuit for an electret microphone and I want to build a bandpass filter around the op-amp. I was using this source until I have found that most sources ( e.g. here and here) indicate ...
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3answers
823 views

How were windows originally conceived?

I am aware of the common types of windows, (Hamming, Hanning, Kaiser, Tukey, etc etc). However while many books describe them - almost none tell me just how exactly they were derived. What is so ...
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4answers
14k views

Designing High Pass Filter to Remove DC Offset from a Signal

I want to design a high pass filter (f0*S/1+f0*S) in order to remove the dc offset from a signal ? (i.e. how to choose the frequency f0). Thanks, Hacene Ps : the filter is for MATLAB implementation, ...
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3answers
2k views

How to select the center frequency and filter type for 5-band digital audio equalizer?

I have a small team project: use Matlab implement a Gui 5-band digital equalizer using FIR and IIR filters (2 cases) - The sample rate of all audio files is standard audio 44.1kHz. Coding is not my ...
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1answer
1k views

Filter - Spatial Domain Versus Frequency Domain

Note: This is a followup question of How can I create a custom band-pass filter?. Can you explain the following python source code?. Bandpass filter bank. Suppose, I have an equation ...