Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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Designing FIR filter

I'm asked to design FIR filter, an ideal low pass filter. I'm given with pass band frequency (wp) and stop band frequency (ws). Now, I have to choose Ns points from stop band and Np points from pass ...
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96 views

What are the tradeoffs for more coefficients for fractional delay filter?

Is there a rule of thumb for how many coefficients to choose for a fractional delay filter? I figure there might be a trade off based on the signal frequency as well as my sample rate (this is ...
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1answer
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How to determine Hilbert coefficients

I want to take a real signal on my FPGA and create a complex one. I have FIR core implemted for a Hilbert Transform, and Xilinx's datasheet on it shows coefficients of: (-819,0,-1365,0,-4096,0,4096,0,...
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28-Band Graphic Equalizer in Matlab

I'm trying to design a MATLAB audio player that includes something like a 31-band equalizer. Right now I have the filters working by using the FIR1() function to ...
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3answers
2k views

Filtering with STFT

I have sucessfully implemented a STFT (+inverseSTFT with perfect reconstruction of original signal, with overlap add, etc.) in order to work on audio files. Using this STFT / iSTFT framework, I have ...
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1answer
23 views

How to detect if the given signal is Data or Clock signal type?

I have the edge information of the signal. I need to find if it's a Clock or Data type signal without iterating through all the points in edge vector... Please provide your valuable inputs
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1answer
284 views

A little bit better than zeroing bins in order to filter

The topic of zeroing frequency bins before IFFT has been discussed here : Why is it a bad idea to filter by zeroing out FFT bins? It was very helpful to understand why it is a bad idea. What do you ...
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1answer
2k views

Design a comb notch filter when sampling rate is not an integer multiplicative of the cutoff frequency

Let's say I have a signal sampled at 976 Hz and I want to remove network interference noise, i.e. 50 Hz and all the harmonics. Is it possible theoretically to design a comb filter to do that? I have ...
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2answers
141 views

How to do a spectral fade out?

I'd like to do a fade out of a sound in a few seconds, but in a "spectral way", with a special curve : (here x-axis is time, y-axis is frequency from a few hz to 20khz). All the part "to be removed"...
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1answer
685 views

Synthesize Digital Filter with Arbitrary Frequency-Response

fdesign.arbmagnphase is a MATLAB built-in function that will create a filter that best fits the frequency-response that it is given. I'd like to implement a similar function in Python. What ...
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143 views

Causal bandstop filter

I have a vector of zeros and ones with length ~2000. I need to detect and filter out the once, that appear periodically. There may be none to 3 such periodic signal's components. Typical periods are ...
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Difference between state space and transfer function model response (in Simulink)

Why I get a different response from the same system (e.g. three phase inverter with LC filter) in state space form and in transfer function (Laplace) form when using the same PI controller values ($...
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1answer
372 views

Very first values of y[n] in an IIR filter

I use a simple biquad IIR filter : ...
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1answer
153 views

Guess the frequency response of a filter with little information

I have a processing unit that does a nice EQ filtering on soundfiles. The only thing I can do is : I give the original .wav file, and the processing unit gives me back the filtered .wav. I have no ...
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3answers
2k views

Draw a frequency response graph from a simple equation

How is it possible to draw an audio frequency response graph (x-axis : freq from 20 to 20 000hz, y-axis : dB) from a simple equation like y[n] = x[n] - x[n-486] ...
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2answers
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IIR issues when order is high

Thanks to all the people who helped me here, I finally achieved to implement a notch filter (*), with coefficients given by : http://www-users.cs.york.ac.uk/~fisher/mkfilter/trad.html When I do order=...
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6answers
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Simplest way to implement FIR filter in C/C++

What's the simplest way to implement a FIR filter in C/C++ ? (I don't want to have to use a third-party software like Matlab or Octave in order to compute the coefficiens because I have to use ...
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1answer
2k views

simplest Notch filter in the world

What do you think of this notch / band reject filter for a very very narrow band ? ...
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1answer
240 views

Scaling of a notch filter

Let's say I have an IIR filter that is a notch type with a peak at -20db. How do I modify the coefficients of the filter so that the peak is at -10 db for example? I know I can scale the denominator ...
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1answer
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Design of 10th order high pass filter using 2nd order biquads with Matlab and further substitution on to TI AIC3204

I am involved in designing a 10th order high pass filter with cutoff frequency 300Hz. My hardware AIC3204 has a 5 biquads of direct form I, 24bit coefficient so I have chosen the FDA (filter design ...
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351 views

Sine-weighted moving average on non-periodic, stationary signal

I am looking for -or trying to create- a filter with a piece-wise monotonic second derivative in such a way that when placed on a non-periodic input signal, changes in sign of the second derivative ...
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2answers
1k views

MACE correlation filter - any suggested values of of u?

I am trying to understand the MACE correlation filter as per the paper Mahalanobis, A., Kumar, B.V.K.V., Casasent, D.: Minimum average correlation energy filters. Appl. Opt. 26, 3630–3633 (1987) ...
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1answer
192 views

Turn acausal Butterworth output into output of a causal Butterworth without input signal

I have a dataset which contains a large number of time-series which have been filtered with acausal Butterworth filters. For a real-time application, I can only use causally filtered data. Is it ...
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3answers
926 views

MATLAB: filter nonperiodic noise in the signal

I have gone through the below related posts of filtering noise from the signals MATLAB, How to filter a discrete signal? MATLAB: Filter a noisy EKG signal But the signal I want to filter contains ...
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2answers
168 views

Digital implementation of type I compensation network

I'm looking for advice or references on how to implement a "type I" compensation network of digitally. An analog example is shown in the following diagram: I am familiar with the basics of digital ...
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2answers
4k views

Mathematical Proof of Complementary Filter?

Having found some unofficial sources on Complementary Filter (Thousand Thoughts Sensor Fusion and The Balance Filter by Shane Colton), I wish to work out its rigorous mathematical proof. The ...
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1answer
205 views

Group cascading IIR filters of different orders

Is it possible to make one IIR filter from various cascading IIR filters of different orders? I know it is possible to do it from filters of the same order, but what about filters with different ...
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2answers
5k views

Removing the overshoot in step response

First of all I'm new to DSP so excuse my simplified words & I'm sorry if it's a duplicated question. Now, I'm trying to filter a signal but as you all know, the filter step response overshoot the ...
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4answers
3k views

parametric eq with FIR filters

Are there any good published approach for designing parametric EQs as FIR filters. The standard techniques discussed in DSP textbooks are frequency-domain-sampling or using Parks-McClellan, which ...
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1answer
2k views

Designing an allpass IIR filter with arbitrary phase response

could you tell me step by step the algorithm to design an arbitrary phase response IIR allpass filter design ? I know Matlab uses a "brute force" algorithm to find an arbitrary group delay response ...
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1answer
976 views

Complex Least Squares FIR filter design using Levinson' s algorithm

i'm trying to implement an arbitrary IIR filter design. I'm decomposing all the algorithm into smaller pieces that i can test. I currently am on the part where it designs a FIR filter from an ...
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2answers
660 views

How does a FIR equiripple filter behave close to DC?

I am multiplying two sine waves with the same frequency (f), but might have a phase difference 0 <-> 90 degrees. The product is a two frequency sinusoidal wave with f1 = f-f = DC and f2 = f+f = 2f. ...
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2answers
1k views

Least-squares digital IIR filter design (with arbitrary responses)

I'm studying the IIR filter design that is described in the book: Algorithms for the constrained design of digital filters with arbitrary phase and magnitude responses. You can get the code at page ...
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2answers
354 views

Filter design for frequencies in range 0-60Mhz

First of all excuse me for asking a seemingly vague question. But i am going nowhere in the problem i am trying to solve. I am working on a powerline communication system where i need to upconvert ...
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2answers
1k views

polyphase sample rate conversion with non-integer factor

I want to do sample rate conversion by subsequently upsampling with factor I=5, and then downsampling with factor D=9. I have designed a nyquist sample rate conversion filter h() of length M, with ...
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2answers
2k 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
108 views

Build discrete IIR from a transfer function which is not in terms of $s$

Firstly I should state that my maths knowledge is limited. I'm currently designing an acoustics modelling application which uses a rectilinear FDTD grid for modelling pressure variations - but I've ...
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1answer
126 views

Apply a filter according to some function

I'm fairly new to digital signal processing. I've been trying with limited success to apply a time-varying bandpass filter to a signal (question at stackoverflow). Instead I thought I'd try a ...
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1answer
2k views

with SciPy, how to generate waveform with common pulse shapes?

With Python/Scipy, if I have a simply BPSK signal sequence [1, -1, -1, 1, ...], how can I generate a real-valued baseband waveforms with oversampling rate ...
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1answer
248 views

What does the stability margin have to do with the quality of a filter?

I designed a 2nd type Chebyshev bandpass filter with passband between 1 and 6 Hz and 30 dB attenuation in the stopband. When I looked at its Bode plot in Mathematica, I noticed "stability margins" (...
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1answer
123 views

How to report how a filter was designed (scientific papers) + how to calculate filter attenuation?

I am looking for a way to completely describe which kind of digital filters I used to process my data, although it seems only few publications in general report all the details. I think I should ...
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2answers
993 views

About designing digital filters [duplicate]

I'm currently using MATLAB's fdatool for filter design. Using that tool, I can easily design different kind of filters. For example, let's take a band-pass FIR filter with 10-40 Hz passband, and 5-10 ...
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2answers
505 views

Implementing a phase-neutral band-pass filter using SciPy

Does the following code properly implement a phase-neutral band-pass filter using SciPy? ...
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1answer
706 views

What are the effects on filtering upsampled data without interpolation?

I have an application that displays several signals from different sensors at various sample rates. In order to display the data, I "stretch" the signal by repeating samples to match the highest ...
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2answers
5k views

Which IIR filters approximate a Gaussian filter?

So it recently dawned on me that Bessel filters, despite being listed along with the other common types, are really an oddball that belongs in a different "class", and I'm trying to learn more about ...
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1answer
1k views

DFT Polyphase Filter bank

What is the purpose of DFT & IDFT blocks which are placed after the Analysis filter bank & before the Synthesis filter bank. The reason i am asking this question is because I tried to design ...
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1answer
149 views

Digital filter performance with noncontinuous signal?

First of all, I'm new to DSP so excuse my simplified words. I'm testing the performance of a digital filter on a (partly) noncontinuous signal: As you can see, the signal is not continuous at some ...
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1answer
5k views

Digital filter design basic principles (IIR/FIR)

Although I have a solid experience in designing audio engines and such, I am fairly new to the realm of Digital Filter Design, particularly IIR and FIR filters. In other words, I'm trying to learn as ...
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818 views

adaptive filter does not converge for all inputs

I am trying to make a frequency domain adaptive filter in matlab. It uses matlab adaptfilt.fdaf to create the filter parameters like step size and initializing initial filter weight values. Then I ...
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3answers
2k views

Detecting a single frequency in noisy data

I am not very experienced in signal processing, and it seems like this should be a simple problem. I have two computers, one plays a tone a 19 kHz at a sampling rate of 44.1 kHz, and the other listens ...