Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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247 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 ...
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16 views

Moving average filter using Blockram

I implemented ADC on FPGA. Sampling frequency of ADC = 1.563MHz. Reconstructed signal frequencies range =0-3khz ENOB achieved for 100hz signal is 7.0844 and SINAD is 44.81db. Now, I designed a ...
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2answers
112 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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1answer
23 views

Removing vocal but retaining all background ambient noises

I have one audio file containing human speeches, a lot of ambient noises like audience laughter, birds chirping, sounds of natures etc.. Now I want to separate it into two audio files where one ...
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1answer
45 views

Real Fast Fourier Transformation (FFT) in 2D changes with changing the axes order

I am using Python to calculate the real FFT for a 2D array. I found that the real FFT function does not return an array with the same size as the input array, rather, it returns an array with the same ...
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12 views

Phase concentration for a broad range of frequency over time

I have local-field potential signals from two animals and I am trying to do phase analysis on these signals. What I am interested in is how phase concentration (or interatrial phase coherence) changes ...
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1answer
22 views

Parallel form filter design

I have a transfer function coefficients as following: [b,a] = ellip(4,.2,40,[.41 .47]); I'd like to factor this system into its PFE using ...
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40 views

pressure decay filtering

So, I've been dealing with a series of pressure decay curves from a pressure transducer. Each one has a waterhammer effect as a consequence of shutting down pumps. Here are examples of 3 curves: I'm ...
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1answer
76 views

2D Multi Scale Dot Enhancement Filter based on Gaussian Filter and Hessian Matrix

I'm trying to implement an algorithm that enhance dot and line like structures based on Hessian matrix, the algorithm uses Gaussian filter with different scales before calculating the hessian matrix ...
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1answer
29 views

Cut-off frequency in Fourier-Spectrum of a noise message

I have a problem involving cut-off frequency. I have a voice message in an audio file .mat which have an eletromagnetic noise. I am trying to recover the original message by applying digital ...
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1answer
32 views

How to make signal smooth using filter?

I have measured the output of a mechanical system and obtained the signal shown at the attached image. I want to make it smoother by removing the chattering (I think it is called chattering). In order ...
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2answers
309 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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35 views

Bandwidth in the filter-Hilbert method

I understand that the Hilbert transform on broadband data produces uninterpretable results. And so we first bandpass the data, and then run the Hilbert transform. My question is, how wide can the ...
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1answer
55 views

Is gradient vector of pole zero carry usful information?

Consider a transfer function (TF) with $n$ number of poles $(p_1,..p_n)$ and $m$ number of zeros $(z_1,..z_n)$. One can write the magnitude of the frequency responce of the TF in terms of poles and ...
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2answers
141 views

Wavelet Filter Coefficients from Scaling Filter Coefficients

I am trying to develop a new type of wavelets and I found out a function that following a particular two scale relation. The function at a scale $t$ say $x(t)$ can be related in a finer scale $x(2t)$ ...
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52 views

Convolution of a non-symmetrical window function by a cosine signal in the frequency domain

A have a time signal: The associated DFT spectrum of this signal: The time signal can be considered as a non-symmetrical rectangular window function multiplied by a cosine signal with a frequency $...
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6answers
23k views

Why is a linear phase important?

If symmetry conditions are met, FIR filters have a linear phase. This is not true for IIR filters. However, for what applications is it bad to apply filters that do not have this property and what ...
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1answer
58 views

What are the limitations of FIR filters?

I know that for type II FIR filter it adds a zero at $z=-1$ so it cannot design a high-pass filter. My question: what type of limitations the other types have (type I, III, IV)?
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1answer
67 views

How to perform spectral inversion in the frequency domain to convert a low-pass filter into a high-pass filter?

To convert a linear-phase FIR low-pass filter into a high-pass filter with the same cut-off frequency, we can invert the sign of the low-pass filter's impulse response $h(n)$ and then add one to the ...
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1answer
4k views

What is the most efficient type of software bandpass filter?

I'm writing a C++ application to extract four modulated carrier signals at four separate frequencies, from a single input signal. I've designed four FIR filters at t-filter.appspot.com but they run ...
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1answer
88 views

Use recursive least square as an “actual” filter?

I am currently studying recursive least square (RLS) and as far as I understand, the setup is (for example) that we are given a process with uncertain parameters/noise and we want to adapt parameters ...
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4answers
28k views

What is the true meaning of a minimum phase system?

What is the true meaning of a minimum phase system? Reading the Wikipedia article and Oppenheim is some help, in that, we understand that for an LTI system, minimum phase means the inverse is causal ...
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1answer
34 views

To find the filter Co-efficients of the following equation

Given $y[n]=h_1[n]x[n]+h_2[n]x[n-1]+h_3[n]x[n-2]$ is an LTI system with unity gain at $\omega =0$ and zero gain at $\omega =\pi$. $h[n]\neq0$ .Also given that the system has a linear phase. Compute $...
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2answers
52 views

Where should I set my anti-aliasing filter corner frequency for this signal?

I have a Fourier transform of an arbitrary digital signal. The transform has For 1024 frequency bins. The magnitude plot of the first 100 is below, with no surprises in the unplotted 924 bins. Now, ...
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0answers
22 views

Bandpass Experiment: Weird graph! amplitude mismatch?

Why is this generic BP filter seem to magnifying amplitudes when I run it at lower sample rates? The bass range seems to 32 times louder at 1275 Hz (44.1 kHz / 32) than at 44.1kHz? Here is a graph of ...
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1answer
44 views

How do I interpret this block diagram correctly?

I am reading notes ahead of the class and have encountered this particular slide: While I completely agree with the first block diagram, I am at a loss trying to understand how the second block ...
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2answers
6k views

Applying Lowpass (LPF) and Highpass (HPF) Filters to an Image in Frequency Domain in MATLAB

What I want is simply to apply low pass and highpass filters to an image in the frequency domain. Does MATLAB image processing toolbox have any commands for this?
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30 views

Wiener Filtering to perform minimum mean square estimation

As the Title suggests, I am trying to design a Weiner Filter for getting the lower bound on my LTE channel estimation algorithm. To give you a high level on the chain, First after generating the ...
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1answer
45 views

Are MATLAB function zp2tf() and tf2zp() are complementary or not?

I was under impression that given, pole, zero and gain the transfer function (filter coefficients b and a) is fixed. Therefore, ...
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1answer
54 views

How to smooth filter response?

I have a filter designed in matlab with the function cheby2( N, Rs, Ws, 'stop'). The filter would give nice frequency response with for a given parameter set when the filter order is 2 or 4 (N=4). But ...
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1answer
24 views

Bessel filters: linear passband phase characteristic only for low pass filters?

I've found a test (internet) from the contest to signal processing teacher and it has a question (true) or (false). I cant find any book or paper to argue this statement. Is it really true?
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42 views

FIR bandpass filter for complex baseband signals

I am DSP newbie and I need to apply a bandpass filter to the complex baseband signals of bandwidth 20MHz and sampling rate 61.44 MHz. Below is the octave code for the same and I have obtained ...
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1answer
37 views

How would I calculate the DC Gain for a Chebyshev Type 2 filter?

I'm currently working on an implementation of a Chebyshev Type 2 filter but the gain is all over the place when playing around with the order and ripple. I haven't found anything online but (as far as ...
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1answer
214 views

Filtering on time or frequency domain?

I'm applying an ideal filter on the frequency domain. My code below: ...
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2answers
55 views

How would one design a (quasi) linear phase adaptive notch filter for a single complex tone?

While IIR notch filters are attractive, I need to retain phase linearity at the filter output. I imagine that it's possible to use a standard IIR notch filter: https://www.researchgate.net/...
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1answer
48 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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0answers
22 views

How to design a filter with smoothness regularisation?

I am rather new to the DSP topic. However, I would like to filter a signal with lowest error and highest smoothness. Until now, I was using tikhonov regularisation inside batch regression with some ...
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2answers
169 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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4answers
26k views

How many taps does an FIR filter need?

I am looking to design a set of FIR filters to implement a low pass filter. I am also trying to reduce the latency of the signal through the filter so I am wondering what the minimum number of taps I ...
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62 views

Can Kaiser's formula be used to estimate the filter length of an FIR filter for all windowing methods?

I came across this problem while designing a FIR filter for given $\omega_p = 0.4\pi$ and $\omega_s = 0.6\pi$ along with the minimum stop-band attenuation $\delta_s$. I have worked out some examples ...
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3answers
112 views

Matlab IIR cheby2 bandpass, problems with Magnitude

I am trying to implement an IIR bandpass in Matlab. There are two things I don't understand. But first of all, let me post the code. ...
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2answers
60 views

Parallel equalizer, working for boosts but not cuts

I am trying to replicate the behaviour of a parallel equalizer. The approach is to use a parallel sum of bandpass filters like so: $Response = 1 + bandpass1 * (gain1 - 1) + bandpass2 * (gain2 - 1) + ...
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1answer
281 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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2answers
2k 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
186 views
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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 ...
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18 views

What is the meaning of these Fixed Point Properties e.g.CustomSectionInputDataType?

I was trying tom implement a Butterworth filter and i came across a example on Mathworks site which emphasizes to customize various terms related to the filter. Following are some of the terms i could ...
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1answer
170 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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0answers
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Vary poles/zeros of digital IIR filter

Many applications that produce sound, such as software synthesizers, are able to apply a filter that varies with time, such as applying a low pass filter that varies with an LFO. I currently have a ...
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1answer
102 views

Low-Pass Filtering of not evenly sampled signal

I have a signal sampled unevenly over 1 million ns. the signal is sampled over 1GHZ clock and the samples are as the following: 0-100 ns - sample every 1 ns. 100-1000 ns - sample every 10 ns. 1000-...