Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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removing noise using Scipy.signal.butter:

I am going to remove the noise from a brain recorded signal. It was a continuous recording and with sample rate=30kHz, it was digitized. So now it is a digital signal. I have written the code here ...
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phase response of an analog filter

I am trying to understand the relation between phase response of an analog filter and its influence on the quality of a signal with PSK modulation. For example I want to design a BPF for HF ...
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C-Weighting filter

I'm trying to implement "accurate" C-weighting filter for range 20Hz-20kHz by using other than bi-linear transformation method. What I've achieved so far is 10th order filter by using Magnitude ...
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Why is the length of smoothed data from filtfilt always double of the input?

I am trying to filter the data I have using python with the code : ...
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best technique to remove noise from bio signal data

I'm still learning about signal processing so sorry if this is an intro problem. I want to remove noise from the following data so I can separate each wave and run analysis on it. The data is sampled ...
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How is the formula of the cutoff frequency of a 2nd order Butterworth low-pass filter derived?

According to articles and notes, the cutoff frequency of a 2nd order Butterworth lowpass filter: is expressed by the formula: $\omega _c = {1\over\sqrt{R_1R_2C_1C_2}}$ and since $\omega _c = 2 \pi ...
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Low pass filter with different non linear phases [closed]

I want to design low pass filters with different non linear phases. Means the magnitude response of all low pass filters will be same but having different non linear phases.
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High Order Buttersworth Filter

Let's suppose i want to implement a Buttersworth filter of order 12 (N=12). First of all ,I can decompose the Buttersworth polynomial in elemental polynomials: Now, if i know how to physically ...
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Transfer Function vs Frequency Response

Talking about Butterworth filters my prof said that a filter is described by its frequency response $H(\omega )$. What I think is that filter is really described by its transfer function, since there ...
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Is a notch filter for 50 Hz without delay possible?

I am processing an EEG signal (2 kHz sampling frequency) from hardware and the built-in filter doesn't work as intended. FIR gives a delay, and IIR is unstable. The signal of interest is from 4&...
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37 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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What is effect of fractional sample delay effect on FIR Filters?

In the DSP lecture slides we have this slide: And So why this part of FIR Type 2 and Type 4 filter impulse response equation bolded in this slide? Thanks for your attention.
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What is importance of the conjugate reciprocal and none reciprocal zeros in FIR Type 2 Filters?

In DSP class we have this slide: So i like to know why this classification is so imoortant? I have seen this post : Conjugate reciprocal pairs of zeros and poles in ... Assuming the impulse ...
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Justification of bilinear transform

I would like to understand the "justification" for the bilinear transform. The basic idea as I understand it is that by integration rule of Laplace transform we have for continuous $y(t)$: $$\mathcal{...
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Where can I find more information about how to create filters on frequency domain?

I'm newbie and I'm very interested in filtering images on frequency domain. I haven't found a lot filters on frequency domain and I wondering how can I create new ones. How can I create new ones? ...
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Constraints for Type 3 Linear Phase Filter

This might be a strange question - but most of the Type 3 LPF I'm seeing are having this in common for $h[n]$, assuming $h[n]$ is real: We cannot have outermost element of h[n] lower than middle ...
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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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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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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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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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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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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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36 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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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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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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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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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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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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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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56 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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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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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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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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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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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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55 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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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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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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63 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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129 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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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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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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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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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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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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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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Interpolation FIR filter Remez design criteria

I want to upsample an audio signal by a factor of 8. The technique I was going to use was to upsample by zero stuffing and then interopolate using a low pass anti-aliasing FIR filter designed using ...
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Why does sampling frequency affect the filter results?

I have a big issue, basically my goal is to create a low-pass butterworth filter around 0-35Hz with a sampling frequency of 5000Hz. My Matlab code is ...
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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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Downsides of higher order filtfilt's

I understand that with a standard filter, higher order will yield sharper frequency selection at the cost of increased phase delay. However, with so-called zero-phase-distortion filters like that ...

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