Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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Derivative filter in Python

In Alaa Kharbouch, Ali Shoeb, John Guttag, Sydney S. Cash, An algorithm for seizure onset detection using intracranial EEG, Epilepsy & Behavior, Volume 22, Supplement 1, 2011 (section 2.1, 3rd ...
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How does the 'phase' of a cosine wave become 'pi' when it transitions from positive to negative?

I was watching this video lecture on filters where this professor says that: The phase of a cosine wave $\cos(\omega)$ is 0 till $\omega=\pi/2$. As soon as $\omega$ goes beyond $\pi/2$, i.e when $\...
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An explanation of “phase” of a filter

I always read the word "phase" (like linear phase, phase shift...) in DSP but I'm still not sure what it supposes to mean, in intuition and also in practice.
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Minimal step response overshoot IIR notch filter design

Is there any IIR notch filter design algorithm, strategy or rule of thumb, that provides control for amount of overshoot/undershoot in step response of a filter?
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Convolution with an impulse response does not give the same result as the frequency response

Context I am trying to design a linear-phase FIR filter with the following frequency and phase responses: Designing the filter Given its characteristics (peak filter at 1kHz with 9dB gain and Q=7), ...
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Design a linear-phase FIR filter approximating the magnitude of a given IIR filter

I have a biquad IIR filter from which I want to get a linear-phase FIR. I saw that related question but the OP does not care much about phase. I have troubles getting the exact same magnitude curve ...
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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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What is the reason for this 'linear “error”'

Here's MIM (Magnitude Invariance Method) thesis (press the "View PDF" to get the theis with MATLAB code in hands) and here's a plot showing the issue I'm trying to solve: Magnitude responses for one ...
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Filter problems in python

I am recording a 30 second 1000Hz Sine wave, saving it to a list, splitting the list into 5 second sublists and I would like to apply a notch filter to each sublist to check when the signal drops out ...
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How to initialize shift registers in polyphase filter based resamplers to avoid or compensate for group delay?

I have a signal a = [a(t0),a(t1),a(t2),..,a(tN-2),a(tN-1)] I'm using a polyphase filter based resampler with filter length $L$ of $8$, fixed interpolation rate $I$ of $10000$ and decimation rate of $...
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Signal filtering using Python

I'm trying to create an application using python that is capable of recording an audio signal and detecting short glitches in the signal. For example, I could be recording a 1000Hz sine wave and ...
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Whats is the difference between FIR/IIR filters and Chebyshev/Butterworth filters

I am new to Signal Processing. From my understanding -- FIR/IIR just refer to the placement of poles and zeros in the z-domain helping us achieve convolution, if FIR and ??? in IIR. Chebyshev and ...
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Downconverting real signal to IQ baseband

Despite reading various answers on this site and on other sites I am still confused about how to convert a real-valued signal back down to its original IQ baseband values. Note I am still a beginner ...
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1st order filter output

Two 1st order filters : ...
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Design narrow bandpass filter for signal with high sampling rate

I have to bandpass-filter a signal which has been sampled with 4000 Hz. Only frequencies around 15 Hz shall remain after filtering (let's say in a band between 10 Hz and 20 Hz; the narrower the ...
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Which equation should I use to compute the Extended Kalman Filter?

Compute the Extended kalman filter can be done in several ways. The first one compute the convariance matrix $P(t)$ from the Riccati Equation: $$\dot{P} = F(t)P(t) + P(t)F^T(t) - P(t)H^T(t) R^{-1}H(...
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why butterworth bandpass filter changes signal in the passband?

I want to apply a butterworth bandpass filter to EEG signals and it's not working so I tried with a simple signal: t=[0:0.01:10]; s3= 9*sin(2*pi*4*t); Still ...
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Extended Kalman Filter in mechanics, electronics and hydraulics?

Extended Kalman Filter is most used in GPS and navigation systems. But how much do I gain to switch to Extended Kalman Filter to the linjear Kalman Filter if I do LQGI controller for the industry? ...
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Digital filter for sharp impulse in data

I am looking for an algorithm or a filter to remove sharp peaks from my data. I am collecting raw data after every 1 second from an sensor through an ADC. The ADC has an internal FIR filter. I use ...
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214 views

How to analyse anti-causal discrete transfer function using matlab?

Consider a discrete transfer function that represents an anti causal filter such as a derivative filter: $$H(z) = (-z^{-2} -2z^{-1} +2z +z^2) (1/8T)$$ Where T is the sampling period. Normally in ...
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What is limit on performance of edge-preserving filters for low signal-to-noise regimes?

Let's say I have a signal, $x(t)$, defined as such, $ x(t) = \begin{cases} 0 &\text{if} \,\,\, t < -\alpha/2 \\ \frac 1\alpha t+\frac12 &\text{if} \,\,\, -\alpha/2 \leq t \leq \alpha/2 \...
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Complex bandpass filter giving quadrature signal

I have found a technique detailed on page 34 & 35 of this thesis. In this, the author states a complex bandpass filter can be derived by first making a prototype real valued IIR filter, then ...
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332 views

Why did a digital HPF increase the gain of high-frequency signal?

Background: I was trying to remove the DC component of a load cell signal using a high-pass filter. I don't care about that anymore. I'm curious why a digital filter and an analog filter behave ...
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933 views

Find smoothed first derivative from signal with noisy slope

How can I get filtered first derivative from a noisy signal that has slowly changing slope in form of y=kx+b? k can slowly ...
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54 views

Resolving two nanosecond pulses with filter

Problem: I have to design a filter that should be able to resolve two ~20ns pulses as well as get rid of some ringing. I was able to find a filter kernel that if convolved with the noise signal is ...
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Are all least square filters adaptive?

Are least square filters, or filters that minimize error energy, the same as least mean square adaptive filters?
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Band edge filter for raised cosine impulse

I'm hoping to use an implementation of a frequency-locked loop for rough frequency synchronization in a PSK31 demodulator. The approach is to define a filter that is the derivative of the matched ...
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Filter design - Rules of thumb regarding ratio of sampling and stop/pass frequency

The issue: I have a signal in the time domain sampled at 51.2KHz. I want to apply a highpass filter with a pass frequency of 20 Hz. The problem: Using Matlab to do that, the design of the filter ...
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Confusion regarding a library of Digital Signal Controller

I'll be using a NXP DSC with 56800Ex core and my aim is to implement a Elliptic IIR Filter in it. The User Reference Manual of the 'General Digital Filters Library' (which is available here) has ...
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Filter design with zero - pole placement method

I have some question for you about Filter design. Α. Calculate the transfer function in order to stop the frequency of $300\textrm{ Hz}$ for sampling frequency at $12\textrm{ kHz}$. Use the zero – ...
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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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FIR filter design in matlab using Kaiser window (using 'fir1' and 'designfilt')

Here are the specs for FIR filter using Kaiser window. ...
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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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Filters using equiripple approximations (different from type I lowpass)

In Oppenheim & Schafer's Discrete-Time Signal Processing, only type I lowpass filters are explained in detail in the chapter "Optimum Approximations of FIR filters". I want to know what happens ...
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How to implement bandpass filter on complex valued signal?

I am using Scipy to implement bandpass filter but it assumes that positive normalized frequency is passed but I & Q samples range from [Fc-Fs/2,Fc+Fs/2] where Fs is sampling frequency & Fc is ...
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683 views

How to Design Nonlinear Phase FIR Filter?

I've recently encountered a rather clever low-pass filter that appears to have been designed to sacrifice uniform group delay in the pass-band to get better frequency response relative to the filter ...
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High-pass filter in Python (Scipy)

This code is taken from a pitch detection algorithm. It is called before pitch detection to remove low-frequency noises. ...
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Latency handling at filter output

In MATLAB, latency at the start of the output of a filter can be handled by discarding those bits and starting the signal where the latency ends. but how can the garbage value at the end be handled?
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239 views

FIR filter design with flat passband but equi-ripple stop-band

I am looking for a lowpass FIR filter with flat passband but equi-ripple stopband. In other words, it likes Chebyshev_filter Type II except that it is FIR instead of IIR. Linear-phase is preferred. ...
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333 views

How to Add Group Delay?

I was confused about the value of group delay, I am not sure whether my opinion is correct. For example, if I want to add a group delay of $0.25$ time units to a zero-phase signal, does it mean that I ...
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240 views

FIR Filters - Type 3

So I understand that a type 3 filter is not suitable for a highpass filter design, but is there any reason why it isnt suitable for a lowpass filter? So ultimately, can a type 3 linear-phase FIR ...
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155 views

IIR filtering vs IFFT filtering by zero binning and making bandpass IIR filters with high sample rates?

I am having some issues making thin IIR filters that work at the sample rate of my data and was wondering if there is a way to improve my method? I am also curious about the deficiencies in using the ...
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Why would one use a Hann or Bartlett window?

Suppose we're designing a low-pass FIR filter, and I want to use one of these three windows: Bartlett, Hann or Hamming. From Oppenheim & Schafer's Discrete-Time Signal Processing, 2nd Ed, p. 471:} ...
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Filter image in Fourier space imaginary components after ifft (R)

I want to high-pass filter an image in Fourier space. It works but the retransformed image has an imaginary part, what shouldn't be the case http://math.stackexchange.com/questions/867337/how-to-...
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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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213 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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Is there a tool for automatically converting filter implementation forms?

I am looking for alternate tools that will automatically convert filter implementation forms (or signal flow network diagrams in general) to simplest or alternate forms; for example for going between ...
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Why low-pass FIR Equiripple filter is designed with 3 dB of ripple in the pass-band and at least 60 dB of attenuation in the stop-band?

Please tell me, why low-pass FIR Equiripple filters are designed with 3 dB of ripple in the pass-band and at least 60 dB of attenuation in the stop-band and not other values ? I want to know are ...
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415 views

Rectangular or trapezoidal signal filtering. Beginner

I have the following signal: I know, that there are 2 states ('High' and 'Low') in my real signal. I need to convert my signal to the sequence of these states (Low from 2 to 5 second, High from 5.5 ...
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Eigenfilters in Matalb [closed]

I have to implement an eigenfilter for an arbitrary frequency response in MATLAB. I have this algorithm: $N$ - order of the filter $M = N/2$ $c(\omega) ...