Questions tagged [filter-design]

Fiter design is the process of designing a filter.

Filter by
Sorted by
Tagged with
2
votes
1answer
349 views

When does a Window function's sidelobe level become out of band attenuation?

Another question I'm struggling to answer by myself. For a Window function, say Blackman (response pictured), what is the out of band attenuation? If I'm designing a filter with out of band ...
3
votes
0answers
166 views

What is the meaning of a inner system/transfer function?

In an academic paper I'm currently reading [1], the notion of an inner system is central. It is defined as follows A (linear) system $G$ (with $m$ inputs and $p$ outputs) is called inner if it ...
2
votes
1answer
115 views

Filter response is changed according to its order

I want to build a band-reject filter for the 60 Hz noise. So, I tried to use Butterworth filter to implement this specification. However, when I chose an order of 8, I got the following response, ...
-1
votes
1answer
131 views

Properties of Optimization Techniques in Filter Design

I have this design: Can you tell me about the type of filter and the algorithm just viewing this design, or should there be other information?
3
votes
0answers
237 views

Least Squares Approximation for FIR Filter Design

Following this paper , I am trying to make a least-squares algorithm in MATLAB, but for type I (I know about firls()). ...
2
votes
0answers
263 views

Applying Jake's remez exchange algorithm for filter design

I have found C implementation for remez exchange algorithm(https://github.com/janovetz/remez-exchange/blob/master/remez.c) and I want to use this to implement a generic prototype lowpass filter. ...
5
votes
1answer
289 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 ...
0
votes
1answer
43 views

Determine proper coefficients for the transfer function

The problem says that we have the information (actually a .mat file that contains the data) of 1 pixel from an IR camera from an object with 26° of temperature. Because of the noise, the output is ...
-1
votes
1answer
353 views

DWT versus band-pass filter

Which technique gives better result in extracting a band from a signal, let's say the alpha band (8-13 Hz): using a band-pass-filter or a DWT (Discrete Wavelet Transformation)? P.S.: I am working in ...
1
vote
1answer
411 views

Frequency Translation in MATLAB

I have to design a generic baseband filter. I have a C++ code which gives me filter coefficients based on the input parameters such as no. of taps, no. of bands, band edges, amplitudes and weights(...
1
vote
2answers
428 views

Looking for a simple and effective notch filter

I ran a finite-difference simulation and the behavior of an output signal, $s$, in time, $t$ (sampled with period $\Delta t$) behaves approximately as in the figure below. It is well-described by a ...
0
votes
1answer
148 views

Is initializing a digital filter's output with no “momentum” a non-trivial task?

I'm working on implementing a filter with a very slow step response. This filter is implemented as a cascaded second-order-section filter (transposed direct form 2). I'm using the output of this ...
0
votes
2answers
211 views

Is there anything called uncorrelated filter?

I understand to an extent various filter like low pass filter, high pass filer, Wiener filter Kalman filter etc. I also understand some of this filter will decorrelate/uncorrelate the signal. The ...
0
votes
1answer
71 views

can $\frac{1}{H(z)}$ be causal and stable? [duplicate]

if we have linear phase FIR filter $H(z)$ which is causal and stable can $\frac{1}{H(z)}$ be causal and stable ? can it be causal without been stable ?
1
vote
0answers
154 views

Speaker equalization for white noise

I need a speaker to play broadband white noise for my scientific application. I would like to know whether my relative naive implementation is correct or not. The speakers I use have a large frequency ...
0
votes
2answers
2k views

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 ...
0
votes
1answer
72 views

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 $\...
0
votes
1answer
47 views

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.
1
vote
0answers
200 views

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?
0
votes
1answer
272 views

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), ...
2
votes
2answers
532 views

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 ...
0
votes
3answers
778 views

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 ...
2
votes
0answers
109 views

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 ...
0
votes
1answer
341 views

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 ...
1
vote
0answers
54 views

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 $...
5
votes
1answer
3k views

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 ...
0
votes
1answer
764 views

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 ...
0
votes
1answer
1k views

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 ...
0
votes
1answer
186 views

1st order filter output

Two 1st order filters : ...
1
vote
1answer
1k views

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 ...
0
votes
0answers
44 views

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(...
1
vote
1answer
2k views

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 ...
1
vote
1answer
69 views

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? ...
1
vote
0answers
135 views

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 ...
0
votes
1answer
236 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 ...
2
votes
2answers
99 views

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 \...
3
votes
0answers
283 views

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 ...
1
vote
2answers
345 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 ...
2
votes
2answers
1k 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 ...
0
votes
1answer
55 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 ...
3
votes
2answers
84 views

Are all least square filters adaptive?

Are least square filters, or filters that minimize error energy, the same as least mean square adaptive filters?
2
votes
1answer
389 views

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 ...
1
vote
3answers
2k views

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 ...
0
votes
1answer
43 views

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 ...
2
votes
1answer
7k views

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 – ...
1
vote
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 ...
1
vote
0answers
2k views

FIR filter design in matlab using Kaiser window (using 'fir1' and 'designfilt')

Here are the specs for FIR filter using Kaiser window. ...
4
votes
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 > ...
1
vote
1answer
68 views

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 ...
3
votes
1answer
2k views

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 ...