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Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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3answers
153 views

Why make the phase of an FIR filter linear?

I am new to DSP and filter design. It is easy to design an FIR filter with linear phase by making the coefficient sequence symmetric. However, why make the phase linear?
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0answers
40 views

Discrete Hilbert transform enhancements?

Consider the Discrete Hilbert Transform, which takes a sequence $u[n]$ defined on the interval $-M\leq n\leq M$ and returns the discrete convolution $$\tilde u[n]=\sum_{i=-M}^Mu[i]h[n-i]$$ where the ...
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1answer
101 views

Filter bank for complex spectrum

I'm using a complex-valued baseband signal as my input for subband adaptive filtering. Therefore i have a complex input spectrum. My problem is now, how to split my complex spectrum into $N$ equally ...
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1answer
625 views

Implementation of the filter function (Matlab) C/C++

I am trying to apply some filtering in C/C++ code, I have designed an algorithm using Matlab and it works well, so now I am moving it to native code. I am ...
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1answer
169 views

Best way to process this signal

I am novice in signal processing, and i have to aplogise for my english level. The problem is the next: I work for foundries, in this sector the cast parts are manufactured pouring liquid metal in ...
2
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2answers
104 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
85 views

Design of a filter with given bandwidth and sidelobe level using Parks–McClellan algorithm

I want to design a filter (beamformer) with a given bandwidth (beamwidth) and sidelobe level using Parks-McClellan algorithm. I use "firpm" function of MATLAB. however, I cannot reach a solution for ...
2
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1answer
292 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 ...
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0answers
156 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
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1answer
112 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, ...
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1answer
124 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?
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0answers
200 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()). ...
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0answers
236 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
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1answer
257 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
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1answer
38 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
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1answer
302 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
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1answer
343 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(...
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2answers
354 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 ...
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1answer
124 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 ...
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2answers
203 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 ...
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1answer
67 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 ?
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0answers
140 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
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2answers
2k views

Derivative filter in Python

In a research paper (section 2.1, 3rd paragraph) about EEG, the authors note that the spectral amplitude profile of a signal is inversely proportional to frequency. To correct for this trend, they ...
0
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1answer
71 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 $\...
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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.
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0answers
187 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?
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1answer
236 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
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2answers
489 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 ...
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3answers
484 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
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0answers
107 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
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1answer
300 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 ...
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0answers
48 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 $...
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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
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1answer
707 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 ...
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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 ...
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1answer
176 views

1st order filter output

Two 1st order filters : ...
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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 ...
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0answers
43 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(...
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1answer
1k 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 ...
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1answer
65 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? ...
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0answers
121 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
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1answer
204 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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2answers
90 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 \...
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0answers
240 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 ...
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2answers
316 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
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2answers
849 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
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1answer
51 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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2answers
77 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?
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1answer
364 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 ...
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3answers
1k 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 ...