Questions tagged [filter-design]

Fiter design is the process of designing a filter.

153 questions with no upvoted or accepted answers
Filter by
Sorted by
Tagged with
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 > ...
4
votes
0answers
302 views

Design Nonlinear (Arbitrary) Phase FIR Filter using Iterative Reweighted Least Squares (IRLS)?

How would you use the MATLAB function below to create a FIR filter with a custom magnitude and phase response? It is from the Statistical Modeling and Inference MATLAB Toolbox ...
3
votes
0answers
173 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 ...
3
votes
0answers
253 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()). ...
3
votes
0answers
301 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 ...
3
votes
0answers
2k views

Difference between state space and transfer function model response (in Simulink)

Why I get a different response from the same system (e.g. three phase inverter with LC filter) in state space form and in transfer function (Laplace) form when using the same PI controller values ($...
2
votes
0answers
21 views

Is it possible to create a causal filter with a positively-sloped phase response?

I am currently prototyping a set of noise-cancelling headphones that utilize active EQ to modify the noise-cancelling signal, allowing the user to modify the frequencies that they hear or don't hear. ...
2
votes
0answers
102 views

Practical implementation of IIR/biquad filters following non-textbook transfer function of real-wold analog filter

I am trying to program a DSP filter to study prototypes of loudspeaker filters, which will be implemented as passive analog circuits in the final speaker system. This process involves simulation of ...
2
votes
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 "...
2
votes
1answer
162 views

Estimate smartphone accelerometer bias

I was planning to develop Android / iOS applications that enable users to measure 3D length using their smartphones. According to this question, you need to know at least the time-varying bias that ...
2
votes
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 &\...
2
votes
0answers
273 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. ...
2
votes
0answers
111 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 ...
2
votes
0answers
70 views

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 ...
2
votes
0answers
38 views

how to design wavelets from splines

Can any one tell me how to design wavelets from splines using matlab. whether we can make wavelets from higher order splines or only with B splines
2
votes
0answers
106 views

Can you apply an audio filter/effect to only part of an audio signal and not get audio glitches?

I'm entertaining the idea of making a music-creating application. I'm a total newb at signal processing. The question that concerns me most right now is, if I apply a filter/effect such as convolution,...
2
votes
0answers
105 views

How many different filters are needed for in an iterated filter bank?

I'm trying to implement an iterated filter bank as described in the Subband coding chapter of the Really Friendly Guide to Wavelets. However, the text seems to suggest that The advantage of this ...
2
votes
0answers
487 views

Doppler shift using PLL

I am trying to apply phase lock loop to calculate phase shift due to doppler effect and finally calculate the doppler frequency($\Delta f$). I am assuming that the received wave is of the form, $r(t) =...
2
votes
0answers
89 views

How to get translation and roation data using 6DOF MPU6150

I am trying to get the position data from an IMU 6 Degrees of Freedom (MPU-6150) board using an Arduino Uno. I am trying Jeff Rowberg’s example MPU6050_DMP6.ino that may be found here. This is the ...
2
votes
1answer
374 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 ...
1
vote
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 ...
1
vote
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 ...
1
vote
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 ...
1
vote
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/...
1
vote
0answers
33 views

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 ...
1
vote
0answers
61 views

CIC compensation filter and total response

UPDATE: New plot with properly scaled CIC comp response. I'm studying about CIC filters for interpolation/decimation and I'm reading about CIC compensation filters due to CIC filters not having a ...
1
vote
0answers
30 views

Concept of scales in Gabor filter

I've been reading about Gabor filters, and there is a concept of scale. While I understand that Gabor features help in analyzing both the frequency and orientations of an image, what is the idea of ...
1
vote
1answer
283 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 ...
1
vote
0answers
31 views

Frequency warping while creating filter coefficients

If I have an audio signal at 16kHz with frequencies band-limited to 8kHz. I want to low-pass filter this signal such that fc=4000Hz. Meaning when I reconstruct the signal I do not want any content ...
1
vote
0answers
52 views

Using a Passive 1st Order RC filter to Extract DC Component

I am trying to extract the DC component out of a PWM signal produced by the MCU using a LPF. I understand that 1st order RC filter is looked upon as being amateurish because the gain in the passband ...
1
vote
0answers
102 views

Generation of Noise-shape filter from Power Spectrum Density

This is my very first time in dealing with signal processing, so I am sorry if I will not use a rigorous terminology. I am dealing with some issues about noise modeling in matlab. I'm trying to ...
1
vote
2answers
264 views

How to compare many bandpass filters (for their signal propogation delay)?

Let's say we have an audio signal sampled at 96 kHz, and we want to compare several bandpass filters to find the one with the lowest rectangular-envelope-propagation-delay (see graph below). The band ...
1
vote
0answers
47 views

How far can numerical differentiation be accurate for sinusoids?

Suppose you have samples of uniform interval of a sinusoidal signal that is some sum of sinusoids. Forget about aliasing, as the signal is bandlimited. If we have infinite number of samples at ...
1
vote
0answers
145 views

Issue with the Massberg Analog-Matched Low Pass Filter

Implemented the filter mentioned in title using Octave and couple sources Vicanek and Zeitlin (LPF.cpp). Got filter working otherwise but, ... when tested the responses against 'analog model' of the ...
1
vote
0answers
161 views

Binomial Approximation of Gaussian Distribution

It is said that we may use the binomial coefficients ( a layer from Pascal's triangle) to approximate the 1-D Gaussian kernel with certain $\sigma$, where $\frac{n}{4} = \sigma$ and $n$ is the index ...
1
vote
2answers
2k views

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 ...
1
vote
0answers
47 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 ...
1
vote
0answers
164 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 ...
1
vote
0answers
209 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?
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 $...
1
vote
0answers
139 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 ...
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. ...
1
vote
0answers
368 views

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-...
1
vote
0answers
103 views

Can picking an inappropriate roll-off frequency for a digital filter just alias the roll-off value?

I believe a related question is asked here, but my question deals more with what filter is realized if one fails to consider design parameters relative to sampling rate. In designing an IIR filter we ...
1
vote
0answers
202 views

Zero STFT bins (and not FFT bins)

Filtering by taking FFT, zeroing bins, and inverse FFT is a bad idea, as discussed here. But what about: take a STFT, (i.e. multiply the input signal by moving window function, and take FFT) zero ...
1
vote
0answers
908 views

FIR filter design sampling rate

Let's assume we have an $x(n)$ time sequence, whose $f_s$ sample rate is 20 kHz. We are required to design a linear-phase lowpass FIR filter that will attenuate the undesired high-frequency noise ...
1
vote
0answers
160 views

Filtering reject two bands (multiband bandstop FIR filter)

I know the existence of Remez Parks-McClellan, but I want to exploring results with Math.... I have five frequencies on input (check blue plot): ...
1
vote
0answers
81 views

Anti tremor motion detection

I'm programming in c# using opencv (emguCV). I use motion detection approach to detect some changes in a paper wall. For example it's may be colored by a laser pointer. My problem is that when I use ...
1
vote
0answers
3k views

Designing bandpass FIR filter MATLAB

I am trying to design a FIR bandpass filter for a STM32F407 microcontroller with a passband of 1kHz between 59500 Hz and 60500 Hz or something similar (since I don't really know how tight I can ...