We’re rewarding the question askers & reputations are being recalculated! Read more.

Questions tagged [filter-design]

Fiter design is the process of designing a filter.

Filter by
Sorted by
Tagged with
3
votes
2answers
720 views

How do can we cancel out noises while listening voice on headphones?

It's a common problem that while talking with the another person on Smart phones with the help of regular headphones,we get very annoying sounds at background . The opposite person's voice gets mixed ...
0
votes
1answer
609 views

Help choosing coefficients for a simple FIR filter to extract DC component

I am a beginner in design of digital filters... I have just completed Coursera's Linear Circuits course! I have a signal which is sampled at 50 Hz where the DC component has a value of 800000 and ...
1
vote
1answer
177 views

What is sound output delay of a band pass filter at different freqencies?

I measured that a 2-pole band pass filter, given imput audio of different frequencies, outputs an audio signal with different time delay. At 440Hz, the time delay between of the input audio and the ...
0
votes
1answer
45 views

Concepts Surrounding First Two Samples in Difference Equation

I'm attempting to construct a difference equation in matlab for the function u(n) and find the auto-correlation matrix R. I'm following the technique as is defined by my text below: The system I'm ...
2
votes
2answers
2k views

Design Filter with Arbitrary Magnitude and Phase Response in Matlab

Does anyone know of a way to design a filter with arbitrary magnitude and phase response in Matlab?
1
vote
0answers
401 views

Phase Linearization

Does anyone know of some simple algorithms to linearize the phase response of a filter. I have a notch filter with a non-linear phase response. I basically need an all pass filter (so that the ...
0
votes
1answer
525 views

digital low pass Butterworth filter

A digital low pass Butterworth filter that has been designed using Bi-linear transformation has been a pole at $z=0.6$. It is also known that the filter's attenuate (at digital frequency) $\omega = 1....
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 ...
0
votes
2answers
171 views

Matlab: is a highpass filter what I want, to obtain my required spectrogram? How do I design/apply this? (pic included)

The axes are wrong in the spectrogram. I have an audio wav file, and I know the sampling rate. this is read into the audioData variable. The audio is 1 channel. selected window length I believe (can't ...
0
votes
1answer
2k views

FIR design using firpm (Parks-Mclellan)

Can you please help me with some ideas on how to implement an FIR filter in Matlab using Parks-Mclellan algorithm? My specifications are the passband cut-off frequency, the transition bandwidth. The ...
0
votes
1answer
2k views

FIR filter coefficients from frequency responses

I have frequency responses of FIR filter of order 13 from obtained 50 frequency samples. How can I get FIR filter coefficients of order 13? For example, in MATLAB a lot of possible ways I tried ; I ...
0
votes
0answers
100 views

Filtering an image with a helmholtz-type equation

I'm attempting to apply the following PDE as an image filter to smooth a discrete heightmap with a helmholtz-type equation as described in this paper. It seemed like an interesting alternative to a ...
0
votes
1answer
96 views

Why this two FIR filters have the same amplitude but different phase response

Why a filter of the form $$ 1+a_0 z^{-1} +a_1 z^{-2}+ a_2 z^{-3} $$ has the same amplitude response when in the reversed form $$ a_2+a_1 z^{-1} +a_0 z^{-2}+z^{-3} $$ but the phase response is ...
0
votes
2answers
428 views

my Butterworth lowpass formulas do not agree with Fisher webpage

I want to implement Butterworth low-pass filter. Thanks to this question, I have found out that the filter coefficients can be generated using Tony Fisher web-site or using his code. But the problem ...
-1
votes
1answer
226 views

Spectral leakage issue cannot solved using windowing

I am designing a frequency lowpass filter using FFT&IFFT. My test data is a time series with a large DC component (the pink curve). Before the FFT, the time series was detrended (curve fitting) to ...
1
vote
1answer
677 views

transfer function of wiener filter

I am trying to understand the function of a Wiener filter. I get that the Wiener filter minimizes the mean square error between the estimated random process and the desired process. Does this mean ...
0
votes
1answer
517 views

How to account for sample rate in a FIR Digital Filter?

As a learning exercise I am trying to simulate a simple notch filter with Matlab. For test purposes I use Matlab to simulate sinewaves of various frequencies and plot frequency vs. gain so I can see ...
1
vote
0answers
87 views

scaling issue when using scypy's firwin2 to get coefficients for a filter

I am trying to use SciPy's firwin2 command to get the coefficients that correspond to a filter of my creating. I took white noise in Audacity, applied an ...
0
votes
1answer
679 views

Tradeoff between Filter Order and Sections

I have designed a bandpass IIR butterworth filter with desired response using the MATLAB FDAtool with the configuration shown in the figure below Since this is multistage implementation, when I ...
0
votes
1answer
227 views

1D filter for speed input and noisy position sensor

I was tasked with designing a filter to smooth out a translating table. The system, we command a speed and a desired position, and measure it with two noisy position sensors. The commanded speed is ...
1
vote
0answers
138 views

Basic “filtering” in digital communications

I am completely stuck on solving some problems for my digital communications class. Junior level undergrad class, so fairly basic. I have been searching around all day for related articles etc but can'...
0
votes
2answers
688 views

Matlab FIlter coefficients to CMSIS api

I am trying to implement iir Band Pass filter design in STM32F429i discovery kit. I konw that it needs coefficients to be generated from matlab and feed them into cmsis api to achivi filtered data. ...
0
votes
0answers
142 views

MATLAB band-stop filter design for room reverberation?

I took acoustic measurements of a room using sine swept method, and I also used Pioneer sc-37 to obtain more accurate data. I used RoomEQ wizard software and measured the impulse response of the room,...
0
votes
1answer
391 views

How would I go about designing a difference equation for a given type of filter?

I have experience using design tools such as Xilinx's CoreGen and MatLab's DSP tool kit, but I have not really had much experience designing filters from the ground up. Reinventing the filter as it ...
0
votes
1answer
317 views

Custom nearly-linear-phase IIR filter computation gone wrong

I'm using the code in this book: Algorithms for the constrained design of digital filters with arbitrary phase and magnitude responses to compute nearly-linear-phase IIR EQs. ...
3
votes
1answer
847 views

Common Filter Types for Audio Applications

I'm beginning to write a basic "virtual analog" synthesizer and want to implement a lowpass with variable cutoff frequency and resonance. Does anyone have insight into filter types commonly used to ...
2
votes
1answer
154 views

Given Gain function, how to design a causal, stable and minimum phase IIR filter?

I am given $|H(\omega)|$, I wonder if minimum phase stable causal filter is unique and how to calculate it.
0
votes
1answer
401 views

How can slope of low-pass biquad be configured?

I have a biquad filter, configuread as a low-pass filter (using Zolzer's formulas, similar to these). It does not overshoot and decays nicely in what I assume is -40dB/dec. How can I device a ...
0
votes
1answer
646 views

Recursive Implementation of the Gaussian Filter

I'm trying to implement an IIR form to approximate the Gaussian Blur Filter. I'm working with the article "Recursive Implementation of the Gaussian Filter" by Ian T. Young and Lucas J. van Vliet. ...
0
votes
1answer
39 views

How to equalize frequency contents using filter?

Suppose there is a bounded continuous signal $f(t)$ that can be fourier-transformed into $F(\omega)$ of frequency contents. $|F(\omega)|$ is non-zero for only finite number of frequencies: $f_1,f_2,.....
0
votes
2answers
457 views

How to take acoustic measurements of a clasroom

I'm an electrical engineering major and I have a question regarding a project I'm working on. I'm supposed to design 3 band-stop filters on MATLAB for echo cancellation/suppression. My problem is ...
0
votes
1answer
160 views

How to manipulate attenuation in Parks MClellan algorithm?

I am using GNU Radio C++ source code in my app (pm_remez()) for generating FIR filters coefficients. Input parameters here are: ...
14
votes
2answers
2k views

Is a high-passed signal the same as a signal minus a low-passed signal?

My question is, if I want to high-pass a signal, is it the same as low-passing a signal and subtracting it from the signal? Is it theoretically the same? Is it practically the same? I have searched (...
0
votes
1answer
96 views

2D IIR filter design (x-axis width proportional to y-axis)

I want to design a 2D IIR filter (that is, created line-by-line progressively, in this case from left-to-right). What is the best way to go about this? I would like the response to look something ...
2
votes
1answer
637 views

Multi-Stage digital FIR filter vs Single stage FIR filter. Which is better?

Can anyone provide an insight to select a filter structure out of multi-stage and single stage FIR filter?(it will be great help if references are also provided) Let's say I have to make 50 tap filter....
3
votes
1answer
1k views

How to test the FIR filter in C++?

I am almost done with design and implementation of my first FIR filter using C++. Its a Sinc-Windowed FIR filter designed for a cut off frequency of 100 Hz with sampling rate of 500, with a filter ...
0
votes
1answer
3k views

Why should the cut off frequency and the transition bandwidth (BW) be expressed as a fraction of sampling rate?

I came across this confusion in Chapter 16,"Book - The Scientist and Engineers guide to DSP" and Topic is "Windowed Sinc-Filters". I am doing some calculations for designing the sinc filter and came ...
0
votes
1answer
154 views

Microphone Array null steering unexpected results after IFFT

I am writing an open source microphone array simulation program:SharpEar. When I give uniform weight to microphone array elements, the result in the angle is a sinc function as expected. There are 41 ...
4
votes
1answer
2k views

What is the relation of the transition band's width and the filter order for the FIR windowing method

When designing an FIR filter using the windowing method, how can one estimate the filter order ? It's obvious that the type of window and the transitions band's width has some effect on the order.
0
votes
1answer
982 views

Obtain transfer function from signal flow graph

I have a pretty complex signal flow graph with all the blocks and the branches completely describing an analog filter, in the Laplace complex variable s. The signal $X(s)$ is the input and $Y(s)$ is ...
1
vote
1answer
144 views

Question regarding filter implementation - Audio EQ Cookbook

Some of you may know the infamous Audio EQ Cookbook. You can take a look here. We tried to implement the functions there to achieve a implementation for a Lowpassfilter. However, when we try to ...
2
votes
2answers
7k views

How to produce a high-pass filter from a low-pass one?

I already have a low-pass filter and want to remove the DC offset of a signal with a high-pass one. I guess you can subtract a low-pass filtered signal from the original signal and the result is a ...
2
votes
3answers
2k views

How to select the center frequency and filter type for 5-band digital audio equalizer?

I have a small team project: use Matlab implement a Gui 5-band digital equalizer using FIR and IIR filters (2 cases) - The sample rate of all audio files is standard audio 44.1kHz. Coding is not my ...
0
votes
1answer
596 views

Apply FIR filters to Extract Stego from WAV data

I appreciate anyone that takes a moment to help me with this problem. I've been banging my head against the keyboard for a while, searching forums and DSP tutorials and I can't figure this problem out....
0
votes
2answers
192 views

In filter design, why isn't it possible to have a frequency response with phase 0?

A linear phase filter will cause a delay in the output signal of the filter. If I'm not mistaken, a filter with phase equal 0 would cause no delays in the output signal. Can someone explain me why ...
0
votes
0answers
254 views

Why Matlab fdatool designed butterworth filter and butter command's co-efficients are different

I met an great confusion. Using below command I have got co-efficents for a 300Hx cut-off high pass filter of order two. [z p k] = butter(2,(300/44100),'high') [sos,g] = zp2sos(z,p,k) Now this ...
1
vote
2answers
6k views

Applying Lowpass (LPF) and Highpass (HPF) Filters to an Image in Frequency Domain in MATLAB

What I want is simply to apply low pass and highpass filters to an image in the frequency domain. Does MATLAB image processing toolbox have any commands for this?
0
votes
2answers
128 views

FIR Filter length Upper Bound

Say I want to design a low pass filter with stop and pass band ripple constrained to $\epsilon$ and transition region $\delta$, what is the smallest FIR Filter length that suffices? Can you please ...
0
votes
1answer
96 views

Filter to speed up convergence of geometric series

I want to speed up convergence of geometric series $\ \ lim + c^n \ \ $ where $|c| < 1$ and $lim$ is not known. A basic low-pass filter, here Butterworth, is good for $-1 < c < 0$ but not $0 ...
1
vote
2answers
2k views

Calculating Savitsky-Golay filter coefficients - normalization

Using the matrix equations found on Wikipedia I wrote some Python code to compute the Savitsky-Golay coefficients for $m$ data-points, using a polynomial of degree $k$. Here it is: ...