A low-pass filter is an electronic filter that passes low-frequency signals but attenuates signals with frequencies higher than the cutoff frequency.

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Transform a continuous time differential equation of a Low Pass Filter to discrete time

Assuming I have a linear differential equation of first order low pass filter: $$RC*y(t)'+y(t) = x(t) $$ And Assuming i know now that the input x(t) is being sampled at a rate of $T_s$ how can now ...
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40 views

Lag in financial trading vs Group Delay in signal processing

I'm a newbie to Signal Processing - and so I'm providing a "background" and a "question" section separately. Background: Suppose I have a linear filter (w.r.t. given as $\theta_{t+1} = \theta_t ...
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19 views

Denoising approach for a combination of several ADC voltage channles

I have 2 ADC channels of constant voltage measurement with small amount of high frequency noise only and no low frequency oscillations. The final signal should be the simple sum of those and denoised ...
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50 views

Possible to implement simple IIR filter with greater than 6 db/octave slope?

I am using the following IIR code as a low-pass filter in a software project, and it works great. However, I would like the slope of the filter to be much steeper as the one here seems to be very ...
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1answer
29 views

Implementation of a cascaded IIR filter

I am attempting to implement a cascaded IIR filter in MATLAB. I used fdatool to design the filter and exported it as an object, Hd. In my script, I import a log file of data I captured and store my ...
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1answer
66 views

Why are ideal filters not particularly useful/ hard to do?

Why is a LPF ideal filter: a) Not useful? b) Hard to make? I was asked to make a Low pass filter (simulink block) that will take an input of a cutoff frequency, and for a signal with a frequency ...
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1answer
31 views

Transfer function, amplitude response and difference equation for a filter

I've found a paper with a filter described in terms of transfer function, amplitude response and difference equation: transfer function of the second-order low-pass filter: $$ H(z) = ...
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2answers
57 views

Find Stopband Edge Frequency

I tried to find the stopband edge frequency.I know that the signal amplitude falls to -3 dB below the passband edge frequency.I examine a lot of DSP books so i don't any information about how to ...
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45 views

FFT “Brick Wall” Low Pass Filter Not Smoothing Output

I'll try to be as brief as possible. I have very limited signal processing experience. I have a data set that consists of essentially a big old sine function (with deviations from ideality), and lots ...
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1answer
57 views

How does sampling rate affect discrete filters?

I am just starting to learn discrete filters and I could use some help. I understand continuous signals and filters. I am trying to understand the math behind discrete filtering. For example in the ...
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1answer
54 views

Minimum Energy Delay Property

I have 4 systems which are represented in z domain like this: Where I know the poles and zeros. Then I compute Partial Energy Sequence of impulse response of all these system using this formula: ...
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1answer
56 views

What is the low-pass filter that gives minimal transient response?

As title says, what is the low-pass filter that gives minimal transient response? By minimal transient response, I mean minimal settling time with reference to very tiny amount of divergence of ...
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27 views

FIR Filter matlab command

i'm trying to understand the fir1 filter but i still don't get it. For example here i got an audio signal that i consider noise, i'm passing it through a low pass filter. ...
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17 views

Calculating the network function of a filter

Given a system, that behaves as a 1st order filter with network function $H(s)$. We input: $$v_1 (t)=1+3\cos(10^4 t)$$ And we obtain as output: $$v_2(t)=1+1.5\cos \left(10^4 t ...
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3answers
120 views

Which anti aliasing filter algorithm for efficient audio resampling?

I'm building a real-time audio resampler (think pitch bend) that needs to have a several different performance vs quality configuration options. I understand that I'll need to apply a low-pass filter ...
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1answer
40 views

Convert low pass continuous time filter design to bandpass, discrete time

I am trying to convert the continuous time transfer function of a second order lowpass Butterworth filter is given by: To a bandpass fourth order bandpass digital filter, I first apply the mapping ...
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31 views

transient response of simple digitalized RC low-pass filter

In http://en.wikipedia.org/wiki/Bilinear_transform#Example, digital version of simple RC low-pass filter is presented: $$\frac{1 + z^{-1}}{(1 + 2RC / T) + (1 - 2RC / T) z^{-1}}$$ where $T$ is ...
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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 ...
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1answer
72 views

low pass filter with minimal ringing

I am trying to filter out some square wave signal to within a limited band (1/4 or 1/8 of the original), I realized that there's a lot of ringing in the wave when I use my filter (elliptical), I also ...
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208 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 ...
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0answers
47 views

extrapolate turning points for a sine weighted moving average?

On a stationary signal a sine-weighted moving average is calculated (SWMA: the coefficient vector looks like the first (>0) part of a sinusoid). The SWMA looks like this: very smooth: The future is ...
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1answer
58 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 ...
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72 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 ...
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63 views

Understanding my Butterworth filter

My input is x and my filtered output is y. The sampling rate of x is ...
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4answers
170 views

How to design very narrow band, very sharp low pass filters - Only DC needed?

I am using a DSP processor to sample signals at 108kHz and I only want to get the DC part out of it. Such a high sampling rate is used because I want to use oversampling to reduce quantization noises. ...
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1answer
121 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 ...
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45 views

Given Gain function, How to design a causual, 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.
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39 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 ...
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59 views

Playing back an audio file at an arbitrary rate with upsampling and downsampling

Background: My overall goal is to create an audio file player which can playback an audio file at an arbitrary rate. The rate can change continuously as the file plays. This means upsampling to ...
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1answer
29 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: ...
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1answer
209 views

How is the lowpass to bandpass transformation derived?

I'm using a lowpass to bandpass filter transformation: $$ s \leftarrow Q \left( \frac{s}{\omega_0} + \frac{\omega_0}{s} \right) $$ Based on the wiki article: ...
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1answer
265 views

Don't understand coefficients in MATLAB “butter” low pass result

I don't unstand the nature of the output in Matlab's "butter" command. Calling [b,a] = butter(4, 0.5, 'low'); gives me two vectors, which the documentation says ...
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469 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 ...
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1answer
290 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 ...
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1answer
177 views

IIR low pass filter: floating point and fixed point

I have implemented an IIR low pass filter in direct form 2, the input to the filter is a unit impulse $\delta[n]$. I have calculated the output of the filter for the mentioned input using the ...
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1answer
91 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 ...
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1answer
74 views

How to concatenate IIR filter (Butterworth) with a Matlab function and evaluate the integral

I have a function of $j\Omega$ and $v$ (velocity), i.e.: $$A(j\Omega,v)=\frac{(j\Omega)^2}{p + \frac{j\Omega}{v}}$$ where $p$ is the pole. I want know how $A(j\Omega,v)$ modify its behavior, when I ...
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65 views

Can someone please explain or describe me the behaviour of equation 2 in my question, relevant to filter response (Sinc Interpolation)

At present, I am learning the theory of operation of resampling for bandlimited periodic discrete signals using sinc interpolation.I am developing a design flow and having difficulty in understanding ...
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1answer
51 views

Periodic Signal Fourier Transform

If I have a signal x(t) as follow x(t)=$\sum\limits_{k=-∞}^∞ R_{T}(t-2kT)$ $R_{T}(t) = \begin{cases}1 ;&[\dfrac{-T}{2},\dfrac{T}{2}] \\ 0; & \text{otherwise} \end{cases}$ ...
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Output of a band pass filter

Signal x(t) = | cos(2πft)|, f != 0 is the input of an ideal band-pass filter with bandwidth B = 3f/2. How can i Find the ...
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1answer
77 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 ...
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2answers
106 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 ...
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2answers
178 views

Selecting the Low Pass FIR Filter, Decimation, DSP

I'm studying DSP, but I have weakness about it. I was reading this: http://www.eetimes.com/document.asp?doc_id=1275556 But, I don't know how select the Low Pass Filter. I am assuming that this ...
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Using bilinear method while filtering a signal - Matlab

i have to current signal in matlab: Fs = 8000; T = 1/Fs; N=512; t = (0:N-1)*T; ...
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90 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: ...
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2answers
79 views

up down sampling and low-pass filters

I am trying to figure out what seems very trivial thing about down sampling and up sampling. The problem that I am facing is how to find the effects of up sample on a signal followed by a filter ...
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24 views

Sampling: how to determine a “good” sampling rate based on a given filter

I assume that the sampling rate is w_s = 2w_max if we have an ideal low-pass filter that nulls frequencies whose absolute value is higher than that of w_max. Therefore, that the only reason for ...
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1answer
52 views

filter design for ECG

I would like to ask how to design a digital filter to remove $50$ Hz interference on an ECG signal. Do I need to use notch filter? Thanks.
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64 views

Why do we use Hilbert Transform in discrete time domain?

I have few questions regarding hilbert transform, Questions are as follows: 1)What are the benifits of using hilbert transform hardware and software point of view? 2)Cannot we use any other digital ...
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117 views

Digital low-pass filters; best practice application

I have time domain strain gauge data from a structure which is subjected to low frequency, random loading. The signal / noise ratio is pretty good, but I would like to try to improve on it as much as ...