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Questions tagged [lowpass-filter]

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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1answer
682 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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1answer
4k 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
98 views

Periodic Signal Fourier Transform

If I have a signal x(t) as follow x(t)=$\sum\limits_{k=-∞}^∞ R_{T}(t-2kT)$ $R_{T}(t) = \begin{cases}1 ;&[\dfrac{-T}{2},\dfrac{T}{2}] \\ 0; & \text{otherwise} \end{cases}$ $\mathfrak{F}\{R_{...
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2answers
524 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 and/...
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0answers
163 views

Is it possible from 1 order cascading low pass filters to make a butterworth filter

Today I have heard, that is "impossible" to make low pass butterworth filter from cascading 1 order low pass filters(RC). I am trying to discover is that true, but I can find little to none ...
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1k views

Why does the inverse fourier transform of a lowpass filter have complex components in matlab?

I am quite confused whether the following numerical differences i find are just severe round off errors made by matlab, or something i am doing wrong. The following happened when trying to see what a ...
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0answers
66 views

Low pass butterworth filter is distorting [duplicate]

my low pass filter keeps distorting and I'm at a loss to explain why. It calculates the correct coefficients, but distorts everything. The code is as follows. It's driving me mad. What's weird is my ...
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0answers
387 views

Sine-weighted moving average on non-periodic, stationary signal

I am looking for -or trying to create- a filter with a piece-wise monotonic second derivative in such a way that when placed on a non-periodic input signal, changes in sign of the second derivative ...
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2answers
233 views

How to pad a windowed sinc filter in the frequency domain

I'm creating windowed sinc filters to apply them to certain signals that I'm dealing with. To design the filters, I'm using the approach described in the book "The Scientist and Engineer's Guide to ...
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2answers
2k views

Audio signal processing

I am trying to code audio signal processing software in java. Purpose: It will use audio signal processing library to filter out noise from the raw PCM based input (.wav) of heartbeat recording. ...
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1answer
88 views

A one-pole LPF (6 dB/oct) has a step response to $1/e$ amplitude of ${\tt time} = 1/(2 \pi f)$. What would the response time be of a 3 dB/oct filter?

I was instructed in another question here that if a one pole (6 dB/oct) low pass filter is fed a discontinuity (ie. going from 1 to 0) it will take ${\tt time} = \frac{1}{2\pi f}$ to reach $1/e$ ...
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1answer
415 views

AM demodulation from a real RF signal

I'm capturing an RF signal transmitted at 433.8MHz, with a bandwidth of about 50KHz. My goal is to be able to save it as a .wav file. I'm using an RTL-SDR dongle and a software named HDSDR, where I ...
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1answer
632 views

Forward Backward filter Scipy - $\tt signal.filtfilt$ changes the amplitude when $\tt signal.butter$ is used with $\tt btype='high'$

In the plot below the blue wave is the original signal. The orange wave is after the low pass filter. The green wave is after the high pass filter. I have a noisy ECG signal stored in the ...
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1answer
548 views

ifft doesn't recreate original signal

I want to convert given signal into frequency domain and do some calculation and then convert that signal into time domain. Here is the code which I am trying out ...
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1answer
337 views

Filtering everything outside 20 - 20000 Hz

With an audio .wav file (sampled at 44.1 or 96 Khz), I would like to filter out everything outside the 20 - 20000 Hz range. I tried a Butterworth bandpass filter and there were unstability issues, ...
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1answer
526 views

How to design and implement a low-pass AR1 filter in Matlab?

I have a data series y(n) that I want to filter using a low-pass AR1 filter. I found that this filter has a transfer function ...
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3answers
715 views

Continuous low-pass filter (LPF) on a periodic signal

I have a continuous-time signal $x(t)$ that has the following properties: It is periodic, with a periodicity of 1 (sec). It has no discontinuities or infinite values at any time. I want to come up ...
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1answer
258 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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1answer
262 views

Low Pass Filter to Maintain Edge Information

I am looking for a kernel acting as a low pass filter that satisfies these conditions: $$K(-\mathbf{u})=K(\mathbf{u}) \tag{1}$$ $$K(\mathbf{u}) \ge K(\mathbf{v}), \;\; \text{if} \;\; |\mathbf{u}|<|...
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0answers
22 views

Low-pass filter on PDM data provides data that does not have a FFT

I'm trying to convert PDM data into analog form by passing the data through a low-pass filter in Matlab. My approach is fairly simple and matches the answer given in this Electrical Engineering Stack ...
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2answers
109 views

Can I use a transfer function to filter noise?

I want to make it simple! I want to filter white noise with a transfer function and it's going to be zero phase. Assume that we have our low pass filter. $$G(s) = \frac {1}{1 + Ts} $$ Where $T$ is ...
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1answer
94 views

How to keep low frequencies in DCT

How can I keep a percentage r of the low frequencies of a Nx1 signal x using the Discrete Cosine Transform of it? Is it simply zeroing all but the first rN elements of DCT(x)?
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1answer
203 views

Cepstrum analysis - Cut off frequencies displayed

My inquiry is regarding the so called cepstrum analysis. By conducting some experiments, i have real time data at my disposal. The acquisition rate is 1600Hz. I wanted to try cepstrum analysis just ...
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2answers
232 views

How do you initialize the output of a second-order-section filter?

I've got a 4 stage second-order-section filter. When I turn this filter on, it assumes it is starting from zero and takes forever to converge on the actual signal. For example, let's say I have a ...
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1answer
224 views

Function in the Fourier domain of a low-pass filter

I have to represent a low-pass filter in the Fourier domain having a cut-off frequency of $f_c={3}\ kHz$. Could all low-pass filters could be represented with this function: $H(f)=\frac{1}{1+j\left(\...

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