Questions tagged [filters]

In signal processing, a filter is a device or process that transforms a signal by selectively choosing specific frequencies and leaving others with appropriate gain or as it is.

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Can you do temporal frequency filtering for a spatially moving signal in image sequence?

Context : A static camera captures video footage of a still indoor environment at given framerate (1080p@30fps or 720p@60fps) A low power line laser projects a line shaped beam in the camera field of ...
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50 views

Laplace Transform: zeros and corresponding impulse response $h(t)$

Poles and the impulse response If our impulse response is in the form : $$h(t) = e^{-\sigma_0 t}\cos(\omega_0 t) \, u(t)$$ (where $u(t)$ is the unit step function) And its Laplace transform is : $$H(s)...
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Zero-ISI Filter Bandwidth, Raw Data Bandwidth and Symbol Rate Relation

Please can you assist I am confused about the relation between Sinc and Rectangle transform pair and how that relates the Bandwidth of Pulses, Bandwidth of Zero-ISI Filter and the Symbol Rate. My ...
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Matlab FIR filter numerator coefficients to python

I created a equiripple FIR filter in Matlab using filterDesigner and exported it to an ASCII file. The file contains the numerator values as a list of 51 numbers ...
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1answer
39 views

Recreating matlab second order filter in python

I want to create some custom filters in Matlab filterDesigner tool and then apply those filters to a large set of audio files. I want to use python to do the ...
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High-pass filter with quantized (fixed-point) coefficients

I have a high pass IIR filter with numerator b = [0.997688673703782,-1.995377347407564,0.997688673703782]; and denominator ...
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39 views

How to remove noise from frequency spectrum?

I have a signal acquired at 40 MHz sampling rate, from an ultrasound transducer with center frequency of 5 MHz. The frequency spectrum is shown in attached images (whole spectrum and zoomed). There is ...
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2answers
158 views

How can initial conditions be taken into account to calculate a system's terminal value using the final value theorem or some other technique

I wish to calculate the Final Value of the following system given initial conditions not at rest, and assuming the X will not change from its initial condition value. The transfer function is $$H(z) = ...
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66 views

The Wiener Filter for the Time Derivative of the Signal

Assume the observation of the original signal $s(t)$ is $x(t) = s(t) +n(t)$, where the signal and noise are independent. Then we need to design the Wiener filter $g(t)$ to estimate $\frac{d s(t)}{dt}$ ...
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48 views

Why is there a dip in this Chebyshev filter response?

I want to build and simulate a 6 GHz, 4th order lumped-element Chebyshev bandpass filter with a maximum passband ripple of 0.1 dB and a bandwidth of 1 GHz, and as I understand it, there should be 3 ...
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Low-pass filters not producing smooth output

I'm relatively new to this material, so apologies in advance if I'm missing something obvious. I'm trying to smooth some very noisy signals. What I observe is that if I apply an ewma, even with a ...
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Box function signal filtering in python

I am trying to produce a box function filter of a signal in python. I expected to find this functionality in scipy.signal, but I can't find any solutions. What I am trying to do is this. I have 2 ...
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Most generally-accepted audio term, for “pre” and “post” lengths of a symmetric filter's impulse response?

Consider a FIR audio filter with symmetric impulse response. In my case, the filter kernel width is 15 frames on each side (so including the center frame, 31 frames total). When examining the audio ...
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33 views

What does poles in unit circles center mean?

Suppose i have all my poles in unit circle center. What kind of information this gives me? Can i determine if this filter is IIR or FIR?
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Scipy Butterworth highpass filter results in distortion and bloated output

Trying to use Scipy Butterworth filter to filter frequencies below 54 Hz from audio files. Audio files are stored as WAVs. Sample rate is 44100. When I try using this filter the result is distorted (...
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Need to filter very small signal

I am measuring ECG signals on subjects and I am having issues filtering a very small signal from the inherent noise of the collecting device. I am using an 18 bit ADC to digitize the signal Here is an ...
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How to ampfify/reduce certain frequencies in time data

currently I am trying to filter time data so that certain containing frequencies are getting amplified/reduced. I am dealing with a system which has a certain characteristic that I would like to get ...
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1answer
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Third octave band zero-phase forwards-backwards Butterworth filter adds amplitude to complex signals?

Overall aim: to take a short signal (likely to be noisy with some tonal components) and filter it into third-octave bands (TOBs), apply frequency-dependent spectral adjustments, then recombine the ...
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Using MNE-Python for EEG trimming and Filtering

I am working on an EEG Signal analysis problem with python. I need to remove the recordings below 1st minute and above 6th minute of the signal, and pass it through a bandpass filter. I am not ...
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1answer
25 views

Finding the equivalent filter H(u,v) in the frequency domain of a 3x3 spatial mask

I'm trying to find the equivalent frequency domain filter, $H(u,v)$, of a 3x3 spatial mask that averages all neighbours of a point $(x,y)$ in said 3x3 neighbourhood excluding the point itself. So far, ...
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1answer
63 views

State-space filters in fixed-point?

I am having difficulties implementing state-space filters in (32-bit) fixed-point. This is because the coefficients have a huge dynamic range, for example (MATLAB code): ...
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1answer
76 views

Approximation and IIR Filters

Consider the discrete-time time-invariant system with input $x[n]$ and output $y[n]$ satisfying $$y[n] = \sum_{k=1}^5{x[n-k]}$$ Consider approximating the desired system with a second-order IIR system ...
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33 views

Efficient parallelized polyphase decimation FIR filter

I am struggling to derive a parallelized polyphase decimation FIR filter structure. I wonder if anyone can point me to some references? Normally a polyphase decimation FIR filter's decimation factor (...
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55 views

Matlab filter implementation

I have a filter response in the $z$-domain. $$H_0(z) = 2 + 6z^{−1} + z^{−2} + 5z^{−3} + z^{−5}$$ How do I implement this filter on MATLAB? I can do this if I have the coefficients but I do not know ...
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Finding the Correlation Matrix of Wiener Filter using Matlab

I have the problem below and need to solve it using Matlab. I not a big expert in Matlab and just don't know how to implement it. Basically, it says that \begin{equation} R\:=\:\left[X\left(n\right)X^...
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44 views

Cascading first order digital filters in C++

In a related question a probable solution was given to build a first-order digital filter and then cascade three of them in order to turn white noise into pink. I have applied the C++ as follows but ...
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Matlab Filter designer not working, how come?

Before reading any further, to give you on some insight on the output level of the oscilloscope picture's the ~340mV green signal is normal. If you know me I posted a question as to why its 340mV but ...
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54 views

Given a -3dB/ octave filter that makes Pink Noise, how can I make a +3dB/ octave filter that makes Blue/ Azure Noise?

I learnt from here that the following filter can turn white noise into pink (-3dB/ octave). ...
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39 views

Modulate Bipolar(?) signal

I want to modulate this signal if possible with an Arduino, but have no idea what circuit can help me here. Let's ignore the timing and the measurements from the image for a moment, my problem is more ...
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1answer
50 views

Is Unsharp Mask (USM) Equivalent to Applying Laplacian of Gaussian Filter Directly on the Image?

There are some educational materials, like Alex Pan - CS 194-26: Image Manipulation and Computational Photography - Fun With Frequencies and Gradients, which demonstrate the unsharp masking technique ...
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52 views

Help implementing digital filter from tansfer function

I'm trying to implement a digital filter, which is given by the following transfer function: $$ 1+2V K \frac{K+c_m+2Kz^{-1}+(K-c_m)z^{-2}}{1+2Kc_m+K^2+(2K^2-2)z^{-1}+(1-2Kc_m+K^2)z^{-2}} $$ $$ +V^2K^...
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44 views

Applying 2D Image Convolution in Frequency Domain with Replicate Border Conditions in MATLAB

I have created a function that filters an image (250x250) with a gaussian blur kernel (5x5) using FFT and IFFT. I am trying to get my filtered image to equal exactly the filtered image created by the '...
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Extrapolation of sinusoidal signal

Let there be a discrete signal that is the sum of sinusoidals and can be described by $$s(n)=\sum_i A_i \mathrm{sin}(2\pi f_i \frac{n}{f_s}+\phi_i)$$ where $A_i, f_i, \phi_i$ are unknown but fixed ...
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45 views

Designing filters vs. editing fft components

What is the difference between designing high-complex filters to reduce/remove frequencies in a signal and calculating the fft, reducing/removing those frequencies manually, and reconstructing the ...
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44 views

Design FIR with linear phase response

What is the best way to design a FIR filter with a given linear phase response, such that each frequency inside the passband is phase shifted according to: $$H(f)=e^{i2\pi f k}$$ where $k$ is factor ...
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Output signal is being attenuated after the Biquad filter direct Form 1, why?

Thought this would deserve a separate post on its own. What's going on is that I finally got my low pass digital filter working on the STM32H753ZI using the PMODI2S2 peripheral and I noticed the ...
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Alternatives to the median filter

I have a embedded system where I get some type of sensor readings with varying frequency between 0.3 to 1 Hz. Sometimes these readings may have impulse like noise. To filter out this noise, I've ...
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2answers
35 views

Bandpass with large group delay

I read that a simple delay filter that delays all frequencies requires at least $\frac {N-1} {2f_s}$ taps. In my case this would result in way too many taps (20000). Can I achieve this with ...
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DTMf decoding using Matlab

I have to identify the individual keys presses from a DTMF signal. There are three keys presses together as seen in the image. The signal has a sampling frequency of 8kHz. Each tone lasts between 0.1 ...
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2answers
164 views

DSP low pass filter (IIR) no longer works when changed to a new MCU

Having troubles understanding why a DSP low pass filter was that working on the M4 is no longer working on an M7. I recently switched over to a STM32H753ZI from a STM32L432KC. In addition to switching ...
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1answer
45 views

How to filter a signal using a bandpass filter consisted of two moving average filters?

I want to filter a PPG signal on a microcontroller. I have limited memory and a was searching for low computational methods. I found the work of Kazuhiro Taniguchi, Earable POCER: Development of a ...
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72 views

Multiple peaks in a same signal?

I am working on peak detection in different signals, the signal plot looks like this: After applying peak detection algorithm and tuning it for each signal, final output looks like this: As you can ...
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Why is Fourier space not adequate for (theoretical or digital) filters?

As far as I have seen, almost all theoretical filter design occurs in Laplace or Z-space. Also, there is a pervasive connection to real life analog filters in the design. If one is just thinking in a ...
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27 views

What is the purpose of the recurrence relation in low and high pass (audio) filters?

Newbie here. I'm familiar with time and frequency domains from math. I think I understand what low and high pass filters do, sonically. But digging into an implementation of ...
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Which one is better method and why? Manually Handcrafted sound features vs spectrogram + convolution

I am working on classifying different sounds ( not speech or words exactly something like ambulance alarm, police alarm, cough sounds etc) I read few paper which suggested to extract dsp features such ...
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46 views

Apply filter before or after Downsampling?

Context: I'm trying to do formant estimation. My first attempt didn't work out all that well so I'm trying to follow closely Praat since their formant estimation is remarkably accurate. Question: I ...
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1answer
46 views

what is OQAM (Offset QAM) and why it is required?

Currently I am studying about a modulation technique FBMC (Filter Bank Multicarrier) which is a predecessor of OFDM. The difference with OFDM is that, FBMC does not have a CP (Cyclic Prefix) which is ...
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1answer
46 views

How to get rid of discontinuities at the start of a signal caused by the filter

I'm processing an audio signal using an overlap add method.Everything at the block boundaries is fine. However I have a discontinuity at the very start of the first block and I'm not sure how to (or ...
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1answer
39 views

Bandwidth of a bandpass signal

If the Fourier transform of an aperiodic continuous time signal has signal components between the minimum frequency w1 and the maximum frequency w2, but not all the frequencies between w1 and w2, is ...
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How to identify the vocal folds in a mel spectrogram image?

I am quite new in the dsp and trying to learn How to identify the vocal folds in a mel spectrogram image? My image data looks like this. How to find the vocal folds in this image? I am using Praat ...

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