Questions tagged [digital-filters]

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A filter to remove f0 and lower harmonics from the signal

I'm studying the perception of vowel formants (resonances of the vocal tract) and need to create stimuli where the signal below the first (lowest) formant is removed. I have some synthesised vowels ...
Renata Koch's user avatar
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How can it be possible that I hear people voice from my computer speaker?

I searched on web and apparently can be due to truck drivers communicating at CB band which is around 20MHz. And apparently this communication may interference with the frequencies used by computer's ...
allexj's user avatar
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How can I make use of Saramäki and MATLAB implementation of the Remez multiple exchange algorithm for FIR filter design

Seeking assistance with an optimized and well-documented implementation of the Parks-McClellan algorithm, I explored beyond the original FORTRAN code provided in A Computer Program for Designing ...
Fadi EID's user avatar
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2 answers
66 views

Mapping PAM-17 levels to PAM-5 levels

A $17$-level $\left\{ -4,-3.5,-3,\ldots,0, 0.5, 1,\ldots, 3.5, 4 \right\}$ digital signal coming at $125 \, \textrm{MHz}$ is passed through an FIR filter ($10$ taps) operating at $1250 \, \textrm{MHz}$...
Isha's user avatar
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1 answer
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Quasi-static-detector operation principle

Im currently working on a project to get navigation information of a robot from Data-Fusion through different IMUs. First step is to make a calibration and I picked a method where I need to detect ...
Orthos's user avatar
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1 answer
53 views

Unstable high-pass filter on mouse-generated signal

I am making a Desktop application using PyQT as the app is a Digital Filter You can add zeros and poles to a unit circle and apply the filter to the data the data can be of two types 1- saved data on ...
Mohamed Hazem's user avatar
1 vote
2 answers
75 views

Unstable high-pass filter on mouse-generated signal using Python

I am practicing some DSP techniques along with desktop application development using Python and PyQt. In my filter design application, I am able to create a filter by placing zeros and poles on a unit ...
Kafka's user avatar
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3 votes
0 answers
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Implementing a non-causal Wiener filter

I'm attempting to implement a non-causal Wiener filter for removing additive white noise from a measured signal. I've had some trouble and I have reduced my problem to a demonstration on simulated ...
Marc's user avatar
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3 answers
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Expression for stopband deviation of a digital filter

Beginning from the basic definition of decibel that expresses the ratio of two amplitudes as $20\log_{10}(A_{2}/A_{1}) $, how do we arrive at the expression $-20\log_{10}(\delta_{s})$ measured in dB, ...
Mukhtar Baqar's user avatar
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24 views

Cutoff Frequency Calculation for Basic Multimode / State-Variable Filter

I am working with a basic multimode / state-variable filter topology which combines highpass, lowpass, and bandpass outputs. I am wondering if there is a formula I can use to calculate cutoff ...
Emmett Palaima's user avatar
3 votes
2 answers
735 views

Confusion on when a filter is considered band-pass

As part of an assignment, I was given a filter with a specific set of poles and zeros, namely poles at $0.51 \pm 0.68i$ and zeroes at $1.5$ and $−0.7$. This part of the assignment is titled as "...
Nyquist-er's user avatar
2 votes
2 answers
182 views

What is the meaning of the filter coefficients in case of an analog filter?

I'm trying to improve my poor understanding of analogue filters. Similarly to this question, I've always wondered why libraries like scipy offer an option to design an analogue filter, and when I ...
bluenote10's user avatar
1 vote
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50 views

IIR filter design for "data-aided" least-squares filtering

Consider a discrete-time signal consisting of a desired signal component and noise, as follows. $$y_k=d_k + n_k, \qquad k\in\{0,1,\dots,K-1\}$$ This signal is applied to a linear time-invariant (LTI) ...
rhz's user avatar
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1 answer
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Performing experiment on a digital control system

I want to perform experiment(s) on a digital control system. I have several questions: In analog control systems, the adder is usually an operational amplifier. However, since here the system will be ...
Root Groves's user avatar
2 votes
4 answers
317 views

3-point highpass filter

I came across an interesting question from Strang & Nguyen$^\color{magenta}{\dagger}$ where it asks to design a $3$-point (or 4-point) high-pass filter with better performance than the $2$-point ...
Sami Al-Dalahmah's user avatar
3 votes
1 answer
149 views

Total energy of a digital filter's output from the current state

Let's say I've been feeding some digital IIR filter input samples $x[n]$ to get output samples $y[n]$. If after some time $t$ the input becomes all 0s, i.e. $$x[n] = 0, \; \; \forall n ≥ t$$ How can ...
cloudfeet's user avatar
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Reducing 1/f-noise

I have some mems accelerometer (ADXL355 and IIS2ICLX) and i want to get a accuracy of 0,005 ° incline measuring by using the axis of it. I am measuring with arduino over I2C and have a Samplerate ...
Rayjay's user avatar
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2 votes
0 answers
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Implementation of an IIR filter with integer coefficients

I'm trying to figure out the implementation of IIR filters with integer coefficients, but I've run into a problem that I can't figure out. There is an excellent post on this topic, and I took it as ...
red15530's user avatar
2 votes
2 answers
129 views

Design of digital filters with negative group delay

I am currently working on digital filters that can predict my input signal(assume that input signal is bandlimited). In other words, I want my filter to have a flat magnitude response in bandwidth of ...
wanghr's user avatar
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1 answer
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Understanding how to implement high-pass filter with cascaded biquad filters

I'm trying to understand how to implement a high-pass filter using, quote: a cascasded biquad direct form II IIR filter with a cut-off of 8kHz. DSP is a complete new field for me, and the ...
DannyNiu's user avatar
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0 answers
47 views

Mixing multiplied audio signals without clipping

I'm building a vocoder on an FPGA, and now trying to mix $N$ signals that are each the product of some carrier signal in one of $N$ bands and an envelope of a modulator in that same band (essentially ...
Andrew Li's user avatar
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0 answers
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The output length of upfirdn in MATLAB doesn't match that of the theory

I have some doubt about the upfirdn built-in function in MATLAB. First I generate the root raised cosine transmit filter as tRRC = rcosdesign(0.25, 20,4,'sqrt'); ...
Math_Novice's user avatar
1 vote
2 answers
85 views

Truncating the output of a digital filter. Which part to discard?

Suppose I have a signal $\mathbf{x}\in \mathbb{C}^{N}$ and a digital filter with impulse response $\mathbf{h}\in\mathbb{C}^L$, where $L<N$. If we pass the signal through the filter, the output will ...
Math_Novice's user avatar
2 votes
0 answers
47 views

Trying to understand scipy.signal.kaiserord

I am trying to understand scipy.signal.kaiserord example. The example says We will use the Kaiser window method to design a lowpass FIR filter for a signal that is sampled at 1000 Hz. We want at ...
Thoth's user avatar
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1 vote
1 answer
253 views

How to design low pass filter for this case

From control systems perspective, I have two subsystems that are connected in a cascade structure. An outer loop represents a joystick that provides position that runs at 1kHz and an inner loop ...
CroCo's user avatar
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1 vote
2 answers
110 views

Why will the output be imprinted with delay effect or echo after convolving two direct dry signals' FFT values (convolution filtering)

I am recently working on the reproduction of the filtering effect of target play back devices, like phone, or speakers, etc using convolution techniques in MATLAB. I firstly created a function called &...
CPK's user avatar
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1 vote
2 answers
58 views

Matlab - dfilt filter produces different results for SOS and normalized SOS

I think the title says it all. Anyone could provide a solid answer to why it happens? Here is a sample code to replicate the issue. ...
Tərlan Əhəd's user avatar
1 vote
1 answer
50 views

Can the standard deviation of the Gaussian window in a Gabor filter be made infinitesimally small?

My understanding is that the standard deviation of the Gaussian window in a Gabor filter dictates the temporal resolution. Wouldn't it always be better then to make the window smaller, thus achieving ...
SNIreaPER's user avatar
2 votes
1 answer
266 views

What is the difference between STFT and Gabor filter?

My understanding is that a Gabor filter gives us information about when in the input signal does the frequency of interest (equal to the frequency of the sinusoid in the Gabor Filter) occur, thus ...
SNIreaPER's user avatar
0 votes
1 answer
238 views

How to implement a filter for a sigma delta modulator?

I am working on a project involving an accelerometer that outputs a pdm signal, which needs to be decoded to an acceleration. The device itself outputs a clock and data line, we just have to sample ...
Jonathan Lundlee's user avatar
0 votes
1 answer
125 views

What does a Gabor filter "filter" out?

Almost all of the searches online have returned results related to image-processing, but I am looking for a more rudimentary example with some physical intuition (eg; a high-pass filter only lets the ...
SNIreaPER's user avatar
0 votes
0 answers
76 views

Lattice Structure of system $H(z)=1+0.7z^{-1}+1.2z^{-2}-z^{-3}$

When solving $$A_{m-1}(k) = \dfrac{A_m(k)-A_m(m)*A_m(m-k)}{1-A_m(m)^2}\,,$$ the $A_m(m)$ here is $A_3(3)=-1$, which results in infinity if put in above equation. How to solve it?
Simple Man's user avatar
1 vote
2 answers
72 views

Modeling an Acoustic Reflection from a Wall - a Paradox?

I am trying to simulate the reflection of a sound ray, that goes from a sound source, bounces off a wall, and is received by a microphone. The wall has a an absorption coefficient, and a specular ...
Mason Wang's user avatar
0 votes
0 answers
35 views

How to preserve constant segments in the signal during digital filtering

I have a sampled signal, which has constant segments(I pad the signal if some of the BLE packets were lost). I want to apply a digital filter to the signal, say Butterworth and I don't want it to ...
Michael Shevelin's user avatar
1 vote
0 answers
302 views

How to recover the LTI system step response by the known output and input signals?

Having the input signal as a step-like pulse and the output as its distorted version after passing through the system: is it possible to somehow recover the step response of the system? In the Figure ...
Curious's user avatar
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0 answers
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Cfar Algorithm Implementation

I am trying to implementation Cfar algorithm to our radar signal processing algorithm. I obtained some output but I want to be sure that is right. Cfar algorithm's output below like, I'm interpreting ...
Paroz's user avatar
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0 votes
1 answer
60 views

Decoding an analogue signal for a given sampling and signal frequency that aren't in sync

I am trying to decode an analogue signal, that is intended to be converted to a digital one and decoded as USART communication. In the image shown the blue trace is the analogue signal, and the ...
Suva23's user avatar
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1 vote
0 answers
50 views

Z-Transform of a Precoder with XOR

I am reading a paper called "Design and Comparison of Three 20-Gb/s Backplane Transceivers for Duobinary, PAM4, and NRZ Data" and got stuck on seemingly easy thing, how the Z-domain ...
Mohamed Osama's user avatar
0 votes
1 answer
94 views

How to trade poles of All pole filter with zeroes?

I have filter of order 2000 IIR all-pole. I want to implement it via cascade IIR filter. Is there any way to reduces poles order and increase zeroes as its cost. To balance zero and poles. Since all ...
mohammadsdtmnd's user avatar
0 votes
1 answer
116 views

Converting transfer function to cascaded IIR filter [closed]

I have two function, But I don't know which one is correct to convert my Transfer Function(tf) to secound order sections(sos). I'm using GNU octave, and it have intrinsic ...
mohammadsdtmnd's user avatar
0 votes
1 answer
85 views

How to find system output by its step response?

Inspired by this post, I tried to reproduce the procedure described in the answer in Python considering rectangular pulse: ...
Curious's user avatar
  • 355
3 votes
2 answers
123 views

How to create a filter that is a single period of a time-warped raised cosine?

I am attempting to implement a filter described in the paper Functional Count-Comparison Model for Binaural Decoding by Pulkki et al 2009: Here is Figure 9: However, I am quite certain that the ...
Bryn's user avatar
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2 votes
1 answer
125 views

What are the tradeoffs between shifting a lowpass FIR filter vs using a bandpass FIR filter?

Is there a difference between shifting a lowpass FIR filter to an arbitrary frequency offset versus creating an equivalent bandpass filter, such that the passband and cutoff frequencies are identical? ...
BigBrownBear00's user avatar
2 votes
0 answers
82 views

multi h CPM implementation

I have understood the math behind CPM. but recently I came across multi h CPM. I haven't found any good document to explain about multi h CPM. Can anybody explain the math behind it. and how it ...
Mike's user avatar
  • 31
3 votes
1 answer
149 views

How to detect decrease of a digital signal?

I have below given digital signal (sampled with period $T = 1\,\mathrm{ms}$) for which I need to evaluate its decrese. Namely I am interested in regions where the signal decreases faster than $-1000\,\...
Steve's user avatar
  • 395
1 vote
1 answer
464 views

Is this a leaky integrator?

We have implemented a way of approximating a data set with the following lines of code based on the dx/dt of wikipedia of leaky integrator. ...
NonIntellego's user avatar
1 vote
1 answer
109 views

How to initialize states of second order direct form 2 transposed filter

Background This question is an extension to this post. I need to calculate states ($S_1$ and $S_2$) of a second order direct form 2 transposed (DF2T) filter that result in minimum 'transient' when ...
Alborz's user avatar
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0 votes
1 answer
136 views

How to initialize a second order direct form 1 filter for no transient upon filter reset

I need to calculate initial states of a second order direct form 1 (DF1) filter that result in zero transient upon reset (initialization). I'm starting with setting filter input (x) equal to filter ...
Alborz's user avatar
  • 13
1 vote
0 answers
66 views

Tap and Impulse Response of MATLAB Filter

This is a simplified version of the Savitzky-Golay MATLAB functions (sgolayfilt, sgolay). ...
Moses Kim's user avatar
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0 answers
48 views

Jointly determining the weighting matrix and target vector in weighted least square

I have the same weighted least square form weighted least square composed of diagonal weighting matrix $ W $, IDFT matrix $F({\omega})$, desired response vector $D({\omega})$ and FIR Filter ...
simon Liao's user avatar

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