Questions tagged [digital-filters]

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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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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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34 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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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
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1 answer
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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
38 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 answer
151 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
107 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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2 answers
52 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
37 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
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1 answer
142 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
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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
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1 answer
92 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
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0 answers
57 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
66 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
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33 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
271 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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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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1 answer
45 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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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
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1 answer
67 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
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1 answer
94 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
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1 answer
72 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
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3 votes
2 answers
102 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
112 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
66 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
143 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
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1 vote
1 answer
263 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
67 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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1 answer
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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
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1 vote
0 answers
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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
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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
1 vote
1 answer
93 views

Denoising Signal With Butterworth-Filter

im trying to denoise a signal to which i added AWGN. Here is what ive done so far: ...
Tom's user avatar
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1 vote
1 answer
208 views

Filtering with scipy sosfiltfilt generates flat edge artifact

I need to filter some array of data before differentiating it, I do this so I don't drown the signal in noise. It's also important that no delay is applied to the signal, that's why I decided to use a ...
Guille's user avatar
  • 29
1 vote
2 answers
161 views

Correct baseline for slow sinusoidal noise signal

I acquired some noise signal (5GHz samplerate, 520 samples) data from a device and recognized that it is not gaussian distributed. There seems to be some slow sinusoidal signal underlying. I tried to ...
Tim Buktu's user avatar
0 votes
1 answer
64 views

Poor Attenuation With High Pass Filter

I am having issues with my implementation of a FIR high pass filter. The low pass filter has an acceptable level of attenuation in the stopband, but the high pass filter has poor attenuation in the ...
expr_champ2's user avatar
0 votes
2 answers
120 views

Why do inverted impulse responses sound awful when applied to an audio system?

In this answer, the responder claims that acquiring a measured impulse response and straightforwardly inverting it (e.g. by directly in the spectrum, after zero-padding, or by determining the inverse ...
oliver's user avatar
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0 answers
326 views

Calculating the filter coefficients for 2nd order bandpass and bandstop Butterworth filter

I want to calculate the filter coefficients for both bandpass and bandstop filter for 2nd order Butterworth filter, like ...
Devil_____03's user avatar
1 vote
1 answer
221 views

Fast-settling CIC/FIR filter design

I am investigating the properties of the cascaded integrator comb (CIC) filter (for a $\Sigma\Delta$-modulator). I have two questions: What's the correct method to design a fast settling decimation ...
Ring's user avatar
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-3 votes
2 answers
268 views

"There's no ideal lowpass filter" - really?

Sinc is $\propto 1/t$. If $x(t)$ is bounded, then there exists $t: |x(t)/t| < \epsilon_M$, where $\epsilon_M$ is machine epsilon. If $x(t)$ is also time-limited, it also means there's $\tau$ such ...
OverLordGoldDragon's user avatar
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0 answers
79 views

Real coefficient linear phase FIR filter design based on least square

I want to build up linear phase filter by least square approach. I have formulated a least square problem similar to Matt's blog. To approximate the desired complex response $F \times 1 $ vector $d$ ...
simon Liao's user avatar
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0 answers
62 views

How to remove PWM cycles present in a signal?

I have applied PWM across a DC motor. The voltage measured across the motor is shown above. I want to count the ripples present in the data. The ripples exist at a different frequency than the PWM ...
Nitin Singh's user avatar
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0 answers
48 views

transformation methods for digital filters

i have the following question: in digital filter design what's the difference between the methods of transformations : bilinear vs impulse invariance vs Euler vs step invariance. thank you! Here is a ...
imene's user avatar
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0 votes
2 answers
125 views

Band pass filter = Band stop filter if you switch the cutoff frequencies?

The normalized frequency response of a N points bandpass digital filter is: $\frac{sin(\omega_{c2}(n-M))}{\pi(n-M)}-\frac{sin(\omega_{c1}(n-M))}{\pi(n-M)}$ where $\omega_{c1}$ is the normalized lower ...
Volpina's user avatar
  • 295
1 vote
1 answer
317 views

Difference between FIR and IIR filters

I understand that FIR filters have a finite impulse response and IIR filters have an infinite impulse response. Here's the issue: A signal is finite in time if and only if it is infinite in frequency. ...
Levi's user avatar
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1 answer
105 views

FIR filter for flipping the phase in negative phase

I have built up a FIR filter based on least square approach. I deploy the IDFT matrix $$F_{inv}$$ in above document page 7 and the desired response for compensated is H_F, the objective is $$ min_{...
simon Liao's user avatar
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1 answer
163 views

Fixed point implementation of IIR filter

I am trying to implement an IIR filter followed by downsampler with the following specifications: fs=18000,order=8,3db freq=[500 1000],downsampling factor=6. ...
Deepa's user avatar
  • 293
1 vote
1 answer
239 views

Using Zero-Phase Anti-Causal Filters in Real-Time Embedded Systems

Wanted to know the feasibility and usefulness of implementing Zero-Phase Anti-Causal filters such as those mentioned at this link in modern embedded signal processing applications given the ...
malik12's user avatar
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-1 votes
2 answers
708 views

Implementing a high pass filter in MATLAB

I am trying to implement a high pass digital filter in MATLAB. High pass filter: ...
Volpina's user avatar
  • 295
0 votes
2 answers
83 views

Behaviour of difference equation does not match that of its z-domain transfer function

I have obtained the following z-domain transfer function: $\frac{Y(z)}{U(z)}=\frac{3.3641×10^{-7}×z^{6}+1.5584×10^{-5}×z^{5}+6.7263×10^{-5}×z^{4}+ 5.5016×10^{-5}×z^{3}+8.525×10^{-6}×z^{2}+1.2303×10^{-...
Daniel Tork's user avatar

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