Questions tagged [digital-filters]
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308
questions
0
votes
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19 views
Richardson-Lucy iterative deconvolution is producing erroneous deconvoluted images
I am working on producing deconvoluted image from degraded image where the process of degradation is known to be linear and space invariant. I have used Richardson-Lucy iterative deconvolution method ...
0
votes
1answer
33 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 ...
0
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0answers
9 views
short frame subband coding
I am implementing a subband ADPCM codec, Fs=16 kHz, frame length 3ms, band1=50Hz-2kHz, band2= 2kHz-8kHz. I used the FIR filter for subband coding and %50 overlap. My reconstructed signal does not ...
0
votes
1answer
28 views
Is the sampling period arbitrary for a bilinear transform, and why?
I have been given the specifications for a digital highpass filter (stopband, passband, stopband attenuation and maximum passband ripple).
I am expected to design a prototype lowpass filter in the ...
1
vote
1answer
33 views
Output of IIR filter not matching the frequency response - Proximity to fs/4
After not having dealt with digital filters for a long time, I have now been playing around with filter design in octave and I am observing a behavior which I do not understand.
After designing a low-...
2
votes
1answer
37 views
How to modify zeros and poles in a delta-sigma modulator loop?
In this specific case, I am modeling a delta-sigma analog-to-digital converter with a basic loop in which the output $V$ is fed back with unity gain to be subtracted from the previously sampled input $...
0
votes
0answers
21 views
Filter using frequency and magnitude
I'm used to filtering using FIR filters with a cutoff and dB attenuations etc.
The issue I have is that I have pulses in my signal which contain similar frequencies as the noise. I'd like to totally ...
0
votes
1answer
24 views
Why does Pre-warping work with a notch filter but not a low pass filter?
Hey guys a little bit of a continuation from a previous post I made here, I found the solution as why the pre-warping was not working at the time,
However I was just playing around and found out pre ...
0
votes
2answers
68 views
Pre-Warping is not working on bilinear transform
New here and I just wanna ask the question of what happens when your pre warping doesnt work on your bilinear transform?
I am assuming my isnt working as part of my research I have read that the ...
-3
votes
2answers
143 views
Can digital filters actually separate signals?
I've come to realize what appears to be a hard limit in convolution theorem: to avoid time-domain aliasing, we must pad the signal/filter, but the padding distorts the spectrum.
Consider a ...
-1
votes
1answer
223 views
Is there really no perfect digital filter?
In context of transition width vs. stopband attenuation, different windows (Blackman, Hamming, etc) are compared in terms of tradeoffs between the two, always noting that one cannot perfect both.
Why ...
2
votes
2answers
31 views
Finding the coefficients of the block diagram
I was looking at the question from a textbook of mine and it doesn't make too much sense to me:
Design a first-order filter, linearly extrapolating the current and
previous measurement. This is ...
0
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1answer
25 views
Detect static in audio with Python
I have a program running in python. It samples the demodulated FM output from an SDR. When there is no transmission, the input to the program is completely static.
How can I go about squelching the ...
1
vote
1answer
54 views
How to implement a series of second-order, digital state-variable filters in MATLAB?
How to implement a series of second-order, digital state-variable filters, using the high-pass output from each section to effect the low-pass to high-pass transformation in MATLAB?
Hello, I am trying ...
1
vote
2answers
52 views
How can we measure the quality of the FIR filter?
The process of designing filters includes too many parameters like bandwidth, transition, degree of attenuation in side-bands, phase delay, causality, the magnitude of ripples, and so on. First of all,...
0
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0answers
27 views
Interpolationg phase and magnitudes, Transformation function
I am trying to filter signal x(n), n = 150. I made a filter with few frequency points on the x axis , [-11., -9., -3., -2., -1., 1., 2., 3., 9., 11.])
...
0
votes
1answer
41 views
Digital filter as physical device
I'd like to find some book or books to get information about how digital filter is built depending on specifications. Like depending on price, or speed and etc. As I know it could be built by some D ...
2
votes
2answers
64 views
How Come the Low Pass Filter in Sobel Operator Isn't Normalized?
I am relatively new to the field of computer vision and I have just learnt about the sobel operator. The sobel operator in the x direction is a convolution of the finite difference kernel $[1,0,-1]$ ...
0
votes
1answer
61 views
How to evaluate fixed-point implementation of LMS filter is correct?
I am having an LMS block with 6 filter coefficients.
The value of filter coefficients are
0.0001
0.00045
0.2535
0.546536
0.0000243
0.3423
I have tried to ...
2
votes
0answers
54 views
Filtering a square signal with a median filtering to preserve the edges
If needed, you can find my first post for this problem here.
I am trying to clean the following signal:
As proposed in the comment, I tried to use this post proposing 2 methods: median filtering and ...
0
votes
1answer
47 views
Applying a lowpass filter to a noisy square signal leads to a shift of the signal
My knowledge of analog and digital signal analysis is behind me.. thus I do not have enough confidence in my procedure to analyze my signal, and I do not know which direction to pursue to improve. I'm ...
0
votes
2answers
55 views
0
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1answer
30 views
Design of Parks-McClellan Lowpass Filter
Can we design Parks-McClellan Lowpass Filter with only sampling frequency of 12KHz and cutoff frequency of 4KHz and order of 30? I went through Matlab help, where we need parameter like passband ...
3
votes
1answer
61 views
Advice on designing a digital filter that doesn't have phase-sensitive edge artifacts?
I'm pretty well versed in statistics, but not really digital signal filtering. I have a data scenario where I expected to be able to pretty easily filter out some noise (human pulse) that's at a known ...
1
vote
2answers
63 views
Question about ideal filters
Following shows the discrete time Fourier transform of an ideal low pass filter with cutoff frequency $\omega_c$:
$$H\left(e^{j\omega}\right) =
\begin{cases}
1, & \text{if $|\omega| \le \omega_c$...
1
vote
1answer
40 views
Which one can be accompanied by linear filters?
I have a matrix :
\begin{bmatrix}
1 & 2 & 3\\
1 & 4 & 5\\
2 & 6 & 7
\end{bmatrix}
After doing operation 1 ,
I get
\begin{bmatrix}
0 & 0 & 0\\
1 & 2 & 3\\
1 &...
0
votes
1answer
39 views
How to find minimum length of a FIR symmetric filter if I am given DTFT
I am practising for upcoming exams and came across this question.
Let $h[n]$ be an FIR filter such that $h[n] = 0$ when $|n| > M$ and $h[n] = h[−n]$. A plot of $H(e^{j\omega})$ (DTFT of $h[n]$) is ...
1
vote
1answer
59 views
Resampling FIR coefficients
I have received from a client a 32-coefficient FIR filter running at 1 kHz. I would like to adapt the filter to a 24-coefficient FIR filter running at 750 Hz while preserving the 0-375 Hz frequency ...
4
votes
1answer
199 views
Digital filter coefficients from low-pass to high-pass
Given I have coefficients a0, a1, a2, b1, and b2, defining the difference equation for a digital filter as:
...
0
votes
2answers
106 views
Complex Samples (IQ) - Baseband Filtering
We are currently analyzing a large set of IQ samples in a desktop application and we are interested in implementing many different bandpass filters dynamically.
We realized, that working with scipy ...
0
votes
1answer
47 views
How to calculate DTFT of cosine function divided by n
I'm having a hard time to calculate the next function, and I don't really know Matlab good enough to calculate it there.
Help would be appreciated:
$$h[n]=\frac{A_1 \cos[\theta_1(n-N/2)]}{n-N/2}$$
0
votes
0answers
18 views
cascade realisation from direct realisation
I am trying to convert filter realisation from Direct form 1 to cascade using function provided from Digital signal processing by Proakis. I designed lowpass IIR filter and extracted numerator and ...
1
vote
1answer
74 views
Frequency conversion in IIR lowpass-to-lowpass transformation
We use
$\ z^{-1} \xrightarrow{} \frac{z^{-1}-\alpha}{1-{\alpha}z^{-1}}$
and
$ \alpha = \frac{\sin(\omega_c^{'}-\omega_c)/2}{\sin(\omega_c^{'}+\omega_c)/2}$
for a lowpass-to-lowpass frequency ...
2
votes
0answers
25 views
Effect of time duration in bandpass filtering
I am new in DSP, but I am trying to process EEG data to decompose into the bands, theta, alpha, and others.
The raw data is processed:
Remove the DC offset by subtracting the average value from the ...
0
votes
1answer
64 views
How to quantize filter coefficients to 10 bits
I am trying to learn digital filter design. One of the concept is quantization of filter coefficients. I used Designfilt to design an lowpass IIR filter and obtained coefficients. Now I want to ...
1
vote
0answers
42 views
The size of an FIR filter for PDM-PCM conversion
I have been trying to understand how the conversion from PDM to PCM happens so that I could interface with a microphone. I have been doing some research online and I found that the most common way to ...
3
votes
1answer
40 views
3 dB bandwidth of a filter
I am solving one of the question from signal processing about calculating the 3dB bandwidth of the filter. But I am stuck once I got the magnitude response of the filter. Can someone help me with this?...
-2
votes
1answer
46 views
Stop band ripple vs stop band attenuation
What is difference between stop band ripple and stop band attenuation in context of filters?
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0answers
24 views
Fixed Point, Time Domain and Freq Domain Filter Difference
I am trying to compare my IIR filter in time domain and freq domain, but i am getting different results which i dont understand why ?. So please help me out in understanding the issue.
My code is :
<...
1
vote
1answer
45 views
Finding number of alternations in parks mcclellan
I'm given this question and asked to find number of alternations for filter 1 and 2, using parks mcclellan algorithm.
I search for solution and find this answer :
My question is why the number of ...
1
vote
1answer
49 views
Impulse Invariant method for digital filter design
One of the known methods for discretizing analog filters is impulse response invariant. We get the impulse response in time domain, discretize it and then get the Z transform.
What I am trying to ...
0
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0answers
27 views
Remove noise from distance calculated from accelerometer data
I got distance calculated from accelerometer raw datas using double integral accelerometer.I attached the image below.But it shows noise with it.Please help me to remove this noise.
3
votes
2answers
238 views
Resources for learning about filters from scratch?
Topic of filters is very important in signal processing
So i want to learn about filters from scratch and then eventually reaching FIR and IIR filters
I have some idea about low pass,high pass,band ...
1
vote
2answers
102 views
Ideal Lowpass Filters [closed]
I am a beginner to Digital Signal Processing. I took this course of DSP from Coursera in which I am stuck with this question. Since, I am new, please be descriptive. Any help will be appreciated. ...
0
votes
0answers
30 views
Increase signal to noise ratio
I work with the K-LC6 microwave module. Previously, when the microwave module was assembled on discrete elements, the signal-to-noise ratio was 5. Now, with the transition to specialized microcircuits,...
1
vote
2answers
133 views
Converting cascaded biquads to parallel biquads
I need to convert a fourth-order IIR filter made using two cascaded second-order sections with the transfer function:
$$
H(z) = \frac{b2_{1}z^{-2} + b1_{1}z^{-1} + b0_{1}}{a2_{1}z^{-2} + a1_{1}z^{-1} ...
2
votes
2answers
133 views
3 dB cut-off frequency of first-order IIR high-pass filter
As descipbed in this great answer (thank you Matt L.), the suggested exact formula was very precise.
I need a first order high-pass with similar precision.
EDIT: (clarification)
The low-pass filter ...
0
votes
0answers
21 views
Threshold Estimation from a Linear FM Pulse Position Modulated signal
I have a signal chain to detect multi channel LFM (Linear Fm - Hann window being applied as shaping filter for the pulses) pulses as shown in picture below. (Numbers in the picture is in Hz., for ...
0
votes
2answers
93 views
Coefficient Scaling of IIR filter to obtain unity gain response
I am designing a 2nd order IIR digital filter :
My tf equation with coefficients is
...
1
vote
2answers
86 views
Calculating the 3dB frequency of an nth order low pass filter and nth order high pass filter
I am using the 6th order high pass filter in the C code. And the cut-off frequency selected for it is 150 Hz and the sampling frequency being used is 32000Hz. Now, I want to calculate the 3 dB ...
