Questions tagged [digital-filters]
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324
questions
1
vote
2answers
22 views
Scipy butter filter - remove DC offset w/ high sample rate
I am having a hard time figuring out how to employ a high pass filter to remove the DC offset of my data signal with the "scipy butter" function because my sample rate is quite high. The ...
1
vote
1answer
24 views
Transposed Direct Form II
In the book digital Signal Processing by Proakis 4th Edition, at page 593 a table is drawn about some second order modules for discrete time systems. In that table there Transposed Form II drawn But ...
1
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0answers
25 views
filter single radio station with sdr [closed]
I want to write a program to filter a single radio station using SDR. Not for any application just a stepping stone to learn. I am going to use C so it is easier for me to see the steps and things ...
0
votes
1answer
32 views
Echo Removal from PCM 16 bit arrays (time frames)?
I am keen to know the basics to remove echo knowing FarEnd Signal and NearEnd Signal. I have read research papers in which its commonly said:
y(n) = (hT)x
where h is the room impulse response and T ...
4
votes
2answers
107 views
DSP Book with More Applications
The process of my DSP class used the Oppenheimer DSP book as text. Although I can easily understand the math in the book, I found it lack examples and applications, hence I still cannot build a ...
1
vote
2answers
52 views
Summation of two IIR filters
Actually my question is how to get the weighted average of two or three IIR filters given their transfer functions, which is nearly equal to this question. How to calculate the total transfer function ...
0
votes
0answers
38 views
FIR filter design using Kaiser window - passband ripple and stopband attenuation
I have been given the following parameters :
lower cutoff frequency = $\pi/3$
lower cutoff freq = $2\pi/3$
beta = 5.6533
M = 51
Now I have to get magnitude and phase response of the filter using ...
0
votes
4answers
47 views
Filtering Angular Measurements
I am trying to apply a filter to reduce noise in angular measurements. Since the data comes with noise, the filter needs to be able to handle measurements that are not smoothly progressing; for ...
0
votes
1answer
38 views
How to remove frequency offset with RRC as pulse shaping filter
I am working on a project with QPSK modulation format and a RRC filtering at the transmitter (TX) side. The channel introduces a constant dropper shift ($f_d$) to the received signal. To make the ...
0
votes
1answer
46 views
Design of discrete time second order all pass filter
I want to design a second order digital all pass filter with the transfer function given by
$$H(z)=\frac{a_2 +a_1 z^{-1}+z^{-2}}{1+a_1 z^{-1}+a_2 z^{-2}}$$
The input to this filter has frequencies ...
0
votes
1answer
89 views
Digital Signal Processing Filters Problem
Hello I need some help with this question.
$$š¦[š] = š_1š„[š ā š_1] + š_2š„[šā š_2] + š_1 š¦[šā š_3]$$
Set $a_1, a_2$, and $b_1$ to any positive or negative real numbers of your choice. Set $...
1
vote
0answers
20 views
interpreting the post filter features in a sensor data
I am new to signal processing and I am working on Filtring signals(using basic analog filters, FIR filters, and IRR filters), for Human activity recognition, using sensor data, I Extract the features ...
4
votes
2answers
103 views
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
...
0
votes
0answers
14 views
How to make 2 finger rake receiver to counter intersymbol inferference?
I have to transmit Walsh sequence (spreading codes) modulated with 8 psk with a channel impaired by intersymbol interferences, fast fading .
Suppose i have a stream of data for instance 1 0 1 1, 0 0 0 ...
0
votes
1answer
53 views
High pass filter not attenuating signal at start of input
I have implemented digital high pass filter with below specs:
Sampling frequency = 5333 Hz
Cutoff frequency = 10 Hz
Order = 3
We are facing problem like below:
As ...
1
vote
1answer
56 views
Matlab filter implementation
I have a filter response in the $z$-domain.
$$H_0(z) = 2 + 6z^{ā1} + z^{ā2} + 5z^{ā3} + z^{ā5}$$
How do I implement this filter on MATLAB?
I can do this if I have the coefficients but I do not know ...
1
vote
1answer
64 views
How to determine the type of a digital filter given its expression?
I'm trying to understand a variant of the filter presented in http://www.stockspotter.com/files/PredictiveIndicators.pdf in the "Code Listing 1.". Its expression is given as follow:
$$y_n = ...
2
votes
2answers
225 views
A path to get the foundation of SP and DSP for a pure math
I have a Ph.D. in pure math (interested in Harmonic analysis and operator theory). I am looking forward some proper references to lead me get the foundation of discrete/signal processing more and ...
0
votes
0answers
53 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
51 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
votes
1answer
37 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
37 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
39 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
35 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
91 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
151 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
251 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 ļ¬rst-order ļ¬lter, linearly extrapolating the current and
previous measurement. This is ...
0
votes
1answer
36 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
75 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
56 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
votes
0answers
28 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
42 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 ...
4
votes
2answers
148 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
71 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
59 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
61 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
66 views
0
votes
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
65 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
74 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
45 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 $h[n] = h[ān]$. A plot of $H(e^{j\omega})$ (DTFT of $h[n]$) is ...
1
vote
1answer
83 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
632 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
143 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
52 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:
$$h[n]=\frac{A_1 \cosā”[\theta_1(n-N/2)]}{n-N/2}$$
0
votes
0answers
23 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
117 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 ...
