Questions tagged [digital-filters]
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339
questions
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2answers
44 views
Should we use a digital low pass or high pass filter to remove sensor bias?
Consider a continuous signal oversampled at, say $2 \;kHz$, and then system digital low pass filtered to a $100\;Hz$ frequency which is the control loop frequency. It is known that there is some bias ...
1
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2answers
52 views
why GMSK use a gaussian filter instead of raised cosine filter?
I'm wondering why GMSK use a gaussian filter instead of raised cosine filter as a pulse shaping filter? Is there special reason?
0
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1answer
35 views
Echo generation of FIR system using Matlab
How can I generate some echo to the system below?
Here is my code. I would appreciate any help.
...
0
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2answers
42 views
OFDM use a pulse shaping filter or not?
I've heard in various literature that band limited channels use pulse shaping filters. By the way, In OFDM it use a cyclic prefix to overcome ISI, so I wonder if OFDM does not use pulse shaping filter....
0
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0answers
32 views
How to find the minimum no of bits required to express the filter coefficients in fixed point format from Histogram plot?
I have designed a filter in MATLAB and now I have to convert the filter coefficient in fixed point format from floating point format. For that my professor asked me to find histogram first and then ...
0
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1answer
42 views
Digital lowpass filter causing aliasing
I'm writing a biquad filter program in C++, using the LBJ Cookbook as the source for the formulas. https://webaudio.github.io/Audio-EQ-Cookbook/audio-eq-cookbook.html
The lowpass filter that I've made ...
0
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2answers
40 views
Designing digital low pass filter with low pass-band group delay
I'm interested in specifying a digital low pass filter with low (and near constant) pass band group delay. I tried python's spicy.signal.minimum_phase to transform ...
0
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0answers
50 views
Hardware implementation of multi rate FIR filter
I have designed a multi rate fir filter in MATLAB and I want to implement it in verilog.This is the block diagram.
Total number of taps are 391. Is there any way to automatically generate verilog ...
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0answers
36 views
understanding FIR SRRC filter with memory implementation in C
I translated the Matlab QPSK receiver/transmitter into real-time C code that I have running on a PlutoSDR platform. QPSK receiver/transmitter
I wrote the square root raised cosine filter for the TX ...
0
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1answer
38 views
Digtial FIR Impulse response & transfer function
I am currently working the figure through below.
as it is an FIR Filter i have worked out using convolution that the output is 4,2,4,6,0,0.
i am trying to obtain the 'z' domain transfer function of ...
1
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0answers
12 views
Calculate ScaleVlues of IIR Filter manually from sosMatrix of MATLAB
while using "design" in matlab to design filters, at the output we can see a ScaleValues matrix that is used to compensate for overflow problem. my question is that how this matrix is ...
1
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1answer
65 views
Take FFT of audio represented in PDM
I have a PDM data input from the MEMS microphone on the Pynq-Z1 development board. My eventual goal is to take the FFT of the audio samples and display the frequency bins on some LED.
I am struggling ...
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4answers
76 views
What is the correct gain of an RRC Filter?
Breaking my brain all morning with this reading previous questions and googling ...
I have made an RRC filter from the equation on wikipedia. It works fine and I compared it to commpy library in ...
3
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0answers
58 views
Wave Digital Filter Diode Equation?
I'm trying to understand how diode circuits are implemented in wave digital filters, particularly for clippers. The research papers and other sources I've looked at use the equation
$$I(V) = 2I_s \...
0
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1answer
79 views
Sub-Band Extraction Techniques for Complex (I/Q) Signal
There is an application which processes Complex I/Q signal of 100 MHz bandwidth and it is required to extract independent sub-bands from this bandwidth at different center frequencies and bandwidths. ...
1
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1answer
56 views
Setting up RRC filter and useful equations
I have looked at previous answers but cant seem to get everything I need in one place to help me and my textbook doesnt explain this in practical terms.
Signal Parameters:
$F_s = 44100$
$R_s = 300$
$...
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0answers
48 views
How to break a second-order filter into two first-order filter
Let's assume the transfer function of a continuous-domain filter consists of two poles and one zero:
$H(s) = \frac{k_c (s-\omega_{z_1})}{(s-\omega_{p_1})(s-\omega_{p_2})}$.
Let's consider we do the bi-...
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0answers
59 views
The optimal estimator using FIR Wiener filter
The received signal $y(n)$ is defined as
$$y(n) = h(n)*x(n) +v(n)$$
where $h(n)$ is unknown impulse response and $v(n)$ is a white Gaussian noise.
The value of $x(n)$ for $n=1, ..., 32$ are given. The ...
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0answers
37 views
How to design IIR digital filters
Practical Infinite-Impulse-Response (IIR) filters are usually based upon analogue equivalents (Butterworth, Chebyshev, etc.), using a transformation known as the bilinear transformation which maps the ...
1
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2answers
48 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
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1answer
43 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
28 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
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1answer
38 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
124 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 ...
0
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2answers
61 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 ...
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0answers
53 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
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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
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1answer
44 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
49 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}}$$
The input to this filter has frequencies ...
0
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1answer
93 views
Digital Signal Processing Filters Problem
Hello I need some help with this question.
$$𝑦[𝑛] = 𝑎_1𝑥[𝑛 − 𝑘_1] + 𝑎_2𝑥[𝑛− 𝑘_2] + 𝑏_1 𝑦[𝑛− 𝑘_3]$$
Set $a_1, a_2$, and $b_1$ to any positive or negative real numbers of your choice. Set $...
0
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0answers
22 views
interpreting the post filter features in a sensor data
I am new to signal processing and I am working on Filtring signals(using basic analog filters, FIR filters, and IRR filters), for Human activity recognition, using sensor data, I Extract the features ...
3
votes
2answers
118 views
High-pass filter with quantized (fixed-point) coefficients
I have a high pass IIR filter with numerator
b = [0.997688673703782,-1.995377347407564,0.997688673703782];
and denominator
...
0
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0answers
15 views
How to make 2 finger rake receiver to counter intersymbol inferference?
I have to transmit Walsh sequence (spreading codes) modulated with 8 psk with a channel impaired by intersymbol interferences, fast fading .
Suppose i have a stream of data for instance 1 0 1 1, 0 0 0 ...
0
votes
1answer
55 views
High pass filter not attenuating signal at start of input
I have implemented digital high pass filter with below specs:
Sampling frequency = 5333 Hz
Cutoff frequency = 10 Hz
Order = 3
We are facing problem like below:
As ...
1
vote
1answer
58 views
Matlab filter implementation
I have a filter response in the $z$-domain.
$$H_0(z) = 2 + 6z^{−1} + z^{−2} + 5z^{−3} + z^{−5}$$
How do I implement this filter on MATLAB?
I can do this if I have the coefficients but I do not know ...
1
vote
1answer
69 views
How to determine the type of a digital filter given its expression?
I'm trying to understand a variant of the filter presented in http://www.stockspotter.com/files/PredictiveIndicators.pdf in the "Code Listing 1.". Its expression is given as follow:
$$y_n = ...
2
votes
2answers
242 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
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0answers
56 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
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1answer
58 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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1answer
41 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
41 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
46 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
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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
74 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
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2answers
131 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
156 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
270 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
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2answers
33 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
49 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
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1answer
101 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 ...
