Questions tagged [digital-filters]
The digital-filters tag has no usage guidance.
538
questions
2
votes
1
answer
62
views
A recursive filter without an infinite impulse response
I'm reading Richard Hamming's book, The Art of Doing Science and Engineering, and in the chapter Digital Filters IV he says
"These recursive filters are often called 'infinite impulse response ...
1
vote
1
answer
43
views
Ringing relation to phase response
I’ve been reading lately about analog filters, and in “Electronic Filter Design Handbook”, Williams & Taylor state that the fact that the Bessel filter exhibits no overshoot in its step response, ...
1
vote
0
answers
80
views
Digital signal processing from FMCW rangefinder
I am developing a FMCW rangefinder and have encountered a number of questions regarding digital signal processing.
So I have a sawtooth modulating signal with a frequency of 1 kHz, which is fed to the ...
0
votes
1
answer
34
views
How to interpret the transfer function to determine if the filter is highpass or lowpass
A digital filter $y_t = x_t - a_1 x_{t-1}$ has the absolute value of its transfer function $\left| \mathscr{H}(z) \right|$ as $\left|z - a_1 \right|$. A Digital Signal Processing Primer says $z = -1$ ...
0
votes
2
answers
57
views
Why $z = e^{jω}$ can be a constant a1 < 1 in the transfer function of a dsp?
A section in the book A Digital Signal Processing Premier by Ken Steiglitz examines the transfer function of a digital filter to analyze the filter behavior. It says $z = e^{jω}$ can be ...
0
votes
1
answer
32
views
Signal excision of known modulation type using costas loop for carrier recovery? or is there a better approach?
I am trying to excise a modulated signal (selectively remove it as best possible). could be BPSK, QPSK ect but the modulation type is known on the receiver. I have been trying a combination of lowpass ...
0
votes
2
answers
40
views
Recovering phase response from frequency magnitude of a minimum-phase system: Practical considerations
My problem is this: Given a frequency magnitude response $|\hat{h}(\omega)|$ of a minimum phase system, how can one reliably recover its phase response?
In my case, the magnitude is quite short (~30-...
0
votes
1
answer
67
views
Scipy butterworth filter sos (second order sections) output - what it is and what is the benefit?
scipy butterworth filter has sos output option. Please help understand what it is and what the benefit in layman's term.
output{‘ba’, ‘zpk’, ‘sos’}, optional
Type of output: numerator/denominator (‘...
0
votes
1
answer
48
views
s to z domain transformation
I have $2^{nd}$ order bessel filter transfer function. I want to model this analog bessel filter in FPGA. For this I need its digital or discrete form. I am trying to get digital filter's coefficients ...
1
vote
2
answers
54
views
Runtimes of FIR for various number formats
I read ST's application note regarding the implementation of DSP operations using their dedicated DSP HW using CMSIS implementation.
They compare performance for an FFT and an FIR filter, each using 3 ...
0
votes
2
answers
112
views
Oversampling and decimation leads to slightly different result in frequency domain
I am trying to explore the effect of oversampling and decimation with the following Matlab code
...
0
votes
2
answers
110
views
What is the numerical error of the signal.iirdesign methods in scipy?
How large are the numerical errors of the signal.iirdesign method from SciPy?
I am trying to implement a digital filter on an FPGA, but at low frequencies, the filter is often unstable. I am not sure ...
2
votes
1
answer
352
views
Is it possible to normalize FIR filter coefficients to "perserve gain on average"?
Let's say we are given a random discrete time FIR filter.
I'm wondering if there is a standard way to normalize its filter coefficient in a way to make the filter "energy preserving on average&...
0
votes
1
answer
198
views
Gaussian Pulse shaping filter
I need expert advice on implementation of Gaussian Pulse Shaping filter for generating GMSK signal in further steps. I have implemented as follows, Can anyone confirm that the Pulse would be shaped as ...
0
votes
2
answers
113
views
Butterworth filter vs Elliptic filter - artefacts and huge transient
I'm working on a project involving digital filters and DFT. I want to use sampled ECG data to determine the heart rate using power spectral density. While filtering the ECG samples (samples are from ...
0
votes
0
answers
88
views
Low pass Thiran filter difference equation in Python
I'm attempting to build a time domain low-pass Thiran filter (in Python if that is relevant) based on the original article:
https://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=1083363&tag=...
2
votes
2
answers
156
views
How do you interpret the sign (positive, negative or zero) of the phase spectrum?
How do you interpret the sign (positive, negative or zero) of the phase spectrum of:
1)a digital filter (the phase spectrum of its frequency response aka phase response of the filter).
2)a signal in ...
0
votes
1
answer
98
views
algorithm for second order butterworth filter
I want to implement an algorithm for a second order butterworth filter on the form
$
H(s) = \dfrac{Y(s)}{U(s)}= \dfrac{1}{\left(\frac{s}{w_0}\right)^2+2\zeta\frac{s}{w_0}+1}
$
I want to get it on the ...
0
votes
1
answer
34
views
Software based energy meter
I have been implementing a software energy meter. In other words my software has to integrate immediate electrical power over time. The immediate electrical power is calculated based on following ...
2
votes
5
answers
518
views
Why does the frequency sampling method for FIR filter design operate in this manner?
I'm studying FIR filter design and it's time for the frequency sampling method, my teacher said that to use this method you need to follow the following steps:
Sample the periodic frequency response ...
0
votes
1
answer
67
views
16-QAM with RRC filter shows a point in the middle (bad EVM)
I build a PCB with the DAC3174 from TI and I'm using it initially to test it as a channel, like shown in the diagram from the datasheet:
The analog signal is mixed with the PLL carrier frequency (at ...
2
votes
0
answers
70
views
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 ...
1
vote
0
answers
58
views
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 ...
0
votes
1
answer
38
views
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 ...
0
votes
2
answers
78
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 $1250 \, \textrm{MHz}$...
0
votes
1
answer
45
views
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 ...
0
votes
1
answer
66
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 ...
1
vote
2
answers
87
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 ...
3
votes
0
answers
122
views
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 ...
1
vote
3
answers
78
views
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, ...
0
votes
0
answers
31
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 ...
3
votes
2
answers
765
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 "...
2
votes
2
answers
265
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 ...
1
vote
0
answers
52
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) ...
0
votes
1
answer
103
views
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 ...
2
votes
4
answers
353
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 ...
3
votes
1
answer
168
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 ...
0
votes
1
answer
64
views
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 ...
2
votes
0
answers
121
views
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 ...
2
votes
2
answers
207
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 ...
1
vote
1
answer
127
views
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 ...
0
votes
0
answers
71
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 ...
1
vote
0
answers
45
views
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');
...
1
vote
2
answers
130
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 ...
2
votes
0
answers
54
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 ...
1
vote
1
answer
312
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 ...
1
vote
2
answers
114
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 &...
1
vote
2
answers
71
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.
...
1
vote
1
answer
69
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 ...
2
votes
1
answer
403
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 ...