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Questions tagged [digital-filters]

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1answer
97 views

Computation of parameter filter to match a given frequency response

I'm looking for the practical way to compute the parameters ($a_1$, $a_2$) of a digital filter to match a certain frequency response. I'm studying the 12-poles filter of the vintage component SP0256 ...
0
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1answer
32 views

What is the formula to convert from continuous-time transfer function to discrete-time TF and vice versa?

Let's say I have a continuous time transfer function which I know its numerator coefficients $(B^c = [b^c_m, ..., b^c_1, b^c_0])$ and denominator coefficients $(A^c = [a^c_m, ..., a^c_1, a^c_0])$. ...
1
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1answer
12 views

How does cascading low pass modules affect pass band ripples?

This question 10.21 is taken from GATE IN paper 2015. I don't understand the answer given, I mean why is the frequency response at the pass band frequency considered to be $(1-\delta)$, isn't the ...
0
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1answer
27 views

Low pass to low pass transformation coefficient?

I am unable to solve this question, 10.10 from GATE IN 2004 (a previous year question paper for an exam targeted at engineering graduates in India.) So I tried to solve the 10.10 by considering the ...
1
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1answer
24 views

Does the output signal from the FIR filter have transient response?

In matlab, I have shown the signal passed through the FIR filter in the time domain. Then the beginning of the signal is like the transient response. Is the result appropriate?
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1answer
24 views

Why is interpolation a time varying system

I was reading about interpolation (Interpolation and Decimation of Digital Signals - A tutorial Review, Ronald E. Crochiere) and found that Interpolation filter is a time varying system. Can someone ...
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2answers
42 views

Digital Filter Design using difference equations

I am reading a chapter on digital filter design from analog filter design using difference equations. What they do first of all is that they map $s$ (Laplace variable) to $z$ ($z$-transform) by the ...
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0answers
54 views

Digital or analogue filtering?

I'm trying to self learn the art of signal processing whilst moving through my third year pure maths degree. Sorry if my terminology is incorrect however I hope I am understandable! I am looking at ...
2
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1answer
62 views

Sobel and Prewitt operators decomposition

I was reading in the wikipedia page of Sobel operator and Prewitt operator that is possible to decompose these two operators (quote form the Formulation paragraph): "as the products of an ...
3
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1answer
72 views

What's logic behind the constuction of Sobel's filter in image processing?

I am basically very new to this image processing field. I am presently working on edge detection on colour images. While learning the basics of edges and edge detection in images, I encountered image ...
0
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1answer
36 views

Backward finite difference differentiation filter frequency response

As title says what would be frequency response of backward finite difference differential filter, or what would be error of this differential filter, analyzed upon frequency of a signal?
2
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1answer
94 views

How does a digital filter work?

To introduce my situation: I'm developing a digital synthesizer in a form of a C++ library, working with low level APIs like WASAPI, ASIO, ALSA etc. It's probably not very practical and I'm mostly "...
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3answers
63 views

Using white noise to test filter freq. response?

If I want to test a digital filter, is using white noise a good way? I mean input white noise to the filter, and visualize the frequency out from the output. Is that procedure the same as testing a ...
2
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2answers
221 views

Alias-free digital nonlinear filter design

@Jazzmaniac has a good answer to the question of how to design an alias-free digital nonlinear time-invariant filter here: https://dsp.stackexchange.com/a/28787/18276 Basically, according to that ...
1
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1answer
74 views

Design 2nd order sections IIR filter for high orders (more than 500)

I am currently building a Reverb that needs to fit on a DSP. I am using the model proposed in this link https://ccrma.stanford.edu/~jos/pasp/Freeverb.html. It involved to create some Feedback Comb-...
2
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2answers
110 views

Detect valleys of a signal

I have raw values that are coming directly from a sensor (on the fly). These values are coming from a wearable sensor installed on a cows neck. When the cow ruminates (meaning that it chews food that ...
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1answer
32 views

Analog equivalent for transition width in Digital radios

I have started off with GNUradio and I am not sure if I understand the transition width parameter available in GNU radio companion filter blocks . As per online documentation transition width of a ...
1
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1answer
36 views

Best digital noise reduction in optical spectra measurement

I am using an optical spectrometer to measure the reflectance of some surfaces; the output is essentially a set of photodiodes' signals: for each photodiode I have a discrete time signal every $T$. ...
0
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1answer
52 views

Band pass filter (digital resonance)

Q1:  Consider a discrete-time band-pass filter (BPF) that uses only two poles as $$ H(z) = G \frac{1}{\big(1-r \, e^{j \omega_0} z^{-1}\big)\big(1-r \, e^{-j \omega_0} z^{-1}\big)} $$ where this ...
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0answers
41 views

Can we sample the signal with any period as long as the Nyquist rate is respected

I have a question regarding the sampling rate. Suppose that we transmitted a signal $m(t)$ then at receiver side, we received the continuous signal $x(t)$, which is time varying and sampled with a ...
2
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1answer
54 views

How can I get the polyphase components of a signal

My question, What is the polyhase components of the signal? What does mean a polyphase component for a signal? Is it true that example : ...
0
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1answer
36 views

designing FIR narrowband pass filter using windowing technique

I m designing FIR narrowband pass filter using windowing technique. is bandpass filter is same as narrowband pass filter? I m not finding any good article on this.
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2answers
67 views

High/Low Pass Filters from Zero

I everyone, I'm a computer scientist but not from the signal processing community. I want to implement a real time digital low pass filter (to begin with) for a sound processing application. I've ...
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0answers
35 views

I get strange BER when I build a code of OFDM in MATLAB with multipath

I have built the code for OFDM using Multi-path channel, but when I plot the BER, I get it abnormal results . (please, any help) .. It's urgent please here is the code I made: ...
1
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1answer
196 views

How to calculate the delay generated by a 2nd order all-pass filter

I'm trying to simulate the dispersion effect of a stiff string in a digital waveguide system, by using second-order all-pass filters. I use a second order all-pass filter with this transfer function:...
2
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0answers
63 views

How can I oversample the received signal by double of sampling time and have two signals

I have a question, how can I make the sampling in Matlab for the received the signal by half of time duration of symbol (Ts), and have two copies of signal (that's the process of fractional sampling ...
1
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1answer
257 views

How to plot the poles and zeros of this Bandpass Filter in MATLAB?

So I'm trying to design a band pass filter in MATLAB (with a center frequency of 10kHz and a sampling frequency of 44kHz). I have calculated the transfer function but I'm not sure how to enter this ...
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0answers
21 views

“Necessary condition for lpf to lpf transformation is that a magnitude response must keep its original values at $\omega=0$ and $\omega=\pi$”?

"Necessary condition for lpf to lpf transformation is that a magnitude response must keep its original values at $\omega=0$ and $\omega=\pi$"? Why is it so? But does that not essentially read that ...
3
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2answers
123 views

What happens when linear phase is critical? [duplicate]

I am studying the topic digital filters. And in the slides it said,"if linear phase is not critical, Infinite Impulse Response (IIR) filters yield a much smaller filter order for a given application". ...
0
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1answer
257 views

Z-domain transfer function to difference equation

So I have a transfer function $ H(Z) = \frac{Y(z)}{X(z)} = \frac{1 + z^{-1}}{2(1-z^{-1})}$. I need to write the difference equation of this transfer function so I can implement the filter in terms of ...
1
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2answers
135 views

what is the frequency selective channel and doubly-selective channel

Could anyone please help me to explain the frequency-selective channel and doubly-selective channel and the difference between them in brief and easy way. I have tried to check on internet, I couldn'...
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0answers
23 views

Why can Generalized eigenvector algorithm for blind equalization be used in SIMO?

I have a question as mentioned the title. I don't know if it's possible to use the Generalized eigenvector algorithm for blind equalization in SISO, but in most of papers I've read, they are talking ...
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2answers
84 views

Analog vs digital

If there are means to process all signals after they have been captured and digitilized, why are we still designing ICs to do analog signal processing? For any giving case, can not we just digitilize ...
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0answers
83 views

Simple Pole-Zero placement bandpass filter doesn't work in MATLAB

I'm trying to make a simple bandpass filter using Pole-Zero Placement method, which has unity gain at frequency w=0.37pi and 3 dB cutoff frequency wc=0.42pi (has a gain of 1/sqrt(2) at that frequency)....
2
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3answers
329 views

What is the filter with the less phase shift?

I have to analyze a dynamic signal but there is too much noise so I applied low pass filter but then there is too much phase shift.So what is the most reactive filter I can apply to my signal ? Best ...
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4answers
268 views

Make a signal that fits another the best possible with a limitation in the 2nd derivative

Consider this step function: The signal that "fits" this should look like the following (in green): The corners are now smooth because the maximum second derivative allowed is not infinite anymore. ...
1
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1answer
45 views

Show that $S[n]= \frac1M \sum e^{\frac{2\pi l n}M}$ for $l = 0$ to $M - 1$

I am learning about image processing. Currently, I am reading about upsampling and down sampling. Here is some background what I am reading: I was wondering if someone show me a proof of how ...
0
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1answer
49 views

is there a way to apply a median filter diagonally?

For example, some studies show how to smooth a spectrogram with a 2D median filter vertically and horizontally (along time or frequency). My question is how would I apply 2D median filter along the ...
0
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1answer
86 views

Understanding and Correcting phase shift in General Linear Phase FIR filters

I have been learning about General Linear Phase Filters and the four standard FIR implementations. I was wondering in the case $\beta = \frac{\pi}{2}$ or $\beta = \frac{3\pi}{2}$ if there was a way ...
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1answer
301 views

What do the filter coefficients in digital filter represent?

Could you help me? How can I understand the function of the filter coefficients practically? In the simple case, it is the impulse response of the LTI system. but how do they work?
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0answers
118 views

Online Frequency and Amplitude Estimation with an Extended Kalman Filter

I'm trying to reproduce some results from a research paper that uses an Extended Kalman filter to estimate the instantaneous frequency and amplitude from accelerometry data. However, I don't ...
0
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1answer
304 views

Why is the second-order-section cascade approach giving “worse” results?

It seems that as a general rule, for digital IIR filters of order more than 2, one is encouraged to use a cascade of second order sections, for reasons connected to the poor stability of high-order ...
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2answers
510 views

What is the largest “safe” order for the digital Butterworth filter of a given signal?

Say I want to apply a zero-phase Butterworth bandpass filter, implemented in MATLAB using butter and filtfilt. So, for an order-$...
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0answers
85 views

Why does noise prevent a (kalman) filter from diverging?

I'm using a filter (not exactly kalman) of the following form to estimate angles by fusing gyroscope with accelerometer and gyro with magnetometer: $(1)\quad \hat{\theta}_k = \hat{\theta}^-_{g,k} + \...
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1answer
494 views

Transition bands and passband gain in digital filter design

In an ideal design, a digital filter has a target gain in the passband and a zero gain (−∞ dB) in the stopband. In a real implementation, a finite transition region between the passband and the ...
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5answers
598 views

Why can't a causal digital filter have an infinitely sharp transition between the passband and the stopband?

In DSP book by Proakis and as well as in this pdf, it is mentioned that practical causal digital filters cannot have an infinitely sharp transition from Pass-band to Stop-band. Why is it so? Can you ...
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1answer
139 views

Biquad Treble Shelf filter gain being weird

Trying to compute Biquad Coefficients for use in a codec chip TLV320AIC3100. Bass Shelf(boost and attenuation) work as expected. Treble Shelf attenuation works as expected but Treble Shelf boost is ...
4
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1answer
329 views

Polyphase Filter decomposition. It is not working

So, I want to do a polyphase implementation of a filter bank (I had problems with it since the beginning and I already asked for help to the community in the past, indeed: Norm MPEG-1 Layer III (Mp3) ...
1
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1answer
68 views

Digital Lowpass Filters, basic questions

Forgive the newbie questions, I'm just starting out and wanted some basic information. I have a signal coming from a digital gyro, most of the useful information is below 90hz but I am seeing ...
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1answer
92 views

First Differentatior and Integrator

For discrete time series $Y$, first differentiation ($D_i=Y_i-Y_{i+1}$) and integrator ($S_i=Y_i+Y_{i+1}$) can be defined as two highpass and lowpass LTI digital filters. Where, transfer function for ...