Questions tagged [digital-filters]

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3answers
289 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
865 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 ...
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2answers
329 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 ...
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1answer
418 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-...
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3answers
275 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
92 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 ...
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1answer
86 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 $T$. ...
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1answer
131 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
54 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 ...
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1answer
128 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 : ...
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1answer
235 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
203 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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1answer
682 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:...
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0answers
185 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 ...
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1answer
952 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
25 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 ...
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2answers
461 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". ...
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1answer
7k 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 ...
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2answers
768 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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2answers
295 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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3answers
3k 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
331 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. ...
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1answer
48 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 ...
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1answer
105 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 ...
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1answer
289 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
2k 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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1answer
2k 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
2k 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
264 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
2k 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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6answers
1k 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
194 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 ...
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1answer
939 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) ...
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1answer
92 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
99 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 ...
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1answer
160 views

When simulating a voltage divider using wave digital filters I get inversed polarity outputs

I'm studying wave digital filters for my bachelor's thesis and I'm finding problems in implementing a voltage divider to test my knowledge. Here's the circuit that I'm trying to simulate and the WDF ...
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1answer
475 views

Same z transformed function, but different answers of inverse z transform?

Given a $\mathcal{Z}$ transformed function $E(z)=\frac{1}{z+4}$. I know there are several ways to get the inverse $\mathcal{Z}$ transform of this function : Using partial fraction $$E(z)=\frac{1}{z+...
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2answers
5k views

How does transition width of a low-pass filter affect a message which is well within the pass-band?

I'm taking a communication systems course and we use GNURadio in the lab. In an exam, we had a simple question where we were asked to demodulate a Dual Side Band (with carrier) AM signal in GNURadio. ...
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1answer
252 views

Apply LTI filter to incoming signal stream [duplicate]

Typically, when I want to filter a signal with an LTI filter (in this case a low-pass filter), I just convolve the signal with the filter. However, what do I do if I have a stream of data that I want ...
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1answer
318 views

Confusion in FFT of Resampled Data

I have the following code in Matlab: ...
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2answers
4k views

Transposed Direct Form II VS Direct Form II IIR filters?

I read in "Discrete-Time Signal Processing" Reference, in transposed Direct Form II filters section, that it implements zeros first then poles(unlike Direct form II which implements poles first), ...
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1answer
311 views

Comparison between two Digital Filters

Suppose you have two designs of IIR filters with different specifications which are meant to be implemented on a DSP. On what parameters will one compare the two filters so that we can choose which ...
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3answers
723 views

Digital filter selection for sensor

I am having a LM35 sensor which gives $10\textrm{ mV}$ signal per degree rise of temperature. Now I have one question, which digital filter will be the best for this ? This sensor will be use in very ...
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0answers
81 views

Digital filter: selection

I am going through digital filter design. But I really get confused with so many Digital filters : IIR FIR Moving Average Linear phase filter Allpass filter Comb filter Analog filter: ...
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2answers
272 views

Is ripples in frequency response of a filter ,result of periodic extension of a signal in time domain thereby producing discontinuities?

I am aware that due to gibbs phenomenon, ripples are induced at the vicinity of transition in the frequency response of a filter and ripples occur in frequency domain when there are discontinuities in ...
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2answers
3k views

What is the optimal filter?

I am currently working on a problem on filtering. In fact, let's suppose a signal that is expressed as : $r(t) = z(t) + x_i(t)$, where $z(t)$ is a smooth function and $x_i(t)$ is white Gaussian noise. ...
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5answers
6k views

What Is the Transfer Function of a Moving Average (FIR Filter)?

To make post-processing easier, I export scope measurements as CSV files, which are then post-processed (mostly in Microsoft Excel, which is not the best tool for the job, but it is all I have at my ...
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2answers
3k views

highpass filter design in Matlab

I hope to design 1st order highpass filter from transfer function. In the example of 1st order lowpass filter, I first get the coefficients of numerator and denominator in the variable 'b' and 'a'. In ...
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1answer
89 views

FFT plot not matching the output wave

I had implemented a $4^{\rm th}$ order Chebyshev I Band Pass IIR Digital Filter in Simulink to separate out $148\textrm{ Hz}$ frequency from the sum of the following waves. Input Waves $$ \begin{...
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1answer
2k views

How to design bandpass filter in python when centre frequency is greater than sampling frequency?

I implemented basic bandpass filter using Scipy resource. But how to implement the same when signal is centred at non-zero centre frequency with centre frequency > sampling frequency. I trying to ...

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