Questions tagged [digital-filters]

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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
102 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
3k 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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2answers
1k views

Different way to separate a particular frequency from a signal

Let's say I have a signal which is composed of 2Hz,10Hz,17Hz,19Hz and 25Hz discrete time sinusoidal waves and I need to allow only 17Hz component to pass. As far as I know, the standard way to ...
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0answers
334 views

How use FIR filter for simulating an integral implemented using the trapezoidal integration?

Suppose I have this equation $$ \phi = \frac{60}{T^5} \int_0^T \left( T^2 - 6T \tau + 6\tau^2\right) y(t-\tau) d\tau - \frac{30 \alpha}{T^5} \int(T-\tau)^2\tau^2 u(t-\tau) d\tau $$ and I want to use ...
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1answer
265 views

Why do digital modulation when the signal entering the channel is analog?

I am learning about digital filters and came across the link Why do digital filters work? In real world communications, I think the input to the channel is in analog after being modulated by digital ...
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1answer
697 views

Allpass Filters - Causal and Stable

So I have been learning about how to test systems for causality and stability but I am confused about the implications on their unit circle representation. Would it be safe to say that a causal and ...
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1answer
78 views

Filtering signal with added noise (frequencies known)

I have been assigned a project to take a given noisy sound file, with added broadband and narrowband noise. The goal is to reach a certain mean-squared error between the filtered signal and the ...
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2answers
2k views

How to remove spike from the data series?

I am getting load cell reading from my adc ADS1231 every 100 ms or say at 10Hz i.e. 10 samples per seconds. I have obtained the readings as shown in the image. As you can see i am getting spike on ...
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2answers
2k views

How to keep phase response of IIR filter?

We know that IIR digital filter has nonlinear phase response in general. How can I linearize the phase response of IIR filter without altering its magnitude response?
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1answer
444 views

Which type of filter gives the smallest filter order?

In a lowpass digital filter if I have passband and stopband ripples, passband and stopband edge frequencies. Which type of filter, IIR or FIR, will give the smaller filter order?
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1answer
181 views

Can error in digital signal be zero after filtering? [closed]

Can error in digital signal be zero after filtering? I have looked up online to know about this but I could not find a solution.
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1answer
261 views

Optimization of Filter Coefficients to Given Word Length

It seems to me that this is a difficult mathematical problem where research is still carried on: Is there an efficient way to find fixed-point coefficients whose zeros are closest to those calculated ...
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2answers
4k views

Digital filters in LabVIEW

I am very new in signal processing and using digital filters. I have to use a low-pass filter to analyze my data in LabVIEW and have a question about it. Any help and advice is appreciated. I am ...
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2answers
15k views

determining type of filter given its pole zero plot

How can I classify a filter given its pole-zero map. For example I've got my zero's located at $\pm j$ and my poles located at $\pm\frac{1}{2}j$.
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1answer
928 views

How to include phase in a sinusoidal Kalman Filter

I start with the equation for sinusoidal motion with an offset and differentiate to get the 2nd order ODE describing the motion of the object. \begin{align} x &= A\sin(\omega t + \phi) + O\\ \dot{...
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1answer
537 views

How to treat noise (in the acceleration) in a Kalman filter when tracking position?

I have an equation of motion for which I am wanting to track the position. The equation looks like this: $$\ddot{x}(t) = -a\dot{x}(t) - bx(t) + F(t)$$ where $F(t)$ is an external Stochastic force ...
4
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1answer
332 views

Problems with implementation of a band-stop filter on an MCU (dsPIC) using fixed-point arithmetic

I'm trying to implement a band-stop (notch) filter on a microcontroller dsPIC33EP64GS506 to filter out a $100\textrm{ Hz}$ component from an input signal. The ...
3
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1answer
960 views

How does parallel structure implementation of IIR speed up the process ? What happens in case of FIR?

I was learning about digital filter structures and there I came to know that parallel implementation of IIR filters speeds up the process and I understand that.But what I am not convinced of is that ...
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1answer
152 views

How to interpret these filter coefficients?

I am trying to make sense of the filter coefficients. The author claims that they are running a 3rd order lowpass Butterworth filter and were willing to give their design. They are not the most ...
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2answers
289 views

Implementation of a filter in a software application

I have to embed a bandpass filter into a software application. All the signal processing will be done in 32 or 64-bit double precision. The filter coefficients for an IIR Butterworth filter are ...
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2answers
4k views

Why analog anti aliasing filter is used before analog to digital converter when there is already a digital filter after ADC?

Normal data acquisition consist of: Analog anti aliasing filter( Sampling frequency : $5\textrm{ kHz}$) ADC - Digital Filter - (Sampling : 200K samples /sec) Digital low pass filters Filters DAC ...
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1answer
100 views

Digital frequency for images

From this Wikipedia link, the fastest fequency for digital frequency is 0.5cycle/sample. It can be rewritten as 1cycle/2sample. What does it mean? Does it mean that at least two pixles are required ...
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1answer
172 views

Filter long audio data using cascading biquad filter

I would like to lowpass filter a long pcm wav file. I designed a 20 order butterworth lowpass filter in Octave and transform the zero pole gain data to a series of second order sections. Then I am ...
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2answers
213 views

Find the transfer function of the difference between IIR and FIR filter

I am using a filter following equations from papers. It is basically the difference between exponential moving average and simple moving average. $$L[n]=\frac{1}{\alpha_L}f[n]+\left(1-\frac{1}{\...
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2answers
491 views

Question about zero-phase filter bank

I found what I believe is a valid solution to the following assignment problem but the answer is bothering me. Let $$H_0(z) = 1+2z^{-1}+3z^{-2}+2z^{-3}+z^{-4}\quad\textrm{and}\quad H_1(z)=H_0(-z).$$...
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1answer
1k views

Problem with scipy.signal.butter for designing filter bank

I need to design a filter bank of band pass filters (code for implementation is shown below). There is a problem in that if my center frequencies fc are much ...
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1answer
2k views

Slow and fast filter

According to my knowledge a filter with low cut off frequency is called slow and with hgih cut off frequency is called fast filter. I am trying to understand more about these two. I know what happens ...
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1answer
41 views

Assess quality of a bandpass filter without human intervention

I wanted to know if there is a way to assess the quality of a bandpass filter created by the Remez exchange algorithm. I am using the Remez algorithm to determine the best bandpass filter at a number ...
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1answer
1k views

Why is there little emphasis on phase distortion?

I have a question regarding the frequency response of digital filters. Specifically the phase aspect of the frequency response. Why do we not seem to care about the nonlinear phase response in ...
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0answers
51 views

Is there any audio DSP without biquad constraint?

I found a codec with audio DSP, ADAU1772. It supports 32 biquads, that is $32\cdot5=160$ parameters. I need to calculate filter coefficients on-line. The result is in transfer function form, i.e., $...
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1answer
3k views

How do you convert angular frequency to $\textrm{Hz}$ from MATLAB's $\tt freqz$ function?

Let's say I have used MATLAB's butter function to generate a 3rd order, low-pass, Buttersworth, digital filter. I have a sample rate of $2000\textrm{ samples/second}$, and I want the cutoff frequency ...
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1answer
1k views

Does a matched filter give a gain in SNR?

I am trying to know if filtering a sequence with a matched filters gives a gain equal to the sequence length, he are the calculus, is it correct ? In my model $s$ is a deterministic complex sequence (...
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1answer
152 views

Errors when converting from digital filter object to transfer function

I am trying to use the designfilt tools in MATLAB to create a bandpass filter, but to be compatible with other code I need the resultant filter in transfer function ...
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1answer
506 views

Is calculus background a must for learn DSP? [closed]

I'm beginner in DSP and in the literature some Authors suggest Analog (Continuous) Signal Processing a prerequisite for learn DSP. But it demands calculus background, that I do not have. Is calculus ...
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1answer
327 views

Weighting function for output error IIR filter design

I noticed, when I try to fit filter coefficients to a given complex transfer function with the output error method, implemented for example in the MATLAB function ...
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1answer
720 views

Multiple biquads vs higher order filtering

I've heard that you can stack biquads on top of each other to get the same (or similar) effect as higher order filters. For instance a biquads is order 2, so putting something through 3 biquads is ...
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1answer
3k views

Butterworth band pass filter

I am trying to build a Butterworth bandpass filter. My intent is to have a pass band between $250\textrm{ Hz}$ and $1000\textrm{ Hz}$. However the plot doesn't look right, as can be seen from the ...
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1answer
143 views

Why can adding delay improve the phase fit in fitting complex transfer functions?

When you fit an IIR filter to a complex transfer function you can use a delay to get a stable filter and improve the fitting results in your phase response. Can anyone explain me the reason for this ...
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5answers
2k views

Why do digital filters work?

So I just started reading on FIR and IIR filters and am amazed by how "simple" the theory seems, so far. But what confuses me is, why does filtering work by creating a weighted sum of previous ...
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4answers
37k views

How many taps does an FIR filter need?

I am looking to design a set of FIR filters to implement a low pass filter. I am also trying to reduce the latency of the signal through the filter so I am wondering what the minimum number of taps I ...
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2answers
25k views

Transfer function of second order notch filter

Can anyone mention the transfer function of second order notch filter to remove the line frequency of 50 Hz, in terms of frequency and sampling rate. Just like for Low pass Butterworth filter as, $$ ...
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1answer
80 views

Digital Filters deal only with samples, right?

I'm new to digital filters. So I'm trying to get things right and I can't find an explicit answer to my question on the internet. Question: Digital filters only accept samples as input? I mean the ...
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2answers
321 views

Question about decimation/downsampling

I am using an on-chip SAR adc where for some reason I can only get accurate signal reconstruction and filtering when I sample at almost 100 times my desired sampling frequency. Specifically, the ...
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0answers
195 views

Combining IIM and MZT transformation

Split from thread: Correcting Impulse Invariance Method Back to this subject. Matt L suggested combining IIM and MZT b-coeffs to get the filter response improved at hihger frequency area. Couldn't ...
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1answer
295 views

Shall a $1^{\rm st}$ order IIR filter include $x[n-1]$ term?

I have seen a $1^{\rm st}$ order IIR filter in the following form: $$y[n] = \alpha x[n] + (1 - \alpha) y[n - 1]$$ Such as: another post in this forum However, the filters generated by Matlab, such ...
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1answer
86 views

Real digital filter property

I am a beginner to study about the filter notion and property Being a real digital filter, (here "real filter" I means that its impulse response is real-valued) this formula is established. But I ...
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1answer
520 views

Correcting Impulse Invariance Method

I'm trying to find out if the correction (Jackson, Nelatury, Mecklenbräuker) could improve the (IIM based) filter response near Nyqvist. Here's my c++ routine (which is used for to calculate various ...
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2answers
379 views

DAC to Analog Filter to ADC

As the order of filter needed for digital information gets higher, it takes longer to process and requires more memory. Maybe analog filtering could be of use here. I know that the words analog and ...
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3answers
680 views

Filter Design - Step Response

I'm looking for a tool\method which allows me to design a filter without involving the frequency response at all. Specifically, I'm trying to design a simple FIR smoothing filter for an emmbedded ...

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