Questions tagged [digital-filters]

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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
39 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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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
941 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
151 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
474 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
325 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
694 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
141 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
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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
36k 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
24k 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
311 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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191 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
280 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
85 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
519 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
377 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
671 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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1answer
740 views

Lowpass filter impulse response in frequency domain

A Linear Time-Invariant system with impulse response $h_1[n]$ is an ideal lowpass filter with cutoff frequency $\omega_c =\pi/2$. The frequency response of the system is $H_1\left(e^{j\omega}\right)$. ...
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2answers
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Exponential weighted moving average time constant

I think it is an easy question but I am stuck. I want to derive that $$ \alpha = -e^{\frac{-T}{\tau}}\,. $$ Can someone provide me with an answer? [EDIT] Sorry guys, this was really a senseless ...
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2answers
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Resampling a digital sound signal

As a non maths-pro, I'm looking for some pointers. I am rewriting the audio core in my emulator to improve accuracy, and am getting a bit stuck on the specifics of the technique. I am trying to ...
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0answers
376 views

Time-domain filtering to match a target time-varying frequency response

I would like to implement a filter in the time domain based on a desired frequency response that changes over time. The initial goal is a bandpass filter with a time-varying center frequency, where ...
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1answer
253 views

dealing with Huge Length (Order) FIR filters

I'm studying a project. My signal is audio 8KHz Mono, 8 bits in Low frequencies 70Hz until 1200Hz. When I perform the calculus with the transition band is very narrow (Transition Frequency) my Order ...
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0answers
1k views

buidling a bandpass filter without using inbuilt functions

I got an assignment which require building a bandpass filter without using any inbuilt functions. I wanted to create a third order Butterworth bandpass filter ...
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2answers
631 views

I do not understand this simple bandstop filter example

I have a periodic signal like this one with period $T=120$: I would like to apply a bandstop filter to the signal, the stop band should go from 90% of $\frac{1}{T}$ to 110% of $\frac{1}{T}$. So I ...
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1answer
570 views

Concepts: Ideal and Non Ideal Channel Resulting in Inter Symbol Interference

I am following the book: Blind equalization and Identification by Zhi Ding and Ye Li. In Chapter 2, the concept of T spaced equalizers is presented. It is mentioned that the output of the channel is $...
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1answer
265 views

Precise Centre frequency of an All-pole digital filter

I'm not an engineer and have essentially taught myself all I know - this present problem is giving me some problems. I have an all-pole filter (a gammatone filter - impulse response is essentially a ...
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1answer
1k views

Minimum phase FIR method

I am trying to make a minimum phase filter (in wxMaxima) according to these steps: first create a "normal" FIR (a simple sinc, wc=0.4, random example, but in the pictures a remez with Octave) ...
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2answers
7k views

Complex low pass filters

I am searching for documentation on the behaviour of complex low pass filters, and their application in telecommunications. Thank you
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1answer
2k views

Difference b/w freqz and fft

Since both fft ,freqz are used for finding the frequency response of digital filters. the frequency response using freqz function is given below and frequency response using fft function is it seems ...
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2answers
1k views

The history of FIR filters

Does anyone know the name of the first DSP (sampled data) practitioner who discovered that real-valued symmetrical-coefficient FIR filters exhibit linear phase in the frequency domain?
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2answers
115 views

Estimating an Image Filter from Examples

I am sure this question must be fairly easy and could have been answered on this site but perhaps I am not finding the right keywords to search for it. I would like to know what are the common ...
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1answer
220 views

For linear IIR digital filter, what happens for negative frequencies?

By negative frequency, I refer to Fourier transform. Often, the frequency response of a digital filter is only displayed for positive frequencies. For a linear IIR digital filter, what happens for ...
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3answers
2k views

ensure stability against coefficient quantization

for a filter: $$ H(z)=\frac{1+0.1z^{-1}}{1+0.1z^{-1}+0.9998z^{-2}} $$ Which precautions could be taken to ensure that the filter does not become unstable because of coefficient quantization? I've ...
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0answers
93 views

State definition in Viterbi Algorithm?

An ISI channel of length 'L+1' $\mathbf{h}=[h_0 \space h_1 \space … \space h_L]$can always be modeled as a finite state machine(FSM) of memory 'L'. For a binary input $x\in\{+1,-1\}$,a state can be ...
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1answer
55 views

When sinusoidal input starts at n=0, why are transient response associated with z-transform poles of digital filter?

In http://www.eng.ucy.ac.cy/cpitris/courses/ECE623/presentations/DSP-LECT-10-11-12.pdf, it says that when sinusoidal input $X(z)$ starts at n=0 (with n<0 having zero input) and the input passes ...
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1answer
5k views

Why are ideal filters not particularly useful/ hard to do?

Why is a LPF ideal filter: a) Not useful? b) Hard to make? I was asked to make a Low pass filter (simulink block) that will take an input of a cutoff frequency, and for a signal with a frequency ...
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1answer
88 views

What kind of filter is this?

Is this filter a BPF? $$\dfrac{z}{z-a}$$ where $a$ is some complex number? If we put a pole somewhere on the unit circle it will emphasise a certain frequency, is that right?
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229 views

FIR Filter Design - from slope and cutoff frequency [duplicate]

I am stuck with the following design requirements for FIR filter: Low-pass -6dB @ 1.6kHz Slope: 24 dB/octave. Ideally I would like to get also FIR coefficients, but any help how to design this ...
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2answers
1k views

FIR Filter Design - input parameters conversion

I am new to DSP and filter design. I have developed a code in C++ to calculate FIR coefficients using Parks-McClellan algorithm. The inputs to calculations are: Filter type (Low-Pass, High-Pass) ...
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1answer
168 views

transient response of simple digitalized RC low-pass filter

In http://en.wikipedia.org/wiki/Bilinear_transform#Example, digital version of simple RC low-pass filter is presented: $$\frac{1 + z^{-1}}{(1 + 2RC / T) + (1 - 2RC / T) z^{-1}}$$ where $T$ is ...
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1answer
50 views

Concepts Surrounding First Two Samples in Difference Equation

I'm attempting to construct a difference equation in matlab for the function u(n) and find the auto-correlation matrix R. I'm following the technique as is defined by my text below: The system I'm ...
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1answer
173 views

Advantage of using nonideal filter in purely digital application [duplicate]

Say I'd like to bandpass filter a signal with corner frequencies 1 kHz and 2 kHz (i.e., passes signals between 1 kHz and 2 kHz and suppresses other frequencies) using software like MATLAB. Intuition ...
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1answer
132 views

Visualising a Z-transformed Transfer Function?

For designing any analog filter and various other outputs of filter we use laplace transform,I can visualise a laplace transform like for ex. s[X(s)] can be ...

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