Questions tagged [digital-filters]

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1answer
807 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
135 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
250 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
78 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 ...
0
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1answer
146 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 ...
3
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2answers
193 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}{\...
3
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2answers
372 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 $$H_0(z) = 1+2z^{-1}+3z^{-2}+2z^{-3}+z^{-4}\quad\textrm{and}\quad H_1(z)=H_0(-z).$$...
2
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1answer
961 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
1k 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 ...
0
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1answer
37 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 ...
3
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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
50 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 ...
0
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1answer
749 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
141 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 ...
0
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1answer
380 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
302 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 ...
0
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1answer
553 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 ...
3
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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
136 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 ...
22
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4answers
32k 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
21k 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, $$ ...
1
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1answer
77 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 ...
3
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2answers
266 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
185 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
206 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
68 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
474 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
365 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
553 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 ...
2
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1answer
680 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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1answer
4k views

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
1k views

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
352 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 ...
1
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1answer
160 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
562 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 ...
2
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1answer
481 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 $...
1
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1answer
235 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 ...
2
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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) ...
2
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2answers
6k 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
1k 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
106 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
175 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
1k 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
92 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
44 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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