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Questions tagged [digital-filters]

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1answer
22 views

ROC and impulse response

For the LTI system given below, there are three regions of convergence. $$H(z)=\frac{5-3z^{-1}}{1-\frac53z^{-1}-\frac23z^{-2}}$$ a) Find all possible regions of convergence for this filter. b) For ...
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1answer
33 views

To find the filter Co-efficients of the following equation

Given $y[n]=h_1[n]x[n]+h_2[n]x[n-1]+h_3[n]x[n-2]$ is an LTI system with unity gain at $\omega =0$ and zero gain at $\omega =\pi$. $h[n]\neq0$ .Also given that the system has a linear phase. Compute $...
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1answer
44 views

How do I interpret this block diagram correctly?

I am reading notes ahead of the class and have encountered this particular slide: While I completely agree with the first block diagram, I am at a loss trying to understand how the second block ...
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1answer
20 views

Whats the technical term for a filter that allows a non unity stride when sliding it's window?

So a 1D mean filter of size two with a sliding window defined as W below could be considered as having a stride of one. Wn=(Xn + Xn+1) / 2 Vn=W2n where as if I were to define a 1D 'stepped ...
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0answers
43 views

GNU Radio - Bandwidth and Band Reject Filter

I am currently trying to generate a psk modulated carrier with gnu radio. Attach is an image of my flow graph. As can be seen I am generating a bit stream using the Random Source module and ...
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1answer
122 views

DC Spur in DAC output

I have a design in which a signal from ADC is filtered and then played by DAC. In FPGA the real signal is converted to quadrature using DDS with 70 MHz frequency. After filtering, the baseband signal ...
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1answer
69 views

What is the peak resonance of convolving with a sine FIR filter?

I'm trying to improve my understanding of FIR filters. As an experiment, I've manually created an FIR filter, whose coefficients follow exactly one period of a sine wave. I'm wondering what is the ...
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1answer
55 views

How to perform spectral inversion in the frequency domain to convert a low-pass filter into a high-pass filter?

To convert a linear-phase FIR low-pass filter into a high-pass filter with the same cut-off frequency, we can invert the sign of the low-pass filter's impulse response $h(n)$ and then add one to the ...
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0answers
27 views

Phase lag changes cross spectrum

I have a same noise passing through two different ARMA filter. The autospectrum of first filter is given by S11=G1*G1*S_in where S_in is input spectrum created by set of random numbers. Similarly the ...
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1answer
44 views

DTFT of even and odd samples

Here to find DTFT of $h(2n)$ they have scaled omega, while in RHS to find DTFT $x(2n+1)$ they didn't, why is that?
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1answer
107 views

CIC Filter implementation for Normalized Correlation

This is a follow-on to the original post ... I am trying to implement real-time correlation of two signals. I noticed the CIC Filter discussion here: How to find correlation/cross-correlation of two ...
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1answer
29 views

Inverse Discrete Time Fourier Transform of $1$

$\textrm{DTFT}(\delta[n]) =1$, but $\textrm{IDTFT(1)} = \frac{\sin(\pi n)}{\pi n}$. Why it is not equal to the unit impulse $\delta[n]$?
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1answer
44 views

High order IIR Comb Notch Filter with small Bandwidth - Low atenuation of harmonics

Im filtering an EMG signal, more precisely the 50 Hz frequency and its harmonics. For that, I've constructed a comb notch filter (on filterdesigner in Matlab), with its fundamental frequency being 50 ...
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2answers
68 views

How Butterworth low-pass filter can be applied on a digital signal (i.e. discrete representation of a signal)?

I would like to know how to apply a low pass filter (Butterworth) to a digital signal. So, I have some values of the signal, let's say $S(t)$. Those values are equally spaced in time. I have read the ...
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1answer
33 views

Calculate transfer function of two parallel transfer functions in a feedback loop

I have rational transfer functions: $$H_1(z) = \frac{ b_0 + b_1z^{-1} + b_2z^{-2} }{a_0 + a_1z^{-1} + a_2z^{-2}}$$ $$H_2(z) = \frac{ q_0 + q_1z^{-1} + q_2z^{-2} }{p_0 + p_1z^{-1} + p_2z^{-2}}$$ And ...
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2answers
78 views

How to filter data with given numerator and denominator coefficients of an IIR Filter, in MATLAB Implementation

Coefficients are: b = [9.02201251375918 0.485095309463300 -0.571759308660462 3.63774928924158 4.42592447788520 1.30093840271037 -0.898592924453941 -9.08965268561448 -12.4939309921467 ...
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1answer
33 views

Bringing the signal into the same level [duplicate]

I have a signal, shown in the attachment (blue one). As you can see that the latter part of my signal has a higher dc offset. I want to bring the signal into the same level. I have tried eliminate the ...
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0answers
47 views

Determining window Size for Savitsky-Golay filter (Savgol_filter)

First, I am aware that an ideal optimal window task is really signal/task-specific, i.e. nature of signal/sampling rate/characteristics of interest etc. What I am interested in: "Is there an optimal ...
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2answers
36 views

Downsides of higher order filtfilt's

I understand that with a standard filter, higher order will yield sharper frequency selection at the cost of increased phase delay. However, with so-called zero-phase-distortion filters like that ...
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1answer
43 views

MATLAB fir2 - npt and lap

This might not be the right place to ask this, but I'm hoping someone can explain two of the arguments in the MATLAB fir2 function. It is a function for designing ...
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3answers
107 views

Reverse engineering a digital filter

I have 2 simultaneous signals that both are designed to measure eye movements. They are sampled at 250 Hz. We have 12 subject recordings. For 3 stable periods in each subject, we choose 256 points and ...
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0answers
30 views

What is the threshold to prefer polyphase channelizer over FIR filtering?

A wideband signal need to be processed to extract several narrowband channels from it. There are two straightforward approaches to do so: For each narrowband channel shift frequency and filter it ...
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2answers
214 views

Numerically Stable IIR filter

I am making an IIR Filter and I am running into problems with stability with higher orders. The values go off to infinity as soon as it starts. So far, I've been using MATLAB to generate my filters ...
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1answer
38 views

Adaptive Band pass filter for extracting respiratory waveform from accelerometer data

I am recording data from an accelerometer attached to the chest (1000Hz). I need to extract the respiratory waveform. I tried an adaptive bandpass filter based on a dominant frequency in my signal ...
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1answer
135 views

How to design a digital filter in python that will run over an uC?

I am trying to implement a digital filter over a uC (it doesn't really matter which filter and which micro controller because I'm looking forward to learn how to do it in the future with different ...
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0answers
19 views

Converting the equalizer coefficients from VSA at 2 sample/Symbol to 1.5 sample/Symbol

I was looking for some feedback on conversion of equalizer/filter coefficients at different sampler rate. Problem Statement: We have built a transmitter system whose output is at 1.5 sample/symbol (...
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2answers
54 views

Where can I find information about how to implement IIR filters on micro-controllers?

I am trying to implement an IIR filter on a LPC1769 (I don't know if it's relevant). I already have the H(z) so all the design part is prety much solved. What I am looking for is information about: ...
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1answer
49 views

bibliography about implementation of IIR filters

I need to implement a 6th order IIR filter. The filter I need to implement is the A-weighting. I've simply used the bilinear function of Matlab with an fs of 20Khz. I need to implement this over the ...
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3answers
208 views

How to determine if an IIR filter is causal or non-causal?

It may sound impractical, as only causal system are passively realizable, but what I am facing is a general statement made regarding recursive digital filters: for them to be stable, the poles should ...
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1answer
27 views

How to find out that a wav file is edited or original in Python? [closed]

I want to verify that if a audio (.wav or any format) file is original or edited. How can it possible in python? Here, "edited" means modification of audio file using software. For example, for an ...
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0answers
66 views

Why do additional harmonics arise after the filter bank?

I'm trying to implement direct implementation of MDCT filter bank. So, some additional artefacts have appeared after all of the routines. The spectrum of the initial signal looks like: ...
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1answer
62 views

Can CV or DS be seriously considered DSP roles? [closed]

This is not a technical question but can have a yes/no answer, and a why statement. My understanding about digital signal processing (DSP) is that it allows to take real-world analog signals (1D, 2D, ...
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0answers
30 views

Correctly interpret Butterworth filter frequencies

I am using the butter function from the signal package in R to filter my data. The data is Fourier-transformed (complex numbers) - and I want to filter out specific frequencies (I have a 256Hz ...
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2answers
143 views

Floating point to fixed point in C

I've been trying to make a digital filter. I've used the following transfer function to get my digital coefficients https://gyazo.com/00195ef119f0b2ff93fa5855715abdc9 Which i've then transfered to ...
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1answer
47 views

How to design a a filter and how to determine the poles and zeros on the basis of following informations?

A sinusoidal signal of 300Hz frequency,3V amplitude is contaminated with line frequency (50Hz,1V). Design a digital filter including poles and zeros to remove the interference component. Consider the ...
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1answer
58 views

FIR filter design with time and frequency domain constraints

I deal with the problem of the synthesis of the filter with the requirements of both the frequency and time characteristics. Now there is a filter that meets the requirements. Roughly speaking the ...
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1answer
64 views

Not able to reach minimum phase using Hilbert transform

My problem is pretty simple, I've designed a magnitude response and I would like to find the corresponding minimum phase filter. I'm using the code below and unless there is a bug my eyes don't want ...
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1answer
214 views

How to prove a filter has linear phase response?

I have designed an FIR filter to have linear phase response using odd-symmetry design. The coefficients of this filter are {2,1,3,1,0,-1,-3,-1,-2}. I am now being asked to prove it has linear phase ...
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2answers
94 views

Interpolation and harmonics

A real valued causal sequence $x1[n]$ exists with length of the sequence being $N$. Valid indices of x conform to $0 \le n \le N-1 $ The DFT of x[n] is: $$ X1[k] = \sum_{n=0}^{N-1} x1[n].e^{-j.2.\...
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1answer
144 views

Using Matlab's filter() function for solving difference equation with initial conditions

I am trying to solve a difference equation with initial conditions using Matlab's filter function: $$y[n]=\frac12y[n-1]+x[n],\qquad y[-1]=1\tag{1}$$ For $x[n]=6u[...
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1answer
50 views

What is the type number of a discrete time system given $H(z)$?

Given a continuous time impulse response $h(t)$, if I take the Laplace transform and count the no. of poles at origin, that gives the type number of the system. For e.g., $$H(s) = \frac{2}{s(s+2)}$$ ...
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0answers
35 views

How to find the filter coefficient of matched filter to detect the pulse of a signal?

I would to understand how to find the coefficient of a matched filter. Let's consider we have a signal and the signal is buried in the following noise. we want to find the coefficient of a matched ...
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0answers
28 views

IFFT, negative frequencies and noncausal impulse

When using the IFFT to compute an approximation to the IDTFT, frequency range is 0 to 2$\pi$ versus $-\pi$ to $\pi$. This produces a different impulse sequence(causal I think). How do you calculate ...
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1answer
46 views

Overshoot when using Bessel filter

Here is simple code to illustrate overshoot using bessel filter: ...
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0answers
30 views

What is a simple smoothing filter for signals sampled at non-uniform times?

An answer: A simple smoothing filter for signals that has a DC gain of one, is a simple RC low pass filter. This filter has a exponentially rising step response, $$O(t) = 1 - e^{\frac{-t}{\tau}},$$ ...
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0answers
45 views

persist state of digitalFilter

Suppose I created a digitalFilter using designfilt (in matlab). I am interested in using it to filter windows (I dont have the whole data set offline). When I call filter() with this filter and a ...
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0answers
29 views

Frequency warping while creating filter coefficients

If I have an audio signal at 16kHz with frequencies band-limited to 8kHz. I want to low-pass filter this signal such that fc=4000Hz. Meaning when I reconstruct the signal I do not want any content ...
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1answer
35 views

confused in practical example of shift invariance?

I am reading book of w smith ,"The scientist and engineer guide to digital signal processing" 2 edition I am trying to understand topic of shift invariance from theory given on pg 92 I have attached ...
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1answer
104 views

How do I initialize the state of a digital filter in Direct From II?

Suppose I have a digital filter implemented in Direct Form II. How do I initialize the state of the filter as if the input $x[n]$ had a fixed value $x_0$ for all $n<0$?
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1answer
62 views

How to characterize this whitening filter?

A little bit of background: I am interested in whitening in the least-square estimation. In this sense, consider a univariate signal $y$. Assuming the covariance of $y$ is $\mathbf{A}\sigma^2$. ...