Questions tagged [digital-filters]

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Wave Digital Filter Diode Equation?

I'm trying to understand how diode circuits are implemented in wave digital filters, particularly for clippers. The research papers and other sources I've looked at use the equation $$I(V) = 2I_s \...
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74 views

Filtering a square signal with a median filtering to preserve the edges

If needed, you can find my first post for this problem here. I am trying to clean the following signal: As proposed in the comment, I tried to use this post proposing 2 methods: median filtering and ...
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36 views

Effect of time duration in bandpass filtering

I am new in DSP, but I am trying to process EEG data to decompose into the bands, theta, alpha, and others. The raw data is processed: Remove the DC offset by subtracting the average value from the ...
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197 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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301 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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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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58 views

Problems discretizing a notch filter - Chebyshev Type I

I'm trying to implement and discretize a notch filter (Chebyshev type I) these specifications: Ws = 2pi[2000 2500]; Wp = 2pi[1000 7000]; Rp = 0.5; Rs = 40; But when I try to use discretization methods,...
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14 views

Calculate ScaleVlues of IIR Filter manually from sosMatrix of MATLAB

while using "design" in matlab to design filters, at the output we can see a ScaleValues matrix that is used to compensate for overflow problem. my question is that how this matrix is ...
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1answer
133 views

How to implement a series of second-order, digital state-variable filters in MATLAB?

How to implement a series of second-order, digital state-variable filters, using the high-pass output from each section to effect the low-pass to high-pass transformation in MATLAB? Hello, I am trying ...
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91 views

The size of an FIR filter for PDM-PCM conversion

I have been trying to understand how the conversion from PDM to PCM happens so that I could interface with a microphone. I have been doing some research online and I found that the most common way to ...
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56 views

Help on understanding Gardner's original paper for QPSK symbol timing recovery?

I am fairly new to DSP and RF receivers in general, and I was recommended to read Gardner's original paper for symbol timing error recovery for QPSK signals. I have a question regarding an equation ...
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171 views

Early-Late Symbol Synchronizer on Non-Triangular Pulses

I've been studying the basics of digital communications using the free book at desktopsdr.com. One of the examples is a 4-QAM demodulator which where symbol timing recovery is done with some Early-...
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341 views

Choosing Right Type of Filter for time series data

I am working on a temperature time-series data which is very noisy. I am trying to measure true low and ...
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355 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
61 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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641 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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27 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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76 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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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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72 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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195 views

Band Edge Filters

I am interested in using Band Edge Filters to do carrier frequency recovery but this is kind of secondary to my question which is: How do you build an FIR filter which isn't symmetric about the ...
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1answer
505 views

How can I get the continuous-time transfer function coefficients (or poles and zeros) from the corresponding discrete-time TF and vice versa?

Let's say I have a continuous time transfer function which I know its numerator coefficients $(B^c = [b^c_m, ..., b^c_1, b^c_0])$ and denominator coefficients $(A^c = [a^c_m, ..., a^c_1, a^c_0])$. ...
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93 views

Digital or analogue filtering?

I'm trying to self learn the art of signal processing whilst moving through my third year pure maths degree. Sorry if my terminology is incorrect however I hope I am understandable! I am looking at ...
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324 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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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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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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167 views

Basic “filtering” in digital communications

I am completely stuck on solving some problems for my digital communications class. Junior level undergrad class, so fairly basic. I have been searching around all day for related articles etc but can'...
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31 views

How to calculate analog cutoff frequency to design a digital filter?

$\Omega_c$=cutoff analog frequency In this example, what is going to be value of $\Omega_c$? Some teachers in youtube are teaching this method-: Is this method correct? If I use this formula, then ...
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7 views

Processing accelerometer and gyroscope data

I'm working on an instrumented headform to measure linear and rotation acceleration of the on impact, for which I'm using ADXL377 accelerometer and ICM-20649 angular rate sensor. What kind of ...
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12 views

Determine the lattice coefficients corresponding to FIR filter with system function

And sketch lattice structure $H(z)=1+ \frac {13} {24} Z^{-1}+ \frac {5} {8} Z^{-2}+ \frac {1} {8} Z^{-3}$ This is my engineering assignment question from Digital Signal Processing subject on the topic ...
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23 views

Can digital filters be arranged as a hierarchy?

I am developing a software framework for image processing. So, I need to arrange the digital filters as a hierarchy. Digital Filters The above link lists the following digital filters: Mean Filter -...
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32 views

How to find the minimum no of bits required to express the filter coefficients in fixed point format from Histogram plot?

I have designed a filter in MATLAB and now I have to convert the filter coefficient in fixed point format from floating point format. For that my professor asked me to find histogram first and then ...
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53 views

Hardware implementation of multi rate FIR filter

I have designed a multi rate fir filter in MATLAB and I want to implement it in verilog.This is the block diagram. Total number of taps are 391. Is there any way to automatically generate verilog ...
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40 views

understanding FIR SRRC filter with memory implementation in C

I translated the Matlab QPSK receiver/transmitter into real-time C code that I have running on a PlutoSDR platform. QPSK receiver/transmitter I wrote the square root raised cosine filter for the TX ...
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1answer
39 views

Digtial FIR Impulse response & transfer function

I am currently working the figure through below. as it is an FIR Filter i have worked out using convolution that the output is 4,2,4,6,0,0. i am trying to obtain the 'z' domain transfer function of ...
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48 views

How to break a second-order filter into two first-order filter

Let's assume the transfer function of a continuous-domain filter consists of two poles and one zero: $H(s) = \frac{k_c (s-\omega_{z_1})}{(s-\omega_{p_1})(s-\omega_{p_2})}$. Let's consider we do the bi-...
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60 views

The optimal estimator using FIR Wiener filter

The received signal $y(n)$ is defined as $$y(n) = h(n)*x(n) +v(n)$$ where $h(n)$ is unknown impulse response and $v(n)$ is a white Gaussian noise. The value of $x(n)$ for $n=1, ..., 32$ are given. The ...
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40 views

How to design IIR digital filters

Practical Infinite-Impulse-Response (IIR) filters are usually based upon analogue equivalents (Butterworth, Chebyshev, etc.), using a transformation known as the bilinear transformation which maps the ...
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93 views

FIR filter design using Kaiser window - passband ripple and stopband attenuation

I have been given the following parameters : lower cutoff frequency = $\pi/3$ lower cutoff freq = $2\pi/3$ beta = 5.6533 M = 51 Now I have to get magnitude and phase response of the filter using ...
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24 views

interpreting the post filter features in a sensor data

I am new to signal processing and I am working on Filtring signals(using basic analog filters, FIR filters, and IRR filters), for Human activity recognition, using sensor data, I Extract the features ...
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15 views

How to make 2 finger rake receiver to counter intersymbol inferference?

I have to transmit Walsh sequence (spreading codes) modulated with 8 psk with a channel impaired by intersymbol interferences, fast fading . Suppose i have a stream of data for instance 1 0 1 1, 0 0 0 ...
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59 views

Richardson-Lucy iterative deconvolution is producing erroneous deconvoluted images

I am working on producing deconvoluted image from degraded image where the process of degradation is known to be linear and space invariant. I have used Richardson-Lucy iterative deconvolution method ...
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1answer
63 views

Is the sampling period arbitrary for a bilinear transform, and why?

I have been given the specifications for a digital highpass filter (stopband, passband, stopband attenuation and maximum passband ripple). I am expected to design a prototype lowpass filter in the ...
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21 views

Filter using frequency and magnitude

I'm used to filtering using FIR filters with a cutoff and dB attenuations etc. The issue I have is that I have pulses in my signal which contain similar frequencies as the noise. I'd like to totally ...
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33 views

Interpolationg phase and magnitudes, Transformation function

I am trying to filter signal x(n), n = 150. I made a filter with few frequency points on the x axis , [-11., -9., -3., -2., -1., 1., 2., 3., 9., 11.]) ...
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63 views

cascade realisation from direct realisation

I am trying to convert filter realisation from Direct form 1 to cascade using function provided from Digital signal processing by Proakis. I designed lowpass IIR filter and extracted numerator and ...
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54 views

Remove noise from distance calculated from accelerometer data

I got distance calculated from accelerometer raw datas using double integral accelerometer.I attached the image below.But it shows noise with it.Please help me to remove this noise.
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48 views

IIR filter relaxation time computation

Let's define the relaxation time of a filter as the delay needed for the impulse response of a filter to reach and remain under an amplitude threshold. Is there a closed-form solution to the ...
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1answer
54 views

How to design a filter that can filter out noise accurately, after setting the parameters of the filter using standard signal?

I meet a problem with designing a filter. I have two different instruments that could measure the same AC signal (usually ~200hz, always <1kHz), A and B. A can carry out signal measurement during ...
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1answer
56 views

How can I find the transform function, amplitude and frequency response of a digital filter in python?

I have applied a laplace filter mask to an image and now I want to find the amplitude and freqency response of a laplacian filter: [[1,1,1], [1,-8,1], [1,1,1] ]. I know I need to first find the ...