Lecture #5 Microwave Filters
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Emotion Perception and Recognition: an Exploration of Cultural Differences and Similarities
Emotion Perception and Recognition: An Exploration of Cultural Differences and Similarities Vladimir Kurbalija Department of Mathematics and Informatics, Faculty of Sciences, University of Novi Sad Trg Dositeja Obradovića 4, 21000 Novi Sad, Serbia +381 21 4852877, [email protected] Mirjana Ivanović Department of Mathematics and Informatics, Faculty of Sciences, University of Novi Sad Trg Dositeja Obradovića 4, 21000 Novi Sad, Serbia +381 21 4852877, [email protected] Miloš Radovanović Department of Mathematics and Informatics, Faculty of Sciences, University of Novi Sad Trg Dositeja Obradovića 4, 21000 Novi Sad, Serbia +381 21 4852877, [email protected] Zoltan Geler Department of Media Studies, Faculty of Philosophy, University of Novi Sad dr Zorana Đinđića 2, 21000 Novi Sad, Serbia +381 21 4853918, [email protected] Weihui Dai School of Management, Fudan University Shanghai 200433, China [email protected] Weidong Zhao School of Software, Fudan University Shanghai 200433, China [email protected] Corresponding author: Vladimir Kurbalija, tel. +381 64 1810104 ABSTRACT The electroencephalogram (EEG) is a powerful method for investigation of different cognitive processes. Recently, EEG analysis became very popular and important, with classification of these signals standing out as one of the mostly used methodologies. Emotion recognition is one of the most challenging tasks in EEG analysis since not much is known about the representation of different emotions in EEG signals. In addition, inducing of desired emotion is by itself difficult, since various individuals react differently to external stimuli (audio, video, etc.). In this article, we explore the task of emotion recognition from EEG signals using distance-based time-series classification techniques, involving different individuals exposed to audio stimuli. -
IIR Filterdesign
UNIVERSITY OF OSLO IIR filterdesign Sverre Holm INF34700 Digital signalbehandling DEPARTMENT OF INFORMATICS UNIVERSITY OF OSLO Filterdesign 1. Spesifikasjon • Kjenne anven de lsen • Kjenne designmetoder (hva som er mulig, FIR/IIR) 2. Apppproksimas jon • Fokus her 3. Analyse • Filtre er som rege l spes ifiser t i fre kvens domene t • Også analysere i tid (fase, forsinkelse, ...) 4. Realisering • DSP, FPGA, PC: Matlab, C, Java ... DEPARTMENT OF INFORMATICS 2 UNIVERSITY OF OSLO Sources • The slides about Digital Filter Specifications have been a dap te d from s lides by S. Mitra, 2001 • Butterworth, Chebychev, etc filters are based on Wikipedia • Builds on Oppenheim & Schafer with Buck: Discrete-Time Signal Processing, 1999. DEPARTMENT OF INFORMATICS 3 UNIVERSITY OF OSLO IIR kontra FIR • IIR filtre er mer effektive enn FIR – færre koe ffisi en ter f or samme magnitu de- spesifikasjon • Men bare FIR kan gi eksakt lineær fase – Lineær fase symmetrisk h[n] ⇒ Nullpunkter symmetrisk om |z|=1 – Lineær fase IIR? ⇒ Poler utenfor enhetssirkelen ⇒ ustabilt • IIR kan også bli ustabile pga avrunding i aritme tikken, de t kan ikke FIR DEPARTMENT OF INFORMATICS 4 UNIVERSITY OF OSLO Ideal filters • Lavpass, høypass, båndpass, båndstopp j j HLP(e ) HHP(e ) 1 1 – 0 c c – c 0 c j j HBS(e ) HBP (e ) –1 1 – –c1 – – c2 c2 c1 c2 c2 c1 c1 DEPARTMENT OF INFORMATICS 5 UNIVERSITY OF OSLO Prototype low-pass filter • All filter design methods are specificed for low-pass only • It can be transformed into a high-pass filter • OiOr it can be p lace -
13.2 Analog Elliptic Filter Design
CHAPTER 13 IIR FILTER DESIGN 13.2 Analog Elliptic Filter Design This document carries out design of an elliptic IIR lowpass analog filter. You define the following parameters: fp, the passband edge fs, the stopband edge frequency , the maximum ripple Mathcad then calculates the filter order and finds the zeros, poles, and coefficients for the filter transfer function. Background Elliptic filters, so called because they employ elliptic functions to generate the filter function, are useful because they have equiripple characteristics in the pass and stopband regions. This ensures that the lowest order filter can be used to meet design constraints. For information on how to transform a continuous-time IIR filter to a discrete-time IIR filter, see Section 13.1: Analog/Digital Lowpass Butterworth Filter. For information on the lowest order polynomial (as opposed to elliptic) filters, see Section 14: Chebyshev Polynomials. Mathcad Implementation This document implements an analog elliptic lowpass filter design. The definitions below of elliptic functions are used throughout the document to calculate filter characteristics. Elliptic Function Definitions TOL ≡10−5 Im (ϕ) Re (ϕ) ⌠ 1 ⌠ 1 U(ϕ,k)≡1i⋅ ⎮――――――dy+ ⎮―――――――――dy ‾‾‾‾‾‾‾‾‾‾‾‾2‾ ‾‾‾‾‾‾‾‾‾‾‾‾‾‾‾‾‾‾2‾ ⎮ 2 ⎮ 2 ⌡ 1−k ⋅sin(1j⋅y) ⌡ 1−k ⋅sin(1j⋅ Im(ϕ)+y) 0 0 ϕ≡1i sn(uk, )≡sin(root(U(ϕ,k)−u,ϕ)) π ― 2 ⌠ 1 L(k)≡⎮――――――dy ‾‾‾‾‾‾‾‾‾‾2‾ ⎮ 2 ⌡ 1−k ⋅sin(y) 0 ⎛⎛ ⎞⎞ 2 1− ‾1‾−‾k‾2‾ V(k)≡――――⋅L⎜⎜――――⎟⎟ 1+ ‾1‾−‾k‾2‾ ⎝⎜⎝⎜ 1+ ‾1‾−‾k‾2‾⎠⎟⎠⎟ K(k)≡if(k<.9999,L(k),V(k)) Design Specifications First, the passband edge is normalized to 1 and the maximum passband response is set at one. -
Advanced Electronic Systems Damien Prêle
Advanced Electronic Systems Damien Prêle To cite this version: Damien Prêle. Advanced Electronic Systems . Master. Advanced Electronic Systems, Hanoi, Vietnam. 2016, pp.140. cel-00843641v5 HAL Id: cel-00843641 https://cel.archives-ouvertes.fr/cel-00843641v5 Submitted on 18 Nov 2016 (v5), last revised 26 May 2021 (v8) HAL is a multi-disciplinary open access L’archive ouverte pluridisciplinaire HAL, est archive for the deposit and dissemination of sci- destinée au dépôt et à la diffusion de documents entific research documents, whether they are pub- scientifiques de niveau recherche, publiés ou non, lished or not. The documents may come from émanant des établissements d’enseignement et de teaching and research institutions in France or recherche français ou étrangers, des laboratoires abroad, or from public or private research centers. publics ou privés. advanced electronic systems ST 11.7 - Master SPACE & AERONAUTICS University of Science and Technology of Hanoi Paris Diderot University Lectures, tutorials and labs 2016-2017 Damien PRÊLE [email protected] Contents I Filters 7 1 Filters 9 1.1 Introduction . .9 1.2 Filter parameters . .9 1.2.1 Voltage transfer function . .9 1.2.2 S plane (Laplace domain) . 11 1.2.3 Bode plot (Fourier domain) . 12 1.3 Cascading filter stages . 16 1.3.1 Polynomial equations . 17 1.3.2 Filter Tables . 20 1.3.3 The use of filter tables . 22 1.3.4 Conversion from low-pass filter . 23 1.4 Filter synthesis . 25 1.4.1 Sallen-Key topology . 25 1.5 Amplitude responses . 28 1.5.1 Filter specifications . 28 1.5.2 Amplitude response curves . -
T/HIS 15.0 User Manual
For help and support from Oasys Ltd please contact: UK The Arup Campus Blythe Valley Park Solihull B90 8AE United Kingdom Tel: +44 121 213 3399 Email: [email protected] China Arup 39/F-41/F Huaihai Plaza 1045 Huaihai Road (M) Xuhui District Shanghai 200031 China Tel: +86 21 3118 8875 Email: [email protected] India Arup Ananth Info Park Hi-Tec City Madhapur Phase-II Hyderabad 500 081, Telangana India Tel: +91 40 44369797 / 98 Email: [email protected] Web:www.arup.com/dyna or contact your local Oasys Ltd distributor. LS-DYNA, LS-OPT and LS-PrePost are registered trademarks of Livermore Software Technology Corporation User manual Version 15.0, May 2018 T/HIS 0 Preamble 0.1 Text conventions used in this manual 0.1 1 Introduction 1.1 1.1 Program Limits 1.1 1.2 Running T/HIS 1.2 1.3 Command Line Options 1.4 2 Using Screen Menus 2.1 2.1 Basic screen menu layout 2.1 2.2 Mouse and keyboard usage for screen-menu interface 2.2 2.3 Dialogue input in the screen menu interface 2.4 2.4 Window management in the screen interface 2.4 2.5 Dynamic Viewing (Using the mouse to change views). 2.5 2.6 "Tool Bar" Options 2.6 3 Graphs and Pages 3.1 3.1 Creating Graphs 3.1 3.2 Page Size 3.2 3.3 Page Layouts 3.2 3.3.1 Automatic Page Layout 3.2 3.4 Pages 3.6 3.5 Active Graphs 3.6 4 Global Commands and Pages 4.1 4.1 Page Number 4.1 4.2 PLOT (PL) 4.1 4.3 POINT (PT) 4.2 4.4 CLEAR (CL) 4.2 4.5 ZOOM (ZM) 4.2 4.6 AUTOSCALE (AU) 4.2 4.7 CENTRE (CE) 4.2 4.8 MANUAL 4.2 4.9 STOP 4.2 4.10 TIDY 4.2 4.11 Additional Commands 4.3 5 Main Menu 5.1 5.0 Selecting Curves -
Classic Filters There Are 4 Classic Analogue Filter Types: Butterworth, Chebyshev, Elliptic and Bessel. There Is No Ideal Filter
Classic Filters There are 4 classic analogue filter types: Butterworth, Chebyshev, Elliptic and Bessel. There is no ideal filter; each filter is good in some areas but poor in others. • Butterworth: Flattest pass-band but a poor roll-off rate. • Chebyshev: Some pass-band ripple but a better (steeper) roll-off rate. • Elliptic: Some pass- and stop-band ripple but with the steepest roll-off rate. • Bessel: Worst roll-off rate of all four filters but the best phase response. Filters with a poor phase response will react poorly to a change in signal level. Butterworth The first, and probably best-known filter approximation is the Butterworth or maximally-flat response. It exhibits a nearly flat passband with no ripple. The rolloff is smooth and monotonic, with a low-pass or high- pass rolloff rate of 20 dB/decade (6 dB/octave) for every pole. Thus, a 5th-order Butterworth low-pass filter would have an attenuation rate of 100 dB for every factor of ten increase in frequency beyond the cutoff frequency. It has a reasonably good phase response. Figure 1 Butterworth Filter Chebyshev The Chebyshev response is a mathematical strategy for achieving a faster roll-off by allowing ripple in the frequency response. As the ripple increases (bad), the roll-off becomes sharper (good). The Chebyshev response is an optimal trade-off between these two parameters. Chebyshev filters where the ripple is only allowed in the passband are called type 1 filters. Chebyshev filters that have ripple only in the stopband are called type 2 filters , but are are seldom used. -
Electronic Filters Design Tutorial - 3
Electronic filters design tutorial - 3 High pass, low pass and notch passive filters In the first and second part of this tutorial we visited the band pass filters, with lumped and distributed elements. In this third part we will discuss about low-pass, high-pass and notch filters. The approach will be without mathematics, the goal will be to introduce readers to a physical knowledge of filters. People interested in a mathematical analysis will find in the appendix some books on the topic. Fig.2 ∗ The constant K low-pass filter: it was invented in 1922 by George Campbell and the Running the simulation we can see the response meaning of constant K is the expression: of the filter in fig.3 2 ZL* ZC = K = R ZL and ZC are the impedances of inductors and capacitors in the filter, while R is the terminating impedance. A look at fig.1 will be clarifying. Fig.3 It is clear that the sharpness of the response increase as the order of the filter increase. The ripple near the edge of the cutoff moves from Fig 1 monotonic in 3 rd order to ringing of about 1.7 dB for the 9 th order. This due to the mismatch of the The two filter configurations, at T and π are various sections that are connected to a 50 Ω displayed, all the reactance are 50 Ω and the impedance at the edges of the filter and filter cells are all equal. In practice the two series connected to reactive impedances between cells. -
Performance Analysis of Analog Butterworth Low Pass Filter As Compared to Chebyshev Type-I Filter, Chebyshev Type-II Filter and Elliptical Filter
Circuits and Systems, 2014, 5, 209-216 Published Online September 2014 in SciRes. http://www.scirp.org/journal/cs http://dx.doi.org/10.4236/cs.2014.59023 Performance Analysis of Analog Butterworth Low Pass Filter as Compared to Chebyshev Type-I Filter, Chebyshev Type-II Filter and Elliptical Filter Wazir Muhammad Laghari1, Mohammad Usman Baloch1, Muhammad Abid Mengal1, Syed Jalal Shah2 1Electrical Engineering Department, BUET, Khuzdar, Pakistan 2Computer Systems Engineering Department, BUET, Khuzdar, Pakistan Email: [email protected], [email protected], [email protected], [email protected] Received 29 June 2014; revised 31 July 2014; accepted 11 August 2014 Copyright © 2014 by authors and Scientific Research Publishing Inc. This work is licensed under the Creative Commons Attribution International License (CC BY). http://creativecommons.org/licenses/by/4.0/ Abstract A signal is the entity that carries information. In the field of communication signal is the time varying quantity or functions of time and they are interrelated by a set of different equations, but some times processing of signal is corrupted due to adding some noise in the information signal and the information signal become noisy. It is very important to get the information from cor- rupted signal as we use filters. In this paper, Butterworth filter is designed for the signal analysis and also compared with other filters. It has maximally flat response in the pass band otherwise no ripples in the pass band. To meet the specification, 6th order Butterworth filter was chosen be- cause it is flat in the pass band and has no amount of ripples in the stop band. -
Comparison of a Low-Frequency Butterworth Filter with a Symmetric SE-Filter
Comparison of a low-frequency Butterworth filter with a symmetric SE-filter K S Medvedeva1 1Saratov State University, Astrakhanskaya Street 83, Saratov, Russia, 410012 Abstract. The article compares two filters: a Butterworth filter and an asymmetric SE- filter.The experimental studydetermines their advantages and disadvantages.Also,experiments based on the peak signal-to-noise ratio(PSNR) metricshowvisual evaluation. The results of experimentsshow that a symmetric filter better restores images usinga small set of continuous function parametersthatare distorted by a low-frequency Gaussian filter. 1. Introduction Due to the imperfection of forming and recording systems, images recorded by systemsare distorted (fuzzy) copiesof the original images. The main causes of distortions that resultindegradation of clarity includethe limited resolution of the forming system, refocusing, the presence of a distorting medium (for example, the atmosphere), and movement of the camera on the object being registered. Eliminating or reducing distortion for clarity is the task of image recovery. Automatic control systems, measuring equipment, signal processing systems, and various filters with different characteristics are used to filter signalsin telecommunications. Depending on the frequency band associated with the bandwidth and the suppression band, there are odd, band, high- frequency, and low-frequency filters. Also, all-pass filters have a constant amplitude-frequency response in the required frequency range, and their phase-frequency response is a given frequency function [1]. The simplest way to restoreimage clarity is to process the observed image in the spatial frequency domain with an inverse filter [2].The drawbacks of this filter are the occurrence of edge effects, which take the form of an oscillating hindrance of high power that completely masksthe reconstructed image. -
Introduction to Signals & Systems
A very Brief Introduction to Signals & Systems Outline • Signals & Systems • Continuous and discrete time signals • Properties of Systems • Input- Output relation : Convolution • Frequency domain representation of signals & systems • Analog to digital Conversion • Sampling – Nyquist Sampling Theorem • Basic Filter Theory • Types of filters • Designing practical filters in Labview and Matlab • What is a signal? – A signal is a function defined on the continuum of time values • What is a system ? – a system is a black box that “takes in” one or more input signals and “produces” one or more output signals Continuous time Vs Discrete time Signals • Most of the modern day systems are discrete time systems. E.g., A computer. • A computer can’t directly process a continuous time signal but instead it needs a stream of numbers, which is a discrete time signal. • Discrete time signals are obtain by sampling the continuous time signals • How fast should we sample the signal? Examples • Signals – Unit Step function – Continuous time impulse function – Discrete time • Systems – A simple circuit Basic System Properties • Linearity – System is linear if the principle of superposition holds • Time- Invariance – The system does not change with time Convolution • Linear & Time invariant (LTI) sytems are characterized by their impulse response • Impulse response is the output of the system when the input to the system is an impulse function • For Continuous time signals • For Discrete time signals Frequency domain representation of signals • In most of -
Filters Matthew Spencer Harvey Mudd College E157 – Radio Frequency Circuit Design
Department of Engineering Lecture 09: Filters Matthew Spencer Harvey Mudd College E157 – Radio Frequency Circuit Design 1 1 Department of Engineering Filter Specifications and the Filter Prototype Function Matthew Spencer Harvey Mudd College E157 – Radio Frequency Circuit Design 2 In this video we’re going to start talking about filters by defining a language that we use to describe them. 2 Department of Engineering Filters are Like Extended Matching Networks Vout Vout Vout + + + Vin Vin Vin - - - Absorbs power Absorbs power at one Absorbs power at one ω ω, but can pick Q in a range of ω 푉 푗휔 퐻 푗휔 = 푉 푗휔 ω ω ω 3 Filters are a natural follow on after talking about matching networks because you can think of them as an extension of the same idea. We showed that an L-match lets us absorb energy at one frequency (which is resonance) and reflect it at every other frequency, so we could think of an L match as a type of filter. That’s particularly obvious if we define a transfer function across an L match network from Vin to Vout, which would look like a narrow resonant peak. Adding more components in a pi match allowed us to control the shape of that peak and smear it out over more frequencies. So it stands to reason that by adding even more components to our matching network, we could control whether a signal is passed or reflected over a wider frequency. That turns out to be true, and the type of circuit that achieves this frequency response is referred to as an LC ladder filter. -
Butterworth Filter
Experiment / Circuit Simulation 0 Butterworth Filters All pages have been marked “for evaluation only” to prevent unauthorized use while still allowing customers to evaluate them for suitability in their courses. Your patience in recognizing this necessity is sincerely appreciated. All stamps are removed before your custom manual is paginated and printed. Objectives © 2002, Formulations Media Inc. 1. To use basic filter circuits to design multi-order filter response. 2. To recognize the interchangeability of components to alter design and create high pass or low pass filters. 3. To apply Butterworth polynomial coefficients in the design of filters. 4. To control operational amplifier gain in filter circuits where the gain is often critical to proper design. 5. To cascade filters to create different characteristics than exist with the individual filters. 6. To simulate various multi-order filters using Electronics Workbench™ software running in a Macintosh™ or Windows™ environment, or other similar software. Equipment 8 Resistors: 680 W (2), 3.3 kW (2), 5.6 kW (2), 68 kW (2) 4 Capacitors: 47 nF (4) 2 LM741 Operational Amplifiers Information There are a number of different multi-order filter circuit configurations, one of which is Butterworth. Amulti-order filter is one which has slope rates exceeding ±20 dB/decade. Athird order filter, for example, will have a slope rate of ±60 dB/decade. At the filter break frequency or critical frequency, fc, the filter response may be peaked or attenuated, depending on the circuit damping factor, zeta (z). The effects of z will not be considered in this experiment as z will be set to 0.707, thus providing -3 dB response at the critical frequency.