Chapter 16: Windowed-Sinc Filters
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Generalizing Sampling Theory for Time-Varying Nyquist Rates Using Self-Adjoint Extensions of Symmetric Operators with Deficiency Indices (1,1) in Hilbert Spaces
Generalizing Sampling Theory for Time-Varying Nyquist Rates using Self-Adjoint Extensions of Symmetric Operators with Deficiency Indices (1,1) in Hilbert Spaces by Yufang Hao A thesis presented to the University of Waterloo in fulfillment of the thesis requirement for the degree of Doctor of Philosophy in Applied Mathematics Waterloo, Ontario, Canada, 2011 c Yufang Hao 2011 I hereby declare that I am the sole author of this thesis. This is a true copy of the thesis, including any required final revisions, as accepted by my examiners. I understand that my thesis may be made electronically available to the public. ii Abstract Sampling theory studies the equivalence between continuous and discrete representa- tions of information. This equivalence is ubiquitously used in communication engineering and signal processing. For example, it allows engineers to store continuous signals as discrete data on digital media. The classical sampling theorem, also known as the theorem of Whittaker-Shannon- Kotel'nikov, enables one to perfectly and stably reconstruct continuous signals with a con- stant bandwidth from their discrete samples at a constant Nyquist rate. The Nyquist rate depends on the bandwidth of the signals, namely, the frequency upper bound. Intuitively, a signal's `information density' and ‘effective bandwidth' should vary in time. Adjusting the sampling rate accordingly should improve the sampling efficiency and information storage. While this old idea has been pursued in numerous publications, fundamental problems have remained: How can a reliable concept of time-varying bandwidth been defined? How can samples taken at a time-varying Nyquist rate lead to perfect and stable reconstruction of the continuous signals? This thesis develops a new non-Fourier generalized sampling theory which takes samples only as often as necessary at a time-varying Nyquist rate and maintains the ability to perfectly reconstruct the signals. -
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. -
Fourier Analysis and Sampling Theory
Reading Required: Shirley, Ch. 9 Recommended: Ron Bracewell, The Fourier Transform and Its Applications, McGraw-Hill. Fourier analysis and Don P. Mitchell and Arun N. Netravali, “Reconstruction Filters in Computer Computer sampling theory Graphics ,” Computer Graphics, (Proceedings of SIGGRAPH 88). 22 (4), pp. 221-228, 1988. Brian Curless CSE 557 Fall 2009 1 2 What is an image? Images as functions We can think of an image as a function, f, from R2 to R: f(x,y) gives the intensity of a channel at position (x,y) Realistically, we expect the image only to be defined over a rectangle, with a finite range: • f: [a,b]x[c,d] Æ [0,1] A color image is just three functions pasted together. We can write this as a “vector-valued” function: ⎡⎤rxy(, ) f (,xy )= ⎢⎥ gxy (, ) ⎢⎥ ⎣⎦⎢⎥bxy(, ) We’ll focus in grayscale (scalar-valued) images for now. 3 4 Digital images Motivation: filtering and resizing In computer graphics, we usually create or operate What if we now want to: on digital (discrete)images: smooth an image? Sample the space on a regular grid sharpen an image? Quantize each sample (round to nearest enlarge an image? integer) shrink an image? If our samples are Δ apart, we can write this as: In this lecture, we will explore the mathematical underpinnings of these operations. f [n ,m] = Quantize{ f (n Δ, m Δ) } 5 6 Convolution Convolution in 2D One of the most common methods for filtering a In two dimensions, convolution becomes: function, e.g., for smoothing or sharpening, is called convolution. -
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. -
Filter Effects and Filter Artifacts in the Analysis of Electrophysiological Data
GENERAL COMMENTARY published: 09 July 2012 doi: 10.3389/fpsyg.2012.00233 Filter effects and filter artifacts in the analysis of electrophysiological data Andreas Widmann* and Erich Schröger Institute of Psychology, University of Leipzig, Leipzig, Germany *Correspondence: [email protected] Edited by: Guillaume A. Rousselet, University of Glasgow, UK Reviewed by: Rufin VanRullen, Centre de Recherche Cerveau et Cognition, France Lucas C. Parra, City College of New York, USA A commentary on unity gain at DC (the step response never FILTER EFFECTS VS. FILTER ARTIFACTS returns to one). These artifacts are due to a We also recommend to distinguish between Four conceptual fallacies in mapping the known misconception in FIR filter design in filter effects, that is, the obligatory effects time course of recognition EEGLAB1. The artifacts are further ampli- any filter with equivalent properties – cutoff by VanRullen, R. (2011). Front. Psychol. fied by filtering twice, forward and back- frequency, roll-off, ripple, and attenuation 2:365. doi: 10.3389/fpsyg.2011.00365 ward, to achieve zero-phase. – would have on the data (e.g., smoothing With more appropriate filters the under- of transients as demonstrated by the filter’s Does filtering preclude us from studying estimation of signal onset latency due to step response), and filter artifacts, that is, ERP time-courses? the smoothing effect of low-pass filtering effects which can be minimized by selection by Rousselet, G. A. (2012). Front. Psychol. could be narrowed down to about 4–12 ms of filter type and parameters (e.g., ringing). 3:131. doi: 10.3389/fpsyg.2012.00131 in the simulated dataset (see Figure 1 and Appendix for a simulation), that is, about an CAUSAL FILTERING In a recent review, VanRullen (2011) con- order of magnitude smaller than assumed. -
Control System Design Methods
Christiansen-Sec.19.qxd 06:08:2004 6:43 PM Page 19.1 The Electronics Engineers' Handbook, 5th Edition McGraw-Hill, Section 19, pp. 19.1-19.30, 2005. SECTION 19 CONTROL SYSTEMS Control is used to modify the behavior of a system so it behaves in a specific desirable way over time. For example, we may want the speed of a car on the highway to remain as close as possible to 60 miles per hour in spite of possible hills or adverse wind; or we may want an aircraft to follow a desired altitude, heading, and velocity profile independent of wind gusts; or we may want the temperature and pressure in a reactor vessel in a chemical process plant to be maintained at desired levels. All these are being accomplished today by control methods and the above are examples of what automatic control systems are designed to do, without human intervention. Control is used whenever quantities such as speed, altitude, temperature, or voltage must be made to behave in some desirable way over time. This section provides an introduction to control system design methods. P.A., Z.G. In This Section: CHAPTER 19.1 CONTROL SYSTEM DESIGN 19.3 INTRODUCTION 19.3 Proportional-Integral-Derivative Control 19.3 The Role of Control Theory 19.4 MATHEMATICAL DESCRIPTIONS 19.4 Linear Differential Equations 19.4 State Variable Descriptions 19.5 Transfer Functions 19.7 Frequency Response 19.9 ANALYSIS OF DYNAMICAL BEHAVIOR 19.10 System Response, Modes and Stability 19.10 Response of First and Second Order Systems 19.11 Transient Response Performance Specifications for a Second Order -
Archived: Labview Digital Filter Design Toolkit User Manual
LabVIEWTM Digital Filter Design Toolkit User Manual Digital Filter Design Toolkit User Manual February 2005 371353A-01 Support Worldwide Technical Support and Product Information ni.com National Instruments Corporate Headquarters 11500 North Mopac Expressway Austin, Texas 78759-3504 USA Tel: 512 683 0100 Worldwide Offices Australia 1800 300 800, Austria 43 0 662 45 79 90 0, Belgium 32 0 2 757 00 20, Brazil 55 11 3262 3599, Canada (Calgary) 403 274 9391, Canada (Ottawa) 613 233 5949, Canada (Québec) 450 510 3055, Canada (Toronto) 905 785 0085, Canada (Vancouver) 604 685 7530, China 86 21 6555 7838, Czech Republic 420 224 235 774, Denmark 45 45 76 26 00, Finland 385 0 9 725 725 11, France 33 0 1 48 14 24 24, Germany 49 0 89 741 31 30, India 91 80 51190000, Israel 972 0 3 6393737, Italy 39 02 413091, Japan 81 3 5472 2970, Korea 82 02 3451 3400, Malaysia 603 9131 0918, Mexico 01 800 010 0793, Netherlands 31 0 348 433 466, New Zealand 0800 553 322, Norway 47 0 66 90 76 60, Poland 48 22 3390150, Portugal 351 210 311 210, Russia 7 095 783 68 51, Singapore 65 6226 5886, Slovenia 386 3 425 4200, South Africa 27 0 11 805 8197, Spain 34 91 640 0085, Sweden 46 0 8 587 895 00, Switzerland 41 56 200 51 51, Taiwan 886 2 2528 7227, Thailand 662 992 7519, United Kingdom 44 0 1635 523545 For further support information, refer to the Technical Support and Professional Services appendix. -
Constant K Low Pass Filter
Constant k Low pass Filter Presentation by: S.KARTHIE Assistant Professor/ECE SSN College of Engineering Objective At the end of this section students will be able to understand, • What is a constant k low pass filter section • Characteristic impedance, attenuation and phase constant of low pass filter. • Design equations of low pass filter. Low Pass Ladder Networks • A low-pass network arranged as a ladder or repetitive network. Such a network may be considered as a number of T or ∏ sections in cascade. Low Pass Ladder Networks • a T section may be taken from the ladder by removing ABED, producing the low-pass filter section shown A B L/2 L/2 L/2 L/2 D E Low Pass Ladder Networks • Similarly a ∏ -network is obtained from the ladder network as shown L L L C C C C 2 2 2 2 Constant- K Low Pass Filter • A ladder network is shown in Figure below, the elements being expressed in terms of impedances Z1 and Z 2. Z1 Z1 Z1 Z1 Z1 Z1 Z2 Z2 Z Z 2 2 Z2 Constant- K Low Pass Filter • The network shown below is equivalent one shown in the previous slide , where ( Z1/2) in series with ( Z1/2) equals Z1 and 2 Z2 in parallel with 2 Z2 equals Z2. AB F G Z1/2 Z1/2 Z1/2 Z1/2 Z1 Z1 Z1 2Z Z2 Z2 2Z 2 2Z 2 2Z 2 2Z 2 2 D E J H Constant- K Low Pass Filter • Removing sections ABED and FGJH from figure gives the T & ∏ sections which are terminated in its characteristic impedance Z OT &Z0∏ respectively. -
Digital Signal Processing Tom Davinson
First SPES School on Experimental Techniques with Radioactive Beams INFN LNS Catania – November 2011 Experimental Challenges Lecture 4: Digital Signal Processing Tom Davinson School of Physics & Astronomy N I VE R U S I E T H Y T O H F G E D R I N B U Objectives & Outline Practical introduction to DSP concepts and techniques Emphasis on nuclear physics applications I intend to keep it simple … … even if it’s not … … I don’t intend to teach you VHDL! • Sampling Theorem • Aliasing • Filtering? Shaping? What’s the difference? … and why do we do it? • Digital signal processing • Digital filters semi-gaussian, moving window deconvolution • Hardware • To DSP or not to DSP? • Summary • Further reading Sampling Sampling Periodic measurement of analogue input signal by ADC Sampling Theorem An analogue input signal limited to a bandwidth fBW can be reproduced from its samples with no loss of information if it is regularly sampled at a frequency fs 2fBW The sampling frequency fs= 2fBW is called the Nyquist frequency (rate) Note: in practice the sampling frequency is usually >5x the signal bandwidth Aliasing: the problem Continuous, sinusoidal signal frequency f sampled at frequency fs (fs < f) Aliasing misrepresents the frequency as a lower frequency f < 0.5fs Aliasing: the solution Use low-pass filter to restrict bandwidth of input signal to satisfy Nyquist criterion fs 2fBW Digital Signal Processing D…igi wtahl aSt ignneaxtl? Processing Digital signal processing is the software controlled processing of sequential data derived from a digitised analogue signal. Some of the advantages of digital signal processing are: • functionality possible to implement functions which are difficult, impractical or impossible to achieve using hardware, e.g. -
User Defined Filter Tool 5 Series/6 Series B MSO Option 5-UDFLT/6-UDFLT Application Datasheet
User Defined Filter Tool 5 Series/6 Series B MSO Option 5-UDFLT/6-UDFLT Application Datasheet In the broad sense, any system that processes a signal can be thought Supported filter types: of as a filter. For example, an oscilloscope channel operates as a low • Low-pass pass filter where its 3 dB down point is referred to as its bandwidth. Given a waveform of any shape, a filter can be designed that can • High-pass transform it into defined shape within the context of some basic rules, • Band-pass assumptions, and limitations. • Band-stop • All-pass Digital filters have some significant advantages over analog filters. For example, the tolerance values of analog filter circuit components are • Hilbert usually large, such that high-order filters are difficult or impossible to • Differentiator implement. With digital filters, high-order filters are easily realized. Supported filter responses: 5 Series and 6 Series MSO allow users to apply filters to math waveforms through a MATH Arbitrary function. The User Defined Filter • Butterworth (UDF) tool, Option 5/6-UDFLT takes this functionality a level deeper, • Chebyshev I providing more than MATH arbitrary basic functions and adds flexibility • Chebyshev II to support standard filters and can be used for application centric filter • Elliptical designs. • Gaussian • Bessel-Thomson Select from the various Filter Types General applications of filters Low-pass filters (LPF) are used to remove background and high- frequency noise. High-pass filters (HPF) can be used to remove DC and low-frequency components. Both LPF and HPF are commonly used in high-speed serial and data communication applications. -
And Elliptic Filter for Speech Signal Analysis Design And
Design and Implementation of Butterworth, Chebyshev-I and Elliptic Filter for Speech Signal Analysis Prajoy Podder Md. Mehedi Hasan Md.Rafiqul Islam Department of ECE Department of ECE Department of EEE Khulna University of Khulna University of Khulna University of Engineering & Technology Engineering & Technology Engineering & Technology Khulna-9203, Bangladesh Khulna-9203, Bangladesh Khulna-9203, Bangladesh Mursalin Sayeed Department of EEE Khulna University of Engineering & Technology Khulna-9203, Bangladesh ABSTRACT and IIR filter. Analog electronic filters consisted of resistors, In the field of digital signal processing, the function of a filter capacitors and inductors are normally IIR filters [2]. On the is to remove unwanted parts of the signal such as random other hand, discrete-time filters (usually digital filters) based noise that is also undesirable. To remove noise from the on a tapped delay line that employs no feedback are speech signal transmission or to extract useful parts of the essentially FIR filters. The capacitors (or inductors) in the signal such as the components lying within a certain analog filter have a "memory" and their internal state never frequency range. Filters are broadly used in signal processing completely relaxes following an impulse. But after an impulse and communication systems in applications such as channel response has reached the end of the tapped delay line, the equalization, noise reduction, radar, audio processing, speech system has no further memory of that impulse. As a result, it signal processing, video processing, biomedical signal has returned to its initial state. Its impulse response beyond processing that is noisy ECG, EEG, EMG signal filtering, that point is exactly zero. -
Performance Comparison of Ringing Artifact for the Resized Images Using Gaussian Filter and OR Filter
ISSN: 2278 – 909X International Journal of Advanced Research in Electronics and Communication Engineering (IJARECE) Volume 2, Issue 3, March 2013 Performance Comparison of Ringing Artifact for the Resized images Using Gaussian Filter and OR Filter Jinu Mathew, Shanty Chacko, Neethu Kuriakose developed for image sizing. Simple method for image resizing Abstract — This paper compares the performances of two approaches which are used to reduce the ringing artifact of the digital image. The first method uses an OR filter for the is the downsizing and upsizing of the image. Image identification of blocks with ringing artifact. There are two OR downsizing can be done by truncating zeros from the high filters which have been designed to reduce ripples and overshoot of image. An OR filter is applied to these blocks in frequency side and upsizing can be done by padding so that DCT domain to reduce the ringing artifact. Generating mask ringing artifact near the real edges of the image [1]-[3]. map using OD filter is used to detect overshoot region, and then In this paper, the OR filtering method reduces ringing apply the OR filter which has been designed to reduce artifact caused by image resizing operations. Comparison overshoot. Finally combine the overshoot and ripple reduced between Gaussian filter is also done. The proposed method is images to obtain the ringing artifact reduced image. In second computationally faster and it produces visually finer images method, the weighted averaging of all the pixels within a without blurring the details and edges of the images. Ringing window of 3 × 3 is used to replace each pixel of the image.