Fourier Series, Fourier Transforms and the Delta Function Michael Fowler, Uva
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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 -
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. -
Image Deconvolution Ringing Artifact Detection and Removal Via PSF Frequency Analysis
Image Deconvolution Ringing Artifact Detection and Removal via PSF Frequency Analysis Ali Mosleh1, J.M. Pierre Langlois1, and Paul Green2 1 Ecole´ Polytechnique de Montr´eal, Canada 2 Algolux, Canada {ali.mosleh,pierre.langlois}@polymtl.ca, [email protected] Abstract. We present a new method to detect and remove ringing artifacts produced by the deconvolution process in image deblurring tech- niques. The method takes into account non-invertible frequency com- ponents of the blur kernel used in the deconvolution. Efficient Gabor wavelets are produced for each non-invertible frequency and applied on the deblurred image to generate a set of filter responses that reveal ex- isting ringing artifacts. The set of Gabor filters is then employed in a regularization scheme to remove the corresponding artifacts from the deblurred image. The regularization scheme minimizes the responses of the reconstructed image to these Gabor filters through an alternating algorithm in order to suppress the artifacts. As a result of these steps we are able to significantly enhance the quality of the deblurred images produced by deconvolution algorithms. Our numerical evaluations us- ing a ringing artifact metric indicate the effectiveness of the proposed deringing method. Keywords: deconvolution, image deblurring, point spread function, ringing artifacts, zero-magnitude frequency. 1 Introduction Despite considerable advancements in camera lens stabilizers and shake reduc- tion hardware, blurry images are still often generated due to the camera motion during the exposure time. Hence, effective restoration is required to deblur cap- tured images. Assuming that the imaging system is shift invariant, it can be modeled as b = l ⊕ k + ω, (1) where b ∈ RMN is the blurred captured image, l ∈ RMN is the latent sharp image, k ∈ RMN×MN is the point spread function (PSF) that describes the degree of blurring of the point object captured by the camera, ω ∈ RMN is the additive noise, and ⊕ denotes the 2D convolution operator. -
Gibbs Phenomenon in Engineering Systems
Gibbs Phenomenon and its Applications in Science and Engineering Josué Njock Libii Engineering Department Indiana University-Purdue University Fort Wayne Fort Wayne, Indiana, 46805-1499 [email protected] Abstract Gibbs phenomenon arises in many applications. In this article, the author first discusses a brief history of this phenomenon and several of its applications in science and engineering. Then, using the Fourier series of a square-wave function and computer software in a classroom exercise, he illustrates how Gibbs phenomenon can be used to illustrate to undergraduate students the concept of nonuniform convergence of successive partial sums over the interval from 0 to B. 1. Introduction Gibbs Phenomenon is intimately related to the study of Fourier series. When a periodic function f(x) with a jump discontinuity is represented using a Fourier series, for example, it is observed that calculating values of that function using a truncated series leads to results that oscillate near the discontinuity [12]. As one includes more and more terms into the series, the oscillations persist but they move closer and closer to the discontinuity itself. Indeed, it is found that the series representation yields an overshoot at the jump, a value that is consistently larger in magnitude than that of the actual function at the jump. No matter how many terms one adds to the series, that overshoot does not disappear. Thus, partial sums that approximate f(x) do not approach f(x) uniformly over an interval that contains a point where the function is discontinuous [23]. This behavior, which appears in many practical applications, is known as Gibbs Phenomenon; it is a common example that is used to illustrate how nonuniform convergence can arise [3]. -
FIR Filter Design Techniques
M.Tech. credit seminar report,Electronic Systems Group, EE Dept, IIT Bombay, submitted November2002 FIR Filter Design Techniques Arojit Roychowdhury (Roll No: 02307424) Supervisor: Prof P.C. Pandey Abstract This report deals with some of the techniques used to design FIR filters. In the beginning, the windowing method and the frequency sampling methods are discussed in detail with their merits and demerits. Different optimization techniques involved in FIR filter design are also covered, including Rabiner’s method for FIR filter design. These optimization techniques reduce the error caused by frequency sampling technique at the non-sampled frequency points. A brief discussion of some techniques used by filter design packages like Matlab are also included. Introduction FIR filters are filters having a transfer function of a polynomial in z- and is an all-zero filter in the sense that the zeroes in the z-plane determine the frequency response magnitude characteristic. The z transform of a N-point FIR filter is given by N 1 H(z) = h(n)z n (1) n 0 FIR filters are particularly useful for applications where exact linear phase response is required. The FIR filter is generally implemented in a non-recursive way which guarantees a stable filter. FIR filter design essentially consists of two parts (i) approximation problem (ii) realization problem The approximation stage takes the specification and gives a transfer function through four steps. They are as follows: (i) A desired or ideal response is chosen, usually in the frequency domain. (ii) An allowed class of filters is chosen (e.g. the length N for a FIR filters). -
Analyzing Radar Signals ––
Analyzing Radar Signals –– PRIMER Analyzing Radar Signals PRIMER Contents Measuring Methods ...................................................................... 3 Frequency and Phase Measurements ................................13 Amplitude-vs-Time ..........................................................3 Carrier Frequency ..........................................................13 Phase-vs-Time .................................................................4 Pulse-to-Pulse Measurements ......................................13 Frequency-vs-Time ..........................................................4 Pulse-to-Pulse Carrier Phase Difference .......................13 Digital Modulation ............................................................5 SNR Effects on Pulse-to-Pulse Phase ...........................14 Short Frame (Single Pulse) .............................................5 Phase Noise Effects on Pulse-to-Pulse Phase ..............14 Long Frame (Multiple Pulses) ..........................................5 Pulse Leading Edge Effects on Pulse-to-Pulse Phase ..15 Center Frequency Offset ..................................................6 Overshoot Effects on Pulse-to-Pulse Phase .................15 Pulse-to-Pulse Carrier Frequency Difference ................15 Choosing Measurement Parameters ...................................... 6 Measurement Filter Type ..................................................6 Chirp Measurements ...................................................................16 Pulse Detection Threshold and -
The Scientist and Engineer's Guide to Digital Signal Processing
CHAPTER 14 Introduction to Digital Filters Digital filters are used for two general purposes: (1) separation of signals that have been combined, and (2) restoration of signals that have been distorted in some way. Analog (electronic) filters can be used for these same tasks; however, digital filters can achieve far superior results. The most popular digital filters are described and compared in the next seven chapters. This introductory chapter describes the parameters you want to look for when learning about each of these filters. Filter Basics Digital filters are a very important part of DSP. In fact, their extraordinary performance is one of the key reasons that DSP has become so popular. As mentioned in the introduction, filters have two uses: signal separation and signal restoration. Signal separation is needed when a signal has been contaminated with interference, noise, or other signals. For example, imagine a device for measuring the electrical activity of a baby's heart (EKG) while still in the womb. The raw signal will likely be corrupted by the breathing and heartbeat of the mother. A filter might be used to separate these signals so that they can be individually analyzed. Signal restoration is used when a signal has been distorted in some way. For example, an audio recording made with poor equipment may be filtered to better represent the sound as it actually occurred. Another example is the deblurring of an image acquired with an improperly focused lens, or a shaky camera. These problems can be attacked with either analog or digital filters. Which is better? Analog filters are cheap, fast, and have a large dynamic range in both amplitude and frequency. -
Design of a No-Reference Perceptual Ringing Artifact Metric Master Thesis in Media & Knowledge Engineering
Design of a No-Reference Perceptual Ringing Artifact Metric Master Thesis in Media & Knowledge Engineering Thesis Committee: Prof. Dr. I. Heynderickx Prof. Dr. R.L. Lagendijk Dr. W.P. Brinkman Dr. K.V. Hindriks MSc. H. Liu Author N.C.R. Klomp Email [email protected] Student number 1099957 Supervisor Prof. Dr. I. Heynderickx Group MMI Author N.C.R. Klomp Email [email protected] Keywords: objective metric, ringing artifact, perceptual edge detection, luminance masking, texture masking This report is made as a part of a Master thesis project at the faculty of Computer Science at the Delft University of Technology. All rights reserved. Nothing from this publication may be reproduced without written permission by the copyright holder. No rights whatsoever can be claimed on grounds of this report. Printed in The Netherlands Master Thesis N.C.R. Klomp __________________________________________________________________________________________________________________ Index 1 Introduction ................................................................................................................................................................................................................................................... 5 1.1 Research Question ............................................................................................................................................................................................................................ 7 1.2 Ringing Artifact ................................................................................................................................................................................................................................ -
Fourier Analysis
I V N E R U S E I T H Y T O H F G E R D I N B U School of Physics and Astronomy Fourier Analysis Prof. John A. Peacock [email protected] Session: 2014/15 1 1 Introduction Describing continuous signals as a superposition of waves is one of the most useful concepts in physics, and features in many branches { acoustics, optics, quantum mechanics for example. The most common and useful technique is the Fourier technique, which were invented by Joseph Fourier in the early 19th century. In many cases of relevance in physics, the equations involved are linear: this means that different waves satisfying the equation can be added to generate a new solution, in which the waves evolve independently. This allows solutions to be found for many ordinary differential equations (ODEs) and partial differential equations (PDEs). We will also explore some other techniques for solving common equations in physics, such as Green's functions, and separation of variables, and investigate some aspects of digital sampling of signals. As a reminder of notation, a single wave mode might have the form (x) = a cos(kx + φ): (1.1) Here, a is the wave amplitude; φ is the phase; and k is the wavenumber, where the wavelength is λ = 2π=k. Equally, we might have a function that varies in time: then we would deal with cos !t, where ! is angular frequency and the period is T = 2π=!. In what follows, we will tend to assume that the waves are functions of x, but this is an arbitrary choice: the mathematics will apply equally well to functions of time. -
Signals & Systems Interaction in the Time Domain
Signals & Systems interaction in the Time Domain (Systems will be LTI from now on unless otherwise stated) Course Objectives Specific Course Topics: -Basic test signals and their properties -Basic system examples and their properties -Signals and systems interaction (Time Domain: Impulse Response and convolution, Frequency Domain: Frequency Response) -Applications that exploit signal & systems interaction: system id, audio effects, filtering, AM / FM radio -Signal sampling and reconstruction (time permitting) Signals & Systems interaction in the TD Goals I. Impulse Response (IR) and Convolution Formula -Definition of IR and its use for system identification -Convolution formula and its graphical interpretation II. Properties of systems from IR and convolution -Impulse response as a measure of system memory/stability -Alternative measures of memory/stability: step response III. Applications of convolution -Audio effects: reverberation -Noise removal (i.e. signal filtering or smoothing) The Impulse response When no model is available to describe a system, then we can try to use measured data in order to build one (this process is called “system identification.”) The Impulse Response When no model is available to describe a system, then we can try to use measured data in order to build one (this process is called “system identification.”) The measured data comes from input/output experiments. δ(t) h(t) x(t) h(t) y(t) € € € € € Measuring the Impulse Response (IR) of a system is one of such experiments. By definition, the IR h ( t ) of a system is its response to the unit impulse signal. € The Impulse Response When no model is available to describe a system, then we can try to use measured data in order to build one (this process is called “system identification.”) The measured data comes from input/output experiments. -
Controlled Envelope Single Sideband
David L. Hershberger, W9GR 10373 Pine Flat Way, Nevada City, CA 95959; [email protected] Controlled Envelope Single Sideband Introducing Controlled Envelope SSB; greatly increase your SSB “talk power” by accurately limiting envelope peaks in the SSB modulator. Generate SSB without the big overshoot peaks that make ALC necessary with conventional SSB modulators. Watch your wattmeter read higher than before. Abstract Achieving simultaneous accurate control 1.5 of both amplitude and bandwidth is a difficult problem. When amplitude-limited 1.0 audio is filtered to limit its bandwidth, the filter may overshoot substantially. It loses 0.5 its amplitude limiting ability. If the resulting overshoots are clipped, the amplitude 0 is controlled but the signal’s bandwidth increases because of the clipping distortion. –0.5 The signal loses its bandwidth limiting. Systems exist for correcting audio low-pass Overshooting Squarewave filter overshoot. But single sideband (SSB) –1.0 is a more difficult problem, because of the inevitable Hilbert transform regardless –1.5 of the method of SSB generation. ALC 0.0 0.002 0.004 0.006 0.008 0.01 systems reduce the amplitude of an SSB signal in response to overshooting envelope QX1411-Hershberger01 Time (seconds) peaks. Fast ALC may result in clipping and splatter. Slow ALC will significantly reduce Figure 1 — 100 Hz square wave filtered by 3 kHz elliptic low-pass filter. transmitted power. This paper presents a method for generating SSB without system overshoots. The result is higher transmitted Benjamin Franklin expressed it well: “As necessary to prevent crosstalk between power without audible distortion. we enjoy great advantages from the inven- the left plus right and left minus right tions of others, we should be glad of an subchannels. -
Digital Overshoot: Causes, Prevention, and Cures
PEAK MODULATION CONTROL IN DIGITAL TRANSMISSION SYSTEMS Digital Overshoot: Causes, Prevention, and Cures David L. Hershberger, Greg J. Ogonowski, and Robert A. Orban PORTIONS COPYRIGHT © 1998 CONTINENTAL ELECTRONICS, INC. GRASS VALLEY, CA USA PORTIONS COPYRIGHT © 1998 MODULATION INDEX DIAMOND BAR, CA USA PORTIONS COPYRIGHT © 1998 ORBAN, INC. SAN LEANDRO, CA USA ALL RIGHTS RESERVED. NON-COMMERCIAL DISTRIBUTION ENCOURAGED. ABSTRACT Overmodulation can occur in both analog and digi- tal systems. Just because a system is digital does What are the causes of overshoot, and how is over- not make it immune to overshoot. Like analog sys- shoot different in digital systems? This tutorial an- tems, a digital audio system also contains subsys- swers these questions and will help you avoid the tems that can cause overshoot, like clippers and pitfalls in building a digital audio processing and filters. transmission system for FM broadcasting. DIAGNOSING OVERSHOOT INTRODUCTION How does one find overshoot in a system, and how Overshoot means at least two different things. In a can we tell the difference between overshoot and device designed to control instantaneous levels sloppy peak control? (like a peak limiter) overshoot occurs when the in- stantaneous output amplitude exceeds the nominal Overshoot detection methods are the same for both threshold of peak limiting. In a linear system de- analog and digital systems because both technolo- signed to pass a peak-controlled signal, overshoot gies ultimately produce the same kind of output: an occurs when the instantaneous system output am- analog FM signal. plitude exceeds the instantaneous input amplitude, assuming that the system has unity gain when One way to detect the problem is to use a good measured with a sinewave.