7.1 Digital Filter Response
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J-DSP Lab 2: the Z-Transform and Frequency Responses
J-DSP Lab 2: The Z-Transform and Frequency Responses Introduction This lab exercise will cover the Z transform and the frequency response of digital filters. The goal of this exercise is to familiarize you with the utility of the Z transform in digital signal processing. The Z transform has a similar role in DSP as the Laplace transform has in circuit analysis: a) It provides intuition in certain cases, e.g., pole location and filter stability, b) It facilitates compact signal representations, e.g., certain deterministic infinite-length sequences can be represented by compact rational z-domain functions, c) It allows us to compute signal outputs in source-system configurations in closed form, e.g., using partial functions to compute transient and steady state responses. d) It associates intuitively with frequency domain representations and the Fourier transform In this lab we use the Filter block of J-DSP to invert the Z transform of various signals. As we have seen in the previous lab, the Filter block in J-DSP can implement a filter transfer function of the following form 10 −i ∑bi z i=0 H (z) = 10 − j 1+ ∑ a j z j=1 This is essentially realized as an I/P-O/P difference equation of the form L M y(n) = ∑∑bi x(n − i) − ai y(n − i) i==01i The transfer function is associated with the impulse response and hence the output can also be written as y(n) = x(n) * h(n) Here, * denotes convolution; x(n) and y(n) are the input signal and output signal respectively. -
Moving Average Filters
CHAPTER 15 Moving Average Filters The moving average is the most common filter in DSP, mainly because it is the easiest digital filter to understand and use. In spite of its simplicity, the moving average filter is optimal for a common task: reducing random noise while retaining a sharp step response. This makes it the premier filter for time domain encoded signals. However, the moving average is the worst filter for frequency domain encoded signals, with little ability to separate one band of frequencies from another. Relatives of the moving average filter include the Gaussian, Blackman, and multiple- pass moving average. These have slightly better performance in the frequency domain, at the expense of increased computation time. Implementation by Convolution As the name implies, the moving average filter operates by averaging a number of points from the input signal to produce each point in the output signal. In equation form, this is written: EQUATION 15-1 Equation of the moving average filter. In M &1 this equation, x[ ] is the input signal, y[ ] is ' 1 % y[i] j x [i j ] the output signal, and M is the number of M j'0 points used in the moving average. This equation only uses points on one side of the output sample being calculated. Where x[ ] is the input signal, y[ ] is the output signal, and M is the number of points in the average. For example, in a 5 point moving average filter, point 80 in the output signal is given by: x [80] % x [81] % x [82] % x [83] % x [84] y [80] ' 5 277 278 The Scientist and Engineer's Guide to Digital Signal Processing As an alternative, the group of points from the input signal can be chosen symmetrically around the output point: x[78] % x[79] % x[80] % x[81] % x[82] y[80] ' 5 This corresponds to changing the summation in Eq. -
Designing Filters Using the Digital Filter Design Toolkit Rahman Jamal, Mike Cerna, John Hanks
NATIONAL Application Note 097 INSTRUMENTS® The Software is the Instrument ® Designing Filters Using the Digital Filter Design Toolkit Rahman Jamal, Mike Cerna, John Hanks Introduction The importance of digital filters is well established. Digital filters, and more generally digital signal processing algorithms, are classified as discrete-time systems. They are commonly implemented on a general purpose computer or on a dedicated digital signal processing (DSP) chip. Due to their well-known advantages, digital filters are often replacing classical analog filters. In this application note, we introduce a new digital filter design and analysis tool implemented in LabVIEW with which developers can graphically design classical IIR and FIR filters, interactively review filter responses, and save filter coefficients. In addition, real-world filter testing can be performed within the digital filter design application using a plug-in data acquisition board. Digital Filter Design Process Digital filters are used in a wide variety of signal processing applications, such as spectrum analysis, digital image processing, and pattern recognition. Digital filters eliminate a number of problems associated with their classical analog counterparts and thus are preferably used in place of analog filters. Digital filters belong to the class of discrete-time LTI (linear time invariant) systems, which are characterized by the properties of causality, recursibility, and stability. They can be characterized in the time domain by their unit-impulse response, and in the transform domain by their transfer function. Obviously, the unit-impulse response sequence of a causal LTI system could be of either finite or infinite duration and this property determines their classification into either finite impulse response (FIR) or infinite impulse response (IIR) system. -
Control Theory
Control theory S. Simrock DESY, Hamburg, Germany Abstract In engineering and mathematics, control theory deals with the behaviour of dynamical systems. The desired output of a system is called the reference. When one or more output variables of a system need to follow a certain ref- erence over time, a controller manipulates the inputs to a system to obtain the desired effect on the output of the system. Rapid advances in digital system technology have radically altered the control design options. It has become routinely practicable to design very complicated digital controllers and to carry out the extensive calculations required for their design. These advances in im- plementation and design capability can be obtained at low cost because of the widespread availability of inexpensive and powerful digital processing plat- forms and high-speed analog IO devices. 1 Introduction The emphasis of this tutorial on control theory is on the design of digital controls to achieve good dy- namic response and small errors while using signals that are sampled in time and quantized in amplitude. Both transform (classical control) and state-space (modern control) methods are described and applied to illustrative examples. The transform methods emphasized are the root-locus method of Evans and fre- quency response. The state-space methods developed are the technique of pole assignment augmented by an estimator (observer) and optimal quadratic-loss control. The optimal control problems use the steady-state constant gain solution. Other topics covered are system identification and non-linear control. System identification is a general term to describe mathematical tools and algorithms that build dynamical models from measured data. -
Finite Impulse Response (FIR) Digital Filters (II) Ideal Impulse Response Design Examples Yogananda Isukapalli
Finite Impulse Response (FIR) Digital Filters (II) Ideal Impulse Response Design Examples Yogananda Isukapalli 1 • FIR Filter Design Problem Given H(z) or H(ejw), find filter coefficients {b0, b1, b2, ….. bN-1} which are equal to {h0, h1, h2, ….hN-1} in the case of FIR filters. 1 z-1 z-1 z-1 z-1 x[n] h0 h1 h2 h3 hN-2 hN-1 1 1 1 1 1 y[n] Consider a general (infinite impulse response) definition: ¥ H (z) = å h[n] z-n n=-¥ 2 From complex variable theory, the inverse transform is: 1 n -1 h[n] = ò H (z)z dz 2pj C Where C is a counterclockwise closed contour in the region of convergence of H(z) and encircling the origin of the z-plane • Evaluating H(z) on the unit circle ( z = ejw ) : ¥ H (e jw ) = åh[n]e- jnw n=-¥ 1 p h[n] = ò H (e jw )e jnwdw where dz = jejw dw 2p -p 3 • Design of an ideal low pass FIR digital filter H(ejw) K -2p -p -wc 0 wc p 2p w Find ideal low pass impulse response {h[n]} 1 p h [n] = H (e jw )e jnwdw LP ò 2p -p 1 wc = Ke jnwdw 2p ò -wc Hence K h [n] = sin(nw ) n = 0, ±1, ±2, …. ±¥ LP np c 4 Let K = 1, wc = p/4, n = 0, ±1, …, ±10 The impulse response coefficients are n = 0, h[n] = 0.25 n = ±4, h[n] = 0 = ±1, = 0.225 = ±5, = -0.043 = ±2, = 0.159 = ±6, = -0.053 = ±3, = 0.075 = ±7, = -0.032 n = ±8, h[n] = 0 = ±9, = 0.025 = ±10, = 0.032 5 Non Causal FIR Impulse Response We can make it causal if we shift hLP[n] by 10 units to the right: K h [n] = sin((n -10)w ) LP (n -10)p c n = 0, 1, 2, …. -
The Scientist and Engineer's Guide to Digital Signal Processing Properties of Convolution
CHAPTER 7 Properties of Convolution A linear system's characteristics are completely specified by the system's impulse response, as governed by the mathematics of convolution. This is the basis of many signal processing techniques. For example: Digital filters are created by designing an appropriate impulse response. Enemy aircraft are detected with radar by analyzing a measured impulse response. Echo suppression in long distance telephone calls is accomplished by creating an impulse response that counteracts the impulse response of the reverberation. The list goes on and on. This chapter expands on the properties and usage of convolution in several areas. First, several common impulse responses are discussed. Second, methods are presented for dealing with cascade and parallel combinations of linear systems. Third, the technique of correlation is introduced. Fourth, a nasty problem with convolution is examined, the computation time can be unacceptably long using conventional algorithms and computers. Common Impulse Responses Delta Function The simplest impulse response is nothing more that a delta function, as shown in Fig. 7-1a. That is, an impulse on the input produces an identical impulse on the output. This means that all signals are passed through the system without change. Convolving any signal with a delta function results in exactly the same signal. Mathematically, this is written: EQUATION 7-1 The delta function is the identity for ( ' convolution. Any signal convolved with x[n] *[n] x[n] a delta function is left unchanged. This property makes the delta function the identity for convolution. This is analogous to zero being the identity for addition (a%0 ' a), and one being the identity for multiplication (a×1 ' a). -
2913 Public Disclosure Authorized
WPS A 13 POLICY RESEARCH WORKING PAPER 2913 Public Disclosure Authorized Financial Development and Dynamic Investment Behavior Public Disclosure Authorized Evidence from Panel Vector Autoregression Inessa Love Lea Zicchino Public Disclosure Authorized The World Bank Public Disclosure Authorized Development Research Group Finance October 2002 POLIcy RESEARCH WORKING PAPER 2913 Abstract Love and Zicchino apply vector autoregression to firm- availability of internal finance) that influence the level of level panel data from 36 countries to study the dynamic investment. The authors find that the impact of the relationship between firms' financial conditions and financial factors on investment, which they interpret as investment. They argue that by using orthogonalized evidence of financing constraints, is significantly larger in impulse-response functions they are able to separate the countries with less developed financial systems. The "fundamental factors" (such as marginal profitability of finding emphasizes the role of financial development in investment) from the "financial factors" (such as improving capital allocation and growth. This paper-a product of Finance, Development Research Group-is part of a larger effort in the group to study access to finance. Copies of the paper are available free from the World Bank, 1818 H Street NW, Washington, DC 20433. Please contact Kari Labrie, room MC3-456, telephone 202-473-1001, fax 202-522-1155, email address [email protected]. Policy Research Working Papers are also posted on the Web at http://econ.worldbank.org. The authors may be contacted at [email protected] or [email protected]. October 2002. (32 pages) The Policy Research Working Paper Series disseminates the findmygs of work mn progress to encouirage the excbange of ideas about development issues. -
ELEG 5173L Digital Signal Processing Ch. 5 Digital Filters
Department of Electrical Engineering University of Arkansas ELEG 5173L Digital Signal Processing Ch. 5 Digital Filters Dr. Jingxian Wu [email protected] 2 OUTLINE • FIR and IIR Filters • Filter Structures • Analog Filters • FIR Filter Design • IIR Filter Design 3 FIR V.S. IIR • LTI discrete-time system – Difference equation in time domain N M y(n) ak y(n k) bk x(n k) k 1 k 0 – Transfer function in z-domain N M k k Y (z) akY (z)z bk X (z)z k 1 k 0 M k bk z Y (z) k 0 H (z) N X (z) k 1 ak z k 1 4 FIR V.S. IIR • Finite impulse response (FIR) – difference equation in the time domain M y(n) bk x(n k) k 0 – Transfer function in the Z-domain M Y (z) k H (z) bk z X (z) k 0 – Impulse response h(n) [b ,b ,,b ] 0 1 M • The impulse response is of finite length finite impulse response 5 FIR V.S. IIR • Infinite impulse response (IIR) – Difference equation in the time domain N M y(n) ak y(n k) bk x(n k) k1 k0 – Transfer function in the z-domain M k bk z Y(z) k0 H (z) N X (z) k 1 ak z k1 – Impulse response can be obtained through inverse-z transform, and it has infinite length 6 FIR V.S. IIR • Example – Find the impulse response of the following system. Is it a FIR or IIR filter? Is it stable? 1 y(n) y(n 2) x(n) 4 7 FIR V.S. -
Time Alignment)
Measurement for Live Sound Welcome! Instructor: Jamie Anderson 2008 – Present: Rational Acoustics LLC Founding Partner & Systweak 1999 – 2008: SIA Software / EAW / LOUD Technologies Product Manager 1997 – 1999: Independent Sound Engineer A-1 Audio, Meyer Sound, Solstice, UltraSound / Promedia 1992 – 1997: Meyer Sound Laboratories SIM & Technical Support Manager 1991 – 1992: USC – Theatre Dept Education MFA: Yale School of Drama BS EE/Physics: Worcester Polytechnic University Instructor: Jamie Anderson Jamie Anderson [email protected] Rational Acoustics LLC Who is Rational Acoustics LLC ? 241 H Church St Jamie @RationalAcoustics.com Putnam, CT 06260 Adam @RationalAcoustics.com (860)928-7828 Calvert @RationalAcoustics.com www.RationalAcoustics.com Karen @RationalAcoustics.com and Barb @RationalAcoustics.com SmaartPIC @RationalAcoustics.com Support @RationalAcoustics.com Training @Rationalacoustics.com Info @RationalAcoustics.com What Are Our Goals for This Session? Understanding how our analyzers work – and how we can use them as a tool • Provide system engineering context (“Key Concepts”) • Basic measurement theory – Platform Agnostic Single Channel vs. Dual Channel Measurements Time Domain vs. Frequency Domain Using an analyzer is about asking questions . your questions Who Are You?" What Are Your Goals Today?" Smaart Basic ground rules • Class is informal - Get comfortable • Ask questions (Win valuable prizes!) • Stay awake • Be Courteous - Don’t distract! TURN THE CELL PHONES OFF NO SURFING / TEXTING / TWEETING PLEASE! Continuing -
Digital Signal Processing Filter Design
2065-27 Advanced Training Course on FPGA Design and VHDL for Hardware Simulation and Synthesis 26 October - 20 November, 2009 Digital Signal Processing Filter Design Massimiliano Nolich DEEI Facolta' di Ingegneria Universita' degli Studi di Trieste via Valerio, 10, 34127 Trieste Italy Filter design 1 Design considerations: a framework |H(f)| 1 C ıp 1 ıp ıs f 0 fp fs Passband Transition Stopband band The design of a digital filter involves five steps: Specification: The characteristics of the filter often have to be specified in the frequency domain. For example, for frequency selective filters (lowpass, highpass, bandpass, etc.) the specification usually involves tolerance limits as shown above. Coefficient calculation: Approximation methods have to be used to calculate the values hŒk for a FIR implementation, or ak, bk for an IIR implementation. Equivalently, this involves finding a filter which has H.z/ satisfying the requirements. Realisation: This involves converting H.z/ into a suitable filter structure. Block or flow diagrams are often used to depict filter structures, and show the computational procedure for implementing the digital filter. 1 Analysis of finite wordlength effects: In practice one should check that the quantisation used in the implementation does not degrade the performance of the filter to a point where it is unusable. Implementation: The filter is implemented in software or hardware. The criteria for selecting the implementation method involve issues such as real-time performance, complexity, processing requirements, and availability of equipment. 2 Finite impulse response (FIR) filter design A FIR filter is characterised by the equations N 1 yŒn D hŒkxŒn k kXD0 N 1 H.z/ D hŒkzk: kXD0 The following are useful properties of FIR filters: They are always stable — the system function contains no poles. -
Mathematical Modeling of Control Systems
OGATA-CH02-013-062hr 7/14/09 1:51 PM Page 13 2 Mathematical Modeling of Control Systems 2–1 INTRODUCTION In studying control systems the reader must be able to model dynamic systems in math- ematical terms and analyze their dynamic characteristics.A mathematical model of a dy- namic system is defined as a set of equations that represents the dynamics of the system accurately, or at least fairly well. Note that a mathematical model is not unique to a given system.A system may be represented in many different ways and, therefore, may have many mathematical models, depending on one’s perspective. The dynamics of many systems, whether they are mechanical, electrical, thermal, economic, biological, and so on, may be described in terms of differential equations. Such differential equations may be obtained by using physical laws governing a partic- ular system—for example, Newton’s laws for mechanical systems and Kirchhoff’s laws for electrical systems. We must always keep in mind that deriving reasonable mathe- matical models is the most important part of the entire analysis of control systems. Throughout this book we assume that the principle of causality applies to the systems considered.This means that the current output of the system (the output at time t=0) depends on the past input (the input for t<0) but does not depend on the future input (the input for t>0). Mathematical Models. Mathematical models may assume many different forms. Depending on the particular system and the particular circumstances, one mathemati- cal model may be better suited than other models. -
Finite Impulse Response (FIR) Digital Filters (I) Types of Linear Phase FIR Filters Yogananda Isukapalli
Finite Impulse Response (FIR) Digital Filters (I) Types of linear phase FIR filters Yogananda Isukapalli 1 Key characteristic features of FIR filters 1. The basic FIR filter is characterized by the following two equations: N -1 N -1 y(n) = å h(k)x(n - k) (1) H (z) = å h(k)z -k (2) k =0 k =0 where h(k), k=0,1,…,N-1, are the impulse response coefficients of the filter, H(z) is the transfer function and N the length of the filter. 2. FIR filters can have an exactly linear phase response. 3. FIR filters are simple to implement with all DSP processors available having architectures that are suited to FIR filtering. 2 Linear phase response • Consider a signal that consists of several frequency components passing through a filter. 1. The phase delay (T p ) of the filter is the amount of time delay each frequency component of the signal suffers in going through the filter. 2. The group delay (T g ) is the average time delay the composite signal suffers at each frequency. 3. Mathematically, T p = -q (w) / w (3) Tg = -dq (w) / dw (4) where q(w) is the phase angle. 3 • A filter is said to have a linear phase response if, J(w) = -aw (5) J(w) = b -aw (6) where a and b are constants. Example) Ideal Lowpass filter k Magnitude 0 wc p w Phase 0 a wc p w 4 k e-jwa passband jw H(e ) = 0 otherwise Magnitude response = |H(ejw)| = k Phase response (q(w)) = < H(ejw) = -wa Follows: y[n] = kx[n-a] : Linear phase implies that the output is a replica of x[n] {LPF} with a time shift of a - p -wu -wl 0 wl wu p w - p -wu -wl 0 wl wu p w 5 Linear phase FIR filters • Symmetric impulse response will yield linear phase FIR filters.