Lecture 7 Discrete Systems

Similar documents
EE 521: Instrumentation and Measurements

Digital Signal Processing Lecture 9 - Design of Digital Filters - FIR

Digital Signal Processing

Discrete-Time Signals and Systems. Frequency Domain Analysis of LTI Systems. The Frequency Response Function. The Frequency Response Function

ELC 4351: Digital Signal Processing

Review of Discrete-Time System

Digital Signal Processing Lecture 8 - Filter Design - IIR

Digital Signal Processing Lecture 10 - Discrete Fourier Transform

Like bilateral Laplace transforms, ROC must be used to determine a unique inverse z-transform.

Discrete-Time Fourier Transform (DTFT)

Lecture 9 Infinite Impulse Response Filters

EE482: Digital Signal Processing Applications

EEL3135: Homework #4

Lecture 16: Filter Design: Impulse Invariance and Bilinear Transform

Filter Analysis and Design

Digital Signal Processing IIR Filter Design via Bilinear Transform

Review of Fundamentals of Digital Signal Processing

ELEG 305: Digital Signal Processing

Discrete-Time Signals and Systems. The z-transform and Its Application. The Direct z-transform. Region of Convergence. Reference: Sections

Chapter 7: IIR Filter Design Techniques

EE 225D LECTURE ON DIGITAL FILTERS. University of California Berkeley

Frequency-Domain C/S of LTI Systems

Z Transform (Part - II)

Stability Condition in Terms of the Pole Locations

UNIVERSITY OF OSLO. Please make sure that your copy of the problem set is complete before you attempt to answer anything.

Discrete Time Systems

-Digital Signal Processing- FIR Filter Design. Lecture May-16

DIGITAL SIGNAL PROCESSING. Chapter 6 IIR Filter Design

VALLIAMMAI ENGINEERING COLLEGE. SRM Nagar, Kattankulathur DEPARTMENT OF INFORMATION TECHNOLOGY. Academic Year

Signals and Systems. Problem Set: The z-transform and DT Fourier Transform

DHANALAKSHMI COLLEGE OF ENGINEERING DEPARTMENT OF ELECTRICAL AND ELECTRONICS ENGINEERING EC2314- DIGITAL SIGNAL PROCESSING UNIT I INTRODUCTION PART A

Review of Fundamentals of Digital Signal Processing

Lecture 14: Windowing

DSP. Chapter-3 : Filter Design. Marc Moonen. Dept. E.E./ESAT-STADIUS, KU Leuven

Lecture 8 - IIR Filters (II)

APPLIED SIGNAL PROCESSING

SIDDHARTH GROUP OF INSTITUTIONS:: PUTTUR Siddharth Nagar, Narayanavanam Road QUESTION BANK (DESCRIPTIVE)

Lecture 4: FT Pairs, Random Signals and z-transform

Digital Signal Processing Lecture 3 - Discrete-Time Systems

ESE 531: Digital Signal Processing

ECE 410 DIGITAL SIGNAL PROCESSING D. Munson University of Illinois Chapter 12

UNIVERSITY OF OSLO. Faculty of mathematics and natural sciences. Forslag til fasit, versjon-01: Problem 1 Signals and systems.

Discrete Time Systems

Digital Signal Processing. Lecture Notes and Exam Questions DRAFT

Convolution. Define a mathematical operation on discrete-time signals called convolution, represented by *. Given two discrete-time signals x 1, x 2,

Discrete-time signals and systems

ESE 531: Digital Signal Processing

LECTURE NOTES DIGITAL SIGNAL PROCESSING III B.TECH II SEMESTER (JNTUK R 13)

/ (2π) X(e jω ) dω. 4. An 8 point sequence is given by x(n) = {2,2,2,2,1,1,1,1}. Compute 8 point DFT of x(n) by

ECSE 512 Digital Signal Processing I Fall 2010 FINAL EXAMINATION

Use: Analysis of systems, simple convolution, shorthand for e jw, stability. Motivation easier to write. Or X(z) = Z {x(n)}

Multimedia Signals and Systems - Audio and Video. Signal, Image, Video Processing Review-Introduction, MP3 and MPEG2

# FIR. [ ] = b k. # [ ]x[ n " k] [ ] = h k. x[ n] = Ae j" e j# ˆ n Complex exponential input. [ ]Ae j" e j ˆ. ˆ )Ae j# e j ˆ. y n. y n.

DSP-CIS. Chapter-4: FIR & IIR Filter Design. Marc Moonen

Digital Control & Digital Filters. Lectures 21 & 22

Transform Analysis of Linear Time-Invariant Systems

ECE503: Digital Signal Processing Lecture 4

Your solutions for time-domain waveforms should all be expressed as real-valued functions.

Discrete-Time David Johns and Ken Martin University of Toronto

IT DIGITAL SIGNAL PROCESSING (2013 regulation) UNIT-1 SIGNALS AND SYSTEMS PART-A

INFINITE-IMPULSE RESPONSE DIGITAL FILTERS Classical analog filters and their conversion to digital filters 4. THE BUTTERWORTH ANALOG FILTER

Filter structures ELEC-E5410

Digital Signal Processing. Midterm 1 Solution

Digital Signal Processing Lecture 4

Very useful for designing and analyzing signal processing systems

INF3440/INF4440. Design of digital filters

Question Bank. UNIT 1 Part-A

Filter Design Problem

SEL4223 Digital Signal Processing. Inverse Z-Transform. Musa Mohd Mokji

z-transform Chapter 6

Discrete Time Fourier Transform

Lecture 3 - Design of Digital Filters

The Z transform (2) 1

Lecture 19 IIR Filters

Signals & Systems Handout #4

There are two main classes of digital lter FIR (Finite impulse response) and IIR (innite impulse reponse).

2. Typical Discrete-Time Systems All-Pass Systems (5.5) 2.2. Minimum-Phase Systems (5.6) 2.3. Generalized Linear-Phase Systems (5.

DEPARTMENT OF EI DIGITAL SIGNAL PROCESSING ASSIGNMENT 1

Lecture 8 - IIR Filters (II)

Z - Transform. It offers the techniques for digital filter design and frequency analysis of digital signals.

Signals and Systems. Spring Room 324, Geology Palace, ,

! Introduction. ! Discrete Time Signals & Systems. ! Z-Transform. ! Inverse Z-Transform. ! Sampling of Continuous Time Signals

VI. Z Transform and DT System Analysis

Lecture 7 - IIR Filters

(Refer Slide Time: 01:28 03:51 min)

ECGR4124 Digital Signal Processing Final Spring 2009

Vel Tech High Tech Dr.Ranagarajan Dr.Sakunthala Engineering College Department of ECE

2.161 Signal Processing: Continuous and Discrete Fall 2008

EC Signals and Systems

Signals and Systems Lecture 8: Z Transform

DIGITAL SIGNAL PROCESSING UNIT III INFINITE IMPULSE RESPONSE DIGITAL FILTERS. 3.6 Design of Digital Filter using Digital to Digital

Design of IIR filters

! z-transform. " Tie up loose ends. " Regions of convergence properties. ! Inverse z-transform. " Inspection. " Partial fraction

UNIT - III PART A. 2. Mention any two techniques for digitizing the transfer function of an analog filter?

ECE503: Digital Signal Processing Lecture 5

Introduction to Digital Signal Processing

X (z) = n= 1. Ã! X (z) x [n] X (z) = Z fx [n]g x [n] = Z 1 fx (z)g. r n x [n] ª e jnω

Lecture 5 - Assembly Programming(II), Intro to Digital Filters

The Z-Transform. Fall 2012, EE123 Digital Signal Processing. Eigen Functions of LTI System. Eigen Functions of LTI System

Discrete-Time Signals and Systems

Transcription:

Lecture 7 Discrete Systems EE 52: Instrumentation and Measurements Lecture Notes Update on November, 29 Aly El-Osery, Electrical Engineering Dept., New Mexico Tech 7. Contents The z-transform 2 Linear Time-Invariant System 2 3 Filter Design 4 3. IIR Filters........................................ 4 3.2 FIR Filters........................................ 5 7.2 The z-transform Definition The z-transform of a discrete-time signal x(n) is defined as the power series X(z) n= x(n)z n () where z is a complex variable, and hence, can be represented as a magnitude and phase. z = re jθ (2) It exists only for those values of z for which this series converges. 7.3 Summation Formula { N a M a n a N+ = a, if a n=m N M +, if a = (3) 7.4 Right-Sided Sequence x(n) = a n u(n) Z X(z) = az (4) This definition is not complete without stating the region-of-convergence (ROC). See Figure 2. 7.5

I m(z).8 a x(n).6.4 Re(z).2... ROC - 2 3 4 5 n... Figure : Region-of-convergence of right-sided sequence left-sided Sequence x(n) = a n Z u( n ) X(z) = az (5) This definition is not complete without stating the region-of-convergence (ROC). See Figure 2. I m(z) n -5-4 -3-2 -... -. a -.2 x(n) Re(z) -.3 -.4 ROC -.5 Figure 2: Region-of-convergence of left-sided sequence 7.6 Poles and Zeros If X(z) is rational function, then X(z) = B(z) A(z) = b + b z + +b M z M a + a z + +a N z N = b a z M+N (z z )(z z 2 ) (z z M ) (z p )(z p 2 ) (z p N ) (6) z = z,z 2,...,z M and p = p, p 2,..., p N are the zeros and poles of X(z), respectively. 7.7 2 Linear Time-Invariant System A relaxed linear time-invariant system with input x(n), an impulse response h(n), and an output y(n) can be expressed in the z-domain as Y(z) = H(z)X(z) (7) where H(z) is the transfer function of the system. 7.8 2

Difference Equation and hence, y(n) = N k= a k y(n k)+ M k= b k x(n k) (8) Y(z) X(z) = H(z) = M k= b kz k + N k= a kz k (9) 7.9 Causality and Stability Causality A system is causal if the ROC is the exterior of a circle. Stability A system is stable if the ROC includes the unit circle. Causal and Stable A system is both causal and stable if the poles are inside the unit circle. 7. Response to Complex Exponential Using the convolution equation y(n) = k= If this system is excited by a complex exponential where ω is an arbitrary frequency, then h(k)x(n k) () x(n) = Ae jωn, < n < () y(n) = h(k)ae jω(n k) k= = A[ k= = AH(ω)e jωn h(k)e jωk ] e jωn (2) Assuming of course that the system is stable so that H(ω) = k= h(k)e jωk (3) exists. 7. Relationship to Fourier Transform If the ROC includes the unit circle, then H(ω) = H(z) z=e jω = k= h(n)e jωn (4) 7.2 3

L p k ω z k Figure 3: Frequency response using the location of poles and zeros Effect of Pole-Zero Location See Figure 3. mag = product o f distances f rom all zeros to L product o f distances f rom all poles to L (5) 7.3 3 Filter Design Filter Specs See Figure 4. Figure 4: Filter Specifications 7.4 3. IIR Filters By Pole-Zero Placement Place zeros at locations you want the magnitude to decrease. Place Poles at locations you want the magnitude to increase. This method has the advantage of being simple and doesn t require a lot of tools. The disadvantage is that it doesn t provide much control in the specs of the filter. 7.5 4

Transformation from Analog Filters Use a transformation to convert analog filters to digital filters. The bilinear transformation is frequently used. s = 2 ( ) z T +z (6) 7.6 Transformation from Analog Filters The bilinear transformation causes a frequency warping described by ΩT ω = 2tan 2 (7) 7.7 Transformation from Analog Filters Some common analog filter includes Butterworth: No ripple in either passband or stopband. Type I Chebyshev: ripple in passband only. Type II Chebyshev: ripple in stopband only. Elliptic: ripple in both passband and stopband. 7.8 Quantization Effects Due to quantization, the location of poles and zeros will be different than designed. This will cause the frequency response to be modified. To minimize the effects of quantization, build your filter as a cascade of second order systems. 7.9 3.2 FIR Filters A system using FIR filters has the output characterized by M y(n) = k= h(k)x(n k) (8) FIR filters with symmetric and antisymmetric impulse responses have linear phase. 7.2 FIR Filter Design Using Windows. Compute the infinite impulse response from the an ideal impulse response. h d (ω) = π H d (ω)e jωn dω (9) 2π π 2. Compute a finite impulse response using a finite length window. h(n) = h d (n)w(n) (2) 3. Windows include rectangular, Hamming, Hanning, Bartlet, Blackman, etc. 4. The wider the window, the sharper the cut-off. 7.2 See Figure 5. 7.22 5

Figure 5: c Proakis and Manolakis, Digital Signal Processing, 4th Edition, Prentice Hall, 27 Figure 6: c Proakis and Manolakis, Digital Signal Processing, 4th Edition, Prentice Hall, 27 See Figure 6. 7.23 FIR Filter Design Using Frequency-Sampling Method. Design the required filter in the Frequency domain. 2. Sample the designed magnitude response. 3. Add the linear phase. 4. Compute the IDFT to obtain h(n). 7.24 See Figure 7..2.2.8.8 Magnitude.6 Magnitude.6.4.4.2.2 2 3 4 5 6 7 2 3 4 5 6 7 k k Figure 7: Frequency sampling approach with not transition points (left) and with transition points (right). 7.25 6

See Figure 8. 2 no transition points transition points -2 Magnitude -4-6 -8 - -2 2 3 4 5 6 7 frequency Figure 8: Magnitude response using frequency sampling approach with not transition points (left) and with transition points (right). 7.26 FIR Filter Design Using Equiripple Method Uses the Alternation Theorem and solves an optimization problem resulting in an equal ripple. Design is accomplished by specifying bandedges and a weighting function. 7.27 7