This Lecture Covers the Following Topics

Total Page:16

File Type:pdf, Size:1020Kb

This Lecture Covers the Following Topics ECE4330 Lecture 24 Discrete-Time Fourier Transform (DTFT): Application to LTID System Analysis and Digital Filter Design Prof. Mohamad Hassoun This Lecture covers the following topics Background: Connections with continuous-time systems The Sampling of 푓(푡) to obtain the sequence 푓[푘]: A brief review Fourier transform of discrete-time signals (DTFT) Connections between the DTFT and the Z-transform Discrete-time linear system analysis using the DTFT Response of a discrete-time system to an everlasting sinusoid Digital filter design based on analog filter approximations: o Impulse-invariance method: Numerical integration example o Impulse-invariance (direct) poles mapping method o The bilinear transformation Frequency domain-based digital filter design Matlab’s digital filter design tool: fdatool Nonlinear LSE-based digital filter design Numerical methods as recursive filters: Error analysis o Trapezoidal integration rule o Simpson’s 1/3 integration rule Appendix: General summary Background: Connections with Continuous-Time Systems In previous lectures we have introduced the Fourier transform that allowed us to analytically represent a continuous-time (CT) aperiodic signal in terms of the radian frequency, 휔, as +∞ 푓(푡) ↔ 퐹(휔) = ∫ 푓(푡)푒−푗휔푡푑푡 −∞ We will refer to this transformation as the CTFT. For example, we have derived earlier the CTFT pair, 1 푓(푡) = 푒−푎푡푢(푡) ↔ 퐹(휔) = (푎 > 1) 푎 + 푗휔 Tables of Fourier transform pairs and Fourier transform properties are available. The Fourier transform was also applied to a stable, continuous- time linear system which allowed us to transform the system’s linear differential equation representation (with zero initial state) into a rational function representation (frequency response function) of the form, 퐾(푗휔 − 푧 )(푗휔 − 푧 ) … (푗휔 − 푧 ) 퐻(휔) = 1 2 푚 (푗휔 − 훾1)(푗휔 − 훾2) … (푗휔 − 훾푛) where the 푧푖 and the 훾푖 are the zeros and poles of 퐻(휔), respectively. This rational function is the Fourier transform of the system’s unit-impulse response, ℎ(푡) ↔ 퐻(휔). The frequency response 퐻(휔) can also be obtained (for stable linear systems) by simply making the substituting 푠 = 푗휔 in the expression for the system’s transfer function, 퐻(푠). Given a causal input signal 푓(푡)푢(푡), with Fourier transform 퐹(휔), the zero-state frequency-domain response, 푌푧푠(휔), of a stable linear system with frequency response 퐻(휔) is given by, 푌푧푠(휔) = 퐻(휔)퐹(휔) with a corresponding time-domain zero-state response given by −1 푦푧푠(푡) = 퐹 {퐻(휔)퐹(휔)} If the input signal 푓(푡) is the everlasting sinusoid 푓(푡) = 퐶cos(휔0푡 + 휃) [푓(푡) = 퐶0 is a special case where 휔0 = 0] or, more generally, periodic with Fourier series ∞ 푓(푡) = 퐶0 + ∑ 퐶푛 cos(푛휔0푡 + 휃푛) 푛=1 then, the zero-state response becomes the steady-state response, 푦푧푠(푡) = 푦푠푠(푡). The steady-state response can be obtained using the formula ∞ 푦푠푠(푡) = 퐻(0)퐶0 + ∑|퐻(푛휔0)|퐶푛 cos[푛휔0푡 + ∠퐻(푛휔0) + 휃푛] 푛=1 This lecture answers the following questions in connection with a discrete- time signal 푓[푘] and a stable LTDI system with impulse response ℎ[푘] (or, equivalently, a stable system with a known LTI difference equation): Is there a discrete-time Fourier transform (DTFT) for 푓[푘]? How can one obtain the DTFT for 푓[푘]? How can the inverse transformation be obtained? What are the properties of the DTFT? How can one obtain the system’s frequency response function? How can one obtain the zero-state and steady-state responses? How can we design a discrete-time filter that is equivalent to a continuous-time (analog) filter? The following is a summary of the three transformation methods that we had utilized in this course along with the DTFT that will be covered in the rest of this lectures. Refer to the appendix for an extended summary. The Sampling of 푓(푡) to Obtain the Sequence 푓[푘]: A Brief Review The discrete-time signal (or sequence) 푓[푘] will be assumed to have been obtained by uniformly sampling a continuous-time signal 푓(푡) at uniform intervals of duration 푇푠, or with a corresponding sampling frequency 푓푠 = 1 . We will assume that the sampling frequency satisfies the (equivalent) 푇푠 inequalities 1 푓 > 2퐵, 휔 > 4휋퐵 and 푇 < 푠 푠 푠 2퐵 where 퐵 is the effective bandwidth (in Hz) of the signal 푓(푡). (Refer to Lectures 18 and 23 for bandwidth and sampling rate, respectively.) For example, for the bandlimited signal 푓(푡) = cos (휔0푡) the sampled signal is obtained by a (discretizing) variable transformation, 푡 → 푇푠푘, to ( ) ( )| ( ) ( ) obtain 푓 푇푠푘 = 푓 푡 푡=푇푠푘 = cos 휔0푇푠푘 = cos Ω0푘 , where we define Ω0 = 푇푠휔0 and 푘 = ⋯ , −2, −1, 0, 1, 2, … . We will refer to Ω0 as the discrete-time angular frequency, measured in rad/sample. Since for a pure 휔0 휋 sinusoid the bandwidth is 퐵 = , we must employ 푇푠 < . Now, we may 2휋 휔0 generate the sequence 푓[푘] by setting its values equal to 푓(푇푠푘). Example. Find a proper sampled signal 푓[푘] for the continuous-time signal 휋 푓(푡) = 3 cos (1500푡 + ) 3 1500 1 This sinusoid has a bandwidth 퐵 = . Any 푇 satisfying 푇 < = 2휋 푠 푠 2퐵 휋 ≅ 0.0021 would allow for adequate reconstruction (by the Sampling 1500 Theorem). So, we can set 푇푠 = 0.001 sec (푓푠 = 1000Hz) to arrive at the sampled signal 휋 휋 푓[푘] = 3 cos (1500(0.001)푘 + ) = 3 cos (1.5푘 + ) 3 3 Notice how the sampling rate 푇푠 = 0.001 leads to a scaled the frequency, 휔0 휔0 = 1500 rad/sec → Ω0 = 푇푠휔0 = = 1.5 rad/sample 푓푠 휋 Your turn: Plot 푓[푘] = 3 cos (1.5푘 + ) and compare it to 푓(푡). 3 We should avoid making 푇푠 unnecessarily small because that would lead to a reduction in speed when digital processors are used to process the sampled signal. It should be emphasized that once the samples are taken from 푓(푡) and stored in memory (in digital/quantized form), the time scale is lost. The discrete-time signal is just a sequence of numbers, and these numbers carry no information about the sampling period, 푇푠. Therefore, the process that generates the discrete-time sequence, 푓[푘], from the continuous-time signal, 푓(푡), performs two operations: 1. It samples the signal 푓(푡) every 푇푠 seconds and assigns the value 푓(푇푠푘) to the 푘th sample of the sequence, 푓[푘]. 2. It scales (normalizes) the time axis by 푘 = 푡/푇푠, so that the distance between successive samples is always unity. Thus, it changes the time axis from 푡 to its normalized integer time, 푘. In Lecture 23, we showed that the CT Fourier transform of the sampled signal 푓(푇푠푘) is, ∞ ∞ 1 휔 퐺(휔) = 퐹(휔) ∗ [휔 ∑ 훿(휔 − 푘휔 )] = 푠 ∑ 퐹(휔 − 푘휔 ) 2휋 푠 푠 2휋 푠 푘=−∞ 푘=−∞ In the above expression, 퐹(휔) is the Fourier transform of 푓(푡) and 휔푠 = 2휋 is the radian sampling frequency. The expression can be rewritten as, 푇푠 ∞ 퐺(푓) = 푓푠 ∑ 퐹(푓 − 푘푓푠) 푘=−∞ Therefore, the process that generates the discrete-time sequence 푓[푘] from the continuous-time signal 푓(푡) can be viewed, in the frequency domain, as performing the following operations: 1. It scales the spectrum of 푓(푡) by 푓푠. 2. It duplicates the scaled spectrum at all integer multiples of the sampling frequency, 푘푓푠. Fourier Transform of Discrete-Time Signals (DTFT) In analogy to using the Fourier transform to arrive at a frequency-domain characterization of a continuous signal, 푓(푡), we define the discrete-time Fourier transform (DTFT) of the sampled signal (sequence) 푓[푘] as the infinite sum ∞ 퐹(Ω) = ∑ 푓[푘]푒−푗Ω푘 푘=−∞ 휔 where Ω = 푇푠휔 = is the discrete-time angular frequency (measured in 푓푠 rad/sample). As with other transforms, we will describe this relationship between the discrete-time signal and its DTFT as a transform pair, 푓[푘] ↔ 퐹(Ω) It is interesting to note how one can obtain the DTFT of a sampled signal from the (continuous-time) Fourier transform of 푓(푡), as shown next. As we did in the previous lecture, the sampled signal 푓[푘] is obtained by multiplying 푓(푡) by the impulse train 훿푠(푡), ∞ 푓(푡)훿푠(푡) = 푓(푡) ∑ 훿(푡 − 푇푠푘) 푘=−∞ Applying the Fourier transformation to the above sampled signal we get +∞ +∞ ∞ −푗휔푡 −푗휔푡 ∫ 푓(푡)훿푠(푡)푒 푑푡 = ∫ [푓(푡) ∑ 훿(푡 − 푇푠푘)] 푒 푑푡 −∞ −∞ 푘=−∞ ∞ +∞ ∞ −푗휔푡 −푗휔푇푠푘 = ∑ ∫ 푓(푡)푒 훿(푡 − 푇푠푘)푑푡 = ∑ 푓(푇푠푘)푒 푘=−∞ −∞ 푘=−∞ Upon defining Ω = 푇푠휔 and replacing 푓(푇푠푘) by its sample values 푓[푘], in the above expression we arrive at the DTFT, ∞ 퐹(Ω) = ∑ 푓[푘]푒−푗Ω푘 푘=−∞ Example. Find the DTFT of the causal signal 푓[푘] = 훾푘푢[푘] and plot its spectra for 훾 = 0.8. ∞ ∞ ∞ 푘 퐹(Ω) = ∑ 훾푘푢[푘]푒−푗Ω푘 = ∑ 훾푘푒−푗Ω푘 = ∑(훾푒−푗Ω) 푘=−∞ 푘=0 푘=0 is the sum of a geometric series with common ratio 훾푒−푗Ω. The sum converges to 1 푒푗Ω 퐹(Ω) = = 1 − 훾푒−푗Ω 푒푗Ω − 훾 if |훾푒−푗Ω| < 1, but since |푒−푗Ω| = 1, then |훾| < 1 is required for the transform to exist. Therefore, we arrive at our first DTFT pair, 1 푒푗Ω 훾푘푢[푘] ↔ = |훾| < 1 1 − 훾푒−푗Ω 푒푗Ω − 훾 Setting 훾 = 0.8 in the DTFT we obtain 1 퐹(Ω) = 1 − 0.8푒−푗Ω or, equivalently, by employing Euler’s identity we may write 1 퐹(Ω) = 1 − 0.8[cos(Ω) − 푗sin(Ω)] The magnitude and angle responses are then (your turn: derive them) 1 |퐹(Ω)| = √1 + 0.64 − 1.6 cos(Ω) 0.8 sin(Ω) angle(퐹(Ω)) = − tan−1 ( ) 1 − 0.8 cos(Ω) The following two plots show the spectra of 퐹(Ω).
Recommended publications
  • Dsp Notes Prepared
    DSP NOTES PREPARED BY Ch.Ganapathy Reddy Professor & HOD, ECE Shaikpet, Hyderabad-08 Ch Ganapathy Reddy, Prof and HOD, ECE, GNITS id:[email protected],9052344333 1 DIGITAL SIGNAL PROCESSING A signal is defined as any physical quantity that varies with time, space or another independent variable. A system is defined as a physical device that performs an operation on a signal. System is characterized by the type of operation that performs on the signal. Such operations are referred to as signal processing. Advantages of DSP 1. A digital programmable system allows flexibility in reconfiguring the digital signal processing operations by changing the program. In analog redesign of hardware is required. 2. In digital accuracy depends on word length, floating Vs fixed point arithmetic etc. In analog depends on components. 3. Can be stored on disk. 4. It is very difficult to perform precise mathematical operations on signals in analog form but these operations can be routinely implemented on a digital computer using software. 5. Cheaper to implement. 6. Small size. 7. Several filters need several boards in analog, whereas in digital same DSP processor is used for many filters. Disadvantages of DSP 1. When analog signal is changing very fast, it is difficult to convert digital form .(beyond 100KHz range) 2. w=1/2 Sampling rate. 3. Finite word length problems. 4. When the signal is weak, within a few tenths of millivolts, we cannot amplify the signal after it is digitized. 5. DSP hardware is more expensive than general purpose microprocessors & micro controllers. Ch Ganapathy Reddy, Prof and HOD, ECE, GNITS id:[email protected],9052344333 2 6.
    [Show full text]
  • Noise Will Be Noise: Or Phase Optimized Recursive Filters for Interference Suppression, Signal Differentiation and State Estimation (Extended Version) Hugh L
    Available online at https://arxiv.org/ 1 Noise will be noise: Or phase optimized recursive filters for interference suppression, signal differentiation and state estimation (extended version) Hugh L. Kennedy Abstract— The increased temporal and spectral resolution of oversampled systems allows many sensor-signal analysis tasks to be performed (e.g. detection, classification and tracking) using a filterbank of low-pass digital differentiators. Such filters are readily designed via flatness constraints on the derivatives of the complex frequency response at dc, pi and at the centre frequencies of narrowband interferers, i.e. using maximally-flat (MaxFlat) designs. Infinite-impulse-response (IIR) filters are ideal in embedded online systems with high data-rates because computational complexity is independent of their (fading) ‘memory’. A novel procedure for the design of MaxFlat IIR filterbanks with improved passband phase linearity is presented in this paper, as a possible alternative to Kalman and Wiener filters in a class of derivative-state estimation problems with uncertain signal models. Butterworth poles are used for configurable bandwidth and guaranteed stability. Flatness constraints of arbitrary order are derived for temporal derivatives of arbitrary order and a prescribed group delay. As longer lags (in samples) are readily accommodated in oversampled systems, an expression for the optimal group delay that minimizes the white-noise gain (i.e. the error variance of the derivative estimate at steady state) is derived. Filter zeros are optimally placed for the required passband phase response and the cancellation of narrowband interferers in the stopband, by solving a linear system of equations. Low complexity filterbank realizations are discussed then their behaviour is analysed in a Teager-Kaiser operator to detect pulsed signals and in a state observer to track manoeuvring targets in simulated scenarios.
    [Show full text]
  • Overview Textbook Prerequisite Course Outline
    ESE 337 Digital Signal Processing: Theory Fall 2018 Instructor: Yue Zhao Time and Location: Tuesday, Thursday 7:00pm - 8:20pm, Javits Lecture Center 101 Contact: Email: [email protected], Office: 261 Light Engineering Office Hours: Tuesday, Thursday 1:30pm - 3:00pm, or by appointment Teaching Assistants, and Office Hours: • Jiaming Li ([email protected]): THU 12:30pm - 2:00pm, or by appointment, Location: 208 Light Engineering (Changes of hours, if any, will be updated on Blackboard.) Overview Digital Signal Processing (DSP) lies at the heart of modern information technology in many fields including digital communications, audio/image/video compression, speech recognition, medical imaging, sensing for health, touch screens, space exploration, etc. This class covers the basic principles of digital signal processing and digital filtering. Skills for analyzing and synthesizing algorithms and systems that process discrete time signals will be developed. 3 credits. Textbook • A.V. Oppenheim and R.W. Schafer, Discrete Time Signal Processing, Prentice Hall, Third Edition, 2009 Prerequisite • ESE 305, Deterministic Signals and Systems Course Outline Week 1 DT signals, DT systems and properties, LTI systems Week 2 Convolution, properties of LTI systems Week 3 Examples of LTI systems, eigen functions, frequency response of DT systems, DTFT 1 ESE 337 Syllabus Week 4 Convergence and properties of DTFT Week 5 Theorems of DTFT, useful DTFT pairs, examples, Z-transform Week 6 Examples of Z-transform, properties of ROC Week 7 Inverse Z-transform,
    [Show full text]
  • Understanding Digital Signal Processing
    Understanding Digital Signal Processing Richard G. Lyons PRENTICE HALL PTR PRENTICE HALL Professional Technical Reference Upper Saddle River, New Jersey 07458 www.photr,com Contents Preface xi 1 DISCRETE SEQUENCES AND SYSTEMS 1 1.1 Discrete Sequences and Their Notation 2 1.2 Signal Amplitude, Magnitude, Power 8 1.3 Signal Processing Operational Symbols 9 1.4 Introduction to Discrete Linear Time-Invariant Systems 12 1.5 Discrete Linear Systems 12 1.6 Time-Invariant Systems 17 1.7 The Commutative Property of Linear Time-Invariant Systems 18 1.8 Analyzing Linear Time-Invariant Systems 19 2 PERIODIC SAMPLING 21 2.1 Aliasing: Signal Ambiquity in the Frequency Domain 21 2.2 Sampling Low-Pass Signals 26 2.3 Sampling Bandpass Signals 30 2.4 Spectral Inversion in Bandpass Sampling 39 3 THE DISCRETE FOURIER TRANSFORM 45 3.1 Understanding the DFT Equation 46 3.2 DFT Symmetry 58 v vi Contents 3.3 DFT Linearity 60 3.4 DFT Magnitudes 61 3.5 DFT Frequency Axis 62 3.6 DFT Shifting Theorem 63 3.7 Inverse DFT 65 3.8 DFT Leakage 66 3.9 Windows 74 3.10 DFT Scalloping Loss 82 3.11 DFT Resolution, Zero Padding, and Frequency-Domain Sampling 83 3.12 DFT Processing Gain 88 3.13 The DFT of Rectangular Functions 91 3.14 The DFT Frequency Response to a Complex Input 112 3.15 The DFT Frequency Response to a Real Cosine Input 116 3.16 The DFT Single-Bin Frequency Response to a Real Cosine Input 117 3.17 Interpreting the DFT 120 4 THE FAST FOURIER TRANSFORM 125 4.1 Relationship of the FFT to the DFT 126 4.2 Hints an Using FFTs in Practice 127 4.3 FFT Software Programs
    [Show full text]
  • ESE 531: Digital Signal Processing Today IIR Filter Design Impulse
    Today ESE 531: Digital Signal Processing ! IIR Filter Design " Impulse Invariance " Bilinear Transformation Lec 18: March 30, 2017 ! Transformation of DT Filters IIR Filters and Adaptive Filters ! Adaptive Filters ! LMS Algorithm Penn ESE 531 Spring 2017 – Khanna Penn ESE 531 Spring 2017 - Khanna 2 IIR Filter Design Impulse Invariance ! Transform continuous-time filter into a discrete- ! Want to implement continuous-time system in time filter meeting specs discrete-time " Pick suitable transformation from s (Laplace variable) to z (or t to n) " Pick suitable analog Hc(s) allowing specs to be met, transform to H(z) ! We’ve seen this before… impulse invariance Penn ESE 531 Spring 2017 - Khanna 3 Penn ESE 531 Spring 2017 - Khanna 4 Impulse Invariance Impulse Invariance ! With Hc(jΩ) bandlimited, choose ! With Hc(jΩ) bandlimited, choose j ω j ω H(e ω ) = H ( j ), ω < π H(e ω ) = H ( j ), ω < π c T c T ! With the further requirement that T be chosen such ! With the further requirement that T be chosen such that that Hc ( jΩ) = 0, Ω ≥ π / T Hc ( jΩ) = 0, Ω ≥ π / T h[n] = Thc (nT ) Penn ESE 531 Spring 2017 - Khanna 5 Penn ESE 531 Spring 2017 - Khanna 6 1 IIR by Impulse Invariance Example jω ! If Hc(jω)≈0 for |ωd| > π/T, no aliasing and H(e ) = H(jω/T), ω<π jω ! To get a particular H(e ), find corresponding Hc and Td for which above is true (within specs) ! Note: Td is not for aliasing control, used for frequency scaling. Penn ESE 531 Spring 2017 - Khanna 7 Penn ESE 531 Spring 2017 - Khanna 8 Example Example 1 eat ←⎯L→ s − a Penn ESE 531 Spring
    [Show full text]
  • Chap 4 Sampling of Continuous-Time Signals Introduction
    Chap 4 Sampling of Continuous-Time Signals Introduction 4.1 Periodic Sampling 4.2 Frequency-Domain Representation of Sampling 4.3 Reconstruction of a Bandlimited Signal from its Samples 4.4 Discrete-Time Processing of Continuous-Time Signals 4.5 Continuous-Time Processing of Discrete-Time Signals 4.6 Changing the Sampling Rate Using Discrete-Time Processing 4.7 Multirate Signal Processing 4.8 Digital Processing of Analog Signals 4.9 Oversampling and Noise Shaping in A/D and D/A Conversion 2018/9/18 DSP 2 Periodic Sampling Sequence of samples x[n] is obtained from a continuous-time signal xc(t) : x[n] = xc(nT), - infinity < n < infinity T : sampling period fs = 1/T : sampling frequency, samples/second C/D Continuous-to-discrete-time Xc(t) converter X[n] = Xc(nT) T In a practical setting, the operation of sampling is often implemented by an analog-to-digital (A/D) converter which can be approximated to the ideal C/D converter. The sampling operation is generally not invertible. The inherent ambiguity in sampling is of primary concern in signal processing. 2018/9/18 DSP 3 Sampling with a Periodic Impulse Train s(t) C/D Converter Conversion from impulse train x[n] = x (nT) x (t) X C C xS(t)to discrete-time sequence xC(t) xC(t) xS(t) 2 Sampling Rates xS(t) -4T 0 2T -4T 0 2T x[n] 2 Output Sequences x[n] -4 0 2 -4 0 2 2018/9/18 DSP 4 Frequency-domain representation of sampling The conversion of xc(t) to xs(t) through modulating signal s(t) which is a periodic impulse train s(t) (t nT ) n xs (t) xc (t)s(t) xc (t) (t nT ) n by shifting property of the impulse xs (t) xc (nT)(t nT) n The Fourier transform of a periodic impulse train is a periodic impulse train.
    [Show full text]
  • Digital Signal Processing Lecture 8
    Lecture 8 Recap Introduction CT->DT Digital Signal Processing Impulse Invariance Lecture 8 - Filter Design - IIR Bilinear Trans. Example Electrical Engineering and Computer Science University of Tennessee, Knoxville Overview Lecture 8 1 Recap Recap Introduction CT->DT 2 Introduction Impulse Invariance 3 Bilinear Trans. CT->DT Example 4 Impulse Invariance 5 Bilinear Trans. 6 Example Roadmap Lecture 8 Introduction Discrete-time signals and systems - LTI systems Unit sample response h[n]: uniquely characterizes an Recap LTI system Introduction Linear constant-coefficient difference equation CT->DT Frequency response: H(ej!) Impulse Invariance Complex exponential being eigenfunction of an LTI j! j! Bilinear Trans. system: y[n] = H(e )x[n] and H(e ) as eigenvalue. Example z transform P1 −n The z-transform, X(z) = n=−∞ x[n]z Region of convergence - the z-plane System function, H(z) Properties of the z-transform The significance of zeros 1 H n−1 The inverse z-transform, x[n] = 2πj C X(z)z dz: inspection, power series, partial fraction expansion Sampling and Reconstruction Transform domain analysis - nwz Review - Design structures Lecture 8 Different representations of causal LTI systems LCDE with initial rest condition H(z) with jzj > R+ and starts at n = 0 Recap Block diagram vs. Signal flow graph and how to determine Introduction system function (or unit sample response) from the graphs CT->DT Design structures Impulse Invariance Direct form I (zeros first) Bilinear Trans. Direct form II (poles first) - Canonic structure Example Transposed form (zeros first) IIR: cascade form, parallel form, feedback in IIR (computable vs. noncomputable) FIR: direct form, cascade form, parallel form, linear phase Metric: computational resource and precision Sources of errors: coefficient quantization error, input quantization error, product quantization error, limit cycles Pole sensitivity of 2nd-order structures: coupled form Coefficient quantization examples: direct form vs.
    [Show full text]
  • IIR Filters (II)
    Lecture 8 - IIR Filters (II) James Barnes ([email protected]) Spring 2009 Colorado State University Dept of Electrical and Computer Engineering ECE423 – 1 / 27 Lecture 8 Outline ● Introduction ● Digital Filter Design by Analog → Digital Conversion ● (Probably next lecture) ”All Digital” Design Algorithms ● (Next lecture) Conversion of Filter Types by Frequency Transformation Colorado State University Dept of Electrical and Computer Engineering ECE423 – 2 / 27 ❖ Lecture 8 Outline Introduction ❖ IIR Filter Design Overview Method: Impulse Invariance for IIR FIlters Approximation of Derivatives Bilinear Transform Matched Z-Transform Introduction Colorado State University Dept of Electrical and Computer Engineering ECE423 – 3 / 27 IIR Filter Design Overview ● Methods which start from analog design ✦ Impulse Invariance ✦ Approximation of Derivatives ✦ Bilinear Transform ✦ Matched Z-transform All are different methods of mapping the s-plane onto the z-plane ● Methods which are ”all digital” ✦ Least-squares ✦ McClellan-Parks Colorado State University Dept of Electrical and Computer Engineering ECE423 – 4 / 27 ❖ Lecture 8 Outline Introduction Method: Impulse Invariance for IIR FIlters ❖ Impulse Invariance ❖ Impulse Invariance (2) ❖ Impulse Invariance (3) ❖ Impulse Invariance (5) ❖ Impulse Invariance Procedure ❖ Impulse Invariance Example Method: Impulse Invariance for IIR FIlters ❖ Impulse Invariance Example (2) Approximation of Derivatives Bilinear Transform Matched Z-Transform Colorado State University Dept of Electrical and Computer Engineering ECE423 – 5 / 27 Impulse Invariance We start by sampling the impulse response of the analog filter: ha(t) h[n]= ha(nt0) t0 Sampling Theorem gives relation between Fourier Transform of sampled and continuous ”signals”: ∞ 1 ω 2πk H(z)|z=ejω = Ha(j − j ), (1) t0 t0 t0 k=X−∞ where ω =Ωt0 = 2πf/fs and f is the analog frequency in Hz.
    [Show full text]
  • IIR Filters (II)
    Lecture 8 - IIR Filters (II) James Barnes ([email protected]) Spring 2014 Colorado State University Dept of Electrical and Computer Engineering ECE423 – 1 / 29 Lecture 8 Outline ● Introduction ● Digital Filter Design by Analog → Digital Conversion ● (Probably next lecture) ”All Digital” Design Algorithms ● (Next lecture) Conversion of Filter Types by Frequency Transformation Colorado State University Dept of Electrical and Computer Engineering ECE423 – 2 / 29 ❖ Lecture 8 Outline Introduction ❖ IIR Filter Design Overview Method: Impulse Invariance for IIR FIlters Approximation of Derivatives Bilinear Transform Matched Z-Transform Introduction Colorado State University Dept of Electrical and Computer Engineering ECE423 – 3 / 29 IIR Filter Design Overview ● Methods which start from analog design ✦ Impulse Invariance ✦ Approximation of Derivatives ✦ Bilinear Transform ✦ Matched Z-transform All are different methods of mapping the s-plane onto the z-plane ● Methods which are ”all digital” ✦ Least-squares ✦ McClellan-Parks Colorado State University Dept of Electrical and Computer Engineering ECE423 – 4 / 29 ❖ Lecture 8 Outline Introduction Method: Impulse Invariance for IIR FIlters ❖ Impulse Invariance ❖ Impulse Invariance (2) ❖ Impulse Invariance (3) ❖ Impulse Invariance (4) ❖ Impulse Invariance Procedure ❖ Impulse Invariance Example Method: Impulse Invariance for IIR FIlters ❖ Impulse Invariance Example (2) Approximation of Derivatives Bilinear Transform Matched Z-Transform Colorado State University Dept of Electrical and Computer Engineering ECE423 – 5 / 29 Impulse Invariance We start by sampling the impulse response of the analog filter: ha(t) h[n]= ha(nt0) t0 Sampling Theorem gives relation between Fourier Transform of sampled and continuous ”signals”: ∞ 1 ω 2πk H(z)|z=ejω = Ha(j − j ), (1) t0 t0 t0 k=X−∞ where ω =Ωt0 = 2πf/fs and f is the analog frequency in Hz.
    [Show full text]
  • Dsp Course File
    DSP COURSE FILE Contents Course file Contents: 1. Cover Page 2. Syllabus copy 3. Vision of the Department 4. Mission of the Department 5. PEOs and POs 6. Course objectives and outcomes 7. Instructional Learning Outcomes 8. Prerequisites, if any 9. Brief note on the importance of the course and how it fits into the curriculum 10. Course mapping with PEOs and POs 11. Class Time Table 12. Individual Time Table 13. Micro Plan with dates and closure report 14. Detailed notes 15. Additional topics 16. University Question papers of previous years 17. Question Bank 18. Assignment topics 19. Unit wise Quiz Questions 20. Tutorial problems 21. Known gaps ,if any 22. References, Journals, websites and E-links 23. Quality Control Sheets 24. Student List 25. Quality measurement sheets a) Course end survey b) Teaching evaluation 26. Group-Wise students list for discussion topics 1. COVER PAGE GEETHANJALI COLLEGE OF ENGINEERING AND TECHNOLOGY DEPARTMENT OF Electronics and Communication Engineering (Name of the Subject / Lab Course) : Digital Signal Processing JNTU CODE -56027 Programme : UG Branch: ECE Version No : 02 Year: III Year ECE ( C ) Document Number: GCET/ECE/DSP/02 Semester: II No. of pages : Classification status (Unrestricted / Restricted ) Distribution List : Prepared by : 1) Name : 1) Name : M.UMARANI 2) Sign : 2) Sign : 3) Design : Asst.Prof 3) Design :Asst.Prof 4) Date : 11/11/13 4) Date :18/11/15 Verified by : 1) Name : * For Q.C Only. 2) Sign : 1) Name : 3) Design : 2) Sign : 4) Date : 3) Design : 4) Date : Approved by : (HOD ) 1) Name : Dr.P.SRIHARI 2) Sign : 3) Date : 2.
    [Show full text]
  • Sampling and Quantization for Optimal Reconstruction by Shay Maymon Submitted to the Department of Electrical Engineering and Computer Science
    Sampling and Quantization for Optimal Reconstruction by Shay Maymon Submitted to the Department of Electrical Engineering and Computer Science in partial fulfillment of the requirements for the degree of MASSACHUSETTS INSTITUTE OF TECHNOLOGY Doctor of Philosophy in Electrical Engineering JUN 17 2011 at the LIBRARIES MASSACHUSETTS INSTITUTE OF TECHNOLOGY ARCHiVES June 2011 @ Massachusetts Institute of Technology 2011. All rights reserved. Author ..... .......... De 'artment of'EYectrical Engineerin /d Computer Science May 17,2011 Certified by.. Alan V. Oppenheim Ford Professor of Engineering Thesis Supervisor Accepted by............ Profeksor Lelif". Kolodziejski Chairman, Department Committee on Graduate Theses 2 Sampling and Quantization for Optimal Reconstruction by Shay Maymon Submitted to the Department of Electrical Engineering and Computer Science on May 17, 2011, in partial fulfillment of the requirements for the degree of Doctor of Philosophy in Electrical Engineering Abstract This thesis develops several approaches for signal sampling and reconstruction given differ- ent assumptions about the signal, the type of errors that occur, and the information available about the signal. The thesis first considers the effects of quantization in the environment of interleaved, oversampled multi-channel measurements with the potential of different quan- tization step size in each channel and varied timing offsets between channels. Considering sampling together with quantization in the digital representation of the continuous-time signal is shown to be advantageous. With uniform quantization and equal quantizer step size in each channel, the effective overall signal-to-noise ratio in the reconstructed output is shown to be maximized when the timing offsets between channels are identical, result- ing in uniform sampling when the channels are interleaved.
    [Show full text]
  • Understanding Digital Signal Processing Third Edition This Page Intentionally Left Blank Understanding Digital Signal Processing Third Edition
    Understanding Digital Signal Processing Third Edition This page intentionally left blank Understanding Digital Signal Processing Third Edition Richard G. Lyons Upper Saddle River, NJ • Boston • Indianapolis • San Francisco New York • Toronto • Montreal • London • Munich • Paris • Madrid Capetown • Sydney • Tokyo • Singapore • Mexico City Many of the designations used by manufacturers and sellers to distinguish their products are claimed as trade- marks. Where those designations appear in this book, and the publisher was aware of a trademark claim, the designations have been printed with initial capital letters or in all capitals. The author and publisher have taken care in the preparation of this book, but make no expressed or implied war- ranty of any kind and assume no responsibility for errors or omissions. No liability is assumed for incidental or con- sequential damages in connection with or arising out of the use of the information or programs contained herein. The publisher offers excellent discounts on this book when ordered in quantity for bulk purchases or special sales, which may include electronic versions and/or custom covers and content particular to your business, training goals, marketing focus, and branding interests. For more information, please contact: U.S. Corporate and Government Sales (800) 382-3419 [email protected] For sales outside the United States please contact: International Sales [email protected] Visit us on the Web: informit.com Library of Congress Cataloging-in-Publication Data Lyons, Richard G., 1948- Understanding digital signal processing / Richard G. Lyons.—3rd ed. p. cm. Includes bibliographical references and index. ISBN 0-13-702741-9 (hardcover : alk. paper) 1.
    [Show full text]