On % = 1010100010841 701 Qe >-@ I - vy 5) yi age Hy 4 xe ‘C. Ramesh Babu Durai 7Published by : j LAXMI PUBLICATIONS (P) LTD 22, Golden House, Daryaganj, | New Delhi-110002. 011-23 26 23 68 Phones: { 011-28 26 23 70 (011-23 25 25.72 | rows |e Branches : 129/1, [ir Main Road, IX Cross Chamrajpet, Bangalore (Phone : 080-26 61 1 61) 26, Damodaran Street, T. Nagar, Chennai (Phone : 044-24 34 47 26) St. Bonedict’s Road, Cochin (Phone : 0484-239 70 04) Pan Bazar, Rani Bari, Guwahati (Phones : 0361-254 36 69, 251 38 81) 4-2-453, Ist Floor, Ramkote, Hyderabad (Phone : 040-2475 02 47) Adda Tanda Chowk, N.D, 365, Jalandhar City (Phone : 0181-222 12 72) 106/A, Ist Floor, S.N. Banerjee Road, Kolkata (Phones : 033-22 27 87 78, 22 274 52 47) 18, Madan Mohan Malviya Marg, Lucknow (Phone : 0522-220 95 78) 128A, Block 3, First Floor, Noorani Building, L.J. Road, Mumbai (Phone : 022}24 46 39 98) Radha Govind Street, Tharpagna, Ranchi (Phone : 0651-230 77 64) EMAIL: [email protected] WEBSITE : www.laxmipublications.com @Ali Rights Reserved with the Author and the Publishers. | EDS-0652-125-DIGITAL SIGNAL PROCESSING First Edition Price : Rs. 125.00 Only. C—9743/04/12 Laser Typesetted at : Goswami Printers, Delhi-110053 Printed at : Ajit Prinfors, Dolhi-110053Preface This book has been primarily written with the objective of providing a single text, cover- ing the syllabus on Digital Signal processing for the B.E degree of Electrical and Electronics Engineering students of Anna University, Tamilnadu. The book will also be useful to the stu- dents of ECE, EIE, CSE and IT as it covers most of the syllabus of these disciplines. The book can be used by student in other states as well as student preparing for AMIE. ‘This book has evolved from the lecture notes prepared for teaching the courses in engi- neering to undergraduate students. This book is divided into 9 chapters. The first chapter deals with basic concept of signals and classification of signals. Chapter 2 deals with application of digital signal processing. Chap- ter 3 concentrates on different types of classification of discrete time system and analysis of discrete time system. Chapter 4 explains frequency domain characterization of discrete time systems and introduces the concept of Z-Transform. Chapter 5 deals with frequency analysis of signals. It concentrates on Fourier Transform and Discrete Time Fourier Transform. Chap- ter 6 covers Discrete Fourier Transform and Fast Fourier Transform. Chapter 7 introduces the sampling process and design of analog low pass filter. Chapter 8 discusses the digital filter structures. Chapter 9 deals with design of Digital Filter from Analog filter. Asset of questions and exercises has been included at the end of each chapter with a view of helping the reader to increase the understanding of the subject and to encourage further reading. Suggestions for improvement of the book will be gratefully acknowledged C. RAMESH BABU DURAIContents Chapters Pages L.__Inntroduction sesssesssssssnntansnansssnsenssnssaninansiasssnnssansnssannossevamssnssansassessssioes 128 1.1 Classification of Signals 12_Multi Channel 1.3 Multi Dimensional Signals. 1.4 Continuous-time Versus Discrete-time Signals . 1.5 Frequency Concept is Continuous Time and Discrete Time Signals... 15.1 Continuous-time Sinusoidal signals 1.5.2 Discrete-time Sinusoidal Signals . 15.3 Harmonically Related Complex Exponentials 1.6 Energy and Power Signals (Continuous time-instants) 1.7 Singularity Functions.......... 1.7.1 Unit-Impulse Function 1.7.2 Unit-Step Function .... 1.7.3 Unit-Ramp Function 1.74 Unit-Pulse Function Co 1.8 Energy Signals and Power Signals (Discrete-time instants) 1.9 Signal Processing... 1.10 Analog Versus Digital Signal Processing Review Questions... Exercises 2. Applications of Digital Signal Processing.. 2.1 Introduetion . 2.2 Application to Speech Processing . 2.2.1 Vocal Mechanism . 2.2.2 Speech Technology 2.23 Parameters of Speech 2.2.4 Speech Analysis 2.25 Speech Coding... 2.3 Application to Image Processing 2.3.1 Image Formation and Recording 2.3.2 Image Sampling and Quantization., 00 lo» be leoChapters 2.3.3. Image Compression 2.3.4 Image Restoration ...... 2.3.5 Image Enhancement... Review Questions. 8. Discrete Time Systems... 3.1 Discrete-time Signals and Systems 3.1.1 Definition .. 3.1.2. Representations 3.1.3 Some Elementary Sequence 3.1.4 Representation of Arbitrary Sequence 3.2 Classification of Discrete-time Signal 3.3 Sampling . 3.4 Real and Complex Sequence .. 3.5 Finite and Infinite Sequence 3.6 Types of Infinite-length Sequence 3.7 Operations on Sequences .. 3.8 Sampling Rate Alteration . 3.9 Classification Based on Symmetry Problem 3.9.1 Periodic Conjugate-symmetric Part and Periodic Conjugate Anti-symmetric Part ... 3.10 Sampling Process. 3.11 Classification of Discrete-time Systems .. 3.12 Time-domain Characterization 3.12.1 Representation of a Discrete-time Signal in Terms of 3.12.2 Discrete-time Unit Impulse Response and Convolution $um Representation of LTI System 8.18 The Convolution Process 3.14 Properties of Linear Time-invariant System ........ 3.15 Causality and Stability Condition for LTI Diserete-time Syste 3.16 Classification of LTI System .. 3.17 Systems Described by Difference Equation 3.18 Recursive and Non-recursive Discrete-time System 3.19 Linear Constant Co-efficient Difference Equation 3.19.1 IIR and FIR System 3.20 Solution of Linear Constant Co-efficient Equation .. 3.20.1 The Homogeneous Solution of a Difference Equation 3.20.2 The Particular Solution of the Difference Equation 3.20.3 The Total Solution the Difference Equation .. 3.21 The Impulse Response of a LTI Recursive SystemChapters Pages 3.22 Impulse Response Review Questions .. Exercis 4, Frequency Donain Characterization or Discrete-Time System. 4.1 Fourier Transform of discrete-time Signals 4.1.1 Fourier Series for Discrete-time Periodic Signal 4.1.2 Condition for convergence of Fourier Transform 4.2. Frequency response of Discrete-time Systems 4.3. Properties of Frequency Response 44 Polar form of Frequency Response 4.5 Frequency Response of First order System 4.6 Properties of Frequency Response. 4.7 Z-Transform 4.7.1 Definition of Z-transform ... 4.7.2 Region of Convergence 4.7.3 Properties... 92-130 4.8.1 The Inverse Z-transform Using Contour Integration . 4.8.2 The Inverse Z-transform by Power Series Expansion or Via Long Division... 4.83 The Inverse Z-transform by Partial Fraction Expansion 118 4.9 Solution of Difference Equation Using Z-Transform... Review Questions Exercises .. 5.__ Frequency Analysis of Signals.. 5.1 Frequency Analysis of Continuous-time (Analog) Signals... 6.2 Evaluation of Fourier Co-efficients 8s 5.3 Symmetry Conditions for Periodic Signals 5.4 Exponential Fourier Series .. 6.4.1 Existence of Fourier Series cites eeeeseeeeeneste LST 5.5 Fourier Spectrum 5.6 Properties of Continuous-time Fourier Series . 5.7 Continuois-Time Fourier Transform 5.8 Fourier Transform ofa Periodic Signal 5.9 Properties of Continuous Time Fourier Transform ... 5.10 Frequency Domain Representation of Discrete Time Signal and System .. 5.10.1 Frequency Analysis of Discrete Time SignalsChapters Pages 6. lL 5.10.2 Fourier Series for Discrete Time Periodic Signals... 5.10.3 Expression for the Values of the Co-efficient ay. 5.11 Discrete Time Fourier Transform ...... 5.11.1 Inverse Discrete Time Fourier Transform 5.11.2 Condition for Convergence of Fourier Transform 5.11.3 Energy Density Spectrum . 5.11.4 Properties of Discrete-Time Fourier Transform Review Questions... Exercises... Discrete Fourier Transform 6.1 Introduction 6.2 The Discrete Fourier Transform 6.3 Properties of the DFT. 6.4 Linear Convolution ints 6.5 Circular Convolution... conseensee sass 6.5.1 Methods of Performing Circular Convolution... 6.6.1 Overlap Add Method 6.6.2 Overlap Save Method 6.7 Computation of the DFT of Real Sequences 6.7.1 N-point DFTs of Two Real Sequences using a Single N-point DFT. 6.7.2 2N-point DFT of a Real Sequence using a Single N-point DFT .. 6.8 Fast Fourier Transforms Algorithms se 6.8.1 Tnbroduction eens 6.8.2 Radix of FFT Algorithms 6.8.3 Radix-2 Algorithm .. 6.9 Decimation-in-time FFT Algorithms 6.10 The 8-point DFT using Radix-2 DIT FFT. 6.10.1 Flow Graph for 8-point DIT Radix-2 FFT ....... . 6.11 Decimation in Frequency (DIF) Radix-2 FFT 6.11.1 The 8-point DFT using Radix-2 DIF FFT. 6.12 Comparison of DIT and DIF .. Review Questions... Exercises... Digital Processing of Continuous Signals 7.1 Introduction ... 7.2 Sampling Process. 7.2.1 Analysis of Sampling Process in Frequency Domain. 165 181 182Chapters 7.3 Sampling Theorem .. 7.4 Anti Aliasing Filter. 7.5 Signal Reconstruction 76 Zero-order Hold ..... 7.6.1 ‘Transfer Function of Zero Order Hold 7.7 Sampling of Band Pass Signals . 7.8 Frequency Selective Filters and Filter Specifications. 7.8.1 Filtor Specifications 7.9 Analog Lowpass Filter Design 7.10 Analog Lowpass Butterworth Filter. 7.11 Analog Lowpass Chebyshev Filters... 7.11.1 ‘Type-I Chebyshev Approximation . 7.11.2 Pole Locations for Chebyshev Filter 7.11.3 Chebyshev Type-II Filter .. 7.12 Analog Frequency Transformation 7.18 Design Procedure for Analog Butterworth Lowpass Filte 7.14 Design Procedure for Analog Chebyshev Lowpass Filter .. 7.15 Sample and Hold Circuit 7.16 Analog-to-Digital Convertor 7.16.1 Flash A/D Converters 7.16.2 Serial-Parallel A/D Converter... 7.16.3 Successive-approximation A/D Converter 7.16.4 Counting A/D Converter. 7.16.5 Oversampling Sigma-Delta A/D Converter . 7.17 Digital-to-Analog Converter 7.17.1 Weighted-Resistor D/A Converter 7.172 Resistor Ladder D/A Converter ... 7.17.3 Oversampling Signal-delta D/A Converter Review Questions... Exercises . 8. Digital Filter Structures. 8.1 Introduction .. 8.2 System Describing Equations 8.3 Recursive and Non-recursive Structures 84 Block Diagram Representations 8.4.1 First Order System Block Diagram Representation .... 8.5 Structure For IIR System .... 8.5.1 Direct Form Structures .. 8.5.2 Cascade Form Structure 8.5.3 Parallel Form Structure ..Chapters 8.6 Structures For FIR Systems . 8.6.1 Direct Form FIR Structure .. 8.6.2 Cascade Form FIR Structure 8.6.3 Linear Phase FIR Structure. Review Questions ... Exercises . . 9. Digital Filter Desi; 320-340 91 Untrod action een 9.2. Selection of the Filter eae 9.2.1 IIR Filter Design by Impulse Invariance ...... 9.3 Bilinear Transform Method ... 9.3.1 Development of Transformation 9.3.2 Characteristics of Bilinear Transformation .. 324 9.4 Warping Effect 9.5 Pre-Warping Review Questions Exercises Examination Question Papers .. IndexDIGITAL SIGNAL PROCESSING Chapters: 1. Introduction 2, Applications of Digital Signal ProcessingIntroduction Characterization and Classification of Signals Signal A‘signal’ is defined as any physical quantity that varies with time, space and any other independent variable or variables. More precisely a signal is a function of a set of independent variables. The signal itself carries some kind of information available for observation. Processing By ‘processing’ we mean operating in some fashion on signal to extract some useful information. Digital ‘The word ‘digital’ shall mean that the processing is done with a digital computer or special purpose digital hardware. Digital Signal Processing Digital signal processing is concerned with the representation of signals by sequence of numbers or symbols and the processing of these sequence. The purpose of such processing may be to estimate characteristic parameters or trans- form a signal into form which is in some sense more desirable. Application Bio-medical engineering, acoustics, radar, speech communication, data communication, image processing, nuclear science and many others. 1.1. CLASSIFICATION OF SIGNALS ‘There are five methods of classifying signals based on different features : (a) Based on independent variable. (6) Depending upon the number of independent variable. (c) Depending upon the certainity by which the signal can be uniquely described. (d) Based on repetition nature. (e) Based on reflection.Digital_Signal! Processing (a) Based on independent variable. Independent variables can be crete. Gntinuous or dis- i 1. Continuous Time Signal, It is also referred as analog signal i.e., th sented continuously in time. In simple words, a signal x(t) is said time signal if it is defined for all time. 2. Discrete Time Signal. Signals are represented as sequence at discre ‘Thus, the independent variable has discrete values only. signal is repre- be a continuous e time intervals, x(t} x(n): T r (a) Continuous time signal (b) Diserete time signal Fig. 1.1 e.g. Speech signal is an example of analog signal. A discrete time signal which discrete-valued represented by a finite number of digits is referred to as a “digital signal” e.g. Digitized music signal stored in CD-ROM disk. (6) Depending upon the number of independent variable. (i) L-D Signals. It is a function of a single independent variable. e.g. (a) speech signal-independent variable is time. (b) musie signal. (ii) 2-D Signal. Itis a function of two independent variables. e.g. Photographic image signal-two independent variables a vafiables. | Each frame of a black and white video signal is a 2D-image signal that is a func- tion of two discrete spatial variable, with cach frame occurring squentially at discrete instants of time. (iii) M-D Signal. It is a function of ‘M’ independent variable in tin eg. Video signal. ‘The black and white video signal can be considered an example of a 3D signal where the three independent variables are two spatial variables and time. A colour video signal is a three-channel signal composed of three 3-D| ing the three primary colours : red, green and blue (RGB). For transmission purpose, the RGB television signal is transformed ito another type of 3-channel signal is composed of luminance component and two chrominande components. (©) Depending upon the certainity by which the signal can be unjquely described the two spatial Fenhls represent- as (i) Deterministic Signal, A signal that can be uniquely determine by a well-defined process such as a mathematical expression or rule, or table Jook-up is called aIntroduction e.g. (a) A sinusoidal signal can be represented as, a) vi) =V,, sin of fort > 0. (®) A square signal can be defined as x(t) for 00 Odd [x(n)} =- 1/2 forn<0 =0 forn=0 =V2 forn >0. Thos, Bven fe(n)] = Zeta) +2) as) Odd [x(n)] = Pein) 2 nl (1.4)Properties of even and odd signal : 1. The sum of two even signals are even signal. 2. The sum of two odd signals are odd. 3, The sum of an even signal and an odd signal is neither even no1 4, The product of two even signal is even. 5, The product of two odd signal is even. 6. The product of even signal and an odd signal is odd. 1.2 MULTI CHANNEL A signal can be generated by a single source or by multiple sources} In the former case, it is a (single) scalar signal and in the later case itis called a multichannel signal. ‘These type of signals can be represented in vector form as, [rye x(t) =| x(t) Px0) Equation represents a 3-channel signal. eg. Inclectrocardiography [ECG] for example 3-lead and 12-lead are often used in practice, which result in 3-channel and 12-channel sign 1.3 MULTI DIMENSIONAL SIGNALS. Ifa signal is a function of a single independent variable, then it is sional signal, Similarly, if signal is a function of N-independent variabl dimensional signal. eg. « Picture signal is a two dimensional signal, since the intensity J *wo independent variables x and y. « Black and white television picture is an example of 3-dimens brightness I(x, y, #) is a function of three independent variables| «tis also possible to have multichannel and multidimensional si For example, a colour TV picture is described by three inten 1.x, 9, #) [redl, I, (xy, #) [green], and I, (x,y, t) [blue]. Hence colour TV picture is a three dimensional and three channel represented by the vector. Lay, A Tee, yt) =| Iga 950 I(x, 9,0). 1.4 CONTINUOUS-TIME VERSUS DISCRETE-TIME SIGNA (2) Signals can be further classified into different categories depe| teristics of the time (independent) variables and the values they take. Continuous jodd signal. lor multiple sensors. vector signal, often (1.5) ‘lectrocardiographs nls. alled as one-dimen- 5, it is called as N- .¥) is a function of pnal signal because y and t (time), als simultaneously. ity functions of form signal, which can be (1.6) S \ding on the charac- * Continuous-time signals or analog signals are defined for evegy value of time and they take on values in the continuous interval (a, b).Introduction 8 where acanbe ban be +<, ‘Mathematically, these signals can be described by functions of a continuous variable. www) —— eg. Speech signals x,(t) = cos at xf)eeltl, -acten, Discrete « Discrete time signals are defined only at certain specific values of time, These time instant need not be equidistant, but generally they are taken at equally spaced inter- vals for convenience eg. x(t,)2e'*! n=0,4122.... index ‘n’ of the discrete-time instants as the independent variables. In applications, discrete-time signal may arise in two ways ‘« In practical setting, such sequence (1) can often arise from periodic sampling of an analog signal. In this case, the numeric value of the n‘* number in the sequence is equal to the value of analog signal x,(t) at time nTie., x(n) =a(nT). ‘The quantity Tis called sampling period and its reciprocal is the sampling frequency. « By accumulating a variable over a pericd of time. For example, counting the number of cars in a given street every hour, or recording the value of gold every day, results in discrete-time signals. (2) Continuous-valued and Discrete-valued Signals. A signal is said to be continu- ous valued signal if it takes on all possible values on a finite or infinite range. On the other hand, if the signal allowed to take on values from the given set, it is said to discrete-valued signal. Normally, these values are equidistance and hence can be expressed as an integer multiple of the distance between two successive values. If the signal to be processed is in analog form, it is converted to a digital signal by sampling the analog signal at discrete instants in time, obtaining a discrete-time signal, and then by quantizing its values to a set of discrete values. Quantization. The process of converting a continuous-valued signal into a discrete- valued signal, called quantization. (3) Deterministic Versus Random Signals Depending upon the certainity by which tho signal can be uniquely described as (i) Deterministic Signal. A signal that can be uniquely determined by a well-defined process such as a mathematical expression or rule, or table look-up is called a deter- ministic signal. xt eg. (a) A sinusoidal signal can be represented as, v(t) =V,, sin ot for t2 0.(6) A square signal can be defined as x)=A for 00,f S@dt a. ‘The integral of the impulse function is also a sin- 1 gularity function and called the unit-step function and is represented as, =f0 , t<0 t wo= {1850 Fig. 1.10 (a), Continuous time ‘The value att = 0 is taken to be finite and in most unit step signal cases it is unspecified. The discrete-time unit-step sig- ut) nal is defined as fO , n 259. 1.7.3 Unit-Ramp Function ‘Tho unit-ramp function, r(¢) can be obtained by in- tegrating the unit-impulse function twice or integrating ) the unit-step function once, ie, ro=f f/ sidrde. =f wade, Sore = matin nef SE recta A ramp signal starts at ¢ = 0 and increases linearly with time ‘. In discrete-time domain, the unit-ramp sig- nal is defined as, rin) 0, n<0 rn={) ,n>0 Fig. 1.11 (6). Diserete time 1.7.4 Unit-Pulse Function ramp signal. An unit-pulso function, n(¢), is obtained from unit-step signal as shown below. n(t) = ult + /2)- u(t - 72)Fig. 1.12. Unit pulse signal. ‘The signal u(t + 1/2) and u(t — 1/2) are the unit-step signals shifted by f/2 units in the time axis towards the left and right respectively. Advantage. The advantage of the singularity function is that any arbitfary|signal that is made up of straight line segments can be represented in terms of step and famp functions. Properties of &(t) (y fo@dt =1 | (2) [xo swat = x(0) | Proof for (2) : £ x(0) lim, 8 4(0 de Be) = Jim 84(0) = sim + [Lx Proae = tim, 2 Py + him Pro jim, aE. x(t)dt =x(0). | According to pulse funtion property, Py(t) = 1 (3) fm) 5e-t) dt =x(t) [xO 3e-W ar =x) 1 lel &t) (6) x(t) &{t —t,) = x(t,) (7) alty) 8(¢ ~ te) = x(¢) (5) (at) = (8) f° x8" -t) dt = IPH). Proof for (8): Sato a1) = x(t) 3 (tt) + 4 () Blt —t,) = X(E)5(t — ty) + 2g) A(t - fy), ty < by 1, the signal is compressed. The signal width becomes 1/2 with unity amplitude. ax) (b) ait) = 2nte - 1/4). Here the signal is shifted to the right, with centre at 1/4. Since a = 1, the signal width is land amplitude is 2. ax < 4 > =a O14 t (c) x(t) = cos (20 xt — 5x) = cos [2x (¢ — 1/4)] Here the signal x(t) is shifted by quarter cycle to the right. (d) x(t) =r(- 0.5¢ + 2) = if- os(t-2)] =r[-05(t-4)] The given ramp signal is reflected through the origin and shifted to the right at ¢ = 4.Digital Signs ‘The signal is expanded by = = 2. When ¢= 0, the magnitude of the signpl x) = 2. x(t) + Problem 4. Write down the corresponding equation for the given signal, xt) oe Sol. Representation through addition of two unit step functions, the signal x(¢) can be obtained by adding both the pulses, ie., x(t) = 2u(e) - u(t-2)] + (w(t - 3)—wle - 5)] Representation through multiplication of two unit step functions, x(t) = Q[ult) w(t + 2)) + [ule 3) u(t + 5) = Dfu(t) u(2 - 2] + fu(t—3) ul5 -#)) Problem 5. Plot the following signals for the given x(n) = (5—n) {u(n) u(n ~ 5) Wyn) = 34-1) (i) yofn) = @n-3). Sol. The given x(n) = (6 ~ n) [u(n) ~ u(n - 5)] is plotted as shown below, u(n 5). un). ov “> re ol re o Fig. (a) Fig. (6)Introduction 23 ] u(r) -uin~ 5) x(n) 5 10 | ces eS 8 OS ee Fig. © Fig. @) Now Fig. (c) is multiplied by scale factor (5 — n) thus we get wave Fig. (d). y(n) = a(4—n) where, x(n) = (5 ~ n) [u(n) - u(n - 5)] x(4—n) = (6 ~ 4 +n) [ul4—n)—u(4—n—5)] = (1 +n) lul4—n)-ul-1-n)] u (4 ~n) means the sequence will exist between - « capt =f | oer om xt) ed T or ee ee ae sn computer OP convortor signal Fig. 1.15. Digital signal processing system Following are the main elements of digital signal processing system. 1. Sampler 2. Quantizer 8. Digital Signal Processor 4, Decoder (D/A converter). _X continuous time signal, when sampled at regular intervals is converted into the dis- crete-time signals by means of sampler. The output of the sampler consists of a sequence of ample values of original analog signal. Note that amplitudes of the sampled signals are not restricted, in principles, any amplitude is permissible. However if the sampled signal is to be processed in digital computer, its values must be represented by a certain number of bits, so only finite amplitude level is possible. This results in quantization of amplitude which is done by quantizer. The quantized discrete signal x(n) is called a digital signal. This signal is applied to the input of the digital signal processor. ‘The digital signal processor may be a large programmable digital computer to perform the desired operations on the input signal. It may also be hardwired digital processor configured to perform a specific set of operations such as filtering frequency analysis and so on. In some applications where the digital output is to be given in analog form, such as speech signal, we must convert the digital signal into the analog signal. Such a operation is performed by Digital to Analog converter (D/A converter). However, there are other applications, involving signal analysis, where the desired in- formation is conveyed in digital form, therefore, no D/A converter is required. For example, in radar applications, the information extracted from the signal, such as its speed and position of aircraft, may simply be printed on the paper.1.10 ANALOG VERSUS DIGITAL SIGNAL PROCESSING Advantages: (1) Flexibility. Digital signal processing operations are flexible as the| bpetations can be changed by changing the program. (2) Tolerance. Unlike analog circuits the operation of the digital circujts does not de- pend on precise values of the digital signals. As a result, the digital cireuits are fess gensitive to tolerance component values. (3) Component drift with temperature and time. Digital systems re fairly inde- pendent of temperature, aging (time), and most other external parameters, Fdr example, due to change in temperature, the internal resistance R may change in analog sfstems. On the other hand, digital systems use logic 1 or logic 0 which are independent of ten}perature. (4) System Size. Analog systems normally use L, C and R, therefore sizd of hardware is large as compared to digital system. | (5) Storage. Digital signals are easily stored on magnetic media (e.g.}tape and disc) without deterioration or loss of signal fidelity, therefore the signal becomes trahsportable and can be processed off-line in a remote laboratory. On the other hand, storedfanalog signals deteriorate rapidly as time progresses and cannot be recovered in their origingl form. (6) Implementation. It is very difficult to perform precise mathematical operations on signal in analog form but these same operations can be routinely implementefl on the digital computer using hardware. (7) Cost. Digital signal processing allows the sharing of a given prodessor among a number of signals by time sharing. Thus reducing the cost of processing pex]signal. This is done by “time-division multiplexing”. Disadvantages : (1) System Complexity. Digital signal processing of analog signals iq more complex because of the need for additional pre-and post processing devices such as A/P and D/A con- verters and their associated filters, (2) Band Width. The second disadvantage associated with digital signgl prpcessing is the limited range of frequencies available for processing. This property limitsits application particularly in the DSP of analog signals. The signals having extremely wide bandwidth re- quire fast sampling rate A/D coverters, Hence, there are many analog signals with large band- width for which the digital signal processing approach is beyond the state of the art of the digital hardware. (3) Power. The another disadvantage of DSP is that signal systems dre constructed using active devices (transistor) that consumes power. On the other hand, a variety of analog processing algorithm can be implemented using passive circuits employing indpctor, capacitor and resistor that do not need any power. Also active devices are less reliable than passive devices.Introduction [ 27 ] 1 8. a 10. uw L 4 REVIEW QUESTIONS Write the major classification of signals, Explain the difference between deterministic signal and random signal with suitable example. Define periodic and aperiodic signals with the help of examples. Explain even and odd signals with the help of examples. Explain enorgy and power signal with the help of examples. Define the following elementary signals (1) unit impulse signal. (2) unit step signal (8) unit ramp signel. Explain the following manipulations for independent variable of a signal (1) Time shifting (2) Time sealing (3) Time inversion or folding. What are the advantages of Digital signal processing compared to Analog signal processing. Briefly explain multichannel and multidimensional signals. Define continuous time exponential and discrete time exponential signal. Write the properties of impulse response signals. EXERCISES Determine which of the following signals are periodic and determine the fundamental period also. (1) aft) = 20 sin 25 at (2) x(t) = 20 sin J5 xt (8) aft) = 10 cos 10 xe (A) xtt) = 8 cos (5 t+ 1/6) (5) xin) = 3 cos (5n + 3/6) (6) x(n) = 2 exp (j (n/6 - x) (7) afm) = c08 (n/8) cos (xn/8) Determine the even and odd components of each of the following signals : (1) aft) = cos t + sint + sin t cos ¢ (2) alt) = 1+ t+ 22 + 5t5 + Be. Consider the sinusoidal signal (0) = A.cos (at + 8) Determine the average power of x(t). Sketch the waveforms of following signals (1) a(t) = u(t)—ult - 2) (2)a(t) = u(t + 1) 2u(t) + ult 1) (8) Mt) =r (t+ I) rl) + Xt 2) Given a sinusoidal signal Determine the fundamental period of x(n). Given a complex valued exponential signal x(t) = Aet 4 for a >0 Evaluate the real and imaginary components of x(¢).Signal Processing . Determine the power and rms value for each of the following signals : @ 200s 100+ 5] (3) 10 cos 5¢ cos 10 (4) e**cos wet. Figure below shows a signal =(¢). For this signal sketch. (2) 20 sin 5¢ cos 10¢ () x= 4) (2) x(e/10) (8) x(t - 2) (4) 3-2).Applications of Digital Signal Processing 2.1 INTRODUCTION Because of the availability of high resolution spectral analysis, DSP has various appli- cation areas, which requires high speed processors to implement the FFT algorithm. It is also popular due to availability of custom made DSP chip which is highly reliable, Speech process- ing, Audio processing, Radar signal processing and Image processing would be discussed in this chapter. 2.2 APPLICATION TO SPEECH PROCESSING ‘The signals of speech are one dimensional. DSP is applied to a wide range of problem in speech such as channel vocoders, spectrum analysis ete. Problems in speech processing can generally be divided into three classes, first is the speech analysis. The speech analysis is performed to extract some desirable information of speech. This system starts with analysis of speech waveform and the desired result is used for speech recognization and speaker indentification. Second type of problem is speech synthesis. Init, input is in written text form and the output is a speech signal. For example, an automatic reading machine for which the input is written text and the output is speech. Pinally the third type is speech compression which involves speech analysis followed by speech synthesis. If the speech is transmitted by simply sampling and digitizing, the data rate required is in the order of 90,000 bits per second of speech. Through the use of appropriate coding this can be reduced by factor of 50, depending on the type of system used. 2.2.1 Vocal Mechanism Production of speech. The two important part responsible for human speech are (a) vocal cord and (6) vocal tract. (a) Vocal cord. It has two bands of tough, elastic tissue, which is located at the opening of the larynx. It vibrates when the air from the lungs passes between them producing sound waves which are emitted from the lips and to some extent from the nose ; these are sound waves heard as speech. (b) Vocal tract. It includes larynx, the pharnx and the nasal cavity. 29Kinds of Sounds | (@) Voiced sound (Gi) Unvoiced (fricative) sound. | Voiced sounds are produced by quasi-periodic pulses of air exciting the vocal tract. Unvoiced sounds are produced at some point along the vocal tract, usually towards the mouth. There are some important speech technology areas. viz., speech coding, speech enhance- 2.2.2 Speech Technology (a) Speech coding. “Speech Coding’ is the process of capturing the speech of a person and processing it to transmit over a communication channel. ‘The application of “speech coding” is in the area of telephony, narrow-band cellular radio, military communication ete. (6) Speech enhancement. This is the process of minimizing the derogatory effects of noise on the performance of speech communication, source coding etc. The application of ‘speech enhancement’ is in the areas where the perforhance of equip- ment is improved in noisy atmosphere like factories ete. (c) Speech analysis and synthesis. Analysing speech is done by studying its spec- trum and extracting time-varying parameters from the signal for the productipn of speech. Synthesizing speech lies in creating speech like waveforms from textual words or sym- bols, using a model for speech production and time-varying parameters. The application of this are in voice alarms, reading machines for the ftumb or blind, data-base enquiry services etc. (d) Speech recognition. The process of deriving the meaning from p speech input whereby a request can be made for information or service from a machinery by conversing with it. Application of “speech recognition” could be Banking from distant locatipn, information retrieval systems ete. (e) Speaker recognition. It means to recognize a particular person's identity with the sample speech dipping. 2.2.3 Parameters of Speech (i Pitch : Corresponds to frequency of sound (in Hz). (i) Loudness : This relates to intensity of sound (in dB). (iii) Quality : This relates to harmonic constant of sound (in timbre). ‘Phonemes’ are the smallest unit of sound that are recognized by contrast! with their environment, these are forming the basic units of speech. ‘Dipones’ are sourgis that stretch from the middle of one phoneme to the centre of the next, there by spanning the transition region. 2.2.4 Speech Analysis ‘The most common methods of speech analysis are as follows : (a) Short-time fourier analysis (6) Linear prediction (c) Homonorphic filtering. Let us discuss about these three methods of speech analysis.Applications of (a) Short-time fourier analysis. The short-time Fourier transform ofa sampled speech signal represented by the sequence x(n) is given by tal Signal Processing a(n) = Dx) h(n - er oAZD) wera Fig. 2.1 shows the short-time fourier analysis. hin)a 0} n Fig. 2.1, Short-time fourier analysis. There are two methods of obtaining the short-time fourier analysis, (i) Through analog implomentation of a filter bank. (i) By digitally computing short-time fourier transform either by using a filter bank or by using the FFT algorithm (®) Linear prediction. This method is based on “Auto regressive moving average” or pole-zero model. If H@) is an all-pole transfer function given by A H@)=—-—_ w(2.2) 1-¥ ae ma We have the time-series discrete time signal as P x(n) = Da x(n- 2) +A Bn) (2.3) aA where A is the gain factor, r for n>0, x(n) = Yay x(n &) (2.4) athis approximate value, 2 Hin) = J) ay Xn), n>0 | (2.6) rest | ‘The corresponding error is given by a= x(n) - X(n) = x(n) — y a, x(n—k),n>0 | +26) it Mean squared error is given by P 2 y (n= = |- Vaan -»| 2D aed 1 im The parameters {a,] that minimize Bug are determined by partial dgrivating Eyyg with respect to each co-efficient a,, # = 1, 2, ...... p and equating to zero, i.e., Ey: Ex 3a, 1,2, that gives . Oy On = 9%» E= 1,2, (2.8) i N-1 | with = LL, ea-Dx(n-2), (2.9) mo (c) Homonorphie filtering (Cepstral Analysis). Since the excitation function and vocal tract impulse response are convolved to produce speech, this problem }s thought of as a separation or deconvolution of speech into these two components. ‘The deconvolution speech is carried out by the non-linear filtering technique, described here as “Homonorphic filtering”. The convolution operation is converted intp addition which gives the output called the “complex cepstrum”. Please note that the “Cepstrum” of a signal represents the fourier transform of its power spectrum, Input Output 4s a a > xin) Sa) Ym Le Ye) System Linear inverso System System i ee — | tz) =togx(z) V2) =109 Viz) Fig. 2.2. Homonorphic systems for deconvolution. | From the Fig. 2.2, we have \ x(n) = x(n)* x(n) \ (2.10) from the system definition S. |Applications of Digital Signal Processi 33] ‘The system S has the property Ke) = log Xz) -(2.1) where X(z) is the z-transform of 4(n). ‘Therefore Xiz)=X1(2)+ Xo) evo 2.12) Hence x(n) = 24(n) + £4(n) + (2.13) From equation 2.13, it is seon that a convolution of components is done by thoir addi- tion. Fig. 2.3 below shows 2 layout of homonorphic system. Input speach ‘signal x(nT) nat) Seu) & a FT Jog meant] Oi 500 Satter Fig. 2.3 (a). Analyser portion of Homonorphic system. Min-phase Low time cepstum sequence c(nT) value oinT) ee ees Fig. 2.3 (6). Synthesizer portion of Homonorphic system. 2.2.5 Speech Coding ‘The process of representing the speech signal in digital form at a low bit rate, with which it can be understood by a listener is called “Speech coding”. ‘There are four important parameters of a speech coder. (a) Bit rate. It measures the number of special properties of speech exploited. (6) Quality. Implies the degradation of the coded speech signal. (c) Delay. It implies the amount of speech signal needed to find the parameters of a speech coder reliably. (d) Complexity. It measures the computational necessity for coder implementation in signal processing hardware.‘There are different kinds of speech coding which is tabulated as belo ‘Table 2.1. Speech coding methods Wave form coding Base band coding Narrowbayd cording (@) Pulse Code Modulation (PCM) (©) Linear predictive coding (d) Frequency domain coding (6) Adaptive pulse code modulation | (APCM) No specific | (b) Segment Quantiza type available (a) Pitch-oxcited co input signal x(n), sample by sample. ‘The quantization error e(n) = y(n) — x4n) the decoder in the receiver. Then an inverse digital transformation is perfokmed for mapping the signal back into the time domain. Input signal Forward ‘Transformation | i} decoding Encoder Lprt Input @ LPF a *| Lpr2 { Encoder unication| ® Encoder Decoder ® Fig. 2.5. Sub-band coding.Applications of Digital Signal Processing 35 ] additive recombination of theset of sub-band signals, the original speech signal can be generated. Each band is separately quantized and coded using pulse code modulation and transmitted. ‘The schematic is shown in Fig. 2.5. 2.3 APPLICATION TO IMAGE PROCESSING Any function which bears two-dimensional information is called an image. Image can be represented by an array of real or complex (real and imaginary) numbers with finite number of bits with respect to speech signal (which are one-dimensional signals), image signals are two dimensional. Image can be divided into picture elements or pixels (smallest element of image). Manipulation of two-dimensional signal with the help of digital computer is called “{mage Processing”. Its purpose is to improve the visual appearance of Image A Digital Image is digitalization of picture. Normally two-dimensional imaye nas reso- lution 128 x 128, 256 x 256, 512 x 512. So image can be processed using two-dimensional signal processing, The image processing including the following steps : (a) Image Formation and Recording. (b) Image Sampling and Quantization. (c) Image Compression. (d) Image Restoration. (e) Image Enhancement. All the operations are possible on advanced software artificial intelligence and high- tech digital computers. Let us discuss all the operations one by one. 2.3.1 Image Formation and Recording ‘The two-dimensional signal of image ean be expressed by image function as a6, = [0 [7 bean y= 9D Fler yd dey dr (2.15) Eqn. (2.15) governs a 2D linear time invariant system. Here system impulse response function h(x -x,,y—y,) is commonly referred as point-spread function which is usually associ- ated with optical image. The funetion f(x, y) is the accumulation of energy from the objects radiant energy distribution. ‘Two major technologies are used for image sensing and recording, which are photo- chemical recording and photo-electronic recording. Both of the technologies are exemplified by readily available products which are photo-graphic films and television respectively. (Here “television” is used in generic sense not commercial broad casting television). 2.3.2 Image Sampling and Quantization After formation and recording of an image, it is sampled and quantized for the suitabil- ity of digital processing. In a system project, a spot of light with intensity J, incident on a film and intensity 1, reflected from the film and collected by photo-multiplier. The transmittance is defined by 1 Te (2.16) 1pnal Processing This eqn. (2.16) can be used to compute optical density. The mathenjatical model can also be described as a spot of light, moves in a raster to sample the film is given by Bx = i f Aig ~ 4,9 91) 6 (Xp Wy) dey dy +217) Here hi, is the intensity profile of the spot of light projected an film. f is the image on film and finallyg, is actual sampled image. The sample matrixg,(k Ax, Ay)}s the sampled or digital image. 2.3.3 Image Compression which can be reduced by image compression. So we can say that image comprdssion is a seience of efficiently coding a digital image to reduce the number of bits, which required to represent ‘it. Uncompressed image consumes memory space in a large amount so |t intreases com- plexity in computational and need a very large transmission bandwidth. A chmpressed image reduces the redundancy in image. There are mainly three types of redunflancy which are discussed one by one. The first type redundancy is Spatial Redundancy which arises due fo correlation be- tween neighbouring pixels. Second type is Spectral Redundancy which is corfelation between various colour plans. And finally is Temporal Redundancy is the correlation Hetween different frames in an image sequence. Inan Image Compression System, the original continuous time image s (Analog to Digital) converter, which converts it into digital signal. Now a seri converter decomposed signal into parallel channels which fed to a quanti al is fed to A/D to parallel (S/P) pr. The S/P con- diagram of image compression system is shown in Fig. 2.6. Digital Decomposed ‘Quantized signal signal into output parallol channols Input (original ‘continuous tims image signal) AD SP *| convertor *) Convertor >) Quantizer *| git image eignal) Pig. 2.6. Image compression system. Applications of digital signal processing image compression system ‘There are mainly three types of compression technique based on the shethod of redun- dancy detection :Appli ions of Digital Signal Proces: 19 (a) Direct data compression method (b) Transformation method (©) Parametric extraction method. 2.3.4 Image Restorat ‘The process of image restoration is used for correcting imaging effect to recover an original signal. ‘This type of effect (imaging effect) is due to variety of intermixing factors, which are defocusing imaging camera, relative motion between object and camera, noise in sensors cte,, All types of imaging offects deteriorate image quality. ‘The process of image restoration is to attempt a image which should be sharp, clean and free from the degradation. The restoration process is also called Image Doblurring. The proc- ess of image formation and recording can be modelled as n aey= RL Pie-ay = 91) Flas, 94) dey yy | 4 nlx, y) --A2.18) Here g(x, y) is the actual image, R is the response characteristic of the recording process and n(x, y) is additive noise source. In the restoration of digital image following equation can be expressed in discrete form . N-1N-1 8.a= YY fi Dh(p-ia-) (2.19) i A large set of simultaneous linear equations can be solved by DSP techniques such as linear filters and FFT algorithms which are computationally efficient tools for solving these. 2.3.5 Image Enhancement This technique improves the appearance of image for human perception by choosing some image foatures like edges or contrast ete. Its main application is in biomedical engineer- ing field for computer aided mammographics studies. In image enhancement spatial filtering is mainly used whose operation is done on im- age to reduce noise contamination of the image signal. Image enhancement is composed of a variety of methods whose suitability depends upon the goals at hand when enhancement is. originally applied. REVIEW QUESTIONS 1, Give the areas in which signal processing find its application. 2, Explain the various stages in voice processing. 3. How is a speech signal generated ? 4, Give the model of speech production system ? 5. What is the noed for short time spectral analysis ? 6, What is a vocoder ? Explain with a block diagram ? 7. Describe how targets can be detected using radar. 8 Give an expression for the following parameters related to radar (@) beam width, and (6) maximum unambiguous range.9. 10. 1. 12, 13. 14, 15. 16. Digital Explain with the block diagram the modern radar system. Give the various image processing applications. Give the various coding techniques for images. What is the need for image compression ? Give the block diagram of basic restoration process. What is sub-band coding ? Explain the process of digital FM stereo signal generation. Explain how privacy can be acheieved in telephone communications. jignal_ ProcessingDiscrete Time Systems 3.1 DISCRETE-TIME SIGNALS AND SYSTEMS 3.1.1 Definition 1. A discrete-time signal is a sequence, that is a function defined on the positive and negative integers. 2. A discrete-time system is a mapping from the set of acceptable discrete-time signals called the input set, to a set of discrete-time signals called output set. 8. A discrete-time signal whose values are from a finite set is called a digital signal. 4, A digital system is a mapping which assigns a digital output signal to every accept- able digital input signal. 3.1.2 Representations 1, Graphical. In digital signal processing, signals are represented as sequence of num- bers called samples. A sampled value of typical discrete-time signal or sequence is denoted by x(n) which is a function of independent variable that is an integer. It is graphically repre- sented in Fig. 3.1. oY x(6) Fig. 3.1. Graphical representation. aignal Processing Itis important to note that x(n) is defined only for integer values of nland| undefined for non-integer values of n. In the signal we have assumed that a discrete-time sequence is define for every integer value of for ~= N, ie, Fig. 3.4. Left-sided sequence.x(n) =0 forn > Ny where, Ny is a finite integer which can be positive or negative. In general ‘Two sided-sequence is defined for all values of n in the range—= it is a delaying operation. If N < 0, it is an advancing operationDiscrete Time Systems [93] The device implementing the delay operation xin) wie) by one sample is called a “Unit delay” and its. |= >| 7" |} schematic representation is shown in Fig. 3.8 Fig. 3.8. Unit delay. wn) =xln— 1) ‘Theschematic representation of the unit advance —_*(") 2 wale) operation is shown in Fig. 3.9 Fig. 8.9. Advance operation. win] = xin + 1) (v) Time-reversed or Folding. The time reversal operation, also called the folding operation, is another useful scheme to develop a new sequence. wn) =x—n) (3.20) which is the time-reversed version of the sequence x(7). (vi) Pick-off node. It is used to provide multiple copies of a x(n) x(n) sequence. Problem 3. Consider the following two sequence of length 5 defined forOsn<4: c(n) = (3.2, 41, 36, - 9.5, 0) x(0) dn) = (1.7, - 0.5, 0, 0.8, 1). Fig. 8.10, Pick-off node Determine w, (n), ws (n) and w, (n) = = e(n). Sol. (1) w(n) = dn). dln). w,(n) = (5.44, - 20.5, 0, - 7.6, 0}. @) w,{n) = o(n) + din) = (4.9, 40.5, 36, - 8.7, 11 1 @) w, (0) = 5 ln) w,(n) = (11.2, 143.8, 126, ~ 33.5, 0}. Problem 4. Consider a sequence [g(n)] of length 3 defined for 0 1, the process is called “interpolation” and results in a sequence with a higher sampling rate. | * On the other hand, if R < 1, the sampling rate is decreased Hy a process called “decimation”. | The basic operations employed in the sampling rate alteration pro sampling and down-sampling. These operations play important roles in time systems. Up-sampling. In up-sampling by an integer factor tess are called up- ultirate discrete L> 1,L~ 1 equidistant zero-valued samples are inserted x00) bythe up-sampler between each two consequtive samples of the input sequence x{n] to develop an output sequence y(n) according to the relation, slWiL], n=0,+L,42L...0. x, In) = ” a (8,22) oe Sie | eam Sampling rate of (7) is‘L’ times larger than that of the original sequpne¢ x(n). Down-sampling, Conversely, the down sampling operation by an intpger factor M > 1, ona sequence x{n] consists of keeping every M™ sample of x[n] and removing)M —1 in between samples, generating an output sequence y(n) according to the relation, (n) = x {nM ‘The result in a sequence y(n) whose sampling rate is (1/m)*" that of x(h). =-(8.23) 3.9 CLASSIFICATION BASED ON SYMMETRY PROBLEM Problem 6. Consider the finite length sequence of length 7 defined fo lg(n)) = (0, 1 + j4, - 2 + j3, 4-72, - 5 - j6, -j2, 3} t 3Ens3: To determine its g., and 8. Sol. (1) To determine conjugate symmetric part g., (n) [g*(n)] = (0, 1 -j4,- 2—j8, 4 + j2, - 5 + j6 , j2, 3) TDiscrete Time Systems 51] Whose time-reversed version is given by, (g* (—n)) = (3,32, —5 +6, 4+j2,~ 2-8, 1 -j4, 0) T Fle) +x¥(—n)]. eq (g,,(n)} = (1.5, 0.5 + j3, - 8.5 +j4.5, 4,-3.5 ~j4.5, 0.5 —j3, 1.5) t 2. To determine conjugate anti-symmetrie part g,, (n) aq (0) = Flen) —24-nd) (g,, @)= + 1.5, 05 +j1, 1.5 -J1.5, ~j2, - 1.5 -J1.5, - 0.5 jl, 5) t It can be easily verified that, Beg (2) = Bg (—) and Bq (2) = ~ Beg" (2) 3.9.1 Periodic Conjugate-Symmetric Part and Periodic Conjugate Anti-symmetric Part Periodic conjugate-symmetric part defined by, Spel ; bx(n) +24 n)gll. 0S ¢N-1 (9.24) Periodic conjugate anti-symmetric part is defined by, X peal) = i (x(n) -x*((-n)y]], OSn=N-1 (3.25) So that xin)=x,,,(0)+%,.,(n), OSnSN-1 «[email protected]) Note. A length N sequence xin] defined for 0 )}. We observe that, u*|(-0),] = ut (0) = 1-j4. u*((— 1),) =u* (3) =-5 +76. u* ((—2),] = u*(2) =4 +j2 u*(—3),) = u() =- 2-73.{u*{(—n),)} = (1 —j4, —5 + 76, 4 + j2,—2—-j3} 1 Ugey (M) = Fluln) + uC n),) Uyeg(M) = (1,-3.5 +) 4.5 4,-3.5 -j 4.5) Upeg () = (94 , 1.5~j1.5, ~ 2, ~ 1.5 -J1.5). It can be easily verified that, Uy. (n)=U",,, ((-n),) and Upag (M) =—U,., (nr). 3.10 SAMPLING PROCESS ‘The discrete-time sequence is developed by uniformly sampling a continudus time signal x,(t) as illustrated in Fig. (3.11). | x0) x(t > t xT) Fig. 3.11. Continuous time signal. The rolation betwcon tho two signals is given by eqn. (0) = (|p ogg = Hq (NT) 10 = 0, —2,-1, 0,1 creo = (B.27) where, t-time variable of the continuous time signal is related to the time variable n of the discrete time signal only at discrete-time instants t, given by, «+ (3.28) F 1 ‘ . with, Fy = 7 denoting the sampling frequency and Qy = 2nFy denoting the sampling angular frequency For example, if continuous — time signal is, x,(t)= A cos (On fit +0) x,(0)= A cos (gt +4) the corresponding discrete-time signal is gives by, x(n) = x,{nT) cos [Q, nT + 6] = Aon [20 | Oy x(n) = A cos [0,,, + 6] (3.29)Discrete Time Systems 53] where, (3.30) It is the normalised angular frequency of the discrete-time signal x(n) Units The unit of the normalised digital angular frequency @ is radians per sample. While, the unit of the normalised analog angular frequency ©, is radians per sample and the unit analog frequency /, is hertz is the unit of the sampling period T is in seconds. Problem 8. Consider the three sequence generated by uniformly sampling the three co- sine functions of frequencies 3Hz, 7Hz and 13H2 respectively :g ,¢) = cos( 614), g.{t) = cos (I4nt), and &ft) = cos (26zt) with sampling rate of 10 Hz. ie., with T = 0.1 sec. Find the derived sequence or discrete sequence. Sol. a(t) = cos (Q, £) ; gin) = c08 (O, nT) (7) = cos (6 xn xT) x(n) = cos (0.6 mn) Similarly, Bo (n) = cos (1.4nn) and Bf) = cos (2.6 nn). Problem 9. Determine the discrete time signal v(n) obtained by uniformly sampling at sampling rate of 200 Hz., a continuous time signal v,(t) composed of a weighted sum of five sinusoidal of frequencies 30 Hz, 150 Hz, 170 Hz, 250 Hz and 330 Hz as given below : v4 (t) = 6 cos (60nt) + 3 sin(BO0nt) + 2 cos (340nt) + 4 cos (500nt) + 10 sin (660x2). Sol. To find the sampling period (T): T 1 aw = 0.005 sec. ‘The generated discrete-time signal u(n) is given by, v(n) = 6 cos (0.3nn) + 3 sin (1.5mn) + 2 cos (1.7xn) + 400s (2.51) + 10 sin (3.310). = 6 cos (0.3nn) +8 sin [(2x — 0.5n)n] + 2 cos [(2x ~ 0.3) n] +4 cos [(2x + 0.51)n] + 10 sin ((4x-0.75)n] = 6 cos [0.3nn] —3 sin [0.5 nn] +2 cos [0.3nn] +4 cos [0.5xn] - 10 sin[0.7nn} u(n) = [8 cos (0.31) + 5 cos (0.5m + 0.6435) — 10 sin (0.7nn)]} The discrete-time signal v(n) is composed of a weighted sum of three-discrete-time sinusoidal signals of normalised angular frequencies : 0.32, 0.5% and 0.7n}. 3.11 CLASSIFICATION OF DISCRETE-TIME SYSTEMS Discrete-time systems are classified according to their general properties and charac- teristics. They are (D Static and Dynamic systems. (2) Time-variant and time-invariant systems.