Digital Signal Processing by Ramesh Babu PDF

tH t it : | CN iL 1010109810041701 r ‘¢ F C. Ramesh Babu DuraiPublished by : j LAXMI PUBLICATIONS (P) LTD 22, Golden House, Daryaganj, i New Delhi-110002. Phones { 01-28 26 23 68 ones) 011-23 26 23 70 011.23 25 25 72 Faxes : { 011-23 26 22 79 Branches : 129/1, IIrd Main Road, IX Cross Chamrajpet, Bangalore (Phone : 080-26 61 15 61) 26, Damodaran Street, T. Nagar, Chennai (Phone : 044-24 34 47 26) St. Benedict's Road, Cochin (Phone : 0484-239 70 04) Pan Bazar, Rani Bari, Guwahati (Phones ; 0361-254 36 69, 251 38 81) 4-2-453, [st 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 37 73, 22 24.52 47) 18, Madan Mohan Malviya Marg, Lucknow (Phone ; 0522-220 95 78) 128A, Block 3, First Floor, Noorani Building, L.d. Road, Mumbai (Phone : 022{24 46 39 98) Radha Govind Street, Tharpagna, Ranehi (Phone : 0651-230 77 64) EMAIL : colaxmi@hotmail.com WEBSITE : www.laxmipublications.com All 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 Printers, Delhi-110053Contents Chapters Pages nie 1.1 Classification of Signals ... 1.2 Muiti Channel 1.3 Multi Dimensional Signals 14 Continuous-time Versus Discrete-time Signals ... F 1.5 Frequency Concept is Continuous Time and Diserets Time Signals. 1.5.1 Continuous-time Sinusoidal signals 15.8 Harmonically Related Uliala Renciaa ais 1.6 Energy and Power Signals (Continuous time-instants) L7 Singularity Functions 17.1 Unit-Impulse Function, 1.7.2 Unit-Step Function... 1.2.3 Unit-Ramp Function. sscaaeeoeeeees L8 Energy Signals and Power Signals (Discrete-time instants) ... LY Signal Processing 1.10 Analog Versus Digital Signal Processing... Review Questions... 2 3 Parameters of Speech . 224 Speech Analysis 2.2.5 Speech Coding... 2. Applieation to Image Processing seuss Image Sampling and Quantization. Copyrighted materialid Infinite uence - 3.6 Types of Infinite-length Sequence 7 Obweatlons id FIR Syster lution of Linear Constant Co-vfficic 3.20.1 Th 3.20.2 The Particular Solution of the Difference Equation 3.20.3 The Total Solution the Difference Equation .... 3.21 The Impuise Homogencous Selution of a Difference Equation Copyrighted materialChapters Pages 3.22 Impulse Response Review Questions Exercises 4, Frequency Donain Characterization or Discrete-Time 5. System.. 4.1 42 43 44 45 4.6 47 48 49 Frequency Analysis of Signals 5.1 5.2 53 5A 5.5 56 5.7 5.8 5.9 5.10 Fourier Transform of discrete-time Signals. 4.1.1 Fourier Series for Diserete-time Periodic Signal .. 4.1.2 Condition for convergence of Fourier Transform, Frequency response of Discrete-time Systems ... Properties of Frequeney Response. Polar form of Frequency Response .. Frequency Response of First order System Properties of Frequency Response... 2-Transform 4.7.1 Definition of Z-transtorm 4.7.2 Region of Convergence 4.7.3 Properties 4.7.4 Some Cammon One Sided Z-transforin Pairs . Inverse Z-transform 48.1 The Inverse Z-transform Using Contour Intogration .. 4.8.2 The Inverse Z-transform by Power Series Expansion or Via Long Division...... 4.8.3 The Inverse Z-transform by Partial Fraction Expansion Solution of Difference Equation Using Z-Transform Review Questions... Exercises... Seeee8 100 107 108 ooo 115 116 127 Frequency Analysis of Continuous-time (Analog) Signals .. ov BB Evaluation of Fourier Co-efficients 188 Symmetry Conditions for Periodic Signals . 136 Exponential Fourier Series .. 5.4.1 Existence of Fourier Series Fourier Spectrum . Properties of Continuous-time Fourier Series .. 137 137 137 139 Continuods-Time Fourier Transform 140 Fourier Transform of a Periodic Signal 143 Properties of Continuous Time Fourier Transform .. 143 Frequency Domain Representation of Discrete Time Signal and System . 144 5.10.1 Frequency Analysis of Discrete Time Signals 144Chapters 6. 1 5.10.2 Fourier Series for Discrete Time Periodic Signals... 5.10.3 Expression for the Values of the Co-efficient a, 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... $.11.4 Properties of Discrete-Time Fourier Transform Review Exercises... Discrete Fourier Transform 6.1 Introduction 6.2 The Discrete Fourier Transform..... 6.3 Properties of the DFT... 64 Linear Convolution 6.5 Circular Convolution . 6.5.1 Methods of Performing Circular Convolution 6.6 Sectioned Convolutions . 6.6.1 Overlap Add Method 6.6.2 Overlap Save Method 6.7 Computation of the DFT of Real Sequences .. a 6.7.1 N-point DFTs of Two Real Sequences using a Single N-pofat DFT. 6.7.2 2N-point DFT of a Real Sequence using a Single N-point 6.8 Fast Fourier Transforms Algorithms 6.8.1 Introduction 6.8.2 Radix of FFT Algorithm snes i 6.8.3 Radix-2 Algorithm 5 5: 6.9 Decimation-in-time FFT Algorithms . 6.10 The 8-point DFT using Radix-2 DIT FFT 6.10.1 Flaw Graph for 8-point DIT Radix-2 FFT 6.11 Decimation in Frequency (DIF) Radix-2 FFT...... i 6.11.1 The 8-point DFT using Radix-2 DIF FFT... 6.12 Comparison of DIT and DIF Review Exercises... Digital Processing of Continuous Signals.. 7.1 Introduction 7.2 Sampling Process.. 7.2.1 Analysis of Sampling Process in Frequency Domain. uestions ..Chapters 7.3 Sampling Theorem .... 7.4 Anti Aliasing Filter. 7.8 Signal Reconstruction .. 71.6 Zero-order Hold .. 7.6.1 Transfer Function of Zero Order Hold 7.7 Sampling of Band Pass Signals .. 78 Frequency Selective Filters and Filter Specifications... 7.8.1 Filter 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 7.11.8 Chebyshev Type-II Filter 7.12 Analog Frequency Transformation . 7.13. Design Procedure for Analog Butterworth Lowpass Filter 7.14. Design Procedure for Analog Chebyshev Lowpass Filter .. 7.15 Sample and Hold Cireuit. 7.16 Analog-to-Digital Convertor . 7.16.1 Flash A/D Converters 7.16.2 Serial-Parallel A/D Converter 7.16.8 Successive-approximation A/D Converter 7.16.4 Counting A/D Converter. 7.16.5 Oversampling Sigma-Delta A/D Converter TAT Digital-to-Analog Converter .. 17.1 Weighted-Resistor D/A Converter .. 7.17.2 Resistor Ladder D/A Converter ..... 7.17.3 Oversampling Signal-delta D/A Converter ... Review Questions stem Describing Equations .... a 83 Reoarsie and Non-recursive Structures 8.4 Block Diagram Representations... 8.4.1 First Order System Block Diagram Rapiesecitatioa, 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 Structur 8.6.3 Linear Phase FIR Structure Review Questions Brercises.... 9% Digital Filter Design... 9.2 Selection of the Filter Type ise 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 . 9.4 Warping Effect 9.6 Pre-Warping.... Review Questions Exercises... Examination Question Papers ..Chapters : 1. Introduction 2. Applications of Digital Signal Processing DIGITAL SIGNAL PROCESSING4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Introduction e.g. (a) A sinusoidal signal can be represented as, xt) v(t) = V,, sin of for t2 0. (6) A square signal can be defined as xt)=A for Ox or |f| > 1/2, are identical to the sequence obtained from the sinusoid with frequency |@ |< or | fj <2. (8) In discrete-time sinusoids, highest rate of oscillations is attained when w = x (or ~ x) or equivalently f = 1/2 (or - 1/2), To investigate the characteristic of the sinusoids, let us vary the frequency from 0 to x. @ @, = 05d, ie,, no oscillations ; N= a x¢0) @y=0 * (ii) @)= 1g, or f,= V/16 ; 16 samples in one cycle ; N = 16. x0)4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Introduction 7] for any value oft < 0, the integral becomes zero and for ut) a t>0,| St)dt=1 ‘The integral of the impulse function is also a sin- 1 gularity function and called the unit-step function and is represented as, w= {91 £2 : 1,#>0 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- un) nal is defined as 1.7.3 Unit-Ramp Function Fig. 1.10 (). Discrete time unit ‘The unit-ramp function, r(t) can be obtained by in- step signal, tegrating the unit-impulse function twice or integrating aa the unit-step function once, ke, y= ff amaraa. = fi uaa, | t That is, Hoe i" ise Fig. 1. fg Cones time A ramp signal starts at ¢ = 0 and increases linearly with time ‘. In discrete-time domain, the unit-ramp signal is defined as, rin) rn) jo, n0" Fig. 1.11 (6). Discrete time 1.7.4 Unit-Pulse Function ramp signal. An unit-pulse function, n(¢), is obtained from unit-step signal as shown below. nt) = ult + V/2)~ u(t ~ 1/2)4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Introduction Sol. (a) x(t) = n(2t + 3) = w[2 (¢ + 3/2) Here, the signal is shifted to left, with centre at - 3/2. Sincea = 2i.e., is compressed. The signal width becomes 1/2 with unity amplitude. >1, the signal ay (b) aft) = 2nt — 1/4), Here the signal is shifted to the right, with centre at 1/4. Land amplitude is 2 ince a = 1, the signal width is rer) f 1 ee (ec) x(t) = cos (20 nt — 5x) = cos [2n (¢ - /4)] Here the signal x(¢) is shifted by quarter cycle to the right. x(t) - 1 (d) x(t) = r(- 0.5 + a-/[-0s(:-2)] =r[-05(¢-4)] The given ramp signal is reflected through the origin and shifted to the right at t = 4,4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Fig. 1.14. Simple pole filter. Digital Signal Processing AID converter ret xen) x0) Ao) input (a (9) J Digtar signa |_ vio aap ae —_F = Paced x(t) i T or os as ee computer OP . ma ‘signal convertae Fig. 1.15, Digital signal processing system. Following are the main elements of digital signal processing system. 1. Sampler 2. Quantizer 3. Digital Signal Processor 4, Decoder (D/A converter), _# continuous time signal, when sampled at regular intervals is converted into the discrete-time signals by means of sampler. The output of the sampler consists of a sequence of sample 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 lovel 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 information 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 theJoperations can be changed by changing the program. (2) Tolerance. Unlike analog circuits the operation of the digital circujts does not depend on precise values of the digital signals. As a result, the digital circuits are Jess sensitive to tolerance component values, (3) Component drift with temperature and time. Digital systems pre fairly independent of temperature, aging (time), and most other external parameters, Far example, due to change in temperature, the internal resistance R may change in analog systems. 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 dise) 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, stored fanalog signals deteriorate rapidly as time progresses and cannot be recovered in their origin}l form. (6) Implementation. It is very difficult to perform precise mathematical operations on signal in analog form but these same operations can be routinely implemented on the digital computer using hardware. (1 Cost. Digital signal processing allows the sharing of a given progessor among a number of signals by time sharing. Thus reducing the cost of processing perfsignal. 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/D and D/A converters 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 limitsjits application particularly in the DSP of analog signals. The signals having extremely widd bandwidth re- quire fast sampling rate A/D coverters, Hence, there are many analog signals with large bandwidth 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 gre 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.Introduct Py Pease a 9. 10. ue a 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 energy and power signal with the help of examples. Define the following elementary signals (2) unit impulse signal. (2) unit step signal (3) unit ramp signal. Explain the following manipulations for independent variable of a signal (2) 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. Determine which of the following signals are periodic and determine the fundamental period also. (1) x(t) = 20 sin 25 nt (2) x0) = 20 sin JB xt (3) (0) = 10 cos 10 nt (4) x(t) = 3 cos (5 t + 1/6) (5) x(n) = 3 cos (5n + 1/6) (6).x(n) = 2 exp (j (nf — 0) (1D) x62) = cos (nf) cos (xn/B). Determine the even and odd components of each of the following signals : () x) = cost + sine + sin t cost (2) ait) = 14 0+ Be + BE + Bet Consider the sinusoidal signal x (0 = Acos (ox +8) Determine the average power of x(t). Sketch the waveforms of following signals : (2) 0) = u(t) — ult -2) (2) x(t) = ut + 1) 2 (8) + ult - 1) Ba =r E41) rs) + t-2) An gz sn)= 20.0 a8] Determine the fundamental period of x(n). Given @ complex valued exponential signal at) = Ae + for a> 0 Evaluate the real and imaginary components of xt),7. Determine the power and rms value for each of the following signals : (1) 20 on [300% « (2) 20 sin ¢ cos 10¢ (8) 10 co 64 cos 10¢ (4) e* cos cat. 8. Figure below shows a signal x(¢). For this signal sketch. () xt ~ 4) (2) x(t/10) ¢3) x13¢~2) (A) x18 = 0.Applications of Digital Signal Processing 2.1INTRODUCTION Beeause of the availability of high resolution spectral analysis, DSP has various application 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 processing, 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 etc. 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. Finally 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) voeal 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) Vocat traet, It includes larynx, the pharnx and the nasal cavity. 29Kinds 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 spepch of a person and processing it to transmit over a communication channel. ‘The application of “speech coding” is in the area of telephony, narro¥-band cellular radio, military communication ete. (b) Speech enhancement. This is the process of minimizing the derogatory effects of on the performance of speech communication, source coding etc. The application of ‘speech enhancement is in the areas where the perforshance of equip- ment is improved in noisy atmosphere like factories etc. (c) Speech analysis and synthesis, Analysing speech is done by studying its spectrum and extracting time-varying parameters from the signal for the productibn 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 Bumb or blind, data-base enquiry services etc. (d) Speech recognition. The process of deriving the meaning from ft 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 locatign, information retrieval systems etc. (¢) Speaker recognition. It means to recognize a particular person's identity with the sample speech dipping. ni 2.2.3 Parameters of Speech (i) Pitch : Corresponds to frequency of sound (in Hz). (ii) 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 conteast with their environment, these are forming the basic units of speech. ‘Dipones’ are soungls 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 (5) Linear prediction. (c) Homonorphic filtering. Let us discuss about these three methods of speech analysis.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Applications of Digital Signal Processing a] The system S has the property R(z) = log Xz) = (Q11) where X(z) is the z-transform of in). Therefore X@)=X,(2) +X.) Hence x(n) = 3,(n) + Z_(n) From equation 2.13, it is seen that a convolution of components is done by their addition. Fig. 2.3 below shows a layout of homonorphic system. Input speech sere a) i Ke R x 0, (ow) } Xen ~ Low time copstrum value (rT) Fig. 2.3 (a). Analyser portion of Homonorphic system. in-phase Low time cepstrum ‘Sequence o{nT) value c{aT) rt f+ expenentas |} +f er | oo Excitation _ [Generation of | | Excitation Output Parameters signal 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 apecial 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.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Applications of Digital Signal Processing G7 (a) Direct data compression method (b) Transformation method (c) Parametric extraction method. 2.3.4 Image Restoration ‘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 ete., All types of imaging effects 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 alsa called Image Deblurring. The process of image formation and recording can be modelled as ae, y) = af [ Max, y- 9) fen rd dxy én] +nlx,y) 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 : NatNal He.d= YL Moh (p-ig- (2.19) A large set of simultancous linear equations can be solved by DSP techniques such as linear filters and FFT algorithms which are computationally efficient tools for solving these. 2.18) 2.3.5 Image Enhancement This technique improves the appearance of image for human perception by choosing some image features like edges or contrast etc. Its main application is in biomedical engincer- ing field for computer aided mammographies studies. In image enhancement spatial filtering is mainly used whose operation is dene on image 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 Give the areas in which signal processing find its application. Explain the various stages in voice processing. How is a speech signal generated ? Give the model of speech production system ? What is the need for short time spectral analysis ? What is a vocoder ? Explain with a block diagram ? Deseribe how targets can be detected using radar. Give an expression for the following parameters related to radar (a) beam width, and () maximum unambiguous range PrP ReePE4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.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. 3. 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 acceptable digital input signal. 3.1.2 Representations 1, Graphical. In digital signal processing, signals are represented as sequence of numbers called samples. A sampled value of typical discrete-time signal or sequence is denoted by x(n) which is a funetion of independent variable that is an integer. It is graphically represented in Fig. 3.1 x(n) x8) Fig. 3.1. Graphical representation. a4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Therefore, periodic signals are power signals. 3. Symmetric (even) and antisymmetric (odd) signals A real valued signal x(n) is called symmetric if x(n) =x(-n), on the other hand, a signal x(n) is ealled antisymmetric it sf-n) = a(n). Even signal, Xm) =X, (n) = 31Kin) +m} 23.) Odd signal, Kel) =X y(n) = 3 Xt) -X(-n)]} (3.8) nea X(n)is defined as, [ Xin) = Xn) + Xin) 43,9) Problem 1. Consider a sequence defined by, 4-L" n20 xm) i otherwise determine the power and energy of x(n). Sol. B= Yc ao Say" > aa = Diatar|=16 Y 1-16) 5 a0 a E== ‘, woh, I6(N+ 1) Power, Pun in ee = aed P,,=16 | which is finite Problem 2. Determine the response of the following systems to the input Xin} = wy 0, 8, 2, 2,0, 1, 2,3, 0.0.3 T (a) y(n)=X(n +) (oy yind =F tein + 1) + x00) + ain — DI) (edyin)= YS Xie). reo4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Time Systems [9] The device implementing the delay operation x(n) wan) by one sample is called a “Unit delay” and its of} schematic representation is shown in Fig. 3.8 Fig. 3.8, Unit delay, wn) =2{n— 1 The schematic representation of the unit advance (") 7 watn) ation is shown in Fig. 3.9 Fig, 3.9. Advance operation. wen] = x{n + 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(n) (vi) Pick-off node, It is used to provide multiple copies of a xin) xta) sequence. Problem 3. Consider the following two sequence of length 5 defined forOsns4: cn) = 18.2, 41, 36, - 9.5, 0) x(n) d(n) = [1.7 ~ 0.5, 0, 0.8, 1) Fig. 3.10. Pick-off node. Determine w, (n), w, (n) and w, (n) = Z e(n). Sol. (1) w,(n) = en). dn). w,(n) = (5.44, - 20.5, 0, - 7.6, 0}. @) w,{n) = e(n) + din) 4,9, 40.5, 36, -8.7, 1) 8) w,(n) = z e(n) w,(n) = 111.2, 143.5, 126, - 33.25, 0}. Problem 4. Consider a sequence [g(n}] of length 3 defined for 0 } xh) v9 - &) aoe ne-”) ¥9) anoana T2Fsse788 oves ts ty (9) = x(0) A(Q) + x(1) A(B) + x(2) A(T) + x(3) h(6) + x4) (5) + 25) f(A) + 2(6) ACB) + x(7) A(2) + x(8) AG) + (9) ACO), =0+0404040+0+040+0. Similarly, 4-1) =0. Now we summarize the entire response for—= 2 mS = (1/2) [1 + 2? + 24+ 28 +. ary tey =(u2y =(yay [ @P-1 | ‘ [ el vm-any| #4] Problem 20. Find the impulse response of the cascade systems, if hy (n) =~ 112)" ula) ha (nd) = (1120" utn). Use the convolution sum to find the response to x(n) = (114)" ufn). Sol. (i) To find the impulse response of the system. Kin) =hyn)*hy(n) 9 cascade connection. =D Adohin-& =| > (-12)" war} un) i i= (5 ] [oder z [var ¥ (ny on| Assume that w(n) = 1.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Time System: [ss] Let us consider an LT! system having an o/p atn = ng yl) = Dy AUR) x Og - &) se ‘The above systems can be subdivided into the two terms, one having present and past values (n < n,) and other having future values (n 2 n,). Thus = <1 Hig) = [z AUR) x (9 -»)-| zr wo ai-0] mo ies = (A(0) x(q) + AUD) ag = D+ ond + [Al- Dxlngt 1) 4 AE 2) xing +2) + 1 (3.50) y(ng) = 14. The first term in the sum withx(79) x(7tg ~ 1) the term (ID) in the sum with x(n + 1), x(ng +2) hn}=0 forn<0. the output y(n) depends on present and past values of inputs but does not depend on future values of inputs because the second term in sum becomes zero once h(n) = 0 forn <0. Hence an LTT system is causal if and only if its impulse response is zero for n < 0. The limit of summation of the convolution formula may be modified. Thus we have, re the present and past input. Whereas are the future values of input. Now if, yin)= D Alk)x(n-b) putn-k=m Fer) = Yxmrn-m yand= x An- 2) (3.51) ¥ If the input to a causal linear time invariant system is causal i.c., x(n) = 0 for n <0, the limit on the convolution formula can be further modified. Thus we obtain, y= 3 akin = Snir -k) yin) = xy Ak) x (n-k) & It is clear from above eqn. that the response of the causal system to a causal input usal because y(n) = 0 for n <0. ity. A discrete time system is to be stable, bounded input, bounded output stable, if the output sequence s(n) remains bounded for all bounded input sequence x(n). ‘We now develop the stability condition for an LTI discrete-time system. We shall dem- onstrate that an LTI system is BIBO stable if and only if its impulse response is absolutely summable, ic,4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Time Systems | iza| 1 wn) = 25 layin — D+ x00) n 1 yn) = 4 ain D+ ata) (8.60) Eqn. (8.60) suggested that the computation of y(n) requires two multiplication, one addition and one memory location, The block diagram representation is shown in Fig. 3.18. This system is known as reeur- sive system and y(n ~ 1) is called the initial condition of the system. wet Fig. 3.18. Block diagram representation. In general the o/p of a causal can be expressed as, yin) =f y(n = 1), y(n = 2) esses ol. = N), x40), (= 1) on (2 = MY) (3.61) ‘The block diagram representation of causal system is shown in Fig. 3.19. yin) xo) Ayia 1) xn), xf = 1) (0 m)) Fig. 3.19. Block diagram representation of causal system. If y(n) is only function of present and past inputs, then Ha) =f lx(n), x(n — 1) oa... (2 — ma) ++(3.62) Such a system is known as non-recursive system. ‘The block diagram representation of causal is shown in Fig. 3.204@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.rete Time Systems eq] Distinct roots. For the case where eqn. (3.70) has N distinct roots c,, ct, then the most general solution is of the form, gC = Ay + Ay Oey escent Ay Oly” A871) where, Aj, A, Ay are weighting co-efficient. ‘Those co-efficients are determined from the initial condition specified by the term. Problem 26. Determine the homogeneous solution of the system described by the first order difference equation. in) + 3y(n — L) = x(n), with initial condition y—1) = 1. Sol. For the homogeneous solution, x(n) = 0 thus, y,(n) + 3y, (a — 1) = 0. We assume solution of the form of Thus, the general form of solution of homogeneous difference equation is, y(n) = Act =A 3y" Using the initial condition y(~ 1) = 1. We have, yin) == 3y,(n = 1) Putn=0 (0) =-3y,(0-1)=-33y,@)=A A=-3 Therefore the homogeneous solution is given by, yy, (n= 8-3)" = (3), Problem 27. Determine the homogeneous solution of 2% order difference equation. y(n) -y(n- D-yn- 2) = 0 with initial condition y(0) = 0, y(D = 1. Sol. Let us assume the solution of the form anh) a1 ar-2 20, ot? [o?-a-1) =0 ‘Therefore, the roots are, a= ‘The general solution to the homogeneous eqn. is. (n= Ay ew! 14(55) Using the initial condition 0) = 0, (1) = 1, we have, nt Ay4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Time Systems [5] ‘Therefore, Ypin)= ‘ n 2 u(n). The total solution is, yn) 2" + Ag Ines n2"u(n) oD) where the constant A, and A, are determined such that the initial conditions are satisfied. To accomplish this, evaluate the given eqn. atn 20, 1. 90) —y— 1) ~ 2y(— 2) = x(0) = 1. yO)=1 and atn = 1. y(1) — (0) - 2y(— 1) = 2(1) + 2x(0) = 24+ 2=4, ya)=5 Using the value of y(0) and y(1) in eqn. (1), we have Atal 2A,-A,+ 37 5. ‘These two eqn. give A, = 3 Aye 2 Thus the final solution forx(n) = 2" u(n) is given by, 3.21 THE IMPULSE RESPONSE OF A LTI RECURSIVE SYSTEM Impulse response of the linear time-invariant system was defined as a response of the syatem toa unit impulse i.e., x(n) = Bin). Now consider the problem of determining the impulse response A(n) given a linear constant co-efficient difference equation. In the proceeding subsection, we have described that the total response of the system to any input consists of solution to the homogeneous equation plus the particular solution. In case when the input is an impulse, then the particular solution is zero because x(n) = 0 for n> Vie, yp(n) = 0. Therefore, the response of the system to an impulse consists of homogeneous solution. Problem 31. Determine the impulse response of the system described by, yn) — ay(n - I) = x(n) with ¥(- 1) = 0. Sol. For x(n) = &(n) above equation reduced to, yn) — ay(n - 1) = &n), for n > 0, this reduces the homogeneous eqn. i.e., yn) ayn - 1) = 0.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Time. Systems [es] Determine its values and sketch the signal x(n) Determine x(4 —n) [ane. (a) x(n) uaanoh wxe—me foon.403 1.0] tT tT 2. From the two sequencea xn)«(3) vy =[-2 Prove that (a) xin) smn to can (1P (8) ain) yn) = [- 2 3. Express the sequence defined by -2 n=-10,1 an)=) 4 w=-2,2 0 otherwise as a weighted sum of unit-sample sequences (Ams, x(n) = 48(n + 2) — 25(n + 1) - 28{n) ~ 25( — 1) + 48(n - 2)} 4. Prove or disprove that x(n) = 6 sin (20+2) is period. If the sequence is periodic, determine its period. (Ans. Not periodic} 5. A discrete time system can be (Q) Static or dynamic (2) Linear or non-linear (3) Time invariant or time variant (4) Causal or non-causal (5) Stable or unstable with respect to above properties examine the following systems (a) x(n) = cos [y(n)] tAns. Static, non-linear, time-invariant causal, stable] (b) y(n) = x40) sin (@, n) (Ans, Static, linear, time variant, causal, stable] (e) yn) = f—n + 8) (Ans, Dynamic, Linear, time-invariant, non-causal, stable] (d) y(n) = | x(0) | [Ans Static, non-linear, time-invariant, causal, stable) (ed y(n) = xin) uin) (Ans. Static, linear, time-invariant, causal-stable) ya) =e, (Ans, Stable, causal, non-linear, time-invariant] @. Compute the convolution y(n) = x(n) * A(n) of the following signals : (a) x(n) = (0, 1,-2, 3,-4) 1 nind={ [ane{o.2. 32" (b) x(n) =10,0,1, 1,1, 0) fT hin) = (1,-2, 3) tans. (0, 0, 1,-1, Tt eo" -3sns5 oy xn= {9 a 1 Osns4 : wo ft Bn (ame 3 xn-k) -3sns9}4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Frequency Domain Characterization of Discrete-time System: [23] a, mee N k=0,1,2,......N=1. ‘The fourier series representation of x(n) consists of N harmonically related exponential functions. (44) Now multiply e-™"-" to both sides, Naa eAtennl® gin)e Yay eM gn temIN ie Rat Natea 2 4 #a-mn Semaine Sy Dae’ ao a0 ko GY tan -ya ye A4.5) ms x N_ if a=1 Here, ¥-( 1-a% l-a if avi 2s x ar ayes _— j si = cANY seok F ne jsink Fn, =14+0. (fork = 0, +N, +2N . > eta mir IN, k-m=0,£N,,22N, s 0 otherwise Eqn, (4.5) is reduces to Nh inn LD xe "%" =a,N. Ferd cant 4° Fy 2 ne a (4.6) imo Sats Synthesis eqn. xind= Fae 8 a Not ax 1 en Analysis eqn. a= Lame Nn". azo4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Frequency Domain Characterization of Discrete-time Systems [27] Or Plot the magnitude and phase response of a system whose impulse response h(n) =a" u(n) a=05. & Sol. Given y(n) = ay(n — 1) +2(n) ‘Take fourier transform on both sides, ye) = ae Y(o™) + X(e) yler| [1 - act] = Xie) Ye”) 1 H=- ce le") Another method : Consider the difference eqn. of a 1" order systems, xn) = ayn - 1) + x(n), with Xin) = The particular solution to this eqn. is i y,{n) = Cem, Substituting this in original complete eqn. ,{n) ~ ay,(n ~ 1) = x(n) Cette — acelt®~2 = etm, Col [1 -aev] =0™, 1 Therefore, C= oe Thus, the steady state solution of the system is, 9,(n) = em, This solution is of the form He) .e™. Therefore, the frequency response of this first order system is, 1 He = Ie To plot H(e*), we find the magnitude and phase term as, 1 He") = ge ___, T= aleos © - jsin ©) ~ 1-acos o+ jasin@ ‘The magnitude response, 1 H(e*) | = —_—___,*>_,_,__, [ees | (.-@cos @)? +a? sin? eo)”4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.= y Lda r*]+ [and * | =D bet my rts 4.12) If X(z) converges in some region of the complex plane, both summations in eqn. (4.12) must be finite. If the first sum of eqn. (4,12) converges, there must exist values of r small enough for x(—n)r" to be absolutely summable. Hence the ROC for the first sum consists of all points in a circle of radius r, as shown in Fig. (4.2) where r, > r. « = figs Fig. 4.2. R00 for > x Ar | Fig. 43. ROC for |S” a! net | If the second sum of eqn. (4.12) converges, there must exist large values of r for which x(nr" is absolutely summable. Hence the ROC for the second sum consists of all points in a circle of radius, r, as shown in Fig. (4.3) where r, , Gey neo X@)= x(n) Fig. 4.5 Problem 5, An anticawsal signal x(n) is given by, x(n) =-a" w-n-D = {%. mee a” n (@y - ) u(—n~1) u(—n — 1) is a discrete unit step sequence, which is defined as, u(-n-1)=0 for k20. =1 for ks-1t x(n)=0 for 20. =o" for k<-1 zlx(a =X) = DY xe" <1 kena = Y@tay =) @ta-1 cost fo Using infinite geometric series sum, we get X@) = -1s 1 l-o'z4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Frequency Domain Characterization of Discrete-time Systems [es] wt 2 yd alee) + alee} 2 e- eT Bre Oe ez eT) ze7FT 4 2? _ zeint © Me? = 26° geht 5 eT e- Hit) 222 ale +e MT 2? -zleT +e TVD Ba? = ze +e] 41° 2? ~2fe* +e] 41 z|z-cos wT] 2? 22003 @T +1 Xz) = Problem 10. Find the one sided z-transform of the discrete sequences generated by math- ematically sampling the following continuous time function. (a)e cost (6) e~* sin wt eT (ze9T (ze? — cos oT) Sol. (a) == 2 e™™ — 22e"7 cos @T +1 ®) ze" sin oT 2* oT _ 22087 cos wT +1 Problem 11. Find the z-transform of the sequence x(n Sol. 2 (3) ws) According to time shifting property, 2le(n — 1)] = 271 X(z) wo 4 Similarly, z (3) un-D)= 2 sr ) win =D. o Problem 12. Find the z-transform of x(n) = r* cos (n@) u(n). Sol. We know that 1-z cos 1-227 cos0+ 2 [a"x(n)] = K(a-!z) 1-(772)" cos 6 1-20-12)? cos 8 + (r42)* = ___1+ ret cos 1-2re cos 0 +772 2 [cos n@ w(n)] = 2 [r* cos nO u(n)] =4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Frequency Domain Characterization of Discrete-Time Systems [7] Sol. (a) Since the ROC is exterior of a circle, we except x(n) to be a causal sequence. ‘Thus we divide so as to obtain a series in negative power of z. Carrying out the long division, we obtain, 14 42% +72 +102" +. 1-227 +27? )1+22 421-827 +425 Iz? 425 42? ~ 1429 4724 102"* -72* 102" -2024 +1026 Thus X(z) = 1+ 42-1 + 72% + 1025 +. By comparing the relation, we have, xn) = (1,4, 7,10, oreny BND ced . (6) When the ROC is the interjor of the circle, the signal x(n) is anticausal signal. Thus we divide so as to obtain a series in power of z as follows. 22 +52? +829 +1124 gt -2eteijaete1 221-4422 5-22 5-102 +52” 82-52" Bz - 162" +82" liz? - 824 liz? 922" + 1124 142° -112* Thus, X(z) = 22 + 52? + 829+ Let +... In this casex(n) = 0 forn 2 0, thus by comparing the result with eqn. X(z)= )) x(n)2"", we get, a(n) = n+) . 11, 8, 5, 2, 0}. T4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Frequency Domain Characterization of Discrete-Time Systems 721] 1 @ X@)= hae Xe)=—_" oO" Gael X@) z (2+V2-1* X@) Ar, Ag z z+1 GD) (@-1? A= evn] = 0.25 A= 2 e-n?| 205 z 2fX@), y2]] .¢]_ 2 gy? -alee v')| Glee i d +122 ~2? (+)x2-1_ 3 + oS | oS 0.75 "Eloi her @tD |, XG) _ 0.25 0.75 0.5 Se OB OB, r z 241 2-1 @-1* 0.252 0.752, 0.62 Mabe 41 e-1 Go x(n) = 0.25(- 1) u(n) + 0.75 u(n) + 0.50 (1)" for n20. Problem 17. Determine the inverse z-transform of the following z-domain functions : Be? +2241 2a 0 -04 (a) Xt) = OX@) =z 2-4 Xt) = ae 3 2 * 34 2)aeh Bed Sol. (2) Xte) = 3 +2z+1 2 Se 4 a}ae + 2z+1 zt B242 ‘Bz*-927+6 1k-5 a+ iw-5 1-5 “Posse ~°* Gi ne-2 By partial expansion, we get AL, A = 1 2 X@)=3+ a ae Me-8_@_p| --6 Ge i 2) Ln4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Frequency Domain Characterization of Discrete-Time Systems [5] =-0616. z+081|, 95, -1 =o = 1.616 2-081), os 0.154 0.404 Y@)= Tole? 1087 xn) = (0.404 (— 0.81)" - 0.154 (0.317) utn). System function : Let us defined the funetion Ht) by, 4 A419) From the eqn, (4.19) Y¥() = Hz) Xz) The function H() is known as the transfer function of a system or system function. When the input to the system is impulse signal x(n) = &(n) then, Y@) = Ht) H@) = z{hn)). Problem 20. Determine the system function H(z) of (a) in) +3 yin D+ 2 yin—2) = ath) +n D. (b) yn) = do (n — 1) + 2x{n). Sol. (a) Takingz-transform of both sides, ¥ea)+ Set Ye) + Fe? Vi) = Ke) +21 Kl) ee 4 oy yin) = 3 yin—2) +234n) Taking z-transform of both sides, Yl) = 5 2} Y@) + 2X2) ‘Taking inverse z-transform, ay hin) = a(-2) un).4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Frequency Domain Characterization of Discrete-Time Systems [729] 9. Find H(@) and determine its poles and zeros if (a) y(n) = x(n) + 3x(n — 1) + 2x - 2) [ Ha) oS 4IKE+ zeros s2-1-2,Polen: += 0,0] z y(n) +4 yn) 4 2 yin 2) =afn) xf. x2+) _— que [ames Grimes 0,- 1, Poles:z v2, x] 10. By using one sided z-transform, determine y(n), n 2 0 in the following case : 1 1 (a) yin) + 4 yin- 1) = Ayn 2) = 0; (-2)= (a) y(n) + 5 yn ~V~ yin - 2) = 05 A- = A-2)= 1. 15-0527 Ans. ¥"(2) = S72, = I~ Lbs" +0527 1- (yin = 2 yin 10+ xm) sn) = (3) un), X- D1.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Frequency Analysis of Signals In the previous chapter, we studied that any arbitrary signal can be represented as a weighted sum of delayed unit sample sequences &n ~ k). In this chapter we consider an alternate de- scription of a signal (sequence) in terms of complex exponential signal (sequence) of the form e% (of), First we study the Fourier representation of periodic and aperiodic continuous time (analog) signal and then we discuss about Fourier representation of discrete time signals. 5.1 FREQUENCY ANALYSIS OF CONTINUOUS-TIME (ANALOG) SIGNALS A periodic signal is one which repeat itself periodically aver ~~ a, cos (n Og) de + D bf sin (n Qt) dt , Anh Ad 6.2) 4334@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Digital Bignal Processing 8. Differentiation dxit) Periodie signal = “7. Fourier series coefficients =n Q,C,, 9. Integration “ Periodic signal = f x(t) de Fourier series coefficients : {_1_} c, dMQo 10. Parseval’s Theorem Consider two periodic signals x,(t) and x,(t) with the same period T. f the fourier series co-eflicients of these two signals are C, and d,, then If x,(t) = x,(t) = x(¢), then above equation can be written as, Af“ lawr ae= Sic The above equation can also be written as, Yicr-cz+ Hier wcgs Be 5.7 CONTINUOUS-TIME FOURIER TRANSFORM The fourier transform of a continuous time aperiodic signal is given X(Q) = £ x(t)e™ gi for all The inverse fourier transform of XQ) is given by, x)= = [ixum ef do, for all t, The above two equations are known as Fourier transform pair and XO) = Fixte}] and xO=F XY) (5.16) can be denoted as,4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Digital Signa} Processing 3. Time reversal. If Flx<¢)] = X(j, then Fix(-0)) = X(-j0). 4, Frequency shifting property. If Fix(t)] = X(i0), then Fix(t) ej = XY -2,) 5. Time sealing. If F[x(#)] = XO), then Fixtat) = 2x 2), fat La@ 6. Differentiation in time. If Fix(¢)] = XQ), then d As xo] = jQX yo). 7. Differentiation in frequency. If F[x(t)] = X(j0), then d Flétt)} = § — XK (0). Heh = Tag 8. Time integration. If F[x(t)] = X(/Q), then . ¥ *| x(t) dt | = <= K(jO0) + x X(0) 60). iP ] ia (YQ), then 9. Conjugation. If Fix()] = Fix"(o)] = X* (-J0). 10, Duality. If Flx(s)) = X(j@), then Flx(s)] = 2n X(— jo). 11. Convolution property. If F[x(t)] = XQ) and F(hit)] = H(i), then Fla(e).Ate)] = X(j2) HGQ), 12, Multiplication property. If Fix(¢)] = XQ) and FIA(e)} = HGQ), then Fixit) Ate) (iQ) * HYQ). 13. Parseval's theorem. [| x(t)/? de = = fiixcay? aa. 5.10 FREQUENCY DOMAIN REPRESENTATION OF DISCRI NAL AND SYSTEM For LTI systems, we ean represent any input sequence as a weighted sum which loads to the output sequence as a weighted sum of unit impulses. ‘When a signal is decomposed in terms of sinusoidal and exponential the signal is said to be represented in FREQUENCY DOMAIN. TIME SIG- 5.10.1 Frequency Analysis of Discrete Time Signals In this section, we consider the Fourier series representation of discre| signals. Since the Fourier series representation of the discrete time periodic s series, the frequency range for the discrete time signals is unique over the in (0, 2m), A diserete time signal with fundamental period N consists of freque time periodic als is a finite val (~ x, x) or icy components4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.oil Perio 16 ae! 2 Page =3/* (e) Xe”) deo ag LIX »P de 5 Pat [Tx (emp Be Dizm =a [Ix dw 5.27) called Energy density spectrum denoted a! Sg Sy (00) = | Xl") |? ‘The above relation eqn. (5.27) is known as Parseval’s relation for di signals. ‘The function | X(e®) |? Sol. X= Y xine, = Sor umes Fate = F (och iat wm mt a if | ae i; Tis | forja}<1 = ~ 4 ; Xlxl= diet "Tai <*: flel -w-amn __1 fF = “ze [x:@a0 Y xalne " [X:@d0x,0 8) le “> [so Xy(@- 0) do. (12) Modulation theorem, If x(n) <—! + X(o) then x(11) 08 ©) n 2+ 3 [Xi@ + @,) + X(w- @,)]. Proof. Kw) =D) x(n) cos one“ Eel] 22S xanterterom gerteron =} X(@- 04) + Xo +o). REVIEW QUESTIONS 1. Define Fourier seri 2, What are Dirichlet condition ? 8. If the period signal x(¢) is having even symmetry, when the Fourier expansion contains... terms only ? 4. ‘The Fourier series expansion of an odd periodic functions contains .. 5. Define Fourier transform of a sequence ? 6. Write down the sufficient condition for the existence of Discrete Time Fourier Transform (DTFT) * 7. State Parseval's theorem for discrete time signals ? 8. What is DTFT pair? 9, State the properties of Fourier Transform of a discrete time aperiodic sequence. 10. Find the Fourier transform of a sequence given by sin)=1, -25n52 =0, otherwise. 11. Find the Fourier transform of xin) = (0.8Y,n =0, 21,22 12. What is the frequency shifting property of Discrete Time Fourier Transform ? 13, What is the time shifting property of Discrete Time Fourier Transform ? 14, What is the linear property of DTFT ? 15, Find the transfer function of the 3-sample average : .. terms only ? 1 hi= 3G, -1snsl =0, otherwise.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.(6.2) The eqn. (6.2) further can be subdivided into an infinite number of summations, where each sum contain N terms. Thus, aN je + DY xe 8" 40. (6.3) mate n=m By changing the index in the inner summation from n to n—-mN, we obtain (i.e,, n=n-mN)ic., mN =0. om) ee 1h mts x(74)- X EY «a-mnye * maaan =O Xt arn = YL YLan-mme *" (since ct = 1) (6.4) m=n~ noo If we change the order of the summation, we obtain, * eal = _ i, x(n) n= S| ¥ xa-mnye * 46.5) a0 [mane ‘Qnk Sth where, X( “F°n | is replaced by X(4) for simplicity. We defined the signal, x,(n)= J) xin-mN) 46.8) which is obtained by periodic repetition of x(n) every N samples. The signal x,(n) is clearly periodic with fundamental period N. Since any periodic sequence can be expanded in a Fourier representation, therefore the signal x,(n) can also be expanded in a Fourier series as,4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Fourier Transform [ies] whenk=2, X(2)= 31+ cos x-jsin x-+c0s 2n-j sin 2] 1 e 1 oi =z il-1-jo+1-sl= 5-5 20. when k= 3, x@-2 in + ean jsindn] 1 1 og ll+04j-1-J0l= ‘The 4-point DFT sequence of x(n) is given by, lj-r 1 1 x 24120,- 22,2 20,2224, aie { 3° 2°38" 3 i} Magnitude function, lida | Xt) | =f. 333) not Phase function, 2 X(h) = fo. ~£,0,5} ‘The three sample sequence and its periodic extension are shown in Fig. (6.2) xin) ot 2 n Fig. 6.2. Three sample sequence. Periodic extension in) of x(n) for N= 4 “4S Ot 2 8 r Fig. 6.3 Note. £(n) is periodic extension of x(n). If x(n) is of length L and if N 2 L, then x(n) can be periodically extended 1X{0) |4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Fourier Transform [is7] Here we wish to determine the sequence x,(r:) for which the DFT is, XA = XR K A) B= O12 =D The IDFT of X,(h) is, ry Jish, 2) = > Xe * a 1M Lah, *y 2a Xp(ke % Using eqn. (6.19) and (6.20) in (6.21), we have, ¥ [Same E20 [m= Dest Nat ft =n) = ¥ xem) xq) Y, e/2ateom 8 mao ro ded The brackets term in eqn. (6.22) has the form, N for a=1 1-a® I-a where, we have defined ‘a’ as, a =62%™-"-1YN, See that for a = 1, the (n ~ m 1) must be integer multiple of N. MD gy No for a#1 io aN=e 1 ie., aN = 1 for any value ofa #0. Therefore, eqn. (6.23) reduces to, Ss ’ fs n-m-l=qN; [l= ~m~-qN)=((a-m))y 0 otherwise q an integer mm If we substitute the eqn. (6,24) into eqn, (6.23), we have, 1X Noa ayin= sD alm) Y xl IN; 12 (r= nig. im Net x,(n)= Y xn) x (a - my] mo m 20,1) ....N=1 (6.21) -(6.22) ++(6.23) + 46.24) (6.25) ‘The eqn. (6.25) has the form of convolution sum. However it differs from a linear convolution of x,(n) and x,(n) as defined in unit-IL. In linear convolution the computation of the sequence x,(n) involves multiplying one sequence by a folder and linear shifted version of the other and then summing the values of the product x,(m) x,(n — m) for all values of n Instead the convolution sum in eqn. (6.25), the second sequence is circularly time reversed and circularly shifted w-r. finite duration sequences. of first. The eqn. (6.25) is called circular convolution of two4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Fourier Transform [7] Der then TD 7 By ) = XG) YOR) where r,,(1) is the circular cross-correlation sequence, given as, 7, =D Aady*n- De ° Nol Proof. Since x) Ny*)= J) x(m) y*(U-m))y, moo then, rfl) = xd) Ny"—D RI =X xm y4 (= (=m) mo Net ry(D= > xm) yn - Dy nao ra(t) =x) NyM-D. We know circular convolution of the two sequences is just equal to the multiy the their DFTs and from complex conjugate property orr yD ms yh) ‘Then N-point DFT of r,, (0) is, R,, (1) = XG) YR). (10) Parseval’s theorem ; For the complex-valued sequence x(n) and y(n), if DFT x(n) > Xi) pyr and yin) a Yk). Not Not then (in) y*(n)= Ck) y * de) Proof. From the cbrrelation property, Net r= 2, x) y*n-Dy neo Nz then ry (OV= DY a(n) ¥*(n) 0 ie Bast and O= 5 z R, (be § (6.34) +-(8.35)4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Fourier Transform [75] eh am 0 Note. Since W,' #1 ie, e 4 #-Lifk=2then ¢ 4 = ~1 then Xk) = > (undeterminable) =1-W2+W,-Ws -CD+C1P-C ps4 Therefore solution of (5) is Kh) = 48(k — 2). (c) For N odd we=-1ir 22 or N=2k. Since N is odd, therefore no k exist ao we can write, X(h) = Wy (d) For N even, Wyt=-1 if k= N 2 Therefs a= IZCDS Oye ee N refore, rime br aed But for k = N/2. = No N=1 X(NR2) = yw = Yrenr- Eien. " neo ano Problem 7. Compute the DFT of sequence x(n) =|-2, 2, 1,-1). Ber wet Sol. Xk) = SY) xi) Wy = Ye WA", k= 0, 1,28. azo nao Xk) =~ 2420 + WW ote. ~ X(0)=-24241- X(1) =~2~2)-1-j=-3-3 X(2)=-2-2414+1=-2 X(3)=-249-1+j=-3+9. X(k) = (0,-3-9),-2,~3 + 3). Problem 8. Consider a finite length sequence, x(n) = 28(n) + 8(n = 1) + Bin - 3) we perform the following operation on this sequence, @ we first compute 5-point DFT X(h). (i) we compute 5-point IDFT of ¥(k) = (X(k)}* to obtain a sequence y(n). Determine the sequence y(n) for n = 0, 1, 2, 3, 4. 0 Wy =e") W227)4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Fourier Transform [7] or y= SY x0) ha-h) Ste = x(— 1) A(2) + x(0) ACL) + (1) ACO) + x(2) A{— 1) + x(3) Af{- 2) + ......... =14+4+3=8. when n=2 nate) x(k) 2 2 => ir (hi + op 1 2 3 : 0123 ‘ k Si or y(2y= Y, th) A(2-8) iene = x(~ 1) A(3) + 2(0) A(2) + x(1) A(1) + x(2) ACO) +268) A= 1) 4 x(— 4) B= 2) 4 oe =-1424641=8 when n=3 nf) x8) yeh) - 3 2 => 2 ity thy 2 3 4 , ot es ‘ oy 23 x a 2 or 33)= YH A3-2 aoe = x(— 1) A(A) + x(0) A(3) + x(1) A(2) 4+ x(2) A(L) + x(3) A(0) + x4) AD =-24+34+2=3.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Fourier Transform Problem 12. Perform circular convolution of the two sequences x, (a) =(2,1,2, 1) and x,(n)=(, 2, 3,4). Sol. Method 1: Graphical method of computing circular convolution The circular convolution of x,(1) and x,(n) is given by, Nat Nat 5 (m) = D 24(0) xq =n) = D200) 2m). ao aso where y(t) = x,(m ~ a). ‘The given sequences can be represented as points on a circle as shown below, the folded sequence x,(- n) is also represented on the circle. x(=4 w(t)=2 x,(2) = 2 x (0) = 2 4(2)=3. (0) = 1 4)=1 =4 nf) =4 x2) =3; (0) =1 x(1)=2 when m=0 Nz} a (0) J) 2100) x,mq (m= J) 2462) x2, 9(0) a io when m =1 Not 2 x)= Dm A-m= Yn (nx. ila), wo ery when m =2 Nz) x, (2)= pa ¥,(2-n) 2 saan)No 243) Dy x00) xy B-n)= x6) x2 60), a0 ao ‘The graphical method of computing each sample of x,(m) is shown below. when m =0 2 x0) = 2am Xp,9(”) x1) 0) 1 +) a) 1 2x3 0) no > Bg 7 *3) it) x(0)=2444+6+2=14 when m= 1 2 xD) = S xy(1) x9, y(n) fer) 1 1 2 2 x 4 2 =p 2x x2 =8 =4 7 3 x1) =4+148+3=16 when m =24@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.| > y(n) x3,4(n) nao 2 9 2 1 3, ° + x > 3 a 2 1 x,(4)=0+0-6+6-1=-1 whenm=4 = x,(4) a xm) = {8,-2,-1,-4,— 1). Matrix method ; Given x(n) = (1,-1,-2,3,-) and x4n)= (1,2, 3). By adding two zero (zero padding technique) to the sequence x,(n), we go x5{n) = (1, 2, 3, 0, 0). ‘The matrix form can be written by substituting N = 5, gO) xp(4) x:(8) xp(2) xq(D][x10)] [2900 xD x9(0) xq(4) x9(3) ¥_(2)|] x,(D| | xg(D xp(2) xg) xg(0) xg(4) 29(3)|| x12)|=|x5(2) 2,3) x22) x,(D x20) x2(4)||x,@)] | x9(3) (4) x9(8) x9(2) x(1) 2(0)}[2,4)] [x9(4) 100327 7 8 2100 3}j-1| |-2 : 3 210 Off-2]=|-1 032100] a} |-4 lo 0 3 2 ajl-3) [-1 “ x,{m) = (8,~2,—1,-4,— 1h. Problem 14. The input x(n) and impulse response of a LTI system are x(n) =(-1, 1,2,-2); h(n) =(0.5, 1,-1, 2, 0.75) tT Determine the response of the system. (a) Using linear convolution and (6) Using circular convolution. Sol. (a) Response of LTI system using linear convolution Let y(n) be the response of LTI system By convolution sum formula, yin) = xin) © hin) = J) xh) Alin) ‘The sequence x(n) starts at n = O and A(n) starts at n= — 1 Hence, y(n) will start at n = 0+ (- 1) =-1.Discrete Fourier Transform [793] ‘The length of x(n) is 4 and the length of h(n) is 5. Hence the length of y(n) is (4+ 5-1) =8. Also y(n) ends at n = 0 + (- 1) + (4+ 5-2) =6. Let us change the indexn toé in x(n) and ‘A(n). The sequences x(h) and h(&) are represented on the tabular array as shown below. k -4|-3| -2]-1] 0 1 2 3 4 5 | 6 7 th) -1 1 2 -2 ik) 0.5 1 -1 2 0.75, 1 AER) om] 2)]-1' 1 | os | A(-1-k) =h_h) | 0.75) 2 -1 1 05 | Nk) = hy kd om} 2/-1] 1 | 05 ~ AO) = hy) 07} 2 {-1] 1 | 05 R2=k) =hsk) 075) 2 -1 1 05 A(3— k) = A, fk) 0.75 2 -1 1 Os hd ~ hb) = he) 07} 2 | -1; 1 | 05 AUS h) = Ag(k) 0.75 2 -1 L Os AIG — hk) = Ah) 0.75 2 bt 1 05 ie, yin) = > 20h) f,, 00) a“ a whenns-1 y-D= >, x4, aera = x 4) A_{— 4) +200 3) A y(- 3) +2 2)h_(- 2) 4x DAD + {0)A_,(0) + x01) h_,(L) + 22) h_,(2) + 28) h_(3). =0+04040+4(-0.5)+0+0+0=-05 2 when n=0 H0)= Y) xh) dig(k) =04040+4(-1)+0.5+0+0=-05. 3 when n=1 d= Dx) Ah) = 0404141414023. oH 2 when n= 2 2) =D) 2) p(B) = 04 (24-424 2-2, ish “ when n =3 (8) =D) x(k) hy(h) = 0.75 + 24-2) +(-2)40=-2.75 rer) 5 whenn=4 — (4)= > 2th) hy(h) =0 40,75 444240402675 ms4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.end of x,(n). The first sample of x,(n) is placed as overlapping sample at thaend of x,(n). The first sample of x,(n) is placed as overlapping sample at the end of x,(n). Singp there is no fifth section, the overlapping sample of z,(n) is taken as zero xn)=1; n=0 | x{n)=2jn=2 x a) =453n=6 =-1jn=1 -2;n=38 = 44;n=7 2; n=2 Bjnz4 j= 05 n =8. Now perform circular convolution of each section with A(n). Here A(n) starts at n =n, =0 | x,(n) starts at n 0; y(n) will start atn =n, +n,=0+0=0 | =,(n) starts atn =n, = 2; y,(n) will start ata =n, +n, =24+0=2 | x, (n) starts at n =n; = 4; y(n) will start atn =n, +n, =44+0=4 x, {n) starts atn =n, = 6; y,(n) will start atn =n, +n,=6+0=6 Convolution of section 1: n -2 f -1 0 1 zt], xn) 1 f-1 21] ) hin) -1 1 ° H acon foe po poe te poe =n) = hyn) of; a]-2 ° |i | hd=n)= hn) 0 1 | -1 [| yy(m) = x(n) @ hin) Nat = J xl) him =n) 5m =0,1,2 a0 \ Not = Y 0) im ; m= 0, 1,2. mee 2 when m = 0 90) = Y x\(n) hg in) =- 1404221 x0 2 when m=1 y= DY xi) ya) = 1 1022 nao when m =2 y@= Yin) oy) = 0-1-2=-3 msDiscrete Fourier Transform [203] Convolution of section 2: n -2]-1] 0] 1/2] 3] 4 xin) 2|-2| 3 | in) -1[ 1] 0 he-ny=hiny | o | 1 [-1 A(2—n) = h(n) o| 1f-1] 0 AG —n) = hen) o| 1f-1] 0 ACA —n) = hyn) o| 1|-1 ym) = x,(n) @ h(n) xea = DY x2(n) hom =n) Ferd “ “ ym) = Y, zaln) bmn) = Y x90) hig) m= 2,3, 4. at nee whenm=2 — ,(2)= D) xpn) igi) =-24+0+3=1 when m =3 948) =D) xaln) hyln) = 24240=4 when m = 4 94) =D) x9(n) h(n) = 0-2-3 =-5. Convolution of section 3 : n ~2] -1 off 2] 3] «[s] 6 x(n) 3 |-3 4 Ain) ~1 1 o l= n) = hol) of a1] -1 MA ~n)= hn) ; oo ja NG —n)= hyn) ToL Pot a f-i ay [ Re —ny= hws { | 0 1 fn yet yok) = x(n) @ hin) = Y, x90) him —n) ano s ys) =D) x(n) hyn) m= 4,5, 6 ase whenm=4 —y,(4)= x(n) hgin) =-3 4044014@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Fourier Transform [207] when m=4 94) = Sxslm) hylm) = 3 + 0~ whenm=5 ——94(5) = D)xs(m) hsm) = 343406. when m = 6 ¥6) =F x30) hglm) = 0-3 4 2= Convolution of section 4: n -2{-1} of 1}2{;3a}4/5] 6] 7] 8 xn) “| ~[ 4|-4|-3 hin) -1}1]o | (=n) = hg) o} 1} -2 | | MG ~n) = hein) o | ai-1; o| 1] AT =n) = hfnd ee) LB =n) = hin) i fo} apna > x4(0) h(n) ,m =6,7,8 xa din) =F, x4) gl a a 7 whenm=6 — y,(6)= So x4(n) hg(n) =-4 40-32 whenm=7 — ¥(7)= Dt) ya) =-440-3=-7 whenm=8 —,(8)= Lain agin) =0-44+32-1 ‘To combine the output of the convolution of each section : It can be observed that the last sample is an output sequence overlaps with first sample of next output sequence. In overlap save method the overall output is obtained by combining the outputs of the convolution of all section. While combining the outputs the overlapped last. sample of every output sequence is discarded and the remaining samples are simply saved of y(n) as shown in the following table. m ofif2]fsaitafsle6fa ia nim) | -2 ym) yg) 3 yam) -7)8 5 yon) | 1 -7 | 6 [+ Mm) = {- 1, 2,-3, 4, -5,6,-7, 8, *}.x(n) = gin) + jh(n), ‘The DFTs GA) and Hl) ean be calculated from the N-point DFT ¥(k) df the sequence x(n) by means of eqns., | 1 Gtk) = g XUh) + X— By) i = myx HOR) = 5 (RR) —X* (- Wy) wea Now vay= ¥ om wit aed Nal Nat = DY oan) wit + ¥ uaa + wir * neo aso Nel Nal = & atm wat + De) wat Wat nao aso Not Noa = D ain) wi + Why Shiny Wat mo m0 Note that, the first sum on the last expression is simply an N-point PFT Gh) of length N sequence gin), whereas the second sum is an N-point DFT H(k) of lengtl-N sequence h(n). ‘Therefore we can express the 2N-point DFT Vik) as, Vik) = Gli] + Wie Hy], 0S $ 2N— 1 Problem 17. Determine the 8-point DFT Vik) of the length 8 real sequence v{n) given below vf) = 12,220.10 t Sol. We form two length-4 real sequence, gin) = (2n) = {u(2 x 0), v(2 x 1), v(2 x 2), (2 3)) = (v0), v(2), v(4), v(6)} an) =11,2,0,1)| Osnsa. | h(n) = v(2n + 1) = (WD, v2 x 1+ 1), (2 x24 V, (2% 34 DI = (u(2), 0(8), 0(6), 7) A(n) = (2, 2, 1, 1) Osns3 But Vik) = Gilk)yl + Wy HIG)x], 05k $2N—-14@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete The direct computation of eqn. (6.44) requires, (1) 4N real multiplications for each value of k. (2) (4N — 2) real additions for each value of h. (3) 4N? real multiplications for N values of k. (4) NAN ~ 2) real additions for N values of k. We have seen from the above discussion that amount of computations, and thus the computation time, is approximately proportional to N?, itis evident that number of arithmetic operations required to compute the DFT by the direct method becomes very large for large values of N. Most approaches, for improving the efficiency of the computation of DFT, exploits the symmetry and periodicity property of Wy" ie., Wy #82) 3 — Wack symmetry property Wyt 8 = Wy! periodicity property. Exploit of these two basic property results in computational efficient algorithms which are collectively known as Fast Fourier Transform Algorithms. FFT algorithms based on the fundamental principle of decomposing the discrete fourier transform of a sequence of length N into the successively smaller discrete fourier transforms. ‘The manner in which this principle is implemented leads to a variety of different algorithms, all which comparable improvement in computation speed. ‘There are two basic classes of FFT algorithms. (1) The first, called decimation in time, derives its name from the fact that in the process of arranging the DFT, into the smaller DFTs, the sequence x(n) (time sequence) is decomposed into successively smaller subsequences, hence it is called as “Decimation in time FFT”, (2) In second class of algorithms, the sequence of DFT co-efficients X(k) (frequency response) is decomposed into smaller subsequences, hence it is called as “Decimation in frequency FFT”. 6.8.2 Radix of FFT Algorithms In an N-point sequence, if N can be expressed as N = r”, then the sequence can be decimated into r-point sequences. For each r-point sequence, r-point DFT can be computed. From the result of r-point DFT, the r?-point DFTs are computed. From the results of r?-point DFTs, the r>-point of DFTs are computed and so on, until we get r™ point DFT. In computing N-point DFT by this method the number of stages of computation will be m-times, The number ‘”’ is called the radix of the FFT algorithms. 6.8.3 Radix-2 Algorithm In radix-2 FFT the N-point sequence is decimated into 2-point sequences and 2 point DFT for each decimated sequence is computed. From the results of 2 point DFTs, the 4-point DFTs can be computed. From the result of 4-point DFT, the 8 point DFT can be computed and 50 on, until we get N-point DFT. For performing radix-2 FFT, the value of N should be such that, N = 2". Here the decimation can be performed m times when m = log, N. In radix-2 FFT, the total number of complex additions are reduced to N log, N and total number of complex multiplications are reduced to (N/2) log, N. TransformNumber Direct computation Radix-2 FFT of Complex Coraptex Complex points addition | multiplication addition mbttiplications N NW-D Ne Nlog,N { | toy N | 4 4-@4 12 16 4x log, 22= 8 Flow, 2?= 4 8 8= (2) 56 64 8 x log, 29 = 24 2 le 2"= 12 16 = (24) 240 256 16 x log, 24 = 64 = log, 2 = 82 ‘ 82 82 = (25) 992 1,024 2 x log, 2° = 160 @ Pelee 2 = 80 64 = (28) 4,032 4,096 64 * log, 2° = 384 & bx log, 25 = 192 12g 128-2) | 16,256 16, 384 128 x log 427 = 896 x log, 27 = 448 6.9 DECIMATION-IN-TIME FFT ALGORITHMS Consider a sequence x(n) of length N. The N is assumed to be powe where m is integer (Radix 2 FFT). Since N is an even integer, we can consider compu by decomposing +(n) into two N/2 point sequences consisting of the even number point in x(n) and the odd number point in x(n) with X(k) given by, Noi XH) = ZA) We = 0,1, 2, cee NT 46.45) a Separating x(n) into even and odd numbered points, we obtai Xe) = YF x) Wy"* +S x6) Wy (6.46) neven nod or with substitution of variable n = 2m for n even and n = (2m + 1) for n is odd, we have Nast XO)= SY xam) Wy? + $ x(2m. + 1) Wy??? 6.47) er) mao Nast Nast XO) = YF Am Wy? + Wy? DY fal) Wy? | +X6.48) m=0 m=0 where, f,() has elements x(0), x(2), x(4),. and ,(m) has elements x(1), x(3), (5) ... +. = x(2m), = 2(2m + 1).4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Fourier Transform [7] ‘The decomposition represented by eqn. (6.56) and (6.57) are shown in Fig. (6.11) x(0) »@) x) (6) x) x{3) x65) 7) Fig. 6.11 Having performed the decimation in time once, we can repeat the process for each of the sequences f,(m) and f,(m). Thus f,(m) would result in the two N/4 point sequence and f,(m) would result in another two N/4 point sequences. Let the decimated N/4 point sequences of f,(n) be V,,(m) and V(r). V, (2) = f,(2m) : for m = 0, 1, 2, ~(E-1) (6.58) 4 and Vm) = f,(2m + 1) for m= 0, 1, 2, i i 1) 46.59) 4 Let the decimated x point sequences of f,(n) be V,,(n) and V,,(n). Vom) =f{2m) for m= 0,1,2, (6,60) and V,,(m) = f,(2m +1) form =0, 1,2, . -1) (6.61) }Discrete Fourier Transform [219] F,(8) = V2)(8) + Wyo? Voo(3). Here, Vj s(en) = f(2m) = 214m). or V(r) = f(2n) = x(4n) ‘The final decimation-in-time representation for N = 8 as shown in Fig. (6.12) x(0) Vii0) FO) X00) _ Ni4-point xia) OFT xq) C. ° +x(2) x2) ° * tw7a-point x0) OFT x) cs . « We w,? xi) xe) ° - ‘ N/4-point My, x5) ber x(5) * . e Wa xi3) x(6) ° + * Nia-point wy 0) OFT x, - : —* Wy Fig. 6.12 6.10 THE 8-POINT DFT USING RADIX-2 DIT FFT ‘The computation of 8-point DFT using radix-2 FFT, involves three stages of computation. Here N = 8 = 2°, therefore r = 2 and m = 3. The given 8-point sequence is decimated to 2-point sequences. For each 2-point sequence, the 2-point DFT is computed. From the result of 2-point DFT the 4-point DFT can be computed. From the result of 4-point DFT, the 8-point DFT can be computed. Let the given sequence be X(0), X(1), X(2), X(3), X(4), X(5), X(6), X(7} which consists of 8 samples.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Discrete Fourier Transform [ 23 ] when k =0 F,{0) = Vp4(0) + Wyn? Vpa(0) whenk=1 F, (1) = Vay) + Wyyg? Vaal) when k = 2 F 62) = Voi(2) + Wyyn? Vog(2) = Vgy(0) — Wye? Vng(0) when k =8 F,(3) = V21(3) + Wyo? Vq(3) = V2,(1) — Wyyg! Vea). ‘The third stage of computation : In the third stage of computations the 8-point DFTs are computed using the result of second stage as inputs. Let X(h) = DFT {x (n)). The 8-point DFT of x(n) can be computed as follows : X(k) = Fk) + Wy! Fy(k), for k= 0, 1, 2,3, 4,5, 6, and 7. X(0) = F,(0) + W,° F,(0) XCD =F) + Wy! FD X(2) = F,(2) + Wy? F,(2) X() = F\(3) + Wy! F,(3) X(4) = F\(4) + Wy! F,(4) = F,(0) — Wy? F(0) X(5) = F,(5) + Wy? F,(5) = F,(D) - Wy! F,(1) X(6) = F(6) + Wy° F,(6) = F,(2) — Wy? F,(2) X(7) = F(7) + Wy! F() = F (3) - Wy? F,(3) 6.10.1 Flow Graph for 8-Point DIT Radix-2 FFT If we observe the basic computation performed at every stage, we can arrive at the following conclusion : (i) Tn each computation two complex numbers a and b are considered. (ii) The complex number 4 is multiplied by a phase factor W,*. (ii) The product 5Wy" is added to the complex number a to form new complex number A. (i) The product bW," is subtracted from complex numbera to form new complex number The above basie computation can be expressed by a signal flow graph shown in Fig. (6.14) a Asa+bMye Dw 74 Fig. 6.14. Signal flow graph or Butterfly diagram.Butterfly diagram for 8-point DFT : The sequence is arranged in bit reversed order and then decimat sequence as (3) x7) x(4) (6) (5) ‘The first stage of computation : Hl V,,(0) = x(0) + Wyy.° (4) | V11(1) = x(0) = Wy? x(4) | ‘V,2{0) = 2(2) + Wyyq? x(6) ‘Vq1(0) = #1) + Way? (5) ‘Voy(1) = 201) — Wyyy? x5) Voq(0) = 28) + Way? (7) Vyq(1) = 28) — Wyy,? (7) x(0) | (2) | xD (0) Y——pv,,(0) x) We x(a) « + + Vin(t) x(6) ¢ 4 : x) V2i(0) x): 4 Ww, x(5) i +; Vai(t) x) Fig. 6.15 Second stage of computation : F,(0) = V,1(0) + When? Vjo (0) F,(1) = Vy) + Wygn? Vig() 4 1 50 F 0) FC) F (2) F,2) F (3) Fig. 6.164@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.6.11 DECIMATION IN FREQUENCY (DIF) RADIX-2 FFT In decimation in frequency algorithm the frequency domain sequencp X(z) is decimated. In this method, the output DFT sequence X(k) is divided into smafler and smaller sequence. Step 1. Assuming that N = 2". We separate the index set into two spts. In this case the first sot consists of the integers 0, #22) Wan, b= 0,1,......N2=1 (6.74) a Step 3. The G,(k) is N/2 point sequence and it can be decimated into two numbers of ‘N/4 point sequences. Hence the eqn. (6.73) can be modified as shown below. wast 6,@)= Ya) Wyn ao4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.whenn=1, g,(1)= bt) -205)1W,! when n fx(2) - 2(6)]W,? when a =3, (3) -x(7))W,3. The samples of the sequences g,(n) and g,(n) are obtained by the bytterfly operation shown in Fig. (6.20). 9,(0) at) 9112) 93) Fig. 6.20 Second stage of computation : Let us define 4 numbers of 2-point sequences from the samples ofg,(n} and g,(n). The 4- numbers of 2-point sequences are obtained by dj,(n), dyin), dyy(n) and d,4(n) and they are obtained using eqns. (6.80) and (6.81). d.,(n) = gn) + a(n + N/A). din) = ain) +90 +2) for n= 0,1. dyn) = ley) — 2,0 + 2) Wyy_" for n= 0,1. when n=0,d,(0)=g,(0)+ 2,2) ; d,(0) = le,(0)-¢,2W when n=1,di(1)=4,01) +8) 5 dy(1) = le(D-¢,(3)1W,! From the eqn. (6.81) y(n) = g,(n) + gyn + N/4) for n=0,1. din) = Gln) —4,(n + N/A)} Wyg" for n= 0,1. when n= 0, — dyy(0)= lgg(0) +4621; day (0) = le (0) — (21° whenn=1, — dyy(1) = lay(1) +803; dg (1) = Wgy(1)—24(3)1,. 8y4(0) ay) 9yA0) 1)4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.13. “4 15. 16. 7. 18. REVIEW QUESTIONS List any four properties of DFT. List out the properties of DFT. Assume two finite duration sequences x,(n) and x,(n) are linearly combined. Le| bx,(n). What is the DFT of x(n)? IEN-point sequence x(n) has N-point DFT X(k) then what is the DFT of the fol What is zero padding ? What are its uses ? ‘What do you understand by periodic convolution ? Define circular convolution. How the circular convolution is obtained 7 Distinguish between linear and circular convolution of two sequences ? ‘State the differences between (i) overlap-save method (ii) overlap-add method Why FFT is neoded ? What is FFT? How many multiplications and additions are required to compute N-point DFT using radix-2 FFT? What is meant by radix-2 FFT? What is a decimation-in-time algorithm ? What is decimation-in-frequency algorithm 7 ‘What are the differences and similarities between DIF and DIF algorithms 7 Draw the basic butterfly diagram for DIT algorithm. ‘What is meant by ‘in-place’ in DIT and DIF algorithms ? . x,00) = ax(n) + ing? In an LTT system the input x(n) = (2, 2, 1) and the impulse response h(n) the response of LITI system by radix - 2 DIT FFT. 2 FFT. The impulse response of the system is h(n) = Determine the response of LITI system by radix—2 FFT. The input x(n) = (1.5, 9, 1) and impulse response h(n) = (1.5, 0.5). Draw the signal flow graph of decimation-in-frequeney algorithm for N = 16. Compute the DPT of the following sequences (a) x(n) = 0, 1,-1,-1) [Ans Xik) = (9, 2-27, 0,2 + 2/1 t (oye N= D ()xin)zan OSnSN-Discrete Fourier Transform [23] 9 10. IL | Na (Wt sinl 2sinh Xe sh NB i Ke = nsN- Ans. (© xin) =cashan OSnSN-1 | i amyFaahe ‘Show that : kel kaa i 2 @) in) P= =D) 1XUe) wer { (a dl mp as 2! i Given the two sequences x,(n) = a" and x,(n) = B* of length 4, determine the circular convolution. (Ans. x,(0) = 1+ of? + 098? + a°B ;x,(1) =B + @ + 0°87 + a 8? =,(2) = B® + of + 0? + oP; x,(3) = B+ af? + o®B + of") ‘Compute the circular convolution of two sequences using stockhams method x(a) = (2, 1,2, 1) Kiln = Dy + Xn + BD) t x,(m) = (1, 2, 3, 4) fAms. x,(n) = (14, 16, 14, 16)] = Consider a FIR fitter with impulse response hin) = 2" 0 xT) 8¢-nT) A738) which is a sequence of impulses at ¢ = nT with weight (nT). ‘Thus x,(t) can be considered as a continuous time representation of the discrete-time sequence x(nT). Apply fourier transform to eqn. (7.3) yields, Feo = 1 Y xanae- | = Y sen ree-n7) ae we have, Flat—nl= [3e-nTy™ at wT AD PI&t —nT)] = est (7.5) Thus, Xe) =F, (l= YD enMe™ AT.) . i< we have, a= DL, We aes yen AT Qn where 2,2. Substituting eqn. (7.7) in eqn. (7.1), we obtain, so-x0] ye # Y xemow Now, Frtol= 2 3 Fixe) (7.8) If Fiz(¢)] = X(jq) then, Plx(t) e/"9) = X4(Q— may) 79)Substituting eqn. (7.9) in eqn. (7.8) yields, ly P Fil = gD, XU(A- mo, ™ Phe,(¢)] = +z x{fo a 2)| | (7.10) replicas of the Fourier transform of the original signal. Now consider the signal x(t) is band limited to/,,. That is the highes4 frequency component of x(t) is f,. Then, x§Q)=0 for | A] >Q, (0, = 2xf,) is the shifting of X(/Q) fromp2 = 0 toa = a 2x mb 0, #1, £2. Temp Oe In ogn. (7.10) the term X (i (a = Hence X, (/) is the sum of shifted replicas of xa) centering at ‘The Fig. [7.5(a)] shows the plot of xe for different value of &. x0) Fig. 7.5 (a) We see that itt >Q,,, the replicas will not overlap as in Fig. (7.5(6)] and asa result the mR frequency spectrum of TX,UQ) in frequency range (- r 4) is identical to XUp). The frequency spectrum X(jf2) can be recovered from X,(j2) by using a low pass filter whicl| has sharp cut off atOe = ‘The same explanation holds for = =2,.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Substituting eqn. (7.23) is eqn. (7.21) yields £,()= xT) tat nT)-ue-(n+ DD) ing Laplace transform on both sides, Lig, @1= Ys TL (ule -nT)-ue-(2 + DDI X= Sar - ety aaneet 228 5 neo x@ 6 ‘aie The output of zero order hold consists of higher order harmonics. monics the output of zero order hold is applied to a low pass filter as sha This filter is often called as a smoothing filter because it tends to smoot} step approximations generated by ZOH. Problem 1.A signa! having a spectrum ranging from near dc to 10 and converted to discrete form. What is the minimum number of samples p be taken to ensure recovery ? Sol. Given fy = 10 KH, ensure recovery is, f= %p 20,000 samples/sec. Plot the amplitude spectrum for | Q | $30 x. Can the original signal cak be recovered from ‘samples ? Why or why not ? Sol. Given xt) = 10 cos (10 xt). XQ) = Pixé)] = LOfn({Q + 10x) + HQ — 10n)]]} ie, Floos Q, ¢] = £18(2 +2,) + {A -2,)]. ‘The amplitude spectrum of x(¢) is shown in figure 10n x9) 10m.Digital Processing of Continuous Signals Ea The sampling rate, f= 8Hz, Q, = Oxf, = 2n(8) = 16% 2, = Bf, = 100 fra SH, Nyquist rate = 2f,, = 10 Hz. ‘The sampling rate is less than Nyquist rate. So, the original signal cannot be recovered from the samples. ‘The frequency spectra of sampled x(¢) is given by, < “ 2x xum=d Y x[:(0-22)] [ i X,GQ) = 2 DY 10101600 + 10m — n>) + #180 - 102 - 29, )1) an 1 where, O=a and Faf=8 X, G0) = 8 J, LOR|8( + 10m — 16nn) + BCA - 10 16nn)} 80m 5, [8+ 10m — 16nn) + 5(0~ 10n ~ 316n2)] ‘The plot of amplitude spectrum for | @ | < 30x is shown in figure. ‘Aliased component x02) ‘Bon ean =10x Gn ox (10n Zax (26s Problem 3. A signal x(t) = sinc (150nt) is sampled at a rate of (a) 100 Hz (6) 200 Hz (c) 300 Hz. For each of these three cases, explain if you can recover the signal from the sampled signal. Sol. Given x{é) = sine (150 xt). ‘The spectrum of the signalx(t) is a rectangular pulse = with a band width (maximum frequency component) of te 150x rad/sec ia shown in figure. a4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Fig. 7.18. Location of poles in s-plane. Butterworth polynomials for various values of N for 2. = 1 r/s. Table 7.1. List of Butterworth polynomials Denominator of H(s) se s+ V2541 GrD@ern (s? + 0.76537s + 1) (s? + 1.84778 + (s + 1) (6? + 0.61803s + 1) (6? + 1.618035 + 1) (s+ 1, 9918553 + 1) (s? + Vs + 1)(s? + 0.517648 + 1) (5 + 1) (6? + 1.801945 + 1) (6? + 1.2475 + 1) (6? + 0.4455 wa ween a) As we are interested on the poles, which lies in the left half of the s- be found by using the formula, ane the same can K=1,2,...N. A741) known as normalised poles. In general, the unnormalised poles are given b 5, =O, 5. The transfer function of such type of Butterworth filter can be obtain s + 8/Q, in the transfer function of Butterworth filter is shown in table. The two parameters completely characterizing a Butterworth filter bre therefore the 3. dB cut off frequency Q, and the order N. These are determined from the specified passband A742) by substituting edge ©, and minimum passband magnitude —_1 sy+ the stop band edge l+e’ ee mum stop band ripple — (or A yew4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Digital Processing of Continuous Signals [265] Problem 6. Given the specifications o,, = 1 dB ; a, = 30 dB, Q, = 2007/8, Q, = 600 r/s. Determine the order of the filter. Sol. Nz keA_ Tog (WR) 218, To find A: As [ee To find k: log (8) Rounding off N to the next integer, we get, N=4 apt Bias fo" — jn Sol. The magnitude square function of Butterworth analog lowpass filter is given by, WHUDDF = 1{ 2 (1) * oy 1 (2) Problem 7. Prove that 0, = We know, [HYa)?? = 2/2 Le (2) Comparing in eqn. (1) and (2), we get as ‘ Q 2(Q 7 — =1. eas (BY (2) § @(2) (2) Q, a.) ° Simplifying the above eqn. by substituting z= 10-1, we obtain4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Digital Processing of Continuous Signals [z7] Low pass-to-band pass transformation. Consider a band pass filter with lower band edge frequency Q, and upper band edge frequency Q,,. The transformation for converting lowpass analog filter, with band edge frequency 9, into bandpass filte can be accomplished by first converting the lowpass filter into another lowpass filter with band edge frequency 9,’ = 1 and then performing transformation, 2.9 go ht 1.87) ‘a, - 21) we can also obtain the same result in a single step by means of the transformation 2 5? +0, 9, 830, ———L—s (7.88) Oe 0, -2,) : Thus we have of7,89) ‘Low pass-to-band stop transformation. To convert a lowpass analog filter with band edge frequency Q, , into the band stop filter, the tansformation is simply the inverse of eqn. (7.87) with addition factor, serving to normalised for the band edge frequency of the lowpass filter. Thus we have $2, spe 8) (7.90) 740, 2, which gives H,, (8)=H, » Gee wAT.90) +2, A All the above four transformations are summarised in Table (7.2) Table 7.2. Frequency transformations for analog filter Filter type Transformation Band edge frequency of new filter Low pass se, a, a,’ High pass > 228 a, 4 no, 242, 9, Band pass 80, £28, Se 2,2, ‘(Q, ~ Q) Band stop 899, Fa a. 9,9, Problem 9. Transform the single pole lowpass Butterworth filter with system function, Oe. s+9, into a high pass filter with band edge frequency Q,/ and band stop filter with upper and lower band edge frequencies Q, and Q, respectively. H,{s) =4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Q Go" _ pan ~ guN Qin _O1n «= (882) ~ 0.939 H(s) for Q, = 0.106 x can be obtained by substituting S > We know, a= = 0.106" o ci0ex ™ BY). His) = 7 A a aa} . 1 paso7 * 2489+ a te Problem 13. Odtain an analog Chebyshev filter transfer fuction that straints 1 Je s|Hom|s1 ;osase | HGQ) | <01 ;Q24. Sol. From the given data we can find that 1 ia Vive ve 1 20.1 Vise 0.707}----» Q,=2 and 9,=4. from which weean obtaine=1 and 4=9.95. We know, Haga) 1 foeh ee cosh” 995 _ 9 269 = 3, N2 “cosh? 2 0.4] | 2 [eee ~(24u + _g | (241ay¥? +(2a1ay rf] fama b= 2.0874@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Digital Processing of Continuous Signal | followed by a reconstruction filter. The basic idea behind the most commonly used D/A converter can be explained by means of the simplified block diagram representation shown in Fig. 7.30, where we have assumed, without any loss of generality, that the digital sample is positive and represented in a natural binary code. Here the!" switch S, is in its ON position if the! binary bita, = 1, and it is in the OFF position ifa, = 0. The output V, of the D/A converter is then given x Vou Y 2h aVp (7.93) Ve MSE " tse Fig. 7.80. Block diagram representation of an N-bit DVA converter. ‘There are a variety of D/A converters that are used in signal processing applications, We discuss below only the following types : (1) Weighted-resistor D/A converter (2) resistor- ladder D/A converter and (3) Over sampling D/A converter. 7.17.1 Weighted-Resistor D/A Converter ‘The schematic of an N-bit weighted-resistor D/A converter is shown in Fig. 7.31. Here the operation of the switches are as shown in Fig. (7.31). It can be shown that the output V, of the D/A converter is given simply by e R, -¥ 2-14 {— Bi Vom 22a Cw DR, i} ist The full scale output voltage V, ps is obtained when all a,‘s are ONEs. Then from Eq. (7.94) (7.94) q Vo.re* (oe ti) +795) In practice, usually (2% ~ 1)R, >> 1 and as a result, V, »,= Vp Usually a buffer amplifier is placed at the output to provide gain and prevent loading. For a D/A converter with a moderate to high resolution, the spread of the resistor values becomes very large, making this type of converter unsuitable for many applications. Based on the same principle as discussed above, a weighted capacitor D/A converter can be designed. Such circuits are more popular in IC design.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Digital Filter Structures 8.1 INTRODUCTION In this chapter, we consider the realization problem of causal IIR and FIR transfer functions and outline realization methods based on both the time-domain and the transform- domain representations. Here, we describe the mast commonly employed methods to implement the digital filter structure from either its difference equation, its unit sample response, or its z-transform. A structural representation using interconnected basic building blocks is the first step in the hardware or software implementation of an LTI digital filter. The strue- tural representation provides the relations between some pertinent internal variable with the input and the output that in turn provide the keys to the implementation, There are various forms of the structural representations of a digital filter. We review in this chapter two such representations, and then describe some popular schemes for the realization of real causal IIR and FIR digital filters. ‘The digital filter structure determine directly from either the difference equation or the system function is called the direct form-I. An alternative view of the same equation results in the memory efficient structure, called the Direct Form-II. Digital filter structures as cascade, parallel and lattice structures are shown to have the advantages in terms of hardware implementation. 8.2 SYSTEM DESCRIBING EQUATIONS ‘The equation that describe the input and output relationship, in the time and z-transform domain, has been defined in the previous chapters. They are repeated here for reference. ‘The linear time invariant system is described by the difference equation of form, x, x. yin) =— Day n-B)+ D) by xin -) 28.1) ms ms or equivalently, x, yr a, yn -k) = ¥ by x(n -k) +(B.2) it mo where a, and 6, are constants with ag = 1. As we have seen by mean of z-transform such a class of LTI discrete-time systems are also characterized by rational system function. 2954@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.‘This realisation requires (M + N) additions, (M + N + 1) multiplicatipn memory locations. nd (M+N +1) M ‘Therefore, u(n) = )) 6, x(n- 4) non recursive system Fe) 8.13) N xn) = - Y 04 ¥n—h) +n) recursive system (8.14) m If all pole system H,(z) eqn. (8.14) is placed before all zero syst more compact structure is obtained. This structure is known as direct shown in Fig. 8.5 for M = N. This structure is cascade of recursive (all xan) win) yin) nf H(z) eqn. (8.13) a TI realizations m. tye bial © ©) [2] Fig. 8.5. Direct form realization (N = M). Here only a single delay line or a single set of memory locations i the past values of w(n). This structure requires (M + N + 1) multiplic and maximum of (M,N) memory locations. Because direct form II re i mum number of delays for the realization of the system, it is also called ds “Canonic form”. Problem 2. Find the digital network in direct and transposed forfn for the system described by the difference equation y(n) = x(n) + 0.8 x(n - 1) +0.4 x{n - 2) - 0.6y (n- 1 - OF y(n ~2). Sol. Given that, n) = x(n) + 0.5 x(n ~ 1) + 0.4 an ~ 2)— 0.6y(n — 1) - O.Py(n'- 2) on taking z-transform, we get, Y(e) = X(e) + 0.5 27! Xz) + 0.42" Xz) 0.621 Vig) 0.74* Vz) ——..(8.15) ‘The direct form digital network can be realised using equation (8.15}as shown in Fig. 8.6. eu4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Digital Filter Desig 9.1 INTRODUCTION The term filter is commonly used to describe a device that discrimingtes, according to some attribute of the objecte applied as its input, what passes through it. Forfexample, a coffee filter, an oil filter, a photographer's filter or an electronic filter. All these filthrs intended to in some way purify or modify the substance it filters. Now the design of digital filter is often carried out completely independently of analog systems. In the design of frequency selective filter, the desired filter characteribtics are given in the frequency domain in terms of the desired magnitude and phase responde of the filter. In developing the digital filter, we determine the co-efficients of a causal FIR of IR that closely approximately the desired frequency response specifications. 9.2 SELECTION OF THE FILTER TYPE The selection of the digital filter type ie., whether an IIR and FIR digital filter to be requirement for a linear phase characteristic within the abel of the phase is not a requirement, either an ITR or FIR filter can be used. However, order (Npj,) of an FIR filter is considerably higher than the order (Nyjp) of filter meeting the same magnitude specifications. It has been shown that filter specifications, the ratio Nrp/Nnp is typically of order of ten or more anfl as a result, IIR filter is usually computationally more efficient. In this chapter we shall discuss techniques for designing IIR filters| filters, with the restriction that the filters be realizable and, of course, stablf. There are four different methods which are available under IIR filter design, these are, 1. Impulse invariance method 2. Bilinear transformation method 3. Matched z-transform technique 4. Approximation of derivatives. ‘We shall concentrate only the first two methods. 9.2.1 BR Filter Design by Impulse Invariance ‘A technique for digitizing an analog filter is called impulse invariancefransformation. ‘The objective of this method is to develop an IIR filter transfer function whose ifapulse response 3204@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.[Qerestelsne gepen pe lene (Qeos o - I+ j sin |] (Qcos a+ )- jsino eos a+ + jsino || Qesar)- jsina a? +g 20sinw ee nt fat 1407 +22c08@ ~ 14+Q? +20.cos.@ a a? -1 Jw A] S}to therefore, s (9,22) Q- Toews (9.23) Let us consider the mapping. Case L If Q < 1, then o < 0i.e., LHP in s-plane maps into the interior of the unit circle inz-plane. Case Il. If Q > 1, then > 0 ie., RHP in s-plane maps into the exterior of unit circle in z-plane. Case Ii. If Q = 1, then o = 0i.e,, the imaginary axis in the s-plane maps onto the unit circle in2-plane. The mapping of thes-plane into thez-plane via the bilinear transformation is illustrated in Fig. (9.1) and is seen to have all the desired properties, also there is no aliasing due to one-to-one mapping. Fig. 9.1. Bilinear transformation mapping. Now for the case III, we have r = 1,0 =0, and 2 2sin o2cos w2 2 cos” ov2 (9.24) (9.25) ar -2t0 (2) or © ntl The relationship in eqn. (9.24) has been plotted in Fig. (9.2). The eqn. (9.26) shows the relation between the analog frequency and digital frequency. It is clear from the plot that positive imaginary axis (0) in s-plane is mapped into the upper half of the unit circle in the z-plane and negative imaginary axis (— jQ) in the s-plane is mapped into lower half of the z-plane.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Sampling this function produces ~eat, afer" cos(bnT,), nn 20. inT,) { .. \erwise. The z-transform of A(nT,), is equal to He) = ¥ e°" cos (b,T,) 2" ase He) > te cos 7,271)" aso . 1-e°*? cos (6T,) He) = Gar hg A py Problem 5. Convert the analog filter with system function. 8+01 ‘ _ ani, Hs) = oF 7g into digital UR filter by means of bilinear ation. Reso- nant frequency of a digital filter is given as w, = - ‘Sol. (i) We first note that the analog filter H,(s) has a resonant freq OQ, = J16 =4. (i) Let us find T (iti) Now map ia 1-2? =4 } (ES By substituting values of s into H(s), we have, Hi) = Hehe af4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.H,(s) for Q, = 2 rad/sec can be obtained by substituting s+ 9/2in Hs) itt)s ——2-—__- 4. _ fey oS) = (ai)? + f2(e2) +1 9? 42828844" By using bilinear transformation H(z) can be obtained as He)= HO), Thus H@)= < 4a+27)? 4(1- 2°")? + 2.828 (1-27) +4014 271? _ 0.292914 271)? “T+ 0171 . ‘The above system function can be realized in direct form Il as sl in Fig. (9.6). x(a) 1 0.2929 O77 ~ 04 (6) Impulse invariance method. The relationship between andlog and digital frequencies in impulse invariance method is = OT. Form the given data T = 1 sec, ie. @ = Q. => Q, = 0, ;2,=0,. We know = 4898,e=1 log Me ‘The order of the filter, N2 —i- The transfer function of a fourth order normalised Butterworth filfer is HUG) = GF, 076587640? + 184776+D4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.4@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.Time: 12. 18. B. ‘Tech. DEGREE EXAMINATION Fifth semester, November/December-2003 Computer Science and Engineering Three hours Maximum : 100 marks Answer All Questions. . PARTA (10 x 2 = 20 marks) Consider an LSI system of A(n) = (0.5)" u(n). Verify the system for stability. State the initial and final value of theorems of z-transform. Compute Fourier transform of the signal x(n) = u(n)— u(n — 1) and comment on the result. |. Write the relationship betweon system function and the frequency response of LTI system. Give the computation efficiency of 1024 point FFT over 1024 point DFT. -i80 5. Let Hiei) =) > OS1O1S@. 15 it PIR (or) IIR filter ? , @, Sfalsn What are Wiener Filters ? What are the two demerits of Impulse-invariant technique ? |. Why the filters designed for Multi-rate systems are named poly-phase filters ? |. State Wiener-Khintchine theorem. PARTB (6 « 16 = 80 marks) |. Consider an LT! system y(n) = 0.5y(n — 1) + x(n). (i) Determine the step response of the system. (6) (ii) Determine the transient and steady state state response of the system when the 2 input is x(n) = 10 cos (22) u(n). The system is initially at rest. (10) (a) (i) State and prove the Parseval's theorem of Fourier Transform. (6) Gi) Determine Fourier Transform for the following signals. ao) CQ) num) (2) cos wy n u(n). Or () A.LTT system is described by the following difference equation yn) = ayln - 1) + bx(n), (é) Determine the Magnitude and phase response of the system. (8) (éi) Choose the parameter ‘b’ so that the maximum value of | He") | is unity. (8) (@) Derive cight point radix-2 decimation in Time FFT algorithm and compute DFT for xin)= 15 9S"S7 | using Decimation in Time FFT algorithm. ao +6) 0, otherwise’ 3414@ ‘You have either reached 3 page that is unavailable for viewing or reached your viewing limit for this book.About the Book This book deals with the analysis of Digital Signal Pro and precise style. There are about 200 solved problems apart fi This book covers the latest syllabus prescribed by the for Electrical and Electronics engineering students. Exerci review questions are included at the end of each chapter. All th should make this book extremely valuable for engineering st for Anna University examinations as well as for practicing engin: About the Author C. Ramesh Babu Durai graduated from Arulmigu Kalasali of Engineering, Srivilliputhur and did his post graduate studies at Hindustan College of Engineering, Chennai. He is a faculty member of the department of Electrical and Electronics Engineering, Hindustan College of Engineering, Chennai. He has more than six years of teaching experience and the college has honoured him by conferring on him “The Best Teacher Award”. His field of interest includes Control Systems, Advanced Digital Signal Ptocessing and Electromagnetic theory. 700% i LAXMI PUBLICATIONS (P) LTD UU

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