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 About the Author
A. Nagoor Kani is a multifaceted personality with an efficient technical expertise and management skills.
He obtained his BE in EEE from Thiagarajar College of Engineering, Madurai and MS (Electronics and
Control) through Distance Learning program of BITS, Pilani.

He started his career as a self-employed industrialist (1986-1989) and then changed his career to teaching
in 1989. He has worked as lecturer in Dr MGR Engineering College (1989-1990) and as Asst. Professor
in Satyabhama Engineering College (1990-1997). The author started his own coaching centre for BE
students named Institute of Electrical Engineering which was renamed as RBA Tutorials in 2005. The
author started his own companies in 1997. The companies currenly run by him are RBA Engineering
(manufacturing of lab equipments and microprocessor trainer kits), RBA Innovations (involved in developing
projects for engineering students and industries), RBA Tutorials (conducting coaching classes for engineering
and GATE students) and RBA Publications (publishing of engineering books.) His optimistic and innovative
ideas brought up RBA GROUP successfully.

He is an eminent writer and till now he has authored nine engineering books, and his books are very
popular among engineering students. He is known by name through his books in all engineering colleges
in south India and in some colleges in north India.
hsqs„ev2ƒsqxev
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Second Edition

A. Nagoor Kani
Founder, RBA Educational Group
Chennai

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 iv

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v

Dedicated to

My sister : Mrs. A. Rajina Bivi, MA, B.Ed

Brother-in-law : Dr. A. Kalilur Rahman, MBBS, MS(ORTHO)

Their daughter : Er. K. Shajina, BE

Their son : K. Shafiq, (MBBS)

 Contents
Preface......................................................................................................................................................... xviii
Acknowledgement.......................................................................................................................................... xxi
List of Symbols and Abbreviations....................................................................................................................xxiii

Chapter 1 : Introduction to Digital Signal Processing

1.1 Introduction ......................................................................................................................................................... 1. 1

1.2 Signal ................................................................................................................................................................. 1. 2
1.3 Discrete Time System ........................................................................................................................................ 1. 2
1.4 Analysis of Discrete Time System ...................................................................................................................... 1. 3
1.5 Filters ................................................................................................................................................................. 1. 5
1.6 Finite Word Length Effects ................................................................................................................................... 1. 5
1.7 Multirate DSP ..................................................................................................................................................... 1. 6
1.8 Energy and Power Spectrum ............................................................................................................................. 1. 6
1.9 Digital Signal Processors .................................................................................................................................... 1. 6
1.10 Importance of Digital Signal Processing ............................................................................................................... 1. 7
1.11 Use of MATLAB in Digital Signal Processing ...................................................................................................... 1. 8

Chapter 2 : Discrete Time Signals and Systems

2.1 Introduction.................................................................................................................................................2. 1
2.2 Discrete Time Signals................................................................................................................................2. 3
2.2.1 Generation of Discrete Time Signals................................................................................................2. 3
2.2.2 Representation of Discrete Time Signals...........................................................................................2. 4
2.2.3 Standard Discrete Time Signals......................................................................................................2. 5
2.3 Sampling of Continuous Time (Analog) Signals..............................................................................................2. 8
2.3.1 Sampling and Aliasing.................................................................................................................. 2. 8
2.4 Classification of Discrete Time Signals..............................................................................................................2.12
2.4.1 Deterministic and Nondeterministic Signals............................................................................................2.12
2.4.2 Periodic and Aperiodic Signals.............................................................................................................2.12
2.4.3 Symmetric (Even) and Antisymmetric (Odd) Signals.............................................................................2.15
2.4.4 Energy and Power Signals.................................................................................................................2.17
2.4.5 Causal, Noncausal and Anticausal Signals...........................................................................................2.19
2.5 Mathematical Operations on Discrete Time Signals............................................................................................2.20
2.5.1 Scaling of Discrete Time Signals........................................................................................................2.20
 viii

2.5.2 Folding (or Reflection or Transpose) of Discrete Time Signals..................................................................2.21

2.5.3 Time Shifting of Discrete Time Signals............................................................................................2. 22
2.5.4 Addition of Discrete Time Signals...................................................................................................2. 23
2.5.5 Multiplication of Discrete Time Signals.............................................................................................2. 23
2.6 Discrete Time System.............................................................................................................................2. 23
2.6.1 Mathematical Equation Governing Discrete Time System................................................................2. 24
2.6.2 Block Diagram and Signal Flow Graph Representation of Discrete Time System................................2. 25
2.7 Response of LTI Discrete Time System in Time Domain.................................................................................2. 28
2.7.1 Zero-Input Response or Homogeneous Solution...............................................................................2. 29
2.7.2 Particular Solution..........................................................................................................................2. 30
2.7.3 Zero-State Response...................................................................................................................2. 31
2.7.4 Total Response...........................................................................................................................2. 31
2.8 Classification of Discrete Time Systems......................................................................................................2. 35
2.8.1 Static and Dynamic Systems.........................................................................................................2. 35
2.8.2 Time Invariant and Time Variant Systems........................................................................................2. 36
2.8.3 Linear and Nonlinear Systems.......................................................................................................2. 39
2.8.4 Causal and Noncausal Systems................................................................................................... 2. 45
2.8.5 Stable and Unstable Systems.......................................................................................................2. 47
2.8.6 FIR and IIR Systems................................................................................................................2. 50
2.8.7 Recursive and Nonrecursive Systems..........................................................................................2. 51
2.9 Discrete or Linear Convolution...................................................................................................................2. 51
2.9.1 Representation of Discrete Time Signal as Summation of Impulses................................................... 2. 52
2.9.2 Response of LTI Discrete Time System using Discrete Convolution...................................................2. 53
2.9.3 Properties of Linear Convolution..................................................................................................2. 54
2.9.4 Interconnections of Discrete Time Systems.................................................................................... 2. 56
2.9.5 Methods of Performing Linear Convolution......................................................................................2. 61
2.10 Circular Convolution................................................................................................................................2. 68
2.10.1 Circular Representation and Circular Shift of Discrete Time Signal.................................................... 2. 68
2.10.2 Circular Symmetrics of Discrete Time Signal................................................................................. 2. 70
2.10.3 Definition of Circular Convolution...................................................................................................2. 70
2.10.4 Procedure for Evaluating Circular Convolution................................................................................2. 70
2.10.5 Linear Convolution via Circular Convolution................................................................................... 2. 72
2.10.6 Methods of Computing Circular Convolution...................................................................................2. 72
2.11 Sectioned Convolution..............................................................................................................................2. 81
2.11.1 Overlap Add Method...................................................................................................................2. 82
2.11.2 Overlap Save Method.................................................................................................................2. 82
ix

2.12 Inverse System and Deconvolution...........................................................................................................2. 96

2.12.1 Inverse System.........................................................................................................................2. 96
2.12.2 Deconvolution............................................................................................................................2. 97
2.13 Correlation, Crosscorrelation and Autocorrelation..........................................................................................2. 99
2.13.1 Procedure for Evaluating Correlation................................................................................................. 2.100
2.14 Circular Correlation........................................................... ........................................................................... 2.107
2.14.1 Procedure for Evaluating Circular Correlation....................................................................................... 2.108
2.14.2 Methods of Computing Circular Correlation..........................................................................................2.109
2.15 Summary of Important Concepts.....................................................................................................................2.113
2.16 Short Questions and Answers.........................................................................................................................2.114
2.17 MATLAB Programs...................................................................................................................................... 2.118
2.18 Exercises..................................................................................................................................................... 2.122

Chapter 3 : Z - Transform

3.1 Introduction..............................................................................................................................................3. 1
3.2 Region of Convergence............................................................................................................................3. 4
3.3 Properties of Z-Transform...........................................................................................................................3. 11
3.4 Poles and Zeros of Rational Function of z....................................................................................................3. 27
3.4.1 Representation of Poles and Zeros in z-plane..................................................................................3. 28
3.4.2 ROC of Rational Function of z......................................................................................................3. 29
3.4.3 Properties of ROC......................................................................................................................3. 30
3.5 Inverse Z-Transform...................................................................................................................................3. 31
3.5.1 Inverse Z-Transform by Contour Integration or Residue Method......................................................... 3. 31
3.5.2 Inverse Z-Transform by Partial Fraction Expansion Method..............................................................3. 32
3.5.3 Inverse Z-Transform by Power Series Expansion Method................................................................3. 35
3.6 Analysis of LTI Discrete Time System Using Z-Transform.............................................................................. 3. 48
3.6.1 Transfer Function of LTI Discrete Time System.................................................................................3. 48
3.6.2 Impulse Response and Transfer Function........................................................................................3. 49
3.6.3 Response of LTI Discrete Time System Using Z-Transform................................................................3. 49
3.6.4 Convolution and Deconvolution Using Z-Transform..........................................................................3. 50
3.6.5 Stability in z-Domain.......................................................................................................................3. 51
3.7 Relation between Laplace Transform and Z-Transform...................................................................................3. 56
3.7.1 Impulse Train Sampling of Continuous Time Signal...........................................................................3. 56
3.7.2 Transformation From Laplace Transform to Z-Transform................................................................... 3. 57
3.7.3 Relation Between s-Plane and z-Plane...........................................................................................3. 57
 x

3.8 Structures for Realization of LTI Discrete Time Systems in z-Domain...............................................................3. 72

3.9 Structures for Realization of IIR Systems.....................................................................................................3. 74
3.9.1 Direct form-I Structure of IIR System..............................................................................................3. 75
3.9.2 Direct form-II Structure of IIR System.............................................................................................3. 76
3.9.3 Cascade form realization of IIR System...........................................................................................3. 78
3.9.4 Parallel form Realization of IIR System..........................................................................................3. 79
3.10 Structures for Realization of FIR Systems...................................................................................................3. 99
3.10.1 Direct form Realization of FIR System...........................................................................................3. 100
3.10.2 Cascade form Realization of FIR System.......................................................................................3. 100
3.10.3 Linear Phase Realization of FIR System.......................................................................................3. 101
3.11 Summary of Important Concepts...................................................................................................................3. 107
3.12 Short Questions and Answers..................................................................................................................3.109
3.13 MATLAB Programs....................................................................................................................................3. 118
3.14 Exercises..............................................................................................................................................3. 123

Chapter 4 : Fourier Series and Fourier Transform of Discrete Time Signals

4.1 Introduction............................................................................................................................................4. 1
4.2 Fourier Series of Discrete Time Signals (Discrete Time Fourier Series)..........................................................4. 2
4.2.1 Frequency Spectrum of Periodic Discrete Time Signals..................................................................4. 3
4.2.2 Properties of Discrete Time Fourier Series.................................................................................... 4. 4
4.3 Fourier Transform of Discrete Time Signals (Discrete Time Fourier Transform)................................................4. 9
4.3.1 Development of Discrete Time Fourier Transform from Discrete Time Fourier Series............................4. 9
4.3.2 Definition of Discrete Time Fourier Transform..................................................................................4. 10
4.3.3 Frequency Spectrum of Discrete Time Signal.................................................................................4. 11
4.3.4 Inverse Discrete Time Fourier Transform.......................................................................................4. 12
4.3.5 Comparison of Fourier Transform of Discrete and Continuous Time Signals..........................................4. 12
4.4 Properties of Discrete Time Fourier Transform..............................................................................................4. 13
4.5 Discrete Time Fourier Transform of Periodic Discrete Time Signals............................................................... 4. 20
4.6 Analysis of LTI Discrete Time System Using Discrete Time Fourier Transform................................................4. 22
4.6.1 Transfer Function of LTI Discrete Time System in Frequency Domain.............................................. 4. 22
4.6.2 Response of LTI Discrete Time System Using Discrete Time Fourier Transform...................................4. 23
4.6.3 Frequency Response of LTI Discrete Time System.........................................................................4. 23
4.6.4 Frequency Response of First Order Discrete Time System...............................................................4. 25
4.6.5 Frequency Response of Second Order Discrete Time System.........................................................4. 31
xi

4.7 Aliasing in Frequency Spectrum Due to Sampling.......................................................................................4. 36

4.7.1 Signal Reconstruction ( Recovery of Continuous Time Signal )........................................................4. 38
4.7.2 Sampling of Bandpass signal.......................................................................................................4. 39
4.8 Relation Between Z-Transform and Discrete Time Fourier Transform.............................................................4. 40
4.9 Summary of Important Concepts.............................................................................................................. 4. 63
4.10 Short Questions and Answers...................................................................................................................4. 64
4.11 MATLAB Programs...................................................................................................................................4. 69
4.12 Exercises...............................................................................................................................................4. 74

Chapter 5 : Discrete Fourier Transform (DFT) and Fast Fourier Transform

(FFT)

5.1 Introduction ....................................................................................................................................................... 5. 1

5.2 Discrete Fourier Transform (DFT) of Discrete Time Signal ................................................................................. 5. 2
5.2.1 Development of DFT From DTFT ....................................................................................................... 5. 2
5.2.2 Definition of Discrete Fourier Transform (DFT) ...................................................................................... 5. 2
5.2.3 Frequency Spectrum using DFT ......................................................................................................... 5. 3
5.2.4 Inverse DFT ....................................................................................................................................... 5. 3
5.3 Properties of DFT ............................................................................................................................................. 5. 4
5.4 Relation Between DFT and Z-Transform ............................................................................................................ 5. 9
5.5 Analysis of LTI Discrete Time Systems using DFT ............................................................................................ 5. 10
5.6 Fast Fourier Transform (FFT) ........................................................................................................................... 5. 19
5.7 Decimation In Time (DIT) Radix-2 FFT ............................................................................................................. 5. 22
5.7.1 8-Point DFT using Radix-2 DIT FFT .................................................................................................. 5. 24
5.7.2 Flow Graph for 8-Point DFT using Radix-2 DIT FFT ............................................................................ 5. 27
5.8 Decimation In Frequency (DIF) Radix-2 FFT ................................................................................................... 5. 29
5.8.1 8-Point DFT using Radix-2 DIF FFT ................................................................................................. 5. 32
5.8.2 Flow Graph for 8-Point DFT using Radix-2 DIF FFT ........................................................................... 5. 35
5.8.3 Comparison of DIT and DIF Radix-2 FFT ............................................................................................ 5. 37
5.9 Computation of inverse DFT Using FFT ............................................................................................................ 5. 37
5.10 Summary of Important Concepts ....................................................................................................................... 5. 53
5.11 Short Questions and Answers ........................................................................................................................... 5. 54
5.12 MATLAB Programs .......................................................................................................................................... 5. 58
5.13 Exercises ......................................................................................................................................................... 5. 63
 xii

Chapter 6 : FIR Filters

6.1 Introduction ....................................................................................................................................................... 6. 1

6.2 LTI system as Frequency Selective Filters ........................................................................................................ 6. 2
6.3 Ideal Frequency Response of Linear Phase FIR Filters ..................................................................................... 6. 4
6.4 Characteristics of FIR Filters with Linear Phase ................................................................................................ 6. 6
6.5 Frequency Response of Linear Phase FIR filter ................................................................................................. 6. 8
6.6 Design Techniques for Linear Phase FIR Filters ................................................................................................. 6. 20
6.7 Fourier Series Method of FIR Filter Design ........................................................................................................ 6. 22
6.8 Windows .......................................................................................................................................................... 6. 40
6.8.1 Rectangular Window ........................................................................................................................... 6. 41
6.8.2 Bartlett or Triangular Window ............................................................................................................... 6. 43
6.8.3 Raised Cosine Window ...................................................................................................................... 6. 44
6.8.4 Hanning Window ................................................................................................................................ 6. 46
6.8.5 Hamming Window .............................................................................................................................. 6. 47
6.8.6 Blackman Window .............................................................................................................................. 6. 49
6.8.7 Kaiser Window ................................................................................................................................... 6. 50
6.8.8 Summary of Various Features of Windows .......................................................................................... 6. 54
6.9 FIR Filter Design Using Windows ..................................................................................................................... 6. 54
6.10 Design of FIR Filters by Frequency Sampling Technique ................................................................................... 6. 79
6.11 Summary of Important Concepts ....................................................................................................................... 6. 85
6.12 Short Questions and Answers ........................................................................................................................... 6. 86
6.13 MATLAB Programs .......................................................................................................................................... 6. 95
6.14 Exercises ......................................................................................................................................................... 6. 107

Chapter 7 : IIR Filters

7.1 Introduction ....................................................................................................................................................... 7. 1

7.2 Frequency response of analog and digital IIR filters ............................................................................................ 7. 3
7.3 Impulse invariant transformation ........................................................................................................................ 7. 6
7.3.1 Relation between analog and digital filter poles in impulse invariant transformation ................................. 7. 7
7.3.2 Relation between analog and digital frequency in impulse invariant transformation ................................. 7. 8
7.3.3 Useful impulse invariant transformation ................................................................................................ 7. 9
7.4 Bilinear transformation ....................................................................................................................................... 7. 15
xiii

7.4.1 Relation between analog and digital filter poles in bilinear transformation ................................................ 7. 16
7.4.2 Relation between analog and digital frequency in bilinear transformation ................................................ 7. 17
7.5 Specifications of digital IIR lowpass filter ............................................................................................................ 7. 24
7.6 Design of lowpass digital Butterworth filter ......................................................................................................... 7. 27
7.6.1 Analog Butterworth filter ....................................................................................................................... 7. 27
7.6.2 Poles of Butterworth lowpass filter ....................................................................................................... 7. 28
7.6.3 Transfer function of analog Butterworth lowpass filter ............................................................................ 7. 35
7.6.4 Frequency response of analog lowpass Butterworth filter ..................................................................... 7. 37
7.6.5 Order of the lowpass Butterworth filter ................................................................................................. 7. 37
7.6.6 Cutoff frequency of lowpass Butterworth filter ....................................................................................... 7. 37
7.6.7 Design procedure for lowpass digital Butterworth IIR filter ..................................................................... 7. 38
7.7 Design of lowpass digital Chebyshev filter ........................................................................................................ 7. 40
7.7.1 Transfer function of analog Chebyshev lowpass filter ........................................................................... 7. 42
7.7.2 Order of analog lowpass Chebyshev filter ........................................................................................... 7. 43
7.7.3 Cutoff frequency of analog lowpass Chebyshev filter ........................................................................... 7. 44
7.7.4 Frequency response of analog Chebyshev lowpass filter .................................................................... 7. 44
7.7.5 Design procedure for lowpass digital Chebyshev IIR filter ................................................................... 7. 45
7.8 Frequency transformation .................................................................................................................................. 7. 47
7.8.1 Analog frequency transformation .......................................................................................................... 7. 47
7.8.2 Digital frequency transformation ........................................................................................................... 7. 48
7.9 Summary of Important Concepts ....................................................................................................................... 7. 109
7.10 Short Questions and Answers ........................................................................................................................... 7. 111
7.11 MATLAB Programs .......................................................................................................................................... 7. 121
7.12 Exercises ......................................................................................................................................................... 7. 141

Chapter 8 : Finite Word Length Effects in Digital Filters

8.1 Introduction ....................................................................................................................................................... 8. 1

8.2 Representation of Numbers in Digital System .................................................................................................... 8. 2
8.2.1 Binary Codes ..................................................................................................................................... 8. 2
8.2.2 Radix Number System ....................................................................................................................... 8. 3
8.2.3 Fixed Point Representation .................................................................................................................. 8. 5
8.2.4 Floating Point Representation ............................................................................................................... 8. 9
8.3 Types of Arithmetic in Digital Systems ............................................................................................................... 8. 12
8.3.1 One's Complement Addition ................................................................................................................ 8. 12
 xiv

8.3.2 Two's Complement Addition ................................................................................................................ 8. 13

8.3.3 Floating Point Addition ......................................................................................................................... 8. 14
8.3.4 Floating point Multiplication...........................................................................................................8. 15
8.3.5 Comparison of Fixed Point and Floating Point Arithmetic ...................................................................... 8. 16
8.4 Quantization by Truncation and Rounding .......................................................................................................... 8. 16
8.4.1 Quantization Steps .............................................................................................................................. 8. 16
8.4.2 Truncation ........................................................................................................................................... 8. 18
8.4.3 Rounding ............................................................................................................................................ 8. 21
8.5 Quantization of Input Data .................................................................................................................................. 8. 22
8.6 Quantization of Filter Cofficients ......................................................................................................................... 8. 29
8.7 Product Quantization Error ................................................................................................................................ 8. 37
8.8 Limit Cycles in Recursive Systems ................................................................................................................. 8. 53
8.8.1 Zero Input Limit Cycle ......................................................................................................................... 8. 53
8.8.2 Overflow Limit Cycle .......................................................................................................................... 8. 61
8.8.3 Scaling To Prevent Overflow .............................................................................................................. 8. 62
8.9 Summary of Important Concepts ....................................................................................................................... 8. 73
8.10 Short Questions and Answers ........................................................................................................................... 8. 76
8.11 Exercises ......................................................................................................................................................... 8. 82

Chapter 9 : Multirate DSP

9.1 Introduction ....................................................................................................................................................... 9. 1

9.2 Downsampling (or Decimation) .......................................................................................................................... 9. 2
9.2.1 Spectrum of downsampler ................................................................................................................... 9. 4
9.2.2 Anti-aliasing Filter ................................................................................................................................ 9. 7
9.3 Upsampling (or Interpolation) ............................................................................................................................. 9. 16
9.3.1 Spectrum of Upsampler ...................................................................................................................... 9. 19
9.3.2 Anti-imaging Filter ............................................................................................................................... 9. 20
9.4 Sampling Rate Conversion ............................................................................................................................... 9. 24
9.4.1 Spectrum of Sampling Rate Convertor by a Rational Factor I/D .......................................................... 9. 25
9.5 Multistage implementation of Sampling Rate Conversion ................................................................................... 9. 26
9.6 Identifier in Multirate Digital Signal Processing ................................................................................................... 9. 27
9.7 Implementation of Sampling Rate Conversion in FIR Filters ............................................................................... 9. 32
9.7.1 Implementation of Sampling Rate Conversion using Decimator in FIR Filters ....................................... 9. 32
9.7.2 Implementation of Sampling Rate Conversion using Interpolator in FIR Filters ...................................... 9. 33
xv

9.8 Polyphase Decomposition ................................................................................................................................ 9. 34

9.8.1 Polyphase Decomposition of FIR Filters .............................................................................................. 9. 34
9.8.2 Polyphase Structure of Decimator ....................................................................................................... 9. 37
9.8.3 Polyphase Structure of Interpolator ...................................................................................................... 9. 38
9.8.4 Polyphase Decomposition of IIR Filters ............................................................................................... 9. 40
9.9 Applications of Multirate DSP ............................................................................................................................ 9. 46
9.9.1 Digital Filter Banks .............................................................................................................................. 9. 46
9.9.2 Sub-band Coding of Speech Signals ................................................................................................... 9. 46
9.9.3 Quadrature Mirror Filter (QMF) Bank .................................................................................................. 9. 47
9.10 Summary of Important Concepts ....................................................................................................................... 9. 48
9.11 Short Questions and Answers ........................................................................................................................... 9. 49
9.12 MATLAB Programs .......................................................................................................................................... 9. 53
9.13 Exercises ......................................................................................................................................................... 9. 60

Chapter 10 : Energy and Power Spectrum Estimation

10.1 Introduction ....................................................................................................................................................... 10. 1

10.2 Energy Spectrum of Discrete Time Signal ......................................................................................................... 10. 1
10.3 Random signal and Random Process ............................................................................................................... 10. 4
10.4 Power Spectrum of Random Process ............................................................................................................... 10. 5
10.5 Periodogram ..................................................................................................................................................... 10. 5
10.6 Use of DFT/FFT in Power Spectrum Estimation ................................................................................................ 10. 6
10.7 Nonparametric Methods of Power Spectrum Estimation .................................................................................... 10. 7
10.7.1 Bartlett Method of Power Spectrum Estimation .................................................................................... 10. 7
10.7.2 Welch Method of Power Spectrum Estimation ..................................................................................... 10. 8
10.7.3 Blackman-Tukey Method of Power Spectrum Estimation ..................................................................... 10. 10
10.8 Performance Characteristics of Nonparametric Methods of Power Spectrum Estimation .................................... 10. 23
10.8.1 Performance Characteristics of Periodogram Power Spectrum Estimation ........................................... 10. 24
10.8.2 Performance Characteristics of Bartlett Power Spectrum Estimation .................................................... 10. 26
10.8.3 Performance Characteristics of Welch Power Spectrum Estimation ..................................................... 10. 28
10.8.4 Performance Characteristics of Blackman-Tukey Power Spectrum Estimation ..................................... 10. 29
10.9 Summary of Important Concepts ....................................................................................................................... 10. 33
10.10 Short Questions and Answers ........................................................................................................................... 10. 34
10.11 MATLAB Programs .......................................................................................................................................... 10. 36
10.12 Exercises ......................................................................................................................................................... 10. 43
 xvi

Chapter 11 : Digital Signal Processors

11.1 Introduction ....................................................................................................................................................... 11. 1

11.2 Special Features of Digital Signal Processors .................................................................................................... 11. 3
11.2.1 Fast Data Access ............................................................................................................................... 11. 3
11.2.2 Fast Computation ................................................................................................................................ 11. 5
11.2.3 Numerical Fidelity ............................................................................................................................... 11. 7
11.2.4 Fast Execution Control ....................................................................................................................... 11. 8
11.3 TMS320C5x Family of Digital Signal Processors ............................................................................................. 11. 8
11.3.1 Pin Diagram of TMS320C5x Processors ............................................................................................ 11. 10
11.3.2 Architecture of TMS320C5x Processors ............................................................................................. 11. 14
11.3.3 Functional Units in CPU of TMS320C5x Processors ......................................................................... 11. 15
11.3.4 On-Chip Memory in TMS320C5x Processors .................................................................................... 11. 19
11.3.5 On-Chip Peripherals of TMS320C5x Processors ................................................................................ 11. 19
11.3.6 Addressing Modes of TMS320C5x Processors .................................................................................. 11. 21
11.3.7 Instruction Pipelining in TMS320C5x Processors ................................................................................ 11. 23
11.3.8 Instructions of TMS320C5x Processors .............................................................................................. 11. 24
11.3.9 Assembly Language Programs in TMS320C5x Processors ............................................................... 11. 35
11.4 TMS320C54x Family of Digital Signal Processors ............................................................................................ 11. 44
11.4.1 Pin Diagram of TMS320C54x Processors .......................................................................................... 11. 47
11.4.2 Architecture of TMS320C54x Processors ........................................................................................... 11. 50
11.4.3 Functional Units in CPU of TMS320C54x Processors ....................................................................... 11. 51
11.4.4 On-Chip Memory in TMS320C54x Processors .................................................................................. 11. 56
11.4.5 On-Chip Peripherals of TMS320C54x Processors .............................................................................. 11. 57
11.4.6 Addressing Modes of TMS320C54x Processors ................................................................................ 11. 60
11.4.7 Instruction Pipelining in TMS320C54x Processors .............................................................................. 11. 63
11.4.8 Instructions of TMS320C54x Processors ............................................................................................ 11. 63
11.4.9 Assembly Language Programs in TMS320C54x Processors .............................................................. 11. 72
11.5 Summary of Important Concepts ....................................................................................................................... 11. 76
11.6 Short Questions and Answers ........................................................................................................................... 11. 78
11.7 Exercises ......................................................................................................................................................... 11. 82
xvii

Chapter 12 : Applications of DSP

12.1 Introduction ....................................................................................................................................................... 12. 1

12.2 Speech Processing .......................................................................................................................................... 12. 2
12.2.1 Speech Coding and Decoding ............................................................................................................ 12. 2
12.2.2 Speech recognition .............................................................................................................................. 12. 4
12.2.3 Speech Synthesis .............................................................................................................................. 12. 5
12.2.4 Digital Vocoder ................................................................................................................................... 12. 6
12.3 Musical Sound Processing ............................................................................................................................... 12. 6
12.3.1 Digital Music synthesis ...................................................................................................................... 12. 7
12.3.2 Musical Sound Processing For Recording .......................................................................................... 12. 7
12.4 Digital Radio ..................................................................................................................................................... 12. 8
12.5 Digital Television .............................................................................................................................................. 12. 9
12.6 DTMF in Telephone Dialing ............................................................................................................................... 12. 9
12.7 RADAR ........................................................................................................................................................... 12. 10
12.8 Biomedical Signal Processing ........................................................................................................................... 12. 11

Appendix 1 Important Mathematical Relations ........................................................................................................ A. 1

Appendix 2 MATLAB Commands and Functions ................................................................................................... A. 5
Appendix 3 Summary of Various Standard Transform Pairs ................................................................................... A. 11
Appendix 4 Summary of Properties of Various Transforms ..................................................................................... A. 14
Appendix 5 Summary of Important Equations for FIR Filter Design ......................................................................... A. 19
Appendix 6 Summary of Important Equations for IIR Filter Design .......................................................................... A. 23
Appendix 7 Summary of Properties of Power Spectrum Estimator ......................................................................... A. 24
INDEX ........................................................................................................................................................... I. 1
 Preface
The main objective of this book is to explore the basic concepts of digital signal processing in a simple
and easy-to-understand manner.
This text on digital signal processing has been crafted and designed to meet student’s requirements.
Considering the highly mathematical nature of this subject, more emphasis has been given on the
problem- solving methodology. Considerable effort has been made to elucidate mathematical derivations
in a step-by-step manner. Exercise problems with varied difficulty levels are given in the text to help
students get an intuitive grasp on the subject.
This book with its lucid writing style and germane pedagogical features will prove to be a master text
for engineering students and practitioners.
Salient Features
The salient features of this book on Digital Signal Processing are,
- proof of properties of transforms are clearly highlighted by shaded boxes
- wherever required, problems are solved in multiple methods
- additional explanations for solutions and proofs are provided in separate boxes
- different types of fonts are used for text, proof and solved problems for better clarity
- keywords are highlighted by bold, italic fonts
Organization
In this book, the concepts of discrete time signals and their transforms are organized in four chapters
and two chapters are devoted for digital filter design. One chapter is devoted to each topic in digital
signal processing like finite word length effects, mutirate DSP, spectrum analysis, digital signal processors
and applications of DSP. Each chapter provides the foundations and practical implications with a large
number of solved numerical examples for better understanding.
The important concepts are summarized at the end of each chapter which can help in quick reference.
Another significant aspect of this book is MATLAB based computer exercises with complete explanations
given in each chapter. This will be of great assistance to both instructors and students.
Chapter 1 deals with a general introduction about various aspects of digital signal processing and its
importance in real life. Basic definitions of discrete time signals and systems, mathematical representation
of discrete time systems and significance of time and frequency domain analysis are presented in
brief. Introduction to various topics of digital signal processing like FIR filters, IIR filters, finite word
length effects, multirate DSP, power spectrum, digital signal processors, applications of digital signal
processing and usage of MATLAB in this course are also presented in a brief manner.
Chapter 2 is devoted to concepts of discrete time signals and systems and is more concerned with
generation, representation, classification, mathematical operations of discrete time signals and systems,
block diagram and signal flow graph notations.
xix

The chapter also presents the methods of obtaining responses of LTI discrete time systems and various
convolution methods. The deconvolution, correlation techniques and the inverse systems are clearly
explained with solved numericals. In addition, the concept of sampling and its importance are dealt with
briefly.
Chapter 3 explains Z-transform and its application to discrete time signals and systems. All the important
properties of Z-transform are presented explicitly. Inverse Z-transform and solution of difference
equations describing the discrete time systems are demonstrated with numerical examples. Also, the
structures for realization of IIR and FIR systems are provided.
Chapter 4 is dedicated to discrete time Fourier series and Fourier transform which forms the basics
for frequency domain analysis of discrete time signals and systems. In the first half of this chapter, the
discrete time Fourier series and the frequency spectrum using discrete time Fourier series are discussed
with relevant examples.
The second half of the chapter details the development of discrete time Fourier transform from discrete
time Fourier series, frequency spectrum, various properties of Fourier transform, and Fourier transform
of some standard discrete time signals. In addition, the computation of frequency responses of LTI
discrete time systems using Fourier transform are also explained with examples. The relation between
Fourier transform and Z-transform of discrete time signals is also discussed in the chapter.
Chapter 5 extends the understanding of the concepts of Discrete time Fourier transform(DTFT) to
DFT (Discrete Fourier transform) and FFT (Fast Fourier Transform). Development of DFT from
DTFT, properties of DFT, relation between DFT and Z-transform, analysis of the LTI systems using
DFT and FFT are extensively discussed.
Chapter 6 focuses on frequency response of FIR filters and characteristics various windows used
for FIR filter design. Also, design of linear phase FIR filters by windowing and frequency sampling
techniques are presented with suitable examples.
Chapter 7 explains the techniques for transforming analog filter to digital filter and the characteristics
of analog Butterworth and Chebyshev filters. Also, design of Butterworth and Chebyshev digital IIR
filters are presented with examples.
Chapter 8 discusses the quantization and representation of digital/binary number systems. The effects
due to finite precision of filter coefficients and products, and various types of overflow in recursive
computations are also discussed with appropriate examples.
Chapter 9 focuses on sampling rate conversion by decimation and interpolation and their effects on
frequency spectrum. Implementation of sampling rate conversion in filters and application of multirate
digital signal processing are also discussed in the chapter.
Chapter 10 is concerned with the estimation of energy spectrum of discrete time signals and power
spectrum of random process. The various nonparametric methods power spectrum estimation and
their performance characteristics are presented.
Chapter 11 focuses on architecture and programming of special purpose processors for digital signal
processing with particular concentration to Texas Instruments digital signal processors TMS320C5x
and TMS320C54x processors.
Chapter 12 provide a brief discussion on some applications of digital signal processing in speech,
musical sound, audio/video, communication and biomedical signals.
The author has taken care to present the concepts of Digital Signal Processing in a simple manner and
hope that the teaching and student community will welcome the book. The readers can feel free to
convey their criticism and suggestions to [email protected] for further improvement of the book.

A.Nagoor Ka ni
Kani
xxi

Acknowledgements
I express my heartful thanks to my wife Ms.C. Gnanaparanjothi Nagoor Kani and my sons
N. Bharath Raj alias Chandrakani Allaudeen and N.Vikram Raj for the support, encouragement and
cooperation they have extended to me throughout my career.
It is my pleasure to acknowledge the contributions to our technical editors Ms.K.Jayashree, Ms.
B.Hemavathy, Ms. S. Pavithra for editing and proofreading of the manuscript, and Ms. A. Selvi, Ms.
M. Faritha for type setting and preparing the layout of the book.
My sincere thanks to all reviewers for their valuable suggestions and comments which helps me to
explore the subject to greater depth.

Prateek Kumar Maharana Pratap Engineering College

Kanpur, Uttar Pradesh.
Ashish Suri Shri Mata Vaishno Devi University
Jammu, Jammu and Kashmir.
S.S Prasad National Institute of Technology (NIT)
Jamshedpur, Jharkhand.
Harpal Theti Kalinga Institute of Industrial Technology,
Bhubaneswar, Orissa.
Kishor Kinage D J Sanghvi Engineering College, Mumbai.
S. Moorthi National Insitute of Technology (NIT),
Tiruchirapalli, Tamil Nadu
S. Anand MEPCO SCHLENK Engineering Colledge,
Sivakasi, Tamil Nadu.
R. Prakash School of Electronics Sciences, Vellore Institute of Technology,
Vellore, Tamil Nadu.
J. Vijayraghavan Rajalakshmi Engineering College,
Chennai.
Jagadeshwar Reddy Sri Venkateswara Institute of Science and Technology,
Kadapa, Andhra Pradesh.
P. Biswagar R V College of Engineering,
Bangalore, Karnataka.

I am also grateful to Ms.Vibha Mahajan, Mr.Ebi John, Ms. Koyel Ghosh, Mr. P.L.Pandita and
Ms. Sohini Mukherjee of Tata McGraw Hill Education for their concern and care in publishing this
work.
 xxii

My special thanks to Ms. Koyel Ghosh of McGraw Hill Education for her care in bringing out this
work at the right time.

I thank all my office staff for their cooperation in carrying out my day-to-day activities.

Finally, a special note of appreciation is due to my sisters, brothers, relatives, friends, students and the
entire teaching community for their overwhelming support and encouragement to my writing.

A. Nagoor Kani
 List of Symbols and Abbreviations

Symbols

A - Number of integer digit

As - Gain at stopband edge frequency

Ap - Gain at passband edge frequency

B - Bandwidth in Hz

b - Size of binary excluding sign bit

ck - Fourier coefficients of exponential form of Fourier series of x(t)

D - Sampling rate reduction factor

E - Energy of a signal

Er - Relative error due to rounding

Et - Relative error due to truncation

er - Rounding error

f - Frequency of discrete time signal (or digital frequency) in

cycles/sample

F - Frequency of continuous time signal (or analog frequency) in Hz

fo - Fundamental frequency of discrete time signal in cycles/sample

Fo - Fundamental frequency of continuous time signal in Hz

Fm - Maximum frequency of continuous time signal in Hz

Fs - Sampling frequency of continuous time signal in Hz

I - Sampling rate mulitplication factor

j - complex operator, −1

L - Number of segments
 xxiv

M - Figure of merit

M - Mantissa

N - Fundamental period

N - Order of the filter

Nf - Floating point binary number

Ntf - Truncated floating point number

P - Power of a signal

p - Pole

P xx(f) - Power spectrum

P xxB(f) - Bartlett power spectrum estimate

P xxBT(f) - Blackman-Tukey power spectrum estimate

P xxper (f) - Periodogram power spectrum estimate

P xxW(f) - Welch power spectrum estimate

q - Quantization step size

Q - Quality factor

R - Range of decimal number

r - Radix or base

S - Sign bit

S xx(f) - Energy spectrum

t - Time in seconds

T - Time period in seconds

V - Variabiltiy

W - Phase factor or Twiddle factor

x(n) - Discrete time signal or Ergodic random process

xxv

X(n) - Random process

z - Complex variable (z = u + jv)

z - Unit advance operator or Zero

z–1 - Unit delay operator

Î - Attenuation costant

W - Angular frequency of continuous time signal in rad/sec

Wo - Center frequency

Ws - Stop band edge analog frequency in rad/sec

Wp - Pass band edge analog frequency in rad/sec

w - Angular frequency of discrete time signal in rad/sample

wk - Sampling frequency point

wp - Pass band edge digital frequency in rad/sample

ws - Stop band edge digital frequency in rad/sample

s2 - Variance

s2eoi - Steady state output noise power due to input quantization error

ap - Attenuation at a pass band frequency

as - Attenuation at a stop band frequency

* - Convolution operator

* - Circular convolution operator

z - Integration operator

d
- Differentiation operator
dt
 xxvi

Standard/Input/Output Signals

|A(w)| - Magnitude function

h(n) - Impulse response of discrete time system

h’(n) - Impulse response of inverse system

hd(n) - Desired impulse response

r xy(m) - Crosscorrelation sequence of x(n) and y(n)

r xx(m) - Autocorrelation sequence of discrete time signal

r xx(m) - Autocorrelation sequence of random process with finite data

gxx (m) - Autocorrelation sequence of random process with infinite data

rxx ( m) - Circular autocorrelation sequence of x(n)

rxy ( m) - Circular crosscorrelation sequence of x(n) and y(n)

u(n) - Discrete time unit step signal

w R (n) - Rectangular window sequence

w r (n) - Bartlett or triangular window sequence

w C (n) - Hanning window sequence

w H (n) - Hamming window sequence

w B (n) - Blackman window sequence

w K (n) - Kaiser window sequence

x(n) - Discrete time signal

x(n) - Input of discrete time system

xo(n) - Odd part of discrete time signal x(n)

x e(n) - Even part of discrete time signal x(n)

x(n–m) - Delayed or linearly shifted x(n) by m units

x((n–m)) N - Circularly shifted x(n) by m units, where N is period

xxvii

x(Dn) - Down sampled version of x(n)

x(n/I) - Upsampled version of x(n)

x P(n) - Periodic extension of x(n)

y(n) - Output / Response of discrete time system

y(n – m) - Delayed output / Response of discrete time system

yp(n) - Particular soultion of discrete time system

yn(n) - Homogenous solution of discrete time system

yzs(n) - Zero state response of discrete time system

yzi(n) - Zero input response of discrete time time system

d(n) - Discrete time impulse signal

d(n – m) - Delayed impulse signal

tp - Phase delay

tg - Group delay

q(w) - Phase function

Transform Operators and Functions

DFT - Discrete Fourier transform (DFT)

DFT–1 - Inverse DFT

E{X} - Expected value of random variable

F - Fourier transform

F–1 - Inverse Fourier transform

H - System operator

H –1 - Inverse system operator

H(z) - Transfer function

 xxviii

H(ejw ) - Frequency response of the digital filter

HN(z) - Normalized transfer function

H d(ejw ) - Desired or ideal frequency response

Q[ ] - Quantization operations

X(e jw ) - Discrete time Fourier transform of x(n)

X r (e jw ) - Real part of X(ejw )

X i(e jw ) - Imaginary part of X(ejw )

X(jW ) - Fourier transform of x(t)

X(k) - Discrete Fourier transform of x(n)

X r (k) - Real part of X(k)

X i(k) - Imaginary part of X(k)

X(z) - Z-transform of x(n)

Z - Z-transform

Z–1 - Inverse Z-transform

Abbreviations
BIBO - Bounded Input Bounded Output

DFT - Discrete Fourier Transform

DIF - Decimation In Frequency

DIT - Decimation In Time

DT - Discrete Time

DTFS - Discrete Time Fourier Series

DTFT - Discrete Time Fourier Transform

FFT - Fast Fourier Transform

FIR - Finite Impulse Response

xxix

IIR - Infinite Impulse Response

LSD - Least Significant Digit

LHP - Left Half Plane

LTI - Linear Time Invariant

MSD - Most Significant Digit

NTF - Noise Transfer Function

RHP - Right Half Plane

ROC - Region Of Convergence

Var - Variance

QMF - Quardrature Mirror Filter

LPF - Low Pass Filter

 Chapter 1
Introduction to Digital
Signal Processing

1.1 Introduction
Digital Signal Processing (DSP) refers to processing of signals by digital systems like Personal
Computers (PC) and systems designed using digital Integrated Circuits (ICs), microprocessors and
microcontrollers. DSP gained popularity in the 1960s. Earlier, DSP systems were limited to general purpose
non-real-time scientific and business applications. The rapid advancement in computers and IC fabrication
technology leads to complete domination of DSP systems in both real-time and non-real-time applications
in all fields of engineering and technology.
The basic components of a DSP system are shown in fig 1.1. The DSP system involves conversion of
analog signal to digital signal, then processing of the digital signal by a digital system and then conversion
of the processed digital signal back to analog signal.

Input analog Input digita l O utput digital O utput analog

s ignal s ignal D igital s ignal s ignal
ADC DAC
s ys tem

F ig 1.1 : B a sic c o m p o ne n ts o f a D S P system .

The real-world signals are analog, and only for processing by digital systems, the signals are converted
to digital. For conversion of signals from analog to digital, an ADC (Analog to Digital Converter) is employed.
The various steps in analog to digital conversion process are sampling and quantization of analog signals,
and then converting the quantized samples to suitable binary codes. The digital signals in the form of binary
codes are fed to digital system for processing, and after processing, it generates an output digital signal in
the form of binary codes. The output analog signal is constructed from the output binary codes using a
DAC (Digital to Analog Converter).
The processing of signals are basically spectrum analysis to determine the various frequency
components of a signal and filtering the signal to extract the required frequency component of the signal.
1. 2 Digital Signal Processing
The digital system can be a specially designed programmable hardware for DSP or an algorithm/
software running on a general purpose digital system like Personal Computer (PC).
Advantages of Digital Signal Processing
Some of the advantages of digital processing of signals are,
1. The digital hardware are compact, reliable, less expensive, and programmable.
2. Since the DSP systems are programmable, the performance of the system can be easily upgraded/
modified.
3. By employing high speed, sophisticated digital hardware higher precision can be achieved in
processing of signals.
4. The digital signals can be permanently stored in magnetic media so that they are transportable
and can be processed in non-real-time or off-line.

1.2 Signal
Any physical phenomenon that conveys or carries some information can be called a signal. The
music, speech, motion pictures, still photos, heart beat, etc., are examples of signals that we normally encounter
in day-to-day life.
When a signal is defined continuously for any value of an independent variable, it is called an analog
or continuous signal. Most of the signals encountered in science and engineering are analog in nature.
When the dependent variable of an analog signal is time, it is called a continuous time signal and it is denoted
as “x(t)”.
When a signal is defined for discrete intervals of an independent variable, it is called a discrete signal.
When the dependent variable of a discrete signal is time, it is called discrete time signal and it is denoted by
“x(n)”. Most of the discrete signals are either sampled versions of analog signals for processing by digital
systems or output of digital systems.
The quantized and coded version of the discrete time signals are called digital signals. In digital
signals the value of the signal for every discrete time “n” is represented in binary codes. The process of
conversion of a discrete time signal to digital signal involves quantization and coding.
Normally, for binary representation, a standard size of binary is chosen. In m-bit binary representation,
we can have 2m binary codes. The possible range of values of the discrete time signals are usually divided
into 2m steps called quantization levels, and a binary code is attached to each quantization level. The values
of the discrete time signals are approximated by rounding or truncation in order to match the nearest quantization
level.

1.3 Discrete Time System

Any process that exhibits cause and effect relation can be called a system. A system will have an input
signal and an output signal. The output signal will be a processed version of the input signal. A system is
either interconnection of hardware devices or software / algorithm.
A system which can process a discrete time signal is called a discrete time system, and so the input
and output signals of a discrete time system are discrete time signals.
Chapter 1 - Introduction to Digital Signal Processing 1. 3
A discrete time system is denoted by the letter H. The input of discrete time system is denoted as
“x(n)” and the output of discrete time system is denoted as “y(n)”. The diagrammatic representation of a
discrete time system is shown in fig 1.2.
D isc rete
tim e sy stem
x (n ) y (n )
H
Input signa l O utput sign al
or exc itation or respons e

F ig 1.2 : R ep resen ta tio n of d iscrete tim e system .

The operation performed by a discrete time system on input to produce output or response can be
expressed as,
Response, y(n) = H{x(n)}
where, H denotes the system operation (also called system operator).
When a discrete time system satisfies the properties of linearity and time invariance then it is called
LTI (Linear Time Invariant) discrete time system .
The input-output relation of an LTI discrete time system is represented by constant coefficient
difference equation shown below.
N M
bg
yn = − ∑ a m ybn − mg + ∑ bm xbn − mg
m=1 m=0

where, N = Order of the system, and M £ N.

The solution of the above difference equation is the response y(n) of the discrete time system, for the
input x(n).

1.4 Analysis of Discrete Time System

Mostly, the discrete time systems are designed for analysis of discrete time signals. Physically, the
discrete time systems are realized in time domain. In time domain, the discrete time systems are governed by
difference equations. The analysis of discrete time signals and systems in time domain involves solution of
difference equations. The solution of difference equations are difficult due to assumption of a solution and
then solving the constants using initial conditions.
In order to simplify the task of analysis, the discrete time signals can be transformed to some other
domain, where the analysis may be easier. One such transform exists for discrete time signals is Z-transform.The
Z-transform, will transform a function of discrete time “n” into a function of complex variable “z”, where
z = rejw .Therefore, Z-transform of a discrete time signal will transform the time domain signal into z-domain
signal.
On taking Z -transform of the difference equation governing the discrete time system, it becomes
algebraic equation in “z” and the solution of algebraic equation will give the response of the system as a
function of “z” and it is called z-domain response. The inverse Z -transform of the z-domain response, will
give the time domain response of the discrete time system. Also, the stability analysis of the discrete
systems are much easier in z-domain.
1. 4 Digital Signal Processing

The ratio of Z -transform of output and input is called transfer function of the discrete time system.
The inverse Z -transform of the system gives the impulse response of the system, which is used to study the
characteristics of a system.
Another important characteristic of any signal is frequency, and for most of the applications the
frequency content of the signal is an important criteria. The frequency range of some of the signals are listed
in table 1.1 and 1.2.
Table 1.1 : Frequency Range of Some Electromagnetic Signals

Type of signal Wavelength (m) Frequency range (Hz)

Radio broadcast 104 to 102 3 ´ 104 to 3 ´ 106
Shortwave radio signals 102 to 10–2 3 ´ 106 to 3 ´ 1010
Radar / Space communications 1 to 10–2 3 ´ 108 to 3 ´ 1010
Common-carrier microwave 1 to 10–2 3 ´ 108 to 3 ´ 1010
Infrared 10–3 to 10–6 3 ´ 1011 to 3 ´ 1014
Visible light 3.9´10–7 to 8.1´10–7 3.7 ´ 1014 to 7.7 ´ 1014
Ultraviolet 10–7 to 10–8 3 ´ 1015 to 3 ´ 1016
Gamma rays and x-rays 10–9 to 10–10 3 ´ 1017 to 3 ´ 1018

Table 1.2 : Frequency Range of Some Biological and Seismic Signals

Type of Signal Frequency Range (Hz)

Electroretinogram 0 to 20
Electronystagmogram 0 to 20
Pneumogram 0 to 40
Electrocardiogram (ECG) 0 to 100
Electroencephalogram (EEG) 0 to 100
Electromyogram 10 to 200
Sphygmomanogram 0 to 200
Speech 100 to 4000
Wind noise 100 to 1000
Seismic exploration signals 10 to 100
Earthquake and nuclear explosion signals 0.01 to 10
Seismic noise 0.1 to 1
The frequency contents of a discrete time signal can be studied by taking Fourier transform of the
discrete time signal. The Fourier transform of discrete time signal is a particular class of Z-transform in which
z = ejw ,where “w” is the frequency of the discrete time signals.
Chapter 1 - Introduction to Digital Signal Processing 1. 5
The Fourier transform, will transform a function of discrete time “n” into a function of frequency
“w”. Therefore, Fourier transform of a discrete time signal will transform the discrete time signal into frequency
domain signal. The Fourier transform of the discrete time signal, is also called frequency spectrum of the
discrete time signal. The Fourier transform of the impulse response of a system is called frequency response
of the system. The frequency spectrum is a complex function of “w” and so can be expressed as magnitude
spectrum and phase spectrum. The magnitude spectrum is used to study the various frequency components
of the discrete time signal.
The frequency spectrum obtained via Fourier transform will be a continuous spectrum and so cannot
be computed by digital systems, Therefore, the samples of Fourier transform can be computed at sufficient
number of points by digital systems. The samples of Fourier transform can also be directly computed using
DFT ( Discrete Fourier Transform ). The computation of DFT involves a large number of calculations. In
order to reduce the computational task of DFT, a number of methods/algorithms are developed which are
collectively called FFT (Fast Fourier Transform). The DFT of discrete time signal will give the discrete
frequency spectrum of the signal.

1.5 Filters
The filters are frequency selective devices. The two major types of digital filters are FIR (Finite Impulse
Response) and IIR (Infinite Impulse Response) filters.
Generally, the filter specification will be a desired frequency response. The inverse Fourier transform
of the frequency response will be the impulse response of the filter, and it will be an infinite duration signal.
The digital filters designed by choosing finite samples of impulse rseponse are called FIR filters, and the
filters designed by considering all the infinite samples are called IIR filters.
Since, an FIR filter is designed from the finite samples of impluse response, the direct design of FIR filter
is possible in which the transfer function of the filter is obtained by taking Z -transform of impulse response.
Note : Mathematically, the filter design is design of transfer function of the filter.
Since, an IIR filter is designed by considering / preserving the infinite samples of impulse response,
the direct design of IIR filter is not possible. Therefore, the IIR filter is designed via analog filter. For
designing IIR filter, first the specifications of IIR filter is transformed to specifications of analog filter using
bilinear or impulse-invariant transformation, then an analog filter transfer function is designed using
Butterworth or Chebychev approximation. Finally the analog filter transfer function is transfered to digital
filter transfer function using the transformation chosen for transforming the specifications.

1.6 Finite Word Length Effects

In digital representation the signals are represented as an array of binary numbers, and the digital
system employ a fixed size of binary called “word size or word length” for number representation. This finite
word size for number representation leads to errors in input signals, intermediate signals in computations and
in the final output signals. In general, the various effects due to finite precision representation of numbers in
digital systems are called finite word length effects.
Some of the finite word length effects in digital systems are given below.
· Errors due to quantization of input data.
· Errors due to quantization of filter coefficients.
1. 6 Digital Signal Processing
· Errors due to rounding the product in multiplication.
· Errors due to overflow in addition.
· Limit cycles in recursive computations.

1.7 Multirate DSP

In many communication systems, the sampling rate conversion is a vital requirement. Some of the
systems that employ sampling rate conversion are video receivers that receive both NTSC and PAL signals,
audio systems that can play CDs recorded in different standards, etc.
The processing of discrete time signals at different sampling rates in different parts of a system is
called multirate DSP. In digital systems, the sampling rate conversion is achieved by either decimation or
interpolation. In decimation, the sampling rate is reduced, whereas in interpolation the sampling rate is
increased. The multirate DSP systems leads to reduction in computations, memory requirement and errors
due to finite word length effects.

1.8 Energy and Power Spectrum

There are many situations where the signals are corrupted by noise like sonar signals corrupted by
ambient ocean noise, speech signal from cockpit of an airplane corrupted by engine noise, etc. When the
signals are corrupted by noise, then the energy or power spectrum will be useful to identify the signal from
noise.
The energy spectrum can be computed for deterministic signals, and it is given by square of magnitude
of Fourier transform of the signal. Alternatively, the energy spectrum is given by Fourier transform of the
autocorrelation sequence of the signal.
The power spectrum can be estimated for nondeterministic signals or random process/signals. The
power spectrum estimation methods can be broadly classified into two groups, namely, nonparametric methods
and parametric methods.
In nonparametric methods, first an estimate of autocorrelation of the random process is determined
which represents the average behaviour of the signal, then the Fourier transform of estimated autocorrelation
is determined, which is the power spectrum estimate of the random process.
In parametric methods, first an appropriate model is selected for the given random process, then the
parameters of the model are computed using the available data of the random process. Finally, the power
spectrum is estimated from the constructed model.

1.9 Digital Signal Processors

The digital signal processors are specially designed microprocessors/microcontrollers for DSP
applications.
The importance of special purpose processors for signal processing applications were realised in 1980s,
and many companies started releasing special processors for DSP applications. The pioneers among them are
Texas Instruments and Analog Devices. The Texas Instruments has released a large variety of processors in the
family name TMS320Cxx and Analog Devices has released processors in the family name ADSPxx.
Chapter 1 - Introduction to Digital Signal Processing 1. 7
Some of the special features of digital signal processors are given below.
· Modified Harvard architecture with two or more internal buses for simultaneous access of code
and one or two data.
· Specialized addressing modes like circular addressing and bit reversed addressing suitable for
computations like convolution, correlation and FFT.
· MAC unit for performing multiply-accumulate computations involved in convolution, correlation
and FFT in single clock cycle.
· Larger size ALU and accumulators with guard bits to accommodate the overflow in computation.
· Pipelining of instructions to execute different phases of four or six instructions in parallel.
· VLIW architecture to fetch and execute multiple instructions in parallel.
· Multiprocessor architecture by integrating multiple processors on a single piece of silicon for
parallel processing.

1.10 Importance of Digital Signal Processing

The technology advancement in programmable digital signal processors, helps to implement more and
more real time applications in digital systems.
The digital processing of signal plays a vital role in almost every field of Science and Engineering. Some
of the applications of digital processing of signals in various field of Science and Engineering are listed here.
1. Biomedical
· ECG is used to predict heart diseases.
· EEG is used to study normal and abnormal behaviour of the brain.
· EMG is used to study the condition of muscles.
· X-ray images are used to predict the bone fractures and tuberculosis.
· Ultrasonic scan images of kidney and gall bladder is used to predict stones.
·. Ultrasonic scan images of foetus is used to predict abnormalities in a baby.
· MRI scan is used to study minute inner details of any part of the human body.

2. Speech Processing
· Speech compression and decompression to reduce memory requirement of storage systems.
· Speech compression and decompression for effective use of transmission channels.
· Speech recognization for voice operated systems and voice based security systems.
· Speech recognization for conversion of voice to text.
· Speech synthesis for various voice based warnings or annoucements.

3. Audio and Video Equipments

· The analysis of audio signals will be useful to design systems for special effects in audio systems
like stereo, woofer, karoke, equalizer, attenuator, etc.
· Music synthesis and composing using music keyboards.
· Audio and video compression for storage in DVDs.
1. 8 Digital Signal Processing
4. Communication
· The spectrum analysis of modulated signals helps to identify the information bearing frequency
component that can be used for transmission.
· The analysis of signals received from radars are used to detect flying objects and thier velocity.
· Generation and detection of DTMF signals in telephones.
· Echo and noise cancellation in transmission channels.

5. Power electronics
· The spectrum analysis of the output of coverters and inverters will reveal the harmonics present in
the output, which in turn helps to design suitable filter to eliminate the harmonics.
· The analysis of switching currents and voltages in power devices will help to reduce losses.

6. Image processing
· Image compression and decompression to reduce memory requirement of storage systems.
· Image compression and decompression for effective use of transmission channels.
· Image recognition for security systems.
· Filtering operations on images to extract the features or hidden information.

7. Geology
· The seismic signals are used to determine the magnitude of earthquakes and volcanic eruptions.
· The seismic signals are also used to predict nuclear explosions.
· The seismic noises are also used to predict the movement of earth layers (tectonic plates).

8. Astronomy
· The analysis of light received from a star is used to determine the condition of the star.
· The analysis of images of various celestial bodies gives vital information about them.

1.11 Use of MATLAB in Digital Signal Processing

MATLAB (MATrix LABoratory) is a software developed by The MathWork Inc, USA, which can run
on any windows platform in a PC (Personal Computer). This software has a number of tools for the study of
various engineering subjects. It includes various tools for digital signal processing also. Using these tools,
a wide variety of studies can be made on discrete time signals and systems. Some of the analysis that is
relevant to this particular textbook are given below.
· Sketch or plot of discrete time signals as a function of independent variable.
· Spectrum analysis of discrete time signals.
· Solution of LTI discrete time systems.
· Perform convolution and deconvolution operations on discrete time signals.
· Perform various transforms on discrete time signals like Fourier transform, Z-transform, Fast Fourier
Transform (FFT), etc.
· Design and frequency response analysis of FIR and IIR filters.
· Decimation and interpolation of discrete time signals.
· Estimation of energy and power spectrum of discrete time signals.
 Chapter 2

Discrete Time Signals

and Systems

2.1 Introduction
In today's world, digital systems are employed for almost every application. The digital systems can
process only discrete signals. This chapter deals with time domain analysis of discrete time signals and
systems. In the first part of this chapter, the generation, representation, classification and mathematical
operations on discrete time signals are discussed in detail. In the second part of this chapter, the representation,
classification and response of discrete time systems are discussed in detail. The concept of LTI systems are
highlighted wherever necessary.
Discrete Signal and Discrete Time Signal
The discrete signal is a function of a discrete independent variable. The independent variable is
divided into uniform intervals and each interval is represented by an integer. The letter "n" is used to denote
the independent variable. The discrete or digital signal is denoted by x(n).
The discrete signal is defined for every integer value of the independent variable "n". The magnitude
(or value) of discrete signal can take any discrete value in the specified range. Here both the value of the
signal and the independent variable are discrete. The discrete signal can be represented by a one-dimensional
array as shown in the following example.
Example :

x(n) = { 2, 4, -1, 3, 3, 4 }
Here the discrete signal x(n) is defined for, n = 0, 1, 2, 3, 4, 5
\ x(0) = 2 ; x(1) = 4 ; x(2) = –1 ; x(3) = 3 ; x(4) = 3 ; x(5) = 4 .

When the independent variable is time t, the discrete signal is called discrete time signal. In discrete
time signal, the time is divided uniformly using the relation t = nT, where T is the sampling time period. (The
sampling time period is the inverse of sampling frequency). The discrete time signal is denoted by x(n) or x(nT).
Chapter 2 - Discrete Time Signals and Systems 2. 2
Since the discrete signals have a sequence of numbers (or values) defined for integer values of the
independent variable, the discrete signals are also known as discrete sequence. In this book, the term sequence
and signal are used synonymously. Also in this book, the discrete signal is referred as discrete time signal.
Digital Signal
The digital signal is same as discrete signal except that the magnitude of the signal is quantized. The
magnitude of the signal can take one of the values in a set of quantized values. Here quantization is necessary
to represent the signal in binary codes.
The generation of a discrete time signal by sampling a continuous time signal and then quantizing the
samples in order to convert the signal to digital signal is shown in the following example.
Let, x(t) = Continuous time signal
T = Sampling time
A typical continuous time signal and the sampling of this continuous time signal at uniform interval
are shown in fig 2.1a and fig 2.1b respectively. The samples of the continuous time signal as a function of
sampling time instants are shown in fig 2.1c. (In fig 2.1c, 1T, 2T, 3T, ....etc., represents sampling time instants
and the value of the samples are functions of this sampling time instants).
x (t) x (t) x (n t)
1.0 1.0 1.0
0.9
0.9 0.9 0.9
0.8 0.8
0.8 0.8 0.8

0.7 0.7 0.7

0.6 0.6 0.6 0.55

0.5 0.5 0.5

0.4 0.4 0.4 0.35

0.3
0.3 0.3 0.3

0.2 0.2 0.2

0.1
0.1 0.1 0.1

0 1T 2T 3T 4T 5T 6T 7T t 0 1T 2T 3T 4T 5T 6T 7T t 0 1T 2T 3T 4T 5T 6T 7T t
F ig 2 .1 a . F ig 2 .1 b . F ig 2 .1 c.
F ig 2 .1 : S a m p lin g a co n tin u o us tim e sig n a l to g en era te d iscrete tim e sig na l.

When t=0 ; x(t) = 0 When t = 4T ; x(t) = 0.55

When t = 1T ; x(t) = 0.1 When t = 5T ; x(t) = 0.8
When t = 2T ; x(t) = 0.3 When t = 6T ; x(t) = 0.8
When t = 3T ; x(t) = 0.35 When t = 7T ; x(t) = 0.9
In general, the sampling time instants can be represented as, "nT", where "n" is an integer. When we
drop the sampling time "T" , then the samples are functions of the integer variable "n" alone. Therefore, the
samples of the continuous time signal will be a discrete time signal, denoted as x(n), which is a function of
an integer variable "n" as shown below.
x(n) = { 0, 0.1, 0.3, 0.35, 0.55, 0.8, 0.8, 0.9 }
Here the discrete signal x(n) is defined for, n = 0, 1, 2, 3, 4, 5, 6, 7
2. 3 Digital Signal Processing

\ x(0) = 0 ; x(1) = 0.1 ; x(2) = 0.3 ; x(3) = 0.35 ;

x(4) = 0.55 ; x(5) = 0.8 ; x(6) = 0.8 ; x(7) = 0.9 .
The sample value lies the range of 0 to 1.
Let us choose 3-bit binary to represent the samples in binary code. Now, the possible binary codes are
23 = 8, and so the range can be divided into eight quantization levels, and each sample is assigned, one of the
quantization level as shown in the following table.
Quantization level Binary code Range represented by quantization level
(R = Range = 1) for quantization by truncation

0 × R3 = 0 × 1 = 0 000 0.000 ≤ x(n) < 0.125 ⇒ 0.000

2 8

1 × R3 = 1 × 1 = 0.125 001 0.125 ≤ x(n) < 0.250 ⇒ 0.125

2 8

2 × R3 = 2 × 1 = 0.25 010 0.250 ≤ x(n) < 0.375 ⇒ 0.250

2 8

3 × R3 = 3 × 1 = 0.375 011 0.375 ≤ x(n) < 0.500 ⇒ 0.375

2 8

4 × R3 = 4 × 1 = 0.5 100 0.500 ≤ x(n) < 0.625 ⇒ 0.500

2 8

5 × R3 = 5 × 1 = 0.625 101 0.625 ≤ x(n) < 0.75 ⇒ 0.625

2 8

6 × R3 = 6 × 1 = 0.75 110 0.750 ≤ x(n) < 0.875 ⇒ 0.750

2 8

7 × R3 = 7 × 1 = 0.875 111 0.875 ≤ x(n) ≤ 1.000 ⇒ 0.875

2 8

Let, xq(n) = Quantized discrete time signal.

xc(n) = Quantized and coded discrete time signal.
Now, xq(n) = { 0, 0, 0.25, 0.25, 0.5, 0.75, 0.75, 0.875 }
xc(n) = { 000, 000, 010, 010, 100, 110, 110, 111 }
The quantized and coded discrete time signal xc(n) is called digital signal.

2.2 Discrete Time Signals

2.2.1 Generation of Discrete Time Signals
A discrete time signal can be generated by the following three methods.
The methods 1 and 2 are independent of any time frame but Method 3 depends critically on time.
1. Generate a set of numbers and arrange them as a sequence.
Example :

The numbers 0, 2, 4, ...., 2N form a sequence of even numbers and can be expressed as,

x(n) = 2n ; 0 £ n £ N
Chapter 2 - Discrete Time Signals and Systems 2. 4

2. Evaluation of a numerical recursion relation will generate a discrete signal.

Example :
x(n) = 0.2 x(n − 1) with initial condition x(0) = 1, gives the sequence, x(n) = 0.2n ; 0 ≤ n < ∞
When n = 0 ; x( 0) = 1 ( initial condition) = 0.20
When n = 1 ; x( 1) = 0.2 x(1 − 1) = 0.2 x(0) = 0.2 = 0.21
When n = 2 ; x(2) = 0.2 x(2 − 1) = 0.2 x(1) = 0.2 × 0.2 = 0.22
When n = 3 ; x(3) = 0.2 x(3 − 1) = 0.2 x(2) = 0.2 × 0.22 = 0.23 and so on
∴ x(n) = 0.2n ; 0 ≤ n < ∞

3. A third method is by uniformly sampling a continuous time signal and using the amplitudes of
the samples to form a sequence.
Let, x(t) = Continuous time signal
Now, Discrete signal, x(nT) = x(t) t = nT ; − ∞ < n < ∞
where, T is the sampling interval
The generation of discrete signal by sampling a continuous time signal is shown in fig 2.1.
2.2.2 Representation of Discrete Time Signals
The discrete time signal can be represented by the following methods.
1. Functional representation
In functional representation, the signal is represented as a mathematical equation, as shown in the
following example. x (n )
x(n) = – 0.5 ; n = – 2 1.5
1.2
= 1.0 ; n = – 1 1.0
= – 1.0 ; n = 0 0.6
= 0.6 ; n = 1
= 1.2 ; n = 2
= 1.5 ; n = 3 −2 −1 0 1 2 3 n
= 0 ; other n −0.5
−1.0
F ig 2.2 : G ra phica l rep resenta tio n of a
2. Graphical representation d iscrete tim e sig na l.
In graphical representation, the signal is represented in a two-dimensional plane. The independent
variable is represented in the horizontal axis and the value of the signal is represented in the vertical axis as
shown in fig 2.2.
3. Tabular representation
In tabular representation, two rows of a table are used to represent a discrete time signal. In the first
row, the independent variable "n" is tabulated and in the second row the value of the signal for each value of
"n" are tabulated as shown in the following table.

n ........... –2 -1 0 1 2 3 ..............
x(n) ........... –0.5 1.0 –1.0 0.6 1.2 1.5 ..............
2. 5 Digital Signal Processing
4. Sequence representation
In sequence representation, the discrete time signal is represented as a one-dimensional array as
shown in the following examples.
An infinite duration discrete time signal with the time origin, n = 0, indicated by the symbol - is represented as,
x(n) = { ..... – 0.5, 1.0, –1.0, 0.6, 1.2, 1.5, ..... }
-

An infinite duration discrete time signal that satisfies the condition x(n) = 0 for n < 0 is represented as,
x(n) = { –1.0, 0.6, 1.2, 1.5, ... } or x(n) = {–1.0, 0.6, 1.2, 1.5, ... }
-

A finite duration discrete time signal with the time origin, n = 0, indicated by the symbol - is represented as,
x(n) = { – 0.5, 1.0, –1.0, 0.6, 1.2, 1.5 }
-

A finite duration discrete time signal that satisfies the condition x(n) = 0 for n < 0 is represented as,
x(n) = { –1.0, –0.6, 1.2, 1.5 } or x(n) = { –1.0, 0.6, 1.2, 1.5}
-

2.2.3 Standard Discrete Time Signals

δ(n ) u (n )
1. Digital impulse signal or unit sample sequence
1 1
Impulse signal, δ( n) = 1 ; n = 0
=0 ; n ≠ 0 0 n 0 1 2 3 4 5 n
F ig 2.3 : D ig ita l im pu lse F ig 2.4 : U nit step sig n a l.
2. Unit step signal sig na l.
u r(n )
Unit step signal, u( n) = 1 ; n ≥ 0 5

= 0; n < 0 4
3
2
3. Ramp signal 1

Ramp signal, u r ( n ) = n ; n ≥ 0 0 1 2 3 4 5 n
= 0 ;n < 0 F ig 2.5 : R a m p sig na l.
4. Exponential signal
n
Exponential signal, g( n) = a ; n ≥ 0
= 0 ;n < 0

0 1 2 3 4 5 6 n 0 1 2 3 4
F ig 2.6a : D e c re asin g e x po n en tia l sign a l. F ig 2.6b : In cre a sin g ex p o ne n tial sig n al.
F ig 2.6 : E xp o n en tia l sig n a l.
Chapter 2 - Discrete Time Signals and Systems 2. 6
5. Discrete time sinusoidal signal
The discrete time sinusoidal signal may be expressed as,
bg b g
x n = A cos ω 0n + θ ; for n in the range -¥ < n < +¥
xb ng = A sin bω n + θg ; for n in the range -¥ < n < +¥
0

where, w 0 = Frequency in radians/sample ; q = Phase in radians

ω0
f0 = = Frequency in cycles/sample
2π
x (n )

−9 −8 −7 −6 −5 −4 4 5 6 7 8 9
−3 −2 −1 0 1 2 3
n

x (n )
F ig 2 .7 a : D iscrete tim e sin u so id a l sig n a l rep re se n ted
b y e q u a tio n x(n ) = A c o s( ω0 n ).

−6 −5 −4 −3 −2 −1 7 8 9 10 11 12
x (n ) 0 1 2 3 4 5 6 n

F ig 2 .7 b : D iscrete tim e sin u so id a l sig n a l rep re se n ted

1 2 3 4 5 6 7 b y e q u a tio n x(n ) = A sin (ω0 n ).
−5 −4 −3 −2 −1 0 n
π
3

F ig 2 .7 c : D iscrete tim e sin u so id a l sig n a l rep re se n ted b y eq u a tio n ,

x (n ) = A c o s π n + π ; ω0 = π ; θ = π
e6 j
3 6 3
F ig 2 .7 : D iscrete tim e sin u so id a l sig n a ls.
Properties of Discrete Time Sinusoid

1. A discrete time sinusoid is periodic only if its frequency f0 is a rational number, (i.e., ratio of two
integers).
2. Discrete time sinusoids whose frequencies are separated by integer multiples of 2p are identical.
∴ x(n) = A cos[(w 0 + 2pk ) n + q], for k = 0,1,2...........are identical in the interval
-p£w0 £ p and so they are indistinguishable.
Proof :

cos[( w 0 + 2pk) n + q] = cos(w 0n + 2pnk + q) = cos[(w 0n + q) + 2pnk]

= cos(w 0n + q) cos 2pnk - sin (w 0n + q) sin 2pnk

Since n and k are integers, cos 2pnk =1 and sin 2pnk = 0

∴ cos[(w 0 + 2pk) n + q] = cos(w 0n + q), for k = 0, 1, 2, 3, .....

2. 7 Digital Signal Processing

Conclusion

1.The sequences of any two sinusoids with frequencies in the range, -p £ w o £ p

(or -1/2 £ f0 £ 1/2), are distinct.
[-p £ w £ p divide by 2π
→ -1/2 £ f £ 1/2]
2. Any discrete time sinusoid with frequency w0 > |p| (or f0 > |1/2|) will be identical to another discrete
time sinusoid with frequency w 0 < |p| (or f0 < |1/2|).
6. Discrete time complex exponential signal
The discrete time complex exponential signal is defined as,
x(n) = a n e j(ω 0 n + θ) = an [cos(w 0n + q) + j sin(w 0n + q)]
= an cos(w 0n + q) + j an sin(w 0n + q) = xr(n) + j xi(n)
where, xr(n) = Real part of x(n) = an cos(w 0n + q)
xi(n) = Imaginary part of x(n) = an sin(w 0n + q)
The real part of x(n) will give an exponentially increasing cosinusoid sequence for a > 1 and exponentially
decreasing cosinusoid sequence for 0 < a < 1.
x r (n) x r(n)
0<a<1
a>1

n n

F ig 2.8 a : T h e d isc rete tim e se q u en c e re presen ted b y the F ig 2.8 b : T h e d isc rete tim e se q u en c e re presen ted b y the
n n
e qu a tio n, x r (n) = a c o s ω0 n for 0 < a < 1 . e qu a tio n, x r (n) = a c o s ω0 n for a > 1 .
F ig 2 .8 : R ea l p a rt o f co m p lex exp o n en tia l sig na l.
The imaginary part of x(n) will give rise to an exponentially increasing sinusoid sequence for a > 1 and
exponentially decreasing sinusoid sequence for 0 < a < 1.
x i (n ) x i (n )
0<a<1 a>1

n n

F ig 2.9a : T he d isc re te tim e seq u e nc e F ig 2.9b : T he d isc re te tim e seq u e nc e

re p rese n te d b y th e eq u a tion , re p rese n te d b y th e eq u a tion ,
x i(n ) = a n sin ω0 n for 0 < a < 1. x i(n ) = a n sin ω0 n for a > 1 .
F ig 2 .9 : Im a g in a ry p a rt of co m p lex e xp o n en tia l sig n a l.
Chapter 2 - Discrete Time Signals and Systems 2. 8

2.3 Sampling of Continuous Time (Analog) Signals

The sampling is the process of conversion of a continuous time signal into a discrete time signal.
The sampling is performed by taking samples of continuous time signal at definite intervals of time. Usually,
the time interval between two successive samples will be same and such type of sampling is called periodic
or uniform sampling.
The time interval between successive samples is called sampling time (or sampling period or sampling
interval), and it is denoted by “T”. The unit of sampling period is second (s). [The lower units are millisecond (ms)
and microsecond (ms)].
The inverse of sampling period is called sampling frequency (or sampling rate), and it is denoted by Fs.
The unit of sampling frequency is hertz (Hz). (The higher units are kHz and MHz).
Let, xa(t) = Analog / Continuous time signal.
x(n) = Discrete time signal obtained by sampling xa(t).
Mathematically, the relation between x(n) and xa(t) can be expressed as,

x(n) = xa (t) = x a (nT) = xa n

FG IJ ; for n in the range -¥ < n < ¥
t = nT Fs H K
where, T = Sampling period or interval in seconds

Fs = 1 = Sampling rate or sampling frequency in hertz

T
bg b
Example : Let, x a t = A cos Ω 0 t + θ g b
= A cos 2 πF0 t + θ g
where, W 0 = Frequency of analog signal in rad/s
Ω0
F0 = = Frequency of analog signal in Hz
2π
1
Let xa (t) be sampled at intervals of T seconds to get x(n), where T =
Fs
∴ x(n) = x a (t) t = nT
= A cos(Ω 0 t + θ) t = nT

F 2πF n + θI = A cos b2πf n + θg = A cos bω n + θg

= A cos(Ω nT + θ) = A cos G 0
0
HF JK s
0 0

F0
where, f 0 = = Frequency of discrete sinusoid in cycles/sample
Fs
w0 = 2pf 0= Frequency of discrete sinusoid in radians/sample

2.3.1 Sampling and Aliasing

In Section 2.2, it is observed that any two sinusoid signals with frequencies in the range
-1/2 £ f £ +1/2 are distinct and a discrete sinusoid with frequency, f > |±1/2|will be identical to another
discrete sinusoid with frequency, f < |±1/2|. Therefore, we can conclude that range of frequency of
discrete time signal is -1/2 to +1/2 . But the range of frequency of analog signal is -¥ to +¥ . While sampling
analog signals, the infinite frequency range continuous time signals are mapped (or converted) to finite
frequency range discrete time signals.
The relation between frequency of analog and discrete time signal is,
F .....(2.1)
f=
Fs
2. 9 Digital Signal Processing
The range of frequency of discrete time signal is,
1 1 .....(2.2)
− ≤f ≤
2 2
On substituting for f from equation (2.1) in equation (2.2) we get,
1 F 1 .....(2.3)
− ≤ ≤
2 Fs 2
On multiplying equation (2.3) by Fs we get,
Fs F
− ≤ F≤ s .....(2.4)
2 2
From equation (2.4) we can say that when an analog signal is sampled at a frequency Fs, the highest
analog frequency that can be uniquely represented by a discrete time signal will be F s /2.
The continuous time signal with frequency above Fs/2 will be represented as a signal within the range
+ Fs/2 to - Fs/2 . Hence the signal with frequency above Fs/2 will have an identical signal with frequency
below Fs/2 in the discrete form.
Hence infinite number of high frequency continuous time signals will be represented by a single
discrete time signal. Such signals are called alias.
The phenomenon of high-frequency component getting the identity of low-frequency component
during sampling is called aliasing.
Sampling an analog signal with frequency F by choosing a sampling frequency Fs such that Fs/2 > F
will not result in alias. But sampling frequency is selected such that Fs/2 < F that the frequency above Fs/2 will
have alias with frequency below Fs/2. Hence the point of reflection is Fs/2, and the frequency Fs/2 is called
folding frequency.
The discrete time sinusoids, A sin [(2pf0 + 2pk)n], will be alias for integer values of k. It is also
observed that, a sinusoidal signal with frequency F1 will be an alias of sinusoidal signal with frequency F2 if
it is sampled at a frequency Fs = F1 - F2. In general, if the sampling frequency is any multiple of F1 - F2,
[i.e., Fs = k(F1 - F2) where k = 1, 2, 3, ........] the signal with frequency F2 will be an alias of the signal with
frequency F1.
Let, Fmax be maximum frequency of analog signal that can be uniquely represented as discrete time
signal when sampled at a frequency Fs.
Fs .....(2.5)
Now, Fmax =
2
∴Fs = 2Fmax .....(2.6)
The equation (2.6) gives a choice for selecting sampling frequency. From equation (2.6) we can say
that for unique representation of analog signal with maximum frequency Fmax, the sampling frequency should
be greater than 2Fmax.
i.e., to avoid aliasing Fs ³ 2Fmax ..... (2.7)
When sampling frequency Fs is equal to 2Fmax, the sampling rate is called Nyquist rate.
It is observed that a nonshifted sinusoidal signal when sampled at Nyquist rate, will produce zero
sample sequence (i.e., discrete sequence with all zeros), (because the sinusoidal signal is sampled at its zero
crossings, Refer example 2.3). Hence to avoid zero sampling of sinewave, the sampling frequency Fs should be
greater than 2Fmax, where Fmax is the maximum frequency in the analog signal.
Chapter 2 - Discrete Time Signals and Systems 2. 10
A discrete signal obtained by sampling can be reconstructed to an analog signal, only when it is
sampled without aliasing. The above concepts of sampling analog signals are summarized as the sampling
theorem, given below.
Sampling Theorem : A band limited continuous time signal with highest frequency (bandwidth)
Fm hertz can be uniquely recovered from its samples provided that the sampling
rate Fs is greater than or equal to 2Fm samples per second.
Note : The effects of aliasing in frequency spectrum are discussed in Chapter-4.

Example 2.1
Consider the analog signals, x1(t) = 3 cos 2p(20t) and x2(t) = 3 cos 2p(70t).
Find a sampling frequency so that 70Hz signal is an alias of the 20Hz signal?

Solution
Let, the sampling frequency, Fs = 70 - 20 = 50Hz.

∴ x1(n) = x1(t) n = 3 cos 2π(20t) = 3cos 2π

FG 20 × n IJ = 3 cos 4π n
t = nT =
Fs t =
n
Fs
H 50 K 5

x 2 (n) = x 2(t) n = 3 cos 2π(70t) = 3cos 2π

FG 70 × nIJ
t = nT =
Fs t =
n
Fs
H 50 K
For integer values of n
= 3cos
14π
n = 3cos 2πn +
4π FG
n = 3 cos
4π
n
IJ
5 5 H 5 K cos(2pn + q) = cos q
From the above analysis, we observe that x1(n) and x2(n) are identical , and so x2(t) is an alias of x1 (t) when
sampled at a frequency of 50 Hz.

Example 2.2
Let an analog signal, xa(t) = 10 cos 200 pt. If the sampling frequency is 150Hz, find the discrete time signal
x(n). Also find an alias frequency corresponding to Fs = 150 Hz.

Solution
n
x(n) = x a (t) n = 10 cos 200πt n
= 10 cos 200 π ×
t = nT =
Fs t = Fs
Fs

= 10 cos
200π × n
= 10 cos
4π
n = 10 cos 2π −
2π FG
n = 10 cos
2π IJ 1
n = 10 cos 2π n
150 3 3 H3 K 3

We know that the discrete time sinusoids whose frequencies are separated by integer multiples of 2p are
identical.

∴ 10 cos
2π
n = 10 cos
2π FG
+ 2π n = 10 cos
8π IJ 4
n = 10 cos 2π n
3 3 H 3 K 3
4 2π
Now, 10 cos 2π n is an alias of 10 cos n.
3 3
4
Here the frequency of the signal, 10 cos 2π n is,
3
4
f= cycles / sample
3
F 4
We know that, f = ⇒ F = f Fs = × 150 = 200Hz
Fs 3
∴ when, Fs = 150Hz, F = 200Hz is an alias frequency.
2. 11 Digital Signal Processing

Example 2.3
Consider the analog signal, xa(t) = 6 cos50pt + 3 sin 200pt – 3 cos100pt .
Determine the minimum sampling frequency and the sampled version of analog signal at this frequency.
Sketch the waveform and show the sampling points. Comment on the result.
Solution
The given analog signal can be written as shown below.
xa(t) = 6 cos50pt + 3 sin 200pt – 3 cos100pt = 6 cos 2p F1t + 3 sin 2p F2t – 3cos 2p F3t
Where, 2p F1 = 50p ; ÞF1 = 25Hz
2p F2 = 200p ; Þ F2 = 100Hz
2p F3 = 100p ;Þ F3 = 50Hz
The maximum analog frequency in the signal is 100Hz. The sampling frequency should be twice that of
this maximum analog frequency.
i.e., Fs ³ 2 Fmax Þ Fs ³ 2 ´ 100
Let, sampling frequency, Fs = 200Hz

∴ x a (nT) = x a ( t ) t = nT = x a (t) n
t =
Fs

50 πn 200 πn 100 πn πn πn
= 6 cos + 3 sin − 3cos = 6 cos + 3 sin πn − 3 cos
200 200 200 4 2
For integer values of n, sinpn = 0.
πn πn
∴ x a (nT ) = 6 cos
− 3 cos
4 2
The components of analog waveform and the sampling points are shown in fig1.
Comment : In the sampled version of analog signal xa (nT), the component 3 sin 200pt will give always zero
samples when sampled at 200Hz for any value of n. This is the drawback in sampling at Nyquist rate (i.e.,
sampling at Fs = 2Fmax).

1
6 c os 50 πt ; F1 = 25 H z ; T1 = = 0.04 s ec
F1

1
3 sin 200 πt ; F 2 = 100 H z ; T 2 = = 0.01 sec
F2

1
3 c os 100 πt ; F3 = 50 H z ; T3 = = 0.02 sec
F3
0.005
1
0.01 Fs = 200 H z ; T = = 0.005 sec
0.02 Fs
0.04

F ig 1 : S a m p lin g p o ints of th e co m po n en ts o f the sig n al x a (t).

Chapter 2 - Discrete Time Signals and Systems 2. 12
2.4 Classification of Discrete Time Signals
The discrete time signals are classified depending on their characteristics. Some ways of
classifying discrete time signals are,
1. Deterministic and nondeterministic signals
2. Periodic and aperiodic signals
3. Symmetric and antisymmetric signals
4. Energy and power signals
5. Causal and noncausal signals
2.4.1 Deterministic and Nondeterministic Signals
The signals that can be completely specified by mathematical equations are called deterministic
signals.The step, ramp, exponential and sinusoidal signals are examples of deterministic signals.
The signals whose characteristics are random in nature are called nondeterministic signals.
The noise signals from various sources are best examples of nondeterministic signals.
2.4.2 Periodic and Aperiodic Signals
When a discrete time signal x(n), satisfies the condition x(n + N) = x(n) for integer values of N, then the
discrete time signal x(n) is called periodic signal. Here N is the number of samples of a period.
i.e, if, x(n + N) = x(n), for all n, then x(n) is periodic.
The smallest value of N for which the above equation is true is called fundamental period. If there is
no value of N that satisfies the above equation, then x(n) is called aperiodic or nonperiodic signal.
When N is the fundamental period, the periodic signals will also satisfy the condition x(n + kN) = x(n),
where k is an integer. The periodic signals are power signals. The discrete time sinusoidal and complex
exponential signals are periodic signals when their fundamental frequency, f0 is a rational number.
x(n )
N=5
2 2 2 2 2 2 2 2

1 1 1 1

0 n
−1 −1 −1 −1 −1 −1 −1 −1

x (n) = {. . . . . . .2 , 1 , 2, − 1, − 1, 2 , 1 , 2, − 1, − 1, 2 , 1, 2 , − 1 , − 1 , . . . . . . . }
A
F ig 2.1 0 : P erio d ic d iscrete tim e sig n a l.
When a discrete time signal is a sum or product of two periodic signals with fundamental periods N1
and N2, then the discrete time signal will be periodic with period given by LCM of N1 and N2.

Example 2.4
Determine whether following signals are periodic or not. If periodic find the fundamental period.

a) x(n) = cos
FG 5π n + 1IJ b) x(n) = sin
FG n − πIJ c) x(n) = sin
π 2
n
H9 K H9 K 8
j7 πn 3 πn
5πn j
d) x(n) = e 4 e) x(n) = 2cos + 3e 4
3
2. 13 Digital Signal Processing
Solution

a) Given that, x(n) = cos

FG 5π n +1IJ
H9 K
Let N and M be two integers.

Now, x(n + N) = cos G

F 5π (n + N) + 1IJ = cos FG 5πn + 1 + 5π NIJ
H9 K H9 9 K
5π
Since, cos(q + 2pM) = cos q, for periodicity N should be integral multiple of 2π.
9
5π
Let , N = M × 2π
9
9 18M
∴ N = M × 2π × =
5π 5
Here N is an integer if, M = 5, 10, 15, 20, .......
Let, M = 5 ; \ N = 18

When N = 18 ; x(n + N) = cos

FG 5πn + 1+ 5π × 18IJ = cos FG 5πn + 1+ 10πIJ = cos FG 5πn + 1IJ = x(n)
H9 9 K H9 K H9 K
Hence x(n) is periodic with fundamental period of 18 samples.

b) Given that, x(n) = sin

FG n − πIJ
H9 K
Let N and M be two integers.

Now, xbn + Ng = sin G

F n + N − πIJ = sin FG n + N − πIJ = sin FG n − π + NIJ
H 9 K H9 9 K H9 9K
N
Since, sin(q + 2pM) = sin q, for periodicity should be equal to integral multiple of 2p.
9
N
Let, = M × 2π
9
\ N = 18 pM
Here N cannot be an integer for any integer value of M and so x(n) will not be periodic.

c) Given that, x(n) = sin

FG π n IJ
2
H8 K
π π
(n + N)2 = sin (n2 + N2 + 2nN) = sin
π 2 πN2 πNF I
∴ x(n + N) = sin
8 8 8
n +
8
+ GH
4
n JK
πN2 πN
Let , = 2πM1 Let , = 2πM2
8 4

∴ N = 4 M1 \ N = 8 M2
Now, N is integer for M1 = 12, 22, 32, 42 ..... Now, N is integer for M2 = 1, 2, 3, 4 .....
2
When M1 = 2 and M2 = 1, we get a common value for N as, N = 8.
F π n + π8 + π8 nI
2
2
For interger M,
When N = 8 ; x(n + N) = sin GH 8 8 4 JK sin (q + 2p M) = sinq
FF π I I Fπ
= sin G G n + 2πnJ + 4 × 2πJ = sin G n
2 2
+ 2πn
IJ
HH 8 K K H8 K
π 2
= sin
n = x(n)
8
\ x(n) is periodic with fundamental period, N = 18 samples.
Chapter 2 - Discrete Time Signals and Systems 2. 14
j7 πn
d) Given that, x(n) = e 4

Let N and M be two integers.

j7 π (n+N) j7 πn j7 πN
Now, x(n + N) = e 4 =e 4 e 4

7 πN
Since, ej2pM = 1, for periodicity should be an integral multiple of 2p.
4
7 πN
Let , = M × 2π,
4
4 8M
∴ N = M × 2π × =
7π 7
Here, N is integer, when M = 7, 14, 21, ......
When M = 7 ; N = 8
\ x(n) is periodic with fundamental period of 8 samples.

3π n
5πn j
e) Given that, x(n) = 2 cos + 3e 4
3
Let , x(n) = x1(n) + x 2 (n)
5πn
where, x1(n) = 2 cos
3
3πn
j
x 2 (n) = 3e 4

5πn j
3 πn
Consider, x1(n) = 2 cos Consider, x 2 (n) = 3 e 4
3

b g
∴ x1 n + N1 = 2 cos
b
5 π n + N1 g j
3π(n +N2 )

3 b g
∴ x 2 n + N2 = 3 e 4

FG 5πn + 5πN IJ 1 .....(1) j

FG 3πn + 3πN2 IJ .....(2)
=3eH 4 K
= 2 cos
H3 3 K 4

5πN1 6 3πN2 8
Let , = 2πM1 ⇒ N1 = M1 Let , = 2πM2 ⇒ N2 = M2
3 5 4 3
Let, M1 = 5 ; \ N1 = 6 Let, M2 = 3 ; \ N2 = 8
Substitute N1 = 6 in equation (1), Substitute N2 = 8 in equation (2),

b g
∴ x1 n + N1 = 2 cos
FG 5πn + 5π × 6IJ j
FG 3πn + 3π × 8 IJ
H4 4 K
H3 3 K b g
∴ x 2 n + N2 = 3e
F 5πn + 5 × 2πIJ
= 2 cos G j
FG 3πn +3× 2πIJ
H4 K
For integer M,
H3 K For integer M,
= 3e
5πn 3πn
cos(q +2pM) = cosq = 2cos = x1(n) ej(q + 2pM) = ejq j
4 =
3 = 3e x 2 (n)

\ x1(n) is periodic with fundamental period, \ x2(n) is periodic with fundamental period,
N1 = 6 samples. N2 = 8 samples.
Here, x(n) = x1(n) + x2(n), and x1(n) is periodic with period N1 = 6, and x2(n) is periodic with period N2 = 8.
Therefore, x(n) is periodic with period N, where N is LCM of N1 and N2.
The LCM of 6 and 8 is 24.
\ N = 24
\ x(n) is periodic with fundamental period, N = 24.
2. 15 Digital Signal Processing

2.4.3 Symmetric (Even) and Antisymmetric (Odd) Signals

The discrete time signals may exhibit symmetry or antisymmetry with respect to n = 0. When a
discrete time signal exhibits symmetry with respect to n = 0 then it is called an even signal. Therefore, the
even signal satisfies the condition,
x(-n) = x(n)
When a discrete time signal exhibits antisymmetry with respect to n = 0, then it is called an odd signal.
Therefore the odd signal satisfies the condition,
x(-n) = -x(n)
x (n ) x (n )
3 3

2 2 2 2
2
1 1
1 1 1 1

−4 −3 −2 −1 0 1 2 3 4 n
−4 −3 −2 −1 0 1 2 3 4 n
−1 −1

−2 −2
x (n) = {1 , 2 , 3 , 1 , 2 , 1 , 3 , 2 , 1} x (n ) = {1, 2, − 2, − 1 , 0 , 1 , 2 , − 2, − 1 }
A
F ig 2.11 a : S y m m e tric (or e v en ) sig na l.
A
F ig 2.11 b : A ntisym m etric (o r o dd ) sig n al.
F ig 2.11 : S y m m etric a nd a ntisym m etric d iscrete tim e sig n al.

A discrete time signal x(n) which is neither even nor odd can be expressed as a sum of even and
odd signal.
Let, x(n) = xe (n) + xo (n)
bg
where, x e n = Even part of x(n)
xo(n) = Odd part of x(n)
Note : If x(n) is even then its odd part will be zero. If x(n) is odd then its even part will be zero.
Now, it can be proved that,
1
Even part, xe (n) = x(n) + x(− n)
2
1
Odd part, xo (n) = x(n) − x(− n)
2
Proof :
Let, x(n) = xe(n) + xo(n) ......(2.8)

On replacing n by –n in equation (2.8) we get,

x(–n) = xe(–n) + xo(–n) ......(2.9)

Since xe(n) is even, xe(–n) = xe(n)

Since xo(n) is odd, xo(–n) = – xo(n)

Hence the equation (2.9) can be written as,

x(–n) = xe(n) – xo(n) ......(2.10)

Chapter 2 - Discrete Time Signals and Systems 2. 16
On adding equation (2.8) and (2.10) we get,
x(n) + x(–n) = 2 xe(n)

1
∴ x e(n) = x(n) + x(− n)
2
On subtracting equation (2.10) from equation (2.8) we get,

x(n) – x(–n) = 2 xo(n)

1
∴ x o(n) = x(n) − x(− n)
2

Example 2.5
Determine the even and odd parts of the signals.
π
j n
a) x(n) = 3n b) x(n) = 3 e 5 c) x(n) = {2, − 2, 6, − 2}
Solution
a) Given that, x(n) = 3n
∴ x(−n) = 3−n
1 1 n
Even part, x e (n) = [x(n) + x( −n)] = [3 + 3−n ]
2 2
1 1 n
Odd part, x o (n) = [x(n) − x( −n)] = [3 − 3−n ]
2 2
π
j n
b) Given that, x(n) = 3 e 5

π
j n π π
x(n) = 3 e 5 = 3 cos n + j3 sin n
5 5
π
−j n π π
∴ x(−n) = 3 e 5 = 3 cos n − j3 sin n
5 5
1
Even part, x e (n) = [x(n) + x(−n)]
2
=
1 πLM π π π
3 cos n + j3sin n + 3 cos n − j3 sin n =
1 π OP
π
6 cos n = 3 cos n
LM OP
2 5 N 5 5 5 2 5 Q
5 N Q
1
Odd part, x o (n) = [x(n) − x( −n)]
2

=
LM 1 π π π π
3 cos n + j3sin n − 3 cos n + j3sin n
OP
N 2 5 5 5 5 Q
1L π O π
= Mj6 sin nP = j3 sin n
2N 5 Q 5

c) Given that, x(n) = {2, − 2, 6, − 2}

A
Given that, x(n) = {2, –2, 6, –2}, \ x(0) = 2 ; x(1) = –2 ; x(2) = 6 ; x(3) = –2
-
x(–n) = {–2, 6, –2, 2}, \ x(–3) = –2 ; x(–2) = 6 ; x(–1) = –2 ; x(0) = 2
-
­
2. 17 Digital Signal Processing
1 1
Even part, x e (n) = [x(n) + x( −n)] Odd part, x o (n) = [x(n) − x( −n)]
2 2

At n = – 3 ; x(n) + x(–n) = 0 + (–2) = – 2 At n = – 3 ; x(n) – x(–n) = 0 – (–2) = 2

At n = – 2 ; x(n) + x(–n) = 0 + 6 = 6 At n = – 2 ; x(n) – x(–n) = 0 – 6 =–6

At n = – 1 ; x(n) + x(–n) = 0 + (–2) = – 2 At n = – 1 ; x(n) – x(–n) = 0 – (–2) = 2

At n = 0 ; x(n) + x(–n) = 2 + 2 = 4 At n = 0 ; x(n) – x(–n) = 2–2 = 0

At n = 1 ; x(n) + x(–n) = – 2 + 0 =–2 At n = 1 ; x(n) – x(–n) = – 2 – 0 =–2

At n = 2 ; x(n) + x(–n) = 6 + 0 = 6 At n = 2 ; x(n) – x(–n) = 6–0 = 6

At n = 3 ; x(n) + x(–n) = – 2 + 0 =–2 At n = 3 ; x(n) – x(–n) = – 2 – 0 =–2

∴ x(n) + x(−n) = {−2, 6, − 2, 4, − 2, 6, − 2} \ x(n) – x(–n) = {2, –6, 2, 0, –2, 6, –2}

-
A
1 1
∴ x e (n) = [x(n) + x(−n)] ∴ x o (n) = [x(n) − x( −n)]
2 2
= {−1, 3, − 1, 2, − 1, 3, − 1} = {1, − 3, 1, 0, − 1, 3, − 1}
A -

2.4.4 Energy and Power Signals

The energy E of a discrete time signal x(n) is defined as,
∞
2 .....(2.11)
Energy, E = ∑ x( n)
n= −∞

The energy of a signal may be finite or infinite, and can be applied to complex valued and
real valued signals.
If energy E of a discrete time signal is finite and nonzero, then the discrete time signal is called an
energy signal. The exponential signals are examples of energy signals.
The average power of a discrete time signal x(n) is defined as,
N
1 2 .....(2.12)
Power , P = lim
N→∞ 2N + 1 ∑ x( n)
n =− N

If power P of a discrete time signal is finite and nonzero, then the discrete time signal is called a power
signal. The periodic signals are examples of power signals.
For energy signals, the energy will be finite and average power will be zero. For power signals the
average power is finite and energy will be infinite.

\ For energy signal, 0 < E < ∞ and P = 0

For power signal, 0 < P < ∞ and E = ∞

Chapter 2 - Discrete Time Signals and Systems 2. 18
Example 2.6
Determine whether the following signals are energy or power signals.

a) x(n) =
FG 1IJ n
u(n) b) x(n) = sin
FG π nIJ c) x(n) = u(n)
H 4K H3 K
Solution

a) Given that, x(n) =

FG 1 IJ u(n) n

H 4K
F 1I n

Here, x(n) = G J u(n) for all n.

H 4K
F 1I
∴ x(n) = G J = 0.25 ;
n
n
n≥0
H 4K Infinite geometric series
+∞ ∞ 2 ∞ sum formula.
(0.25)n =
2
Energy, E = ∑ x(n) = ∑ ∑ (0.252 )n ∞
n 1
n = −∞ n=0 n=0 ∑ C =
1 − C
n = 0
∞
1
= ∑ (0.0625)n = = 1.067 joules Using infinite geometric series sum formula.
n=0
1 − 0.0625

+N N
1 2 1 2
Power, P = Lt ∑ x(n) = Lt ∑ (0.25)n
N→∞ 2N + 1 n = −N
N→ ∞ 2N + 1 n = 0

N N
1 1
= Lt ∑ (0.252 )n = Lt ∑ (0.0625)n
N→∞ 2N + 1 n = 0
N→∞ 2N + 1 n = 0

1 (0.0625)N +1 − 1 Using finite geometric series sum formula.

= Lt
N→∞ 2N + 1 0.0625 − 1
Finite geometric series
1 0.0625∞ − 1 sum formula.
= × =0 N
∞ 0.0625 − 1 n CN + 1 − 1
∑ C =
C −1
n = 0
Here E is finite and P is zero and so x(n) is an energy signal.

b) Given that, x(n) = sin

FG π nIJ 1 − cos2θ
H3 K sin2θ =
2
2π
1 − cos n
Energy, E =
+∞

∑ x(n) =
2
∑
+∞
sin2
FG π nIJ = ∑ +∞
3
n = −∞ n = −∞
H3 K n = −∞ 2

1 F FG1 − cos 2π nIJ I = 1 F

+∞ +∞ +∞
2π 1I
=
2 GH ∑ H
n = −∞ 3 K JK 2 GH ∑ n = −∞
1n − ∑
n = −∞
cos
3 JK
n = (∞ − 0) = ∞
2

Note : Sum of infinite 1's is infinity. Sum of samples of one period of cosinusoidal signal is zero.
N N
1 2 1 πn
Power, P = Lt
N→ ∞
2N + 1
∑ x(n) = Lt
N→∞
2N + 1
∑ sin2
3
n = −N n = −N
2. 19 Digital Signal Processing

FG1 − cos 2π nIJ

∴ P = Lt
1
∑
H N
3 K
N→∞ 2N + 1 n = −N
2

1 L
MM ∑ 1 − ∑ cos 23π nOPP
N N
1 n
= Lt
N→∞ 2N + 1 2 N n = −N Q n = −N

= Lt
1 1 L
M1+ 14 +.......+
O
1+31 − 0P
31+ 1+ 11+.......+
N→∞ 2N + 1 MN
2 144 2444 44
N terms
42444
PQ N terms

1 1 1 1
= Lt 2N + 1 = Lt = watts
N→∞ 2N + 1 2 N→∞ 2 2

Since P is finite and E is infinite, x(n) is a power signal.

2π 2π
Note: The term cos n is periodic with periodicity of 3 samples. Samples of cos n for two periods are
3 3
given below. It can be observed that sum of samples of a period is zero.
2π 2π 2π
When n = 0 ; cos n = 1, When n = 1 ; cos n = −0.5, When n = 2 ; cos n = −0.5
3 3 3
2π 2π 2π
When n = 3 ; cos n = 1, When n = 4 ; cos n = −0.5, When n = 5 ; cos n = −0.5
3 3 3

c) Given that, x(n) = u(n)

+∞ +∞
E= ∑ x(n) =
2
∑ bu(n)g 2

n = −∞ n =0
+∞
= ∑ u(n) = 1+ 1+ 1 . ........ ∞ = ∞
n =0

1 N
1 N
1 F I
G 3JJ
2
P = Lt ∑ x(n) = Lt ∑ u(n) = Lt
G 1+ 1+ 1+.........+1
N→∞
2N + 1 n = −N
N→∞
2N + 1 n =0
N→∞
2N + 1 H
1444 424444
N + 1 terms K
FG 1 IJ
N 1+ 1
= Lt
1
(N + 1) = Lt
H NK = 1+ ∞ = 1+ 0 = 1 watts
N→∞
2N + 1 N→∞ F 1I 2 + 1 2 + 0 2
N G2 + J
H NK ∞

Since P is finite and E is infinite, x(n) is a power signal.

2.4.5 Causal, Noncausal and Anticausal signals

A discrete time signal is said to be causal, if it is defined for n ³ 0. Therefore if x(n) is causal, then
x(n) = 0 for n < 0.

A discrete time signal is said to be noncausal, if it is defined for either n ≤ 0, or for both n ≤ 0 and
n > 0. Therefore if x(n) is noncausal, then x(n) ≠ 0 for n < 0. A noncausal signal can be converted to causal
signal by multiplying the noncausal signal by a unit step signal, u(n).

When a noncausal discrete time signal is defined only for n ≤ 0, it is called an anticausal signal.
Chapter 2 - Discrete Time Signals and Systems 2. 20
Examples of Causal and Noncausal Signals

x(n) = {1, –1, 2, –2, 3, –3}

123
3
-

3 12
Causal signals
x(n) = {2, 2, 3, 3,.............}
-
x(n) = {1, –1, 2, –2, 3, –3}

123

123
-

12
Anticausal signals
x(n) = {............,2, 2, 3, 3}
-
Noncausal signals
x(n) = {2, 3, 4, 5, 4, 3, 2}
-
x(n) = {......, 2, 3, 4, 5, 4, 3, 2,......}
-

2.5 Mathematical Operations on Discrete Time Signals

Some of the mathematical operations that can be performed on discrete time signals are,

1. Scaling : Amplitude scaling and time scaling

2. Folding

3. Shifting : Right shift (or advance) and left shift (or delay)

4. Addition

5. Multiplication

2.5.1. Scaling of Discrete Time Signals

Amplitude Scaling (or Scalar Multiplication)

Amplitude scaling of a discrete time signal by a constant A is accomplished by multiplying the value
of every signal sample by the constant A.

Example :

Let y(n) be amplitude scaled signal of x(n), then y(n) = A x(n)

Let, x(n) = 10 ; n=0 and A = 0.2, When n = 0 ; y(0) = A x(0) = 0.2 ´ 10 = 2.0

= 16 ; n=1 When n = 1 ; y(1) = A x(1) = 0.2 ´ 16 = 3.2

= 20 ; n=2 When n = 2 ; y(2) = A x(2) = 0.2 ´ 20 = 4.0

Time Scaling (or Downsampling and Upsampling)

There are two ways of time scaling a discrete time signal. They are downsampling and upsampling.

In a signal x(n), if n is replaced by Dn, where D is an integer, then it is called downsampling.

n
In a signal x(n), if n is replaced by , where I is an integer, then it is called upsampling.
I
2. 21 Digital Signal Processing
Example :
If x(n) = bn ; n ³ 0 ; 0 < b < 1, then

x1(n) = x(2n) will be a down sampled version of x(n) and

x2(n) = x n will be an up sampled version on x(n).

e j
2
When n = 0 ; x1(0) = x(0) = b0 When n = 0 ; x 2 (0) = x 0 = x(0) = b 0
ej
2
When n = 1 ; x1(1) = x(2) = b2 When n = 1 ; x (1) = x e 1 j = 0
2
2
When n = 2 ; x1(2) = x(4) = b4 and so on.
When n = 2 ; x (2) = x e 2 j = x(1) = b
2
1
2

When n = 3 ; x (3) = x e 3 j = 0 and so on.

2
2

x (n ) x 1 (n) x 2 (n)
b
0

1
b0 x 1 (n) = x(2n) b
0
x 2 (n) = x ej
n
2
b
b1
b2 b2
2
b
3
4
b
b 3
b4 b
b5 6 b6
b
0 1 2 3 4 5
0 1 2 3 6 n 0 1 2 3 4 5 6 n
n
F ig 2.12 a : A d isc re te tim e sig n al x (n ). F ig 2.12 b : D o w n sa m p le d sig na l o f x (n). F ig 2.12 c : U p sa m p le d sig n al x (n ).
F ig 2.1 2 : A d iscrete tim e sign a l a n d its tim e sca led versio n .

2.5.2. Folding (or Reflection or Transpose) of Discrete Time Signals

The folding of a discrete time signal x(n) is performed by changing the sign of the time base n in x(n).
The folding operation produces a signal x(–n) which is a mirror image of the signal x(n) with respect to time
origin n = 0.

Example :

Let x(n) = 0.8n ; –2 £ n £ 2. Now the folded signal, x1(n) = x(–n) = –0.8n ; –2 £ n £ 2

x (n ) x 1 (n)
x 1 (n) = x ( −n)
1.6 1.6
0.8 0.8

−2 −1 0 1 2 n −2 −1 0 1 2 n
−0.8 −0.8
−1.6 −1.6

F ig 2.13a : A d isc re te tim e sig n al x (n ). F ig 2.13b : F o ld e d sig n al o f x(n ).

F ig 2.1 3 : A d iscrete tim e sign a l a n d its fo ld ed v ersio n .
Chapter 2 - Discrete Time Signals and Systems 2. 22
2.5.3. Time Shifting of Discrete Time Signals
A signal x(n) may be shifted in time by replacing the independent variable n by n – m, where m is an integer.
[i.e, x(n–m) is shifted version of x(n)]. If m is a positive integer, the time shift results in a delay by m units of time. If
m is a negative integer, the time shift results in an advance of the signal by |m| units in time. The delay results in
shifting each sample of x(n) to the right. The advance results in shifting each sample of x(n) to the left.
Example :
Let, x(n) = 3 ; n = 2
=2 ; n=3
=1 ; n=4
= 0 ; for other n
Let, x1(n) = x(n – 2), where x1(n) is delayed Let, x2(n) = x(n + 2), where x2(n) is an advanced
signal of x(n) signal of x(n)
When n = 4 ; x1(4) = x(4 – 2) = x(2) =3 When n = 0 ; x2(0) = x(0 + 2) = x(2) = 3
When n = 5 ; x1(5) = x(5 – 2) = x(3) =2 When n = 1 ; x2(1) = x(1 + 2) = x(3) = 2
When n = 6 ; x1(6) = x(6 – 2) = x(4) =1 When n = 2 ; x2(2) = x(2 + 2) = x(4) = 1
The sample x(2) is available at n = 2 in The sample x(2) is available at n = 2 in the original
the original sequence x(n), but the same sample sequence x(n), but the same sample is available at n = 0
is available at n = 4 in x1(n). Similarly every sample in x2(n). Similarly every sample of x(n) is advanced by two
of x(n) is delayed by two sampling times. sampling times. Hence the signal x2(n) is an advanced
version of x(n).
x (n )
3 x 1 (n) x1(n) = x(n−2) x 2 (n)
3
2 3
1 2 2
1 1
0 1 2 3 4 5 6 n
0 1 2 3 4 5 6 n 0 1 2 3 4 5 6 n
VI sampling time
II sampling time

Vth sampling time

Ist sampling time

III sampling time

IV sampling time
Time origin

F ig 2.14b : D e la y ed sig n a l o f x (n ). F ig 2.14c : A dv a n ce d sig n al o f x(n ).

th
nd

th
rd

F ig 2.14a : A disc re te tim e sig n a l x (n) .

F ig 2.1 4 : A d iscrete tim e sign a l a n d its sh ifted version .

Delayed Unit Impulse Signal x 3 (n)

x 3 (n) = δ(n −m )
The unit impulse signal is defined as, 1

d(n) = 1 ; for n = 0
0 m n
= 0 ; for n ¹ 0 F ig 2.1 5 : D elayed u nit im pu lse.
The unit impulse signal delayed by m units of time is denoted as d(n – m).
Now, d(n – m) = 1 ; n = m
= 0 ; n ¹m
2. 23 Digital Signal Processing
Delayed Unit Step Signal x 4 (n)
x 4 (n) = u(n −m )
1
The unit step signal is defined as,

u(n) = 1 ; for n ³ 0 n

m +1
0

m + 2

m + 4
m +3
m
= 0 ; for n < 0 F ig 2.1 6 : D elay ed un it step sig n a l.
The unit step signal delayed by m units of time is denoted as u(n – m).

Now, u(n – m) = 1 ; n ³ m
=0;n<m

2.5.4. Addition of Discrete Time Signals

The addition of two discrete time signals is performed on a sample-by-sample basis.
The sum of two signals x1(n) and x2(n) is a signal y(n), whose value at any instant is equal to the sum
of the samples of these two signals at that instant.
i.e., y(n) = x1(n) + x2(n) ; -¥ < n < ¥ .
Example :

Let, x1(n) = {2, 2, –1} and x2(n) = {–1, 1, 2}

When n = 0 ; y(0) = x1(0) + x2(0) = 2 + (–1) = 1

When n = 1 ; y(1) = x1(1) + x2(1) = 2 + 1 =3

When n = 2 ; y(2) = x1(2) + x2(2) = –1 + 2 = 1

\ y(n) = x1(n) + x2(n) = {1, 3, 1}

2.5.5. Multiplication of Discrete Time Signals

The multiplication of two discrete time signals is performed on a sample-by-sample basis.The
product of two signals x1(n) and x2(n) is a signal y(n), whose value at any instant is equal to the product of
the samples of these two signals at that instant. The product is also called modulation.
Example :

Let, x1(n) = { 2, 2, –1 } and x2(n) = { –1, 1, 2 }

When n = 0 ; y(0) = x1(0) ´ x2(0) = 2 ´ (–1) = –2

When n = 1 ; y(1) = x1(1) ´ x2(1) = 2 ´ 1 = 2

When n = 2 ; y(2) = x1(2) ´ x2(2) = –1 ´ 2 = –2

\ y(n) = x1(n) ´ x2(n) = {–2, 2, –2}

2.6 Discrete Time System

A discrete time system is a device or algorithm that operates on a discrete time signal, called the input
or excitation, according to some well-defined rule, to produce another discrete time signal called the output
or the response of the system. We can say that the input signal x(n) is transformed by the system into a
signal y(n), and the transformation can be expressed mathematically as shown in equation (2.13).The
diagrammatic representation of discrete time system is shown in fig 2.17.
Chapter 2 - Discrete Time Signals and Systems 2. 24
Response, y(n) = H{x(n)} .....(2.13)
where, H denotes the transformation (also called an operator).
D isc re te tim e
syste m
In p u t sign a l O u tp u t sign a l
x (n ) y (n )
or H or
E x cita tion R e sp o n se
F ig 2.1 7 : R ep resen tation of d iscrete tim e system .
LTI System

A discrete time system is linear if it obeys the principle of superposition and it is time invariant if its
input-output relationship does not change with time. When a discrete time system satisfies the properties of
linearity and time invariance then it is called an LTI system (Linear Time Invariant system).

Impulse Response

When the input to a discrete time system is a unit impulse d(n) then the output is called an impulse
response of the system and is denoted by h(n).

\ Impulse Response, h(n) = H{d(n)} .....(2.14)

δ(n ) h (n )
H
F ig 2 .18 : D iscrete tim e system w ith im p u lse in p u t.

2.6.1 Mathematical Equation Governing Discrete Time System

The mathematical equation governing the discrete time system can be developed as shown below.
The response of a discrete time system at any time instant depends on the present input, past inputs
and past outputs.
Let us consider the response at n = 0. Let us assume a relaxed system and so at n = 0, there is no past
input or output. Therefore the response at n = 0, is a function of present input alone.
i.e., y(0) = F[x(0)]
Let us consider the response at n =1. Now the present input is x(1), the past input is x(0) and past
output is y(0). Therefore the response at n = 1, is a function of x(1), x(0), y(0).
i.e., y(1) = F[y(0), x(1), x(0)]
Let us consider the response at n = 2. Now the present input is x(2), the past inputs are x(1) and x(0),
and past outputs are y(1) and y(0). Therefore the response at n = 2, is a function of x(2), x(1), x(0), y(1), y(0).
i.e., y(2) = F[y(1), y(0), x(2), x(1), x(0)]
Similarly, at n = 3, y(3) = F[y(2),y(1), y(0), x(3), x(2), x(1), x(0)]
at n = 4, y(4) = F[y(3),y(2), y(1), y(0), x(4), x(3), x(2), x(1), x(0)] and so on.
In general, at any time instant n,
y(n) = F[y(n – 1), y(n – 2), y(n – 3), .....y(1), y(0), x(n), x(n – 1),
x(n – 2), x(n – 3) ..... x(1), x(0)] .....(2.15)
2. 25 Digital Signal Processing
For an LTI system, the response y(n) can be expressed as a weighted summation of dependent terms.
Therefore the equation (2.15) can be written as,
y(n) = – a1 y(n – 1) – a2 y(n – 2) – a3 y(n – 3) – ...........
+ b0 x(n) + b1 x( n – 1) + b2 x(n – 2) + b3 x(n – 3) +........ ..... (2.16)
where, a1, a2, a3, .... and b0, b1, b2, b3, ..... are constants.
Note : Negative constants are inserted for output signals, because output signals are feedback
from output to input. Positive constants are inserted for input signals, because input
signals are feed forward from input to output.
Practically, the response y(n) at any time instant n, may depend on N number of past outputs,
present input and M number of past inputs where M £ N. Hence the equation (2.16) can be written as,
y(n) = – a1 y(n – 1) – a2 y(n – 2) – a3 y(n – 3) – ........ – aN y(n – N)
+ b0 x(n) + b1 x(n – 1) + b2 x(n – 2) + b3 x(n – 3) + ....... +bM x(n – M)
N M
∴ b g ∑ a ybn − mg + ∑ b xbn − mg
y n =− m m
.....(2.17)
m=1 m=0

The equation (2.17) is a constant coefficient difference equation, governing the input-output relation
of an LTI discrete time system.
In equation (2.17) the value of "N" gives the order of the system.
If N = 1, the discrete time system is called 1st order system
If N = 2, the discrete time system is called 2nd order system
If N = 3, the discrete time system is called 3rd order system , and so on.
The general difference equation governing 1st order discrete time LTI system is,
y(n) = – a1 y(n – 1) + b0 x(n) + b1 x(n – 1)
The general difference equation governing 2nd order discrete time LTI system is,
y(n) = – a2 y(n – 2) – a1 y(n – 1) + b0 x(n) + b1 x(n – 1) + b2 x(n – 2)

2.6.2 Block Diagram and Signal Flow Graph Representation of Discrete Time System
The discrete time system can be represented diagrammatically by block diagram or signal flow
graph. These diagrammatic representations are useful for physical implementation of discrete time system in
hardware or software.
The basic elements employed in block diagram or signal flow graph are adder, constant multiplier, unit
delay element and unit advance element.
Adder : An adder is used to represent addition of two discrete time signals.
Constant Multiplier : A constant multiplier is used to represent multiplication of a scaling factor
(constant) to a discrete time signal.
Unit Delay Element : A unit delay element is used to represent the delay of samples of a discrete
time signal by one sampling time.
Chapter 2 - Discrete Time Signals and Systems 2. 26
Unit Advance Element : A unit advance element is used to represent the advance of samples of a
discrete time signal by one sampling time.
The symbolic representation of the basic elements of block diagram and signal flow graph are listed in
table 2.1.
Table 2.1 : Basic Elements of Block Diagram and Signal Flow Graph

Element Block diagram Signal flow

representation graph representation

x1 ( n )
x1 (n ) x1 ( n ) + x 2 ( n )
Adder +
x1 ( n ) + x 2 ( n )

x2 (n)
x2 (n)

x (n ) a x(n ) a
a x (n ) a x(n )
Constant multiplier

x (n ) x (n − 1 ) z −1
z −1 x (n ) x (n − 1 )
Unit delay element

x (n ) x (n + 1 ) z
z x (n ) x (n + 1 )
Unit advance element

Example 2.7
Construct the block diagram and signal flow graph of the discrete time systems whose input-output
relations are described by the following difference equations.
a) y(n) = 0.7 x(n) + 0.7 x(n – 1)
b) y(n) = 0.4 y(n – 1) + x(n) – 3 x(n – 2)
c) y(n) = 0.2 y(n – 1) + 0.7 x(n) + 0.9 x(n – 1)

Solution
a) Given that, y(n) = 0.7 x(n) + 0.7 x(n – 1)

The individual terms of the given equation are 0.7 x(n) and 0.7 x(n – 1). They are represented by basic
elements as shown below.
2. 27 Digital Signal Processing
Block diagram representation Signal flow graph representation

0.7
x(n) 0.7 0.7 x(n) x(n) 0.7 x(n)

−1
−1
z
0.7 x(n) z 0.7 x(n − 1) 0.7 x(n) 0.7 x(n − 1)

The input to the system is x(n) and the output of the system is y(n). The above elements are connected
as shown below to get the output y(n).

x(n) 0.7 x(n) 0.7 x(n − 1) y(n) −1

x(n) 0.7 0.7 x(n) z y(n)
−1
0.7 z +
1
0.7 x(n)

F ig 1 : B lo c k d ia g ra m o f th e syste m F ig 2 : S ig n al flow grap h of th e syste m

y (n ) = 0.7 x(n ) + 0 .7 x (n − 1 ). y (n ) = 0.7 x(n ) + 0 .7 x (n − 1 ).

b) Given that, y(n) = 0.4 y(n – 1) + x(n) – 3 x(n – 2)

The individual terms of the given equation are 0.4 y(n – 1) and – 3 x(n – 2). They are represented by
basic elements as shown below.
Block diagram representation Signal flow graph representation

−3 x(n − 2) 0.4 y (n − 1)
x (n) y (n)
x (n) y (n)

−1
z 0.4 −1
−1 −1 z
z z

x (n − 1) y (n − 1) x (n − 1)
−3 y (n − 1)
−1 0.4 y (n − 1)
z −1
0.4 z
x (n − 2)
−3 −3 x(n − 2)
x (n − 2)

The input to the system is x(n) and the output of the system is y(n). The above elements are connected
as shown below to get the output y(n).
x (n) y (n) x (n) 1 1 1 1 1 y (n)
+ +

−1
−1 z
−1 −1
z
z z 0.4

−3

−1
z −1
0.4 z

−3

F ig 3 : B lo c k d ia g ra m o f th e syste m F ig 4 : S ig n al flow grap h of th e syste m

describ e d b y th e e qu a tio n describ e d b y th e e qu a tio n
y(n ) = 0 .4 y (n − 1 ) + x(n ) − 3 x(n − 2 ). y(n ) = 0 .4 y (n − 1 ) + x(n ) − 3 x(n − 2 ).
Chapter 2 - Discrete Time Signals and Systems 2. 28
c) Given that, y(n) = 0.2 y(n – 1) + 0.7 x(n) + 0.9 x(n – 1)
The individual terms of the given equation are 0.2 y(n – 1), 0.7 x(n) and 0.9 x(n – 1). They are represented
by basic elements as shown below.
Block diagram representation Signal flow graph representation

x(n) 0.7 x(n) 0.7

0.7 x(n) 0.7 x(n)

x(n) x(n)
0.9 x(n − 1)

−1
z
−1
z
0 .9
0.9 x(n − 1)
0.9
x(n − 1) x(n − 1)

y(n)
0.2 y(n − 1)
y(n)
−1
z −1
z
0.2 y(n − 1) y(n − 1) 0.2
0.2
y(n − 1)

The input to the system is x(n) and the output of the system is y(n). The above elements are connected
as shown below to get the output y(n).

y(n) x(n) 1 0.7 1 1 y(n)

x(n)
0.7 + +

−1 −1 −1
z z
−1 z 0.9 0.2 z

0.9 0.2

F ig 5 : B lo c k d ia g ra m o f th e syste m d e scrib e d F ig 6 : S ig n al flow gra ph o f th e syste m d e scrib e d

b y th e eq u a tio n b y th e eq ua tio n
y (n) = 0 .2 y(n − 1 ) + 0.7 x (n )+ 0 .9 x (n − 1). y(n ) = 0 .2 y (n − 1 ) + 0 .7 x(n ) + 0.9 x (n − 1).

2.7 Response of LTI Discrete Time System in Time Domain

The general equation governing an LTI discrete time system is,
N M
bg
y n =− ∑
m=1
b
am y n − m + g ∑ b xbn − mg
m=0
m

N M
∴ y(n) + ∑ a m ybn − mg = ∑ bm xbn − mg
m=1 m= 0
N M
.....(2.18)
( or ) ∑ a m ybn − mg = ∑ b m xbn − mg with a o = 1
m= 0 m= 0

The solution of the difference equation (2.18) is the response y(n) of LTI system, which consists of
two parts. In mathematics, the two parts of the solution y(n) are homogeneous solution yh(n) and particular
solution yp(n).
2. 29 Digital Signal Processing
\ Response, y(n) = yh(n) + yp(n) .....(2.19)
The homogeneous solution is the response of the system when there is no input.The particular
solution yp(n) is the solution of difference equation for specific input signal x(n) for n ³ 0.
In signals and systems, the two parts of the solution y(n) are called zero-input response yzi(n) and
zero-state response yzs(n).
\ Response, y(n) = yzi(n) + yzs(n) .....(2.20)
The zero-input response is mainly due to initial conditions (or initial stored energy) in the system.
Hence zero-input response is also called free response or natural response. The zero-input response is given
by homogeneous solution with constants evaluated using initial conditions.
The zero-state response is the response of the system due to input signal and with zero initial
condition. Hence the zero-state response is called forced response.The zero-state response or forced response
is given by the sum of homogeneous solution and particular solution with zero initial conditions.

2.7.1 Zero-Input Response or Homogeneous Solution

The zero-input response is obtained from homogeneous solution yh(n) with constants evaluated
using initial condition.
∴ Zero - input response, y zi ( n) = y h ( n) with constants evaluated using initial conditions

The homogeneous solution is obtained when x(n) = 0. Therefore the homogeneous solution is the
solution of the equation,
N

∑ a m y( n − m) = 0 .....(2.21)
m= 0

Let us assume that the solution of equation (2.21) is in the form of an exponential.
i.e., y(n) = ln
On substituting y(n) = ln in equation (2.21) we get,
N

∑ a m λn − m = 0
m= 0
On expanding the above equation (by taking a0 = 1), we get,
ln + a1 ln – 1 + a2 ln – 2 + ... + aN – 1 ln – (N – 1) + aN ln – N = 0
ln – N (lN + a1 lN – 1 + a2 lN – 2 + ... + aN – 1 l + aN) = 0
Now, the characteristic polynomial of the system is given by,
lN + a1 lN – 1 + a2 lN – 2 + ... + aN – 1 l + aN = 0
The characteristic polynomial has N roots, which are denoted as l1, l2,...lN.
The roots of the characteristic polynomial may be distinct real roots, repeated real roots or complex.
The assumed solutions for various types of roots are given below.
Distinct Real Roots
Let the roots l1, l2, l3, ... lN be distinct real roots. Now the homogeneous solution will be in the form,
y h ( n) = C1 λn1 + C2 λn2 + C3 λn3 + ...... + C N λnN
where, C1 , C2 , C3 ,......C N are constants that can be evaluated using initial conditions.
Chapter 2 - Discrete Time Signals and Systems 2. 30
Repeated Real Roots
Let one of the real roots l1 repeats p times and the remaining (N – p) roots are distinct real roots. Now,
the homogeneous solution is in the form,

y h ( n) = (C1 + C2 n + C3 n2 + ..... + C p n p − 1 ) λn1 + C p + 1 λnp + 1 + .... + C N λnN

where, C1 , C2 , C3 ,......C N are constants that can be evaluated using initial conditions.
Complex Roots
Let the characteristic polynomial has a pair of complex roots l and l* and the remaining (N – 2) roots
be distinct real roots. Now, the homogeneous solution will be in the form,
yh(n) = rn [C1 cos nq + C2 sin nq] + C3 l3n + C4 l4n + ... + CN lN n
b
where, λ = a + jb, λ∗ = a − jb, r = a 2 + b2 , θ = tan−1
a
C1, C2, C3 ... CN are constants that can be evaluated using initial conditions.
2.7.2 Particular Solution
The particular solution, yp(n) is the solution of the difference equation for specific input signal x(n)
for n ³ 0. Since the input signal may have different form, the particular solution depends on the form or type
of the input signal x(n).
If x(n) is constant, then yp(n) is also a constant.
Example :

Let, x(n) = u(n) ; now, yp(n) = K u(n)

If x(n) is exponential, then yp(n) is also an exponential.

Example :

Let, x(n) = an u(n) ; now, yp(n) = K an u(n)

If x(n) is sinusoid, then yp(n) is also a sinusoid.

Example :

Let, x(n) = A cos w 0n ; now, yp(n) = K1 cos w 0n + K2 sin w 0n

The general form of particular solution for various types of inputs are listed in table 2.2.
Table 2.2 : Particular Solution

Input signal, x(n) Particular solution, yp(n)

A K
n
AB K Bn
A nB K0 nB + K1 n(B–1) + ..... + KB
An nB An (K0 nB + K1 n(B–1) + .....+ KB )
A cos w 0n K1 cos w 0n + K2 sin w 0n
A sin w 0n K1 cos w 0n + K2 sin w 0n
2. 31 Digital Signal Processing

2.7.3 Zero-State Response

The zero-state response or forced response is obtained from the sum of homogeneous solution and
particular solution and evaluating the constants with zero initial conditions.

∴ Zero - state response, y zs ( n) = y h ( n) + y p ( n)

with constants C1 , C 2 ,.... C N evaluated with zero initial conditions

2.7.4 Total Response

The total response of discrete time system can be obtained by the following two methods.
Method-1
The total response is given by sum of homogeneous solution and particular solution.
\ Total response, y(n) = yh(n) + yp(n)
Procedure to Determine Total Response by Method-1
1. Determine the homogeneous solution yh(n) with constants C1, C2, .....CN.
2. Determine the particular solution y p(n) and evaluate the constants K for any value of
n ³ 1 so that no term of y(n) vanishes.
3. Now the total response is given by the sum of yh(n) and yp(n).
\ Total response, y(n) = yh (n) + yp(n)
4. The total response will have N number of constants C1, C2, .....CN. Evaluate the given difference
equation for n = 0, 1, 2, ....N – 1 and form one set of N number of equations. Then evaluate the total
response for n = 0, 1, 2, ..... N – 1 and form another set of N number of equations. Now
solve the constants C1, C2, .....CN using the two sets of N number of equations.
Method-2
The total response is given by sum of zero-input response and zero-state response.
\ Total response, y(n) = yzi(n) + yzs(n)
Procedure to Determine Total Response by Method-2
1. Determine the homogeneous solution yh(n) with constants C1, C2, ....CN.
2. Determine the zero-input response, which is obtained from the homogeneous solution yh(n)
and evaluating the constants C1, C2, ....CN using the initial conditions.
3. Determine the particular solution yp(n) and evaluate the constants K for any value of n ³ 1 so
that no term of y(n) vanishes.
4. Determine the zero-state response, yzs(n) which is given by sum of homogeneous solution
and particular solution and evaluating the constants C1, C2, ....CN with zero initial conditions.
5. Now, the total response is given by sum of zero input response and zero state response.

\ Total response, y(n) = yzi(n) + yzs(n)

Chapter 2 - Discrete Time Signals and Systems 2. 32
Example 2.8
Determine the response of first order discrete time system governed by the difference equation,
y(n) = – 0.8 y(n – 1) + x(n)
When the input is unit step, and with initial condition a) y(–1) = 0 b) y(–1) = 2/9.
Solution
Given that, y(n) = – 0.8 y(n – 1) + x(n)
\ y(n) + 0.8 y(n – 1) = x(n) .....(1)
Homogeneous Solution
The homogeneous equation is the solution of equation (1) when x(n) = 0.
\ y(n) + 0.8 y(n – 1) = 0 .....(2)
n
Put, y(n) = l in equation (2).
\ ln + 0.8 l(n – 1) = 0
(n – 1)
l (l + 0.8) = 0 Þ l = – 0.8
The homogeneous solution yh(n) is given by,
yh(n) = C ln = C (– 0.8)n ; for n³³
³0 .....(3)
Particular Solution
Given that the input is unit step and so the particular solution will be in the form,
y(n) = K u(n) .....(4)
On substituting for y(n) from equation (4) in equation (1) we get,
y(n) + 0.8 y(n – 1) = x(n) Þ K u(n) + 0.8 K u(n – 1) = u(n) .....(5)
In order to determine the value of K, let us evaluate equation (5) for n = 1, ( Q we have to evaluate equation
(5) for any n ³1, such that none of the term vanishes).
From equation (5) when n = 1, we get,
1 10 5
K + 0.8 K = 1 Þ 1.8 K = 1 Þ K= = =
1. 8 18 9
The particular solution yp(n) is given by,
5
yp (n) = K u(n) = u(n) ; for all n
9
5
= ; for n ≥ 0
9
Total Response
The total response y(n) of the system is given by sum of homogeneous and particular solution.
\ Response, y(n) = yh(n) + yp(n)
n5
\ y(n) = C(−0.8) + ; for n ≥ 0 .....(6)
9
When n = 0, from equation (1), we get, y(0) + 0.8 y(–1) = 1
\ y(0) = 1 – 0.8 y(–1) .....(7)
5
When n = 0, from equation (6), we get, y(0) = C + .....(8)
9
5
On equating (7) and (8) we get, C+ = 1 − 0.8 y(−1)
9
5
∴ C = 1 − 0.8 y(−1) −
9
4 .....(9)
= − 0.8 y( −1)
9
2. 33 Digital Signal Processing
On substituting for C from equation (9) in equation (6) we get,

y(n) =
FG 4 − 0.8 y(−1)IJ (−0.8) n
+
5
H9 K 9
a) When y(–1) = 0
4 5
∴ y(n) = ( −0.8)n + ; for n ≥ 0
9 9

=
LM b
4
g
−0.8 +
n 5 OP
u(n)
N
9 9 Q
b) When y(–1) = 2/9

∴ y(n) =
FG 4 − 0.8 × 2IJ (−0.8) n
+
5
=
2.4 5 24
( −0.8)n + = (−0.8)n +
5
H9 9K 9 9 9 90 9
5 12
∴ y(n) = + (−0.8)n ; for n ≥ 0
9 45

=
LM 5 + 12 (−0.8)nOP u(n)
N 9 45 Q
Example 2.9
Determine the response y(n), n ³ 0 of the system described by the second order difference equation,
y(n) – 0.2 y(n – 1) – 0.03 y(n – 2) = x(n) + 0.4 x(n – 1),
when the input signal is, x(n) = 0.2n u(n) and with initial conditions y( – 2) = 0, y( –1) = 0.5.

Solution
Given that, y(n) – 0.2 y(n – 1) – 0.03 y(n – 2) = x(n) + 0.4 x(n – 1) .....(1)
Homogeneous Solution
The homogeneous equation is the solution of equation (1) when x(n) = 0.
\ y(n) – 0.2 y(n – 1) – 0.03 y(n – 2) = 0 .....(2)
n
Put y(n) = l in equation (2). The roots of quadratic,
\ ln – 0.2 ln – 1 – 0.03 ln – 2 = 0 λ2 − 0.2λ − 0.03 = 0 are,
ln – 2 (l2 – 0.2l – 0.03) = 0 0.2 ± 0.22 + 4 × 0.03
λ=
The characteristic equation is, 2
l2 – 0.2l – 0.03 = 0 Þ (l – 0.3) (l + 0.1) = 0 0.2 ± 0.4
= = 0.3, − 0.1
\ The roots are, l = 0.3, –0.1 2
The homogeneous solution, yh(n) is given by,

yh (n) = C1 λn1 + C 2 λn2

= C1(0.3)n + C 2( −0.1)n ; for n ≥ 0 .....(3)
Particular Solution
Given that the input is an exponential signal, 0.2n u(n) and so the particular solution will be in the form,
y(n) = K 0.2n u(n) .....(4)
On substituting for y(n) from equation (4) in equation (1) we get,
K0.2n u(n) –0.2 K 0.2(n – 1) u(n – 1) – 0.03 K 0.2(n – 2) u(n – 2) = 0.2n u(n) + 0.4 ´ 0.2(n – 1) u(n) .....(5)
In order to determine the value of K, let us evaluate equation (5) for n = 2, ( Q we have to evaluate equation
(5) for any n ³ 1, such that none of the term vanishes).
Chapter 2 - Discrete Time Signals and Systems 2. 34
From equation (5) when n = 2, we get,
K 0.22 – 0.2K ´ 0.21 – 0.03K ´ 0.20 = 0.22 + 0.4 ´ 0.21
0.04K – 0.04K – 0.03K = 0.04 + 0.08
– 0.03K = 0.12
0.12
∴ K=− = −4
0.03
The particular solution yp(n) is given by,
yp (n) = K 0.2n u(n) = (–4) 0.2n u(n)
Total Response

The total response y(n) of the system is given by sum of homogeneous and particular solution.
\ Response, y(n) = yh (n) + yp (n)
= C1 0.3n + C2 (–0.1)n + (– 4) 0.2n ; for n ³ 0 .....(6)
To find y(0) and y(1)
When n = 0,
From equation (1) we get,
y(0) – 0.2 y(–1) – 0.03 y(–2) = x(0) + 0.4 x(–1) .....(7)
Given that, y(–1) = 0.5, y(–2) = 0
x(n) = 0.2n u(n), \ x(0) = 0.20 = 1
x(–1) = 0
On substituting the above conditions in equation (7) we get,
y(0) – 0.2 ´ 0.5 – 0.03 ´ 0 = 1 + 0
\ y(0) = 1.1 .....(8)
When n = 1,
From equation (1) we get,
y(1) – 0.2 y(0) – 0.03 y(–1) = x(1) + 0.4 x(0) .....(9)
We know that, y(0) = 1.1, y(–1) = 0.5, y(–2) = 0
Given that, x(n) = 0.2n u(n), \ x(0) = 0.20 = 1
x(1) = 0.21 = 0.2
On substituting the above conditions in equation (9) we get,
y(1) – 0.2 ´ 1.1 – 0.03 ´ 0.5 = 0.2 + 0.4 ´ 1
\ y(1) = 0.6 + 0.235 = 0.835 .....(10)
To solve constants C1 and C2

When n = 0,
From equation (6) we get,
y(0) = C1 0.30 + C2 (–0.1)0 + (–4) 0.20 = C1 + C2 – 4 .....(11)
From equations (8) and (11) we can write,
C1 + C2 – 4 = 1.1
\ C1 + C2 = 5.1 .....(12)
2. 35 Digital Signal Processing
When n = 1,
From equation (6) we get,
y(1) = C1 ´ 0.3 + C2 (–0.1) + (–4) 0.2 = 0.3 C1 – 0.1C2 – 0.8 .....(13)
From equations (10) and (13) we can write,
0.3 C1 – 0.1C2 – 0.8 = 0.835
\ 0.3 C1 – 0.1C2 = 1.635 .....(14)
Equation (12) ´ 0.1 Þ 0.1C1 + 0.1C2 = 0.51
Equation (13) Þ 0.3C1 – 0.1C2 = 1.635
Add 0.4C1 = 2.145
2.145
∴ C1 = = 5.3625
0.4
From equation(12),
C2 = 5.1 – C1 = 5.1 – 5.3625
= – 0.2625
Total Response
y(n) = [5.3625(0.3)n – 0.2625(–0.1)n + (–4) 0.2n] u(n) ; for all n

2.8 Classification of Discrete Time Systems

The discrete time systems are classified based on their characteristics. Some of the classifications of
discrete time systems are,
1. Static and dynamic systems
2. Time invariant and time variant systems
3. Linear and nonlinear systems
4. Causal and noncausal systems
5. Stable and unstable systems
6. FIR and IIR systems
7. Recursive and nonrecursive systems

2.8.1 Static and Dynamic Systems

A discrete time system is called static or memoryless system if its output at any instant n depends at
most on the input sample at the same time but not on the past or future samples of the input. In any other case,
the system is said to be dynamic or to have memory.
Example :
123

y(n) = a x(n) Static systems

y(n) = n x(n) + 6 x3(n)
123

y(n) = x(n) + 3 x(n – 1)

123

N Finite memory is required

y(n) = ∑
m= 0
x(n − m)
Dynamic systems
123

∞
Infinite memory is required
y(n) = ∑ x(n − m)
m= 0
Chapter 2 - Discrete Time Signals and Systems 2. 36
2.8.2 Time Invariant and Time Variant Systems
A system is said to be time invariant if its input-output characteristics do not change with time.
Definition : A relaxed system H is time invariant or shift invariant if and only if
H{x(n)} = y(n) implies that, H{x(n – m)} = y(n – m)
for every input signal x(n) and every time shift m.
i.e., in time invariant systems, if y(n) = H{x(n)} then y(n – m) = H{x(n – m)}.
Alternative Definition for Time Invariance
A system H is time invariant if the response to a shifted (or delayed) version of the input is identical
to a shifted (or delayed) version of the response based on the unshifted (or undelayed) input.
i.e., In a time invariant system, H{x(n - m)} = z-m H{x(n)}; for all values of m .....(2.22)
-m
The operator z represents a signal delay of m samples.
The diagrammatic explanation of the above definition of time invariance is shown in fig 2.19.
Procedure to Test for Time Invariance
1. Delay the input signal by m units of time and determine the response of the system for this
delayed input signal. Let this response be y(n – m).
2. Delay the response of the system for undelayed input by m units of time. Let this delayed
response be yd(n).
3. Check whether y (n – m) = yd(n). If they are equal then the system is time invariant.
Otherwise the system is time variant.
x (n) x (n − m ) y (n − m )
−m

Input signa l
z H
D elayed input R espons e for
D elay S y stem delay ed in put

x (n) y (n) y d (n)

H z
−m

Input signa l R espons e for D elayed

S y stem undelay ed input D elay respons e
If, y (n − m ) = y d (n), then the sy s tem is tim e inv ariant

F ig 2.1 9 : D ia g ra m m a tic exp la n a tio n o f tim e in va ria n ce .

Example 2.10
Test the following systems for time invariance.
a) y(n) = x(n) + x(n – 1) b) y(n) = 2n x(n) c) y(n) = x(–n) d) y(n) = x(n) – b x(n – 1)

Solution
a) Given that, y(n) = x(n) + x(n – 1)
Test 1 : Response for delayed input
Let, y(n – m) = Response for delayed input.
x(n) y(n − m) = x(n − m) + x(n − m − 1)
−m
x(n − m)
Input signal
z
Delayed input
H Response for
Delay System delayed input
2. 37 Digital Signal Processing
Test 2 : Delayed response
Let, yd(n) = Delayed response.
x(n) y(n) = x(n) + x(n −1) y d (n) = x(n − m ) + x(n − m − 1)

Input sign al
H z
−m

R esponse for D elayed response

S ystem undelayed input D elay

Conclusion : Here, y(n – m) = yd(n), therefore the system is time invariant.

b) Given that, y(n) = 2n x(n)

Test 1 : Response for delayed input
Let, y(n – m) = Response for delayed input.
x (n) y (n − m ) = 2(n − m ) x(n − m )
−m
x (n − m )
Input signa l
z H R espons e for
D elayed input
D elay S y stem delay ed in put
Test 2 : Delayed response
Let, yd(n) = Delayed response.
x (n) y (n) = 2n x (n) y d (n) = 2n x(n − m )
H z
−m

Input signa l R espons e for D elayed res ponse

S y stem undelay ed input D elay

Conclusion : Here, y(n – m) ¹ yd(n), therefore the system is time variant.

c) Given that, y(n) = x(–n)

Test 1 : Response for delayed input
Let, y(n – m) = Response for delayed input.
x (n) y (n − m ) = x ( −(n − m )) = x( −n + m )
−m x (n − m )
z H
Input signa l D elayed input R espons e for
D elay S y stem delay ed in put
Test 2 : Delayed response
Let, yd(n) = Delayed response.
x (n) y (n) = x( −n) y d (n) = x ( −n − m )
H z
−m

Input signa l R espons e for D elayed res ponse

S y stem undelay ed input D elay

Conclusion : Here, y(n – m) ¹ yd(n), therefore the system is time variant.

d) Given that, y(n) = x(n) – b x(n – 1)

Test 1 : Response for delayed input
Let, y(n – m) = Response for delayed input.
x (n) x (n − m ) y (n − m ) = x (n − m ) −b x (n − m − 1)
−m
z H
In put sign a l D elayed input R espons e for
D elay S y stem delay ed in put
Test 2 : Delayed response
Let, yd(n) = Delayed response.
x (n ) y d (n ) = x (n − m ) − b x(n − m − 1)
y (n ) = x(n ) − b x(n − 1 )
H z
−m

In p ut sign a l R esp on s e for D elayed res p on se

S y ste m u nd elay e d inp u t D elay

Conclusion : Here, y(n – m) = yd(n), therefore the system is time invariant.

Example 2.11
Test the following systems for time invariance.
M N
a) y(n) = x(n) + B b) y(n) = n x3(n) c) y(n) = bx(n) d) y(n) = ∑
k = 0
bk x(n − k) − ∑
k =1
a k y(n − k)
Chapter 2 - Discrete Time Signals and Systems 2. 38
Solution
a) Given that, y(n) = x(n) + B
Test 1 : Response for delayed input
Let, y(n – m) = Response for delayed input.
y (n − m ) = x (n − m ) + B
x (n) x (n − m )
−m

Input signa l
z H R espons e for
D elayed input
D elay S y stem delay ed inp ut
Test 2 : Delayed response
Let, yd(n) = Delayed response.
x (n) y (n) = x(n) + B y d (n) = x (n − m ) + B
H z
−m

Input signa l R espons e for D elayed res ponse

S y stem undelay ed input D elay

Conclusion : Here, y(n – m) = yd(n), therefore the system is time invariant.

b) Given that, y(n) = n x3(n)

Test 1 : Response for delayed input
Let, y(n – m) = Response for delayed input.
x (n) y (n − m ) = (n − m ) x 3 (n − m )
x (n − m )
H
−m
z
Input signa l D elayed input R espons e for
D elay S y stem delay ed in put
Test 2 : Delayed response
Let, yd(n) = Delayed response.
y d (n) = n x 3 (n − m )
x (n) y (n) = n x 3 (n)
H z
−m

Input signa l R espons e for D elayed res ponse

S y stem undelay ed input D elay

Conclusion : Here, y(n – m) ¹ yd(n), therefore the system is time variant.

c) Given that, y(n) = bx(n)

Test 1 : Response for delayed input
Let, y(n – m) = Response for delayed input.
x(n − m)
x (n) x (n − m ) y (n − m ) = b
−m

Input signa l
z
D elayed input
H R espons e for
D elay S y stem delay ed inp ut
Test 2 : Delayed response
Let, yd(n) = Delayed response.
x(n − m)
x (n) y (n) = b
x(n) y d(n) = b
H
−m
z
Input signa l R espons e for D elayed res ponse
S y stem undelay ed input D elay

Conclusion : Here, y(n – m) = yd(n), therefore the system is time invariant.

M N
d) Given that, y(n) = ∑b
k=0
k x(n − k) − ∑
k=1
ak y(n − k)

Test 1 : Response for delayed input

Let, y(n – m) = Response for delayed input.
x (n) y (n − m )
x (n − m )
H
−m
z
Input signa l D elayed input R espons e for
D elay S y stem delay ed inp ut
M N
Response for delayed input, y(n – m) = H{x(n – m)} = ∑b
k=0
k x(n − m − k) − ∑a
k=1
k y(n − m − k)
2. 39 Digital Signal Processing
Test 2 : Delayed response
Let, yd(n) = Delayed response.
x (n) y (n) y d (n)

Input signa l
H z
−m

R espons e for D elayed res ponse

S y stem D elay
undelay ed input
M N
Response for undelayed input = H{x(n)} = y(n) = ∑
k =0
bk x(n − k) − ∑
k =1
a k y(n − k)
–m
Delayed response, yd(n) = z H{x(n)}
LM b M N OP
= z −m
MN ∑
k =0
k x(n − k) − ∑
k =1
ak y(n − k)
PQ
M N
= ∑
k =0
bk x(n − m − k) −
k =1
∑ ak y(n − m − k)

Conclusion : Here, y(n – m) = yd(n), therefore the system is time invariant.

2.8.3 Linear and Nonlinear Systems

A linear system is one that satisfies the superposition principle. The principle of superposition
requires that the response of the system to a weighted sum of the signals is equal to the corresponding
weighted sum of the responses of the system to each of the individual input signals.
Definition : A relaxed system H is linear if
H{a1 x1(n) + a2 x2(n)} = a1 H{x1(n)} + a2 H{x2(n)} .....(2.23)
for any arbitrary input sequences x1(n) and x2(n) and for any arbitrary constants a1 and a2.
If a relaxed system does not satisfy the superposition principle as given by the above definition, then
the system is nonlinear.The diagrammatic explanation of linearity is shown in fig 2.20.
x 1 (n) a 1 x 1 (n)
a1

a 1 x 1 (n) + a 2 x 2 (n) H{a 1 x 1 (n) + a 2 x 2 (n)}

+ H
x 2 (n) a 2 x 2 (n)
a2

x 1 (n) H{x 1 (n)} a 1 H{x 1 (n)}

H a1

a 1 H{x 1 (n)} + a 2 H{x 2 (n)}

+

x 2 (n) H{x 2 (n)} a 2 H{x 2 (n)}

H a2

T he s ys tem , H is linear if an d only if, H{a 1 x 1(n ) + a 2 x 2 (n)} = a 1 H{x 1 (n)} + a 2 H{x 2 (n)}

F ig 2.2 0 : D ia g ra m m a tic exp la n a tio n o f lin ea rity.

Procedure to test for linearity

1. Let x1(n) and x2(n) be two inputs to system H, and y1(n) and y2(n) be corresponding responses.
2. Consider a signal, x3(n) = a1 x1(n) + a2 x2(n) which is a weighed sum of x1(n) and x2(n).
3. Let y3(n) be the response for x3(n).
4. Check whether y3(n) = a1 y1(n) + a2 y2(n). If they are equal then the system is linear, otherwise it is
nonlinear.
Chapter 2 - Discrete Time Signals and Systems 2. 40

Example 2.12
Test the following systems for linearity.
a) y(n) = n x(n) b) y(n) = x(n2) c) y(n) = x2(n) d) y(n) = B x(n) + C

Solution
a) Given that, y(n) = n x(n)
Let H be the system represented by the equation, y(n) = nx(n).
The system H operates on x(n) to produce, y(n).

x(n)
H y (n) = H{x (n)} = n x (n )

Consider two signals, x1(n) and x2(n).

Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.

x 1 (n)
H y 1 (n ) = H {x 1 (n )} = n x 1 (n)

x 2 (n)
H y 2 (n ) = H {x 2 (n )} = n x 2 (n)

\ a1 y1(n) + a2 y2(n) = a1 n x1(n) + a2 n x2(n) .....(1)

Consider a linear combination of inputs, a1 x1(n) + a2 x2(n) = x3(n).
Let, y3(n) be the response for x3(n).

x 3 (n)
H y 3 (n ) = H {x 3 (n )}

\ y3(n) = H{a1 x1(n) + a2 x2(n)} = n[a1 x1(n) + a2 x2(n)] = a1 n x1(n) + a2 n x2(n) .....(2)
The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (2) we can say that, y3(n) = a1 y1(n) + a2 y2(n). Hence the system is linear.

b) Given that, y(n) = x(n2)

Let, H be the system represented by the equation, y(n) = x(n2).
The system H operates on x(n) to produce, y(n).
x(n)
y (n ) = H {x (n)} = x (n )
2
H

Consider two signals, x1(n) and x2(n).

Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.

x 1 (n)
y 1 (n ) = H {x 1 (n )} = x 1 (n )
2
H

x 2 (n)
y 2 (n ) = H {x 2 (n )} = x 2 (n )
2
H

\ a1 y1(n) + a2 y2(n) = a1 x1(n2) + a2 x2(n2) .....(1)

Consider a linear combination of inputs, a1 x1(n) + a2 x2(n) = x3(n).
Let, y3(n) be the response for x3(n).
2. 41 Digital Signal Processing

x 3 (n)
H y 3 (n ) = H {x 3 (n )}

\ y3(n) = H{a1 x1(n) + a2 x2(n)} = a1 x1(n2) + a2 x2(n2) .....(2)

The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (2) we can say that, y3(n) = a1 y1(n) + a2 y2(n). Hence the system is linear.
c) Given that, y(n) = x2(n)
Let, H be the system represented by the equation, y(n) = x2(n).
The system H operates on x(n) to produce, y(n).
x(n)
y (n ) = H {x (n)} = x (n)
2
H
Consider two signals, x1(n) and x2(n).
Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.
x 1 (n)
y 1 (n ) = H {x 1 (n )} = x 1 (n )
2
H

x 2 (n)
y 2 (n ) = H {x 2 (n )} = x 2 (n )
2
H

\ a1 y1(n) + a2 y2(n) = a1 x12(n) + a2 x22(n) .....(1)

Consider a linear combination of inputs, a1 x1(n) + a2 x2(n) = x3(n).
Let, y3(n) be the response for x3(n).
x 3 (n)
H y 3 (n ) = H {x 3 (n )}

\ y3(n) = H{a1 x1(n) + a2 x2(n)} = [a1 x1(n) + a2 x2(n)]2

= a12 x12 (n) + a22 x22(n) + 2 a1 a2 x1(n)x2(n) .....(2)
The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (2) we can say that, y3(n) ¹¹a1 y1(n) + a2 y2(n). Hence the system is nonlinear.
d) Given that, y(n) = B x(n) + C
Let, H be the system represented by the equation, y(n) = B x(n) + C.
The system H operates on x(n) to produce, y(n).
x(n)
H y (n) = H{x (n)} = B x (n) + C

Consider two signals, x1(n) and x2(n).

Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.
x 1 (n)
H y 1 (n ) = H{x 1 (n )} = B x 1 (n ) + C

x 2 (n)
H y 2 (n ) = H{x 2 (n )} = B x 2 (n ) + C

\ a1 y1(n) + a2 y2(n) = a1[B x1(n) + C] + a2[B x2(n) + C]

= B a1 x1(n) + C a1 + B a2 x2(n) + C a2 .....(1)
Consider a linear combination of inputs, a1 x1(n) + a2 x2(n) = x3(n).
Let, y3(n) be the response for x3(n).
x 3 (n)
H y 3 (n ) = H {x 3 (n )}

\ y3(n) = H{a1 x1(n) + a2 x2(n)} = B[a1 x1(n) + a2 x2(n)] + C = Ba1 x1(n) + B a2 x2(n) + C .....(2)
The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (2) we can say that, y3(n) ¹ a1 y1(n) + a2 y2(n). Hence the system is nonlinear.
Chapter 2 - Discrete Time Signals and Systems 2. 42
Example 2.13
Test the following systems for linearity.

a) y(n) = ex(n) b) y(n) = bx(n) c) y(n) = n x2(n)

Solution
a) Given that, y(n) = ex(n)

Let, H be the system represented by the equation, y(n) = ex(n).

The system H operates on x(n) to produce, y(n).

x(n) x( n)
H y (n ) = H {x (n)} = e

Consider two signals, x1(n) and x2(n).

Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.

x 1 (n) x 1 (n )
H y 1 ( n ) = H { x 1(n )} = e

x 2 (n)
H y 2 ( n ) = H { x 2 ( n )} = e x 2 ( n )

∴ a1 y1(n) + a 2 y 2(n) = a1 ex1(n) + a 2 ex 2 (n) .....(1)

Consider a linear combination of inputs, a1 x1(n) + a2 x2(n) = x3(n).

Let, y3(n) be the response for x3(n).

x 3 (n)
H y 3 (n ) = H {x 3 (n )}

\ y3(n) = H{a1 x1(n) + a2 x2(n)} = e[ a1 x1(n) + a 2 x 2 (n)] = ea1 x1(n) ea 2 x 2 (n) .....(2)
The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (2) we can say that, y3(n) ¹ a1 y1(n) + a2 y2(n). Hence the system is nonlinear.

b) Given that, y(n) = bx(n)

Let, H be the system represented by the equation, y(n) = bx(n).
The system H operates on x(n) to produce, y(n).
x(n)
y (n) = H {x(n)} = b
x(n)
H

Consider two signals, x1(n) and x2(n).

Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.

x 1 (n)
H y 1 ( n ) = H { x 1(n )} = b x 1 ( n )

x 2 (n) x 2 (n )
H y 2 ( n ) = H { x 2 ( n )} = b

∴ a1 y1(n) + a 2 y 2(n) = a1 bx1(n) + a 2 bx 2 (n) .....(1)

Consider a linear combination of inputs, a1x1(n) + a2x2(n) = x3(n).

2. 43 Digital Signal Processing
Let, y3(n) be the response for x3(n).
x 3 (n)
H y 3 (n ) = H {x 3 (n )}

∴ y 3 (n) = H a1x1(n) + a 2x 2 (n) = b[a 1x 1(n) + a 2x 2 (n)] = b a 1x 1(n) b a 2x 2 (n)

l q .....(2)
The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (2) we can say that, y3(n) = a1 y1(n) + a2 y2(n). Hence the system is nonlinear.

c) Given that, y(n) = n x2(n)

Let, H be the system represented by the equation, y(n) = n x2(n).
The system H operates on x(n) to produce, y(n).
x(n) 2
H y (n) = H {x (n)} = n x (n )

Consider two signals, x1(n) and x2(n).

Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.
x 1 (n)
y 1 (n ) = H {x 1 (n )} = n x 1 (n)
2
H
x 2 (n)
y 2 (n ) = H {x 2 (n )} = n x 2 (n)
2
H
\ a1 y1(n) + a2 y2(n) = a1 n x12(n) + a2 n x22(n) .....(1)
Consider a linear combination of inputs, a1 x1(n) + a2 x2(n) = x3(n).
Let, y3(n) be the response for x3(n).
x 3 (n)
H y 3 (n ) = H {x 3 (n )}

\ y3(n) = H{a1 x1(n) + a2 x2(n)} = n[a1 x1(n) + a2 x2(n)]2

= n a12 x12(n) + n a22 x22(n) + 2 n a1 a2 x1(n) x2(n) .....(2)
The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (2) we can say that, y3(n) ¹ a1 y1(n) + a2 y2(n). Hence the system is nonlinear.

Example 2.14
Test the following systems for linearity.
M N
1
g
a y(n) = 3 x(n) +
x(n − 2)
g
b y(n) = x(n) − 2 x(n − 1) g
c y(n) = ∑
m = 0
bm x(n − m) − ∑
m = 1
cm y(n − m)

Solution
1
a) Given that, y(n) = 3 x(n) +
x(n − 2)
1
Let, H be the system represented by the equation, y(n) = 3x(n) + .
x(n − 2)
The system H operates on x(n) to produce, y(n).

x (n) 1
H y ( n ) = H { x (n )} = 3 x (n ) +
x (n − 2 )
Consider two signals, x1(n) and x2(n).
Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.
x 1 (n) 1
H y 1( n ) = H { x 1( n )} = 3 x 1(n ) +
x 1( n − 2 )
x 2(n) 1
H y 2 (n ) = H { x 2 ( n )} = 3 x 2 ( n ) +
x 2 (n − 2 )
Chapter 2 - Discrete Time Signals and Systems 2. 44

F
∴ a1 y1(n) + a 2 y 2(n) = a1 3x1(n) +
1 I F
+ a 2 3x 2(n) +
1 I .....(1)
GH x1(n − 2) JK GH x 2(n − 2) JK
Consider a linear combination of inputs, a1 x1(n) + a2 x2(n) = x3(n).
Let, y3(n) be the response for x3(n).
x 3 (n)
H y 3 (n ) = H {x 3 (n )}
1 .....(2)
\ y3(n) = H{a1 x1(n) + a2 x2(n)} = 3[a1 x1(n) + a 2 x 2(n)] +
a1 x1(n − 2) + a 2 x 2(n − 2)
The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (2) we can say that, y3(n) ¹ a1 y1(n) + a2 y2(n). Hence the system is nonlinear.

b) Given that, y(n) = x(n) − 2 x(n − 1)

Let, H be the system represented by the equation, y(n) = x(n) – 2 x(n–1).
The system H operates on x(n) to produce, y(n).
x(n)
H y (n ) = H {x (n)} = x (n) −2x (n − 1)

Consider two signals, x1(n) and x2(n).

Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.
x 1 (n)
H y 1 (n ) = H {x 1 (n )} = x 1 (n ) − 2 x 1 (n − 1)

x 2 (n)
H y 2 (n ) = H {x 2 (n )} = x 2 (n ) − 2 x 2 (n − 1)

\ a1 y1(n) + a2 y2(n) = a1 x1(n) – a1 2 x1(n – 1) + a2 x2(n) – a2 2 x2(n – 1) .....(1)

Consider a linear combination of inputs, a1 x1(n) + a2 x2(n) = x3(n).
Let, y3(n) be the response for x3(n).
x 3 (n)
H y 3 (n ) = H {x 3 (n )}

\ y3(n) = H{a1 x1(n) + a2 x2(n)}= a1 x1(n) + a2 x2(n) – 2[a1 x1(n – 1) + a2 x2(n – 1)]
= a1 x1(n) – a1 2 x1(n – 1) + a2 x2(n) – a2 2 x2(n – 1) .....(2)
The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (2) we can say that, y3(n) = a1 y1(n) + a2 y2(n). Hence the system is linear.
M N
c) Given that, y(n) = ∑
m=0
bm x(n − m) − ∑
m=1
cm y(n − m)

Let, H be the system represented by the equation,

M N
y(n) = ∑
m=0
bm x(n − m) − ∑
m =1
cm y(n − m)

The response of the system |UV = H{ x(n)} = y(n) = ∑ b M

m x(n − m) −
N

∑ cm y(n − m)
H for the input x(n) |W m=0 m=1

Consider two signals, x1(n) and x2(n).

Let, y1(n) and y2(n) be the response of the system H for inputs x1(n) and x2(n) respectively.
M N
\ y1(n) = H{x1(n)} = ∑
m=0
bm x1(n − m) − ∑
m =1
cm y1(n − m)

M N
y2(n) = H{x2(n)} = ∑
m =0
bm x 2 (n − m) − ∑
m = 1
cm y 2 (n − m)
2. 45 Digital Signal Processing

F b x (n − m) − c y (n − m)I
M N
∴ a1 y1(n) + a 2 y 2(n) = a1 GH ∑ m =0
m ∑ 1 JK m=1
m 1

F M
+ a G ∑ b x (n − m) − ∑ c y (n − m)J
I N
.....(1)
2 m 2 m 2
H m =0 K m =1

Consider a linear combination of inputs, a1 x1(n) + a2 x2(n) = x3(n).

Let, y3(n) be the response for the input x3(n).
\ y3(n) = H{x3(n)} = H{a1 x1(n) + a2 x2(n)}
M N
= ∑
m =0
bm (a1 x1(n − m) + a 2 x 2 (n − m)) −
m=1
∑ cm y 3(n − m)

M M N
.....(2)
= a1 ∑ bm x1(n − m) + a 2 ∑ bm x 2 (n − m) − ∑ cm y 3(n − m)
m=0 m=0 m=1

By time invariant property,

If y3(n) = H{a1 x1(n) + a2 x2(n)} then y3(n – m) = H{a1 x1(n – m) + a2 x2(n – m)}
If y2(n) = H{x2(n)} then y2(n – m) = H{x2(n – m)}
If y1(n) = H{x1(n)} then y1(n – m) = H{x1(n – m)}
\ y3(n – m) = H{a1 x1(n – m) + a2 x2(n – m)} = a1 H{x1(n – m)} + a2 H{x2(n – m)}
= a1 y1(n – m) + a2 y2(n – m)] .....(3)
Using equation (3), the equation (2) can be written as,
M M N
y3(n) = a1 ∑b
m = 0
m x1(n − m) + a 2 ∑
m = 0
bm x 2(n − m) − ∑
m = 1
cm [a1 y1(n − m) + a 2 y 2 (n − m)]

M M N N
= a1 ∑
m = 0
bm x1(n − m) + a 2 ∑
m = 0
bm x 2(n − m) − a1 ∑
m = 1
cm y1(n − m) − a 2 ∑
m = 1
cm y 2(n − m)

F b
M N I F M N I .....(4)
= a1 GH ∑
m = 0
m x1(n − m) − ∑c
m = 1
m y1(n − m) + a 2 JK GH ∑
m = 0
bm x 2(n − m) − ∑
m = 1
cm y 2(n − m)JK
The condition to be satisfied for linearity is, y3(n) = a1 y1(n) + a2 y2(n).
From equations (1) and (4) we can say that the condition for linearity is satisfied. Therefore the system is
linear.

2.8.4 Causal and Noncausal Systems

Definition : A system is said to be causal if the output of the system at any time n depends only on the
present input, past inputs and past outputs but does not depend on the future inputs and outputs.
If the system output at any time n depends on future inputs or outputs then the system is called
noncausal system.
The causality refers to a system that is realizable in real time. It can be shown that an LTI system is
causal if and only if the impulse response is zero for n < 0, (i.e., h(n) = 0 for n < 0).
Let, x(n) = Present input and y(n) = Present output
\ x(n - 1), x(n - 2), ......, are past inputs
y(n - 1), y(n - 2), ......, are past outputs
In mathematical terms the output of a causal system satisfies the equation of the form,
y(n) = F [x(n), x(n - 1), x(n - 2), .... , y(n - 1), y(n - 2) ....]
where, F[.] is some arbitrary function.
Chapter 2 - Discrete Time Signals and Systems 2. 46
Example 2.15
Test the causality of the following systems.
n
a) y(n) = x(n) – x(n – 2) b) y(n) = ∑ x(k) c) y(n) = b x(n) d ) y(n) = n x(n)
k = −∞
Solution
a) Given that, y(n) = x(n) – x(n – 2)
When n = 0, y(0) = x(0) – x(–2) Þ The response at n = 0, i.e., y(0) depends on the present input
x(0) and past input x(–2)
When n = 1, y(1) = x(1) – x(–1) Þ The response at n = 1, i.e., y(1) depends on the present input
x(1) and past input x(–1).
From the above analysis we can say that for any value of n, the system output depends on present and
past inputs. Hence the system is causal.
n
b) Given that, y(n) = ∑ x(k)
k = −∞

0
When n = 0, y(0) = ∑ x(k)
k = −∞
= ... x(–2) + x(–1) + x(0) Þ The response at n = 0, i.e., y(0) depends on the
present input x(0) and past inputs x(–1), x(–2),.....
1

When n = 1, y(1) = ∑ x(k)

k = −∞

= ... x(–2) + x(–1) + x(0) + x(1) Þ The response at n = 1, i.e., y(1) depends on the
present input x(1) and past inputs x(0), x(–1), x(–2),..
From the above analysis we can say that for any value of n, the system output depends on present and
past inputs. Hence the system is causal.
c) Given that, y(n) = b x(n)
When n = 0, y(0) = b x(0) Þ Þ The response at n = 0, i.e., y(0) depends on the present input x(0).
When n = 1, y(1) = b x(1) Þ Þ The response at n = 1, i.e., y(1) depends on the present input x(1).
From the above analysis we can say that the response for any value of n depends on the present input.
Hence the system is causal.
d) Given that, y(n) = n x(n)
When n = 0, y(0) = 0 ´ x(0) Þ The response at n = 0, i.e., y(0) depends on the present input x(0).
When n = 1, y(1) = 1 ´ x(1) Þ The response at n = 1, i.e., y(1) depends on the present input x(1).
When n = 2, y(2) = 2 ´ x(2) Þ The response at n = 2, i.e., y(2) depends on the present input x(2).
From the above analysis we can say that the response for any value of n depends on the present input.
Hence the system is causal.

Example 2.16
Test the causality of the following systems.
a) y(n) = x(n) + 2 x(n + 3) b) y(n) = x(n2) c) y(n) = x(3n) d) y(n) = x(–n)
Solution
a) Given that, y(n) = x(n) + 2 x(n + 3)
When n = 0, y(0) = x(0) + 2 x(3) Þ The response at n = 0, i.e., y(0) depends on the
present input x(0) and future input x(3).
2. 47 Digital Signal Processing
When n = 1, y(1) = x(1) + 2 x(4) Þ Þ The response at n = 1, i.e., y(1) depends on the
present input x(1) and future input x(4).
From the above analysis we can say that the response for any value of n depends on present and future
inputs. Hence the system is noncausal.

b) Given that, y(n) = x(n2)

When n = –1 ; y(–1) = x(1) Þ The response at n = –1, depends on the future input x(1).
When n = 0 ; y(0) = x(0) Þ Þ The response at n = 0, depends on the present input x(0).
When n = 1 ; y(1) = x(1) Þ Þ The response at n = 1, depends on the present input x(1).
When n = 2 ; y(2) = x(4) Þ Þ The response at n = 2, depends on the future input x(4).
From the above analysis we can say that the response for any value of n (except n = 0 and n = 1) depends
on future inputs. Hence the system is noncausal.
c) Given that, y(n) = x(3n)
When n = –1 ; y(–1) = x(–3) Þ Þ The response at n = –1, depends on the past input x(–3).
When n = 0 ; y(0) = x(0) Þ Þ The response at n = 0, depends on the present input x(0).
When n = 1 ; y(1) = x(3) Þ Þ The response at n = 1, depends on the future input x(3).
From the above analysis we can say that the response of the system for n > 0, depends on future inputs.
Hence the system is noncausal.
d) Given that, y(n) = x(–n)
When n = –2 ; y(–2) = x(2) Þ Þ The response at n = –2, depends on the future input x(2).
When n = –1 ; y(–1) = x(1) Þ Þ The response at n = –1, depends on the future input x(1).
When n = 0 ; y(0) = x(0) Þ Þ The response at n = 0, depends on the present input x(0).
When n = 1 ; y(1) = x(–1) Þ Þ The response at n = 1, depends on the past input x(–1).
From the above analysis we can say that the response of the system for n < 0 depends on future inputs.
Hence the system is noncausal.

2.8.5 Stable and Unstable Systems

Definition : An arbitrary relaxed system is said to be BIBO stable (Bounded Input-Bounded Output stable)
if and only if every bounded input produces a bounded output.
Let x(n) be the input of discrete time system and y(n) be the response or output for x(n).The term
bounded input refers to finite value of the input signal x(n) for any value of n. Hence if input x(n) is bounded
then there exits a constant Mx such that |x(n)| £ Mx and Mx < ¥ , for all n.
Examples of bounded input signal are step signal, decaying exponential signal and impulse signal.
Examples of unbounded input signal are ramp signal and increasing exponential signal.

The term bounded output refers to finite and predictable output for any value of n. Hence if output
y(n) is bounded then there exists a constant My such that |y(n)| £ My and My < ¥ , for all n.
In general, the test for stability of the system is performed by applying specific input. On applying a
bounded input to a system if the output is bounded then the system is said to be BIBO stable.For LTI (Linear
Time Invariant) systems the condition for BIBO stability can be transformed to a condition on impulse
response as shown below.
Condition for Stability of LTI System
The condition for stability of an LTI system is,
+∞

∑ h( n ) < ∞ .....(2.24)
n =−∞
i.e., an LTI system is stable if the impulse response is absolutely summable.
Chapter 2 - Discrete Time Signals and Systems 2. 48
Proof
Let, x(n) = Input to LTI system.
y(n) = Response of LTI system for the input x(n).
Now, by convolution sum formula,
y(n) = x(n) * h(n) = h(n) * x(n) Convolution satisfy
commutative property.
+∞
= ∑
m = −∞
h(m) x( n − m)

+∞ Taking absolute
∴ y(n) = ∑ h(m) x( n − m) value on both sides.
m =−∞
+∞
For linear system the order summation
= ∑ h(m) x( n − m) and absolute value can be interchanged.
m = −∞
+∞
For linear system the order of multiplication
= ∑ h(m) x( n − m)
and absolute value can be interchanged.
m = −∞
+∞
= ∑ h(m) M X If input is bounded, then
|x(n – m)| = constant = MX
m = −∞
+∞
= MX ∑ h(m) MX is indepentent of
m =−∞ summation index m.
+∞
= MX ∑ h( n) Change index m to n.
n = −∞

In the above equation, if

+∞
.....(2.25)
∑ h(n) < ∞
n = −∞

then the response y(n) is bounded.

Example 2.17
Test the stability of the following systems.
a) y(n) = cos[x(n)] b) y(n) = x(–n – 3) c) y(n) = n x(n)
Solution
a) Given that, y(n) = cos [x(n)]
The given system is a nonlinear system, and so the test for stability should be performed for specific inputs.
The value of cos q lies between –1 to +1 for any value of q. Therefore the output y(n) is bounded for any
value of input x(n). Hence the given system is stable.
b) Given that, y(n) = x(–n – 3)
The given system is a time variant system, and so the test for stability should be performed for specific
inputs.
The operations performed by the system on the input signal are folding and shifting. A bounded input
signal will remain bounded even after folding and shifting. Therefore in the given system, the output will be
bounded as long as input is bounded. Hence the given system is BIBO stable.
c) Given that, y(n) = n x(n)
The given system is a time variant system, and so the test for stability should be performed for specific
inputs.
2. 49 Digital Signal Processing
Case i : If x(n) tends to infinity or constant, as "n" tends to infinity, then y(n) = n x(n) will be infinite as "n"
tends to infinity. So the system is unstable.
Case ii : If x(n) tends to zero as "n" tends to infinity, then y(n) = n x(n) will be zero as "n" tends to infinity.
So the system is stable.
Example 2.18
Determine the range of values of "p" and "q" for the stability of LTI system with impulse response,
h(n) = pn ; n < 0
= qn ; n ≥ 0
Solution
∞

The condition to be satisfied for the stability of the system is, ∑

n = −∞
|h(n)| < ∞.

Given that, h(n) = pn ; n < 0

= qn ; n ³ 0
∞ −1 ∞ ∞ ∞
∴ ∑
n = −∞
h(n) = ∑ p +∑ q
n = −∞
n

n= 0
n
= ∑p
n=1
−n
+ ∑
n=0
qn

∞ ∞ ∞ ∞ n is always positive.
1 1 n
= ∑
n=1
+
pn n= 0 ∑
qn = ∑
n =1 p
n
+ ∑
n= 0
q

n
∞ F 1I ∞
n |p|0 = 1
= ∑ GH p JK
n=0
− 1+ ∑
n= 0
q

The summation of infinite terms in the above equation converges if, 0 < 1/|p| < 1 and 0 < |q| < 1. Hence
by using infinite geometric series formula,
+∞
1 1 Infinite geometric
∑
|h(n)| =
1
−1+
1 − | q| series sum formula
n = −∞ 1−
p ∞
1
Cn = ∑
= constant 1 − C
n = 0
Therefore, the system is stable if |p| > 1 and |q| < 1. if 0 < C < 1

Example 2.19
Test the stability of LTI systems, whose impulse responses are,

a) h(n) = 0.2n u(n) b) h(n) = 0.3n u(n) + 2n u(n)

n
c) h(n) = 4 u(-n) d) h(n) = 0.2n u(-n) + 3n u(-n)

Solution
a) h(n) = 0.2n u(n)

+∞ +∞ ∞ Infinite geometric
∴ ∑ h(n) = ∑ 0.2n u(n) = ∑ 0.2n series sum formula
n= −∞ n = −∞ n= 0
∞
1 1
= = 1. 25 ∑ Cn =
1 − 0.2 n = 0
1− C
+∞ if 0 < C < 1
Since, ∑ h(n) < ∞, system is stable.
n= −∞
Chapter 2 - Discrete Time Signals and Systems 2. 50
b) h(n) = 0.3n u(n) + 2n u(n)
+∞ +∞
∴ ∑ h(n) = ∑ 0.3n u(n) + 2n u(n)
n= −∞ n = −∞ ∞
∞
n
∞
n 1 ∑ Cn = ∞
= ∑ 0.3 + ∑ 2 u(n) =
1 − 0.3
+∞=∞ n = 0
if C > 1
n=0 n= 0
+∞
Since, ∑ h(n) = ∞, system is unstable.
n= −∞

c) h(n) = 4n u(-
-n)
+∞ +∞ 0 +∞
∴ ∑ h(n) = ∑ 4n u( −n) = ∑ 4n = ∑ 4−n
n = −∞ n = −∞ n = −∞ n= 0
∞ ∞ n ∞
=
1
∑ 4n = ∑ GH 4 JK
F 1I = ∑ 0.25 n
=
1
= 1. 3333
n=0 n= 0 n= 0
1 − 0.25
+∞
Since, ∑ h(n) < ∞ , system is stable.
n= −∞

d) h(n) = 0.2n u(-

-n) + 3n u(-
-n)

+∞ +∞
∴ ∑ h(n) = ∑ 0.2n u( −n) + 3n u( −n)
n= −∞ n= −∞
0 0 +∞ +∞
= ∑ 0.2n + ∑ 3n = ∑ 0.2−n + ∑ 3−n
n= −∞ n= −∞ n= 0 n=0
∞ ∞ ∞ n ∞ n
= ∑
1
+
1
= ∑
1
∑ GH
F IJ + ∑ FG 1IJ
n=0 0.2n n=0 3n n=0 0.2 K H 3K n=0
∞ ∞
n n 1
= ∑5 +∑ 0.333 = ∞ +
1 − 0.333
=∞
n=0 n= 0
+∞
Since, ∑ h(n) = ∞, system is unstable.
n= −∞

2.8.6 FIR and IIR Systems

In FIR system (Finite duration Impulse Response system), the impulse response consists of finite
number of samples. The convolution formula for FIR system is given by,
N −1
.....(2.26)
y( n ) = ∑ h( m) x( n − m)
m=0

where, h(n) = 0 ; for n < 0 and n ³ N

From equation (2.26) it can be concluded that the impulse response selects only N samples of the input
signal.In effect, the system acts as a window that views only the most recent N input signal samples in
forming the ouput. It neglects or simply forgets all prior input samples. Thus a FIR system requires memory
of length N. In general, a FIR system is described by the difference equation,
N −1
.....(2.27)
y( n) = ∑ bm x( n − m)
m= 0

where, bm = h(m) ; for m = 0 to N -1

2. 51 Digital Signal Processing
In IIR system (Infinite duration Impulse Response system), the impulse response has infinite number
of samples. The convolution formula for IIR systems is given by,
∞
.....(2.28)
y( n ) = ∑ h( m) x( n − m)
m=0
Since this weighted sum involves the present and all the past input sample, we can say that the IIR
system requires infinite memory. In general, an IIR system is described by the difference equation,
N M
y( n) = − ∑ a m y( n − m) + ∑ bm x( n − m)
m =1 m= 0

2.8.7 Recursive and Nonrecursive Systems

A system whose output y(n) at time n depends on any number of past output values as well as present
and past inputs is called a recursive system. The past outputs are y(n – 1), y(n – 2), y(n – 3), etc.,.
Hence for recursive system, the output y(n) is given by,
y(n) = F [y(n - 1), y(n - 2),...y(n - N), x(n), x(n - 1),...x(n - M)]
A system whose output does not depend on past output but depends only on the present and past
input is called a nonrecursive system.
Hence for nonrecursive system, the output y(n) is given by,
y(n) = F [x(n), x(n – 1) ,....., x(n – M)]
In a recursive system, in order to compute y(n0), we need to compute all the previous values y(0),
y(1) ,......., y(n0 – 1) before calculating y(n0). Hence the output samples of a recursive system has to be
computed in order [i.e., y(0), y(1), y(2), ....]. The IIR systems are recursive systems.
In nonrecursive system, y(n 0 ) can be computed immediately without having y(n 0 - 1),
y(n0-2)..... Hence the output samples of nonrecursive system can be computed in any order [i.e. y(50), y(5),
y(2), y(100),....]. The FIR systems are nonrecursive systems.

2.9 Discrete or Linear Convolution

The Discrete or Linear convolution of two discrete time sequences x1(n) and x2(n) is defined as,
+∞ +∞
x3 ( n) = ∑
m = −∞
x1 ( m) x2 (n − m) or x 3 ( n) = ∑
m = −∞
x2 (m) x1 (n − m) .....(2.29)

where, x3(n) is the sequence obtained by convolving x1(n) and x2(n)

m is a dummy variable
The convolution relation of equation (2.29) can be symbolically expressed as,
x3(n) = x1(n) * x2(n) = x2(n) * x1(n) ..... (2.30)
where, the symbol * indicates convolution operation.
In linear convolution, the sequences x1(n) and x2(n) are nonperiodic sequences and the sequence x3(n)
obtained by convolution is also nonperiodic. Hence this convolution is also called aperiodic convolution.
Procedure For Evaluating Linear Convolution
Let, x1(n) = Discrete time sequence with N1 samples
x2(n) = Discrete time sequence with N2 samples
Now, the convolution of x1(n) and x2(n) will produce a sequence x3(n) consisting of N1+N2–1 samples.
Each sample of x3(n) can be computed using the equation (2.29). The value of x3(n) at n = q is obtained by
replacing n by q, in equation (2.29).
Chapter 2 - Discrete Time Signals and Systems 2. 52

+∞
.....(2.31)
∴ x3 ( q ) = ∑ x1( m) x2 (q − m)
m =−∞
The evaluation of equation (2.31) to determine the value of x3(n) at n = q, involves the following five
steps.
1. Change of index : Change the index n in the sequences x 1 (n) and x 2(n), to get the
sequences x1(m) and x2(m).
2. Folding : Fold x2(m) about m = 0, to obtain x2(-m).
3. Shifting : Shift x2(-m) by q to the right if q is positive, shift x2(-m) by q to the left
if q is negative to obtain x2(q - m).
4. Multiplication : Multiply x1(m) by x2(q - m) to get a product sequence. Let the product
sequence be vq(m). Now, vq(m) = x1(m) × x2(q - m).
5. Summation : Sum all the values of the product sequence vq(m) to obtain the value of
x3(n) at n = q. [i.e., x3(q)].
The above procedure will give the value of x3(n) at a single time instant say n = q. In general, we are
interested in evaluating the values of the sequence x3(n) over all the time instants in the range -¥ < n < ¥ .
Hence the steps 3, 4 and 5 given above must be repeated, for all possible time shifts in the range
-¥ < n < ¥ .
Convolution of finite duration sequences
In convolution of finite duration sequences it is possible to predict the length of resultant sequence.
If the sequence x1(n) has N1 samples and sequence x2(n) has N2 samples then the output sequence
x3(n) will be a finite duration sequence consisting of "N1+N2–1" samples.
i.e., if, Length of x1(n) = N1
Length of x2(n) = N2
then, Length of x3(n) = N1 + N2 – 1
In the convolution of finite duration sequences it is possible to predict the start and end of the
resultant sequence. If x1(n) starts at n = n1 and x2(n) starts at n = n2 then, the initial value of n for x3(n) is
"n = n1 + n2". The value of x1(n) for n < n1 and the value of x2(n) for n < n2 are then assumed to be zero.The final
value of n for x3(n) is "n = (n1 + n2) + (N1 + N2 – 2)".
i.e., if, x1(n) start at n = n1
x2(n) start at n = n2
then, x3(n) start at n = n1 + n2
and x3(n) end at n = (n1 + n2) + (N1 + N2 – 1) – 1
= (n1 + n2) + (N1 + N2 – 2)

2.9.1 Representation of Discrete Time Signal as Summation of Impulses

A discrete time signal can be expressed as summation of impulses and this concept will be useful to
prove that the response of discrete time LTI system can be determined using discrete convolution.
Let, x(n) = Discrete time signal
d(n) = Unit impulse signal
d(n - m) = Delayed impulse signal
2. 53 Digital Signal Processing
We know that, d(n) = 1 ; at n = 0
= 0 ; when n ¹ 0
and, d(n – m) = 1 ; at n = m
= 0 ; when n ¹ m
If we multiply the signal x(n) with the delayed impulse d(n - m) then the product is nonzero only at
n = m and zero for all other values of n. Also at n = m, the value of product signal is mth sample x(m) of the
signal x(n).
\ x(n) d(n - m) = x(m)
Each multiplication of the signal x(n) by an unit impulse at some delay m, in essence picks out the
single value x(m) of the signal x(n) at n = m, where the unit impulse is nonzero. Consequently if we repeat this
multiplication for all possible delays in the range -¥ < m < ¥ and add all the product sequences, the result will
be a sequence that is equal to the sequence x(n).
For example, x(n) d(n - (-2)) = x(-2)
x(n) d(n - (-1)) = x(-1)
x(n) d(n) = x(0)
x(n) d(n - 1) = x(1)
x(n) d(n - 2) = x(2)
From the above products we can say that each sample of x(n) can be expressed as a product of the
sample and delayed impulse, as shown below.
\ x(-2) = x(-2) d(n-(-2))
x(-1) = x(-1) d(n - (-1))
x(0) = x(0) d(n)
x(1) = x(1) d(n - 1)
x(2) = x(2) d(n - 2)
\ x(n) = ..... + x(-2) + x(-1) + x(0) + x(1) + x(2) + ...........
= ..... + x(-2) d(n - (-2)) + x(-1) d(n - (-1)) + x(0) d(n) + x (1) d(n - 1)
+ x(2) d(n - 2) + ...........
+∞
.....(2.32)
= ∑ x( m) δ( n − m)
m =−∞

In equation (2.32) each product x(m) d(n – m) is an impulse and the summation of impulses gives the
sequence x(n).

2.9.2 Response of LTI Discrete Time System Using Discrete Convolution

In an LTI system, the response y(n) of the system for an arbitrary input x(n) is given by convolution
of input x(n) with impulse response h(n) of the system. It is expressed as,
+∞
.....(2.33)
y ( n) = x ( n ) * h ( n ) = ∑ x( m) h( n − m)
m =−∞
where, the symbol * represents convolution operation.
Chapter 2 - Discrete Time Signals and Systems 2. 54
Proof :
Let y(n) be the response of system H for an input x(n)
\ y(n) = H{x(n)} .....(2.34)

From equation (2.32) we know that the signal x(n) can be expressed as a summation of impulses,
+∞
.....(2.35)
i.e., x(n) = ∑
m = −∞
x(m) δ(n − m)

where, d(n – m) is the delayed unit impulse signal.

From equations (2.34) and (2.35) we get,

y(n) = H
|RS ∑
+∞
x(m) δ(n − m)
|UV .....(2.36)
|T
m = −∞ |W
The system H is a function of n and not a function of m. Hence by linearity property the equation
(2.36) can be written as,
+∞
.....(2.37)
y(n) = ∑
m = −∞
x(m) H {δ(n − m)}

Let the response of the LTI system to the unit impulse input d(n) be denoted by h(n),
\ h(n) = H{d(n)}
Then by time invariance property the response of the system to the delayed unit impulse input
d(n – m) is given by,
h(n – m) = H{d(n – m)} .....(2.38)
Using equation (2.38), the equation (2.37) can be expressed as,
+∞
y( n) = ∑
m = −∞
x( m) h(n − m)

The above equation represents the convolution of input x(n) with the impulse response h(n) to yield
the output y(n). Hence it is proved that the response y(n) of LTI discrete time system for an
arbitrary input x(n) is given by convolution of input x(n) with impulse response h(n) of the system.

2.9.3 Properties of Linear Convolution

The Discrete or Linear convolution will satisfy the following properties.
Commutative property : x1(n) * x2(n) = x2(n) * x1(n)
Associative property : [x1(n) * x2(n)] * x3(n) = x1(n) * [x2(n) * x3(n)]
Distributive property : x1(n) * [x2(n) + x3(n)] = [x1(n) * x2(n)] + [x1(n) * x3(n)]
Proof of Commutative Property :
Consider convolution of x1(n) and x2(n).
By commutative property we can write,
x1(n) * x2(n) = x2(n) * x1(n)
(LHS) (RHS)
LHS = x1(n) * x2(n)
+∞
= ∑
m = −∞
x1( m) x 2(n − m) .....(2.39)

where, m is a dummy variable used for convolution operation.

2. 55 Digital Signal Processing
Let, n–m=p when m = –¥ , p = n – m = n + ¥ = +¥
\ m=n–p when m = +¥ , p = n – m = n – ¥ = –¥
On replacing m by (n – p) and (n – m) by p in equation (2.39) we get,
+∞ +∞
LHS = ∑
p = −∞
x1( n − p) x2( p) = ∑
p = −∞
x 2( p) x1( n − p)

= x2 (n) * x1(n) p is a dummy variable used for convolution operation.

= RHS

Proof of Associative Property :

Consider the discrete time signals x1(n), x2(n) and x3(n).
Let, y1(n) = x1(n) * x2(n) .....(2.40)
Let us replace n by p
\ y1(p) = x1(p) * x2(p)
+∞
= ∑
m = −∞
x1( m) x 2(p − m) .....(2.41)

Let, y2(n) = x2(n) * x3(n) .....(2.42)

+∞
∴ y2( n) = ∑
q = −∞
x1( q) x 2(n − q)

+∞
∴ y2(n − m) = ∑
q = −∞
x1( q) x2(n − q − m) .....(2.43)

where p, m and q are dummy variables used for convolution operation.

By associative property we can write,
[x1 (n) * x2(n)] * x3(n) = x1(n) * [x2 (n) * x3(n)]
LHS RHS
LHS = [x1(n) * x2(n)] * x3(n)
= y1(n) * x3(n) Using equation (2.40)
+∞
= ∑
p = −∞
y1( p) x 3(n − p)

+∞ +∞
= ∑
p = −∞
∑
m = −∞
x1( m) x 2(p− m) x 3 (n − p) Using equation (2.41)

+∞ +∞
= ∑
m = −∞
x1(m) ∑
p = −∞
x 2 (p − m) x 3(n − p) .....(2.44)

Let, p – m = q when p = –¥ , q = p – m = –¥ – m = –¥
\p=q+m when p = +¥ , q = p – m = +¥ – m = +¥
On replacing (p – m) by q, and p by (q + m) in the equation (2.44) we get,
+∞ +∞
LHS = ∑ x1( m) ∑ x 2( q) x 3 ( n − q − m)
m = −∞ q = −∞
+∞
= ∑ x1( m) y2 ( n − m) Using equation (2.43)
m = −∞

= x1 (n) * y2(n)
= x1(n) * [x2(n) * x3(n)] Using equation (2.42)
= RHS
Chapter 2 - Discrete Time Signals and Systems 2. 56
Proof of Distributive Property :
Consider the discrete time signals x1(n), x2(n) and x3(n). By distributive property we can write,
x1 (n) * [x2(n) + x3(n)] = [x1(n) * x2 (n)] + [x1(n) * x3 (n)]
LHS RHS
LHS = x1(n) * [x2(n) + x3(n)]
= x1(n) * x4(n) x4(n) = x2(n) + x3(n)
+∞
= ∑
m = −∞
x1( m) x 4(n − m) m is a dummy variable used for convolution operation.
+∞
x4(n – m) = x2(n – m) + x3(n – m)
= ∑
m = −∞
x1( m) [x 2( n − m) + x 3(n − m)]

+∞ +∞
= ∑
m = −∞
x1( m) x 2(n − m) + ∑
m = −∞
x1( m) x 3(n − m)

= [x1 (n) * x2(n)] + [x1(n) * x3 (n)]

= RHS

2.9.4 Interconnections of Discrete Time Systems

Smaller discrete time systems may be interconnected to form larger systems. Two possible basic ways
of interconnection are cascade connection and parallel connection. The cascade and parallel connections
of two discrete time systems with impulse responses h1(n) and h2(n) are shown in fig 2.21.
h1(n)
x(n) y1(n) y(n) x(n) y(n)
h1(n) h 2 (n) +

h 2 (n)

F ig 2.21 a : C a sca d e c on n e ctio n . F ig 2.21 b : P a ralle l co n n ec tio n.

F ig 2.2 1 : In terco n n ec tio n o f d iscrete tim e system s.
Cascade Connected Discrete Time Systems
Two cascade connected discrete time systems with impulse response h1(n) and h2(n) can be replaced
by a single equivalent discrete time system whose impulse response is given by convolution of individual
impulse responses.
x(n) y1(n) y(n) x(n) y(n)
h1(n) h 2 (n) ⇒ h1(n) ∗ h 2 (n)

F ig 2.2 2 : C a sca de c o n ne cted d iscrete tim e syste m a n d th eir e q u iv a lent.

Proof:
With reference to fig 2.22 we can write,
y1(n) = x(n) * h1(n) .....(2.45)
y(n) = y1(n) * h2(n) .....(2.46)
Using equation (2.45), the equation (2.46) can be written as,
y(n) = x(n) * h1(n) * h2(n)
= x(n) * [h1(n) * h2(n)]
= x(n) * h(n) .....(2.47)
where, h(n) = h1(n) * h2(n)
From equation (2.47) we can say that the overall impulse response of two cascaded discrete
time systems is given by convolution of individual impulse responses.
2. 57 Digital Signal Processing
Parallel Connected Discrete Time Systems
Two parallel connected discrete time systems with impulse responses h1(n) and h2(n) can be replaced
by a single equivalent discrete time system whose impulse response is given by sum of individual impulse
responses.
y 1 (n)
h 1 (n)
y (n) x (n) y (n)
x (n)
+ ⇒ h 1 (n) + h 2 (n)
y 2 (n)
h 2 (n)

F ig 2.2 3 : P a ra llel c o n ne cted d iscrete tim e syste m s an d th eir eq u iv a len t.

Proof:
With reference to fig 2.23 we can write,
y1(n) = x(n) * h1(n) .....(2.48)
y2(n) = x(n) * h2(n) .....(2.49)
y(n) = y1(n) + y2(n) .....(2.50)
On substituting for y1(n) and y2(n) from equations (2.48) and (2.49) in equation (2.50) we get,
y(n) = [ x(n) * h1(n)] + [ x(n) * h2(n)] .....(2.51)
By using distributive property of convolution, the equation (2.51) can be written as shown below,
y(n) = x(n) * [h1(n) + h2(n)]
= x(n) * h(n) .....(2.52)
where, h(n) = h1(n) + h2(n)
From equation (2.52) we can say that the overall impulse response of two parallel connected
discrete time systems is given by sum of individual impulse responses.

Example 2.20
Determine the impulse response for the cascade of two LTI systems having impulse responses,
n n
h1(n) =
FG 2IJ u(n) and h2 (n) =
FG 1IJ u(n).
H 5K H 5K
Solution
Let h(n) be the impulse response of cascade system. Now h(n) is given by convolution of h1(n) and h2(n).
+∞
\ h(n) = h1(n) * h2(n) = ∑
m = −∞
h1(m) h2 (n − m)

where, m is a dummy variable used for convolution operation

m m n−m
h1(m) =
FG 2IJ ; h2 (m) =
FG 1IJ ; h2 (n − m) =
FG 1IJ
H 5K H 5K H 5K
The product h1(m) h2(n – m) will be nonzero in the range 0 £ m £ n. Therefore the summation index in the
above equation is changed to m = 0 to n.
n n m n − m n m n −m n n m
∴ h(n) = ∑ h1(m) h2 (n − m) = ∑
FG 2IJ FG 1IJ = ∑
FG 2IJ FG 1IJ FG 1IJ =
FG 1IJ ∑ FG 2IJ 5m
m= 0 m= 0
H 5K H 5K m= 0
H 5K H 5 K H 5 K H 5K H 5K
m= 0
n n m n n
=
FG 1IJ ∑
FG 2 × 5IJ = FG 1IJ ∑ 2 m
Finite geometric series
H 5K m= 0
H 5 K H 5K m= 0 sum formula
n n
F 1I F 2 − 1I = FG 1IJ (2 − 1)
n+1 N
CN + 1 − 1
=G J
H 5K GH 2 − 1 JK H 5K n+1
; for n ≥ 0 ∑
n = 0
Cn =
C −1
n
F 1I
=G J (2n + 1 − 1) u(n) ; for all n Using finite geometric series sum formula.
H 5K
Chapter 2 - Discrete Time Signals and Systems 2. 58

Example 2.21
Determine the overall impulse response of the interconnected discrete time systems shown below,
a) b)

h1(n) h1(n) h2 (n)

x(n) y(n) y(n)
+ x(n)
+
h 2 (n) + h 3 (n)
h 3 (n) h1(n)

n n
h1(n) =
FG 1IJ u(n); h2 (n) =
FG 1IJ u(n); h1(n) = an u(n) ; h2 (n) = δ(n − 1) ; h3 (n) = δ(n − 2)
H 3K H 2K
n
F 1I
h (n) = G J u(n)
3
H 5K
Solution
a) The given system can be redrawn as shown below.

h1(n)

y(n)
+
x(n)
h1(n)

+ h 3 (n)

h2 (n)

The above system can be reduced to single equivalent system as shown below.

y1(n)
h1(n)
y(n)
x(n) h1(n) + [(h1(n) + h2 (n)) ∗ h 3 (n)] y(n)
x(n)
+ ⇒
⇓
h1(n) + h2 (n) h 3 (n)
x(n) h(n) y(n)

Here, h(n) = h1(n) + [(h1(n) + h2(n)) * h3(n)]

= h1(n) + [h1(n) * h3(n)] + [h2(n) * h3(n)] Using distributive property.
Let us evaluate the convolution of h1(n) and h3(n).
∞
h1(n) ∗ h3 (n) = ∑
m = −∞
h1(m) h3 (n − m)

The product of h1(m) h3(n – m) will be nonzero in the range 0 £ m £ n. Therefore the summation index in
the above equation can be changed to m = 0 to n.
n
∴ h1(n) ∗ h3(n) = ∑ h (m) h (n
m = 0
1 3 − m)

n m n − m n m n −m

∑ FGH 3IJK FGH 5IJK ∑ FGH 3IJK FGH 5IJK FGH 5IJK
1 1 1 1 1
= =
m = 0 m = 0
n n m n n m
=
FG 1IJ ∑ FG 1IJ 5m =
FG 1IJ ∑ FG 5IJ
H 5K H 3K
m = 0
H 5K H 3K
m = 0
2. 59 Digital Signal Processing
n +1
FG 5IJ − 1 n Using finite geometric series sum formula.
F 1I H 3 K
∴ h (n) ∗ h (n) = G J
1 3
H 5K 5 − 1 Finite geometric series
3
n sum formula
FG 5IJ 5 − 1
n LM 3 FG 5IJ 5 − 3 OP
n n N
CN+1 − 1
1I H 3 K 3
F
=G J
F 1I
H 5K 5 − 3 = GH 5JK ∑ Cm =
C −1
3
MN 2 H 3K 3 2 PQ m = 0

n n n n n
=
FG 1IJ
5 FG 5IJ − 3 FG 1IJ = 5 FG 1IJ − 3 FG 1IJ ; for n ≥ 0
H 5K
2 H 3 K 2 H 5K 2 H 3K 2 H 5K
n n
5 F 1I 3 F 1I
= G J u(n) − G J u(n) ; for all n
2 H 3K 2 H 5K

Let us evaluate the convolution of h2(n) and h3(n).

+∞
h2 (n) ∗ h3 (n) = ∑
m = −∞
h2 (m) h3(n − m)

The product of h2(m) and h3(n–m) will be nonzero in the range 0 £ m £ n. Therefore the summation index
in the above equation can be change to m = 0 to n.
n
∴ h2 (n) ∗ h3 (n) = ∑
m = 0
h2 (m) h3(n − m)

n m n − m n m n −m
= ∑
FG 1IJ FG 1IJ = ∑
FG 1IJ FG 1IJ FG 1IJ Finite geometric series
m = 0
H 2K H 5 K m = 0
H 2K H 5K H 5K sum formula
n m n m N
FG 1IJ n
F 1I FG 1IJ ∑ FG 5IJ n
CN+1 − 1
=
H 5K ∑ GH 2JK 5 m
=
H 5K H 2K ∑
m = 0
Cm =
C −1
m = 0 m = 0
n + 1

n
FG 5IJ − 1
= GHF 51IJK H 2K Using finite geometric series sum formula.
5
−1
2
n
F 5I 5
=
FG 1IJ GH 2JK 2 − 1 = FG 1IJ LM 2 FG 5IJ 5 − 2 OP
n n n

H 5K 5 − 2 H 5K MN 3 H 2K 2 3 PQ
2
n n n n n
= G J G J − 32 FGH 51IJK = 35 FGH 21IJK − 32 FGH 51IJK
5 F 1I F 5 I
3 H 5K H 2K
for n ≥ 0
n n
5 F 1I
= G J u(n) − 32 FGH 51IJK u(n) for all n
3 H 2K
Now, the overall impulse response h(n) is given by,

h(n) = h1(n) + h1(n) ∗ h3(n) + h2 (n) ∗ h3 (n)

n n n n n
=
FG 1IJ u(n) + 5 FG 1IJ u(n) − 3 FG 1IJ u(n) + 5 FG 1IJ u(n) −
2 FG 1IJ u(n)
H 3K 2 H 3K 2 H 5K 3 H 2K 3 H 5K
n n n
=
FG1+ 5IJ FG 1IJ u(n) − FG 3 + 2IJ FG 1IJ u(n) + 5 FG 1IJ u(n)
H 2 K H 3K H 2 3 K H 5K 3 H 2K

=
LM 7 FG 1IJ − 13 FG 1IJ + 5 FG 1IJ OP u(n)
n n n

MN 2 H 3K 6 H 5K 3 H 2K PQ
Chapter 2 - Discrete Time Signals and Systems 2. 60
b) The given system can be reduced to single equivalent system as shown below.

h1(n) ∗ h 2 (n)

x(n) y(n)
+

h3 (n) ∗ h1(n)

⇓
x(n) [h1(n) ∗ h2 (n)] + [h 3 (n) ∗ h1(n)] y(n)

⇓
x(n) h(n) y(n)

Here, h(n) = [h1(n) * h2(n)] + [h3(n) * h1(n)]

Let us evaluate the convolution of h1(n) and h2(n).

∞
h1(n) ∗ h2 (n) = ∑
m = −∞
h1(m) h2 (n − m)

∞
= ∑
m = −∞
h2 (m) h1(n − m) Using commutative property.

∞ ∞
= ∑
m = −∞
δ(m − 1) a(n − m) = ∑
m = −∞
δ(m − 1) an a −m

∞
=a n
∑
m = −∞
δ(m − 1) a −m

The product of d(m – 1) and a–m in the above equation will be nonzero only when m = 1.

\ h1(n) * h2(n) = an a–1 = an – 1 ; for n ³ 1

= an – 1 u(n – 1) ; for all n.

Let us evaluate the convolution of h3(n) and h1(n).

∞
h3(n) ∗ h1(n) = ∑
m = −∞
h3(m) h1(n − m)

∞ ∞
= ∑
m = −∞
δ(m − 2) a (n − m) = ∑
m = −∞
δ(m − 2) an a −m

∞
=a n
∑ δ(m − 2) a −m

m = −∞

The product of d(m – 2) and a–m in the above equation will be nonzero only when m = 2.

\ h1(n) * h2(n) = an a–2 = an – 2 ; for n ³ 2

= an – 2 u(n – 2) ; for all n

Now, the overall impulse response h(n) is given by,

h(n) = [h1(n) * h2(n)] + [h3(n) * h1(n)]

= a(n – 1) u(n – 1) + a(n – 2) u(n – 2)

2. 61 Digital Signal Processing

2.9.5 Methods of Performing Linear Convolution

Method 1: Graphical Method

Let x1(n) and x2(n) be the input sequences and x3(n) be the output sequence.

1. Change the index "n" of input sequences to "m" to get x1(m) and x2(m).
2. Sketch the graphical representation of the input sequences x1(m) and x2(m).
3. Let us fold x 2 (m) to get x 2(–m). Sketch the graphical representation of the folded
sequence x2(–m).
4. Shift the folded sequence x2(–m) to the left graphically so that the product of x1(m) and
shifted x2(–m) gives only one nonzero sample. Now multiply x1(m) and shifted x2(–m) to get
a product sequence, and then sum up the samples of product sequence, which is the first
sample of output sequence.
5. To get the next sample of output sequence, shift x2(–m) of previous step to one position right
and multiply the shifted sequence with x1(m) to get a product sequence. Now the sum of the
samples of product sequence gives the second sample of output sequence.
2. To get subsequent samples of output sequence, the step 5 is repeated until we get a nonzero
product sequence.
Method 2: Tabular Method
The tabular method is same as that of graphical method, except that the tabular representation of the
sequences are employed instead of graphical representation. In tabular method, every input sequence, folded
and shifted sequence is represented by a row in a table.
Method 3: Matrix Method
Let x1(n) and x2(n) be the input sequences and x3(n) be the output sequence. In matrix method one of
the sequences is represented as a row and the other as a column as shown below.
Multiply each column element with row elements and fill up the matrix array.
Now the sum of the diagonal elements gives the samples of output sequence x3(n). (The sum of the
diagonal elements are shown below for reference).
x 2 (0) x 2 (1) x 2 (2) x 2 (3)

x 1 (0) x 1 (0) x 2 (0) x 1 (0) x 2 (1) x 1 (0) x 2 (2) x 1 (0) x 2 (3)

x 1 (1) x 1 (1) x 2 (0) x 1 (1) x 2 (1) x 1 (1) x 2 (2) x 1 (1) x 2 (3)

x 1 (2) x 1 (2) x 2 (0) x 1 (2) x 2 (1) x 1 (2) x 2 (2) x 1 (2) x 2 (3)

x 1 (3) x 1 (3) x 2 (0) x 1 (3) x 2 (1) x 1 (3) x 2 (2) x 1 (3) x 2 (3)

Chapter 2 - Discrete Time Signals and Systems 2. 62

......

x3(0) = ..... + x1(0) x2(0) + .....

x3(1) = ..... + x1(1) x2(0) + x1(0 ) x2(1) + .....
x3(2) = ..... + x1(2) x2(0) + x1(1) x2(1) + x1(0) x2(2) + .....
x3(3) = ..... + x1(3) x2(0) + x1(2) x2(1) + x1(1) x2(2) + x1(0) x2(3) + .....
......

\ x3(n) = {..... x3(0), x3(1), x3(2), x3(3), .....}

Example 2.22
Determine the response of the LTI system whose input x(n) and impulse response h(n) are given by,
x(n) = {1, 2, 0.5, 1} and h(n) = {1, 2, 1, –1}
­ - -

Solution
The response y(n) of the system is given by convolution of x(n) and h(n).
+∞
y(n) = x(n) ∗ h(n) = ∑
m = −∞
x(m) h(n − m)

In this example the convolution operation is performed by three methods.

The Input sequence starts at n = 0 and the impulse response sequence starts at n = –1. Therefore the
output sequence starts at n = 0 + (–1) = –1.
The input and impulse response consists of 4 samples, so the output consists of 4 + 4 – 1 = 7 samples.
Method 1 : Graphical Method
The graphical representation of x(n) and h(n) after replacing n by m are shown below. The sequence h(m)
is folded with respect to m = 0 to obtain h(–m).

x (m ) h (m ) h ( −m )

2 2 2

1 1 1 1 1 1
0.5

0 1 2 3 m −1 0 1 2 m −2 −1 0 1 m
−1 −1
F ig 1 : In p u t sequ e n ce. F ig 2 : Im p u lse resp o n se . F ig 3 : F o ld ed im p ulse respo nse.

The samples of y(n) are computed using the convolution formula,

+∞ +∞
y(n) = ∑
m = −∞
x(m) h(n − m) = ∑
m = −∞
x(m) hn (m) ; where hn (m) = h(n − m)

The computation of each sample using the above equation are graphically shown in fig 4 to fig 10. The
graphical representation of output sequence is shown in fig 11.
2. 63 Digital Signal Processing
+∞ +∞ +∞
When n = −1 ; y(−1) = ∑
m = −∞
x(m) h( −1 − m) = ∑
m = −∞
x(m) h−1(m) = ∑
m = −∞
v −1(m)

h −1 (m ) x (m ) v −1 (m )

2
X 2 ⇒
1 1 1 1 1
0.5

−3 −2 −1 0 m 0 1 2 3 m −3 −2 −1 0 1 2 3 m
T he s um of pro du c t s e qu en c e v −1(m )
−1 F ig 4 : C o m p u ta tio n of y ( −1 ).
g iv e s y ( −1 ). ∴ y ( −1) = 1
+∞ +∞ +∞
When n = 0 ; y(0) = ∑
m = −∞
x(m) h(0 − m) = ∑
m = −∞
x(m) h0 (m) = ∑
m = −∞
v 0 (m)

h 0 (m ) x (m ) v 0 (m )

2 2 2
X 2 ⇒
1 1 1 1
0.5

−2 −1 0 1 m 0 1 2 3 m −2 −1 0 1 2 3 m
T h e s u m o f p rod uc t s eq ue nc e v 0 ( m )
−1 F ig 5 : C o m p u ta tion of y (0 ).
giv es y(0 ) . ∴ y (0) = 2 + 2 = 4
+∞ +∞ +∞
When n = 1 ; y(1) = ∑
m = −∞
x(m) h(1 − m) = ∑
m = −∞
x(m) h1(m) = ∑
m = −∞
v1(m)

h 1 (m ) x (m ) v 1 (m )
4

X ⇒
2 2

1 1 1 1 1
0.5 0.5
−1 0 1 2 m 0 1 2 3 m −1 0 1 2 3 m
−1 T he s um of pro du c t s e qu en c e v 1 (m )
F ig 6 : C o m p u ta tio n of y (1 ).
g iv e s y (1). ∴ y(1 ) = 1 + 4 + 0.5 = 5 .5
+∞ +∞ +∞
When n = 2 ; y(2) = ∑
m = −∞
x(m) h(2 − m) = ∑
m = −∞
x(m) h2 (m) = ∑
m = −∞
v 2 (m)

h 2 (m ) x (m ) v 2 (m )

X ⇒
2 2
2
1 1 1 1 1 1
0.5
0 1 2 3 m 0 1 2 3 m 0 1 2 3 m
−1
−1 T he s um of pro du c t s e qu en ce v 2 (m )
F ig 7 : C o m p u ta tio n of y (2 ). g iv e s y (2). ∴ y (2 ) = −1 + 2 + 1 + 1 = 3
Chapter 2 - Discrete Time Signals and Systems 2. 64
+∞ +∞ +∞
When n = 3 ; y(3) = ∑
m = −∞
x(m) h(3 − m) = ∑
m = −∞
x(m) h3 (m) = ∑
m = −∞
v 3 (m)

h 3 (m ) x (m ) v 3 (m )

X ⇒
2 2
2
1 1 1 1
0.5 0.5
1 2 3 4 m 2 m
0 0 1 3 0 1 2 3 4 m
−1
−2
F ig 8 : C o m p u ta tio n of y (3 ). T he su m o f pro du ct s e qu en ce v 3 (m )
giv es y (3). ∴ y (3) = −2 + 0.5 + 2 = 0 .5
+∞ +∞ +∞
When n = 4 ; y(4) = ∑
m = −∞
x(m) h(4 − m) = ∑
m = −∞
x(m) h4 (m) = ∑
m = −∞
v 4 (m)

h 4 (m ) x (m ) v 4 (m )

2 X ⇒
2
1 1
1 1 1
0.5
0 1 2 3 4 5 m
−1
0 1 2 3 m 0 1 2 3 4 5 m
−0.5
T h e s u m o f p rod uc t s eq ue nc e v 4 (m )
F ig 9 : C o m p u ta tio n of y (4 ).
give s y(4 ). ∴ y (4 ) = −0 .5 + 1 = 0 .5

+∞ +∞ +∞
When n = 5 ; y(5) = ∑
m = −∞
x(m) h(5 − m) = ∑
m = −∞
x(m ) h5 (m) = ∑
m = −∞
v 5 (m)

h 5 (m ) x (m ) v 5 (m )

2 X 2
⇒
1 1 1 1
0.5
0 1 2 3 4 5 6 m 0 1 2 3 m 0 1 2 3 4 5 6 m
−1
−1
F ig 1 0 : C o m p u ta tio n of y (5 ). T h e s um of p rod uc t s eq ue nc e v 5 (m )
g ive s y(5 ). ∴ y (5 ) = −1

y (n )
The output sequence, y(n) = {1, 4, 5.5, 3, 0.5, 0.5, − 1
A
} 5.5

1
0.5 0.5

−2 −1 0 1 2 3 4 5 6 7 n
−1
F ig 11 : G ra ph ica l rep resen ta tio n of y (n ).
2. 65 Digital Signal Processing
Method 2 : Tabular Method
The given sequences and the shifted sequences can be represented in the tabular array as shown below.

Note : The unfilled boxes in the table are considered as zeros.

m –3 –2 –1 0 1 2 3 4 5 6
x(m) 1 2 0.5 1
h(m) 1 2 1 –1
h(–m) –1 1 2 1
h(–1 – m) = h–1(m) –1 1 2 1
h(0 – m) = h0(m) –1 1 2 1
h(1 – m) = h1(m) –1 1 2 1
h(2 – m) = h2(m) –1 1 2 1
h(3 – m) = h3(m) –1 1 2 1
h(4 – m) = h4(m) –1 1 2 1
h(5 – m) = h5(m) –1 1 2 1
Each sample of y(n) is computed using the convolution formula,
+∞ +∞
y(n) = ∑ x(m) h(n − m)
m = −∞
= ∑ x(m) h (m),
m = −∞
n where hn (m) = h(n − m)

To determine a sample of y(n) at n = q, multiply the sequence x(m) and hq(m) to get a product sequence
(i.e., multiply the corresponding elements of the row x(m) and hq(m)). The sum of all the samples of the product
sequence gives y(q).
3
When n = −1 ; y( −1) = ∑
m = −3
x(m) h −1(m) Q The product is valid only for m = −3 to + 3.

= x(–3) h–1(–3) + x(–2)h–1(–2) + x(–1)h–1(–1) + x(0) h–1(0) + x(1) h–1(1)

+ x(2) h–1(2) + x(3) h–1(3)
=0+0+0+1+0+0+0=1
The samples of y(n) for other values of n are calculated as shown for n = –1.
3
When n = 0 ; y(0) = ∑ x(m) h0 (m) = 0 + 0 + 2 + 2 + 0 + 0 = 4
m = −2
3
When n = 1 ; y(1) = ∑ x(m) h1(m) = 0 + 1+ 4 + 0.5 + 0 = 5.5
m = −1
3
When n = 2 ; y(2) = ∑ x(m) h2(m) = −1+ 2 + 1+ 1= 3
m = 0
4
When n = 3 ; y(3) = ∑ x(m) h3 (m) = 0 − 2 + 0.5 + 2 + 0 = 0.5
m = 0
5
When n = 4 ; y(4) = ∑ x(m) h4 (m) = 0 + 0 − 0.5 + 1+ 0 + 0 = 0.5
m = 0
6
When n = 5 ; y(5) = ∑ x(m) h5 (m) = 0 + 0 + 0 − 1+ 0 + 0 + 0 = −1
m = 0

The output sequence, y(n) = l 1, 4, 5.5, 3, 0.5, 0.5, − 1q

A
Chapter 2 - Discrete Time Signals and Systems 2. 66
Method 3 : Matrix Method
The input sequence x(n) is arranged as a column and the impulse response is arranged as a row as shown
below. The elements of the two-dimensional array are obtained by multiplying the corresponding row element
with the column element. The sum of the diagonal elements gives the samples of y(n).

h(n) h(n)
x(n) 1 2 1 −1 x(n) 1 2 1 −1

1 1 2 1 −1
1 1 ×1 1 ×2 1 ×1 1 × (−1)
2 2 4 2 −2
2 2 ×1 2×2 2 ×1 2 ×(−1) ⇒
0.5 0.5 1 0.5 −0.5
0.5 0.5 × 1 0.5 × 2 0.5 × 1 0.5 × (−1)
1 1 2 1 −1
1 1 ×1 1×2 1 ×1 1 × (−1)

y(–1) = 1 y(3) = 2 + 0.5 + (–2) = 0.5

y(0) = 2 + 2 = 4 y(4) = 1 + (–0.5) = 0.5 \ y(n) = {1, 4, 5.5, 3, 0.5, 0.5, –1}
-
y(1) = 0.5 + 4 + 1 = 5.5 y(5) = –1
y(2) = 1 + 1 + 2 + (–1) = 3

Example 2.23
Determine the output y(n) of a relaxed LTI system with impulse response,
h(n) = an u(n) ; where |a| < 1 and
When input is a unit step sequence, i.e., x(n) = u(n).

Solution
The graphical representation of x(n) and h(n) after replacing n by m are shown below. Also the sequence
x(m) is folded to get x(–m).

h (m ) x (m ) x ( −m )
1 1 1
a
2
a
3
a

0 1 2 3 m 0 1 2 3 m −3 −2 −1 0 m
F ig 1 : Im p u lse resp o n se. F ig 2 : Im p u lse sequ en ce. F ig 3 : F o ld ed inp u t sequ en ce.

Here both h(m) and x(m) are infinite duration sequences starting at n = 0. Hence the output sequence y(n)
will also be an infinite duration sequence starting at n = 0.
By convolution formula,
∞ ∞
y(n) = ∑ h(m) x(n − m) = ∑ h(m) x (m) ;
m = −∞ m =0
n where xn (m) = x(n − m)

The computation of some samples of y(n) using the above equation are graphically shown below.
2. 67 Digital Signal Processing
∞ ∞ ∞
When n = 0 ; y(0) = ∑
m = 0
h(m) x(0 − m) = ∑
m = 0
h(m) x 0 (m) = ∑
m = 0
v 0 (m)

h (m ) x 0 (m ) v 0 (m )

1 1 1
a
a
2
X ⇒
3
a

0 1 2 3 m −3 −2 −1 m 0 1 2
0 1 m
F ig 4 : C o m p u ta tio n of y (0 ). y (0) = 1

∞ ∞ ∞
When n = 1 ; y(1) = ∑
m = 0
h(m) x(1 − m) = ∑
m = 0
h(m) x 1(m) = ∑
m = 0
v1(m)

h (m ) x 1 (m ) v 1 (m )

1 1 1
a a
a
2 X ⇒
3
a

0 1 2 3 m −2 −1 m
0 1 −1 0 1 2 m
y (1) = 1 + a
F ig 5 : C o m p u ta tio n of y (1 ).
∞ ∞ ∞
When n = 2 ; y(2) = ∑
m = 0
h(m) x(2 − m) = ∑
m = 0
h(m) x 2 (m) = ∑
m = 0
v 2 (m)

h (m ) x 2 (m ) v 2 (m )
1 1 1
a a
2
a ⇒ a
2

a
3 X

0 1 2 3 m −1 0 1 2 m 0 1 2 3 m
2
y(2) = 1 + a +a
F ig 6 : C o m p u ta tio n of y (2 ).
Solving similarly for other values of n, we can write y(n) for any value of n as shown below.
n
y(n) = 1 + a + a 2 +......+ an = ∑a
p=0
p
; for n ≥ 0

y (n )
3
1+a+a +a
2
1 + a + a2
1+a

0 1 2 3 m
F ig 7 : G ra ph ica l rep rese n ta tio n o f y (n).
Chapter 2 - Discrete Time Signals and Systems 2. 68

2.10 Circular Convolution

2.10.1 Circular Representation and Circular Shift of Discrete Time Signal
Consider a finite duration sequence x(n) and its periodic extension xp(n). The periodic extension of x(n)
can be expressed as xp(n) = x(n + N), where N is the periodicity. Let N = 4. The sequence x(n) and its periodic
extension are shown in fig 2.24.

Let, x(n) = 1 ; n=0

=2; n=1
=3; n=2
=4; n=3
x p (n )
x (n )
4 4 4 4

3 3 3 3

2 2 2 2

1 1 1 1

0 1 2 3 n −4 −3 −2 −1 0 1 2 3 4 5 6 7 n
F ig 2.2 4 a : F in ite d u ra tio n seq u en c e x(n ). F ig 2.2 4 b : P erio d ic e xten sio n o f x (n ).
F ig 2.2 4 : A fin ite d u ra tion seq ue n ce a n d its p erio d ic e xten sio n .

Let us delay the periodic sequence xp(n) by two units of time as shown in fig 2.25(a). (For delay the
sequence is shifted right). Let us denote one period of this delayed sequence by x1(n). One period of the
delayed sequence is shown in fig 2.25(b).
x p (n −2 )
x 1 (n ) x1 (n) = xp ((n − 2))4
4 4 4 4

3 3 3 3

2 2 2 2
1 1 1 1

−2 −1 0 1 2 3 4 5 6 7 8 9 n 0 1 2 3 n
F ig 2 .2 5 a: x p (n ) d e la y ed b y tw o un its o f tim e. F ig 2 .2 5 b: O ne period of x p (n −2 ).
F ig 2 .2 5 : D elay ed versio n o f x p (n).
The sequence x1(n) can be represented by xp(n – 2, (mod 4)), or xp((n – 2))4, where mod 4 indicates that
the sequence repeats after 4 samples. The relation between the original sequence x(n) and one period of the
delayed sequence x1(n) are shown below.

x1(n) = xp(n – 2, (mod 4)) = xp((n – 2))4

\ When n = 0; x1(0) = xp((0 – 2))4 = xp((– 2))4 = x(2) = 3

When n = 1; x1(1) = xp((1 – 2))4 = xp((– 1))4 = x(3) = 4

When n = 2; x1(2) = xp((2 – 2))4 = xp((0))4 = x(0) = 1

When n = 3; x1(3) = xp((3 – 2))4 = xp((1))4 = x(1) = 2

2. 69 Digital Signal Processing
The periodic sequences xp(n) and x1(n) can be represented as points on a circle as shown in fig 2.26.
From fig 2.26 we can say that, x1(n) is simply xp(n) shifted circularly by two units in time, where the counter
clockwise (anticlockwise) direction has been arbitrarily selected for right shift or delay.
2 1 4

x (1) = 2 x 1 (1) = 4
3 1 ⇒2 4 ⇒1 3

x (n) 4 3 2
x (2) = 3 x (0) = 1 x 1 (2) = 1 x 1 (n) x 1 (0) = 3
R otate x p(n) antic loc kw ise tw o tim es to get x 1(n)
= x p ((n − 2)) 4

x (3) = 4 x 1 (3) = 2
F ig 2.2 6 a: C ircu la r rep rese n ta tio n o f x (n). F ig 2.2 6 b: C ircu la r rep rese n ta tio n o f x 1 (n ).
F ig 2.2 6 : C ircu la r rep rese n ta tio n o f a sig n a l a nd its dela yed version .
Let us advance the periodic sequence xp(n) by three units of time as shown in fig 2.27(a). Let us denote
one period of this advanced sequence by x2(n). One period of the advanced sequence is shown in fig 2.27(b).

x p (n + 3 ) x 2 (n ) x 2 (n) = x p ((n + 3)) 4

4 4 4 4

3 3 3 3

2 2 2 2
1 1 1 1

−3 −2 −1 0 1 2 3 4 5 6 7 8 n 0 1 2 3 n
F ig 2 .2 7 a: x p (n ) a d v an c ed by th ree u n its o f tim e. F ig 2 .2 7 b: O ne p e rio d of x p (n + 3 ).
F ig 2 .2 7 : A d va n c ed v ersion of x p (n ).
The sequence x2(n) can be represented by xp(n + 3, (mod 4)) or xp((n + 3))4, where mod 4 indicates that the
sequence repeats after 4 samples. The relation between the original sequence x(n) and one period of the
advanced sequence x2(n) are shown below.
x2(n) = xp(n + 3, (mod 4)) = xp((n + 3))4
\ When n = 0; x2(0) = xp((0 + 3))4 = xp((3))4 = x(3) = 4
When n = 1; x2(1) = xp((1 + 3))4 = xp((4))4 = x(0) = 1
When n = 2; x2(2) = xp((2 + 3))4 = xp((5))4 = x(1) = 2
When n = 3; x2(3) = xp((3 + 3))4 = xp((6))4 = x(2) = 3
The periodic sequences xp(n) and x2(n) can be represented as points on a circle as shown in fig 2.28.
From fig 2.28 we can say that x2(n) is simply xp(n) shifted circularly by three units in time where clockwise
direction has been selected for left shift or advance.
2 3 4 1
x (1) = 2 x 2 (1) = 1
3 1 4 2 ⇒1 3 ⇒2 4

x p (n) 4 1 2 3 x 2 (n)
x (2) = 3 x (0) = 1 x 2 (2) = 2 x 2 (0) = 4
R otate x p(n) cloc k w is e three tim es to get x 2(n)

x (3) = 4 x 2 (3) = 3
F ig 2.2 8 a: C ircu la r rep rese nta tio n o f x (n). F ig 2.2 8 b: C ircu la r rep rese nta tio n o f x 2 (n ).
F ig 2.2 8 : C ircu la r rep rese nta tio n o f a sig n a l a n d its ad v a nced versio n.
Chapter 2 - Discrete Time Signals and Systems 2. 70
Thus we conclude that a circular shift of an N-point sequence is equivalent to a linear shift of its
periodic extension and viceversa. If a nonperiodic N-point sequence is represented on the circumference of
a circle then it becomes a periodic sequence of periodicity N. When the sequence is shifted circularly, the
samples repeat after N shifts. This is similar to modulo-N operation. Hence, in general, the circular shift may
be represented by the index mod-N. Let x(n) be an N-point sequence represented on a circle and x¢(n) be its
circularly shifted sequence by m units of time.
Now, x¢(n) = x(n – m, mod N) º x((n – m))N ..... (2.53)
When m is positive, the equation (2.53) represents delayed sequence and when m is negative, the
equation (2.53) represents advanced sequence.

2.10.2 Circular Symmetries of Discrete Time Signal

The circular representation of a sequence and the resulting periodicity gives rise to new definitions for
even symmetry, odd symmetry and the time reversal of the sequence.
An N-point sequence is called even if it is symmetric about the point zero on the circle. This implies
that,
x(N - n) = x(n) ; for 0 £ n £ N - 1 ...... (2.54)
An N-point sequence is called odd if it is antisymmetric about the point zero on the circle.
This implies that,
x(N - n) = - x(n) ; for 0 £ n £ N - 1 ...... (2.55)
The time reversal of a N-point sequence is obtained by reversing its sample about the point zero on the
circle. Thus the sequence x(–n, (mod N) ) is simply written as,
x (-n, (mod N)) = x(N - n) ; for 0 £ n £ N - 1 ...... (2.56)
This time reversal is equivalent to plotting x(n) in a clockwise direction on a circle, as shown in
fig 2.29.
x (2) x (6)

x (1) x (5) x (7)

x (3)

x (n) x (0) x ( −n)

x (4) x (4) x (0)

x (5) x (7) x (3) x (1)

x (6) x (2)
F ig 2.2 9 : C ircu la r rep rese n ta tio n o f a n 8 -p o in t seq ue n ce a n d its fo ld ed seq u en ce.
2.10.3 Definition of Circular Convolution
The circular convolution of two periodic discrete time sequences x1(n) and x2(n) with periodicity of
N samples is defined as,
N −1 N −1
.....(2.57)
x3 ( n) = ∑ x1( m) x2 (( n − m)) N or x 3 ( n) = ∑ x2 ( m) x1(( n − m)) N
m=0 m= 0

where, x3(n) is the sequence obtained by circular convolution,

x1((n – m))N represents circular shift of x1(n)
x2((n – m))N represents circular shift of x2(n)
m is a dummy variable.
2. 71 Digital Signal Processing
The output sequence x3(n) obtained by circular convolution is also a periodic sequence with periodicity
of N samples. Hence this convolution is also called periodic convolution.
The convolution relation of equation (2.57) can be symbolically expressed as
x3(n) = x1(n) * x2(n) = x2(n) * x1(n) ..... (2.58)
where, the symbol * indicates circular convolution operation.
The circular convolution is defined for periodic sequences. But circular convolution can be performed
with nonperiodic sequences by periodically extending them.The circular convolution of two sequences
requires that, at least one of the sequences should be periodic. Hence it is sufficient if one of the sequences
is periodically extended in order to perform circular convolution.
The circular convolution of finite duration sequences can be performed only if both the sequences
consist of the same number of samples. If the sequences have different number of samples, then convert the
smaller size sequence to the length of larger size sequence by appending zeros.
Circular convolution basically involves the same four steps as that for linear convolution, namely,
folding one sequence, shifting the folded sequence, multiplying the two sequences and finally summing the
values of the product sequence. Like linear convolution, any one of the sequence is folded and rotated in
circular convolution.
The difference between the two is that in circular convolution the folding and shifting (rotating)
operations are performed in a circular fashion by computing the index of one of the sequences by modulo-N
operation. In linear convolution there is no modulo-N operation.
2.10.4 Procedure for Evaluating Circular Convolution
Let, x1(n) and x2(n) be periodic discrete time sequences with periodicity of N-samples. If x1(n) and
x2(n) are non-periodic then convert the sequences to N-sample sequences and periodically extend the sequence
x2(n) with periodicity of N-samples.
Now the circular convolution of x1(n) and x2(n) will produce a periodic sequence x3(n) with periodicity
of N-samples. The samples of one period of x3(n) can be computed using the equation (2.57). The value of
x3(n) at n = q is obtained by replacing n by q, in equation (2.57).
N −1
.....(2.59)
∴ x3 (q ) = ∑ x1( m) x2 ((q − m)) N
m=0
The evaluation of equation (2.59) to determine the value of x3(n) at n = q involves the following five
steps.
1. Change of index : Change the index n in the sequences x1(n) and x2(n), in order to get the
sequences x1(m) and x2(m). Represent the samples of one period of the
sequences on circles.
2. Folding : Fold x2(m) about m = 0, to obtain x2(-m).
3. Rotation : Rotate x2(-m) by q times in anti-clockwise if q is positive, rotate x2(-m) by
q times in clockwise if q is negative to obtain x2((q – m))N.
4. Multiplication : Multiply x1(m) by x2((q – m))N to get a product sequence. Let the product
sequence be vq(m). Now, vq(m) = x1(m) × x2((q – m))N.
5. Summation : Sum up the samples of one period of the product sequence vq(m) to
obtain the value of x3(n) at n = q. [i.e., x3(q)].
The above procedure will give the value of x3(n) at a single time instant say n = q. In general we are
interested in evaluating the values of the sequence x3(n) in the range 0 < n < N -1. Hence the steps 3 , 4 and
5 given above must be repeated, for all possible time shifts in the range 0 < n < N - 1.
Chapter 2 - Discrete Time Signals and Systems 2. 72

2.10.5 Linear Convolution via Circular Convolution

When two numbers of N-point sequences are circularly convolved, it produces another N-point
sequence. For circular convolution, one of the sequence should be periodically extended. Also the resultant
sequence is periodic with period N.
The linear convolution of two sequences of length N1 and N2 produces an output sequence of length
N1 + N2 -1. To perform linear convolution via circular convolution both the sequences should be converted
to N 1 + N 2 - 1 point sequences by padding with zeros. Then perform circular convolution of
N1 + N2 -1 point sequences. The resultant sequence will be same as that of linear convolution of N1 and N2
point sequences.

2.10.6 Methods of Computing Circular Convolution

Method 1 : Graphical Method
In graphical method, the given sequences are converted to same size and represented on circles. In
case of periodic sequences, the samples of one period are represented on circles. One of the sequence is
folded and shifted circularly. Let x1(n) and x2(n) be the given sequences. Let x3(n) be the sequence obtained by
circular convolution of x1(n) and x2(n). The following procedure can be used to get a sample of x3(n) at n = q.
1. Change the index n in the sequences x1(n) and x2(n) to get x1(m) and x2(m) and then represent the
sequences on circles.
2. Fold one of the sequence. Let us fold x2(m) to get x2(–m).
3. Rotate (or shift) the sequence x2(–m), q times to get the sequence x2((q – m))N. If q is positive then
rotate (or shift) the sequence in anticlockwise direction and if q is negative then rotate (or shift) the
sequence in clockwise direction.
4. The sample of x3(q) at n = q is given by,
N −1 N −1
x3 ( q ) = ∑ x1 ( m) x2 (( q − m)) N = ∑ x1( m) x2,q ( m)
m=0 m= 0

where, x2, q(m) = x2((q – m))N

Determine the product sequence x1 ( m) x 2,q ( m) for one period.
5. The sum of all the samples of the product sequence gives the sample x3(q) [i.e., x3(n) at n = q].
The above procedure is repeated for all possible values of n to get the sequence x3(n).
Method 2 : Tabular Method
Let x1(n) and x2(n) be the given N-point sequences. Let x3(n) be the N-point sequence obtained by
circular convolution of x1(n) and x2(n). The following procedure can be used to obtain one sample of x3(n)
at n = q.
1. Change the index n in the sequences x1(n) and x2(n) to get x1(m) and x2(m) and then represent the
sequences as two rows of tabular array.
2. Fold one of the sequence. Let us fold x2(m) to get x2(–m).
3. Periodically extend x2(–m). Here the periodicity is N, where N is the length of the given sequences.
4. Shift the sequence x2(–m), q times to get the sequence x2((q – m))N. If q is positive then shift the
sequence to the right and if q is negative then shift the sequence to the left.
2. 73 Digital Signal Processing
N −1 N −1
5. The sample of x3(q) at n = q is given by, x3 (q) = ∑ x1 (m) x 2 ((q − m)) N = ∑ x1(m) x2,q (m)
m=0 m=0
where x2, q(m) = x2((q – m))N
Determine the product sequence x1 ( m) x 2,q ( m) for one period.
6. The sum of the samples of the product sequence gives the sample x3(q) [i.e., x3(n) at n = q].
The above procedure is repeated for all possible values of n to get the sequence x3(n).
Method 3: Matrix Method
Let x1(n) and x2(n) be the given N-point sequences. The circular convolution of x1(n) and x2(n) yields
another N-point sequence x3(n).
In this method an (N ´ N) matrix is formed using one of the sequence as shown below. Another
sequence is arranged as a column vector (column matrix) of order (N ´ 1). The product of the two matrices
gives the resultant sequence x3(n).
LMx (0) x ( N − 1)
2 2 x2 ( N − 2) ..... x 2 ( 2) x2 (1) OP LM xx ((10)) OP LM xx ((10)) OP
1 3

MMx (1) x (0)

2 2 x2 ( N − 1) ..... x 2 ( 3) x 2 ( 2) PP MM x (2) PP MM x (2) PP
1

1
3

3
MMx M(2) x (1M)
2 2 x 2 ( 0)
M
..... x 2 ( 4)
M
x (3)
2
M
PP × MM M PP = MM M PP
MMx (N − 2) x ( N − 3) x2 ( N − 4) ..... x 2 ( 0) x ( N − 1) P
MM M PP MM M PP
2 2 2
MNx (N − 1) x ( N − 2) x 2 ( N − 3) ..... x 2 (1) x (0)
PP Mx (N − 2)P M x ( N − 2)P
1 3
2 2 2 Q MNx (N − 1) PQ MN x ( N − 1) PQ
1 3

Example 2.24
Perform circular convolution of the two sequences, x1(n) = {2, 1, 2, –1} and x2(n)= {1, 2, 3, 4}
- -
Solution
Method 1:Graphical Method of Computing Circular Convolution
Let x3(n) be the sequence obtained by circular convolution of x1(n) and x2(n).
The circular convolution of x1(n) and x2(n) is given by,
N − 1 N − 1
x3(n) = ∑
m = 0
x1(m) x 2((n − m))N = ∑ x (m) x
m = 0
1 2,n (m)

where x 2,n (m) = x 2((n − m))N and m is the dummy variable used for convolution.
The index n in the given sequences are changed to m and each sequence is represented as points on a
circle as shown below. The folded sequence x2(–m) and circularly shifted sequences x2(n– m) are also represented
on the circle.
x 1 (1) = 1 x 2 (1) = 2 x 2 (3) = 4

x 1 (2) = 2 x 1 (m ) x 1 (0) = 2 x (2) = 3 x 2 (m ) x 2 (0) = 1 x 2 (2) = 3 x 2 ( −m ) x 2 (0) = 1

2

x 1 (3) = −1 x 2 (3) = 4 x 2 (1) = 2

F ig 1 . F ig 2 . F ig 3 .
4 1 2 3

x 2((0 − m )) 4 1 ⇒ x 2((1 − m )) 4 x 2((2 − m )) 4 x 2 ((3 − m )) 4

3 4 2 ⇒ 1 3 ⇒2 4
= x 2,0 (m ) = x 2,1 (m ) = x 2,2(m ) = x 2,3 (m )

2 3 4 1
F ig 4 : C ircula rly sh ifted seq u en c es x 2 ( −m ) fo r n = 0 , 1 , 2 , 3 .
Chapter 2 - Discrete Time Signals and Systems 2. 74
The given sequences are 4-point sequences . \ N = 4.
Each sample of x3(n) is given by sum of the samples of product sequence defined by the equation,
3 3
x3 (n) = ∑
m = 0
x1(m) x 2,n (m) = ∑
m = 0
vn (m) ; where vn (m) = x1(m) x 2,n (m) .....(1)

Using the above equation (1), graphical method of computing each sample of x3(n) are shown in fig 5 to fig 8.
3 3 3
When n = 0 ; x 3 (0) = ∑
m = 0
x1(m) x 2 ((0 − m))4 = ∑
m = 0
x1(m) x 2,0 (m) = ∑
m = 0
v 0 (m)

1 4 1 ×4 = 4

2 x 1 (m ) 2 X 3 x 2, 0 (m ) 1 ⇒2 × 3 = 6 v 0 (m ) 2 ×1 = 2

−1 2 −1 × 2 = −2
T he su m o f sa m p les of v 0 (m ) giv es x 3 (0)
F ig 5: C o m p u ta tio n of x 3 (0 ). ∴ x (0 ) = 2 + 4 + 6 − 2 = 1 0
3

3 3 3
When n = 1 ; x 3 (1) = ∑
m = 0
x1(m) x 2 ((1 − m))4 = ∑
m = 0
x1(m) x 2,1(m) = ∑
m = 0
v1(m)

1 1 1 ×1 = 1

x 1 (m ) x 2, 1 (m ) ⇒ v 1 (m )
2 2 X 4 2 2 ×4 = 8 2 ×2 = 4

−1 3 −1 × 3 = −3
T he su m o f sa m ples of v 1 (m ) giv es x 3 (1)
F ig 6: C o m p u ta tio n of x 3 (1 ). ∴ x (1 ) = 4 + 1 + 8 − 3 = 1 0
3

3 3 3
When n = 2 ; x 3 (2) = ∑
m = 0
x1(m) x 2 ((2 − m))4 = ∑
m = 0
x1(m) x 2,2 (m) = ∑
m = 0
v 2 (m)

1 2 1 ×2 = 2

2 x 1 (m ) 2 X 1 x 2 , 2 (m ) 3 ⇒ 2 ×1 = 2 v 2 (m ) 2 ×3 = 6

−1 4 −1 × 4 = −4
T h e s u m o f s a m p le s of v 2 (m ) giv es x 3 (2)
F ig 7 : C o m p u ta tio n of x 3 (2 ). ∴ x 3 (2) = 6 + 2 + 2 − 4 = 6
3 3 3
When n = 3 ; x 3 (3) = ∑
m = 0
x1(m) x 2 ((3 − m))4 = ∑
m = 0
x1(m) x 2,3 (m) = ∑
m = 0
v 3 (m)

1 3 1 ×3 = 3

2 x 1 (m ) 2 X 2 x 2, 3 (m ) 4 ⇒ 2 ×2 = 4 v 3 (m ) 2 ×4 = 8

−1 1 −1 × 1 = −1

F ig 8 : C o m p u ta tio n of x 3 (3 ). T h e s u m o f s a m p le s of v 3 (m ) giv es x 3 (3)

∴ x 3 (3) = 8 + 3 + 4 − 1 = 1 4
\ x3(n) = {10, 10, 6, 14}
-
2. 75 Digital Signal Processing
Method 2 : Circular Convolution Using Tabular Array

The index n in the given sequences are changed to m and then, the given sequences can be represented
in the tabular array as shown below. Here the shifted sequences x2, n(m) are periodically extended with a
periodicity of N = 4. Let x3(n) be the sequence obtained by convolution of x1(n) and x2(n). Each sample of x3(n) is
given by the equation,

N − 1 N − 1
x 3 (n) = ∑
m = 0
x1(m) x 2((n − m))N = ∑
m = 0
x1(m) x 2,n (m), where x 2,n (m) = x 2((n − m))N

Note : The boldfaced numbers are samples obtained by periodic extension.

m –3 –2 –1 0 1 2 3

x1(m) 2 1 2 –1

x2(m) 1 2 3 4

x2((–m))4 = x2,0(m) 4 3 2 1 4 3 2

x2((1– m))4 = x2,1(m) 4 3 2 1 4 3

x2((2 – m))4 = x2,2(m) 4 3 2 1 4

x2((3 – m))4 = x2,3(m) 4 3 2 1

To determine a sample of x3(n) at n = q, multiply the sequence, x1(m) and x 2,q (m), to get a product
sequence x1(m) x 2,q (m). [i.e., multiply the corresponding elements of the row x1(m) and x2, q(m)]. The sum of all
the samples of the product sequence gives x3(q).
3
When n = 0 ; x 3(0) = ∑
m =0
x1(m) x 2,0 (m)

= x1(0) x 2,0 (0) + x1(1) x 2,0 (1) + x1(2) x 2,0 (2) + x1(3) x 2,0 (3)
= 2 × 1 + 1 × 4 + 2 × 3 + (−1) × 2 = 2 + 4 + 6 − 2 = 10
The samples of x3(n) for other values of n are calculated as shown for n = 0.
3
When n = 1; x3(1) = ∑
m =0
x1(m) x 2,1(m) = 4 + 1 + 8 − 3 = 10

3
When n = 2; x 3(2) = ∑
m =0
x1(m) x 2,2(m) = 6 + 2 + 2 − 4 = 6

3
When n = 3; x3 (3) = ∑
m =0
x1(m) x 2,3 (m) = 8 + 3 + 4 − 1 = 14

l
∴ x3 (n) = 10, 10, 6, 14 q
A
Method 3 : Circular Convolution Using Matrices
The sequence x1(n) can be arranged as a column vector of order N ´ 1 and using the samples of x2(n) the
N ´ N matrix is formed as shown below. The product of the two matrices gives the sequence x3(n).

LMx (0)
2 x 2(3) x 2(2) x 2(1)OP LMx (0)OP
1 LMx (0)OP
3

MMx (1)
2 x 2 (0) x 2(3) x (2) P
2 MMx (1) PP
1
= MMx (1) PP
3
x (3) P
MMxx ((32))
2 x 2(1)
x 2(2)
x 2 (0)
x 2(1)
2
x (0)PQ
P MMxx ((32))PP
1
MMxx ((32)) PP
3

N2 2 N Q
1 N Q
3
Chapter 2 - Discrete Time Signals and Systems 2. 76

LM1 4 3 2 OP LM 2OP LM1× 2 + 4 × 1+ 3 × 2 + 2 × −1OP LM10OP

MM2 1 4 3 P MM 1PP = MM2 × 2 + 1 × 1+ 4 × 2 + 3 × −1PP = MM10PP
4P
MM34 2
3
1
2 1 PQ
P MM−21PP MMN34 ×× 22 ++ 23 ×× 1+ 1 × 2 + 4 × −1
P
1+ 2 × 2 + 1 × −1PQ
MM146PP
N N Q N Q
\ x3(n) = {10, 10, 6, 14}
- ­

Example 2.25
Perform the circular convolution of the two sequences x1(n) and x2(n), where,

l
x1(n) = 0.2, 0.4, 0.6, 0.8, 1.0, 1.2, 1.4, 1.6 q
- ­
l
x 2(n) = 0.1, 0.3, 0.5, 0.7, 0.9, 1.1, 1.3, 1.5 q
- ­
Solution
Let x3(n) be the result of the circular convolution of x1(n) and x2(n). The given sequences consists of eight
samples. Then x3(n) will also have 8 samples.

The sequences are represented in the tabular array as shown below after replacing n by m. The sequence
x2(m) is folded and shifted.

The shifted sequences x2,n(m) are periodically extended with a periodicity of N = 8.

Note : The boldfaced numbers are samples obtained by periodic extension

m –7 –6 –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7

x1(m) 0.2 0.4 0.6 0.8 1.0 1.2 1.4 1.6

x2(m) 0.1 0.3 0.5 0.7 0.9 1.1 1.3 1.5

x2((–m))8 = x2,0(m) 1.5 1.3 1.1 0.9 0.7 0.5 0.3 0.1 1.5 1.3 1.1 0.9 0.7 0.5 0.3

x2((1 – m))8 = x2,1(m) 1.5 1.3 1.1 0.9 0.7 0.5 0.3 0.1 1.5 1.3 1.1 0.9 0.7 0.5

x2((2 – m))8 = x2,2(m) 1.5 1.3 1.1 0.9 0.7 0.5 0.3 0.1 1.5 1.3 1.1 0.9 0.7

x2((3 – m))8 = x2,3(m) 1.5 1.3 1.1 0.9 0.7 0.5 0.3 0.1 1.5 1.3 1.1 0.9

x2((4 – m))8 = x2,4(m) 1.5 1.3 1.1 0.9 0.7 0.5 0.3 0.1 1.5 1.3 1.1

x2((5 – m))8 = x2,5(m) 1.5 1.3 1.1 0.9 0.7 0.5 0.3 0.1 1.5 1.3

x2((6 – m))8 = x2,6(m) 1.5 1.3 1.1 0.9 0.7 0.5 0.3 0.1 1.5

x2((7 – m))8 = x2,7(m) 1.5 1.3 1.1 0.9 0.7 0.5 0.3 0.1

Each sample of x3(n) is given by the equation,

7 7
x 3 (n) =
m =0
∑ x1(m) x 2 ((n − m))8 = ∑
m =0
x1(m) x 2,n (m) ; where x 2,n (m) = x 2 ((n − m))8

The samples of x3(0) are calculated as shown below.

2. 77 Digital Signal Processing
7 7
When n = 0 ; x 3 (n) = ∑ x (m) x
m= 0
1 2 ((0 − m))8 = ∑ x (m) x
m= 0
1 2, 0 (m)

= x1(0) x 2, 0 (0) + x1(1) x 2, 0 (1) + x1(2) x 2, 0 (2) + x1(3) x 2, 0 (3)

+ x1(4) x 2, 0 (4) + x1(5) x 2, 0 (5) + x1(6) x 2, 0 (6) + x1(7) x 2, 0 (7)
= 0.02 + 0.6 + 0.78 + 0.88 + 0.9 + 0.84 + 0.7 + 0.48 = 5.20
The samples of x3(n) for other values of n are calculated as shown for n = 0.
7 7
When n = 1; x3(1) = ∑
m=0
x1(m) x 2((1 − m))8 = ∑
m=0
x1(m) x 2,1(m) = 6.00

7 7
When n = 2; x3 (2) = ∑ x (m) x ((2 − m))
m=0
1 2 8 = ∑ x (m) x
m =0
1 2,2 (m) = 6.48

7 7
When n = 3; x3(3) = ∑ x (m) x ((3 − m))
m =0
1 2 8 = ∑ x (m) x
m =0
1 2,3 (m) = 6.64

7 7
When n = 4; x3 (4) = ∑ x (m) x ((4 − m))
m =0
1 2 8 = ∑
m=0
x1(m) x 2,4 (m) = 6.48

7 7
When n = 5; x3 (5) = ∑
m=0
x1(m) x 2((5 − m))8 = ∑
m=0
x1(m) x 2,5(m) = 6.00

7 7
When n = 6; x3(6) = ∑
m=0
x1(m) x 2 ((6 − m))8 = ∑ x (m) x
m =0
1 2,6 (m) = 5.20

7 7
When n = 7; x3(7) = ∑
m=0
x1(m) x 2 ((7 − m))8 = ∑ x (m) x
m =0
1 2,7 (m) = 4.08

lA
∴ x3(n) = 5.20, 6.00, 6.48, 6.64, 6.48, 6.00, 5.20, 4.08 q
Example 2.26
Find the linear and circular convolution of the sequences, x(n) = 1, 0.5 l q l q
and h(n) = 0.5, 1 .
­ ­ A A
Solution
Linear Convolution by Tabular Array
∞
Let , y(n) = x(n) * h(n) = ∑
m = −∞
x(m) h(n − m) ; where m is a dummy variable for convolution.

Since both x(n) and h(n) starts at n = 0, the output sequence y(n) will also start at n = 0.
Since the length of x(n) and h(n) is 2, the length of y(n) is 2 + 2 – 1 = 3.
Let us change the index n to m in x(n) and h(n). The sequences x(m) and h(m) are represented in the
tabular array as shown below.

Note : The unfilled boxes in the table are considered as zeros.

m –1 0 1 2
x(m) 1 0.5
h(m) 0.5 1
h(–m) = h0(m) 1 0.5
h(1 – m) = h1(m) 1 0.5
h(2 – m) = h2(m) 1 0.5
Chapter 2 - Discrete Time Signals and Systems 2. 78
Each sample of y(n) is given by the relation,
∞ ∞
y(n) = ∑
m = −∞
x(m) h(n − m) = ∑
m = −∞
x(m) hn (m) ; where hn (m) = h(n − m)

∞ 1
When n = 0 ; y(0) = ∑
m = −∞
x(m) h( −m) = ∑ x(m) h (m) = x(−1) h (−1) + x(0) h (0) + x(1) h (1)
m = −1
0 0 0 0

= 0 × 1 + 1 × 0.5 + 0.5 × 0 = 0 + 0.5 + 0 = 0.5

∞ 1
When n = 1 ; y(1) = ∑
m = −∞
x(m) h(1 − m) = ∑
m = 0
x(m) h1(m) = 1 + 0.25 = 125
.

∞ 2
When n = 2 ; y(2) = ∑
m = −∞
x(m) h(2 − m) = ∑
m= 0
x(m) h2(m) = 0 + 0.5 + 0 = 0.5

l
∴ y(n) = 0.5, 1.25, 0.5 q
A
Circular Convolution by Tabular Array
N− 1
Let, y(n) = x(n) ∗ h(n) =
m=0
∑ x(m) h((n − m)) N ; where m is a dummy variable for convolution.

The index n in the sequences are changed to m and the sequences are represented in the tabular array as
shown below. The shifted sequence hn(m) is periodically extended with periodicity N = 2.
Note : The boldfaced number is the sample obtained by periodic extension.

m –1 0 1
x(m) 1 0.5
h(m) 0.5 1
h((–m))2 = h0(m) 1 0.5 1
h((1 – m))2 = h1(m) 1 0.5

Each sample of y(n) is given by the equation,

N − 1 N − 1
y(n) = ∑
m = 0
x(m) h((n − m))N = ∑
m = 0
x(m) hn (m); where hn (m) = h((n − m))N

N − 1 1
When n = 0 ; y(0) = ∑
m= 0
x(m) h((0 − m))2 = ∑ x(m) h (m)
m = 0
0

= x(0) h0 (0) + x(1) h0 (1) = 1 × 0.5 + 0.5 × 1 = 0.5 + 0.5 = 10

.
N − 1 1
When n = 1 ; y(1) = ∑ x(m) h((1 − m))2 = ∑ x(m) h1(m)
m= 0 m = 0
= x(0) h1(0) + x(1) h1(1) = 1 × 1 + 0.5 × 0.5 = 1 + 0.25 = 125
.
∴ y(n) = 10l
. , 1.25 q
A
Example 2.27
The input x(n) and impulse response h(n) of a LTI system are given by,
x(n) = {–1, 1, 2, –2) ; h(n) = {0.5, 1, –1, 2, 0.75}
A A ­
Determine the response of the system a) using linear convolution and b) using circular convolution.
2. 79 Digital Signal Processing
Solution
a) Response of LTI system using linear convolution
Let y(n) be the response of LTI system. By convolution sum formula,
+∞
y(n) = x(n) ∗ h(n) = ∑
m = −∞
x(m) h(n − m) ; where m is a dummy variable used for convolution.

The sequence x(n) starts at n = 0 and h(n) starts at n = –1. Hence y(n) will start at n = 0 + (–1) = –1.
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 index n to m in x(n) and h(n). The sequences x(m) and h(m) are represented on the
tabular array as shown below. Let us fold h(m) to get h(–m) and shift h(–m) to perform convolution operation.
Note : The unfilled boxes in the table are considered as zeros.
m –4 –3 –2 –1 0 1 2 3 4 5 6 7
x(m) –1 1 2 –2
h(m) 0.5 1 –1 2 0.75
h(–m) 0.75 2 –1 1 0.5
h(–1 – m) = h–1(m) 0.75 2 –1 1 0.5
h(0 – m) = h0(m) 0.75 2 –1 1 0.5
h(1 – m) = h1(m) 0.75 2 –1 1 0.5
h(2 – m) = h2(m) 0.75 2 –1 1 0.5
h(3 – m) = h3(m) 0.75 2 –1 1 0.5
h(4 – m) = h4(m) 0.75 2 –1 1 0.5
h(5 – m) = h5(m) 0.75 2 –1 1 0.5
h(6 – m) = h6(m) 0.75 2 –1 1 0.5

Each sample of y(n) is given by summation of the product sequence, x(m) h(n – m). To determine a
sample of y(n) at n = q, multiply the sequence x(m) and hq(m) to get a product sequence [i.e., multiply the
corresponding elements of the row x(m) and hq(m)]. The sum of all the samples of the product sequence gives
y(q).
+∞ +∞
i. e. , y(n) = ∑ x(m) h(n − m) = ∑ x(m) h (m)
m = −∞ m = −∞
n

3
When n = −1 ; y(−1) = ∑
m = −4
x(m) h−1(m)

= x(–4) h–1(–4) + x(–3) h–1(–3) + x(–2) h–1(–2) + x(–1) h–1(–1) + x(0) h–1(0)
+ x(1) h–1(1) + x(2) h–1(2) + x(3) h–1(3)
= 0 + 0 + 0 + 0 + (–0.5) + 0 + 0 + 0 = –0.5

The samples of y(n) for other values of n are calculated as shown for n = –1.
3
When n = 0 ; y(0) = ∑
m = −3
x(m) h0 (m) = 0 + 0 + 0 + (−1) + 0.5 + 0 + 0 = −0.5

3
When n = 1 ; y(1) = ∑
m = −2
x(m) h1(m) = 0 + 0 + 1+ 1+ 1+ 0 = 3

3
When n = 2 ; y(2) = ∑
m = −1
x(m) h2 (m) = 0 + (−2) + ( −1) + 2 + ( −1) = −2
Chapter 2 - Discrete Time Signals and Systems 2. 80
4
When n = 3 ; y(3) = ∑
m= 0
x(m) h3(m) = −0.75 + 2 + (−2) + (−2) + 0 = −2.75

5
When n = 4 ; y(4) = ∑
m= 0
x(m) h4 (m) = 0 + 0.75 + 4 + 2 + 0 + 0 = 6.75

6
When n = 5 ; y(5) = ∑
m= 0
x(m) h5(m) = 0 + 0 + 1.5 + (−4) + 0 + 0 + 0 = −2. 5

7
When n = 6 ; y(6) = ∑
m= 0
x(m) h6 (m) = 0 + 0 + 0 + (−1.5) + 0 + 0 + 0 + 0 = −1.5

The response of LTI system y(n) is,

y(n) = {–0.5, –0.5, 3, –2, –2.75, 6.75, –2.5, –1.5}
- ­
b) Response of LTI System Using Circular Convolution
The response of LTI system is given by linear convolution of x(n) and h(n). Let y(n) be the response
sequence of LTI system. To get the result of linear convolution from circular convolution, both the sequences
should be converted to the size of y(n) and perform circular convolution of the converted sequences. Also the
converted sequences should start and end at the same value of n as that of y(n).
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. Therefore both
the sequences should be converted to 8-point sequences.
The x(n) starts at n = 0 and h(n) starts at n = –1. Hence y(n) will start at n = 0 + (–1) = –1. The y(n) will
end at n = [0 +(–1)] + (4 + 5 – 2) = 6. Therefore the converted sequences should start at n = –1 and end at n = 6.
\ x(n) = {0, –1, 1, 2, –2, 0, 0, 0} and h(n) = {0.5, 1, –1, 2, 0.75, 0, 0, 0}
- -
The converted sequences x(n) and h(n) are represented on the tabular array after replacing the index n by
m as shown below. The sequence h(m) is folded and shifted.
The shifted sequences hn(m) are periodically extended with a periodicity of N = 8.

Note : The boldfaced numbers are samples obtained by periodic extension of the sequences.
m –7 –6 –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7
x(m) 0 –1 1 2 –2 0 0 0
h(m) 0.5 1 –1 2 0.75 0 0 0
h(–m) 0 0 0 0.75 2 –1 1 0.5
h((–1 – m))8 = h–1(m) 0 0 0 0.75 2 –1 1 0.5 0 0 0 0.75 2 –1 1
h((0 – m))8 = h0(m) 0 0 0 0.75 2 –1 1 0.5 0 0 0 0.75 2 –1
h((1 – m))8 = h1(m) 0 0 0 0.75 2 –1 1 0.5 0 0 0 0.75 2
h((2 – m))8 = h2(m) 0 0 0 0.75 2 –1 1 0.5 0 0 0 0.75
h((3 – m))8 = h3(m) 0 0 0 0.75 2 –1 1 0.5 0 0 0
h((4 – m))8 = h4(m) 0 0 0 0.75 2 –1 1 0.5 0 0
h((5 – m))8 = h5(m) 0 0 0 0.75 2 –1 1 0.5 0
h((6 – m))8 = h6(m) 0 0 0.75 2 –1 1 0.5 0 0 0 0.75 2 –1 1 0.5

Let y(n) be the sequence obtained by circular convolution of x(n) and h(n).
Now, each sample of y(n) is given by,
6 6
y(n) = ∑
m = −1
x(m) h((n − m))8 = ∑
m = −1
x(m) hn (m) ; where hn (m) = h((n − m))8
2. 81 Digital Signal Processing
To determine a sample of y(n) at n = q, multiply the sequence x(m) and hq(m) to get a product sequence
x(m) hq(m), [i.e., multiply the corresponding elements of the row x(m) and hq(m)]. The sum of all the samples of the
product sequence gives y(q).
6
When n = −1 ; y(−1) = ∑
m = −1
x(m) h−1(n) = x( −1) h−1( −1) + x(0) h−1(0) + x(1) h−1(1) + x(2) h−1(2)

+ x(3) h−1(3) + x(4) h−1(4) + x(5) h−1(5) + x(6) h−1(6)

= 0 + (−0.5) + 0 + 0 + 0 + 0 + 0 + 0 = −0.5

The samples of y(n) for other values of n are calculated as shown for n = –1.
6
When n = 0 ; y(0) = ∑ x(m) h0m = 0 + (−1) + 0.5 + 0 + 0 + 0 + 0 + 0 = −0.5
m = −1
6
When n = 1 ; y(1) = ∑ x(m) h1m = 0 + 1+ 1+ 1+ 0 + 0 + 0 + 0 = 3
m = −1
6
When n = 2 ; y(2) = ∑ x(m) h2m = 0 + (−2) + (−1) + 2 + (−1) + 0 + 0 + 0 = −2
m = −1
6
When n = 3 ; y(3) = ∑ x(m) h3m = 0 + (−0.75) + 2 + (−2) + (−2) + 0 + 0 + 0 = −2.75
m = −1
6
When n = 4 ; y(4) = ∑ x(m) h4m = 0 + 0 + 0.75 + 4 + 2 + 0 + 0 + 0 = 6.75
m = −1
6
When n = 5 ; y(5) = ∑ x(m) h5m = 0 + 0 + 0 + 1.5 + (−4) + 0 + 0 + 0 = −2.5
m = −1
6
When n = 6 ; y(6) = ∑ x(m) h6m = 0 + 0 + 0 + 0 + (−1. 5) + 0 + 0 + 0 = −1.5
m = −1

The response of LTI system y(n) is,

y(n) = {–0.5, –0.5, 3, –2, –2.75, 6.75, –2.5, –1.5}
-

Note : 1. Since circular convolution is periodic, the convolution is performed for any one period.
2. It can be observed that the results of both the methods are same.

2.11 Sectioned Convolution

The response of an LTI system for any arbitrary input is given by linear convolution of the input and
the impulse response of the system. If one of the sequences (either the input sequence or impulse response
sequence) is very much larger than the other, then it is very difficult to compute the linear convolution for the
following reasons.

1. The entire sequence should be available before convolution can be carried out. This makes long
delay in getting the output.

2. Large amounts of memory is required to store the sequences.

The above problems can be overcome in the sectioned convolutions. In this technique the larger
sequence is sectioned (or splitted) into the size of smaller sequence. Then the linear convolution of each
section of longer sequence and the smaller sequence is performed. The output sequences obtained from the
convolutions of all the sections are combined to get the overall output sequence. There are two methods of
sectioned convolutions. They are overlap add method and overlap save method.
Chapter 2 - Discrete Time Signals and Systems 2. 82
2.11.1 Overlap Add Method
In the overlap add method, the longer sequence is divided into smaller sequences. Then linear
convolution of each section of longer sequence and smaller sequence is performed. The overall output
sequence is obtained by combining the output of the sectioned convolution.
Let, N1 = Length of longer sequence
N2 = Length of smaller sequence
Let the longer sequence be divided into sections of size N3 samples.
Note : Normally the longer sequence is divided into sections of size same as that of smaller sequence.

N3 + N2 −1

N3 N2 −1
N3 + N2 −1

N2 −1 N 3 − ( N 2 − 1) N2 −1
N3 + N2 − 1

N2 − 1 N 3 − ( N 2 − 1) N2 −1
O v erlapped region

Note : Samples in the shaded region are added. O v erlapped region

F ig 2 .3 0 : O ve rla p p in g of o u tp ut seq u en c e o f sec tio n ed c o n vo lution b y o v erla p a d d m eth o d .

The linear convolution of each section with smaller sequence will produce an output sequence of size
N3 + N2 –1 samples. In this method the last N2 –1 samples of each output sequence overlaps with the first
N2 –1 samples of the next section. [i.e., there will be a region of N2 –1 samples over which the output sequence
of qth convolution overlaps the output sequence of (q +1)th convolution]. While combining the output
sequences of the various sectioned convolutions, the corresponding samples of overlapped regions are
added and the samples of non-overlapped regions are retained as such.

2.11.2 Overlap Save Method

In the overlap save method, the results of linear convolution of the various sections are obtained
using circular convolution. In this method, the longer sequence is divided into smaller sequences. Each
section of the longer sequence and the smaller sequence are converted to the size of the output sequence of
sectioned convolution. The circular convolution of each section of the longer sequence and the smaller
sequence is performed. The overall output sequence is obtained by combining the outputs of the sectioned
convolution.
Let, N1 = Length of longer sequence
N2 = Length of smaller sequence
Let the longer sequence be divided into sections of size N3 samples.
Note : Normally the longer sequence is divided into sections of size same as that of smaller sequence.
In the overlap save method, the results of linear convolution are obtained by circular convolution.
Hence each section of longer sequence and the smaller sequence are converted to the size of output sequence
of size N3 + N2 – 1 samples.The smaller sequence is converted to size of N3 + N2 –1 samples, by appending with
zeros.The convertion of each section of longer sequence to the size N3 + N2 –1 samples can be performed in
two different methods.
2. 83 Digital Signal Processing
Method-1
In this method, the first N2 –1 samples of a section is appended as last N2 –1 samples of the previous
section (i.e., the overlapping samples are placed at the beginning of the section). The circular convolution of
each section will produce an output sequence of size N3 + N2 –1 samples. In this output the first N2 –1 samples
are discarded and the remaining samples of the output of sectioned convolutions are saved as the overall
output sequence.
N3 + N2 −1
N3 N2 −1

N3 + N2 −1

N2 −1 N 3 − ( N 2 − 1) N2 −1

N3 + N2 −1

N2 −1 N 3 − ( N 2 − 1)
A ppended
w ith zero
F ig 2 .3 1 : A p p en d ing o f sec tio n s o f in pu t seq u en c e
in m eth o d 1 o f o ve rlap sa ve m e th o d .
N3 + N2 −1

N2 −1 N 3 − ( N 2 − 1) N2 −1

N3 + N2 −1

N2 −1 N 3 − ( N 2 − 1) N2 −1

N3 + N2 − 1
N2 −1 N3
O v erlapped region

Note : Samples in the shaded region are discarded.

O v erlapped region

F ig 2 .3 2 : O ve rla p p in g of o u tp ut seq u en c e of sec tio n ed c o n vo lution

b y m eth o d 1 o f o ve rla p sa ve m ethod .
Method-2
In this method, the last N2–1 samples of a section is appended as last N2 –1 samples of the next section
(i.e, the overlapping samples are placed at the end of the sections). The circular convolution of each section
will produce an output sequence of size N3 + N2 –1 samples. In this output the last N2 –1 samples are discarded
and the remaining samples of the output of sectioned convolutions are saved as the overall output sequence.

N3 + N2 −1
N 3 −( N 2 − 1) N2 −1 N2 −1
A ppended
w ith zero
N3 + N2 −1

N 3 −( N 2 − 1) N2 −1 N2 −1

N3 + N2 −1

N3 N2 −1

F ig 2 .3 3 : A p p en d ing o f sec tio n s o f in pu t seq u en c e

in m eth o d 2 o f o ve rlap sa ve m eth o d .
Chapter 2 - Discrete Time Signals and Systems 2. 84

N3 + N2 −1

N3 N2 −1
N3 + N2 −1

N2 −1 N 3 − ( N 2 − 1) N2 −1

N3 + N2 −1
N2 − 1 N 3 −( N 2 − 1 ) N2 −1
O v erlapped region

Note : Samples in the shaded region are discarded. O v erlapped region

F ig 2 .3 4 : O ve rla p p ing of o utp ut seq u en c e o f sec tio n ed co n vo lu tio n

b y m eth o d 2 o f o verla p sa v e m eth od .

Example 2.28
Perform the linear convolution of the following sequences by a) Overlap add method, and b) Overlap
save method.

x(n) = {1, –1, 2, –2, 3, –3, 4, –4} ; h(n) = {–1, 1}

Solution
a) Overlap Add Method

In this method the longer sequence is sectioned into sequences of size equal to smaller sequence. Here
x(n) is a longer sequence when compared to h(n). Hence x(n) is sectioned into sequences of size equal to h(n).

Given that, x(n) = {1, –1, 2, –2, 3, –3, 4, –4}

Let x(n) can be sectioned into four sequences, each consisting of two samples of x(n) as shown below.
x1(n) = 1 ; n = 0 x2(n) = 2 ; n = 2 x3(n) = 3 ; n=4 x4(n) = 4 ; n=6
= –1 ; n = 1 = –2 ; n = 3 = –3 ; n=5 = –4 ; n = 7
Let y1(n), y2(n), y3(n) and y4(n) be the output of linear convolution of x1(n), x2(n), x3(n) and x4(n) with h(n)
respectively.
Here h(n) starts at n = nh = 0
x1(n) starts at n = n1 = 0, \ y1(n) will start at n = n1 + nh = 0 + 0 = 0
x2(n) starts at n = n2 = 2, \ y2(n) will start at n = n2 + nh = 2 + 0 = 2
x3(n) starts at n = n3 = 4, \ y3(n) will start at n = n3 + nh = 4 + 0 = 4
x4(n) starts at n = n4 = 6, \ y4(n) will start at n = n4 + nh = 6 + 0 = 6

Here linear convolution of each section is performed between two sequences each consisting of 2
samples. Hence each convolution output will consists of 2 + 2 – 1 = 3 samples. The convolution of each section
is performed by tabular method as shown below.

Note :
1. Here N1 = 8, N2 = 2, N3 = 2. \ (N2 – 1) = 2 – 1 = 1 and (N2 + N3 – 1) = 2 + 2 – 1 = 3
2. The unfilled boxes in the tables are considered as zero.
3. For convenience of convolution operation the index n is replaced by m in x1(n), x2(n), x3(n), x4(n) and h(n).
2. 85 Digital Signal Processing
Convolution of Section 1
+∞

m –1 0 1 2
y1(n) = x1(n) ∗ h(n) = ∑ x (m) h(n − m)
m = −∞
1

+∞
x1(m) 1 –1
= ∑ x (m) h (m) ;
1 n n = 0, 1, 2
h(m) –1 1 m = −∞
where hn (m) = h(n − m)
h(–m) = ho(m) 1 –1
h(1 – m) = h1(m) 1 –1
When n = 0 ; y1(0) = ∑ x1(m ) h0 (m) = 0 − 1+ 0 = − 1

h(2 – m) = h2(m) 1 –1
When n = 1 ; y1(1) = ∑ x (m) h (m) = 1+ 1 = 2 1 1

When n = 2 ; y (2) = ∑ x (m) h (m) = 0 − 1+ 0 = −1

1 1 2
Convolution of Section 2
+∞

m –1 0 1 2 3 4 y 2 (n) = x 2(n) ∗ h(n) = ∑

m = −∞
x 2(m) h(n − m)

x2(m) 2 –2 +∞

h(m) –1 1
= ∑ x (m) h (m)
m = −∞
2 n ; n = 2, 3, 4

where hn (m) = h(n – m)

h(–m) 1 –1
h(2 – m) = h2(m) 1 –1 When n = 2 ; y 2(2) =∑ x (m) h (m) = 0 − 2 + 0 = −2
2 2

h(3 – m) = h3(m) 1 –1 When n = 3 ; y (3) = ∑ x (m) h (m) = 2 + 2

2 = 4
2 3

h(4 – m) = h4(m) 1 –1 When n = 4 ; y (4) = ∑ x (m) h (m) = 0 − 2 + 0 = −2

2 2 4

Convolution of Section 3

m –1 0 1 2 3 4 5 6
x3(m) 3 –3
h(m) –1 1
h(–m) 1 –1
h(4 – m) = h4(m) 1 –1
h(5 – m) = h5(m) 1 –1
h(6 – m) = h6(m) 1 –1
+∞ +∞
y3(n) = x3(n) ∗ h(n) = ∑ x (m) h(n − m) = ∑ x (m) h (m)
m = −∞
3
m = −∞
3 n ; n = 4, 5, 6

where hn (m) = h(n – m)

When n = 4 ; y3(4) = ∑ x (m) h (m) = 0 − 3 + 0 = −3
3 4

When n = 5 ; y (5) = ∑ x (m) h (m) = 3 + 3

3 3 = 6 5

When n = 6 ; y (6) = ∑ x (m) h (m) = 0 − 3 + 0 = −3

3 3 6

Convolution of Section 4
m –1 0 1 2 3 4 5 6 7 8
x4(m) 4 –4
h(m) –1 1
h(–m) 1 –1
h(6 – m) = h6(m) 1 –1
h(7 – m) = h7(m) 1 –1
h(8 – m) = h8(m) 1 –1
Chapter 2 - Discrete Time Signals and Systems 2. 86
+∞ +∞
y 4 (n) = x 4 (n) ∗ h(n) = ∑x
m = −∞
4 (m) h(n − m) = ∑ x (m) h (m)
m = −∞
4 n ; n = 6, 7, 8

where hn (m) = h(n − m)

∑ x (m) h (m) = 0 − 4 + 0 = −4
When n = 6 ; y 4 (6) = 4 6

When n = 7 ; y (7) = ∑ x (m) h (m) = 4 + 4

4 4 7 = 8
When n = 8 ; y (8) = ∑ x (m) h (m) = 0 − 4 + 0 = −4
4 4 8

To Combine the Output of Convolution of Each Section

It can be observed that the last sample in an output sequence overlaps with the first sample of next output
sequence. In this method the overall output is obtained by combining the outputs of the convolution of all
sections. The overlapped portions (or samples) are added while combining the output. The output of all sections
can be represented in a table as shown below. Then the samples corresponding to same value of n are added to
get the overall output.
n 0 1 2 3 4 5 6 7 8
y1(n) –1 2 –1
y2(n) –2 4 –2
y3(n) –3 6 –3
y4(n) –4 8 –4
y(n) –1 2 –3 4 –5 6 –7 8 –4

\ y(n) = x(n) * h(n) = {–1, 2, –3, 4, –5, 6, –7, 8, –4}

b) Overlap Save Method

In this method, the longer sequence is sectioned into sequences of size equal to smaller sequence. The
number of samples that will be obtained in the output of linear convolution of each section is determined. Then
each section of longer sequence is converted to the size of output sequence using the samples of original longer
sequence. The smaller sequence is also converted to the size of output sequence by appending with zeros. Then
the circular convolution of each section is performed.
Here x(n) is a longer sequence when compared to h(n). Hence x(n) is sectioned into sequences of size
equal to h(n). Given that, x(n) = {1, –1, 2, –2, 3, –3, 4, –4}
Let x(n) be sectioned into four sequences, each consisting of two samples of x(n) as shown below.
x1(n) = 1 ; n = 0 x2(n) = 2 ; n = 2 x3(n) = 3 ; n=4 x4(n) = 4 ; n=6
= –1 ; n = 1 = –2 ; n = 3 = –3 ; n=5 = –4 ; n = 7
Let y1(n), y2(n), y3(n) and y4(n) be the output of linear convolution of x1(n), x2(n), x3(n) and x4(n) with h(n)
respectively. Here linear convolution of each section will result in an output sequence consisting of
2 + 2 – 1 = 3 samples.
The sequence h(n) is converted to 3-sample sequence by appending with zero. \ h(n) = {–1, 1, 0}
Method - 1
In method 1, the overlapping samples are placed at the beginning of the sections. Each section of longer
sequence is converted to 3-sample sequences, using the samples of original longer sequence as shown below.
It can be observed that the first sample of x2(n) is placed as overlapping sample at the end of x1(n). The first sample
of x3(n) is placed as overlapping sample at the end of x2(n). The first sample of x4(n) is placed as overlapping
sample at the end of x3(n). Since there is no fifth section, the overlapping sample of x4(n) is taken as zero.
x1(n) = 1 ; n = 0 x2(n) = 2 ; n = 2 x3(n) = 3 ; n = 4 x4(n) = 4 ; n = 6
= –1 ; n = 1 = –2 ; n = 3 = –3 ; n = 5 = –4 ; n = 7
= 2; n=2 = 3 ; n=4 = 4; n=6 = 0 ;n=8
2. 87 Digital Signal Processing
Now perform circular convolution of each section with h(n). The output sequence obtained from circular
convolution will have three samples. The circular convolution of each section is performed by tabular method
as shown below.
Here h(n) starts at n = nh = 0
x1(n) starts at n = n1 = 0, \ y1(n) will start at n = n1 + nh = 0 + 0 = 0
x2(n) starts at n = n2 = 2, \ y2(n) will start at n = n2 + nh = 2 + 0 = 2
x3(n) starts at n = n3 = 4, \ y3(n) will start at n = n3 + nh = 4 + 0 = 4
x4(n) starts at n = n4 = 6, \ y4(n) will start at n = n4 + nh = 6 + 0 = 6

Note : 1. Here N1 = 8, N2 = 2, N3 = 2. \ (N2 – 1) = 2 – 1 = 1 and (N2 + N3 – 1) = 2 + 2 – 1 = 3

2. The boldfaced numbers in the tables are obtained by periodic extension.
3. For convenience of convolution operation, the index n in x1(n), x2(n), x3(n), x4(n) and h(n)
are replaced by m.
mf
Convolution of Section 1 y1(n) = x1(n) ∗ h(n) = ∑
m = mi
x1(m) h((n − m))N
m –2 –1 0 1 2 2

x1(m) 1 –1 2 = ∑ x (m) h (m) ;

m = 0
1 n n = 0, 1, 2,

h(m) –1 1 0 where hn (m) = h((n − m))N

h((–m))3 = h0(m) 0 1 –1 0 1 When n = 0 ; y1(0) = ∑ x (m) h (m) = −1 + 0 + 2 = 1
1 0

h((1 – m))3 = h1(m) 0 1 –1 0 When n = 1 ; y (1) = ∑ x (m) h (m) = 1+ 1+ 0 = 2

1 1 1

h((2 – m))3 = h2(m) 0 1 –1 When n = 2 ; y (2) = ∑ x (m) h (m) = 0 − 1 − 2 = −3

1 1 2

Convolution of Section 2

m –2 –1 0 1 2 3 4
x2(m) 2 –2 3
h(m) –1 1 0
h(–m) 0 1 –1
h((2 – m))3 = h2(m) 0 1 –1 0 1
h((3 – m))3 = h3(m) 0 1 –1 0
h((4 – m))3 = h4(m) 0 1 –1

mf 4
y 2(n) = x 2(n) ∗ h(n) = ∑
m = mi
x 2(m) h((n − m))N = ∑ x (m) h (m) ;
m = 2
2 n n = 2, 3, 4

where hn (m) = h((n − m))N

When n = 2 ; y 2(2) =∑ x (m) h (m) = −2 + 0 + 3 = 1
2 2

When n = 3 ; y (3) = ∑ x (m) h (m) = 2 + 2 + 0 = 4

2 2 3

When n = 4 ; y (4) = ∑ x (m) h (m) = 0 + −2 − 3 = −5

2 2 4

Convolution of Section 3

mf 6
y 3(n) = x 3 (n) ∗ h(n) = ∑ x (m) h((n − m))
m = mi
3 N = ∑ x (m) h (m) ;
m = 4
3 n n = 4, 5, 6

where hn (m) = h((n − m))N

Chapter 2 - Discrete Time Signals and Systems 2. 88
m –2 –1 0 1 2 3 4 5 6
x3(m) 3 –3 4
h(m) –1 1 0
h(–m) 0 1 –1
h((4 – m))3 = h4(m) 0 1 –1 0 1
h((5 – m))3 = h5(m) 0 1 –1 0
h((6 – m))3 = h6(m) 0 1 –1

When n = 4 ; y3(4) = ∑ x (m) h (m) = −3 + 0 + 4 = 1

3 4

When n = 5 ; y (5) = ∑ x (m) h (m) = 3 + 3 + 0 = 6

3 3 5

When n = 6 ; y (6) = ∑ x (m) h (m) = 0 − 3 − 4 = −7

3 3 6

Convolution of section 4

m –2 –1 0 1 2 3 4 5 6 7 8
x4(m) 4 –4 0
h(m) –1 1 0
h(–m) 0 1 –1
h((6 – m))3 = h6(m) 0 1 –1 0 1
h((7 – m))3 = h7(m) 0 1 –1 0
h((8 – m))3 = h8(m) 0 1 –1

mf 8
y 4 (n) = x 4 (n) ∗ h(n) = ∑ x (m) h((n − m))
m = mi
4 N = ∑ x (m) h (m) ; n = 6, 7, 8
m = 6
4 n

where hn (m) = h((n – m))N

∑ x (m) h (m) = −4 + 0 + 0 = −4
When n = 6 ; y 4 (6) = 4 6

When n = 7 ; y (7) = ∑ x (m) h (m) = 4 + 4 + 0 = 8

4 4 7

When n = 8 ; y (8) = ∑ x (m) h (m) = 0 − 4 + 0 = −4

4 4 8

To Combine the Output of the Convolution of Each Section

It can be observed that the last sample in an output sequence overlaps with the first sample of next output
sequence. In overlap save method the overall output is obtained by combining the outputs of the convolution
of all sections. While combining the outputs, the overlapped first sample of every output sequence is discarded
and the remaining samples are simply saved as samples of y(n) as shown in the following table.
n 0 1 2 3 4 5 6 7 8
y1(n) 1 2 –3
y2(n) 1 4 –5
y3(n) 1 6 –7
y4(n) –4 8 –4
y(n) * 2 –3 4 –5 6 –7 8 –4

y(n) = x(n) * h(n) = {*, 2, –3, –4, –5, 6, –7, 8, –4}

Note : Here y(n) is linear convolution of x(n) and h(n). It can be observed that the results of both the methods
are same, except the first N2 – 1 samples.
2. 89 Digital Signal Processing
Method 2
In method 2, the overlapping samples are placed at the end of the section.Each section of longer
sequence is converted to 3-sample sequence, using the samples of original longer sequence as shown below.
It can be observed that the last sample of x1(n) is placed as overlapping sample at the end of x2(n). The last
sample of x2(n) is placed as overlapping sample at the end of x3(n). The last sample of x3(n) is placed as
overlapping sample at the end of x4(n). Since there is no previous section for x1(n), the overlapping sample of
x1(n) is taken as zero.

x1(n) = 1 ; n = 0 x2(n) = 2 ; n = 2 x3(n) = 3 ; n=4 x4(n) = 4 ; n = 6

= –1 ; n = 1 = –2 ; n = 3 = –3 ; n=5 = –4 ; n = 7

= 0; n=2 = –1 ; n = 4 = –2 ; n = 6 = –3 ; n = 8

Now perform circular convolution of each section with h(n). The output sequence obtained from circular
convolution will have three samples. The circular convolution of each section is performed by tabular method as
shown below.

Here h(n) starts at n = nh = 0

x1(n) starts at n = n1 = 0, \ y1(n) will start at n = n1 + nh = 0 + 0 = 0

x2(n) starts at n = n2 = 2, \ y2(n) will start at n = n2 + nh = 2 + 0 = 2

x3(n) starts at n = n3 = 4, \ y3(n) will start at n = n3 + nh = 4 + 0 = 4

x4(n) starts at n = n4 = 6, \ y4(n) will start at n = n4 + nh = 6 + 0 = 6

Note : 1. Here N1 = 8, N2 = 2, N3 = 2. \ (N2 – 1) = 2 – 1 = 1 and (N2 + N3 – 1) = 2 + 2 – 1 = 3

2. The boldfaced numbers in the tables are obtained by periodic extension.
3. For convenience of convolution the index n is replaced by m in x1(n), x2(n), x3 (n), x4 (n) and h(n).
mf
Convolution of Section 1
y1(n) = x1(n) ∗ h(n) = ∑
m = mi
x1(m) h((n − m))N
m –2 –1 0 1 2
2
x1(m) 1 –1 0 = ∑ x (m) h (m) ;
m = 0
1 n n = 0, 1, 2
h(m) –1 1 0
where hn (m) = h((n − m))N
h((–m))3 = h0(m) 0 1 –1 0 1
When n = 0 ; y1(0) = ∑ x (m) h (m) = −1 + 0 + 0 = –1
1 0
h((1 – m))3 = h1(m) 0 1 –1 0
When n = 1 ; y (1) = ∑ x (m) h (m) = 1+ 1 + 0 = 2
1 1 1
h((2 – m))3 = h2(m) 0 1 –1
When n = 2 ; y (2) = ∑ x (m) h (m) = 0 − 1 + 0 = −1
1 1 2
Convolution of Section 2

m –2 –1 0 1 2 3 4
x2(m) 2 –2 –1
h(m) –1 1 0
h(–m) 0 1 –1
h((2 – m))3 = h2(m) 0 1 –1 0 1
h((3 – m))3 = h3(m) 0 1 –1 0
h((4 – m))3 = h4(m) 0 1 –1
Chapter 2 - Discrete Time Signals and Systems 2. 90
mf 4
y 2 (n) = x 2(n) ∗ h(n) = ∑ x (m) h((n − m))
m = mi
2 N = ∑ x (m) h (m); n = 2,
m = 2
2 n 3, 4,

where hn (m) = h((n − m))N

When n = 2 ; y 2(2) =∑ x 2 (m) h2(m) = −2 + 0 − 1 = – 3
When n = 3 ; y (3) = ∑ x (m) h (m) =
2 2 3 2+ 2+ 0 = 4
When n = 4 ; y (4) = ∑ x (m) h (m) =
2 2 4 0 − 2 + 1 = −1

Convolution of Section 3

m –2 –1 0 1 2 3 4 5 6
x3(m) 3 –3 –2
h(m) –1 1 0
h(–m) 0 1 –1
h((4 – m))3 = h4(m) 0 1 –1 0 1
h((5 – m))3 = h5(m) 0 1 –1 0
h((6 – m))3 = h6(m) 0 1 –1

mf 6
y3(n) = x3 (n) ∗ h(n) = ∑ x (m) h((n − m))
m = mi
3 N = ∑ x (m) h (m) ;
m = 4
3 n n = 4, 5, 6

where hn (m) = h((n − m))N

When n = 4 ; y3(4) = ∑ x3(m) h4 (m) = −3 + 0 − 2 = –5
When n = 5 ; y (5) = ∑ x (m) h (m) =
3 3 5 3 +3+ 0 = 6
When n = 6 ; y (6) = ∑ x (m) h (m) =
3 3 6 0 − 3 + 2 = −1

Convolution of Section 4

m –2 –1 0 1 2 3 4 5 6 7 8
x4(m) 4 –4 –3
h(m) –1 1 0
h(–m) 0 1 –1
h((6 – m))3 = h6(m) 0 1 –1 0 1
h((7 – m))3 = h7(m) 0 1 –1 0
h((8 – m))3 = h8(m) 0 1 –1

mf 8
y 4 (n) = x 4 (n) ∗ h(n) = ∑ x (m) h((n − m))
m = mi
4 N = ∑ x (m) h (m) ; n = 6, 7, 8
m = 6
4 n

where hn (m) = h((n – m))N

∑
When n = 6 ; y 4 (6) = x 4 (m) h6 (m) = −4 + 0 − 3 = −7
When n = 7 ; y (7) = ∑ x (m) h (m) =
4 4 7 4+4+0 = 8
When n = 8 ; y (8) = ∑ x (m) h (m) =
4 4 8 0 − 4 + 3 = −1
2. 91 Digital Signal Processing
To Combine the Output of the Convolution of Each Section
It can be observed that the last sample in an output sequence overlaps with the first sample of next output
sequence. In overlap save method the overall output is obtained by combining the outputs of the convolution of
all sections. While combining the outputs, the overlapped last sample of every output sequence is discarded and
the remaining samples are simply saved as samples of y(n) as shown in the following table.

n 0 1 2 3 4 5 6 7 8
Note :
y1(n) –1 2 –1
Here y(n) is linear convolution
y2(n) –3 4 –1 of x(n) and h(n). It can be
y3(n) –5 6 –1 observed that the results of both
the methods are same except the
y4(n) –7 8 –1 last N2–1 samples.
y(n) –1 2 –3 4 –5 6 –7 8 *

\ y(n) = x(n) * h(n) = {–1, 2, –3, 4, –5, 6, –7, 8, *}

Example 2.29
Perform the linear convolution of the following sequences by a) Overlap add method and b) Overlap
save method.

x(n) = {1, 2, 3, –1, –2, –3, 4, 5, 6} and h(n) = {2, 1, –1}

Solution
a) Overlap Add Method

In this method the longer sequence is sectioned into sequences of size equal to smaller sequence. Here
x(n) is a longer sequence when compared to h(n). Hence x(n) is sectioned into sequences of size equal to h(n).

Given that x(n) = {1, 2, 3, –1, –2, –3, 4, 5, 6}. Let x(n) can be sectioned into three sequences, each
consisting of three samples of x(n) as shown below.

x1(n) = 1 ; n = 0 x2(n) = –1 ; n = 3 x3(n) = 4 ; n = 6

=2;n=1 = –2 ; n = 4 =5;n=7
=3;n=2 = –3 ; n = 5 =6;n=8

Let y1(n), y2(n) and y3(n) be the output of linear convolution of x1(n), x2(n) and x3(n) with h(n) respectively.

Here h(n) starts at n = nh = 0

x1(n) starts at n = n1 = 0, \ y1(n) will start at n = n1 + nh = 0 + 0 = 0

x2(n) starts at n = n2 = 3, \ y2(n) will start at n = n2 + nh = 3 + 0 = 3
x3(n) starts at n = n3 = 6, \ y3(n) will start at n = n3 + nh = 6 + 0 = 6
Here linear convolution of each section is performed between two sequences each consisting of three
samples. Hence each convolution output will consists of 3 + 3 – 1 = 5 samples. The convolution of each section
is performed by tabular method as shown below.

Note : 1.Here N1 = 9, N2 = 3, N3 = 3, \ (N2 – 1) = 3 – 1 = 2 and (N2 + N3 – 1) = 3 + 3 – 1 = 5.

2.The unfilled boxes in the table are considered as zero.
3.For convenience of convolution operation, the index n is replaced by m in x1(n), x2(n), x3(n)
and h(n).
Chapter 2 - Discrete Time Signals and Systems 2. 92
Convolution of Section 1

m –2 –1 0 1 2 3 4
+∞

x1(m) 1 2 3 y1(n) = x1(n) ∗ h(n) = ∑ x (m) h(n − m)

m = −∞
1

h(m) 2 1 –1 +∞

h(–m) = h0(m) –1 1 2
= ∑ x (m) h (m)
m = −∞
1 n

h(1 – m) = h1(m) –1 1 2 for n = 0, 1, 2, 3, 4

where hn (m) = h(n − m)
h(2 – m) = h2(m) –1 1 2
h(3 – m) = h3(m) –1 1 2
h(4 – m) = h4(m) –1 1 2

When n = 0 ; y1(0) = å x (m) h (m) = 0 + 0 + 2 + 0 + 0 = 2

1 o

When n = 1 ; y (1) = å x (m) h (m) = 0 + 1 + 4 + 0

1 1 1
= 5
When n = 2 ; y (2) = å x (m) h (m) = –1 + 2 + 6
1 1 2
= 7
When n = 3 ; y (3) = å x (m) h (m) = 0 – 2 + 3 + 0
1 1 3
= 1
When n = 4 ; y (4) = å x (m) h (m) = 0 + 0 – 3 + 0 + 0 = –3
1 1 4

Convolution of Section 2

m –2 –1 0 1 2 3 4 5 6 7
x2(m) –1 –2 –3
h(m) 2 1 –1
h(–m) = h0(m) –1 1 2
h(3 – m) = h3(m) –1 1 2
h(4 – m) = h4(m) –1 1 2
h(5 – m) = h5(m) –1 1 2
h(6 – m) = h6(m) –1 1 2
h(7 – m) = h7(m) –1 1 2

∞ ∞
y 2(n) = x 2(n) ∗ h(n) = ∑ x (m) h(n − m) = ∑ x (m) h (m); n = 3, 4, 5, 6, 7
m = −∞
2
m = −∞
2 n

where hn (m) = h(n − m)

∑ x (m) h (m) = 0 + 0 − 2 + 0 + 0 = –2
When n = 3 ; y 2 (3) = 2 3

When n = 4 ; y (4) = ∑ x (m) h (m) = 0 − 1 − 4 + 0

2 2 4= −5
When n = 5 ; y (5) = ∑ x (m) h (m) = 1 − 2 − 6
2 2 5= −7
When n = 6 ; y (6) = ∑ x (m) h (m) = 0 + 2 − 3 + 0
2 2 6= −1
When n = 7 ; y (7) = ∑ x (m) h (m) = 0 + 0 + 3 + 0 + 0 = 3
2 2 7
2. 93 Digital Signal Processing
Convolution of Section 3
m –2 –1 0 1 2 3 4 5 6 7 8 9 10
x3(m) 4 5 6
h(m) 2 1 –1
h(–m) = h0(m) –1 1 2
h(6 – m) = h6(m) –1 1 2
h(7 – m) = h7(m) –1 1 2
h(8 – m) = h8(m) –1 1 2
h(9 – m) = h9(m) –1 1 2
h(10 – m) = h10(m) –1 1 2
∞ ∞
y3(n) = x3(n) ∗ h(n) = ∑ x (m) h(n − m) = ∑ x (m) h (m) ;
m = –∞
3
m = –∞
3 n n = 6, 7, 8, 9, 10

where hn (m) = h(n − m)

When n = 6 ; y3 (6) =∑ x (m) h (m) = 0 + 0 + 8 + 0 + 0 = 8
3 6

When n = 7 ; y (7) = ∑ x (m) h (m) = 0 + 4 + 10 + 0

3 3 7 = 14
When n = 8 ; y (8) = ∑ x (m) h (m) = –4 + 5 + 12
3 3 8 = 13
When n = 9 ; y (9) = ∑ x (m) h (m) = 0 – 5 + 6 + 0
3 3 9 = 1
When n = 10 ; y (10) = ∑ x (m) h (m) = 0 + 0 – 6 + 0 + 0 = – 6
3 3 10

To Combine the Output of the Convolution of Each Section

It can be observed that the last N2 – 1 sample in an output sequence overlaps with the first N2 – 1 sample
of next output sequence. In this method, the overall output is obtained by combining the outputs of the convolution
of all sections. The overlapped portions (or samples) are added while combining the output.
The output of all sections can be represented in a table as shown below. Then the samples corresponding
to same value of n are added to get the overall output.
n 0 1 2 3 4 5 6 7 8 9 10
y1(n) 2 5 7 1 –3
y2(n) –2 –5 –7 –1 3
y3(n) 8 14 13 1 –6
y(n) 2 5 7 –1 –8 –7 7 17 13 1 –6

\ y(n) = x(n) * h(n) = {2, 5, 7, –1, –8, –7, 7, 17, 13, 1, – 6}

b) Overlap Save Method
In this method the longer sequence is sectioned into sequences of size equal to smaller sequence. The
number of samples that will be obtained in the output of linear convolution of each section is determined. Then
each section of longer sequence is converted to the size of output sequence using the samples of original longer
sequences. The smaller sequence is also converted to the size of output sequence by appending with zeros.
Then the circular convolution of each section is performed.
Here x(n) is a longer sequence when compared to h(n). Hence x(n) is sectioned into sequences of size
equal to h(n). Given that x(n) = {1, 2, 3, –1, –2, –3, 4, 5, 6}.
Let x(n) be sectioned into three sequences each consisting of three samples as shown below.
Let, N1 = Length of longer sequence
N2 = Length of smaller sequence
N3 = N2 = Length of each section of longer sequence.
Chapter 2 - Discrete Time Signals and Systems 2. 94
x1(n) = 1 ; n = 0 x2(n) = –1 ; n = 3 x3(n) = 4 ; n = 6
=2;n=1 = –2 ; n = 4 =5;n=7
=3;n=2 = –3 ; n = 5 =6;n=8
Let y1(n), y2(n) and y3(n) be the output of linear convolution of x1(n), x2(n) and x3(n) with h(n) respectively.
Here linear convolution of each section will result in an output sequence consisting of 3 + 3 – 1 = 5 samples.
Hence each section of longer sequence is converted to five sample sequence, using the samples of
original longer sequence as shown below. It can be observed that the first N2 – 1 samples of x2(n) is placed as
overlapping sample at the end of x1(n). The first N2 – 1 samples of x3(n) is placed as overlapping sample at the end
of x2(n). Since there is no fourth section, the overlapping samples of x3(n) are considered as zeros.
x1(n) = 1 ; n = 0 x2(n) = –1 ; n = 3 x3(n) = 4 ; n = 6
= 2;n=1 = –2 ; n = 4 =5;n=7
= 3;n=2 = –3 ; n = 5 =6;n=8
= –1 ; n = 3 = 4; n=6 =0;n=9
= –2 ; n = 4 = 5; n=7 = 0 ; n = 10
The sequence h(n) is also converted to five sample sequence by appending with zeros.
\ h(n) = {2, 1, –1, 0, 0}
Now perform circular convolution of each section with h(n). The output sequence obtained from circular
convolution will have five samples. The circular convolution of each section is performed by tabular method as
shown below.
Here h(n) starts at n = nh = 0
x1(n) starts at n = n1 = 0, \ y1(n) will start at n = n1 + nh = 0 + 0 = 0
x2(n) starts at n = n2 = 3, \ y2(n) will start at n = n2 + nh = 3 + 0 = 3
x3(n) starts at n = n3 = 6, \ y3(n) will start at n = n3 + nh = 6 + 0 = 6

Note : 1. Here N1 = 9, N2 = 3, N3 = 3 \ (N2 – 1) = 3 – 1 = 2 and [N2 + N3 – 1] = 3 + 3 – 1 = 5 samples.

2. The boldfaced numbers in the table are obtained by periodic extension.
3. For convenience of convolution operation the index n is replaced by m in x1(n), x2(n), x3(n) and h(n).
Convolution of Section 1

m –4 –3 –2 –1 0 1 2 3 4
x1(m) 1 2 3 –1 –2
h(m) 2 1 –1 0 0
h((–m))5 = h0(m) 0 0 –1 1 2 0 0 –1 1
h((1 – m))5 = h1(m) 0 0 –1 1 2 0 0 –1
h((2 – m))5 = h2(m) 0 0 –1 1 2 0 0
h((3 – m))5 = h3(m) 0 0 –1 1 2 0
h((4 – m))5 = h4(m) 0 0 –1 1 2

mf 4
y1(n) = x1(n) ∗ h(n) = ∑ x (m) h((n − m))
m = mi
1 N = ∑ x (m) h (m); n = 0, 1, 2, 3, 4
m = 0
1 n

where hn (m) = h((n − m))N

2. 95 Digital Signal Processing

When n = 0 ; y1(0) = å x (m)h (m) = 2 + 0 + 0 + 1 – 2 = 1

1 o

When n = 1 ; y (0) = å x (m)h (m) = 1 + 4 + 0 + 0 + 2 = 7

1 1 1

When n = 2 ; y (2) = å x (m)h (m) = –1 + 2 + 6 + 0 + 0 = 7

1 1 2

When n = 3 ; y (3) = å x (m)h (m) = 0 – 2 + 3 – 2 + 0 = – 1

1 1 3

When n = 4 ; y (4) = å x (m)h (m) = 0 + 0 – 3 – 1 – 4 = – 8

1 1 4

Convolution of Section 2

m –4 –3 –2 –1 0 1 2 3 4 5 6 7
x2(m) –1 –2 –3 4 5
h(m) 2 1 –1 0 0
h(–m) = h0(m) 0 0 –1 1 2
h((3 – m))5 = h3(m) 0 0 –1 1 2 0 0 –1 1
h((4 – m))5 = h4(m) 0 0 –1 1 2 0 0 –1
h((5 – m))5 = h5(m) 0 0 –1 1 2 0 0
h((6 – m))5 = h6(m) 0 0 –1 1 2 0
h((7 – m))5 = h7(m) 0 0 –1 1 2
mf 7
y 2 (n) = x 2(n) ∗ h(n) = ∑ x (m) h((n − m))
m = mi
2 N = ∑ x (m) h (m); n = 3, 4, 5, 6, 7
m = 3
2 n

where hn (m) = h((n − m))N

∑
When n = 3 ; y 2 (3) = x 2(m) h3 (m) = −2 + 0 + 0 – 4 + 5 = – 1
When n = 4 ; y 2(4) =∑ x (m) h (m) = −1 − 4 + 0 + 0 – 5 = −10
2 4

When n = 5 ; y (5) = ∑ x (m) h (m) = 1 − 2 − 6 + 0 + 0 = − 7

2 2 5

When n = 6 ; y (6) = ∑ x (m) h (m) = 0 + 2 − 3 + 8 + 0 = 7

2 2 6

When n = 7 ; y (7) = ∑ x (m) h (m) = 0 + 0 + 3 + 4 + 10 = 17

2 2 7

Convolution of Section 3

m –4 –3 –2 –1 0 1 2 3 4 5 6 7 8 9 10
x3(m) 4 5 6 0 0
h(m) 2 1 –1 0 0
h(–m) = h0(m) 0 0 –1 1 2
h((6 – m))5 = h6(m) 0 0 –1 1 2 0 0 –1 1
h((7 – m))5 = h7(m) 0 0 –1 1 2 0 0 –1
h((8 – m))5 = h8(m) 0 0 –1 1 2 0 0
h((9 – m))5 = h9(m) 0 0 –1 1 2 0
h((10 – m))5 = h10(m) 0 0 –1 1 2

mf 10
y3 (n) = x 3(n) ∗ h(n) = ∑ x (m) h((n − m))
m = mi
3 N = ∑ x (m) h (m)
m = 6
3 n ; n = 6, 7, 8, 9, 10

where hn (m) = h((n − m))N

Chapter 2 - Discrete Time Signals and Systems 2. 96

When n = 6 ; y3 (6) = ∑ x (m) h (m) = 8 + 0 + 0 + 0 + 0 = 8

3 6

When n = 7 ; y (7) = ∑ x (m) h (m) = 4 + 10 + 0 + 0 + 0 = 14

3 3 7

When n = 8 ; y (8) = ∑ x (m) h (m) = – 4 + 5 + 12 + 0 + 0 = 13

3 3 8

When n = 9 ; y (9) = ∑ x (m) h (m) = 0 – 5 + 6 + 0 + 0 = 1

3 3 9

When n = 10 ; y (10) = ∑ x (m) h (m) = 0 + 0 – 6 + 0 + 0

3 3 10 = –6
To Combine the Output of Convolution of Each Section
It can be observed that the last N2–1 samples in an output sequence overlaps with the first N2–1 samples
of next output sequence. In overlap save method the overall output is obtained by combining the outputs of the
convolution of all sections. While combining the outputs, the overlapped first N2–1 samples of every output
sequence is discarded and the remaining samples are simply saved as samples of y(n) as shown in the following
table.
n 0 1 2 3 4 5 6 7 8 9 10
y1(n) 1 7 7 –1 –8
y2(n) –1 10 –7 7 17
y3(n) 8 14 13 1 –6
y(n) * * 7 –1 –8 –7 7 17 13 1 –6

\ y(n) = x(n) * h(n) = {*, *, 7, –1, –8, –7, 7, 17, 13, 1, –6}
Note : Here y(n) is linear convolution of x(n) and h(n). It can be observed that the results of both the
methods are same except the first N2 – 1 samples.

2.12 Inverse System and Deconvolution

2.12.1 Inverse System
The inverse system is used to recover the input from the response of a system. For a given system, the
inverse system exists, if distinct inputs to a system leads to distinct outputs. The inverse systems exists for
all LTI systems.
The inverse system is denoted by H–1. If x(n) is input and y(n) is the output of a system, then y(n) is
the input and x(n) is the output of its inverse system.
x (n) y (n) y (n) w (n) = x(n)
H H −1

F ig 2.3 5 a : System . F ig 2.3 5 b : In verse system .

F ig 2.3 5 : A system and its inverse system .

Let h(n) be the impulse response of a system and h¢(n) be the impulse response of inverse system. Let
us connect the system and its inverse in cascade as shown in fig 2.36.
Identity sy s tem

H H
-1

y (n)
x(n) h(n) h ’(n) w (n) = x(n)

F ig 2.3 6 : C a sca d e co n n ectio n o f a system a n d its in v erse.

2. 97 Digital Signal Processing
Now it can be proved that,
h(n) * h¢(n) = d(n) .....(2.60)
Therefore the cascade of a system and its inverse is identity system.
Proof :
With reference to fig 2.36 we can write,
y(n) = x(n) * h(n) .....(2.61)
w(n) = y(n) * h¢(n) .....(2.62)
On substituting for y(n) from equation (2.61) in equation (2.62) we get,
w(n) = x(n) * h(n) * h¢(n) .....(2.63)
In equation (2.63),
if, h(n) * h¢(n) = d(n), then, x(n) * d(n) = x(n)
In a inverse system, w(n) = x(n), and so,

h(n) * h¢(n) = d(n). Hence proved.

2.12.2 Deconvolution
In an LTI system the response y(n) is given by convolution of input x(n) and impulse response h(n).
i.e., y(n) = x(n) * h(n)
The process of recovering the input from the response of a system is called deconvolution. (or the
process of recovering x(n) from x(n) * h(n) is called deconvolution).
When the response y(n) and impulse response h(n) are available, then the input x(n) can be computed
using the equation (2.64).

x(n) =
1 LM n −1
y(n) − ∑ x(m) h(n − m)
OP .....(2.64)
h(0) MN m= 0 PQ
Proof :
Let x(n) and h(n) be finite duration sequences starting from n = 0. Consider the matrix method
of convolution of x(n) and h(n) shown below.

h(0) h(1) h(2) h(3)

x(0) x(0)h(0) x(0)h(1) x(0)h(2) x(0)h(3)

x(1) x(1)h(0) x(1)h(1) x(1)h(2) x(1)h(3)

x(2) x(2)h(0) x(2)h(1) x(2)h(2) x(2)h(3)

x(3) x(3)h(0) x(3)h(1) x(3)h(2) x(3)h(3)

Chapter 2 - Discrete Time Signals and Systems 2. 98
From the above two-dimensional array we can write,

y(0)
y( n) = x(0) h(0) ⇒ x(0) =
h(0)
y(1) − x(0) h(1)
y(1) = x(1) h(0)+ x(0) h(1) ⇒ x(1) =
h(0)
y(2) − x(0) h(2) − x(1) h(1)
y(2) = x(2) h(0)+ x(1) h(1)+ x(0) h(2) ⇒ x(2) =
h(0)
y(3) − x(0) h(3) − x(1) h(2) − x(2) h(1)
y(3) = x(3) h(0)+ x(2) h(1)+ x(1) h(2)+ x(0) h(3) ⇒ x(3) =
h(0)
and so on.
From the above analysis, in general for any value of n, the x(n) is given by,

y(n) − x(0) h(n) − x(1) h(n − 1) − ...... − x(n − 1) h(1)

x( n) =
h(0)
1 LM n−1 O
∴ x(n) =
h(0)
y(n) −
MN ∑ x(m) h(n − m)PP
m =0 Q
Example 2.30
n
A discrete time system is defined by the equation, y(n) = ∑
m =0
x(m) ; for n ≥ 0. Find the inverse system.

Solution
n
Given that, y(n) = ∑ x(m)
m=0

0
When n = 0; y(0) = ∑ x(m) = x(0)
m =0
1
When n = 1; y(1) =
m=0
∑ x(m) = x(0) + x(1) = y(0) + x(1)

2
When n = 2; y(2) = ∑
m =0
x(m) = x(0) + x(1) + x(2) = y(1) + x(2)

3
When n = 3; y(3) = ∑
m=0
x(m) = x(0) + x(1) + x(2) + x(3) = y(2) + x(3)

and so on,

From the above analysis we can write,

x(0) = y(0) ; x(1) = y(1) – y(0) ; x(2) = y(2) – y(1) ; x(3) = y(3) – y(2) and so on,

In general for any value of n, the signal x(n) can be written as,

x(n) = y(n) – y(n –1)

Therefore the inverse system is defined by the equation,

x(n) = y(n) – y(n –1)

­
2. 99 Digital Signal Processing

Example 2.31
When a discrete time system is excited by an input x(n), the response is y(n) = { 2, 5, 11, 17, 13, 12 }
­--
If the impulse response of the system is h(n) = { 2, 1, 3 }, then what will be the input to the system?
-
Solution
Let N1 be number of samples in x(n) and N2 be number of samples in h(n), then the number of samples N3
in y(n) is given by,
N3 = N1 + N2 – 1
\ N1 = N3 – N2 + 1 = 6 – 3 + 1 = 4 samples
Therefore x(n) is 4 sample sequence.
Each sample of x(n) is given by,

1 LMy(n) − x(m) h(n − m)OP

n − 1
x(n) =
h(0) MN ∑m=0 PQ
y(0) 2
When n = 0 ; x(0) = = =1
h(0) 2
1 LMy(1) − x(m) OP
When n = 1 ; x(1) =
h(0) MN ∑
m=0
h(1 − m)
PQ
1 1
= y(1) − x(0) h(1) = 5 − 1× 1 = 2
h(0) 2
1 LMy(2) − x(m) h(2 − m) OP
1
When n = 2 ; x(2) =
h(0) MN ∑
m =0 PQ
1 1
= y(2) − x(0) h(2) − x(1) h(1) = 11 − 1 × 3 − 2 × 1 = 3
h(0) 2
1 LMy(3) − x(m) h(3 − m) OP
2
When n = 3 ; x(3) =
h(0) MN ∑
m=0 PQ
1 1
= y(3) − x(0) h(3) − x(1) h(2) − x(2) h(1) = 17 − 1 × 0 − 2 × 3 − 3 × 1 = 4
h(0) 2
∴ x(n) = {x(0), x(1), x(2), x(3)} = {1, 2, 3, 4}
A

2.13 Correlation, Crosscorrelation and Autocorrelation

The correlation of two discrete time sequences x(n) and y(n) is defined as,
+∞
.....(2.65)
rxy (m) = ∑ x(n) y(n − m)
n = −∞
where rxy(m) is the correlation sequence obtained by correlation of x(n) and y(n) and m is the variable used
for time shift. The correlation of two different sequences is called crosscorrelation and the correlation of a
sequence with itself is called autocorrelation. Hence autocorrelation of a discrete time sequence is defined as,
+∞
rxx (m) = ∑ x(n) x(n − m) .....(2.66)
n = −∞
If the sequence x(n) has N1 samples and sequence y(n) has N2 samples then the crosscorrelation
sequence rxy(m) will be a finite duration sequence consisting of N1 + N2 – 1 samples.If the sequence x(n) has
N samples, then and the autocorrelation sequence rxx(m) will be a finite duration sequence consisting of
2N – 1 samples.
Chapter 2 - Discrete Time Signals and Systems 2. 100
In the equation (2.65), the sequence x(n) is unshifted and the sequence y(n) is shifted by m units of
time for correlation operation. The same results can be obtained if the sequence y(n) is unshifted and the
sequence x(n) is shifted opposite to that of earlier case by m units of time, hence the crosscorrelation
operation can also be expressed as,
+∞
.....(2.67)
rxy (m) = ∑ x(n + m) y(n)
n = −∞

2.13.1 Procedure for Evaluating Correlation

Let, x(n) = Discrete time sequence with N1 samples
y(n) = Discrete time sequence with N2 samples
Now the correlation of x(n) and y(n) will produce a sequence rxy(m) consisting of N1+N2–1 samples.
Each sample of rxy(m) can be computed using the equation (2.65). The value of rxy(m) at m = q is obtained by
replacing m by q, in equation (2.65).
+∞
.....(2.68)
∴ rxy (q) = ∑ x(n) y(n − q)
n = −∞

The evaluation of equation (2.68) to determine the value of rxy(m) at m = q involves the following three
steps.
1. Shifting : Shift y(n) by q times to the right if q is positive, shift y(n) by q times to the
left if q is negative to obtain y(n - q).
2. Multiplication : Multiply x(n) by y(n - q) to get a product sequence. Let the product
sequence be vq(n). Now, vq(n) = x(n) × y(n - q).
3. Summation : Sum all the values of the product sequence vq(n) to obtain the value of
rxy(m) at m = q. [i.e., rxy(q)].
The above procedure will give the value rxy(m) at a single time instant say m = q. In general we are
interested in evaluating the values of the sequence rxy(m) over all the time instants in the range -¥ < m < ¥ .
Hence the steps 1, 2 and 3 given above must be repeated, for all possible time shifts in the range -¥ < m < ¥ .
In the correlation of finite duration sequences it is possible to predict the start and end of the resultant
sequence. If x(n) is N-point sequence and starts at n = n1 and if y(n) is N2-point sequence and starts at n = n2 then,
the initial value of m = mi for rxy(m) is mi = n1 – (n2 + N2 – 1). The value of x(n) for n < n1 and the value of y(n) for
n < n2 are then assumed to be zero.The final value of m = mf for rxy(m) is mf = mi + (N1+N2– 2).
The correlation operation involves all the steps in convolution operation except the folding.
Hence it can be proved that the convolution of x(n) and folded sequence y(-n) will generate the crosscorrelation
sequence rxy(m).
i.e., r (m) = x(n) * y(-n) .....(2.69)
xy

The procedure given above can be used for computing autocorrelation of x(n). For computing
autocorrelation using equation (2.68) replace y(n – q) by x(n – q). Similarly when equation (2.69) is used,
replace y(–n) by x(–n).
The autocorrelation of N-point sequence x(n) will give 2N –1 point autocorrelation sequence.
If x(n) starts at n = nx then initial value of m = mi for rxx(m) is mi = – (N –1). The final value of m = mf for rxx(m)
is mf = mi + (2N–2).
2. 101 Digital Signal Processing
Properties of Correlation
1. The crosscorrelation sequence rxy(m) is simply a folded version of ryx(m),
i.e., rxy(m) = ryx(-m)
Similarly for autocorrelation sequence,
rxx(m) = rxx(-m)
Hence autocorrelation is an even function.
2. The crosscorrelation sequence satisfies the condition,

rxy (m) ≤ rxx ( 0) ryy ( 0) = Ex Ey

where, Ex and Ey are energy of x(n) and y(n) respectively.

On applying the above condition to autocorrelation sequence we get,
rxx (m) ≤ rxx (0) = E x

From the above equations we infer that the crosscorrelation sequence and autocorrelation
sequences attain their respective maximum values at zero shift/lag.
3. Using the maximum value of crosscorrelation sequence, the normalized crosscorrelation sequence
is defined as,
rxy (m)
ρxy (m) ≤
rxx (0) ryy (0)
Using the maximum value of autocorrelation sequence, the normalized autocorrelation sequence
is defined as,
rxx (m)
ρxx (m) ≤
rxx ( 0)

Methods of Computing Correlation

Method 1: Graphical Method
Let x(n) and y(n) be the input sequences and rxy(m) be the output sequence.
1. Sketch the graphical representation of the input sequences x(n) and y(n).
2. Shift the sequence y(n) to the left graphically so that the product of x(n) and shifted y(n) gives
only one nonzero sample. Now multiply x(n) and shifted y(n) to get a product sequence, and then
sum up the samples of product sequence, which is the first sample of output sequence.
3. To get the next sample of output sequence, shift y(n) of previous step to one position right
and multiply the shifted sequence with x(n) to get a product sequence. Now the sum of the
samples of product sequence gives the second sample of output sequence.
4. To get subsequent samples of output sequence, the step 3 is repeated until we get a nonzero
product sequence.
Method 2: Tabular Method
The tabular method is same as that of graphical method, except that the tabular representation of the
sequences are employed instead of graphical representation. In tabular method, every input sequence and
shifted sequence is represented on a row in a table.
Chapter 2 - Discrete Time Signals and Systems 2. 102
Method 3: Matrix Method
Let x(n) and y(n) be the input sequences and rxy(m) be the output sequence. We know that the
convolution of x(n) and folded sequence y(-n) will generate the crosscorrelation sequence rxy(m). Hence fold
y(n) to get y(-n), and compute convolution of x(n) and y(-n) by matrix method.
In matrix method one of the sequence is represented as a row and the other as a column as shown
below.
y (0) y (−1 ) y (−2 ) y (−3 )

x (0) x (0)y(0) x (0)y( −1) x (0)y( −2) x (0)y( −3)

x (1) x (1)y(0) x (1)y( −1) x (1)y( −2) x (1)y( −3)

x (2) x (2)y(0) x (2)y( −1) x (2)y( −2) x (2)y( −3)

x (3) x (3)y(0) x (3)y( −1) x (3)y( −2) x (3)y( −3)

Multiply each column element with row elements and fill up the matrix array.
Now the sum of the diagonal elements gives the samples of output sequence rxy(m). (The sum of the
diagonal elements are shown below for reference).
:
:
rxy(0) = ..... + x(0) y(0) + .....
rxy(1) = ..... + x(1) y(0) + x(0 ) y(-1) + .....
rxy(2) = ..... + x(2) y(0) + x(1) y(-1) + x(0) y(-2) + .....
rxy(3) = ..... + x(3) y(0) + x(2) y(-1) + x(1) y(-2) + x(0) y(-3) + .....
:
:

Example 2.32
Perform crosscorrelation of the sequences, x(n) = {1, 1, 2, 2} and y(n) = {1, 0.5, 1}.

Solution
Let rxy(m) be the crosscorrelation sequence obtained by crosscorrelation of x(n) and y(n).
The crosscorrelation sequence rxy(m) is given by,
+∞
rxy = ∑ x(n) y(n − m)
n = −∞

The x(n) starts at n = 0 and has 4 samples.

2. 103 Digital Signal Processing
\ n1 = 0, N1 = 4
The y(n) starts at n = 0 and has 3 samples.
\ n2 = 0, N2 = 3
Now, rxy(m) will have N1 + N2 – 1 = 4 + 3 – 1 = 6 samples.
The initial value of m = mi = n1 – (n2 + N2 –1)
= 0 – (0 + 3 – 1) = – 2
The final value of m = mf = mi + (N1 + N2 – 2)
= –2 + (4 + 3 – 2) = 3
In this example the correlation operation is performed by three methods.
Method 1 : Graphical Method
The graphical representation of x(n) and y(n) are shown below.

x (n ) y (n )
2 2
1 1 1 1
0.5
0 1 2 3 n 0 1 2 n
F ig 1. F ig 2.
The 6 samples of rxy(m) are computed using the equation,
+∞ +∞
rxy (m) = ∑
n = −∞
x(n) y(n − m) = ∑
n = −∞
x(n) y m (n) ; where y m (n) = y(n − m)

The computation of each sample of rxy(n) using the above equation are graphically shown in fig 3 to fig 8.
The graphical representation of output sequence is shown in fig 9.
+∞ +∞ +∞
When m = −2 ; rxy ( −2) = ∑
n = −∞
x(n) y(n − ( −2)) = ∑
n = −∞
x(n) y −2 (n) = ∑
n = −∞
v − 2 (n)

x (n ) y −2 (n) v −2 (n)

2 2
X ⇒
1 1 1 1 1
0.5
0 1 2 3 n −2 −1 0 n −2 −1 0 1 2 3 n
T he su m o f pro du ct s eq ue n ce
v −2 (n ) giv es rxy ( −2)
F ig 3 : C o m p uta tio n o f r x y ( −2).
∴ rxy ( −2) = 0 + 0 + 1 + 0 + 0 + 0 = 1

+∞ +∞ +∞
When m = −1 ; rxy (−1) = ∑
n = −∞
x(n) y(n − ( −1)) = ∑
n = −∞
x(n) y −1(n) = ∑
n = −∞
v − 1(n)

x (n ) y −1 (n) v −1 (n )

2 2
X ⇒ 1
1 1 1 1
0.5 0.5
0 1 2 3 n −1 0 1 n −1 0 1 2 3 n
T he s um of pro du c t s e qu en c e
v −1(n ) g iv e s rx y ( −1 )
F ig 4 : C om p u ta tio n of rx y ( −1 ).
∴ rxy ( −1) = 0 + 0 .5 + 1 + 0 + 0 = 1 .5
Chapter 2 - Discrete Time Signals and Systems 2. 104
+∞ +∞ +∞
When m = 0 ; rxy (0) = ∑ x(n) y(n) = ∑ x(n) y 0 (n) = ∑ v 0 (n)
n = −∞ n = −∞ n = −∞

x (n ) y 0 (n ) v 0 (n )

2 2 2
X ⇒ 1
1 1 1 1
0.5 0.5
0 1 2 3 n 0 1 2 n 0 1 2 3 n
T he su m o f pro du ct s eq u en ce

F ig 5 : C o m p u ta tio n o f rxy (0 ). v 0 (n ) g ive s rx y (0)

∴ rx y (0) = 1 + 0 .5 + 2 + 0 = 3.5
+∞ +∞ +∞
When m = 1 ; rxy (1) = ∑
n = −∞
x(n) y(n − 1)) = ∑
n = −∞
x(n) y1(n) = ∑
n = −∞
v1(n)

x (n ) y 1 (n ) v 1 (n )

2 2 2
X ⇒
1 1 1 1 1 1
0.5
0 1 2 3 n 0 1 2 3 n 0 1 2 3 n
T he su m o f pro du ct s eq u en ce
F ig 6 : C o m p u ta tio n of rxy (1 ). v 1 (n) giv es rxy (1 )
∴ rx y (1 ) = 0 + 1 + 1 + 2 = 4

+∞ +∞ +∞
When m = 2 ; rxy (2) = ∑
n = −∞
x(n) y(n − 2) = ∑
n = −∞
x(n) y 2 (n) = ∑
n = −∞
v 2 (n)

x (n ) y 2 (n ) v 2 (n )

2 2 2
X ⇒ 1
1 1 1 1
0.5
0 1 2 3 n 0 1 2 3 4 n 0 1 2 3 4 n
T he su m o f pro du ct s eq u en ce
F ig 7 : C o m pu ta tio n o f rx y (2). v 2 (n ) giv es rx y (2 )
∴ rx y (2 ) = 0 + 0 + 2 + 1 + 0 = 3

+∞ +∞ +∞
When m = 3 ; rxy (3) = ∑
n = −∞
x(n) y(n − 3) = ∑
n = −∞
x(n) y 3(n) = ∑
n = −∞
v 3(n)

x (n ) y 3 (n ) v 3 (n )

2 2
X ⇒ 2
1 1 1 1
0.5
0 1 2 3 n 0 1 2 3 4 5 n 0 1 2 3 4 5 n
T he su m o f pro du ct s eq u en ce
F ig 8 : C o m p uta tio n o f rxy (3 ). v 3 (n ) giv es rx y (3 )
∴ rx y (3) = 0 + 0 + 0 + 2 + 0 + 0 = 2
2. 105 Digital Signal Processing
The crosscorrelation sequence, rxy(m) = {1, 1.5, 3.5, 4, 3, 2}
-
r x y (m )

4 ­
3.5
3

2
1.5
1

−2 −1 0 1 2 3 m
F ig 9 : G ra p h ic a l rep resenta tio n o f rxy (m ).
Method 2: Tabular Method
The given sequences and the shifted sequences can be represented in the tabular array as shown below.

n –2 –1 0 1 2 3 4 5
x(n) 1 1 2 2
y(n) 1 0.5 1
y(n –(–2)) = y–2(n) 1 0.5 1
y(n –(–1)) = y–1(n) 1 0.5 1
y(n) = y0(n) 1 0.5 1
y(n – 1) = y1(n) 1 0.5 1
y(n – 2) = y2(n) 1 0.5 1
y(n – 3) = y3(n) 1 0.5 1

Note: The unfilled boxes in the table are considered as zeros.

Each sample of rxy(m) is given by,
+∞ +∞
rxy (m) = ∑
n = −∞
x(n) y(n − m) = ∑
n = −∞
x(n) y m (n) ; where ym (n) = y(n − m)

To determine a sample of rxy(m) at m = q, multiply the sequence x(n) and yq(n) to get a product sequence
[i.e., multiply the corresponding elements of the row x(n) and yq(n)]. The sum of all the samples of the product
sequence gives rxy(q).
3
When m = −2 ; rxy (−2) = ∑
n = −2
x(n) y −2(n) = 0 + 0 + 1+ 0 + 0 + 0 = 1

3
When m = −1 ; rxy (−1) = ∑ x(n) y
n = −1
−1(n) = 0 + 0.5 + 1+ 0 + 0 = 1.5

3
When m = 0 ; rxy (0) = ∑ x(n) y
n =0
0 (n) = 1+ 0.5 + 2 + 0 = 3.5

3
When m = 1 ; rxy (1) = ∑
n=0
x(n) y1(n) = 0 + 1+ 1+ 2 =4

4
When m = 2 ; rxy (2) = ∑ x(n) y (n)
n =0
2 = 0 + 0 + 2 + 1+ 0 =3

5
When m = 3 ; rxy (3) = ∑ x(n) y (n)
n=0
3 = 0 +0 + 0 + 2+ 0 +0 = 2

∴ Crosscorrelation sequence, rxy (m) = {1, 1.5, 3.5, 4, 3, 2}

A
Chapter 2 - Discrete Time Signals and Systems 2. 106
Method 3: Matrix Method
Given that, x(n) = {1, 1, 2, 2} ; y(n) = {1, 0.5, 1} ; \ y(–n) = {1, 0.5, 1}
- - -
The sequence x(n) is arranged as a column and the folded sequence y(–n) is arranged as a row as shown
below. The elements of the two-dimensional array are obtained by multiplying the corresponding row element
with column element. The sum of the diagonal elements gives the samples of the crosscorrelation sequence,
rxy(m).

y ( −n ) y ( −n )
x (n) 1 0.5 1 x (n) 1 0.5 1

1 1 ×1 1 × 0.5 1 ×1 1 1 0.5 1

1 1 ×1 1 × 0.5 1 ×1 ⇒ 1 1 0.5 1

2 2 ×1 2 × 0.5 2 ×1 2 2 1 2

2 2 ×1 2 × 0.5 2 ×1 2 2 1 2

rxy (–2) = 1 ; rxy (–1) = 1 + 0.5 = 1.5 ; rxy(0) = 2 + 0.5 + 1 = 3.5

rxy(1) = 2 + 1 + 1 = 4 ; rxy(2) = 1 + 2 = 3 ; rxy(3) = 2
\ rxy(m) = {1, 1.5, 3.5, 4, 3, 2}
-

Example 2.33
Determine the autocorrelation sequence for x(n) = {1, 2, 3, 4}.
Solution
Let, rxx(m) be the autocorrelation sequence.
The autocorrelation sequence rxx(m) is given by,
+∞
rxx (m) = ∑
n = −∞
x(n) x(n − m)

The x(n) starts at n = 0 and has 4 samples.

\ nx = 0 and N = 4
Now, rxx(m) will have, 2N – 1 = 2 ´ 4 – 1 = 7 samples.
The initial value of m = mi = – (N – 1) = – (4 – 1) = –3
The final value of m = mf = mi + (2N – 2) = –3 + (2 ´ 4 – 2) = 3
The autocorrelation is computed by tabular method. Hence the sequence x(n) and the shifted sequences
of x(n) are tabulated in the following table.

n –3 –2 –1 0 1 2 3 4 5 6
x(n) 1 2 3 4
x(n –(–3)) = x–3(n) 1 2 3 4
x(n –(–2)) = x–2(n) 1 2 3 4
x(n –(–1)) = x–1(n) 1 2 3 4
x(n) = x0(n) 1 2 3 4
x(n – 1) = x1(n) 1 2 3 4
x(n – 2) = x2(n) 1 2 3 4
x(n – 3) = x3(n) 1 2 3 4
2. 107 Digital Signal Processing
Each sample of rxx(m) is given by,
+∞ +∞
rxx (m) = ∑
n = −∞
x(n) x(n − m) = ∑
n = −∞
x(n) xm (n) ; where xm (n) = x(n − m)

To determine a sample of rxx(m) at m = q, multiply the sequence x(n) and xq(n) to get a product sequence
[i.e., multiply the corresponding elements of the row x(n) and xq(n)]. The sum of all the samples of the product
sequence gives rxx(q).
3
W hen m = − 3 ; rxx ( − 3) = ∑
n = −3
x(n) x − 3 (n ) = 0 + 0 + 0 + 4 + 0 + 0 + 0 = 4

3
W hen m = − 2 ; rxx ( − 2) = ∑
n = −2
x(n) x − 2 (n ) = 0 + 0 + 3 + 8 + 0 + 0 = 11

3
W hen m = − 1 ; rxx ( − 1 ) = ∑
n = −1
x(n) x − 1(n ) = 0 + 2 + 6 + 12 + 0 = 20

3
W hen m = 0 ; rxx ( 0 ) = ∑
n = 0
x(n) x 0 (n ) = 1 + 4 + 9 + 16 = 30

4
W hen m = 1 ; rxx (1 ) = ∑
n = 0
x(n) x 1(n ) = 0 + 2 + 6 + 12 + 0 = 20

5
W hen m = 2 ; rxx ( 2) = ∑
n = 0
x(n) x 2 (n ) = 0 + 0 + 3 + 8 + 0 + 0 = 11

6
W hen m = 3 ; rxx (3) = ∑
n = 0
x(n) x 3 (n ) = 0 + 0 + 0 + 4 + 0 + 0 + 0 = 4

∴ Autocorrelation sequence, rxx (m ) = {4, 11, 20, 30, 20, 11, 4}

A
2.14 Circular Correlation
The circular correlation of two periodic discrete time sequences x(n) and y(n) with periodicity of N
samples is defined as,
N−1
rxy (m) = ∑ x(n) y* ((n − m))
n=0
N
.....(2.70)

where, rxy ( m) is the sequence obtained by circular correlation

y*((n – m))N represents circular shift of y*(n)
m is a variable used for circular time shift
The circular correlation of two different sequences is called circular crosscorrelation and the circular
correlation of a sequence with itself is called circular autocorrelation. Hence circular autocorrelation of a
discrete time sequence is defined as,
N−1
rxx ( m) = ∑ x( n) x* ((n − m))
n=0
N
.....(2.71)

The output sequence obtained by circular correlation is also periodic sequence with periodicity of N
samples. Hence this correlation is also called periodic correlation. The circular correlation is defined for
periodic sequences. But circular correlation can be performed with non-periodic sequences by periodically
extending them.The circular correlation of two sequences requires that, at least one of the sequences should
be periodic. Hence it is sufficient if one of the sequences is periodically extended in order to perform circular
correlation.
Chapter 2 - Discrete Time Signals and Systems 2. 108
The circular correlation of finite duration sequences can be performed only if both the sequences
consists of same number of samples. If the sequences have different number of samples, then convert the
smaller size sequence to the size of larger size sequence by appending zeros.

In the equation (2.70), the sequence x(n) is unshifted and the sequence y*(n) is circularly shifted by
m units of time for correlation operation. The same results can be obtained if the sequence y*(n) is unshifted
and the sequence x(n) is circularly shifted opposite to that of earlier case by m units of time, hence the circular
correlation operation can also be expressed as,
N−1
rxy ( m) = ∑ x(( n + m))
n=0
N y* (n) .....(2.72)

Circular correlation basically involves the same three steps as that for correlation, namely shifting one
of the sequence, multiplying the two sequences and finally summing the values of product sequence. The
difference between the two is that in circular correlation the shifting (rotating) operations are performed in a
circular fashion by computing the index of one of the sequences by modulo-N operation. In correlation, there
is no modulo-N operation.
2.14.1 Procedure for Evaluating Circular Correlation
Let, x(n) and y(n) be periodic discrete time sequences with periodicity of N-samples. If x(n) and y(n)
are non-periodic then convert the sequences to N-sample sequence and periodically extend the sequence
y(n) with periodicity of N-samples.
Now the circular correlation of x(n) and y(n) will produce a periodic sequence rxy ( m) with periodicity
of N-samples. The samples of one period of rxy ( m) can be computed using the equation (2.70).

The value of rxy ( m) at m = q is obtained by replacing m by q, in equation (2.70), as shown below.

N−1
rxy (q ) = ∑ x( n) y* ((n − q))
n=0
N
.....(2.73)

The evaluation of equation (2.73) to determine the value of rxy ( m) at m = q involves the following
four steps.
1. Conjugation : Take conjugate of y(n) to get y*(n). If y(n) is a real sequence then y*(n)
will be same as y(n). Represent the samples of one period of the sequences
x(n) and y*(n) on circles.
2. Rotation : Rotate y*(n) by q times in anticlockwise if q is positive, rotate y*(n) by
q times in clockwise if q is negative to obtain y*((n – q))N.
3. Multiplication : Multiply x(n) by y*((n – q))N to get a product sequence. Let the product
sequence be vq(m). Now, vq(m) = x(n) × y*((n – q))N.
4. Summation : Sum up the samples of one period of the product sequence vq(m) to
obtain the value of rxy ( m) at m = q. [i.e., rxy (q ) ].

The above procedure will give the value of rxy ( m) at a single time instant say m = q. In general, we are
interested in evaluating the values of the sequence rxy ( m) in the range 0 < m < N - 1. Hence the steps 2 ,
3 and 4 given above must be repeated, for all possible time shifts in the range 0 < m < N - 1.
2. 109 Digital Signal Processing

2.14.2 Methods of Computing Circular Correlation

Method 1 : Graphical Method
In graphical method the given sequences are converted to same size and represented on circles. In
case of periodic sequences, the samples of one period are represented on circles. Let x(n) and y(n) be the
given real sequences. Let rxy ( m) be the sequence obtained by circular correlation of x(n) and y(n). The
following procedure can be used to get a sample of rxy ( m) at m = q.
1. Represent the sequences x(n) and y(n) on circles.
2. Rotate (or shift) the sequence y(n), q times to get the sequence y((n – q))N. If q is positive then
rotate (or shift) the sequence in anticlockwise direction and if q is negative then rotate (or shift)
the sequence in clockwise direction.

3. The sample of rxy (q ) at m = q is given by,

N−1 N−1
rxy (q ) = ∑ x(n) y((n − q )) N = ∑ x(n) yq ( n)
n= 0 n=0
where, yq ( n) = y(( n − q )) N

Determine the product sequence x(n)yq(n) for one period.

4. The sum of all the samples of the product sequence gives the sample rxy (q ) [i.e., rxy ( m) at m = q].

The above procedure is repeated for all possible values of m to get the sequence rxy ( m).
Method 2 : Using Tabular Array
Let x(n) and y(n) be the given real sequences. Let rxy ( m) be the sequence obtained by circular
correlation of x(n) and y(n). The following procedure can be used to get a sample of rxy ( m) at m = q.
1. Represent the sequences x(n) and y(n) as two rows of tabular array.
2. Periodically extend y(n). Here the periodicity is N, where N is the length of the given sequences.
3. Shift the sequence y(n), q times to get the sequence y((n – q))N. If q is positive then shift the
sequence to the right and if q is negative then shift the sequence to the left.
4. The sample of rxy (q ) at m = q is given by,
N−1 N−1
rxy (q) = ∑ x(n) y((n − q)) N = ∑ x(n) yq ( n)
n=0 n=0
where, yq (n) = y((n − q)) N
Determine the product sequence x(n)yq(n) for one period.
5. The sum of all the samples of the product sequence gives the sample rxy (q ) [i.e., rxy ( m)
at m = q].
The above procedure is repeated for all possible values of m to get the sequence rxy ( m).
Method 3: Using Matrices
Let x(n) and y(n) be the given N-point sequences. The circular correlation of x(n) and y(n) yields
another N-point sequence rxy ( m).
Chapter 2 - Discrete Time Signals and Systems 2. 110
In this method an N ´ N matrix is formed using the sequence y(n) as shown below. The sequence x(n)
is arranged as a column vector (column matrix) of order N ´ 1. The product of the two matrices gives the
resultant sequence rxy ( m).
L r (0) O
OP LM xx((10)) OP MM r (1) PP
xy
LMy(0) y(1) y ( 2) ..... y( N − 1) y( N )
xy
MMy(N) y( 0) y (1) ..... y ( N − 2) y( N − 1) P MM P M r ( 2) P
x( 2) P
MMy(MN − 1) y( N ) y(0) ..... y ( N − 3) y( N − 2) P
P × M
M M PP = MM M PP xy

MMy(2)
M M M M PP M M P MM M PP
y (3) y (4) ..... y( 0) y(1)
P MMx(N − 2)PP M r (N − 2)P
MNy(1) y( 2) y( 3) ..... y( N ) y( 0) PQ MNx(N − 1) PQ MM r (N − 1) PP xy

N Q xy

Example 2.34
Perform circular correlation of the two sequences, x(n) = {1, 1, 2, 1} and y(n)= {2, 3, 1, 1}
- -
Solution
Method 1:Graphical Method of Computing Circular Correlation
The given sequences are represented as points on circles as shown in fig 1 and 2.
x (1) = 1 y (1) = 3

x (2) = 2 x (n) x (0) = 1 y(2) = 1 y (n) y (0) = 2

x (3) = 1 y (3) = 1
F ig 1. F ig 2.
3 2 1 1

y ((n −0))4 y ((n −1) ) 4 y ((n −2) ) 4 y ((n −3) ) 4

1 2 ⇒ 3 1 ⇒ 2
= y 2 (n)
1 ⇒1 3
= y 0 (n) = y 1 (n) = y 3 (n)

1 1 3 2
F ig 3 : C ircu la rly sh ifted seq u e nces y (n -m ), for m = 0 , 1 , 2, 3 .
Let rxy (m) be the sequence obtained by circular correlation of x(n) and y(n). The given sequences are 4
sample sequences and so N = 4. Each sample of rxy (m) is given by the equation,
N − 1 N − 1
rxy (m) = ∑
n = 0
x(n) y((n − m))N = ∑
n = 0
x(n) y m (n), where y m (n) = y((n − m))N

Using the above equation, graphical method of computing each sample of rxy (m) are shown in fig 4 to fig 7.
3 3 3
When m = 0 ; rxy (0) = ∑
n = 0
x(n) y((n − 0))4 = ∑
n = 0
x(n) y 0 (n) = ∑ v (n)
n = 0
0

1 3 1 ×3 = 3

y 0 ( n)
2 x (n) 1 X 1 2 ⇒ 2 ×1 = 2 v 0 (n) 1 ×2 = 2

1 1 1 ×1 = 1

T h e s um of s am ple s o f v 0 (n) giv es rxy (0 )

F ig 4 : C om p u tation o f rxy (0 ).
∴ rx y (0 ) = 2 + 3 + 2 + 1 = 8
2. 111 Digital Signal Processing
3 3 3
When m = 1 ; rxy (1) = ∑ x(n) y((n − 1))
n = 0
4 = ∑ x(n) y (n) = ∑ v (n)
n = 0
1
n = 0
1

1 2 1 ×2 = 2

2 x (n) 1 X 3 y 1( n) 1 ⇒ 2 ×3 = 6 v 1( n) 1 ×1 = 1

1 1 1 ×1 = 1

T h e s um of s a m p le s o f v 1(n ) g ive s rx y (1)

F ig 5 : C o m p u ta tio n of rxy (1 ).
∴ rxy (1) = 1 + 2 + 6 + 1 = 10

3 3 3
When m = 2 ; rxy (2) = ∑ x(n) y((n − 2))
n = 0
4 = ∑ x(n) y (n) = ∑ v (n)
n = 0
2
n = 0
2

1 1 1×1 = 1

2 x (n) 1 X 2 y 2 (n) 1 ⇒ 2 ×2 = 4 v 2 ( n) 1 ×1 = 1

1 3 1 ×3 = 3
T he su m o f sa m ples of v 2 (n) g iv e s rxy (2)
F ig 6 : C o m pu ta tio n o f rxy (2 ).
∴ rxy (2) = 1 + 1 + 4 + 3 = 9

3 3 3
When m = 3 ; rxy (3) = ∑ x(n) y((n − 3))
n = 0
4 = ∑ x(n) y (n) = ∑ v (n)
n = 0
3
n = 0
3

1 1 1 ×1 = 1

2 x (n) 1 X 1 y 3 (n) 3 ⇒ 2 ×1 = 2 v 3 (n) 1 ×3 = 3

1 2 1 ×2 = 2

T h e s um of s am ple s o f v 3 (n) g iv e s rxy (3)

F ig 7 : C o m p uta tio n o f rxy (3 ).
∴ rx y (3) = 3 + 1 + 2 + 2 = 8

\ rxy (m) = {8, 10, 9, 8}

-­

Method 2 : Circular Correlation Using Tabular Array

The given sequences are represented in the tabular array as shown below. Here the shifted sequences
ym(n) are periodically extended with a periodicity of N = 4. Let rxy (m) be the sequence obtained by circular
correlation of x(n) and y(n). Each sample of rxy (m) is given by the equation,

N − 1 N − 1
rxy (m) = ∑
n = 0
x(n) y((n − m))N = ∑
n = 0
x(n) y m (n), where y m (n) = y((n − m))N

Note : The boldfaced numbers are samples obtained by periodic extension.

Chapter 2 - Discrete Time Signals and Systems 2. 112

n 0 1 2 3 4 5 6

x(n) 1 1 2 1

y(n) 2 3 1 1

y0((n – 0))4 = y0(n) 2 3 1 1 2 3 1

y1((n – 1))4 = y1(n) 1 2 3 1 1 2 3

y2((n – 2))4 = y2(n) 1 1 2 3 1 1 2

y3((n – 3))4 = y3(n) 3 1 1 2 3 1 1

To determine a sample of rxy (m) at m = q, multiply the sequence, x(n) and y q (n), to get a product sequence
x(n) xq(n) [i.e., multiply the corresponding elements of the row x(n) and yq(n)]. The sum of all the samples of the
product sequence gives rxy (m).
3
When m = 0 ; rxy (0) = ∑ x(n) y
n=0
0 (n)

= x(0) y 0 (0) + x(1) y 0 (1) + x(2) y 0 (2) + x(3) y 0 (3)

= 1× 2 + 1× 3 + 2 × 1+ 1× 1 = 2 + 3 + 2 + 1 = 8

The samples of rxy (m) for other values of m are calculated as shown for m = 0.
3
When m = 1; rxy (1) = ∑
n=0
x(n) y1(n) = 1 + 2 + 6 + 1 = 10

3
When m = 2; rxy (2) = ∑ x(n) y 2(n) = 1 + 1 + 4 + 3 = 9
n =0
3
When m = 3; rxy (3) = ∑
n =0
x(n) y3(n) = 3 + 1 + 2 + 2 = 8

l
∴ rxy (m) = 8, 10, 9, 8 q
A
Method 3 : Circular Correlation Using Matrices

The sequence rxy (m) can be arranged as a column vector of order N ´ 1 and using the samples of y(n) the
N ´ N matrix is formed as shown below. The product of the two matrices gives the sequence rxy (m).

LMy(0) y(1) y(2) y(3) OP LMx(0)OP LMr xy (0) OP

MMy(3) y(0) y(1) y(2)P MMx(1) PP =
MMrxy (1) P
y(1) P (2) PP
MMy(y(12)) y(3)
y(2)
y(0)
y(3) y(0)PQ
P MMx(x(32))PP MMr
xy

(3) PQ
N N Q Nr
xy

LM2 3 1 1OP LM1OP LM8 OP

MM1 2 3 1 PP MM1PP = MM10PP
MMN13 11 2
1
3
2
PPQ MMN12PPQ MMN98 PPQ
\ rxy (m) = {8, 10, 9, 8}
-
2. 113 Digital Signal Processing

2.15. Summary of Important Concepts

1. The discrete signal is a function of a discrete independent variable.
2. In a discrete time signal, the value of discrete time signal and the independent variable time are discrete.
3. The digital signal is same as discrete signal except that the magnitude of the signal is quantized.
4. A discrete time sinusoid is periodic only if its frequency is a rational number.
5. Discrete time sinusoids whose frequencies are separated by an integer multiple of 2p are identical.
6. The sampling is the process of conversion of continuous time signal into discrete time signal.
7. The time interval between successive samples is called sampling time or sampling period.
8. The inverse of sampling period is called sampling frequency.
9. The phenomenon of high frequency component getting the identity of low-frequency component during
sampling is called aliasing.
10. For analog signal with maximum frequency Fmax, the sampling frequency should be greater than 2Fmax.
11. When sampling frequency Fs is equal to 2Fmax, the sampling rate is called Nyquist rate.
12. The signals that can be completely specified by mathematical equations are called deterministic signals.
13. The signals whose characteristics are random in nature are called nondeterministic signals.
14. A signal x(n) is periodic with periodicity of N samples if x(n + N) = x(n).
15. When a signal exhibits symmetry with respect to n = 0 then it is called an even signal.
16. When a signal exhibits antisymmetry with respect to n = 0, then it is called an odd signal.
17. When the energy E of a signal is finite and nonzero, the signal is called energy signal.
18. When the power P of a signal is finite and nonzero, the signal is called power signal.
19. For energy signals, the energy will be finite and average power will be zero.
20. For power signals the average power is finite and energy will be infinite.
21. A signal is said to be causal, if it is defined for n ³ 0.
22. A signal is said to be noncausal, if it is defined for both n ≤ 0 and n > 0.
23. A signal is said to be anticausal, if it is defined for n ≤ 0.
24. A discrete time system is a device or algorithm that operates on a discrete time signal.
25. When a system satisfies the properties of linearity and time invariance, it is called an LTI system.
26. When the input to a discrete time system is unit impulse d(n), the output is called impulse response, h(n).
27. In a static or memoryless system, the output at any instant n depends on input at the same time.
28. A system is said to be time invariant if its input-output characteristics do not change with time.
29. A linear system is one that satisfies the superposition principle.
30. A system is said to be causal if the output does not depends on future inputs/outputs.
31. When a system output at any time n depends on future inputs/outputs, it is called a noncausal system.
32. System is said to be BIBO stable if and only if every bounded input produces a bounded output.
33. When a system output at any time n depends on past outputs, it is called a recursive system.
34. A system whose output does not depends on past outputs is called a nonrecursive system.
35. The convolution of N1 and N2 sample sequences produce a sequence consisting of N1+N2–1 samples.
36. In an LTI system, response for an arbitrary input is given by convolution of input with impulse response.
37. The output sequence of circular convolution is also periodic sequence with periodicity of N samples.
38. The inverse system is used to recover the input from the response of a system.
39. The process of recovering the input from the response of a system is called deconvolution.
40. The correlation of two different sequences is called crosscorrelation.
41. The correlation of a sequence with itself is called autocorrelation.
Chapter 2 - Discrete Time Signals and Systems 2. 114

2.16. Short Questions and Answers

Q2.1 Perform addition of the discrete time signals, x1(n) = {2, 2, 1, 2} and x2(n) = {–2, –1, 3, 2}.
Solution

In addition operation, the samples corresponding to same value of n are added.

When n = 0, x1(0) + x2(0) = 2 + (–2) = 0 When n = 2, x1(2) + x2(2) = 1 + 3 = 4
When n = 1, x1(1) + x2(1) = 2 + (–1) = 1 When n = 3, x1(3) + x2(3) = 2 + 2 = 4
\ x1(n) + x2(n) = {0, 1, 4, 4}

Q2.2 Perform multiplication of discrete time signals, x1(n) = {2, 2, 1, 2} and x2(n) = {–2, –1, 3, 2}.
Solution

In multiplication operation, the samples corresponding to same value of n are multiplied.

When n = 0, x1(0) × x2(0) = 2 ×(–2) = –4 When n = 2, x1(2) × x2(2) = 1 × 3 = 3
When n = 1, x1(1) × x2(1) = 2 ×(–1) = –2 When n = 3, x1(3) × x2(3) = 2 × 2 = 4
\ x1(n) × x2(n) = {–4, –2, 3, 4}

Q2.3 Express the discrete time signal x(n) as a summation of impulses.

If we multiply a signal x(n) by a delayed unit impulse d(n – m), then the product is x(m), where x(m)
is the signal sample at n = m [because d(n – m) is 1 only at n = m and zero for other values of n].
Therefore, if we repeat this multiplication over all possible delays in the range –¥ < m < ¥ and sum
all the product sequences, then the result will be a sequence that is equal to the sequence x(n).
\ x(n) = ... x(–2) d(n + 2) + x(–1) d(n + 1) + x(0) d(n) + x(1) d(n – 1) + x(2) d(n – 2) + ...
+∞
= ∑ x(m) δ(n − m)
m =−∞

Q2.4 What are the basic elements used to construct the block diagram of discrete time system?
The basic elements used to construct the block diagram of discrete time system are adder, constant
multiplier and unit delay element.
x 1 (n) x 1(n ) + x 2 (n ) x 1 (n) ax 1 (n) x (n) x (n − 1)
−1
+ a z

x 2 (n)

F ig b : C o n sta nt m ultip lier. F ig c : U nit d ela y elem en t.

F ig a : A d d er.
2 4
Q2.5 Let, x(n) = {1, 2, 3, 4}, be one period of a periodic
sequence. What is x(n – 2, mod4)?
The x(n) can be represented on the circle as shown 3 x (n) 1 1 x (n − 2 , m od4) 3
in fig Q2.5a. The x(n – 2, mod4) is circularly shifted
sequence of x(n) by two units of time as shown in
fig Q2.5b.(Here, mod 4 stands for periodicity of 4). 4 2

\ x(n – 2, mod4) = {3, 4, 1, 2} F ig Q 2.5 a . F ig Q 2.5 b .

2. 115 Digital Signal Processing
Q2.6 Why linear convolution is important in digital signal processing ?
The response or output of an LTI discrete time system for any input x(n) is given by linear
convolution of the input x(n) and the impulse response h(n) of the system. (This means that if
the impulse response of a system is known, then the response of the system for any input can
be determined by convolution operation.)
Q2.7 In y(n) = x(n)* h(n), how will you determine the start and end point of y(n)? What will be
the length of y(n)?
Let, length of x(n) be N1 and starts at n = nx. Let, length of h(n) be N2 and starts at n = nh.
Now, y(n) will start at n = nx + nh
y(n) will end at n = (nx + nh) + (N1 + N2 – 2)
The length of y(n) is N1 + N2 – 1.
Q2.8 What is zero padding? Why is it needed?
Appending zeros to a sequence in order to increase the size or length of the sequence is called
zero padding.
In circular convolution, when the two input sequences are of different size, then they are
converted to equal size by zero padding.
Q2.9 List the differences between linear convolution and circular convolution.
Linear convolution Circular convolution
1. The length of the input sequence 1. The length of the input sequences
can be different should be same.
2. Zero padding is not required. 2. If the length of the input sequences are
different, then zero padding is required.
3. The input sequences need not be 3. Atleast one of the input sequence
periodic. should be periodic or should be
periodically extended.
4. The output sequence is nonperiodic. 4. The output sequence is periodic.
The periodicity is same as that of input
sequence.
5. The length of output sequence will 5. The length of the input and output
be greater than the length of input sequences are same.
sequences.

Q2.10 Perform the circular convolution of the two sequences x1(n) = {1, 2, 3} and x2(n) = {4, 5, 6}.
Solution
Let x3(n) be the sequence obtained from circular convolution of x1(n) and x2(n). The sequence
x1(n) can be arranged as a column vector of order 3 ´1 and using the samples of x2(n) a 3 ´ 3 matrix
is formed as shown below. The product of two matrices gives the sequence x3(n).

LMx (0)
2 x2 (2) x 2 (1) OP LMx (0)OP LMx (0)OP
1 3 LM4 6 5 OP LM1OP LM31OP
⇒
MMNxx (1)
2
2 (2)
x2 (0) x 2 (2)
x2 (1)
P Mx (1)P = MMNxx (1)
1
x (0) PQ MN x (2) PQ
2 1
3
P
(2) PQ
3
MMN56 4 6
5
P M2P = MMN2831PPQ
4PQ MN 3PQ

\ x3(n) = x1(n) * x2(n) = {31, 31, 28}.

Chapter 2 - Discrete Time Signals and Systems 2. 116
Q2.11 Perform the linear convolution of the two sequences x1(n) = {1, 2} and x2(n) = {3, 4} via circular
convolution.
Solution
Let x3(n) be the sequence obtained from linear convolution of x1(n) and x2(n). The length of
x3(n) will be 2 + 2 – 1 = 3. Let us convert x1(n) and x2(n) into three sample sequences by padding
with zeros as shown below.
x1(n) = {1, 2, 0} and x2(n) = {3, 4, 0}
Now the circular convolution of x1(n) and x2(n) will give x3(n). The sequence x1(n) is arranged as
a column vector and using the sequence x2(n), a 3 ´3 matrix is formed as shown below. The
product of the two matrices gives the sequence x3(n).
LMx (0)
2 x 2 (2) x 2 (1) OP LMx (0)OP LMx (0)OP
1 3 LM3 0 4 OP LM1OP LM 3 OP
⇒
MMNxx (1)
2
2 (2)
x 2 (0) x 2 (2)
x 2 (1)
P Mx (1)P = MMNxx (1)
1
x (0) PQ MN x (2) PQ
2 1
3
P
(2) PQ
3
MMN40 3 0
4
P M2P = MMN108PPQ
3PQ MN0PQ

\ x3(n) = x1(n) * x2(n) = {3, 10, 8}

Q2.12 Compare the overlap add and overlap save method of sectioned convolutions.
Overlap add method Overlap save method
1. Linear convolution of each section of 1. Circular convolution of each section of
longer sequence with smaller sequence longer sequence with smaller sequence
is performed. is performed. (after converting them to
the size of output sequence).
2. Zero padding is not required. 2. Zero padding is required to convert
the input sequences to the size of
output sequence.
3. Overlapping of samples of input 3. The N2–1 samples of an input section of
sections are not required. longer sequence is overlapped with next
input section.
4. The overlapped samples in the output 4. Depending on method of overlapping
of sectioned convolutions are added input samples, either last N2–1 samples
to get the overall output. or first N2–1 samples of output sequence
of each sectioned convolution are
discarded.

Q2.13 In what way zero padding is implemented in overlap save method?

In overlap save method, the zero padding is employed to convert the smaller input sequence to
the size of the output sequence of each sectioned convolution. The zero padding is also employed
to convert either the last section or the first section of the longer input sequence to the size of the
output sequence of each sectioned convolution. (This depends on the method of overlapping
input samples).
Q2.14 List the similarities and differences in convolution and correlation of two sequences.
Similarities
1. Both convolution and correlation operation involves shifting, multiplication and summation of
product sequence.
2. Both convolution and correlation operation produce same size of output sequence.
2. 117 Digital Signal Processing
Differences
1. Correlation operation does not involve change of index and folding of one of the input
sequence.
2. The convolution operation is commutative, [i.e., x(n) * y(n) = y(n) * x(n)], whereas in correlation
operation in order to satisfy commutative property, while performing correlation of y(n) and
x(n), the shifting has to performed in opposite direction to that of performing correlation of x(n)
and y(n).
Q2.15 Let rxy(m) be the correlation sequence obtained by correlation of x(n) and y(n), how will you
determine the start and end point of rxy(m)? What will be the length of rxy(m) ?
Let, length of x(n) be N1 and starts at n = n1. Let length of y(n) be N2 and starts at n = n2.
Now, rxy(m) will start at mi = n1 – (n2 + N2 – 1)
rxy(m) will end at mf = mi + (N1 + N2 – 2)
The length of rxy(m) is N1 + N2 – 1.
Q2.16 What are the differences between crosscorrelation and autocorrelation?
1. Crosscorrelation operation is correlation of two different sequences, whereas autocorrelation
is correlation of a sequence with itself.
2. Autocorrelation operation is an even function, whereas crosscorrelation is not an even function.
Q2.17 Perform the correlation of the two sequences, x(n) = {1, 2, 3} and y(n) = {2, 4, 1}.
Solution
Given that, x(n) = {1, 2, 3 } and y(n) = {2, 4, 1}. \ y(–n) = {1, 4, 2 }
- - -
The sequence x(n) is arranged as a column and the folded sequence y(–n) is arranged as a row as
shown below. The elements of the two dimensional array are obtained by multiplying
the corresponding row element with column element. The sum of the diagonal elements gives the
samples of the crosscorrelation sequence, rxy(m).
y(−n) y(−n)
x(n) 1 4 2 x(n) 1 4 2

1 1 ×1 1 ×4 1 ×2 1 1 4 2
⇒
2 2 ×1 2 ×4 2 ×2 2 2 8 4

3 3×1 3×4 3×2 3 3 12 6

rxy (–2) = 1 ; rxy (–1) = 2 + 4 = 6 ; rxy(0) = 3 + 8 + 2 = 13 ; rxy(1) = 12 +4 = 16 ; rxy(2) = 6 ;

\ rxy(m) = {1, 6, 13, 16, 6}
-
Q2.18 Perform the circular correlation of the two sequences, x(n) = {1, 2, 3} and y(n) = {2, 4, 1}.
Solution
Let rxy (m) be the sequence obtained from circular correlation of x(n) and y(n). The sequence x(n)
can be arranged as a column vector of order 3 ´1 and using the samples of y(n) a 3 ´3 matrix is
formed as shown below. The product of two matrices gives the sequence rxy (m).

LMy(0) y(1) y(2) OP LMx(0)OP LMrxy (0)

OP LM2 4 1 OP LM1OP LM13OP
⇒
MMNy(2)
y(1)
y(0) y(1)
y(2)
P Mx(1)P = MMrr
y(0) PQ MN x(2) PQ
xy (1)
(2) PQ
P MMN41 2 4
1
P M2P = MMN1712PPQ
2PQ MN 3PQ
N xy

\ rxy (m) = {13, 17, 12}

Chapter 2 - Discrete Time Signals and Systems 2.118
Q2.19 Perform circular autocorrelation of the sequence, x(n) = {1, 2, 3, 4}.

Solution

Let rxx ( m) be the sequence obtained from circular autocorrelation of x(n). The sequence x(n) can
be arranged as a column vector of order 4´1 and again by using the samples of x(n) a 4´4 matrix
is formed as shown below. The product of two matrices gives the sequence rxx ( m) .
LMx(0) x(1) x(2) x(3) OP LMx(0)OP LMr xx (0) OP LM1 2 3 OP LM1OP LM 30OP
4

MMx(3)
x(2)
x(0)
x(3)
x(1)
x(0)
x(2)
x(1)
PP MMx(1) P = MMrr
x(2) P
xx (1)
xx (2) P
P ⇒ MM43 1
4
2
1
3
2
PP MM23PP = MM 2422PP
MNx(1) x(2) x(3) x(0) PQ MNx(3)PQ MNr xx (3) PQ MN2 3 4 1 PQ MN4PQ MN 24PQ
\ rxx ( m) = {30, 24, 22, 24 }

Q2.20 What is the difference between circular crosscorrelation and circular autocorrelation?
Circular crosscorrelation operation is circular correlation of two different sequences, whereas
circular autocorrelation is circular correlation of a sequence with itself.

2.17 MATLAB Programs

Program 2.1
Write a MATLAB program to generate the standard discrete time signals unit
impulse, unit step and unit ramp signals.

%******************* program to plot some standard signals

n=-20 : 1 : 20; %specify the range of n

%******************* unit impulse signal

x1=1;
x2=0;
x=x1.*(n==0)+x2.*(n~=0); %generate unit impulse signal
subplot(3,1,1);stem(n,x); %plot the generated unit impulse signal
xlabel(‘n’);ylabel(‘x(n)’);title(‘unit impulse signal’);

%******************* unit step signal

x1=1;
x2=0;
x=x1.*(n>=0)+x2.*(n<0); %generate unit step signal
subplot(3,1,2);stem(n,x); %plot the generated unit step signal
xlabel(‘n’);ylabel(‘x(n)’);title(‘unit step signal’);

%******************* unit ramp signal

x1=n;
x2=0;
x=x1.*(n>=0)+x2.*(n<0); %generate unit ramp signal
subplot(3,1,3);stem(n,x); %plot the generated unit ramp signal
xlabel(‘n’);ylabel(‘x(n)’);title(‘unit ramp signal’);
OUTPUT
The output waveforms of program 2.1 are shown in fig P2.1.
2. 119 Digital Signal Processing

F ig P 2 .1 : O u tp u t w av efo rm s o f pro g ra m 2 .1. F ig P 2 .2 : O u tp u t w av efo rm s o f pro g ra m 2 .2.

Program 2.2
Write a MATLAB program to generate the standard discrete time signals exponential
and sinusoidal signals.

%******************* program to plot some standard signals

n=-20 : 1 : 20; %specify the range of n

%******************* exponential signal

A=0.95;
x=A.^n; %generate exponential signal
subplot(2,1,1);stem(n,x); %plot the generated exponential signal
xlabel(‘n’);ylabel(‘x(n)’);title(‘exponential signal’);

%******************* sinusoidal signal

N=20; %declare periodicity
f=1/20; %compute frequency
x=sin(2*pi*f*n); %generate sinusoidal signal
subplot(2,1,2);stem(n,x); %plot the generated sinusoidal signal
xlabel(‘n’);ylabel(‘x(n)’);title(‘sinusoidal signal’);

OUTPUT
The output waveforms of program 2.2 are shown in fig P2.2.

Program 2.3
Write a MATLAB program to find the even and odd parts of the signal x(n)=0.8n.

%To find the even and odd parts of the signal, x(n)= 0.8^n

n= -5 :1 :5; %specify the range of n

A=0.8;
x1=A.^n; %generate the given signal
x2=A.^(-n); %generate the folded signal
Chapter 2 - Discrete Time Signals and Systems 2.120
if(x2==x1)
disp(‘“The given signal is even signal”’);
else if (x2==(-x1))
disp(‘“The given signal is odd signal”’);
else
disp(‘“The given signal is neither even nor odd signal”’);
end
end

xe=(x1+x2)/2; %compute even part

xo=(x1-x2)/2; %compute odd part

subplot(2,2,1);stem(n,x1);
xlabel(‘n’);ylabel(‘x1(n)’);title(‘signal x(n)’);

subplot(2,2,2);stem(n,x2);
xlabel(‘n’);ylabel(‘x2(n)’);title(‘signal x(-n)’);

subplot(2,2,3);stem(n,xe);
xlabel(‘n’);ylabel(‘xe(n)’);title(‘even part of x(n)’);

subplot(2,2,4);stem(n,xo);
xlabel(‘n’);ylabel(‘xo(n)’);title(‘odd part of x(n)’);

F ig P 2 .3 : O u tp u t w av efo rm s o f pro g ra m 2 .3. F ig P 2 .4 : O u tp u t w av efo rm s o f pro g ra m 2 .4.

OUTPUT

“The given signal is neither even nor odd signal”

The output waveforms of program 2.3 are shown in fig P2.3.

Program 2.4
Write a MATLAB program to perform amplitude scaling and time shift on the
signal x(n) = 1+n; for n = 0 to 2.

Program to declare the given signal as function y(n)

% declare the given signal as function y(n)

function x = y(n)
x=(1.0 + n).*(n>=0 & n<=2);
2. 121 Digital Signal Processing
Note: The above program should be stored as a separate file in the current
working directory

Program to perform amplitude scaling and time shift on y(n)

%To Perform Amplitude scaling and Time shift on signal x(n)=1+n;

%for n= 0 to 2
%include y.m file in current work directory which declare given signal as
%function y(n)

n=-5:1:5; %specify range of n

y0 =y(n); %assign the given signal as y0

y1 =1.5*y(n); %compute the amplified version of x(n)
y2 =0.5*y(n); %compute the attenuated version of x(n)
y3 =y(n-2); %compute the delayed version of x(n)
y4 =y(n+2); %compute the advanced version of x(n)

%plot the given signal and amplitude scaled signal

subplot(2,3,1);stem(n,y0);
xlabel(‘n’);ylabel(‘x(n)’);title(‘Signal x(n)’);
subplot(2,3,2);stem(n,y1);
xlabel(‘n’);ylabel(‘x1(n)’);title(‘Amplified signal 1.5x(n)’);
subplot(2,3,3);stem(n,y2);
xlabel(‘n’);ylabel(‘x2(n)’);title(‘Attenuated signal 0.5x(n)’);

%plot the given signal and time shifted signal

subplot(2,3,4);stem(n,y0);
xlabel(‘n’);ylabel(‘x(n)’);title(‘Signal x(n)’);
subplot(2,3,5);stem(n,y3);
xlabel(‘n’);ylabel(‘x3(n)’);title(‘Delayed signal x(n-2)’);
subplot(2,3,6);stem(n,y4);
xlabel(‘n’);ylabel(‘x4(n)’);title(‘Advanced signal x(n+2)’);

OUTPUT
The input and output waveforms of program 2.4 are shown in fig P2.4.

Program 2.5
Write a MATLAB program to perform convolution of the following two discrete
time signals.
x1(n)=1; 1<n<10 x2(n)=1; 2<n<10
%******************Program to perform convolution of two signals
%******************x1(N)=1; n= 1 to 10 and x2(n)=1; n= 2 to 10

n = 0 : 1 : 15; %specify range of n

x1=1.*(n>=1 & n<=10); %generate signal x1(n)

x2=1.*(n>=2 & n<=10); %generate signal x2(n)
N1=length(x1);
N2=length(x2);
x3=conv(x1,x2); %perform convolution of signals x1(n) and x2(n)
n1=0 : 1 : N1+N2-2; %specify range of n for x3(n)
Chapter 2 - Discrete Time Signals and Systems 2.122
subplot(3,1,1);stem(n,x1);
xlabel(‘n’);ylabel(‘x1(n)’);
title(‘signal x1(n)’);

subplot(3,1,2);stem(n,x2);
xlabel(‘n’);ylabel(‘x2(n)’);
title(‘signal x2(n)’);

subplot(3,1,3);stem(n1,x3);
xlabel(‘n’);ylabel(‘x3(n)’);
title(‘signal, x3(n) =
x1(n)*x2(n)’);

OUTPUT
F ig P 2 .5 : O u tp u t w av efo rm s o f pro g ra m 2 .5.
The input and output waveforms of
program 2.5 are shown in fig P2.5.

2.18 Exercises
I. Fill in the blanks with appropriate words
1. A signal x(n) may be shifted in time by m units by replacing the independent variable n by _______.
2. The _______ of a signal x(n) is performed by changing the sign of the time base n.
3. If the average power of a signal is finite then it is called _______.
4. The smallest value of N for which x(n + N) = x(n) is true is called _______.
5. In a discrete time signal x(n), if x(n) = x(–n) then it is called _______ signal.
6. In a discrete time signal x(n), if x(–n) = –x(n) then it is called _______ signal.
7. The output of the system with zero input is called _______.
8. A discrete time system is _______ if it obeys the principle of superposition.
9. A discrete time system is _______ if its input-output relationship do not change with time.
10. The response of an LTI system is given by _______ of input and impulse response.
11. If the output of a system depends only on present input then it is called _______.
12. A system is said to be _______ if the output does not depends on future inputs and outputs.
13. An LTI system is causal if and only if its impulse response is _______ for negative values of n.
14. When a system output at any time n depends on past output values, it is called _______ system.
15. An N-point sequence is called _______ if it is symmetric about point zero on the circle.
16. An N-point sequence is called _______ if it is antisymmetric about point zero on the circle.
17. The _______ is called aperiodic convolution.
18. The _______ is called periodic convolution.
19. Appending zeros to a sequence in order to increase its length is called _______.
20. The two methods of sectioned convolutions are _______ and _______method.
2. 123 Digital Signal Processing
21. In _______ method of sectioned convolution, overlapped samples of output sequences are _______.
22. In _______ method, the overlapped samples in one of the output sequences are discarded.
23. The correlation of two different discrete time sequences is called _______ .
24. The cascade of a system and its inverse is _______.
25. The process of recovering the input from the response of a system is called ______ .

Answers
1. n – m 8. linear 15. even 22. overlap save
2. folding 9. time invariant 16. odd 23. cross correlation
3. power signal 10. convolution 17. linear convolution 24. identity system
4. fundamental period 11. memoryless or static 18. circular convolution 25. deconvolution
5. symmetric 12. causal 19. zero padding
6. antisymmetric 13. zero 20. overlap add, overlap save
7. natural response 14. recursive 21. overlap add, added

II. State whether the following statements are True/False

1. The discrete signals are continuous function of an independent variable.
2. In digital signal the magnitudes of the signal are unquantized.
3. A discrete time signal x(n) is defined for noninteger values of n.
4. An impulse signal has a nonzero sample only for one value of n.
5. When we multiply a discrete time signal by unit step signal, the signal is converted to one-sided signal.
6. Shifting a signal to left is called delay and shifting to right is called advance.
7. Any discrete time signal can be expressed as a summation of impulses.
8. Periodic signals are power signals.
9. When the energy of a signal is infinite, it is called energy signal.
10. The output of a system for impulse input is called impulse response.
11. A system can be realized in real time only if it is noncausal and stable.
12. Dynamic systems does not require memory but static systems require memory.
13. A system is time invariant if the response to a shifted version of the input is identical to a shifted version
of the response based on the unshifted input.
14. An LTI system is unstable if the impulse response is absolutely summable.
15. A system whose output depends only on the present and past input is called a recursive system.
16. The circular shift of an N-point sequence is equivalent to a linear shift of its periodic extension.
17. For an N-point sequence represented on a circle, the time reversal is obtained by reversing its sample
about the point zero on the circle.
18. When a nonperiodic N-point sequence is represented on a circle then it becomes periodic with
periodicity N.
19. In linear convolution the length of the input sequences should be same.
20. In circular convolution the length of the input sequences need not be same.
21. In circular correlation the length of the input and output sequences are same.
Chapter 2 - Discrete Time Signals and Systems 2. 124
∞ ∞
22. The correlation operation, rxy (m) = ∑ x(n) y(n − m) is not same as rxy (m) = ∑ x(n + m) y(n) .
n =−∞ n =−∞
23. The cross correlation sequence rxy(m) is folded version of ryx(m).
24. The inverse systems exist for all LTI systems.
25. The final value of m in autocorrelation sequence of N-point sequence is, mf = mi + (2N – 1).
Answers
1. False 6. False 11. False 16. True 21. True
2. False 7. True 12. False 17. True 22. False
3. False 8. True 13. True 18. True 23. True
4. True 9. False 14. False 19. False 24. True
5. True 10. True 15. False 20. False 25. False

III. Choose the right answer for the following questions

x(n − 1)
1. x(n) = with initial condition x(0) = –1, gives the sequence,
4

a) x(n) =
FG 1 IJ n
b) x(n) = −
FG 1 IJ n
c) x(n) =
FG 1 IJ −n
d) x(n) =
FG −1IJ −n

H 4K H 4K H 4K H4K
2. The process of conversion of continuous time signal into discrete time signal is known as,
a) aliasing b) sampling c) convolution d) none of the above
3. If Fs is sampling frequency then the relation between analog frequency F and digital frequency f is,
F F F 2F
a) f = b) f = s c) f = d) f =
2Fs F Fs Fs

4. If Fs is sampling frequency then the highest analog frequency that can be uniquely represented in its
sampled version of discrete time signal is,
Fs 1
a) b) 2Fs c) Fs d)
2 Fs

5. The sampling frequency of the following analog signal, x(t) = 4 sin150pt + 2 cos50pt should be,
a) greater than 75 Hz b) greater than 150 Hz c) less than 150 Hz d) greater than 50 Hz
6. Which of the following signal is the example for deterministic signal?
a) step b) ramp c) exponential d) all of the above
7. For energy signals, the energy will be finite and the average power will be,
a) infinite b) finite c) zero d) cannot be defined
n
8. In a signal x(n), if 'n' is replaced by , then it is called,
3
a) upsampling b) folded version c) downsampling d) shifted version
9. The unit step signal u(n) delayed by 3 units of time is denoted as,
a) u(n + 3) = 1; n ≥ 3 b) u(3 − n) = 1; n ≥ 3 c) u(n − 3) = 1; n ≥ 3 d) u(3n) = 1; n > 3
= 0; n < 3 = 0; n < 3 = 0; n < 3 = 0; n < 3
2. 125 Digital Signal Processing
10. The zero input response (or) natural response is mainly due to,
a) Initial stored energy in the system b) Initial conditions in the system
c) Specific input signal d) both a and b
11. If x(n) = an u(n) is the input signal, then the particular solution yp(n) will be,
a) Kn an u(n) b) K an u(n)
c) K1 an u(n) + K2 an u(n) d) K a–n u(n)
12. The discrete time system, y(n) = x(n–3) – 4x(n–10) is a,
a) dynamic system b) memoryless system c) time varying system d) none of the above
13. An LTI discrete time system is causal if and only if,
a) h(n) ¹ 0 for n < 0 b) h(n) = 0 for n < 0 c) h(n) ¹ ¥ for n < 0 d) h(n) ¹ 0 for n > 0
14. Which of the following system is causal?

a) h(n) = n
FG 1 IJ n
u(n + 1) b) y(n) = x2(n) – x(n+1) c) y(n) = x(–n) +x(2n–1) d) h(n) = n
FG 1 IJ n
u(n)
H 2K H 2K
15. An LTI system is stable, if the impulse response is,
∞ ∞ ∞
a) ∑ h(n) = 0 b) ∑ h(n) < ∞ c) ∑ h(n) ≠ 0 d) either a or b
n= −∞ n= −∞ n= −∞

16. The system y(n) = sin[x(n)] is,

a) stable b) BIBO stable c) unstable d) none of the above
17. Two parallel connected discrete time systems with impulse responses h1(n) and h2(n) can be replaced
by a single equivalent discrete time system with impulse response,
a) h1(n) * h2(n) b) h1(n) + h2(n) c) h1(n) – h2(n) d) h1(n) * [h1(n) + h2(n)]
18. Sectioned convolution is performed if one of the sequence is very much larger than the other in order
to overcome,
a) long delay in getting output b) larger memory space requirment
c) both a and b d) none of the above
19. In overlap save method, the convolution of various sections are performed by,
a) zero padding b) linear convolution c) circular convolution d) both b and c
20. If x(n) is N1-point sequence,if y(n) is N 2-point sequence, if r xy(m) is the correlation sequence
starts at m = mi , then the value of m corresponding to last sample of rxy(m) is,
a) mf = mi + (N1 + N2 – 2) b) mf = mi + (2N – 2) c) mf = mi + (N1 + N2 – 1) d) mf = mi + (2N + 1)
21. For a system, y(n) = nx(n), the inverse system will be,

a) y 1
nej b) 1 y n
n
bg c) ny(n) d) n –1y(n)

22. For a system y(n) = x(n–3) the impulse response of the system and the inverse system will be ––––––
and –––––– respectively.
a) h(n) = d(n + 3), x(n) = y(n – 3) b) h( n) = δ(3n), x(n) = y n ej
3
c) h(n) = d(n – 3), x(n) = y(n + 3) d) h(n) = d(n + 3), x(n) = y(3n)
Chapter 2 - Discrete Time Signals and Systems 2. 126
23. The circular correlation rx 1 x 2 (q) of the sequence x1(n) and x2(n) of length 'N' can be defined by the
equation,
∞ N −1
a) ∑ x1(n) x2 (n − q) b) ∑ x1(n) x∗2 (n − q)
n= −∞ n=0
N −1 ∞
c) ∑ x1(n) x∗2 b(n − q)gN d) ∑ x1(n) x∗2 b(n − q)gN
n=0 n=−∞

24. The evaluation of correlation involves,

a) shifting, rotating and summation b) shifting, multiplication and summation
c) change of index, folding and summation d) change of index, folding, shifting & multiplication
25. The circular correlation of N-point sequences is evaluated in the range,
a) – N < m < N b) –N<m<0 c) 0<m<N d) 0 < m < N– 1

Answers
1. b 6. d 11. b 16. a 21. b
2. b 7. c 12. a 17. b 22. c
3. c 8. a 13. b 18. c 23. c
4. a 9. c 14. d 19. c 24. b
5. b 10. a 15. d 20. a 25. d

IV. Answer the following questions

1. Define discrete and digital signal.
2. Explain briefly, the various methods of representing discrete time signal with examples.
3. Define sampling and aliasing.
4. What is Nyquist rate?
5. State sampling theorem.
6. Define the impulse and unit step signal.
7. Express the discrete time signal x(n) as a summation of impulses.
8. How will you classify the discrete time signals?
9. What are energy and power signals?
10. When a discrete time signal is called periodic?
11. What is discrete time system?
12. What is impulse response? Explain its significance.
13. Write the difference equation governing the Nth order LTI system.
14. Write the expression for discrete convolution.
15. List the various methods of classifying discrete time systems.
16. Define time invariant system.
17. What is linear and nonlinear systems?
18. What is the importance of causality?
19. What is BIBO stability? What is the condition to be satisfied for stability?
20. What are FIR and IIR systems?
21. Write the convolution sum formula for FIR and IIR systems.
22. What are recursive and nonrecursive systems? Give examples.
2. 127 Digital Signal Processing
23. Write the properties of linear convolution.
24. Prove the distributive property of linear convolution.
25. What are the two ways of interconnecting LTI systems?
26. Define circular convolution.
27. What is the importance of linear and circular convolution in signals and systems?
28. How will you perform linear convolution via circular convolution?
29. What is sectioned convolution? Why is it performed?
30. What are the two methods of sectioned convolution?
31. What is inverse system? What is its importance?
32. Define deconvolution.
33. Define cross correlation and autocorrelation?
34. What are the properties of correlation?
35. What is circular correlation?
V. Solve the following problems
E2.1 Determine whether the following signals are periodic or not. If periodic, find the fundamental
period.

a) x(n) = sin
5π
n+6
FG IJ
b) x(n) = sin
7n
+π
FG IJ
c) x(n) = cos
4πn FG IJ
8 H K 3 H K 12 H K
d) x(n) = cos
π 2
n
FG IJ
e) x(n) = e j9 n f) x(n) = 4 sin
3πn
+ 5 cos
3πn
32 H K 2 4
E2.2 Determine the even and odd parts of the signals.
π
1 −j n
a) x(n) = 2n b) x(n) = 8e 6 l
c) x(n) = 6, 4, 2, 2 q
a
A
E2.3 a) Consider the analog signal x(t) = 2 sin80pt. If the sampling frequency is 60 Hz, find the sampled
version of discrete time signal x(n). Also find an alias frequency corresponding to Fs = 60 Hz.
b) Consider the analog signals, x1 (t) = 4 cos2π (30t) and x2 (t) = 4 cos 2π (5t). Find a sampling
frequency so that 30 Hz signal is an alias of 5 Hz signal.
c) Consider the analog signal, x(t) = 3 sin40π t − sin100π t + 2cos 50π t. Determine the minimum
sampling frequency and the sampled version of analog signal at this frequency. Sketch the
waveform and show the sampling points. Comment on the result.
E2.4 Determine whether the following signals are energy or power signals.

a) x(n) =
FG 5 IJ n
u( n)
FG
b) x(n) = cos
3π IJ
n
H 9K H 4 K c) x(n) = u(2n) d) x(n) = 2 u(3 – n)

E2.5 Construct the block diagram and signal flow graph of the discrete time systems whose input-
output relations are described by the following difference equations.
a) y(n) = 2y(n – 1) + 2.1 x(n – 1) + 0.5 x(n – 2)
b) y(n) = 1.6 x(n –2) + 0.7 x(n) + 3y(n – 1) + 0.3y(n – 2)
E2.6 Determine the response of the discrete time systems governed by the following difference equations.
a) y(n) = 0.1y(n – 1) + x(n – 1) + 0.7x(n) ; x(n) = 2–n u(n) ; y(–1) = –1
b) y(n) + 2.1y(n – 1) + 0.2y(n – 2) = x(n) + 0.56x(n –1) ; x(n) = u(n) ; y(–2) = 1; y(–1) = –3
Chapter 2 - Discrete Time Signals and Systems 2. 128
E2.7 Test the following systems for time invariance.
a) y(n) = x(n + 1) + x(n + 2) b) y(n) = nax(n) c) y(n) = x2(n + 2) + C d) y(n) = (n –1) x2(n) + C
E2.8 Test the following systems for linearity.
a) y(n) = x2(n) + x3(n – 1) b) y(n) = bx(n + 2) + nex(n) c) y(n) = a x(n) + b x( n)
1 N M
d) y(n) = x(n) + e) y(n) = ∑b m x( n + m) + ∑ c m y( n + m)
x(n) m= −1 m=0

E2.9 Test the causality of the following systems.

a) y(n) = x(n) – x(–n – 2) + x(n –1) b) y(n) = a x(2n) + x(n2)
n n 4
c) y(n) = ∑ x( m) + ∑ x(2 m)
m= −1 m= −∞
d) y(n) = (0.3)n u(n + 2) e) y(n) = ∑ x( n − k )
k= −4

E2.10 Test the stability of the following discrete time systems.

a) y(n) = x2(n) + x(n + 1) b) y(n) = nx(n –1) c) h(n) = (0.4)n u(n + 3)
d) h(n) = (8)n u(4 – n) e) y(n) = x(n – 3)
E2.11 Determine the range of values of 'a' and 'b' for the stability of an LTI system with impulse response,

h(n) =
R|S( −4a) n
; n≥0
|T 2b −n
; n<0
E2.12 a) Determine the impulse response for the cascade of two LTI systems having impulse responses,

h1 (n) =
FG 1 IJ n
u(n) and h2 (n) = d(n − 3) h2 (n)
H7 K x (n )
b) Determine the overall impulse response + y (n )
of the interconnected discrete time
system shown in fig E2.12.
h1(n) + h 3 (n)

F ig E 2 .1 2.
Take, h1 (n) =
FG 1 IJ n
u(n) ; h2 (n) =
FG 1 IJ n
u(n) ; h3 (n) =
FG 1 IJ n
u(n)
H 3K H6K H 9K
E2.13 Determine the response of an LTI system whose impulse response h(n) and input x(n) are given by,
l
a) h( n) = 1, 4, 1, −2, 1 q , l
x( n) = 1, 3, 5, −1, −2 q
A A
b) h( n) =
RS1 ; 0≤n≤2
, x( n) = a n u( n); a <1
T0 ; n≥3
E2.14 Perform circular convolution of the two sequences,
l
a) x1 ( n) = 1, 2, −1, 1 ; q l
x2 ( n) = 2, 4, 6, 8 q
b) x ( n) = l0,
1 0.6, −1, 15
., 2 ; q x ( n) = l−2,
2 3, 0.2, 0.7, 0.8 q
E2.15 The input x(n) and impulse response h(n) of an LTI system are given by,
l
x(n) = −1, 1, −1, 1, −1, 1 ; q l
h(n) = −0.5, 0.5, −1, 0.5, −1, −2 q
A A
Find the response of the system using a) Linear convolution, b) Circular convolution.
E2.16 Perform linear convolution of the following sequences by,
a) Overlap add method b) Overlap save method
l
x( n) = 1, −1, 2, 1, −1, 2, +1, −1, +2 q ; l
h( n) = 2, 3, −1 q
2. 129 Digital Signal Processing
E2.17 Perform crosscorrelation of the sequences,
l
x( n) = −1, 2 3, −4, ; q l
h( n) = 2, −1, −3, q
A A
E2.18 Determine the autocorrelation sequence for x(n) = 1, 4, 3, −5, 2 . l q
A
E2.19 Find the inverse system for the following discrete time system,
n
y(n) = ∑ c p x(p − 2) ; for n ≥ 0
p =0

E2.20 A discrete time system is excited by an input x(n), and the response is, y(n) = 4, 3, 6, 7.5, 3, 30, − 8 . m r
A
l
If the impulse response of the system is h(n) = 2, 4, −2 , then what will be the input to the system? q
A
E2.21 Perform circular correlation of the sequence, x(n) = −1, 1, 2, 6 and y(n) = 4, −2, −1, 2 . l q l q
Answers
E2.1 a) periodic; N=16 b) nonperiodic c) periodic; N=6 d) periodic; N=32 e) nonperiodic. f) periodic; N=8

π
E2.2 a) xe ( n) =
1 −2n
a + a2n b) xe ( n) = 8 cos n l q
c) x e ( n) = 1, 1, 2, 6, 2, 1, 1
2 6 A
x (n) = l−1, − 1, − 2, 0, 2, 1, 1q
1 π o
xo (n) = a −2n − a 2n
2
xo (n) = − j8 sin n
6
A
4πn
E2.3 a) x( n) = 2 sin ; Alias frequency = 100Hz b) Fs = 25Hz
3
2 πn πn
c) Fs,min = 100 Hz ; x( nT) = 3sin + 2 cos (sin πn = 0, for integer n)
5 2
The component sin100pt will give always zero samples when
sampled at 100Hz for any value of n (Refer fig E2.3c).
E2.4 a) E = 1.435J ; P = 0 ; Energy signal.
b) E = ¥ ; P = 0.5W ; Power signal.
c) E = ¥ ; P = 0.25W ; Power signal.
d) E = ¥ ; P=2W ; Power signal.

E2.5 a)
x (n ) 2.1x(n −1) y (n ) F ig E 2 .3 c : S a m p lin g p o ints.
−1 2.1
z + +
)
)

−1
−2

(n
x (n

2y

F ig E 2 .5 a.1 : B lo ck −1
0.5

−1
z z
0.5 2
d ia g ra m . x (n ) −1
1 z 2.1 1 1 1 y (n )

−1 −1
z 0.5 2 z
F ig E 2 .5 a.2 : S ig n al flow grap h .
Chapter 2 - Discrete Time Signals and Systems 2. 130
E2.5 b)
x (n ) y (n ) x (n ) 1 0.7 1 1 y (n )
0.7 + +
−1 3 −1
z z
−1 −1
z + 3 z

0.
6

3
1.
−1
−1 z
z
−1
z 1.6 0.3 z
−1

F ig E 2 .5 b.2 : S ig n al flow grap h .

F ig E 2 .5 b.1 : B lo c k d ia g ra m .

E2.6
LM
a) y(n) = −2.775(0.1) n + 3.375
FG 1 IJ OP u(n)
n
b) y(n) = 0.47 − 0.02 ( −0.1) n + 6.65 ( −2) n u(n)
MN H 2 K PQ
E2.7 a) c) Time invariant b) d) Time variant
E2.8 a) e) Linear b) c) d) Nonlinear
E2.9 a) b) c) d) e) Noncausal
E2.10 a) c) d) e) Stable system b) Unstable system
1 1
E2.11 For stability, 0 < a < and 0 < b <
4 2
FG 1 IJ ( n − 3) LM F 1 I n
FG IJ + FG 3 IJ FG 1IJ OP u(n)
3 1
n n
E2.12 a) h( n) =
H 7K u(n − 3) b) h( n) = 4
MN GH 6 JK −
H K H 2 K H 3K PQ
2 9
n
E2.13 a) y( n) = l1, 7, 18, 20, −6, −16, 5, 3, −2 q b) y( n) = ∑ ak ; for n = 0, 1, 2
A k =0
n
= ∑ a k ; for n > 2
k = n−2

E2.14 a) x3 ( n) = 8, −6, 4, 14 l q l
b) x3 ( n) = 6.08, −0.55, 6.4, −4.28, 0.72 q
A A
E2.15 l
y( n) = 0.5, −1, 2, −2.5, 3.5, −1.5, 1, −0.5, −0.5, 1, −2 q
A
l
E2.16 a) Overlap add method : y( n) = 2, 1, 0, 9, −1, 0, 9, −1, 0, 7, −2 q
b) Overlap save method : y( n) = l*, *, 0, 9, −1, 0, 9, −1, 0, 7, −2q
E2.17 r ( m) = l3, −5, −13, 13, 10, −8q
xy
A
1
E2.18 r ( m) = l2, 3, − 11, − 9, 55, − 9, − 11, 3, 2q
xx E2.19 x( n) = [ y( n + 2) − y( n + 1)] ; for n ≥ −1
cn + 2
A with initial condition x( −2) = y(0)

E2.20 l
x( n) = 2, −2.5, 10, −18.75, 49 q E2.21 rxy (m) = 4, −8, −1, 29 l q
A
Solution for Exercise Problems E2. 1

Digital Signal Processing - A. Nagoor Kani Chapter 2 - Discrete Time Signals and Systems

Solution for Exercise Problems

E2.1. Determine whether the following signals are periodic or not. If periodic, find the fundamental period.

a) x(n) = sin
FG 5π n + 6IJ
H8 K
Solution

b g
Now, x n + N = sin
FG 5π bn + Ng + 6IJ = sin FG 5πn + 6 + 5πN IJ
H8 K H8 8 K

5πN
Since sin (q + 2pM) = sin q, for periodicity, should be integral multiple of 2p.
8

5πN
Let, = M × 2π , M and N are integers.
8
8 16
∴ N = M × 2π × = M
5π 5
16
N = M , if M = 5, 10, 15, 20 ..... N will be a integer.
5

When M = 5 , N = 16.

b
x n + N = sing FG 5π n + 6 + 5π × 16IJ = sin FG 5π n + 6 + 10πIJ = sinFG 5π n + 6IJ = x(n).
H8 8 K H8 K H8 K
\ x(n) is periodic.

Fundamental period is 16 samples.

b) x(n) = sin
FG 7n + π IJ
H3 K
Solution

b g
x n + N = sin
FG 7(n + N) + πIJ = cos
FG 7n + π + 7NIJ
H 3 K H3 3 K

7N
Since cos (q + 2pM) = cos q, for periodicity, should be equal to integral multiple of 2p.
3
7N 2π × 3 6π
Let, = M × 2π ⇒ N = M = M
3 7 7
Here, N cannot be an integer for any integer value of M, and so, x(n) will not be periodic.

c ) x(n) = cos
FG 4π n IJ
H 12 K
Solution

bg FG π nIJ
x n = cos
H3 K
F π I F nπ + Nπ IJ
xbn + Ng = cos G (n + N)J = cos G
H3 K H 3 3 K
Nπ
= 2πM ⇒ N = 6M
3
For M = 1, 2, 3, ..... N will be integer.

For M = 1, N = 6.

\ x(n) is periodic.

Fundamental period is 6 samples.

E2. 2 DSP, Chapter 2 -Discrete Time Signals and Systems

d) x(n) = cos
FG π n IJ
2
H 32 K
Solution

Given that, x(n) = cos

FG π n IJ
2
H 32 K
π
∴ x(n + N) = cos (n + N)2
32
π 2
= cos (n + N2 + 2nN)
32
Fπn 2 πN2 πN I
= cos GH 32 +
32
+
16
n JK
πN2 πN
Let, = 2πM1 Let, = 2πM2
32 16

∴ N = 8 M1 \ N = 32 M2

Now, N is integer for M1 = 12, 22, 32, 42 ..... Now, N is integer for M2 = 1, 2, 3, 4 .....

When M1 = 42 and M2 = 1, we get a common value for N as, N = 32.

F π n + π32 + π32 nI 2
When N = 32 ; x(n + N) = cos GH 32 32 16 JK 2

FF π I I
= cosG G n + 2πnJ + 16 × 2πJ
2
H H 32 K K
Fπ
= cosG n + 2πnJ
I 2
H 32 K For integer M,
π 2 cos(q + 2pM) = cosq
= cos n = x(n)
32
\ x(n) is periodic with fundamental period, N = 32 samples.

e) x(n) = e j9n

Solution

b g
x n + N = e j9(n+N) = e j9n . e j9N

Since, e j2 πM = 1
2π
Let, b9Ng = M × 2π ⇒ N=
9
M

For any integer value of M, N will not be an integer.

Hence x(n) is non periodic.

3πn 3πn
f) Given that, x(n) = 4 sin + 5 cos
2 4
Solution
3πn 3πn
Let, x1(n) = 4 sin Let, x2 (n) = 5 cos
2 4

b g
∴ x1 n + N1 = 4 sin
b
3 π n + N1 g b g
∴ x 2 n + N2 = 5 cos
b
3 π n + N2 g
2 4

= 4 sin
FG 3πn + 3πN IJ 1 .....(1) = 5 cos
FG 3πn + 3πN IJ 2 .....(2)
H2 2 K H4 4 K
3πN1 4 3πN2 8
Let, = 2πM1 ⇒ N1 = M1 Let, = 2πM2 ⇒ N2 = M2
2 3 4 3
Let, M1 = 3 ; \ N1 = 4 Let, M2 = 3 ; \ N2 = 8
Solution for Exercise Problems E2. 3
substitute N1 = 4 in equation (1), substitute N2 = 8 in equation (2),

b
∴ x1 n + N1 = 4 sin g FG 3πn + 3π × 4IJ FG 3πn + 3π × 8IJ
H2 2 K b g
∴ x 2 n + N2 = 5 cos
H4 4 K
F 3πn + 3 × 2πIJ
= 4 sin G F 3πn + 3 × 2πIJ
H2 K = 5 cos G
For integer M, For integer M, H4 K
sin(q + 2pM) = sinq 3πn cos(q + 2pM) = cosq 3πn
= 4 sin = x1(n) = 5 cos = x 2 (n)
2 4
\ x1(n) is periodic with fundamental period, N1 = 4 samples. \ x2(n) is periodic with fundamental period, N2 = 8 samples.

Here, x(n) = x1(n) + x2(n), and x(n) is periodic with period N1 = 4, and x2(n) is periodic with period N2 = 8.

Therefore, x(n) is periodic with period N, where N is LCM of N1 and N2.

The LCM of 4 and 8 is 8.

\ x(n) is periodic with fundamental period, N = 8.

E2.2. Determine the even and odd parts of the signals.

1
a) x(n) = ⇒ x(n) = a −2n
a 2n

Solution

1
x(−n) = ⇒ x(−n) = a 2n
a −2n
Even part of the signal,

1 1 −2n
xe (n) = x(n) + x(−n) = a + a2n
2 2
Odd part of the signal is,

1 1
x0 (n) = x(n) − x(−n) = a −2n − a 2n
2 2

π
−j n
b) x(n) = 8 e 6

Solution

x(n) = 8 e
π
−j n
6
= 8 cos
LM π π
n − j sin n
OP
N 6 6 Q
x( −n) = 8 e
π
− j ( − n)
6 = 8 cos
LM π π
n + j sin n
OP
N 6 6 Q
xe (n) =
LM1 π π π OP π π
× 8 cos n − j sin n + cos n + j sin n = 8 cos n
N2 6 6 6 Q 6 6

1 L π π π π O π
x (n) = × 8 Mcos n − j sin n − cos n − j sin nP = − j8 sin n
0
2 N 6 6 6 6 Q 6

l
c) x(n) = 6, 4, 2, 2 q
A
Solution

l
Given that, x(n) = 6, 4, 2, 2 q
A
x(0) = 6, x(1) = 4, x(2) = 2, x(3) = 2
l
x( −n) = 2, 2, 4, 6 q
A
x(0) = 6; x( −1) = 4; x(−2) = 2; x( −3) = 2
E2. 4 DSP, Chapter 2 -Discrete Time Signals and Systems
1 1
Even part, x e (n) =
2
b
x(n) + x( −n) g Odd part, x 0 (n) =
2
b
x(n) − x( −n) g
at n = –3 ; x(n) + x(–n) = 0 + 2 = 2 n = –3 ; x(n) – x(–n) = 0 – 2 = –2
n = –2 ; 0+2 = 2 n = –2 ; = 0 – 2 = –2
n = –1 ; 0+4 = 4 n = –1 ; = 0 – 4 = –4
n= 0; 6 + 6 = 12 n= 0; = 6–6 = 0
n= 1; 4+0 = 4 n= 1; = 4–0 = 4
n= 2; 2+0 = 2 n= 2; = 2–0 = 2
n= 3; 2+0 = 2 n= 3; = 2–0 = 2

1 1
xe (n) = x(n) + x( −n) x0 (n) = x(n) − x( −n)
2 2
l
xe (n) = 1, 1, 2, 6, 2, 1, 1 q l
x0 (n) = −1, − 1, − 2 , 0, 2, 1, 1 q
A A
E2.3. a) Consider the analog signal x(t) = 2sin80pt. If the sampling frequency is 60 Hz, find the sampled version of
discrete time signal x(n). Also find an alias frequency corresponding to Fs = 60 Hz.

Solution

x(n) = x( t) n
t = nT =
Fs

∴ x(n) = 2 sin 80 πt = 2 sin 80 π ×

n
= 2 sin
4 πn FG IJ
t=
n
60
60 3 H K
Now, 2 sin
FG 4π n + 2πnIJ = 2 sinFG 10πn IJ
H3 K H 3 K
F 10πn IJ is, f = 5
The frequency of 2sin G
H 3 K 3

F 5
Also, f = ⇒ F = fFs = × 60 = 100
Fs 3

Hence for Fs = 60 Hz, F = 100 Hz is an alias frequency.

b) Consider the analog signals x1(t) = 4 cos 2p (30t), x2(t) = 4 cos 2p (5t). Find a sampling frequency so that
30 Hz signal is an alias of 5 Hz signal.

Solution

Let the sampling frequency be, Fs = 30 – 5 = 25 Hz

∴ x1(n) = x1(t) = 4 cos 2π 30 ×

FG n IJ
t = nT =
n
Fs
H 25 K
∴ x1(n) = 4 cos
12π
n = 4 cos 2πn +
2 πnFG
= 4 cos
2π
n
IJ
5 5 H 5 K
x 2 (n) = x 2 (t) = 4 cos 2π 5 ×
FG n IJ = 4 cos
2π
n
t = nT =
n
Fs
H 25 K 5

c) Consider the analog signal, x(t) = 3sin40pt – sin100pt + 2cos50pt

Determine the minimum sampling frequency and the sampled version of analog signal at this frequency. Sketch the
waveform and show the sampling points.Comment on the result.

Solution

x(t) = 3 sin 40πt − sin100 πt + 2 cos 50πt ≡ x(t) = 3 sin 2πF1t − sin 2πF2 t + 2cos 2πF3 t
40 100 50
∴ F1 = = 20 Hz ; F2 = = 50 Hz ; F3 = = 25 Hz
2 2 2
Solution for Exercise Problems E2. 5
The maximum analog frequency in the signal is 50 Hz.

The minimum sampling frequency should be twice that of this maximum analog frequency.

Fs ≥ 2 Fmax ⇒ Fs ≥ 2 × 50

Let, Fs = 100Hz

∴ x(nT) = x(t) n
t=nT=
Fs

n n n 2πn π
x(nT) = 3 sin 40π × − sin 100π + 2cos 50π × = 3 sin − sin πn + 2 cos n
100 100 100 5 2
sin πn = 0, for integer values of n.
2πn πn
∴ x(nT) = 3 sin + 2cos
5 2

3 sin 40πt
1
⇒ F1 = 20 Hz, T1 = = 0.05 sec
20

sin 100πt

⇒ F2 = 50 Hz, T2 = 0.02 sec

2cos 50πt

⇒ F3 = 25 Hz, T3 = 0.04

Fs = 100 Hz

Ts = 0.01sec

In the analog signal x(nT), the component sin 100pt will give always zero samples when sampled at 100Hz for any value of n.
This is the drawback in sampling at nyquist rate, which is Fs = 2 Fmax.

E2.4. Determine whether the following signals are energy or power signals.

a) x(n) =
FG 5 IJ n

u(n)
H 9K
Solution

x(n) =
FG 5 IJ u(n)
n
for all n.
H 9K
∴ x(n) = (0.55)n ; n ≥ 0
+∞ ∞ ∞ ∞
2
2 n
∑ d(0.55) i = ∑ b0. 302g
2 n
Energy, E = ∑
n = −∞
x(n) = ∑
n=0
(0.55)n =
n= 0 n= 0

∞
1
∴ E= ∑ (0.302)
n= 0
n
=
1 − 0.302
= 1. 43 Joules

N
1 2
Power, P = Lt
N→∞ 2N + 1
∑ x(n)
n = −N
N N
1 n 1
= Lt
N→∞ 2N + 1
∑ d(0.55) i 2
= Lt
N→∞
∑
2N + 1 n = 0
(0.302)n
n= 0

1 (0.302)N+1 − 1 1 0−1
= Lt = × =0
N→ ∞ 2N + 1 0.302 − 1 ∞ −0.698
P is zero and E is finite.
So x(n) is energy signal.
E2. 6 DSP, Chapter 2 -Discrete Time Signals and Systems
3π 1 + cos 2θ
b) x(n) = cos n cos2 θ =
4 2
Solution
F 1+ cos 2 × 3π n I
∑ GG 4 J
+∞ +∞ 2 +∞
2 3π
Energy, E = ∑ x(n) = ∑ cos JJ n =
n = −∞ GH
n= −∞ 2 4
K n = −∞

F 1I F 3π I
+∞
1 F 3π I 1 +∞ +∞
= G J ∑ G 1 + cos
H 2K H nJ =
2 K 2 GH
∑ 1 + ∑ cos 2 nJK = 2 b∞ + 0g = ∞ n

n = −∞ n = −∞ n = −∞

Power, P = Lt
1 1 N

∑ x(n) = Lt 2N + 1 ∑ cos GH 4 nJK

F 3π I
2
N
2
2N + 1N→ ∞
n = −N
N→ ∞
n = −N

FG1+ cos 2 × 3π nIJ

1 H 4 K
N
1 1 L 3π O +N +N
= Lt
2N + 1
∑
N→ ∞ 2
= Lt
n = −N 2N + 1 2 NM
M ∑ 1 + ∑ cos nP
2 PQ N→ ∞
n = −N
n

n = −N

1 1 L O
= Lt M11+414+2144
2N + 1 2 MN
N→ ∞
.....31 + 1 + 11+44
1 + 12.....
44+ 31 + 0P
N termsPQ N terms

1 1 1
= Lt × 2N + 1 = = 0.5
N→ ∞ 2N + 1 2 2
Since P is finite and E is infinite, x(n) is power signal.

3π
It can be shown that cos n is periodic and sum of samples of one period of periodic cosine signal is zero.
2

cos
3π
b
n + N = cos
3πn 3πN
g+
FG IJ 3π 3π
2 2 2 H K n = 0 ; cos
2
n=1 n = 4 ; cos
2
n=1

3 πN 3π 3π
Let, = 2 πM n = 1 ; cos n=0 n = 5 ; cos n=0
2 2 2
4M 3π 3π
∴ N= n = 2 ; cos n = −1 n = 6 ; cos n = −1
3 2 2
Let, M = 3, Now, N = 4 3π 3π
n = 3 ; cos n=0 n = 7 ; cos n=0
2 2
3π
∴ cos n is periodic with
2
period 4 samples.

c) x(n) = u(2n)
Solution
+∞ ∞
2
E = ∑ x(n) = ∑ u(2n) 2
= ∑ u(n) = 1+ 1+ 1+ 1 ..... ∞ = ∞
n = −∞ n= 0 n = even

N N
1 2 1 1
P =
N→∞
Lt
2N + 1 ∑
n = −N
x(n) = Lt
N→∞ 2N + 1 ∑ u(2n)
n= 0
2
= Lt
N→∞
= 1+ 1+......+1
2N + 1 144244 3
N
1+ terms
2

N
FG 1 + 1IJ 1 1
+
1
= Lt
1
1+
N
=
FG IJ Lt
H N 2K = 2 ∞ = 2 =
1
.
N→∞ (2N + 1) 2 H K N→∞ F 1I
NG 2 + J 2+
1 2 4
H NK ∞
Since P is finite, E is infinite, x(n) is power signal.
d ) x(n) = 2 u(3 − n)
Solution
+∞ −3 −3

∑ b2 u(3 − n)g
2 2
E = ∑ x(n) = = ∑ 4 .......1 + 1 + 1 = ∞
14 4244 3 u (3 −n )
n = −∞ n = −∞ n = −∞ inf inite terms

+N −3
1 2 1
P =
N→∞
Lt
(2N + 1) ∑
n = −N
x(n) = Lt
N→∞ (2N + 1) ∑ 4 u(3 − n)
n=N

1 4
= Lt 4 1+ 1+ 1......+1 = Lt N− 2 n −3 −2 −1 0
N→∞ (2N + 1) 1442443 N→∞ (2N + 1)
N − 2 terms
Solution for Exercise Problems E2. 7

N × 4 1−
2 FG IJ 4 1−
FG 2 IJ
∴ P = Lt
N H K =
H ∞ 4
= =2
K
N→∞
N 2+
1 FG IJ 2+
1 2
N H K ∞

Since P is finite, E is infinite x(n) is power signal.

E2.5. Construct the block diagram and signal flow graph of the discrete time systems whose input-output relations are
described by the following difference equations.
a) y(n) = 2y(n − 1) + 2.1 x(n − 1) + 0.5 x(n − 2).

Solution

B loc k D ia gra m S ign a l F lo w G ra p h

x (n )
x (n ) z
−1
z
−1
2.1 x (n −1)

x (n − 1) 2.1 x (n −1)
−1
z 2.1

x (n − 1 )
x (n − 1 )
0.5 x (n − 2)

−1
z z
−1
0.5

x (n − 2) 0.5 0.5 x (n − 2) x (n − 2)

y (n )
y (n )
2 y(n −1 )
−1
z z
−1

y (n − 1)
2
2 y(n −1 )

B loc k D ia gra m S ign a l F lo w G ra p h

−1
x (n ) 2.1x(n −1) y (n ) x (n ) 1 z 2.1 1 1 1 y (n )
−1 2.1
z + +
)
)

−1
−2

−1 −1
(n

z
x(n

z 2
2y

0.5
0 .5

−1 −1
z 0.5 2 z

b) y(n) = 1.6x(n − 2) + 0.7 x(n) + 3y(n − 1) + 0.3 y(n − 2).

Solution

B lo c k D ia gra m S ig n a l F lo w G ra p h

0.7
x (n ) 0.7 0.7x(n) x (n ) 0.7x(n)

x (n )
x (n )

−1 −1 1.6 x (n −2)
z z

1.6
−1
−1
z
z
x (n −2 )
x (n − 2 ) 1.6 1.6 x (n − 2)
E2. 8 DSP, Chapter 2 -Discrete Time Signals and Systems
B loc k D ia gra m S ign a l F lo w G ra p h

y (n ) 3 y(n −1 ) y (n )

−1
z
−1 3
z

3 y(n − 1 ) y (n −1 )
y (n − 1 )
3

y (n − 1 ) 0.3 y (n −2) y (n −1)

−1
z
−1 z
0.3

0.3 y (n − 2 )
y (n −2 )
0.3

B loc k D ia gra m S ign a l F lo w G ra p h

x (n ) y (n ) y (n )
x (n ) 1 0.7 1 1
0.7 + +
−1 3 −1
−1
z z
−1
z + 3 z

0 .3
6
1.
−1
−1 z
z
−1 −1
z 1.6 0.3 z

E2.6. Determine the response of the discrete time systems governed by the following difference equations.

a) y(n) = 0.1 y(n − 1) + x(n − 1) + 0.7 x(n) ;

x(n) = 2 − n u(n) ; y( −1) = − 1

Solution

y(n) = 0.1 y(n – 1) + x(n – 1) + 0.7 x(n)

\ y(n) – 0.1 y(n –1) = 0.7 x(n) + x(n –1) .....(1)

Homogeneous solution

When the input is zero the equation (1) can be written as,

y(n) – 0.1 y(n –1) = 0 .....(2)

On substituting y(n) = ln in equation (2) we get,

ln – 0.1 l(n – 1) = 0

\ l(n – 1) (l – 0.1) = 0 Þ l = 0.1

The homogeneous solution yh(n) is given by,

yh(n) = Cln = C(0.1)n for n ³ 0 = C(0.1)n u(n) .....(3)

Particular solution

Given that, x(n) =

FG 1IJ u(n)
n
; ∴ y(n) = K
FG 1IJ u(n)
n

H 2K H 2K
Using the above values for x(n) and y(n) in equation(1) we get,

K
FG 1IJ u(n) − 0.1KFG 1IJ
n (n −1)
u(n − 1) = 0.7 ×
FG 1IJ u(n) + FG 1IJ
n n −1
u(n − 1) .....(4)
H 2K H 2K H 2K H 2K
To determine the value of ‘K’ evaluate equation(4) for n = 1.

K
FG 1IJ u(1) − 0.1KFG 1IJ u(0) = 0.7FG 1IJ u(1) + FG 1IJ u(0)
1 0 1 0

H 2K H 2K H 2K H 2K
1.35
0.5K − 0.1K = 0.35 + 1 ⇒ 0.4K = 1.35 ⇒ K= = 3.375
0.4
Solution for Exercise Problems E2. 9
\ The particular solution yp(n) is given by,

y p (n) = K
FG 1IJ u(n) = 3.375 × FG 1IJ u(n)
n n

.....(5)
H 2K H 2K
Total response

∴ Response, y(n) = yh (n) + y p (n)

LM
y(n) = C(0.1)n + 3.375 ×
FG 1IJ OP u(n)
n
(or) y(n) = C(0.1)n + 3.375
FG 1IJ n
; for n ≥ 0 .....(6)
MN H 2 K PQ H 2K
At n = 0, from equation (1) we get,
y(0) − 0.1y(−1) = 0.7 x(0) + x( −1) .....(7)

Given that : y(−1) = − 1 and x(n) = 2 −n u(n)

∴ x(0) = 1 and x(−1) = 0

On substituting the above values in equation (7) we get,

y(0) + 0.1 = 0.7

y(0) = 0.7 – 0.1

\ y(0) = 0.6

Put n = 0 and y(0) = 0.6 in equation (6).

y(0) = C(0.1)0 + 3 . 375 ×

FG 1IJ 0
= C + 3 . 375
H 2K
0.6 = C + 3 . 375
C = 0.6 − 3 . 375
C = −2 . 775
\ The total response is given by,
LM
y(n) = −2 . 775 (0.1)n + 3.375
FG 1IJ OP u(n)
n

MN H 2 K PQ
b) y(n) + 2.1 y(n – 1) + 0.2 y(n –2) = x(n) + 0.56 x(n – 1) ; x(n) = u(n) ; y(–2) = 1 ; y(–1) = –3.
Solution
y(n) + 2.1 y(n – 1) + 0.2 y(n – 2) = x(n) + 0.56 x(n – 1) .....(1)

Homogeneous Solution

When the input is zero, the equation(1) can be written as,

y(n) + 2.1 y(n –1) + 0.2 y(n –2) = 0 .....(2)

n
substitute y(n) = l in equation (2)

\ ln + 2.1 ln – 1 + 0.2 ln – 2 = 0

l(n –2) [l2 + 2.1 l + 0.2] = 0

The characteristic equation is,

λ2 + 2 .1 λ + 0.2 = 0 ⇒ bλ + 0.1g bλ + 2g = 0
\ The roots are, l1 = –0.1, l2= –2.

The homogenous solution yh(n) is given by,

yh (n) = C1 λn1 + C2 λn2

b g + C b−2g
yh (n) = C1 −0.1
n
2
n
for n ≥ 0

= C b −0.1g + C b −2g
n n .....(3)
1 2 u(n)
E2. 10 DSP, Chapter 2 -Discrete Time Signals and Systems
Particular Solution

Given that , x(n) = u(n) ; \ y(n) = K u(n). .....(4)

Using the above values for x(n) and y(n) in equation(1) we get,

K u(n) + 2.1 K u(n –1) + 0.2 K u(n –2) = u(n) + 0.56 u(n –1) .....(5)

To find ‘K’ evaluate equation (5), for n = 2.

\ K u(2) + 2.1 K u(1) + 0.2 K u(0) = u(2) + 0.56 u(1)

K + 2.1 K + 0.2 K = 1 + 0.56

1.56
3.3K = 1. 56 ⇒ K= = 0.47
3.3
\ yp(n) = 0.47 u(n)
\ Total response,
y(n) = yh(n) + yp(n)
y(n) = [C1(–0.1)n + C2(–2)n + 0.47] u(n)
y(n) = C1(–0.1)n + C2(–2)n + 0.47 for n ³ 0. .....(6)
At n = 0 from equation (1) we get,
y(0) + 2.1 y(–1) +0.2 y(–2) = x(0) + 0.56 x(–1) .....(7)
Given, y(–1) = –3, Also, x(n) = u(n)
y(–2) = 1 \ x(0) = 1 and x(–1) = 0.
On substituting the above values in equation (7),
y(0) + 2.1 (–3) + 0.2 (1) = 1 + 0.
y(0) – 6.3 + 0.2 = 1 Þ y(0) – 6.1 = 1
\ y(0) = 1 + 6.1 = 7.1
At n = 1 from equation (1) we get,
y(1) + 2.1 y(0) + 0.2 y(–1) = x(1) + 0.56 x(0)
We know that, y(0) = 7.1 , x(0) = 1
y(–1) = –3 , x(1) = 1
\ y(1) + 2.1 (7.1) + 0.2 (–3) = 1 + 0.56 Þ y(1) + 14.31 = 1.56
\ y(1) = 1.56 – 14.31 = –12.75
Put n = 0 and y(0) = 7.1 in eqaution(6).
y(0) = C1(–0.1)0 + C2(–2)0 + 0.47
7.1 = C1 + C2 + 0.47
\ C1 + C2 = 7.1 – 0.47
\ C1 + C2 = 6.63 .....(8)
Put n= 1 and y(1) = –12.75 in equation(6).
y(1) = C1(–0.1)1 + C2(–2)1 + 0.47
–12.75 = –0.1C1 – 2 C2 + 0.47
0.1C1 + 2C2 = 12.75 + 0.47
\ 0.1C1 + 2 C2 = 13.22 .....(9)

Equation (8) ´ 2 Þ 2 C1 + 2 C2 = 13.26

Equation (9) Þ 0.1 C1 + 2 C2 = 13.22

(–) (–) (–)

1.9 C1 = –0.04
−0.04
C1 = = − 0.02
1. 9
∴ C2 = 6.63 − C1 = 6.63 + 0.02 = 6.65
\ y(n) = –0.02 (–0.1)n + 6.65 (–2)n + 0.47, for n ³ 0.
= [0.47 – 0.02 (–0.1)n + 6.65 (–2)n] u(n).
Solution for Exercise Problems E2. 11
E2.7. Test the following systems for time invariance.
a) y(n) = x(n + 1) + x(n + 2)
Solution
Given that, y(n) = H {x(n)} = x(n + 1) + x(n + 2)
Response for Delayed Input
y(n –m) = H {x(n – m)} = x(n –m + 1) + x(n – m + 2)
Response for Unshifted Input
y(n) = H {x(n)} = x(n + 1) + x(n + 2)
Delayed Response

l q
y d (n) = z −m H x(n) = z −m x(n + 1) + x(n + 2) = z −m x(n + 1) + z −m x(n + 2)
= x(n − m + 1) + x(n − m + 2)

Here, y(n − m) = y d (n)

Hence system is time invariant.
b) y(n) = n ax(n)
Solution
Given that, y(n) = H{x(n)} = y(n) = n ax(n)
Response for Delayed Input
y(n – m) = H {x(n – m)} = (n–m) ax(n – m)
Delayed Response

l q
y d (n) = z −m H x(n) = z −m n a x(n) = n a x(n − m)

Here, y(n − m) ≠ y d (n)

Hence the system is time variant.
c) y(n) = x2(n + 2) + C
Solution
Given that, y(n) = H{x(n)} = x2(n + 2) + C
Response for Delayed Input
y(n –m) = x2(n – m + 2) + C
Delayed Response

l q
y d (n) = z −m H x(n) = z −m x 2 (n + 2) + C = z −m x 2 (n + 2) + C = x 2 (n − m + 2) + C

Here, y(n − m) = y d (n)

Hence the systems is time invariant.
d) y(n) = (n – 1) x2(n) + C
Solution
Given that, y(n) = H{x(n)} = (n - 1)x2(n) + C
Response for Delayed Input
y(n –m) = (n – m – 1)x2(n – m) + C
Delayed Response

l q
y d (n) = z −m H x(n) = z −m (n − 1) x 2 (n) + C = (n − 1) x 2 (n − m) + C.

Here, y(n − m) ≠ y d (n)

Hence the systen is time variant.

E2.8. Test the following systems for linearity.

a) y(n) = x2(n) + x3(n – 1)
Solution
Let, ‘H ’ be the system,
\ y(n) = H{x(n)} = x2(n) + x3(n – 1)
Consider two signals, x1(n) and x2(n).
E2. 12 DSP, Chapter 2 -Discrete Time Signals and Systems
Let y1(n) and y2(n) be responses of system ‘H’ for inputs x1(n) and x2(n).
\ y1(n) = H {x1(n)} = x12(n) + x13(n –1)
y2(n) = H {x2(n)} = x22(n) + x23(n –1)
\ a1 y1(n) + a2 y2(n) = a1 [x12(n) + x13(n –1)] + a2 [x22(n) + x23(n –1)] .....(1)
Consider a linear combination of inputs.
a1 x1(n) + a2 x2(n) = x3(n)
\ y3(n) = H {x3(n)} = x32(n) + x3(n – 1)
= [a1 x1(n) + a2 x2(n)]2 + [a1 x1(n – 1) + a2 x2(n – 1)]3 .....(2)
From equations (1) and (2) we can say that,
y3(n) ¹ a1 y1(n) + a2 y2(n)
Hence the system is non-linear.
b) y(n) = bx(n + 2) + n e x(n)
Solution
Let ‘H ’ be the system.
\ y(n) = H{x(n)} = b x(n + 2) + n ex(n)
Consider two signals x1(n) and x2(n).
Let y1(n) and y2(n) be their respective outputs.
l q
∴ y1(n) = H x1(n) = b x1(n + 2) + ne x1(n)

y (n) = H lx (n)q = b x (n + 2) + ne
2 2 2
x 2 ( n)

∴ a y (n) + a y (n) = a eb x (n + 2) + ne .....(1)

1 1 2 2 1j + a eb x (n + 2) + ne j
1
x1 ( n )
2 2
x 2 (n)

Consider a linear combination of inputs.

∴ a1 x1(n) + a 2 x 2 (n) = x 3 (n)

m r
∴ y 3 (n) = H x 3 (n) = b x 3 (n + 2) + ne x 3 (n)

= b a1 x1 (n + 2) + a 2 x 2 (n + 2) + ne[ a1 x1(n) + a 2 x 2 (n)]

= a1 b x1(n + 2) + a 2 b x 2 (n + 2) + nea1x1(n) ea 2 x 2 (n) .....(2)

From equations (1) and (2) we get,

y 3 (n) ≠ a1y1(n) + a 2y 2 (n)

Hence the system is non-linear system.

c) y(n) = a x(n) + b x(n)

Solution
Let ‘H ’ be the system.
l q
∴ y(n) = H x(n) = a x(n) + b x(n)

Consider two signals x1(n) and x2(n).

Let y1(n) and y2(n) be their respective outputs.

l q
∴ y1(n) = H x1(n) = a x1(n) + b x1(n)

y (n) = H lx (n)q = a
2 2 x 2 (n) + b x 2 (n)

∴ a1 y1(n) + a2 y2 (n) = a1 a x1(n) + b x1(n) + a2 a x2 (n) + b x2 (n) .....(1)

Consider a linear combination of inputs.

∴ a1 x1(n) + a2 x 2 (n) = x3 (n)

m r
∴ y 3 (n) = H x 3 (n) = a x 3 (n) + b x 3 (n)

= a a1 x1(n) + a 2 x 2 (n) + b a1 x1(n) + a 2 x 2 (n) .....(2)

From equations (1) and (2) we can say that,

y3(n) ¹ a1 y1(n) + a2 y2(n)
Hence the system is non-linear system.
Solution for Exercise Problems E2. 13
1
d) y(n) = x(n) +
x(n)

Solution
Let ‘H ’ be the system.

1
∴ y(n) = H x(n) = x(n) +l q x(n)

Consider two signals x1(n) and x2(n).

Let y1(n) and y2(n) be their respective outputs.

1 1
l q
∴ y1(n) = H x1(n) = x1(n) +
x1(n)
l
; y 2 (n) = H x 2 (n) = x 2 (n) + q x 2 (n)

a1 a2
∴ a1 y1(n) + a 2 y 2 (n) = a1 x1(n) + + a 2 x 2 (n) + .....(1)
x1(n) x 2 (n)

Consider linear combination of inputs.

∴ a1x1(n) + a2 x2 (n) = x3 (n)

1 1
m
∴ y 3 (n) = H x 3 (n) = x 3 (n) + r x 3 (n)
= a1x1(n) + a 2 x 2 (n) +
a1x1(n) + a 2 x 2 (n)
.....(2)

From equations (1) and (2) we get,

y3(n) ¹ a1 y1(n) + a2 y2(n)

Hence the system is non-linear system.

N M
e) y(n) = ∑b
m = −1
m x(n + m) + ∑c
m =0
m y(n + m)

Solution

Let, H be the system represented by the given equation.

N M
∴ y(n) = H x(n) = l q ∑b m x(n + m) + ∑c m y(n + m)
m = −1 m=0

Consider two signals, x1(n) and x 2 (n). Let y1(n) and y 2 (n) be the respective outputs.

N M
l q ∑ b x (n + m) + ∑ c y (n + m)
y1(n) = H x1(n) = m 1 m 1
m= −1 m=0

N M
l q ∑ b x (n + m) + ∑ c y (n + m)
y 2 (n) = H x 2 (n) = m 2 m 2
m= −1 m=0

L N O L M O
a y (n) + a y (n) = a M ∑ b x (n + m) + ∑ c y (n + m)P + a M ∑ b x (n + m) + ∑ c y (n + m)P
N M

.....(1)
1 1 2
MN
2 1
m= −1 PQ MN
m 1
m=0 PQ m 1 2
m= −1
m 2
m=0
m 2

Now consider linear combination of inputs

a1x1(n) + a 2 x 2 (n) = x 3 (n)

N M
∴ y 3 (n) = H x 3 (n) = m r ∑b m = −1
m x 3 (n + m) + ∑c
m=0
m y 3 (n + m)

N M
= ∑b
m = −1
m a1x1(n + m) + a 2 x 2 (n + m) + ∑c
m= 0
m y 3 (n + m)

N M M
.....(2)
= a1 ∑b
m = −1
m x1(n + m) + a 2 ∑b
m = −1
m x 2 (n − m) + ∑c
m= 0
m y 3 (n + m)

By time invariant property,

If y3(n) = H {a1 x1(n) + a2 x2(n)} ; then, y3(n + m) = H {a1 x1(n + m) + a2 x2(n + m)}

If y2(n) = H {x2(n)}, then y2(n + m) = H {x2(n + m)}

E2. 14 DSP, Chapter 2 -Discrete Time Signals and Systems
If y1(n) = H {x1(n)}, then y1(n + m) = H {x1(n + m)}

\ y3(n + m) = H {a1 x1(n + m) + a2 x2(n + m)} = a1 H {x1(n + m)} + a2 H {x2(n + m)}

= a1 y1(n + m) + a2 y2(n + m) .....(3)

Using equation (3), the equation(2) can be written as,
N N M
y 3 (n) = a1 ∑b
m = −1
m x1(n + m) + a 2 ∑b
m = −1
m x 2 (n + m) + ∑c
m=0
m a1y1(n + m) + a 2 y 2 (n + m)

N N M M
= a1 ∑b
m = −1
m x1(n + m) + a 2 ∑b
m = −1
m x 2 (n + m) + a1 ∑c
m=0
m y1(n + m) + a 2 ∑c
m=0
m y 2 (n + m)

F b N M I
= a1 GH ∑
m = −1
m x1(n + m) + ∑c
m= 0
m y1(n + m) JK
F b N M I
+ a2 GH ∑
m = −1
m x 2 (n − m) + ∑c
m= 0
m JK
y 2 (n + m) .....(4)

From equations (1) and (4) we can say that,

y 3 (n) = a1 y1(n) + a 2 y2 (n)

Hence the system is linear.
E2.9. Test the causality of the following systems.
a) y(n) = x(n) – x(–n – 2) + x(n –1)
Solution
When, n = –2, y(–2) = x(–2) – x(0) + x(–3)
n = –1, y(–1) = x(–1) – x(–1) + x(–2)
n = 0, y(0) = x(0) –x(–2) + x(–1)
n = 1, y(1) = x(1) –x(–3) + x(0)
n=2, y(2) = x(2) – x(–4) + x(1)
For n £ –2, the system response depends on future input.
Hence the system is noncausal.

b) y(n) = a x(2n) + x(n2)

Solution
When, n = –1, y(–1) = a x(–2) + x(1) ; Response depends on future input
n = 0, y(0) = a x(0) + x(0)
n = 1, y(0) = a x(2) + x(1) ; Response depends on future input.
Except n = 0 for all other values of n, the response depends on future input.
Hence the system is noncausal.
n n
c) y(n) = ∑ x(m) + ∑ x(2m)
m = −1 m = −∞

Solution
0 0
n = 0, y(0) = ∑ x(m) + ∑ x(2m) ⇒ y(0) = x( −1) + x(0) + ..... + x(−4) + x(−2) + x(0).....
m = −1 m = −∞

∴ y(0) depends on present and past inputs.

1 1
n = 1, y(1) = ∑ x(m) + ∑ x(2m) ⇒ y(1) = x( −1) + x(0) + x(1) + ..... + x( −4) + x( −2) + x(0) + x(2)
m = −1 m = −∞

y(1) depends on future input x(2).

The system response depends on future input for n > 0.

Hence it is noncausal system.
Solution for Exercise Problems E2. 15
d) y(n) = (0.3)n u(n + 2) Note : For causality y(n) = 0 ; for n < 0.

Solution
1
( 0 .3 ) 3
1 1
(0 .3)
n
( 0 .3 ) 2 ( 0 .3 ) 2
1 y (n )
1
0 .3 u (n + 2 ) 0 .3
1 1
X ⇒
0.3
2
( 0 .3 )
−3 −2 −1 0 1 2 −3 −2 −1 0 1 2 −3 −2 −1 0

Here y(n) ¹ 0 for n < 0. Therefore the system is noncausal.

4
e) y(n) = ∑ x(n − k)
k = −4

Solution
4
y(n) = ∑ x(n − k) = x(n + 4)
k = −4
+ x(n + 3) + x(n + 2) + x(n + 1) + x(n) + x(n − 1) + x(n − 2) + x(n − 3) + x(n − 4)

For any value of n the system response depends on future inputs.

Hence, the system is noncausal.

E2.10. Test the stability of the following discrete time systems.

a) y(n) = x2(n) + x(n + 1)
Solution
The given system involves squaring operation and so it is nonlinear.
The operations performed by system is squaring and shifting.
A bounded input signal will remain bounded even after squaring and shifting.
Hence the system is BIBO stable.
b) y(n) = n x(n – 1)
Solution
The given system involves multiplication by n and so it is time variant system.
If x(n) doesnot tend to “0” as n tends to infinity then the system is unstable.
c) h(n) = (0.4)n u(n + 3)
Solution
Here, h(n) = 0.4n u(n + 3) = 0.4n ; For n = –3 to + ¥
+∞ ∞ −1 ∞
1
∑ h(n) = ∑ (0.4) = ∑ (0.4) n n
+ ∑ (0.4) n
= (0.4)−3 + (0.4)−2 + (0.4)−1 +
1 − 0.4
= 26.04 = Constant
n = −∞ n = −3 n = −3 n= 0

Hence it is stable system.

d) h(n) = (8)n u(4 – n)

Solution
Here, h(n) = 8n u(4 – n) = 8n ; for n = – ¥ to +4
∞
For stability, ∑ |h(n)|
n = −∞
< ∞

+∞ 4 0 4
∴ ∑ h(n)
n = −∞
= ∑ (8)
n = −∞
n
= ∑ (8) + ∑ (8)
n = −∞
n

n =1
n

∞
= ∑ (8)
n= 0
−n
+ 81 + 8 2 + 8 3 + 84

=
∞
F 1I
∑ GH 8 JK
n
+ 4680 =
∞

∑ (0.125) n
+ 4680 =
1
+ 4680
n= 0 n= 0 1 − 0.125
= 4681.14 = Constant
Hence it is stable system.
E2. 16 DSP, Chapter 2 -Discrete Time Signals and Systems
e) y(n) = x(n – 3)
If x(n) = d(n), then y(n) = h(n)
d(n – 3) = 1, only
\ h(n) = d(n – 3) when n = 3, and
∞ +∞ zero for all other
∴ ∑
n = −∞
h(n) = ∑
n = −∞
δ(n − 3) = 1
values of n

Hence the system is stable.

E2.11. Determine the range of values of ‘a’ and ‘b’ for the stability of LTI system with impulse response,

h(n) =
|RS ( −4a) n
; n ≥ 0
T|(2b) −n
; n < 0

Solution
The condition to be satisfied for the stability of the system is,
+∞ −1 ∞

∑ h(n) = ∑ (2b)−n + ∑ (−4a)n

n = −∞ n= −∞ n= 0
−1 ∞
= ∑ |2b|−n + ∑ |4a|n
n = −∞ n= 0

∞ ∞
= ∑ |2b| n
+ ∑ |4a| n

n =1 n= 0

∞ ∞
= ∑ |2b| n
− |2b|0 + ∑ |4a| n

n= 0 n= 0
∞
1
If 0 <|2b| < 1, then ∑|2b| n
=
1 − |2b|
n= 0

∞
1
If 0 <|4a| < 1, then ∑|4a| n
=
1 − |4a|
n= 0

+∞
1 1
∴ ∑ h(n) = 1 − |2b| − 1+ 1 − |4a| = Constant
n = −∞

∴ Condition for stability is,

1
0 < |2b| < 1 ⇒ 0 < |b| <
2
1
0 < |4a| < 1 ⇒ 0 < |a| <
4

E2.12. a) Determine the impulse response for the cascade of two LTI systems having impulse responses,

h1 (n) =
FG 1 IJ n

u(n) and h2 (n) = δ (n − 3)

H7 K
Solution
The impulse response of the cascade system is given by,
h(n) = h1(n) ∗ h2 (n) = h2 (n) ∗ h1(n)
∞
= ∑ h (m) h (n − m) ;
m= −∞
2 1 'm' is dummy variable

∴ h(n) = ∑
∞
h2 (m) h1(n − m) =
∞

∑ δ(m − 3)
FG 1IJ n −m
=
∞

∑ δ(m − 3)
FG 1IJ FG 1IJ
n −m

m= 0 m=0
H 7K m=0
H 7K H 7K
FG 1IJ ∑ δ(m − 3) FG 1IJ
=
n ∞ −m

H 7K H 7K m=0

F 1I will be nonzero, only when m = 3.

The product of d(m – 3) and GH JK
−m

7
F 1I F 1I for n ≥ 3
∴ h(n) = G J G J
n −3

H 7K H 7K
F 1I
h(n) = G J u(n − 3) for all n.
n− 3

H 7K
Solution for Exercise Problems E2. 17
b) Determine the overall impulse response of the interconnected discrete time system shown in fig E2.12.

Take, h1 (n) =
FG 1 IJ n

u(n) ; h2 (n) =
FG 1 IJ n

u(n) ; h3 (n) =
FG 1 IJ n

u(n)
H 3K H6K H 9K
h 2 ( n)

x (n )
+ y (n )

h 1 ( n) + h 3 ( n)

Solution F ig E 2 .1 2
The given system can be redrawn as,

h 2 (n)

x (n ) y (n ) x (n ) y (n )
h 2 (n) + h 2 (n) + [(h 2 (n) + h1(n)) ∗ h3 (n)]

+ h 3 (n)

h1(n) x (n ) h(n) y (n )

b g
h(n) = h2 (n) + h1(n) + h2 (n) ∗ h3 (n) = h2 (n) + h1(n) ∗ h3 (n) + h2 (n) ∗ h3 (n) b g b g
Evaluation of h1(n) * h3(n)
∞
h1(n) ∗ h3 (n) = ∑ h (m) h (n − m)
m= −∞
1 3

=
n

∑ h (m) h (n − m) =
n

∑ GH 3 JK
F 1I FG 1IJ
m n −m
=
FG 1IJ ∑ FG 1IJ
n n m
9m =
FG 1IJ ∑ FG 9 IJ
n n m

m= 0
1 3
m= 0
H 9K H 9K H 3K m= 0
H 9K H 3K m= 0

FG 9 IJ − 1 n +1

F 1I H 3 K
n
= G J
H 9K 9 − 1
3
FG 9 IJ 9 − 1 n
FG 9 IJ 9 − 1 n

F 1I H 3 K 3
= G J
n
= G J
F 1I H 3 K 3 F 1I L 1 F 9 I 9 − 1 OP
= G J M G J
n n n

H 9K 9
−1
H 9K 2 H 9 K MN 2 H 3 K 3 2 PQ
3

G J G J − 21 FGH 91IJK = 32 FGH 31IJK − 21 FGH 91IJK for n ≥ 0.

3 F 1I F 9 I
n n n n n
=
2 H 9K H 3K

3 F 1I
G J u(n) − 21 FGH 91IJK u(n) for all 'n'
n n
=
2 H 3K
Evaluation of h2(n) * h3(n)

h2 (n) ∗ h3 (n) =
+∞

∑ h (m) h (n − m) = ∑ h (m) h (n − m)
F 1I FG 1IJ = FG 1IJ ∑ FG 1IJ FG 1IJ
n
=
n

∑ GH 6 JK
m n −m n n m −m

m= −∞
2 3
m= 0
H 9K H 9K H 6K H 9K
2 3
m= 0 m= 0

FG 3 IJ − 1 LM FG 3 IJ − 1OP n+1 n+1

F 1 I F 1I n
F 1I F 3 I = FG 1IJ H 2 K
n
= G J ∑ G J 9 = G J ∑ G J
m n
F 1I M H 2 K P
n m n n

H 9 K H 2 K H 9 K 3 − 1 = GH 9 JK MM 1 PP
m
H 9K H 6K m=0 m=0
2 MN 2 PQ
F 1I L F 3 I 3 − 2OP = FG 1IJ LMFG 3 IJ 3 − 2OP = FG 1IJ FG 3 IJ 3 − 2 FG 1IJ
= G J M2G J
n n n n n n n

H 9 K MN H 2 K 2 PQ H 9 K MNH 2 K PQ H 9 K H 2 K H 9K
F 3I F 1In
F 1 I F 1I F 1 I n
= G J 3 − 2 G J = 3 G J − 2 G J = 3 G J u(n) − G J u(n) for all n.
F 1I n n n n

H 18 K H 9K H 6K H 9K H 6K H 9K
Overall Impulse Response

Now the overall impulse response h(n) is given by,

b
h(n) = h2 (n) + h1(n) ∗ h3 (n) + h2 (n) ∗ h3 (n) g b g
E2. 18 DSP, Chapter 2 -Discrete Time Signals and Systems

h(n) =
FG 1IJ u(n) + LMFG 3 IJ FG 1IJ u(n) − FG 1IJ FG 1IJ u(n)OP + LM3 FG 1IJ u(n) − FG 1IJ u(n)OP
n n n n n

H 6K MNH 2 K H 3 K H 2 K H 9 K PQ MN H 6 K H 9 K PQ
FG 1IJ u(n) − 3 FG 1IJ u(n) + 3 FG 1IJ u(n) =
n n n
LM4 F 1I n
FG IJ
3 1
n
3 FG 1IJ OP u(n)
n
⇒ h(n) = 4
H 6K 2 H 9K 2 H 3K MN GH 6 JK −
H K
2 9
+
2 H 3 K PQ

E2.13. Determine the response of an LTI system whose and impulse response h(n) and input x(n) are given by,

a) l
h(n) = 1, 4, 1, − 2, 1 q
A
l
x(n) = 1, 3, 5, − 1, − 2 q
A
Solution

The response y(n) of the system is given by convolution of x(n) and h(n).
+∞
y(n) = x(n) ∗ h(n) = ∑ x(m) h(n − m)
m= −∞

Input sequence starts at n = –1

Impulse response starts at n = –2

Therefore the output sequence start at, n = –1 + (–2) = –3

The output consists of 5 + 5 –1 = 9 samples.

The 9 samples of output sequence are computed by table method as shown below.

m –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7

x(m) 1 3 5 –1 –2

h(m) 1 4 1 –2 1

h(–m) 1 –2 1 4 1

h(–3–m) 1 –2 1 4 1

h(–2–m) 1 –2 1 4 1

h(–1–m) 1 –2 1 4 1

h(0 – m) 1 –2 1 4 1

h(1 – m) 1 –2 1 4 1

h(2 – m) 1 –2 1 4 1

h(3 – m) 1 –2 1 4 1

h(4 – m) 1 –2 1 4 1

h(5 – m) 1 –2 1 4 1

3
When n = −3 ; y(−3) = ∑ x(m) h(−3 − m) = 0 + 0 + 0 + 0 + 1+ 0 + 0 + 0 + 0 = 1
m = −5

3
When n = −2 ; y(−2) = ∑ x(m) h(−2 − m) = 0 + 0 + 0 + 4 + 3 + 0 + 0 + 0
m = −4
=7

3
When n = −1 ; y(−1) = ∑ x(m) h(−1 − m) = 0 + 0 + 1+ 12 + 5 + 0 + 0 = 18
m = −3

3
When n = 0 ; y(0) = ∑ x(m) h(0 − m)
m = −2
= 0 − 2 + 3 + 20 − 1+ 0 = 20

3
When n = 1 ; y(1) = ∑ x(m) h(1 − m)
m = −1
= 1 − 6 + 5 − 4 − 2 = −6
Solution for Exercise Problems E2. 19
4
When n = 2 ; y(2) = ∑ x(m) h(2 − m) = 0 + 3 − 10 − 1− 8 + 0 = −16
m = −1
5
When n = 3 ; y(3) = ∑ x(m) h(3 − m) = 0 + 0 + 5 + 2 − 2 + 0 + 0 = 5
m = −1
6
When n = 4 ; y(4) = ∑ x(m) h(4 − m) = 0 + 0 + 0 − 1+ 4 + 0 + 0 + 0 = 3
m = −1
7
When n = 5 ; y(5) = ∑ x(m) h(5 − m) = 0 + 0 + 0 + 0 − 2 + 0 + 0 + 0 + 0 = −2
m = −1

l
∴ y(n) = 1, 7, 18, 20, − 6, − 16, 5, 3, − 2 q
A

E2.13. b) h(n) =
|RS1 ; 0 ≤n≤ 2

T|0 ; n ≥ 3
n
x(n) = a u(n) ; |a| < 1
Solution
x (m ) h (m ) h ( −m )
0
a =1 1 1 1
a
2
a
3
a

0 1 2 3 m 0 1 2 3 6 m −3 −2 −1 0
4 5 −6 −5 −4 m
By convolution formula,
∞
y(n) = ∑ x(m) h(n − m)
m = −∞

∞ ∞
When n = 0 ; y(0) = ∑ x(m) h(0 − m) = ∑ v0 (m)
m=0 m=0

x (m ) h( −m ) v 0 (m )
0 0
a =1 1 1=a
a
a
2
x ⇒
3
0
a ∴ y (0) = a

0 1 2 3 m −3 −2 −1 0 1 m 0 1 2 m
∞ ∞
When n = 1 ; y(1) = ∑ x(m) h(1 − m) = ∑ v1(m)
m=0 m=0

x (m ) h (1 − m ) v 1 (m )
0
0
a =1 1=a
1 1
a 1
a
a
2
⇒
3
x y (1) = a
0
+ a
1
a

0 1 2 3 m −2 −1 0 1 m −1 0 1 2 m
Similarly,
When, n = 2 ; y(2) = a0 + a1 + a2
When, n = 3 ; y(3) = a1 + a2 + a3
When, n = 4 ; y(4) = a2 + a3 + a4
When, n = 5 ; y(5) = a3 + a4 + a5
n
∴ y(n) = ∑a
k =0
k
; for n = 0, 1, 2

n
= ∑a
k =n − 2
k
; for n > 2
E2. 20 DSP, Chapter 2 -Discrete Time Signals and Systems
E2.14. Perform circular convolution of the two sequences,
l
a) x1 (n) = 1, 2, − 1 −1 q and x2 (n) = 2, 4, 6, 8 l q
A A
Solution
N −1
Let x 3 (n) = x1(n) ∗ x 2 (n) = ∑ x (m) x
m=0
1 2,n (m) ; x 2,n (m) = x 2 ((n − m))N ; N = 4

x 1 (1) = 2 x 2 (1) = 4 x2 (3) = 8

x 1 ( 2 ) = −1 x 1( m ) x 1( 0 ) = 1 x2 (2) = 6 x 2 (m ) x 2 (0 ) = 2 x 2 (2 ) = 6 x 2 ( −m ) x 2 (0) = 2

x 1 ( 3 ) = −1 x 2 (3 ) = 8 x 2 (1) = 4

3 3
When n = 0 ; x 3 (0) = ∑ x (m) x
m=0
1 2,0 (m) = ∑ v (m) = 2 + 16 − 6 − 4 = 8
m=0
0

2 8 16

−1 x 1 (m ) 1 x 6 x 2 ,0 ( m ) 2 ⇒ −6 v 0 (m ) 2

−1 4 −4

3 3
When n = 1 ; x 3 (1) = ∑ x (m) x
m=0
1 2,1(m) = ∑ v (m) = 4 + 4 − 8 − 6 = −6
m=0
1

2 2 4

−1 x 1(m) 1 x 8 x 2, 1(m ) 4 ⇒ −8 v 1(m) 4

−1 6 −6

3 3
When n = 2; x 3 (2) = ∑ x (m) x
m=0
1 2,2 (m) = ∑ v (m) = 6 + 8 − 2 − 8 = 4
m=0
2

2 4 8

−1 x1(m) 1 x 2 x 2 , 2 (m) 6 ⇒ −2 v 2 (m ) 6

−1 8 −8

3 3
When n = 3 ; x 3 (3) = ∑ x (m) x
m=0
1 2,3 (m) = ∑ v (m) = 8 + 12 − 4 − 2 = 14
m=0
3

2 6 12

−1 x 1(m) 1 x 4 x 2, 3 (m ) 8 ⇒ −4 v 3 (m) 8

−1 2 −2

l
x 3 (n) = 8, − 6, 4, 14 q
A
b) Perform the circular convolution of the two sequences,
l
x1 (n) = 0, 0.6, − 1, 1.5, 2 ; x2 (n) = −2, 3, 0.2, 0.7, 0.8 q l q
A A
Solution
The response x3(n) of the system is given by convolution of x1(n) and x2(n).
Solution for Exercise Problems E2. 21

N −1 4
x 3 (n) = x1(n) ∗ x 2 (n) = ∑ x (m) x ((n − m))
m=0
1 2 N = ∑ x (m) x ((n − m))
m=0
1 2 5

4
= ∑ x (m) x
m=0
1 2,n (m)

m –4 –3 –2 –1 0 1 2 3 4

x1(m) 0 0.6 –1 1.5 2

x2(m) –2 3 0.2 0.7 0.8

x2((–m))5 = x2,0(m) 0.8 0.7 0.2 3 –2 0.8 0.7 0.2 3

x2((1–m))5 = x2,1(m) 0.8 0.7 0.2 3 –2 0.8 0.7 0.2

x2((2–m))5 = x2,2(m) 0.8 0.7 0.2 3 –2 0.8 0.7

x2((3–m))5 = x2,3(m) 0.8 0.7 0.2 3 –2 0.8

x2((4–m))5 = x2,4(m) 0.8 0.7 0.2 3 –2

When n = 0 ;
4
x 3 (0) = ∑ x (m) x
m=0
1 2,0 (m) = x1(0) x 2,0 (0) + x1(1) x 2,0 (1) + x1(2) x 2,0 (2) + x1(3) x 2,0 (3) + x1(4) x 2,0 (4)

b g b
= (0 × −2) + 0.6 × 0.8 + −1 × 0.7 + 1.5 × 0.2 + 2 × 3 = 6.08 g b g b g
Similarly

g b g b b g b g
When n = 1 ; x 3 (1) = (0 × 3) + 0.6 × −2 + −1 × 0.8 + 1.5 × 0.7 + 2 × 0.2 = −0.55

When n = 2 ; x (2) = (0 × 0.2) + b0.6 × 3g + b −1 × −2g + b1.5 × 0.8g + b2 × 0.7g = 6.4

3

When n = 3 ; x (3) = (0 × 0.7) + b0.6 × 0.2g + b −1 × 3g + b1.5 × −2g + b2 × 0.8g = − 4.28

3

When n = 4 ; x (4) = (0 × 0.8) + b0.6 × 0.7g + b −1 × 0.2g + b1.5 × 3g + b2 × −2g = 0.72

3

∴ x (n) = l6.08, − 0.55, 6.4, − 4 . 28, 0.72q

3

A
E2.15. The input x(n) and impulse response h(n) of an LTI system are given by,

x(n) = l − 1, 1, − 1, 1, − 1, 1 q and l
h(n) = −0.5, 0.5, − 1, 0.5, − 1, − 2 q
A A
Find the response of the system using,
a) Linear Convolution
b) Circular Convolution

Solution

a) Response of LTI System Using Linear Convolution

+∞
Let, y(n) = x(n) ∗ h(n) = ∑ x(m) h(n − m)
m= −∞
+∞
= ∑ x(m) h (m) ; where h (m) = h(n − m)
n n
m= −∞

x(n) starts at n = −1 and h(n) starts at n = 0.

∴ y(n) will start at n = 0 + ( −1) = − 1

Length of x(n) is 6 and h(n) is 6.

Hence length of y(n) is 6 + 6 –1 = 11. Also y(n) ends at n = 0 + (–1) + (6 + 6 –2) = 9

E2. 22 DSP, Chapter 2 -Discrete Time Signals and Systems

The 11 samples of y(n) computed by table method as shown below.

m –6 –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7 8 9

x(m) –1 1 –1 1 –1 1

h(m) –0.5 0.5 –1 0.5 –1 –2

h(–m) –2 –1 0.5 –1 0.5 –0.5

h–1(–m) –2 –1 0.5 –1 0.5 –0.5

h0(m) –2 –1 0.5 –1 0.5 –0.5

h1(m) –2 –1 0.5 –1 0.5 –0.5

h2(m) –2 –1 0.5 –1 0.5 –0.5

h3(m) –2 –1 0.5 –1 0.5 –0.5

h4(m) –2 –1 0.5 –1 0.5 –0.5

h5(m) –2 –1 0.5 –1 0.5 –0.5

h6(m) –2 –1 0.5 –1 0.5 –0.5

h7(m) –2 –1 0.5 –1 0.5 –0.5

h8(m) –2 –1 0.5 –1 0.5 –0.5

h9(m) –2 –1 0.5 –1 0.5 –0.5

4
When n = −1, y( −1) = ∑ x(m) h
m = −6
−1(m)

= x( −6) h−1(−6) + x( −5) h−1( −5) + x( −4)h−1(−4) + x( −3) h−1(−3) + x( −2) h−1(−2) + x( −1)h−1(−1)
+ x(0) h−1(0) + x(1) h−1(1) + x(2)h−1(2) + x(3) h−1(3) + x(4) h−1(4)
= 0 + 0 + 0 + 0 + 0 + (−1 × − 0.5 ) + 0 + 0 + 0 + 0 + 0 = 0.5

4
When n = 0 ; y(0) = ∑ x(m) h (m) 0 b g b
= 0 + 0 + 0 + 0 + −1 × 0.5 + 1 × −0.5 + 0 + 0 + 0 + 0 = −1 g
m= −5

4
When n = 1 ; y(1) = ∑ x(m) h (m) 1 b g b g b
= 0 + 0 + 0 + −1 × −1 + 1 × 0.5 + −1 × −0.5 + 0 + 0 + 0 = 2 g
m= −4

4
When n = 2 ; y(2) = ∑ x(m) h (m) 2 b g b g b
= 0 + 0 + −1 × 0.5 + 1 × −1 + −1 × 0.5 + 1 × −0.5 + 0 + 0 = −2.5 g b g
m= −3

4
When n = 3 ; y(3) = ∑ x(m) h (m) 3 b g b g b g b
= 0 + −1 × −1 + 1 × 0.5 + −1 × −1 + 1 × 0.5 + −1 × −0.5 + 0 = 3.5 g b g
m= −2
4
When n = 4 ; y(4) = ∑ x(m) h (m) 4 b g b g b g b
= −1 × −2 + 1 × −1 + −1 × 0.5 + 1 × −1 + −1 × 0.5 + 1 × −0.5 = −1.5 g b g b g
m= −1
5
When n = 5 ; y(5) = ∑ x(m) h (m) 5 b g b g b g b
= 0 + 1 × −2 + −1 × −1 + 1 × 0.5 + −1 × −1 + 1 × 0.5 + 0 = 1 g b g
m= −1
6
When n = 6 ; y(6) = ∑ x(m) h (m) 6 b g b g b
= 0 + 0 + −1 × −2 + 1 × −1 + −1 × 0.5 + 1 × −1 + 0 + 0 = −0.5 g b g
m= −1
7
When n = 7 ; y(7) = ∑ x(m) h (m) 7 b g b g b
= 0 + 0 + 0 + 1 × −2 + −1 × −1 + 1 × 0.5 + 0 + 0 + 0 = −0.5 g
m= −1
8
When n = 8 ; y(8) = ∑ x(m) h (m) 8 b g b
= 0 + 0 + 0 + 0 + −1 × −2 + 1 × −1 + 0 + 0 + 0 + 0 = 1 g
m= −1
9
When n = 9 ; y(9) = ∑ x(m) h (m) 9 b g
= 0 + 0 + 0 + 0 + 0 + 1 × −2 + 0 + 0 + 0 + 0 + 0 = − 2
m= −1

The response of LTI system y(n) is,

l
y(n) = 0.5, − 1, 2, − 2.5, 3.5, − 1. 5, 1, − 0.5, − 0.5, 1, − 2 q
A
Solution for Exercise Problems E2. 23
b) Response of LTI system using circular convolution
The response y(n) is 11-point sequence. The y(n) start at n = –1 and end of n = 9. Hence both x(n) and h(n)
should be converted to 11-point sequence such that they start at n = –1 and end at n = 9 by appending zeros for
missing samples.

l
∴ x(n) = −1, 1, − 1, 1, − 1, 1, 0, 0, 0, 0, 0 q
A
l
h(n) = 0, − 0.5, 0.5, − 1, 0.5, − 1, − 2, 0, 0, 0, 0 q
A
9 9
Now, y(n) = x(n) ∗ h(n) = ∑ x(m) h((n − m))
m= −1
11 = ∑ x(m) h (m) ;
m= −1
n where hn (m) = h((n − m))11

The 11 samples of y(n) are computed by table method as shown below.

m –10 –9 –8 –7 –6 –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7 8 9

x(m) –1 1 –1 1 –1 1 0 0 0 0 0

h(m) 0 –0.5 0.5 –1 0.5 –1 –2 0 0 0 0

h(–m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0

h–1(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0 0 0 0 –2 –1 0.5 –1 0.5

h0(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0 0 0 0 –2 –1 0.5 –1

h1(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0 0 0 0 –2 –1 0.5

h2(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0 0 0 0 –2 –1

h3(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0 0 0 0 –2

h4(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0 0 0 0

h5(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0 0 0

h6(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0 0

h7(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0

h8(m) 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0

h9(m) 0 0 0 –2 –1 0.5 –1 0.5 –0.5 0 0 0 0 0 –2 –1 0.5 –1 0.5 –0.5

9
When n = −1 ; y( −1) = ∑ x(m) h −1(m) b g
= −1 × −0.5 + 0 + 0 + 0 + 0 + 0 + 0 + 0 + 0 + 0 + 0 = 0.5
m= −1

9
When n = 0 ; y(0) = ∑ x(m) h (m) 0 b g b g
= −1 × 0.5 + 1 × −0.5 + 0 + 0 + 0 + 0 + 0 + 0 + 0 + 0 + 0 = −1
m= −1
9
When n = 1 ; y(1) = ∑ x(m) h (m) 1 b g b g b g
= −1 × −1 + 1 × 0.5 + −1 × −0.5 + 0 + 0 + 0 + 0 + 0 + 0 + 0 + 0 = 2
m= −1
9
When n = 2 ; y(2) = ∑ x(m) h (m) 2 b g b g b g b g
= −1 × 0.5 + 1 × −1 + −1 × 0.5 + 1 × −0.5 + 0 + 0 + 0 + 0 + 0 + 0 + 0 = −2.5
m= −1
9
When n = 3 ; y(3) = ∑ x(m) h (m) 3 b g b g b g b g b
= −1 × −1 + 1 × 0.5 + −1 × −1 + 1 × 0.5 + −1 × −0.5 + 0 + 0 + 0 + 0 + 0 + 0 = 3.5 g
m= −1
9
When n = 4 ; y(4) = ∑ x(m) h (m) 4 b g b g b g b g b
= −1 × −2 + 1 × −1 + −1 × 0.5 + 1 × −1 + −1 × 0.5 + 1 × −0.5 + 0 + 0 + 0 + 0 + 0 = −1.5 g b g
m= −1
9
When n = 5 ; y(5) = ∑ x(m) h (m) 5 b g b g b g b
= 0 + 1 × −2 + −1 × −1 + 1 × 0.5 + −1 × −1 + 1 × 0.5 + 0 + 0 + 0 + 0 + 0 = 1 g b g
m= −1
9
When n = 6 ; y(6) = ∑ x(m) h (m) 6 b g b g b g b
= 0 + 0 + −1 × −2 + 1 × −1 + −1 × 0.5 + 1 × −1 + 0 + 0 + 0 + 0 + 0 = −0.5 g
m= −1
9
When n = 7 ; y(7) = ∑ x(m) h (m) = 0 + 0 + 0 + b1× −2g + b−1 × −1g + b1× 0.5g + 0 + 0 + 0 + 0 + 0 = −0.5
m= −1
7
E2. 24 DSP, Chapter 2 -Discrete Time Signals and Systems
9
When n = 8 ; y(8) = ∑ x(m) h (m) 8 b g b g
= 0 + 0 + 0 + 0 + −1 × −2 + 1 × −1 + 0 + 0 + 0 + 0 + 0 = 1
m= −1
9
When n = 9 ; y(9) = ∑ x(m) h (m) 9 b g
= 0 + 0 + 0 + 0 + 0 + 1 × −2 + 0 + 0 + 0 + 0 + 0 = −2
m= −1

The response of LTI system y(n) is,

l
y(n) = 0.5, − 1, 2, − 2.5, 3.5, − 1. 5, 1, − 0.5, − 0.5, 1, − 2 q
A
E2.16. Perform linear convolution of the following sequences by,
i) Overlap add method
ii) Overlap save method
x(n) = 1, l − 1, 2 1, − 1, 2, 1, − 1, 2 q
l
h(n) = 2, 3, − 1 q
Solution
Overlap Add Method
l
x(n) = 1, − 1, 2, 1, − 1, 2, 1, − 1, 2 q
x1(n) = 1, n = 0 x 2 (n) = 1, n = 3 x 3 (n) = 1, n = 6
= −1, n = 1 = −1, n = 4 = −1, n = 7
= 2, n = 2 = 2, n = 5 = 2, n = 8
l
h(n) = 2, 3, − 1 q
Let, y1(n), y2(n), y3(n) be output of linear convolution of x1(n), x2(n), x3(n) with h(n) respectively.
Here, h(n) starts at nh = 0.
x1(n) starts at, n = n1 = 0 \ y1(n) starts at, n = 0 + 0 = 0
x2(n) starts at, n = n2 = 3 \ y2(n) starts at, n = 3 + 0 = 3
x3(n) starts at, n = n3 = 6 \ y3(n) starts at, n = 6 + 0 = 6
Here, N1 = 9, N2 = 3, N3 = 3
N2 – 1 = 2
N2 + N3 – 1 = 5
Convolution output of each section will consists of 3 + 3 – 1 = 5 samples.
Convolution of Section - 1

m –2 –1 0 1 2 3 4

x(m) 1 –1 2

h(m) 2 3 –1

h(–m)=h0(m) –1 3 2

h1(m) –1 3 2

h2(m) –1 3 2

h3(m) –1 3 2

h4(m) –1 3 2
+∞
y1(n) = x1(n) ∗ h(n) = ∑ x (m) h(n − m)
m= −∞
1

+∞
= ∑ x (m) h (m) ;
m= −∞
1 n n = 0, 1, 2, 3, 4

where hn (m) = h(n − m)

When n = 0 ; y1(0) = ∑ x (m) h (m) = 0 + 0 + 2 + 0 + 0 = 2
1 0

When n = 1 ; y (1) = ∑ x (m) h (m) = 0 + 3 − 2 + 0 = 1

1 1 1

When n = 2 ; y (2) = ∑ x (m) h (m) = − 1 − 3 + 4 = 0

1 1 2

When n = 3 ; y (3) = ∑ x (m) h (m) = 0 + 1+ 6 + 0 = 7

1 1 3

When n = 4 ; y (4) = ∑ x (m) h (m) = 0 + 0 − 2 + 0 + 0 = −2

1 1 4
Solution for Exercise Problems E2. 25
∴ y1(n) = l 2, 1, 0, 7, − 2 q
An=0

Convolution of sections 2 and 3

The convolution of section –2 and 3 are identical to that of section -1 except the starting value of n.

∴ y 2 (n) = l 2, 1, 0, 7, − 2 q
An= 3

∴ y 3 (n) = l 2, 1, 0, 7, − 2 q
An= 6

Overall Output

n 0 1 2 3 4 5 6 7 8 9 10

y1(n) 2 1 0 7 –2

y2(n) 2 1 0 7 –2

y3(n) 2 1 0 7 –2

y(n) 2 1 0 9 –1 0 9 –1 0 7 –2

l
y(n) = 2, 1, 0, 9, − 1, 0, 9, − 1, 0, 7, − 2 q
Overlap save Method

l
x(n) = 1, − 1, 2, 1, − 1, 2, 1, − 1, 2 q
h(n) = l2, 3, − 1 q
N1 = 9, N2 = 3, Let N3 = 3
x1(n) = 1, n = 0 x 2 (n) = 1 , n = 3 x 3 (n) = 1, n = 6
= −1, n = 1 = −1, n = 4 = −1, n = 7
= 2, n = 2 = 2, n = 5 = 2, n = 8

Let, y1(n), y2(n) and y3(n) be output of linear convolution of x1(n), x2(n) and x3(n) with h(n) respectively.
Now each output will consists of 3 + 3 – 1 = 5 samples. Hence convert x1(n), x2(n), x3(n) and h(n) to 5 sample sequence as shown
below.
x1(n) = 1, n = 0 x 2 (n) = 1, n = 3 x 3 (n) = 1, n = 6
= −1, n = 1 = −1, n = 4 = −1, n = 7
= 2, n = 2 = 2, n = 5 = 2, n = 8
= 1, n = 3 = 1, n = 6 = 0, n = 9
= −1, n = 4 = −1, n = 7 = 0, n = 10

h(n) = {2, 3, –1, 0, 0}

Now perform circular convolution of each section with h(n).
x1(n) starts at, n = 0 ; \ y1(n) starts at, n = 0.
x2(n) starts at, n = 3 ; \ y2 (n) starts at, n = 3.
x3(n) starts at, n = 6 ; \ y3(n) starts at, n = 6.

Convolution of Section 1

m –4 –3 –2 –1 0 1 2 3 4

x1(m) 1 –1 2 1 –1

h(m) 2 3 –1 0 0

h(–m)= h0(m) 0 0 –1 3 2 0 0 –1 3

h1(m) 0 0 –1 3 2 0 0 –1

h2(m) 0 0 –1 3 2 0 0

h3(m) 0 0 –1 3 2 0

h4(m) 0 0 –1 3 2
E2. 26 DSP, Chapter 2 -Discrete Time Signals and Systems
N −1 4
y1(n) = x1(n) ∗ h(n) = ∑ x (m) h((n − m)) = ∑ x (m) h (m) ;
m=0
1 n
m=0
1 n where hn (m) = h((n − m))5

4
When n = 0 ; y1(0) = ∑ x (m) h (m) =
m=0
1 0 2 + 0 + 0 − 1 − 3 = −2

4
When n = 1 ; y1(1) = ∑ x (m) h (m) =
m=0
1 1 3 − 2 + 0 + 0 + 1= 2

4
When n = 2 ; y1(2) = ∑ x (m) h (m) =
1 2 − 1− 3 + 4 + 0 + 0 = 0
m=0

4
When n = 3 ; y1(3) = ∑ x (m) h (m) =
m=0
1 3 0 + 1+ 6 + 2 + 0 = 9

4
When n = 4 ; y1(4) = ∑ x (m) h (m) =
m=0
1 4 0 + 0 − 2 + 3 − 2 = −1

∴ y1(n) = l − 2, 2, 0, 9, − 1 q
A
n=0

Convolution of Section 2

The output of convolution of section -2 will be identical to that of section-1 except the starting value of n.

∴ y 2 (n) = l − 2, 2, 0, 9, − 1 q
A
n= 3

Convolution of Section 3

m –4 –3 –2 –1 0 1 2 3 4 5 6 7 8 9 10

x3(m) 1 –1 2 0 0

h(m) 2 3 –1 0 0

h0(m) 0 0 –1 3 2

h6(m) 0 0 –1 3 2 0 0 –1 3

h7(m) 0 0 –1 3 2 0 0 –1

h8(m) 0 0 –1 3 2 0 0

h9(m) 0 0 –1 3 2 0

h10(m) 0 0 –1 3 2

10 10
y 3 (n) = x 3 (n) ∗ h(n) = ∑ x (m) h((n − m)) = ∑ x (m) h (m) ;
m=6
3 5
m=6
3 n where hn (m) = h((n − m))5

10
When n = 6 ; y 3 (6) = ∑ x (m) h (m) =
m=6
3 6 2+0+0+0+0 = 2

10
When n = 7 ; y 3 (7) = ∑ x (m) h (m) =
3 7 3−2+0+0+0=1
m=6

10
When n = 8 ; y 3 (8) = ∑ x (m) h (m) =
m=6
3 8 − 1− 3 + 4 + 0 + 0 = 0

10
When n = 9 ; y 3 (9) = ∑ x (m) h (m) =
3 9 0 + 1+ 6 + 0 + 0 = 7
m=6

10
When n = 10 ; y 3 (10) = ∑ x (m) h
m=6
3 10 (m) = 0 + 0 − 2 + 0 + 0 = −2

∴ y 3 (n) = l 2, 1, 0, 7, − 2 q
A n= 6
Solution for Exercise Problems E2. 27
Overall Output
n 0 1 2 3 4 5 6 7 8 9 10

y1(n) –2 2 0 9 –1

y2(n) –2 2 0 9 –1

y3(n) 2 1 0 7 –2

y(n) * * 0 9 –1 0 9 –1 0 7 –2

y(n) = {*, *, 0, 9, –1, 0, 9, –1, 0, 7, –2}

Hence both the results are same except the first (N2 – 1) samples.
E2.17. Perform crosscorrelation of the sequences,
l
x(n) = −1, 2, 3 − 4 q and l
y(n) = 2, − 1, − 3 q
A A
Solution
The crosscorrelation sequence rxy(m) is given by,
∞
rxy (m) = ∑ x(n) y(n − m)
n= −∞

The x(n) starts at n = 0, and has 4 samples.

\ n1 = 0, N1 = 4
The y(n) start at, n = –1 and has 3 samples.
\ n2 = –1 , N2 = 3

Now rxy(m) will have, N1 + N2 – 1 = 4 + 3 –1 = 6 samples.

The initial value of m = mi = n1 – (n2 + N2 – 1) = 0 – (–1 + 3 – 1) = –1

The final value of m = mf = mi + (N1 + N2 – 2) = –1 + (4 + 3 –2) = 4

The 6 samples of crosscorrelation sequence are computed using table method as shown below.

n –2 –1 0 1 2 3 4 5

x(n) –1 2 3 –4

y(n) 2 –1 –3

y(n–(–1))= y–1(n) 2 –1 –3

y(n–0)= y0(n) 2 –1 –3

y(n–1)= y1(n) 2 –1 –3

y(n–2)= y2(n) 2 –1 –3

y(n–3)= y3(n) 2 –1 –3

y(n–4)= y4(n) 2 –1 –3

Each sample of rxy(m) is given by,

+∞ +∞
rxy (m) = ∑ x(n) y(n − m) = ∑ x(n) y
n= −∞ n= −∞
m (n) ; where ym (n) = y(n − m)

3
When m = −1 ; rxy ( −1) = ∑ x(n) y
n= −2
−1(n) = 0+0+3+0+0+0 = 3

3
When m = 0 ; rxy (0) = ∑ x(n) y (n) =
n= −1
0 0 + 1 − 6 + 0 + 0 = −5

3
When m = 1 ; rxy (1) = ∑ x(n) y (n) =
n=0
1 − 2 − 2 − 9 + 0 = −13

3
When m = 2 ; rxy (2) = ∑ x(n) y (n) =
n=0
2 0 + 4 − 3 + 12 = 13
E2. 28 DSP, Chapter 2 -Discrete Time Signals and Systems
4
When m = 3 ; rxy (3) = ∑ x(n) y (n) =
n=0
3 0 + 0 + 6 + 4 + 0 = 10

5
When m = 4 ; rxy (4) = ∑ x(n) y
n=0
4 (n) = 0 + 0 + 0 − 8 + 0 + 0 = −8

∴ rxy (m) =
RS 3, −5, − 13, 13, 10, − 8
UV
T A W
E2.18. Determine the autocorrelation sequence for x(n) = {1, 4, 3, –5, 2}
-
Solution
The autocorrelation sequence rxx(m) is given by,
+∞
rxx (m) = ∑ x(n) x(n − m)
n= −∞

The x(n) starts at n = –1, and has 5 samples.

\ nx = –1, and N = 5
The rxx(m) will have, 2N –1 = 10 – 1 = 9 samples
The initial value of m = mi = – (N – 1) = – (5 – 1) = –4
The final value of m = mf = mi + (2 N – 2) = –4 + (10 – 2) = 4

n –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7

x(n) 1 4 3 –5 2

x–4(n) 1 4 3 –5 2

x–3(n) 1 4 3 –5 2

x–2(n) 1 4 3 –5 2

x–1(n) 1 4 3 –5 2

x0(n) 1 4 3 –5 2

x1(n) 1 4 3 –5 2

x2(n) 1 4 3 –5 2

x3(n) 1 4 3 –5 2

x4(n) 1 4 3 –5 2

Each sample of autocorrelation sequence rxx(m) is given by,

+∞ +∞
rxx (m) = ∑ x(n) x(n − m) = ∑ x(n) x
n= −∞ n= −∞
m (n) ; where xm (n) = x(n − m)

3
When m = −4 ; rxx ( −4) = ∑ x(n) x
n= −5
−4 (n) = 0+0+0+0+2+0+0+0+0 = 2

3
When m = −3 ; rxx ( −3) = ∑ x(n) x
n= −4
−3 (n) = 0+0+0− 5+8+0+0+0 = 3

3
When m = −2 ; rxx ( −2) = ∑ x(n) x
n= −3
−2 (n) = 0 + 0 + 3 − 20 + 6 + 0 + 0 = −11

3
When m = −1 ; rxx ( −1) = ∑ x(n) x
n= −2
−1(n) = 0 + 4 + 12 − 15 − 10 + 0 = −9

3
When m = 0 ; rxx (0) = ∑ x(n) x (n) =
n= −1
0 1+ 16 + 9 + 25 + 4 = 55

4
When m = 1 ; rxx (1) = ∑ x(n) x (n) =
n= −1
1 0 + 4 + 12 − 15 − 10 + 0 = −9

5
When m = 2 ; rxx (2) = ∑ x(n) x (n) =
n= −1
2 0 + 0 + 3 − 20 + 6 + 0 + 0 = −11
Solution for Exercise Problems E2. 29
6
When m = 3 ; rxx (3) = ∑ x(n) x (n) = 3 0+0+0− 5 + 8+0+0+0= 3
n= −1
7
When m = 4 ; rxx (3) = ∑ x(n) x (n) = 4 0+0+0+0+2+ 0+0+0+0= 2
n= −1

∴ rxx (m) = l 2, 3, − 11, − 9, 55, − 9, − 11, 3, 2 q

A
E2.19. Find the inverse system for the following discrete time system
n
y(n) = ∑ c p x(p − 2) ; for n ≥ 0.
p=0

Solution n
Given that, y(n) = ∑ c x(p − 2)
p=0
p
; for n ≥ 0

0
When n = 0 ; y(0) = ∑ c x(p − 2) = c x(−2) = x(−2)
p=0
p 0
⇒ x(−2) = y(0)

1
When n = 1 ; y(1) = ∑ c x(p − 2) = c x(−2) + c x(−1)
p=0
p 0 1

= x( −2) + c x( −1)
1
= y(0) + c x(−1) ⇒ x( −1) = y(1) − y(0)
c
2
When n = 2 ; y(2) = ∑ c x(p − 2) = c x(−2) + c x(−1) + c x(0)
p=0
p 0 1 2

= x(−2) + cx(−1) + c 2 x(0)

= y(0) + y(1) − y(0) + c 2 x(0)

1
= y(1) + c 2 x(0) ⇒ x(0) = y(2) − y(1)
c2
1
Therefore, in general, x(n) = n+2
y(n + 2) − y(n + 1) ; for n ≥ −1 with initial condition x( −2) = y(0).
c

E2.20. A discrete time system is excited by an input x(n), and the response is, y(n) = 4, 3, 6, 7.5, 3, 30, − 8 . If the l q
A
l q
impulse response of the system is h(n) = 2, 4, − 2 , then what will be the input to the system?
A
Solution
Let, N1 = Number of samples in x(n)

N2 = Number of samples in h(n)

N3 = Number of samples in y(n)

Now, N3 = N1 + N2 – 1 Þ N1 = N3 – N2 + 1 = 7 – 3 + 1 = 5 samples

Each sample of x(n) is given by,

1 LM n −1 O
x(n) = y(n) −
MN ∑ x(m) h(n − m)PP
h(0) m=0 Q
y(0) 4
When n = 0 ; x(0) = = =2
h(0) 2
LM
1 OP 1 1
When n = 1 ; x(1) =
MN ∑
h(0)
y(1) −
m= 0
x(m) h(1 − m) =
PQ h(0)
y(1) − x(0) h(1) =
2
3 − (2 × 4) = −2.5

1 L O 1 1
1
When n = 2 ; x(2) = M
h(0) MN
y(2) − ∑ x(m) h(2 − m)P =
PQ h(0) y(2) − x(0) h(2) − x(1) h(1) =
2
b g
6 − (2 × −2) − −2.5 × 4 = 10
m=0
E2. 30 DSP, Chapter 2 -Discrete Time Signals and Systems

1 LM 2 O= 1
When n = 3 ; x(3) = y(3) −
MN ∑ x(m) h(3 − m)PP y(3) − x(0) h(3) − x(1) h(2) − x(2) h(1)
h(0) m=0 Q h(0)
1
=
2
b g
7.5 − (2 × 0) − −2.5 × −2 − (10 × 4) = −18 . 75

1 LM 2 O= 1
When n = 4 ; x(4) = y(4) −
MN ∑ x(m) h(4 − m)PP y(4) − x(0) h(4) − x(1) h(3) − x(2) h(2) − x(3) h(1)
h(0) m=0 Q h(0)
1
=
2
b g
3 − (2 × 0) − −2.5 × 0 − (10 × −2) − (−18.75 × 4) = 49

l q l
∴ x(n) = x(0), x(1), x(2), x(3), x(4) = 2, − 2.5, 10, − 18.75, 49 q
A
E2.21. Perform circular correlation of the sequences, x(n) = −1, 1, 2, 6 l q and l q
y(n) = 4, − 2, − 1, 2

Solution

Let rxy (m) be the sequence obtained by circular correlation of x(n) and y(n).

The circular correlation is given by,

N −1 N −1
rxy (m) = ∑ x(n) y((n − m)) = ∑ x(n) y
n= 0
N
n= 0
m (n), where y m (m) = y((n − m))N

Here, N = 4, The circular correlation is performed by table method as shown below.

q 0 1 2 3 4 5 6 7

x(n) –1 1 2 6

y(n) 4 –2 –1 2

y0(n) 4 –2 –1 2 4 –2 –1 2

y1(n) 2 4 –2 –1 2 4 –2 –1

y2(n) –1 2 4 –2 –1 2 4 –2

y3(n) –2 –1 2 4 –2 –1 2 4

3
When m = 0 ; rxy (0) = ∑ x(n) y (n) = −4 − 2 − 2 + 12 = 4
0
n= 0

3
When m = 1 ; rxy (1) = ∑ x(n) y (n) = −2 + 4 − 4 − 6 = −8
1
n=0

3
When m = 2 ; rxy (2) = ∑ x(n) y (n) = 1+ 2 + 8 − 12 = −1
2
n= 0

3
When m = 3 ; rxy (3) = ∑ x(n) y (n) = 2 − 1+ 4 + 24 = 29
3
n= 0

l
∴ rxy (m) = 4, − 8, − 1, 29 q
 Chapter 3

Z-Transform

3.1 Introduction
Transform techniques are an important tool in the analysis of signals and systems. The Laplace
transforms are popularly used for analysis of continuous time signals and systems. Similarly Z-transform
plays an important role in analysis and representation of discrete time signals and systems. The Z-transform
provides a method for the analysis of discrete time signals and systems in the frequency domain which is
generally more efficient than its time domain analysis.
The Z-transform of x(n) will convert the time domain signal x(n) to z-domain signal X(z), where the
signal becomes a function of complex variable z.
jv z -plan e
The complex variable z is defined as, z1
r1
z = u + jv = r ejw ω1
−u u
where, u = Real part of z ; v = Imaginary part of z
r = u 2 + v 2 = Magnitude of z −jv
F ig 3 .1 : z-p la ne .
ω = tan −1 v = Phase or Argument of z
u
The u and v takes value from –¥ to +¥ . A two dimensional complex plane with values of u on
horizontal axis and values of v on vertical axis as shown in fig 3.1 is called z-plane. A circle with radius r1 in
z-plane represents all values of z1 having same magnitude r1 with variable phase w 1, where w 1 = 0 to 2p.

History of Z-Transform
A transform of a sampled signal or sequence was defined in 1947 by W. Hurewicz as,
∞
z [ f ( kT)] = ∑ f ( kT) z− k
k =0
Chapter 3 - Z - Transform 3. 2
which was later denoted in 1952 as Z-transform by a sampled-data control group at Columbia University led
by professor John R. Raggazini and including L.A. Zadeh, E.I. Jury, R.E. Kalman, J.E. Bertram, B. Friedland
and G.F. Franklin, (Source : www.ling.upenn.edu).
Definition of Z-Transform
Let, x(n) = Discrete time signal
X(z) = Z-transform of x(n)
The Z-transform of a discrete time signal, x(n) is defined as,
∞
X( z) = ∑ x( n ) z − n ; where, z is a complex variable. .....(3.1)
n = −∞

The Z-transform of x(n) is symbolically denoted as,

Z{x(n)} ; where, Z is the operator that represents Z-transform.
+∞
∴ X( z) = Z x(n) = l q ∑ x ( n) z −n

n = −∞

Since the time index n is defined for both positive and negative values, the discrete time signal x(n) in
equation (3.1) is considered to be two-sided and the transform is called two-sided Z-transform. If the signal
x(n) is one-sided signal, [i.e., x(n) is defined only for positive value of n] then the Z-transform is called
one-sided Z-transform.
The one-sided Z-transform of x(n) is defined as,
+∞
.....(3.2)
X( z) = Z{x( n)} = ∑ x( n) z − n
n= 0

The computation of X(z) involves summation of infinite terms which are functions of z. Hence it is
possible that the infinite series may not converge to finite value for certain values of z. Therefore for every
X(z) there will be a set of values of z for which X(z) can be computed. Such a set of values will lie in a particular
region of z-plane and this region is called Region Of Convergence (ROC) of X(z).
Inverse Z-Transform
Let, X(z) be Z-transform of x(n). Now the signal x(n) can be uniquely determined from X(z) and its
region of convergence (ROC).
The inverse Z-transform of X(z) is defined as,

x( n) =
1
2 πj z
c
X(z) z n −1 dz .....(3.3)

The inverse Z-transform of X(z) is symbolically denoted as,

Z–1 {X(z)} ; where, Z–1 is the operator that represents the inverse Z-transform

∴ x(n) = Z −1 {X( z)} =

1
2 πj z
c
X(z) z n −1 dz

We also refer x(n) and X(z) as a Z-transform pair and this relation is expressed as,
Z
x(n) ¬ ® X(z)
Z-1
3. 3 Digital Signal Processing

Proof :

Consider the definition of Z-transform of x(n),

+∞ +∞
X( z) = ∑ x(n) z −n
= ∑ x(k) z −k
Let n → k
n = −∞ k = −∞
+∞
X( z) zn −1 = ∑ x(k) z −k
zn −1 Multiply both sides by zn −1
k = −∞
Let us integrate the above equation on both sides over a closed contour "C" within the ROC of X(z) which
encloses the origin.

∴
zc
X(z) zn −1 dz =
zc k = −∞
+∞

∑ x(k) z n −1− k
dz
Interchanging the order of
=
+∞

∑
k = −∞
x(k)
z c
zn −1− kdz summation and integration.
Multiply and divide by 2pj.

By Cauchy integral theorem,

= 2πj
+∞

∑
k = −∞
x(k)
1
2πj z
c
z n −1− k
dz .....(3.4)

1
2πj c z
zn −1− k dz = 1 ; k = n
= 0 ; k ≠ n
On applying Cauchy integral theorem the equation (3.4) reduces to,

z c
X(z) zn −1 dz = 2πj x(n)
+∞

∑ x(k)
k = −∞ n=k
= x(n)

∴ x(n) =
1
2πj zc
X(z) zn −1 dz

Geometric Series
The Z-transform of a discrete time signal involves convergence of geometric series. Hence the following
two geometric series sum formula will be useful in evaluating Z-transform.
1. Infinite geometric series sum formula.
If C is a complex constant and 0 < |C|< 1, then,
∞
1
∑= Cn = 1− C
.....(3.5)
n 0

2. Finite geometric series sum formula.

If C is a complex constant and,

N −1 N
1 − CN CN − 1 C N +1 − 1
When C ≠ 1, ∑ Cn =
1− C
=
C −1
or ∑ Cn =
C −1
.....(3.6)
n= 0 n= 0

N−1 N
When C = 1, ∑= Cn = N or ∑= Cn = N +1 .....(3.7)
n 0 n 0

Note : The infinite geometric series sum formula requires that the magnitude of C be strictly less than unity,
but the finite geometric series sum formula is valid for any value of C.
Chapter 3 - Z - Transform 3. 4

3.2 Region of Convergence

Since the Z-transform is an infinite power series, it exists only for those values of z for which the series
converges. The region of convergence, (ROC) of X(z) is the set of all values of z, for which X(z) attains a finite
value. The ROC for the following six types of signals are discussed here.
Case i : Finite duration, right-sided (causal) signal
Case ii : Finite duration, left-sided (anticausal) signal
Case iii : Finite duration, two-sided (noncausal) signal
Case iv : Infinite duration, right-sided (causal) signal
Case v : Infinite duration, left-sided (anticausal) signal
Case vi : Infinite duration, two-sided (noncausal) signal
Case i : Finite duration, right-sided (causal) signal
Let, x(n) be a finite duration signal with N-samples, defined in the range 0 £ n £ (N – 1).
\ x(n) = {x(0), x(1), x(2),.....x(N–1)} jv
Now, the Z-transform of x(n) is, z -p la n e
N−1
u
X( z) = ∑ x(n) z− n R O C is e ntire
z -p la n e ex c e pt
n=0
z=0
= x(0) + x(1)z −1 + x(2)z −2 + ........+ x(N − 1) z− (N −1)
F ig 3 .2 : R O C o f fin ite
x(1) x(2) x(N − 1) d u ra tio n c ausa l sig n a l.
= x(0) + + + ..........+
z z2 zN − 1
In the above summation, when z = 0, all the terms except the first term become infinite. Hence the X(z)
exists for all values of z, except z = 0. Therefore, the ROC of finite duration right-sided (or causal signal) is
entire z-plane except z = 0.
Case ii : Finite duration, left-sided (anticausal) signal
Let, x(n) be a finite duration signal with N-samples, defined in the range –(N–1) £ n £ 0.
jv
\ x(n) = {x(–(N–1)),.....,x(–2), x(–1), x(0)} z -p la n e
Now, the Z-transform of x(n) is,
R O C is en tire u
0
z -p la n e ex c ep t
X( z ) = ∑ x(n) z −n
z=∞
n = − (N −1)

= x( − (N − 1)) z(N −1) + ....... + x( −2)z2 + x( −1) z + x(0) F ig 3.3 : R O C o f fin ite
d u ra tio n a ntica u sa l sig n a l.
In the above summation, when z = ¥ , all the terms except the last term become infinite. Hence the X(z)
exists for all values of z, except, z = ¥ . Therefore, the ROC of X(z) is entire z-plane, except z = ¥ .
Case iii : Finite duration, two-sided (noncausal) signal
Let, x(n) be a finite duration signal with N-samples, defined in the range –M £ n £ + M,
N −1
where, M =
2
l q
∴ x(n) = x( − M),......., x( −2), x( −1), x(0), x(1), x(2), ........x( M)
3. 5 Digital Signal Processing
Now, the Z-transform of x(n) is,
+M
X( z) = ∑− x(n) z− n
n= M
= x( − M ) z M + ....... + x( −2) z2 + x( −1) z + x(0) + x(1)z−1 + x(2)z −2 +......+ x( M) z − M
x(1) x(2) x( M)
= x( − M ) z M + ........ + x( −2) z2 + x( −1) z + x(0) + + + ...... +
z z2 zM
jv
z -p la n e
In the above summation, when z = 0, the terms with negative
R O C is e ntire u
power of z attain infinity and when z = ¥ , the terms with positive
z -p la n e ex c ep t
power of z attain infinity. Hence X(z) converges for all values of z, z = 0 an d z = ∞
except z = 0 and z = ¥ . Therefore, the ROC is entire z-plane, except F ig 3.4 : R O C o f fin ite
z = 0 and z = ¥ . d u ra tio n tw o -sid ed sig n a l.

Case - iv : Infinite duration, right-sided (causal) signal

Let, x(n) = r1n ; n ³ 0 Using infinite geometric
series sum formula
Now, the Z-transform of x(n) is,
∞
+∞ ∞ ∞ 1
X( z) = ∑ x(n) z −n
= ∑ r1n z −n
= ∑ er1 z j −1 n ∑ Cn =
1 − C
n=0
n = −∞ n= 0 n= 0
∞
if , 0 < |C | < 1
1
If, 0 < |r1 z −1| < 1, then ∑ (r1 z −1 ) n =
1 − r1 z −1 jv
n=0
z -p la n e
1 1 1 z
∴ X(z) = −1
= = = r1
1 − r1 z r z − r z − r1
1− 1 1
z z u
Here the condition to be satisfied for the convergence of X(z) is, R O C of
0 < |r1 z–1| < 1 x (n ) = r1n ; n ≥ 0
F ig 3.5 : R O C o f infinite
| r1| d u ra tio n rig h t-sid ed sig n a l.
\ |r1 z–1| < 1 Þ < 1 ⇒ |z| > |r1|
| z|
The term |r1| represents a circle of radius r1 in z-plane as shown in fig 3.5. From the above analysis we
can say that, X(z) converges for all points external to the circle of radius r1 in z-plane. Therefore, the ROC of
X(z) is exterior of the circle of radius r1 in z-plane as shown in fig 3.5.
Case v : Infinite duration, left-sided (anticausal) signal
Let, x(n) = r2n ; n £ 0
Now, the Z-transform of x(n) is,
+∞ 0 +∞ +∞
X( z) = ∑ x(n) z − n = ∑ r2n z− n = ∑ r2− n z n = ∑ (r2−1 z) n
n = −∞ n = −∞ n= 0 n= 0
∞
1 Using infinite geometric
If , 0 < |r2−1 z| < 1, then ∑
(r2−1 z) n =
1 − r2−1 z series sum formula
n=0
1 1 1 r2 r ∞
1
∴ X(z) = = = = = − 2
1 − r2−1 z 1−
z r2 − z r2 − z z − r2 ∑ Cn 1
=
− C
n=0
r2 r2
if , 0 < |C | < 1
Chapter 3 - Z - Transform 3. 6

Here the condition to be satisfied for the convergence of X(z) is, jv z -p lan e
0 < |r 2
–1
z| < 1 r2

| z|
\ |r2–1 z| < 1 Þ < 1 ⇒ |z| < |r2 |
| r2 | R O C of u
n
The term |r2| represents a circle of radius r2 in z-plane as shown in fig 3.6. x (n) = r 2 ; n ≤ 0
From the above analysis we can say that X(z) converges for all points internal to
the circle of radius r2 in z-plane. Therefore, the ROC of X(z) is interior of the F ig 3.6 : R O C o f infinite
circle of radius r2 as shown in fig 3.6. d u ratio n left-sid ed sign a l.

Case vi: Infinite duration, two-sided (noncausal) signal

Let, x(n) = r1n u(n) + r2n u(– n)
Now, the Z-transform of x(n) is,
+∞ 0 +∞ +∞ +∞
X( z) = ∑ x(n) z − n = ∑ r2n z− n + ∑ r1n z − n = ∑ r2− n z n + ∑ r1n z − n
n = −∞ n = −∞ n= 0 n= 0 n= 0
+∞ +∞ Infinite geometric series sum formula
= ∑ (r2− 1 z) n + ∑ (r1 z −1 ) n ∞
Cn =
∑
1
if , 0 < |C | < 1
n= 0 n= 0
n=0
1 − C
1 1
= + Using infinite geometric series sum formula
1 − r2−1 z 1 − r1 z −1
if, 0 < |r2−1 z| < 1, and, 0 < |r1 z−1| < 1
∞ ∞
The term ∑ (r2− 1 z) n converges if, The term ∑ (r1 z−1 ) n converges if,
n= 0 n= 0
0 < r2− 1 z < 1 0 < r1 z− 1 < 1
z r1
∴ r2− 1 z < 1 ⇒ < 1 ⇒ |z |<|r2 | ∴ r1 z−1 < 1 ⇒ < 1 ⇒ |z | > |r1|
r2 z
The term |r2| represents a circle of radius r2 and |r1| jv z -p la n e
represents a circle of radius r1 in z-plane. If |r2| > |r1| then there will R O C of n
r2
be a region between two circles as shown in fig 3.7. Now the X(z) x (n) = r1 u(n) + r2n u( −n)
r1 if |r 2 |> |r1 |
will converge for all points in the region between two circles
(because the first term of X(z) converges for |z| < |r2| and the u
second term of X(z) converges for |z| > |r1|). Hence the ROC is the
region between two circles of radius r1 and r2 as shown in fig 3.7. F ig 3.7 : R O C o f in fin ite
Table 3.1 : Summary of ROC of Discrete Time Signals d u ra tio n tw o -sid ed sign al.

Sequence ROC
Finite, right-sided (causal) Entire z-plane except z = 0
Finite, left-sided (anticausal) Entire z-plane except z = ¥
Finite, two-sided (noncausal) Entire z-plane except z = 0 and z = ¥
Infinite, right-sided (causal) Exterior of circle of radius r1, where |z| > r1
Infinite, left-sided (anticausal) Interior of circle of radius r2, where |z| < r2
Infinite, two-sided (noncausal) The area between two circles of radius r2 and r1
where, r2 > r1, and r1< |z| < r2, (i.e., |z| >r1, and, |z| < r2)
3. 7 Digital Signal Processing
Table 3.2 : Characteristic Families of Signals and Corresponding ROC

Signal ROC in z-plane

Finite Duration Signals
E n tire z -p la n e
x (n ) jv e xc e pt z = 0
R ig h t-side d z -p la n e
(o r c au sal)
u

0
n
x (n ) E n tire z -p la n e
L e ft-sid ed jv
(o r a n tic a usa l) e xc e pt z = ∞
z -p la n e

0
n
x (n ) E n tire z -p la n e
Tw o -side d
(o r n o nc a u sa l) jv e xc e pt z = 0
z -p la n e a nd z = ∞

0
n
Infinite Duration Signals
x (n ) jv
r1 z -p la n e
R ig h t-side d
(o r c au sal)
u

|z|> r 1
0
n
jv z-p la n e
x (n )
L e ft-sid ed
(o r a n tic a usa l)

u
r2

0
n
|z| < r 2
jv z -p la n e
x (n )
Tw o -side d r 1 < |z | < r 2
(o r n o nc a u sa l) r1
[|z | > r 1
a nd |z |< r 2 ]
r2
u

0
n
Chapter 3 - Z - Transform 3. 8

Example 3.1
Determine the Z-transform and their ROC of the following discrete time signals.
a) x(n) = {3, 4, 2, 7} b) x(n) = {6, 8, 9, 3} c) x(n) = {2, 4, 6, 8, 10}
- - -

Solution
a) Given that, x(n) = {3, 4, 2, 7}
-
i.e., x(0) = 3 ; x(1) = 4 ; x(2) = 2 ; x(3) = 7 ; and x(n) = 0 for n < 0 and for n > 3.
By the definition of Z-transform,
∞
Z {x(n)} = X(z) = ∑ x(n) z −n
n = −∞

The given sequence is a finite duration sequence defined in the range n = 0 to 3, hence the limits of
summation is changed to n = 0 to n = 3.
3 jv
∴ X ( z) = ∑
n = 0
x(n) z −n z -p la n e

= x(0) z0 + x(1) z −1 + x(2) z −2 + x(3) z −3 R O C is e ntire u

−1 −2 −3 z -p la n e ex c e pt
= 3 + 4z + 2z + 7z
4 2 7 z=0
= 3 + + 2 + 3
z z z
In X(z), when z = 0, except the first terms all other terms will become infinite. Hence X(z) will be finite for
all values of z, except z = 0. Therefore, the ROC is entire z-plane except z = 0.

b) Given that, x(n) = {6, 8, 9, 3}

-
i.e, x(–3) = 6 ; x(–2) = 8 ; x(–1) = 9 ; x(0) = 3 ; and x(n) = 0 for n < –3 and for n > 0.
By the definition of Z-transform,
∞
Z {x(n)} = X(z) = ∑ x(n) z −n
n = −∞

The given sequence is a finite duration sequence defined in the range n = –3 to 0, hence the limits of
summation is changed to n = –3 to 0.
jv
0
z -p la n e
∴ X(z) = ∑
x(n) z
n = −3
−n

R O C is e ntire u
= x(−3) z3 + x( −2) z 2 + x(−1) z + x(0)
z -p la n e ex c e pt
= 6z3 + 8z2 + 9z + 3 z=∞
In X(z), when z = ¥ , except the last term all other terms become infinite. Hence X(z) will be finite for all
values of z, except z = ¥ . Therefore, the ROC is entire z-plane except z = ¥.

c) Given that, x(n) = {2, 4, 6, 8, 10}

A
i.e, x(–2) = 2 ; x(–1) = 4 ; x(0) = 6 ; x(1) = 8 ; x(2) = 10 and x(n) = 0 for n < –2 and for n > 2.

By the definition of Z-transform,

∞
Z{x(n)} = X(z) = ∑ x(n) z −n
n = −∞
3. 9 Digital Signal Processing
The given sequence is a finite duration sequence defined in the range n = –2 to +2, hence the limits of
summation is changed to n = –2 to n = 2.
2 jv
∴ X(z) = ∑
n = −2
x(n) z −n z -p la n e

= x(−2) z2 + x(−1) z1 + x(0) z0 + x(1) z −1 + x(2) z −2 R O C is e ntire u

= 2z 2
+ 4z + 6 + 8z + 10z −1 −2 z -p la n e ex c e pt
8 10 z = 0 an d z = ∞
= 2z2 + 4z + 6 + + 2
z z
In X(z), when z = 0, the terms with negative power of z will become infinite and when z = ¥, the terms with
positive power of z will become infinite. Hence X(z) will be finite for all values of z except when z = 0 and
z = ¥ .Therefore, the ROC is entire z-plane except z = 0 and z = ¥.

Example 3.2
Determine the Z-transform and their ROC of the following discrete time signals.
a) x(n) = u(n) b) x(n) = 0.3n u(n) c) x(n) = 0.8n u(–n –1) d) x(n) = 0.3n u(n) + 0.8n u(–n–1)
Solution
a) Given that, x(n) = u(n)
The u(n) is a discrete unit step signal, which is defined as,
u(n) = 1 ; for n ³ 0 Infinite geometric series sum formula
= 0 ; for n < 0 ∞
1
By the definition of Z-transform,
∑
n = 0
Cn =
1− C
; if , 0 <|C|< 1

∞ ∞
Z{x(n)} = X(z) = ∑ x(n) z −n
= ∑ u(n) z −n

n = −∞ n = 0
∞ ∞
1 Using infinite geometric series sum formula.
= ∑ z −n = ∑ (z −1 n
) =
1 − z −1
n = 0 n = 0

1 z jv
= = 1 z -p la n e
1− 1 / z z − 1 |Z |=

Here the condition for convergence is, 0 < |z–1| < 1. u

1 ROC
∴ |z −1| < 1 ⇒ < 1 ⇒ |z| > 1
|z|
The term |z| = 1 represents a circle of unit radius in z-plane. Therefore, the ROC is exterior of unit circle
in z-plane.
b) Given that, x(n) = 0.3n u(n)
The u(n) is a discrete unit step signal, which is defined as,
jv
u(n) = 1 ; for n³0 |Z |= 0.3
z -p la n e
=0 ; for n<0
\ x(n) = 0.3n ; for n ³ 0
u
= 0 ; for n < 0 ROC
By the definition of Z-transform,
∞ ∞
Z{x(n)} = X(z) = ∑ x(n) z −n = ∑ 0.3 n
z −n
n = −∞ n = 0
∞
−1 n 1 Using infinite geometric series sum formula.
= ∑ d0.3z i =
1 − 0.3z −1
n = 0
Chapter 3 - Z - Transform 3. 10
1 z
∴ X(z) = =
1 z − 0.3
1 − 0.3
z
Here the condition for convergence is, 0 < |0.3 z–1| < 1.
0.3
∴ |0.3 z−1| < 1 ⇒ < 1 ⇒ |z|> 0.3
|z|
The term |z| = 0.3 represents a circle of radius 0.3 in z-plane. Therefore, the ROC is exterior of circle with
radius 0.3 in z-plane.

c) Given that, x(n) = 0.8n u(–n –1) jv

z -p la n e
The u(–n –1) is a discrete unit step signal, which is defined as,

.8
=0
u(–n – 1) = 0 ; for n ³ 0

|z |
=1 ; for n £ –1 u
ROC
\ x(n) = 0 ; for n ³ 0
n
= 0.8 ; for n £ –1
By the definition of Z-transform,
∞ −1
Z{x(n)} = X(z) = ∑ x(n) z −n = ∑ 0.8n z −n
n = −∞ n = −∞
∞ ∞ ∞
= ∑ 0.8 −n zn = ∑ (0.8 −1 z)n = ∑ (0.8 −1 z)n − 1 (0.8–1 z)0 = 1
n = 1 n = 1 n = 0

1 Using infinite geometric

= −1
−1 series sum formula.
1 − (0.8 z)
1 0.8 0.8 − 0.8 + z z z
= − 1= − 1= = =−
z 0.8 − z 0.8 − z 0.8 − z z − 0.8
1 −
0.8
Here the condition for convergence is, 0 < |0.8–1 z| < 1.
|z|
∴ |0.8 −1 z|< 1 ⇒ < 1 ⇒ |z|< 0.8
0.8
The term |z| = 0.8, represents a circle of radius 0.8 in z-plane. Therefore, the ROC is interior of the circle
of radius 0.8 in z-plane.
jv z -p la n e
d) Given that, x(n) = 0.3n u(n) + 0.8n u(–n – 1)
n n
X(z) = Z {x(n)} = Z{0.3 u(n) + 0.8 u(–n –1)}
|z| =0 .3

n n |z |
= Z{0.3 u(n)} + Z{0.8 u(−n − 1)} Using linearity property. =0 u
.8
z z
= − Using the results of (b) and (c). ROC
z − 0.3 z − 0.8
2 2
z(z − 0.8) − z(z − 0.3) z − 0.8z − z + 0.3z −0.5z
= = 2 = 2
(z − 0.3) (z − 0.8) z − 0.8z − 0.3z + 0.24 z − 1.1z + 0.24

Here the condition for convergence of 0.3n u(n) is,

0 < |0.3 z–1| < 1 Þ |z| > 0.3
and the condition for convergence of 0.8n u(–n –1) is,
0 < |0.8–1 z| < 1 Þ |z| < 0.8
The term |z| = 0.8, represents a circle of radius 0.8 in z-plane and the term |z| = 0.3 represents a circle of
radius 0.3 in z-plane. Hence the common region of convergence for both the terms of x(n) is the region in
between the circles of radius |z| = 0.8 and |z| = 0.3 in the z-plane.
3. 11 Digital Signal Processing
3.3 Properties of Z-Transform
1. Linearity property
The linearity property of Z-transform states that the Z-transform of linear weighted combination of
discrete time signals is equal to similar linear weighted combination of Z-transform of individual discrete time
signals.
Let, Z{x1(n)} = X1(z) and Z{x2(n)} = X2(z) then by linearity property,
Z{a1x1(n) + a2x2(n)} = a1X1(z) + a2X2(z) ; where, a1 and a2 are constants.
Proof :
By definition of Z-transform,
+∞
X1(z) = Z{x1(n)} = ∑ x (n) z
n = −∞
1
−n
.....(3.8)

+∞
X 2(z) = Z{x 2(n)} = ∑ x (n) z
n = −∞
2
−n
.....(3.9)

+∞ +∞
∴ Z{a1x1(n)+a2 x 2(n)} = ∑
n = −∞
a1x1(n)+a2 x 2 (n) z− n = ∑
n = −∞
a1x1(n) z− n +a2 x2 (n) z− n

+∞ +∞ +∞ +∞
= ∑
n = −∞
a1x1(n) z− n + ∑
n = −∞
a2 x2(n)z− n = a1 ∑
n = −∞
x1(n) z− n +a2 ∑ x (n)z
n = −∞
2
−n

= a1 X1(z) + a2 X 2(z) Using equations (3.8) and (3.9).

2. Shifting property
Case i: Two-sided Z-transform
The shifting property of Z-transform states that, Z-transform of a shifted signal shifted by m-units of
time is obtained by multiplying zm to Z-transform of unshifted signal.
Let, Z{x(n)} = X(z)
Now, by shifting property,
Z{x(n–m)} = z–m X(z)
Z{x(n+m)} = zm X(z)
Proof :
By definition of Z-transform,
+∞
X(z) = Z{x(n)} = ∑ x(n) z −n
.....(3.10)
n = −∞
+∞
∴ Z{x(n − m)} = ∑ x(n − m) z −n

n = −∞
+∞
Let, n – m = p, \ n = p + m
= ∑ x(p) z −(m + p)
when n ® -¥, p ® -¥
p = −∞
+∞ when n ® +¥, p ® +¥
= ∑ x(p) z −m
z− p
p = −∞
+∞ +∞
= z− m ∑ x(p) z
p = −∞
−p
= z− m ∑ x(n) z
n = −∞
−n
Let, p → n

= z− m X(z) Using equation (3.10).

Chapter 3 - Z - Transform 3. 12
By definition of Z-transform,
+∞
Z{x(n + m)} = ∑ x(n + m) z
n = −∞
−m
Let, n + m = p, \ n = p – m
+∞ when n ® -¥, p ® -¥
= ∑ x(p) z
p = −∞
−(p − m)
when n ® +¥, p ® +¥

+∞
= ∑ x(p) z
p = −∞
−p
zm
+∞ +∞
= zm ∑
p = −∞
x(p) z− p = zm ∑ x(n) z
n = −∞
−n
Let, p → n

= zm X(z) Using equation (3.10).

Case ii: One-sided Z-transform

Let x(n) be a discrete time signal defined in the range 0 < n < ¥ .
Let, Z{x(n)} = X(z)
Now by shifting property,
m
l q
Z x(n − m) = z − m X(z) + ∑ x( − i) z− ( m − i)
i=1
m−1
l q
Z x(n + m) = z m X(z) − ∑ x( i) z( m − i)
i=0

Proof :
By definition of one-sided Z-transform,
+∞
X( z) = Z {x(n)} = ∑ x(n) z −n
.....(3.11)
n= 0
+∞
∴ Z {x(n − m)} = ∑
n=0
x(n − m) z− n

+∞ Multiply by zm and z –m
= ∑
n=0
x(n − m) z− n zm z− m

+∞
= z− m ∑
n=0
x(n − m) z−( n − m)
Let, n – m = p,
+∞ when n ® 0, p ® –m
= z −m
∑ x(p) z −p
when n ® +¥, p ® +¥
p = −m

+∞ −1
= z− m ∑ x(p) z− p + z− m ∑ x(p) z− p
p= 0 p = −m
+∞ m
= z− m ∑ x(p) z− p + z− m ∑ x(− p) zp Let p = n, in first summation.
p= 0 p=1 Let p = i, in second summation.
+∞ m
= z− m ∑ x(n) z− n + z− m ∑ x(− i) zi Using equation (3.11).
n=0 i=1
m
= z− m X(z) + ∑ x( − i) z−(m − i) .....(3.12)
i=1

Note : In equation (3.12) if x(–i) for i = 1 to m are zero then the shifting property of one-sided Z-transform
for delayed signal will be same as that for two-sided Z-transform.
3. 13 Digital Signal Processing
By definition of one-sided Z-transform,
+∞
Z {x(n+m)} = ∑
n=0
x(n + m) z− n
Multiply by zm and z –m
+∞
= ∑
n=0
x(n + m) z− n zm z− m

+∞ Let, n + m = p,
= zm ∑ x(n + m) z −( n + m)
when n ® 0 , p ® m
n=0
+∞
when n ® + ¥ , p ® + ¥
= z m
∑ x(p) z− p
p=m
+∞ m −1
= zm ∑ x(p) z− p − zm ∑ x(p) z− p
p=0 p=0 Let p = n, in first summation.
+∞ m −1 Let p = i, in second summation.
= zm ∑ x(n) z− n − zm ∑ x(i) z− i
n=0 i=0 Using equation (3.11).
m −1
m
= z X(z) − ∑ x(i) z m−i
.....(3.13)
i=0

Note : In equation (3.13) if x(i) for i = 0 to m –1 are zero then the shifting property of one-sided Z-transform
for advanced signal will be same as that for two-sided Z-transform.
3. Multiplication by n (or Differentiation in z-domain)
If Z{x(n)} = X(z)
d
m
then Z nx(n) = − zr dz
X(z)

In general,

t FGH IJ X(z) m
d
o
Z n m x(n) = − z
Kdz
d F d F F d F IJ I ..... IJ I
= −z −z ..... G − z
dz GH dz GH H
G −z dzd
dz H
X(z)
K JK K JK
1444444424444444
3
m − times

Proof :
By definition of Z-transform,
+∞
X( z) = Z{x(n)} = ∑ x(n) z −n
.....(3.14)
n = −∞
+∞
∴ Z {n x(n)} = ∑
n = −∞
n x(n) z− n

+∞
= ∑
n = −∞
n x(n) z− n z z−1 Multiply by z and z –1
+∞
= −z ∑
n = −∞
x(n) − n z− n − 1

= −z ∑
+∞
x(n)
LM d z OP
−n d −n
z = − n z− n −1
n = −∞ N dz Q dz
+∞
d Interchanging summation
= −z
dz ∑ x(n) z
n = −∞
−n
and differentiation.
d
= −z X(z) Using equation (3.14).
dz
Chapter 3 - Z - Transform 3. 14
4. Multiplication by an exponential sequence, an (or Scaling in z-domain)
If Z{x(n)} = X(z)
o t
then Z a n x(n) = X(a −1z)

Proof :

By definition of Z-transform,
+∞
Z{x(n)} = ∑ x(n) z −n
.....(3.15)
n = −∞
+∞
∴ Z {a n x(n)} = ∑
n = −∞
a n x(n) z− n

+∞
= ∑ x(n) (a −1z)− n .....(3.16)
n = −∞
The equation (3.16) is similar
= X(a −1z) to the form of equation (3.15).

5. Time reversal
If Z{x(n)} = X(z)
then Z{x(–n)} = X(z–1)
Proof :
By definition of Z-transform,
+∞
Z{x(n)} = ∑ x(n) z −n
.....(3.17)
n = −∞
+∞
∴ Z {x( − n)} = ∑
n = −∞
x( − n) z− n
Let, p = –n
+∞ when n ® -¥, p ® +¥
= ∑
p = −∞
x(p) zp when n ® +¥, p ® -¥

+∞
= ∑ x(p) (z−1 )− p .....(3.18)
p = −∞ The equation (3.18) is similar
= X(z−1) to the form of equation (3.17).

6. Conjugation
If Z{x(n)} = X(z)
then Z{x*(n)} = X*(z*)
Proof :
By definition of Z-transform,
+∞
X(z) = Z{x(n)} = ∑ x(n) z −n
.....(3.19)
n = −∞
+∞
∴ Z {x ∗(n)} = ∑
n = −∞
x∗(n) z− n

LM+∞ OP ∗

=
MN ∑
n = −∞
x(n) (z∗ )− n
PQ The equation (3.20) is similar
.....(3.20)

∗
= X(z∗ ) to the form of equation (3.19).

= X ∗( z∗ )
3. 15 Digital Signal Processing
7. Convolution theorem
If Z{x1(n)} = X1(z)
and Z{x2(n)} = X2(z)
then Z{x1(n) * x2(n)} = X1(z) X2(z)
+∞
where, x1 (n) ∗ x2 (n) = ∑
m = −∞
x1 (m) x2 (n − m) .....(3.21)

Proof :

By definition of Z-transform,
+∞
X1(z) = Z{x1(n)} = ∑ x (n) z
1
−n
.....(3.22)
n = −∞
+∞
X 2(z) = Z{x 2 (n)} = ∑ x (n) z 2
−n
.....(3.23)
n = −∞
+∞
∴ Z {x1(n) ∗ x 2 (n)} = ∑
n = −∞
x1(n) ∗ x 2 (n) z− n

+∞ LM +∞ OP Using equation (3.21).

= ∑
n = −∞ MN ∑
m = −∞
x1(m) x 2 (n − m) z− n
PQ
+∞ +∞
Multiply by zm and z –m
= ∑ ∑ x1(m) x 2(n − m) z− n z− m zm
n = −∞ m = −∞
+∞ +∞
= ∑ x1(m) z− m ∑ x 2(n − m) z−( n − m)
Let, n – m = p
m = −∞ n = −∞
+∞ +∞ when n ® -¥, p ® -¥
= ∑ x1(m) z− m ∑ x 2(p) z− p when n ® +¥, p ® +¥
m = −∞ p = −∞

LM +∞ OP LM+∞ OP Let m = n, in first summation.

=
MN ∑
n = −∞
x1(n) z− n
PQ MN ∑
n = −∞
x 2(n) z− n
PQ Let p = n, in second summation.

= X1( z) X 2( z) Using equations (3.22) and (3.23).

8. Correlation property
If Z{x(n)} = X(z) and Z{y(n)} = Y(z)
then Z{rxy(m)} = X(z) Y(z –1)
+∞
where, rxy (m) = ∑ x(n) y(n − m)
n= −∞
.....(3.24)

Proof :
By definition of Z-transform,
+∞
X(z) = Z {x(n)} = ∑ x(n) z −n
.....(3.25)
n = −∞
+∞
Y(z) = Z{y(n)} = ∑ y(n) z −n
.....(3.26)
n = −∞
Chapter 3 - Z - Transform 3. 16

+∞
∴ Z { rxy (m)} = ∑
m = −∞
rxy (m) z− m

+∞ LM +∞ OP
= ∑
m = −∞ MN ∑
n = −∞
x(n) y(n − m) z− m
PQ Using equation (3.24).

+∞ +∞
= ∑ ∑
m = −∞ n = −∞
x(n) y(n − m) z− m z− n zn Multiply by zn and z –n

+∞ +∞
= ∑ x( n) z− n ∑ y( n − m) z( n − m)
n = −∞ m = −∞ Let, n – m = p \ m = n – p
+∞ +∞
when m ® -¥, p ® +¥,
= ∑
n = −∞
x( n) z −n
∑
p = −∞
y(p) z p
when m ® +¥, p ® -¥.

LM +∞ OP LM +∞ OP
=
MN ∑
n = −∞
x( n) z− n
PQ MN ∑
p = −∞
y(p) (z−1)− p
PQ Using equations (3.25)
−1
= X(z) Y(z ) and (3.26).

9. Initial value theorem

Let x(n) be an one-sided signal defined in the range 0 £ n £ ¥.

Now, if Z{x(n)} = X(z),

then the initial value of x(n) [i.e., x(0)] is given by,

x(0) = Lt X(z)
z→∞

Proof :

By definition of one-sided Z - transfrom,

+∞
X( z) = ∑ x(n) z −n

n=0

On expanding the above summation we get,

X( z) = x(0) + x(1) z−1 + x(2) z−2 + x(3) z−3 + ......

x(1) x(2) x(3)
∴ X( z) = x(0) + + 2
+ + ......
z z z3

On taking limit z ® ¥ in the above equation we get,

Lt X( z) = Lt
LMx(0) +
x(1)
+
x(2)
+
x(3)
+ ......
OP
z→∞ z→∞ N z z2 z3 Q
= x(0) + 0 + 0 + 0 + ......
∴ x(0) = Lt X( z)
z→∞
3. 17 Digital Signal Processing
10. Final value theorem
Let x(n) be a one-sided signal defined in the range 0 £ n £ ¥.
Now, if Z {x(n)} = X(z),
then the final value of x(n) [i.e., x(¥ )] is given by,

x(∞) = Lt (1 − z −1 ) X(z) or x( ∞) = Lt
FG z − 1IJ X(z)
z→1 z→1 H zK
Proof :
By definition of one-sided Z-transfrom,
+∞
m r ∑ x(n) z
Z x(n) =
n=0
−n
.....(3.27)

+∞
m
∴ Z x(n − 1) − x(n) = r ∑ n=0
x(n − 1) − x(n) z− n
(LHS) (RHS)

m
LHS = Z x(n − 1) − x(n) r
m r m r
= Z x(n − 1) − Z x(n) Using linearity property.
−1
= z X( z) + x( −1) − X(z) Using shifting property and equation (3.27).
−1
= x( −1) − (1 − z ) X(z)
= Lt x(−1) − (1− z−1) X(z) Taking limit z → 1
z→1

= x( −1) − Lt (1 − z−1) X(z) .....(3.28)

z→1

+∞
RHS = ∑
n=0
x(n − 1) − x(n) z− n

+∞
= Lt
z→1 ∑
n=0
x(n − 1) − x(n) z− n Taking limit z ® 1

+∞
On applying limit z ® 1, the term z–n
= ∑
n=0
x(n − 1) − x(n) becomes unity.
p Changing the summation index from
= Lt
p→∞ ∑
n=0
x(n − 1) − x(n) 0 to p and then taking limit p ® ¥.

LM x(−1) − x(0) + x(0) − x(1) + x(1) − x(2) + ..... OP

= Lt
p→∞ MN ..... + x(p − 2) − x(p −1) + x(p −1) − x(p) PQ
= Lt x(−1) − x(p)
p→∞

= x( −1) − x(∞) .....(3.29)

On equating equation (3.29) with (3.28) we get,

x( − 1) − x(∞) = x( −1) − Lt (1 − z−1) X(z)
z→1

∴ x(∞) = Lt (1 − z−1) X(z)

z→1
Chapter 3 - Z - Transform 3. 18
11. Complex convolution theorem (or Multiplication in time domain)
Let, Z {x1(n)} = X1(z) and Z {x2(n)} = X2(z).
Now, the complex convolution theorem states that,

m
Z x1 (n) x2 (n) = r 1
2 πj z
C
X (v) X
1 2
FH vz IK v −1
dv

where, v is a dummy variable used for contour integration

Proof :

Let, Z {x1(n)} = X1(z) and Z {x2(n)} = X2(z).

Now, by definition of inverse Z-transform,

x1(n) =
1
2πj z
C
X1( z) zn − 1 dz =
1
2πj z
C
X1( v) v n − 1 dv let, z = v .....(3.30)

Now, by definition of Z-transform,

+∞
X 2( z) = ∑ x ( n) z
2
−n
.....(3.31)
n = −∞

Using the definition of Z-transform, the Z {x1(n) x2(n)} can be written as,
+∞
m
Z x1( n) x 2( n) = r ∑ x (n) x (n) z
1 2
−n

n = −∞
L1 OP
=
+∞

∑ MM 2πj
n = −∞ N z
C
X1( v ) v n − 1 dv x 2( n) z− n
PQ Using equation (3.30).

=
1
2πj z
C
X1( v )
+∞

∑
n = −∞
x 2( n) z− n v n v −1 dv Interchanging the order of
summation and integration.
L OP
` =
1
2πj z
C
1
MN
+∞

n = −∞
F zI
X ( v ) M ∑ x ( n) G J
H vK 2
−n

PQ v
−1
dv

=
1
2πj z
C
X (v ) X FH z IK v dv
1
v 2
−1
Using equation (3.31).

12. Parseval’s relation

If Z {x1(n)} = X1(z) and Z {x2(n)} = X2(z).

Then the Parseval’s relation states that,

∑
+∞

n = −∞
x1 (n) x∗2 (n) =
1
2πj z
C
FH IK
X1(z) X∗2 1∗ z−1 dz
z
3. 19 Digital Signal Processing
Proof :

Let, Z {x1(n)} = X1(z) and Z {x2(n)} = X2(z).

Now, by definition of inverse Z-transform,

x1(n) =
1
2πj z
C
X1( z) zn − 1 dz =
1
2πj z
C
X1( v) v n − 1 dv let, z = v .....(3.32)

Now, by definition of Z-transform,

+∞
m
Z x 2( n) = r ∑ x2(n) z− n .....(3.33)
n = −∞

Using the definition of Z - transform, the Z x1(n) x∗2( n) can be written as, { }
+∞
{
Z x1( n) x∗2( n) = } ∑ x1(n) x∗2(n) z− n
n = −∞
.....(3.34)

On substituting for x1(n) from equation (3.32) in equation (3.34) we can write,

LM 1 OP
+∞

∑ x (n) x (n) z
n = −∞
1
∗
2
−n
=
+∞

∑
n = −∞ MN 2πj z
C
X1( v ) v n − 1 dv x∗2( n) z− n
PQ
LM x (n) z v OP v dv
=
1
2πj z
C
X1( v)
MN ∑ n = −∞
+∞

PQ
∗
2
−n n −1
Interchanging the
order of summation
and integration.

L F zI O
=
1
2πj z
C
1
MN
+∞
X ( v ) M ∑ x ( n) G J P v dv
H v K PQ
n = −∞
∗
2
−n
−1

L Fz I O −n
∗

=
1
2πj z
C
1
MN
+∞
X ( v ) M ∑ x ( n) G J P v dv
H v K PQ
n = −∞
2
∗