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EE6403 DISCRETE TIME SYSTEMS AND SIGNAL PROCESSING L T P C3 0 0 3

UNIT I INTRODUCTION 9
Classification of systems: Continuous, discrete, linear, causal, stable, dynamic, recursive, time
variance; classification of signals: continuous and discrete, energy and power; mathematical
representation of signals; spectral density; sampling techniques, quantization, quantization error,
Nyquist rate, aliasing effect.

UNIT II DISCRETE TIME SYSTEM ANALYSIS 9

Z-transform and its properties, inverse z-transforms; difference equation – Solution by z-transform,
application to discrete systems - Stability analysis, frequency response – Convolution – Discrete Time
Fourier transform , magnitude and phase representation.

UNIT III DISCRETE FOURIER TRANSFORM & COMPUTATION 9

Discrete Fourier Transform- properties, magnitude and phase representation - Computation of DFT
using FFT algorithm – DIT &DIF using radix 2 FFT – Butterfly structure.

UNIT IV DESIGN OF DIGITAL FILTERS 9

FIR & IIR filter realization – Parallel & cascade forms. FIR design: Windowing Techniques – Need and
choice of windows – Linear phase characteristics. Analog filter design – Butterworth and Chebyshev
approximations; IIR Filters, digital design using impulse invariant and bilinear transformation -
Warping, pre warping.

UNIT V DIGITAL SIGNAL PROCESSORS 9

Introduction – Architecture – Features – Addressing Formats – Functional modes - Introduction to
Commercial DSProcessors.

TOTAL : 45 PERIODS
OUTCOMES:
Ability to understand and apply basic science, circuit theory, Electro-magnetic field
theory control theory and apply them to electrical engineering problems.
TEXT BOOKS:
1. J.G. Proakis and D.G. Manolakis, ‗Digital Signal Processing Principles, Algorithms
and Applications‘, Pearson Education, New Delhi, PHI. 2003.
2. S.K. Mitra, ‗Digital Signal Processing – A Computer Based Approach‘, McGraw
Hill Edu, 2013.
3. Robert Schilling & Sandra L.Harris, Introduction to Digital Signal Processing using Matlab‖,
Cengage Learning,2014.
REFERENCES:
1. Poorna Chandra S, Sasikala. B ,Digital Signal Processing, Vijay Nicole/TMH,2013.
2. B.P.Lathi, ‗Principles of Signal Processing and Linear Systems‘, Oxford University Press, 2010
3. Taan S. ElAli, ‗Discrete Systems and Digital Signal Processing with Mat Lab‘, CRC Press, 2009.
4. Sen M.kuo, woonseng…s.gan, ―Digital Signal Processors, Architecture, Implementations &
Applications, Pearson,2013
5. Dimitris G.Manolakis, Vinay K. Ingle, applied Digital Signal Processing,Cambridge,2012
6. Lonnie C.Ludeman ,‖Fundamentals of Digital Signal Processing‖,Wiley,2013
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DISCRETE SYSTEMS & SIGNAL PROCESSING

1. BASICS OF DIGITAL SIGNALS & SYSTEMS 7

2. ANALYSIS OF LSI SYSTEMS- Z TRANSFORM 27

3. ANALYSIS OF FT, DFT AND FFT 32

4. DIGITAL FILTER CONCEPTS & DESIGN 69

5. DSP ARCHITECTURES AND APPLICATIONS OF DSP 101

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CONTENT PG.NO
CHAPTER -1

SIGNALS AND SYSTEMS

Introduction 7
Classification of signals processing 8
 Advantages of DSP over ASP
 Disadvantages of DSP over ASP
Classification of signals 11
Discrete time systems 11

Standard signal sequences 12

Properties Discrete time signals 12
Symbols used in Discrete time systems 12
Classification of Discrete time systems 13
Stability for LTI systems 13
Analysis of discrete LTI systems 16
 Types of convolution
 Properties of convolution
 Causality of LSI systems
 Stability for LTI systems
Correlation 19
 Types of correlation
 Properties of correlation
22
A/D Conversion
Differential Equations 23

CHAPTER – 2

DISCRETE TIME SYSTEMS ANALYSIS

2.0 Introduction to Z Transform 26

 Advantages of Z Transform
 Advantages of ROC
2.1 Z Transform plot 26
 Properties of Z -Transform
2.2 Relationship between FT & Z Transform 29
2.3 Inverse Z – Transform 31
2.4 Pole Zero Plot 34
2.5 One sided Z – Transform 36
2.6 Solution of Differential Equations 37
2.
38
CHAPTER – 3
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DISCRETE FOURIER TRANSFORM & COMPUTATION

3.0 Introduction 39
3.1 Difference between FT & DFT 39
3.2 Calculation of DFT
3.3 Difference between DFT & IDFT 41
3.4 Properties of DFT
3.5 Application of DFT 47
3.6 FFT 50
 Radix 2 FFT Algorithm
3.7 GEOERZEL Algorithm 59

CHAPTER – 4

DESIGN OF DIGITAL FILTERS

Introduction 61
Types of digital filters 65
Structure for FIR system 65
Structure for IIR system 68
IIR Filter design (Impulse invariance) 73
IIR Filter design (BZT) 76
Butterworth approximation 80
Frequency Transformation 87
FIR Filter design 89
Design filter for pole zero placement 93

CHAPTER – 5

DIGITAL SIGNAL PROCESSOR

Requirements of DSP Processor 97

Microprocessor Architecture 100
Core Architecture of ADSP - 21XX 101
 ADSP- 21XX Development tools
Application of DSP 106
Figure for finite duration 110
Figure for infinite duration 111
Figure for pole zero plot 112

 References
 Glossary Terms
 Tutorial Problems
 Worked Out Problems
 Question Bank
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CHAPTER 1

SIGNALS AND SYSTEM

PREREQISTING DISCUSSION
A Signal is defined as any physical quantity that changes with time, distance, speed, position, pressure,
temperature or some other quantity. A Signal is physical quantity that consists of many sinusoidal of
different amplitudes and frequencies.
Ex x(t) = 10t
X(t) = 5x2+20xy+30y
A System is a physical device that performs an operations or processing on a signal. Ex Filter or
Amplifier.

CLASSIFICATION OF SIGNAL PROCESSING

1) ASP (Analog signal Processing) : If the input signal given to the system is analog then system does
analog signal processing. Ex Resistor, capacitor or Inductor, OP-AMP etc.

Analog ANALOG Analog

Input SYSTEM Output

2) DSP (Digital signal Processing) : If the input signal given to the system is digital then system does
digital signal processing. Ex Digital Computer, Digital Logic Circuits etc. The devices called as ADC
(analog to digital Converter) converts Analog signal into digital and DAC (Digital to Analog Converter)
does vice-versa.

Analog ADC DIGITAL DAC Analog

signal SYSTEM signal

Most of the signals generated are analog in nature. Hence these signals are converted to digital form by
the analog to digital converter. Thus AD Converter generates an array of samples and gives it to the
digital signal processor. This array of samples or sequence of samples is the digital equivalent of input
analog signal. The DSP performs signal processing operations like filtering, multiplication,
transformation or amplification etc operations over this digital signals. The digital output signal from the
DSP is given to the DAC.

ADVANTAGES OF DSP OVER ASP

1. Physical size of analog systems are quite large while digital processors are more compact and light
in weight.
2. Analog systems are less accurate because of component tolerance ex R, L, C and active components.
Digital components are less sensitive to the environmental changes, noise and disturbances.
3. Digital system are most flexible as software programs & control programs can be easily modified.
4. Digital signal can be stores on digital hard disk, floppy disk or magnetic tapes. Hence becomes
transportable. Thus easy and lasting storage capacity.
5. Digital processing can be done offline.
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6. Mathematical signal processing algorithm can be routinely implemented on digital signal processing
systems. Digital controllers are capable of performing complex computation with constant accuracy
at high speed.
7. Digital signal processing systems are upgradeable since that are software controlled.
8. Possibility of sharing DSP processor between several tasks.
9. The cost of microprocessors, controllers and DSP processors are continuously going down. For some
complex control functions, it is not practically feasible to construct analog controllers.
10. Single chip microprocessors, controllers and DSP processors are more versatile and powerful.

Disadvantages Of DSP over ASP

1. Additional complexity (A/D & D/A Converters)
2. Limit in frequency. High speed AD converters are difficult to achieve in practice. In high frequency
applications DSP are not preferred.

CLASSIFICATION OF SIGNALS
1. Single channel and Multi-channel signals
2. Single dimensional and Multi-dimensional signals
3. Continuous time and Discrete time signals.
4. Continuous valued and discrete valued signals.
5. Analog and digital signals.
6. Deterministic and Random signals
7. Periodic signal and Non-periodic signal
8. Symmetrical(even) and Anti-Symmetrical(odd) signal
9. Energy and Power signal

1) Single channel and Multi-channel signals

If signal is generated from single sensor or source it is called as single channel signal. If the signals are
generated from multiple sensors or multiple sources or multiple signals are generated from same source
called as Multi-channel signal. Example ECG signals. Multi-channel signal will be the vector sum of
signals generated from multiple sources.

2) Single Dimensional (1-D) and Multi-Dimensional signals (M-D)

If signal is a function of one independent variable it is called as single dimensional signal like speech
signal and if signal is function of M independent variables called as Multi-dimensional signals. Gray
scale level of image or Intensity at particular pixel on black and white TV are examples of M-D signals.

3) Continuous time and Discrete time signals.

S. No Continuous Time (CTS) Discrete time (DTS)
1 This signal can be defined at any time This signal can be defined only at certain
instance & they can take all values in specific values of time. These time instance need
the continuous interval(a, b) where a not be equidistant but in practice they are
can be -∞ & can be ∞ usually takes at equally spaced intervals.
2 These are described by differential These are described by difference equation.
equations.
3 This signal is denoted by x(t). These signals are denoted by x(n) or notation
x(nT) can also be used.
4 The spepedacdonetreopl ozf na dectmotor Microprocessors and computer based systems
using a techogenerator feedback or Sine or
uses discrete time signals.
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exponential waveforms.

4) Continuous valued and Discrete Valued signals.

S. No Continuous Valued Discrete Valued
1 If a signal takes on all possible values on If signal takes values from a finite set of
a finite or infinite range, it is said to be possible values, it is said to be discrete valued
continuous valued signal. signal.
2 Continuous Valued and continuous time Discrete time signal with set of discrete
signals are basically analog signals. amplitude are called digital signal.

5) Analog and digital signal

S. No Analog signal Digital signal
1 These are basically continuous time & These are basically discrete time signals &
continuous amplitude signals. discrete amplitude signals. These signals are
basically obtained by sampling &
quantization process.
2 ECG signals, Speech signal, Television All signal representation in computers and
signal etc. All the signals generated from digital signal processors are digital.
various sources in nature are analog.

Note: Digital signals (DISCRETE TIME & DISCRETE AMPLITUDE) are obtained by sampling the
ANALOG signal at discrete instants of time, obtaining DISCRETE TIME signals and then by
quantizing its values to a set of discrete values & thus generating DISCRETE AMPLITUDE signals.
Sampling process takes place on x axis at regular intervals & quantization process takes place along y
axis. Quantization process is also called as rounding or truncating or approximation process.

6) Deterministic and Random signals

S. No Deterministic signals Random signals

1 Deterministic signals can be represented or Random signals that cannot be

described by a mathematical equation or lookup represented or described by a
table. mathematical equation or lookup table.
2 Deterministic signals are preferable because for Not Preferable. The random signals can
analysis and processing of signals we can use be described with the help of their
mathematical model of the signal. statistical properties.
3 The value of the deterministic signal can be The value of the random signal cannot
evaluated at time (past, present or future) without be evaluated at any instant of time.
certainty.
4 Example Sine or exponential waveforms. Example Noise signal or Speech signal

7) Periodic signal and Non-Periodic signal

The signal x(n) is said to be periodic if x(n+N)= x(n) for all n where N is the fundamental
period of the signal. If the signal does not satisfy above property called as Non-Periodic signals.
Discrete time signal is periodic if its frequency can be expressed as a ratio of two integers. f= k/N where
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Tutorial problems:
a) cos (0.01 ∏ n) Periodic N=200 samples per cycle.
b) cos (3 ∏ n) Periodic N=2 samples
c) sin(3n) Non-Periodic
d) cos(n/8) cos( ∏n/8) Non-Periodic

8) Symmetrical(Even) and Anti-Symmetrical(odd) signal

A signal is called as symmetrical(even) if x(n) = x(-n) and if x(-n) = -x(n) then signal is odd. X1(n)=
cos(ωn) and x2(n)= sin(ωn) are good examples of even & odd signals respectively. Every discrete
signal can be represented in terms of even & odd signals.
X(n) signal can be written as

𝑋 𝑛 + (𝑛) 2
𝑋 𝑛 =

Rearranging the above terms we have 𝑋 𝑛 − (𝑛) 2

𝑋 𝑛 =
Thus X(n)= Xe(n) + Xo(n)
Even component of discrete time signal is given by

𝑋 𝑛 + (𝑛) 2
𝑋𝑒 𝑛 =

Odd component of discrete time signal is given by

𝑋 𝑛 − (𝑛) 2

𝑋𝑜 𝑛 =
Test whether the following CT waveforms is periodic or not. If periodic find out the fundamental period.
a) 2 sin(2/3)t + 4 cos (1/2)t + 5 cos((1/3)t Ans: Period of x(t)= 12
b) a cos(t √2) + b sin(t/4) Ans: Non-Periodic

a) Find out the even and odd parts of the discrete signal x(n)={2,4,3,2,1}
b) Find out the even and odd parts of the discrete signal x(n)={2,2,2,2}

9) Energy signal and Power signal

Discrete time signals are also classified as finite energy or finite average power signals.
The energy of a discrete time signal x(n) is given by

∞
E= 𝑛=−∞ 𝑥2(𝑛)

The average power for a discrete time signal x(n) is defined as

1
𝑃 = lim 𝑁→∞ 2𝑁+1 ∞ 𝑛=−∞ 𝑥 (𝑛)
2

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If Energy is finite and power is zero for x(n) then x(n) is an energy signal. If power is finite and energy
is infinite then x(n) is power signal. There are some signals which are neither energy nor a power signal.

Tutorial problems:
a) Find the power and energy of u(n) unit step function.
b) Find the power and energy of r(n) unit ramp function.
c) Find the power and energy of an u(n).

DISCRETE TIME SIGNALS AND SYSTEM

There are three ways to represent discrete time signals.
1) Functional Representation
4 for n=1,3
x(n)= -2 for n =2
0 elsewhere
2) Tabular method of representation
n -3 -2 -1 0 1 2 3 4 5
x(n) 0 0 0 0 4 -2 4 0 0
3) Sequence Representation
X(n) = { 0 , 4 , -2 , 4 , 0 ,……}

n=0

STANDARD SIGNAL SEQUENCES

1) Unit sample signal (Unit impulse signal)

δ (n) = 1 n=0
0 n=0 i.e δ(n)={1}

2) Unit step signal

u(n) = 1 n≥0
0 n<0

3) Unit ramp signal

ur (n) = n n≥0
0 n<0
4) Exponential signal
x(n) = a n = (re j Ø ) n = r n e j Ø n = r n (cos Øn + j sin Øn)

4) Sinusoidal waveform
x(n) = A Sin wn
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1) Shifting : signal x(n) can be shifted in time. We can delay the sequence or advance the sequence.
This is done by replacing integer n by n-k where k is integer. If k is positive signal is delayed in time by
k samples (Arrow get shifted on left hand side) And if k is negative signal is advanced in time k samples
(Arrow get shifted on right hand side)
X(n) = { 1, -1 , 0 , 4 , -2 , 4 , 0 ,……}

n=0
Delayed by 2 samples : X(n-2)= { 1, -1 , 0 , 4 , -2 , 4 , 0 ,……}

n=0
Advanced by 2 samples : X(n+2) = { 1, -1 , 0 , 4 , -2 , 4 , 0 ,……}

n=0
2) Folding / Reflection : It is folding of signal about time origin n=0. In this case replace n by –n.
Original signal:
X(n) = { 1, -1 , 0 , 4 , -2 , 4 , 0}

n=0
Folded signal:
X(-n) = { 0 , 4 , -2 , 4 , 0 , -1 , 1}

n=0
3) Addition : Given signals are x1(n) and x2(n), which produces output y(n) where y(n) = x1(n)+ x2(n).
Adder generates the output sequence which is the sum of input sequences.

4) Scaling: Amplitude scaling can be done by multiplying signal with some constant. Suppose original
signal is x(n). Then output signal is A x(n)

5) Multiplication : The product of two signals is defined as y(n) = x1(n) * x2(n).

SYMBOLS USED IN DISCRETE TIME SYSTEM

1. Unit delay
Z-1
x(n) y(n) = x(n-1)

2. Unit advance
Z+1
x(n) y(n) = x(n+1)

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3. Addition
x1(n)
+
y(n) =x1(n)+x2(n)
x2(n)

4. Multiplication
x1(n)
×
y(n) =x1(n)*x2(n)
x2(n)
5. Scaling (constant multiplier)
A
x(n) y(n) = A x(n)

CLASSIFICATION OF DISCRETE TIME SYSTEMS

1) STATIC v/s DYNAMIC

S. No STATIC DYNAMIC
(Dynamicity property)
1 Static systems are those systems whose output at any Dynamic systems output
instance of time depends at most on input sample at same depends upon past or future
time. samples of input.
2 Static systems are memory less systems. They have memories for
memorize all samples.

It is very easy to find out that given system is static or dynamic. Just check that output of the system
solely depends upon present input only, not dependent upon past or future.
S. No System [y(n)] Static / Dynamic
1 x(n) Static
2 A(n-2) Dynamic
3 X2(n) Static
2
4 X(n ) Dynamic
5 n x(n) + x2(n) Static
6 X(n)+ x(n-2) +x(n+2) Dynamic

2) TIME INVARIANT v/s TIME VARIANT SYSTEMS

S. No TIME INVARIANT (TIV) / TIME VARIANT SYSTEMS /
SHIFT INVARIANT SHIFT VARIANT SYSTEMS
(Shift Invariance property)
1 A System is time invariant if its input output A System is time variant if its input output
characteristic do not change with shift of time. characteristic changes with time.
2 Linear TIV systems can be uniquely No Mathematical analysis can be
characterized by Impulse response, frequency performed.
response or transfer function.
3 a. Thermal Noise in Electronic components a. Rainfall per month
b. wPrwint.ipngaddoecuempeznt.snbeyta printer b. Noise Effect
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It is very easy to find out that given system is Shift Invariant or Shift Variant.
Suppose if the system produces output y(n) by taking input x(n)
x(n)  y(n)
If we delay same input by k units x(n-k) and apply it to same systems, the system produces output y(n-k)
x(n-k)  y(n-k)

3) LINEAR v/s NON-LINEAR SYSTEMS

S.No LINEAR NON-LINEAR
(Linearity Property)
1 A System is linear if it satisfies superposition theorem. A System is Non-linear if it
does not satisfies
superposition theorem.
2 Let x1(n) and x2(n) are two input sequences, then the system
is said to be linear if and only if T[a1x1(n) +
a2x2(n)]=a1T[x1(n)]+a2T[x2(n)]

a1
x1(n)
SYSTEM
y(n)= T[a1x1[n] + a2x2(n) ]
x2(n)
a2

x1(n) SYSTEMa1

y(n)=T[a1x1(n)+a2x2(n)]

x2(n) SYSTEM a2

hence T [ a1 x1(n) + a2 x2(n) ] = T [ a1 x1(n) ] + T [ a2 x2(n) ]

It is very easy to find out that given system is Linear or Non-Linear.
Response to the system to the sum of signal = sum of individual responses of the system.
S. No System y(n) Linear or Non-Linear

1 ex(n) Non-Linear

2 x2 (n) Non-Linear

3 m x(n) + c Non-Linear

4 cos [ x(n) ] Non-Linear

5 X(-n) Linear

6 Log 10 (|x(n)|) Non-Linear

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4) CAUSAL v/s NON CAUSAL SYSTEMS

S.No CAUSAL NON-CAUSAL
(Causality Property)
1 A System is causal if output of system at any A System is Non causal if output of
time depends only past and present inputs. system at any time depends on future
inputs.
2 In Causal systems the output is the function of In Non-Causal System the output is the
x(n), x(n-1), x(n-2)….. and so on. function of future inputs also. X(n+1)
x(n+2) .. and so on
3 Example Real time DSP Systems Offline Systems

It is very easy to find out that given system is causal or non-causal. Just check that output of the system
depends upon present or past inputs only, not dependent upon future.
S.No System [y(n)] Causal /Non-Causal
1 x(n) + x(n-3) Causal
2 X(n) Causal
3 X(n) + x(n+3) Non-Causal
4 2 x(n) Causal
5 X(2n) Non-Causal
6 X(n)+ x(n-2) +x(n+2) Non-Causal

5) STABLE v/s UNSTABLE SYSTEMS

S.No STABLE UNSTABLE
(Stability Property)
1 A System is BIBO stable if every bounded A System is unstable if any bounded input
input produces a bounded output. produces a unbounded output.
2 The input x(n) is said to bounded if there exists
some finite number Mx such that |x(n)| ≤ Mx <
∞
The output y(n) is said to bounded if there
exists some finite number My such that |y(n)| ≤
My < ∞

STABILITY FOR LTI SYSTEM

It is very easy to find out that given system is stable or unstable. Just check that by providing input
signal check that output should not rise to ∞.

The condition for stability is given by

∞
𝑘=−∞ 𝑕(𝑘) < ∞

S.No System [y(n)] Stable / Unstable

1 Cos [ x(n) ] Stable
2 x(-n+2) Stable
3 | Stable
4
zx(n)
.net u(n) Stable
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5 X(n) + n x(n+1) Unstable

ANALYSIS OF DISCRETE LINEAR TIME INVARIANT (LTI/LSI) SYSTEM

1) CONVOLUTION SUM METHOD

2) DIFFERENCE EQUATION

LINEAR CONVOLUTION SUM METHOD

1. This method is powerful analysis tool for studying LSI Systems.
2. In this method we decompose input signal into sum of elementary signal. Now the elementary
input signals are taken into account and individually given to the system. Now using linearity property
whatever output response we get for decomposed input signal, we simply add it & this will provide us
total response of the system to any given input signal.
3. Convolution involves folding, shifting, multiplication and summation operations.
4. If there are M number of samples in x(n) and N number of samples in h(n) then the maximum
number of samples in y(n) is equals to M+n-1.
Linear Convolution states that
y(n) = x(n) * h(n)
∞ ∞

y(n) = ∑ x (k) h(n – k ) = ∑ x (k) h[ -(k-n) ]

k= -∞ k= -∞

Example 1: h(n) = { 1 , 2 , 1, -1 } & x(n) = { 1, 2, 3, 1 } Find y(n)

METHOD 1: GRAPHICAL REPRESENTATION

Step 1) Find the value of n = nx+ nh = -1 (Starting Index of x(n)+ starting index of h(n))

Step 2) y(n)= { y(-1) , y(0) , y(1), y(2), ….} It goes up to length(xn)+ length(yn) -1.

i.e n=-1 y(-1) = x(k) * h(-1-k)

n=0 y(0) = x(k) * h(0-k)
n=1 y(1) = x(k) * h(1-k) ….
ANSWER : y(n) ={1, 4, 8, 8, 3, -2, -1 }

METHOD 2: MATHEMATICAL FORMULA

Use Convolution formula
∞

y(n) = ∑ x (k) h(n – k )

k= -∞

k= 0 to 3 (start index to end index of x(n))

y(n) = x(0) h(n) + x(1) h(n-1) + x(2) h(n-2) + x(3) h(n-3)

METHOD 3: VECTOR FORM (TABULATION METHOD)

X(n)= {x1,x2,x3} & h(n) ={ h1,h2,h3}
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X1 x2 x3
h1
h1x1 h1x2 h1x3

h2

h3 h2x1 h2x2 h2x3

y(-1) = h1 x1
y(0) = h2 x1 + h1 x2
y(1) = h1 x3 + h2x2 + h3 x1 …………

METHOD 4: SIMPLE MULTIPLICATION FORM

X(n)= {x1,x2,x3} & h(n) ={ h1,h2,h3}
x1 x2 x3
y(n) = ×

y1 y2 y3

PROPERTIES OF LINEAR CONVOLUTION

x(n) = Excitation Input signal
y(n) = Output Response
h(n) = Unit sample response
1. Commutative Law: (Commutative Property of Convolution)
x(n) * h(n) = h(n) * x(n)

X(n) Unit Sample Response = y(n) = x(n) *h(n)

Response =h(n)

h(n) Unit Sample Response = y(n) = h(n) * x(n)

Response =x(n)

2. Associate Law: (Associative Property of Convolution)

[ x(n) * h1(n) ] * h2(n) = x(n) * [ h1(n) * h2(n) ]

X(n) Unit Sample Unit Sample h(n) Response

Response=h1(n) Response=h2(n)

X(n) Unit Sample Response Response

h(n) = h1(n) * h2(n)

3 Distribute Law: (Distributive property of convolution)

x(n) * [ h1(n) + h2(n) ] = x(n) * h1(n) + x(n) * h2(n)
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CAUSALITY OF LSI SYSTEM

The output of causal system depends upon the present and past inputs. The output of the causal system
at n= n0 depends only upon inputs x(n) for n≤ n0. The linear convolution is given as
∞

y(n) = ∑ h(k) x(n–k)

k=-∞

At n= n0 ,the output y(n0) will be

∞

y(n0) = ∑ h(k) x(n0–k)

k=-∞

Rearranging the above terms...

∞ -∞

y(n0) = ∑ h(k) x(n0–k) + ∑ h(k) x(n0–k)

k=0 k=-1

The output of causal system at n= n0 depends upon the inputs for n< n0 Hence
h(-1)=h(-2)=h(-3)=0
Thus LSI system is causal if and only if
h(n) =0 for n<0
This is the necessary and sufficient condition for causality of the system.
Linear convolution of the causal LSI system is given by
n

y(n) = ∑ x (k) h(n – k )

k=0

STABILITY FOR LTI SYSTEM

A System is said to be stable if every bounded input produces a bounded output.

The input x(n) is said to bounded if there exists some finite number Mx such that |x(n)| ≤ Mx < ∞. The
output y(n) is said to bounded if there exists some finite number My such that |y(n)| ≤ My < ∞.
Linear convolution is given by
∞

y(n) = ∑ x (k) h(n – k )

k=- ∞

Taking the absolute value of both sides

∞

|y(n)| = ∑ h(k) x(n-k)

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The absolute values of total sum is always less than or equal to sum of the absolute values of
individually terms. Hence
∞

|y(n)| ≤ ∑ h(k) x(n–k)

k=-∞

|y(n)| ≤ ∑ |h(k)| |x(n–k)|

k=-∞

The input x(n) is said to bounded if there exists some finite number Mx such that |x(n)| ≤ Mx < ∞.
Hence bounded input x(n) produces bounded output y(n) in the LSI system only if ∞
∑ |h(k)| < ∞
k=-∞

With this condition satisfied, the system will be stable. The above equation states that the LSI system is
stable if its unit sample response is absolutely summable. This is necessary and sufficient condition for
the stability of LSI system.

SELF-STUDY: Exercise No. 1

Q1) Show that the discrete time signal is periodic only if its frequency is expressed as the ratio of two
integers.
Q2) Show that the frequency range for discrete time sinusoidal signal is -∏ to ∏ radians/sample or -½
cycles/sample to ½ cycles/sample.
Q3) Prove δ (n)= u(n)= u(n-1).
n
Q4) Prove u(n)= ∑ δ(k)
k=-∞
∞
Q5) Prove u(n)= ∑ δ(n-k)
k=0
Q6) Prove that every discrete sinusoidal signal can be expressed in terms of weighted unit impulse.
Q7) Prove the Linear Convolution theorem.

CORRELATION:
It is frequently necessary to establish similarity between one set of data and another. It means we
would like to correlate two processes or data. Correlation is closely related to convolution, because the
correlation is essentially convolution of two data sequences in which one of the sequences has been
reversed.

Applications are in
1) Images processing for robotic vision or remote sensing by satellite in which data from
different image is compared
2) In rad.arandsonarsystems for range and position finding in which transmitted and reflected
waveforms are compared.
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3) Correlation is also used in detection and identifying of signals in noise.

4) Computation of average power in waveforms.
5) Identification of binary codeword in pulse code modulation system.
DIFFERENCE BETWEEN LINEAR CONVOLUTION AND CORRELATION

S.No Linear Convolution Correlation

1 In case of convolution two signal sequences In case of Correlation, two signal sequences are
input signal and impulse response given by the just compared.
same system is calculated
2 Our main aim is to calculate the response given Our main aim is to measure the degree to which
by the system. two signals are similar and thus to extract some
information that depends to a large extent on the
application
3 Linear Convolution is given by the equation Received signal sequence is given as
y(n) = x(n) * h(n) & calculated as Y(n) = α x(n-D) + ω(n)
∞ Where α= Attenuation Factor
D= Delay
y(n) = ∑ x (k) h(n – k ) ω(n) = Noise signal

k= -∞
4 Linear convolution is commutative Not commutative.

TYPES OF CORRELATION
Under Correlation there are two classes.
1) CROSS CORRELATION: When the correlation of two different sequences x(n) and y(n) is
performed it is called as Cross correlation. Cross-correlation of x(n) and y(n) is rxy(l) which can
be mathematically expressed as

rxy(l) = ∑ x (n) y(n – l )

n= -∞

OR
∞

rxy(l) = ∑ x (n + l) y(n)
n= -∞

2) AUTO CORRELATION: In Auto-correlation we correlate signal x(n) with itself, which can be
mathematically expressed as
∞

rxx(l) = ∑ x (n) x(n – l )

n= -∞

OR

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rxx(l) = ∑ x (n + l) x(n)
n= -∞

PROPERTIES OF CORRELATION
1) The cross-correlation is not commutative.
rxy(l) = ryx(-l)
2) The cross-correlation is equivalent to convolution of one sequence with folded version of another
sequence.
rxy(l) = x(l) * y(-l).
3) The autocorrelation sequence is an even function.
rxx(l) = rxx(-l)

Examples:
Q) Determine cross-correlation sequence
x(n)={2, -1, 3, 7,1,2, -3} & y(n)={1, -1, 2, -2, 4, 1, -2 ,5}
Answer: rxy(l) = {10, -9, 19, 36, -14, 33, 0,7, 13, -18, 16, -7, 5, -3}
Q) Determine autocorrelation sequence
x(n)={1, 2, 1, 1} Answer: rxx(l) = {1, 3, 5, 7, 5, 3, 1}

A/D CONVERSION
BASIC BLOCK DIAGRAM OF A/D CONVERTER

Analog signal Samplerx(n) Quantizer Encoder

Xa(t)

Discrete time Quantized Digital

signal signal signal

1. SAMPLING THEOREM
It is the process of converting continuous time signal into a discrete time signal by taking samples of the
continuous time signal at discrete time instants.

X[n]= Xa(t) where t= nTs = n/Fs ….(1)

When sampling at a rate of fs samples/sec, if k is any positive or negative integer, we cannot distinguish
between the samples values of fa Hz and a sine wave of (fa+ kfs) Hz. Thus (fa + kfs) wave is alias or
image of fa wave.

Thus Sampling Theorem states that if the highest frequency in an analog signal is Fmax and the
signal is sampled at the rate fs > 2Fmax then x(t) can be exactly recovered from its sample values. This
sampling rate is called Nyquist rate of sampling. The imaging or aliasing starts after Fs/2 hence folding
frequency is fs/2. If the frequency is less than or equal to 1/2 it will be represented properly.

Example:
Case 1: X1(t)=cos2.(10) t
Fs= 40 Hz i.e t= n/Fs
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x1[n]= cos 2∏(n/4)= cos (∏/2)n

Case 2: X1(t) = cos 2∏ (50) t Fs= 40 Hz i.e t= n/Fs

x1[n]= cos 2∏(5n/4)= cos 2∏( 1+ ¼)n
= cos (∏/2)n

Thus the frequency 50 Hz, 90 Hz , 130 Hz … are alias of the frequency 10 Hz at the sampling rate of 40
samples/sec

2. QUANTIZATION
The process of converting a discrete time continuous amplitude signal into a digital signal by expressing
each sample value as a finite number of digits is called quantization. The error introduced in
representing the continuous values signal by a finite set of discrete value levels is called quantization
error or quantization noise.

Example: x[n] = 5(0.9)n u(n) where 0 <n < ∞ & fs= 1 Hz

N [n] Xq [n] Rounding Xq [n] Truncating eq [n]

0 5 5.0 5.0 0
1 4.5 4.5 4.5 0
2 4.05 4.0 4.0 -0.05
3 3.645 3.6 3.6 -0.045
4 3.2805 3.2 3.3 0.0195

Quantization Step/Resolution : The difference between the two quantization levels is called quantization
step. It is given by Δ = XMax – xMin / L-1 where L indicates Number of quantization levels.

3. CODING/ENCODING
Each quantization level is assigned a unique binary code. In the encoding operation, the quantization
sample value is converted to the binary equivalent of that quantization level. If 16 quantization levels are
present, 4 bits are required. Thus bits required in the coder is the smallest integer greater than or equal to
Log2 L. i.e b= Log2 L
Thus Sampling frequency is calculated as fs=Bit rate / b.

4. ANTI-ALIASING FILTER
When processing the analog signal using DSP system, it is sampled at some rate depending upon the
bandwidth. For example if speech signal is to be processed the frequencies upon 3khz can be used.
Hence the sampling rate of 6khz can be used. But the speech signal also contains some frequency
components more than 3khz. Hence a sampling rate of 6khz will introduce aliasing. Hence signal should
be band limited to avoid aliasing.
The signal can be band limited by passing it through a filter (LPF) which blocks or attenuates all
the frequency components outside the specific bandwidth. Hence called as Anti aliasing filter or pre-
filter. (Block Diagram).

5. SAMPLE-AND-HOLD CIRCUIT:
The sampling of an analogue continuous-time signal is normally implemented using a device called an
analo guew-tow- wdig.iptaal cdoenveeprtezr.(nAe/Dt ). The continuous-time signal is first passed
through a device called a sample-and-hold (S/H) whose function is to measure the input signal value at
the clock instant and hold
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it fixed for a time interval long enough for the A/D operation to complete. Analogue-to-digital conversion
is potentially a slow operation, and a variation of the input voltage during the conversion may disrupt the
operation of the converter. The S/H prevents such disruption by keeping the input voltage constant during
the conversion. This is schematically illustrated by Figure.

After a continuous-time signal has been through the A/D converter, the quantized output may differ from
the input value. The maximum possible output value after the quantization process could be up to half the
quantization level q above or q below the ideal output value. This deviation from the ideal output value is
called the quantization error. In order to reduce this effect, we increases the number of bits.

Tutorial problems:
Q) Calculate Nyquist Rate for the analog signal x(t)
1) x(t)= 4 cos 50 ∏t + 8 sin 300∏t –cos 100∏t Fn=300 Hz
2) x(t)= 2 cos 2000∏t+ 3 sin 6000∏t + 8 cos 12000∏t Fn=12KHz
3) x(t)= 4 cos 100∏t Fn=100 Hz

Q) The following four analog sinusoidal are sampled with the fs=40Hz. Find out corresponding time
signals and comment on them
X1(t)= cos 2∏(10)t
X2(t)= cos 2∏(50)t
X3(t)= cos 2∏(90)t
X4(t)= cos 2∏(130)t
Q) Signal x1(t).=10coe2∏(100n0)t+5 cos2∏(5000)t. Determine Nyquist rate. If the signal is sampled at
4khz will the signal be recovered from its samples.
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Q) Signal x1(t)=3 cos 600∏t+ 2cos800∏t. The link is operated at 10000 bits/sec and each input sample
is quantized into 1024 different levels. Determine Nyquist rate, sampling frequency, folding frequency
& resolution.

DIFFERENCE EQUATION

S.No Finite Impulse Response (FIR) Infinite Impulse Response (IIR)

1 FIR has an impulse response that is zero outside of some IIR has an impulse response on
finite time interval. infinite time interval.
2 Convolution formula changes to Convolution formula changes to
M ∞

y(n) = ∑ x (k) h(n – k ) y(n) = ∑ x (k) h(n – k )

n= -M n= -∞
For causal FIR systems limits changes to 0 to M. For causal IIR systems limits
changes to 0 to ∞.
3 The FIR system has limited span which views only most The IIR system has unlimited span.
recent M input signal samples forming output called as
―Windowing‖.
4 FIR has limited or finite memory requirements. IIR System requires infinite
memory.
5 Realization of FIR system is generally based on Realization of IIR system is
Convolution Sum Method. generally based on Difference
Method.

Discrete time systems has one more type of classification.

1. Recursive Systems
2. Non-Recursive Systems
S. No Recursive Systems Non-Recursive systems
1 In Recursive systems, the output depends upon past, In Non-Recursive systems, the
present, future value of inputs as well as past output. output depends only upon past,
present or future values of inputs.
2 Recursive Systems has feedback from output to No Feedback.
input.
3 Examples y(n) = x(n) + y(n-2) Y(n) = x(n) + x(n-1)

First order Difference Equation

y(n) = x(n) + a y(n-1)
where y(n) = Output Response of the recursive system
x(n) = Input signal
a= Scaling factor
y(n-1) = Unit delay to output.
Now we will start at n=0
n=0 y(0) = x(0) + a y(-1) ….(1)
n=1 y(1) = x(1) + a y(0) ….(2)
= x(1) + a [ x(0) + a y(-1) ]
=ea2pyz(-.1n)e+ta x(0) + x(1) ….(3)
hence
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n
y(n) = a n+1
y(-1) + ∑ a k x (n -k) n≥0

k= 0

1) The first part (A) is response depending upon initial condition.

2) The second Part (B) is the response of the system to an input signal.

Zero state response (Forced response) : Consider initial condition are zero. (System is relaxed at time
n=0) i.e y(-1) =0

Zero Input response (Natural response) : No input is forced as system is in non-relaxed initial
condition. i.e y(-1) != 0
Total response is the sum of zero state response and zero input response.

Q) Determine zero input response for y(n) – 3y(n-1) – 4y(n-2)=0; (Initial Conditions are y(-1)=5 &
y(-2)= 10) Answer: y(n)= 7 (-1)n + 48 (4)n
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CHAPTER 2

DISCRETE TIME SYSTEM ANALYSIS

2.0 PREREQISTING DISCUSSION ABOUT Z TRANSFORM

For analysis of continuous time LTI system Laplace transform is used. And for analysis of
discrete time LTI system z transform is used. Z transform is mathematical tool used for conversion of
time domain into frequency domain (z domain) and is a function of the complex valued variable Z. The
z transform of a discrete time signal x(n) denoted by X(z) and given as
∞

X(z) = ∑ x (n) z –n z-Transform.……(1)

n=-∞

Z transform is an infinite power series because summation index varies from -∞ to ∞. But it is useful
for values of z for which sum is finite. The values of z for which f (z) is finite and lie within the region
called as ―region of convergence (ROC).

ADVANTAGES OF Z TRANSFORM
1. The DFT can be determined by evaluating z transform.
2. Z transform is widely used for analysis and synthesis of digital filter.
3. Z transform is used for linear filtering. z transform is also used for finding Linear convolution,
cross-correlation and auto-correlations of sequences.
4. In z transform user can characterize LTI system (stable/unstable, causal/anti- causal) and its
response to various signals by placements of pole and zero plot.

ADVANTAGES OF ROC(REGION OF CONVERGENCE)

1. ROC is going to decide whether system is stable or unstable.
2. ROC decides the type of sequences causal or anti-causal.
3. ROC also decides finite or infinite duration sequences.

2.1 Z TRANSFORM PLOT

Imaginary Part of z
Im (z)
Z-Plane
|z|>a

|z|<a

Re (z) Real part of z

Fig showthep.lot ofztransforms. The z transform has real and imaginary parts. Thus a plot of imaginary
part versus real part is called complex z-plane. The radius of circle is 1 called as unit circle.
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This complex z plane is used to show ROC, poles and zeros. Complex variable z is also expressed in
polar form as Z= rejω where r is radius of circle is given by |z| and ω is the frequency of the sequence in
radians and given by ∟z.

S.No Time Domain Property z Transform ROC

Sequence
1 δ(n) (Unit sample) 1 complete z plane
2 δ(n-k) Time shifting z-k except z=0
3 δ(n+k) Time shifting zk except z=∞
4 u(n) (Unit step) 1/1- z-1 = z/z-1 |z| > 1
5 u(-n) Time reversal 1/1- z |z| < 1
6 -u(-n-1) Time reversal z/z- 1 |z| < 1
7 n u(n) (Unit ramp) Differentiation z-1 / (1- z-1)2 |z| > 1
8 an u(n) Scaling 1/1- (az-1) |z| > |a|
9 -an u(-n-1)(Left side 1/1- (az-1) |z| < |a|
exponential sequence)
10 n an u(n) Differentiation a z-1 / (1- az-1)2 |z| > |a|
11 -n an u(-n-1) Differentiation a z-1 / (1- az-1)2 |z| < |a|
12 an for 0 < n < N-1 1- (a z-1)N/ 1- az-1 |az-1| < ∞
except z=0
13 1 for 0<n<N-1 or Linearity 1- z-N/ 1- z-1 |z| > 1
u(n) – u(n-N) Shifting
14 cos(ω0n) u(n) 1- z-1cosω0 |z| > 1

1- 2z-1cosω0+z-2
15 sin(ω0n) u(n) z-1sinω0 |z| > 1

1- 2z-1cosω0+z-2
16 an cos(ω0n) u(n) Time scaling 1- (z/a)-1cosω0 |z| > |a|

1- 2(z/a)-1cosω0+(z/a)-2

17 an sin(ω0n) u(n) Time scaling (z/a)-1sinω0 |z| > |a|

1- 2(z/a)-1cosω0+(z/a)-2

Tutorial problems:
Q) Determine z transform of following signals. Also draw ROC.
i) x(n) = {1,2,3,4,5}
ii) x(n)={1,2,3,4,5,0,7}
Q) Determine z transform and ROC for x(n) = (-1/3)n u(n) –(1/2)n u(-n-1).
Q) Determine z transform and ROC for x(n) = [ 3.(4 n)–4(2n)] u(n).
Q) Determine z transform and ROC for x(n) = (1/2) n u(-n).
Q) Determine z transform and ROC for x(n) = (1/2) n {u(n) – u(n-10)}.
Q) Find linear convolution using z transform. X(n)={1,2,3} & h(n)={1,2}
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2.1.1 PROPERTIES OF Z TRANSFORM (ZT)

1) Linearity
The linearity property states that if
z
x1(n) X1(z) And
z
x2(n) X2(z) Then

Then z
a1 x1(n) + a2 x2(n) a1 X1(z) + a2 X2(z)
z Transform of linear combination of two or more signals is equal to the same linear combination of z
transform of individual signals.

2) Time shifting
The Time shifting property states that if
z
x(n) X(z) And
z
Then x(n-k) X(z) z–k
–k
Thus shifting the sequence circularly by ‗k‘ samples is equivalent to multiplying its z transform by z

3) Scaling in z domain
This property states that if
z
x(n) X(z) And
z
Then an x(n) x(z/a)
Thus scaling in z transform is equivalent to multiplying by a n in time domain.

4) Time reversal Property

The Time reversal property states that if
z
x(n) X(z) And
z
Then x(-n) x(z-1)
It means that if the sequence is folded it is equivalent to replacing z by z -1 in z domain.

5) Differentiation in z domain
The Differentiation property states that if

z
x(n) X(z) And
z
Then n x(n) -z d/dz (X(z))
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6) Convolution Theorem
The Circular property states that if
z
x1(n) X1(z) And
z
x2(n) X2(z) Then
z
Then x1(n) * x2(n) X1(z) X2(z)
N
Convolution of two sequences in time domain corresponds to multiplication of its Z transform sequence
in frequency domain.

7) Correlation Property
The Correlation of two sequences states that if
z
x1(n) X1(z) And
z
x2(n) X2(z) Then
∞ z

then ∑ x1 (l) x2(-l) X1(z) x2(z-1)

n=-∞

8) Initial value Theorem

Initial value theorem states that if
z
x(n) X(z) And
then
x(0) = lim X(Z)
z∞

9) Final value Theorem

Final value theorem states that if
z
x(n) X(z) And
then
lim x(n) = lim(z-1) X(z)
z∞ z1

RELATIONSHIP BETWEEN FOURIER TRANSFORM AND Z TRANSFORM.

There is a close relationship between Z transform and Fourier transform. If we replace the complex
variable z by e –jω, then z transform is reduced to Fourier transform.
Z transform of sequence x(n) is given by
∞

X(z) = ∑ x (n) z –n (Definition of z-Transform)

n=-∞
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Fourier transform of sequence x(n) is given by

∞

X(ω) = ∑ x (n) e –jωn (Definition of Fourier Transform)

n=-∞

Complex variable z is expressed in polar form as Z= rejω where r= |z| and ω is ∟z. Thus we can be
written as
∞

X(z) = ∑ [ x (n) r–n] e–jωn

n=-∞

X(z) z =e
jw
= ∑ x (n) e–jωn

n=-∞
jw
X(z) z =e = x(ω) at |z| = unit circle.

Thus, X(z) can be interpreted as Fourier Transform of signal sequence (x(n) r –n). Here r–n grows with n if
r<1 and decays with n if r>1. X(z) converges for |r|= 1. hence Fourier transform may be viewed as Z
transform of the sequence evaluated on unit circle. Thus The relationship between DFT and Z transform
is given by

X(z) z=ej2∏kn = x(k)

The frequency ω=0 is along the positive Re(z) axis and the frequency ∏/2 is along the positive Im(z)
axis. Frequency ∏ is along the negative Re(z) axis and 3∏/2 is along the negative Im(z) axis.

Im(z)
ω=∏/2
z(0,+j)

z=rejω

ω=∏ ω=0
z(-1,0) z(1,0) Re(z)

ω=3∏/2
z(0,-j)
Frequencyscale on unit circle X(z)= X(ω) on unit circle
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INVERSE Z TRANSFORM (IZT)

The signal can be converted from time domain into z domain with the help of z transform (ZT). Similar
way the signal can be converted from z domain to time domain with the help of inverse z
transform(IZT). The inverse z transform can be obtained by using two different methods.
1) Partial fraction expansion Method (PFE) / Application of residue theorem
2) Power series expansion Method (PSE)

1. PARTIAL FRACTION EXPANSION METHOD

In this method X(z) is first expanded into sum of simple partial fraction.
a0 zm+ a1 zm-1+ …….+ am
X(z) = for m ≤ n

b0 zn + b1 znn-1+ …….+ bn
First find the roots of the denominator polynomial
a0 zm+ a1 zm-1+ …….+ am
X(z) =

(z- p1) (z- p2)…… (z- pn)

The above equation can be written in partial fraction expansion form and find the coefficient AK and
take IZT.

SOLVE USING PARTIAL FRACTION EXPANSION METHOD (PFE)

S.No Function (ZT) Time domain sequence Comment

1 an u(n) for |z| > a causal sequence

1 -an u(-n-1) for |z| < a anti-causal sequence
1- a z-1
1 (-1)n u(n) for |z| > 1 causal sequence
2 -(-1)n u(-n-1) for |z| < a anti-causal sequence
1+z-1

-2(3)n u(-n-1) + (0.5)n u(n) stable system

for 0.5<|z|<3
2(3)n u(n) + (0.5)n u(n) causal system
3 for |z|>3
3-4z-1 -2(3)n u(-n-1) - (0.5)n u(-n-1) for |z|<0.5 anti-causal system
1- 3.5 z-1+1.5z-2

-2(1) n u(-n-1) + (0.5)n u(n) stable system

for 0.5<|z|<1
4 1 2(1)n u(n) + (0.5)n u(n) causal system
for |z|>1
1- 1.5 z-1+0.5z-2 -2(1)n u(-n-1) - (0.5)n u(-n-1) for |z|<0.5 anti-causal system
t
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5 1+2 z-1+ z-2 2δ(n)+8(1)n u(n)- 9(0.5)n u(n) causal system

for |z|>1
1- 3/2 z-1+0.5z-2

1+ z-1 (1/2-j3/2) (1/2+j1/2)n u(n)+ causal system

6 (1/2+j3/2) (1/2+j1/2)n u(n)
1- z-1 + 0.5z-2

1 –(0.5) z-1 4(-1/2)n u(n) – 3 (-1/4)n u(n) for |z|>1/2 causal system
7
1-3/4 z-1+1/8 z-2

1- 1/2 z-1 (-1/2)n u(n) for |z|>1/2 causal system

8
1- 1/4 z-2

z+1 δ(n)+ u(n) – 2(1/3)n u(n) causal system

9 for |z|>1
3z2 - 4z + 1

5z 5(2n-1) causal system

10 for |z|>2
(z-1) (z-2)

z3 4-(n+3)(1/2)n causal system

11 for |z|>1
(z-1) (z-1/2)2

2. RESIDUE THEOREM METHOD

In this method, first find G(z)= zn-1 X(Z) and find the residue of G(z) at various poles of X(z).

SOLVE USING ―RESIDUE THEOREM― METHOD

S. No Function (ZT) Time domain Sequence
1 z For causal sequence (a)n u(n)

z–a
2 z (2n -1 ) u(n)

(z–1)(z-2)

3 z2 + z (2n+1) u(n)

(z – 1)2

4 z3 4 – (n+3)(0.5)n u(n)

(z-1) (z–0.5)2
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3. POWER-SERIES EXPANSION METHOD

The z transform of a discrete time signal x(n) is given as
∞

X(z) = ∑ x (n) z –n (1)

n=-∞

Expanding the above terms we have

x(z) = …..+x(-2)Z2+ x(-1)Z+ x(0)+ x(1) Z-1 + x(2) Z2 +….. (2)
This is the expansion of z transform in power series form. Thus sequence x(n) is given as
x(n) ={ ….. ,x(-2),x(-1),x(0),x(1),x(2),… ............. }.
Power series can be obtained directly or by long division method.
SOLVE USING ―POWER SERIES EXPANSION― METHOD
S.No Function (ZT) Time domain Sequence
1 z For causal sequence an u(n)
For Anti-causal sequence -an u(-n-1)
z-a

2 1 {1,3/2,7/4,15,8,… ........ } For |z| > 1

{… 14,6,2,0,0} For |z| < 0.5
1- 1.5 z-1+0.5z-2

3 z2+z {0,1,4,9,…..} For |z| > 3

z3 -3z2+3z -1

4 z2(1-0.5z-1)(1+z-1) (1-z-1) X(n) ={1,-0.5,-1,0.5}

5 log(1+az-1)
(-1)n+1an/n for n≥1 and |z|>|a|

4. RECURSIVE ALGORITHM
The long division method can be recast in recursive form.
a0 + a1 z-1+ a2 z-2
X(z) =
b0 + b1 z-1+ b2 z-2

Their IZT is give as

n
x(n) = 1/b0 [ an - ∑ x(n-i) bi] for n=1,2,…………….
i=1
Thus
X(0) = a0/b0
X(1) = 1/b0 [ a1- x(0) b1]
X(2) = 1/b0 [ a1- x(1) b1 - x(0) b2] ……………
Click Here for Discrete Time Systems and Signal Processing full study material.

SOLVE USING ―RECURSIVE ALGORITHM― METHOD

S. No Function (ZT) Time domain Sequence
1 -1
1+2z +z -2
X(n) = {1,3,3.6439,….}

1-z-1 +0.3561z2

2 1+z-1 X(n) = {1,11/6,49/36,….}

1-5/6 z-1+ 1/6 z-2

3 z4 +z2 X(n) = { 23/16,63/64,………}

z2-3/4z+ 1/8

POLE –ZERO PLOT

1. X(z) is a rational function, that is a ratio of two polynomials in z -1 or z.
The roots of the denominator or the value of z for which X(z) becomes infinite, defines locations
of the poles. The roots of the numerator or the value of z for which X(z) becomes zero, defines
locations of the zeros.

2. ROC dos not contain any poles of X(z). This is because x(z) becomes infinite at
the locations of the poles. Only poles affect the causality and stability of the system.

3. CASUALTY CRITERIA FOR LSI SYSTEM

LSI system is causal if and only if the ROC the system function is exterior to
the circle. i. e |z| > r. This is the condition for causality of the LSI system in terms of z transform.
(The condition for LSI system to be causal is h(n) = 0 ….. n<0 )

4. STABILITY CRITERIA FOR LSI SYSTEM

Bounded input x(n) produces bounded output y(n) in the LSI system only if
∞
∑ |h(n)| < ∞
n=-∞

With this condition satisfied, the system will be stable. The above equation states that the LSI
system is stable if its unit sample response is absolutely summable. This is necessary and
sufficient condition for the stability of LSI system.

H(z) = ∑ h (n) z –n Z-Transform.……(1)

n=-∞
Click Here for Discrete Time Systems and Signal Processing full study material.

Taking magnitude of both the sides

∞

|H(z)| = ∑ h(n) z –n ..............................................................................(2)

n=-∞

Magnitudes of overall sum is less than the sum of magnitudes of individual sums.
∞

|H(z)| ≤ ∑ h(n) z-n

n=-∞

|H(z)| ≤ ∑ |h(n)| | z-n | ….(3)

n=-∞

5. If H(z) is evaluated on the unit circle | z-n|=|z|=1.

Hence LSI system is stable if and only if the ROC the system function includes the unit circle. i.e
r < 1. This is the condition for stability of the LSI system in terms of z transform. Thus
For stable system |z| < 1
For unstable system |z| > 1
Marginally stable system |z| = 1

Im(z)
z-Plane

Re(z)

Fig: Stable system

Poles inside unit circle gives stable system. Poles outside unit circle gives unstable system.
Poles on unit circle give marginally stable system.
6. A causal and stable system must have a system function that converges for
|z| > r < 1.
 EE6403 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

STANDARD INVERSE Z TRANSFORMS

S. No Function (ZT) Causal Sequence Anti-causal sequence

|z| > |a| |z| <|a|
n n
1 z (a) u(n) -(a) u(-n-1)

z–a

2 z u(n) u(-n-1)

z–1

3 z2 (n+1)an -(n+1)an

(z – a)2

4 zk 1/(k-1)! (n+1) (n+2)……an -1/(k-1)! (n+1) (n+2)………an

(z – a)k

5 1 δ(n) δ(n)

6 Zk δ(n+k) δ(n+k)

7 Z-k δ(n-k) δ(n-k)

ONE SIDED Z TRANSFORM

S.No z Transform (Bilateral) One sided z Transform (Unilateral)

1 z transform is an infinite power series One sided z transform summation index varies from 0 to
because summation index varies from ∞ ∞. Thus One sided z transform are given by
to -∞. Thus Z transform are given by ∞
∞
X(z) = ∑ x (n) z –n
X(z) = ∑ x (n) z –n

n=-∞ n=0

2 z transform is applicable for relaxed One sided z transform is applicable for those systems
systems (having zero initial condition). which are described by differential equations with non zero
initial conditions.
3 z transform is also applicable for non- One sided z transform is applicable for causal systems
causal systems. only.
4 ROC of x(z) is exterior or interior to ROC of x(z) is always exterior to circle hence need not to
circle hence need to specify with z be specified.
trawnswforwm.opfasidgneaelsp.
EE6403 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Properties of one sided z transform are same as that of two sided z transform except shifting property.
1) Time delay
z+
x(n) X+(z) and

z+ k
Then x(n-k) z –k
[ X (z) + ∑ x(-n) zn]
+
k>0
n=1
2) Time advance
z+
x(n) X+(z) and

z+ k-1
Then x(n+k) z k [ X+(z) - ∑ x(n) z-n] k>0
n=0
Examples:
Q) Determine one sided z transform for following signals
1) x(n)={1,2,3,4,5} 2) x(n)={1,2,3,4,5}

SOLUTION OF DIFFERENTIAL EQUATION

One sided Z transform is very efficient tool for the solution of difference equations with nonzero
initial condition. System function of LSI system can be obtained from its difference equation.

∞
Z{x(n-1)} = ∑ x(n-1) z-n (One sided Z transform)
n=0
= x(-1) + x(0) z-1 + x(1) z-2 + x(2) z-3 +………………

= x(-1) + z-1 [x(0) z-1 + x(1) z-2 + x(2) z-3 + .......................]

Z{ x(n-1) } = z-1 X(z) + x(-1)

Z{ x(n-2) } = z-2 X(z) + z-1 x(-1) + x(-2)

Similarly Z{ x(n+1) } = z X(z) - z x(0)

Z{ x(n+2) } = z2 X(z) - z1 x(0) + x(1)

1. Difference equations are used to find out the relation between input and output sequences. It
is also used to relate system function H(z) and Z transform.

2. The transfer function H(ω) can be obtained from system function H(z) by putting z=e jω.
Magnitude and phase response plot can be obtained by putting various values of ω.

Tutorial problems:
Q) A difwferwenwce.peqauadtieonepofzth.ne esytstem is given below
Y(n)= 0.5 y(n-1) + x(n)
EE6403 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Determine a) System function

b) Pole zero plot
c) Unit sample response
Q) A difference equation of the system is given below
Y(n)= 0.7 y(n-1) – 0.12 y(n-2) + x(n-1) + x(n-2)
a) System Function
b) Pole zero plot
c) Response of system to the input x(n) = nu(n)
d) Is the system stable? Comment on the result.
Q) A difference equation of the system is given below
Y(n)= 0.5 x(n) + 0.5 x(n-1)
Determine a) System function
b) Pole zero plot
c) Unit sample response
d) Transfer function
e) Magnitude and phase plot
Q) A difference equation of the system is given below
a. Y(n)= 0.5 y(n-1) + x(n) + x(n-1)
b. Y(n)= x(n) + 3x(n-1) + 3x(n-2) + x(n-3)
a) System Function
b) Pole zero plot
c) Unit sample response
d) Find values of y(n) for n=0,1,2,3,4,5 for x(n)= δ(n) for no initial condition.
Q) Solve second order difference equation
2x(n-2) – 3x(n-1) + x(n) = 3n-2 with x(-2)=-4/9 and x(-1)=-1/3.
Q) Solve second order difference equation
x(n+2) + 3x(n+1) + 2x(n) with x(0)=0 and x(1)=1.
Q) Find the response of the system by using Z transform
x(n+2) - 5x(n+1) + 6x(n)= u(n) with x(0)=0 and x(1)=1.

JURY'S STABILITY CRITERIA / ALGORITHM:

Jury's stability algorithm says
1. Form the first rows of the table by writing the coefficients of D(z).
B0 B1 B2 BN
BN BN-1 BN-2 ------------- B0
2. Form third and fourth rows of the table by evaluating the determinant CJ
B0 BN-J

BN BJ
3. This process will continue until you obtain 2N-3 rows with last two having 3
elements. Y0,Y1,Y2

A digital filter with a system function H(z) is stable, if and only if it passes the following terms.
a. D(Z)|Z=1 >0
b. (-1)N D(Z)|Z=-1 >0
c. |b0|>|bN|, |C0|
EE6403 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

CHAPTER 3

DISCRETE FOURIER TRANSFORM AND COMPUTATION

3.1 PREREQISTING DISCUSSION

Any signal can be decomposed in terms of sinusoidal (or complex exponential) components. Thus
the analysis of signals can be done by transforming time domain signals into frequency domain and
vice-versa. This transformation between time and frequency domain is performed with the help of
Fourier Transform(FT) But still it is not convenient for computation by DSP processors hence Discrete
Fourier Transform(DFT) is used.

Time domain analysis provides some information like amplitude at sampling instant but does not
convey frequency content & power, energy spectrum hence frequency domain analysis is used.

For Discrete time signals x(n) , Fourier Transform is denoted as x(ω) & given by

X(ω) = ∑ x (n) e –jωn FT ............(1)

n=-∞

DFT is denoted by x(k) and given by (ω= 2 ∏ k/N)

N-1

X(k) = ∑ x (n) e –j2 ∏ kn / N

DFT ........(2)

n=0

IDFT is given as
N-1

x(n) =1/N ∑ X (k) e j2 ∏ kn / N

IDFT……(3)

k=0

DIFFERENCE BETWEEN FT & DFT

S. No Fourier Transform (FT) Discrete Fourier Transform (DFT)

1 FT x(ω) is the continuous function DFT x(k) is calculated only at discrete values of ω.
of x(n). Thus DFT is discrete in nature.

2 The range of ω is from - ∏ to ∏ or 0 Sampling is done at N equally spaced points over

to 2∏
. period 0 to 2∏. Thus DFT is sampled version of FT.
3 FT is given by equation (1) DFT is given by equation (2)
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

4 FT equations are applicable to most DFT equations are applicable to causal, finite
of infinite sequences. duration sequences
5 In DSP processors & computers In DSP processors and computers DFT‘s are mostly
applications of FT are limited used.
because x(ω) is continuous function APPLICATION
of ω. a) Spectrum Analysis
b) Filter Design

Tutorial problems:
Q) Prove that FT x(ω) is periodic with period 2∏.
Q) Determine FT of x(n)= an u(n) for -1< a < 1.
Q) Determine FT of x(n)= A for 0 ≤ n ≤ L-1.

Q) Determine FT of x(n)= u(n)

Q) Determine FT of x(n)= δ(n)

Q) Determine FT of x(n)= e–at u(t)

CALCULATION OF DFT & IDFT

For calculation of DFT & IDFT two different methods can be used. First method is using mathematical
equation & second method is 4 or 8 point DFT. If x(n) is the sequence of N samples then consider W N=
e –j2 ∏ / N (twiddle factor)

Four POINT DFT ( 4-DFT)

S. No WN=W4=e –j ∏/2 Angle Real Imaginary Total
1 W4 0 0 1 0 1
2 W4 1 - ∏/2 0 -j -j
3 W4 2 -∏ -1 0 -1
4 W4 3 - 3 ∏ /2 0 J J

n=0 n=1 n=2 n=3

k=0 W4 0 W4 0 W4 0 W4 0
[WN] =k=1 W4 0 W4 1 W4 2 W4 3
0 2 4
k=2 W4 W4 W4 W4 6
k=3 W4 0 W4 3 W4 6 W4 9

Thus 4 point DFT is given as XN = [WN ] XN

1 1 1 1
[WN] = 1 –j -1 j
-1 1 -1
1 j -1 -j
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

EIGHT POINT DFT ( 8-DFT)

S. No WN = W8= e –j ∏/4 Angle Magnitude Imaginary Total

1 W8 0 0 1 ---- 1
2 W8 1 - ∏/4 1/√2 -j 1/√2 1/√2 -j 1/√2
3 W8 2 - ∏/2 0 -j -j
4 W8 3 - 3 ∏ /4 -1/√2 -j 1/√2 - 1/√2 -j 1/√2
5 W8 4 -∏ -1 ---- -1
6 W8 5 - 5∏ / 4 -1/√2 +j 1/√2 - 1/√2 + j 1/√2
7 W8 6 - 7∏ / 4 0 J J
8 W8 7 - 2∏ 1/√2 +j 1/√2 1/√2 + j 1/√2

Remember that W8 0 = W8 8 = W8 16 = W8 24 = W8 32 = W8 40 (Periodic Property)

Magnitude and phase of x(k) can be obtained as,
|x(k)| = sqrt ( Xr(k)2 + XI(k)2)
Angle x(k) = tan -1 (XI(k) / XR(k))

Tutorial problems:
Q) Compute DFT of x(n) = {0,1,2,3} Ans: x4=[6, -2+2j, -2, -2-2j ]
Q) Compute DFT of x(n) = {1,0,0,1} Ans: x4=[2, 1+j, 0, 1-j ]
Q) Compute DFT of x(n) = {1,0,1,0} Ans: x4=[2, 0, 2, 0 ]
Q) Compute IDFT of x(k) = {2, 1+j, 0, 1-j } Ans: x4=[1,0,0,1]

DIFFERENCE BETWEEN DFT & IDFT

S.No DFT (Analysis transform) IDFT (Synthesis transform)

1 DFT is finite duration discrete frequency IDFT is inverse DFT which is used to calculate time
sequence that is obtained by sampling one domain representation (Discrete time sequence) form
period of FT. of x(k).
2 DFT equations are applicable to causal IDFT is used basically to determine sample response
finite duration sequences. of a filter for which we know only transfer function.
3 Mathematical Equation to calculate DFT is Mathematical Equation to calculate IDFT is given by
given by

N-1
N-1
x(n) = 1/N ∑ X (k)e j2 ∏ kn / N

X(k) = ∑ x (n) e –j2 ∏ kn / N

n=0 n=0

4 Thus DFT is given by In DFT and IDFT difference is of factor 1/N & sign
X(k)= [WN][xn] of exponent of twiddle factor.
Thus
t x(n)= 1/N [ WN]-1[XK]
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

PROPERTIES OF DFT
DFT
x(n) x(k)
N
1. Periodicity
Let x(n) and x(k) be the DFT pair then if
x(n+N) = x(n) for all n then
X(k+N) = X(k) for all k
Thus periodic sequence xp(n) can be given as
∞

xp(n) = ∑ x(n-lN)

l=-∞

2. Linearity
The linearity property states that if
DFT
x1(n) X1(k) And
N
DFT
x2(n) X2(k) Then
N
Then DFT
a1 x1(n) + a2 x2(n) a1 X1(k) + a2 X2(k)
N
DFT of linear combination of two or more signals is equal to the same linear combination of DFT of
individual signals.

3. Circular Symmetries of a sequence

A) A sequence is said to be circularly even if it is symmetric about the point zero on the circle. Thus
X(N-n) = x(n)

B) A sequence is said to be circularly odd if it is anti symmetric about the point zero on the circle. Thus
X(N-n) = - x(n)

C) A circularly folded sequence is represented as x((-n))N and given by x((-n))N = x(N-n).

D) Anticlockwise direction gives delayed sequence and clockwise direction gives advance sequence.
Thus delayed or advances sequence x`(n) is related to x(n) by the circular shift.

4. Symmetry Property of a sequence

A) Symmetry property for real valued x(n) i.e xI(n)=0

This propertyst.ates thaix(n.) is real then X(N-k) = X*(k)=X(-k)

 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

B) Real and even sequence x(n) i.e xI(n)=0 & XI(K)=0

This property states that if the sequence is real and even x(n)= x(N-n) then DFT becomes
N-1

X(k) = ∑ x(n) cos (2∏kn/N)

n=0

C) Real and odd sequence x(n) i.e xI(n)=0 & XR(K)=0

This property states that if the sequence is real and odd x(n)=-x(N-n) then DFT becomes
N-1

X(k) = -j ∑ x(n) sin (2∏kn/N)

n=0

D) Pure Imaginary x(n) i.e xR(n)=0

This property states that if the sequence is purely imaginary x(n)=j X I(n) then DFT becomes
N-1

XR(k) = ∑ xI(n) sin (2∏kn/N)

n=0

N-1

XI(k) = ∑ xI(n) cos (2∏kn/N)

n=0

5. Circular Convolution
The Circular Convolution property states that if
DFT
x1(n) X1(k) And
N

DFT
x2(n) X2(k) Then
N
DFT
Then x1(n) x2(n) x1(k) x2(k)
N
It meansthat circular convolutionof x1(n) & x2(n) is equal to multiplication of their DFT‘s. Thus
circular convolution of two periodic discrete signal with period N is given by
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

N-1

y(m) = ∑ x1 (n) x2 (m-n)N.............................................................. (4)

n=0

Multiplication of two sequences in time domain is called as Linear convolution while Multiplication of
two sequences in frequency domain is called as circular convolution. Results of both are totally different
but are related with each other.

DFT
x(n) x(k) x(k)*h(k) x(n)*h(n)
IDFT
*
DFT h(n) h(k)

There are two different methods are used to calculate circular convolution
1) Graphical representation form
2) Matrix approach

DIFFERENCE BETWEEN LINEAR CONVOLUTION & CIRCULAR CONVOLUTION

S. No Linear Convolution Circular Convolution

1 In case of convolution two signal sequences input Multiplication of two DFT‘s is called as
signal x(n) and impulse response h(n) given by the circular convolution.
same system, output y(n) is calculated
2 Multiplication of two sequences in time domain is Multiplication of two sequences in frequency
called as Linear convolution domain is called as circular convolution.

3 Linear Convolution is given by the equation y(n) Circular Convolution is calculated as

= x(n) * h(n) & calculated as N-1
∞
y(m) = ∑ x1 (n) x2 (m-n)N
y(n) = ∑ x (k) h(n – k )
n=0
k= -∞

4 Linear Convolution of two signals returns N-1 Circular convolution returns same number of
elements where N is sum of elements in both elements that of two signals.
sequences.

Tutorial problems:
Q) The two sequences x1(n)={2,1,2,1} & x2(n)={1,2,3,4}. Find out the sequence x3(m) which is equal
to circular conv.olutioetwo.sequences. Ans: X3(m)={14,16,14,16}
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Q) x1(n)={1,1,1,1,-1,-1,-1,-1} & x2(n)={0,1,2,3,4,3,2,1}. Find out the sequence x3(m) which is equal to
circular convolution of two sequences. Ans: X3(m)={-4,-8,-8,-4,4,8,8,4}

Q) Perform Linear Convolution of x(n)={1,2} & h(n)={2,1} using DFT & IDFT.

Q) Perform Linear Convolution of x(n)={1,2,2,1} & h(n)={1,2,3} using 8 Pt DFT & IDFT.

6. Multiplication
The Multiplication property states that if
DFT
X1(n) x1(k) and
N

DFT
X2(n) x2(k) then
N

DFT
Then x1(n) x2(n) 1/N x1(k) x2(k)
N
It means that multiplication of two sequences in time domain results in circular convolution of their
DFT‘s in frequency domain.

7. Time reversal of a sequence

The Time reversal property states that if
DFT
X(n) x(k) and
N

DFT
Then x((-n))N = x(N-n) x((-k))N = x(N-k)
N
It means that the sequence is circularly folded its DFT is also circularly folded.

8. Circular Time shift

The Circular Time shift states that if
DFT
X(n) x(k) And
N
DFT
Then x((n-l))N x(k) e –j2 ∏ k l / N
N

Thus shifting the sequence circularly by ‗l‘ samples is equivalent to multiplying its DFT by
e –j2 ∏ k l / N
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

9. Circular frequency shift

The Circular frequency shift states that if
DFT
X(n) x(k) And
N
DFT
j2 ∏ l n / N
Then x(n) e x((n-l))N
N
Thus shifting the frequency components of DFT circularly is equivalent to multiplying its time domain
sequence by e –j2 ∏ k l / N

10. Complex conjugate property

The Complex conjugate property states that if
DFT
X(n) x(k) then
N
DFT
x*(n) x*((-k))N = x*(N-k) And
N
DFT
x*((-n))N = x*(N-k) x*(k)
N
11. Circular Correlation

The Complex correlation property states

DFT
rxy(l) Rxy(k)= x(k) Y*(k)
N
Here rxy(l) is circular cross correlation which is given as

N-1

rxy(l) = ∑ x (n) y*((n – l ))N

n=0

This means multiplication of DFT of one sequence and conjugate DFT of another sequence is equivalent
to circular cross-correlation of these sequences in time domain.

12. Parseval’s Theorem

The Parseval‘s theorem states

N-1 N-1

∑ X(n) y*(n) = 1/N ∑ x (k) y*(k)

n=0 .n=
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

This equation give energy of finite duration sequence in terms of its frequency components.

APPLICATION OF DFT

1. DFT FOR LINEAR FILTERING

Consider that input sequence x(n) of Length L & impulse response of same system is h(n) having M
samples. Thus y(n) output of the system contains N samples where N=L+M-1. If DFT of y(n) also
contains N samples then only it uniquely represents y(n) in time domain. Multiplication of two DFT‘s is
equivalent to circular convolution of corresponding time domain sequences. But the length of x(n) &
h(n) is less than N. Hence these sequences are appended with zeros to make their length N called as
―Zero padding‖. The N point circular convolution and linear convolution provide the same sequence.
Thus linear convolution can be obtained by circular convolution. Thus linear filtering is provided by
DFT.

When the input data sequence is long then it requires large time to get the output sequence. Hence other
techniques are used to filter long data sequences. Instead of finding the output of complete input
sequence it is broken into small length sequences. The output due to these small length sequences are
computed fast. The outputs due to these small length sequences are fitted one after another to get the
final output response.

METHOD 1: OVERLAP SAVE METHOD OF LINEAR FILTERING

Step 1> In this method L samples of the current segment and M-1 samples of the previous segment
forms the input data block. Thus data block will be

X1(n) ={0,0,0,0,0,………………… ,x(0),x(1), ................... x(L-1)}

X2(n) ={x(L-M+1),..................... x(L-1),x(L),x(L+1),,,,,,,,,,,,,x(2L-1)}
X3(n) ={x(2L-M+1), ..................... x(2L-1),x(2L),x(2L+2),,,,,,,,,,,,,x(3L-1)}

Step2> Unit sample response h(n) contains M samples hence its length is made N by padding zeros.
Thus h(n) also contains N samples.

h(n)={ h(0), h(1), …………….h(M-1), 0,0,0,…...........................(L-1 zeros)}

Step3> The N point DFT of h(n) is H(k) & DFT of mth data block be xm(K) then corresponding DFT of
output be Y`m(k)

Y`m(k)= H(k) xm(K)

Step 4> The sequence ym(n) can be obtained by taking N point IDFT of Y` m(k). Initial
(M-1) samples in the corresponding data block must be discarded. The last L samples are the correct
output samples. Such blocks are fitted one after another to get the final output.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

X(n) of Size N

Size L
X1(n)

M-1 Size L
Zeros
X2(n)
Size L

X3(n)
Y1(n)
Discard M-1 Points
Y2(n)
Discard M-1 Points

Y3(n)

Discard M-1 Point

Y(n) of Size N

METHOD 2: OVERLAP ADD METHOD OF LINEAR FILTERING

Step 1> In this method L samples of the current segment and M-1 samples of the previous segment
forms the input data block. Thus data block will be

X1(n) ={x(0),x(1),…………….x(L-1),0,0,0,…........ }
X2(n) ={x(L),x(L+1),x(2L-1),0,0,0,0}
X3(n) ={x(2L),x(2L+2),,,,,,,,,,,,,x(3L-1),0,0,0,0}

Step2> Unit sample response h(n) contains M samples hence its length is made N by padding zeros.
Thus h(n) also contains N samples.

h(n)={ h(0), h(1), …………….h(M-1), 0,0,0,…...........................(L-1 zeros)}

Step3> The N point DFT of h(n) is H(k) & DFT of mth data block be xm(K) then corresponding DFT of
output be Y`m(k)

Y`m(k)= H(k) xm(K)

Step 4> The sequence ym(n) can be obtained by taking N point IDFT of Y` m(k). Initial
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

(M-1) samples are not discarded as there will be no aliasing. The last (M-1) samples of current output
block must be added to the first M-1 samples of next output block. Such blocks are fitted one after
another to get the final output.

X(n) of Size N

Size L
M-1
X1(n) Zeros
Size L
M-1
X2(n) Zeros
Size L
M-1
Zeros X3(n)

Y1(n)

M-1
Points add Y2(n)
together

Y3(n)

Y(n) of Size N

DIFFERENCE BETWEEN OVERLAP SAVE AND OVERLAP ADD METHOD

S. No OVERLAP SAVE METHOD OVERLAP ADD METHOD

1 In this method, L samples of the current In this method L samples from input
segment and (M-1) samples of the previous sequence and padding M-1 zeros forms data
segment forms the input data block. block of size N.

2 Initial M-1 samples of output sequence are There will be no aliasing in output data
discarded which occurs due to aliasing blocks.
effect.

3 To avoid loss of data due to aliasing last M- Last M-1 samples of current output block
1 samples of each data record are saved. must be added to the first M-1 samples of
next output block. Hence called as overlap
add method.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

2. SPECTRUM ANALYSIS USING DFT

DFT of the signal is used for spectrum analysis. DFT can be computed on digital computer or digital
signal processor. The signal to be analyzed is passed through anti-aliasing filter and samples at the rate
of Fs≥ 2 Fmax. Hence highest frequency component is Fs/2.

Frequency spectrum can be plotted by taking N number of samples & L samples of waveforms. The
total frequency range 2∏ is divided into N points. Spectrum is better if we take large value of N & L But
this increases processing time. DFT can be computed quickly using FFT algorithm hence fast processing
can be done. Thus most accurate resolution can be obtained by increasing number of samples.

FAST FOURIER ALGORITHM (FFT)

1. Large number of the applications such as filtering, correlation analysis, spectrum analysis require
calculation of DFT. But direct computation of DFT require large number of computations and hence
processor remain busy. Hence special algorithms are developed to compute DFT quickly called as Fast
Fourier algorithms (FFT).

3. The radix-2 FFT algorithms are based on divide and conquer approach. In this method, the N-point
DFT is successively decomposed into smaller DFT‘s. Because of this decomposition, the number of
computations are reduced.

3.7.1 RADIX-2 FFT ALGORITHMS

1. DECIMATION IN TIME (DITFFT)

There are three properties of twiddle factor WN

1) W Nk+N = W NK (Periodicity Property)

2) W Nk+N/2 = -W NK (Symmetry Property)
3) W N2= WN/2.

N point sequence x(n) be splitted into two N/2 point data sequences f1(n) and f2(n). f1(n) contains even
numbered samples of x(n) and f2(n) contains odd numbered samples of x(n). This splitted operation is
called decimation. Since it is done on time domain sequence it is called ―Decimation in Time‖. Thus
f1(m)=x(2m) where n=0,1,… ........... N/2-1
f2(m)=x(2m+1) where n=0,1,… ............ N/2-1
N point DFT is given as

N-1

X(k) =∑ x (n) WNkn (1)

n=0
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Since the sequence x(n) is splitted into even numbered and odd numbered samples, thus

N/2-1 N/2-1

X(k) =∑ x (2m) W N2mk + ∑ x (2m+1) W Nk(2m+1) (2)

m=0 m=0

X(k) =F1(k) + WNk F2(k) (3)

X(k+N/2) =F1(k) - WNk F2(k) (Symmetry property) (4)

Fig 1 shows that 8-point DFT can be computed directly and hence no reduction in computation.

x(0) X(0)
x(1) X(1)
x(2) X(2)
x(3) X(3)
8 Point DFT

x(7) X(7)

Fig 1. DIRECT COMPUTATION FOR N=8

x(0) f1(0) X(0)

x(2) f1(1) X(1)
x(4) N/2 Point f1(2) X(2)
x(6) DFT f1(3) X(3)

x(1) f2( w80 X(4)

x(3) f2(1) w81 X(5)
x(5) N/2 Point f2(2) w82 X(6)
x(7) DFT f2(3) w83 X(7)

Fig 2. FIRST STAGE FOR FFT COMPUTATION FOR N=8

Fig 3 shows N/2 point DFT base separated in N/4 boxes. In such cases equations become
g1(k) =P1(k) + WN2k P2(k) ------- (5)
2k
g1(k+N/2) =p1(k) - WN P2(k) -------- (6)
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

x(0) F(0)

x(4) N/4 Point F(1)

DFT

x(2) w80 F(2)

N/4 Point
x(6) w82 F(3)
DFT

Fig 3. SECOND STAGE FOR FFT COMPUTATION FOR N=8

a A= a + WNr b

b WNr B= a - WNr b

Fig 4. BUTTERFLY COMPUTATION (THIRD STAGE)

x(0) A X(0)

x(2) w80 B X(1)

x(1) C w80 X(2)

x(3) w80 D w81 X(3)

Fig 5. SIGNAL FLOW GRAPH FOR RADIX- DIT FFT N=4

x(0) A1 A2 X(0)

x(4) w80 B1 B2 X(1)

x(2) C1 w80 C2 X(2)

x(6) w80 D1 w82 D2 X(3)

x(1) E1 E2 w80 X(4)

x(5) w80 F1 F2 w81 X(5)

x(3) G1 w80 G2 w82 X(6)

x(7) w80 H1 w82 H2 w83 X(7)

 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

2 Point
Combine 2
DFT
Point DFT’s

2 Point
DFT Combine 4
Point DFT

2 Point
DFT Combine 2
Point DFT’s

2 Point
DFT

Fig 7. BLOCK DIAGRAM FOR RADIX- DIT FFT N=8

COMPUTATIONAL COMPLEXITY  FFT V/S DIRECT COMPUTATION

For Radix-2 algorithm value of N is given as N= 2V

Hence value of v is calculated as

V= log10 N / log10 2

= log2 N

Thus if value of N is 8 then the value of v=3. Thus three stages of decimation. Total number of
butterflies will be Nv/2 = 12.

If value of N is 16 then the value of v=4. Thus four stages of decimation. Total number of butterflies
will be Nv/2 = 32.

Each butterfly operation takes two addition and one multiplication operations. Direct computation
requires N2 multiplication operation & N2 – N addition operations.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

N Direct computation DIT FFT algorithm Improvement in

processing speed for
Complex Complex Complex Complex multiplication
Multiplication Addition Multiplication Addition

N2 N2 – N N/2 log2 N N log2 N

8 64 52 12 24 5.3 times

16 256 240 32 64 8 times

256 65536 65280 1024 2048 64 times

MEMORY REQUIREMENTS AND IN PLACE COMPUTATION

a A= a + WNr b

b WNr B= a - WNr b

Fig. BUTTERFLY COMPUTATION

From values a and b new values A and B are computed. Once A and B are computed, there is no need to
store a and b. Thus same memory locations can be used to store A and B where a and b were stored
hence called as In place computation. The advantage of in place computation is that it reduces memory
requirement.

Thus for computation of one butterfly, four memory locations are required for storing two complex
numbers A and B. In every stage there are N/2 butterflies hence total 2N memory locations are required.
2N locations are required for each stage. Since stages are computed successively these memory
locations can be shared. In every stage N/2 twiddle factors are required hence maximum storage
requirements of N point DFT will be (2N + N/2).

BIT REVERSAL

For 8 point DIT DFT input data sequence is written as x(0), x(4), x(2), x(6), x(1), x(5), x(3), x(7) and the
DFT sequence X(k) is in proper order as X(0), X(1), X(2), X(3), X(4), x(5), X(6), x(7). In DIF FFT it is
exactly opposite. This can be obtained by bit reversal method.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Decimal Memory Address x(n) in Memory Address in bit reversed New Address
binary (Natural Order) order in decimal

0 0 0 0 0 0 0 0

1 0 0 1 1 0 0 4

2 0 1 0 0 1 0 2

3 0 1 1 1 1 0 6

4 1 0 0 0 0 1 1

5 1 0 1 1 0 1 5

6 1 1 0 0 1 1 3

7 1 1 1 1 1 1 7

Table shows first column of memory address in decimal and second column as binary. Third column
indicates bit reverse values. As FFT is to be implemented on digital computer simple integer division by
2 method is used for implementing bit reversal algorithms. Flow chart for Bit reversal algorithm is as
follows,
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

DECIMAL NUMBER
B TO BE REVERSED

I=1

B1=B

B2=Int(B1/2)

BR=(2*BR)+ B1- (2 * B2)]

B1=B2;

Is

I > log2N

Store BR as Bit
reversal of B

FLOW CHART

2. DECIMATION IN FREQUENCY (DIFFFT)

In DIF N Point DFT is splitted into N/2 points DFT‘s. X(k) is splitted with k even and k odd this is
called Decimation in frequency(DIF FFT).

N point DFT is given as

N-1

X(k) =∑ x (n) WNkn (1)

=0
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Since the sequence x(n) is splitted N/2 point samples, thus

N/2-1 N/2-1

X(k) =∑ x (n) WN kn + ∑ x (n + N/2) W Nk(n+N/2) (2)

m=0 m=0

N/2-1 N/2-1

X(k) =∑ x (n) WN kn + W NkN/2 ∑ x (n + N/2) W Nkn

m=0 m=0

N/2-1 N/2-1

X(k) =∑ x (n) W Nkn + (-1)k ∑ x (n + N/2) W Nkn

m=0 m=0

N/2-1

X(k) =∑ x (n) + (-1)k x(n + N/2)W Nkn (3)

m=0

Let us split X(k) into even and odd numbered samples

N/2-1

X(2k) =∑ x (n) + (-1)2k x(n + N/2) WN2kn (4)

m=0

N/2-1

X(2k+1) =∑ x (n)+(-1)(2k+1) x(n + N/2)W N(2k+1)n (5)

m=0

Equation (4) and (5) are thus simplified as

g1(n) = x (n) + x(n + N/2)
g2(n) = x (n) - x(n + N/2) W Nn

Fig 1 shows Butterfly computation in DIF FFT.

a A= a + b

b WN r B= (a –b)WNr

BUTTERFLY COMPUTATION
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Fig 2 shows signal flow graph and stages for computation of radix-2 DIF FFT algorithm of N=4

x(0) A X(0)

X(1) x(1)

X(2) B w40 x(2) X(3)

0
w4 C x(3) w41 D
0
w4

Fig 2. SIGNAL FLOW GRAPH FOR RADIX- DIF FFT N=4

Fig 3 shows signal flow graph and stages for computation of radix-2 DIF FFT algorithm of N=8

x(0) A1 A2 X(0)

x(1) B1 B2 w8X(4)

x(2) C1 w80 C2 X(2)

x(3) D1 w82 D2 w80X(6)

x(4) w80 E1 E2 X(1)

x(5) w81 F1 F2 w80X(5)

x(6) w82 G1 w80 G2 X(3)

x(7) w83 H1 w82 H2 w8X(7)

Fig 3. SIGNAL FLOW GRAPH FOR RADIX- DIF FFT N=8

DIFFERENCE BETWEEN DITFFT AND DIFFFT

S. No DIT FFT DIF FFT

1 DITFFT algorithms are based upon DIFFFT algorithms are based upon
decomposition of the input sequence into decomposition of the output sequence into
smaller and smaller sub sequences. smaller and smaller sub sequences.

2 In this input sequence x(n) is splitted into In this output sequence X(k) is considered to
even and odd numbered samples be splitted into even and odd numbered
samples

3 Splitting operation is done on time domain Splitting operation is done on frequency

sequence. domain sequence.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

4 In DIT FFT input sequence is in bit In DIFFFT, input sequence is in natural order.
reversed order while the output sequence is And DFT should be read in bit reversed order.
in natural order.

DIFFERENCE BETWEEN DIRECT COMPUTATION & FFT

S. No Direct Computation Radix -2 FFT Algorithms

1 Direct computation requires large number Radix-2 FFT algorithms requires less number
of computations as compared with FFT of computations.
algorithms.

2 Processing time is more and more for large Processing time is less hence these algorithms
number of N hence processor remains busy. compute DFT very quickly as compared with
direct computation.

3 Direct computation does not requires Splitting operation is done on time domain
splitting operation. basis (DIT) or frequency domain basis (DIF)

4 As the value of N in DFT increases, the As the value of N in DFT increases, the
efficiency of direct computation decreases. efficiency of FFT algorithms increases.

5 In those applications where DFT is to be Applications

computed only at selected values of
1) Linear filtering
k(frequencies) and when these values are
2) Digital filter design
less than log2N then direct computation
becomes more efficient than FFT.

Tutorial problems:
Q) x(n)={1,2,2,1} Find X(k) using DITFFT.
Q) x(n)={1,2,2,1} Find X(k) using DIFFFT.
Q) x(n)={0.3535,0.3535,0.6464,1.0607,0.3535,-1.0607,-1.3535,-0.3535} Find X(k) using DITFFT.
Q) Using radix 2 FFT algorithm, plot flow graph for N=8.

3.8 GOERTZEL ALGORITHM

FFT algorithms are used to compute N point DFT for N samples of the sequence x(n). This requires N/2
log2N number of complex multiplications and N log2N complex additions. In some applications DFT is
to be computed only at selected values of frequencies and selected values are less than log 2N, then direct
computations of DFT becomes more efficient than FFT. This direct computations of DFT can be
realized through linear filtering of x(n). Such linear filtering for computation of DFT can be
implemented using Goertzel algorithm.
.
By definitionNpoint DFTis given as
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

N-1

X(k) =∑ x (m) WNkm (1)

m=0

Multiplying both sides by WN-kN (which is always equal to 1).

N-1

X(k) =∑ x (m) W Nk(N-m) (2)

m=0

Thus for LSI system which has input x(n) and having unit sample response

hk(n)= WN-kn u(n)

X(n) hk(n)= WN-kn u(n) yk(n)

Linear convolution is given by

∞

y(n) = ∑ x (k) h(n – k )

k=- ∞

yk(n) = ∑ x (m) W N
-k(n-m)
u(n–m) (3)
m=-∞

As x(m) is given for N values

N-1

yk(n) = ∑ x (m) W N-k(n-m) (4)

m=0

The output of LSI system at n=N is given by

yk(n)|n=N = ∑ x (m) W N
-k(N-m)
(5)
m=-∞

Thus comparing equation (2) and (5),

X(k) = yk(n)| n=N

Thus DFT can be obtained as the output of LSI system at n=N. Such systems can give X(k) at selected
values owf k Thus DaFdTeies cpozmpnuetetd as linear filtering operations by Goertzel Algorithm.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

CHAPTER 4

DESIGN OF DIGITAL FILTERS

4.1 PREREQISTING DISCUSSION

To remove or to reduce strength of unwanted signal like noise and to improve the quality of
required signal filtering process is used. To use the channel full bandwidth we mix up two or more
signals on transmission side and on receiver side we would like to separate it out in efficient way. Hence
filters are used. Thus the digital filters are mostly used in

1. Removal of undesirable noise from the desired signals

2. Equalization of communication channels
3. Signal detection in radar, sonar and communication
4. Performing spectral analysis of signals.

Analog and digital filters

In signal processing, the function of a filter is to remove unwanted parts of the signal, such as random
noise, or to extract useful parts of the signal, such as the components lying within a certain frequency
range.
The following block diagram illustrates the basic idea.

There are two main kinds of filter, analog and digital. They are quite different in their physical makeup
and in how they work.
An analog filter uses analog electronic circuits made up from components such as resistors,
capacitors and op amps to produce the required filtering effect. Such filter circuits are widely used in
such applications as noise reduction, video signal enhancement, graphic equalizers in hi-fi systems, and
many other areas.
In analog filters the signal being filtered is an electrical voltage or current which is the direct analogue of
the physical quantity (e.g. a sound or video signal or transducer output) involved.
A digital filter uses a digital processor to perform numerical calculations on sampled values of
the signal. The processor may be a general-purpose computer such as a PC, or a specialized DSP
(Digital Signal Processor) chip.
The analog input signal must first be sampled and digitized using an ADC (analog to digital
converter). The resulting binary numbers, representing successive sampled values of the input signal, are
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

transferred to the processor, which carries out numerical calculations on them. These calculations
typically involve multiplying the input values by constants and adding the products together. If
necessary, the results of these calculations, which now represent sampled values of the filtered signal,
are output through a DAC (digital to analog converter) to convert the signal back to analog form.
In a digital filter, the signal is represented by a sequence of numbers, rather than a voltage or
current.
The following diagram shows the basic setup of such a system.

BASIC BLOCK DIAGRAM OF DIGITAL FILTERS

Quantizer Digital Filter

Analog signal Sampler & Encoder
Xa (t)
Discrete
Time signal Digital signal
1. Samplers are used for converting continuous time signal into a discrete time signal by taking
samples of the continuous time signal at discrete time instants.
2. The Quantizer are used for converting a discrete time continuous amplitude signal into a digital
signal by expressing each sample value as a finite number of digits.
3. In the encoding operation, the quantization sample value is converted to the binary equivalent of
that quantization level.
4. The digital filters are the discrete time systems used for filtering of sequences. These digital filters
performs the frequency related operations such as low pass, high pass, band pass and band reject etc.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

These digital Filters are designed with digital hardware and software and are represented by
difference equation.
DIFFERENCE BETWEEN ANALOG FILTER AND DIGITAL FILTER
S. No Analog Filter Digital Filter
1 Analog filters are used for filtering Digital filters are used for filtering digital sequences.
analog signals.
2 Analog filters are designed with various Digital Filters are designed with digital hardware like FF,
components like resistor, inductor and counters shift registers, ALU and software‘s like C or
capacitor assembly language.
3 Analog filters less accurate & because of Digital filters are less sensitive to the environmental
component tolerance of active changes, noise and disturbances. Thus periodic
components & more sensitive to calibration can be avoided. Also they are extremely
environmental changes. stable.
4 Less flexible These are most flexible as software programs & control
programs can be easily modified. Several input signals
can be filtered by one digital filter.
5 Filter representation is in terms of Digital filters are represented by the difference equation.
system components.
6 An analog filter can only be changed by A digital filter is programmable, i.e. its operation is
redesigning the filter circuit. determined by a program stored in the processor's
memory. This means the digital filter can easily be
changed without affecting the circuitry (hardware).

FILTER TYPES AND IDEAL FILTER CHARACTERISTIC

Filters are usually classified according to their frequency-domain characteristic as lowpass, highpass,
bandpass and bandstop filters.
1. Lowpass Filter
A lowpass filter is made up of a passband and a stopband, where the lower frequencies
Of the input signal are passed through while the higher frequencies are attenuated.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

|H (ω)|
1

ω
-ωc ωc
2. Highpass Filter
A highpass filter is made up of a stopband and a passband where the lower frequencies
of the input signal are attenuated while the higher frequencies are passed.
|H(ω)|
ω
-ωc ωc
3. Bandpass Filter
A bandpass filter is made up of two stopbands and one passband so that the lower and
higher frequencies of the input signal are attenuated while the intervening
frequencies are passed.
|H(ω)|
1

ω
-ω2 -ω1 ω2 ω1
4. Bandstop Filter
A bandstop filter is made up of two passbands and one stopband so that the lower and higher
frequencies of the input signal are passed while the intervening frequencies are attenuated. An idealized
bandstop filter frequency response has the following shape.

|H(ω)|
1

www.BrainKart.com
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

ω
5. Multipass Filter
A multipass filter begins with a stopband followed by more than one passband. By default, a multipass
filter in Digital Filter Designer consists of three passbands and four stopbands. The frequencies of the
input signal at the stopbands are attenuated while those at the passbands are passed.
6. Multistop Filter
A multistop filter begins with a passband followed by more than one stopband. By default, a
multistop filter in Digital Filter Designer consists of three passbands and two stopbands.
7. All Pass Filter
An all pass filter is defined as a system that has a constant magnitude response for all frequencies.
|H(ω)| = 1 for 0 ≤ ω < ∏
The simplest example of an all pass filter is a pure delay system with system function
H(z) = Z-k. This is a low pass filter that has a linear phase characteristic.
All Pass filters find application as phase equalizers. When placed in cascade with a system that
has an undesired phase response, a phase equalizers is designed to compensate for the poor phase
characteristic of the system and therefore to produce an overall linear phase response.
IDEAL FILTER CHARACTERISTIC
1. Ideal filters have a constant gain (usually taken as unity gain) passband characteristic and
zero gain in their stop band.
2. Ideal filters have a linear phase characteristic within their passband.
3. Ideal filters also have constant magnitude characteristic.
4. Ideal filters are physically unrealizable.
4.2 TYPES OF DIGITAL FILTER
Digital filters are of two types. Finite Impulse Response Digital Filter & Infinite Impulse Response
Digital Filter.
DIFFERENCE BETWEEN FIR FILTER AND IIR FILTER
S. No FIR Digital Filter IIR Digital Filter
1 FIR system has finite duration unit sample IIR system has infinite duration unit sample
response. i.e h(n) = 0 for n<0 and n ≥ M response. i. e h(n) = 0 for n<0
Thus the unit sample response exists for the Thus the unit sample response exists for the
duration from 0 to M-1. duration from 0 to ∞.
2 FIR sypstaemdseaer zno.nn eretcursive. Thus IIR systems are recursive. Thus they use
output of
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

FIR filter depends upon present and past inputs. feedback. Thus output of IIR filter depends
upon present and past inputs as well as past
outputs
3 Difference equation of the LSI system for FIR Difference equation of the LSI system for IIR
filters becomes filters becomes
M N M
y(n)=∑ bk x(n–k) y(n)=-∑ ak y(n–k)+∑ bk x(n–k)
k=0 k=1 k=0

4 FIR systems has limited or finite memory IIR system requires infinite memory.
requirements.
5 FIR filters are always stable Stability cannot be always guaranteed.
6 FIR filters can have an exactly linear phase IIR filter is usually more efficient design in
response so that no phase distortion is terms of computation time and memory
introduced in the signal by the filter. requirements. IIR systems usually requires less
processing time and storage as compared with
FIR.
7 The effect of using finite word length to Analogue filters can be easily and readily
implement filter, noise and quantization errors transformed into equivalent IIR digital filter.
are less severe in FIR than in IIR. But same is not possible in FIR because that
have no analogue counterpart.
8 All zero filters Poles as well as zeros are present.
9 FIR filters are generally used if no phase IIR filters are generally used if sharp cutoff and
distortion is desired. high throughput is required.
Example: Example:
System described by System described by
Y(n) = 0.5 x(n) + 0.5 x(n-1) is FIR filter. Y(n) = y(n-1) + x(n) is IIR filter.
h(n)={0.5,0.5} h(n)=an u(n) for n≥0
4. 3 STRUCTURES FOR FIR SYSTEMS
FIR Systems are represented in four different ways
1. Direct Form Structures
2.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

3. Frequency-Sampling Structures
4. Lattice structures.
1. DIRECT FORM STRUCTURE OF FIR SYSTEM
The convolution of h(n) and x(n) for FIR systems can be written as
M-1
y(n)=∑ h(k) x(n–k) (1)
k=0
The above equation can be expanded as,
Y(n)= h(0) x(n) + h(1) x(n-1) + h(2) x(n-2) + …………… + h(M-1) x(n-M+1) (2)
Implementation of direct form structure of FIR filter is based upon the above equation.
x(n) Z-1 x(n-1) Z-1 Z-1 x(n-M+1)

h(0) h(1) h(M-1)

+ +
+ +
h(0)x(n) h(0)x(n)+
h(1)x(n) y(n)
FIG - DIRECT FORM REALIZATION OF FIR SYSTEM
1) There are M-1 unit delay blocks. One unit delay block requires one memory location.
Hencedirect form structure requires M-1 memory location
2) The multiplication of h(k) and x(n-k) is performed for 0 to M-1 terms. Hence M multiplications
and M-1 additions are required.
3) Direct form structure is often called as transversal or tapped delay line filter.
2. CASCADE FORM STRUCTURE OF FIR SYSTEM
In cascade form, stages are cascaded (connected) in series. The output of one system is input to another.
Thus total K number of stages are cascaded. The total system function 'H' is given by
H= H1(z) . H2(z)……………………. Hk(z) (1)
H= Y1(z)/X1(z). Y2(z)/X2(z). ……………Yk(z)/Xk(z) (2)
k
H(z)=π Hk(z) (3)
k=1
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

x(n)=x1(n) y1(n)=x2(n) y2(n)=x3(n) yk(n)=y(n)

H1(z) H2(z) Hk(z)

FIG- CASCADE FORM REALIZATION OF FIR SYSTEM

Each H1(z), H2(z)… etc is a second order section and it is realized by the direct form as shown in below
figure.
System function for FIR systems
M-1
H(z)=∑ bk z-k (1)
k=0
Expanding the above terms we have
H(z)= H1(z) . H2(z)… ............................ Hk(z)
where HK(z) = bk0 + bk1 z-1 + bk2 z-2 (2)
Thus Direct form of second order system is shown as

bk0 bk1 x(n-1) bk2

-1 -1
x(n) Z Z

+ +
y(n) y(n-1)
FIG - DIRECT FORM REALIZATION OF FIR SECOND ORDER SYSTEM
4. 4 STRUCTURES FOR IIR SYSTEMS
IIR Systems are represented in four different ways
1. Direct Form Structures Form I and Form II
2. Cascade Form Structure
3. Parallel Form Structure
4. Lattice and Lattice-Ladder structure.
DIRECT FORM STRUCTURE FOR IIR SYSTEMS
IIR systems can be described by a generalized equations as
N M
y(n)=-∑ ak y(n–k)+∑ bk x(n–k) (1)
k=1 a
k=d0
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Z transform is given as
M N
H(z) = ∑ bk z–k / 1+ ∑ ak z–k (2)
K=0 k=1
M N
Here H1(z) = ∑ bk z–k And H2(z) = 1+ ∑ ak z–k
K=0 k=0
Overall IIR system can be realized as cascade of two function H1(z) and H2(z). Here H1(z) represents
zeros of H(z) and H2(z) represents all poles of H(z).
1. Direct form I realization of H(z) can be obtained by cascading the realization of H1(z)
which is all zero system first and then H2(z) which is all pole system.
2. There are M+N-1 unit delay blocks. One unit delay block requires one memory location.Hence direct
form structure requires M+N-1 memory locations.
3. Direct Form I realization requires M+N+1 number of multiplications and M+N numberof
additions and M+N+1 number of memory locations.
DIRECT FORM – I
X(n) bo y(n)

+ +

Z-1 Z-1

b1 -a1
+ +
Z-1 Z-1

b2 -a2
+ +

bM-1 -aN-1

+ +
-1
Z Z-1
bM -aN

FIG - DIRECT FORM I REALIZATION OF IIR SYSTEM

WW
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

DIRECT FORM - II
1. Direct form realization of H(z) can be obtained by cascading the realization of H1(z) which is all
pole system and H2(z) which is all zero system.
2. Two delay elements of all pole and all zero system can be merged into single delay element.
3. Direct Form II structure has reduced memory requirement compared to Direct form I structure. Hence
it is called canonic form.
4. The direct form II requires same number of multiplications(M+N+1) and additions (M+N) as that of
direct form I.
b0
X(n) + Y(n)
+

-a1
Z-1 b1

+
+
Z-1
-a2 b2

+ +

-aN-1 bN-1

+
+
Z-1
-aN bN

FIG - DIRECT FORM II REALIZATION OF IIR SYSTEM

CASCADE FORM STRUCTURE FOR IIR SYSTEMS
In cascade form, stages are cascaded (connected) in series. The output of one system is input to another.
Thus total K number of stages are cascaded. The total system function 'H' is given by
H= H1(z) . H2(z)……………………. Hk(z) (1)
H= Y1(z)/X1(z). Y2(z)/X2(z). ……………Yk(z)/Xk(z) (2)
k
H(z)=π Hk(z) (3)
k=1
WW
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

x(n)=x1(n) y1(n)=x2(n) y2(n)=x3(n) yk(n)=y(n

H1(z) H2(z) Hk(z)

FIG - CASCADE FORM REALIZATION OF IIR SYSTEM

Each H1(z), H2(z)… etc is a second order section and it is realized by the direct form as shown in below
figure.
System function for IIR systems
M N
H(z) = ∑ bk z–k / 1+ ∑ ak z–k (1)
K=0 k=1
Expanding the above terms we have
H(z)= H1(z) . H2(z)… ............................ Hk(z)
where HK(z) = bk0 + bk1 z-1 + bk2 z-2 / 1 + ak1 z-1 + ak2 z-2 (2)
Thus Direct form of second order IIR system is shown as
X(n) bk0
+ y(n)

Z-1

+
+
-ak1 bk1
Z-1

+
+

-ak2 bk2
+

FIG - DIRECT FORM REALIZATION OF IIR SECOND ORDER SYSTEM (CASCADE)

WW
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

PARALLEL FORM STRUCTURE FOR IIR SYSTEMS

System function for IIR systems is given as
M N
H(z) = ∑ bk z–k / 1+ ∑ ak z–k (1)
K=0 k=1
= b0 + b1 z-1 + b2 z-2 + ……..+ bM z-M / 1 + a1 z-1 + a2 z-2 +……+ aN z-N (2)
The above system function can be expanded in partial fraction as follows
H(z)= C + H1(z) + H2(z)…………………….+ Hk(z) (3)
Where C is constant and Hk(z) is given as
Hk(z) = bk0 + bk1 z-1 / 1 + ak1 z-1 + ak2 z-2 (4)
C

H1(z) +

H2(z) +

X(n)
k1(z) +

FIG - PARALLEL FORM REALIZATION OF IIR SYSTEM

Thus Direct form of second order IIR system is shown as
X(n) bk0 +
+ y(n)

Z-1

+
-ak1
Z-1
-ak2
+
+
FIG - DIRECT FORM REALIZATION OF IIR SECOND ORDER SYSTEM (PARALLEL)

WW
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

IIR FILTER DESIGN

1. IMPULSE INVARIANCE
2. BILINEAR TRANSFORMATION
3. BUTTERWORTH APPROXIMATION
IIR FILTER DESIGN - IMPULSE INVARIANCE METHOD
Impulse Invariance Method is simplest method used for designing IIR
Filters. Important Features of this Method are
1. In impulse variance method, Analog filters are converted into digital filter just by replacing unit
sample response of the digital filter by the sampled version of impulse response of analog filter.
Sampled signal is obtained by putting t=nT hence
h(n) = ha(nT) n=0,1,2. …………
where h(n) is the unit sample response of digital filter and T is sampling interval.
2. But the main disadvantage of this method is that it does not correspond to simple algebraic
mapping of S plane to the Z plane. Thus the mapping from analog frequency to digital frequency is
many to one. The segments
(2k-1)∏/T ≤ Ω ≤ (2k+1) ∏/T of jΩ axis are all mapped on the unit circle ∏≤ω≤∏. This
takes place because of sampling.
3. Frequency aliasing is second disadvantage in this method. Because of frequency aliasing, the
frequency response of the resulting digital filter will not be identical to the original analog frequency
response.
4. Because of these factors, its application is limited to design low frequency filters like LPF or a limited
class of band pass filters.
RELATIONSHIP BETWEEN Z PLANE AND S PLANE
Z is represented as rejω in polar form and relationship between Z plane and S plane is given as Z=e ST
where s= σ + j Ω.
Z= eST (Relationship Between Z plane and S plane)
Z= e (σ + j Ω) T
= e σ T. e jΩ T
Comparing Z value with the polar form we have.
r= e σ T
and ω = Ω T
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

1) If σ = 0 then r=1
2) If σ < 0 then 0 < r < 1
3) If σ > 0 then r> 1
Thus
1) Left side of s-plane is mapped inside the unit circle.
2) Right side of s-plane is mapped outside the unit circle.
3) jΩ axis is in s-plane is mapped on the unit circle.

Im(z)
1 jΩ

Re(z) σ

Im(z)
1 jΩ

Re(z) σ

CONVERSION OF ANALOG FILTER INTO DIGITAL FILTER

 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

n
Ha(s)= Σ Ck / s-pk (1)
k=1
where pk are the poles of the analog filter and ck are the coefficients of partial fraction expansion. The
impulse response of the analog filter ha(t) is obtained by inverse Laplace transform and given as
n
ha(t) = Σ Ck epkt (2)
k=1
The unit sample response of the digital filter is obtained by uniform sampling of ha(t).
h(n) = ha(nT) n=0,1,2. ………….
n
h(n) =Σ Ck epknT (3)
k=1
System function of digital filter H(z) is obtained by Z transform of h(n).
N ∞
H(z) =Σ Ck Σ epkT z-1 n
(4)
k=1 n=0
Using the standard relation and comparing equation (1) and (4) system function of digital filter is given
as
1 1
s - pk 1- epkT z-1
STANDARD RELATIONS IN IIR DESIGN
S. No Analog System Function Digital System function
1
1
1
1- eaT z-1
s–a

2
1- e-aT (cos bT) z-1
s+a
1-2e-aT (cos bT)z-1+ e-2aTz-2
(s+a)2 + b2
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

EXAMPLES - IMPULSE INVARIANCE METHOD

S. No Analog System Function Digital System function
1
1 - (e -0.1Tcos3T)z-1
s + 0.1
2
1-2e-0.1T (cos 3T)z-1+ e-0.2Tz-2
(s+0.1) + 9

2 1
(s+1) (s+2) 0.148 z
(for sampling frequency of 5 z2 - 1.48 z + 0.548
samples/sec)
3 10
(s+2) 10
(for sampling time is 1 - z-1
0.01 sec)

IIR FILTER DESIGN - BILINEAR TRANSFORMATION METHOD (BZT)

The method of filter design by impulse invariance suffers from aliasing. Hence in order to overcome this
drawback Bilinear transformation method is designed. In analogue domain frequency axis is an
infinitely long straight line while sampled data z plane it is unit circle radius. The bilinear transformation
is the method of squashing the infinite straight analog frequency axis so that it becomes finite.
Important Features of Bilinear Transform Method are
1. Bilinear transformation method (BZT) is a mapping from analog S plane to digital Z plane. This
conversion maps analog poles to digital poles and analog zeros to digital zeros. Thus all poles and
zeros are mapped.
2. This transformation is basically based on a numerical integration techniques used to simulate an
integrator of analog filter.
3. There is one to one correspondence between continuous time and discrete time frequency points.
Entire range in Ω is mapped only once into the range -∏≤ω≤∏.
4. Frequency relationship is non-linear. Frequency warping or frequency compression is due to non-
linearity. Frequency warping means amplitude response of digital filter is expanded at the lower
frequencies and compressed at the higher frequencies in comparison of the analog filter.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

5. But the main disadvantage of frequency warping is that it does change the shape of the desired filter
frequency response. In particular, it changes the shape of the transition bands.
CONVERSION OF ANALOG FILTER INTO DIGITAL FILTER
Z is represented as rejω in polar form and relationship between Z plane and S plane in BZT method is
given as
2z-1
S=
T z+1
2 rejω - 1
S=
T rejω + 1
2 r (cos ω + j sin ω) -1
S=
T r (cos ω + j sin ω) +1
2 r2 -1 + 2r j 2 r sin ω
S
T 1+r2+2r cos ω p11+r2+2r cos ω
Comparing the above equation with S= σ + j Ω. We have

σ= 2 r2 -1
T 1+ r2+2r cos ω

Ω= 2 2 r sin ω
T 1+ r2+2r cos ω
Here we have three condition
1) If σ < 0 then 0 < r < 1
2) If σ > 0 then r > 1
3) If σ = 0 then r=1
When r =1

Ω=
2 sin ω
T 1+cos ω
Ω=
= (2/T) tan (ω/2)
ω=
2 tan -1 (ΩT/2)
The above equations shows that in BZT frequency relationship is non-linear. The frequency relationship
is plotted as
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Ω 2 tan -1 (ΩT/2)

ΩT

FIG - MAPPING BETWEEN FREQUENCY VARIABLE 𝝎 AND Ω IN BZT METHOD.

DIFFERENCE - IMPULSE INVARIANCE Vs BILINEAR TRANSFORMATION
S. No Impulse Invariance Bilinear Transformation
1 In this method IIR filters are designed having a This method of IIR filters design is based on the
unit sample response h(n) that is sampled trapezoidal formula for numerical integration.
version of the impulse response of the analog
filter.
2 In this method small value of T is selected to The bilinear transformation is a conformal mapping
minimize the effect of aliasing. that transforms the j Ω axis into the unit circle in
the z plane only once, thus avoiding aliasing of
frequency components.
3 They are generally used for low frequencies like For designing of LPF, HPF and almost all types of
design of IIR LPF and a limited class of Band pass and band stop filters this method is used.
bandpass filter
4 Frequency relationship is linear. Frequency relationship is non-linear. Frequency
warping or frequency compression is due to non-
linearity.
5 All poles are mapped from the s plane to the z All poles and zeros are mapped.
plane by the relationship
Zk= epk.T. Buttheze.rosin two domain does not
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

satisfy the same relationship.

LPF AND HPF ANALOG BUTTERWORTH FILTER TRANSFER FUNCTION

Sr Order of the Low Pass Filter High Pass Filter
No Filter
1 1 1 / s+1 s / s+1
2 2 1 / s2+ √2 s + 1 s2 / s2+ √2 s + 1

3 3 1 / s3 + 2 s2 + 2s +1 s3 / s3 + 2 s2 + 2s +1
METHOD FOR DESIGNING DIGITAL FILTERS USING BZT
step 1. Find out the value of ωc*.

ωc* = (2/T) tan (ωc Ts/2)

step 2. Find out the value of frequency scaled analog transfer function
Normalized analog transfer function is frequency scaled by replacing s by s/ωp*.
step 3. Convert into digital filter
Apply BZT. i.e Replace s by the ((z-1)/(z+1)). And find out the desired transfer function of digital
function.
Example:
Q) Design first order high pass butterworth filter whose cutoff frequency is 1 kHz at sampling frequency
of 104 sps. Use BZT Method
Step 1. To find out the cutoff frequency
ωc = 2∏f
= 2000 rad/sec
Step 2. To find the prewarp frequency
ωc* = tan (ωc Ts/2)
= tan(∏/10)
Step 3. Scaling of the transfer function
For First order HPF transfer function H(s) = s/(s+1)
Scaled transfer function H *(s) = H(s) |s=s/ωc*
H*(s)= s/(s + 0.325)
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Step 4. Find out the digital filter transfer function. Replace s by (z-1)/(z+1)

H(z)= z-1
1.325z -0.675

Tutorial problems:
Q) Design second order low pass butterworth filter whose cutoff frequency is 1 kHz at sampling
frequency of 104 sps.
Q) First order low pass butterworth filter whose bandwidth is known to be 1 rad/sec . Use BZT method
to design digital filter of 20 Hz bandwidth at sampling frequency 60 sps.
Q) Second order low pass butterworth filter whose bandwidth is known to be 1 rad/sec . Use BZT
method to obtain transfer function H(z) of digital filter of 3 DB cutoff frequency of 150 Hz and
sampling frequency 1.28 kHz.
Q) The transfer function is given as s2+1 / s2+s+1 The function is for Notch filter with frequency 1
rad/sec. Design digital Notch filter with the following specification
(1) Notch Frequency= 60 Hz
(2) Sampling frequency = 960 sps.

BUTTERWORTH FILTER APPROXIMATION

The filter passes all frequencies below Ωc. This is called passband of the filter. Also the filter
blocks all the frequencies above Ωc. This is called stopband of the filter. Ωc is called cutoff frequency or
critical frequency.
No Practical filters can provide the ideal characteristic. Hence approximation of the ideal
characteristic are used. Such approximations are standard and used for filter design. Such three
approximations are regularly used.
a) Butterworth Filter Approximation
b) Chebyshev Filter Approximation
c) Elliptic Filter Approximation
Butterworth filters are defined by the property that the magnitude response is maximally flat in
the passband.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

|H(𝜔)|2

fC f
1

|Ha(Ω)|2=
1 + (Ω/Ωc)2N

The squared magnitude function for an analog butterworth filter is of the form.
1
|Ha(Ω)|2=
1 + (Ω/Ωc)2N
N indicates order of the filter and Ωc is the cutoff frequency (-3DB frequency).
At s = j Ω magnitude of H(s) and H(-s) is same hence

Ha(s) Ha(-s) = 1
1 + (-s2/Ωc2)N
To find poles of H(s). H(-s) , find the roots of denominator in above equation.
-s2 = (-1)1/N
Ωc2
As e j(2k+1) ∏ = -1 where k = 0,1,2,… ...... N-1.
-s2 = (e j(2k+1) ∏ 1/N
)
Ωc2
s2 = (-1) Ωc2 e j(2k+1) ∏ / N
Taking the square root we get poles of s.
j(2k+1) ∏ / N 1/2
pk = + √-1 Ωc [ e ]
Pk = + j Ωc e j(2k+1) ∏ / 2N

As e j∏/2 = j
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Pk = + Ωc e j(N+2k+1) ∏ / 2N (1)
This equation gives the pole position of H(s) and H(-s).
FREQUENCY RESPONSE CHARACTERISTIC
The frequency response characteristic of |Ha(Ω)|2 is as shown. As the order of the filter N increases, the
butterworth filter characteristic is more close to the ideal characteristic. Thus at higher orders like N=16
the butterworth filter characteristic closely approximate ideal filter characteristic. Thus an infinite order
filter (N  ∞) is required to get ideal characteristic.
|Ha(Ω)|2 N=18

N=6

N=2

|Ha(Ω)|

Ap

0.5

Ω
Ωp Ωc Ωs
Ap= attenuation in passband.
As= attenuation in stopband.
Ωp = passband edge frequency
Ωs = stopbandedge frequency
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Specification for the filter is

|Ha(Ω)| ≥ Ap for Ω ≤ Ωp and |Ha(Ω)| ≤ As for Ω ≥ Ωs. Hence we have
1 ≥ Ap2
1 + (Ωp/Ωc)2N
1 ≤ As2
1 + (Ωs/Ωc)2N
To determine the poles and order of analog filter consider equalities.
(Ωp/Ωc)2N = (1/Ap2) - 1

(Ωs/Ωc)2N = (1/As2) – 1

Ωs 2N = (1/As2)-1

Ωp (1/Ap2)-1
Hence order of the filter (N) is calculated as

(1/As2)-1
log
(1/Ap2)-1
(2)
N= 0.5
log (Ωs/ Ωp)
log((1/As2) -1) (2A)
N= 0.5
log (Ωs/ Ωc)
And cutoff frequency Ωc is calculated as

Ωc = Ωp (3)
[(1/Ap2) -1]1/2N
If As and Ap values are given in DB then
As (DB) = - 20 log As
log As = -As /20
As = 10 -As/20
(As)-2 = 10 As/10
(As)-2 = 10 0.1 As DB
Hence equation (2) is modified as
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

100.1 As -1
log
100.1 Ap -1
(4)
N= 0.5
log (Ωs/ Ωp)
Q) Design a digital filter using a butterworth approximation by using impulse invariance.
Example
|Ha(Ω)|

0.89125

0.1778
0.2∏ 0.3∏ Ω

Filter Type - Low Pass Filter

Ap - 0.89125
As - 0.17783
Ωp - 0.2∏
Ωs - 0.3∏
Step 1) To convert specification to equivalent analog filter.
(In impulse invariance method frequency relationship is given as ω= Ω T while in Bilinear
transformation method frequency relationship is given as Ω= (2/T) tan (ω/2) If Ts is not specified
consider as 1)
|Ha(Ω)| ≥ 0.89125 for Ω ≤ 0.2∏/T and |Ha(Ω)| ≤ 0.17783 for Ω ≥ 0.3∏/T.
Step 2) To determine the order of the filter.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

(1/As2)-1
log
(1/Ap2)-1
N= 0.5
log (Ωs/ Ωp)
N= 5.88
A) Order of the filter should be integer.
B) Always go to nearest highest integer vale of N.
Hence N=6
Step 3) To find out the cutoff frequency (-3DB frequency)

Ωc = Ωp
[(1/Ap2) -1]1/2N
cutoff frequency Ωc = 0.7032
Step 4) To find out the poles of analog filter system function.
Pk = + Ωc e j(N+2k+1) ∏ / 2N
As N=6 the value of k = 0,1,2,3,4,5.
K Poles

-0.182 + j 0.679
0 P0= + 0.7032 e j7∏/12
0.182 - j 0.679

-0.497 + j 0.497
1 P1= + 0.7032 e j9∏/12
0.497 - j 0.497

-0.679 + j 0.182
2 P2= + 0.7032 e j11∏/12
0.679 - j 0.182

3 P3= + 0.7032 e j13∏/12 -0.679 - j 0.182

0.679 + j 0.182
4
-0.497 - j 0.497
P4= + 0.7032 e j15∏/12
0.497 + j 0.497
www.BrainKart.com
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

5
-0.182 - j 0.679
P5= + 0.7032 e j17∏/12
0.182 + j 0.679

For stable filter all poles lying on the left side of s plane is selected. Hence
S1 = -0.182 + j 0.679 S1* = -0.182 - j 0.679
S2 = -0.497 + j 0.497 S2* = -0.497 - j 0.497
S3 = -0.679 + j 0.182 S3* = -0.679 - j 0.182
Step 5) To determine the system function (Analog Filter)

Ha(s) = Ωc6
(s-s1)(s-s1*) (s-s2)(s-s2*) (s-s3)(s-s3*)
Hence

Ha(s) = (0.7032)6
(s+0.182-j0.679)(s+0.182+j0.679) (s+0.497-j0.497)
(s+0.497+j0.497) (s+0.679-j0.182)(s+0.679-j0.182)

Ha(s) = 0.1209
[(s+0.182)2 +(0.679)2] [(s+0.497)2+(0.497)2] [(s+0.679)2-(0.182)2]

Ha(s) = 1.97 × 0.679 × 0.497 × 0.182

[(s+0.182)2 +(0.679)2] [(s+0.497)2+(0.497)2] [(s+0.679)2-(0.182)2]
Step 6) To determine the system function (Digital Filter)
(In Bilinear transformation replace s by the term ((z-1)/(z+1)) and find out the transfer function of
digital function)

0.5235 z-1 0.29 z-1 0.09 z-1

H(z)=1.97 × ×× ×
1-1.297z-1+0.695z-2 1-1.07z-1+0.37z-2 1-0.99z-1+0.26z-2

Step 7) Represent system function in cascade form or parallel form if asked.

Tutorial problems:
Q) Given for low pass butterworth filter
Ap= -1 db at 0.2∏
As= -15 w atw0..3p∏BrainKart.com
dbw
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

1) Calculate N and Pole location

2) Design digital filter using BZT method.
Q) Obtain transfer function of a lowpass digital filter meeting specifications
Cutoff 0-60Hz
Stopband > 85Hz
Stopband attenuation > 15 db
Sampling frequency= 256 Hz . use butterworth characteristic.
Q) Design second order low pass butterworth filter whose cutoff frequency is 1 kHz at sampling
frequency of 104 sps. Use BZT and Butterworth approximation.

FREQUENCY TRANSFORMATION
When the cutoff frequency Ωc of the low pass filter is equal to 1 then it is called normalized filter.
Frequency transformation techniques are used to generate High pass filter, Bandpass and bandstop filter
from the lowpass filter system function.
FREQUENCY TRANSFORMATION (ANALOG FILTER)
S. No Type of transformation Transformation ( Replace s by)

s
1 Low Pass
ωlp
ωlp - Password edge frequency of another LPF

ωhp
2 High Pass
s
ωhp = Password edge frequency of HPF

(s2 + ωl ωh )
s (ωh - ωl )
3 Band Pass
ωh - higher band edge frequency
ωl - Lower band edge frequency

4 wBand Stop
s (ωh - ωl)
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

s2+ ωh ωl
ωh - higher band edge frequency
ωl - Lower band edge frequency

FREQUENCY TRANSFORMATION (DIGITAL FILTER)

S. No Type of transformation Transformation ( Replace z-1 by)

z-1 - a
1 Low Pass
1 - az-1

- (z-1+ a)
2 High Pass
1 + az-1

- (z-2 - a1z-1 + a2)

3 Band Pass
a2z-2 - a1z-1 + 1

z-2 - a1z-1 + a2
4 Band Stop
a2z-2 - a1z-1 + 1

Example:
Q) Design high pass butterworth filter whose cutoff frequency is 30 Hz at sampling frequency of 150
Hz. Use BZT and Frequency transformation.
Step 1. To find the prewarp cutoff frequency
ωc* = tan (ωcTs/2)
= 0.7265
Step 2. LPF to HPFtransformation
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

For First order LPF transfer function H(s) = 1/(s+1)

Scaled transfer function H*(s) = H(s) |s=ωc*/s

H*(s)= s/(s + 0.7265)

Step 4. Find out the digital filter transfer function. Replace s by (z-1)/(z+1)

H(z)= z-1
1.7265z - 0.2735

Tutorial problems:
Q) Design second order band pass butterworth filter whose passband of 200 Hz and 300 Hz and
sampling frequency is 2000 Hz. Use BZT and Frequency transformation.
Q) Design second order band pass butterworth filter which meet following specification
Lower cutoff frequency = 210 Hz
Upper cutoff frequency = 330 Hz
Sampling Frequency = 960 sps
Use BZT and Frequency transformation.

FIR FILTER DESIGN

Features of FIR Filter
1.FIR filter always provides linear phase response. This specifies that the signals in the pass band
will suffer no dispersion Hence when the user wants no phase distortion, then FIR filters are preferable
over IIR. Phase distortion always degrade the system performance. In various applications like
speech processing, data transmission over long distance FIR filters are more preferable due to this
characteristic.
2 FIR filters are most stable as compared with IIR filters due to its non feedback nature.
3. Quantization Noise can be made negligible in FIR filters. Due to this sharp cutoff FIR filters can be
easily designed.
4. Disadvantage of FIR filters is that they need higher ordered for similar magnitude response of IIR
filters.
FIR SYSTEM ARE ALWAYS STABLE. Why?
Proof: Difference equation of FIR filter of length M is given as

M-1
y(n)=∑ bk x(n–k) (1)
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

k=0
And the coefficient bk are related to unit sample response as
H(n) = bn for 0 ≤ n ≤ M-1
= 0 otherwise.
We can expand this equation as
Y(n)= b0 x(n) + b1 x(n-1) + …….. + bM-1 x(n-M+1) (2)
System is stable only if system produces bounded output for every bounded input. This is stability
definition for any system.
Here h(n)={b0, b1, b2, } of the FIR filter are stable. Thus y(n) is bounded if input x(n) is bounded. This
means FIR system produces bounded output for every bounded input. Hence FIR systems are always
stable.
Symmetric and Anti-symmetric FIR filters
1. Unit sample response of FIR filters is symmetric if it satisfies following condition
h(n)= h(M-1-n) n=0,1,2… ................M-1
2. Unit sample response of FIR filters is Anti-symmetric if it satisfies following condition
h(n)= -h(M-1-n) n=0,1,2…………….M-1
FIR Filter Design Methods
The various method used for FIR Filer design are as follows
1. Fourier Series method
2. Windowing Method
3. DFT method
4. Frequency sampling Method. (IFT Method)
GIBBS PHENOMENON
Consider the ideal LPF frequency response as shown in Fig 1 with a normalizing angular cut off
frequency Ωc.

Impulse response of an ideal LPF is as shown in Fig 2.

 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

1. In Fourier series method, limits of summation index is -∞ to ∞. But filter must have finite terms.
Hence limit of summation index change to -Q to Q where Q is some finite integer. But this type of
truncation may result in poor convergence of the series. Abrupt truncation of infinite series is equivalent
to multiplying infinite series with rectangular sequence. i.e at the point of discontinuity some oscillation
may be observed in resultant series.
2. Consider the example of LPF having desired frequency response H d (ω) as shown in figure. The
oscillations or ringing takes place near band-edge of the filter.
3. This oscillation or ringing is generated because of side lobes in the frequency response W(ω) of the
window function. This oscillatory behavior is called "Gibbs Phenomenon".
Truncated response and ringing effect is as shown in fig 3.

WINDOWING TECHNIQUE
Windowing is the quickest method for designing an FIR filter. A windowing function simply
truncates the ideal impulse response to obtain a causal FIR approximation that is non causal and
infinitely long. Smoother window functions provide higher out-of band rejection in the filter response.
However this smoothness comes at the cost of wider stopband transitions.
Various windowing method attempts to minimize the width of the main lobe (peak) of the
frequency response. In addition, it attempts to minimize the side lobes (ripple) of the frequency
responsew
.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Rectangular Window: Rectangular This is the most basic of windowing methods. It does not require
any operations because its values are either 1 or 0. It creates an abrupt discontinuity that results in sharp
roll-offs but large ripples.

Rectangular window is defined by the following equation.

=1 for 0 ≤ n ≤ N
W[n]
=0 otherwise
Triangular Window: The computational simplicity of this window, a simple convolution of two
rectangle windows, and the lower sidelobes make it a viable alternative to the rectangular window.

Kaiser Window: This windowing method is designed to generate a sharp central peak. It has reduced
side lobes and transition band is also narrow. Thus commonly used in FIR filter design.

Hamming Window: This windowing method generates a moderately sharp central peak. Its ability to
generate a maximally flat response makes it convenient for speech processing filtering.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Hanning Window: This windowing method generates a maximum flat filter design.

DESIGNING FILTER DESIGN FROM POLE ZERO PLACEMENT

Filters can be designed from its pole zero plot. Following two constraints should be imposed while
designing the filters.
1. All poles should be placed inside the unit circle on order for the filter to be stable. However zeros can
be placed anywhere in the z plane. FIR filters are all zero filters hence they are always stable. IIR filters
are stable only when all poles of the filter are inside unit circle.
2. All complex poles and zeros occur in complex conjugate pairs in order for the filter coefficients to be
real.
In the design of low pass filters, the poles should be placed near the unit circle at points corresponding to
low frequencies ( near ω=0)and zeros should be placed near or on unit circle at points corresponding to
high frequencies (near ω=∏). The opposite is true for high pass filters.
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

NOTCH AND COMB FILTERS

A notch filter is a filter that contains one or more deep notches or ideally perfect nulls in its frequency
response characteristic. Notch filters are useful in many applications where specific frequency
components must be eliminated. Example Instrumentation and recording systems required that the
power-line frequency 60Hz and its harmonics be eliminated.
To create nulls in the frequency response of a filter at a frequency ω0, simply introduce a pair of
complex-conjugate zeros on the unit circle at an angle ω0.
comb filters are similar to notch filters in which the nulls occur periodically across the frequency band
similar with periodically spaced teeth. Frequency response characteristic of notch filter |H(ω)| is as
shown in figure.

ωo ω1 ω
DIGITAL RESONATOR
A digital resonator is a special two pole bandpass filter with a pair of complex conjugate poles
located near the unit circle. The name resonator refers to the fact that the filter has a larger magnitude
response in the vicinity of the pole locations. Digital resonators are useful in many applications,
including simple bandpass filtering and speech generations.
IDEAL FILTERS ARE NOT PHYSICALLY REALIZABLE. Why?
Ideal filters are not physically realizable because Ideal filters are anti-causal and as only causal systems
are physically realizable.
Proof:
Let take example of ideal lowpass filter.
H(ω) = 1 for - ωc ≤ ω ≤ ωc
= 0 elsewhere
The unit sample response of this ideal LPF can be obtained by taking IFT of H(ω).

h(n)=. 𝐻(𝑗𝑤
∞
−∞
)𝑒 𝑗𝑤𝑛𝑑𝑤 (1)
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

h(n)= 𝑤𝑐 1𝑒𝑗𝑤𝑛 𝑑𝑤 (2)

−𝑤𝑐
ωc
1_ ejωn
h(n) =
2∏ jn
-ωc
1
[ejωcn - e-jωcn ]
2∏jn

Thus h(n)= sin ωcn / ∏n for n≠0

Putting n=0 in equation (2) we have
h(n) = 𝑤𝑐
−𝑤𝑐
1 dω (3)

1_
[ω]ωc
2∏ -ωc

and h(n) = ωc / ∏ for n=0

sin (ωcn )
h(n) = for n≠0
∏n

𝑤𝑐
for n=0
𝑛

Hence impulse response of an ideal LPF is as shown in Fig

LSI system is causal if its unit sample response satisfies following condition.
h(n) = 0 for n<0
In above figure h(n) extends -∞ to ∞. Hence h(n) ≠0 for n<0. This means causality condition is not
satisfied by the ideal low pass filter. Hence ideal low pass filter is non causal and it is not physically
realizable.
EXAMPLESOF SIM
PLED
IG
ITAL FILTERS:
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The following examples illustrate the essential features of digital filters.

1.UNITY GAIN FILTER: yn = xn
Each output value yn is exactly the same as the corresponding input value xn:
2.SIMPLE GAIN FILTER: yn = Kxn (K = constant) Amplifier or attenuator)
This simply applies a gain factor K to each input value:
3. PURE DELAY FILTER: yn = xn-1
The output value at time t = nh is simply the input at time t = (n-1)h, i.e. the signal is delayed by time h:
4.TWO-TERM DIFFERENCE FILTER: yn = xn - xn-1
The output value at t = nh is equal to the difference between the current input xn and the previous
inputxn-1:
5.TWO-TERM AVERAGE FILTER: yn = (xn + xn-1) / 2
The output is the average (arithmetic mean) of the current and previous input:
6.THREE-TERM AVERAGE FILTER: yn = (xn + xn-1 + xn-2) / 3
This is similar to the previous example, with the average being taken of the current and two
previousinputs.
7.CENTRAL DIFFERENCE FILTER: yn = (xn - xn-2) / 2
This is similar in its effect to example (4). The output is equal to half the change in the input signal over
the previous two sampling intervals.
ORDER OF A DIGITAL FILTER
The order of a digital filter can be defined as the number of previous inputs (stored in the processor's
memory) used to calculate the current output.
This is illustrated by the filters given as examples in the previous section.
Example (1): yn = xn
This is a zero order filter, since the current output yn depends only on the current inputxn and not on any
previous inputs.
Example (2): yn = Kxn
The order of this filter is again zero, since no previous outputs are required to give the current output
value.
Example (3): yn = xn-1
This is a first order filter, as one previous input (xn-1) is required to calculate yn. (Note that this filter is
classed as first-order because it uses one previous input, even though the current input is not used).
Examplw
e
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

This is again a first order filter, since one previous input value is required to give the current output.
Example (5): yn = (xn + xn-1) / 2
The order of this filter is again equal to 1 since it uses just one previous input value.
Example (6): yn = (xn + xn-1 + xn-2) / 3
To compute the current output yn, two previous inputs (xn-1 and xn-2) are needed; this is therefore a
second-order filter.
Example (7): yn = (xn - xn-2) / 2
The filter order is again 2, since the processor must store two previous inputs in order to compute the
current output. This is unaffected by the absence of an explicitxn-1 term in the filter expression.
Q) For each of the following filters, state the order of the filter and identify the values of its coefficients:
(a) yn = 2xn - xn-1 A) Order = 1: a0 = 2, a1 = -1
(b) yn = xn-2 B) Order = 2: a0 = 0, a1 = 0, a2 = 1
(c) yn = xn - 2xn-1 + 2xn-2 + xn-3 C) Order = 3: a0 = 1, a1 = -2, a2 = 2, a3 = 1
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

CHAPTER 5

DIGITAL SIGNAL PROCESSOR

5.1 REQUIREMENTS OF DSP PROCESSORS

The most fundamental mathematical operation in DSP is sum of products also called as dot of products.
Y(n)= h(0)*x(n) + h(1)*x(n-1) +………+ h(N-1)*x(n-N)
This operation is mostly used in digital filter designing, DFT, FFT and many other DSP applications. A
DSP is optimized to perform repetitive mathematical operations such as the dot product. There are five
basic requirements of s DSP processor to optimize the performance They are
1) Fast arithmetic
2) Extended precision
3) Fast Execution - Dual operand fetch
4) Fast data exchange
5) Fast operations
6) Circular buffering
S. No Requirements Features of DSP processor
1 Fast Arithmetic Faster MACs means higher bandwidth.
Able to support general purpose math functions, should have ALU
and a programmable shifter function for bit manipulation.
Powerful interrupt structure and timers
2 Extended precision A requirement for extended precision in the accumulator register.
High degree of overflow protection.
3 Fast Execution Parallel Execution is required in place of sequential.
Instructions are executed in single cycle of clock called as True
instruction cycle as oppose to multiple clock cycle.
Multiple operands are fetched simultaneously. Multi-processing
Ability and queue, pipelining facility
Address generation by DAG's and program sequencer.
4 Fast data Exchange Multiple registers, Separate program and data memory and Multiple
operands fetch capacity
5 Fast Operations Program and data memories are on chip memory and extendable off-
chip
No multiplexing is used. Separate address, data and control bus.
Powerful instruction set and various addressing modes.
6 Circular shift operations Circular Buffers

5.2 MICROP ROCESSORAR CHITECTURES

 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

There are mainly three types of microprocessor architectures present.

1. Von-Neumann architecture
2. Harvard architecture
3. Analog devices Modified Harvard architecture.
1) Von-Neumann Architecture
General purpose microprocessors uses the Von-Neumann Architectures. (named after the American
mathematician John Von Neumann)
1. It consists of ALU, accumulator, IO devices and common address and data bus. It also consists of a
single memory which contains data and instructions, a single bus for transferring data and instructions
into and out of the CPU.
2. Multiplying two numbers requires at leased three cycles, two cycles are required to transfer the two
numbers into the CPU and one cycle to transfer the instruction.

3. This architecture is giving good performance when all the required tasks can be executed serially.
4.For large processing applications like DSP applications Von-Neumann architecture is not suitable as
processing speed is less. Processing speed can be increased by pipelining up to certain extend which is
not sufficient for DSP applications. In order to perform a single FIR filter multiply-accumulate, an
instruction is fetched from the program memory, and during the same cycle, a coefficient can be fetched
from the data memory. A second cycle is required to fetch the data word from data memory

MEMORY Address Bus

A:
VON NEUMANN Data and CPU
Instruction Data Bus

HARVARD
B: PROGRAM PM Address Bus DM Address Bus DATA
MEMORY:
CPU MEMORY:
Instruction
Only PM Data Bus DM Data Bus Data Only

ADI MODIFIED
C: PROGRAM PM Address Bus CPU DM Address Bus DATA
MEMORY: MEMORY:
HARVARD Instructions Data Only
and Data PM Data Bus Instructio
Cache
n DM Data Bus
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Figure: MICROPROCESSOR ARCHITECTURES
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

2) Harvard Architecture (named for the work done at Harvard university)

1. Data and program instructions each have separate memories and buses as shown.
Program memory address and data buses for program memory and data memory
address and data buses for data memory.

2. Since the buses operate independently, program instructions and data can be
fetched at the same time. Therefore improving speed over the single bus Von
Neumann design.

3. In order to perform a single FIR filter multiply-accumulate, an instruction is

fetched from the program memory, and during the same cycle, a coefficient can be
fetched from the data memory. A second cycle is required to fetch the data word
from data memory.

CORE ARCHITECTURE OF ADSP-21xx

ADSP-21xx family DSP's are used in high speed numeric processing applications.

ADSP-21xx architecture consists of Five Internal Buses

Program Memory Address(PMA)

Data memory address (DMA)

Program memory data(PMD)

Data memory data (DMD) and Result (R)

PROCESSOR CORE

Address Bus 1

MEMORY A
Data Bus 1
Address Bus 2

Data Bus 2

MEMORY A
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MEMORY B
SCE 100
 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

 Three Computational Units are

Arithmetic logic unit (ALU)
Multiply-accumulate (MAC)
Shifter
 Two Data Address generators (DAG)
 Program sequencer
 On chip peripheral Options
 Data Memory
 Timer
 Serial Port

Instruction
Data Register
Address Cache Program
Generator 1 Generator 2 Memory Sequencer

PMA BUS

DMA BUS

PMD BUS

Bus
Exchange
DMD BUS

Input Regs Input Regs I*put Regs

ALU MAC Shifter

Output Reg s Output Regs Output Regs

R BUS

BUSES: Block Diagram of Architecture

The ADSP-21xx processors have five internal buses to ensure data transfer.

1. PMA and DMA buses are used internally for addresses associated with Program
and data memory. The PMD and DMD are used for data associated with memory
spaces. Off Chip, the buses are multiplexed into a single external address bus and a
single external data bus. The address spaces are selected by the appropriate
control signal.

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 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

2. The result (R) bus transfers the intermediate results directly between various
computational units.
3. PMA bus is 14-bits wide allowing direct access of up to 16k words of code and
data. PMD bus is 24 bits wide to accommodate the 24 bit instruction width.
The DMA bus 14 bits wide allowing direct access of up to 16k words of data.The
DMD bus is 16 bit wide.
4. The DMD bus provides a path for the contents of any register in the processor to
be transferred to any other register or to any external data memory location in a
single cycle. DMA address comes from two sources. An absolute value specified in
the instruction code (direct addressing) or the output of DAG (Indirect addressing).
The PMD bus can also be used to transfer data to and from the
computational units thro direct path or via PMD-DMD bus exchange unit.
COMPUTATIONAL UNITS:
The processor contains three -independent computational units. ALU, MAC
(Multiplier-accumulator) and the barrel shifter. The computational units process 16-
bit data directly. ALU is 16 bits wide with two 16 bit input ports and one output
port. The ALU provides a standard set of arithmetic and logic functions.
ALU Features
1. Add, subtract, Negate, increment, decrement, Absolute value, AND, OR, EX-OR,
Not etc.
2. Bitwise operators, Constant operators
3. Multi-precision Math Capability
4. Divide Primitives and overflow support.
MAC:
The MAC performs high speed single-cycle multiply/add and multiply/subtract
operations. MAC has two 16 bit input ports and one 32 bit product output port. 32
bit product is passed to a 40 bit adder/subtractor which adds or subtracts the new
product from the content of the multiplier result (MR). It also contains a 40 bit
accumulator which provides 8 bit overflow in successive additions to ensure that no
loss of data occurs. 256 overflows would have to occur before any data is lost. A set
of background registers is also available in the MAC for interrupts service routine.

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SHIFTER:
The shifter performs a complete set of shifting functions like logical and arithmetic
shifts (circular or linear shift) , normalization (fixed point to floating point
conversion), demoralization (floating point to fixed point conversion) etc
ALU, MAC and shifter are connected to DMD bus on one side and to R bus on
other side. All three sections contains input and output registers which are
accessible from the internal DMD bus. Computational operations generally take the
operands from input registers and load the result into an output register.
DATA ADDRESS GENERATORS (DAG):
Two DAG's and a powerful program sequencer ensure efficient use of these
computational units. The two DAG's provides memory addresses when memory
data is transferred to or from the input or output registers. Each DAG keeps track of
up to four address pointers. Hence jumps, branching types of instructions are
implemented within one cycle. With two independent DAG's, the processor can
generate two address simultaneously for dual operand fetches.
DAG1 can supply addresses to data memory only. DAG2 can supply addresses to
either data memory or program memory. When the appropriate mode bit is set in
mode status register (MSTAT), the output address of DAG1 is bit-reversed before
being driven onto the address bus. This feature facilitates addressing in radix-2 FFT
algorithm.
PROGRAM SEQUENCER:
The program sequencer exchanges data with DMD bus. It can also take from
PMD bus. It supplies instruction address to program memory. The sequencer is
driven by the instruction register which holds the currently executing instruction.
The instruction register introduces a single level of pipelining into the program
flow. Instructions are fetched and loaded into the instruction register during one
processor cycle, and executed during the following cycle while the next instruction
is pre-fetched. The cache memory stores up to 16 previously executed instructions.
Thus data memory on PMD bus is more efficient because of cache memory. This
also makes pipelining and increase the speed of operations.

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 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

FEATURES OF ADSP-21xx PROCESSOR

1. 16 bit fixed DSP microprocessor
2. Enhanced Harvard architecture for three bus performance.
3. Separate on chip buses for program and data memory.
4. 25 MIPS, 40 ns maximum instruction set 25Mhz frequency.
5. Single cycle instruction execution i.e True instruction cycle.
6. Independent computational units ALU, MAC and shifter.
7. On chip program and data memories which can be extended off chip.
8. Dual purpose program memory for instruction and data.
9. Single cycle direct access to 16K × 16 of data memory.
10. Single cycle direct access to 16K × 24 of program m
ADSP-21xx DEVELOPMENT TOOLS
Various development tools such as assembler , linker, debugger, simulator are
available for ADSP-21xx family.
SYSTEM BUILDER
The system builder is the software development tool for describing the
configuration of the target system's memory and I/O. The ranges for program
memory(PM) and data memory(DM) are described.
The program memory space is allotted with 16 K of memory for instructions and
16K for data. The data memory space is also divide into two blocks. The lowest 2K
is the dual -port memory that is shared by both processors. Each processor has an
additional 14K of private data memory.
.SYSTEM
.ADSP2108
{PROGRAM MEMORY SECTION}
.SEG/ROM/ABS=0/PM/CODE CODE_AREA
.SEG/RAM/ABS=0X4000/PM/DATA DATA_AREA
{DATA MEMORY SECTION
.SEG/RAM/ABS=0/DM/DATA COMMON
MEMORY

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 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

.SEG/RAM/ABS=0X800/DM/DATA PRIVATE
MEMORY
.PORT/RAM/ABS=0x100 DATA MEMORY IO
PORT
.PORT/RAM/ABS=0x200 AD PORT
.PORT/RAM/ABS=0x300 DA PORT
.ENDSYS
The sample system configuration file (.SYS) is shown. .SEG directive is used to
declare the system's physical memory segments and its characteristics. .PORT
directive declares the memory mapped IO port. .ENDSYS indicates the end of .SYS
file.
ASSEMBLER
The assembler translated source code into object code modules. The source code is
written in assembly language file (.DSP) Assembler reads .DSP file and generates
four
output filed with the same root name. Object file(.OBJ), Code File(.CDE),
Initialization File (.INT), List File(.LST) etc. The file can be assembled by the
following command.
ASM21FILTER
S.No File Extension Application
1 .OBJ Binary codes for instruction, Memory allocation
and symbol declaration, Addresses of the
instructions
2 .CDE Instruction opcodes
3 .INT Initialization information
4 .LST Documentation

LINKER
The linker is a program used to join together object files into one large
object file. The linker produces a link file which contains the binary codes for all
the combined modules. The linker uses .INT, .CDE, .OBJ and .ACH file and

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 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

generates three files. Map listing file (.MAP), Memory Image File (.EXE) and
Symbol table (.SYM).
LD21 FILTER -a SAMPLE -e FILTER
Here Filter is the input file name, sample.ACH file is system architecture file name
and FILTER.EXE is output file name.
DEBUGGER
A debugger is a program which allows user to load object code program into system
memory, execute the program and debug it. The debugger allows the user to look at
the contents of registers and memory locations etc. We can change the contents of
registers and memory locations at run time, generates breakpoints etc.
SIMULATOR
The multiprogramming environment can be tested using the simulator. When
simulated, the filter program produces output data and stores in common data
memory. Simulator command is used to dump form data memory to store the dual
port data memory image on disk which can be reloaded and tested.
APPLICATIONS OF DSP
1. SPEECH RECOGNITION
Basic block diagram of a speech recognition system is shown in Fig 1
1. In speech recognition system using microphone one can input speech or voice.
The analog speech signal is converted to digital speech signal by speech digitizer.
Such digital signal is called digitized speech.
2. The digitized speech is processed by DSP system. The significant features of
speech such as its formats, energy, linear prediction coefficients are extracted.The
template of this extracted features are compared with the standard reference
templates.The closed matched template is considered as the recognized word.
3. Voice operated consumer products like TV, VCR, Radio, lights, fans and voice
operated telephone dialing are examples of DSP based speech recognized
devices.

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 DISCRETE TIME SYSTEMS & SIGNAL PROCESSING

Impulse Voiced
Synthetic
Train speech
Generator Time
× varying
Random digital filter
number
generator Unvoiced

2. LINEAR PREDICTION OF SPEECH SYNTHESIS

Fig shows block diagram of speech synthesizer using linear prediction
1. For voiced sound, pulse generator is selected as signal source while for unvoiced
sounds noise generator is selected as signal source.
2. The linear prediction coefficients are used as coefficients of digital filter.
Depending upon these coefficients , the signal is passed and filtered by the digital
filter.
3. The low pass filter removes high frequency noise if any from the synthesized
speech. Because of linear phase characteristic FIR filters are mostly used as digital
filters.
Pitch Period

Pulse
Voiced
Generator Synthetic
speech
Digital Time
filter varying
digital
White
filter
Noise
generator Unvoiced
Filter Coefficients

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3. SOUND PROCESSING
1. In sound processing application, Music compression(MP3) is achieved by
converting the time domain signal to the frequency domain then removing
frequencies which are no audible.
2. The time domain waveform is transformed to the frequency domain using a filter
bank. The strength of each frequency band is analyzed and quantized based on how
much effect they have on the perceived decompressed signal.
3. The DSP processor is also used in digital video disk (DVD) which uses MPEG-2
compression, Web video content application like Intel Indeo, real audio.
4. Sound synthesis and manipulation, filtering, distortion, stretching effects are also
done by DSP processor. ADC and DAC are used in signal generation and recording.
4. ECHO CANCELLATION
In the telephone network, the subscribers are connected to telephone exchange by
two wire circuit. The exchanges are connected by four wire circuit. The two wire
circuit is bidirectional and carries signal in both the directions. The four wire circuit
has separate paths for transmission and reception. The hybrid coil at the exchange
provides the interface between two wire and four wire circuit which also provides
impedance matching between two wire and four wire circuits. Hence there are no
echo or reflections on the lines. But this impedance matching is not perfect because
it is length dependent. Hence for echo cancellation, DSP techniques are used as
follows.

Figure : Echo canceller principle

1. An DSP based acoustic echo canceller works in the following fashion: it records
the sound going to the loudspeaker and substract it from the signal coming from the

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microphone. The sound going through the echo-loop is transformed and delayed,
and noise is added, which complicate the substraction process.
2. Let be the input signal going to the loudspeaker; let be the signal picked up
by the microphone, which will be called the desired signal. The signal after
substraction will be called the error signal and will be denoted by . The adaptive
filter will try to identify the equivalent filter seen by the system from the
loudspeaker to the microphone, which is the transfer function of the room
the loudpeaker and microphone are in.
3. This transfer function will depend heavily on the physical characteristics of the
environment.In broad terms, a small room with absorbing walls will origninate
just a few, first order reflections so that its transfer function will have a short
impulse response. On the other hand, large rooms with reflecting walls will have
a transfer function whose impulse response decays slowly in time, so that
echo cancellation will be much more difficult.
5. VIBRATION ANALYSIS
1. Normally machines such as motor, ball bearing etc systems vibrate depending
upon the speed of their movements.
2. In order to detect fault in the system spectrum analysis can be performed. It
shows fixed frequency pattern depending upon the vibrations. If there is fault in the
machine, the predetermined spectrum is changes. There are new frequencies
introduced in the spectrum representing fault.
3. This spectrum analysis can be performed by DSP system. The DSP system can
also be used to monitor other parameters of the machine simultaneously.

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FIGURE FOR FINITE DURATION

Entire Z Plan e except

Z=0

Entire Z Plan e except

Z=∞

Entire z Plane except Z

=0 & Z=∞

 There are one sided(right and left sequences) and two sided sequences are
present in the Z Transform pair.
 It cosists of two duration they are
 Finite duration
 Infinite duration
 IF ITS RIGHT SIDED SEQUENCE ------ROC LIES BETWEEN Z = ∞
 IF ITS LEFT SIDED SEQUENCES ----- ROC LIES BETWEEN Z = 0
 IF ITS BOTH SIDED SEQUENCES ----- ROC LIES BETWEEN Z= 0 &
Z = ∞.

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FIGURE FOR INFINITE DURATION

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FIGURE FOR POLE ZERO PLOT

Match the Pairs.

REFERENCES:
1. Poorna Chandra S, Sasikala. B ,Digital Signal Processing, Vijay
Nicole/TMH,2013.
2. B.P.Lathi, ‗Principles of Signal Processing and Linear Systems‘, Oxford
University Press, 2010
3. Taan S. ElAli, ‗Discrete Systems and Digital Signal Processing with Mat Lab‘,
CRC Press, 2009.
4. Sen M.kuo, woonseng…s.gan, ―Digital Signal Processors, Architecture,
Implementations &
Applications, Pearson,2013
5. Dimitris G.Manolakis, Vinay K. Ingle, applied Digital Signal
Processing,Cambridge,2012
6. Lonnie C.Ludeman ,‖Fundamentals of Digital Signal Processing‖,Wiley,2013
7. R.L Lekha , Digital Signal Processing by Lakshmi publications.

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GLOSSARY TERMS:
DDC:
Digital Down ConverterTuner Mix a signal down to baseband. A DDC operates in
the complex domain and consists of a frequency mixer, a LPF, and a decimator.
SeethedownmixerinInputDevicewindow.
Decimation:
A decimator reduces the sample rate by discarding samples. It is important to low
passfilterinordertopreventaliasing.
DSP:
Digital Signal Processing. The art of manipulating signals in the discrete digital
domain.
FFT:
Fast Fourier Transform. A mathematical operation that converts between the time
andthefrequencydomains. O(n*logn).
FIR:
Finite Impulse Response filter. Performs convolution in the time domain. A DSP
algorithmic construct made up of cascaded MAC units and filter taps. FIR filters
are stable and have linear phase. Both the decimator in Input Devices and Hilbert
filtersareimplementedwithFIRfilters.
Address: The logical location of program code or data stored in memory.
Addressing mode: The method by which an instruction interprets its operands to
acquire the data it needs.
Address stage: The second stage of the parallel processor‘s fetch, address, execute
(FAE) pipeline during which addresses are calculated and supplied to the crossbar.
(TMS320C8x)
Address unit: Hardware on the parallel processor that computes a bit address
during each cycle. Each parallel processor has two address units: a global address
unit and a local address unit. (TMS320C8x)
Address unit arithmetic: The parallel processor‘s use of the local and global
address units to perform general-purpose arithmetics in parallel with the data unit.
The computed address is not used for memory access, but is stored in the
destination register. (TMS320C8x)
Address visibility (AVIS) bit: A bit field that allows the internal program address
to appear at the external address pins. This enables the internal program address to
be tracpedadanedethpenteerrupt vector to be decoded in conjunction with the interrupt

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acknowledge (IACK) signal when the interrupt vectors reside in on-chip memory.
At reset, AVIS = 0. (TMS320C5x, TMS320C54x, TMS320C2xx)
Administrative privileges: Authority to set software and hardware access; includes
access and privileges to install, manage, and maintain system and application
software and directories on a network server or individual computer systems.
ADTR: Asynchronous data transmit and receive register. See also receive (ADTR)
register.
AFB: See auxiliary register file bus.
Aggregate type: A C data type, such as a structure or an array, in which a variable
is composed of multiple other variables, called members.
AIC: See analog interface circuit.
A-Law companding: See companded.
lias disambiguation: A technique that determines when two pointer expressions
cannot point to the same location, allowing the compiler to freely optimize such
expressions. (TMS320C6200)
Aliasing: 1) A method of customizing debugger commands; aliasing provides a
shorthand method for entering often-used command strings. 2) A method of
accessing a single data object in more than one way, as when a pointer points to a
named object. The optimizer has logic to detect aliasing, but aliasing can cause
problems for the optimizer. 3) Aliasing occurs when a single object can be accessed
in more than one way, such as when two pointers point to a single object. It can
disrupt optimization, because any indirect reference could refer to any other object.
Alignment: A process in which the linker places an output section at an address
that falls on an n-byte boundary, where n is a power of 2. You can specify
alignment with the SECTIONS linker directive.
Allocation: A process in which the linker calculates the final memory addresses of
output sections.
Allocation node: The processor node into which an internode message is allocated.
ALU: See arithmetic logic unit.
ALU function: For the parallel processor, an action performed on the three inputs
to the arithmetic logic unit (ALU), which includes any arithmetic or Boolean

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combination of the three inputs, as well as mixed arithmetic and Boolean functions.
(TMS320C8x)
ALU function modifier: For the parallel processor, a 4-bit code that specifies
modifications to the functions performed by the arithmetic logic unit (ALU) data
path (such as carry-in or multiple arithmetic). These function modifiers are
specified in the opcode or in the D0 register, depending on the application.
(TMS320C8x)
ALU operation: For the parallel processor, an action performed by the arithmetic
logic unit (ALU) data path (that is, the result of the ALU function, the operation
class, and any function modifiers). (TMS320C8x) analog interface circuit (AIC):
Integrated circuit that performs serial analog- to-digital (A/D) and digital-to-analog
(D/A) conversions.
Analog mixing: The mixing together of two analog signals; the multiplexing of two
analog signals into one.
Analog repeater: Analog Repeater is telecommunications equipment used to boost
and clarify analog signals on telephone lines.
Analog-to-digital (A/D): Conversion of continuously variable electrical signals to
discrete or discontinuous electrical signals.
Analog-to-digital (A/D) converter: A converter that changes real-world analog
signals such as light and sound into digital code.
ANSI: American National Standards Institute. An organization that establishes
standards voluntarily followed by industries.
ANSI C: A version of the C programming language that conforms to the C
standards defined by the American National Standards Institute (ANSI).
annul: Any instruction that is annulled does not complete its pipeline stages.
API: See application programming interface.
application programming interface (API): Used for proprietary application
programs to interact with communications software or to conform to protocols from
another vendor‘s product. (TMS320C8x)
Application-Specific ICs (ASICs): Application-Specific ICs (ASICs) are chips
designed by our customers for specific applications by integrating cells from a TI

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standard library of pre-tested code. ASIC design is faster than designing a chip from
scratch, and design changes can be made more easily.
AR0–AR7: Auxiliary Registers 0–7. Eight registers that are used as pointers to an
address within the data space address range. The registers are operated on by the
auxiliary register arithmetic unit (ARAU) and are selected by the auxiliary register
pointer (ARP).
AR: See auxiliary register.
ARAU: See auxiliary register arithmetic unit.
ARB: See auxiliary register pointer buffer.
Architecture: The software or hardware structure of all or part of a computer
system; includes all the detailed components of the system.
Archive library: A collection of individual files grouped into a single file by the
archiver.
Archiver: A software program that collects several individual files into a single file
called an archive library. With the archiver, you can add, delete, extract, or replace
members of the archive library.
ARCR: See auxiliary register compare register.
Argument buffer: A memory block into which argument values are placed that
accompany a command to a server parallel processor.
Arithmetic logic unit (ALU): The section of the computer that carries out all
arithmetic operations (addition, subtraction, multiplication, division, or comparison)
and logic functions.
ARP: See auxiliary register pointer.
ARR: See BSP address receive register.
ARSR: See asynchronous serial port receive shift register.
ASCII: American Standard Code for Information Interchange, 1968. The standard
set of 7-bit coded characters ( 8-bit including parity check) used for information
interchange among data processing systems, communications systems, and
associated equipment. The ASCII set consists of control characters and graphics
characters.
ASPCR: See asynchronous serial port control register.

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Assemble: To prepare a machine-language program from a symbolic language

program by substituting absolute operation codes for symbolic operation codes and
absolute or relocatable addresses for symbolic addresses.
Assembler: A software program that creates a machine-language program from a
source file that contains assembly language instructions and directives. The
assembler substitutes absolute operation codes for symbolic operation codes, and
absolute or relocatable addresses for symbolic addresses.
Assembly language: A low-level symbolic programming language, closely
resembling machine code language and composed of groups of letters — each
group representing a single instruction; allows a computer user to write a program
using mnemonics instead of numeric instructions.
Assembly language instructions: The language in which computer operations are
represented by mnemonics.

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