DSP

Elective 1 (EE 504) Communication Engineering (Digital Signal Processing) By Engr. Elemuwa E. P.
Elective I (EE- 504)

Communication Engineering (Digital Signal Processing)
1.0 Introduction
Digital signal Processing (DSP) is an exciting new growth field in electronics.
These devices allow calculations to be performed quickly in order to digitally
emulate the operation of many analog filter circuits. DSP is responsible for many
of the recent advances in high-fidelity audio, high- definition TV, and
telecommunications by providing methods to eliminate random noise, Crosstalk,
and interference without adversely affecting the desired signal.
1.1
Signals, Systems and Signal processing
A signal is defined as any physical quantity that varies with time, space,
or any other independent variable or variables. Mathematically, a signal is
defined as a function of one or more independent variables.
For example,
S1(t)
=
St
2
S2(t) = 20t
(1.0)
the functions describe two signals, one that varies linearly with the independent
variable t and a second that varies quadratically with t.
A speech signal is a segment of speech which may be represented to a
high degree of accuracy as a sum of several sinusoids of different amplitudes,
frequencies and phases, i.e. as
N
A (t ) Sin [2 fi(t)t + (t)]

i 1
....
(1.1)
Where {Ai(t)}, {2fi(t)}, {i(t)} are the sets of amplitudes, frequencies, and phases
respectively of the sinusoids. This is one way to interpret the information content
or message conveyed by any short time segment of the speech signal is to
measure the amplitudes, frequencies, and phases contained in the short time
segment of the signal. Electrocardiogram (ECG) is another example of natural
signal. This signal provides a doctor with information about the condition of the
patients heart. Hence, speech, electrocardiogram, and electroencephalogram
signal are examples of information-bearing signals that evolve as functions of a
single independent variable namely, time. While image signal is an example of
signals that has a function of two independent variables.
Systems: A system is defined as a physical device that performs an
operation on a signal. For instance, a filter used to reduce the noise and
interference corrupting a desired information- bearing signal is known as a
system. The system performs some operation on the signal.
Signal processing: When we pass signal through a system, as in
filtering, we say that we have processed the signal. In this case, the processing
1
of the signal involves filtering the noise and interference from the desired signal.
Normally, the system is characterized by the type of operation that it performs on
the signal. For instance, if the operation is linear, the system is called linear. A
system is linear if it exhibits properties of homogeneity and additively. (called
super position). On the other hand, if the operation on the signal is nonlinear, the
system is said to be nonlinear. This kind of operations is known as signal
processing.
However, in DSP a system covers both physical devices and software
realizations of operations on a signal.
1.2
Basic Elements of a Digital Signal Processing System
Most of the signals in countered in Science are analog in nature. This

implies that the signals are functions of a continuous variable, such as time or
space, hence, takes on values in a continuous range. This signal is processed
directly by appropriate analog systems such as filters or frequency analyzers or
frequency multipliers for the purpose of changing their characteristics or
extracting some desired information. So, this kind of signals is said to be
processed in analog form as shown in fig 1.0. Both the input Signal and the
output signal are in an analog form.
Analog input
Signal
Analog Signal
Processor
(ASP)
Analog output
Signal
Fig 1.0 Analog signal Processing (ASP)

Digital signal processing (DSP) provides an alternative and better method for
processing the analog signal as shown in fig 1.1.
Analog
input
Signal
A/D
Converter
Digital
Signal
Processor
Digital Input signal
D/A
Converter
Analog
output
Signal
Digital output signal
Fig. 1.1. A Block diagram of a digital signal processing system.

To perform the processing digitally, an interface known as an analog-to-digital
(A/D) converter is place between the analog signal and the digital signal
processor. The output of the A/D converter is a digital signal which serves an
input to the digital processor.
The digital signal processor may be a large programmable digital
computer or a small microprocessor programmed to perform the desired
operations on the input signal. Also, it can be a hardwired digital processor

configured to perform a specified set of operations on the input signal.
Programmable machines are used to change the signal processing operations
through a change in the software, since hardwired machines are difficult to recon
figure. However, in some cases, where the digital output from the digital signal
processor is to be given to the user in analog form, such as speech
communications, we must use another interface known as digital to-Analog
(D/A) Converter, which convert the signal from digital domain to analog domain.
Thus, the signal is provided to the user in analog form.
Although, in some signal analysis, the desired information could be in
digital form. In such case, the D/A converter is not necessary. e.g. radar signal.
The information extracted from the radar signal, such as the position of the
aircraft and its speed can simply be printed out on paper.
Merits of DSP Over ASP
1. Reconfiguration: DSP can be easily programmed to change their
function and operation, without changing the logic circuit (Hardware
circuit) which is not possible in ASP.
2. Accuracy: Tolerance in Analog circuit components makes it difficult for the
system designer to control the accuracy of ASP systems. Thus, a DSP
system provides much better control of accuracy requirements.
3. Implementation: More sophisticated signal processing algorithms is
easily implemented in a DSP.
4. Portability: Digital signals are easily stored on magnetic media (tape or
disk) without deterioration or loss of signal fidelity. Hence, DSP is portable
and can easily be processed off-line.
5. Cost: It is cheaper than the ASP.
6. Digital signals can be easily delayed and/or compressed, an effect that is
difficult to achieve with analog signals.
7. Digital systems are less sensitive to additive noise.
8. A digital system doesnt have impedance-Matching requirements.
Limitations of DSP.
1. The speed of operation of A/D converters and digital signal processors
may not have the same signals.
1.3
Applications of DSP
1. Information systems
(a) Speech processing (b) Audio processing (c) Voice mail (d) Facsimile
(FAX) (e) modem (f) cellular telephones (g) Digital & LAN communications
(h) Line equalizers (i) Data encryption etc.
2. Control
(a) Printer control (b) Disk Control (C) Servo control (d) Engine Control
(e) Robots (f) Power system monitor (g) Vibration (modal) control etc.
3. Instrumentation
(a) Waveform generation
(b) Transient analysis
(c) Steady-state analysis

(d) Scientific instrumentation
4. General-Purpose
(a) Detection (correlation) (b) Filtering (convolution) (c) Spectral analysis
(Fourier transforms)
5. Graphics
(a) Image transmission and Compression (b) Image recognition
(c) Image Enhancement
6. Other areas of application
(a) Radio and Television (b) Radar and sonar (c) Oil exploration (d) music and
sound reproduction (e) Entrapment (f) Biomedical signal processing etc.
1.4
Classification of Signals
In processing a signal or in analyzing the response of a system to a signal
depends heavily on the chrematistic attributes of the specific signal. They are:
1. Multi channel and multidimensional signals: A signal can be defined as
a function of one or more independent variables. The value of the function
(i.e.
the
dependent
variable)
can
be
a
real-valued
scalar
quantity,
Complexvalued
quantity,
or
a
vector
Example
S1(t) =
A Sin 3t
Is a real-valued signal
j3t
Si(t) = Ae = ACos3t + jA Sin 3t is a complex-Valued Signal
In some applications, Signals are generated by multiple sources or multiple
sensors. Hence, such signals in turn can be represented in vector form if we
consider acceleration due to earthquake, this acceleration is the result of
three basic types of elastic waves. The Primary (P) and the secondary (S)
waves propagate within the body of rock and are longitudinal and transversal
respectively. The third type of elastic wave is called the surface wave,
because it propagates near the ground surface. If S k(t) , K =1,2,3, denotes the
electrical signal from the kth sensor as a function of time, the set of P =3
signals can be represented by a vector S3(t), where
S3(t)
S1(t)

= S (t )
2
S3 (t)
We refer to such a vector of signals as a multi channel signal.

For example, in electrocardiography; 3-lead and 12 lead electrocardiograms
(ECG) are used in practice, which result in 3-channel and 12-channel signals.
Therefore, if the signal is a function of a single independent variable, the

signal is called a one-dimensional signal. On the other hand, a signal is
called M-dimensional signal if its value is a function of M independent
variables.
The black and white television is an example of two-dimensional signal.
Hence, the black and white television picture may be represented as I(x, y, t),
since the brightness is a function of time. In contrast, a colour TV known as a
three dimensional signal. The colour TV picture may be described by three
intensity functions of the form Ir(x,y,t), Ig(x,y,t), and Ib(x,y,t), corresponding to
the brightness of the three principal colours (red, green, blue) as functions of
time. Thus, the colour TV picture is a 3-channel, 3-dimensional signal, which
can be represented by the vector.
I r ( x, y , t )
I(x,y,t) =
I g ( x, y , t )
I ( x, y , t )
b
However, in this course we will deal mainly with single-channel, onedimensional real or complex-valued signals, unless otherwise stated.
Signals can be further classified into four different categories depending
on the characteristics of the time (independent) variable and the values they
take.
2. Continuous- Time Signal
Continuous-time or analog signals e.g. Speech waveforms are defined on a
continuum of points in both the independent and dependent variables. A
signal is said to be continuous if its derivative is defined everywhere, and it is
said
to
be
discontinuous
if
it
is
not.
Mathematically, the average or mean value of an arbitrary continuous time
signal
x(t)
is
given
by
x(t) = lim
1
2T
x (t ) dt
..............................
(1.2)
T
3. Discrete- time signals:- Discrete time Signals are defined only at certain
specific values of time. These time instants need not be equidistant, but in
practice they are usually taken at equally spaced intervals for
computational
convenience
and
mathematical
tractability.
x(tn)
lim
1 N
x (k )
2N 1 k N
. . . (1. )3
N
5
In real applications, discrete time signals may arise in two ways:

(a) Sampling: This is by selecting values of an analog signal at discrete
time
instants.
All measuring instruments that take measurements at a regular interval of
time
provide
discrete-time
signals.
(b) Accumulation: By accumulating a variable over a period of time. For
example, counting the number of cars using a given street every hour, or
recording the value of gold every day, which results in discrete-time
signals.
4. Continuous valued and Discrete valued signals.

The value of a continuous time or discrete-time signal can be continuous or
discrete. If a signal takes on all possible values on a finite or an infinite range, it
is said to be continuous valued signal. Alternatively, if the signal takes on
values from a finite set of possible values, it is said to be a discrete-valued
signal. Usually, these values are equidistant and hence can be expressed as an
integer multiple of the distance between two successive values.
A discrete-time signal having a set of discrete values is called a digital
signal. For a signal to be processed digitally, it must be discrete in time and its
values must be discrete in time and its values must be discrete (i.e. it must be a
digital signal). If the signal to be processed is in analog form, it is converted to a
digital signal by sampling the analog signal at discrete instants in time, obtaining
a discrete-time signal, and then by quantizing its values to a set of discrete
values. The process of converting a continuous- valued signal into a discrete
valued signal is called quantization, which is basically an approximation process
which is achieved by rounding or truncation.
5. Deterministic and Random Signals
The mathematical analysis and processing of signals requires the availability of a
mathematical description (signal model) for the signal itself. Hence, any signal
that can be uniquely described by an explicit mathematical expression, a table of
data, or a well defined rule is called deterministic. This term is used to emphasize
the fact that all past, present and future values of the signal are known precisely,
without
any
uncertainty.
In many practical applications, there are signals that either cannot be
described to any reasonable degree of accuracy by explicit mathematical
formulas, or such a description is too complicated to be of any practical use. The
lack of such a relationship implies that such signals evolve in time in an
unpredictable manner. We refer to these signals as random. Examples of random
signals are speech signal, seismic signal etc.
1.5
The Concept of Frequency in Continuous Time and Discrete Time

Signals
The concept of frequency is directly related to the concept of time. Actually, it has
the dimension of inverse time. The nature of the time whether continuous or
discrete will affect the nature of the frequency equally.
Continuous Time Signal:
A Simple harmonic oscillation is mathematically represented by the following
continuous Time sinusoid signal.
xa(t) = Acos(t + ),
-< t<
(1.4)
as shown in fig 1.3, where A is the amplitude of the sinusoid, is the frequency
and is the phase. The subscript a used with x(t) is the analog signal.
Replacing with frequency F, which is in cycles per second or hertz (Hz). Thus,
= 2f
. (1.5)
Substituting equation 1.4 becomes,
xa(t) = A cos (2ft + ),
-< t<
... (1.6)
Fig.1.3.
Example of an analog Sinusoidal Signal
Properties of continuous- Time Signal

1. For every fixed value of the frequency F, x a(t) is periodic. i.e.
xa(t
+TP)
=
xa(t)
where TP = 1/F is the fundamental period of the sinusoidal Signal.
2. Continuous time sinusoidal signal with distinct (different) frequencies are
themselves distinct.
3. Increasing the frequency F, results in an increase in the rate of oscillation of
the Signal.
F = 1/Tp
As a result of Continuity of the time variable t, we can increase the frequency F,
without limit, with a corresponding increase in the rate of oscillation.
By definition, frequency is an inherently positive physical quantity.
Although, for mathematical convenience, we introduce negative frequencies as
shown in equ. 1.6.
xa (t) = A cos (t +Q) =
A j ( Rt Q )
A j ( Rt Q )
e
e
........1.6
+
2
2
Recall that the equ.1.6 is generated from equ. 1.4.

Note that , Xa (t) = Ae j t
is a complex conjugate exponential signals. Also, a sinusoidal signal can be

obtained by adding two equal amplitude complex- conjugate exponential
signals, sometimes called phasor as shown in fig 1.4.
Im
Re
Fig.1.4. Representation of a Cosine function by a pair of complex-conjugate

exponentials (Phasor)
Thus, as time progresses the phasors rotate in opposite directions with angular
frequencies radians per second. Since,
Positive frequency
Anton clockwise (unit form Angular motion)
Negative
Clockwise angular motion
Discrete time signals
A discrete time sinusoidal signal is expressed as;
x(n) = A Cos (n + ),
-< n<
Where,
n
=
integer variable, called the sample number
A
=
Amplitude
=
Frequency
0
=
Phase
Instead of , we use the frequency variable f defined by
=
2f
..
Substituting in equ. 1.7., we have,
x(n) =
A Cos (2fn + ), -< n<
(1.7)
(1.8)
(1.9)
The frequency f has dimensions of cycles per sample.

Properties of discrete time signals
1. It is periodic only if its frequency is a rational number.
A discrete time signal x(n) is periodic with period N(N > 0) if and only if
x(n + N) = x(n),
for all n
(1.10)
Thus, the
smallest value
of N for which
equ
(1.10)
is true is called the fundamental period. For a sinusoid with frequency f o to
be periodic, is when, Cos [2 fo (N+n) + ] = Cos (2fon + ). This relation is
true if and only if there exists an integer k such that,
2foN = 2k
or, equivalently,
fo =
k
N
(1.11)
2. Discrete time sinusoids whose frequencies are separated by an integer

multiple of 2 are identical. To show this, let us consider the sinusoid cos
(on+). It implies that
cos [(o + 2)n+] = Cos (on + 2)n+] = cos (on+). (1.12)
As a result, all sinusoidal sequences,
x k(n) =
A Cos (k n + ),
k = 0, 1, 2, 3,.
(1.13)
where
k
=
o + 2k,
- o
are indistinguishable ( i.e. identical).
3. The highest rate of Oscillation in a discrete time sinusoid is attained when
= ( or = -) or equivalently; f = (or f = -1/2).
Example 1. Stored in the memory of a digital signal processor is one cycle of
the
sinusoidal
signal
2n

x(n)
=
Sin
where
=
2q/N, Where q and N are integers.
(a) Determine how this table of values can be used to obtain values of
harmonically
related
sinusoids
having
the
same
phase.
(b) Determine how this table can be used to obtain sinusoids of the same
frequency
but
different
phase.
Solution
(a) let xk(n) denote the sinusoidal signal sequence, then,
2n
xk(n)
=
Sin
This is a sinusoid with frequency fk = K/N which is harmonically related to x(n).

But
,
xk(n)
xk(n)
maybe
expressed
as
2 ( K n )
=
x(kn)
Thus, Observe that xk(0) = x(0), xK(1) = x(k), xk(2) = x(2K), etc .
So the sinusoidal sequence x k(n) can be obtained from the table of values of
x(n) by taking every kth value of x(n), beginning with x(0). In this manner, we
can generate the values of all harmonically related sinusoids with frequencies
fk = k/N,
for
k
=
0,1,,
N-1.
(b) We can control the phase of the sinusoid with frequency f k = K/N by
taking the first value of the sequence from memory location q = N2, where
q is an integer. Thus, the initial phase controls the starting location with the
table and we wrap around the table each time the index (k n) exceeds N.
xk(n)=
1.5
Sin
Analog To- Digital Conversion
Since most of the signals such as speech, biological, Seismic, radar, sonar,
and various communications signals such as audio and video signals, are
analog. So, to process analog signals by digital means, it is first necessary to
convert them into digital form, i.e. to convert them to a sequence of numbers
having finite precision. This process is known as analog-to-digital (A/D)
conversion, and the corresponding devices are called A/D converters
(ADCs).
From the diagram shown in fig. 1.5, A/D has three main parts, namely:
A/D Converter
Digital
xa(t)
Signal
Sampler
x(n)
Quantizer
Discrete time Signal

Fig 1.5
xq(n)
Coder
Digital
Signal
Quantized Signal
Block diagram of an analog to-Digital (A/D) Converter
Sampling : This is the conversion of a Continuous- time signal into a discretetime signal obtained by taking samples of the continuous- time signal at
discrete time instants. If xa(nT) x(n), where T is known as the Sampling
interval.
Quantization: Is the conversion of a discrete time continuous-valued signal into
a discrete time, discrete valued (digital) signal. The value of each signal sample
is represented by a value selected from a finite set of possible values. The
difference between the unquantized sample x(n) and the quantized output x q(n) is
10
known as Quantization error. It is an irreversible process that results in signal

distortion.
Coding: In this case, each describe value x q(n) is represented by a b-bit binary
sequence. In coding, the quantized signal is converted to digital signal.
It is important to note that in practice the A/D conversion is performed by a
single device that takes xa(t) and produces a binary coded number.
The factors that determines the accuracy of the A/D converter are:
(i) Cost
and
(ii) Sampling rate.
Sampling of Analog signals
Periodic or Uniform sampling is one of the major ways to sample an analog
signal. As shown by Shannons,
It is expressed as
x(n) = xa(nT),
-<n<
...
(1.14)
where x(n) is the discrete-time signal obtained by taking Samples of the analog
signals xa(t) every T seconds. The time interval T between successive samples is
known as sampling period or sample interval and its reciprocal 1/T=Fs is
called the sampling rate or the sampling frequency.
Periodic sampling establishes a relationship between the time variables t
and n of continuous time and discrete-time signals respectively. These variables
are linearly related through the sampling period T or equivalently, through the
sampling rate Fs= 1/T, as
t = nT = n/Fs
(1.15)
.
The fundamental difference between continuous time and discrete time
signals is in their range of values of the frequency variables F and f, or and .
Periodic sampling of a continuous time signal implies a mapping of the infinite
frequency range for the variable F( or ) into a finite frequency range for the
variable f (or ). Thus, the highest frequency in a discrete-time signal is = or
f = . Hence, with a sampling rate F s, the corresponding highest values of F and
are:
Fmax = FS/2 = 1/2T
.
(1.16)
and max = Fs = /T
Therefore, sampling introduces an ambiguity.
Example 2: Consider the analog signal
xa(t) =3 Cos 100 t
(i)
Determine the minimum sampling rate required to avoid aliasing.
(ii)
Suppose that the signal is sampled at the rate F s=200Hz. What is the
discrete-time signal obtained after sampling?
(iii)
Suppose that the signal is sampled at the rate F s= 75 Hz. What is the
discrete-time signal obtained after sampling?
(iv)
What is the frequency 0<F<Fs/2 of a sinusoid that yields samples
identical to those obtained in part (iii)?
Solution
(i)
The freq. of the analog signal is F=50H z. Hence, the minimum
sampling rate required to avoid liaising is Fs= 100Hz.
11
(ii)
When the signal is sampled at the rate F s= 200Hz, the discrete- time
signal
is
100
n 3Cos n
x(n) = 3 cos
(iii)
IF
200
FS
x(n)
75Hz,
the
3
discrete-time
Cos
=
= 3 Cos
signal
100
75
2
n
3
cos
is
4
n
3
2
(2)n
3
n 3Cos
(iv)
For the sampling rate of F s = 75 Hz, we obtained,

F
=
fFs
=
75f
The freq. of the sinusoid in part (c) is f
= 1/3
Hence,
F=
75x
1/3
F=
25Hz
So, the sinusoidal signal is
ya(t)
=
3
cos
2
Ft
= 3 Cos 50 t
Sampled at Fs= 75 samples yields identical Samples. Hence, F=50Hz is an alias
of F= 25Hz for the sampling rate Fs= 75 Hz.
Sampling Theorem:
The theorem states that if the highest frequency components of the analog signal
are represented in sampled form without ambiguity, and hence the analog signal
can then be reconstructed without distortion from the sample values using an
appropriate interpolation (Digital-to-Analog conversion) method.
OR
If the highest frequency contained in an analog signal x a(t) is Fmax = B and the
signal is sampled at a rate F s > 2Fmax 2B, then xa(t) can be exactly recovered
from its sample values using the interpolation function.
Sin 2Bt
g(t) = 2 B t
(1.17)
This is the ideal D/A conversion process using the interpolation functions.
Hence, xa(t) may be expressed as
xa(t) =
n
Fs
X a
g (t
n
)
Fs
Where xa(n/Fs) = xa(nT) x(n) are samples of xa(t)

When the sampling of xa(t) is performed at the minimum sampling rate
Fs= 2B, the reconstruction formula in equ 1.18 becomes,
n Sin 2 B (t n / 2 B )
xa(t) = X a
(1.19)
2B ( n / 2 B )
2B
n
The Sampling rate Fn=2B = 2Fmax is known as Nyquist rate.
12
(1.18)
It is important to note that equ. 1.18 and equ 1.19 used for the reconstruction of
xa(t) from the sequence x(n) is a complicated process, which involves a weighted
sum of the interpolation function g(t) and its time-shifted versions g(t - nT)
for -<n<, where the weighting factors are the samples x(n).
Example 3: Consider the analog signal
xa(t) = 3 Cos 50t + 10 sin 300t - cos 100t.
What is the Nyquist rate for this signal.
Solution:
The freq. present in the signal is
f1 = 25 Hz, F2 = 150Hz. F3 = 50Hz
Therefore,
Fmax = 150Hz, since,
Fs> 2Fmax = 300Hz;
The Nyquist rate is FN = Fmax
FN = 300Hz
Quantization of Continuous Amplitude signals
Since, a digital signal is a sequence of numbers (samples) in which each number
is represented by a finite number of digits (finite precision). Therefore, the
process of converting a discrete-time continuous amplitude signal into a digital
signal by expressing each sample value as a finite (instead of an infinite) number
of digits is known as quantization. The error introduced in representing the
continuous-valued signal by a finite set of discrete value levels is known as
quantization error or quantization noise.
Denoting the quantizer operation on the Samples x(n) as Q[x(n)] and let x q(n)
denote the sequence of quantized samples at the output of the quantizer. Thus,
xq(n) = Q[x(n)]
The quantization error is a sequence eq(n) defined as the difference between the
quantized value and the actual sample value. Thus,
eq(n) = xq(n) - x(n)
(1.20)
However, the quality of the output of the A/D converter is usually
measured by the signal to- quantization noise ratio (SQNR). This provides the
ratio of the signal power to the noise power:
P
2b
x
SQNR = P 2 .2
q
where Px is the average power of the signal xa(t) & is given as:
Px =
1
TP
Tp
( A cos o t ) 2 dt
A2
2
and
Pq is the mean-square error power is given as
Pq=
A2
22b
Defining SQNR in decibels (dB) is given as

13
SQNR(dB) = 10 log10 SQNR = 1.76 + 6.02b

where b is the bits of accuracy.
1.6
Digital to-Analog Conversion
D/A Converter are used to convert a digital signal to an analog signal. The task of
D/A converter is to interpolate between samples.
The simplest D/A converter are the zero-order hold, which simply holds constant
the value of one sample until the next one is received. Better interpolation can be
achieved by using more sophisticated higher-order (linear) interpolation
techniques. Like the suboptimum interpolation techniques which result in passing
frequencies above the folding frequency. Thus, the D/A conversion usually
involve a suboptimum interpolator followed by a post filter or smoothing filter.
2.0 Discrete- Time Signals and Systems

Now, we will look at how discrete-time signals are used as basis functions or
building blocks to describe more complex signals.
2.1 Discrete- time Signals

In discrete-time signals, a Signal flow diagram or signal flow graph can be used
to represent the information flow and constraining paths found within a system. A
simple signal flow graph is shown in fig 2.1 and has the following elements:
i)
Node: This is equivalent to an adder in a block diagram
e.g. 1,2,3,4,
ii)
Branch : It is a direct line segment connecting two nodes e.g. [ 1,2],
[2,3] etc
iii)
Source: A node possessing only outgoing branches e.g. [ 1]
iv)
Sink (output node): A node possessing only incoming branches e.g.
[4].
v)
Internal node: Both branches directed toward it and branches directed
away from it e.g. [2], [3].
vi)
Path: Any Continuously connected set of branches e.g. [1,2,3,4]
vii)
Loop (Feedback): This is a path that originates and terminates on a
common node and through which all interconnected nodes are
transverse only once e.g. [2, 3, 2].
viii)
Feed forward path: A path connecting an input node to an a output
node in which all interconnected nodes are traversed only once e.g. [1,
2, 3, 4 ] .
ix)
Path gain: this is the product of all branch gains along a connected
path e.g. [1, 2, 3, 4] that has gain ABC.
(x)
Loop gain: The product of all branch gains along a closed loop.
e.g.[2, 3,2], has a loop gain of BD.
14
D
Fig 2.1 Signal flow graph
Apart from graphical representation of a discrete time signal there are
other alternative representations that are often more convenient to use. These
are:
1. Functional representation e.g.
1 for n 1, 3
x(n ) = 4 for n 2
0 else where
.........................
(2.1)
2. Tabular representation, such as

n
x(n)
... 2 1 0 1 2 3 4 5 . . .
. .. 0 0 0 1 4 1 0 0 . . .
3. Sequence representation
There are made up of two:
1. Infinite - duration signal or sequence: if it is with the time or ig in (n
= 0 ) medicated by the symbol is represented as
x(n) = (. 0, 0, 1, 4 , 1, 0, 0, . . .)
(2.2)
For n < 0 can be represented as
x(n) = (0, 1, 4, 1, 0, 0 . . . )
. . . . .
(2, 3)
2. Finite duration sequence: It can be represented as
x(n) = (3, -1, -2, 5, 0, 4, -1 )
.
(2, 3)
This signal consists of seven samples or points (in time), so it is known as
seven- point sequence.
But, a finite duration sequence that satisfies the condition x(n) = 0
for n< 0 can be represented as
x( n) = (0, 1, 4, 1 )
..............
(2, 4)
This is a four- point sequence.
Basic times of discrete Time signals
1. Unit sample sequence (Unit impulse): It is denoted as (n) and is defined
as
15
(n)
1 for n 0
=
0 for n 0 . . . .(2.5)
This is a signal that is zero everywhere except at n = 0 where its value is

unity.
2. unit step signal: It is denoted as U(n) and defined as
u(n)
1 for n 0
0 for n 0
. . . (2.6)
3. Unit Ramp signal: It is denoted as ur (n ) and is defined as

ur=
n, for n 0
(n)
0, for n 0
..
(2.7)
4. Exponential signal : It is a sequence of the form

x(n) = an for all n
..
(2.8)
This is a real signal, but if the parameter a is complex valued, it becomes
a rej where r and are now the parameter
x(n) = rnej = rn(Cos n+ j Sinn)
(2.9)
Alternatively, this signal x(n) can be represented graphically by the amplitude
function
x(n) = A(n) rn
.
(2.10)
and the phase function
<x(n) =(n) n
.
(2.11)
Classification of Discrete- Time Signals
1. Energy signals and power signals: The energy E of a signal x(n) is defined
as
E
x ( n)
(2.12)
The energy of a signal can be finite or infinite. If E is finite (i.e. 0<E< ), then x(n)
is called an energy signal.
However, many signals that posses infinite energy, have a finite average
power. The average power of a discrete time signal x(n) is defined as
N
1
2
x ( n)
P= in
..
(2.13)
2N 1 n N
Obviously, if E is finite, p = 0. While if E is infinite, the average power P may be
either finite or infinite. If p is finite (and nonzero), the signal is called a power
signal.
2. Periodic signals and Aperiodic Signals: A signal x(n) is periodic with period
N(N > 0) if and only if x(n+N) = x(n) for all n
..
(2.14)
16
The smallest value of N for which equ. 2.14 holds is called the period
(fundamental). If no value of N that satisfies equ. 2.14., the signal is known as
non periodic or Aperiodic.
Periodic signals are power signals with
2
1 N 1
P=
1x(n) .......................................(2,14)
N n 0
3. Symmetric and non symmetric signals: A real- valued signal x(n) is called
Symmetric (even) if
x(-n) = x(n)
.
(2.15)
While, a signal x(n) is non-symmetric (odd) if
x(-n) = -x(n)
(2.16)
Manipulations of Discrete Time signals
Transformation of the in dependent. Variable (time).
A signal x(n) can be shifted in time by replacing the independent variables n by nk, where k is an integer. If k is a positive integer, the time shift results in the
delay of the signal by k units of time
But, if x is a negative integer, the time shift results in an advance of the
signal by (k) units in time,
Example 1: A signal x(n) is graphically shown in fig. 2.2a. Show a graphical
representation of the signals x(n-3) and x (n+2).
Solution:
The signal x (n-3) is obtained by delaying x(n) by three units in time and the
result is shown in fig 2.26. While, the signal x(n+2) is obtained by advancing x(n)
by two units in time and the result is in fig. 2,2,c. Mote that delay corresponds to
shifting a signal to the right, whereas advance in plies shifting the signal to the
left on the time axis.
X(n)
4
(a)
-5
-4 -3 -2 -1 0 1
17
x(n - 3)
-2
-1
x(n+ 2)
4
-7
n
-6 -5 -4 -3
-2 -1 0 1
Fig. 2.2. Graphical representation of a signal delayed and advanced.

Addition, multiplication and scaling of sequences
All these involve amplitude modifications.
Amplitude scaling of a signal by a constant A is accomplished by
multiplying the value of every signal sample by A. i.e.
y(n) = Ax(n)
-<n< .
The sum of two signals x 1(n) and x2(n)is a signal y(n), whose value at any instant
is equal to the sum of the values of these two signals at that instant, i.e.
n
y(n) = x1(n) + x2(n)
While, the products of two signals are
n
y(n) = x1(n) x2(n)
2.2. Discrete Time systems.
x(n)
Input signal
Discrete Time
System
y(n)
Output signal (Response)
Fig. 2.3. Block diagram representation of discrete time sys.

A Discrete Time System is a device or algorithm that operates on a discretetime signal known as the input signal or excitation, according to some well
18
defamed rule, to produce another discrete time signal called the olp signal
(Response) of the system. This implies that the input signal x(n) is transformed
by the system to yield an output signal y(n). i.e.
y(n) T[x(n)]
..
(2.17)
Where
T = transformation (or an operator )
Alternatively,
x(n)
T
y(n)
(2.18)
Example: Determine the response of the following systems to the input signal.
x(n) =
3 n 3
0,
otherwise
n,
(a)
(b)
(c)
(d)
(e)
y(n) = x (n)
y (n) = x (n -1)
y (n) = x(n +1)
y (n) = 1/3 [x(n+1)+x(n)+x(n-1)]
y (n) = max {x(n+1), x(n), x(n-1)}
(f)
y (n) =
x(k ) x(n) x(n 1) x(n 2) ...........
Solution
The sample values of the input signal is
x(n) = { . . . , 0, 3, 2, 1, 0, 1, 2, 3, 0, . . . }
Then, we determine the o/p of each system;
(a) Output of the system = input to the system: Hence, the system is identity
system.
(b) Delays the input by one sample. Its output is given by
x(n) = {. . . , 0 ,3, 2, 1, 0, 1, 2, 3, 0, . . .}
(c)
Here, the system advances the input one sample into the future.
x(n) = {. . . , 0, 3, 2, 1, 0, 1, 2, 3, 0, . . . }
(d)
The o/p of this system at anytime is the mean value of the present, the
immediate past, and the immediate past, and the immediate future samples e.g.
the o/p at time n = o is
y(o) =
1
x 1 x(0) x(1) 1 1 0 1 2
3
3
3
Repeating this computation for every value of n, we obtain the output signal as
y(n) = {.. 0, 1, 5/3, 2, 1, 2/3, 1, 2, 5/3, 1, 0, ,,,,,,,,,}
(e)
This system selects as its output at time n the maximum value of the three
input samples x(n-1), x(n), and x(n+1). Thus, the response of the system is
19
y(n) = {0, 3, 3, 3, 2, 1, 2, 3, 3, 3, 0, ,, ,, ,, }
(f)
This system is basically an accumulator that computes the running sum of
all the past values upto present time. Mathematically,
y(n)
=
n
n 1
x ( k ) X ( k ) X ( n)
.....................
2.19
y ( n 1) x (n)
So, the response of the system to the given input is

y(n) = {, , , , 0, 3, 5, 6, 6, 7, 9, 12, 0 ..}
Block diagram Representation of Discrete time Systems
1. An adder: it is memoryless
x1(n)
y(n) = x1(n)+x2(n)
x2(n)
2. Constant multiplier: It is memoryless
x(n)
y(n) = ax(n)
3. Signal multiplier : It is memoryless

x1(n)
y(n) = y(n) = x1x2(n)
4. Unit delay element: It is a special system that singly delays the signal
passing through it by one sample. It has memory. z-1 is used to denote
x2(n)
the unit of delay
x(n)
Z-1
y(n) = (n-1)
5. A unit advance element: It can be stored in the memory. It moves the

input x(n) ahead by one sample in time to yield x(n+1). It is denoted by z.
x(n)
y(n)=x(n+1)
20
Example 1: Sketch the block diagram representation of the discrete time system
described by the input - output relation.
y(n) = (n-1)+ x(n)+ x(n-1)
Solution:
y(n) = 1/4y(n-1) + x(n) + x(n-1)
Rearranging, we have,
y(n) = (n-1) + [x(n) + x(n-1)]
The block diagram of the system we becomes
Z-1
0.5
x(n)
y(n)
+
Z-1
0.25
Classification of Discrete- Time Systems

In the analysis as well as in the design of systems, it is better to classify the
systems according to the general properties that they satisfy. However, for a
system to possess a given property, the property must hold for the entire input
signal to the system. These are:
1. Static and dynamic systems: It is static or memoryless if its output at
any instant in depends at most on the 1/p sample at the same time, but
not on past or future samples of the input. While, it is dynamic or memory,
if the o/p of a system at time n is completely determined by the 1/p
samples in the interval from n-N to n(N ). If N=0, the system is static.
The general input- Out put relationship equations of a static systems is
y(n) = T[x(n),n]
.
(2.20)
While, the systems described by the following input-output relations
y(n) =x(n) + 3x(n-1)
(Finite memory)
..
(2.21)
y(n) =
x(n k )
(Finite memory)
..
(2.22)
(Infinite memory)
(2.33)
k 0
y(n) =
x(n k )
k 0
are dynamic systems.

2. Time- Invariant and Time-Variant Systems:
A system is known as time-invariant if its input-output characteristics do
not change with time. Hence, a relaxed system T is time invariant or shift
invariant if and only if
x(n) T y(n)
implies that
21
x(n - k) T y(n - k)
..
(2.24)
for every input signal x(n) and every time shift k.
If the output y(n,k) = y(n - k), for all possible values of k, the system is time
invariant, while if the output y(n,k) y(n - k), even for one value of k, it is time
variant system.
3. Linear and nonlinear systems: A linear systems is one that satisfies the
superposition principle. The superposition principle states that the
response of the system to a weighted sun of signal is equal to the
corresponding weighted sum of the responses of the system to each of
the individual input signals.
Hence, a system is linear if and only if T[a,x,(n)] + [a 2x2(n)] = [a1T[x1(n)] +
[a2T[x(n)]
for any arbitrary input sequences x1(n) and x2(n), and any arbitrary
constants a1 and a2.
If a2 = 0, then equation 2.25 reduces to
T[a1x1(n)] = a1T[x1(n)] = a1y1(n)
(2.26)
where y1(n) = T[x1(n)]
Equ. 2.26 is the multiplicative or scaling property of a linear system. But, if
a1 = a2 = 1 in equ. 2.25,
Then, T[x1(n) + x2(n)] = T[x1(n)] + T[x1(n)] = y1(n) + y2(n) .
(2.27)
This kind of relationship is the additive property of a linear system.
From equ 2.26, if a, = 0, then y(n) =0. This implies a relaxed situation. So,
linear system with zero input produces a zero output. But, if a system
produces a nonzero output with a 0 input, the system is a nonlinear
system.
4. Causal and Non-causal systems: A system is said to be causal if the
output of the system at anytime n [i.e., y(n)] depends only on present and
past inputs [i.e. x(n)], x(n-1),x(n-2)..], but does not depend on future
inputs [i.e. x(n+1), x(n+2)..]. Mathematically, the output of a causal
system is given as
y(n) = F[x(n),x(n-1),x(n-2),..]
..
(2.28)
where F [ .] is an arbitrary function.
While, a system which does not satisfy this definition is known as non
causal. The system depends both on present, past and future inputs.
5. Stable and unstable systems: The theorem states that an arbitrary
relaxed system is said to be bounded input- bounded. Output (BlBO)
stable if and only if every bounded input produces a bounded output.
While, a if abounded input sequence, x(n) produces an unbounded
(infinite) is known as an unstable systems.
2.3.
Analysis of Discrete-Time Linear Time-invariant systems
22
In order to determine the output of any linear, time-invariant system to any

arbitrary input signal, two techniques are used. These are:
a. Direct solution of the input-output equation for the system: In
general, it has the form
y(n) = F[y(n-1),y(n-2),,y(n-N),x(n),X(n-1), .,x(n-m)].
Where F is the function of the quantities in brackets. Although, the
general form of the input-output relationship is
m
y(n)=- ak y (n k ) bk x( n k ) ..(2.29) where (ak) and (bk)

k 0
are constant parameters and are independent of x(n) and y(n). Equ.
2.29 is known as difference equation.
b. In this method, firstly decompose the input signal into, a sum of
elementary signals. Then, using the linearity property of the system,
the responses of the system to the elementary signals are added to
obtain the total response of the system to the given input signal.
Response of LTI Systems to Arbitrary Inputs: (The Convolution sum)
The convolution sum provides us with a means for computing the response of a
relaxed, LTI system of any arbitrary input signal x(n).
Let the response y(n,k) of the system to the input unit sample sequence at n=k
by the special symbol h(n,k), -<k< . i.e.
y(n,k) h (n,k)= T[85-k)](2.30)
where n is the time index and k is a parameter showing the location of the input
impulse. If the impulse at the input is scaled by an amount C kx(k), the response
of the system becomes
Ckh (n,k)= x(k) h(n,k).(2.31)
But, if the input is the arbitrary signal x(n)i.e
expressed as a sum of weighted impulses, i.e.
X(n)=
x(k )6(n k ) ..(2.32)
Them the response of the system to x(n) is the corresponding sum of weighted
outputs, i.e.
y(n)= T[x(n)]= T
=
=
x(k )
6 (n-k) .(2.33)
x (k )T [6 (n-k)(2.34)
x ( k ) h ( n, k )
This is known as superposition summation. equ. 2.34 is applies to any relaxed

linear (time variant)system. If in addition, the system is time invariant, the formula
in equ. 2.34 simplifies considerably. But if the response of the LTI system to the
unit sample sequence (n) is denoted as h(n), i.e. h(n) T [ (n)]
Then by the time invariante property, the response of the system to the delayed
unit sample sequence (n-k) is
h(n-k)= T [ (n-k) ] . (2.35)
23
Consequently, equ. 2.34 is reduces to

y(n) = =
x(k )h(n k ) ..(2.36)
As shown in equ. , the response y(n) of the LTI system as a function of theinput
signal x(n) and the unit sample (impulse) response h(n) is called a convolution
sum. In constr at, the general characteristic of the output of a time variant,
hinear system requires an infinite number of unit sample response functions,
h(n,k), one for each possible delay.
Suppose, we wish to compute the output of the system at sometime nistnt,
say n = no,
Applying the above equ, the response at n = no be comes
x ( k )h( n
y ( no )
k)
the process of computing the convolution between x(k) and h(k) involves the
following form steps
1. Folding: fold h(k) arbout k = 0 to obtain h(-k).
2. Shifting: Shift h(-k) by n 0 to the right (left) it n 0 to the right (left) if n 0 is positive
(negative , to obtain h(n0-k).
3. Multiplication:- multiply x(k) by h (n 0 k) to obtain the product sequence
Vno(K) = x(k) h (n0-k).
4. Summation: sum all the values of the product sequence V no(k) to obtain the
value of the output at time n = n0
Assignment 1:
The imputes response of a him ear time variants system is
h(n) = (1, 2, 1, -1 )
Determine the response of the system to the niput signal (Hint use either
graphically or mathematically method )
x(n) = x (n) * h (n)
x(k )h(n k )...........................(2,37)
Where the asterisk (*) is the convolution operation. and the impulse response
h(n) is folded and shifted.
The input to the system is x(n). so,
y(n) =h (n) x(n)
h( k )
x ( n k )............................( 2,28)
in this case, it is the input signal that is folded and it results from an inter change
of the roles of x(n) and h(n). the above two equs. is known as commutative law;
x(n) * h(n) = h(n) * x(n)
Associative lams:
[x(n) * h1(n)] * h2 (n) = x(n) * [h1(n)*h2(n)]
x(n)
h1(n)
h2(n)
y(n)
x(n)
h2(n)=
h1(n)*h2(n)
y(n)
(a)
x(n)
h1(n)
h2(n)
(b)
y(n)
24
x(n)
h2(n)
h1(n)
y(n)
Distributive law:
x(n)* [h1(n)+h2(n)]=x(n)* h1(n)+x(n)*h2(n)
x(n)
h1(n)
x(n)
+
h2(n)
25
h(n)=
h1(n)+h2(n)
y(n)

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Elective 1 (EE 504) Communication Engineering (Digital Signal Processing) By Engr. Elemuwa E. P.

Elective I (EE- 504)

Signals, Systems and Signal processing

A (t ) Sin [2 fi(t)t + (t)]

Basic Elements of a Digital Signal Processing System

Most of the signals in countered in Science are analog in nature. This

Fig 1.0 Analog signal Processing (ASP)

Digital Input signal

Digital output signal

Fig. 1.1. A Block diagram of a digital signal processing system.

operations on the input signal. Also, it can be a hardwired digital processor

(c) Steady-state analysis

We refer to such a vector of signals as a multi channel signal.

Therefore, if the signal is a function of a single independent variable, the

In real applications, discrete time signals may arise in two ways:

4. Continuous valued and Discrete valued signals.

The Concept of Frequency in Continuous Time and Discrete Time

Example of an analog Sinusoidal Signal

Properties of continuous- Time Signal

Recall that the equ.1.6 is generated from equ. 1.4.

is a complex conjugate exponential signals. Also, a sinusoidal signal can be

Fig.1.4. Representation of a Cosine function by a pair of complex-conjugate

The frequency f has dimensions of cycles per sample.

2. Discrete time sinusoids whose frequencies are separated by an integer

This is a sinusoid with frequency fk = K/N which is harmonically related to x(n).

Analog To- Digital Conversion

Discrete time Signal

Block diagram of an analog to-Digital (A/D) Converter

known as Quantization error. It is an irreversible process that results in signal

For the sampling rate of F s = 75 Hz, we obtained,

Where xa(n/Fs) = xa(nT) x(n) are samples of xa(t)

Defining SQNR in decibels (dB) is given as

SQNR(dB) = 10 log10 SQNR = 1.76 + 6.02b

Digital to-Analog Conversion

2.0 Discrete- Time Signals and Systems

2.1 Discrete- time Signals

2. Tabular representation, such as

This is a signal that is zero everywhere except at n = 0 where its value is

3. Unit Ramp signal: It is denoted as ur (n ) and is defined as

4. Exponential signal : It is a sequence of the form

Fig. 2.2. Graphical representation of a signal delayed and advanced.

Fig. 2.3. Block diagram representation of discrete time sys.

x(k ) x(n) x(n 1) x(n 2) ...........

So, the response of the system to the given input is

3. Signal multiplier : It is memoryless

y(n) = y(n) = x1x2(n)

5. A unit advance element: It can be stored in the memory. It moves the

Classification of Discrete- Time Systems

are dynamic systems.

Analysis of Discrete-Time Linear Time-invariant systems

In order to determine the output of any linear, time-invariant system to any

y(n)=- ak y (n k ) bk x( n k ) ..(2.29) where (ak) and (bk)

x(k )6(n k ) ..(2.32)

This is known as superposition summation. equ. 2.34 is applies to any relaxed

Consequently, equ. 2.34 is reduces to

x(k )h(n k ) ..(2.36)

x(k )h(n k )...........................(2,37)

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