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Lecture Notes
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UNIT I
CLASSIFICATION OF SIGNALS AND SYSTEMS
Continuous time signals (CT signals), discrete time signals (DT signals)
A signal is a function representing a physical quantity or variable, and typically it
contains information about the behavior or nature of the phenomenon. For instance, in a
RC circuit the signal may represent the voltage across the capacitor or the current flowing
in the resistor. Mathematically, a signal is represented as a function of an independent
variable t. Usually t represents time. Thus, a signal is denoted by x(t).
A. Continuous-Time and Discrete-Time Signals:
A signal x(t) is a continuous-time signal if t is a continuous variable. If t is a discrete
variable, that is, x(t) is defined at discrete times, then x(t) is a discrete-time signal. Since a
discrete-time signal is defined at discrete times, a discrete-time signal is often identified as
a sequence of numbers, denoted by {x,) or x[n], where n = integer. Illustrations of a
continuous-time signal x(t) and of a discrete-time signal x[n] are shown in Fig. 1-1.
A discrete-time signal x[n] may represent a phenomenon for which the independent
variable is inherently discrete. For instance, the daily closing stock market average is by its
nature a signal that evolves at discrete points in time (that is, at the close of each day). On
the other hand a discrete-time signal x[n] may be obtained by sampling a continuous-time
signal x(t) such as
or in a shorter form as
and xn s are called samples and the time interval between them is called the sampling
interval. When the sampling intervals are equal (uniform sampling), then
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. Any signal x(t) or x[n] can be expressed as a sum of two signals, one of which is even
and one of which is odd. That is,
signal x(t) (known as a dc signal). For a constant signal x(t) the fundamental period is
undefined since x(t) is periodic for any choice of T (and so there is no smallest positive
value). Any continuous-time signal which is not periodic is called a nonperiodic (or
aperiodic ) signal.
Periodic discrete-time signals are defined analogously. A sequence (discrete-time
signal) x[n] is periodic with period N if there is a positive integer N for which
x[n + N] =x[n] all n (1.9)
An example of such a sequence is given in Fig. 1-3(b). From Eq. (1.9) and Fig. 1-3(b) it
follows that
for all n and any integer m. The fundamental period No of x[n] is the smallest positive
integer N for which Eq. (1.9) holds. Any sequence which is not periodic is called a
nonperiodic (or aperiodic sequence.
Note that a sequence obtained by uniform sampling of a periodic continuous-time
signal may not be periodic (Probs. 1.12 and 1.13). Note also that the sum of two
continuous-time periodic signals may not be periodic but that the sum of two periodic
sequences is always periodic .
G. Energy and Power Signals:
Consider v(t) to be the voltage across a resistor R producing a current dt). The
instantaneous power p( t) per ohm is defined as
For an arbitrary continuous-time signal x(t), the normalized energy content E of x(t) is
defined as
Similarly, for a discrete-time signal x[n], the normalized energy content E of x[n] is
defined as
Based on definitions (1.14) to (1.17), the following classes of signals are defined:
1. x(t) (or x[n]) is said to be an energy signal (or sequence) if and only if 0 < E < m, and
so P = 0.
2. x(t) (or x[n]) is said to be a power signal (or sequence) if and only if 0 < P < m, thus
implying that E = m.
3. Signals that satisfy neither property are referred to as neither energy signals nor power
signals.
Note that a periodic signal is a power signal if its energy content per period is finite, and
then the average power of this signal need only be calculated over a period.
A. The Unit Step Sequence:
The unit step sequence u[n] is defined as
which is shown in Fig. 1-10(a). Note that the value of u[n] at n = 0 is defined [unlike the
continuous-time step function u(f) at t = 01 and equals unity. Similarly, the shifted unit step
sequence ii[n - k] is defined as
which is shown in Fig. 1-ll(a). Similarly, the shifted unit impulse (or sample) sequence
6[n - k] is defined as
Thus x[n] is a complex sequence whose real part is cos Ron and imaginary part is sin
Ron
. General Complex Exponential Sequences:
The most general complex exponential sequence is often defined as
where C and α are in general complex numbers. Note that Eq. (1.52) is the special case
of
Eq. (1.57) with C = 1 and
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As we observed in the case of the complex exponential sequence in Eq. (1.52), the same G.
observations [Eqs. (1.54) and (1.5611 also hold for sinusoidal sequences. For instance, the
sequence in Fig. 1-13(a) is periodic with fundamental period 12, but the sequence in Fig.
l-13( b) is not periodic.
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Note that all memoryless systems are causal, but not vice versa
. Linear Systems and Nonlinear Systems:
If the operator T in Eq. (1.60) satisfies the following two conditions, then T is called a
linear operator and the system represented by a linear operator T is called a linear system:
1. Additivity:
Given that Tx, = y, and Tx, = y,, then
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where α1 and α2 are arbitrary scalars. Equation (1.68) is known as the superposition
property. Examples of linear systems are the resistor [Eq. (1.6111 and the capacitor [Eq.
( 1.62)]. Examples of nonlinear systems are
Note that a consequence of the homogeneity (or scaling) property [Eq. (1.6711 of linear
systems is that a zero input yields a zero output. This follows readily by setting α = 0 in
Eq.(1.67). This is another important property of linear systems.
F. Time-Invariant and Time-Varying Systems:
A system is called rime-inuariant if a time shift (delay or advance) in the input signal
causes the same time shift in the output signal. Thus, for a continuous-time system, the
system is time-invariant if
for any real value of T. For a discrete-time system, the system is time-invariant (or
shift-incariant ) if
for any integer k. A system which does not satisfy Eq. (1.71) (continuous-time system) or
Eq. (1.72) (discrete-time system) is called a time-varying system. To check a system for
time-invariance, we can compare the shifted output with the output produced by the
shifted input (Probs. 1.33 to 1.39).
G. Linear Time-Invariant Systems
If the system is linear and also time-invariant, then it is called a linear rime-invariant
(LTI) system.
H. Stable Systems:
A system is bounded-input/bounded-output (BIBO) stable if for any bounded
input x defined by
where k , and k, are finite real constants. Note that there are many other definitions of
stability. (See Chap. 7.)
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where K is a (gain) constant. Thus, the corresponding impulse response h(f) is simply
Therefore, if h(tJ # 0 for I,, # 0, the continuous-time LTI system has memory.
B. Causality:
As discussed in Sec. 1.5D, a causal system does not respond to an input event until that
event actually occurs. Therefore, for a causal continuous-time LTI system, we have
Applying the causality condition (2.16) to Eq. (2.101, the output of a causal continuous-time
LTI system is expressed as
Equation (2.18) shows that the only values of the input x(t) used to evaluate the output
y( t ) are those for r 5 t.
Based on the causality condition (2.161, any signal x(t) is called causal if
Then, from Eqs. (2.17), (2. I8), and (2. Iga), when the input x(t) is causal, the output y(t )
of a causal continuous-time LTI system is given by
C. Stability:
The BIBO (bounded-input/bounded-output) stability of an LTI system (Sec.
1.5H) is readily ascertained from its impulse response. It can be shown (Prob. 2.13) that a
continuous-time LTI system is BIBO stable if its impulse response is absolutel integrable,
that is,
Applying the causality condition (2.44) to Eq. (2.391, the output of a causal discrete-time
LTI system is expressed as
Equation (2.46) shows that the only values of the input x[n] used to evaluate the output
y[n] are those for k I n.
As in the continuous-time case, we say that any sequence x[n] is called causal if
Then, when the input x[n] is causal, the output y[n] of a causal discrete-time LTI
system is given by
C. Stability:
It can be shown (Prob. 2.37) that a discrete-time LTI system is BIB0 stable if its
impulse response is absolutely summable, that is,
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UNIT II
Analysis of continuous time signals
A two parts tutorial on Fourier series. In the first part an example is used
to show how Fourier coefficients are calculated and in a second part you may
use an applet to further explore Fouries series of the same function.
where n = 1 , 2 , 3 , ... and T is the period of function f(t). an and bn are called
Fourier coefficients and are given by
Example : Find the Fourier series of the periodic function f(t) defined by
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Coefficient a0 is given by
Coefficients an is given by
is
where is defined as
Time integral over in equation (b) becomes over the entire time axis:
a non-periodic signal :
distributed along the frequency axis. We can only speak of the energy contained in
a particular frequency band :
Note on notations:
Example 0:
Example 1:
i.e.,
Example 2:
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The spectrum is
Note that the height of the main peak is and it gets taller and narrower as gets
larger. Also note
The integral in the above transform is an important formula to be used frequently later:
Example 3:
Again, these spectra represent the energy density distribution of the sinusoids, while the
corresponding Fourier coefficients
and
Inverse Transforms
If we have the full sequence of Fourier coefficients for a periodic signal, we can
reconstruct it by multiplying the complex sinusoids of frequency ω0k by the weights Xk
and summing:
p1
x(t) X k e ik0t
x(n) X k e ik0n
k
k 0
We can perform a similar reconstruction for aperiodic signals
1 1
x(n) X ()ein d x(t) X ()eit d
2 2
These are called the inverse transforms.
X () x(t)eit dt (t)eit dt ei0 1
Fourier Transform of Impulse Functions
Find the Fourier transform of the Dirac delta function:
Laplace Transform
► Lapalce transform is a generalization of the Fourier transform in the sense that it allows
“complex frequency” whereas Fourier analysis can only handle “real frequency”. Like
Fourier transform, Lapalce transform allows us to analyze a “linear circuit” problem, no
matter how complicated the circuit is, in the frequency domain in stead of in he time
domain.
►Mathematically, it produces the benefit of converting a set of differential equations into
a corresponding set of algebraic equations, which are much easier to solve. Physically, it
produces more insight of the circuit and allows us to know the bandwidth, phase, and
transfer characteristics important for circuit analysis and design.
►Most importantly, Laplace transform lifts the limit of Fourier analysis to allow us to
find both the steady-state and “transient” responses of a linear circuit. Using Fourier
transform, one can only deal with he steady state behavior (i.e. circuit response under
indefinite sinusoidal excitation).
►Using Laplace transform, one can find the response under any types of excitation (e.g.
switching on and off at any given time(s), sinusoidal, impulse, square wave excitations,
etc.
Laplace Transform
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3
UNIT III
LINEAR TIME INVARIANT –CONTINUOUS TIME SYSTEMS
Differential equation, Block diagram representation, Impulse response, Convolution
integral, frequency response, State variableequations and matrix representation of
systems.
System:
A system is an operation that transforms input signal x into output signal y.
LTI Systems
• Time Invariant
• Linearity
– a1x1(t)+ a2x2(t)
a1y1(t)+ a2y2(t)
Differential equation:
• This is a linear first order differential equation with constant coefficients
(assuming a and b are constants)
Impulse Response
LTI System
δ (t) H[.]
h(n)=H[ δ (t)]
A very important way to analyse a system is to stud y the output signal when a unit
impulse signal is used as an input.
This impulse response signal can be used to infer properties about the system’s structure
(LHS of difference equation or unforced solution).
The system impulse response, h(t) completely characterises a linear, time invariant
system
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Convolution Integral
• An approach (available tool or operation) to describe the input-output relationship
for LTI Systems
• In a LTI system
– d(t)
h(t)
• Many familiar properties, such as stability, can be derived from this common
representation. It forms the basis for the theoretical analysis of linear systems.
• The state variables often represent internal elements of the system such as
voltages across capacitors and currents across inductors.
• They account for observable elements of the circuit, such as voltages, and also
account for the initial conditions of the circuit, such as energy stored in
capacitors. This is critical to computing the overall response of the system.
• Let us define the state of the system by an N-element column vector, x(t):
x (t)
x 1(t)
x(t) 2 x (t) x2 (t) x (t)t
1 N
Note that inxth(its) development, v(t) will be the input, y(t) will be the
N
output, and x(t) is used for the state variables.
• Any system can be modeled by the following state equations:
x t y(i1) t i
i0
The state variables are defined as N
i ( )
, 1, 2, ...,
xN 1 (t) xN (t)
N 1
0 1 0 0 0
0
0 1 0
A B
0 0 0 1 0
a a a a b
0 1 2 N 1 0
C 1 0 0 0 D 0
UNIT IV
The sampling frequency or sampling rate fs is defined as the number of samples obtained
in one second, or fs = 1/T. The sampling rate is measured in hertz or in samples per
second.
The frequency equal to one-half of the sampling rate is therefore a bound on the highest
frequency that can be unambiguously represented by the sampled signal. This frequency
(half the sampling rate) is called the Nyquist frequency of the sampling system.
Frequencies above the Nyquist frequency fN can be observed in the sampled signal, but
their frequency is ambiguous. That is, a frequency component with frequency f cannot be
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distinguished from other components with frequencies NfN + f and NfN – f for nonzero
integers N. This ambiguity is called aliasing. To handle this problem as gracefully as
possible, most analog signals are filtered with an anti-aliasing filter (usually a low-pass
filter with cutoff near the Nyquist frequency) before conversion to the sampled discrete
representation.
► The theory of taking discrete sample values (grid of color pixels) from functions
defined over continuous domains (incident radiance defined over the film plane)
and then using those samples to reconstruct new functions that are similar to the
original (reconstruction).
► Sampling theory
Sampling Theorem:
bandlimited signal can be reconstructed exactly if it is sampled at a rate atleast twice the
maximum frequencycomponent in it."
► Consider a signal g(t) that is bandlimited.
► Sampling theory
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The maximum frequency component of g(t) is fm. To recover the signal g(t) exactly from
its samples it has to be sampled ata rate fs _ 2fm.
The minimum required sampling rate fs = 2fm is called nyquist rate
Aliasing
Aliasing is a phenomenon where the high frequency components of the sampled signal
interfere with each other because of inadequate sampling ωs < 2ωm.
Aliasing leads to distortion in recovered signal. This is the reason why sampling
frequency should be atleast twice the bandwidth of the signal .
DISCRETE TIME FOURIER TRANSFORM
To understand this, consider the Poisson summation formula, which indicates that a
periodic summation of function can be constructed from the samples of function
The result is:
(Eq.2)
The right-hand sides of Eq.2 and Eq.1 are identical with these associations:
Inverse transform
The integrals span one full period of the DTFT, which means that the x[n] samples are
also the coefficients of a Fourier series expansion of the DTFT. Infinite limits of
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Properties
This table shows the relationships between generic discrete-time Fourier transforms. We
use the following notation:
Remark
Property Time domain Frequency domain s
Linearity
Shift in time integer k
Shift in
frequency real
(modulation number a
)
Time
reversal
Time
conjugation
Time
reversal &
conjugation
Derivative
in frequency
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Integral in
frequency
Convolve in
time
Multiply in
time
Correlation
Parseval's
theorem
SYMMETRY PROPERTIES
The Fourier Transform can be decomposed into a real and imaginary part or into an even
and odd part.
or
Z-transforms
► Definition: The Z – transform of a discrete-time signal x(n) is defined as the power
series:
X ( z) x(n) z
k
H[z] hnz n
Bilateral inversn eZ transform
1
h[n] H[z] zn1dz
2 j R
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Z-transform Pairs
► h[n] = d[n]
H[z] nz n
nz n 1
n n0
► h[n] = d[n-1]
► Inverse z-transform
cj
f n 1 F z z n1dz
2j c
j
► Using the definition requires a contour integration in the complex z-plane.
► Fortunately, we tend to be interested in only a few basic signals (pulse, step, etc.)
Virtually all of the signals we’ll see can be built up from these basic signals.
For these common signals, the z-transform pairs have been tabulated (see
Lathi, Table 5.1)
Z-transform Properties
Properties of z - transform
1. Linearity
Z (x1(nT) x2 (nT)) Z (x1(nT)) Z (x2 (nT))
2. Initial Value x(0) lim X (z)
z
s 0 z 1
1
1
s 1 z 1
s 1 z 1
sX (s) (1 z 1 ) X (z)
lim sX (s) lim (1 z 1 ) X (z)
s 0 z 1
UNIT V
– input
– output
– intermediate results
– addition
– multiplication
Direct Form I
General form of difference equation
N M
â yn k bˆ xn k
k 0
k
k 0
k
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Direct Form II
► Cascade form
z 1g z 1g z
H z A1f
M1 M2
1 1 1
k k k
k 1 k 1
1c z 1d z 1d z
N1 N2
1 1 1
k k k
k 1 k 1
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0
Parallel form
► Represent system function using partial fraction expansion
Bk 1ek z 1
HEzSPONSCEk z 1
ankdzwe
NP NP NP
ynxnhn
► The z-transform of impulse response is called transfer or system function H(z).
CONVOLUTIO N SUM
The convolution sum provides a concise, mathematical way to express
the output of an LTI system based on an arbitrary discrete-time input signal and the system's
response. The convolution sum is expressed as
Linear time-yi[nnv]ariax[nnt ]syhst[enm
] s can be described by the convolution sum
• Convolution is conmutative
h[k]
k
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h[n] = 0 n < 0
LTI SYSTEMS ANALYSIS USING DTFT
X (e j) X (e j) e jX ( e j)
• Consider and H (e j) H (e j) e jH (e j)
, then
– magnitude
a yn k b xn k
k 0
k
k 0
k
N
k
a z Y (z) M b zk X z
k k
• Compute thekz0-transform k 0
M
Y z
k
bk z
H
z kN0
X z
a
k 0
k k
z
• Location of zeros and poles determines the frequency response and phase
1c z
M
1
k
H z 0
b k 1
1d z
N
a0
k 1
k 1