|A| > 1 → Compression of the signal

|A| < 1 → Expansion of the signal

Note: u(at) = u(t) time scaling is not applicable for unit step function.

Time Reversal

x(-t) is the time reversal of the signal x(t).

Classification of Systems:

Systems are classified into the following categories:

• Liner and Non-linerSystems

• Time Variant and Time InvariantSystems
• Liner Time variant and Liner Time invariantsystems
• Static and DynamicSystems
• Causal and Non-causalSystems
• Invertible and Non-InvertibleSystems
• Stable and UnstableSystems
Linear and Non-linear Systems

A system is said to be linear when it satisfies superposition and homogenate principles.

Consider two systems with inputs as x1(t), x2(t), and outputs as y1(t), y2(t) respectively. Then,
according to the superposition and homogenate principles,

T [a1 x1(t) + a2 x2(t)] = a1 T[x1(t)] + a2 T[x2(t)]

∴ T [a1 x1(t) + a2 x2(t)] = a1y1(t) + a2 y2(t)

From the above expression, is clear that response of overall system is equal to response of
individual system.

Example:

y(t) = x2(t)

Solution:

y1 (t) = T[x1(t)] = x21(t)

y2 (t) = T[x2(t)] = x22(t)

T [a1 x1(t) + a2 x2(t)] = [ a1 x1(t) + a2 x2(t)]2

Which is not equal to a1y1(t) + a2 y2(t). Hence the system is said to be non linear.

Time Variant and Time Invariant Systems

A system is said to be time variant if its input and output characteristics vary with time.
Otherwise, the system is considered as time invariant.

The condition for time invariant system is:

y (n , t) = y(n-t)

The condition for time variant system is:

y (n , t) ≠ y(n-t)

Where y (n , t) = T[x(n-t)] = inputchange

y (n-t) = output change

Example:

y(n) = x(-n)

y(n, t) = T[x(n-t)] = x(-n-t)

y(n-t) = x(-(n-t)) = x(-n + t)

∴ y(n, t) ≠ y(n-t). Hence, the system is time variant.

Liner Time variant (LTV) and Liner Time Invariant (LTI) Systems

If a system is both liner and time variant, then it is called liner time variant (LTV) system.

If a system is both liner and time Invariant then that system is called liner time invariant (LTI)
system.

Static and Dynamic Systems

Static system is memory-less whereas dynamic system is a memory system.

Example 1: y(t) = 2 x(t)

For present value t=0, the system output is y(0) = 2x(0). Here, the output is only dependent upon
present input. Hence the system is memory less or static.

Example 2: y(t) = 2 x(t) + 3 x(t-3)

For present value t=0, the system output is y(0) = 2x(0) + 3x(-3).

Here x(-3) is past value for the present input for which the system requires memory to get this
output. Hence, the system is a dynamic system.

Causal and Non-Causal Systems

A system is said to be causal if its output depends upon present and past inputs, and does not
depend upon future input.

For non causal system, the output depends upon future inputs also.

Example 1: y(n) = 2 x(t) + 3 x(t-3)

For present value t=1, the system output is y(1) = 2x(1) + 3x(-2).

Here, the system output only depends upon present and past inputs. Hence, the system is causal.
Example 2: y(n) = 2 x(t) + 3 x(t-3) + 6x(t + 3)

For present value t=1, the system output is y(1) = 2x(1) + 3x(-2) + 6x(4) Here, the system output
depends upon future input. Hence the system is non-causal system.

Invertible and Non-Invertible systems

A system is said to invertible if the input of the system appears at the output.

Y(S) = X(S) H1(S) H2(S)

= X(S) H1(S) · 1(H1(S))

Since H2(S) = 1/( H1(S) )

∴ Y(S) = X(S)

→ y(t) = x(t)
Hence, the system is invertible.

If y(t) ≠ x(t), then the system is said to be non-invertible.

Stable and Unstable Systems

The system is said to be stable only when the output is bounded for bounded input. For a
bounded input, if the output is unbounded in the system then it is said to be unstable.

Note: For a bounded signal, amplitude is finite.

Example 1: y (t) = x2(t)

Let the input is u(t) (unit step bounded input) then the output y(t) = u2(t) = u(t) = bounded
output.

Hence, the system is stable.

 DISCRETE TIME FOURIER TRANSFORM

Discrete Time Fourier Transforms (DTFT)

Here we take the exponential signals to be where is a real number.The

representation is motivated by the Harmonic analysis, but instead of following the historical
development of the representation we give directly the defining equation.Let be

discrete time signal such that that is sequence is absolutely summable.

Thesequence can be represented by a Fourier integral of theform.

(1)
Where

(2)

Equation (1) and (2) give the Fourier representation of the signal. Equation (1) is referred
as synthesis equation or the inverse discrete time Fourier transform (IDTFT) and equation (2)is
Fourier transform in the analysis equation. Fourier transform of a signal in general is a complex
valued function, we can write

where is magnitude and is the phase of. We also use the term Fourier
spectrumorsimply,thespectrumtoreferto.Thus iscalledthemagnitudespectrumand is
called the phase spectrum. From equation (2) we can see that is a periodic function
with period i.e.. We can interpret (1) as Fourier coefficients in the representation of a
periodic function. In the Fourier series analysis our attention is on the periodic function, here we
are concerned with the representation of the signal. So the roles of the two equation are
interchanged compared to the Fourier series analysis of periodicsignals.

Now we show that if we put equation (2) in equation (1) we indeed get the signal.
Let

where we have substituted from (2) into equation (1) and called the result as.
Since we have used n as index on the left hand side we have used m as the index variable forthe
sum defining the Fourier transform. Under ourassumptionthat sequence isabsolutely
summable we can interchange the order of integration and summation.Thus

Example: Let

Fourier transform of this sequence will exist if it is absolutely summable. We have

Fourier transform of Periodic Signals

For a periodic discrete-time signal,

its Fourier transform of this signal is periodic in w with period 2∏ , and is given

Now consider a periodic sequence x[n] with period N and with the Fourier series representation

The Fourier transform is

Properties of the Discrete Time Fourier Transform:

Let and be two signal, then their DTFT is denoted by and. Thenotation

is used to say that left hand side is the signal x[n] whose DTFT is is given at right hand
side.

1. Periodicity of theDTFT:

2. Linearity of theDTFT:

3. Time Shifting and FrequencyShifting:

4. Conjugation and Conjugate Symmetry:

From this, it follows that ReX(e jw)is an even function of w and ImX(e jw)is
an odd function of w . Similarly, the magnitude of X(e jw) is an even function and the
phase angle is
an odd function. Furthermore,

5. Differencing andAccumulation
The impulse train on the right-hand side reflects the dc or average value that can result from
summation.

For example, the Fourier transform of the unit step x[n] u[n] can be obtained by using
the accumulation property.

6. TimeReversal

7. Time Expansion

For continuous-time signal, we have

For discrete-time signals, however, a should be an integer. Let us define a signal with k a
positive integer,
For k  1, the signal is spread out and slowed down in time, while its Fourier transform is
compressed.

Example: Consider the sequence x[n] displayed in the figure (a) below. This sequence can be
related to the simpler sequence y[n] as shown in (b).

As can be seen from the figure below, y[n] is a rectangular pulse with 2 1 N  , its Fourier
transform is given by
Using the time-expansion property, we then obtain

8. Differentiation inFrequency

The right-hand side of the above equation is the Fourier transform of jnx[n] .Therefore,
multiplying both sides by j , we see that

9. Parseval’sRelation
A physical transmission system may have amplitude and phase responses as shown below:

FILTERING
One of the most basic operations in any signal processing system is filtering. Filtering is
the process by which the relative amplitudes of the frequency components in a signal are
changed or perhaps some frequency components are suppressed. As we saw in the preceding
section, for continuous-time LTI systems, the spectrum of the output is that of the input
multiplied by the frequency response of the system. Therefore, an LTI system acts as a filter on
the input signal. Here the word "filter" is used to denote a system that exhibits some sort of
frequency-selectivebehavior.

A. Ideal Frequency-Selective Filters:

An ideal frequency-selective filter is one that exactly passes signals at one set of
frequencies and completely rejects the rest. The band of frequencies passed by the filter is
referred to as the pass band, and the band of frequencies rejected by the filter is called the stop
band.
The most common types of ideal frequency-selective filters are the following.

1. Ideal Low-PassFilter:
An ideal low-pass filter (LPF) is specified by

The frequency wcis called the cutoff frequency.

2. Ideal High-PassFilter:
An ideal high-pass filter (HPF) is specified by
3. Ideal BandpassFilter:
An ideal bandpass filter (BPF) is specified by

4. Ideal BandstopFilter:
An ideal bandstop filter (BSF) is specified by

The following figures shows the magnitude responses of ideal filters

Fig: Magnitude responses of ideal filters (a) Ideal Low-Pass Filter (b)Ideal High-Pass Filter

© Ideal Bandpass Filter (d) Ideal Bandstop Filter

 UNIT – V
Z-Transform
Analysis of continuous time LTI systems can be done using z-transforms. It is a powerful
mathematical tool to convert differential equations into algebraic equations.

The bilateral (two sided) z-transform of a discrete time signal x(n) is given as

The unilateral (one sided) z-transform of a discrete time signal x(n) is given as

Z-transform may exist for some signals for which Discrete Time Fourier Transform (DTFT) does
not exist.

Concept of Z-Transform and Inverse Z-Transform

Z-transform of a discrete time signal x(n) can be represented with X(Z), and it is defined as

The above equation represents the relation between Fourier transform and Z-transform
Inverse Z-transform:
Z-Transform Properties:

Z-Transform has following properties:

Linearity Property:

Time Shifting Property:

Multiplication by Exponential Sequence Property

Time Reversal Property

Differentiation in Z-Domain OR Multiplication by n Property

Convolution Property

Correlation Property

Initial Value and Final Value Theorems

Initial value and final value theorems of z-transform are defined for causal signal.

Initial Value Theorem

For a causal signal x(n), the initial value theorem states that

This is used to find the initial value of the signal without taking inverse z-transform
Final Value Theorem
For a causal signal x(n), the final value theorem states that

This is used to find the final value of the signal without taking inverse z-transform

Region of Convergence (ROC) of Z-Transform

The range of variation of z for which z-transform converges is called region of convergence of z-
transform.

Properties of ROC of Z-Transforms

• ROC of z-transform is indicated with circle in z-plane.

• ROC does not contain anypoles.

• If x(n) is a finite duration causal sequence or right sided sequence, then the ROC isentire
z-plane except at z =0.

• If x(n) is a finite duration anti-causal sequence or left sided sequence, then the ROCis
entire z-plane except at z =∞.

• If x(n) is a infinite duration causal sequence, ROC is exterior of the circle with radiusa.
i.e. |z| > a.

• If x(n) is a infinite duration anti-causal sequence, ROC is interior of the circle withradius
a. i.e. |z| < a.

• If x(n) is a finite duration two sided sequence, then the ROC is entire z-plane except atz
= 0 & z = ∞.

The concept of ROC can be explained by the following example:

Example 1: Find z-transform and ROC of a n u[n]+a − nu[−n−1] anu[n]+a−nu[−n−1]

The plot of ROC has two conditions as a > 1 and a < 1, as we do not know a.
In this case, there is no combination ROC.

Here, the combination of ROC is from a<|z|<1/a

Hence for this problem, z-transform is possible when a < 1.

Causality and Stability

Causality condition for discrete time LTI systems is as follows:

A discrete time LTI system is causal when

• ROC is outside the outermostpole.

• In The transfer function H[Z], the order of numerator cannot be grater than the order of
denominator.