Unit 1.

DISCRETE TIME SIGNALS

Introduction to Digital signal processing, Classification of Signal Processing, Advantages of
DSP over ASP, Signal, Classification of Signals, The concept of frequency in continuous time
and discrete-time signals, Discrete Time Harmonics and Sinusoids, Basic block diagram of
A/D converter, Sampling Theorem (natural, Flattop), Nyquist rate, Aliasing effect,
Quantization. Digital-to-Analog Conversion (D/A).

Representation of Discrete Time signals, Common Discrete signals, Simple manipulations on

Discrete Time signals, Decimation and Interpolation, Exercises.
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Introduction to Digital signal processing

Digital signal processing (DSP) technology and its advancements have dramatically impacted
our modern society everywhere. Without DSP, we would not have digital/Internet audio or
video; digital recording; CD, DVD, and MP3 players; digital cameras; digital and cellular
telephones; digital satellite and TV; or wire and wireless networks. Medical instruments would
be less efficient or unable to provide useful information for precise diagnoses if there were no
digital electrocardiography (ECG) analyzers or digital x-rays and medical image systems. We
would also live in many less efficient ways, since we would not be equipped with voice
recognition systems, speech synthesis systems, and image and video editing systems. Without
DSP, scientists, engineers, and technologists would have no powerful tools to analyze and
visualize data and perform their design, and so on.

The concept of DSP is illustrated by the simplified block diagram in Figure 1.1, which consists
of an analog filter, an analog-to-digital conversion (ADC) unit, a digital signal (DS) processor,
a digital-to-analog conversion (DAC) unit, and a reconstruction (anti-image) filter.
As shown in the diagram, the analog input signal, which is continuous in time and amplitude,
is generally encountered in our real life. Examples of such analog signals include current,
voltage, temperature, pressure, and light intensity. Usually a transducer (sensor) is used to
convert the nonelectrical signal to the analog electrical signal (voltage). This analog signal is
fed to an analog filter, which is applied to limit the frequency range of analog signals prior to
the sampling process. The purpose of filtering is to significantly attenuate aliasing distortion,
which will be explained in the next chapter. The band-limited signal at the output of the analog
filter is then sampled and converted via the ADC unit into the digital signal, which is discrete
both in time and in amplitude. The DS processor then accepts the digital signal and processes
the digital data according to DSP rules such as lowpass, highpass, and bandpass digital filtering,
or other algorithms for different applications. Notice that the DS processor unit is a special type
of digital computer and can be a general-purpose digital computer, a microprocessor, or an
advanced microcontroller; furthermore, DSP rules can be implemented using software in
general. With the DS processor and corresponding software, a processed digital output signal
is generated. This signal behaves in a manner according to the specific algorithm used. The
next block in Figure 1.1, the DAC unit, converts the processed digital signal to an analog output
signal. As shown, the signal is continuous in time and discrete in amplitude. The final block in Figure
1.1 is designated as a function to smooth the DAC output voltage levels back to the analog signal via a
reconstruction (anti-image) filter for real-world applications.

In general, the analog signal process does not require software, an algorithm, ADC, and DAC.
The processing relies wholly on electrical and electronic devices such as resistors, capacitors,
transistors, operational amplifiers, and integrated circuits (ICs). DSP systems, on the other
hand, use software, digital processing, and algorithms; thus they have a great deal of flexibility,
less noise interference, and no signal distortion in various applications. However, as shown in
Figure 1.1, DSP systems still require minimum analog processing such as the anti-aliasing and
reconstruction filters, which are musts for converting real-world information into digital form
and digital form back into real-world information. Note that there are many real-world DSP
applications that do not require DAC, such as data acquisition and digital information display,
speech recognition, data encoding, and so on. Similarly, DSP applications that need no ADC
include CD players, text-to-speech synthesis, and digital tone generators, among others. We
will review some of them in the following sections.

Advantages of DSP over ASP

Advantages of DSP
1. Physical size of analog systems is quite large while digital processors are more compact and
lighter in weight.
2. Analog systems are less accurate because of component tolerance. example R, L, C and
active components. Digital components are less sensitive to the environmental changes, noise
and disturbances.
3. Digital system is most flexible as software programs & control programs can be easily
modified.
4. Digital signal can be stored on digital hard disk, digital media (like a compact disc) and
manipulated on digital systems (like the integrated circuit in a CD player). Hence becomes
transportable. Thus, easy and lasting storage capacity.
5. Digital processing can be done offline.
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 they are software controlled.
8. Possibility of sharing DSP processor between several tasks.
9. 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 A/D converters are difficult to achieve in practice. In high
frequency applications DSP are not preferred.
What is a Signal?
A signal is defined as any physical quantity that changes with time, distance, speed, position,
pressure, temperature or some other quantity.
A few of the examples of signals would be;
1. An electrical voltage travelling along copper wires between your telephone and the
local exchange.
2. Pulses of light (though we might not be able to see them) in a fibre-optic cable
3. The radio emissions that are picked up by a mobile telephone or radio receiver.
All these provide the necessary variations to represent the data.

The signal as a function of time, can be represented as x(t) or f(t) or any other notation. For
example, f(t) might denote a voltage level, or the velocity of an object, or the price of a stock
at time t.

In some cases, we might be interested in measuring the quantity as a function of some variable
other than time. For example, suppose we are interested in measuring the water temperature in
the ocean as a function of depth. In this case, the signal is a function of a spatial variable, with
f(x) denoting temperature at depth x.

A signal need not be a function of just a single variable. To continue the example above,
suppose we are interested in the temperature at particular points in the ocean, not simply as a
function of depth. In this case, we might let f (x, y, z) denote the temperature at the point (x, y,
z), so the signal is a function of three variables.
Now, if we are also interested in how the temperature evolves in time, the signal f (x, y, z, t)
would be a function of four variables.

Other examples of signals that we will encounter frequently are audio signals, images, and
video.
An audio signal is created by changes in air pressure, and therefore can be represented by a
function of time f(t) with f representing the air pressure due to the sound at time t.

Figure 1: Graph of someone saying the word “MATLAB”

Signals need not always be functions of time. For example, in the case of an image, the signal
is a function of its position. A pixel in any image is specified by its x- and y-coordinates.
However, most of the signals we encounter are functions of time, as they change with respect
to time.
A black and white image can be represented as a function f(x,y) of two variables. Here (x,y)
denotes a particular point on the image, and the value f(x,y) denotes the brightness (or gray
level) of the image at that point.

Figure 2. A Gray scale Image

A video can be thought of as a sequence of images. Hence, a black and white video signal can
be represented by a function f(x,y,t) of three variables (two spatial variables and time). In this
case, for a fixed t, f (·, ·, t) represents the still image/frame at time t, while for a fixed (x, y), f
(x, y, ·) denotes how the brightness at the point (x, y) changes as a function of time.

Figure 3. Video Frames

Any signal can be represented by weighted sum of sinusoids of different amplitudes and
frequencies.

Here A is the amplitude, and w is the radian frequency.

Figure 4. Sinusoid representation of a signal

An equivalent form of sinusoid equation which is often used is

The frequency f is in units of Hertz (abbreviated Hz) which is sec-1, or often called cycles per
second. Of course, f and w are related by w = 2pf.

Any signal is a combination of sinusoids,

x(t) =

In this example mostly a sinusoid at frequency w2 is dominant, with small contributions from
sinusoids at frequencies w1 and w3. But, it is not obvious by seeing the composite waveform
x(t) represented in the time domain. Therefore, to analyse a signal we have to convert it into a
frequency domain.

Signals can be represented in time-domain or in frequency-domain.

What is frequency domain and Why use frequency domain representation?

It reveals the fundamental characteristics of a signal and therefore, it is easier to design a system
that handles or processes the signal.
If you project/mix a sinewave of one frequency onto another sinewave of a different frequency,
no matter how close they are, each frequency representation is unique.
This is the essence of transforms, and that is why we convert signals from one domain to
another especially time domain to frequency domain. This is fundamental to signal processing.
Depending on what you want to do with the signal, processing in one of the two domains will
prove beneficial.

Example 1:
Listen to the piano chord. You hear several notes being struck, and fading away. This is
waveform is plotted below:
The time series plot shows the time the chord starts, and its decay, but it is difficult tell what the notes are from
the waveform. If we represent the waveform as a sum of sinusoids at different frequencies, and plot the amplitude
at each frequency, the plot is much simpler to understand.

Example 2:
When a signal (sinewave) is corrupt by large noise signal, in time-domain, it looks a mess. In
frequency-domain, we can clearly see the spectrum occupied by noise and hence design/use
proper filters to filter out the noise and get the original signal back.

Signals can be Continuous or Discrete. Continuous Time signals-CTS are called as analog
signals. Analog signal is represented as x(t). Discrete Time Signals-DTS are called as digital
signals. Digital signal is represented as x(n).
CLASSIFICATION OF SIGNALS
Signal is a function of time. Signal is a variation in some property of the medium used to convey
the data. We have variety of signals, Light, Sound, Electronic, Electromagnetic are a few of
them. Analog signal can be mathematically expressed as x(t) or f(t) and digital signal as x(n)
or f(n) or any other notation, meaning it is the function of time.

Signals can be classified according to various properties. Some of these classifications are:
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. Causal and non-causal signals
10. 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 are called as Multi-channel
signal. Example ECG signals. Multichannel 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. Image f(x,y) is a 2D signal, three-dimensional 3D signals are a time-variable image
f(x, y, t) or tomographic space-image data f(x, y, z).

3. Continuous time and Discrete time signals.

Sometimes a signal that starts out as an analog signal needs to be digitized (i.e., converted to a
digital signal). The process of digitizing is called sampling. Sampling converts continuous-
Time signal (CTS) signal to Discrete Time Signal (DTS). For example, if x(t) denotes
temperature as a function of time, and we are interested only in the temperature at 1 second
intervals, we can sample x at the times of interest as shown in Figure below:

Figure 5. Sampling an analog signal

Another example of sampling an image is shown in Figure below. An original image f(x; y) is
shown together with sampled versions of the image.

Figure 6. Sampling an image

In mathematics, a differential equation is an equation that relates one or more functions and
their derivatives. In applications, the functions generally represent physical quantities, the
derivatives represent their rates of change, and the differential equation defines a relationship
between the two.

differential equation of CT systems.

A difference equation is a discrete version of a differential equation and a differential

equation is a continuous version of a difference equation.

difference equation of DT systems

 4. Continuous valued and Discrete valued signals.
Discrete amplitude signals take on only a countable set of values.

5. Analog and digital signal

An analog signal exists throughout a continuous interval of time and takes on a continuous
range of values and sinusoidal signal which has both of these properties is an example.

A digital signal is a sequence of discrete symbols. If these symbols are zeros and ones, we call
them bits.
Note: DIGITAL signals are called as DISCRETE TIME & DISCRETE
AMPLITUDE/VALUED. By sampling the ANALOG signal at discrete instants of time we
obtain 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

Deterministic signals-There is no uncertainty with respect to its value at any time.
Example: sin(3t)

Random signals- There is uncertainty before its actual occurrence.

7.Periodic signal and Non-Periodic signal

Continuous time signal is periodic if and only if there exists a T0 > 0 such that
T0 is the period of x(t) in time.

A discrete-time signal is periodic if and only if there exists an integer N0 > 0 such that

N0 is the period of x[n] in sample spacings.

The smallest T0 or N0 is the fundamental period of the periodic signal.

If the signal does not satisfy above property called as Non-Periodic signals.

Periodic signals

Ø Shifting x(t) by 1 time unit results in the same signal.

Ø Common periodic signals are sines and cosines

Non-periodic signals
On the other hand, a non-periodic signal is any signal that does not repeat itself after any period
of time, however large that period may be. Most of the signals we come across in real life are
non-periodic. For example, speech is a non-periodic signal.

Example of a non-periodic signal

For an aperiodic signal, there is no such value of T.
8. Symmetrical (Even) and Anti-Symmetrical(odd) signal
A signal is called as symmetrical(even) if x(t) = x(-t) and if x(-t) = -x(t) then signal is anti-
symmetric (odd).

xe(n)= cos(ωn) and xo(n)= sin(ωn) are good examples of even & odd signals respectively.

Every signal can be represented in terms of even & odd signals.

Given an analog signal x(t), we can create even and odd functions as given below:

and when added together they create the original function.

Given a discrete signal g[n], we can create even and odd functions as given below:

Example:
Same type of decomposition applies for discrete-time signals.
The decomposition into even and odd components depends on the location of the origin.
Shifting the signal changes the decomposition.

9. Causal and non-causal signals

For Analog case examples,
Ø Causal signals are non-zero only for t ³ 0 (starts at t = 0, or later)

Ø Noncausal signals are non-zero for some t < 0 (starts before t = 0)

Ø Anti-causal signals are non-zero only for t £ 0 (goes backward in time from t = 0)

For Digital case examples,

Ø Causal signal are non-zero for n ³ 0 (starts at n = 0, or later)

Ø Noncausal signals are non-zero for some n < 0 (starts before n = 0)

 Ø Anti-causal signals are non-zero only for n £ 0 (goes backward in time from n = 0)

10. Energy signal and Power signal

Discrete time signals are also classified as finite energy or finite average power signals.

Signal Energy and Power:

If i(t) is the current through a resistor, then the energy dissipated in the resistor is

This is energy in Joules.

The signal energy for i(t) is defined as the energy dissipated in a 1W resistor

The signal energy for a (possibly complex) signal x(t) is

In most applications, this is not an actual energy (most signals aren't actually applied to 1W
resistor).
The average of the signal energy over time is the signal power

Properties of Energy and Power Signals:

The energy of a discrete time signal x(t) is given by

• An energy signal x(t) has zero power

• A power signal has infinite energy
CONCEPT OF FREQUENCY IN CONTINUOUS-TIME AND DISCRETE-TIME
SIGNALS
From Physics we know that,

(i) Trigonometric representation Continuous-Time Sinusoidal Signals

(ii) Trigonometric representation of Discrete-Time Sinusoidal Signals

Basic Block Diagram of A/D Conversion
1. Sampling of Analog Signals

But, as seen before, a discrete time sinusoid can be written as,

Sampling Theorem (Nyquist Sampling Rate):
A continuous time signal can be represented in its samples and can be recovered back
when sampling frequency Fs is greater than or equal to the twice the highest frequency
component of message signal.

If a sinusoidal signal is sampled with a high sampling rate, the original signal can be recovered
exactly by connecting the samples together in a smooth way.
Fs >= 2 fm (fm is the maximum analog signal frequency).
F=fm

Fs=1/T in uniform sampling

In contrast, if a sinusoidal signal is sampled with a low sampling rate, the samples may be too
infrequent to recover the original signal.
Types of Sampling
There are three types of sampling techniques:
i. Impulse sampling.
ii. Natural sampling.
iii. Flat Top sampling.

Impulse sampling is called ideal sampling. You cannot use this practically because pulse width
cannot be zero and the generation of impulse train is not possible practically.

(i) Impulse Sampling

Sampling of input signal x(t) can be obtained by multiplying x(t) with an impulse train δ(t) of
period Ts.

Proof: Consider a continuous time signal x(t). The spectrum of x(t) is band limited to fm Hz
i.e. the spectrum of x(t) is zero for |ω|>ωm. where ω =2pf.
The output of multiplier is a discrete signal called sampled signal which is represented with
y(t) in the following diagrams

Figure

(i)The time domain representation of the band-limited signal x(t)

(ii) multiplier system diagram
(iii) The time domain signal representation of the periodic impulse signal d(t) and
(iv) the obtained time domain signal sampled output y(t) from multiplier
(v) the frequency spectrum of x(t) after taking Fourier Transform is X(w) with bandwidth of w0
(d) the frequency spectrum of the obtained output sampled signal from multiplier y(t) is Y(w)
with a repetition of ws or fs because we multiplied the input x(t) with periodic impulse signal
d(t)

Before proceeding with the mathematical derivation of the process of sampling, let us see the
fundamentals.
 Any periodic signal x(t) may be expanded in a Fourier series. Let the period be T0. Since the
period is T0, we take the fundamental frequency to be ω0=2π/T0.
There are two common forms of the Fourier Series, "Trigonometric" and "Exponential."

Trigonometric

Exponential

Sampling process mathematical formulation:

Signals Sampling Theorem - Tutorialspoint
Signals Sampling Theorem - Tutorialspoint
19/02/20, 2:43 PM
19/02/20, 2:43 PM
Signals Sampling Theorem - Tutorialspoint 19/02/20, 2:43 PM
The process of sampling can be explained by the following mathematical expression:
SignalsSampled signal
Sampling Theorem y(t) = x(t). δ(t) . . . . . . (1)
- Tutorialspoint 19/02/20, 2:43 PM
Sampledsignal
Sampled signaly(t) y(t)= =x(t).
x(t).
δ(t)δ(t)
. . .. .. .. ..(1)
. . (1)
Sampled signal y(t) = x(t). δ(t) . . . . . . (1)
The trigonometric Fourier series representation of
Thetrigonometric
The trigonometricFourier
Fourier series
series representation
representation of of
δ (t) is given by
δδ (t)trigonometric
The (t)isisgiven Fourier series representation of
givenbyby
δ (t) is given by
δ(t) = a0 + Σ ∞ (a cos nω t + b sin nω t) . . . . . . (2)
n
δ(t)==aa0 0++ΣΣ
δ(t) ∞(a(acos
∞n=1
n=1 n n cos
n=1 t ss+t b+n bsin
nωnsω n
n sin
nωn ωss. t). . .. .. . (2)
s t) . . . (2)
δ(t) = a0 + Σ∞
n=1 (an cos nωs t + bn sin nωs t) . . . . . . (2)
Here 𝛿(t) is the periodic T Dirac delta distribution with period Ts
Where a0 = 1 T1 ∫T2 −T2 δ(t)dt =1 T11 δ(0) =
1 T T2 1
1 T1
Where aa
Where 0 0==T1s T∫ ∫2 −T2δ(t)dt
s −T ==
δ(t)dt 1Ts δ(0) = 1=
s δ(0)
Ts T
s
Where a0 = T ∫s −T δ(t)dt = Ts δ(0)
2 2
s =
Ts
T s
s
2

T
= 2 22 ∫T22 −TT22δ(t) 2 2
T
aaa
annnn=
==TT2s TT∫∫s −T
δ(t)coscos
∫ −T2δ(t)δ(t)
−T cos cos
nωdts dt
nnωωsnsdt = 2=
ωs=dt =
T
2 2 δ(0) cos nωs20 2=
δ(0) cos
δ(0)
T2 cos
δ(0) nsω0sn0=ω=s 0 T=
cos
nω T2
s s 2 2 T 2 T
2 T T
2 2

T
TT
2 2 ∫22 −TT2 δ(t) sin nωs t dt 2= 2 2 δ(0) sin nωs 0 = 0
bbb ==
bnnnn==Ts TT2∫ss −T 2δ(t)sin
∫22 −T2δ(t)
−T sinsin
δ(t) nnωωsnstω
tdtdtt=dt
=T= δ(0)
δ(0)
sTs
sin
T2s sin
δ(0)nωnsω0sn0=ω=
sin 000= 0
s Ts s
2
[ Aside..
Substitute above
Substitute
Substitute above values
abovevalues in
valuesin equation
in 2.2. 2.
equation
equation
Substitute above values in equation 2.
1 ∞ 2
∴∴ δ(t)
δ(t)===TT1sT11s+
∴ δ(t) ++Σ Σ∞
∞
Σn=1 (( T2s( cos
2 nωs t + 0)
coscos
nωnω
t +t 0)
+ 0)
∴ δ(t) = T + Σn=1 ( T2s cos nsωss t + 0)
s
∞
n=1
n=1 T s T
s s

Substitute δ(t) in equation 1.

Substitute
Substituteδ(t)
δ(t)ininequation
equation1. 1.
Substitute δ(t) in equation 1.
→ y(t) = x(t). δ(t)
→
→y(t)
y(t)==x(t).
x(t).δ(t)
δ(t)
 n=1
δ(t) = a0 + Σ∞
n=1 (an cos nωs t + bn sin nωs t) . . . . . . (2)

Where a = 1 ∫ 2 δ(t)dt 1= T1 δ(0)1 = T1

T

Where a0 0= T1 T∫s −T2 −T

δ(t)dt = T δ(0)s = T s
s 2 s s
2

T T
anan==T2s T∫2 −T∫
2
δ(t)
δ(t)
2
−T coscos
nωs ndtωs=dtT22= 2
δ(0)
T2
δ(0)
cos nωcos nωT2s 0 = T2
s0 =
s
2 2

T
2 T
bn = ∫2 −T2 δ(t) sin nωs t dt = T2 δ(0)2 sin nωs 0 = 0
bn = s T 2∫ −T2 δ(t)
T sin nωs t dt s = T δ(0) sin nωs 0 =0
s s
Aside ends…..] 2
Signals Sampling Theorem - Tutorialspoint 19/02/20, 2:43 PM

Substitute above values in equation 2.

Substitute abovey(t)
Sampled signal values in δ(t)
= x(t). equation
. . . . . .2.(1)
1 2
∴ δ(t) = Ts 1
+ Σ∞
n=1 ( T cos nωs t + 0)
Σ∞ ( T2s cos
s
∴ trigonometric
The δ(t) = Ts + n=1series
Fourier nωs t + 0)of
representation
δ (t) is given
Substitute δ(t) in
byequation 1.
Substitute δ(t) in equation 1.
→ y(t) = x(t). δ(t)
δ(t) = a0 + Σ∞
n=1 (an cos nωs t + bn sin nωs t) . . . . . . (2)
→ y(t) = x(t). δ(t)
= x(t)[ T1s T + Σ∞ 2
n=1 ( Ts cos nωs t)]
Where a0 = T1 ∫ −T21 δ(t)dt ∞
= T1s δ(0)
2 = T1s
s = x(t)[2Ts + Σn=1 ( Ts cos nωs t)]
1
= Ts
[x(t) + 2Σ∞
n=1 (cos nωs t)x(t)]
T
= T21 [x(t)
an = ∫ −T2 δ(t)
+ cos
2Σ ∞nωs(cos 2
dt =nω δ(0) cos nωs 0 = T2
T2s t)x(t)]
s
Ts 2 n=1
y(t) = T1 [x(t) + 2 cos ωs t. x(t) + 2 cos 2ωs t. x(t) + 2 cos 3ωs t. x(t) . . . . . . ]
s

y(t) is the sampled

T
version of the input continuous signal x(t) which can be denoted as
21
x(n). bn =transform
y(t)
Take Fourier ∫ −T2 δ(t)
TTs [x(t)
sin
+ both
on nsides.
2 cosωsωt dt x(t)T2s δ(0)
s t. = + 2 sin
cosn2ωω
s 0s t.
=x(t)
0 + 2 cos 3ωs t. x(t) . . . . . . ]
s 2

In Y(ω)
order =to see
1 the frequency
[X(ω) + X(ω − ωspectrum of y(t), we have to Transform the signal to frequency
s ) + X(ω + ωs ) + X(ω − 2ωs ) + X(ω + 2ωs )+ . . . ]
Ts
domain using
Substitute the
above
Take Fourier Frequency
values
transform Transform
in equation
on both sides. method call Fourier Transform.
2.

1 ∞ 2
∴ δ(t) = T1s + Σn=1 ( Ts cos nωs t + 0)
[Aside….
Y(ω) = T [X(ω) + X(ω − ωs ) + X(ω + ωs ) + X(ω − 2ωs ) + X(ω + 2ωs )+ . . . ]
s
Fourier Transform formula given below:
https://www.tutorialspoint.com/signals_and_systems/signals_sampling_theorem.htm Page 2 of 3

IfSubstitute
x(t) is a δ(t) time domain 1.
in equation signal, X(w) is the frequency domain conversion. The conversion is
done by applying Fourier transform.
→ y(t) = x(t). δ(t)
https://www.tutorialspoint.com/signals_and_systems/signals_sampling_theorem.htm Page 2 of 3

= x(t)[ T1 + Σ∞ 2
n=1 ( T cos nωs t)]
s s

1
= Ts
[x(t) + 2Σ∞
n=1 (cos nωs t)x(t)]

1
y(t) = [x(t) + 2 cos ωs t. x(t) + 2 cos 2ωs t. x(t) + 2 cos 3ωs t. x(t) . . . . . . ]
Aside ends] Ts

Take Fourier transform on both sides.

1
Y(ω) = Ts
[X(ω) + X(ω − ωs ) + X(ω + ωs ) + X(ω − 2ωs ) + X(ω + 2ωs )+ . . . ]

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Signals Sampling Theorem - Tutorialspoint 19/02/20, 2:43 PM

Signals Sampling Theorem - Tutorialspoint 19/02/20, 2:43

1
∴ Y(ω) = Ts
Σ∞
n=−∞ X(ω − nωs ) where n = 0, ±1, ±2, . . .
1 ∞
∴ Y(ω) = Σ
Ts n=−∞
X(ω − nωs ) where n = 0, ±1, ±2, . . .
To reconstruct x(t), you must recover input signal
spectrum X(ω) from sampled signal spectrum
Y(ω), which is x(t),
To reconstruct possible whenrecover
you must there input
is no signal
overlapping
spectrum between
X(ω) from the cycles of Y(ω).
sampled signal spectrum
To reconstruct
Y(ω), which
Possibility the
is original
of sampledpossible signal
frequencywhen x(t), there
spectrum youwith
must
is recover
no input signal spectrum X(ω) from
sampled conditions
overlapping
different signal spectrum
betweenis theY(ω).byThis
cycles
given of is
the possible using appropriate filters and only when there
Y(ω).
following
is no overlapping
diagrams: between the cycles of Y(ω). If they overlap it is not possible and it is called
Aliasing effect.
Possibility of sampled frequency spectrum with
different conditions is given by the following
Aliasing Effect
diagrams:
Possibility of sampled frequency spectrum with different conditions is given by the following
diagrams:
Signals Sampling Techniques - Tutorialspoint

Signals Sampli
Techniques

There are three types of sampling tec

Aliasing Effect Impulse sampling.
The overlapped region in case of under sampling Natural sampling.
represents aliasing effect, which can be removed
The overlapped region in case of under sampling represents aliasing effect,Flat
by Topcan
which sampling.
be
removedconsidering
by
Aliasing Effectfs >2fm
• considering fs >2fm
• By using
By using anti-aliasing
anti aliasing filters.
filters.
Impulse Sampling
The overlapped region in case of under sampling
represents aliasing effect, which can be removed Impulse sampling can be performed
(ii) Natural Sampling
by input signal x(t) with imp
Natural sampling is similar to impulse sampling, except the impulse train is replaced by pulse
considering
train of period T. i.e. fyou
>2f
multiply input signal x(t) to pulse train Σ∞
n=−∞ δ(t − nT) asof period 'T
s m
shown below:
By using anti aliasing filters. amplitude of impulse changes wit
amplitude of input signal x(t). Th
sampler is given by

Aliasing Effect
https://www.tutorialspoint.com/signals_and_systems/signals_sampling_theorem.htm Page 3 of 3

y(t) = x(t)× impulse train

Aliasing Effect
https://www.tutorialspoint.com/signals_and_systems/signals_sampling_theorem.htm ∞ Page 3
 n=−∞

The output of sampler is

y(t) = x(t) × pulse train

= x(t) × p(t)
Natural Sampling
= x(t) × Σ∞
n=−∞ P(t − nT) . . . . . . (1)

The exponential Fourier series representation of

(iii)Flat Top Sampling
p(t) can be given as
During transmission, noise is introduced at top of the transmission pulse which can be easily
p(t) = Σ∞
removed ejnωst .is. .in
if theFnpulse
n=−∞ . . .the
(2) form of flat top. Here, the top of the samples are flat i.e. they
have constant amplitude. Hence, it is called as flat top sampling or practical sampling. Flat
top sampling
∞
makesj2πnfuse
st
of sample and hold circuit.
= Σn=−∞ Fn e

T
1
Where Fn = T
∫ −T2 p(t)e−jnωs t dt
2

1
= TP
(nωs )
Comparison
Substitute Fn value in equation 2

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Exercises involving sampling

Example 1:

Tips:
xa(t) = 3cos w t (continuous time signal)
w = 2pF =100p =2*50* p
F=50
Fs >= 2 F
x(n) is got after sampling and it is a discrete time signal.

[Aside…
Relation between analog frequency and digital frequency is given in (1.4.5)

Aside ends]

!"p "p
= 3 cos 𝑛 = -3 cos 𝑛
# #
Example 2:

Tips:
Hz to KHz conversion
1 kHz = 1000 Hz

Take the highest frequency component F in the given signal.

w = 2pF =12000p =2*6000* p
Therefore, F=6000Hz= 6kHz is highest

So for the given signal Hz can be represented in KHz as,

2000 Hz= 2 kHz
6000Hz = 6kHz
12000 Hz = 12 kHz
Tips:
Fs sampling frequency is 2 * the maximum of the signal frequency F
Given Fs = 5kHz
5kHz = 2* F
Therefore, F=5 kHz/2 =2.5 kHz
So, the maximum signal frequency component that can be recovered by sampling at 5kHz is
2.5kHz.

sampled signal.

The sampled signal is x(n) with frequency f is got by diving the analog signal frequency F by the sampling
frequency Fs,

Exercises to do yourself…
2. Quantization of Continuous-Amplitude Signals
3. Coding of Quantized Samples

REPRESENTING DISCRETE TIME SIGNALS

There are three ways to represent discrete time signals.

1) Functional Representation

2) Tabular method of representation

3) Sequence Representation

n=0

4) Graphical representation
Another example for the four types of discrete signal representations:

SIMPLE MANIPULATION OF SIGNALS

Amplitude Scaling
The scaled signal a x(t) is x(t) multiplied by the constant ‘a’
a=2

The scaled signal a x[n] is x[n] multiplied by the constant a

Time dependence of a signal

1. Time scaling
A signal x(t) is scaled in time by multiplying the time variable by a positive constant b, to
produce x(bt). A positive factor of b either expands (0 < b < 1) or compresses (b > 1) the
signal in time.

The discrete-time sequence x[n] is compressed in time by multiplying the index n by an integer
k, to produce the time-scaled sequence x[nk].
Ø This extracts every kth sample of x[n].
Ø Intermediate samples are lost.
Ø The sequence is shorter.
Ø Called as down sampling, or decimation.
The discrete-time sequence x[n] is expanded in time by dividing the index n by an integer m,
to produce the time-scaled sequence x[n/m].
Ø This specifies every mth sample.
Ø The intermediate samples must be synthesized (set to zero, or interpolated).
Ø The sequence is longer.
Ø Called up sampling, or interpolation.

Example

2. Time reversal/ Reflection/Folding

Continuous time: replace t with -t, time reversed signal is x(-t)

Discrete time: replace n with -n, time reversed signal is x[-n].

In another representation,
 n=0

n=0
3. Time shift
For a continuous-time signal x(t), and a time t1 > 0,
Ø Replacing t with t - t1 gives a delayed signal x(t - t1)
Ø Replacing t with t + t1 gives an advanced signal x(t + t1)

For a discrete time signal x[n], and an integer n1 > 0

Ø x[n - n1] is a delayed signal.
Ø x[n + n1] is an advanced signal.
Ø The delay or advance is an integer number of sample times.

Example
A signal x(n) is graphically illustrated in Fig below. Show a graphical representation of the
signals x(n-3) and x(n+2).
Example in sequence form of representation

If n-k is delayed in time by k samples (Arrow (n=0) gets shifted on left hand side) and if n+k
(Arrow get shifted on right hand side)

4. Combinations
Ø Time scaling, shifting, and reversal can all be combined.
Ø Operation can be performed in any order, but care is required.
Ø This will cause confusion.

Example.
Show the graphical representation of the signal x(- n) and x(-n +2). where x(n) is the signal
illustrated in Fig below:
Example:
Show the graphical representation of the signal x(n - 2) and x(3n-2), where x (n) is
the signal illustrated in Fig below:

Example.
Show the graphical representation of the signal y(n) = x(2n), where x(n) is the signal illustrated
in Fig. below
Example:
Given the Signal below:

(i) Find x(n) - x(n + 1).

(ii) Find
Try these yourselves ....
1. Solve the following for the signal given below:

Find x(n-3),discrete-time
Important x(n+2), x(-n), x(-n+1),
signals If ax(-n-2)
continuous-time signal x(t) is sampled at T-second
intervals, the result is the sequence {x(nT)}. For convenience we shall drop the T and the braces
and use just x(n) to represent the sequence.
ELEMENTARY DISCRETE TIME SIGNALS
1) The unit sample sequence (discrete-time impulse, aka Kronecker delta)
1. The unit impulse signal/sequence (discrete-time impulse, aka Kronecker delta)
= 1, n = 0
0, n 0
Whereas is somewhat similar to the continuous-time impulse function the
Dirac delta we note that the magnitude of the discrete impulse is finite. Thus there are no
analytical difficulties in defining . It is convenient to interpret the delta function as follows:

(argument) = 1 when argument = 0

0

k)

1 1
analytical difficulties in defining . It is convenient to interpret the delta function as follows:

(argument) = 1 when argument = 0

0

k)

1 1

n n
1 0 1 1 0 1 k

2) The unit step sequence

u(n) = 1, n 0
0, n<0

u(argument) = 1, if argument 0
0, if argument < 0

u(n) u(n k)

1 1

n n
1 0 1 1 0 k

a) The discrete delta function can be expressed as the first difference of the unit step function:

= u(n) u(n 1)

b) The sum from to n of the function gives the unit-step:

16 of 77
2. Unit step signal:

Unit step signal represented as a Unit impulse sequence

Unit impulse signal represented using unit step response signals

3. Unit ramp signal:

4.Exponential signals
The exponential signal is a sequence of the form

If the parameter a is real, then x(n) is a real signal.

When the parameter a is complex valued, it can be expressed as

Hence, we can express x(n) as

Definition of discrete time signal

A discrete-time signal is a sequence, that is, a function defined on the positive and negative
integers.
The sequence x(n) = xR(n) + j xI(n) is a complex (valued) sequence if xI(n) is not zero for all
n. Otherwise, it is a real (valued) sequence.

Examples of discrete-time signals represented in functional form are given below.

x1(n) = 2 cos 3n
x2(n) = 3 sin (0.2πn)

Definition of digital signal

A discrete-time signal whose values are from a finite set is called a digital signal.
Try Yourself…