DSP Chapter One

Digital Signal Processing
Chapter-1
Discrete-Time Signals and Systems

Overview
❖Introduction to digital signal processing
❖Time domain Sampling
✓ Sampling and Reconstruction
✓ Quantization and Encoding
✓Analog to Digital Converters(ADC) and Digital
to Analog Converters (DAC)
❖Discrete time systems
2
Introduction to digital signal
processing
3
Analog vs. Digital Signal Processing
Analog input Signal x(t) Analog output Signal y(t)

Analog
Signal Processor
Analog Signal Processing
Analog input Analog output

Signal x(t) Signal y(t)
Analog-to-Digital Digital D/A
converter Signal Processor converter
Digital Signal Processing

4
Typical Digital Signal Processing System
It consists of
• an analog filter called (anti-imaging) filter,
• an analog-to-digital conversion (ADC) unit,
• a digital signal (DS) processor,
• a digital-to-analog conversion (DAC) unit,
• and an analog filter called reconstruction (anti-image) filter.
5
Typical Digital Signal Processing System
6
Digital vs analog processing Digital
7
DSP…
8
Applications
9
Time domain Sampling
10
A/D & D/A Conversion
11
Analog to Digital (A/D) Conversion
• Most signals of practical interest are analog in nature

Examples: Voice, Video, RADAR signals, Transducer/Sensor
output, Biological signals etc
• So in order to utilize those benefits, we need to convert our

analog signals into digital
• This process is called Analog-to-Digital (A/D) conversion
12
Analog to Digital Conversion
A/D conversion can be viewed as a three step process
13
A/D conversion can be viewed as a three step process
14
Sample & Hold (Sampler)
• Analog signal is continuous in time and continuous in

amplitude.
• It means that it carries infinite information of time and infinite

information of amplitude.
• Analog (continuous-time) signal has some value defined at

every time instant, so it has infinite number of sample points.
15
• It is impossible to digitize an infinite number of points.
• The infinite points cannot be processed by the digital signal

(DS) processor or computer, since they require an infinite
amount of memory and infinite amount of processing power
for computations.
• Sampling is the process to reduce the time information or

sample points.
16
• The first essential step in analog-to-digital (A/D) conversion is

to sample an analog signal.
• This step is performed by a sample and hold circuit, which

samples at regular intervals called sampling intervals.
• Sampling can take samples at a fixed time interval.
• The length of the sampling interval is the same as the

sampling period, and the reciprocal of the sampling period is
the sampling frequency fs.
17
• After a brief acquisition time, during which a sample is

acquired, the sample and hold circuit holds the sample steady
for the remainder of the sampling interval.
• The hold time is needed to allow time for an A/D converter to
generate a digital code that best corresponds to the analog
sample.
• If x(t) is the input to the sampler, the output is x(nT), where T
is called the sampling interval or sampling period.
• After the sampling, the signal is called “discrete time
continuous signal” which is discrete in time and continuous in
amplitude.
18
Figure below shows an analog (continuous-time) signal (solid
line) defined at every point over the time axis (horizontal line)
and amplitude axis (vertical line).
Hence, the analog signal contains an infinite number of points.
19
• Each sample maintains its voltage level during the sampling
interval 𝑻 to give the ADC enough time to convert it.
• This process is called sample and hold.
20
Nyquist–Shannon Sampling Theorem
The sampling theorem guarantees that an analogue signal can be

perfectly recovered as long as the sampling rate is at least twice
as large as the highest-frequency (Fmax) component of the
analogue signal to be sampled.
21
22
Examples
23
Example: For the following analog signal, find the Nyquist sampling
rate, also determine the digital signal frequency and the digital signal
24
Example: Find the sampling frequency of the following signal.
So sampling frequency should be
25
Exercise
Determine the Nyquist sampling rate of a signal

x(t) = 3sin(5000t + 17o)
26
Aliasing
• When the minimum sampling rate is not respected, distortion
called aliasing occurs.
• Aliasing causes high frequency signals to appear as lower

frequency signals.
• To be sure aliasing will not occur, sampling is always preceded

by low pass filtering.
• The low pass filter, called the anti-aliasing filter, removes all
frequencies above half the selected sampling rate.
27
Aliasing
• Figure illustrates sampling a 40 Hz sinusoid
• The sampling interval between sample points is T = 0.01 second,
and the sampling rate is thus fs = 100 Hz.
• The sampling theorem condition is satisfied
28
Aliasing
• Figure illustrates sampling a 90 Hz sinusoid
• The sampling interval between sample points is T = 0.01 second,
and the sampling rate is thus fs = 100 Hz.
• The sampling theorem condition is not satisfied
29
Aliasing
30
Anti Aliasing Filter
• A signal with no frequency component above a certain
maximum frequency is known as a band-limited signal.
• In our case we want to have a signal band-limited to ½ Fs.
• Some times higher frequency components (both harmonics

and noise) are added to the analog signal (practical signals are
not band-limited).
• In order to keep analog signal band-limited, we need a filter,

usually a low pass that stops all frequencies above ½ Fs.
• This is called an “Anti-Aliasing” filter. 31

Anti Aliasing Filter
• Anti-aliasing filters are analog filters.
• They process the signal before it is sampled.
• In most cases, they are also low-pass filters unless band-pass

sampling techniques are used.
32
Under Sampling
• If the sampling rate is lower than the required Nyquist rate, that
is fS < 2W, it is called under sampling.
• In under sampling images of high frequency signals incorrectly

appear in the baseband (or Nyquist range) due to aliasing.
33
Sampling of Band Limited Signals
Signals whose frequencies are restricted to a narrow band of

high frequencies can be sampled at a rate similar to twice the
Bandwidth (BW) instead of twice the maximum frequency.
Fs ≥ 2BW
34
• While this under-sampling is normally avoided, it can be
exploited.
• For example, in the case of band limited signals all of the
important signal characteristics can be deduced from the copy
of the spectrum that appears in the baseband through
sampling.
• Depending on the relationship between the signal frequencies
and the sampling rate, spectral inversion may cause the shape
of the spectrum in the baseband to be inverted from the true
spectrum of the signal.
35
Figure: Signal recovered

From Nyquist range are
Base band versions of the
Original signal. Sampling rate is
Important to make sure no aliasing
and spectral inversion occurs.
(a) Fs = 80 kHz, signal spectrum

is Inverted in the baseband.
(b) Fs = 100 kHz, the lowest

Frequencies In the signal alias
to the highest frequencies.
(c) Fs = 120 kHz, No spectral

36
Inversion occurs.
Over Sampling
• Oversampling is defined as sampling above the minimum
Nyquist rate, that is, fS > > 2fmax.
• Oversampling is useful because it creates space in the

spectrum that can reduce the demands on the analog anti-
aliasing filter.
37
Over Sampling
• In the example below, 2x oversampling means that a low order analog filter is
adequate to keep important signal information intact after sampling.
• After sampling, higher order digital filter can be used to extract the information.
38
Over Sampling
• The ideal filter has a flat pass-band and the cut-off is very sharp,
since the cut-off frequency of this filter is half of that of the
sampling frequency, the resulting replicated spectrum of the
sampled signal do not overlap each other. Thus no aliasing
occurs.
• Practical low-pass filters cannot achieve the ideal
characteristics.
• Firstly, this would mean that we have to sample the filtered
signals at a rate that is higher than the Nyquist rate to
compensate for the transition band of the filter
39
Spectra of Sampled signals
Figure: Signal ‘s Spectra

(i) Over sampled
(ii) Nyquest Rate
(iii) Under Sampled
40
Sampling Low Pass Signals
41
Exercise
Exercise-1: If the 20 kHz signal is under-sampled at 30 kHz, find the aliased
frequency of the signal.
Exercise-2: A voice signal is sampled at 8000 samples per second.

i. What is the time between samples?
ii. What is the maximum frequency that will be recovered from the signal?
Exercise-3: An analog Electromyogram (EMG) signal contains useful

frequencies up to 3000 Hz.
i. Determine the minimum required sampling rate to avoid aliasing.
ii. Suppose that we sample this signal at a rate of 6500 samples/s. what is
the highest frequency that can be represented uniquely at this sampling
rate?
42
Exercise
Exercise-4: Humans can hear sounds at frequencies between 0 and 20 kHz.
What minimum sampling rate should be chosen to permit perfect recovery
from samples?
Exercise-5: An ECG signal is sampled at 250 samples per second.

i. What is the time between samples?
ii. What is the maximum frequency that will be recovered from the signal?
Exercise-6: An ultrasound signal ranging in frequency from 900 kHz to 900.5

kHz is under-sampled at 200 kHz. If a 200 Hz target appears in the baseband,
what is the actual frequency of the target?
43
Quantizer
• After the sampling, the discrete time continuous signal still

carry infinite information (can take any value) in terms of
amplitude.
• Quantization is the process to reduce infinite information of

the amplitude.
• Quantizer do the conversion of discrete time continuous

valued signal into a discrete-time discrete-value signal.
• The value of each signal sample is represented by a value

selected from a finite set of possible values.
44
Quantizer
• The A/D converter chooses a quantization level for each

analog sample.
• Number of levels of quantizer is equal to L = 2N
• An N-bit converter chooses among 2N possible quantization

levels.
• So 3 bit converter has 8 quantization levels, and 4 bit

converter has 16 quantization levels.
45
The quantization step size or resolution is calculated as:
Δ = Q = R/2N
where
R is the full scale range of the analog signal (i.e. Ymax - Ymin)
N is the number of bits used by the converter
• Resolution of a quantizer is the distance between two

successive quantization levels
• More quantization levels, a better resolution!
• What's the downside of more quantization levels?
The strength of the signal compared to that of the quantization

errors is measured by dynamic range and signal-to-noise ratio.
46
4-bit Quantizer
47
4-bit Quantizer
48
Quantization Error
• The error caused by representing a continuous-valued signal
(infinite set) by a finite set of discrete-valued levels.
• The larger the number of quantization levels, the smaller the

quantization errors.
• The quantization error is calculated as the difference between

the quantized level and the true sample level.
• Most quantization errors are limited in size to half a

quantization step Q or Δ .
49
Quantization Error
• Suppose a quantizer operation given by Q(.) is performed on
continuous-valued samples x[n] is given by Q(x[n], xq[n]), then
the quantization error is given by
50
• Lets consider the signal which is to be quantized.
In the figure, we can see that there is a difference between the

original signal (Blue Line) and the quantized signal (Red Lines).
This is the error produced while quantization
51
Quantization error can be reduced, however, if the number of
quantization levels is increased as illustrated in the figure
52
Quantization of unipolar data (maximum error = full step)
53
Quantization of unipolar data (maximum error = half step)
54
Example: Analog pressures are recorded using a pressure transducer as
voltages between 0 and 3 V. The signal must be quantized using a 3-bit
digital code. Indicate how the analog voltages will be covered to digital
values.
The quantization step size is

Q = 3 V/23 = 0.375 V
The half of quantization step is

0.1875 V
55
Quantization of bipolar data (maximum error = half step)
56
Three-bit A/D Conversion
57
Dynamic Range
• Quantization errors can be determined by the quantization
step.
• Quantization errors can be reduced by increasing the number
of bits used to represent each sample.
• Unfortunately these errors can not be entirely eliminated and
their combined effect is called quantization noise.
• The dynamic range of the quantizer is the number of levels it

can distinguish in noise.
• It is a function of the range of signal values and the range of
error values, and is expressed in decibels, dB.
𝑅
• 𝐷𝑦𝑛𝑎𝑚𝑖𝑐 𝑅𝑎𝑛𝑔𝑒 = 20𝑙𝑜𝑔2
𝑄
58
Signal-to-Quantization-Noise Ratio
• Provides the ratio of the signal power to the quantization

noise (or quantization error),
• Mathematically the signal-to-quantization noise ratio (SQNR),
where
• Px= Power of the signal ‘x’ (before quantization)
• Pq= Power of the error signal ‘xq’
59
60
2 bit Flash ADC
61
Digital-to-Analog (D/A) Conversion
Block Diagram of D/A Conversion
62
• Once digital signal processing is complete, digital-to-analog

(D/A) conversion must occur.
• This process begins by converting each digital code into an
analog voltage that is proportional in size to the number
represented by the code.
• This voltage is held steady through zero order hold until the next
code is available, one sampling interval later.
• This creates a staircase-like signal that contains frequencies
above W Hz.
• These signals are removed with a smoothing analog low pass
filter, the last step in D/A conversion.
63
• In the frequency domain, the high frequency elements

present in the zero order hold signal appear as images, copies
of the original signal spectrum situated around integer
multiples of the sampling frequency.
• The smoothing analog filter removes these images and so is

given the name of Anti-Imaging Filter.
• Only the frequencies in the baseband, between 0 and fS/2 Hz,

remain.
64
Three bit D/A Conversion
65
Comparing Signals in the A/D & D/A Chain
66
Comparing Signals in the A/D & D/A Chain
67
Discrete-Time Signals and Systems
68
Discrete-Time Signals
• A signal defined only for discrete values of time is called a
discrete-time (DT) signal or simply a discrete signal
• Discrete signal can be obtained by taking samples of an analog

signal at discrete instants of time
• Digital signal is a discrete-time signal whose values are

represented by digits
69
Discrete-Time Signals
• Figure (a): CT Signal
• Figure (b): DT Signal
• Examples of DT signals in nature:

• DNA base sequence
• Number of students in a class
• Population of the nth generation of certain species 70
Digital Functions
The basic digital functions (signal or sequence) are
• Unit Impulse Function

• Unit Step Function
• Unit Ramp Function
• Power Function
• Exponential Function
• Sine Function
71
Unit Impulse Function
The unit impulse function or unit sample sequence is defined as
the sequence with values
72
The unit impulse function δ[n] has an amplitude of zero at all
samples except n = 0, where it has the value 1.
Every digital signal can be written as a sum of impulse functions,

using the amplitude at each sample.
73
Example:
x[n] = 4δ[n] - 2δ[n-1] + 3δ[n-2] - δ[n-3]
74
Unit Step Function
The unit step function or unit step sequence is defined as the
sequence with values
75
Unit Step Function
Example:
x[n] = u[n] - u[n - 3]
76
Unit Step Function
Example: A digital signal is described as x[n] = 4(u[n] - u[n - 1]).
Write the function that describes x[n-3].
Answer
Substituting n = (n – 3) gives
x[n-3] = 4(u[n-3] - u[n - 4])
77
Unit Ramp Function
• The unit-ramp function is defined as
78
Power Function
• Power functions take the form:
x[n] = Aα βn
• In the special case where α = e, such functions are called
exponential functions.
• When β is positive, the function grows.
• When β is negative the function decays.
• When α is negative, the signal samples alternate positive and
negative.
• The value of A is determine the magnitude/amplitude/value
of the function when n = 0
79
Power Function
• Example: x[n] = (-0.6)n
80
Exponential Function
• Exponential functions take the form:
x[n] = Ae βn
• Where e = 2.71828
• When β is positive, the function grows.
• When β is negative the function decays.
• When α is negative, the signal samples alternate positive and
negative.
• The value of A is determine the magnitude/amplitude/value
of the function when n = 0
81
Exponential Function
• Example: x[n] = e-0.5n
82
Complex Exponential Function
• A digital signal of the form
x[n] = Aejβn
is called a complex exponential function.
• For all n, samples of this signal lie in the complex plane on a

circle with radius A.
• By Euler’s identity, a complex exponential may be expressed

as a rectangular-form complex number
eβn= cosβn + jsinβn
83
Sinusoidal Sequence
The sinusoidal functions take the form
x[n] = Asin(nW + q) or x[n] = Acos(nW + q) for all n with real A.
where W is a digital frequency in radians and q is a phase shift.
84
Sinusoidal Sequence
Example: x[n] = 3sin(nπ/8)
85
Linear System
• A linear system is illustrated in the figure.

• The linear system obeys the superposition principle.
• 𝑦1(𝑛) is the system output using an input 𝑥1(𝑛)
• 𝑦2(𝑛) the system output with an input 𝑥2(𝑛)
86
Linear System
• The Linear system output due to the weighted sum inputs

∝𝑥1(𝑛) + 𝛽𝑥2(𝑛) is equal to the same weighted sum of the
individual outputs obtained from their corresponding inputs,
that is, 𝑦(𝑛) = ∝𝑦1(𝑛) + 𝛽𝑦2(𝑛), where ∝ and 𝛽 are constants.
87
Linear System
Example: A digital amplifier is represented by 𝑦(𝑛) = 10𝑥(𝑛), the

input is multiplied by 10 to generate the output.
88
Linear System
89
Linear System
90
Time-Invariant System
A time-invariant system is illustrated in the figure.
91
92
Example: Determine whether the linear system y(𝑛) = 2𝑥(𝑛) − 5

is time invariant.
93
94
Example: Determine whether the linear system y(𝑛) = 2𝑥(3𝑛)

is time invariant.
N:B A Linear Time Invariant (LTI) System obey both the

characteristics of both linearity and Time Invariance.
95
Causal System
96
Causal System
Example: Determine whether the following systems are causal or not.
Solution
1) Causal
2) Non-causal
97
Summary
98
Summary
99
End of Chapter-1
100

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Digital Signal Processing

Discrete-Time Signals and Systems

Analog input Signal x(t) Analog output Signal y(t)

Analog Signal Processing

Analog input Analog output

Digital Signal Processing

• Most signals of practical interest are analog in nature

• So in order to utilize those benefits, we need to convert our

• This process is called Analog-to-Digital (A/D) conversion

• Analog signal is continuous in time and continuous in

• It means that it carries infinite information of time and infinite

• Analog (continuous-time) signal has some value defined at

• It is impossible to digitize an infinite number of points.

• The infinite points cannot be processed by the digital signal

• Sampling is the process to reduce the time information or

• The first essential step in analog-to-digital (A/D) conversion is

• This step is performed by a sample and hold circuit, which

• Sampling can take samples at a fixed time interval.

• The length of the sampling interval is the same as the

• After a brief acquisition time, during which a sample is

The sampling theorem guarantees that an analogue signal can be

Example: Find the sampling frequency of the following signal.

So sampling frequency should be

Determine the Nyquist sampling rate of a signal

• Aliasing causes high frequency signals to appear as lower

• To be sure aliasing will not occur, sampling is always preceded

• In our case we want to have a signal band-limited to ½ Fs.

• Some times higher frequency components (both harmonics

• In order to keep analog signal band-limited, we need a filter,

• This is called an “Anti-Aliasing” filter. 31

• They process the signal before it is sampled.

• In most cases, they are also low-pass filters unless band-pass

• In under sampling images of high frequency signals incorrectly

Signals whose frequencies are restricted to a narrow band of

Figure: Signal recovered

(a) Fs = 80 kHz, signal spectrum

(b) Fs = 100 kHz, the lowest

(c) Fs = 120 kHz, No spectral

• Oversampling is useful because it creates space in the

Figure: Signal ‘s Spectra

Exercise-2: A voice signal is sampled at 8000 samples per second.

Exercise-3: An analog Electromyogram (EMG) signal contains useful

Exercise-5: An ECG signal is sampled at 250 samples per second.

Exercise-6: An ultrasound signal ranging in frequency from 900 kHz to 900.5

• After the sampling, the discrete time continuous signal still

• Quantization is the process to reduce infinite information of

• Quantizer do the conversion of discrete time continuous

• The value of each signal sample is represented by a value

• The A/D converter chooses a quantization level for each

• Number of levels of quantizer is equal to L = 2N

• An N-bit converter chooses among 2N possible quantization

• So 3 bit converter has 8 quantization levels, and 4 bit

• Resolution of a quantizer is the distance between two

The strength of the signal compared to that of the quantization

• The larger the number of quantization levels, the smaller the

• The quantization error is calculated as the difference between

• Most quantization errors are limited in size to half a

In the figure, we can see that there is a difference between the

The quantization step size is