Sana’a Community College

Digital Signal Processing

Dr. Ibrahim Ismail Al-kebsi

[email protected]
[email protected]
2024
 Text Book
• Li Tan, Digital Signal
Processing
Fundamentals and
Applications,
Elsevier Inc, 2008.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 2

 REFERENCES

J. G. Proakis, D. G. Manolakis,

Digital Signal Processing; Principles,
algorithms and applications,
Prentice Hall International Inc, New
Delhi, 2003.

B. A. Shenoi, Introduction to Digital

Signal Processing and Filter Design,
New Jersey , A John Wiley & Sons,
INC., 2006.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 3
 Assessment
• Test 1 15 %
• Test 2 15 %

• Assignments and LOs 30 %

• Attendance 10 %

• Final written Exam 30 %

Communications System Sana'a Community College 4
 Chapter One (section I)
DIGITAL SIGNAL PROCESSING (DSP)
DEFINITION

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 5

 What is Digital Signal Processing?
• Digital: operating by the use of discrete signals to
represent data in the form of numbers.

• Signal: a parameter (electrical quantity or effect) that

can be varied in such a way as to convey information.

• Processing: a series operations performed according

to programmed instructions.

changing or analyzing information, which is

measured as discrete sequences of numbers
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 6
 Embedded system

• An embedded system is a computer system

designed for specific control functions within a
larger system.

• It is embedded as part of a complete device often

including hardware and mechanical parts.

• Embedded systems contain processing cores that

are either microcontrollers or digital signal
processors.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 7

 Embedded system (2)

The internals of an ADSL modem/router.

A modern example of an embedded system.
Labelled parts include a microprocessor (4), RAM (6).

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 8

 Section II
BASIC CONCEPTS OF DIGITAL SIGNAL
PROCESSING

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 9

 Basic Concepts of DSP

• 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 filter.

Figure 1.1 A digital signal processing scheme

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 10

 Basic Concepts of DSP (2)
• 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).
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 11
 Basic Concepts of DSP (3)

• 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.
• 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.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 12
 Basic Concepts of DSP (4)
• 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
Dr. IBRAHIMused.
AL-KEBSI (SCC) Digital Signal Processing 13
 Basic Concepts of DSP (5)

• 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 filter for real-world
applications.

Figure 1.1 A digital signal processing scheme

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 14
 Basic Concepts of DSP (6)
• 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
Dr. IBRAHIMapplications
AL-KEBSI (SCC) Digital Signal Processing 15
 Basic Concepts of DSP (7)
• However, as shown in Figure 1.1, DSP systems still
require minimum analog processing such as
– the anti-aliasing, and
– reconstruction filters.
• They are necessary for:
– converting real-world information into digital form, and
– digital form back into real-world information.

Figure 1.1 A digital signal processing scheme

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 16
 Digital Filter

• A digital filter is a system that performs mathematical

operations on a sampled, discrete-time signal to reduce
or enhance certain aspects of that signal.
– This is in contrast to the other major type of electronic filter,
– the analog filter, which is an electronic circuit operating
on continuous-time analog signals.
• An analog signal may be processed by a digital filter by
– first being digitized and represented as a sequence of
numbers,
– then manipulated mathematically, and then reconstructed
as a new analog signal.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 17

 Digital Filter

• A low-pass filter
– is a filter that passes low frequency signals but attenuates (reduces the amplitude of)
signals with frequencies that are higher than the cut off frequency.

• A high-pass filter,
– is a filter that passes signals containing high frequencies, but attenuates frequencies
lower than the filter's cut off frequency.

• A band-pass filter
– is a device that passes frequencies within a certain range and rejects (attenuates)
frequencies outside that range.

• A band-stop filter or band-rejection filter

– is a filter that passes most frequencies unaltered, but attenuates those in a specific
range to very low levels.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 18

 Digital Filtering (4)

FIGURE 1.3 (Top) Digitized noisy signal, (Bottom) Clean digital signal using the
digital lowpass filter.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 19
 Section III
SAMPLING OF CONTINUOUS SIGNAL

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 20

 Sampling of Continuous Signal

• Figure 1.1 describes a simplified block diagram of a

digital signal processing (DSP) system.
• The analog filter processes the analog input to obtain
the band-limited signal,
– The band-limited signal is sent to the ADC unit.
– The ADC unit samples the analog signal, quantizes the sampled signal,
and encodes the quantized signal levels to the digital signal.

Figure 1.1 A digital signal processing scheme

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 21
 Sampling of Continuous Signal (2)
• Here we first develop concepts of sampling processing in
time domain.
• Figure 1.2 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.

Figure 1.2 Display of

the analog
(continuous) signal
and display of digital
samples
versus the sampling
time instants.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 22

 Sampling of Continuous Signal (3)
• It is impossible to digitize an infinite number of
points.
• Furthermore, the infinite points are not appropriate
to be processed by the digital signal (DS) processor
or computer,
– since they require infinite amount of memory and infinite
amount of processing power for computations.
• Sampling can solve such a problem by taking samples
at the fixed time interval, as shown in Figure 1.2 and
Figure 1.3,
– where the time T represents the sampling interval or
Dr. IBRAHIMsampling
AL-KEBSI (SCC) period in seconds.
Digital Signal Processing 23
 Sampling of Continuous Signal (4)
• As shown in Figure 1.3,
– each sample maintains its voltage level during the
sampling interval T to give the ADC enough time
to convert it.

Figure 1.3
Sample-and-
hold analog
voltage for
ADC.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 24

 Sampling of Continuous Signal (5)
• This process is called sample and hold.
• Since there exists one amplitude level for each
sampling interval,
– we can sketch each sample amplitude level at its
corresponding sampling time instant shown in
Figure 1.2,
– where 14 samples at their sampling time instants
are plotted,
– each using a vertical bar with a solid circle at its
top.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 25

 Sampling of Continuous Signal (6)
• For a given sampling interval T, which is
defined as the time span between two sample
points, the sampling rate is therefore given by:
1
fs  samples per second (Hz)
T
• For example, if a sampling period is T = 125
microseconds,
– the sampling rate is determined as
1
fs  8000 samples per second (Hz)
125μs
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 26
 Sampling of Continuous Signal (7)

• After the analog signal is sampled,

– we obtain the sampled signal whose amplitude
values are taken at the sampling instants,
– thus the processor is able to handle the sample
points.
– Next, we have to ensure that samples are collected
at a rate high enough that the original analog signal
can be reconstructed or recovered later.
– In other words, we are looking for a minimum
sampling rate to acquire a complete reconstruction
of the analog signal from its sampled version.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 27

 Sampling of Continuous Signal (8)

• If an analog signal is not appropriately sampled,

aliasing will occur, which causes unwanted signals in
the desired frequency band.
• The sampling theorem guarantees that an analog
signal can be in theory perfectly recovered as long as
– the sampling rate is at least twice as large as the highest-
frequency component of the analog signal to be sampled.

• The condition is described as

f s 2 f max
where fmax is the maximum-frequency component of the analog
signal to be sampled.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 28
 Sampling of Continuous Signal (9)
For example,
• To sample a speech signal containing
frequencies up to 4 kHz,
– the minimum sampling rate is chosen to be at
least 8 kHz, or 8,000 samples per second;

• To sample an audio signal possessing

frequencies up to 20 kHz,
– at least 40,000 samples per second, or 40 kHz, of
the audio signal are required.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 29

 Sampling of Continuous Signal (10)
• Figure 1.4 illustrates sampling of two sinusoids,
– where the sampling interval between sample points is
T = 0.01 second, thus the sampling rate is fs = 100 Hz.

Figure 1.4 Plots of

the appropriately
sampled signals and
nonappropriately
sampled
(aliased) signals.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 30

 Sampling of Continuous Signal (11)

• The first plot in the figure displays a sine wave with

a frequency of 40 Hz and its sampled amplitudes.
• The sampling theorem condition is satisfied, since 2
fmax = 80 Hz < fs.

• The sampled amplitudes are labeled using the

circles shown in the first plot.
• We notice that the 40-Hz signal is adequately
sampled,
– since the sampled values clearly come from the analog
version of the 40-Hz sine wave.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 31
 Sampling of Continuous Signal (12)

• However, as shown in the second plot, the

sine wave with a frequency of 90 Hz is
sampled at 100 Hz.
• Since the sampling rate of 100 Hz is relatively
low compared with the 90-Hz sine wave,
• the signal is undersampled due to 2 fmax = 180
Hz > fs.
•

• Hence, the condition of the sampling theorem

is not satisfied.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 32
 Sampling of Continuous Signal (13)

• Based on the sample amplitudes labeled with the

circles in the second plot,
• we cannot tell whether the sampled signal comes from
– sampling a 90-Hz sine wave (plotted using the solid line) or
– from sampling a 10-Hz sine wave (plotted using the dot-
dash line).
– They are not distinguishable.
– Thus they are aliases of each other.

• We call the 10-Hz sine wave the aliasing noise in this

case,
– since the sampled amplitudes actually come from sampling
the 90-Hz sine wave.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 33
 Sampling of Continuous Signal (14)

• Now let us develop the sampling theorem in

frequency domain,
– that is, the minimum sampling rate requirement for an
analog signal.

• In practice this can help us design the anti-aliasing

filter to be applied before digitizing.
– a lowpass filter that will reject high frequencies that
cause aliasing.

• The anti-image filter to be applied after the DAC.

– a reconstruction lowpass filter that will smooth the
recovered sample-and-hold voltage levels to an analog
Dr. IBRAHIMsignal.
AL-KEBSI (SCC) Digital Signal Processing 34
 Sampling of Continuous Signal (15)

• Figure 1.5 depicts the sampled signal xs(t) obtained by sampling the
continuous signal x(t) at a sampling rate of fs samples per second.

Figure 1.5 The simplified

sampling process

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 35

 Sampling of Continuous Signal (16)
• Mathematically, this process can be written as the product of the
continuous signal and the sampling pulses (pulse train):
xs (t )  x(t ) p (t ) (1.1)
where p(t) is the pulse train with a period T=1/ fs.

• From spectral analysis,

– the original spectrum (frequency components) X( f ) and the
sampled signal spectrum Xs( f ) in terms of Hz are related as:
1 
X s ( f )   X ( f  nf s ) (1.2)
T n 

where X( f ) is assumed to be the original baseband spectrum,

while Xs( f ) is its sampled signal spectrum, consisting of the original
Dr.baseband
IBRAHIM AL-KEBSIspectrum
(SCC) X( f ) and
Digitalits replicas
Signal Processing X( f ± nf ). 36
s
Sampling of Continuous Signal (17)
• Expanding Equation (1.2) leads to the
sampled signal spectrum in Equation (1.3):
1 1 1
X s ( f ) ...  X ( f  f s )  X ( f )  X ( f  f s )  ... (1.3)
T T T
• Equation (1.3) indicates that:
– the sampled signal spectrum is the sum of
the scaled original spectrum and copies of its
shifted versions, called replicas.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 37

 Sampling of Continuous Signal (18)

• The sketch of Equation (1.3) is given in Figure 1.6,

– where three possible sketches are classified.

• Given the original signal spectrum X( f ) plotted in

Figure 1.6 (a),
– the sampled signal spectrum according to Equation (1.3) is
plotted in1Figure 1.6
1 (b), where
1 the replicas:
X ( f ), X ( f  f s ), X ( f  f s ) , ...
T T T
Have separations between them.
– Figure 1.6 (c) shows that the baseband spectrum and its
replicas, are just connected
– finally, in Figure1.6 (d), the original spectrum and its
replicas are overlapped
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 38
 a

Figure 1.6 Plots of the sampled signal spectrum.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 39
 Sampling of Continuous Signal (19)
• From Figure 1.6 it is clear that
– the sampled signal spectrum consists of the scaled baseband spectrum
centered at the origin and
– its replicas centered at the frequencies of nfs (multiples of the sampling
rate) for each of n =1,2,3, . . . .

• If applying a lowpass reconstruction filter to obtain exact

reconstruction of the original signal spectrum, the following
condition must be satisfied:
f s  f max  f max (1.4)
solving equation (1.4) gives
f s 2 f max (1.5)
• In terms of frequency in radian per second, Eq. (1.5) is equivalent to:
s 2max (1.6)
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 40
 Sampling of Continuous Signal (20)
• This fundamental conclusion is well known as
the Nyquist–Shannon sampling theorem,
which is formally described below:
For a uniformly sampled DSP system, an
analog signal can be perfectly recovered as
long as the sampling rate is at least twice as
large as the highest-frequency component of
the analog signal to be sampled.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 41

 Sampling of Continuous Signal (21)

• We summarize two key points here.

1. Sampling theorem establishes a minimum
sampling rate for a given bandlimited
analog signal with the highest-frequency
component fmax.

• If the sampling rate satisfies Equation (1.5),

– then the analog signal can be recovered via its
sampled values using the lowpass filter, as
described in Figure 1.6(b).
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 42
 Sampling of Continuous Signal (22)

2. Half of the sampling frequency fs/2 is

usually called the Nyquist frequency
(Nyquist limit), or folding frequency.
– The sampling theorem indicates that a DSP
system with a sampling rate of fs can ideally
sample an analog signal with its highest
frequency up to half of the sampling rate
without introducing spectral overlap (aliasing).
•

– Hence, the analog signal can be perfectly

recovered from its sampled version.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 43
 Example 1.1

Suppose that an analog signal is given as

and is sampled at the rate of 8,000 Hz.

a. Sketch the spectrum for the original signal.
b. Sketch the spectrum for the sampled signal from 0 to 20
kHz.
Solution:
• Since the analog signal is sinusoid with a peak value of 5 and
frequency of 1,000 Hz, we can write the cosine wave using
Euler’s identity:

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 44

 Example 1.1 (cont.)

• which is a Fourier series expansion for a continuous

periodic signal in terms of the exponential form.
• We can identify the Fourier series coefficients as

Figure 1.7 (a) Spectrum of the analog signal in Example 1.1.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 45
 Example 1.1 (cont.)

b.
• After the analog signal is sampled at the rate of 8,000 Hz,
– the sampled signal spectrum and its replicas centered at the
frequencies ±nfs,
– each with the scaled amplitude being 2.5/T, are as shown in
Figure 2.7b:

Figure 1.7 (b) Spectrum of the sampled signal in Example 1.1

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 46

 Example 1.1 (cont.)

• Notice that the spectrum of the sampled

signal shown in Figure 2.7b depicts the
images of the original spectrum shown in
Figure 2.7a;
– the images repeat at multiples of the sampling
frequency fs (for our example, 8 kHz, 16 kHz,
24 kHz, . . . ); and
– all images must be removed, since they convey
no additional information.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 47

 Section II
SIGNAL RECONSTRUCTION

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 48

 Signal Reconstruction

• Now, we investigate the recovery of analog signal from its sampled

signal version.
– Two simplified steps are involved, as described in Figure 1.8.

Figure 1.8 Signal

notations at
reconstruction stage.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 49
 Signal Reconstruction (2)

• First, the digitally processed data y(n) are

converted to the ideal impulse train ys(t),
– in which each impulse has its amplitude
proportional to digital output y(n), and two
consecutive impulses are separated by a sampling
period of T;

• second, the analog reconstruction filter is

applied to the ideally recovered sampled
signal ys(t) to obtain the recovered analog
signal.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 50
 Signal Reconstruction (3)
• To study the signal reconstruction,
– we let y(n) = x(n) for the case of no DSP,
– so that the reconstructed sampled signal and the
input sampled signal are ensured to be the same;
that is, ys(t) = xs(t).
• Hence, the spectrum of the sampled signal ys(t)
contains the same spectral content as the
original spectrum X( f ),
– that is, Y( f ) = X( f ), with a bandwidth of fmax= B
Hz (described in Figure 1.8(D) and the images of the
original spectrum (scaled and shifted versions).
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 51
 Signal Reconstruction (4)

• The following three cases are discussed for recovery

of the original signal spectrum X( f ).
1. Case 1: fs = 2fmax
• As shown in Figure 1.9,
– where the Nyquist frequency is equal to the maximum
frequency of the analog signal x(t),
– an ideal lowpass reconstruction filter is required to recover
the analog signal spectrum.
– This is an impractical case.

F IGURE 1.9 Spectrum of the sampled signal when fs= 2fmax.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 52
 Signal Reconstruction (5)
Case 2: fs > 2fmax
• In this case, as shown in Figure 1.10, there is a separation
between the highest-frequency edge of the baseband
spectrum and the lower edge of the first replica.
• Therefore, a practical lowpass reconstruction (anti-image)
filter can be designed to reject all the images and achieve
the original signal spectrum.

Figure 1.10 Spectrum of the sampled signal when fs > 2fmax

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 53
 Signal Reconstruction (6)
Case 3: fs < 2fmax
• Case 3 violates the condition of the Shannon sampling theorem.
• As we can see, Figure 1.11 depicts the spectral overlapping
between the original baseband spectrum and the spectrum of
the first replica and so on.
– Even when we apply an ideal lowpass filter to remove these
images, in the baseband there is still some foldover frequency
components from the adjacent replica.

Figure 1.11 Spectrum of the sampled signal when fs <2 fmax

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 54
 Signal Reconstruction (7)

• This is aliasing, where the recovered baseband spectrum

suffers spectral distortion,
– that is, contains an aliasing noise spectrum;
• In time domain, the recovered analog signal may consist
of the aliasing noise frequency or frequencies.
– Hence, the recovered analog signal is incurably distorted.

• Note that if an analog signal with a frequency f is

undersampled,
– the aliasing frequency component faliasin the baseband is
simply given by the following expression:

falias = fs - f
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 55
 Signal Reconstruction (8)
• The following examples give a spectrum analysis of the
signal recovery.

Example 1.2
Assuming that an analog signal is given by

and it is sampled at the rate of 8,000 Hz,

a. Sketch the spectrum of the sampled signal up to 20 kHz.
b. Sketch the recovered analog signal spectrum if an ideal
lowpass filter with a cutoff frequency of 4 kHz is used to filter
the sampled signal ys(t) = xs(t), to recover the original
signal.
Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 56
 Signal Reconstruction (8)

Solution:
Using Euler’s identity, we get

The two-sided amplitude spectrum for the sinusoids is

displayed in Figure 1.12:

Figure 1.12 Spectrum of the sampled signal in Example 1.2.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 57
 Example 1.2 (cont.)

b. Based on the spectrum in (a), the sampling

theorem condition is satisfied;
• hence, we can recover the original spectrum
using a reconstruction lowpass filter.
• The recovered spectrum is shown in Figure 1.13:

Figure 1.13 Spectrum of the recovered signal in Example 1.2.

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 58

 Example 1.3

Given an analog signal

which is sampled at a rate of 8,000 Hz,

a. Sketch the spectrum of the sampled signal up to 20
kHz.
b. Sketch the recovered analog signal spectrum
if an ideal lowpass filter with a cutoff frequency of 4
kHz is used to recover the original signal (y(n) =
x(n) in this case).

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 59

 Example 1.3

Solution:
a. The spectrum for the sampled signal is sketched in
Figure 1.14:

Figure 1.14 Spectrum of the sampled signal in Example 1.3

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 60
 Example 1.3 (cont.)

b. Since the maximum frequency of the analog signal is

larger than that of the Nyquist frequency
– that is, twice the maximum frequency of the analog signal is
larger than the sampling rate.
– the sampling theorem condition is violated.
• The recovered spectrum is shown in Figure 1.15,
– where we see that aliasing noise occurs at 3 kHz.

Figure 1.15 Spectrum of the recovered signal in Example 1.3

Dr. IBRAHIM AL-KEBSI (SCC) Digital Signal Processing 61