Mustansiriyah University

College of Engineering
Electrical Engineering Department
4th Class

Digital Signal Processing

(EE402)

2022-2023
Digital Signal Processing / 4th Class/ 2022-2023

Topics Covered
Introduction to Digital Signal Processing
Signal Sampling and Reconstructions: Sampling of Continuous Signal,
Signal Reconstruction, Aliasing Noise Level
Digital Signals and Systems: Classification of Systems, Linear System,
Time-Invariant System, Causal System, Stability
Digital Convolution: Graphical Method, Table Lookup Method, Matrix by
Vector Method, Linear Convolution and Circular Convolution,
Deconvolution
Frequency Response and Sinusoidal Steady State Response
Z-Transform (Review), Discrete Fourier Transform, Fast Fourier
Transform
Fast Fourier Transform (FFT) Algorithms
Analog Filter Design: Butterworth Filters , Chebyshev Filters.
Digital Filter Design: Infinite Impulse Response (IIR) filter , Finite Impulse
Response (FIR) filter
Realization of Digital Filters :Realization of IIR Filters , Realization of FIR
Filters
Theoretical: 2 Hrs/Wk
Total hours (60 Theoretical)

Suggested References:
1) "Digital Signal Processing Principles, Algorithms, and Applications", John G. Proakis,
Dimitris G. Manolakis, Third Edition (1996).
2) "Applied Digital Signal Processing Theory and Practice", Dimitris G. Manolakis, Vinay K.
Ingle, First Edition (2011).
Digital Signal Processing / 4th Class/ 2022-2023

Introduction to Digital Signal Processing

Signal (flow of information):
Signal is defined as any physical quantity that varies with Time, Space, or any other
independent variables. For Example:
 Measured quantity that varies with time (or position).
 Electrical signal received from a transducer (Microphone, Thermometer, Accelerometer,
Antenna, etc.)
 Electrical signal that controls a process.
Example:

A. Continuous-Time Signal or Analog Signal:

The analog signal is defined for every value of time and they take on values in the
continuous interval as shown in Fig. 1.

Fig. 1. Continuous or analog signal

 Continuous in time.
 Amplitude may take on any value in the continuous range of (-∞,∞).

 Analog Processing
 Differentiation, Integration, Filtering, Amplification.
 Differential Equations
 Implemented via passive or active electronic circuitry.

B. Discrete-Time signals:
Discrete signals are defined only at certain specific value of time as shown in Fig. 2.

.
Fig. 2. Discrete signal

1
Digital Signal Processing / 4th Class/ 2022-2023

 Continuous in amplitude but discrete in time.

 Only defined for certain time instances.
 Can be obtained from analog signals via sampling.

C. Digital Signal:
Digital signal is the signal that takes on values from a finite set of possible values as
shown in Fig. 3.

Fig. 3. Digital signal with four different amplitude values

 Discrete in amplitude & discrete in time.

 Can be obtained from discrete signals via quantization.

Finite and infinite length signal:

Finite length signal is nonzero over a finite interval t min< t< tmax as shown in Fig. 4.

Fig. 4. Finite length signal

In contrast, the infinite length signal is nonzero over all real numbers.

What is signal processing?

Signals may have to be transformed in order to
 Amplify or filter out embedded information.
 Detect patterns.
 Prepare the signal to survive a transmission channel.
 Undo distortions contributed by a transmission channel.
 Compensate for sensor deficiencies.
 Find information encoded in a different domain.

2
Digital Signal Processing / 4th Class/ 2022-2023

Fig. 5 explains the main idea of the signal processor.

Fig. 5. Signal Processor

Analog signal processing:

Fig. 6 shows a basic block diagram of a typical analog signal processing system.

Fig. 6. A typical analog signal processing system

Where,
ℎ(𝑡): The System Impulse Response
H(𝑠): The System Transfer Function
H(Ω): The System Frequency Response
Analogue signal processing is achieved by using analogue components such as:
 Resistors.
 Capacitors.
 Inductors.

Limitations of analog signal processing:

 Accuracy limitations due to
 Component tolerances
 Undesired nonlinearities
 Limited repeatability due to
 Tolerances
 Changes in environmental conditions
 Temperature
 Vibration
 Sensitivity to electrical noise
 Limited dynamic range for voltage and currents
 Inflexibility to changes

3
Digital Signal Processing / 4th Class/ 2022-2023

 Difficulty of implementing certain operations

 Nonlinear operations
 Time-varying operations
 Difficulty of storing information

Digital signal processing (DSP) system:

Digital signal processing (DSP) is one of the most powerful technologies that will
shape science and engineering in the twenty-first century. Revolutionary changes have already
been made in aboard range of fields: communications, radar and sensor. DSP converts signals
that naturally accrue in analog form (such as sound, video and information from sensors) to
digital form and uses digital techniques to enhance and modify analog signal data for various
applications. Fig. 7 shows a basic block diagram of a typical digital signal processing system.

Fig. 7. A typical digital signal processing (DSP) system

The system consists of an analog filter, an analog-to-digital conversion (ADC) unit, a

digital signal processor (DSP), 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 non-electrical 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.

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 DSP 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 DSP 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 DSP and

4
Digital Signal Processing / 4th Class/ 2022-2023

corresponding software, a processed digital output signal is generated. This signal behaves in a
manner according to the specific algorithm used.

The DAC unit converts the processed digital signal to an analog output signal. The
signal is continuous in time and discrete in amplitude (usually a sample-and-hold signal).

The final stage in Fig. 7 is often another analog filter designated as a function to
smooth the DAC output voltage levels back to the analog signal (i.e. to reconstruct the analog
signal from the DAC output).

In contrast to the above, a direct analog processing of analog signals is much simpler
since it involves only a signal processor. It is therefore natural to ask why we go to use the
DSP systems. There are several good reasons:

1- Rapid advances in integrated circuit design and manufacture are producing more
powerful DSP systems on a single chip at decreasing size and cost.

2- Digital processing is inherently stable and reliable.

3- Good processing techniques are available for digital signals, such as Data compression
(or source coding), Error Correction (or channel coding), Equalization and Security.

4- Easy to mix signals and data using digital techniques known as Time Division
Multiplexing (TDM).

5- It is easy to Change, Correct, or Update applications (software changes), such as-that

needed in implementing adaptive circuits.

6- Sensitivity to electrical noise is minimal.

7- Digital information can be encrypted for security.

The list below by no means covers all DSP applications. Many more areas are
increasingly being explored by engineers and scientists. Applications of DSP techniques will
continue to have profound impacts and improve our lives.

1- Digital audio and speech: Digital audio coding such as CD players, digital crossover,
digital audio equalizers, digital stereo and surround sound, noise reduction systems,
speech coding, data compression and encryption, speech synthesis and speech
recognition.

2- Digital telephone: Speech recognition, high-speed modems, echo cancellation, speech

synthesizers, DTMF (dual-tone multi frequency) generation and detection, answering
machines.
5
Digital Signal Processing / 4th Class/ 2022-2023

3- Automobile industry: GPS, Active Noise Cancellation, Cruise Control, Parking.

4- Electronic communications: Cellular phones, digital telecommunications, wireless LAN

(local area networking), satellite communications.

5- Medical imaging equipment: ECG analyzers, cardiac monitoring, medical imaging and
image recognition, digital x-rays, image processing, magnetic resonance, tomography
and electrocardiogram.

6- Multimedia: Internet phones, audio, and video, hard disk drive electronics, digital
pictures, digital cameras, DVD, JPEG, Movie special effects, video conferencing, text-
to-voice and voice-to-text technologies.

7- Military: Radar, sonar, space photographs, remote sensing.

8- Mechanical: Motor control, process control, oil and mineral prospecting.

6
Digital Signal Processing / 4th Class/ 2022-2023

Signal Sampling and Reconstruction

Analog to digital (A/D) conversion:
The analog-to-digital conversion is basically a 2 step process:
 Sampling
• Converts continuous-time analog signal xa (t) to discrete-time continuous value signal
x(n).
• It is obtained by taking the ”samples” of xa(t) at discrete-time intervals, Ts
 Quantization
• Converts discrete-time continuous valued signal to discrete time discrete valued signal.
These steps are shown in Fig. 8.

Fig. 8. Basic steps of ADC

Sampling of continuous signal

Sampling is the processes of converting continuous-time analog signal, xa(t), into a
discrete-time signal by taking the “samples” at discrete-time intervals.
 Sampling analog signals makes them discrete in time but still continuous valued.
 If done properly (Nyquist theorem is satisfied), sampling does not introduce distortion.
Fig. 9 shows an analog (continuous-time) signal (solid line) defined at every point over
the time axis and amplitude axis. Hence, the analog signal contains an infinite number of
points.

Fig. 9. Display of analog (continuous) signal and digital samples versus the sampling time instants

7
Digital Signal Processing / 4th Class/ 2022-2023

It is impossible to digitize an infinite number of points. Furthermore, the infinite points

are not appropriate to be processed by the digital signal processor or computer, since they
require an 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 Fig. 9 and Fig. 10, where the time T represents the sampling interval or
sampling period in seconds. As shown in Fig. 10, 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.

Fig. 10. Sample-and-hold analog voltage for ADC

For a given sampling interval T, which is defined as the time span between two sample
points, the sampling rate or sampling frequency is the rate at which the signal is sampled,
expressed as the number of samples per second (reciprocal of the sampling interval).

fs=1/T Samples per second (Hz)

 If the signal is slowly varying, then fewer samples per second will be required than if the
waveform is rapidly varying. So, the optimum sampling rate depends on the maximum
frequency component present in the signal.

Nyquist sampling theorem or Nyquist criterion:

If an analog signal is not appropriately sampled, aliasing will occur, which causes
unwanted signals in the desired frequency band (i.e. if the sampling is performed at a proper
rate, no info is lost about the original signal and it can be properly reconstructed later).
”If a signal is sampled at a rate at least, but not exactly equal to twice the max frequency
component of the waveform, then the waveform can be exactly reconstructed from the samples
without any distortion“. The condition is described as

f s  2 f max
Where, fmax is the maximum-frequency component of the analog signal to be sampled.

8
Digital Signal Processing / 4th Class/ 2022-2023

Ts is called the Nyquist interval: It is the longest time interval that can be used for sampling a
band limited signal and still allow reconstruction of the signal at the receiver without
distortion.

Example: Find the Nyquist frequency and Nyquist interval of the following signals:
a) speech signal containing frequencies up to 4 kHz
b) audio signal possessing frequencies up to 20 kHz
Solution:
a) to sample a speech signal containing frequencies up to 4 kHz, the Nyquist rate
(minimum sampling rate fs) is chosen to be at least 8 kHz, or 8,000 samples per
second (fs=2fm) and Nyquist interval (maximum time interval Ts) is 1/fs = 1/8 kHz =
0.125 ms.
b) 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 and Nyquist interval
(maximum time interval Ts) is 1/fs = 1/40 kHz = 25 μs.

Sampled signal spectrum:

Fig. 11 depicts the sampled signal xs(t) obtained by sampling the continuous signal x(t)
at a sampling rate of fs samples per second. 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)

Where, p(t) is the pulse train with a period T = 1/ fs.

Fig. 11. The simplified sampling process

From the spectral analysis shown in Fig. 12, it is clear that the sampled signal spectrum
9
Digital Signal Processing / 4th Class/ 2022-2023

consists of the scaled baseband spectrum centered at the origin and its replicas centered at the
frequencies of ± nfs (± n/Ts) (multiples of the sampling rate) for each of n = 1,2,3, . . .

In Fig. 12, three possible sketches are classified. Given the original signal spectrum
X(f) plotted in Fig. 12(a), the sampled signal spectrum is plotted in Fig. 12(b), where, the
replicas have separations between them. In Fig. 12(c), the baseband spectrum and its replicas
are just connected. In Fig. 12(d), the original spectrum and its replicas are overlapped; that is,
there are many overlapping portions in the sampled signal spectrum.

Fig. 12. Plots of the sampled signal spectrum

If applying a lowpass reconstruction filter to obtain exact reconstruction of the original signal
spectrum,

 As long as fs > 2B, no overlap of repeated replicas X(f - n/Ts) will occur in Xs(f). Hence,
the signal at the output of the filter will be the original signal spectrum without
distortion as shown in Fig. 13.

 If the waveform is undersampled (i.e. fs < 2B), then there will be spectral overlap in the
sampled signal. Hence, the signal at the output of the filter will be different from the
original signal spectrum as shown in Fig. 14. [This is the outcome of aliasing].

 This implies that whenever the sampling condition is not met, an irreversible overlap of
10
Digital Signal Processing / 4th Class/ 2022-2023

the spectral replicas is produced.

Fig. 13. Filter o/p in case of fs > 2B

Fig. 14. Filter o/p in case of fs < 2B

Example:
Suppose that an analog signal is given as
x(t) = 5 cos (2π.1000t), for t > 0, 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.
Sol.
a. Since the analog signal is sinusoid with a peak value of 5 and frequency of 1,000 Hz, we
can write the sine wave using Euler’s identity:

11
Digital Signal Processing / 4th Class/ 2022-2023

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 below.

Notice that the spectrum of the sampled signal contains the images of the original spectrum;
that the images repeat at multiples of the sampling frequency fs (for our example, 8 kHz, 16
kHz, 24 kHz, . . . ); and that all images must be removed, since they convey no additional
information.

Signal reconstruction
Two simplified steps are involved, as described in Fig. 15. 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.

Fig. 15. Signal notations at reconstruction stage

The following three cases are listed for recovery of the original signal spectrum:
12
Digital Signal Processing / 4th Class/ 2022-2023

Case 1: fs = 2fmax: 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.

Case 2: fs > 2fmax: In this case, 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.

Case 3: fs < 2fmax: 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.

Example: Assuming that an analog signal is given by

x(t) = 5cos(2π.2000t) +3cos(2π.3000t) for t ≥ 0, 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 (y(n)=x(n) in this case) to recover the
original signal.
Sol.
a. Using Euler’s identity, we get

The two-sided amplitude spectrum for the sinusoids (sampled signal) is displayed in Fig.

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 the following Fig.

13
Digital Signal Processing / 4th Class/ 2022-2023

Aliasing noise level

Given the DSP system shown in Fig. 16, where we can find the percentage of the
aliasing noise level using the symmetry of the Butterworth magnitude function and its first
replica. Then:

Fig. 16. DSP system with anti-aliasing filter

2n
 f 
1   a 
 fc 
Aliasing noise level % = for 0 ≤ f ≤ fc
2n
 f  fa 
1   s 
 fc 

Where, n is the filter order, fa is the aliasing frequency, fc is the cutoff frequency, and fs is the
sampling frequency.

Example: In a DSP system with anti-aliasing filter, if a sampling rate of 8,000 Hz is used and the
anti-aliasing filter is a second-order Butterworth lowpass filter with a cutoff frequency of 3.4
kHz,
a. Determine the percentage of aliasing level at the cutoff frequency.
b. Determine the percentage of aliasing level at the frequency of 1,000 Hz.

Sol.

14