‫ڪالس جي اسقاتڪار‬

‫‪Class time table:‬‬

‫‪Section I:‬‬
‫‪Mon & Tue 9:00 to 11:00 PST‬‬ ‫سيڪشن ‪:1‬‬
‫‪ 9‬کان ‪11‬‬ ‫سومر ۽ اڱارس‬
‫‪Section II:‬‬
‫‪Wed & Thu 9:00 to 11:00 PST‬‬ ‫سيڪشن ‪:2‬‬
‫‪ 9‬کان ‪11‬‬ ‫اربع ۽ خميس‬

‫‪INTRODUCTION T0 DSP‬‬
‫عددي سگنل پردازڪاريَء جو تعارف‬
‫احسان احمد عرساڻي‬ ‫‪Lecture 1‬‬ ‫ليڪچر ‪1‬‬
 In today‟s class ‫اڄوڪي ڪالس ۾‬
THE TEACHING PLAN ‫دريسي رٿا‬
OTHER DETAILS ‫ٻيا تفصيل‬
INTRODUCTION TO DSP
‫عسپ جو تعارف‬
ANALOG TO DIGITAL CONVERSION
• SAMPLING
‫ايناالگ کان عددي تدبل‬
• SAMPLING THEOREM, ALIASING ‫•جزڪاري‬
• QUANTIZATION ‫ ايليازڱ‬،‫•جزڪاري نظريئڙس‬
• QUANTIZATION NOISE
‫•ايڪي ڪاري‬
‫•ايڪي ڪاري گوڙ‬
 Teaching plan ‫تدريسي رٿا‬

 Introduction 2 lectures

 Preliminaries

 Review of Signals & Systems

 Analog to Digital Conversion

 Convolution 1 lecture
 Correlation 1 lecture
 Z-domain transformation 4 lectures

 Fourier transformation 7 lectures

 Discrete Fourier Transform (Review)
 Fast Fourier Transform
Cont‟d
 Digital Filters
 Finite Impulse Response (FIR) filters 11 lectures

 Infinite Impulse Response (IIR) filters 10 lectures

 Realization of Digital Filters 2 lectures

 Multi-rate Signal Processing 5 lectures

 Adaptive Signal Processing 4 lectures
 Spectrum Estimation & Analysis 5 lectures

Pre-requisite : Signals & Systems (Theory)

 Reading material ‫مطالعاتي مواد‬

 Text book
 Digital Signal Processing:
Principles, Algorithms and
Applications
by
John Proakis,
Dimitris Manolakis
 Reference books
 Discrete-time
Signal Processing by Alan Oppenheim
and Ronald Schaffer
 By Emmanuel Ifeacher

 Web based demos

Marks distribution
Marks Distribution

5 5
10
Test (one at the end)

Assignments/Class
performance
80 Attendence

Examination
Submitting Assignments!
 Assignments will have to be submitted in the form of
groups
 Each group can have 6 members max.
 Every group must have at least one person in the
Top10 students of your class (section)
 Group leader will be the student among Top10
 Submit the name of your group members tomorrow!
 Mention roll no‟s
 Email address of group leader
Digital Signal Processing: What is it?

What is digital signal processing (DSP) anyway?

The term DSP generally refers to the use of digital

computers to process signals. Normally, these signals
can be handled by analog processors but, for a
variety of reasons, we may prefer to handle them
digitally
Why Digital Signal Processing?
 Why Digital Signals & Digital Systems?
 Analog signals can‟t be stored properly
 Processing analog signals is not efficient
 e.g.An analogue filter with sharp cutoff has non-linear
phase response
 Analog systems are usually implemented using
resistors, capacitors, inductors, op-amps, transistors etc
 Potentially consume a lot of power
 Analog systems can get very complex
 An analog system made for one task/purpose may not
work for another task/purpose
 Why Digital Signals & Digital Systems?
 Digital signals can be easily stored and processed
 Digital Signal „Processing‟ is implemented in processors
 Processors in your cell phones, mp3 players etc
 Same processor can be used for many applications
 Multi-purpose
 PC can be used for education, gaming, watching
movies, listening to songs… what else?
 Digital
systems will always consume less power than their
analogue counterparts!
DSP for Telecom engineers: Why?
 DSP has made deep inroads in to Telecommunications
 All modern communication systems are based on
„digital communication‟ principles, which makes them
robust against noise, that is always present in the
channel
 Robustness is due to the signal processing algorithms
that run in the digital system
 Regenerative repeaters
 Equalizers

 Error-correction codes and so on

Cont‟d
 Modern communication systems are based on Field
Programmable Gate Array (FPGAs), Digital Signal
Processors (DSPs) or Application Specific Integrated
Circuits (ASICs)

 All these really implement DSP algorithms

DSP Applications
 Medical
 Diagnostic imaging (MRI, CT, ultrasound, etc.)
 Electrocardiogram (ECG) analysis
 Electroencephalogram (EEG) analysis
 Medical image storage and retrieval
 Scientific
 Data acquisition
 Data extraction (DIP)
 Simulation & modeling
 Spectral analysis
Cont‟d
Commercial
 Sound Processing

 Its no fluke that suddenly every singer has a good voice!!

 MP3, WMA, RM etc

 Image Processing
 Adobe Photoshop
 JPEG, BMP, GIF etc
 Djvu (new compression format for scanned documents)

 Video Processing
 Better video formats that occupy less space
 Video stabilization
Cont‟d
 Military
 More secure information transfer (better encryption)
 Jammers

 RADAR

 GPS guided missiles?

So many more applications!

Need for A/D conversion
 We know by now the benefits of digital signals and
systems
 But most signals of practical interest are still analog
 Voice, Video
 RADAR signals

 Biological signals etc

 So in order to utilize those benefits, we need to
convert our analog signals into digital
 This process is called A/D conversion
Three step process
 Analog to Digital conversion is really a three step
process involving
 Sampling
 Conversion from continuous-time, continuous valued
signal to discrete-time, continuous-valued signal
 Quantization
 Conversion from discrete-time, continuous valued signal
to discrete-time, discrete-valued signal
 Coding
 Conversion from a discrete-time, discrete-valued signal
to an efficient digital data format
 Represent as bit?
 Analog signal Binary bits

SAMPLING QUANTIZATIO CODING

N

CT-CV DT-CV DT-DV DT-DV

7.5

1 1 1
7

0.5 0.5 0.5

6.5

0 0 0
6

-0.5 -0.5 -0.5 5.5

-1 -1 -1 5

2 4 6 8 10 2 4 6 8 10 2 4 6 8 10 4.5
1 2 3 4 5 6 7 8 9 10

Arbitrarily, I‟ve chosen Differential

PCM…. Can you re-create these graphs?
Sampling
 A continuous-time signal has some value „defined‟ at „every‟ time instant
 So it has infinite number of sample points
1 1
sample
0.5 0.5 every
1 sec
0 0

-0.5 -0.5

-1 -1

2 4 6 8 10 2 4 6 8 10

1 1

0.5 0.5

0 sample 0 sample
every every
-0.5 0.1 sec -0.5 1 μsec
-1 -1

2 4 6 8 10 2 4 6 8 10
Aliasing:
 It is impossible to digitize an infinite number of points
because infinite points would require infinite amount
of memory and infinite amount of processing power
 So we have to take some finite number of points
 Sampling can solve such a problem by taking samples
at the fixed time interval

 If an analog signal is not appropriately

sampled, aliasing will occur, where a discrete-time
signal may be a representation (alias) of multiple
continuous-time signals
 Shannon‟s sampling theorem

The sampling theorem guarantees that an analogue 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 analogue signal to be sampled
Fs  2Fmax

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 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
 In order to sample a voice signal containing
frequencies up to 4 KHz, we need a sampling rate
of 2*4000 = 8000 samples/second
 Similarly for sampling of sound with frequencies up
to 20 KHz, we need a sampling frequency of
2*20000 = 40000 samples/second
 What is the sampling rate for CDs?
 Isn‟t it more than the one we just calculated?
Example 1: For the following analog signal, find the Nyquist sampling
rate, also determine the digital signal frequency and the digital signal
x(t )  3 cos(70 )t

The maximum frequency component is x(t) is

70
Fmax   35 Hz
2
Therefore according to Nyquist, we need a sampling rate of
Fs  2Fmax  70 Hz
The digital signal would have a frequency
35
w  2 
70
The digital signal can be represented as

x[n]  3 cos( n)
 Anti-aliasing filters
Anti-aliasing filters are analog filters as they process the signal
before it is sampled. In most cases, they are also low-pass filters
unless band-pass sampling techniques are used

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.
What can be the implications?

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

That‟s why the sampling rate of a CD is 44.1 KHz, much higher than
the 40 KHz we calculated

Go through the assignment… it has some reading task along with

some problems
Example 2: Find the Nyquist‟s rate for the following signal
x(t )  3 cos(50 )t  10 sin(300 )t - cos(100 )t

This composite signal comprises three frequencies

f1 = 25 Hz, f2 = 150 Hz, f3 = 50 Hz

So, according to Nyquist we need to sample at 300 Hz

However, for the „sine‟ term, the sampled signal has values
sin(πn), meaning the samples are taken at the „zero crossings‟, so the
sine term is not counted in the process

Therefore, a solution is to sample at higher than twice the max. freq

component
 Quantization
 Now that we have converted the continuous-
time, continuous-valued signal into a discrete-
time, continuous-valued signal, we STILL need to make
it discrete valued
 This is where Quantization comes into picture
 “The process of converting analog voltage with
infinite precision to finite precision”
 For e.g. if a digital processor has a 3-bit word, the
amplitudes of the signal can be segmented into 8 levels
Quanitization
General rules for Quantization
 Important properties
of the quantizer
include 1
Ymax = 1

 Number of 0.5

quantization levels 0

 Quantization resolution -0.5

 Note the minimum & -1

Ymin = -1

maximum amplitude of 0 1 2 3 4 5 6 7 8 9 10

the input signal

 Ymin & Ymax
 Note the word-length of DSP
m-bits
 Number of levels of quantizer is equal to
L = 2m
 The resolution of the quantizer is given as
( ymax  ymin )
 (volts)
L 1
 Resolution of a quantizer is the distance between two
successive quantization levels
 More quantization levels, better resolution!
 Whats the downside of more quantization levels?
 1
0.9 n n0
0.9 x[n]  
 0 n0
0.8 m = 4, L = 16
Ymin = 0
0.7 Ymax = 1
∆ = 1/15 = 0.0667
0.6

0.5

0.4

0.3

0.2

0.1

0
0 5 10 15 20 25
 Quantization error
The error caused by representing a continuous-valued
signal(infinite set) by a finite set of discrete-valued
levels

Suppose a quantizer operation given by Q(.) is

performed on continuous-valued samples x[n] is given
by Q(x[n]), then the quantization error is given by
eq [n]  x[n]  xq [n]
 0.9 n n0
Lets consider the signal x[n]   , which is to be
 0 n0
quantized.

In the figure (previous slide), we saw that there was a

difference between the original signal and the quantized
signal. This is the error produced while quantization

It can be reduced, however, if the number of quantization

levels is increased as illustrated on next slide
 1 0.14

0.9
0.12
0.8

0.7 0.1

0.6
0.08
0.5

0.4 0.06

0.3
0.04
0.2

0.1 0.02

0
0 5 10 15 20 25 0
0 5 10 15 20 25

3-bit ADC Quant. error

-3
x 10
1 4

0.9
3.5
0.8
3
0.7

0.6 2.5

0.5
2

0.4
1.5
0.3
1
0.2

0.1 0.5

0
0 5 10 15 20 25 0
0 5 10 15 20 25

8-bit ADC Quant. error

 Signal-to-Quantization-noise ratio
 Provides the ratio of the signal power to the
quantization noise (or quanitization error)
1 N 1 1 N 1
Pq   eq n   xn  xq n
2 2

 Mathematically, N n 0 N n 0

SQNRdB  10 log10 
Px
Pq
where
Px = ¨Power of the signal „x‟ (before quantization)
Pq = ¨Power of the error signal „xq‟