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
Convolution 1 lecture
Correlation 1 lecture
Z-domain transformation 4 lectures
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
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?
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
7.5
1 1 1
7
0 0 0
6
-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
-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
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
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
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
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
Number of 0.5
quantization levels 0
maximum amplitude of 0 1 2 3 4 5 6 7 8 9 10
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
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
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
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‟