INTRODUCTION T0 DSP

1 Lecture 1
Class time table:
Section I:
Mon & Tue 9:00 to 11:00 PST
Section II:
Wed & Thu 9:00 to 11:00 PST


1 :
9 11
2 :
9 11
THE TEACHING PLAN
OTHER DETAILS
INTRODUCTION TO DSP
ANALOG TO DIGITAL CONVERSION
SAMPLING
SAMPLING THEOREM, ALIASING
QUANTIZATION
QUANTIZATION NOISE
In todays class








Teaching plan
Introduction
Preliminaries
Review of Signals & Systems
Analog to Digital Conversion
Convolution
Correlation
Z-domain transformation
Fourier transformation
Discrete Fourier Transform (Review)
Fast Fourier Transform
2 lectures
1 lecture
1 lecture
4 lectures
7 lectures
Contd
Digital Filters
Finite Impulse Response (FIR) filters
Infinite Impulse Response (IIR) filters
Realization of Digital Filters
Multi-rate Signal Processing
Adaptive Signal Processing
Spectrum Estimation & Analysis
11 lectures
10 lectures
5 lectures
4 lectures
2 lectures
5 lectures
Pre-requisite : Signals & Systems (Theory)
Reading material
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
Test (one at the end)
Assignments/Class
performance
Attendence
Examination
80
10
5
5
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 nos
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 cant 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 songswhat 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
Contd
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
Contd
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
Contd
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?
SAMPLING QUANTIZATIO
N
CODING
CT-CV DT-CV DT-DV DT-DV
Analog signal Binary bits
2 4 6 8 10
-1
-0.5
0
0.5
1
2 4 6 8 10
-1
-0.5
0
0.5
1
2 4 6 8 10
-1
-0.5
0
0.5
1
1 2 3 4 5 6 7 8 9 10
4.5
5
5.5
6
6.5
7
7.5
Arbitrarily, Ive 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
2 4 6 8 10
-1
-0.5
0
0.5
1
2 4 6 8 10
-1
-0.5
0
0.5
1
2 4 6 8 10
-1
-0.5
0
0.5
1
2 4 6 8 10
-1
-0.5
0
0.5
1
sample
every
1 sec
sample
every
0.1 sec
sample
every
1 sec
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
Aliasing:
Shannons 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
max
2F F
s
>
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 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 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?
Isnt 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
t ) 70 cos( 3 ) ( t = t x
The maximum frequency component is x(t) is
Therefore according to Nyquist, we need a sampling rate of
The digital signal would have a frequency
The digital signal can be represented as
Hz F 35
2
70
max
= =
t
t
Hz F F
s
70 2
max
= =
t t = =
70
35
2 w
) cos( 3 ] [ n n x t =
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
Thats 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 Nyquists rate for the following signal
t ) 100 cos( - t ) 300 sin( 10 t ) 50 cos( 3 ) ( t t t + = t x
This composite signal comprises three frequencies
f
1
= 25 Hz, f
2
= 150 Hz, f
3
= 50 Hz
So, according to Nyquist we need to sample at 300 Hz
However, for the sine term, the sampled signal has values
sin(n), 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
Number of
quantization levels
Quantization resolution
Note the minimum &
maximum amplitude of
the input signal
Ymin & Ymax
0 1 2 3 4 5 6 7 8 9 10
-1
-0.5
0
0.5
1
Ymax = 1
Ymin = -1
Note the word-length of DSP
m-bits
Number of levels of quantizer is equal to
L = 2
m
The resolution of the quantizer is given as
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
) (
min max
volts
L
y y

= A
0 5 10 15 20 25
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1

<
>
=
0 0
0 9 . 0
] [
n
n
n x
n
m = 4, L = 16
Ymin = 0
Ymax = 1
= 1/15 = 0.0667
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
] [ ] [ ] [ n x n x n e
q q
=
Lets consider the signal , which is to be
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

<
>
=
0 0
0 9 . 0
] [
n
n
n x
n
0 5 10 15 20 25
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
0 5 10 15 20 25
0
0.02
0.04
0.06
0.08
0.1
0.12
0.14
0 5 10 15 20 25
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
0 5 10 15 20 25
0
0.5
1
1.5
2
2.5
3
3.5
4
x 10
-3
3-bit ADC
8-bit ADC
Quant. error
Quant. error
Signal-to-Quantization-noise ratio
Provides the ratio of the signal power to the
quantization noise (or quanitization error)
Mathematically,
where
P
x
= Power of the signal x (before quantization)
P
q
= Power of the error signal x
q

( )
q
x
P
P
dB
SQNR
10
log 10 =
| | ( ) | | | | ( )


=

=
= =
1
0
2
1
0
2
1 1
N
n
q
N
n
q
n x n x
N
n e
N
Pq