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Lecture

The document outlines the course objectives and outcomes for a 5G Communication subject at the University Institute of Engineering, focusing on preparing students for careers in Electronics and Communication Engineering. It emphasizes the importance of Digital Signal Processing (DSP) in applications like GSM and ADSL, detailing the technical aspects and methodologies involved in these technologies. Additionally, it covers filter design and realization steps necessary for implementing DSP systems.

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Chahat Pal
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0% found this document useful (0 votes)
41 views48 pages

Lecture

The document outlines the course objectives and outcomes for a 5G Communication subject at the University Institute of Engineering, focusing on preparing students for careers in Electronics and Communication Engineering. It emphasizes the importance of Digital Signal Processing (DSP) in applications like GSM and ADSL, detailing the technical aspects and methodologies involved in these technologies. Additionally, it covers filter design and realization steps necessary for implementing DSP systems.

Uploaded by

Chahat Pal
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PPTX, PDF, TXT or read online on Scribd
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University Institute of Engineering

Department of Electronics & Communication


Engineering
Subject Name 5G_Communication
Subject Code 22ECT325
Prepared By
Dr. Sunil Kumar

DISCOVER . LEARN . EMPOWER


Course
Objectives
S.
Objectives
No.
To prepare students to have successful careers in ECE
1 core companies of national and international repute
and/or to excel in research.
To provide students with solid foundation in ECE so that
2 they will use this knowledge in getting jobs and
maintaining their jobs.
To train the students in core/ advance fields of
electronics and communication using state of the art
3 technology.

2
Course
Outcomes
S.NO Description Blooms’ Taxonomy
Level

The ability to face the test and interview


CO1 conducted by different companies and EASY
succeed.
The ability to recognize the need for
CO2 continuing professional development. MEDIUM

The ability to succeed in competitive exams


CO3 (GATE / GRE / PSU’s) and excel themselves MEDIUM
in the related field of research.
Able to apply the knowledge of Electronics
and communication systems for the
CO4 development of society. HARD

The ability to design cellular and optical


CO5 networks. HARD
3
Digital Signal Processing II
Chapter-1 : Introduction
• Aims/Scope
Why study DSP ?
DSP in applications : GSM, ADSL,…

• Overview

• Activities
Lectures - Course Notes/Literature
Homeworks/Exercise sessions
Paper Review Assignment
Exam
Why• study DSP ?
Analog Systems vs. Digital Systems

IN OUT IN 2 OUT
A/D +2 D/A
=4

- can translate (any) analog (e.g. filter) design into digital


- going `digital’ allows to expand functionality/flexibility/…
(e.g. how would you do analog speech recognition ? analog audio
compression ? …? )
Why study DSP ?
• Will give two `DSP in applications’ examples
• DSP in mobile communications (GSM)
• DSP in wireline communications (ADSL/VDSL)

• Main message:
Consumer electronics products have become
supercomputers (Mops/sec…Gops/sec), packed with DSP
functionalities…
DSP in applications : GSM
Cellular mobile telephony (e.g. GSM)

• Basic network architecture :


-country covered by a grid of cells
-each cell has a base station
-base station connected to land telephone network and
communicates with mobiles via a radio interface
-digital communication format
DSP in applications : GSM

• DSP for digital communications (`physical layer’ ) :


.99,.01,.96,.95,.07,…
• a common misunderstanding is that digital communications
is `simple’….
Transmitter Channel Receiver
1,0,1,1,0,…

decision
x + x
1,0,1,1,0,…
a 1/a
noise

• While in practice…
DSP in applications : GSM
• DSP for digital communications (`physical layer’ ) :

!!
• In practice… .59,.41,.76,.05,.37,…

Transmitter `Multipath’ Receiver


1,0,1,1,0,…
Channel
+

noise
?? 1,0,1,1,0,…

• This calls for channel modeling + compensation (equalization)


DSP• GSM
in Channel
applications : GSM
Estimation/Compensation
• Multi-path channel is modeled with short (3…5 taps) FIR filter

H(z)= a+b.zˉ¹+c.z ˉ²+d.z ˉ³+e.z ˉ 4 (interpretation?)

a
`Multipath’
Channel b
Δ
c
+
≈ Δ
Δ
Δ
Δ Δ
d
+

e
Δ Δ Δ Δ
DSP• GSM
in Channel
applications : GSM
Estimation/Compensation (continued)
• Multi-path channel is modeled with short (3…5 taps) FIR filter

H(z)= a+b.zˉ¹+c.z ˉ²+d.z ˉ³+e.z ˉ 4

a
b
Δ
IN[k] c OUT[k]
 OUT[1]   IN [1] 0 0 0 0 
 OUT[2]   IN [ 2] IN [1] 0 0 0  a Δ Δ +
     d
 OUT[3]   IN [3] IN [2] IN [1] 0 0   b Δ Δ Δ
    
 OUT[4]   IN [ 4] IN [3] IN [2] IN [1] 0  . c  e
 OUT[5]   IN [5]   Δ Δ Δ Δ
IN [4] IN [3] IN [2] IN [1]   d 
   
           e 
 OUT[ N ]  0 0 0 0 IN [ N  4]
  

=convolution
DSP• GSM
in Channel
applications : GSM
Estimation/Compensation (continued)

• Channel coefficients (cfr. a,b,c,d,e) are identified in receiver based on


transmission of pre-defined training sequences (TS), in between data bits.
Problem to be solved at receiver is: `given channel input (=TS) and channel
output (=observed), compute channel coefficients’.
This leads to a least-squares parameter estimation procedure

2
 OUT [1]   TS [1] 0 0 0 0 
 OUT [ 2 ]
  TS [ 2 ] TS [1] 0 0 0
  a
 OUT [ 4 ] 
OUT [ 3]
 TS
TS [ 3] TS [ 2 ] TS [1] 0 0
  b
min  OUT [ 5]  
 TS [[ 54]] TS [ 3] TS [ 2 ] TS [1] 0
  d
. c
a, b, c, d , e TS [ 4 ] TS [ 3] TS [ 2 ] TS [1]
        
  e 
 OUT [ N ]  0 0 0 0 TS [ N  4 ]
 2
See PART-III on optimal filtering
DSP• GSM
in Channel
applications : GSM
Estimation/Compensation (continued)

• Channel coefficients (cfr. a,b,c,d,e) are identified in receiver based on


transmission of pre-defined training sequences (TS), in between data bits.

• Channel model is then used to design suitable equalizer (`channel


inversion’), or (better) to reconstruct transmitted data bits based on
maximum-likelihood sequence estimation (`Viterbi decoding’).

• Channel is highly time-varying (e.g. terminal speed 120 km/hr !)


=> All this is done at `burst-rate’ (+- 100 times per sec).

= SPECTACULAR !!
DSP• GSM
in Channel
applications : GSM
Estimation/Compensation

• GSM Speech Coding


• Original `PCM’-signal has 64kbits/sec = 8 ksamples/sec * 8bits/sample.
• Aim is to reduce this to <11kbits/sec, while preserving quality !
• Coding based on speech generation model (vocal tract,…), where model
coefficient are identified for each new speech segment (e.g. 20 msec).
• This leads to a least-squares parameter estimation (again), executed +- 50
times per second. Fast algorithm is used, e.g. `Levinson-Durbin’ algorithm.
See PART-III on optimal filtering.
• Then transmit model coefficients instead of signal samples.
• Synthesize speech segment at receiver
(should `sounds like’ original speech segment).

= SPECTACULAR !!
DSP• GSM
in Channel
applications : GSM
Estimation/Compensation

• GSM Speech Coding

• GSM Multiple Access Schemes


• Capacity increase by time & frequency `multiplexing’
• FDMA : e.g. 125 frequency channels for GSM/900MHz
• TDMA : 8 time slots(=users) per channel, `burst mode’ communication
(PS: in practice, capacity per cell << 8*125 ! )
See PART-II on filter banks/transmultiplexers
• Etc..

= BOX FULL OF DSP/MATHEMATICS !!


DSP in applications : ADSL

Telephone Line Modems

• voice-band modems : up to 56kbits/sec in 0..4kHz band

• ADSL modems : up to 8Mbits/sec in 30kHz…1MHz band


(3,5…5km)
• VDSL modems : up to 52Mbits/sec in …12MHz band

X 1000
(0.3…1.5km)

How has this been made possible?


DSP in applications : ADSL
Communication Impairments :
• Channel attenuation
• Received signal may be attenuated by more than 60dB
(attenuation increases with line length & larger at high (MHz) frequencies)
PS: this is why for a long time, only the voiceband (up to 4kHz) was used
• Frequency-dependent attenuation introduces ``inter-symbol interference’’ (ISI). ISI channel can (again)
be modeled with an FIR filter. Number of taps will be much larger here (>500!)
DSPCommunication
in applications
Impairments :
: ADSL
• Coupling between wires in same or adjacent binders introduces `crosstalk’
• Near-end Xtalk (NEXT) (=upstream in downstream, downstream in upstream)
• Far-end Xtalk (FEXT) (=upstream in upstream, downstream in downstream)
Meaning that a useful signal may be drowned in (much larger) signals from other users..
…leading to signal separation and spectrum management problems

• Radio frequency interference


(AM broadcast, amateur radio)
• Echo due to impedance mismatch
• Etc..

Conclusion: Need advanced modulation, DSP,etc. !


DSP in applications : ADSL
• ADSL spectrum : divide available transmission band in 256 narrow bands (`tones’), transmit
different sub-streams over different sub-channels (tones) (=DMT, `Discrete Multi-tone
Modulation’)
DSP in applications : ADSL
ADSL-DMT Transmission block scheme :
DFT/IDFT (FFT/IFFT) based modulation/demodulation scheme

pointer : www.adslforum.com PS: do not try to understand details here...


DSPADSL
inspecs
applications : ADSL
• 512-point (I)FFT’s (or `similar’) for DMT-modulation

c
se
FFT-rate = 4.3215 kHz

s/
op
(i.e. >4000 times 512-point FFTs per second !!!!)

0M
>2
• basic sampling rate is 2.21 MHz (=512*4.3215k)
8.84 MHz A/D or D/A (multi-rate structure)

c
se
s/
• fixed HP/LP/BP front-end filtering for frequency duplex

op
0M
See PART-I on filter banks/transmultiplexers

>6
• adjustable time-domain equalization filter (TEQ)
e.g. 32 taps @ 2.21 MHz
filter initialization via least-squares/eigenvalue procedure
• adaptive frequency-domain equalization filters (FEQ)
See PART-III on adaptive filters
= BOX FULL OF DSP/MATHEMATICS !!
DSP in applications : VDSL
VDSL specs
• 4096-point (I)FFT’s (or `similar’) for DMT-modulation

c
se
s/
FFT-rate = 4.3215 kHz

op
M
!!!!)

60
>1
(i.e. >4000 times 4096-point FFTs per second

• etc..

= BOX FULL OF DSP/MATHEMATICS !!


DSP in applications : Other…
• Speech
Speech coding (GSM, DECT, ..), Speech synthesis (text-to-speech),
Speech recognition
• Audio Signal Processing
Audio Coding (MP3, AAC, ..), Audio synthesis
Editing, Automatic transcription, Dolby/Surround, 3D-audio,.
• Image/Video
• Digital Communications
Wireline (xDSL,Powerline), Wireless (GSM, 3G, Wi-Fi, WiMax
CDMA, MIMO-transmission,..)
•…
DSP in applications
Enabling Technology is Signals&Systems course (JVDW)
DSP-I (PW)
• Signal Processing DSP-II
1G-SP: analog filters
2G-SP: digital filters, FFT’s, etc.
3G-SP: full of mathematics, linear algebra,
statistics, etc...
• VLSI
• etc...
PART-II : Filter Design/Realization
• Step-1 : Define filter specs
(pass-band, stop-band, optimization criterion,…)
• Step-2 : Derive optimal transfer function
FIR or IIR design
• Step-3 : Filter realization (block scheme/flow graph)
direct form realizations, lattice realizations,…
• Step-4 : Filter implementation (software/hardware)
finite word-length issues, …
question: implemented filter = designed filter ?
‘You can’t always get what you want’ -Jagger/Richards (?)
Step-1: Filter Specification
1.2
Ex: Low-pass
Passband Ripple
1
1 P
0.8 P S
0.6 Passband Cutoff -> <- Stopband Cutoff

0.4 Stopband Ripple S


0.2

0
0 0.5 1 1.5 2 2.5 3
• FIR filters
• Linear-phase FIR filters
• FIR design by optimization
Weighted least-squares design, Minimax design
• FIR design in practice
`Windows’, Equiripple design, Software (Matlab,…)
• IIR filters
• Poles and Zeros
• IIR design by optimization
Weighted least-squares design, Minimax design
• IIR design in practice
Analog IIR design : Butterworth/Chebyshev/elliptic
Analog->digital : impulse invariant, bilinear transform,…
Software (Matlab)
FIR Filters
FIR filter = finite impulse response filter

• Also known as `moving average filters’ (MA)


• L poles at the origin z=0 (hence guaranteed stability)
• L zeros (zeros of B(z)), `all zero’ filters
• Corresponds to difference equation

• Impulse response
Linear Phase FIR Filters

• Non-causal zero-phase filters :


example: symmetric impulse response (length 2.Lo+1)
h[-Lo],….h[-1], h[0] ,h[1],...,h[Lo]
h[k]=h[-k], k=1..Lo
k
Lo

e  jx  e  jx 2. cos x
frequency response is

i.e. real-valued (=zero-phase) transfer function


DSP-CIS / Chapter-4: FIR & IIR Filter Design / Version 2014-2015 p. 30
Linear
• CausalPhase FIR= non-causal
linear-phase filters Filters zero-phase + delay
example: symmetric impulse response & L even
h[0],h[1],….,h[L]
L=2.Lo
h[k]=h[L-k], k=0..L
k
0 N
frequency response is

= i.e. causal implementation of zero-phase filter, by


introducing (group) delay
Linear
Type-1 Phase
Type-2 FIR Filters
Type-3 Type-4
N=2Lo=even N=2Lo+1=odd N=2Lo=even N=2Lo+1=odd
symmetric symmetric anti-symmetric anti-symmetric
h[k]=h[N-k] h[k]=h[N-k] h[k]=-h[N-k] h[k]=-h[N-k]

zero at zero
at zeroat0,   0
LP/HP/BP LP/BP BP HP

PS: `modulating’ Type-2 with 1,-1,1,-1,.. gives Type-4 (LP->HP)


PS: `modulating’ Type-4 with 1,-1,1,-1,.. gives Type-2 (HP->LP)
PS: `modulating’ Type-1 with 1,-1,1,-1,.. gives Type-1 (LP<->HP)
PS: `modulating’ Type-3 with 1,-1,1,-1,.. gives Type-3 (BP<->BP)

PS: IIR filters can NEVER have linear-phase property ! (proof see literature)
FIR Filter Design by Optimization
(I) Weighted Least Squares Design :
• Select one of the basic forms that yield linear phase
e.g. Type-1

Specify desired frequency N / 2


• H d response
( ) e  j(LP,HP,BP,…)
. Ad ( )

• Optimization criterion is

W ( ) 0
where is a weighting function
FIR Filter Design by Optimization
• …This is equivalent to

= `Quadratic Optimization’ problem


xOPT Q  1. p
FIR Filter Design by Optimization
• Example: Low-pass design 1.2

Passband Ripple
1

0.8

Ad ( ) 1,    P (pass - band) 0.6 Passband Cutoff -> <- Stopband Cutoff

Ad ( ) 0,  S    (stop - band)


0.4 Stopband Ripple

optimization function is 0.2

0
0 0.5 1 1.5 2 2.5 3

i.e.
W ( ) 1,    P W ( ) ,  S   

DSP-CIS / Chapter-4: FIR & IIR Filter Design / Version 2014-2015 p. 35


FIR •Filter Design
A simpler problem by
is obtained by Optimization
replacing the F(..) by…

where the wi’s are a set of (n) selected sample frequencies


This leads to an equivalent (`discretized’) quadratic optimization function:

Q  W (i ).c(i ).cT (i ), p  W (i ). Ad (i ).c(i ),   ...


i i

1
xOPT Q . p Compare to p.10

+++ : simple
- - - : unpredictable behavior in between sample freqs.
FIR Filter Design by Optimization
• This is often supplemented with additional constraints, e.g. for pass-band and stop-band ripple
control : A( P ,i )  1  P , for pass - band freqs.  P ,1 ,  P , 2 ,... ( P is pass - band ripple)
A( S ,i )  S , for stop - band freqs.  S ,1 ,  S , 2 ,... ( S is stop - band ripple)

• The resulting optimization problem is :


minimize : (=quadratic function)

AP .x bP
subject to (=pass-band constraints)
A .x b
S S
(=stop-band constraints)
= `Quadratic Programming’ problem
FIR Filter Design by Optimization
(II) `Minimax’ Design :
• Select one of the basic forms that yield linear phase
e.g. Type-1

• Specify desired frequency


H d (response
) e  jN / 2(LP,HP,BP,…)
. Ad ( )

• Optimization criterion is

W ( ) 0
where is a weighting function
• Leads to `Semi-Definite Programming’ (SDP) problem, for which efficient interior-
point algo’s & software are available.
FIR Filter Design by Optimization
• Conclusion:
(I) weighted least squares design
(II) minimax design
provide general `framework’, procedures to translate filter design problems into
standard optimization problems
• In practice (and in textbooks):
emphasis on specific (ad-hoc) procedures :
- filter design based on `windows’
- equiripple design
FIR Example
Filter: Low-pass
Design using `Windows’
filter design
• Ideal low-pass filter is

1   C
H d ( ) 
0  C     
• Hence ideal time-domain impulse response is

• Truncate hd[k] to L+1 samples (L even):

• Add (group) delay to turn into causal filter


FIR Filter Design using `Windows’
Example : Low-pass filter design (continued)
• PS : It can be shown (use Parceval’s theorem) that the filter obtained by such time-domain truncation is
also obtained by using W ( ) 1 least-squares design procedure with the given Hd, and weighting
a weighted
function
• Truncation corresponds to applying a `rectangular window’ :
h[k ] hd [k ].w[k ]

• +++: Simple procedure (also for HP,BP,…)


• - - - : Truncation in time-domain results in `Gibbs effect’ in frequency domain, i.e. large ripple in pass-band
and stop-band (at band edge discontinuity), which cannot be reduced by increasing the filter order L.
FIR Filter Design using `Windows’
Remedy:
Apply windows other than rectangular window

• Time-domain multiplication with a window function w[k] corresponds to frequency domain convolution
with W(z) : h[k ] hd [k ].w[k ]

H ( z ) H d ( z ) *W ( z )

• Candidate windows : Han, Hamming, Blackman, Kaiser,…. (see textbooks, see DSP-I)
• Window choice/design = trade-off between side-lobe levels (define peak pass-/stop-band ripple) and
width main-lobe (defines transition bandwidth)
FIR Equiripple Design
• Starting point is minimax criterion, e.g.

• Based on theory of Chebyshev approximation and the `alternation theorem’, which (roughly) states that
the optimal d’s are such that the `max’ (maximum weighted approximation error) is obtained at L o+2
extremal frequencies…

…that hence will exhibit the same maximum ripple (`equiripple’)


• Iterative procedure for computing extremal frequencies, etc. (Remez exchange algorithm, Parks-
McClellan algorithm)
• Very flexible, etc., available in many software packages
• Details omitted here (see textbooks)
FIR •Filter Design
FIR Filter design Software
abundantly available in commercial
software
• Matlab:
b=fir1(L,Wn,type,window), windowed linear-phase FIR design, L is filter order,
Wn defines band-edges, type is `high’,`stop’,…
b=fir2(L,f,m,window), windowed FIR design based on inverse Fourier transform
with frequency points f and corresponding magnitude response m
b=remez(L,f,m), equiripple linear-phase FIR design with Parks-McClellan
(Remez exchange) algorithm
REFERENCES
https://ocw.mit.edu/courses/materials-science-and-engineeri
ng/3-091sc-introduction-to-solid-state-chemistry-fall-2010/a
morphous-materials/21-introduction-to-glasses/
https://www3.nd.edu/~amoukasi/CBE30361/Lecture__crysta
llography_A.pdf
http://ctc.xmu.edu.cn/jiegou/ppt/chapter7.pdf
https://swayam.gov.in/nd1_noc20_mm17
eng.libretexts.org/Bookshelves/Materials_Science
Supplemental_Modul_es
(Materials_Science)/Semiconductors/Band_Theory_of_Semic
onduCtors
hyperphysics.phy-astr.gsu.edu/hbase/Solids/band.html
45
REFERENCE BOOKS
• Sedra, A. S., Smith, K. C., and Chandorkar, A. N., “Microelectronic Circuits: Theory and Applications”,
7th edition, Oxford University Press, 2017.
• Haykin, S. and Moher, M., “Communication Systems” 5th edition, Wiley, 2018.
• Sklar, B., “Digital Communication - Fundamentals and Applications”, 2nd edition, Prentice-Hall India,
2017.
• Senior, J, M., “Optical Fiber Communications: Principles and Practice”, 3rd edition, Pearson Education,
2010.
• Theodore and S. Rappaport, “Wireless Communications, Principles, Practice”, 2nd Ed, PHI, 2002.
• Kang, S. M., Leblebici, Y., and Kim, C., “CMOS Digital Integrated Circuits: Analysis and Design”, 4th
edition, McGraw Hill Education, 2016.
• Wolf, M., “Computers as Components: Principles of Embedded Computing System Design”, 3rd
edition, Elsevier, 2013.
• Krishna, C. M. and Shin, K. G., "Real-Time Systems", 1st edition, McGraw Hill Education, 2017.
• Pozar, D. M., “Microwave Engineering”, 4th edition, Wiley, 2013.

46
REFERENCE VIDEOS
https://nptel.ac.in/courses/104/106/104106093/
https://www.coursera.org/lecture/materials-science/crystallograp
hy-and-the-electron-microscope-YhPAB
www.youtube.com/watch?v=mwlWIfNvt1w
www.youtube.com/watch?v=zdmEaXnB-5Q
https://www.youtube.com/watch?v=A56UtkYYExg
 https://www.youtube.com/watch?v=x3LOTaEVLak
https://www.youtube.com/watch?v=1DHCV3LgITY

47
THANK YOU

For queries
Email: [email protected]

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