Redactor ef / Editor in chief

Prof. dr. ing. Ioan Naforni

Colegiul de redacie / Editorial Board:

Prof. dr. ing. Virgil Tiponu
Prof. dr. ing. Alexandru Isar
Conf. dr. ing. Dorina Isar
Prof. dr. ing. Traian Jurc
Prof. dr. ing. Aldo De Sabata
As. ing. Maria Kovaci - secretar de redacie
As. ing. Maria Kovaci - tehnoredactor
DEANS WELCOME SPEECH

Dear colleagues,

I wish you a warm welcome to the sixth edition of the Symposium of Electronics and
Telecommunications.
I am delighted to announce you that, from its first edition of 1994, a continuous increase in both
quantity and quality of papers, submitted by well known researchers from universities and
industry, from Romania and abroad, can be clearly identified. With its 13 sections and over 170
papers, the Symposium Proceedings cover wide areas in its field.
Besides the scientific and informative value of the published volumes, another purpose of our
Symposium is to be, for all participants, a forum for exchange of ideas and socialization, for
emphasizing the feeling of belonging to a highly specialized and expert community.
Although research results are the focus of a scientific conference, we cannot forget that higher
education is in a process of profound transformations. Therefore, we took advantage of the
presence of an important number of high ranked personalities from universities to organize a round
table on the Impact of the Bologna Declaration on Electronics and Telecommunications
Education. We hope that the ideas exposed by the participants and the common conclusions will
constitute guidelines for our future common action.
Our region has been target of important investments in the last 14 years, which led to an
important industrial development in the field of electronics and telecommunications. Recently we
have celebrated the graduation of the 30th generation from our faculty. What these two facts have
in common is the highly trained human resource exiting our faculty and entering the industry.
Fortunately, a feedback mechanism has been created, as industrials understood the necessity of
supporting education. This symposium would not have been possible without the gracious help of
our sponsors, to whom we express our gratitude.
I am also honored to express my thanks to you, all the participants, for attending the Symposium,
to wish you a successful and profitable audience through the sessions, and a nice stay in Timisoara.
I am looking forward of meeting you again in 2006.

Dean,
Prof. Dr. Eng. Marius Otesteanu

Timisoara, October 21st, 2004

 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Table of Contents

Information Theory and Coding

V. Bota, M. Varga, Zs. Polgar

Performances of LDPC-Coded QAM constellations employing non-coded bits... 7

Zs. Polgar, A. Nastase, V. Bota

Performances of the Reed-Solomon codes decoded with the Guruswami-Sudan algorithm. 13

S. V. Halunga, O. Fratu
New interleaver design algorithms with enhanced B.E.R. performances.. 19

R. Radescu, R. Popa
On the performances of symbol ranking text compression method... 25

R. Stoian, L. A. Perisoara
Application of turbo principle to product codes 28

H. Balta, M. Kovaci
A study on turbo decoding iterative algorithms. 33

H. Balta, M. Kovaci
The Performances of Convolutional Codes used in Turbo Codes. 38

S. Popescu
A new approach on delay coding: the receiver.. 44

G. G. Fericean, M. Borda
Selective encryption of image with IDEA algorithm.. 50

L. Scripcariu, P. Duma
Analysis of simple inversable functions defined on Galois fields for cryptography use 55

R. Stoian, L. A. Perisoara
Parallel concatenated convolutional turbo codes: performance analysis for different interleaving
schemes.. 60

R. Radescu, I. Balasan
Evaluation of parameters used in lossless text compression with the Burrows-Wheeler transform. 65

Signal Processing

C. Paleologu, S. Ciochina, A. A. Enescu

A low dynamics fast transversal filter adaptive algorithm 69

C. Partheniu
Gradient algorithms with improved convergence.. 75

1
E. Szopos, N. Toma, M. Topa
Adaptive filtering algorithms. 81

C. Chioncel, J. Gal
Parameter estimation of the chirp signal... 87

G. Budura, C. Botoca
Efficient implementation of the second order Volterra filter. 91

L. Grama
Phase approximation using signals affected by random perturbations. 96

L. Stanciu, L. Banu
Reconstruction methods for missing portions in signals... 102

D. Tarniceriu, V. Munteanu
Information theoretic approach of filterbanks performing energy compaction 106

A. Quinquis, A. Isar, D. Isar

Denoising over-sampled signals 110

M. A. Matin
Invisible watermarking for Copyright Protection.. 114

D. Osborne, D. Abbott, M. Sorell, D. Rogers

Multiple embedding using semi-fragile watermarks for medical images.. 120

C. Nafornita
A wavelet-based watermarking for still images. 126

A. Vlad, A. Luca
The statistical behaviour of the chaotic signals: application to cryptography.. 132

R. Arsinte, C. Ilioaei
Considerations and results in Multimedia and DVB application development on Philips Nexperia
Platform. 138

I. G. Mocanu
Shape similarity measure for k nearest-neighbor queries. 142

I. G. Mocanu
Shape representation and retrieval using centroid radii and turning angle. 146

A. Stoica, C. I. Vizitiu, I. Nicolaescu, L. Anton

Considerations about the design requirements for analog anti-aliasing filters 150

Microcontrollers
P. Duma
Software setting telephone links using ATMEL microcontrollers in time switching network PABX 154

S. Zoican
Variable step size affine projection adaptive algorithm implementation.. 160

A. A. Enescu, C. Paleologu, S. Ciochina

QRD-LSL algorithm suitable for implementation on D.S.P .. 165

D. Silion, D. Panaitopol, M. S. Rusu

An improved LMS algorithm for single and doubletalk echo canceller implemented on Motorola
DSP SC140. 170

2
M. Otesteanu, D. Criste
Precision electronic driver for pneumatic engines 175

A. Gontean, M. Bbi, R. Jibleanu

Hardware Simulation and Debugging For Microchip RISC Microcontrollers. 180

M. Bbi, A. Gontean, R. Jibleanu

General Purpose PIC16F84A Based Development Board 184

P. Duma, L. Scripcariu
Development system equipped with AT89S8252 microcontroller. 187

D. V. Ene, R. M. Udrea, I. Pirnog, C. Sucholotiuc

Digital video broadcasting terrestrial modulator implemented on Motorola MSC8101 DSP.. 193

T. Ionica, C. Balint
Telephone interface for remote control systems with network capabilities... 197

Instrumentation and Measurement

M. S. Crainic
Prepayment gas meter a new trends in natural gas metering technology.. 201

S. Moraru, D. Faur, B. Pantelimon, C. Voina, A. Cosac

Currents and voltages measurements using aquisition board DAQ 6024 E. 205

L. Dragomir, I. Sandu, B. Pantelimon

Considerations regarding the factors involved in calibration of DC voltage standard calibrator... 209

V. Simion, B. Pantelimon, C. Stefanescu

Polarization method for electrical current measurement.. 213

E. Buzac, I. Urdea Marcus, C. Cepisca

General considerations regarding the measurement and the harmonization of the national system
of standards in the field of the public domain of measuring electric energy, according to the
requirements of the European and international standardization organizations.. 217

V. Dogaru Ulieru, C. Cepisca, H. Andrei, A. Husu, C. Dogaru Ulieru, T. Ivanovici

Measuring current-voltage (i-v) characteristics for non-linear electrical devices...
221

D. Belega
Accurate sinewaves implemented with a 16-bit fixed-point digital signal processor 225

S. Mischie
On frequency measurement by using zero crossings. 230

L. Toma Jr., F. Shu, A. Ignea, W. Neddermeyer, M. Schnell

The development of a new system to measure Camber and Toe using stereo cameras. 236

S. C. Ionel
A correlation analysis of measured CO-concentration signals. 240

S. Mereuta
Analysis of biomolecular sequences through spectral based methods.. 244

C. I. Dumitrescu
K - complex Detection using the Continuous Wavelet Transform. 247

3
D. Stoiciu, M. Lascu
PC-based system for automated calibration of a digital voltmeter... 253

A. Ignea, A. Mihaiut
The measurement of dc magnetic field... 255

Microwaves and Optical Communications

A. Rosu-Niculescu, T. Petrescu
The simulation of the effect of the geometry of the Rectangular Double Barrier structure about the
transmission and reflection coefficient.. 259

S. Simion
MEMS based broadband phase shifters 265

F. Toadere
Optical coherent and incoherent systems frequency analyze in Cartesian coordinate. 271

F. Toadere
Cartesian coordinate optical filter analyses.. 275

F. Toadere
Space and frequency analyze of an LSI optical system with different input signals.. 280

M. G. Banciu, G. Lojewski, L. Nedelcu, D. Ghetu, N. Militaru, D. Brinaru, T. Petrescu

Design of small-size planar filters using FDTD and wavelet analysis.. 285

I. Rados,T. Sunaric, P. Turalija

Suggestions for availability improvement of optical cables.. 289

M. Telescu, L. Ghisa, P. Besnard, A. Mihaescu

Simulations of impulse response for diffuse indoor wireless channels.. 294

L. Ghisa, M. Telescu, P. Besnard, A. Mihaescu

Experimental characterization of impulse response for optical indoor wireless channels... 297

E. Teodoru , S. Demeter
A discrete model for reference source noise in indirect frequency synthesis 300

Education in Electrical Engineering

V. Tiponut
A Physical Laboratory for Smart Transducers Education. 305

G. Sirbu, D. Aiordachioaie
On radio spectrum measurements with the ESVB Rohde & Schwarz test receiver... 309

G. Sirbu, D. Aiordachioaie
On GSM mobile phone measurements with the CTS-65 Rohde & Schwarz digital radio tester... 313

A. De Sabata, L. Matekovits
Scattering parameters of symmetrical networks 317

M. Cremene, I. Benta, L. Chira, C. Loghin

A management service for student examination results with nomadic access... 323

4
G. Oltean, E. Sipos, I. Oltean
A new approach of op-amp amplifier biasing 328

S. V. Tiponut
A toolkit for internet based distance laboratory development... 332

R. Popa
A complete laboratory on evolutionary electronics... 335

B.Orza, M. Givan, A. Vlad, A. Olah, A.Vlaicu

IeL Com an integrated module for communication in E-learning.. 341

S. Ionel, M. Daneti
Low-cost electronic board improves electronics laboratory efficiency. 346

D. Stoiciu, C. Dughir
A web-based teaching tool for laboratory classes. 348

C. Popescu, A. Rapa, P. Svasta

Monitoring Network Activities in Web Based Training Courses... 350

Wireless Communications
D. M. Dobrea, N. Cleju, A. T. Sechelea, A. Banar
Mobile accident warning system -The LoRD- 354

A. A. Enescu, S. Ciochina
An improved MIMO-OFDM channel estimator in the tracking phase.. 360

D. Andrei, C. Vladeanu, A. Serbanescu

Performance evaluation of a multiple access dcsk system under a noisy multiuser environment. 366

A. F. Paun, S. G. Obreja
A low complexity decision feedback equalization for sparse wireless channels... 370

S. G. Obreja, A. F. Paun
Terestrial digital video broadcasting (DVB-T). System performances simulation 378

C. Comsa, D. Burdia, D. Chiper

Implementation of an OFDM synchronizer... 382

C. Comsa, F. Beldianu, P. Cotae

Windowing techniques for OFDM systems 385

M. Oltean, E. Marza, M. Nafornita

BER performances of a differential OFDM system in fading channels. 389

C. Vladeanu, R. Lucaciu, D. Andrei

Optimal chaotic asynchronous DS-CDMA communications over frequency-nonselective rician
fading channels.. 394

I. I. Duma
Noise impulse generation with convenient characteristics in time and frequency domain... 398

F. Craciun, C. Mateescu, O. Fratu, S. Halunga

UWB communications systems based on orthogonal waveforms set. 403

M. Moise
Mobility concept for wireless ATM networks 409

5
M. Moise
Lossless handover scheme for Mobile ATM networks... 415

A. Sikora
Design rules for lightweight short-range wireless networks. 421

M. Oltean, A. Vesa, E. Marza

Performance evaluation of single-carrier broadband transmission with frequency domain
equalization 425

P. Bechet, St. Demeter, R. Mitran, S. Miclaus

Some aspects about frequency hopping radio networks 431

M. Salagean, C. Ioana, A. Quinquis

Recognition of OFDM modulations : approach based on high-order time-frequency methods... 434

Index of Authors... 439

6
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Performances of LDPC-Coded QAM Constellations

Employing Non-Coded Bits
Vasile Bota, Mihaly Varga, Zsolt Polgar 1
Abstract The paper deals with the BER and
Throughput vs. SNR performances of the LDPC-coded
QAM signal constellations modulated on OFDM-type
multicarrier transmissions. It analyzes, by means of
computer simulation, the influences of the LDPC-code
parameters (codeword length and rate) and the effects of
mapping various ratios of coded and non-coded bits on
QAM symbols, upon the SNR performances of these
transmissions. Comparisons to the Shannon limits are
presented as well.
Keywords: LDPC codes, Bit Error Rate, Throughput
I. INTRODUCTION
The LDPC codes are powerful block error correcting
codes that provide high coding gains, comparable to
the ones provided by the turbocodes for the same
coding rate, at the expense of a simple encoding and
moderate decoding complexity.
A. Parameters and Types of LDPC Codes
The Low Density Parity Check codes are block codes,
basically defined by three parameters p, j, and k
(integer numbers) that observe the following
conditions [1]:
p is a prime integer; j < k p; (1)
The codeword length N, the number of control bits C,
the number of information bits J and the code ratio RC
are defined, respectively, by [1]:
N = kp; C = jp; J = (k-j) p; RC = (k-j)/k; (2)
The control matrix, H, of a LDPC code is a 4-cycle
free sparse matrix that might take three forms, which
define the three types of LDPC codes: randomly
generated by computer search [2], [3], by complete
array-code control matrices [3] and by triangular-
shaped array-code control matrices [4].
This paper considers only the codes generated by a
triangular-shaped array-code control matrix H (jp x
kp).
The generic form of the triangular-shaped matrix HT
is generated, see (3), by using an elementary matrix ,
pxp, and the unity and null pxp matrices, I and 0:
I I I . I I . I
0 I . j2 j1 . k 2
H T = 0 0 I . 2( j3) 2( j2) . 2(k 3)
M M M M M M M M
0 0 0 .
I j1 . ( j1)(k 1)
(3)
Two different types of elementary matrices that may
be employed; they can be generated starting from the
unity pxp matrix, either by shifting its rows
downwards or upwards with one position.
The code parameter j equals the number of control
equations a codeword bit is involved in, while code
parameter k shows the number of bits involved in a
control equation.
B. Generation of the Control Matrix HT
The regular structure of the triangular-shaped matrix
allows a systematic generation, starting from the code
parameters j, k, p. Using the property of the matrix ,
p = I (pxp), and the rule that gives the power of
inside the HT (3), an algorithm to compute the indexes
(row, column) defining the positions that take the
logical value 1, in terms of the parameters j, k and
p, can be determined. So, the binary matrix HT is
generated by filling with 1 only on the positions
given by that algorithm.
C. Shortening the LDPC Codes
In order to adapt the number of information bits of the
codeword to the information source or to adapt the
codeword length, to the transmitted symbol-packet,
the LDPC codes can be shortened [1].
The shortened code rate R is smaller than the rate of
the parent code R:
R = J/(J+C) < R = J/(J+C); (4)
D. Encoding the LDPC Codes
If the codeword is v = [c0,,cjp-1, i0,,i(k-j)p-1], the
control bits cm are computed in terms of the infor-
mation bits il by solving the C equations system:

1
Technical University Cluj-Napoca, Faculty of Electronics and Telecommunications, Communications Department,
str. G. Baritiu Nr.25, 400427, Cluj-Napoca Romania; e-mail: Vasile.Bota@com.utcluj.ro

7
 Hvt = 0
This approach has two major shortcomings:
- for great values of parameters j and/or p, C
becomes large implying a significant computational
load that increases the processing time and/or the
hardware required by the implementation;
- it requires all information bits il, l = 0,..., (k-j)p-l
at the same time; this requirement induces a one-
codeword additional latency in the system.
These shortcomings may be avoided by a simpler and
faster encoding method, described in [5].
E. Bit-Mapping on the QAM Signal Constellation
When the LDPC-coded bits are to be modulated on a
b-bit/symbol QAM constellation, the b-tuple is
mapped on the I and Q coordinates of the QAM
vector, by splitting the b-tuple into two groups of b/2
bits, each group being assigned to one axis. The bits
that are assigned to an axis are mapped to the
amplitude levels of that axis according to a Gray
encoding [1]. Since the transmission bit-loading might
involve non-coded information bits, they are also
mapped according to a separate Gray encoding, in
order to maximize the distance between levels having
the same non-coded bits. Therefore, the multibit
assigned to an axis of the QAM constellation, coded
and non-coded bits, is mapped according to a 2-level
Gray encoding described in [3]. Fig. 1 presents an
example of mapping b/2 = 4 bits on one axis (I or Q)
of a QAM constellation.
-15 -13 -11 -9 -7 -5 -3 -1 1 3
0000 0001 0011 0010 0100 0101 0111 0110 1100 1101 1111 1110 1000 1001 1011 1010
Coded bits Non-coded bits
Distance between groups, dg Distance within a group dig
Fig. 1. Bit mapping on a QAM-constellation axis using the 2-level
Gray encoding
The amplitude levels employed on each axis belong to
the set A defined by:
A = {Al = 2l-(Lb-1); l = 0,1,Lb-1;}; Lb = 2b/2; (6)
This bit-mapping method, which allows only for the
employment of the square QAM constellations, is
simpler because the mapping is identical on both
coordinates.
F. Decoding the LDPC Codes
The decoding of the LDPC codes is accomplished by
using the message-passing algorithm (MP), as
presented in [2], which will not be described here.
This algorithm, based on the Bayes criterion, requires
the previous computation of the a posteriori
probabilities for every bit of a codeword, Fn0(r/0) and
Fn1(r/1), where r denotes the received vector, n is the
bit index and 0/1 denote the bit logical value.
Basically, the algorithm performs the decoding of a
codeword by using the a posteriori probabilities of a
(5) group of N bits v that is checked by means of
syndrome-computation; if the syndrome equals zero,
the algorithm considers v to be the correct codeword;
otherwise it performs another iteration adjusting the
values of the a posteriori probabilities by using some
internal values computed in the previous iteration.
The maximum number of iterations allowed, B, is a
parameter of the algorithm. The values of the a
posteriori probabilities are previously extracted from
the OFDM-demodulated coordinates I and Q of each
QAM-symbol, by means of the soft-demapping
procedure.
This algorithm does not search for the closest
codeword compared to the received sequence, but
tries to correct every bit. Due to this property, the
number of error bits after the decoding is always
smaller than the one of error bits prior the decoding,
when the algorithm is convergent. Extensive
simulation performed by the authors confirmed this
property, which might lead to the decrease of error-
packet length that should be corrected by the RS code
that follows the LDPC or convolutional codes in
many applications.
G. Soft-Demapping
Because the MP algorithm requires the a posteriori
probabilities of each bit and the received vector
carries more bits, a soft-demapping [3] is required in
order to provide the Fn0(0/r) and Fn1(1/r) probabilities
of each bit mapped the received vector.
For multibit/symbol modulations, the two probabilities
5 7 9 11 13 15 of each bit are extracted, from the received level on the
I or Q branches, by using (7) that gives the probability
of bit bj to be 1 when the demodulated level on a
branch equals r and the channel is AWGN [1]:
2b / 2
(r L(l)) 2
exp( 2 2
) b lj
(7)
Fj1 = l=12b / 2 ; j = 0,..., b / 2 1;
(r L(l)) 2
exp(
2 2
)
l =1
In (7) blj denotes the logical value of j-th bit of the l-th
modulating level of the I or Q branch of the
demodulated vector. A similar expression is derived
for Fj0 and the two values are normalized to their sum.
The soft-demapping requires a previous estimation of
the noise variance ; computer simulations run by the
authors showed that estimation errors of less than 2 dB,
between the actual channel noise variance and the one
stored in the soft-demapper, lead to insignificant
decreases of the decoder performances.
H. Soft Decision of the Non-Coded Information Bits
The information non-coded bits mapped on a QAM
symbol can be decided by two methods, namely:
- hard decision, applying the Bayes criterion to the
probabilities provided by the soft-demapping; this
method does not employ the information provided by
8
decoding the coded bits placed on the same tone It displays the BER values and the BER vs. SNR
during the same symbol period. characteristic for the selected SNR range, the number
- soft decision, that considers the information of coded bits error after the decoding of each
provided by the decoding of the coded bits mapped on codeword, and the number of non-coded bits decided
the same QAM symbol and tone. by soft-demapping. It also displays the throughput of
Basically, the optimal decision memorizes the the transmission for a defined packet dimension.
received level r and, using the decoded bits provided The simulations were performed on a test of 106
by the LDPC decoder, selects the closest (in the dE information bits and the maximum number of B = 15
sense) level that was mapped with the same decoded iterations/codeword for the decoding algorithm.
bits, see fig. 2. This method provides lower BER of
B. Effects of the Coding Rate upon the BER
the non-coded bits, as resulted from simulations Performances of LDPC-Coded QAM Constellations
performed by the authors, but may error the non-
coded bits if the corresponding coded bits were As shown in (2) the coding rate might be changed,
wrongly decoded. without changing the codeword length, by changing
PL PL OD r HD PL PL the parameter j. Considering k=14 and p = 31, so that
-15 -13 -11 -9 -7 -5 -3 -1 1 3 5 7 9 11 13 15 a short codeword N = 434 bits (see (2)) is used, a
0000 0001 0011 0010 0100 0101 0111 0110 1100 1101 1111 1110 1000 1001 1011 1010
family F1 of LDPC codes with RC ranging from 0.78
Coded bits PL possible levels for 10 coded bits to 0.21 is displayed in table 1. The family includes the
Non-coded bits
OD optimal decision non-coded configuration for comparison.
HD hard (Bayes) decision
Distance between groups, dg Distance within a group dig
Table 1. Parameters of F1 LDPC codes; k = 14, p = 31
Fig. 2. Optimal decision of non-coded bits for 10 coded bits
F1 j N C R
I. Bit-loading on Coded-QAM OFDM Employing C11 Non-coded 1
Non-Coded Bits
C12 3 434 93 0.78
Denoting by T the number of available sub-carriers,
by DOFDM the OFDM symbolrate, by Nci and Nni the C13 7 434 217 0.50
numbers of coded and non-coded bits on the i-th C14 9 434 279 0.35
subcarrier and by RC the LDPC code rate, the nominal
payload Dn of the OFDM transmission would be (8). C15 11 434 341 0.21
The number of bits actually carried by each QAM-
symbol equals Nc + Nni. In order to evaluate the correction capability of these
codes, the BER vs. SNR performances were evaluated
D n = D OFDM T ( N ci R C + N ni ); (8) employing a 2-PSK constellation. And are shown in
fig. 3.
The rate of the coded QAM-modulation is computed
by:
N ci R C + N ni ; (9)
R CM =
N ci + N ni

II. BER PERFORMANCES

OF THE LDPC-CODED QAM CONSTELLATIONS
Due to the complexity of the theoretical evaluation of
Fig. 3. BER vs. SNR of 2-PSK coded with the LDPC codes of
the BER provided by the LDPC coded QAM family F1
constellations that are also loaded with non-coded
bits, the authors developed a computer simulation The coding gains provided by the F1 codes, at BER =
environment that was employed to derive the results 110-6, and their rates are displayed in table 2.
shown below.
Table 2. Coding gains provided by the F1 codes
A Simulation Environment and Parameters Code C11 C12 C13 C14 C15
The simulation program that implements the LDPC- CG (dB) 0 6 8.5 9 9.5
coded multi-carrier transmissions, allows the follow- RC 1 0.78 0.5 0.35 0.21
ing parameters to be set: LDPC code parameters (k, j,
As expected, the coding gains provided increases with
p), number of sub-carrier groups G, number of sub-
the decrease of the coding rate.
carriers within a group Ti, bit-loading for each group
The coding gain provided by the parent code (RC =
(number of coded bits - Nci, number of non-coded bits 0.78) is about 6.5 dB, comparable to the ones of
- Nni), maximum number B of iterations/codeword of convolutional codes of R = and K = 5 7,
the LDPC decoder, range and step of SNR, Rayleigh The R = LDPC code provides a coding gain of 8.5
channel model, test length. dB, larger than the one the convolutional codes.

9
Taking into account the fact that the MP decoder has C. Effects of the Codeword Length upon the BER
about the same implementation complexity as the 64- Performances of LDPC-Coded QAM Constellations
state Viterbi decoder we may say that the LDPC
codes provide about the same coding gain as the In order to evaluate the effects of the codeword length
convolutional code, at a higher coding rate. upon its performances we consider a family of codes
Comparing the coding gains with the ones of the turbo F2, see table 4, with RC = 0.214 (as code C15 of F1).
codes, the RC = LDPC code ensures BER =10-6 at The code word length N is modified by means of
a SNR = 2 dB, about 1 dB higher than the turbo-codes parameter p, ranging from 23 to 73.
[6], but it requires only one MAP decoder instead of Table 4. Codes of rate Rc = 0.214 and various
two decoders (MAP + MLD) and a de-interleaver. codeword length N family F2
By decreasing the code rate by changing the code Code k j p N [bits]
parameter j additional coding gains of up to 3.5 dB C151 14 11 23 322
can be obtained. The code C15 (RC = 0.21) requires a C151 14 11 31 434
SNR = 1dB to provide a BER = 10-6. C153 14 11 43 602
In order to evaluate how close these codes are to the C154 14 11 53 742
Shannon limit, we recall that the maximum spectral
C155 14 11 73 1022
efficiency wMi, which could be provided by a code of
rate Rci in an error-free transmission across an fs The BER vs. SNR performances of these codes are
bandwidth, is: displayed in fig.4.
1 S
wMi = lg 2 (1 + R ci ) [bps/Hz]; (10)
2 N
Assuming that the error-free transmission is
accomplished for SNRs > SNR0i, where:
BER (SNR0i) = 110-5; (11)
we may compute the maximum spectral efficiency,
wM0i, that could be accomplished by a code with rate
RCi, by using (10) and SNR0i obtained by simulations Fig. 4 BER vs.SNR of LDPC codes from table 4
(see fig.3). This value, compared to the actual spectral
efficiency wi, provided by the code (which equals The results in fig.3 indicate that by increasing the
Rci) indicates how far is the code from the theoretical codeword length, extra coding gain, can be provided
limit. at the same coding rate, but at the expense of a more
Another evaluation can be performed by computing difficult implementation. The extra coding gain may
the minimum signal/noise ratio, SNRmi, for which wi be as high as 1.5 dB, see codes C152 and C155 in fig.4
could be obtained. The difference SNRi = |SNRmi - on one hand and code C15 in fig.3 on the other. The
SNR0i| shows the quality of the code in terms of the code C155 provides a total 11 dB coding gain.
SNR. Increasing the codeword length would also bring the
The values of these parameters for the codes of table 1 code performances closer to the Shannon limit.
are shown in table 3 Computing the parameters defined in table 3 for the
codes of family F2 we get the values of table 5.
Table 3. Ideal and actual performances for codes of
family F1 Table 5. Ideal and actual performances for codes of
Code wi wMi SNR0i SNRmi SNRi family F2
Ci [bps/Hz] [bps/Hz] [dB] [dB] [dB] Code wi wMi SNR0i SNRmi SNRi
C12 0.785 1.59 4.1 -0.35 4.45 [bps/Hz] [bps/Hz] [dB] [dB] [dB]
C13 0.5 0.80 1.7 -0.82 2.52 C151 0.214 0.352 0.5 -1.26 1.76
C14 0.357 0.546 1.1 -1.04 2.14 C152 0.214 0.291 0.2 -1.26 1.46
C15 0.214 0.29 0.1 -1.26 1.36 C153 0.214 0.280 0.0 -1.26 1.26
C154 0.214 0.274 -0.1 -1.26 1.16
As expected, codes of a certain length come closer to
the theoretical limits as their rate decreases. Results of C155 0.214 0.268 -0.25 -1.26 1.01
tables 2 and 3 show that rather short codes, easy to The longest code of family F2, C155, is about 1 dB
implement, are close enough to the theoretical maxi- away from the Shannon limit, for BER < 110-5.
mum performances. Simulations performed by the authors showed that the
The performances of the LDPC codes might be coding gains provided by the LDPC-coded QAM
improved by increasing the maximum number of constellations are about the same, compared to the
iterations/codeword of the MP decoder; simulations same non-coded QAM constellations, regardless the
showed that increasing B = 25, leads to extra coding constellation employed.
gains of 0.5-1 dB, at the expense of a longer proces-
sing time required.

10
 III. LDPC-CODED QAM CONFIGURATIONS
EMPLOYING NON-CODED BITS
The employment of error-correcting codes leads to a
decrease of throughput provided by the coded QAM
constellation. This occurs due to the control bits
inserted by the code, and the throughput decrease is
higher for low-rate codes that secure a good
correction capability.
In order to reach a reasonable trade-off between the
correction capability and the coding rate, which
affects significantly the throughput, each QAM
symbol is loaded with nci coded bits and with nni non-
coded bits.
Considering a LDPC code with an RC coding rate the
coding rate of the configuration employing coded and
non-coded bits is expressed by (9).
The coded and non-code bits are mapped on the I and
Q coordinates (6) of the QAM symbol using the 2-
level Gray mapping, see fig.1, on each axis.
A. BER Performances of the LDPC-Coded QAM
Configurations Employing Non-Coded Bits
To evaluate the effects of employing non-coded bits, a
family F3 of possible LDPC-coded configurations
using non-coded bits based on the 256-QAM (Nci +
Nni = 8 bits) are presented in table 6 together with
their coding rates RCM and coding gains CG. The
LDPC code employed is (k = 14, j = 3, p = 31; RC =
0.78), which provided a 6.5 dB coding gain on a 2-
PSK modulation (code C12 in table 1 and fig. 3). The
BER vs. SNR performances of this family are
presented in fig. 5.
Table 6. Coded QAM Configurations of Family F3
Cfg. Nci Nni RCM CG
1 0 8 1 -
2 2 6 0.945 5
3 4 4 0.890 6
4 6 2 0.835 6.5
5 8 0 0.780 7
Fig. 5. BER vs. SNR for configurations of F3 defined in table 6
As shown in table 6, the employment of non-coded
bits leads to a significantly increase of the coding rate,
at the expense of a coding gain decrease of about 1-2
dB. The coding rate increases starting from 0.78 up to
0.945, in terms of the proportion of non-coded bits
within the 8 bits loaded on a QAM symbol.
The relatively small decrease of the coding gain
could be explained by the protection of the non-
coded provided by the 2-level Gray mapping and by
their soft decoding. Fig. 6 shows the BER vs. SNR
11
curves of the coded and non-coded bits before the MP
decoding (for the coded bits) and before the soft
decision (for the non-coded bits), and after these
decoders. The BER prior to decoding was obtained
by using a hard Bayes decision that employs the a
posteriori probabilities provided by the soft-
demapping. The curves correspond to configuration
no.3 from table 6.
1
BER
2
4 3
SNR
Fig.6. BER vs. SNR of the coded and non-coded bits of
configuration 3 from table 6; line 1 coded bits Bayes decision;
line 2 non-coded bits Bayes decision; line 3 coded bits MP
decision; line 4 non-coded bits soft decision
As shown in figure 6, the non-coded bits have lower
BER than the coded bits loaded on the same QAM
symbols, both before their decoding (line 1 vs. line 2)
and after it (line 3 vs. line 4). This is due to the 2-level
Gray mapping of the coded and non-coded bits.
Fig. 6 shows that the number of error bits is always
smaller after the decoding process, than before it.
This, combined with additional simulations performed
by the authors indicate that the two decoders might
require some smaller outer codes (small RS or even
BCH) in FEC schemes employing concatenated
codes.
The spectral efficiencies of the configurations
employing non-coded bits are higher than the ones of
the proximity to the Shannon limit of the
configurations from family F3 are displayed in table 7.
Table 7. Ideal and actual performances of
coded configurations from family F3
wi wMi SNR0i SNRmi SNRi
Cfg RCMi
[bps/Hz] [bps/Hz] [dB] [dB] [dB]
1 1 8 10.46 31.5 24 7.5
2 0.945 7.52 9.05 27.5 22.9 4.6
3 0.890 7.12 8.64 26.5 21.9 4.6
4 0.835 6.68 8.21 25.5 20.9 4.6
5 0.780 6.24 8.31 25.0 19.8 5.2
The spectral efficiencies of the coded configurations
carrying non-coded bits (lines 2, 3, 4 in table 7) are
higher than the ones of the coded configuration with
no non-coded bits (line 5). Also they are closer, in the
SNR sense, to the Shannon limit.
C. Throughput Performances of the LDPC-Coded
QAM Configurations Employing Non-Coded Bits
The employment of the non-coded bits decreases the
coding gain leading to a higher BER at a given SNR,
on one hand, and increases the coding rate leading to
more information bits transmitted, on the other. The
effects of these two factors upon the throughput
provided by such configurations are shown below.
For throughput evaluation we considered an OFDM
transmission that is based on a user-bin of Ti tones
and F OFDM symbol periods, thus containing Tsi = Ti
x F QAM symbols (packet length), out of which only
Asi are active symbols being used for the payload.
The cyclic prefix is denoted by G and represents a
fraction of the symbol period, the number of bits per
QAM symbol is ni (it defines the QAM constellation
employed), the bin rate is Db and the CRC (required
for channel estimation-prediction) is t bits long.
Using (8) the nominal payload for a non-coded
constellation i (ni), i.e. the maximum value for the
payload when SNR is very high, is:
1 As t paper allow for a simple encoding and a moderate
Dni = Db Ts n i (1 ); (12)
1 + G Ts As n i
As for the coded configurations, their nominal pay-
load is computed using (8) and is given by (13). There
should be noted that the number of control bits of a
codeword jp should be divided by a constant that
indicates the number of bins on which a codeword is
loaded.
1 A jp
D ci = s D b Ts n i (1 ); (13)
1 + G Ts ki As ni
Considering an adapted version of the values
proposed in [7], namely Db = 1500 bins/sec, Ts = 120
symbols, As = 108 symbols, G = 0.11, t = 8 bits and ni
= 8 the nominal payloads (12, 13) of the
configurations of table 7 are listed in table 8. The
constants ki are respectively 2, 1, 2/3 and .
Table 8. Nominal payload of configurations from
table 6
Cfg 1 2 3 4 5
Dni (kbps) 1156 1104 1041 979 916
The non-coded throughput ni is computed
considering only the correctly received bins given by
the error-bin probability BinERni, and is:
ni = D ni (1 BinER ni )
The throughput ci of the coded configurations is
computed using (14), but the error-rate of the coded
bins BinERci is employed.
The throughput ni or ci vs. SNR curves of the trans-
missions employing the configurations of table 6 were
obtained by simulations and are displayed in fig. 7.
Each configuration exhibits a range of SNR where it
provides the best performance, out the entire family.
Fig. 7. Throughput vs. SNR of configurations from table 6
The SNR ranges of optimum for each configuration
are separated by the thresholds Ui.
The family F3 together with the thresholds Ui may be
employed into an adaptive modulation scheme that
provides the best throughput according to the channel
current SNR.
This scheme is very simple since it changes only the
bit-loading and employs the same QAM constellation
and LDPC code. Despite its simplicity, it provides a
reasonable throughput over a SNR range of about 10-
12 dB.
V. CONCLUSIONS
The array-based LDPC codes employed in the present
complexity decoding, compared to the turbo codes.
The LDPC decoder requires the a priori knowledge of
the channels noise variance.
The BER performances of the LDPC-coded QAM
modulations are close to the ones provided by the
similar modulations coded with turbo codes at the
same rate and the same number of iterations per code
word. A BER = 10-6 at SNR = 2 dB in an AWGN
channel can be obtained by an R = 0.5 LDPC-coded
2-PSK, ensuring a coding gain of about 9 dB.
A very flexible rate changing LDPC-coded scheme
can be obtained by using a bit-loading that combines
coded and non-coded bits. This approach allows for
significant increases of the coding rate at the expense
of rather small coding gain losses.
Due to the behavior of the LDPC-decoding algorithm
and to the soft-decision of the non-coded bits, the
authors estimate that small and high rate RS outer
codes should be employed in FEC schemes based on
concatenated codes.
The LDPC-coded modulation scheme proposed in the
paper ensures coding gains of about 6 dB, compared
to the correspondent non-coded scheme.
It also provides an increased throughput and offers the
possibility of adaptive employment according to the
(14)
channel SNR.
REFERENCES (SELECTED)
[1] ITU-T, LDPC codes for G.dmt.bis and G.lite.bis,
Temporary Document CF-060.
[2] D.J.C. McKay, Good error-correcting codes based on very
sparse matrices, IEEE Trans. on Information Theory, vol. 45, No.
2, March, 1999.
[3] R. Gallagher, Low-density parity-check codes IRE Trans.
Information. Theory, vol. IT-8, January 1962.
[4] ITU-T, Low-density parity-check codes for DSL
transmission, Temporary Document BI-095.
[5] V.Bota, `Zs.Polgar, M.Varga, BER Performance of the QAM
Modulation Coded with LDPC Codes, International Symposium
Etc. 2002, Buletinul Universitii Politehnica, Tom 47 (61),
2002, Fascicola 1, 2, 2002, pp. 104.
[6] - Cl. Berrou, A.Glavieux, Near Optimum Error Correcting
Coding And Decoding: Turbo-Codes, IEEE Transactions on
Communications, vol.44, pp. 1261-1271, October 1996
[7] - W. Wang, M.Sternad, T. Ottosson, A. Ahlen, A. Svensson, -
Impact of Multiuser Diversity and Channel Variability on
Adaptive OFDM, Proceedings of COST 289 Spectral and Power
Efficient Broadband Communications Seminar, Budapest 2004.
12
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Performances of the Reed-Solomon Codes Decoded with

the Guruswami-Sudan Algorithm
Zsolt Polgar, Ana Nstase, Vasile Bota 1
Abstract The Guruswami-Sudan (GS) decoding
algorithm is a list-type decoding algorithm that corrects
more errors than the declared capability, for certain
coding rates of the Reed-Solomon (RS) codes. Using
computer simulations, the paper presents a comparison
between the correction capability and the processing
time of the GS and Berlekamp-Massey (BM) algorithms.
The simulations are employed to establish the optimum
values of the GS parameters that ensure the maximum
performance/processing time ratio. Some methods of
changing the GS parameters, in terms of the packet-
error length, which provide shorter decoding times, are
also presented.
Keywords: RS codes, Guruswami-Sudan decoding
algorithm, Berlekamp-Massey decoding algorithm,
correction capability, decoding time.
I. INTRODUCTION
Considering an RS-type code C, defined over the
Galois field Fq, with the parameters n codeword
length, k information word length, d Hamming
distance, there are three possible definitions for such a
code [1] [2], namely:
Cyclic codes: if a code word cC and is the
cyclic shift operator, then (c)C. The code word can
be expressed as:
n1
RSC(k) = (c0, c1,K, cn1) : c jx j = 0 for , 2,K, nk (1)
j=0
Evaluation codes: the code word is obtained by
evaluating a polynomial f(x), defined by (2),
associated to the information word v (v0, v1, . vk-1),
over the elements of Fq, as shown in (3):
k1
f (x) = vj xj
j=0
with a decoding radius higher than (dmin-1)/2,
delivering a list of possible code words. If the
distances between code words are distributed in such a
manner that the decoding list contains, in most cases,
a single word, then this algorithm-type ensures a
higher correction capability than the classical
algorithms, such as BM.
II. THE GS ALGORITHM. MAIN ASPECTS
The operation steps of list-type GS decoding
algorithm for the RS codes are [3] [4] [5] :
let (0, 1,..., n-1) be the elements of Fq, f(x) the
polynomial corresponding to the information word (2)
and (0, 1,...,n-1) the received code word. If a code
word is correctly received, then relations (4) hold true:
( )
i = f i ; i [0, n 1] (4)
a two-variable interpolation polynomial Q(x,y),
which has an m-order multiplicity zero in every point
(i,i), is built.
the polynomial Q(x,y) is decomposed in (y-f(x))-
type (factorization), whit deg f(x) k; the
polynomials f(x) obtained represent the code words
from the decoding list.
A two variable polynomial, Q(x,y), is an ordered
structure of two-variable monomials, expressed as:
J
Q (x , y ) = a i , j x i y j = a j j (x , y ) ;
i , j 0 j= 0 (5)
1 = 0 (x , y ) < 1 (x , y ) < 2 (x , y ) < K < J (x , y );
J denotes the rank of the Q(x,y) polynomial and
J(x,y) is the leading monomial.
The monomials (x,y) are ordered according to
(2)
their weighted degree, defined by:

{( ( ) ( ) ( )) }
RSG (k ) = f 0 , f 1 , K, f n 1 , deg f < k ; f Fq [x ] (3) deg w x i y j = u i + v j ; w = (u , v ) (6)
There are two possible ordering rules, namely direct
Codes dual to the evaluation codes.
ordering (lex order) and reversed ordering (revlex
Defining the RS codes as evaluation codes, leads
order), defined by:
to the possibility of employing list-type algorithms for
their decoding, algorithms that provide higher per- lex order : xi1 y j1 < xi 2 y j2 if ui1 + vj1 < ui2 + vj2 or
formances than the classical ones, represented in this
paper by the BM algorithm. ui1 + vj1 = ui2 + vj2 and i1 < i 2 (7)
The list-type decoding algorithms [3] [5] operate rev lex order : the same order but for i1 > i2

1
Technical University of Cluj Napoca, Communications Department,
26-28 G. Baritiu Str., 400027 Cluj Napoca, e-mail: Zsolt.Polgar@com.utcluj.ro

13
 A significant theorem that, together with other 2
n k+1 k+1 n
theorems, secures the existence of an interpolation Lmax= m(m+1) + <(m+0.5) (12)
polynomial is [3]: k1 2k 2 2k 2 k 1
Theorem 1: Let {m(,):(,)F2} be the
multiplicity function of the zeros of Q(x,y) and
0<1<.... an arbitrary monomial order. There always
exists a polynomial Q(x,y): n (k 1) m + 1 rd n 1 (k 1) m + 1 k 1
m m m
C (13)
Q(x, y) = a i i (x, y) (8)
i =0
III. ANALYSIS OF THE SIMULATION RESULTS
In (8), C is expressed by:
The main goal of this paper is to compare, by
m(, ) + 1 computer simulations, the correction capability and
C = (9)
, 2 processing time of the GS and BM (representative for
the classical algorithms) RS decoding algorithms. The
The complete proof of the existence of the analysis is intended to establish the optimum values of
interpolation polynomial is to be found in [4] and [5]. the parameters of the GS algorithm, for which a
The existence of a polynomial that could be maximum ratio correction capability/decoding time is
decomposed in (y-f(x)) factors is secured by theorem accomplished and to elaborate some thumb rules for
2 [3]: adapting these parameters, so that shorter decoding
Definition 1: For Q(x,y)F[x,y] and f(x)F[x] the Q- times could be attained.
score of f(x) is defined as: The software simulator, that can operate in the
S Q (f ) = ord zero(Q : , f ( )) (10) Galois fields GF(23), GF(24), GF(25), GF(26) and
GF(28), performs the following functions:
Theorem 2: generation of a symbol-sequence represented on
If f(x)Fk[x], Q(x,y)F[x,y] and SQ(f) > deg1,vQ the number of bits corresponding to the employed
(11) Galois field.
then y-f(x) is a factor of Q(x,y) ; v=k-1. RS encoding (cyclic code for the BM or evaluation
A thorough analysis of the factorization step is code for the GS), depending on the decoding
presented in [4] and [5]. algorithm employed.
One of the most efficient interpolation algorithms serialization of the coded bits, generation of the
is the Koetter algorithm [3], which is defined by the packet-errors and their insertion in the coded bits.
pseudo-code below: GS or BM decoding and computation of the
Koetter interpolation algorithm parameters of the simulated transmission, namely: bit
- input data: L number of code words in the list, (i and symbol error rates, the ratio of the correction
j) ni=1 interpolation points, (mi)ni=1 zeros multi- capability of the GS algorithm versus the correction
plicity order, (1,k-1) monomials weighted degree. capability of the BM algorithm, the numbers of words
in the decoding list and erasures, both for the GS
1. FOR j=0 to L algorithm.
gj=yj The generation of the packet-errors, which
2. FOR i=1 to n DO simulates the transmission channel, is performed
2. FOR (r,s)=(0,0) to (mi-1,0) DO /*lex order according to the impulse noise models employed for
4. . FOR j=0 to L DO the xDSL transmissions [6]. This model was adopted,
5. j=Dr,sgj(i,j) with several simplifications, since it is a representative
6. J={j: j 0} one for transmission systems employing RS codes as
7. IF J outer codes. The main features of algorithm that
8. j*=min_rank {gj:jJ} generates the packet-errors are:
9. f=gj* ; =j* the distance in symbols between two packet-errors
10. FOR jJ DO has a Poisson distribution, with a modifiable average
11. IF (jj*) value . In the simulations performed, the value of
12. gj=gj+jf equaled the number of symbols of two code words, for
13. ELSE IF (j=j*) each GF.
14. gj=(x+i) f the packet-error length, in bits, has a gaussian
15. Q0(x,y)=min_rank{gj(x,y)} /* the interpolation distribution, defined by the average value and
polynomial variance . The value of equaled tbq, tb denoting the
number of error-symbols that could be corrected by
One of the best factorization algorithms, the Roth- the classical decoding algorithms (e.g. BM) and q
Ruckenstein [3], was used in the present analysis. denoting the number of bits/character of the GF
The bounded values of two significant parameters employed. The value of was set according to the
of the GS algorithm, the number of code words in the estimated correction capability of the GS algorithm.
decoding list, L, and the decoding radius, rd, are given the positions of the errors inside the packet are
by [3]: random, being distributed according to a uniform law.

14
A. Decoding Capability of the GS Algorithm
The performances of the GS algorithm were evaluated
for RS codes with the coding rate Rc[0.3, 0.65]. The
parameters that indicate the correction capability are
the minimum and the maximum decoding radius,
computed using (12), and the correction rate Rd
(obtained by simulations). The Rd parameter is defined
as the ratio of the number of error words after the GS
decoding and the number of error words that have a
number of error symbols higher than tb (the decoding
radius of the classical algorithms), before the
decoding. The codes with Rc < 0.3 were not
considered, since they are of low practical importance.
As for the codes with Rc > 0.5 - 0.6 (depending of the
employed GF), the correction capability of the GS
Fig.1.c shows that for RS codes defined in GF(23)
and GF(24), m has to be set to 3 or 4, for Rc close to
0.5, and to 1 or 2 for Rc close (or smaller) than 0.3.
The increase of m above a certain limit does not bring
a performance improvement, but it might lead to a
decrease of performances (see Rc = 0.33). A more
complete evaluation of the GS decoder requires the
consideration of the rmin and rmax, as well; good
decoding performances should be accomplished when
the two parameters take equal or close values. The
optimum values of m can not be established by
considering only the rmin and rmax parameters of the
code, as shown by Rc = 0.46.
14
Fig. 2.a Rc=0.29
Rc=0.35
12

algorithm is the same with the one of the BM 10

Rc=0.42
algorithm. Rc=0.48
Note: In figs. 1-5 m denotes the multiplicity order of 8
Rc=0.55

r_min
the zeros in the GS algorithm; m = 0 is actually 6

equivalent to the BM algorithm; for this algorithm: 4

rmin = rmax =tb = n(1-Rc)/2; Rd = 1; (14) 2

The variance of the packet-errors, depicted in figs. 0

0 1 2 3 4 5 6
1.b, 2.b, 3.b, 4.b, for each Rc and GF, was set in all m

simulations to a value that provides packet lengths 14

close to the rmax of the GS algorithm. Fig. 2.b

13 =5 ; Rc=0.29
9 =3.5 ; Rc=0.35
Rc=0.2 Fig. 1.a 12
8
11
r_max

7
=3.5 ; Rc=0.42
Rc=0.33 10
6
Rc=0.46 =2 ; Rc=0.48
r_min

9
5
=2 ; Rc=0.55
4 8
Rc=0.43*
3 7
0 1 2 3 4 5 6
m
2
0 1 2 3 4 5 6
m
1
9 Fig.2.c
=3.5 ; Rc=0.2 Fig. 1.b 0.9 Rc=0.55
8
0.8
7
Rc=0.48
=2.5 ; Rc=0.33 0.7
6

=2 ; Rc=0.46 0.6
r_max

Rd

5
0.5
4
=2 ; Rc=0.43* 0.4
3
0.3
2
0 1 2 3
m
4 5 6
0.2 Rc=0.42
1
0.1 Rc=0.29 Rc=0.35
Fig. 1.c
0.
Rc=0.43* 0
0 1 2 3 4 5 6
0. m

0. Fig.2 Minimum, rmin (2.a), maximum decoding radius rmax

(2.b), correction rate Rd (2.c) in terms of m ; RS codes in Galois
0.
Rc=0.46 GF(25);
Rd

0.
Rc=0.33
Rc=0.2 The values of Rd, see fig.2.c, indicate that for RS
0.
codes defined over GF(25) the optimum values of m
0. are m = 3 - 4 for Rc close to 0.5, m = 2 - 3 for Rc
0. around 0.3 and m = 4 - 5 for Rc around 0.4. There
0.
should be noticed that for m=6, the performances of
0 1 2 3
m
4 5 6
the GS decoder exhibit a significant decrease,
Fig.1 Minimum, rmin (1.a), maximum decoding radius rmax
especially for high values of the coding rate Rc.
(1.b), correction rate Rd (1.c) in terms of m ; RS codes in For RS codes defined in GF(25) having the
Galois GF(23) and GF(24) ; * denotes codes defined in GF(23); mentioned Rc and for the optimum values of m, the

15
decoding radius of the GS lies between rmin tb, and The performance loss exhibited by the GS
rmax = rmin+1 or rmin+2. The values of parameter of algorithm for high values of m, regardless the coding
the error-packet for the considered rates are given in rate, could be explained by the incomplete
fig.2.b. factorization, see (11), the requirements for the
interpolation being ensured by a proper choice of the
25
Fig.3.a number of words within the decoding list.
Rc=0.33
Rc=0.4 A primary analysis of the interpolation algorithm
20
Rc=0.46 presented in Section II and of the properties of the
two-variable polynomials [3] leads to the following:
15 Rc=0.52
the number of iterations, nit, performed by the
r_min

Rc=0.59
10
interpolation algorithm for n-symbol code words and
multiplicity order of zeros equaling m, is :
5
m (m + 1)
n it = n (15)
2
0
0 1 2 3 4 5 6
m
the initial polynomials of Koetter interpolation
26 algorithm, for maximum L words in the final decoding
Fig.3.b
list, are:
24 =7 ; Rc=0.33
22
=6 ; Rc=0.4 p0(x, y) = 1, p1(x, y) = y, p2(x, y) = y2 ,K, p2(x, y) = yL (16)
20
=4 ; Rc=0.46 supposing that the values of , computed within
r_max

the Koetter algorithm, never equal zero (supposition

18 =4 ; Rc=0.52
that does not always hold true), then after
16
L(L+1)/2(k-1) iterations all polynomials will have
=2 ;Rc=0.59 the same degree L(k-1). The leading monomials of
14
these polynomials are:
lp0 (x, y) = xL(k1) , lp1(x, y) = x(L1)(k1) y ,
12
0 1 2 3 4 5 6
m
(17)
1 lp2 (x, y) = x(L2)(k1) y2 ,K, lpL (x, y) = yL
Fig.3.c
taking into account that each iteration increases the
0.9
Rc=0.59
0.8 degree of the polynomial with the minimum rank and
Rc=0.52
0.7 that a polynomial with a higher degree also has a
0.6
higher rank, then we may assert that the increase of
the degree of each polynomial will require L+1
0.5
iterations. By the end of the algorithm the degree,
Rd

0.4 degmin, of the minimum-degree polynomial would be:

0.3
n it L (L + 1) 2 (k 1)
+ L (k 1)
Rc=0.46
deg min = (18)
0.2 L +1
Rc=0.33
0.1
Rc=0.4 the minimum value of the SQ parameter (10) of an
0
0 1 2 3 4 5 6 interpolation polynomial associated to a n-symbol
m code word and to a multiplicity order of zeros
Fig.3 Minimum, rmin (3.a), maximum decoding radius rmax equaling m and to a decoding radius r, is:
(3.b), correction rate Rd (3.c) in terms of m ; RS codes in Galois
GF(26); SQ min = (n r ) m (19)

Considering the RS codes defined in GF(26), see
figs. 3, they exhibit a clear separation of the optimum
values of m, in terms of the coding rate Rc. For
Rc 0.5, optimum m equals 3 or 4, but for Rc 0.45,
optimum m equals 2 or 3. Sometimes, see Rc = 0.4,
m = 4 provides better performances at the expense of a
longer decoding time.
The codes defined in GF(26) exhibit the same
decrease of performance for higher values of m (e.g.
m = 6), as the ones defined in GF(25); for the
considered values of Rc, the performances secured by
the GS become equal to the ones of the BM. The
values of parameter of the error-packet for the
considered rates are given in fig.3.b.
from the factorization requirements we have:
r (L + 1)
1
SQ min n 2
>1 (20)
deg min (m + 1) + L (L + 1) R
m
The values of the ratio defined in (20), for the
codes of figs. 2 and 3 and for various values of m, are
smaller than 1 (approximately equal, but smaller).
There should be noted that the considerations above
are not complete, since it did not considered that the
evolution of polynomials degrees within the
interpolation algorithm would be different, mostly
because of the fact that might equal zero quite often,
changing the evolution of the polynomials degrees

16
(see the interpolation algorithm in Section II), and 1

decreasing significantly the values of degmin. Also, the 0.9

value of SQ might be higher than the value computed
0.8 Rc=0.47
by (19). Nevertheless, the considerations above show
Rc=0.65
that, for different coding rates and various values of 0.7
Rc=0.61
m, there is a possibility that the GS algorithm would
0.6
not be effective, even for a high decoding radius. The
suppression of this limitation of the value of m may be 0.5
Rc=0.41

accomplished by using different values of m for every
interpolation point, values chosen depending of the
channel characteristics [5]. Obviously, this approach
would complicate the implementation of the decoding
GS algorithm.
Unlike the previous cases, for optimal values of m,
the codes defined in GF(26) have rmin < tb, but the
difference rmax - rmin takes values between 3 and 6. The
difference rmax - tb takes values between 0 and 3. So,
for the codes defined in GF(26) the optimum values of
m cannot be evaluated only be considering the limit
values of the decoding radius, rmin and rmax.
Note: the relation rmin < tb does not imply that the GS
algorithm could not correct tb symbol-errors (the
declared correction capability of the code), so
practically one should consider that rmintb. The values
of rmin and rmax provided by (13) evaluate the
possibility of the GS algorithm to correct more errors
than the classical algorithms. As for the RS codes
defined in GF(28), see figs. 4, the considerations
regarding rmin and rmax, presented above, are still valid.
There should be mentioned that, for the optimal values
of m, rmin tb, and the difference rmax - rmin takes values
higher or equal than 20, and the difference rmax-tb lies
between 2 and 13.
100
Fig.4.a
Rc=0.32
90
Rc=0.36
80
Rc=041
70
Rc=0.47
60
Rc=0.51
50
Rc=0.56
r_min
Rd
Rc=0.32
0.4
Rc=0.56
0.3
0.2
Rc=0.36
Rc=0.51
0.1
Fig.4.c
0
0 1 2 3 4 5
m
Fig.4 Minimum, rmin (4.a), maximum decoding radius rmax
(4.b), correction rate Rd (4.c) in terms of m ; RS codes in Galois
GF(28);
Fig. 4.c shows three optimum values of m,
depending of the coding rate Rc, for the codes defined
in GF(28). For coding rates higher or equal to 0.6 the
optimum value of m is 4, for Rc (0.6, 0.45) the
optimum value of m is 3, and for coding rates ranging
between 0.3 and 0.45, the optimal m equals 2. The
figure also shows that, similar to the codes defined in
GF(26), the maximum limit of m falls to 5, for the
coding rates considered.
The comparison of the results presented in figs. 1.c
- 4.c show that the coding rate for which the GS
decoding algorithm provides better performances than
the classical decoding algorithms increases with the
increase of dimension of the Galois field in which the
RS codes are defined.
Regarding the number of words in the decoding
list, the simulations performed by the authors show
that for the RS codes defined in GF(25) and in the
higher fields, the number of the words in the list
equals 1, with very few exceptions, when then list
contains more than one code word. As for the codes
defined in GF(23) and GF(24), there are more cases

40
Rc=0.61 when the decoding list contains more than one code
30 word, but their percentage is still small, about 1%. As
20
Rc=0.65 a general conclusion, if the GS decoding algorithm
10
can not correct a code word, this fact is owned to an
unsuccessful interpolation or factorization and, quite
0
0 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5 seldom, to the presence of more than one code words
m
in the decoding list.
110
Fig.4.b =50 ; Rc=0.32
B. Evaluation of the GS algorithm decoding time
100 =40 ; Rc=0.36 The references [3] [4] [5] present some considerations
regarding the number of operations performed by the
90 =30 ; Rc=0.41 GS algorithm, which affect significantly the decoding
time, but these considerations do not include a
80 =20 ; Rc=0.47
comparison to the decoding time required by the
r_max

70 =10 ; Rc=0.51 classical RS decoding algorithms. The software

=10 ; Rc=0.56 simulator implemented by the authors includes a RS
60 decoder based on an optimized version of the
=5.5 ; Rc=0.61
Berlekamp-Massey (BM), described in [1]. For
50
comparison, the simulations using the BM decoding
=5.5 ; Rc=0.65
40
algorithm were performed in the same conditions as
0 1 2
m
3 4 5
the ones using the GS algorithm.

17
 The evaluation of the decoding time implied the
measurement, for a certain number of code words, of
the simulation time tsim, and of the time required for
encoding and error-pattern insertion taux ; for the
measurement of taux the decoding procedures were
removed from the simulation program. There should
be noted that the time required to decode a correct
code word differs from the time required to decode an
error code word for both algorithms, especially for the
GS algorithm. The ratio between the average decoding
times, tdec, of the two algorithms is expressed by:
t t t t
t d = decGS = simGS auxGS simGS (21)
t decBM t simBM t auxBM t simBM
Fig. 5 presents the variation of the ratio td
(expressed on a logarithmic scale) between the
average decoding times of the GS and BM algorithms
in terms of m, for various coding rates and for codes
defined in several Galois fields.
4.5
4 GF(28) ; Rc=0.32
3.5
GF(26) ; Rc=0.59
GF(28) ; Rc=0.64
3
lg(td)
2.5
2
GF(26) ; Rc=0.33
GF(24) ; Rc=0.3
1.5
4
GF(2 ) ; Rc=0.46
1 which a maximum correction capability/decoding time
1 2 3 4
m ratio is accomplished, are also presented in the paper.
Fig.5 lg(td) ratio between the average decoding time of the GS
and BM algorithms, in terms of m, for various coding rates and
for RS codes defined in several Galois fields
The results presented in fig. 5 show that the
decoding time required by the GS algorithm is much
larger than the one required by the BM algorithm. The
td ratio increases significantly with the increase of m
and with the increase of dimension of the Galois field
employed. The increase of the coding rate for codes
defined over GF higher than GF(24) also increases the
value of td. By changing the coding rate from 0.3 to
0.6 for these codes, involves an increase of the ratio td
by a factor ranging from 2 to 3. There should be
mentioned that the implementations of the two
algorithms were optimized to the best knowledge of
the authors.
The results displayed in fig. 5 underline the
importance of establishing optimal values for the
parameter m and the necessity of finding some
variants of the GS algorithms (decoding strategies),
which should require a decoding time as small as
possible. The authors have considered three possible
variants to accomplish the GS decoding, namely:
employing the same value of m for the decoding of
every code word; this variant would require a very
large average decoding time, even larger than the ones
presented in fig. 5, because even the correctly received
code words would be decoded in a very long time.
the successive increase of the value of m, from 1 to
a maximum optimal value. The decoding is stopped
when the decoding list contains at least one code
word; this approach would require a smaller average
decoding time for packet-errors with relatively small
lengths, compared to the maximum packet length for
which a successful GS decoding is accomplished.
the employment of two values for parameter m,
namely 1 and an optimum value mopt. The correct code
words and the ones affected by a small number of
errors (equal or higher than tb) would be decoded
using m=1, and the code words with more errors
would be decoded with m = mopt; this last option
should be employed if the decoding with m=1
generates no code word in the decoding list. This
variant of employing the GS algorithm provides a
smaller average decoding time for long error-packets,
compared the maximum packet length for which a
successful GS decoding is accomplished. The results
displayed in fig. 5 were obtained using this decoding
variant.
IV. CONCLUSIONS
The computer simulations performed by the
authors showed that the GS decoding of RS codes,
defined in the GF(24) GF(25) GF(26) and GF(28),
provides a significantly greater correction capability
than the BM decoding, for coding rates ranging
between 0.3 and 0.6. The improvement becomes more
obvious as the coding rate decreases and the
dimension of the Galois field increases. The optimum
values of the factor m (zeroes multiplicity order) for
5
As for the decoding time, the simulations performed
showed that the time required by the GS is
significantly longer than the one required by the BM
algorithm. The paper presents some decoding
strategies for the GS that lead to a significant decrease
of its decoding time for various lengths of the packet-
errors.
REFERENCES
[1] M.E.OSullivan, Coding Theory, http://www-
roham.sdsu.edu/~mosulliv/Courses/coding02.html.
[2] J.I.Hall, Notes on Coding Theory, http://www.mth.msu.edu/
~jhall/classes/codenotes/coding-notes.html.
[3] R.J. McEliece, The Guruswami-Sudan Decoding Algorithm for
Reed-Solomon Codes, IPN Progress Report 42-153,15 May, 2003.
[4] M. Sudan, Decoding of Reed Solomon codes beyond the error-
correcting bound, Journal of Complexity, vol. 13, 1997.
[5] V. Guruswami, M. Sudan, Improved Decoding of Reed-
Solomon Codes and Algebraic Geometry Codes, IEEE Trans.
Inform. Theory, vol. 45, no. 6, September 1999.
[5] W. Gross, Fr. Kschischang, R. Koetter, P.G. Gulak,
Applications of Soft-Decision Decoding of Reed-Solomon Codes,
submitted to IEEE Trans. Comm. July 2003,
http://www.macs.ece.mcgill.ca/~wjgross/papers/gkkg_tc.pdf.
[6] W. Henkel, T. Kessler, A wideband impulsive noise survey in
the German telephone network-Statistical description and
modeling, AEU, vol. 48, no. 6, Nov./Dec. 1994.
18
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
New Interleaver Design Algorithms with Enhanced B.E.R.
Performances
Simona V. HALUNGA1, Octavian FRATU1
Abstract: This paper investigates some aspects of
turbo encoder and decoder parameters design effects
on the overall system performances. A new interleaver
structure is presented, which, combining the block and
hibrid strategies, achieves very good BER
performances using low complexity structure. Based
on an interactive Matlab simulation program that can
accommodate several encoder structures, interleaver
types and decoding algorithms, extended simulations
has been developed and important results are
highlighted with respect to different parameter
influence on the system performances (evaluated in
terms of Bit-Error-Rat) and complexity.
Keywords: turbocodes, interleaver design, Bit Error
Rate performances, complexity.
I. INTRODUCTION
In a digital transmission system, the error control
function is achieved by using a channel encoder
used at the transmitter and a corresponding decoder
at the receiver. A well known result from
Information Theory states that, for any value of the
Bit Error Rate (BER) larger then the Shannon limit
[1], there exists a coding scheme that can ensure that
imposed BER, whatever the channel bandwidth is.
The Shannon Theorem, however, does not give any
indications regarding the type or complexity of code
that has to be used, being, more or less, a theoretical
lower bound in the bit error rate
In the last five decades many code structures have
been developed in order to achieve a BER as close
as possible to the Shannon limit. However, the
optimal decoding complexity, whereas the codes
used are block, convolutional or hybrid, increases
exponentially with BER decrease, up to a point
where decoding becomes physically unrealizable.
Recently, a new class of error correcting block
codes called Turbo codes was introduced [2]. Due to
1 (1
)
Politehnica University of Bucharest,
Electronics , Telecommunications Information Technology Faculty
Telecommunications Department,
Splaiul Independentei 313, 77206, Bucharest 6, Romania,
E-mail: shalunga@elcom.pub.ro
19
concatenation the bloc encoding principles with the
convolutional ones and to the use of a well designed
interleaver, these codes posses the quality of being
able to achieve a BER close to the Shannon limit with
an acceptable structural complexity. Due to their good
behavior in severe distorting and fading channels,
those type of codes are widely used in 3rd generation
mobile communications systems like
UMTS/IMT2000, as well as in modern DVB-S
satellite links.
This paper investigates different encoder
configurations, interleaver structures and decoder
algorithms, and their influence on the overall system
performances, evaluated in terms of BER and system
complexity. Moreover, a new interleaver structure is
proposed, that takes advantage of both the block and
random interleaver properties, by combining them in
order better BER properties with simpler encoder /
decoder structures. The results obtained using those
types of interleaver are compared with the ones
obtained using classical block / random interleaver,
both for SOVA and MAP decoding algorithms, with
different frame lengths, different number of iterations
and different length encoder polynomials.
II. ENCODER STRUCTURE
The Turbo Encoder structure consists of two recursive
Systematic Codes (RSC) that operates on the same
input bits, as shown in figure 1. The first encoder
(RSC1) operates with the systematic encoder
polynomial g1 (D ) while the second encoder (RSC2)
uses a nonsystematic polynomial g 2 (D ) . For the
second encoder the input bits order is changed by
placing an interleaver in front of it, in order to protect
the overall generated codeword against burst errors,
that often appears in mobile communication systems.
Since the most common types of interleavers works
with blocks of bits, the overall code can be
considered as a block code too
v0
x
RSC 1 Puncturing
c
v1 And
Multiplexing
Interleaver RSC 1
~
x w1

Fig.1. Encoder architecture

algorithm, which considerably affects both

. Let us denote performances and complexity of the code. For low
x = [x0 , x1 , ... xn ] (1) Signal-to-Noise Ratios (SNRs) the performances are
the information codeword. Considering that both determined mainly by the size of the interleaver,
codes are of rate 1/2, the first encoder output is while for large SNRs the structure design becomes
[ ] [ ]
v = v ( 0) , v (1) = v0(0 ) , v1(0 ) , ... , vn(0 ) , v0(1) , v1(1) , ... , vn(1) ,
the key factor.

(2) A. Block Interleavers

where v(0) represents the information bits and v(1) the
parity check bits, generated separately since the
code is systematic. Similarly, for the second
encoder, the input is the interleaved data, denoted
with
~x = [~
x0 , ~
x1 , ... ~ xn ] ,
and the output is
[ ][
w = w(0) , w(1) = w0(0) , w1(0) , ... , wn(0) , w0(1) , w1(1) , ... , wn(1) , (4)
where only the parity check bits are transmitted over
the channel. The generator matrix of the turbo code
can be written as
g (D )
G = 1, 1
g 0 ( D )
where g1(D) and g0(D) are the positive and negative
reaction polynomials of the two encoders, havingthe
same degree.
Since the overall code has a much higher rate then
the corresponding classical convolutional code,
generated with a polynomial of the same degree, the
rate can be reduced by using a puncturing operation.
The idea is to transmit all the systematic bits from
the first encoder and half of the parity bits from each
encoder alternately. For a 1/2 rate code, as the one
described above, this operation can be described by
the puncturing matrix
1 1 (6)
P = 1 0
0 1
and, therefore, the corresponding code sequence,
results by combining (2) and (4) in accordance with
the puncturing matrix, resulting the overall
codeword
[
c = v0(0 ) , v0(1) , v1(0 ) , w1(1) , ... (7)
III. INTERLEAVER DESIGN
One of the key elements in designing a
turbo-code is the interleaver size, structure and
A1. The Row-Column Interleaver uses a NxM matrix,
in which the input sequence is written in row-wise
and read out column-wise. The function performed by
this interleaver is described by the function
(3) : A A, (8)
(i ) = [(i 1) mod M ] N + (i 1) / N + 1
]
where i is the index of the input data, (i) is the
corresponding index of the output data, x
represents the integer part of x and A is the set of
integer numbers corresponding to all possible values
(5) of data indexes. The total delay introduced by the
interleaver and de-interleaver blocks is 2 MNTb ,
where Tb is the bit period. The Row-Column
interleavers are often used to break short Hamming
weight error patterns, (i.e. with the length smaller then
the row length). If the errors are extending over
several consecutive rows, the structure is no longer
efficient.
A2. The Even-Odd Interleaver maps the odd indexed
bits on even-indexed positions and vice-versa. It is
mathematically described by the function
: A A, [ (i ) + i ]mod 2 = 0 (9)
This structure is used to break long error patterns that
are not uniformly distributed within the sequence.
A3. The Helical Interleaver is based on a matrix
structure too, but this time the data is written in row-
wise and read out diagonal-wise. This structure
prevents consecutive input bits to have consecutive
positions in the output sequence.
] B. Convolutional Cyclic Shift Interleavers
implements an interleaver ([3], [4]) by writing the
data into a MxN matrix, column-wise. Then, the M
rows are applied to M N-length shift registers, where
the i-th register cyclically left shifts the i-th matrix
row (i-1)B times, where B is an integer number such
that B N / M . Those shifted sequences are then
introduced into a second matrix, from which are read

20
out column-wise. The interleaver has the following
distance property
()i, j A, i j MB 1 (10)
(i ) ( j ) M 1
C. Random Type Interleavers introduces an N
bits input block of data into a memory and reads it
out randomly, in accordance to the following N step
algorithm:
Step 1: choose index i1 from the set A {1,2, ... ,N},
in accordance to a uniform probability function
p(i1 ) = ; the corresponding output index is (1);
1 interleaver identical to the one used in the encoder, as
N shown in figure 2. The first decoder uses the received
Step k: choose index ik from the set
Ak = {i A, i i1 , i2 , ... ik 1 } , in accordance to a
uniform probability function p (i ) = 1 ; the
k improve the estimate of the apriori probabilities for
N k 1
output index is (k);
Since a pure random interleaver is
generally hard to implement, in practice are often
used pseudo-random interleavers, where the indexes
(i) are the outputs of a pseudonoise shift register,
generated by a primitive polynomial.
D. Hybrid Interleavers
Simulations have shown that bloc / even-odd
interleavers are simple to implement, but their
decorrelation properties are low and, therefore, the
overall BER properties are poor. On the other hand,
the random type (pseudorandom) interleavers have
the best decorrelation capabilities, so the BER
results are very good, especially when the PN codes
De-Interleaver
used to control the interleaver are long. One idea was
to use a bloc even-odd interleaver structure, for which
the indexes have been pseudorandomly interleaved in
advance. In this way, using 10 times shorter PN
codes, and the results are close to the ones obtained
using purely random interleavers.
IV. DECODER STRUCTURE
The iterative decoder structure consists of two
component decoders, serially concatenated via an
information bits r0 and the parity bits generated by the
first encoder r1 in order to produce a soft output,
denoted with 1e , which is interleaved and used to
the second decoder. The other two inputs of the
second decoder are also the received information
sequence which are interleaved by the same algorithm
as in the encoder, ~r0 and the received parity sequence
produced by the second encoder r2. This decoder
produces also a soft output, denoted with 2e , that is
de-interleaved and used by the first decoder to
improve its apriori probabilities. This iterative
feedback operation increase the performances of the
overall structure, especially in the first decoding steps.
After a number of iterations the soft outputs from the
decoders will no longer affect significantly the
performances, and, therefore, a hard decision is
applied at the end in order to obtain the decoded data
sequence. Both decoders are Soft Input Soft Output
(SISO) type

1e
Decoder
r0 1 Interleaver
r1
~
r0
Decoder 2
Interleaver 2 De-Interleaver Threshold
r2 2e

Fig.2. Iterative Decoder Architecture

second encoder in order to compute the log-likelihood

The first decoder uses thus the received sequences ratio for the overall code trellis
[ ]
r ' = ... tt ,0 , rt ,1 , tt +1,0 , rt +1,1 , ... (11)
(ct ) = log
P(ct = 1 | r ' , r ' ')

obtained from the received information bits and the P (ct = 0 | r ' , r ' ') (13)
t ,c =1 P(r ' | c )P (r ' ' | c )P(c )
parity check bits generated by the first encoder,
while the second decoder uses the sequence
= log t

[
r ' ' = ... ~
tt ,0 , rt , 2 , ~ ]
tt +1,0 , rt +1, 2 , ... (12) t ,ct =0 P(r ' | c )P (r ' ' | c )P(c )
obtained from the interleaved received information for all the paths in the code trellis, and makes the
bits and the parity check bits generated by the decision

21
 1 ; (ct ) 0
ct = (14)
0 ; (ct ) < 0
The log-likelihood ratio from (6) can be determined
using MAP, log-MAP, Max-Log-MAP and SOVA
algorithms ([5], [6]).

V. SIMULATION RESULTS AND

CONCLUSIONS

In order to analyze the performances obtained by

different turbo-codes structures, an interactive
Matlab program has been developed. User data is
randomly generated and encoded using two
component RSC codes; the first encoder is
terminated with tail bits, while, for the second
encoder the data and tail bits are interleaved and Fig. 3. Bit Error rate versus Eb/N0, different interleaver types,
SOVA decoder, frame length L=200, punctured, 5 iterations, 10
passed through the second encoder, which is left errors to terminate de decoding
open (i.e. no tail bits of itself). The frame size and
generator polynomials are user defined. The
encoded data is transmitted through an AWGN
channel (the signal to-noise ratio at channel level is
also defined by the user) and demodulated at
receiver level using either MAPs or SOVA
algorithms. Both punctured (rate 1/2) and un-
punctured versions might be chosen. The user can
also define the number of iterations for each frame
and the number of frame errors the decoder
terminates. The receiver counts and displays the bit
error rate and the frame error rate at each decoding
algorithm iteration. Several interleaver algorithms
have been developed (i.e. block, even-odd, helical,
random, hibrid) and compared one-another from
BER results point of view.

Using this backbone program, several important Fig. 4. Bit Error rate versus Eb/N0, different interleaver types,
aspects have been studied and compared one SOVA decoder, frame length L=400, punctured, 5 iterations, 10
errors to terminate de decoding
another from the BER point of view. The simulation
of the overall system leaded to a complex and time
The random interleaver gives the best performances
consuming program and the simulation process is
from all (about 10dB improvement at SNR=1dB), for
still under development in order to cover all the
both SOVA and MAP algorithms, and this
problems encountered, to compare all the possible
improvement does not depend on the random
configurations and obtain relevant and
interleaver realization. The hybrid interleavers
comprehensive results. However, from the results
performances are close to the ones obtained using the
obtained till now, several aspects have to be
pure hybrid one, especially when the interleaver
emphasize.
length is large (in figure 3, for frame length L=200,
the hybrid interleaver behaves slightly worse then the
The interleaver type effect: several interleaver random one while in figure 4, for frame length
structures have been studied: block (row-column), L=400, the two curves merely overlap).
even-odd, helical, random and hibrid. The results
are shown in figures 3 and 4. The block interleaver
The interleaver length effect: the interleaver
achieves the worst performances from all; the even-
length gives us the length of the data block that has to
odd and helical have similar performances, better
be processed by the decoder at a certain step in order
then the row-column, with both SOVA and MAP
to recover the data. As it can be seen form figures 3, 4
decoding algorithms (the difference becoming
and 5, as the interleaver length increases, the system
significant at high SNRs). The cyclic-shift (helical)
performances improve also, whatever decoding
interleaver has better performances then the all
algorithm is used.
block ones, especially at high SNRs (about 6dB
improvement at SNR=1dB).

22
 Fig. 5. Bit Error rate versus Eb/N0, hybrid interleaver, SOVA & Fig. 6. Bit Error rate versus Eb/N0, SOVA decoder, different
MAP decoders, frame length L=200/ 400, punctured, 5 iterations, number of iterations, hybrid interleaver, frame length L=200,
10 errors to terminate de decoding punctured, 10 errors to terminate de decoding

However, the frame length increase determines also curves for 2, 3 and 4 degree generator polynomials,
an increase in structure complexity and decoding namely
delay; therefore, the solution has to be chosen as a
compromise between a certain threshold in deg 2 : g1 (D ) = 1 + D + D , g 2 (D ) = 1 + D 2
2

performances that needs to be achieved and the

deg 2 : g1 (D ) = 1 + D + D 2 + D 3 , g 2 (D ) = 1 + D + D 3
complexity of the system.
deg 2 : g1 (D ) = 1 + D + D 2 + D 3 + D 4 , g 2 (D ) = 1 + D 4
The decoding algoritm effect: MAP, log-MAP, (15)
Max-Log-MAP and SOVA decoding algorithms However, a linear increase of the polynomials degree
have been studied and compared one-another. The leads to an exponential increase in the encoder and
MAP and log-MAP have similar behavior, as well decoder structures, as well as in the decoder algorithm
as the Max-Log-MAP and SOVA, both with respect length and complexity; this can cause other
to their performances and complexity, so it is impairments, like important delays in data decoding
sufficient to compare MAP to SOVA. The and substantially cost increase. Therefore, in the
simulation results shown that the MAP algorithm following, we restricted our study to 2 and 3 degree
achieves better performances then SOVA, especially encode polynomials.
for low SNRs, the difference increasing as the
interleaver length and BER are larger (see figure 5).
For large SNRs the difference is nio longer
significant. The main disadvantage of the MAP
algorithm is that it is about three times more
complex then SOVA, and therefore the
computational effort is correspondingly higher.

The Number of Iterations effect: simulations

have shown that, as the number of iterations in the
decoding algorithm increase, the BER performances
improves up to a certain point (see figure 5). For a
large number of iterations the increase is no longer
significant; this threshold depends on the interleaver
length: for a 200 bits interleaver an increase over 8
iterations is no longer useful, while for a 400 bits
interleaver the threshold in number of iterations is Fig. 7. Bit Error rate versus Eb/N0, MAP decoder, different
10. generator polynomials, hybrid interleaver, frame length L=200,
punctured, 5 iterations, 10 errors to terminate de decoding
The memory order effect: as the memory order
(i.e. the degree of the encoder generator Frome the results above, the following conclusions
polynomials) increases, the system BER may be highlighted:
performances improves, especially for large SNRs, the block type interleaver has the worst
while for low SNRs the low order degree encoders performances with respect to the overall system
behaves better. In figure 6 are shown the BER BER, while the random ones achieves the best
BER performances; the even-odd and helical

23
 interleaver are in between, close to one another
from BER point of view;
the new hybrid interleaver achieves BER
performances close to the random ones,
especially when the frame length is large;
as the frame length (block size) increases, the
system performances improves also; however,
as the frame length increases, the system
complexity (and costs) and the delay increases
also; therefore a compromise has to be made
between performances and costs;
the MAP decoding algorithm achieves better
performances then the SOVA one, especially
fol low SNRs; for large SNRs the difference is
no longer significant; however, the MAP
algorithm complexity is 3 times larger then the
SOVA one, and therefore the decoder structure
(and cost) and the associate delays are also
higher;
the number of iterations in decoding algorithm
leads to an increase in BER performances, till
to a certain threshold which depends on the
frame length (as the frame length increases, the
number of necessary iterations decreases);
as the memory encoder / decoder polynomials
degrees increase, the system BER performances
improves, especially for large SNRs, while for
low SNRs the low order degree encoders
behaves better.
It has to be mentioned that simulations are
continuing, in order to determine the system
behavior for low (negative) SNRs. Other
interleaver structures are currently under study, as
well as different types of fading effects on BER
performances. Moreover, in order to obtain more
reliable results, the BER results has to be averaged
on several number of simulations
24
REFERENCES:
[1] C. E. Shannon A mathematical Theory of Communications,
Bell Syst. Techn. Journal Vol. 27, pp. 379-423 (part I) & 623-656
(part II), Oct 1948.
[2] C. Berrou, A. Glavieux and P. Thitimajshima, Near Shannon
limit error-correcting coding and decoding: turbocodes" ICC-1993,
Geneva, Switzerlend, pp. 1064-1070.
[3] J. J. Ramsey, Realization of Optimum Interleavers, IEEE
Trans. on Info. Theory, Vol.16, No.3, May 1970, pp. 338-345
[4] G. D. Forney, Jr., Burst Correcting Codes for Bursty
Channels, IEEE Trans. Comm., Vol. 19, No. 5, Oct 1971, pp. 772-
781.
[5] P. Robertson, E. Villebrun, P. Hocher, A comparison of
Optimal and Sub-Optimal MAP Decoding Algorithms Operation in
the Log-Domain, Proc. ICC95, Seattle, June 1995
[6] B Vucetic, Iterative Decoding Algorithms, PIMRC97, Sept
1997, Finland, pp. 99-120.
[7] S. A. Barbulescu, W. Farrell, P. Gray, and M. Rice, Bandwidth
Efficient Turbo Coding for High Speed Mobile Satellite
Communications", in Proc. International Symposium on Turbo
Codes and Related Topics, Brest, France, pp 119-126, Sep. 1997.
[8] S. S. Pietrobon, Implementation and performance of a
turbo/MAP decoder, International Journal on Satellite
Communications, Vol. 16, pp.23-46, 1998.
[9] S. A. Barbulescu and S. S. Pietrobon, Interleaver design for
turbo codes, Electronics Letters, Vol 30, No 25, Dec. 1994.
[10] W. Feng, J. Yuan and B. Vucetic, A code matched interleaver
design for turbo codes, Proceedings International Symposium on
Personal, Indoor and Mobile radioCcommunications (PIMRC),
Osaka, Japan, pp. 578-582, Sep. 1999.
[11] J. Vogt, K. Koora, A. Finger and G. Fettweis, Comparison of
Different Turbo Decoder Realizations for IMT-2000,
GLOBECOM99, pp.2704-2708, Dec. 1999
[12] J. H. Kang and W. E. Stark, Turbo codes for noncoherent FH-
SS with partial band interference, IEEE Transactions on
Communications, Vol. 46, pp. 1451-1458, Nov. 1998.
[13] D. Roddy Satellite Communications, McGraw-Hill
TELECOM Engineering, 2001.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
On the Performances of Symbol Ranking Text
Compression Method
Radu Rdescu, Rzvan Popa1
Abstract This paper presents an implementation of a
method for text compression, first described by
Shannon in 1951 and later, in 1997, by Fenwick. Unlike
other compressors, which exploit symbol frequencies in
order to assign shorter codes to more frequent symbols,
this technique prepares a list of probable symbols to
follow the current one, ordered from most likely to least
likely. Tests were conducted on various input sources
and the results are shown here. The implementation
supports further optimizations, as it will be explained.
Keywords: symbol ranking, context, number of tries
I. INTRODUCTION
Symbol ranking compression is a method initially
described by Shannon in 1951 [1] and later, in 1996,
by Fenwick [2]. The algorithm is expecting to
encounter repeating strings in the input, which makes
it more suitable for repetitive text, such as human
language. It consists of two blocks: a seeking
function, for both compression and decompression,
which is used to suggest a symbol, and a processing
function, specific to the action being carried out.
Other similar methods do exist, by Bentley et al. [3],
Howard and Vitter [4] or Burrows and Wheeler [5]
(the Burrows-Wheeler Transform), but with little
reference to Shannon's original work. BWT could be
regarded as a symbol ranking compressor, with the
Move-To-Front list acting as a good estimate of
symbol ranking.
II. ALGORITHM DESCRIPTION
The algorithm extends one of Charles Bloom's
methods of offering possible symbols in the
approximate order of probability of their occurring in
the current context. Bloom [6] noted that the longest
earlier context, which matches the current context, is
an excellent predictor of the next symbol. However,
this implementation only uses contexts of a certain
maximal length, as it would be more time-consuming
to leave the context unbound.
1
Facultatea de Electronic i Telecomunicaii, Catedra de
Electronic Aplicat i Ingineria Informaiei, Bd. Iuliu Maniu
nr. 1-3, sector 6, Bucureti, e-mail: rradescu@atm.neuro.pub.ro
The first block of the two mentioned above is
suggesting which the next symbol could be, based on
the current context. It starts by searching for a string
of the maximum permitted length matching the
current context and as soon as it is found, the next
symbol is offered. If the offer is rejected, the search
continues until there are no possible strings of this
length left in the buffer. Then the length is
decremented and the search starts over. The context's
1
length can go as low as 1, meaning a symbol could be
the context, but no lower. When a string is found to
be equal to the current context, the symbol next to it
is first checked to see if it was not offered before and
rejected, in which case it is not used again.
The second block performs as a validation. It reads a
symbol and asks for suggestions, a kind of guess
game. If it receives a good answer, it outputs a "1"
bit, otherwise a "0" bit: this is where the compression
occurs. It does not accept more than a certain number
of guesses, if, until that, the right symbol has not been
offered the search is aborted, the correct symbol is
output, and the scheme moves on. On decompression
things work quite similar, but now the answers are
read from the compressed file, be it a "1" or a "0" bit,
or the correct symbol.
III. EXPERIMENTAL SOLUTION
A software implementation of this compression
method was performed. The program uses a circular
buffer to keep track of processed text, both on
compression and decompression. This means that
when the first symbol is read it is inserted at the first
address in the buffer, and subsequent symbols follow
it, until the end of the buffer is meet, then symbols are
again inserted from the first position, and so on.
Obviously, this way some possible better contexts in
the past are lost, and an ideal approach would store in
the memory all the previous text, instead of a bound
length buffer. That is why the buffer's length can be
varied to some extent, to try to accustom to different
input texts.
25
To find a matching context the search begins in the 0011100100000s1111101000100100000_00000o000
buffer at the previously inserted symbol with the 00f000f000e0001000010001,
maximum context length. It seeks backwards for a
symbol that is equal to the one at the end of the which means that for the 168-bit input, the coder
context. outputs 110 bits, giving a compression ratio of
0.65476. Of course, the result depends a great deal on
Having found one, it tries to match the symbol at the the parameters used in the algorithm and on the
half of the current context to the one at the half of the context (the previous symbols).
possible candidate context and then the first symbols,
in the context and in the candidate. If all these IV. COMPRESSION RESULTS
symbols match, then a complete string comparison is
performed, and, if it succeeds, the symbol is offered The following tests were conducted using 5 files of
and marked in the exclusion table. This approach is various content and dimension:
used to avoid unnecessary comparisons, and many
false contexts fail the half- or start- symbol test. codulpenal.txt Romanian text, legal;
book1.001 English text, fiction book
An example of how the compression sequence could (incomplete, in order to have about the same length as
look like is presented in the following. The text to be the first one a comparison between the two
coded is "the symbol is offered", the context is languages' compressibility was attempted);
"Having found one [...]", from the previous obj1 object code for VAX machine, binary file;
paragraph, all in the buffer. At most 5 unsuccessful pic black & white fax picture (it is supposed to
attempts are allowed. The maximum context size is 4. have a big redundancy and to be very compressible);
In Table 1, the first column is the original text and progl source code in LISP.
the columns to the right are the attempts of guessing
the next symbol. The rightmost cell on every row Every file (except the first one) is part of the Calgary
contains the sequence of output symbols for the Corpus collection [7]. The goal was to try different
corresponding input character, underlined characters scenarios for the use of this algorithm, as it is a
representing a binary value (a "1" bit or a "0" bit) and known fact that a compression method can yield
the numbers between parentheses the context length better results only for certain file types.
at which the right symbol was found, or (n/a) if no
guess was successful. White spaces were converted The program can be tuned by three parameters:
to underscores for reasons of readability.
(a) the length of the buffer;
Table 1 (b) the maximum context length;
Context: Having found one, it tries to match [] (c) the number of tries before outputting the
Text Attempts Output unchanged symbol.
t i a t 001 (2)
h h 1 (3) The following graphics represent the compression
e e 1 (4) ratio of the symbol ranking method when modifying
_ n s _ 001 (4) only one of the parameters, keeping the others to
s c f p h o s 00000s (n/a) some fixed best-ratio values.
y y 1 (4)
m m 1 (4) The columns grouped by five in the histograms below
b b 1 (4) (see Fig. 1, 2, and 3) represent the compression ratio
o o 1 (4) of the files in the above listed order.
l l 1 (4)
_ s _ 01 (4) 1
i a t s i 0001 (1)
s t f s 001 (2)
0.8
Ratio

_ p o y , u _ 00000_ (n/a) 0.6

o p t i s a o 00000o (n/a) 0.4
f n l r m s f 00000f (n/a) 0.2
f _ o i f 000f (na) 0
e _ o i e 000e (na) 4K 8K 16K 32K 64K
r _ d e r 0001 (1)
e f m i s e 00001 (1) Buffer size
d n r _ d 0001 (1)
Fig. 1. Compression ratio as function of buffer size
For the considered text, the output is:

26
As the buffer length is increased, the compression REFERENCES
ratio improves, except for the object code file; still,
the buffer should not be too large, as the running time [1] Shannon, C. E., "Prediction and Entropy of Printed English",
Bell System Technical Journal, Vol. 30, pp. 50-64, January 1951.
can reach high values. [2] Fenwick, P., "Symbol Ranking Text Compression with
Shannon Recordings", Journal of Universal Computer Science,
Vol. 3, No. 2, pp. 70-85, February 1997.
0.8 [3] Bentley, J. L., Sleator, D. D., Tarjan R. E., and Wei, V. K., "A
Locally Adaptive Data Compression Algorithm", Communications
0.6 of the ACM, Vol. 29, No. 4, pp. 320-330, April 1986.
Ratio

[4] Howard, P. G., and Vitter J. S., "Design and Analysis of Fast
0.4 Text Compression Based on Quasi-Arithmetic Coding", Data
0.2 Compression Conference, pp. 98-107, IEEE Computer Society, Los
Alamitos, California, 1993.
0 [5] Burrows, M., and Wheeler, D. J., "A Block-Sorting Lossless
Data Compression Algorithm", SRC Research Report 124, Digital
2 4 6 8 10 Systems Research Center, Palo Alto, California, May 1994,
available at: gatekeeper.dec.com/pub/DEC/SRC/research-
Maximumcontext size reports/SRC-124.ps.Z.
[6] Bloom, C., "LZP: A New Data Compression Algorithm", Data
Compression Conference, Vol. 3, No. 2, pp. 70-85, IEEE Computer
Fig. 2. Compression ratio as function of maximum context size Society, Los Alamitos, California, 1996.
[7] The Calgary Corpus can be found on the Internet at the address:
ftp://ftp.cpsc.ucalgary.ca/pub/projects/text.compression.corpus.
The maximum context size does not appear to make a [8] Fenwick, P. M., "Symbol Ranking Text Compressors: Review
lot of difference; it probably helps occasionally to and Implementation", Software Practice and Experience, Vol. 28,
have a larger context, but overall it seems to be less No. 5, pp. 547-559, April 1998.
important. The default value of this parameter should
be set to 4.

0.8
0.6
Ratio

0.4
0.2
0
3 6 9
Maximumnumber of tries

Fig. 3. Compression ratio as function of maximum number of tries

More tries, more errors; if it does not get right fast,

chances are it will get wrong in the end. The
exception here is represented by the Romanian text,
for which more attempts available is good news.

As a conclusion, a maximum context length of 4-6

characters, with a maximum number of tries between
3 and 6, and a buffer as large as possible should yield
the best ratios.

V. REMARKS

The present implementation of the symbol ranking

text compression method does not completely follow
Fenwick's work. The two omitted steps can increase
compression and speed. Further optimization is
recommended, such as RLE an implementation by
Fenwick [8] uses RLE and hash tables for a fast
compressor. Although the compressor is intended to
process human languages, it performs good on the
object code file.

27
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Application of turbo principle to product codes

Rodica Stoian, Lucian Andrei Perioar 1
Abstract - In this paper we introduce the iterative
decoding principle, the turbo principle, for the
bidimensional Turbo Product Codes (TPCs). The
constituent codes used for encoding on rows and columns
are two concatenated (7,4) Hamming block codes.
Several Soft Input Soft Output (SISO) algorithms can be
used for the iterative decoding process. At each iteration,
the two decoders decode all rows, then all columns. For
particular SISO algorithms, Maximum A Posteriori
(MAP) algorithm and Soft Output Viterbi Algorithm
(SOVA), the system is simulated and performances, in
terms of Bit Error Rate (BER), are evaluated for an
AWGN channel with BPSK modulation.
Keywords: turbo principle, iterative decoding, extrinsic
information
I. INTRODUCTION
Turbo codes were introduced as binary Error
Correcting Codes (ECCs) built up from two Recursive
Systematic Convolutional (RSC) codes concatenated in
parallel. The turbo decoding algorithm, which
processes the data in an iterative way, can achieve very
high coding gain, reaching almost the Shannon limit
[1]. For the decoding of the component codes are used
the Soft Input-Soft Output (SISO) algorithms like
Maximum A Posteriori (MAP) algorithm [1][2] or Soft
Output Viterbi Algorithm (SOVA) [2].
Turbo Product Codes (TPCs), also known as
Block Turbo Codes (BTCs), are based on linear block
codes not on convolutional codes. Here, turbo refers
to the iterative decoding approach and product refers
to the fact that the TPC parameters are the product of
those of its component codes.
Usually, TPCs are built on two or three-
dimensional arrays of block codes. While the encoding
process is done in a single iteration, the decoding
process works with a fixed number of iterations or with
a variable number of iterations and with a stop
criterion.
The turbo principle, more exactly the turbo iterative
decoding algorithm has been successfully applied in
several decoding and detection problems as block
turbo coding [1][2][3], coded modulation [4], multi-
user detection [5], etc.
This paper presents the application of turbo
principle to block array codes, using the BPSK
modulation, the transmission over an AWGN channel
and two SISO algorithms for the decoder.
II. THE SYSTEM MODEL
A. The product code construction
Elias first introduced product codes (or iterated
codes) in 1954 [7]. The concept of product codes is
very simple and relatively efficient for building very
long block codes by using two or more short block
codes.
A product code C = C1 C2 is defined by the serial
concatenation of two block codes C1 (n1 , k1 , d1 ) and
C2 (n2 , k2 , d 2 ) , where ni, ki and di (i = 1, 2) denote the
codeword length, the number of information bits and
the minimum Hamming distance of the code Ci.
n2
k2
Checks
k1 Information bits on rows
n1
Checks
Checks on columns on
checks
Fig. 1. The construction of the product code.
The construction of the product code is shown in
Fig. 1 and can be described by the following steps:
a) the information bits are arranged, line by line, in
an array of k1 rows and k2 columns;
b) the all k1 rows are encoded horizontally using
the code C2;
c) the all n2 columns are then encoded vertically
using the code C1;
d) the bidimensional codewords are transmitted
row by row over the transmission channel.
The parameters of the product code C (n, k , d ) are :
- the matrix codeword dimension n = n1 n2 ;
- the number of information bits k = k1 k2 ;
- the minimum Hamming distance d = d1 d 2 .
The code rate R is given by R = k1k2 n1n2 .

1
Electronics and Telecommunications Faculty, Applied Electronics Departament, Iuliu Maniu 1-3, Bucharest, Romania
e-mail: rodicastoian2003@yahoo.com, lperisoara@yahoo.com
28
 If the two codes can correct t1 = (d1 1) 2 , Lr- = Lec is used as a priori information in conjunction
with Y. The decoder generates the a posteriori log-
respectively t2 = (d 2 1) 2 errors, then the product likelihood ratios Lr+ for all bits. The extrinsic
code C is capable of correcting any combination of y
t = (d1d 2 1) 2 = 2t1t2 + t1 + t2 errors. Thus, we can
build very long block codes with large minimum
Hamming distance by combining short codes with Read entire block
small minimum Hamming distance. and place in array
The parity check matrix H of a product code C is Channel
computed using the parity check matrices H1 and H2 of reliability Y
individual systematic codes C1 and C2 as: computing Lc-
Columns
H T = I n2 H1T | H T2 I n1 (1) Lc
SISO decoder

Lec
where I n1 and I n2 are the unit matrices of order n1 and
n2, respectively. Lr -
Given the construction procedure, it is clear that Rows
(n2-k2) last columns of the matrix are the control bits of SISO decoder
C2. Also, all (n1-k1) last rows of matrix C are the
control bits of C1. Hence, all the rows of matrix C are Ler
Lr+
the codewords of C2 and all the columns of matrix C
are codewords of C1.
Hard
decision
B. The iterative decoder

The Bahl, Cocke, Jelinek and Raviv (BCJR) x

decoding algorithm, used in turbo decoding schemes,
Fig. 2. The iterative decoding principle.
is a Soft-Input/Soft-Output algorithm while the Viterbi
is a Soft-Input/Hard-Output (SIHO) algorithm [6].
McEliece presented a generalized description for information is then defined as Ler = Lr+-Lr--LcY and is
the Viterbi Algorithm (VA), which acts as a unifying used as a priori information for the columns decoder.
concept tying together the Viterbi and BCJR After a fixed number of iterations, the hard decision
algorithms [7]. According McEliece, the Viterbi and is done for each block of received symbols y. Lr+ is
BCJR are the same algorithm, differing only in the computed at the output of the second decoder and the
definition of the semi-ring operation and both original information message u is estimated based on
algorithms can be used to produce an SISO decoder. the sign of the a posteriori values Lr+, as:
To further describe turbo decoding in the context of
TPC, it is helpful to consider trellis description of { }
u = sign L+r (2)
linear block codes (see ANNEX 1). In this description,
the Viterbi algorithm use a metric with (min) and (+) It can be easily seen, from Fig. 2, that the iteration
operations and BCJR algorithm use a metric with principle is applicable to one complete decoding of
(min_log) and (*) operations. columns and one complete decoding of rows. Note that
The iterative turbo decoding can be view as a all the decoding operations are made on all the bits
general Viterbi algorithm used in conjunction with within that block.
MAP or SOVA, with appropriate metric for TPC case For a low complex implementation, we can use the
and specific applications [2][3]. same SISO decoder for rows and columns decoding if
As indicated by Elias [8], the TPC codes can be we add a block interleaver at the input of the rows
decoded by sequentially decoding the rows and decoder and a deinterleaver at the output. Also, when
columns of C in order to reduce decoding complexity. the constituent codes C1 and C2 are identical, the two
However, to achieve optimum performance, one must decoders can be identically.
use soft decoding of the component codes using SISO
decoders. More over, we can iterate the sequential
decoding of C and thus reduce the BER after each
iteration as for turbo codes [1].
The iterative decoding process is described in Fig.
2. The decoding is performed iteratively column-wise
then row-wise using SISO decoders. The column
decoder uses the channel observations Y and the a
priori information Lc- in the form of log-likelihood
ratios to generate the a posteriori log-likelihood ratios
Lc+ for all bits received. The extrinsic information is
defined as Lec = Lc+-Lc--LcY. For the second decoder,

29
 def P (y | x = +1)
- L( y | x) = log (5)
P (y | x = 1)
SISO decoding of Le(i)
rows or columns of + According [9], for AWGN fading channel using
L-(i) matrix Y L+(i) L-(i+1)
- BPSK modulation we can write:
LcY
1 E 2
P( y | x = 1) = exp b ( y m a ) , (6)
N 0 N 0

Y Lc where Eb is the transmitted energy per bit, a is the

fading amplitude and N0/2 is the noise variance.
We can rewrite the equation (5) as following:
Fig. 3. The elementary block turbo decoder.

In this case, the decoding procedure described
above is generalized by cascading elementary decoders
illustrated in Fig. 3. The parameter i indicates the
current decoding step of the iterative process. For the
implementation of SISO decoders, we can use the
MAP algorithm or the SOVA algorithm, which are
described below.
C. The Maximum A Posteriori algorithm
Bahl, Cocke, Jelinek and Raviv proposed the
Maximum A Posteriori (MAP) decoding algorithm for
convolutional codes in 1974 [6]. The iterative decoder
developed by Berrou et al. [1] in 1993 has a greatly
increased attention. In their paper [1], the MAP
algorithm was modified to minimize the sequence error
probability instead of bit error probability for the
original MAP algorithm. Because of its increased
complexity, the MAP algorithm was simplified and the
optimal MAP algorithm called the Log-MAP
algorithm was developed.
The decoder operates based on the Logarithm
Likelihood Ratio (LLR) for the transmitted bits x
which is defined as:
def P( x = +1)
L( x) = log =L
( = 1)
P x
where the sign of the LLR L(x) indicate whether the
each bit of x is more likely to be +1 or -1 and the
magnitude of the LLR gives an indication of the
correct values of x.
In channel coding theory we are interested in the
probability that x = 1 , based or conditioned on
received sequence y. So, we use the conditional LLR:
def P (x = +1| y ) We noted +
L( x | y ) = log = L
P (x = 1| y )
The conditional probabilities P (x = 1| y ) are the
a posteriori probabilities of the decoded bits x and L+
is the a posteriori information about x.
Also, it is used the conditional LLR L( y | x) based
on the probability that the receivers output would be y
when the transmitted bits x were either +1 or -1:
Eb E Noted
( y a ) ( y + a ) = 4a b y = Lc y
2 2
L( y | x) =
N0 N0
(7)
where Lc = 4a Eb N 0 is defined as the channel
reliability value. For non fading AWGN channels a=1
and Lc = 4 Eb N 0 .
In [1], [9] the extrinsic information is defined as:
P( x = +1| y ) P (x = +1)
Le = log log
P( x = 1| y ) P (x = 1)
P(y | x = +1)
log (8)
P(y | x = 1)
= L+ L Lc y
In the iterative decoding procedure the extrinsic
information Le becomes the a priori information L- for
the next decoder. If L- is a large (or small) positive
number, then it would be difficult (or easier) to change
the estimated symbol decision from +1 to -1 between to
consecutive decoding stages [10].
The term Lcy is the soft output of the channel for the
information symbol x. For high SNR, the channel
reliability value Lc will be high and this information
(3)
symbol will have a large influence on L+. Conversely,
for low SNR, the Lc is low and its influence on L+ is
insignificant.
D. The Soft Output Viterbi Algorithm
In practical systems, we quantize the received
channel symbols with one (hard decision) or a few bits
of precision (soft decision) in order to improve the
performances of the Viterbi decoder. For m-bit
quantization, one quantization bit is devoted to the sign
of the decision and m-1 bits are devoted to the signal's
(4)
magnitude. The larger the magnitude, the more
confidence that the sign bit is correct. Decoders that
exploit soft decisions can reduce S/N ratio requirements
by approximately 2 dB over those that use hard
decisions alone [11].
The Viterbi algorithm finds the trellis path or state
sequence s so that the a posteriori probability p(s|y) is
maximized. Accordingly to the Bayes rule, we can
equivalently maximize:

30
 p(s j , y j ) = p (s j 1 , y j 1 ) p(u j ) p( y j | s ', s ) , (9) 1

Bit Error Rate

where s j = ( s1 , s2 ,..., s j ) , y j = ( y1 , y2 ,..., y j ) , s = sj-1 0.1
uncoded
and s = sj. The uj is the source symbol for the state
transition ss of trellis path sj. The path metric Mj(sj) 0.01
associated with the trellis path sj is defined as:
nriter=1
( )
0.001
M j (s j ) = log p(s j , y j ) . (10) nriter=5 nriter=2

0.0001
Obviously, nriter=3

(
p(s j , y j ) = exp M j (s j ) .
Substituting (9) into (10) gives:
) (11)
0.00001
0 2 4 6
Eb/N0 (dB)
Fig. 4. BER(Eb/N0) performance for MAP algorithm.

( ) (
M j (s j ) = M j 1 (s j 1 ) + log p(u j ) + log p( y j | s ', s ) )
(12) The performances of the SOVA algorithm are
illustrated in Fig. 5. In this case, for the same Eb/N0
(
where log p(u j ) ) is the a priori information of the of 3dB the BER is greater, 0.0867 for one iteration,
0.0273 at iteration 2, 0.0016 at iteration 3 and
source symbol uj and log p( y j | s ', s)( ) is the branch 0.0007 at iteration 5.
metric for the state transition ss given the received
signal yj. At time j, for each state s, the path metrics for From Fig. 4 and Fig. 5 we observe that the MAP
all possible paths terminating at state s are calculated. algorithm gives better results, in terms of BER,
compared with SOVA algorithm.
III. PERFORMANCE EVALUATION
1
To simulate the application of iterative decoding uncoded
Bit Error Rate

principle, we applies the described algorithms to TPC 0.1

ensemble which use two identical systematic
Hamming block codes H1(7,4,3), H2(7,4,3) 0.01
concatenated in a serial way. nriter=5 nriter=1
The product code is H1(7,4,3) H2(7,4,3) =
0.001
H(49,16,9) and the output sequence of TPC is BPSK nriter=3
modulated and transmitted over an AWGN channel,
with fading amplitude a = 1 [11]. 0.0001
The most important characteristic of iterative nriter=2
principle is the dependence of BER(Eb/N0) of the 0.00001
number of decoding iterations. 0 2 4 6
Bit Error Rate is computed over 105 blocks, each Eb/N0 (dB)
block of dimension 49 bits. Fig. 5. BER(Eb/N0) performance for SOVA algorithm.

For Maximum A Posteriori algorithm we obtain the

curves plotted in Fig. 4. For each additional
iteration, we obtain a reduction of BER. We
observe that for an Eb/N0 of 3dB the BER is equal
to 0.0633 for one iteration, 0.0105 at iteration 2,
0.00054 at iteration 3 and 0.000043 at iteration 5.
IV. CONCLUSIONS
In this paper, two iterative SISO decoding
algorithms for TPC have been presented. It has been
proved that the two-bit soft decision decoding for TPC
(SISO algorithms) can pick up 2 dB of additional
coding gain compared with SIHO variant [11][12].
For these results, the complexity is low and TPC
systems starts to be available as standard products. Of
major interest are the combination of the TPC coding
with modulation and the development of specific SISO
algorithms, combined with helical data scrambling to
improve burst error performance

31
 ANNEX 1. TRELLIS DESCRIPTION Conference on Communication, vol. 2, pp. 974-978, Dallas, Texas,
June 1996.
OF BLOCK CODES [5] A. Stefanov and T. M. Duman, Turbo-coded modulation for
wireless communications with antenna diversity, in Proc. IEEE
For a Hamming code with control matrix Vehicular Technology Conference, pp. 1565-1569, Amsterdam,
H = [h1 , h 2 ,..., h n ] , where hi is the ith column of H, any Netherlands, September 1999.
[6] L. R. Bahl, J. Cocke, F. Jelinek and J. Raviv, Optimal
codeword ci = (ci1 ,ci 2 ,..., cin ), ci C (n, k ), i = 1, n Decoding of Linear Codes for Minimizing Symbol Error Rate, IEEE
Trans. on Inf. Theory, vol. 20, pp. 284-287, 1974.
must satisfy the condition: [7] R. J. McEliece, On the BCJR trellis for linear block codes,
IEEE Trans. on Inf. Theory, vol. IT-42(4), pp. 1072-1092, 1996.
ci1h1 + ci 2 h 2 + ... + cin h n = 0 (13) [8] P. Elias, Error-free coding, IRE Trans. on Inf. Theory, vol.
IT-4, pp. 29-37, September 1954.
[9] C. Berrou and A. Glavieux, Near Optimum Error Correcting
where cij F2 and h j F2n k . Coding and Decoding: Turbo Codes, IEEE Trans. on Comm. vol.
44, no. 10, pp.1261-1271, October 1996.
For any codeword affected by errors the value of [10] R. Stoian, L.A. Perioar, The reliability of turbo decoders
the syndrome is: over Gaussian channels, in Proc. of Communications 2004,
Technical Military Academy, pp. 545-550, Bucharest, June 2004.
n
[11] J. Hagenauer and P. Hoeher, A Viterbi Algorithm with Soft
s n = yi h i (14) Decision Outputs and Its Applications, in Proc. of GLOBECOM
1989, pp. 1680-1686, Dallas, Texas, November 1989.
i =1
[12] R. M. Pyndiah, Near-optimum decoding of Product Codes:
Block Turbo Codes, IEEE Trans. on Comm., vol. 46, no. 8, pp.
where yi are the components of received vector y. 1003-1010, August 1998.
The BCJR trellis construction for linear block
codes is based on recursive computation of the
syndrome [6]:

si = si 1 + yi h i , s 0 = 0 (15)

which determine the unconstrained trellis. Because for

linear block codes the initial and final states must be 0,
the branches of the unconstrained trellis, which not
start from the state 0 and not end at the state 0, are
removed. Fig. 6 shows the BCJR trellis for the
systematic Hamming block code H(7,4,3).

111

110

101

100

011

010

001

000

symbol 0 symbol 1

Fig. 6. The BCJR trellis for the systematic H(7,4,3).

REFERENCES
[1] C. Berrou, A. Glavieux and P. Thitmajshima, Near Shannon
limit error-correcting coding: Turbo codes, in Proc. of ICC 93,
pp.1064-1070, Geneva, Switzerland, May 1993.
[2] R. Stoian, L. A. Perioar, M. Crngeanu, A comparison of
MAP and SOVA decoding algorithms for Turbo Codes, in Proc. of
the 35th Symposium of ACTTM, Bucharest, March 2004.
[3] J. Hagenauer, E. Offer and L. Papke, Iterative decoding of
binary block and convolutional codes, IEEE Trans. on Inf. Theory,
vol. 42, pp. 429-445, March 1996.
[4] S. Benedetto, D. Divsalar, G. Montorsi and F. Pollara, Parallel
concatenated trellis coded modulation, in Proc. IEEE International

32
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

A study on turbo decoding iterative algorithms

Horia Balt1, Maria Kovaci2
Abstract - In the paper, a study of some turbo decoding
iterative algorithms: MAP, MaxLogMAP, LogMAP, is k(,s) = P({yk Sk = s}| Sk-1 = ) is the probability
presented. For the correction of the approximation used that the encoder trellis took the transition from state
in the MAxLogMAP algorithm two methods are
to state s and the received channel sequence for this
proposed obtaining two LogMAP algorithm variants. All
algorithms variants have been simulated to make transition is yk.
possible a comparison from the bit error rate (BER)
point of view, in order to provide an optimization for k (s , s ) = C e (u k L(u k ) / 2 )
each algorithm.
The simulations were made for AWGN channel. Two E n (4)
component codes with generator matrix: G1 =[1, 5/7] exp b 2 a y ki x ki

and two interleaver types: pseudo-random [1] and S- 2 2 i =1
interleaver (S=29) are used. The interleaving length is
N=1784. The number of emitted blocks in one simulation
depends of signal to noise ration, SNR, to obtain a good In relation (4), uk is the value of information bit for
precision of resulted curve. the trellis branch, L(uk) is the extrinsic information
Keywords: Turbo codes, MAP algorithm, trellis. for the k-th bit, Eb is the energy of the information
bit, 2 is the noise power, yki and xki represent
I. INTRODUCTION corresponding values of all the bits attached to the
branch which makes the liaison of states and s, from
The Maximum A-Posteriori (MAP) algorithm, reception (yki), respectively, emission (xki).
proposed by Bahl, Cocke, Jelinek and Raviv (1974), Fig.1 presents the computation way of forward-
is frequently used after the turbo codes discovery and backward- coefficients, for a part of trellis of
realized by Berrou and al. [1]. Essentially, MAP convolutional code with constraint length K=3.
algorithm, [2], calculate the Log Likelihood Ratio,
LLR, under the form: Sk- Sk-2 Sk-1 Sk Sk+
k -1 (s ) k (s, s ) k (s )

(s,s)
(
L uk y = ln )
uk = +1
k -1 (s ) k (s, s ) k (s ) (1)
( s,s)
u = 1
k

where:k-1() = P(Sk-1 = yj<k) is the probability that

the encoder trellis was in state at instant k-1 and the yj<k yk yj>k
received channel sequence, before this moment, is
yj<k,
k-1() k-1(,s) k(s)
k ( s ) = k ( s,s
) k 1 ( s ) (2)
alls s Fig.1 The code trellis with G = [1, 5/7]. The continuous line
corresponds of the input bit value 1.

k(s) = P(yj>k| Sk = s) is the probability that, having

Due to the exponential and logarithm operators in
been given the trellis state s at instant k, the received
relations (1) and (4), the MAP algorithm is difficult to
channel sequence, after this moment, to be yj>k,
be implemented.
Like an alternative, the MaxLogMAP algorithm is
k 1 ( s ) = k ( s,s
) k ( s ) (3)
easier to be implemented, due to the approximation:
alls s

1,2
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, balta@etc.utt.ro; kmaria@etc.utt.ro

33

ln e xi max(x i )
i
i closing techniques can be used for these coders. In
So, the computing relations, in the MaxLogMAP
algorithm case, are the following:
Ak (s ) =
ln( k (s )) max( Ak 1 (s ) + k (s , s ))
s Variant
B k (s ) =
ln( k (s )) (B k (s ) + k (s , s ))
k (s , s ) =
ln( k (s )) =
1 L n
= C + u k L(u k ) + c y ki x ki
2 2 i =1
L(uk|y) max (Ak-1() + k(,s) + Bk(s))
(s,s )
u k = +1
max (Ak-1() + k(,s) + Bk(s))
(s,s )
u k = 1
The implementation simplification price of the
MaxLogMAP algorithm is the reduction of the
performances (of the BER) with 0,2 dB versus the
MAP algorithm.
LogMAP Algorithm, proposed by Robertson and al.
[3], corrects the approximation used by MaxLogMAP
algorithm and is a little bit more complicated than it.
x x
ln( e x1 + e x 2 )=max(x1,x2)+ln(1+ e 1 2 )
= max(x1,x2) + c( |x1-x2| )
II. THE TRELLIS CLOSING
In function of the trellis closing the alpha and beta
coefficients are initialized. The initial trellis closing
means the coder initialization with a predefined state.
This state is also known by the decoder. So, the
initialization of the alpha and beta coefficients can be
done. With the exception of circular coding, this
initial state is zero. The final trellis closing is more
difficult to be realized. It is done (excepting the
circular code) with the price of insertion of the M (the
code memory) redundant bits in the information
sequence. This fact realizes the reduction of the
transmission rate from 1/2 to the following value:
Rcc = (N M) / 2N.
The trellis closing gives the advantage of the initial
state knowledge (and/or of the final state), fact which
leads to the firm knowledge of the alpha coefficients
(at the beginning of trellis) and beta coefficients (at
the end of the trellis). In the case of the unclosed
trellis these coefficients can only be predicted
probabilistically.
A turbo code implies at least two coders, C1 and C2.
(5) At each coder corresponds a trellis. Different trellis
this paper we investigate few trellis closing methods
for a turbo code (parallel). The table 1 presents these
methods.
(6) Table 1
Start Final Coding rate
C1, C2 C1, C2
(7) 01 0, 0 0, ? (N-M)/3N
11 0, 0 ?, ? 1/3
C Sx, Sy Sx, Sy 1/3
(8) 01. In this case, the first coder closes the trellis on
both extremities, it inserts M redundant bits after the
N-M information bits. The second coder can not do
the same final closing due to the interleaving of the
input sequence. So, the second trellis is not closed.
(9) The first decoder initializes the alpha coefficient,
which corresponds at the front end of the trellis to the
zero state, with the probability 1, and the other alpha
coefficients with the probability zero. The first
decoder treats the beta coefficients in the same way at
the end of the trellis. The second decoder acts in the
same way, like the first, for the alpha coefficients.
The beta coefficients of the second decoder, can be
initialized by the one of the following methods:
01s. The beta coefficients are met equals with
the values of the alpha coefficients obtained at the last
iteration. This is called the soft initialization;
01h. This method initializes the beta coefficient,
that corresponds to the state with the highest alpha
(10)
coefficient, at probability 1 and the others beta
coefficients at the probability zero. This is called the
hard initialization;
01e. This method makes the beta coefficients to
have the same probability. This is called the equal
probability initialization;
11. None of the trellises is final closed. The
advantage, in this case, is a higher coding rate. But
this coding rate increase can not be observed if
N>>M. Both decoders must initialize the beta
coefficients in one of the three ways enounced above.
In this paper we implement only the soft initialization.
The case of both trellis final closing is possible only
with some modifications of the interleaving between
the two coders.
C. There is the possibility, using a pre coding
technique, to find, for any data sequence x, an initial
state So of the coder identically with its final state. So
the coding becomes circular. The decoder does not
(11) know the state S0, but knows that it can use the final
state like initial state. So, it must to do at least a
forward recurrence. We are implemented and
simulated the following variants of circular turbo
code:
C1 the decoder realizes a forward recurrence and
computes a final state, S0. The alpha and beta
coefficients are initialized with S0, the backward
recurrence is made and the forward recurrence is
34
remade. It memorizes the new state S0 for the starting
of the next iteration.
C2 the decoder realizes the both recurrences in the
soft variant and retains, for the next iteration, with the
role of S0, the beta coefficients values from the end of
the backward recurrence.
C3 the decoder realizes the both recurrences in the
soft variant plus one for the alpha coefficients, only.
The initial state for that second recurrence is done by
the finals values of beta coefficients of the last
iteration. The final coefficients of the second forward
recurrence give the state to be stored for the next
iteration.
C4 the decoder realizes the forward recurrence and
build a S0 state in a hard decision (it searches the
alpha coefficients maximum). It retains this state for
the next iteration and also makes the backward
recurrence and remakes the forward recurrence.
III. THE LOGMAP ALGORITHM.
IMPLEMENTATION.
The variants of LogMAP algorithm differs by the
correction term approximation way, described in
equation (10):
fc(x) = ln(1+e-x) , x 0 (12)
The functions that approximate fc(x) must be easy to
implement and they must reproduce the most exactly
possible the form of this function. Two
approximations were proposed in this paper, indicated
in Fig. 2 and Fig.3.
fc(x)
gc(x)
x obviously the performances are similar. The curves
Fig.2. The rectangular approximation way.
fc(x)
hc(x)
Fig.3. The linear approximation way.
The rectangular variant proposed by Robertson and
all. [3] is a zero order extrapolation of the function
35
fc(x). The values of the function gc(x) are in the set
{0.6, 0.3, 0.14, 0.065, 0.03, 0.014, 0.005, 0.002, 0}.
The linear variant corresponds to an approximation of
fc(x) of the form:
0.7
0.7 x, x x o
hc (x ) = xo (13)
0, x > xo
and by numerical approximation was obtained the
value xo = 2,347 for which hc(x) realizes the better
approximation of fc(x).
IV. EXPERIMENTAL RESULTS
In the figure 4 are presented the curves BER(SNR)
obtained with the three MAP algorithms variants 01
plus the MAP algorithm 11s. Despite the fact that
for signal to noise ratios inferior to 1 dB the
performances are identical, up to this value the results
show that the variant 01e is better. It is followed, in
order, by the variants: 01s, 11s and 01h. These results
show that at low signal to noise ratios the errors are
produced exclusively by the bad selection of the path
in the trellis and up 1 dB the errors due to the trellis
non closing have a higher weight.
In figure 5 are represented the BER(SNR) curves
obtained with the four variants of the circular MAP
algorithm already defined in comparison with the best
MAP algorithm: 01e. The first three circular MAP
variants have similar performances, inferior to the
performances of the variant 01e.
Tacking into account all the results already presented
it results that the hard variant is not a good solution.
The simulation results realized with 1/3 rate RSC
turbo code (parallel) with G=[1,5/7], which utilizes in
the variant 00s the LogMAP algorithms are compared
in Fig.6 with the results obtained with the best MAP
variant: 01e. From figure results that all the two
LogMAP variants are better than the MAP at least for
values of the signal to noise ratio inferior to 1 dB. Up
this value the curves are not very accurate but
reduced precision is due to the reduced volume of
simulations.
- Despite the fact that practical implementations of the
LogMAP algorithm are faster than those of the MAP
algorithm the simulation programs work slower in the
case of the LogMAP algorithm.
Between the two LogMAP algorithm variants the
results show that the linear one is better. These
conclusions must be verified also for other component
codes.
 BER

uncoded
MAP01s
MAP01h
MAP01e
o MAP11s

SNR
Fig. 4 The BER curves obtained with: MAP01s, MAP01h, MAP01e, MAP 11s algorithms.
BER

uncoded
MAPC1
MAPC2
MAPC3
o MAPC4
+ MAP01e

SNR
Fig. 5 BER performances of C1, C2, C3, C4 algorithms versus MAP01e algorithm.
BER

uncoded
MAPC1
MAPC2
MAPC3

SNR
Fig. 6 BER performances of rectangular and linear LogMAP algorithms versus MAP01e algorithm.

36
 V. CONCLUSIONS

In the paper, a study of some turbo decoding iterative

algorithms: MAP, MaxLogMAP, LogMAP, was
presented. For the correction of the approximation
used in the MAxLogMAP algorithm, two methods
were proposed, obtaining two LogMAP algorithm
variants. All algorithms variants have been simulated
to make possible a comparison from the bit error rate
point of view, in order to provide an optimization for
each algorithm.

VI. REFERENCES

[1] C. Berrou, A. Glavieux, P. Thitimajshima Near Shannon limit

error-correcting coding and decoding: Turbo-codes, Proc.ICC93,
Geneva, Switzerland, May 1993, pp. 1064-1070.
[2] L.Hanzo, T.H.Liew, B.L.Yeap, Turbo Coding, Turbo
Equalisation and Space-Time Coding for Transmission over Fading
Channels, John Wiley & Sons Ltd, England, 2002
[3] P. Robertson, E.Villebrun, P.Hoeher, A Comparison of
Optimal and Sub-Optimal MAP Decoding Algorithms Operating in
the Log Domain, Proceedings of the International Conference on
Communications, Seattle, USA, pag. 1009-1013, iunie 1995

37
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

The Performances of Convolutional Codes used in Turbo

Codes
Horia Balt1, Maria Kovaci2
Abstract In this paper are presented and compared the LLR1
x0 y0
BER performances obtained by the simulation of a u DEC1
C1 x1 y1

Transmission
transmission system, which utilizes the forward error

Channel
correcting by codes concatenation and iterative decoding Iex21
(turbo coding). There have been investigated all the I I I
DI
systematic convolutional codes having the constraint
C2 x2 y2 DEC2 Iex12
length K less or equal to 6, under three diffrent
concatenated forms: parallel PCCC (pure turbo code),
serial SCCC and hybrid HCCC, at the following rates: a)
u
1/3, 1/4 and 1/3, respectively, all unpunctured.
Two interleaver types were used: pseudo-random
and S-interleaver having the same length, 1784. C1 MUX x , x
1 2 y1, y2
The AWGN channel and the BPSK modulation LLR1

Transmission
were employed. I DMUX

channel
The used iteration number was eight. For
increasing the work speed an iterations stop criterion I DEC1
DEC2 DI DMUX
was used. When the resulting error number from the
decoding of a data block is zero, the remaining iterations C2 I MUX
x3 y3
are not effectuated, passing to the next block.
For decoding, the MAP, MaxLogMAP and Log b)
MAP algorithms were used. In all the cases, a tail off
LLR1
was employed for the first code, with the decreasing x0
u y0
transmission rate price. C1 x1 DEC1
y1
The transmitted data block numbers for a
Transmission

simulation were chosen in function of the signal to noise I

channel

ratio, SNR, i.e. to keep a good precision for obtained I DI

curve. C2
Keywords: convolutional codes, turbo codes, interleaver, I
iteration. x2 y2 DEC2

I. INTRODUCTION
Two procedures which improve the performances of
convolutional codes, CCs, (and block codes), from
point of view of error rate (BER), are concatenation
and iterative decoding.
Fig.1 illustrates the possible ways of convolutional
codes concatenation. Concatenated convolutional
schemes tend to fall into three categories: parallel
concatenated convolutional codes, PCCCs, (as in
Fig.1 a)), serial concatenated convolutional codes,
SCCC, (as in Fig.1 b)) and hybrid concatenated
convolutional codes, HCCC, (as in Fig.1 c)).
The PCCC (or Turbo code) was introduced by Berrou
et al. [1], in 1993, and it was the beginning of turbo
code era.
c)
Fig.1 a) Parallel concatenated convolutional codes, b) Serial
concatenated convolutional codes, c) Hybrid concatenated
convolutional codes
Blocks I and DI realize interleaving and
deinterleaving functions. We used in this paper two
interleavers: pseudo-random [2] and S-interleaver [3].
Our simulations prove that the interleavers have an
essential influence on performances of Turbo codes.
DEC1 and DEC2 are iterative decoder blocks [4],
which implement algorithms like: the MAP algorithm,
the first and the must important, proposed by Bahl and
al. [5], the MAXLogMAP algorithm [6], and the Log
MAP algorithm, proposed by Robertson and al. [7].
The constitutive codes can be convolutional codes or
block codes. In this paper we studied the first
exclusively. The general scheme of a recursive

1,2
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, balta@etc.utt.ro; kmaria@etc.utt.ro

38
systematic convolutional code, RSC, is shown in Fig. The two polynomials attached, a and b, define in
2 a) and an example of RSC is shown in Fig. 2 b). totality the convolutional code. The maximum degree
of a and b polynomials give the coders memory
(equal with K-1) and it is a measure of the complexity
systematic
and of the volume of the effectuated computation of
each component decoder. It increases exponentially
x[n] b1 b2 bM-1 bM with K or M [6].
+ D D D Practically, convolutional codes with K=36 are used.
input
Table 1 shows the generator polynomials with degree
a0 a1 a2 aM-1 aM
w[n] inferior or equal with 5 which can be selected like a
or b. Because of the restriction that a and b to be
+ prime the table shows also the possible divisors for
each polynomial.
y[n]
output Table1

Ireductibil
Primitive
Degree
a) Polynomials in octal and their
divisors
systematic

x[n] 0 * 1 1 0 0 0 0 0 0 0
+ D D 1 * * 3 1 0 0 0 0 0 0 0
input
2 5 1 3 0 0 0 0 0 0
+ * * 7 1 0 0 0 0 0 0 0
3 11 1 3 7 0 0 0 0 0
y[n] * * 13 1 0 0 0 0 0 0 0
output * * 15 1 0 0 0 0 0 0 0
b) 17 1 3 0 0 0 0 0 0
4 21 1 3 0 0 0 0 0 0
Fig.2 Recursive systematic convolutional code: a) general
* * 23 1 0 0 0 0 0 0 0
scheme, b) example
25 1 0 7 0 0 0 0 0
The following ecuations result: 27 1 3 0 0 15 0 0 0
* * 31 1 0 0 0 0 0 0 0
33 1 3 7 0 0 0 0 0
M
y[n] = a k w[n k ] (1) 35 1 3 0 13 0 0 0 0
k =0 * 37 1 0 0 0 0 0 0 0
5 41 1 3 0 0 0 0 0 37
M 43 1 0 7 0 15 0 0 0
w[n] = bk w[n k ] + x[n] (2) * * 45 1 0 0 0 0 0 0 0
k =0 47 1 3 0 13 0 0 0 0
* * 51 1 0 0 0 0 0 0 0
M 53 1 3 0 0 0 0 31 0
Y (D ) = a k D k W (D ) (3) 55 1 3 7 0 0 0 0 0

k =0 * * 57 1 0 0 0 0 0 0 0
61 1 0 7 13 0 0 0 0
M 63 1 3 0 0 0 0 0 0
W (D ) b k D k = X (D ) (4) 65 1 3 0 0 0 23 0 0
k =0 * * 67 1 0 0 0 0 0 0 0
71 1 3 0 0 15 0 0 0
M * * 73 1 0 0 0 0 0 0 0
k
ak D (5) * * 75 1 0 0 0 0 0 0 0
Y (D ) k = 0 77 1 3 7 0 0 0 0 0
=
X (D ) M k
bk D
k =0 II. EXPERIMENTAL RESULTS
The code generator matrix, G(D), is: Table 2 and Table 3 present the simulation results
obtained with parallel concatenated convolutional
a (D )
G (D ) = 1, (6)
code, Fig.1 a), rate R=1/3, RSC, with the pseudo-
b (D ) random interleaver (table 2), and S-interleaver (table
3).

39
Table 2 (pseudo-random interleaver)
1 3 5 7 11 13 15 17 21 23 25 27 31 33 35 37
1 0 2954035 3021300 1935725 2864349 1188340 1177130 1093049 3157698 1073326 1821748 710482 1242932 585061 560538 704876
3 3612356 0 0 1358637 0 527925 498583 0 0 243060 156083 0 318487 0 0 392376
5 4516335 0 0 100348 0 61305 62825 0 0 5311 1471412 0 5259 0 0 20042
7 1262331 919632 5710 0 0 5163 6663 378923 1419 1445 0 8216 1916 0 5840 289992
11 4684497 0 0 0 0 37148 37295 0 0 47924 0 0 37591 0 0 437
13 4684497 33594 6755 7043 1720 0 312 292 220 3144 243 8839 265 1192 0 155
15 108671 34188 9633 6457 1294 235 0 403 204 579 313 0 2910 1077 7302 229
17 445882 0 0 230633 0 9302 9040 0 0 8749 1741 0 10009 0 0 249017
21 4886023 0 0 13579 0 11594 11975 0 0 176919 159130 0 173697 0 0 8037
23 178171 20128 1102 521 2061 1629 230 691 6165 0 578 476 1032 1283 3005 586
25 1420029 5267 966445 0 0 751 1159 926 86082 34388 0 3327 1676 0 1904 1999
27 21498 0 0 6201 0 12060 0 0 0 345 1105 0 373 0 0 546
31 429482 6548 1855 600 2888 418 1823 344 10420 3162 207 828 0 776 534 1443
33 10949 0 0 0 0 2602 2471 0 0 417 0 0 450 0 0 1374
35 10609 0 0 6446 0 0 9675 0 0 241 558 0 368 0 0 270
37 135016 123269 1379 114297 1445 236 449 104996 956 757 10480 1418 737 901 731 0
41 4107943 0 0 2430 0 1991 1651 0 0 2242 2031 0 1643 0 0 0
43 424030 8621 7204 0 0 456 0 2488 6217 551 0 0 2140 0 6606 3117
45 299338 13935 5521 1638 3316 901 555 6442 1873 12670 4607 854 3971 5999 4711 13557
47 38916 0 0 10547 0 0 3771 0 0 1679 9612 0 13089 0 0 6179
51 205797 7216 6101 1502 3468 678 630 2024 1006 3910 5735 3119 17711 5515 3467 15763
53 9846 0 0 599 0 2125 6769 0 0 962 681 0 0 0 0 1890
55 16898 0 0 0 0 3796 3708 0 0 1940 0 0 13015 0 0 16663
57 5160 1513 2592 2879 913 748 4673 879 4271 1351 1768 8903 30764 20257 3587 2084
61 510340 12737 2421 0 0 0 607 721 6592 4919 0 18487 744 0 0 5890
63 125731 0 0 38990 0 29225 38412 0 0 39576 43539 0 36760 0 0 38388
65 28245 0 0 698 0 2550 3709 0 0 0 2436 0 2649 0 0 1497
67 2075 8583 2910 2383 8190 355 2117 607 22152 26281 20200 1009 2018 1191 3883 11581
71 21641 0 0 7941 0 18221 0 0 0 4869 8983 0 945 0 0 2437
73 2587 6523 3030 4425 12835 5061 471 313 114070 997 16231 1540 12697 977 424 3386
75 9336 4653 11238 1249 968 12662 961 2685 6851 104145 2325 7577 781 16290 1835 450
77 94043 0 0 0 0 1404 2169 0 0 1751 0 0 2123 0 0 73523

41 43 45 47 51 53 55 57 61 63 65 67 71 73 75 77
1 2558092 1291785 1193479 523168 1207492 480732 644618 277521 1286689 589732 508488 246833 586229 251741 312032 409436
3 0 145046 50884 0 41916 0 0 31688 119502 0 0 137957 0 139590 32654 0
5 0 6347 50865 0 52682 0 0 20946 5037 0 0 1560 0 1861 23048 0
7 348 0 262 13007 216 329 0 737 0 257 363 18076 11557 14284 351 0
11 0 0 43879 0 44683 0 0 401 0 0 0 416 0 445 662 0
13 7029 416 199 0 250 1679 1008 195 0 392 242 153 215 1689 265 424
15 1396 0 198 122 151 482 1639 499 370 282 1975 2524 0 466 235 359
17 0 8663 1597 0 5878 0 0 8822 9446 0 0 8446 0 8800 9174 0
21 0 155602 9816 0 9225 0 0 1372 159927 0 0 14240 0 12664 3965 0
23 1917 254 616 140 17637 261 604 368 3818 1717 0 1465 13100 471 14498 2881
25 1267 0 865 15090 359 2107 0 2429 0 4654 2191 17575 15184 9580 1931 0
27 0 0 586 0 1273 0 0 4623 23549 0 0 297 0 585 13015 0
31 6477 3924 21692 5640 332 0 490 27704 250 1406 185 379 253 3520 365 14592
33 0 0 13682 0 16959 0 0 6718 0 0 0 1994 0 2053 2180 0
35 0 12497 820 0 265 0 0 15250 0 0 0 547 0 251 3434 0
37 0 617 6125 912 13870 3399 21406 885 723 229 5265 1037 837 972 909 90250
41 0 52386 51113 0 54221 0 0 3930 44754 0 0 74738 0 11696 4972 0
43 12883 0 7156 5074 4128 6676 0 2928 0 8412 18832 18168 0 14370 14586 0
45 7550 7623 0 4591 23549 16302 4950 16167 4498 37402 2142 19979 28922 17354 20076 24917
47 0 29280 13511 0 24500 0 0 13531 0 0 0 13803 0 16293 113221 0
51 8407 4568 39701 11210 0 525 12873 24937 3149 14515 129624 24140 3039 15473 27847 69808
53 0 11421 16531 0 4511 0 0 23816 149866 0 0 13064 0 9364 7155 0
55 0 0 8543 0 6700 0 0 21371 0 0 0 11158 0 14395 11214 0
57 17131 27777 26594 1752 14839 10324 8576 0 23236 15364 10750 15603 6537 6747 23012 6439
61 10064 0 5965 0 2505 23247 0 35109 0 3933 3802 9639 3637 89620 14001 0
63 0 7429 35644 0 29862 0 0 19276 19300 0 0 6196 0 14240 18842 0
65 0 113080 5937 0 20680 0 0 14321 7637 0 0 19730 0 14989 4491 0
67 12965 25181 13423 15669 12916 7387 7400 13738 3109 13093 9074 0 6612 2022 20297 30407
71 0 0 15047 0 13372 0 0 14385 5929 0 0 14642 0 12603 7323 0
73 12240 1979 40387 16590 24624 14869 5216 19117 35211 7683 12892 3100 4207 0 6536 93671
75 15321 20015 23252 40156 9457 106484 6020 112287 78940 18715 2690 26072 2081 9212 0 18490
77 0 0 33231 0 26997 0 0 1144 0 0 0 1291 0 1185 1035 0

We used an AWGN noise and a BPSK modulation.
All the simulations were made for signal/noise ratio
equal with 1 dB and for a number of 500 errors, at
least.
An iterations stop criterion was used for each decoder.
When the resulting errors number for a data block is
zero, the remaining iterations are not effectuated,
passing to the next block.
The tables contain bit error rate (BER108) obtained
for each polynomial pair indicated in octal, at the
beginning of each row (the denominator, b(D), from
relation 6), or column (the numerator, a(D), from
relation 6).

40
Table 3 (S- interleaver)
1 3 5 7 11 13 15 17 21 23 25 27 31 33 35 37
1 0 2782307 2943825 1786715 3066143 1207492 1339953 1123232 2853139 1250243 1811238 611821 1159192 575409 691787 657002
3 3713565 0 0 1513452 0 480592 586229 0 0 188714 158108 0 235426 0 0 515695
5 4532351 0 0 38628 0 54530 45274 0 0 2482 1401345 0 3080 0 0 11095
7 1040109 1022982 3436 0 0 2994 2956 305910 832 817 0 6151 850 0 4025 241462
11 3979820 0 0 0 0 36253 41956 0 0 41470 0 0 35644 0 0 477
13 3979820 24035 5447 4684 3999 0 143 242 167 763 185 1311 479 182 0 100
15 118335 36556 7559 4703 1166 267 0 185 266 738 145 0 707 401 1421 228
17 394618 0 0 162101 0 332 365 0 0 302 3075 0 307 0 0 193675
21 4155989 0 0 594 0 781 830 0 0 42699 46773 0 48925 0 0 10501
23 186127 5812 539 658 1333 752 144 333 24686 0 313 743 6509 1428 2622 794
25 1198150 2598 1069026 0 0 539 162 640 2893 4516 0 731 2887 0 908 758
27 7770 0 0 2034 0 1874 0 0 0 258 732 0 506 0 0 432
31 180019 6177 1006 568 2092 220 939 501 5303 5826 437 1721 0 945 439 1285
33 5135 0 0 0 0 1273 1560 0 0 302 0 0 252 0 0 1161
35 9702 0 0 2503 0 0 3214 0 0 460 487 0 439 0 0 324
37 102566 114577 322 76369 275 385 317 81253 543 397 4446 565 421 570 404 0
41 4212043 0 0 1934 0 1139 824 0 0 639 1355 0 903 0 0 0
43 186387 5332 2908 0 0 1305 0 2480 9810 550 0 0 1105 0 15923 6397
45 359028 10573 6894 2182 7325 413 737 2858 1180 2040 3133 2808 4826 15890 3769 14232
47 9265 0 0 13877 0 0 2067 0 0 992 9972 0 7480 0 0 9044
51 228619 14099 4668 2885 2697 1039 1665 9634 1735 8191 3568 3668 8333 13754 1056 13976
53 12047 0 0 1625 0 8100 2027 0 0 777 1633 0 0 0 0 2386
55 29230 0 0 0 0 4696 6396 0 0 3125 0 0 1994 0 0 6292
57 8908 4020 1333 2419 612 1843 19325 949 4723 739 16047 1396 71654 16720 4934 4592
61 227980 4904 2968 0 0 0 846 703 6697 2943 0 10200 291 0 0 3813
63 10090 0 0 263 0 417 384 0 0 307 167 0 190 0 0 387
65 8205 0 0 817 0 2784 2383 0 0 0 1867 0 2581 0 0 1138
67 3536 4241 1694 2789 12315 190 2025 287 28519 130035 26804 1066 648 1183 1525 2540
71 6549 0 0 2588 0 2093 0 0 0 2634 2889 0 1310 0 0 24800
73 3793 12398 1442 43816 15414 2714 672 788 33805 1276 113225 12719 19350 2428 714 3802
75 12376 2481 3854 2155 1844 10375 687 377 11232 19005 8164 5724 329 15031 934 2206
77 70523 0 0 0 0 2842 2491 0 0 5198 0 0 3437 0 0 48073

41 43 45 47 51 53 55 57 61 63 65 67 71 73 75 77
1 2993273 1207492 1223435 556141 1188807 549547 717488 262103 1134529 583417 583417 260213 494650 250240 333025 367943
3 0 131135 31141 0 28824 0 0 39378 117391 0 0 104040 0 102492 33592 0
5 0 3172 41831 0 40679 0 0 9642 3278 0 0 358 0 727 8419 0
7 276 0 127 9802 74 197 0 373 0 258 125 8786 8039 11305 363 0
11 0 0 41854 0 49803 0 0 508 0 0 0 485 0 519 569 0
13 1649 83 80 0 61 386 473 127 0 147 935 196 121 396 193 287
15 1299 0 286 94 73 1313 499 208 186 128 518 350 0 342 122 575
17 0 333 532 0 533 0 0 335 335 0 0 488 0 782 402 0
21 0 45040 1133 0 588 0 0 1887 35638 0 0 1074 0 2472 2761 0
23 11315 233 1940 136 25743 437 1122 391 3591 2305 0 2710 4540 330 15135 2778
25 1331 0 538 13856 204 1010 0 1071 0 15702 878 13606 19005 9179 708 0
27 0 0 202 0 752 0 0 982 16096 0 0 520 0 249 152923 0
31 13931 2866 13021 16983 425 0 600 69932 269 1091 217 286 167 13009 207 4345
33 0 0 7457 0 14444 0 0 17526 0 0 0 919 0 1416 1690 0
35 0 23136 258 0 336 0 0 7477 0 0 0 218 0 335 2291 0
37 0 731 5433 390 22609 3707 14937 376 331 384 4299 1593 534 923 798 58997
41 0 36471 47788 0 40759 0 0 30444 46317 0 0 18534 0 401227 6071 0
43 9174 0 2550 2094 1518 4200 0 4059 0 5968 20458 20065 0 444 15899 0
45 10600 14247 0 23325 26952 22341 4008 22238 5201 31271 3309 16184 8513 98094 24113 10032
47 0 2094 5453 0 16476 0 0 3385 0 0 0 4066 0 30534 26549 0
51 20778 1979 25027 13750 0 2564 4858 21800 8835 52297 15758 99546 3659 14286 16219 68429
53 0 28429 44149 0 3700 0 0 4375 206608 0 0 14589 0 15480 13301 0
55 0 0 16959 0 14568 0 0 5918 0 0 0 15839 0 923983 6148 0
57 22967 36629 28408 616 8852 1526 3734 0 6688 29861 39126 15876 27662 17594 32804 18244
61 13068 0 3898 0 2653 23247 0 32727 0 2570 3929 3518 686 64571 4821 0
63 0 19832 53766 0 30534 0 0 8299 3021 0 0 7494 0 13151 30265 0
65 0 65910 14435 0 26699 0 0 8136 10792 0 0 33623 0 39829 3974 0
67 61580 63321 111671 5047 66312 7881 24005 29670 21168 10619 13226 0 18438 1958 19668 16879
71 0 0 27579 0 12537 0 0 192134 7742 0 0 22430 0 20300 5443 0
73 27606 3694 31173 17325 108604 14879 4281 17458 122750 99527 2136 3568 14395 0 61241 25116
75 24360 10319 17105 15974 18931 12671 2340 119031 27355 24284 3416 22923 1731 27310 0 48372
77 0 0 27526 0 27316 0 0 64 0 0 0 1055 0 1720 582 0

The zeros in the tables contain mark that the
respective codes have common divisors (see Table 1)
and can not be used together. With little exceptions,
the results obtained with S-interleaver are superior to
the results corresponding the pseudo-random
interleaver. We also remark the superior results
obtained with both interleavers in the case of the use
at denominator, b(D), of the primitive polynomials,
comparing to non-primitive polynomials, at the same
constraint length. Despite of the fact that global
performances increase proportionally with K, it must
be remarked the performances obtained using, for the
Feed Back loop, the polynomials b1=7=111,
b2=13=1011 and b3=15=1101 as in combination with
the polynomials with the same degree (ex. 15/13 or
13/15) as in combination with the polynomials with
superior degree (ex. 51/7, 51/13 or 51/15).
Good performances are obtained using the Feed Back
with b4=23=10011, indicated in [2] (25/23, 33/23,
37/23). The last two combinations (33/23 and 37/23),
correspond to the situation when the pseudo-random
interleaver is superior to S-interleaver.

41
 In the following are presented some practical results.

BER
BER

x uncoded uncoded
MAP, P MAP
o MAP, S MLMAP
MLMAP, P LogMAP
+ MLMAP, S
LogMAP, P
* LogMAP, S SNR

Fig.7 Simulation of rate 1/3 HCCC, 15/13 code, S-interleaver

SNR

Fig.3 Simulation of rate 1/3 PCCC, 5/7 code, pseudo-random

interleaver and S-interleaver.

BER
BER

uncoded
MAP
MLMAP
uncoded LogMAP
5/7 code
15/13 code
37/13 code SNR
o 51/13 code
Fig.8 Simulation of rate 1/3 SCCC, 15/13 code, S-interleaver.

SNR
The results obtained with different decoding
algorithms (MAP, MaxLogMap and LogMAP), with
Fig.4 Simulation of rate 1/3 PCCC, P-interleaver, MaxLogMAP, two types of interleavers: pseudo-random and S-type
for 5/7, 15/13, 37/13 and 51/13 codes.
interleaver, with S=29, are comparted in Fig.3. In all
the cases we used the 5/7 code (the most performant
from all codes which have K=3) with data blocks size
equal with N=1784 bits. We can remark the
superiority of S-interleaver versus the pseudo-random
BER

interleaver. For example, in the case of MAP

algorithm, the S-interleaver brings a SNR
uncoded improvement of 0,1dB, at BER=10-5, versus the
5/7 code
15/13 code pseudo-random interleaver. Surprisingly, in the
o
37/13 code
51/13 code
algorithms performances hierarchy the first place is
took by the LogMAP algorithm, especially under
1dB. The performances of MAP and Log MAP
SNR algorithms are with approximation equals over 1dB.
As we expected, MaxLogMAP algorithm is with 0,2
Fig.5 Simulation of rate 1/3 PCCC, S-interleaver, MaxLogMAP,
for 5/7, 15/13, 37/13 and 51/13 codes.
dB inferior than the two enounced above.
Fig. 4 and Fig. 5 compare the most performant codes
from each constraint length: for K=3 is the 5/7 code,
for K=4 is the 15/13 code, for K=5 is the 37/13 code
and for K=6 is the 51/13 code, in the case of using
MaxLogMAP algorithm (the fastest in simulations)
for pseudo-random interleaver and S-interleaver.
BER

Thought we used 108 transmitted bits (or more)

sometimes it was insufficient to obtain smooth curves.
uncoded
Turbo code
Besides of this observation, we notice that for SNRs
Hibrid code inferior to 1 dB, the 5/7 code is the best. When SNR
Serial code
increases more than 1 dB, the codes with K>3 are
more performants. This fact leads to the idea that the
SNR codes hierarchy can changes at higher SNR. Notice
Fig.6 Simulation of rate 1/3 PCCC, 1/3 HCCC, 1/4 SCCC, S-
that turbo codes which have Feed Back loop realized
interleaver, MaxLogMAP algorithm, 15/13 code. on the basis of polynomial 13, for 1dB, are the best.
Probably, for SNR higher than 1dB, maybe the xx/13

42
codes can lose their supremacy. This verification [5] L.R. Bahl, J. Cocke, F. Jelinek, and J. Raviv, Optimal
Decoding of Linear Codes for Minimising Symbol Error Rate,
constitute the objective of a future study which we IEEE Transactions on Information Theory, Vol. 20, pp. 284-287,
propose us. March 1974.
The diagrams, for the last three figures, show the BER [6] L.Hanzo, T.H.Liew, B.L.Yeap, Turbo Coding, Turbo
performances obtained with different concatenation Equalisation and Space-Time Coding for Transmission over Fading
Channels, John Wiley & Sons Ltd, England, 2002
modes (parallel, hybrid and serial) and with different
[7] P. Robertson, E.Villebrun, P.Hoeher, A Comparison of
algorithms (MAP, MaxLogMAP and LogMAP). In all Optimal and Sub-Optimal MAP Decoding Algorithms Operating in
the cases we used the 15/13 code and the S- the Log Domain, Proceedings of the International Conference on
interleaver. Communications, Seattle, USA, pag. 1009-1013, iunie 1995
Obvious, the serial concatenation is less performant
than parallel and hybrid concatenations. Because of
multiplexing and restriction using of interleavers with
the same length, N=1784, in the case of SCCC code it
results a number of N=1784/2=892 information bits
per block, than N=1784 for PCCC and HCCC codes.
Moreover, because the SCCC transmission rate is of
1/4 versus 1/3 in the case of PCCC and HCCC codes,
for the equivalence of the ratio between the
transmitted energy in information bit (E0) and the
noise power spectral energy (N0/2), for the 1/4
transmission rate case, the channel noise power is
higher than the corresponding power for the 1/3
transmision rate case.
We remark the falling of the SCCC codes, for SNR of
2 dB, falling what its not find in the case of the
PCCC and HCCC codes.
Finally, we also notice the good behavior of the
LogMAP algorithm in the case of SCCC and HCCC,
fact which invite us to make an investigation more
detailed of this algorithm for the future.

III. CONCLUSIONS

In this paper we presented and compared the BER

performances obtained by the simulation of a
transmission system, which utilizes the forward error
correcting by codes concatenation and iterative
decoding (turbo coding). We used two interleaver
types, pseudo-random and S-interleaver with the
length equal with N=1784. The BPSK modulation and
the AWGN channel were employed. The MAP,
MaxLogMAP and Log MAP algorithms were used for
decoding.
There have been investigated all the systematic
convolutional codes having the constraint length K
less or equal to 6, under three different concatenated
forms: parallel PCCC (pure turbo code), serial SCCC
and hybrid HCCC, at the following rates: 1/3, 1/4 and
1/3, respectively, all unpunctured.

REFERENCES
[1] C. Berrou, A. Glavieux, P. Thitimajshima Near Shannon limit
error-correcting coding and decoding: Turbo-codes, Proc.ICC93,
Geneva, Switzerland, May 1993, pp. 1064-1070.
[2] Consultative Committee for Space Data Systems Telemetry
Channel Coding, CCSDS 101.0-B-6, Blue Book, October, 2002.,
http://www.ccsds.org/documents/101x0b6.pdf
[3] P. Ha, A Fast Algorithm for Generating Random Interleavers,
www.sarim.changwon.ac.kr
[4] J.Hagenauer, E. Offer, L. Papke, Iterative Decoding of Binary
Block and Convolutional Codes, IEEE Transactions on
Information Theory, Vol 42 No 2, March 1996 pp 429-445

43
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

A New Approach On Delay Coding: The Receiver

Sorin Popescu1
Abstract 1 - This paper represents the second part of an
issue from theCommunications 2004. Now, herewith
we are presenting the receiving part following the idea to
see the Delay Coded Signal DCS as a QAM signal,
namely a Vestigial SideBand VSB one. One uses the fact
that the DCS is a four phase signal, rotating 00/900
clockwise. The demodulated signal is ternary too and
alternating on the two branches, elsewhere identical.
The carriers have half the signaling rate frequency. One
expose three demodulation methods and two probability
density functions together with the simulation results.
Key words: Miller code, floating threshold, integral
criterion.
INTRODUCTION
1. General considerations
The electrical representation of the data must have
few qualities: restricted bandwidth, lack of energy at
very low frequencies, good noise protection. For
example, the Alternated Mark Inversion (AMI) code
is ternary and so noise sensitive, but allows an easy
synchronization. The differential biphasic (DBPh)
signal is binary and hence robust to noise. But its
bandwidth is double than necessary. A special role is
played by the Miller coded signal (MCS), also called
Delay Coded Signal (DCS, DC). Its bandwidth is
comparable with that of a baseband NRZ signal, but
does it does not possess very low frequency
components. Some drawbacks result from its intrinsic
phase modulation. Unfortunately, a catastrophic
carrier and bit timing ambiguity appears. But the
Miller coding can however be improved. We provide
in this paper some methods to perfect its efficiency.
The second method exploits Gray coded data and
obtain two decision branches, two unknown data
flows and the final DCD. The third method use an
integrating criterion. Both are using an original
decision proceeding device. Comparisons are
presented with the classic Miller detection and the
binary baseband model. The AWG noises are inserted
both in the front and the end of the channel.
In Section III we find the probability density function
PDF d(h) and d(f) for the difference h and sum f of
the absolute values of the inphase and quadature
signals. The angle between them is =acos(2/) for
the difference h and (-) for the sum f.
The timing problem is mooted too. We proved that it
is not necessary a bit timing of 2fs frequency, the only
possible till now, but an fs/2 one. It eliminates the T/2
ambiguity regarding the sampling points.
Section I Previous results
I.1 Main Baseband data signals
In Fig. 1 are illustrated baseband (b.b.) data. The DCS
can be constructed by simply dividing a DBPh code
in a T-flip flop. We will denote vs as the signaling
rate. The periodic signal P, with vs/2 frequency, is one
of two quadrature carriers. We can see the Miller code
as a 4-PSK signal.

2. The contents
The writing has three sections.
In Section I one remember the obtaining of a VSB-
QAM signal using three level quadrature carriers and
two data flows: the first, divided data (DD) is
obtained from the (DCD) by dividing it in a T type
flip-flop; the second is the modulo-2 sum of this two
flows. The two operations result in a flow of Gray-
coded dibits. They possess a structure of Hilbert pairs.
In Section II we present three demodulation methods.
Fig.1. Data representation: a) On-Off Keying (OOK); b) DCD; c)
The first method uses a QAM procedure, but the Bit timing, is the carrier for the DBP code. d) Differential biphase
decision rule is with floating threshold. code, with binary 00/1800 jumps; e) DC Signal: a transition at half-
interval represents a "1" in DCD data ak. f) P is the In phase
carrier, with 00/1800 phases. g) AMI (bipolar) code with "0 Volt"
for logic zero and alternating "1 V" for logic 1;
1
Facultatea de Electronic i Tc., UPB, Catedra de
Telecomunicaii, Bd. Iuliu Maniu 3-5 Bucureti e-mail:
sorin.popescu @comm.pub.ro
44
The OOK data will be denoted bk and called absolute
coded data (ACD). DCS is a quaternary DPSK signal.
Data bk must be differentially coded, leading to the
data sequence ak (DCD). The ACD produce for P a
phase change of -90 when bk=1 and no change when
bk=0. We will have 00/1800 when ak=0 and -900/-2700
when ak=1.
I.2. Plain Old Demodulation Method of DC data
Summing up the values from 2 sub-intervals, we
obtain Mimod(ified) data. If the phase k on that
interval is 0/180 (Fig.3), the sum is 0 (Fig.2) and will
be sampled on the second sub-interval, resulting ak=0.
Fig2 - Standard Miller detector .The detector works in two steps. It
decides Mimod: a. the DCD a k and b. the bit differences ACD b k .
If the phase is e.g. 90/270, it changes during T and the
sum will be ak=1 (Fig.4, Mimod signal). Then one
obtain bk=akak-1. The drawback is that it needs a
clock with double frequency, 2vs . By division it is
possible equally to obtain an inversed clock, =1800,
sampling at the points denoted by xx in Fig.4. The so
decided data are erroneous and the recovery is
difficult, after few hundred of erroneus bits.. This is
the phase ambiguity we mentioned previously
Fig.3 Signal diagram. DCS is the received signal. Mimod is to be
evaluated in the second sub-interval; x are no-matter data
I.3 DC modulation is a VSB -QAM with ternary
carrier (VSB-QAM-TC)
DCS can be built by modulating two quadrature
carriers by two data bit streams at the vs rate, which
are then summed up. Summing displaced binary
carriers result in a ternary one. So it is necessary these
carriers to be ternary to resulting in a binary signal. In
Fig.4 the constellation 4QAM is rotated by 450 w.r.t.
P and Q. In Fig.5 we have the ternary carriers and
K; their sum and difference are P and Q signals.
45
Fig.4 - Signal constellation with rotated carriers. The figures 0,1,3,
... denote the intervals kT
Fig.5 - QAM ternary carrier: carriers with phases 1800 and 2700
are obtained by inverting P and Q. Fig.10c. The resulting
schematic; N1, N2 are BCD numbers. G1, G2 are Gray coded bits
The synthesis of the bit streams G2,G1 is obtained
based on the constellation from fig.5 and the
modulation steps kT. The phases l900 with l=0,1,2,3
written in BCD provide the N2N1 bits. By writing
G2G1 as being the coordinates of the projections of
the signal phasors on the axes, one can obtain Gray
coded data (GC), as illustrated by the QAM scheme in
Fig.6, right (see10c). Projections of the phasors k on
the axes and K take the values on the constellation;
then G2 G1 also takes the necessary values. Fig.6
illustrates the PSD of the VSB signal.
Fig.6- The VSB spectrum; spectra of the Gray quadrature data
flows are visible as odd and even functions
Section II New demodulation methods
II.1 Coherent quadrature detection floating
threshold decision
a. Receiver model
The channel is simulating a cooper pair in a city
network. The noise is placed at the channel entry or
output. The carriers P, Q have fs/2 frequency and are
rectangular. The two low pass post detection filters
have a cut-off frequency of fs/2 too, at the Nyquist error rate is presented in the tables below together
limit. The idea of the method is becomes clear by with standard Miller, denoted Mold and a base band,
observing the three level demodulated signals, NRZ code, Mnyq. In the first hypo thesis the noise is
amazingly the same, in fact complementary. If one of placed at the channel issue, in the second is at the
them is big (1 or 1) the other is 0. The succesion is entry. All upper mentioned names receive a 2 at the
1,0.-1,0,1,-1.. end (e.g. Misnd2).
Table 1
ak differ.data 00 1 0 1 1 000 1 0 1
k sent phases 0 0 90 180 90 90 0 0 0 90 180 270
p=inphase sign. 1 1 0 -1 0 0 1 1 1 0 -1 0
q=quadr. signal 0 0 1 0 -1 -1 0 0 0 1 0 -1
|u| |v| = -a k,+dc 1 1 -1 1 -1 -1 1 1 1 -1 1 -1
|u|-|v| s c = -a k 11 0 1 0 0 111 0 1 0
In the example from Table 1 a k are arbitrary. The
modules difference is finally DCD a k. Let +/-A be the
amplitude of p and q. The decision may be effected by
simple rectifying one of them and comparing it with a
threshold of A/2 value. The noise can be at most A/2.

Fig. 8 Eye diagrams for Miller code demodulation on the branches;

the ternary signal at the upper image is without ISI while this below
has a big amount of ISI.

Fig.7 The transmission system: Miller encoder sends DCS to a Table 2 BER for noise at the issue
low pass filter as a channel with additive white Gaussian noise. Snr/ber 4 dB 6 dB 8 dB 9 dB
Two multipliers translate the line band in the base band, aided by
two l.p.f. abs(p,q) represent modules, is an adder/substractor. Msnd 5e-3 .9e-4 1.6e-4 8.6e-6
The name of this scheme is Misnd. The signals are thresholds each
other and are noisy. is used as an adder only for computations Mold 2.8e-1 1.3e-1 3.5e-2 1.3e-2
purposes.
In our solution (fig.7) a threshold circuit decide Mnyq 1.8e-1 8.6e-2 2.7e-2 1.1e-2
between a voltage A and an electric 0 on the threshold snr/ber 10 dB 11 dB 12 dB 13 dB
entry, usually set at 0 voltages. But herein this entry is
hot because always there exists the noise. In fact it Mold 5.2e-3 1.4e-3 2.6e-4 1.2 e-5
happens on both entries. The sum will be of double Mnyq 4.9 e-3 1.5 e-3 4.2 e -4 7.5e-5
power (3dB) wile the permitted noise can be at most
-4
A. These get the advantage of 6 dB for the signal. The Mnyq has 12dB (11.4) for BER=10 . Figure 4.2 is the
noise increases with only 3 dB. It can be expected a effect of the descrambler. Miller old is equivalent
large advantage, of 3 dB. with a 4 PSK. Remember the penalty of double BER
Our simulations implied a plain old Miller receiver, for DPSK. So, our method is 5 dB superior to the
denoted Milvec(no) for old or new structure concern classic Miller decoder. See for all that the detection
ing the placement of the noise source in the transmi has included two LPFs.
ssion chain. In fig.8 it is put at the receiver entry, at
the end of the channel. The eye diagrams are plotted Table 3 BER for noise at the entry
on the two branches in fig.8. The three levels are snr/ber 4 dB 6 dB 8 dB 9 dB
visible and the sampling points are too. The Msnd e
2.2 -3 e
5.8 -5 e
6.4 -6 -
transitions in the above part figure are producing only
e e
between the neighbouring levels. The intersymbol Mold 2.5e-2 3.4 -3 3.3 -4 2.3e-5
interference ISI is not present. In the figure below Mnyq 4e-3 6.6e-4 8.1e-6 -
there exist transitions between extreme levels. But the
ISI is not them imputable. At a careful insight one
observe on the two figures two adjacent and not When the noise is inserted at the entry (table3) it
identical eyes in every bit interval. They are displaced undergoes a loss of 9.5DB. The hierarchy remains
each other and the second is sampled in a wrong but the differences are smaller. However, an effective
moment. So the two branches are not equally noise consistent advantage of 2.5dB is real. 1dB is due to
protected. This method will be referred as Msnd. The the post detection filters. A 1.5 value results from the

46
peculiar probability density function of the module subtracting the signal is doubled but the noise is only
difference (see later). increased by a sqrt2 factor.

II.2 Coherent detection double flow decision

Remember that the DCS can be view as a VSB-

QAM signal, with two Gray coded data flows. In
table4 one observes the detected u, v signals and their
sum and difference. Dc is double current (polar) and
sc is simple current (biased). The decisions dk, + and dk, -
of u+/-v are the data flows obtained from DCD with a T
flip-flop on the positive and negative edge respectively!
Their sum is desired ak data. With the price of two decision
devices and a sophisticated timing procedure one obtain a Figure 10b The quqdrature signal u-v. Eye pattern with two
significant gain in SNR and BER sampling locations: single point for both branches in a compromise
style (up) and two points, left and right in a perfect solution

Table 5 Double detection, decision and sampling method;

the noise is applied at the entry of the channel

Snr/ber 5 dB 6 dB 7 dB 8dB
e e e
Mddds 8 -5 3.6 -5 3.3 -6 <1e-6
Mold 7e-3 3.4e-3 7e-4 3.3e-4

II.3 Coherent detection integratig criterion

Nyquists third criterion (NTC) statues the condition

Figure9 Miller detector with double flow data decision; the sum and for the lack of interference as:
difference between branches are two level signals; the thresholds
kT + T / 2 1, for k = 0 (2)
are set to 0 and are noiseless.
kT T / 2 h ( t ) dt = 0 , k = 0 , for k 0
Table.4 Relationship between data, phases and divided data; DCD
ak result as a modulo 2 sum of two data flows. The solution of replace a LPF with an integrator
circuit is attractive by reason of the smaller number of
bk absolute data 00 1 1 1 0 100 1 1 1 operations. A certain filtering effect also does exist.
ak DCD 00 1 0 1 1 000 1 0 1 The DCS is of the form: s=pcosct+qsin ct,
k sent phases 0 0 90 180 90 90 0 0 0 90 180 270 p=cos k; q=sin k. The proof of the NTC validity is
immediate. The scheme in fig.11 uses the same
u inphase sign. 1 1 0 -1 0 0 111 0 -1 0 method of as in fig.7. The carriers P, Q realize the
zero-turning of the integrals at end of the interval T.
v quadr. signal 00 1 0 -1 -1 0 0 0 1 0 -1
(u-v) dc 1 1 -1 -1 1 1 1 1 1 -1 -1 1
(u-v)sc=dk,+ 11 0 0 1 1 111 0 0 1
(u+v)sc=dk,- 00 0 1 1 1 000 0 1 1

dk,++dk,-=ak 00 1 0 1 1 000 1 0 1

Figure 11 DCS detector using an integrating proceeding (Mint)

Table 6 BER vs. SNR for the integrating criterion

Snr/ber 4 dB 6 dB 7dB 8 dB 9 dB
Mint 4.4e-3 2.3e-4 5.3e-5 3.3e-6 <1e-6
Mold 2.5e-2 3.4e-3 7.0e-4 3.3e-4 2.3e-5

Mint is 1.5 dB better than plain old Miller, due to the

Figure 10 a The in phase signal u+v
We have 3dB gain. The reason is the binary eye
diagram and the possibility to sample both in their
maximum opening point. By summing and
filtering quality of a LPF with 1/s transfer function.
II.4 Carrier phase and bit timing recovery
Is a matter of fact that the carrier frequency of P, Q
is half the signaling rate. Let a phase appear in the
line signal s=cos(ct+k+). p and q become:

47
p= cos(k+ ), q=-sin(k+ ); f 2
x2
2 f / 2
d( f ) =
2 2
4
e e dx ; (8)
f / 2
2
f
2
(9)
d( f ) = erf ( f / 2 );
2
4
e


h = u v,v > 0,u > h;d(h) = g(v)g(h + v)dv; (10)
0
h2 (v + h / 2) 2
2
d (h) =
2 2
4
e e dv (11)
2 0

h2
2
d (h) = erfc ( h / 2 )
2
4 (12)
e

Figure 12 PLL for data aided carrier recovery 2
e y dy = F ( ) F ( x ); (13)
2
erfc ( x ) =
x

The product is: pq=cos2k sin2=aksin2; so, it is

necessary to use a decizion directed algorythm. Data
ak are taken from the trigger that delivers sgn(u-v).
The control signal in the loop is superposed to an
iniialvalue j0 with the aim to avoid negative delays.

SECTION III Theoretical Results

III.1. Distributions
In order to evaluate the error rate for our detection
method we need to find the probability density
function for the sum and difference of absolute values
of the noise on the two channels, f=u+v and h=u-v.
Let denote by d(f) and d(h). Their values are:
2
f
2
d( f ) = e 4 erf ( f / 2 ); for f (3)
2

2
h2
1 Figure 14 Statistic distributions for the nonlinear processed noises
d (h ) = e 4 erfc ( h / 2 ); for h 0
2

2
(4) The error probability is to be approximated by:
2 x2 / 2 1
pe,new = d(h)dh
2
e dx =. 2 e y / 2 (14)
Both u and v has normal, Gaussian distribution: y y x y
2
u2
2
e 2 2
for u 0 (5) erfc ' ( x ) = F ' ( x ) = e x (15)
2

g ( u ) = d ( u ) = 2

0 for u < 0
We retain the first term from the Laurent series:
2
ex a1 a2 a3
erfc ( x ) = ( + ...) (16)
x x3 x5
where ak are obtained from derivative of erfc and
identifying:
2
ex 1 13 135
erfc ( x ) = (1 + + ....) (17)
x 2x2 4x4 8x 6

So d(h) decreases more rapidly than Gausss curve

with a 1/x term.
2
ex 2 2 2 2 (18)
erfc(x) ; d(h) eh / 2; d(x) / g(x) ;
x x x
Figure13 Half-Gaussian distributions are convoluted from 0 to f.
An approximation for the standard error rate is:

f = u + v > 0, u f ; d( f ) = g(u)g( f u)du; (6)
p e , gauss = g ( x ) dx 1 / 2.erfc ( y / 2 )
1
ey
2
/2
; (19)
y
y 2
The ratio between classical and our error rate is:
u2 ( f u)2
4 f
p e , gauss P signal
22 0
d( f ) = e 2 2
e 2 2
du; (7) = = y = (20)
p e , new 2 2 P noise

48
 1 c) A graphic image for sums and difference of
lg = lg p e , gauss lg p e , new = 0.196 + n signal / noise ; (21)
10 powers of the rectified noises as a.c. phasors; there
virtual angle has peculiar proprieties.
III.2 How to explain some numbers
Bibliography
Now, lets expose a peculiar result. If we denote by:
[1] N.D. Alexandru, Dae Young Kim: Spectral Shaping via
2 = f 2 = f 2d( f )df = 3.27325.175d B; =1.809 (22) Coding , Ed. Cermi, Iasi 2003
0 [2] N.D. Alexandru, G. Morgenstern: Digital Line Codes and
Spectral Shaping Ed. Matrix, Buc. 1998
s = h = h2d(h)dh= 0.7271.39dB;s =0.8525
2 2
; [3] W.R.Bennett, J.Davey: Data Transmission Ed. Mc-Graw Hill

(23) Book Co. 1965

[3] Gilbert Held,: Understanding Data Com munications, Sams
2
= g2 = x 2 g ( x ) dx = 1 0 dB ; = 1; (24) Publishing 1996

[4] M. Stein, Les modems pour transmissions de donnees, Ed.
the ms values of u+v, u-v and p,q respectively, then Masson, Paris, 1987
(see fig.15) the effective values for this noises are [5] S. Popescu, Transmisia Datelor, Ed. Matrix, Buc. 2003
summing as sinusoidal signals and the angle between [6] Popescu, S: "Phase-Precession Modulation, a New Approach for
SSB-AM" - IV International Conference on Reliability of
them is obtained from: cos =2/=0.6366. Semiconductor Devices and Sistems RSDS'96 Chiinu, June 5-7,
2 = 2 + 2 + 2 2 cos = 2 (1 + cos ) = 3 .2732 ; (25) 1996, pp. 182-187
[7] S. Popescu, Modulaia cu precesie de faz, o abordare nou a
s 2 = 2 (1 cos ) = 0 .7268 ; modulaiei de amplitudine cu BLU - Telecomunicaii nr. 3/1997,
5 1 (26) pp. 9-16
cos = 2 / = 0 .6366 !; = 50 0 40 ' a cos ; [8] S. Popescu: A non-Interference Criterion for FSK Data
2 Signals Proceedings of the Symposium on Electronics and
The value for it is: =50040, very close near to: Telecommunications ETc. 2000, Timioara, November 23-24,
5 1 2000, Vol. I pp. 227-231
= a cos = 51 0 50 ' (27) [8] Popescu, S.: Delay Coding a new perspective Proceedings
2 of the IEEE International Conference Communications 2004
This is right the angle with the basement for the faces June 3-5, 2004, vol.2, pp 115-122, Bucharest, Academia Militar
of Keopss great pyramid. And is in visible relation Tehnic, Romnia
with a root of Fibonacis series characteristic
equation: x2-x-1=0.

Figure 15 A new gold triangle: the vectorial sum and difference

of rectified noises is realized whet their angle is a golden one.

The powers 2, s2 and 2 were measured as temporal

quadratic means and the logarithmic values confirm
the calculus.
n calcul = 6.565 dB; n mesur = 6.8 dB (28)

II. 3. More about the DCS reception

1. It is possible to demodulate DCS as a QAM even

though the result is ternary and the SNR is
deteriorating.
2. Here exists a large amount of correlation between
branches.
3. Other methods can be used to improve the noise
protection
Conclusions
a) Three new methods were proposed, with good
and very good results: decision with floating
threshold, detection following an integrating criterion
and decoding by two data flows decision and double
sampling
b) Formulas for two new found distributions in
concordance with good experimental results.

49
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Selective encryption of image with IDEA algorithm
Gabriel G. Fericean1, Monica Borda2
Abstract In this paper, a study of selective image
encryption using IDEA algorithm is presented.
Experimental results show a good image security
using the proposed selective encryption.
Keywords: IDEA, image encryption, selective
encryption, cryptography, encryption.
I. INTRODUCTION
The strongest solutions for security are offered by
computational cryptography. Cryptography is used for
insuring the communication confidentiality in military
and diplomatic fields for a long time. During last
years, cryptography has known a spectacular progress,
many services and devices of security, which are used
in Internet is a proof of this fact. Image encryption is
among the last applications of cryptography. It looks
to have in the near future many applications in
Internet, taking into account that fingerprints and
retina images will replace the numeric passwords.
II. IDEA ALGORITHM
Xuejia Lai and James Massey of the Swiss
Federal Institute of Technology developed the
International Data Encryption Algorithm (IDEA) in
1999 year. The main application for IDEA is PGP
(Pretty Good Privacy). This program is the most
secure and fast encryption system nowadays.
Cryptographic strength of IDEA is given by:
Block length
Key length is long enough to prevent
exhaustive key searches
Confusion the cipher text should depend on
the plain text and key in a complicated way.
Diffusion each plaintext bit should
influence every chipertext bit, and each key
bit should influence every chipertext bit.
1
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Str. Daicoviciu Nr. 2, Cluj-Napoca, e-mail gabifericean@yahoo.com
2
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Str. Daicoviciu Nr. 2, Cluj-Napoca, e-mail Monica.Borda@com.utcluj.ro
50
Confusion is achieved by mixing three different
operations. Each operation is executed on two 16-bit
inputs. These operations are:
Exclusiveor (XOR)
Addition by integer modulo 216, inputs and
output are unsigned 16 bit integer
Multiplication of integers modulo 216+1,
inputs and output is unsigned 16-bit integer.
In this case the blocks of all zeros is treated
as representing 216.
Using this three operations we provide a complex
transformation of the input, making cryptanalysis
much more difficult than DES algorithm, which uses
just XOR operation.
In IDEA, diffusion is provided by the basic
building block of the algorithm, known as
multiplication/addition (MA) structure (figure 2.).
This structure has as inputs two 16-bit values derived
from the plaintext and two 16-bit subkeys derived
from the primary key and produces 16-bit outputs.
This particular structure is repeated eight times in the
algorithm (figure 1.).
IDEA uses a primary key of 128 bits long. This
primary key produces 52 subkeys with 16-bit long.
Encryption and decryption makes on 64-bit blocks.
Subkeys Generation
First 8 subkeys are taken directly from the
primary key through segmentation in 16-bit segments.
Then a circular left shift of 25 bit position is applied
to the primary key and the next eight subkeys are
extracted. This procedure is repeated until all 52
subkeys are generated.
Encryption
Encryption schema for IDEA has two inputs
plaintext (64b) and primary key (128b). IDEA is
making up for 8 rounds and one output
transformation. These algorithms divide plaintext in 4
blocks of 16 bits. Output transformation achieves 4
outputs of 16 bits, which is concatenated and makes
chipertext of 64 bits. Each round uses 6 subkeys of
16-bits, but output transformation uses 4 subkeys
(figure 1.)

Figure 1. Idea algorithm
Figure 2.
Decryption
In this case the inputs is chipertext (64b) and
different selection of subkeys. Decryption subkeys
U1,...,U52 are derived from encryption subkeys as
follows:
First 4 subkeys of decryption round i are
derived from the first 4 subkeys of
encryption round 10-i, where the output
transformation is counted as round 9. First
and fourth subkey are equal to the
multiplicative inverse modulo 216+1 of the
corresponding first and fourth encryption
subkeys. For round 2 through 8, the second
and third decryption subkeys are equal to the
additive inverse modulo 216 corresponding
third and second encryption subkeys. For
round 1 and 9, the second and third
decryption subkeys are equal to the additive
51
inverse modulo 216 of corresponding second
and third encryption subkeys.
For the first 8 rounds, the last two subkeys of
decryption round i are equal to last two
subkeys of encryption round 9-i. [5],[6]
III. SELECTIVE ENCRYPTION
Selective encryption or partial encryption
represents a good idea if we wish to reduce the
volume of calculation necessary for image processing.
The security of selective encryption application is
always lower than using full encryption. The only
reason to use selective encryption is to reduce time
and computational demand.
This method can be applied to binary images or
any other format as: JPEG, BMP, GIF, PNG. If
applied to a binary image, the method consists in
mixing image data and a message (key) that has the
same size as the image. A XOR function is sufficient
when the message is used only once. A generalized
for gray level images, is possible: in this case the
image is divided in bitplanes, which are encrypted
separately and reconstruct a gray level image. The
highest bitplanes exhibit some similarities with the
image, but the least significant bitplanes look random,
adding noise to the image.
Selective encryption can be applied with any
cryptographic algorithm (IDEA, AES, DES). [14],
[15], [16].
IV. EVALUATION OF SELECTIVE
ENCRYPTION
In this chapter we present a study of behavior for
image encryption and implementation of selective
encryption for color planes (RGB), for luminance and
chrominance planes and for bitplanes. The IDEA
algorithm is used for encryption because it offers high
security and we have experience in VHDL
implementation for IDEA algorithm.
Luminance and chrominance planes are obtain
from color planes using the following formulae:
Y = 0.299 * R + 0.587 * G + 0.114 * B
U = 0.493 * ( B Y ) (1)
V = 0.877 * ( R Y )
The color plains are obtain from luminance and
chrominance with the relations:
R = Y + 1.14 * V
G = Y 0.395 * U 0.581 * V (2)
B = Y + 2.032 * U
Difficulties were met in primary key generation
because Matlab uses only 52 bits to represent a
number. For this reason the number of all zero
subkeys is very high, so we looked for another
solution: we generated eight subkeys with values
between 1 and 216, than we transformed in binary and
concatenated them. After these operations we
obtained primary key with 128 bits dimension.
Next we present the results obtained for color
plane encryption on different images.
For a correct function, after the program
compilation, we must import an image and generate a Figure 6 Encryption R plane
primary key. These operations achieve on File menu
(figure 3).

Figure 7 Encryption G plane

Figure 3. Image of application

Now we can achieve the image encryption.

The color planes (RGB) encryption is as follows: for
encrypting all planes the Encryption RGB menu is
accessed, where the Encryption planes RGB option is Figure 8 Encryption B plane
chosen. For encrypting only one of this color planes
one of next options: Encryption plane R, Encryption For color plane encryption, the encrypted image
plane G, Encryption plane B are available. The next is secured only if all color planes are encrypted.
five images present results for color plane encryption.

Figure 9 Image view without

R plane
Figure 4 Original image
This fact is valid for any combination at the color
planes.
Luminance and chrominance planes encryption is
the same with color planes encryption. For a secure
encryption we must encrypt all planes. A presentation
of the luminance and chrominance planes encryption
for the same image is done in figure 10.

Figure 5 Encryption RGB

planes

Figure10 Encryption YUV

planes

52
 The proposed method has as disadvantage the
necessary time for all pixels encryption. Figures 11 to
18 present results obtained for bitplanes encryption.
Each of these figures is followed by the representation
of the initial image without bitplanes encryption. The
most significant bit (msb) plane is considered the
plane 1 and the last significant bit (lsb) plane is
considered the plane 8.
Figure 16 Original image without first,
second and third planes

Because the security is lower for encryption of

lsb planes (8 2) we decided to represent only
encrypted images (figures 19 to 21).

Figure 11 Encryption first plane

Figure 19 Encryption eighth

plane
Figure 12 Original image without first
plane

Figure 20 Encryption second, third, fourth,

Figure 13 Encryption first and second fifth, sixth, seventh and eighth planes
planes

Figure 21 Original image without lsb

Figure 14 Original image without first plane
and second planes

Figure 22 a.Original image

Figure 15 Encryption first, second and
third planes

53

Figure 22 b. Encryption first plane
(msb)
Figura 22 c. Original image without
first plane
Through selective encryption we can offer more
security planes, the minimum for an acceptable
security being represented by the first plane
encryption (msb plane). This affirmation is sustained
by the experiment presented in figure 22.
[4] Bajenescu T., Borda M.: Securitatea in informatica si
telecomunicatii. Editura Dacia, Cluj-Napoca, 2001
[5] Schneier B.: Applied Cryptography, Second Edition, Editura
John Wiley& Sons, Inc.1996
[6] Stallings W.: Cryptography and Network Security: Principles
and Practice Second Edition, Editura Prentice Hall, New Jersey,
1999
[7] Borko Furth, Darko Kirovski, Multimedia Security
Handbook, February 17, 2004
[8] Vlaicu A., Curs multimedia, 2002 (manuscris)
[9] Gibson D. Jerry, Berger T., Lookabaugh T., Lindbergh D.,
Baker R., Digital compresion for multimedia: Principles and
Standards, Editura morgan Kaufmann, 1998
[10] Biryukov A., Nakahara J., Preneel B., Vandewalle, J, New
Weak Key Classes of IDEA, Advances in Cryptology, Eurocrypt
1998
[11] Daemen J., Govaerts R., Vandewalle J.: Weak Keys for
IDEA, Advances in Cryptology, Crypto93, LNCS 773, D.R.
Stinson, Ed., Springer-Verlag, 1994
[12] Marc Van Droogenbroeck and Raphael Benedett.
Techniques for a selective encryption of uncompressed and
compressed images. In Proc. Advanced Concepts for Intelligent
Vision Systems (ACIVS2002), pages 9097, 2002.
[13] Martina Podesser, Hans-Peter Schmidt, and Andreas Uhl.
Selective bitplane encryption for secure transmission of image data
in mobile environments. In Proc. 5th IEEE Nordic Signal
Processing Symposium (NORSIG2002), 2002.
[14] Roland Norcen, Martina Podesser, Andreas Pommer, Hans-
Peter Schmidt, and Andreas Uhl. Confidential storage and
transmission of medical image data. Computers in Biology and
Medicine, 33(3):277292, 2003.

V. CONCLUSIONS

Our paper had as starting point the need of

transmitting safely images on Internet, images such as
fingerprints, medical images, etc.
Such images must guarantee a high security rank
as well as speed and all these can be obtained through
cryptographic algorithm and hard implementation.
The whole paper is a study of encryption tehnics
for images with experimental results.
The concluding remarks are:
For image encryption the first step is the
knowledge of needed security rank.
If a fast connection is needed and less
security, we can choose for selective encryption,
encrypt only first plane (msb plane).
To enhance the security degree, we can
encrypt first and second bitplanes. In this mode
we can grow image security until image security
is relied only on algorithm security. Maximum
security can be provided if all bitplanes are
encrypted.
For color planes encryption, image security
can be obtained only for all color planes
encryption.

REFERENCES
[1] http://www.byte.ro/byte95-03/vic.html [2004]
[2] Vasiu Ioana, Criminalitatea informatic, Editura Nemira,
2001
[3] Patriciu Victor, Monica Pietroanu-Ene, Ion Bica, Costel
Cristea, Securitatea informatic n UNIX i INTERNET, Editura
Tehnic, 1998

54
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Analysis of Simple Inversable Functions Defined On

Galois Fields for Cryptography Use
Luminita Scripcariu1, Petrut Duma2
Abstract The private character of the information
transmitted on a communication channel or network
could be ensured using some cryptography codes.
Different encryption techniques performances depend
on the processing time and the complexity of the
encoding algorithm. We propose a new and
advantageous method for symbol permutation, using
inversable algebraic function defined on Galois Fields
(GF), which minimizes the necessary memory capacity
of the encoding algorithm, ensures a great diversity of
data and is harder to attack.
Keywords: encryption, permutation, Galois Field
I. INTRODUCTION
Secure communications imply encryption methods
use. Different encryption algorithms (DES Data
Encryption Standard, 3DES, MD Message Digest
etc) are known but new cryptography principles are
searched for higher diversity and efficiency of the
communication system [1].
The coded sequence is deduced as:
c = E k (a) (1)
We denote:
a - the data sequence;
c - the coded sequence;
E k - the encryption function.
Symmetric encryption systems (Fig.1) use secret keys
and could be implemented software or hardware as
media-access cards (MAC). Data are extracted from
the received sequence with the same encryption
function:
a = Ek (c) = Ek ( Ek (a)) (2)
Public keys, defined as 'one-way' functions, are
applied for asymmetric encryption (Fig.2) and the
decoder applies the inversed encryption function:
a = E k1 (c) = E k1 ( E k (a)) (3)
The encryption keys are specified in large tables,
which request high-capacity of memory.
The symmetric encryption code could use different
methods based on substitution and/or permutation of
the symbols. The combined methods have better
performances. The coding-rate is about 1:1, so the
transmission rate is not affected.
The permutation order is hardly to inverse and to store
for long sequences. For example, DES, used on
Inetrnet by the SSH (Secure Shell) protocol, is a
powerful encryption algorithm (ANSI X3.92), with a
64 bits secret key, which permutes an input word of
64 symbols.
Symbol or character permutation is a frequently used
cryptography method, hardly to detect if the sequence
length is high [2]. For example, a 20-symbols
sequence could be permuted in 2.4 *1018 different
ways.
Large tables are used for permutation of long
sequences and high memory capacity is required. The
access time to the memory and the processing time of
the encryption algorithm are increased when a longer
permutation length is used.
Therefore an algebraic method for permutation could
reduce the encryption time and the coding complexity.

Fig.1 Secret-Key Cryptosystem

1
Facultatea de Electronic i Telecomunicaii Iasi, Romania, Departamentul
Telecomunicaii Bd. Copou Nr. 11, Iasi, E-mail lscripca@etc.tuiasi.ro
2
***, E-mail pduma@etc.tuiasi.ro

55
The performances of the encryption algorithm will be
improved using algebraic functions to generate the
permutation order.
Some coefficients combinations do not generate all
the symbols of the GF and therefore the decoding
process becomes catastrophic. These sequences could
not be used as encryption keys because there is no
inverse function in these cases. It is necessary to find
out which combination of coefficients ensures the
permutation of the GF symbols. The original sequence

Fig.2 Public-Key Cryptosystem

II. GALOIS FIELDS is considered the reference, so the identity function

The cryptographic methods could be easier
implemented using the Galois Field (GF) theory [3].
A GF(2m) has 2m elements where m is the length of
the binary sequence associated with a symbol of the
field.
Internal addition and multiplication operations are
defined on the GF.
Let us denote by a an element of the GF. It could be
written in an equivalent mode as a binary sequence:
a = a m 1 a m 2 ...a 0 (4)
or as a polynomial:
a a( x) = a 0 + a1 x + ... + a m 1 x m 1 (5)
could not be considered an encryption transform.
Other sequences of coefficients make the same
permutation and the chances of the cryptanalyzer to
find the key are increased. These combinations are
called weak keys. Only those keys which uniquely
generate a permutation of symbols could be used as
strong keys. These keys are classified as symmetric
or asymmetric keys.
On GFs, a large number of simple and inversable
polynomial functions could be used for encryption:
k2
E k (a) = c = k 0 + k1 a , k1 0, k 2 0, (11)
k 2 2 m 1, (k 0 , k1 , k 2 ) (0, 1, 1)

Addition of two symbols is made modulo-2 bit-by-bit. These functions have only three coefficients which
The null element (0) does not change the result of an compose the transmission key of an encryption
addition. system.
The opposite element is the element itself. The inversed functions, defined on the same GF, are:
Multiplication of two elements is defined based on the
polynomials product of the two elements and an
irreducible m-degree polynomial p(x):
[
E k 1 ( c ) = k 1 1 ( c + k 0 ) ]
q

(12)
c = a b c( x) = a( x)b( x) mod[ p ( x)] (6)
The unit element (1) does not change the result of a The integer exponent q is the inverse key component
multiplication. which verifies that:
If the product of two elements is equal to one, than k2
(a ) q = a . (13)
they are named inversed elements:
and
a b = 1 a 1 = b, b 1 = a. (7)
(k 2 q ) mod(2 m 1) = 1 (14)
The substraction and the division are defined based on
the opposite and the inversed elements.
The existence of q for any value of k2 is quaranteed
a b = a +b (8)
1
only if m is a prime number and all the GFs elements
a / b = a b (9) have the maximum order equal to 2m-1. In fact, m and
Polynomial functions defined on GF(2m) could be 2m-1 should be simultaneously prime numbers to
used as permutation transform for different encryption ensure the maximum number of simple encryption
algorithms: functions defined on a GF(2m). We deduce some
M 1 optimum values of m: 3, 5, 7, 13, 17, 19, and 31.
k a , M = 2
i
E k (a) = i
m
1 For example, on GF(8) these couples (k2, q) are:
i =0 (10)
(2 4), (3 5), (4 -2), (5 3) and (6 6).
The sequence of coefficients from the
GF, k = [k 0 k1 ... k m ] , represents the encryption key. Other Galois fields, such as GF(16), GF(64),
GF(256), do not allow any combination of

56
coefficients for simple polynomial inversable 2
c = E1 (a) = 1 + 5a
functions because the order of some elements is less
then 2m-1. But there are some values of the
coefficients which produce inversable functions and After value encryption, it results:
these GFs could be used with few constraints.
C1 = [7 3 5 2 4 1 0 3]
III. ENCRYPTION ALGORITHMS
For position encryption let us use other functions
The polynomial inversable functions defined on GFs defined on GF(4):
could be used for symbol permutation.
2 2
For optimum m, the number of simple polynomial E 2 (a) = 3a , E 3 (a) = 2a + 2
functions defined on GF(2m) is equal to the number of
the generated permutations (except the identity one)
First function permutes the reference sequence of 4
and it is given by:
symbols (0 1 2 3) into (0 3 2 1).
The second function permutes the reference sequence
M = 2 m (2 m 1) ( 2 m 2) 1 (15) into (2 0 3 1).
Each block of four symbols will be permuted
The coefficients of the 3-coefficients functions could according to a different function. The final symbol
be randomly generated to change the permutation sequence is:
order in a fast way. C2 = [7 2 5 3 0 4 3 1].
The function could be applied directly on the data The transmitted bits stream is:
symbols to permute the bits of a symbol or indirectly,
on the position index of each data symbol from a C = [1 1 1 0 1 0 1 0 1 0 1 1 0 0 0 1 0 0 0 1 1 0 0 1].
block of 2m elements, resulting a permutation of
symbols. In this case, if the encryption functions coefficients
We call the direct method the Value Encryption are not changed, the CVPEA permutes 24 bits.
Algorithm (VEA). In a similar way, longer permutation length could be
The indirect method is called the Position obtained.
Encryption Algorithm (PEA). If VEA uses a GF with 2v elements and the PEA uses
Both algorithms could be applied simultaneously on another GF with 2p symbols, then the permutation
the data with different encryption functions, defined length of the CVPEA is:
on different GFs. The last case represents the
Combined Value-Position Encryption Algorithm L = v 2 p (bits ) (16)
(CVPEA) which is robust against the differential
attacks.
For longer permutation length, the GF dimension of
The direct method could be applied in a fast way with
the PEA has to be increased first because the
different encryption functions for short sequences of
dimension of the GF used for VEA affects harder the
symbols.
encryption algorithm complexity then those used for
VEA has no constraints but PEA is constrained to be
PEA.
applied on a sequence of exactly 2m elements.
For a higher diversity of the coded sequence, both
The coefficients of the encryption key could be fast
VEA and PEA must use larger GFs.
and randomly generated to ensure great value
The transmission key contains the GFs dimensions
diversity.
and the coefficients of the encryption functions or the
For high GFs dimensions the efficiency of the
parameters of the coefficients generator.
algorithms is increased but the processing time of the
Fast and random generation of the key components
algorithm does not become very high because only
ensures a large diversity of the encrypted sequence.
arithmetical operations defined on GFs are used.
A pseudorandom sequence generator could be used by
Example:
the VEA for faster permutation of the composing bits
Let us consider the binary data sequence:
of each symbol. In this case, a high dimension GF
should be used.
A = [1 0 1 0 1 0 1 1 1 1 0 0 0 0 1 0 0 0 1 1 0 0 1 0]
IV. NUMERICAL RESULTS
If the GF(8) is chosen for VEA, the binary data
stream is transformed into a sequence of symbols
Different GFs are analyzed to establish the number of
expressed on three bits:
permutations obtained with the simple polynomial
B = [5 2 7 4 1 0 6 2]
functions.
Small dimensions of GFs are sufficient if
A simple inversable function is applied. for the value
combinations of GFs are used to generate high-length
encryption of data:
permutations with CVPE which is very efficient, very

57
fast and hard to attack with an acceptable
computational complexity. The encryption key type is specified:
For example, if both VEA and PEA use functions S symmetric;
defined on GF(16) then the minimum permutation A asymmetric.
length of CVPEA is about 64 bits, but if we change There are 9 symmetric keys different from the identity
randomly the coefficients of the encryption functions, and 14 asymmetric keys.
then longer binary sequences are permuted. For the asymmetric encryption system, it is easy to
For higher GFs dimensions, longer sequences deduce the inverse polynomial functions coefficients
permutation is made but the computational from Table 1. There are 7 couples of direct and
complexity and the processing time are both inverse keys:
increased. (0,2,3,1) - (0,3,1,2);
A. GF(4) (1,2,0,3) - (2,0,1,3);
This is a small algebraic field with 2-bits elements so (1,2,3,0) - (3,0,1,2);
it is not efficient for value encryption but it could be (1,3,0,2) - (2,0,3,1);
used by the PEA. Position permutation is made on 4- (1,3,2,0) - (3,0,2,1);
symbols vectors. (2,1,3,0) - (3,1,0,2);
There are 4!-1 = 23 possible permutations without the (2,3,1,0) - (3,2,0,1).
identity one (0 1 2 3) (Table 1).
All these permutations could be generated using We are not interested to store the inverse function
simple polynomial functions with the maximum coefficients because the inverse algorithm depends
degree equal to 2: only on k0, k1 and q equal to k2.
k2 A decimal identifier of each permutation could be
E k (a) = c = k 0 + k1 a , k1 0,
used as the encryption key.
k 2 {1; 2}, (k 0 , k1 , k 2 ) (0, 1, 1) B. GF(8)
All these functions could be inversed: This field has eight 3-bits symbols of order 7 and it
could be used by VEA and PEA.

[
E k 1 ( c ) = k 1 1 ( c + k 0 ) ]
q
= q 0 + q1 c
q2 There are 8!-1 = 40 319 possible permutations without
the identity one (0 1 2 3 4 5 6 7) but not all these
permutations are generated using simple polynomial
The inverse functions are simple polynomial functions functions defined on GF(8).
with another set of coefficients. There are 335 simple 3-coefficients functions:
On GF(4) we use two couples (k2, q): (1, 1) and (2, 2).
k2
E k (a) = c = k 0 + k1 a , k1 0, k 2 0,
Table 1.
Encryption Key Permutation Inverse Key k 2 7, ( k 0 , k1 , k 2 ) (0, 1, 1)
k0 k1 k2 Permutation Type
0 1 1 (0,1,2,3) (0,1,2,3) Identity On GF(8) the couples (k2, q) are: (1, 1), (2, 4), (3, 5),
1 1 1 (1,0,3,2) (1,0,3,2) S (4, 2), (5, 3) and (6, 6).
2 1 1 (2,3,0,1) (2,3,0,1) S Other polynomial inversable functions defined on
3 1 1 (3,2,1,0) (3,2,1,0) S GF(8) have the following expression:
0 2 1 (0,2,3,1) (0,3,1,2) A 2 3
E k (a) = k 0 + k1 a + k 2 a + k 3 a , k1 k 2 0, k 3 = k16 k 22
1 2 1 (1,3,2,0) (3,0,2,1) A
2 2 1 (2,0,1,3) (1,2,0,3) A These functions generate different permutations then
3 2 1 (3,1,0,2) (2,1,3,0) A those obtained with the simple functions but the
0 3 1 (0,3,1,2) (0,2,3,1) A inverse function is difficult to deduce.
1 3 1 (1,2,0,3) (2,0,1,3) A C. GF(16)
2 3 1 (2,1,3,0) (3,1,0,2) A This field has sixteen 4-bits symbols and it could be
3 3 1 (3,0,2,1) (1,3,2,0) A used by VEA, PEA or CVPEA .
0 1 2 (0,1,3,2) (0,1,3,2) S Only some elements of this field have the maximum
1 1 2 (1,0,2,3) (1,0,2,3) S order 15.
There are 8 couples (k2, q) which can be used:
2 1 2 (2,3,1,0) (3,2,0,1) A
(1, 1), (2, 8), (4, 4), (7, 13), (8, 2), (11, 11), (13, 7)
3 1 2 (3,2,0,1) (2,3,1,0) A
and (14, 14).
0 2 2 (0,2,1,3) (0,2,1,3) S
1 2 2 (1,3,0,2) (2,0,3,1) A There are 16!1 21 1012 possible permutations of
2 2 2 (2,0,3,1) (1,3,0,2) A 16 symbols.
3 2 2 (3,1,2,0) (3,1,2,0) S There are 1920 simple polynomial inversable
0 3 2 (0,3,2,1) (0,3,2,1) S functions defined on GF(16).
The experimental analysis of these functions showed
1 3 2 (1,2,3,0) (3,0,1,2) A
that only the linear and the square functions generate
2 3 2 (2,1,0,3) (2,1,0,3) S
unique permutations. For higher exponents, the
3 3 2 (3,0,1,2) (1,2,3,0) A

58
permutations are repeated. So we can use 479
functions for permutation on GF(16):

k2
E k (a) = c = k 0 + k1 a , k1 0,
k 2 {1, 2}, (k 0 , k1 , k 2 ) (0, 1, 1)
D. GF(32)
This field has all 5-bits symbols of order 31 and it is
optimum to define permutation functions for CVPEA.
There are 32!1 2.6 10 35 possible permutations of
32 symbols except the identity one.
29759 permutations are generated using simple
polynomial functions defined on GF(32):
k2
E k (a) = k 0 + k1 a , k1 0,
k 2 0, k 2 31; (k 0 , k1 , k 2 ) (0, 1, 1)

V. CONCLUSIONS

Encryption is the base of any secure communication

system. Simple polynomial inversable functions
defined on Galois Fields are proposed for symbol
permutation. Efficient and fast cryptography
algorithms are introduced: Value Encryption
Algorithm (VEA), Position Encryption Algorithm
(PEA) and Combined Value-Position Encryption
Algorithm (CVPEA). Different Galois Fields and the
3-coefficients polynomial functions are analyzed.

REFERENCES
[1] R.E Blahut, Digital Transmission of Information, Addison-
Wesley Publishing Co., 1990.
[2] A.Menezes, Handbook of Applied Cryptography, CRC Press,
Inc., 1997.
[3] Scripcariu L., Duma P, About Some Cryptography Functions
Defined on Galois Fields, Buletinul Institutului Politehnic Iasi,
Romania, Sect. III Electrotehnica, Energetica, Electronica, Tom
L(LIV), Fasc.1-2, 2004, pp.65-70.

59
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Parallel Concatenated Convolutional Turbo Codes:
Performance analysis for different interleaving schemes
Rodica Stoian, Lucian Andrei Perioar1
Abstract The Turbo Codes reported to achieve
Shannon limits are concatenated recursive systematic
codes. The performances of the Turbo Codes are
dependent on encoder design, the iterative decoding
procedures and the interleaving scheme. There are two
classic types of interleaving, block and convolutional,
which can be used in Turbo Codes and which operate the
data in a fix or a pseudorandom order.
In this study, we investigate by simulation an iterative
decoding algorithm for Turbo Codes using a classical
convolutional interleaver (A) or a block interleavear (B).
For the same code parameters, the (B) algorithm assures
better results then (A) algorithm, assuming an increased
number of corrected errors.
Keywords turbo codes, convolutional codes, weight,
frame size, generator functions, interleaver.
I. INTRODUCTION
Error control coding, or channel coding, is a
method of adding redundancy to information so that it
can be transmitted over a noisy channel to another
party, and can be checked and corrected for errors that
occurred in the transmission. Channel coding is the
best method for transmitting information with fewer
errors and lower signal power. It is best used for
transmitting media that is sensitive to error, such as
compressed voice and video, or data. Of course, some
types of codes work better than others. To date, the
best practical code that has been found is the Recursive
Systematic Convolutional (RSC) Turbo code. This
uses two RSC encoders, separated by an interleaver,
for encoding and multiple iterations of the MAP
algorithm for decoding. Some advantages to using
Turbo codes are that they significantly outperform
conventional codes and they use interleavers which
reduce burst errors.
Some of the performance characteristics of Turbo
codes include more than 8.5 dB coding gain compared
with an uncoded channel, at BER=10-5 [1]. Such
coding power is extremely important in many
telecommunications applications because it either
decreases the amount of power necessary for
transmitting a signal or increases the range in which
the signal can be received. This coding gain is also
1
Electronics and Telecommunications Faculty, Applied Electronics Departament, Iuliu Maniu 1-3, Bucharest, Romania
e-mail: rodicastoian2003@yahoo.com, lperisoara@yahoo.com
60
favorable in deep space and satellite communications,
digital communication and storage.
The goal of channel coding is to reduce the number
of errors caused by transmission in a power-limited
environment. Theoretically, the best possible
performance that AWGN channel can accomplish is
called the Shannon limit. A code with Shannon limit
performance is ideal, but so far, has not been achieved
in practice. The only practical code that comes close to
the Shannon limit is the RSC Turbo code [1], [2].
II. PARALLEL CONCATENATED
CONVOLUTIONAL CODES
Turbo codes typically use at least two convolutional
component encoders and two MAP algorithm
component decoders in the Turbo codec. This is known
as concatenation. Three different arrangements of
Turbo codes are Parallel Concatenated Convolutional
Codes (PCCC), Serial Concatenated Convolutional
Codes (SCCC), and Hybrid Concatenated
Convolutional Codes (HCCC). Typically, Turbo codes
are arranged like the PCCC. An example of a PCCC
Turbo encoder is as follows in Fig. 1.
It is known that turbo codes are the best practical
codes due to their performance at low SNR. One reason
for their better performance is because Turbo codes
produce high weight codewords [3]. For example, if the
input sequence u is originally low weight, the
systematic u and parity c1 outputs may produce a low
weight codeword. However, the parity output c2 is less
likely to be a low weight codeword due to the
interleaver in front of it. The interleaver shuffles the
input sequence u, in such a way that when introduced
to the second encoder, it is more likely to produce a
high weight codeword. This is ideal for the code
because high weight codewords result in better decoder
performance.
Although the encoder determines the capability for
the error correction, it is the decoder that determines the
actual performance. The performance, however,
depends upon which algorithm is used, the Maximum
A Posteriori algorithm (MAP) or the Soft Output
Viterbi Algorithm (SOVA).
 RSC 1 coder u

D D
u
x=
c1
Interleaver
(u,c1,c2)

D D
u

c2
RSC 2 coder
Fig. 1: Parallel Concatenated Convolutional Code with rate 1/3 and G ( D ) = [ 1 1 + D 2 1 + D + D 2 ] .

III. INTERLEAVER For continuous processing of the data, are needed

two UxV matrix, one for the interleaver and another for
In this paper we will analyze the interleaver the deinterleaver. The parameters defined above are:
influence on Turbo codec performance. We propose to The delay of the interleaver / deinterleaver circuits:
investigate two types of interleavers: block and
convolutional. An interleaver of period T, can be D = 2 U V time units (2)
described by the finite permutation on the numbers: The total memory:

: T T , with T {1, 2,..., T } M = 2 U V (3)

1 2 ... T The period:

= (1)
(1) (2) ... (T ) T = U V (4)
The spreading factors:
The performances of the interleaver are dependent
on the following numerical values:
LR/TB: (s=U-1, t=V), (s=U, t=V-1),
- delay, D the time needed for the memory
LR/BT: (s=U, t=V),
access, read/write operations and permutations. (5)
RL/TB: (s=U, t=V),
- number of memory cells, M the total number of
RL/BT: (s=U-1, t=V), (s=U, t=V-1),
memory cells used by the interleaving and
deinterleaving circuits for the delay operations.
- period, T the time between two identical writing
sequences. LR/TB LR/BT
- spreading factors, s, t an interleaver has the (s,t) 1 2 3
reading

factors if wherever i j < s then 4 5 6

RL/TB RL/BT
(i ) ( j ) t . 7 8 9
Interleaver design should focus on two goals:
maximum scattering of data to counter-time dependent (a) 3x3 block (b) operating modes
channel effects, and maximum disruption of data to
eliminate data structure that lead to low weight parity
sequences [4], [5].
LR/TB LR/BT
A. Block Interleaver 1 2 3 4 5 6 7 8 9 1 2 3 4 5 6 7 8 9

A block interleaver is a matrix UxV, as in Fig. 2.a.
1 4 7 2 5 8 3 6 9 7 4 1 8 5 2 9 6 3
The information bits are writing on rows and reading
on columns. These operations are realized as it is RL/TB RL/BT
described in Fig. 2.b. The four types of block 1 2 3 4 5 6 7 8 9 1 2 3 4 5 6 7 8 9
interleavers are named after the writing and reading
modes: 3 6 9 2 5 8 1 4 7 9 6 3 8 5 2 7 4 1
LR/TB - write Left to Right; read Top to Bottom (c) block permutations
LR/BT - write Left to Right; read Bottom to Top
RL/TB - write Right to Left; read Top to Bottom
Fig. 2: Classical Block Interleaver Schemes.
RL/BT - write Right to Left; read Bottom to Top

61
 1 1

L
2 (K-1)L

2L 2L
3 K-2

in out in out
L
K-1
(K-1)L
K K

(a) (b)

Fig. 3: Block diagram of the convolutional interleaver (a) and convolutional deinterleaver (b).

To increase the dispersion of errors, the columns The period:

must be read in another order that was described. M
and D parameters doesnt change. To realize this, the
matrix columns are permutated before reading. Also,
we can realize different permutations on each row.
B. Convolutional Interleaver
The convolutional interleaver has K delay lines
(shift registers) which work in parallel. On register
inputs and outputs are two switches needed for
matrix reading and writing operations. Each line has a
different number of delay cells: first line is a shortcut
from input to output; the second line has L delay cells;
the following lines have a surplus of L delay cells
each; the last line has (K-1)L delay elements. In Fig. 3,
is shown the block diagram of the convolutional
interleaver and deinterleaver.
For any (K,L) convolutional coder, it is true the
followings:
Total memory register used:
K 1 KL ( K 1)
M = L i =
i =0 2 The decoding process begins by receiving partial
The delay of the interleaver/deinterleaver circuits:
D = 2 M = KL ( K 1)
T =K (8)
The spreading factors:
L=1: (s=K+1, t=K-1)
L>1: (s=K, t=KL-1), (s=K(L-1)+1, t=K), (9)
(s=KL+1, t=K-1)
The values of characteristic parameters are
dependent on interleaver type, block or convolutional,
and operation modes for block interleaver. These values
are given in a synthetic form, in Tabel 1.
IV. TURBO DECODER
The MAP algorithm, also commonly referred to as
the BCJR algorithm, is often used to estimate the most
likely information bit to have been transmitted in a
coded sequence. The MAP algorithm is favored
because it outperforms other algorithms, under low
SNR conditions. The major drawback, however, is that
it is more complex than most algorithms because of its
(6) focus on each individual bit of information.
information from the channel u and c1 and passing it to
the first decoder. The rest of the information, parity c2,
(7) goes to the second decoder and waits for the rest of the
information to catch up. While the second decoder is

Tabel 1. Interleaver parameters

Interleaver Block
Convolutional
Parameters LR/ TB LR/BT RL/TB RL/BT

Delay 2(U-1)(V-1) 2(U-1)V 2U(V-1) 2UV-2 KL ( K 1)

KL ( K 1)
Interleaver memory (U-1)(V-1) (U-1)V U(V-1) UV-1
2

Period T T T T K

s=U-1, t=V s=U, t=V s=U, t=V s=U-1, t=V

Spreading factors see Equations (9)
s=U, t=V-1 s=U, t=V s=U, t=V s=U, t=V-1

62
waiting, the first decoder makes an estimate of the
transmitted information, interleaves it to match the
format of parity c2, and sends it to the second decoder.
The second decoder takes information from both the
first decoder and the channel and re-estimates the
information. This second estimation is looped back to
the first encoder where the process starts again.
This cycle will continue until certain conditions are
met, such as a certain number of iterations or stop
criterion are performed. When the decoder is ready, the
estimated information is finally kicked out of the cycle
and the decisions are made in the threshold block. The
result is the decoded information sequence.

V. SIMULATION RESULTS

The interleavers are specified by the design of a Fig. 6: The output frame of LR/TB block interleaver U=100, V=100.
permutation on the integers {0, 1, 2, , T-1}, where
T=10000 is the frame size. For a LR/TB block
interleaver, we set the matrix dimensions to U=100,
V=100, with U*V=T. The total memory used by the
interleaver and deinterleaver is 20000 and the delay of
interleaving is also 20000 time units. The spreading
factors are s=99, t=100 or s=100, t=99. For this
interleaver, Fig. 6 shows the scatter plot when the input
frame is the identity permutation (i)=i (Fig. 5).
The dispersion for a convolutional interleaver is
shown in the scattered plot (Fig. 7 and Fig. 8), for
different values of the K and L parameters. Setting
K=10 and L=15 we obtain the convolutional interleaver
with memory M=675 and a delay of 1350 time units.
The spreading factors (s,t) are (10, 149), (141, 10) or
(151, 9).With this parameters, the convolutional
interleaver has the scatter plot represented in Fig. 7. If
we change the K and L parameters (K=20, L=30), the
dispersion is bigger, as we see from the scatter plot in Fig. 7: The output frame of convolutional interleaver with K=10
the Fig. 8. The spreading factors (s,t) are (20, 599), and L=15.
(581, 20) or (601, 19). Also, the memory size and the
delay increases at M=5700 and D=11400 time units.
The number of lines from Fig. 7 and Fig. 8 is
given by the number of lines K of the interleaver.
The distance between them varies with the KL
product. It is obvious that the best performance is
obtained for the block interleaver.

Fig. 8: The output frame of convolutional interleaver with K=20

and L=30.

To study the effect of interleaving dimensions on

Turbo coding system, for a Turbo coder with rate 1/3,
two identical RSCs with
Fig. 5: The input frame for dispersion simulations. G ( D ) = [ 1 1 + D 2 1 + D + D 2 ] it was simulated the
function for four different block interleavers with

63
dimensions (U,V): (2,1000), (10,200), (16,125), 1
(40,50). We consider that the transmission was done

Bit Error Rate

on
1 0.1
Bit Error Rate

0.01
0.1

0.001

0.01 uncoded
0.0001 conv
uncoded
block
I1 (2,1000)
0.001 I2 (10,200) 0.00001
I3 (16,125) 0 1 2 3 4
I4 (40,50) Eb/N0 (dB)
0.0001 Fig. 11: BER(Eb/N0) for different types of interleavers.
0 1 2 3 4
Eb/N0 (dB)

Fig. 9: BER(Eb/N0) for block interleavers with different dimensions.

an AWGN channel with signal to noise ratios (Eb/N0)
variable from 0dB to 4dB and the BPSK modulation.
The Turbo decoder operates based on iterative Log-
MAP algorithm.
The results are presented in Fig. 9 and they put in
evidence the BER dependence of Eb/N0, U/V ratio and
st product. For SNR greater then 2 dB, the best
performance (BER = 10-3 10-4) are assure for the
interleaver I4 characterized by almost quadratic matrix
(U,V)=(40,50), U/V ratio tending to 1 and the biggest
values for the st=1950. The poor result was assure by
the interleaver I1 with U/V =2*10-3 and st=1000.
The Turbo coding system performances are
dependent on frame size. The influence of frame size k
on BER for the system implemented with the
interleaver I4 has also been examined. The results of
the simulation are presented in Fig. 10. For frame size
with k=5000, the BER is less than 10-4 for Eb/N0=4 dB.
1
Bit Error Rate
The performances of the Turbo codec system for
different types of interleaving schemes are shown in
Fig. 11. The block interleaver is described by the
matrix dimensions U=40, V=50 and the convolutional
interleaver by the parameters K=20 and L=30. As
expected, the block interleaver is better than the
convolutional interleaver.
VI. CONCLUSIONS
In this paper we analyze the performances of a
Turbo codec with different types of interleavers. There
were presented performance parameters and operation
modes for block and convolutional interleavers. The
performances of four type block interleavers are
compared and analyzed. For the best block interleaver,
a comparison with a convolutional interleaver was
done. For the implemented interleavers in Turbo code
system, the simulation results confirm the better
position of block interleavers in terms of BER and
SNR.
REFERENCES

0.1 [1] G. Berrou, A. Glavieux, Near optimum error-correcting coding

and decoding: Turbo codes, IEEE Trans. Comm, vol. 44, pp. 1261-
1271, Oct. 1996.
[2] L. R. Bahl, J. Cocke, F. Jelinek and J. Raviv, Optimal decoding
0.01 of linear codes for minimizing symbol error rate, IEEE Trans.
Inform. Theory, vol. 20, pp. 284-287, 1974.
[3] H. Jin, R. J. McEliece, Coding Theorems for Turbo Code
uncoded Ensembles, IEEE Trans. Inform. Theory, vol. 48, No. 6, pp. 1451-
0.001
k= 50 1463, June 2002.
k= 200 [4] J. L. Ramsey, Realization of optimum interleavers, IEEE
k= 500 Trans. Inform. Theory, IT-16(3): pp. 338-345, May 1970.
0.0001
k=2000 [5] S. A. Barbulescu and S. S. Pietrobon, Interleaver design for
Turbo codes, Electron. Lett., vol. 30, No. 25, pp. 2107-2108, Dec.
k=5000
1994.
0.00001 [6] C. Heegard, S. B. Wicher, Turbo Coding, Kluwer Academic
0 1 2 3 4 Publishers, Boston, Massachusetts, 1999.
Eb/N0 (dB)

Fig. 10: BER(Eb/N0) for a quadratic interleaver and different

frame sizes.

64
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Evaluation of Parameters Used in Lossless Text
Compression With the Burrows-Wheeler Transform
Radu Rdescu, Ionu Blan1
Abstract This paper presents a study of parameters
involved in the lossless text compression methods using
the Burrows-Wheeler Transform (BWT). BWT (also
known as Block Sorting) is one of the most efficient
techniques used in data compression. Its purpose is to
preprocess the text before applying a compression
algorithm, thus providing a better use of the inner
redundancy of the text. BTW converts the original
blocks of data into a format that is extremely well suited
for compression. This paper deals with the choice of the
range for the block length for different types of text
files, evaluating the compression ratio and compression
time.
Keywords: Block Sorting, Move-to-Front (MFT), Run
Length Encoding (RLE), Huffman coding, arithmetic
coding
1
I. INTRODUCTION
With the Burrows-Wheeler Transform [1], the
compression algorithm is the following. Given a text
file, the Burrows-Wheeler Transform is applied on it.
This produces a new text, which is suitable for a
Move-to-Front encoding [2], [4] (since it has a great
number of sequences with identical letters). The
result is another new text, which is more suitable for
Huffman or arithmetic encoding, usually preceded by
a Run-Length Encoding (RLE), [3] since this text
produces many small numerical values. The only step
that actually performs compression is the third one
(the statistical algorithm). The two other steps are
meant to ensure that the Huffman/arithmetic
encoding [6] is able to compress the data efficiently.
II. BURROWS-WHEELER TRANSFORM
The transform divides the original text into blocks of
the same length, each of them being processed
separately. The blocks are then rearranged using a
sorting algorithm. This is why it is also called Block
Sorting. [1] The resulting block of text contains the
same symbols as the original, but in a different order.
Sorting the rows will be the most complex and time
consuming task in the algorithm, but present
implementations can perform this step with an
1
Facultatea de Electronic i Telecomunicaii, Catedra de
Electronic Aplicat i Ingineria Informaiei, Bd. Iuliu Maniu
nr. 1-3, sector 6, Bucureti, e-mail: rradescu@atm.neuro.pub.ro
65
acceptable complexity. The transformation groups
similar symbols, so the probability of finding a
character close to another instance of the same
character increases substantially. The resulting text
can be easily compressed with fast locally adaptive
algorithms, such as Move-to-Front coding combined
with Huffman or arithmetic coding.
The Block Sorting algorithm transforms the original
string S of N characters by forming all possible
rotations of those characters (cyclic shifts), followed
by a lexicographical sort of all of the resulting strings.
The output of the transform is the last character of the
strings, in the same order they appear after sorting.
All these strings contain the same letters but in a
different order. One of them is the original string S.
An index of the string S is needed, because its
position among the sorted strings has to be known in
order to reverse the transform.
The transform is reversible because the output is a
string containing the same letters as the input. By
performing a lexicographical sorting, a string
identical to the first column of the transform matrix M
is obtained. Starting only with the last column of the
matrix (the transform result), the first column of the
matrix is easily recognizable. The unsorting column
U requires the use of a transformation vector T [5].
The transformation vector gives the correspondence
between the characters of the first column and the
following ones, found in the last column. The
procedure begins with a starting point and thus the
rows contained in the original string are identified.
Since BWT groups closely together symbols with a
similar context, the output can be more than two
times smaller than the output obtained from a regular
compression. Compressing a text file with the
Burrows-Wheeler Transform can reduce its size while
the compression without the transform gave a weaker
output. The compression method used in both cases
consists of the three stages following BWT: Move-to-
Front, Run-Length Encoding and arithmetic coding.
 III. MOVE-TO-FRONT

Move-to-Front encoding technique inputs a string

and outputs a series of numbers, one for each
character in the input string. All the 256 characters in
the ASCII code will have correspondents in a list
with 256 numbers. It can be made an optimization by
putting the most often characters on the first places in
the list so that they will be coded with small numbers.
If the input string has an important number of
sequences containing the same letter in a row, then
the output series will have many small values. This
means that if an input string has numerous sequences
Fig. 1. Compression ratios with and without BWT
containing the same letter in a row, then the Move-to-
Front encoding could be performed first. Huffman or
For both cases (with and without BTW), the
arithmetic encoding could be applied afterwards. For
compression time was estimated using the same set of
the Huffman coding table an adequate choice is to
test files (in the same order, from left to right). The
use less space for small numbers than for greater
results are shown in Figure 2.
numbers. The result should be a shorter sequence
than the original one.

IV. COMPRESSION PARAMETERS

In order to evaluate the performances of BWT (the

compression ratio and the time needed to perform the
compression), it is suitable to perform the tests on
different types of text files and to vary the block
length of the currently processed input. The test files
are text files (.txt, .ppt, and .doc) but also a .bmp file, Fig. 2. Compression time with and without BWT
containing a screen shot.
The obvious conclusion is that BWT improves
The main goal of the test is to evaluate the essentially the compression ratio (two times, in
contribution of BWT in the overall result of the average) with the price of increasing the compression
compression process. The steps of the compression time, but only for certain files from the test set.
method are the following:
Run-Length Encoding; In order to estimate the performances of the transform
Burrows-Wheeler Transform; as function of block length (in the compression
Move-To-Front; process), a test file (.doc) of 858 kB was used. The
Run-Length Encoding; block length represents the information transformed
Arithmetic compression. by BWT and compressed at a time (on a processing
stage). The number of stages performed to obtain the
Initially, the complete 5-step algorithm was compressed file is calculated as the overall dimension
performed for the test file set and then the algorithm of the file divided by the block dimension. Taking
was performed again, omitting the 2nd step (BWT). into account that the file is read binary and the output
Table 1 presents the original dimensions of the test is stored on bytes (characters of 8 bits) it results a
files. number of 858,000 symbols for the test file. It is
recommended to choose the block length sufficiently
Table 1 large in order to exploit the redundancy within. The
File name Dimension (kB) compression results for different values of block
fis1.doc 85 length are presented in Table 2.
fis2.doc 694
fis3.doc 858 The dimension of the compressed file is constantly
fis4.ppt 90 decreasing until the block length exceeds 320,000
fis5.ppt 152 symbols. Beyond this value, it appears a limitation
img.bmp 2305 and then a slight increasing of the resulting archive.

To represent the corresponding graphic a logarithmic

The compression results for both cases are shown in
scale was used in order to largely emphasize the
Figure 1 (on the left with BWT, on the right
range of the values for the block length. The
without BWT).
dimension of the compressed file as function of block
length is shown in Figure 3.

66
Table 2
Block length (103) Compressed file (kB) Therefore, it is obvious that the required time will
0.1 671 increase, depending on the dimension of the block of
1 256 processed data.
2 213
3 196 The compression time as function of block length is
5 180 shown in Figure 4. For large values of the block
10 165 length, the compression time substantially increases
50 145 in the case of applying the BWT. This result could be
explained not only by the presence of the BWT but
100 142
also by the adaptive arithmetic compression, which
250 139
supposes a two-stage processing of the block and an
286 134
adjustment of the codewords, depending on their
300 132 frequencies and, eventually, on the number of
325 131 symbols.
350 132
400 134 V. COMPARISON WITH STANDARD
600 138 COMPRESSORS
800 138
WinRAR, WinAce and WinZip were chosen among
the usual compression programs, in order to compare
the performances of the algorithm presented above,
applied on the same 5-file test set. The dimensions of
the compressed files (using the 3 standard
compressors and the BWT algorithm) are shown in
Figure 5, for the considered test files.

Fig. 3. Compressed file dimension as function of block length

The dimension of the compressed file is constantly

decreasing with block length. For high values of the
block length, the compression ratio (calculated as
original file dimension divided by compressed file
dimension) is stabilized to the value of 6.35.
Generally, the block length could be about 200,000
bytes. In this case, both compression ratio and
compression time reach their optimal values. Fig. 5. Dimension as function of block length for BWT, WinAce,
WinZip, WinRAR compressed files and original files
In order to evaluate the algorithm complexity once
the BWT was introduced, the compression time for The advantages of using the BTW algorithm for all
both cases is compared. The BWT implies the files (and especially for the image file) are obvious.
permutation of the symbols within the file, as well as The compression ratio of the BWT algorithm is very
the sorting of the permutations. close to the average of the compression ratios of the
standard compressors.

V. CONCLUSIONS AND REMARKS

The compression technique based on BWT provides

good results in comparison with the general-purpose
compressors. The algorithm has a high degree of
generality and could be applied on the majority of file
types (text, image or other files).

BWT uses the sorting of symbols in the original file,

and the data processing is performed on blocks
obtained by dividing the source file. An important
issue in optimizing the performances of the BWT
Fig. 4. Compression time as function of block length
algorithm is to choose an adequate value of the block

67
length. From this point of view, one have to take into
account both the compression performances and the
required computing resources. To get a good
compression ratio and an acceptable compression
time, the block length could be situated around
200,000 bytes. For block length less than 100,000
bytes, the compression ratio is sensibly decreased, as
well as the compression time. An excessive
increasing of block length (over 800,000 bytes)
produces an unacceptable compression time.

Generally, one can observe a constancy of the

compression ratio for the recommended value of the
block length (200 kB), due to the high redundancy
within the text or the image file. This value could be
considered an upper bound for the block length, in
order to assure a satisfactory result.

The BWT algorithm can be applied on any type of

data because the inverse transform is performed with
no losses of information. Hence, BTW modifies the
symbol positions but it does not change the
probability distribution. As a result, the complexity of
an implementation of the overall compression
method (including the other four steps) does not
exceed the similar values of the classic lossless
compression standards, based on LZW-type
algorithms.

The BWT represents an efficient data processing

method that could be successfully integrated in any
compression technique for general purpose.

REFERENCES
[1] M. Burrows and D. J. Wheeler, "A Block-Sorting Lossless
Data Compression Algorithm", 1994, report available at:
http://gatekeeper.dec.com/pub/DEC/SRC/research-
reports/abstracts/src-rr-124.html.
[2] M. Nelson, "Data Compression with the Burrows-Wheeler
Transform", September 1996, available at:
http://dogma.net/markn/articles/bwt/bwt.htm.
[3] M. A. Maniscalco, "A Run Length Encoding Scheme for Block
Sort Transformed Data", 2000, available at:
http://www.geocities.com/m99datacompression/papers/rle/rle.html.
[4] P. M. Fenwick, "Block Sorting Text Compression", 1996,
available at: ftp.cs.auckland.ac.nz.
[5] T. C. Tell, J. G. Cleary and I. H. Witten, Text Compression,
Prentice Hall, Englewood Cliffs, NJ, 1990.
[6] R. Rdescu, Compresia fr pierderi: metode i aplicaii,
Matrix Rom, Bucharest, 2003.

68
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

A Low Dynamics Fast Transversal Filter Adaptive

Algorithm
Constantin Paleologu, Silviu Ciochin, Andrei Alexandru Enescu1
Abstract This paper propose a low dynamics version of The organization of this paper is as follows. The
FTF adaptive algorithm using a modified form of the theoretical background of our problem is introduced
cost function, based on an asymptotically unbiased in the next section. The derivation of the low
estimator of the mean square error. The reduced dynamics FTF adaptive algorithm is performed in
dynamics of the modified algorithms parameters could
lead to facility for fixed-point implementation.
section III. Some simulation results are presented in
Keywords: Adaptive filtering algorithms, FTF section IV. Finally, the conclusion remarks are given
algorithm, dynamics of parameters, fixed-point in section V.
implementation.
II. THEORETICAL BACKGROUND
I. INTRODUCTION
The LS cost function is defined as an estimate of the
The Recursive Least Squares (RLS) adaptive mean square-error:
algorithm [1], [2] is one of the most popular adaptive
algorithms, mainly due to its fast convergence rate. n 2 2
Nevertheless, there are some major drawbacks related J ( n ) = n i e ( i ) = J ( n 1) + e ( n ) (1)
to the high computational complexity and the large i =1
dynamic range of the algorithms variables.
The first inconvenient could be solved by using a where 0 < 1 is the exponential weighting factor
fast least squares (LS) algorithm, in sense that the and e(i) is the estimation error at time i. This estimate
computational cost increases linearly with the number of the cost function induces similar estimates for the
of adjustable parameters. There are a variety of fast correlation matrix (n) and the cross-correlation
LS algorithms with widely varying properties. The vector (n):
most known are Recursive Least-Squares Lattice
(RLSL), QR-Decomposition based Least-Squares n
Lattice (QRD-LSL) and Fast Transversal Filter (FTF) ( n ) = n i x ( i ) x H ( i ) = ( n 1) + x ( n ) x H ( n )
algorithms [1], [2]. Of these, only the FTF algorithm i =1
generates the adaptive filter weights. Unfortunately, (2)
FTF algorithm suffers from the numerical instability n
( n ) = n i x ( i ) d * ( i ) = ( n 1) + x ( n ) d * ( n )
problem under a finite precision implementation [1].
i =1
The second drawback concerning the large dynamics
of parameters could cause unwanted finite precision (3)
effects. Especially in a fixed-point arithmetic context, where x(i) is the tap-input vector and d(i) is the
overflow or stalling phenomena could occur mainly desired response, both at time i. The superscript H
due to the inherent scaling operations. The guiltiest denotes Hermitian transposition (transposition and
parameter for that large dynamic range is the complex conjugation) and the superscript * denotes
algorithm cost function. In the case of any LS the complex conjugation.
adaptive algorithm the cost function produces a large The expectations of these functions are
biased estimate of the mean square error. In this paper

{ }
we propose a version of FTF algorithm based on a 1 n
E { J ( n )}
2
modified form of the cost function, using an E e ( n) (4)
1
asymptotically unbiased estimator of the mean square
error [3], [4]. The reduced dynamics of the modified
1 n
algorithms parameters could lead to facility for fixed- E { ( n )} R (5)
point implementation. 1

1
Politehnica University of Bucharest, Electronics and Telecommunications Faculty, 1-3, Iuliu Maniu Bvd., Bucharest, Romania,
e-mail: pale@comm.pub.ro, silviu@comm.pub.ro, aenescu@comm.pub.ro

69
where R is the correlation matrix of input data. We III. LOW DYNAMICS FTF ADAPTIVE
can see that J(n) is a biased estimate of E{|e(n)|2} and ALGORITHM
similarly (n) is a biased estimate of R. It will result
Let us consider a forward linear predictor of order N
E { J ( n )}
n

1
1
E e ( n)
2
{ } (6)
with the vector coefficients at time n denoted by
a N (n) . The forward a posteriori prediction error
produced at the output is
1
E { ( n )} R (7)
n 1 eNf (i ) = a H
N ( n) x N +1 (i ) (13)

Some classes of applications [5] require a high where x N +1 (i ) is the N+1-by-1 the tap-input vector,
memory algorithm, which means that the value of the
with 1 i n .
exponential weighting factor is very close to unity.
The cost function is the sum of weighted forward a
In this case very large values for these parameters can
posteriori prediction-error squares in the modified
result, causing unwanted finite precision effects in a
form according to (8):
practical implementation.
Taking into account the previous discussion we
n 2
propose an unbiased estimator of the matrix (n). So J Nf ( n ) = (1 ) n i eNf ( i ) =
that, we will modify the cost function from equation i =1 (14)
(1) as follows: 2
= J Nf ( n 1) + (1 ) eNf (n)
n 2
J ( n ) = (1 ) n i e (i ) =
i =1 (8) The corresponding backward linear prediction-error
2 filter with the vector coefficients denoted by c N ( n)
= J ( n 1) + (1 ) e ( n )
will produced the backward a posteriori prediction
error:
In this case
ebN (i ) = c H
N ( n) x N +1 (i ) (15)
E { J ( n )} (1 n ) E { e (n) 2
} (9)
In this case, the cost function is the sum of weighted
backward a posteriori prediction-error squares:
is an asymptotically unbiased estimator of the mean
square-error. n
2
Following this idea we have to perform the same ( ) ( ) ni ebN ( i ) =
b n = 1
JN
modification in equations (2) and (3) obtaining i =1 (16)
2
n = JN ( ) ( ) N( )
b n 1 + 1 eb n
( n ) = (1 ) ni x ( i ) x H ( i ) =
i =1 (10)
Let N +1 ( n ) denote the N+1-by-N+1 correlation
= ( n 1) + (1 ) x ( n ) xH (n)
matrix of the tap-input vector x N +1 (i ) , where
n 1 i n , f ( n ) denote the m-by-1 cross-correlation
(n) = n i x (i ) (i ) =
d*
i =1 (11) vector between x(i ) and x N (i 1) , and b ( n )
= ( n 1) + (1 ) x ( n ) d * ( n ) denote the N-by-1 cross-correlation vector between
x N (i ) and x(i N ) . According to (10) and (11)
According, these parameters have the following forms:

( )
n
E { ( n )} 1 n R (12) N +1 ( n) = (1 ) ni x N +1 (i )x H
N +1 (i ) =
i =1
is an asymptotically unbiased estimator of the = N +1 ( n 1) + (1 ) x N +1 (n)x H
N +1 ( n)
correlation matrix. (17)
Most of the expressions in the following section may n
look familiar to readers acquainted with the theory of f (n) = (1 ) ni x m (i 1) x* (i ) =
least-squares transversal filters. However, the i =1 (18)
derivation that follows is developed according to the
= f (n 1) + (1 ) x m ( n 1) x* (n)
new approach.

70
 n g ( n ) = g ( n 1) + k N ( n ) N ( )
b* n (28)
b (n) = (1 ) ni x N (i ) x* (i N ) =
i =1 (19)
= b (n 1) + (1 ) x N (n) x* (n N ) k N ( n ) = (1 ) N1 ( n ) x N ( n ) (29)

In the case of forward linear prediction the normal Nb ( n ) = c H

N ( n 1) x N +1 ( n ) (30)
equation is:
k * ( n ) b
N ( n 1) w ( n ) = f ( n ) (20) c N ( n ) = c N ( n 1) N N (n) (31)
0
where w(n) is the tap-weight vector of the forward
min ( n ) = J N min ( n 1) + (1 ) N ( n ) eN ( n )
b
JN b b b*
linear predictor, or
(32)
J f where:
N +1 ( n ) a N ( n ) = N min (21) b
- JN min is the minimum value of the sum of
0
weighted backward a posteriori prediction-error
squares;
where J Nf min is the minimum value of the sum of - c N (n) is the tap-weight vector of the backward
weighted forward a posteriori prediction-error prediction-error filter;
squares. - g(n) is the tap-weight vector of the backward
Following the classical procedure it is easy to deduce predictor;
the following recursion for updating the tap-weight - k N (n) is the modified gain vector;
vector of the predictor:
- Nb (n) is the backward a priori prediction error.
w ( n ) = w ( n 1) + k N ( n 1) Nf * ( n ) (22) The next step is to define a modified extended gain
vector:

where Nf ( n) is the forward a priori prediction error: k N +1 ( n ) = (1 ) N1+1 ( n ) x N +1 ( n ) (33)

Nf ( n ) = a H
N ( n 1) x N +1 ( n ) (23) It can be demonstrated that the inverse of the
correlation matrix may be expressed as follows:
and k N ( n 1) is the modified gain vector:
0 0 1
N1+1 ( n ) = 1 n 1
+ a ( n) aH
N (n)
N ( ) J N min ( n ) N
0 f
k N ( n 1) = (1 ) N1 ( n 1) x N ( n 1) (24)
(34)
Taking these into account we may write the recursion Using the previous relation we get the following
for updating the tap-weight vector of the prediction- recursion for the modified extended gain vector:
error filter:
0 eNf ( n )
0 f k N +1 ( n ) = (
+ 1 ) N( ) f
a n
a N ( n ) = a N ( n 1) * N (n) (25) k N ( n 1) J N min ( n )
k N ( n 1) (35)
Similarly, using an alternative expression for the
Finally, we get the following recursion for updating inverse of the correlation matrix:
the minimum value of the sum of weighted forward
prediction-error squares: 1 ( n ) 0 1
N1+1 ( n ) = N + b cN ( n) cH
N (n)
0 0 J N min ( n )
J Nf min ( n ) = J Nf min ( n 1) + (1 ) Nf ( n ) eNf * ( n )
(36)
(26) we get the second recursion for the modified extended
In a similar manner we obtain a set of relations for the
gain vector:
backward prediction part of the algorithm:

k ( n ) ebN ( n )
0 k N +1 ( n ) = N + (1 ) c N ( n ) b
N +1 ( n ) c N ( n ) = b (27) 0 J N min ( n )
J N min ( n )
(37)

71
The definition of the modified gain vector from J Nf min ( n 1)
relation (29) may also be viewed as the solution of a N +1 ( n ) = N ( n 1) (46)
special case of the normal equations. It defines the J Nf min ( n )
tap-weight vector of a transversal filter that contains N
taps and that operates of the input data x N ( n) to
min ( n 1)
b
JN
produce a least-squares estimate of a special desired N +1 ( n ) = N ( n ) (47)
min ( n )
b
JN
response:
1, i=n
d (i ) = (38) Finally, we have to put together four distinct tasks
0, 0 < i < n (forward linear prediction, backward linear prediction,
computation of the gain vector and estimation of the
The estimation error (modified conversion factor) is desired response) in order to obtain our modified FTF
defined as follows: adaptive algorithm.
First, let us define the normalized gain vector:
N ( n ) = 1 (1 ) x H
N (n) N (n) xN (n)
1 (39)
k N (n)
k N (n) = (48)
Taking into account the expression of the inverse of N (n)
the correlation matrix from the standard recursive
least-squares estimation problem [1], [2] we get:
According, some simplified recursions can be
obtained:
1
N (n) = (40)
1 + (1 ) 1x H
N (n) N (n) xN (n)
1
(1 ) N ( n )
f
0
k N +1 ( n ) = + a N ( n 1)
k N ( n 1) J N min ( n 1)
f
Three useful interpretations of the conversion factor
are known [1], [2]: (49)
- for recursive least-squares estimation: 0 f
a N ( n ) = a N ( n 1) * eN ( n ) (50)
eN ( n ) k N ( n 1)
N (n) = (41)
N (n)

Nb ( n ) = b
JN min ( n 1) k N +1, N +1 ( n ) (51)
where eN (n) is the a posteriori estimation error (1 )
and N (n) is the a priori estimation error;
N +1 ( n )
- for adaptive forward linear prediction: N (n) = (52)
1N
b
( n ) N +1 ( n ) k *N +1, N +1 ( n )
eNf ( n )
N ( n 1) = (42)
Nf ( n ) where k *N +1, N +1 ( n ) is the last element of the vector
k N +1 ( n ) .
- for adaptive backward linear prediction:
Similarly, in the case of backward prediction we get:

ebN ( n ) k N ( n )
N (n) = (43)
Nb ( n ) = k N +1 ( n ) (1 )k N +1, N +1 ( n ) c N ( n 1)
0
(53)
Taking these into account, the following recursions
k
*
for updating the conversion factor can be obtained: c N ( n ) = c N ( n 1) ebN ( n ) N (54)
0
2
eNf ( n )
N +1 ( n ) = N ( n 1) (1 ) (44) In order to complete the algorithm it is necessary to
J Nf min ( n ) update the tap-weight vector of the adaptive filter as
follows:
2
ebN ( n ) N (n) = d (n) wH
N ( n 1) x N ( n ) (55)
N +1 ( n ) = N ( n ) (1 ) (45)
min ( n )
b
JN
w N ( n ) = w N ( n 1) + *N ( n ) k N ( n ) =
(56)
= w N ( n 1) + e*N ( n ) k N ( n )

72
In this manner we obtain our Low-Dynamics FTF procedure presented in [1] and [2] and we obtain the
(LD-FTF) adaptive algorithm. It is summarized algorithm as follows.
below.
Initialization of LD-FTF adaptive algorithm
LD-FTF adaptive algorithm
a0 (1) = c0 (1) = 1 , k 0 (1) = 0 , 0 (1) = 1
Predictions
d (1)
Nf ( n ) = a H
N ( n 1) x N +1 ( n ) w1 (1) =
x (1)
eNf ( n ) = N ( n 1) Nf ( n) 2
f
J 0min ( n ) = (1 ) x (1) , x (1) 0
J Nf min (n) = J Nf min ( n 1) + (1 ) Nf (n) eNf * (n)
for n = 2:N+1
J Nf min ( n 1)
N +1 ( n ) = N ( n 1) nf 2 ( n ) = aTn 2 ( n 1) x n 1 ( n )
J Nf min (n)
a n 2 ( n 1)
(1 ) N ( n )
f
k N +1 ( n ) =
0
+ a N ( n 1) a n 1 ( n ) = nf 2 ( n )

k N ( n 1) J N min ( n 1)
f
x (1)

0 f enf 2 ( n ) = n 2 ( n 1) nf 2 ( n )
a N ( n ) = a N ( n 1) * eN ( n )
k N ( n 1)
J nf1min ( n ) = J nf 2 min ( n 1)

Nb ( n ) = b
JN min ( n 1) k N +1, N +1 ( n )
(1 ) J nf 2 min ( n ) = J nf1min ( n ) + (1 ) nf 2 ( n ) enf2 ( n )

N +1 ( n ) J nf1min ( n 1)
N (n) = n 1 ( n ) = n 2 ( n 1)
1N ( ) N +1 ( n ) k *N +1, N +1 ( n )
b n
J nf 2 min ( n )

ebN ( n ) = N ( n ) N ( )
b n
0 (1 ) nf2
k n 1 ( n ) = + an 2 ( n 1)
k n 2 ( n 1) J n 2 min ( n 1)
f
min ( n ) = J N min ( n 1) + (1 ) N ( n ) eN ( n )
b
JN b b b*
if n = N+1
k N ( n )
= k N +1 ( n ) (1 )k N +1, N +1 ( n ) c N ( n 1)
cn 1 ( n ) = ( ) n 1 ( ) n 1 ( )
x 1 n k n
0
1
k *
c N ( n ) = c N ( n 1) ebN ( n ) N J nb1min ( n ) = (1 ) n 1 ( n ) x (1)
2
0
Filtering end

N (n) = d (n) wH n 1 (n ) = d (n ) w nH1 (n 1)x n 1 (n )

N ( n 1) x N ( n )
en 1 (n ) = n 1 (n ) n 1 (n )
eN ( n ) = N ( n ) N ( n )
if n = N+1
w N ( n ) = w N ( n 1) + e*N ( n ) k N ( n )
w n 1 (n ) = w n 1 (n 1) + k n 1 (n )en1 (n )

else
The initialization of the algorithm, i.e. 1 n N + 1
period, is quite complex and requires a lot of paper w n 1 (n 1)
w n (n ) = n (n )
space in order to be deduced. The most common
initialization is for the case when the initial condition
is zero. At time n = N, initialization of both the gain x (1)
vector and the adaptive filter is completed. However,
the forward and backward prediction-error filters are end
both one unit longer. So, their initialization is end
completed at time n = N + 1. We have introduced our
modifications into the standard initialization

73
 IV. SIMULATION RESULTS
For the experimental results we consider a system
identification configuration. In this class of
applications dealing with system identification, an
adaptive filter is used to provide a linear model that
represents the best fit (in some sense) to an unknown
system. The adaptive filter and the unknown system
are driven by the same input. The unknown system
output supplies the desired response for the adaptive
filter. These two signals are used to compute the
estimation error, in order to adjust the filter
coefficients.
In our experiments we compare the classical FTF
algorithm and the proposed LD-FTF algorithm. The
input signal is a random sequence with an uniform
distribution on the interval (1;1). The length of the
adaptive filter is N = 5. The results are presented in
Fig. 1 and Fig. 2, using an exponential weighting
factor = 0.999.
Fig.1. Convergence rate for FTF and LD-FTF
adaptive algorithms
Fig.2. Evolution of adaptive filter coefficients for FTF
and LD-FTF adaptive algorithms
74
It can be noticed that the performances of both
adaptive algorithms are the same. Hence, the LD-FTF
algorithm keeps the fast rate of convergence and
specific to the family of fast LS algorithms.
Moreover, the reduced dynamics of the modified
algorithms parameters could lead to facility for fixed-
point implementation.
V. CONCLUSIONS AND PERSPECTIVES
In this paper we have proposed a modified version of
the FTF adaptive algorithm, named LD-FTF, with low
dynamics of the parameters, as a result of a different
approach of the least squares estimation problem.
The basic idea was to use a modified form for the
algorithms cost functions in order to obtain
asymptotically unbiased estimators for the mean
square errors. In this manner we reduce the dynamic
range of the algorithm parameters, preventing the
unwanted overflow or stalling phenomena which may
appear when such an algorithm is implemented using
fixed-point arithmetic.
The simulation results prove that LD-FTF adaptive
algorithm keeps the fast rate of convergence specific
to the family of fast LS algorithms.
This paper represents only the first step of our
research. Future work will focus on fixed-point DSP
implementation of this algorithm. Also, a careful
analysis of numerical stability of LD-FTF algorithm
could be considered in perspective.
REFERENCES
[1] S. Haykin, Adaptive Filter Theory Fourth Edition. Prentice-
Hall, Inc., Upper Saddle River, N.J., 2002.
[2] S. Ciochin, C. Negrescu, Adaptive Systems, Ed. Tehnic,
Bucharest, 1999.
[3] S. Ciochin, C. Paleologu, A.A. Enescu, On the Behaviour of
RLS Adaptive Algorithm in Fixed-Point Implementation, Proc. of
IEEE Int. Symp. on Signals Circuits and Systems, SCS 2003, Iai,
Romania, 2003, vol. 1, pp. 57-60, .
[4] C. Paleologu, S. Ciochin, A.A. Enescu, A Low Dynamics
Recursive Least-Squares Lattice Adaptive Algorithm, Proc. of
IEEE ICSES04, Poznan, Poland, 2004, pp. 195-198.
[5] C. Paleologu, S. Ciochin, A.A. Enescu, A Network Echo
Canceler Based on a SRF QRD-LSL Adaptive Algorithm
Implemented on Motorola StarCore SC140 DSP, Proc. of IEEE
ICT2004, Fortaleza, Brasil, 2004, LNCS Springer-Verlag, vol.
3124, pp. 560-567.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Gradient Algorithms with Improved Convergence
Cezar Partheniu1
Abstract-A generalized normalized gradient descend
(GNGD) algorithm for linear finite-impulse response is
presented and analized. The GNGD is an extension of the
normalized least mean square (NLMS) algorithm by
means of an additional gradient adaptive term in the
denominator of the learning rate of NLMS. GNGD has
better convergence in linear prediction configuration than
other algorithms, good performances in system
identification configuration in some conditions, worse
response in interferences cancelling configuration and
similar results with NLMS in reverse modelling
configuration.
Keywords: Gradient adaptive learning rate, adaptive filter
configuration, generalized normalized gradient descent
I. INTRODUCTION
The generalized normalized gradient descent
(GNGD) algorithm is an extension of the normalized
least mean square (NLMS) algorithm by means of an
additional gradient adaptive term in the denominator of
the learning rate of NLMS. GNGD adapts its learning
rate according to the dynamics of the input signal with
the additional adaptive term compensating for the
simplifications in the derivation of NLMS. GNGD is
robust to the initialisation of its parameters.
The NLMS is described by the following equations:
y[n] = x H [n]w[n] (1)
e[n] = d *[n] y[n] (2)
[ n] = 2
(3)
x[n] + e
w[n + 1] = w[n] + [n]x[n]e[n] (4)
where e[n] is the error of the output signal, d[n] is the
desired signal, x[n]=[x[n-1],,x[n-n]]H is the input
signal vector, N is the filter length, H is the vector
1
Facultatea de Electronica si Telecomunicatii, Catedra de Comunicatii, Bucuresti, Bd.Iuliu Maniu 1-3, phone: 021.331.18.17,
e-mail: cezarpart@yahoo.com
75
transpose operator, w[n] is the filter coefficients vector
and is the learning rate which defines de convergence
speed of the algorithm on the error surface defined with
cost function
2
e [ n]
E[n] = , (5)
2
a very important parameter for the LMS algorithm.
The usual independence assumptions leed to a
unitar for the fastest convergence. Practically the
NLMS rate is smaller.
The input signals with unknown and variate
dynamics, the ill-conditioned self-correlation matrix
and the correlation between signals may determinate the
divergence or low performances for the NLMS
algorithm. As a solution, new algorithms have recently
been developed [1], [2], [3]. The Mathews and
Benveniste algorithms are presented in the Appendix.
These algorithms are based on the E[n]/ estimators.
A major disadvantage of these algorithms is their
sensitivity to the time correlation between the input
signal samples and to the value of the additional
adaptive rate. To this cause, a generalized normalized
gradient descent (GNGD) algorithm has been
developed. The stability and the improved convergence
are introduced by the gradient adaptive compensation
term e from the denominator of the learning rate of
NLMS.
Due to noise, ill-conditioned correlation matrix,
close-to-zero value of the input vector or a large
learning rate, the NLMS algorithm (6) is not optimal for
many practical settings.
w[n + 1] = w[n] + e[n]x[n] =
2
x[n] +
= w[n] + [n]e[n]x[n]
(6)
 To that cause, the parameter from (6) is made A.Adaptive system configurations
gradient adaptive as
There are four adaptive system configurations
e[n + 1] = e[n] [ n 1]e[n] (7) defined by the function realized.
System identification (Fig. 1). We want to create a
model for an unknown system. This system and the
Using the chain rule, the gradient [ n 1]e[n] can adaptive filter have the same test signal x. The output
signal of the unknown system is the desired signal for
be evaluated as the adaptive filter. When y and d are close, the transfer
function of the unknown system is approached with the
E[n] E[n] [n] y[n] transfer function of the adaptive filter. The dynamics of
= * the system determine a time variability for the model.
[n 1] [n] y[n] w[n]
w[n] [n 1]
* = (8)
[n 1] [n 1]
e[n]e[n 1]x H [n]x[n 1]
= 2
( x[n 1] + [n 1]) 2
Fig. 1 System identification
The GNGD algorithm is therefore described by
Reverse modelling (fig. 2). The model also identifies an
unknown system. When the error is zero,the global
y[n] = x [n]w[n]
H
(9)
transfer function of both unknown system and adaptive
e[n] = d [n] y[n] (10) filter is reduced to a delay. The transfer function of the
adaptive filter is the reverse transfer function of the
w[n + 1] = w[n] + [n]e[n]x[n] (11) unknown system with a small difference caused by the
unavoidable noise. The model can also eliminate the
w[n] = 2
(12) result of an unknown function (eg. Automate
x[n] + [n] equalisation of communications channels).

e[n]e[n 1]x H [n]x[n 1]

e[n] = e[n 1] 2
( x[n 1] + [n 1]) 2
(13)

The adaptive rate of GNGD is essentialy bounded by

the stability limits of the NLMS algorithm. The
compensation term is lower bounded according to Fig 2 Reverse modelling
(14)[4] for =1.
2 Linear prediction (fig. 3). The response of the filter for
x[n] a delayed input sequence is compared with the actual
[ n] > (14) sample. The error minimisation realise an optimal
2 prediction of the input signal. The 1 output realise the
prediction error filter and the 2 output realise the
The complexity of GNGD lies in between the prediction.
complexity of Mathews and Benvenistes algorithms
and is roughly twice that of NLMS. To reduce the
complexity and prevent disturbance in the steady state,
it is possible to impose bounds on [n] or to stop its
adaptation aftrer convergence.

Fig. 3 Linear prediction

76
Interferences cancelling (fig. 4). The primary signal is order of the filter ord and the specific parameters ,
the useful signal. It has an unuseful perturbing signal for RLS, and .
overlapped. There must be created a similar signal
which will be substracted from the primary signal using
a reference. This signal results from the adaptive filter.

Fig.4 Interference cancelling

Fig 5 Convergence of GNGD and NLMS algorithms
II. EXPERIMENTS

The analysis of the adaptive algorithms is made

using all the four adaptive systems configurations: linear
prediction, system identification, interferences
cancelling and reverse modelling.

Linear prediction

The first comparison is made with GNGD and

NLMS algorithms. The order of the filter is ord=5 and
the input sequence has N=3500 samples. The other
parameters of the algorithms have usual values which
made possible the comparison.
The experiment is made using a liniar stationary
filtered signal given by

y[n] = 1.79y[n - 1] - 1.85y[n - 2] + Fig 6. Convergence of GNGD, Mathews and Benvenistes algorithms
(15)
1.27y[n - 3] - 0.41y[n - 4] + x[n]. We set N=1000, ord=7, =0.001 and =0.9. The mean
square error for the algorithms is shown in Fig. 7.
where x[n], a white noise with a zero average and
unitary variance, is passed through a AR filter.
We observe in Fig.5 that GNGD converges faster
than NLMS with 500 iterations. This improved
convergence results from the gradient adaptive in the
denominator of the learning rate of the algorithm.
In [4] we find a comparison between GNGD and
Mathews and Benvenistes algorithms. Using usual
values of the parameters, it is evaluated that GNGD has
faster convergence than Mathews and Benvenistes
algorithms. This result is shown in Fig.6.

System identification

The parameters of LMS, NLMS and GNGD used in

system identification are the number of iterations N, the
Fig 7 Mean square error in system identification(dB)

77
 d[n] = sin(n 0 ), 0 = 0.05 * (16)
The optimization of the parameters results in a
faster convergence for GNGD comparing to NLMS.
and a perturbation with the following recursive relation

v1[n] = 0.8v1[n 1] + g[n] (17)

where g[n] is a white noise with zero average and

unitary variance. The following signal results

x[n] = d [n] + v1[n] (18)

We also consider a second signal v2 [n] defined as

v2 [n] = 0.6v2 [n 1] + g[n] (19)

Fig. 8 Mean square error in system identification for optimized
parameters (dB)

A value of close to 0.1 leeds to the same

performances for all the algorithms studied (Fig. 9).

Fig. 10 System identification for <0.1

The error for N=1000 is made with 100 runs of

independent trials performed and averaged. If we have
no noise, we obtain the graphics in Fig. 11.
Fig. 9 System identification for =0.1

A value of less than 0.1 determines a worse

response of the adaptive filter for GNGD and NLMS
algorithms, the output signals being different (Fig.10).
GNGD prooves to be sensitive to the values of its
parameters.The algorithm has a good output signal for
close to 0.1 and less than 5.5.
For an optimal set of parameters values GNGD has a
response better than NLMS.

Interferences cancelling

The algorithms studied here may be used in

interferences cancelling configuration.
We consider a primary signal given by
Fig. 11. Interferences cancelling (n=0)

78
 A noise factor n=0.5 leeds to an inpossible
interferences cancelling for all the algorithms (Fig. 12).

Fig. 14 Channel adaptive equalizor

The length of the input randomly sequence is

Fig. 12 Interferences cancelling (n=0.5)
A value of =0.1 prooves that GNGD and NLMS
have a similar performance better than LMS.
N=1000 and the order of the filter is ord=5. The figures
represent the input data sequence d[n], the channel
output sequence x[n] and y f [n] i y d [n]
sequences obtained after equalization and decision in
histograms for LMS, NLMS and GNGD algorithms. A
noise makes the equalization impossible.

Fig.13 Interfence cancelling Fig. 15 LMS channel equalization

All the algorithms have good results for filter length

less than 25.

Reverse modelling
The reverse modelling configuration cancels the
results of an unknown transfer function (eg. Automate
equalization of communication channels). For this
configuration we use an adaptive channel equalizor with
a general design as Fig.14 describes.
The input signal has the 1 values randomly
distributed. The channel transfer function H c [z ] is
given by (15). The output signal has a white noise
overlapped and the adaptive filter realizes the
equalization. The switch is used in position 1 with
training sequence. Fig.16 NLMS channel equalization

79
 REFERENCES

[1] A.M. Kuzminskiy, Self-adjustement of an adaptive coefficient of

a noise compensator in a nonstationary process, Iszvestiya VUZ
Radioelektron., vol.29, no.3,pp.103-105,1986
[2] V.J.Mathews and Z.Xie, A stochastic gradient adaptive filter with
gradient adaptive step size, IEEE Trans.Signal Processing, vol.41,
pp.2075-2087, June 1993
[3] A.Benveniste, M.Metivier andP.Priouret, Adaptive algorithms
and stochastic approximation, New York: Springer-Verlag, 1990
[4] D.P.Mandic, A generalized normalizede gradient descent
algorithm, IEEE Signal Processing Letters, vol.11, no.2, February
2004
[5] C.Partheniu, Gradient adaptive algorithms with improved
convergence, diploma project, June 2004

Fig. 17 GNGD channel equalization

As Fig.15-17 shows, NLMS and GNGD have better

performances than LMS in reverse modelling.

III. CONCLUSION

The GNGD algorithm has better convergence than

the other LMS algorithms studied. GNGD has better
performances than NLMS in linear prediction
configuration and similar in interferences cancelling
design. In system identification and reverse modelling
GNGD is sensitive to its parameters, conclusion which
makes GNGD unuseful in these situations.

APPENDIX

Mathews algorithm:
y[n] = x H [n]w[n] (20)

e[n] = d *[n] y[n] (21)

[n] = [n 1] + e[n]e[n 1]
(22)
* x H [n]x[n 1]
w[n + 1] = w[n] + [n]x[n]e[n] (23)

Benvenistes algorithm:

y[n] = x H [n]w[n] (24)

e[n] = d *[n] y[n] (25)

[n] = [n 1] + Re{e[n]x H [n][n]}

(26)
[n] = [ I [n 1]x[n 1]
x H [n 1]][n 1] + e*[n 1]x[n 1]
(27)
w[n + 1] = w[n] + [n]x[n]e[n] (28)

80
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Adaptive Filtering Algorithms

Erwin Szopos1, Norbert Toma, Marina opa
Abstract The paper presents some efficient ek
algorithms for adaptive filtering: Wiener filter, Least yk = sk + nk +
Mean Square (LMS) algorithm, Kalman algorithm. These

(signal + noise) (estimated signal)
are used in several applications such as: echo canceller on -
telephone lines, enclosure noise canceller, adaptive xk Wiener
equalization etc. (noise) Filter N 1

The above-mentioned algorithms were implemented nk = w(i) xk i

using the virtual instrumentation program LabVIEW, and
Simulink, respectively. Experiments were carried out and
their utility, limits and efficiency are demonstrated on
different types of signals (Sine wave, audio signals).
Keywords: Wiener filters, mean square error,
performance area, steepest descent, convergence, Kalman
filter.
1. INTRODUCTION
Adaptive filtering is used when is necessary to
realize, simulate or model a system which
characteristics develop with time. This leads to the
use of time variable coefficients filters. The variations
of the coefficients are defined by an optimization
criterion and are realized according to an adaptive
algorithm. In the literature there are many different
criteria and algorithms. The simplest but the most
important in practice is the case where the criterion of
mean square error minimization is associated with the
gradient algorithm.
While the filtering with constant coefficients is
i =
Fig. 1. Wiener filter structure.
At the k moment, the sample yk contains two
components: the main signal sk and a noise component
nk, which is correlated with xk. The Wiener filter
produces an optimum estimation of nk, named nk . It is
presumed that the Wiener filter is a FIR filter with N
coefficients and the estimated error signal ek is
computed by subtraction of noise estimation nk from
the input signal yk:
N 1
ek = yk nk = yk w(i ) x(k i ) , (1)
i =0
where w(i) are the Wiener filter coefficients. Because
it operates with discrete values, the input signal and
the filter coefficients can be represented in matrix
form:
T

generally associated with frequency domain X k = xk xk -1 xk -(N -1)

. (2)
specifications, the adaptive filtering corresponds to W = [ w(0) w(1) w(N - 1)]
T

time domain specifications and is obvious to use it for

the filter coefficient computation [2].
By substituting these matrices in equation (1), the
2. PRESENTATION OF THE ADAPTIVE estimated error signal will be:
ALGORITHMS
ek = yk W T X k = yk X kT W . (3)
2.1 The Wiener algorithm
The instantaneous biquadratic error of the signal can
The principle of an adaptive filter consists in time be obtained by squaring equation (3):
variation and auto fitting of its characteristics.
Usually, an adaptive filter takes the shape of a FIR ek2 = yk2 2W T ( yk X k ) + W T X k X kT W . (4)
filter structure, with an adaptive algorithm
permanently updating the filter coefficients, when the
The square mean error (SME) is defined by the
error signal is minimized in accordance with a
probabilistic operator of the quadratic error from
criterion.
equation (4). Thus SME can be described:
The Wiener filter structure is shown in figure 1.

1
Technical University of Cluj-Napoca, Str. C-tin. Daicoviciu 15, 400020 Cluj-Napoca, Romania,
Phone-fax: +40 264 591340, E-mail: erwin@bel.utcluj.ro

81
 = E ek2 = E yk2 2W T E [ yk X k ] + W T E X k X kT W . (5)
The SME function can be expressed more suitable by
substituting the term E[XkXkT] from (5) with the
autocorrelation matrix RXX. More, the term E[ykXk]
can be substituted with the intercorrelation matrix
RyX. Thus, the SME can be expressed as:
= E {ek2 } = E { yk2 } 2W T RyX + W T RXX W .
From (6) we can observe that is a quadratic function
of the weights of vector W (filters coefficients). When
equation (6) is expanded, the elements of W will be
only of first and second order. This equation is valid
when the input components and the desired response
are stochastic (random) variables.
2.1.1 Performance area
A part of a bidimensional mean error function is
shown in figure 2. The vertical axis represents the
SME and the other two horizontal axes represent the
values of two coefficients of the filter. The square
error or the performance area can be used to
determine the optimum vector of weights Wopt
(Wiener filter coefficients). A quadratic performance
function allows only a unique optimum global value;
a local minimum does not exist. If the graphical
representation is varying with many coefficients, the
shape of the function will be hyper-parabolic.
The gradient method is used in many adaptive
processes to determine the optimum vector of weights
corresponding to the minimum of the performance
area [6].
Square mean error
Fig. 2. Bidimensional quadratic performance area.
The gradient of the SME of the performance area
can be obtained by derivation of (6) with respect to
each component of the vector of weights:
T
= = . (7)
W w( 0 ) w( 1 ) w( N 1 )
Note that SME was obtained with the probability
operator of the square error (equation (6)). In the same
manner, the gradient can be found by the derivation of
the probabilistic operator of the square error function
with respect to the vector of weights.
e 2 e
= E k = E 2ek k =
W W
= 2E { X k yk } + 2E { X k X kT }W = (8)
(6) = 2RyX + 2RXX W .
When the vector of weights (filter coefficients) has
the optimum value Wopt, the SME will be minimum.
So, the gradient will be zero ( = 0). Equating (8)
with zero results
1
Wopt = RXX RyX . (9)
Equation (9) is known as Wiener-Hopf equation in
matrix form and the filter with coefficients given by
Wopt represents the Wiener filter. For non-stationery
signals Wopt must be computed recurrently which
needs a complex computation. The steepest descent
algorithm represents an iterative solution of the
Wiener-Hopf equation.
2.2 Steepest descent algorithm
In practice, it is not usually to compute the
optimum value Wopt using equation (9) because the
-1
evaluation of RXX [N N] implies the inversion of a
matrix that needs complex computations. More, if the
signals are non-stationery (frequent case), the
computations must be performed periodically to
pursue the changes.
An alternative method to compute the optimum
vector of weights Wopt is represented by the steepest
descent algorithm. In according to this method, the
weights are fitted recurrently with respect to gradient:
W p + 1 = W p p , (10)
where Wp is the weights vector at iteration p, p is the
gradient vector at iteration p computed by substitution
of Wp in (8). Parameter is a constant that fits the size
of the step and controls the stability and the
convergence rate.
2.3 The LMS algorithm
This algorithm is very used due to its simplicity
and for the easy computation. The algorithm is based
on the steepest descent method, but it simplifies this
method considering only one iteration per sample and
computing only one estimation of the gradient vector
) in each moment k.
( k
The estimation of the gradient vector at moment
), can be obtained from the error definition,
k, ( k
(equation (3)):
82
 ek = yk X kT W
= e e e
2 2 2
k
k k
= 2ek X k . (12)
k 2.4.1.1 Process estimation
w(0) w(1) w(N - 1)
This gradient estimation can be replaced in equation
(10) and we obtain:
Wk +1 = Wk + 2 ek X k (13)
or:
wk +1 ( i ) = wk ( i ) + 2 ek xk i , (14)
for i = 0,,N - 1 .
Expression (14) represents the LMS algorithm.
Parameter is a constant that controls the stability
and the convergence rate just like in the case of
steepest descent algorithm. According to equation
(13), the hardware implementation of the LMS
algorithm it can be made simpler, because it does not
need square-, averaging- or derivation operations [6].
2.3.1 Convergence of the LMS algorithm
From the definition equation of the LMS
algorithm (equation (14)), we can observe that the
convergence properties of the algorithm depend on:
step size ; stochastic properties of the input signal x;
N - window length (used number cells).
The step size : there is a time depending on optimum
step opt=(i) with decreasing values for with time
increasing (for example (0)=0.01 ()=0.0001).
In practice it is very important to choose because
that controls the convergence rate. If the value of is
too small, it will need a longer time to converge to
min. If the value of is too large, the algorithm
becomes unstable and it will not converge to min.
Generally, the LMS algorithm will converge if the
following condition is true:
1
0<< , (15)
max
where max is the maximum value from the input
covariance matrix.
The input signal x: can be chosen to suit the properties
of the reference signal yk.
The length N: a large value improves the quality of
the estimations (of the convergence), but also
increases the computation effort.
2.4 The Kalman algorithm
The Kalman filters, named after the scientist
Rudolph E. Kalman, represents a set of mathematical
equations, implementing a type of predictor-corrector
estimator, that is optimum in the meaning of
minimizing the estimated error covariance, when
certain presumed conditions are accomplished.
(11) 2.4.1 Discrete Kalman filter
The Kalman filter tries to estimate a state x n
belonging to a controlled process, time discrete,
described by the finite difference linear equation:
xk = Axk -1 + Buk + wk -1 , (16)
with measurement z R m :
zk = Hxk + vk . (17)
The random variables wk and vk represents the
process noise, respectively the measurements noise. It
presumes that they are white noises, independent one
to another and with normal distributions of
probability:
p( w ) ~ N( 0,Q )
. (18)
p( v ) ~ N( 0,R )
In practice, covariance of the process noise Q and
the covariance of the measurement noise R can be
modified in each moment, but in our case it is
presumed that they are constants.
Matrix A[nn] from (16) describes the process
state from the previous moment k-1 at the current
moment k, in default of driver function or the process
noise. In practice, matrix A can be modified at each
moment, but here it is presumed to be constant.
Matrix B[n1] describes the optional control of input
u l about the state x. Matrix H[mn] from
equation, intended to measure (17), describes the
measurement zk. In practice, matrix H can be modified
at each moment or at each measurement, but here it is
presumed to be constant [4].
2.4.1.2 Filter computation
xk n is defined to be the a priori estimated
state at moment k and xk n the a posteriori
estimated state at moment k having the measurement
zk. It can be defined now the a priori and a posteriori
estimated errors:
ek xk xk and ek xk xk . (19)
Under these conditions, the covariances of the a priori
and the a posteriori estimated error, is:
Pk = E ek ekT (20)
respectively
Pk = E ek ekT . (21)
83
 To obtain the equations for the Kalman filter, we
have to find an equation which computes the a
posteriori estimated state xk as a linear combination
of the a priori estimated state xk and of the weighted
difference between the current measurement zk and
the measurement prediction Hx k :
xk = xk + K( zk Hx k ) . (22)
The term ( zk Hx k ) from equation (22) is called
the measurement innovation or residue. This
difference reflects the discrepancy between the
measurement prediction Hx k and the current
measurement zk. If the residue is zero then the two
quantities are equal.
The matrix K[nm] from (22) represents the gain
or the interference factor and its role is to minimize
the a posteriori estimated error covariance Pk from
(21). This can be achieved by substituting equation
(22) in the definition equation of ek; the obtained error
will be replaced in (21) and the probability operator
will be computed. The obtained result will be derived
with respect to K, equated with zero and solved to
compute K. The most encountered form of the
Kalman gain, which minimizes equation (21), is:
Pk H T
K k = Pk H T ( HPk H T + R )1 = (23)
HPk H T + R
When the covariance of the measurement noise R
tends to zero, the gain K weights better the residue:
lim K k = H 1 . (24)
Rk 0
When the a priori estimated error covariance Pk is
achieving zero, the gain K weights very slightly the
residue:
lim K k = H 1 . (25)
Pk 0
So, when the measurement noise covariance R is
achieving zero, the current measurement zk matters
more and the measurement prediction Hx k matters
less. On the other side, when the a priori estimated
error covariance Pk is achieving zero, the current
measurement zk matters less and the measurement
prediction Hx k matters more.
2.4.1.3 Description of the Kalman algorithm
The Kalman filter estimates a process using a
feedback control: the filter estimates the state of a
process at a moment and then obtains a feedback in
the form of the measurements (in noisy conditions).
Thus, equations of the Kalman filter are divided in
two groups: equations for re-update in time and
equations for re-update of measurement. The
equations for re-update in time are responsible for
time designing of the current state and the estimated
error covariance to obtain the a priori estimations for
the next instant. The equations for re-update of the
measurements are responsible for the realization of
the inverse feedback the inclusion of new
measurement in the a priori estimation to obtain an
improved a posteriori estimation. The equations for
re-update in time are also called prediction equations
and the equations for re-update of measurement are
also called the correction equations. So, the final
estimating algorithm is a predictor-corrector
algorithm, shown in figure 3.
Re-update in time Re-update of measurement
(prediction) (correction)
Fig. 3. Discrete Kalman filter cycle.
The specific time and measurements re-update
equations are presented below:
xk = Axk1 + Buk
(26)
Pk = APk 1 A + Q
T
K k = Pk H T ( HPk H T + R ) 1
xk = xk + K ( zk Hxk ) . (27)
P
k = (1 K k H ) Pk
From (27) we notice that the first step consists in
Kalman gain Kk computation. The next step consists
in measurement updating to obtain zk and generating
the a posteriori estimated state. The final step consists
in obtaining the a posteriori estimated error
covariance.
After each re-update of the time and
measurements pairs, the process is repeated using the
a posteriori estimations to compute the new a priori
estimations. That recurrence is one of the most
interesting property of the Kalman filters, that makes
its practical implementation to be more easier to
realize relative to implementation of the Wiener filter,
designed to work directly on all data for each
estimation [4]. Instead, the Kalman filter computes
recursively the current estimation based on all
measurements performed.
3. IMPLEMENTATION AND SIMULATION
In the last years, LabVIEW and Matlabs Simulink
became the most well known software packages used
in education and industry for modeling and simulation
of dynamic systems.
An example for processing signals using the LMS
algorithm is shown in figure 4. The LMS algorithm
structure with 4 coefficients is shown in figure 5.
84
 A _ n o is e
signal is an audio wave signal on 16 bits, with 8 KHz
S a m p le s frequency, on single channel (mono).
S in + N o is e
111

-1

A _ s in
F ilt e re d s ig n a l
S in

d e lt a
100

Hz

Fig. 4. Signal processing with LMS algorithm

S in

S in + Z g

Fig. 8. Signal processing with adaptive LMS algorithm.

S a m p le s

w _ 0 (n )

w _ 1 (n )

w _ 2 (n )

w _ 3 (n )

d e lta

F ilte r_ O u tp u t

d e lta * e (n )

e (n )

w _ 0 (n + 1 ) w _ 1 (n + 1 ) w _ 2 (n + 1 ) w _ 3 (n + 1 )

a)
Fig. 5. The LMS algorithm structure.

b)

Fig. 6. Input signal; Filtered signal; Desired signal (Sine wave).

The results in figure 6 were obtained with the

following program specifications: the algorithm
contains 4 coefficients; amplitude of sine wave = 10V;
frequency of sine wave = 100Hz; number of samples c)
for sine wave = 111; amplitude of noise (white noise) Fig. 9. a) Audio signal with noise; b) filtered audio
signal; c) desired audio signal.
wave = 10V; step value =510-7.
In figure 7 are shown the results obtained using
the same specifications, but the number of the filter The results of the filtering example (figure 11)
coefficients was increased to 16. It is visible that the with the Kalman algorithm (implemented in
filtered signal (continuous black line) is closer to the LabVIEW figure 10) were obtained under the
desired signal (continuous gray line) than in figure 6. following program specifications: amplitude of sine
wave = 10V; noise amplitude = 10V; sine wave
frequency = 100Hz; samples of sine = 111; Q=510-3;
R=110-2. Figure 10 shows the structure of the
Kalman filter.
In p u t filte r
X_k

O u tp u t filte r

X _ k -1
P _ k -1 + Q + R
P _ k -1 Q

P _ k -1 + Q
R * (P _ k -1 + Q ) P_k

g a in K
K_k

Fig. 7. Input signal; Filtered signal; Desired signal (Sine wave). Z _k

Z _ k - X _ k -1

The results of the LMS algorithm figure 9

(implemented in Simulink figure 8) were obtained
Fig. 10. Kalman filter structure
with the following program specifications: LMS filter
with 16 coefficients; the step =510-2; the input

85

Fig. 11. Input signal; Filtered signal; Desired signal (Sine wave).
In the simulation example with Kalman algorithm
(implemented in Simulink), an input audio signal of 8
KHZ frequency, represented on 16 bits, and Q=110-8;
R=110-2 were considered. The obtained results are
shown in figure 12.
a)
b)
c)
Fig. 12. a) Audio signal with noise; b) filtered
audio signal; c) desired audio signal.
4. CONCLUSIONS
A. General conclusions.
In this paper, several techniques for designing and
implementing adaptive filters were presented. These
techniques were based on the gradient algorithm,
being the simplest and the most efficient instrument
for varying the coefficients.
The gradient algorithm leads to slowly modifying
filter coefficients values, when it requests a reduced
residual error and it is used in the simpler form (the
sign algorithm). To find the most rapid adaptation
rate, all of the coefficients can be re-computed
periodically using rapid iterative procedures.
It is possible to consider some other criteria,
which, for other precision applications, are more
suitable than the minimizing SME criterion, thus,
86
more efficient algorithms than the gradient algorithm
can be developed. The gradient algorithm can be
improved, for example, using different coefficient
step variations, which are obtained from statistical
estimations of the signal characteristics. However, due
to implementation imperfections, applying these
algorithms and sensitivity problems can be more
difficult.
The specific noise cancellation case was already
studied since 4 decades but lately hardware
implementation possibilities of the theoretical systems
with signal processors are loomed.
This paper tries to achieve some important sides
of the adaptive systems in noise cancellation.
B. Conclusions concerning the simulation results.
From simulations, we can observe when the noise
amplitude is growing up, the filtered signal is visible
distorted, that reduces the respectively algorithm
performances. Otherwise, increasing the number of
filter coefficients, the accuracy of the filtered signal
increases.
Another important problem is choosing the
optimum parameters of the adaptive filters. For
example, in case of the LMS algorithm, choosing the
step (), which determines the convergence rate is
critical. If is too large, the algorithm will converge
very rapidly but will present oscillations until stability
limit is reached, or, the effect of inverse error
minimizing - another drawback - appears. If a too
small step is chosen, oscillations will not appear
during the convergence process, but the convergence
speed is slower.
The optimum Wiener filter theory was made for
random stationery processes. When the statistical
properties of the random processes are changing in
time, the above description becomes more difficult.
Due to the permanently modifying of the error surface
of which minimum is to be searched, the adaptive
algorithm must ensure not only the convergence to the
optimum solution, but also to follow the continuous
changing of this optimum value. The Kalman filter
theory that allows a model, for the considered
application, based on state equations gives the
solution. The obtained recursive algorithm is more
rapid than the LMS algorithm and less dependent by
the static characteristics of input data, but presumes
more complex computations.
5. REFERENCES
[1] Adelaida Mateescu, Neculai Dumitriu, Lucian Stanciu,
Semnale i Sisteme. Aplicaii n filtrarea semnalelor, Ed. Teora,
2001;
[2] Maurice Bellanger, Digital Processing of Signals. Theory and
Practice, Ed. John Wiley&Sons, 1984;
[3] J.G. Proakis, Advanced Digital Signal Processing, Ed.
McGraw Hill , 1992;
[4] Greg Welch, Garry Bishop, An Introduction to the Kalman
Filter, Ed.ACM, 2001; (http://www.cs.unc.edu/~welch/kalman);
[5] Yifen Tu, Multiple Reference Active Noise Control, March
1997;
[6] Digital Signal Processing Chapter 7: Adaptive Filtering
(http://www.staff.ncl.ac.uk/oliver.hinton/eee305/Chapter7.pdf);
[7] Mathlab Help. Signal Processing Toolbox, Simulink.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Parameter estimation of the chirp signal

Chioncel Cristian1, Gal Janos2
Abstract The paper presents the characteristics of the
parameters of the chirp signal, and the situations where
this signal intervene. Using Kalman filtering, a few
parameters of the chirp signal will be estimated, such as
frequency and phase. The estimation results and the
impact of changes in the Kalman Filters are analysed
graphically, and finally some conclusions are drawn.
Keywords: Kalman Filter, chirp signal, simulation,
estimation
I. INTRODUCTION
where
xk vector state of the process at time tk, xk=x(tk)
k - matrix that relates xk to xk+1
wk - vector whose elements are white sequences
Bk linear connection matrix between output yk
and state xk
Consider a dynamic process described by an n-th
order difference equation of the form:

The solution of the optimal filter problem is a filter y i +1 = a 0 , y i + ... + a n 1, y i n +1 + u i , i 0 (3)

weighting function that tells us how the past values of
the input should be weighted in order to determine the This difference equation can be re-written as:
present value of the output, the optimal estimate. The
Kalman solution has two main features: ones a vector (4)
modeling of the random processes and a recursive yi +1
y
processing of the noisy measurement data. i
The input data makes part of the common case of xi +1 = y y 1 =
noisy sensor measurements. The time-varying ratio of
the pure signal to the electrical noise affects the ...
quantity and the quality of the information. The result yi n + 2
is that the measured information must be qualified as
it is interpreted as part of an overall sequence of a0 a1 L an 1 y1 1
1 0 L 0 y 1 0
estimates. i
The main feature of Kalman filtering is the recursive = 0 1 L 0 y y 2 + 0 ui
operation mode. The key element in any recursive
procedure is the use of the results of the previous step M M M M ... ..
to aid in obtaining the desired result for the current 0 0 L 0 yi n +1 1
step.
which leads to the state space model form from
II. DISCRET-TIME MODEL equations (1) and (2), with Bk=[1 0 0].
Generally, a continuous process can be described by
Discrete-time processes may arise in two ways: the
situation where a sequence of events takes place . (5)
naturally in discrete steps, with a fixed or random x = Fx + Gu
variable for each step length. Another solution is to
sample a continuous process at discrete time. where u is a vector forcing function whose elements
Irrespective of how the discretization arises, the are white noise.
general format is: One special case of evaluating k is in case of fixed
parameters for the dynamical system (i.e., F is a
x k +1 = k x k + wk (1) constant), the state transition matrix (STM) may be
written as an exponential series:
y k = Bk x k (2)

1
Facultatea de Inginerie, Universitatea E.Murgu, P-a Tr.Vuia 1-4 Reia, e-mail c.chioncel@uem.ro
2
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail gal@etc.utt.ro

87
 k = e Ft = 1 + (Ft ) +
(Ft )2 + ...
(6) after setting the derived equal to zero, is called the
Kalman gain:
2!

where t is the step size. (

K k = Pk H kT H k Pk H kT + R k )1 (13)

III. KALMAN FILTER The covariance matrix, associated with the optimal
estimate, is now:
The Kalman filter is essentially a set of mathematical
equations that implements a predictor-corrector type Pk = (I K k H k )Pk (14)
estimator that minimizes the estimated error
covariance. ^
The random process that has to be estimated can be The update estimates x k can be projected ahead via
modeled in form of equation (1). The process
the transition matrix:
measurement, at discrete points is:
^ ^ (15)
z k = H k xk + vk (7) x k +1 = k x k

where The error covariance matrix associated with the

zk vector measurement at time tk updated estimated has the expression:
Hk matrix given the ideal connection between
the measurement and the state vector at time tk
vk measurement error
[ ]
Pk+1 = E e k +1e k +1 = k Pk kT + Q k (16)

The covariance matrices for the vectors vk and wk are

The table 1 offers a complete picture of the Kalman
given by:
filter operation that processes discrete measurements
[ ]
E wk wiT = Q k (8)
(input) into optimal estimates (the output).

Table 1
E [v v ] = R
T
k i k
(9) Predict Correct
(time update) (measurement update)
The estimation error is defined as the difference (1) Project the state (1) Compute the Kalman
between the state and his a priori estimate (- best ahead gain
estimate):
^
x k +1 = k x k
^
(
K k = Pk H kT H k Pk H kT + R k ) 1

(2) Update estimate with

(10) (2) Project the error
^ measurement zk
e k = x k xk covariance ahead ^ ^ ^
Pk+ 1 = k Pk kT + Qk xk = x k + K k K k H k x k

^
(3) Update the
With the assumption of a prior estimate x k the
covariance error
measurement zk can be used to improve a prior
Pk = (I K k H k )Pk
estimate:

(11) IV CHIRP SIGNAL

^ ^ ^
x k = x k + K k K k H k x k
Chirp signals are encountered in many different
engineering applications including radar, active sonar
^ and passive sonar systems. The main characteristic of
where x k is the update estimate and Kk an blending the chirp signals is the linear change of their
factor. The expression for the error covariance matrix instantaneous frequencies, and therefore they have
associated with the update (a posteriori) estimate is: often been used in representing signals with time
varying spectra. Parameter estimation of chirp signals
(12) has been of great interest in the past, and a wide
^
T
[ ] ^
Pk = E ek ekT = E xk x k xk x k

variety of estimation procedures have been proposed
and studied.

The chirp signal can be formed in two sweep modes.
A unidirectional one, Fig. 1, where the cosine
The expressions (7) and (11) will be substituted in frequency is immediately reset to f(0), the initial
(12). The next step is to minimize the expression of P, frequency, after the sweep period is traversed. When
the sum of the mean-square error (12), differentiating the sweep mode is bi-directional, Fig. 2, the frequency
P with respect to K. This particularly Kk solution, sweep reverses direction half way through the period,
and returns to f(0) along a symmetrical trajectory.

88
 1 d (t ) (20)
f i (t ) =
2 dt

VI. SIMULATIONS MODEL USING KALMAN

FILTERING

We consider the problem of estimating the parameters

Fig.1 Chirp signal with unidirectional sweep
Fig.2 Chirp signal with bi-directional sweep
The frequency sweep of a chirp signal can be cosine,
similar to linear, quadratic, or logarithmic. The linear
frequency sweep uses an instantaneous frequency
( )
f t g f (0 )
sweep f(t), f (t ) = f (0 ) + t , where =
tg
tg target time, f(tg) target frequency. Quadratic
frequency sweep uses an instantaneous frequency
sweep f(t) of f (t ) = f (0 ) + t 2 and the logarithmic
once with the frequency sweep f (t ) = f (0 ) + 10 t ,
of a chirp signal observed in additive noise. This
paper presents the results of the computer simulation
for a signal with constant amplitude and linear
frequency modulation defined through the equations
(13) and (14). The considered values are the
instantaneous frequency at time 0, f0=50 Hz,
instantaneous frequency f1 achieved at time t1=1s is f1
=500Hz, p=1. The signal will be discretized in 21
points. The first plot, figure 3, shows the real phase,
the noise affected phase and the estimated one. The
Kalman filter parameters for this estimation are: fi
=0.98, H=1, Q=1, R=1, X=0, P=1, I=1. The
implemented algorithm follows the steps presented in
Table 1.
,
Fig.3 Phase estimation

where =
[( )
log f t g f (0 ) ]. After the first eight steps, the filter settles down to a
tg steady state condition where the Kalman filter gain
is about 0.8274. Fig. 4 shows the evolution of the
V. INSTANTANEOUS FREQUENCY AND PHASE phase estimation corresponding to the first part of the
Kalman algorithm.
The chirp signal to be dealt with is a linear, quadratic
one, given by

= ( f 1 f 0 )t1 p (17)

1+ p (18)
y = cos 2 t + f 0 t + phi / 360
1 + p
Fig. 4 First part of the phase estimation
where p is the polynomial order and phi de initial
phase. The expression of the instantaneous frequency (17)
The model of the signal that is here considered can be was computed based on equation (16).
expressed as:
2 f (21)
y (t ) = A cos (t ) (19) F= t + 0
1+ p 2
with A constant. The instantaneous frequency, fi(t), of
Using the same Kalman filter parameters like those in
the signal is
the phase estimation, but for a greater number of

89
points, 101, the simulation for the frequency
parameter estimation is shown in Fig. 5.
Fig.5 Frequency estimation
After the first ten steps the Kalman filter stabilizes by
0.618. If we repeat the simulation for other filter
parameters like the state transition matrix (scalar in
this case) k, we can observe a different behavior of
the estimated result. For k=1.45, the phase estimation
will have the following allure, Fig 6.
Fig.6 Phase estimation with k > 1
If we choose a sub unitary value for the transition
matrix k and do not change the time at which the
noisy measurements of this process are taken, t, the
resulted frequency estimation is shown in Fig. 7.
Fig.7 Frequency estimation with k < 1
CONLUSIONS
The plot for the estimation of the phase of the
analyzed chirp signal gives us a parabola, a second-
degree function. We observe that, as time goes by, the
filter depends more on the measurements and less on
the initial assumptions. The estimated frequency has a
linear evolution, but also, the simulation result also
shows that the filter parameters need to be adapted for
an optimal result. If this doesnt happen, then the
90
estimation will follow too much the measurement and
the noise that affects it.
The importance of the correct determination of the
filter parameters k state transition matrix, Hk
measurement relationship to x, the noise sequence Qk
and the measurement error Rk, have been evidenced in
the plots form Fig. 6 and 7. The estimation, after a
higher number of steps, doesnt settle down to an
optimum estimation but increases or decreases from
the actual process x(t).
REFERENCES
[1] R.G. Brown, Y.C. Hwang, Introduction to random signals and
applied Kalman filtering, Second edition, John Wiley &Sons, Inc,
1992
[2] G.Welch, G. Bishop, An Introduction to the Kalman Filter,
ACM, INC, 2001
[3] J.Gal, M.Slgean, M.Bianu, I.Nafornita, The Instantaneous
Frequency Determination for Signals with Polynomial Phase using
Kalman Filtering, Buletinul Universitii Politehnica, Seria
Electronic i Telecomunicaii, Tom 47(61), 2002, Fascicola 1-2.
[4] A. Quinquis, A erbnescu, E,Rdoi, Semnale i sisteme.
Aplicaii n MatLab, Ed. Academiei Tehnice Militare, Bucureti,
1998
[5] MatLab R12 Tutorial
[6] S.Saha, S.Kay, A noniterative Maximum Likelihood parameter
estimator of superimposed chirp signals, Department of Electrical
and Computer Engineering, University of Rhode Island
[7] O.Besson, M.Ghogo, Parameter estimation for random
amplitude chirp signals, IEEE Transactions on signal processing,
Vol.47, No.12, December 1999
[8] Chung-Chieh Lin, Petar M. Djuric, Bayesian estimation of
chirplet signals by MCMC sampling, Department of Electrical and
Computer Engineering, State University of New York at Stony
Brook
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Efficient Implementation of The Second Order Volterra
Filter
Georgeta Budura, Corina Botoca1
Abstract This paper investigates a new implementation
of the second order isotropic filter using the Walsh
Hadamard Transform. The new implementation is used
in a second order Volterra filter. Its performances are
evaluated in a typical nonlinear system identification
application. For the second order adaptive Volterra
filter an LMS adaptive algorithm with variable step size
for the linear and the quadratic parts is proposed.
Experimental results show that by using the WHT, the
computational complexity of the adaptive second order
filter is considerably reduced and the convergence rate
of this filter is also significantly improved. Adaption is
working well even for high levels of the input signal.
The mean-squared error of the proposed filter is
compared with those of a classic second order LMS
adaptive filter.
Keywords: Adaptive nonlinear filter, Walsh Hadamard
transform, LMS adaptation algorithm with variable step
size.
I. INTRODUCTION
The second order Volterra filter has been increasing
research interest in nonlinear filtering techniques. It
has been extensively studied and has been employed
in system identification, channel equalization, echo
cancellation and image processing [1]. A second order
Volterra filter consists mainly of a linear and a
quadratic part described as follows:
y[n] = h0 + H 1 X nT + X n H 2 X nT
where h0 is a constant required to make the output
signal y[n] an unbiased estimate, H1 and H2 are the
linear and quadratic kernels respectively, and Xn is
the input vector of the form:
X n = [x[ n], x[ n 1], K , x[n N + 1]] , where N
represents the filter length or filter memory.
The linear kernel is a 1xN vector, and the quadratic, or
second order, kernel is a NxN matrix:
H 1 = [h1 h2 h3 h4 ]
1
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail getab@etc.utt.ro
h11 h12 h13 h14
h21 h22 h23 h24
H2 = (3)
h31 h32 h33 h34
h41 h42 h43 h414
Many researches have been focused on the
implementation of the quadratic filter, considered a
prototype for the nonlinear filters. In the above
representation we consider the same memory for the
linear and the second order filter. The most general
case would allow a different memory for each
nonlinearity order.
A further simplification can be made to (3) by
considering symmetric Volterra kernels. A second
order Volterra kernel, having the elements h2(n1 ,n2 ),
is symmetric if the indices n1 and n2 can be exchanged
without affecting its value. Any asymmetric Volterra
kernel can be easily symmetrised using the method
given by [2]. So, most authors considered symmetric
Volterra kernels which are in facts symmetric
matrices. If the elements of a second order symmetric
kernel also have the property
hi , j = h N i +1, N j +1 , i, j = 1,2, K N , the kernel is called
isotropic. This paper investigates a new
implementation of the second order isotropic
quadratic filter using the Walsh Hadamard Transform
(1) (WHT). This new implementation is used to construct
an adaptive second order Volterra filter whose
performances are evaluated to a typical nonlinear
system identification application.
II. NEW IMPLEMENTATION FOR THE
QUADRATIC KERNEL
To reduce complexity we consider a second order
isotropic filter with the memory dimension N=4. The
implementation of the second order kernel represents
the major problem for this filter. The new
(2) implementation is based on the Walsh-Hadamard
Transform.
The WHT is considered mainly for two reasons:
91
 (i) The use of orthogonal transform can H H 12
improve the performance of LMS H 21 = 11 (9)
adaptive filters. H 12 H 11
(ii) The WHT is a fast orthogonal transform
which only involves the addition where the size of matrices H11 and H12 is half that of
operation. H 2' . In that case the transformed kernel is a block
The WHT matrix is an NxN matrix (N=2k , diagonal matrix:
k=1,2,3,) usually defined recursively using a block-
1 1 1 a b 0 0
matrix decomposition as follows: W1 =
2 1 1 b c 0 0
H 21W = (10)
1 Wn 1 Wn1 0 0 d e
and Wn = . The WHT matrix is
2 Wn 1 Wn 1 0 0 e f
denoted by W in the following discussion. An
important property of the WHT matrix is given in where the variables a,b,c,d, e and f represent the
eq.4: independent elements. The output of the new second
order filter now becomes:
W W T = 1 (4)
' '
y 2 [ n] = X nW H 2W X nW
T
= X nW H 2W ( X nW )T (11)
For the new implementation we consider the second
order kernel given in eq.3 and the isotropic property: This new implementation raises two problems:
-Is the reduction of the computational complexity
h11 h12 h13 h14 accomplished? We have calculated the number of
operations required for a direct implementation and
h12 h22 h23 h13
H2 = (5) those required for the new implementation. The
h13 h23 h22 h12
results listed in Table 1 show that the new
h14 h13 h12 h11 implementation requires less multiplication
operations.
It can be easily seen that the matrix H2 is symmetric
according to its both diagonals. The nonlinear filter Tabel I
produces the output signal y2[n]: Multiplications Additions
Direct N [(3 / 4) N + 1] (3N 2 + 2 N 4) / 4
implementation
y 2 [ n] = X n H 2 X nT = X nWW H 2WW
T T
X nT =
T
X nW H 2W X nW New [N(N/2)+1] N2/2+Nlog2N-1
implementation
(6)
where: X nW = X nW is the Walsh-Hadamard
transform of the input vector and H 2W = W T H 2W is - Can the TWH of the input vector( X nW = X nW ) be
the WHT of the second order Volterra kernel. substituted by the WHT of the rearranged input vector
If we rearrange the input vector as: '
( X nW = X n' W , X n' = [ x(n) x(n 1) x(n 3) x(n 2)] )?
X n = [x[ n], x[ n 1], x[n 3], x[ n 2]] ,
'
then the We easily find the relationship by examining X nW '

corresponding isotropic kernel H 2' is: and X nW . For example, we have the following
relationships between the two transformed vectors
h11 h12 h14 h13 '
X nW '
= [ X nW '
(1) X nW '
( 2) X nW '
(3) X nW ( 4)] and

H2 =
h12 h22 h13 h23
(7) X nW = [ X nW (1) X nW ( 2) X nW (3) X nW ( 4)] :
h14 h13 h11 h12
'
h13 h23 h12 h22 X nW (1) = X n (1)
'
X nW (2) = X n (4)
We can easily demonstrate that: (12)
'
X nW (3) = X n (3)
X n H 2 X nT = X n' H 2' ( X ' ) Tn (8) '
X nW (4) = X n (2)

This new second order kernel can actually be Similar relationships can be established for the input
decomposed into four sub matrices of the form: vectors of size 8, 16, etc.

92
 III. EXPERIMENTAL RESULTS filter weights so that the system output y 1[n] tracks
the desired signal y[n] . The error signal e[n] is
The new implementation was applied to a typical
nonlinear system modeling problem, as shown in defined as the difference between the desired signal
fig.1. An adaptive second-order Volterra filter is used y[n] and the predicted signal y 1[n] , as indicated in
to identify the nonlinear system having the input- fig.1. The input process x[n] is assumed to be a zero
output characteristic given in fig.2. mean independent sequence with covariance, a
positive definite matrix. This simplifying assumption
is often made in literature [1,3].
y[n] The LMS type adaptive algorithm is a gradient search
nonlinear
system algorithm which computes a set of filter weights
+ H 1[ k + 1], H 2 [ k + 1] , at the time moment k+1, that
x[n] e[n]
+ seeks to minimize the error function, E[(e[k ]) 2 ] ,
cosidered at the time moment k.
-
The update equations for the Volterra adaptive filter
1
y [n] weights are well known in the literature[3] and are
adaptive Volterra given in eq. 15:
filter

H 1 [k + 1] = H 1[ k ] + 21e[k ] X [k ]
(15 )
lms H 2 [ k + 1] = H 2 [ k ] + 2 e[k ] X T [k ] * X [ k ]
adaptive
algorithm
where 1 and 2 are in both cases two small positive
Fig1.Nonlinear system identification using adaptive constants (referred to as the step size) that determine
quadratic filter the speed of convergence and also affect the final
error of the filter output.
The nonlinear system output is given in Eq.13. In our simulation we have used the efficient
implementation for the quadratic kernel. In this case
y[n] = A * X n + X nT * B * X n (13) the update equation for that kernel is:

where A = [0.78 1.48 1.39 0.04] and H 2W [ k + 1] = H 2W [k ] + 2 ( X W' [k ]) T * X W' [ k ] (16)

0.54 3.72 1.86 0.76 where H 2W and X W" are the WHT of the quadratic
3.72 1.62 0.76 1.86 kernel respectively the WHT of the rearranged input
B= (14) vector.
1.86 0.76 1.62 3.72
Finally the output of the adaptive Volterra filter,
0.76 1.86 3.72 0.54 y 1[ n] , is:
This system is a slight modification of that used in
[1]. y 1[ n] = H 1 * X n + ( X nW
'
) T * H 2W * X nW
T
(17)
For a linear input signal x[n], the resulted output
signal is plotted in Fig.2. The linear kernel is a 1x4 vector and the quadratic
kernel is a 4x4 matrix. The input sequence is a
random gaussian zero-mean sequence having 1500
values.
The majority of papers examine the LMS algorithm
with a constant step size. The choice of the step size
reflects a tradeoff between misadjustement and the
speed of adaptation. The approximate expressions
derived in [3] showed that a small step size causes
small misadjustement, but also a longer convergence
time constant.
For adaptive Volterra filters the problems seem to be
much more complicated. In [3,5] the problems of step
size for different order kernels are well discussed.
Fig. 2 Input-output characteristic of the nonlinear system
The maximum step size bound is related to the
maximum eigenvalue of the autocorrelation matrix of
The adaptive filtering or system identification the input vector. Because we consider a second order
problem being considered is to try to adjust the set of Volterra filter without DC component included in the

93
estimation algorithm, the step size bounds for 1 and
2 are those given in [3]:
2 2
0 < 1 < ; 0 < 2 < (18)
3tr[ R XX ] 3(tr[ R XX ]) 2

In [6], a variable step size LMS algorithm for linear

filter is proposed. The step size adjustment is
controlled by the square of the prediction error. The
motivation is that a large prediction error will cause
the step size to increase to provide faster tracking
while a small prediction error will result in a decrease
in step size to yield smaller misadjustements.
In this paper we have used this variable step size
algorithm for both, the linear and the quadratic filter. Fig.3 The adaptive filter error with white input signal
The relation for adjusting the step size 1 is that
given in [6]:

1[k + 1] = 1[k ] + e 2 [k ] (19)

with 0 < < 1 and > 0

and

1 max if 1 [k + 1] > 1 max

1[k + 1] = 1 min if 1 [k + 1] < 1 min (20)
[k + 1] otherwise
1

where 0 < min < max . The initial step size 1[0] Fig.4 The adaptive variable step size filter error with white
input signal
was chosen to be 1 max although the algorithm is not
sensitive to this choice. As can be seen from eq.20,
the step size 1 is always positive and is controlled by
the size of the prediction error and the parameters
and . A large prediction error increases the step
size to provide faster tracking. If the prediction error
decreases, the step size will be decreased to reduce the
misadjustements. The constant 1 max is chosen to
ensure that the mean-square error(mse) of the
algorithm remains bounded . A sufficient condition
for 1 max to guarantee bounded mse is:

2
1 max < (21)
3tr ( R XX ) Fig.5 The adaptive variable step size filter error with
colored input signal
Usually 1 min will be near the value that would be
The colored input signal was generated as specified in
chosen for the fixed step size algorithm. In our eq.22.
simulation the value is 1 min = 10 7 .
Parameter must be chosen in the range (0,1) to x[n] = 0.25 * r[n 1] + r[n 2] + 0.25 * r[n 3] (22)
provide exponential forgetting. A typical value of
that was found to work well in simulations is where r[n] is a random, normal distributed sequence.
= 0.97 . The parameter is usually small In Fig.6 we have compared the mean-squared error of
the proposed filter (represented with solid line ) with
( 4,8 * 10 4 was used in our simulations.) those of a classic second order LMS adaptive
filter(represented with doted line ) . We also consider
the case of the colored input signal for the new filter
(represented with dashed line).

94
Fig.6 The mean-squared errors for the compared filters

For high level input signal the filter with variable step size
still adapts, as can be seen in fig.7.

Fig.7. The adaptive variable step size filter error with high
level input signal

IV. CONCLUSIONS

We have proposed a new implementation of the isotropic

second order filter. This new implementation has two
advantages: it requires less operations than the direct
implementation and it has better performances in modelling
a nonlinear system. We have also proposed a variable step
size algorithm which improves the capabilities of the
adaptive Volterra filter.

REFERENCES
[1] V.J. Mathews, Adaptive Polinomial Filters, IEEE Signal
Processing Magazine, No.7, pp. 10-23, July, 1991.
[2] M. Schetzen, The Volterra and Wiener theories of nonlinear
systems, Wiley and Sons, New York, 1980.
[3] J. Tsimbinos, Identification And Compensation Of Nonlinear
Distortion, Thesis, http://www.unisa.edu.au/html
[4] G.Budura, Contributions to the nonlinear systems study using
Volterra series, Thesis, Politehnica University of Timisoara, 30
Sept.,1999.
[5] G.Budura, C. Botoca, Applications of the Volterra Models in
Nonlinear Systems Identification, Buletinul Universitatii
Politehnica, Seria Electrotehnica, Electronica si Telecomunicatii,
Tom 47(61),Fascicola 1-2, pp.196-201, 2002.
[6] R.H. Kwong, E.W. Johnston, A Variable Step Size LMS
Algorithm, IEEE Transactions on Signal Processing, Vol.40, No.7,
pp.1633-1642.

95
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Phase approximation using signals affected by random
perturbations
Lcrimioara Grama1
Abstract
The goal of this paper is to give a comparison between
two methods for phase approximations: non-compact
gain technique for linear frequency domain and the
approach based on logarithmic sampling of gain for
logarithmic frequency domain, using signals affected by
random perturbations. A comparison of the behavior of
these algorithms, considering signals affected by
perturbations, respectively signals that are not affected
by perturbations, will be also presented. For this
purpose we first recall Hilbert transform and Bode
relationships, then the two methods will be discussed.
Numerical examples are provided to emphasize the
advantages and disadvantages of each method and
computer simulations performed using Matlab are also
presented.
Keywords: phase approximation, logarithmic sampling,
linear sampling, Hilbert transform, Bayard-Bode
relationships.
1. INTRODUCTION
The non-compact gain technique cannot be employed
when the gain characteristic has slopes different from
zero at zero and at high frequency. Since the Bode
transfer functions do not satisfy the last requirement,
in order to overpass this inconvenient, we have
proposed the modified Bode transfer functions to be
used in implementation.
We will present three cases of how signals are
affected by random perturbations:
1. signal affected by a complex random
perturbation;
2. system function affected by a real random
perturbation;
3. parameters of the Bode transfer function,
respectively of the modified Bode transfer
function affected by a real random perturbation.
The proposed methods are then tested on some
numerical examples. Our analysis will be
concentrated on minimum-phase functions, since the
1
Department of Electronics and Telecommunications, Technical
University of Cluj-Napoca, Str. George Baritiu 26-28, RO-400027,
Cluj-Napoca, Romania, e-mail Lacrimioara.Grama@bel.utcluj.ro
96
experimental results can be very easily applied to non
minimum-phase functions [6].
Hilbert transform and Bayard-Bode relationships [1]
have been recognized as very important methods in
circuit theory, communications and control science.
Their sampled derivations have been encountered in
different applications from science and engineering.
In some situations the domain is restricted or other
explicit conditions are imposed. A critical issue is
related to the singularities involved in the Hilbert
transform computation, since we are confronted with
an improper integral (Section 2). If the integral cannot
be evaluated in a closed form, as it is the case with
discrete input data, numerical implementation is in
general complicated [2], as localized errors should
lead to localized errors. Hilbert transform has the
advantage of not requiring derivatives, but the serious
disadvantage that it is not a bounded operator from Lr
to L . To solve the problem, different approaches for
gain-phase relationships in logarithmic frequency
domain have been proposed. A suitable change of
variable can give the bounded operator (5) from Lr to
L for any r 1 [3].
The goal of this paper is to give a comparison
between linear and logaritmic frequency domain
phase approximations, using as test signals, signals
that are affected by random perturbations. There is
also presented a comparison of these algorithms,
considering signals that are affected by perturbations,
respectively signals that are not affected by
perturbations.
The paper is organised as follows. In Section 2 we
shortly remind Hilbert transform and Bode
relationships. First we will discuss the logarithmic
sampling of gain approach in Section 3, then in
Section 4 the non-compact gain technique [4] is
addressed. Furthermore, we derive the modified Bode
transfer functions (Section 5) to be used in
implementation. Finally it results a comparison based
on numerical examples (Section 6) and we will drag
conclusions (Section 7).
 2. BAYARD-BODE RELATIONSHIPS AND
( )
pN
p [ ( p ) ( p )],
(6)
HILBERT TRANSFORM >1

The Bayard-Bode relations method is based on the Using quadrature formulae, several approximations
fact that the transform results. Here we shall consider for study that one
derived from Simpson approach (the parabolic rule):
H ( j ) = R ( ) + jI ( ), (1)
1
of a causal function h(t) is uniquely determined in S ( ) = () +

terms of R() or I() (subject to an arbitrary
reactance value if determined from R() and to an 2 ln ( ) ( 1 )
+ +
arbitrary real value, if determined from I()) [1]. 3 1
Proofs based on Cauchys residue theorem [13] or on
convolution [6] establish ( 2 ) ( 2 ) ( 3 ) ( 3 )
+4 2
+2 + (7)

2
3 3
1

I ( y) ( k 1 ) (1 k )
R( ) = R()


y
dy = + + 4
k 1 1 k
+
(2)
2

yI ( y ) I ( ) ( k ) ( k )
+2
= R ( )

0 y2 2
dy k k

or
1 R( y ) 2 R ( y ) R( )
I ( ) =
y
dy =
y2 2
dy (3)
0 S ( ) = S p ( p ),
pZ
One can easily obtain the gain-phase relationships (or
the Bayard-Bode relations) from (2) and (3) directly
by taking logarithms [6], after fulfilling the 2 ln
1 1 + 3 , p = 1
requirements needed to satisfy the right half plane 1
analyticity conditions of the Hilbert transform, i.e. the
stable and minimum phase conditions. Under the 8ln
, p = 2, 2m
3 ( )
assumption that H ( s ) is not only analytic, but has no p p

zeros for Re( s ) 0 , then:

4 ln
S p = S p = , p = 3, ( 2m 1)
3 ( )
p p
ln( H ( j )) = ( ) + j ( ) (4)

2 ln
, p = ( 2m + 1) (8)
will be also analytical in the right-hand plane. Thus 3 ( p p )
the phase ( ) (in nepers), using a change of
0, otherwise
variable u = ln ( y / c ) , where c is a normalizing
frequency, will be:

2 ( y ) ( )
0 y 2 2
( ) = dy = 4. PHASE APPROXIMATION IN LINEAR
FREQUENCY DOMAIN
2

( c eu ) ( c )

eu e u
du = (5)
The formula between the imaginary and real parts of a

complex function of real frequency as expressed in
1 d u |u| equation (3) can be rewritten in many ways [4]. By
= ( c e ) ln coth du
0 du 2 integrating the right member of (3) by parts one can
find:

3. PHASE APPROXIMATION IN LOGARITHMIC 1 y +
FREQUENCY DOMAIN
I ( ) =
R '( y) ln
y
dy (9)

The scope is to find a phase approximation from the

provided
gain samples, given at equally spaced points on the
logarithmic frequency domain:

97
 R( y ) (v) = (v + 1) ln | v + 1 | + (v 1) ln | v 1| 2v ln | v |
lim =0 (10)
y y
Remarks:
Alternatively, we can continue by integrating the right 1. The an numbers are determined by a broken-
member of (9) by parts, i.e. a double integration by line approximation to the gain-versus-
parts of the right member of (3) and the integrand will arithmetic-frequency characteristic.
be: 2. This procedure cannot be employed when the
gain characteristic has slopes different from
y + zero at zero and at high frequency.
R ''( y ) 2 ln 2 y 2 y ln (11) 3. The non-compact support gain method can
y
be easily extended to broken-parabolic (or
higher order curve approximation.
provided

R( y )
lim =0 (12) 5. MODIFIED BODE TRANSFER FUNCTIONS
y y
Previous attempts to test the phase approximations
and approaches have used the Bode transfer functions [1]

lim R '( y ) y < (13) 1| 1| 1| 1|

y H ( s) = + 2
+ + =
| s | sK | s / H |1
Previous relationships are seldom integrated 1 (17)
= =
analytically and in practice it is customary to use 1
approximations to find the relationship between phase s+
1
and gain. An idea is to use straight-line segments so sK 2 +
s / H +1
that the second derivative ''( y ) is a set of impulses
[2]. Gain functions will satisfy the following: The magnitude of the frequency response | H ( j ) | is
Second derivative consists of groups of 2 given by:
impulses;
Each group has a positive impulse at the
origin and a negative impulse at a frequency ( K 2 2 ) 2 + ( K 2 H ) 2
n ; ( K 2 H 2 + H ) 2 + [ K 2 3 ( H + 1) ]2
Only positive n n need to be considered.
and the gain ( ) has slopes different from zero at
_

Thus the second derivative of the gain is given as

high frequency.
''( y ) an [ ( y ) ( y n ) ] (14) We shall slightly modify the Bode transfer functions
n as follows:

It follows succesivelly that2: As + B

H ( s) = (18)
1
(3)
1

( y) 1 (9)
y + (10) s+
1
( ) =
y
dy = '( y ) ln
y
dy = sK 2 +
s / H +1


1 y +
''( y) 2 ln y 2 y ln = (15)
2
and now we are looking what requirements should

y satisfy the parameters A , B , K and H such that
= a ( ) the gain has zero slopes at zero and at high frequency.
We have the following expresions for H ( s ) , H ( j )
Finally we have: and | H ( j ) | respectively:

1 AK 2 s 3 + ( AH + B ) K 2 s 2 + ( A + BK 2 ) Hs + BH
( ) a ( ) =

a n n (16)
n n K 2 s 3 + K 2 Hs 2 + ( H + 1) s + H
[ BH ( AH + B ) K 2 2 ] + j[( A + BK 2 ) H AK 2 3 ]
where
( H K 2 H 2 ) + j[( H + 1) K 2 3 ]

2
An extended form of ( ) can be found in [11]

98
 [ BH ( AH + B ) K 2 2 ]2 + [( A + BK 2 ) H AK 2 3 ]2 K test = K + noise _ real (24)

( H K 2 H 2 ) 2 + [( H + 1) K 2 3 ]2
6. SIMULATIONS
Thus
Now we are going to compare the given approaches.
1 1
( ) = ln U ( ) ln V ( ) , A. Logarithmic Frequency Domain
2 2
For logarithmic frequency domain, the selected
where transfer function is:
1
U ( ) = [ BH ( AH + B) K 2 2 ]2 + H ( s) = (25)
1
+[( A + BK 2 ) H AK 2 3 ]2 ; s+
1
4s +
s
+1
V ( ) = ( H K 2 H 2 ) 2 + [( H + 1) K 2 3 ]2 2
where we used the Bode transfer function (17),
considering K = H = 2 . The phase of the selected
The gain slope is given by3:
transfer function (i) is almost constant for < 0.01
U '( )V ( ) V '( )U ( ) and > 10 [1], consequently the interval of interest
'( ) = = in our experiments was [ 0.01,10] . We select
2U ( )V ( )
(19)
(2 A2 + B 2 K 2 ) 9 + + ( ) H 2 = 2 as sample ratio. Three plots are shown for
= different number of samples: k = 5 (ii), k = 9 (iii)
2 A2 K 8 12 + + B 2 H 4
and k = 17 (iv).
Now, If noise is present, then it can affect the quality of
1. From lim '( ) = 0 , we need phase approximation. The phase (v) and phase

approximations for different number of samples:
A2 K 4 K 4 0 ;
k = 5 (vi), k = 9 (vii) and k = 17 (viii) using as test
2. From lim '( ) = 0 , it follows
0 signal one that is affected by random perturbations are
H 2 B2 H 2 0 . also ploted.
Consequently, the modified Bode transfer functions 1. signal affected by a complex random perturbation
should satisfy the requirements:

A B K H 0 (20)

To see the behaviour of the phase approximation

algorithms under random perturbation conditions in
the logarithmic frequency domain, we have three
cases:
1. signal affected by a complex random
perturbation

H test ( j ) = H ( j ) + noise _ complex (21)

2. system function affected by a real random

perturbation

| H test ( j ) |=| H ( j ) | + noise _ real (22)

3. parameters of the Bode transfer function,
respectively of the modified Bode transfer
function affected by a real random
perturbation
H test = H + noise _ real
Fig. 1. Phase (i) and phase approximation (ii), (iii), (iv) for the
transfer function (25); phase (v) and phase approximation (vi), (vii),
(viii) using signal affected by a complex random perturbation
2. system function affected by a real random
perturbation
(23)

3
An extended form of ( ) can be found in [8]

99
 frequencies varying from 0 to 10. Outside this
interval, both gain and phase of the transfer function
do not exhibit important variations. The phase and the
approximated phase are also shown (ii). If noise is
present, then it can affect the quality of phase
approximation. The gain (iii) and phase
approximation (iv) using as test signal one that is
affected by random perturbations are also ploted.
1. signal affected by a complex random perturbation

Fig. 2. Phase (i) and phase approximation (ii), (iii), (iv) for the
transfer function (25); phase (v) and phase approximation (vi), (vii),
(viii) using system function affected by a real random perturbation

3. parameters of the Bode transfer function affected

by a real random perturbation

Fig. 4. Gain (-) and gain samples (*) (i) used in linear
approximation of gain for (26); phase (-) and phase approximations
(.) with this linear approximation of gain (ii); gain (-) and gain
samples (*) (iii), respectively phase (-) and phase approximations
(.) (iv) for signals affected by complex random perturbations

2. system function affected by a real random

perturbation

Fig. 3. Phase (i) and phase approximation (ii), (iii), (iv) for the
transfer function (25); phase (v) and phase approximation (vi), (vii),
(viii) using parameters affected by a real random perturbation

B. Linear Frequency Domain

For linear frequency domain, the selected transfer
function is:
s +1
H ( s) = (26)
1
s+
1
s+
s +1
Fig. 5. Gain (-) and gain samples (*) (i) used in linear
where we used the modified Bode transfer function approximation of gain for (26); phase (-) and phase approximations
(18), considering A = B = K = H = 1 . Relation (26) (.) with this linear approximation of gain (ii); gain (-) and gain
samples (*) (iii), respectively phase (-) and phase approximations
respects the requirement of initial and final slopes (.) (iv) for signals affected by real random perturbations
given by (20). The gain of the selected transfer
function together with its piecewise-linear 3. parameters of the modified Bode transfer function
aproximation (i) are shown in next figures, for affected by a real random perturbation

100
 REFERENCES
[1] H. W. Bode, Network analysis and feedback amplifier design,
D. Van Nostrand, Princeton, NJ, 1945.
[2] B. D. O. Anderson and M. Green, Hilbert transform and
gain/phase error bounds for rational functions, IEEE Transactions
on Circuits and Systems, vol. 35, no. 5, pp. 528535, May 1988.
[3] M. Green and B. D. O. Anderson, On the continuity of the
Wiener-Hopf factorization operation, J. Austral. Math. Soc. Ser. B,
vol. 28, pp. 443461, 1987.
[4] G. C. Newton, L. A. Gould, and J. F. Kaiser, Analytical
design of linear feedback controls, John Wiley & Sons, Inc.,
London, 1957.
[5] C. Rusu, P. Kuosmanen, and A. Burian, 1-D non-minimum
phase retrieval by gain sampling, in Proc. ECCTD99, Stresa,
Italy, Sept. 1999, vol. 2, pp. 755758.
[6] A. Papoulis, The Fourier integral and its applications,
McGraw-Hill, 1962.
[7] Corneliu Rusu and Pauli Kuosmanen, Phase approximation
by logarithmic sampling of gain, IEEE Transactions on Circuits
and Systems II: Analog and Digital Signal Processing, vol. 50, no.
2, pp. 93101, Feb. 2003.
[8] Lacrimioara Buzan, Phase approximation by gain samples,
M.S. Thesis, Technical University of Cluj-Napoca, 2003.
Fig. 6. Gain (-) and gain samples (*) (i) used in linear
[9] Peter Henrici, Applied and Computational Complex Analysis,
approximation of gain for (26); phase (-) and phase approximations
vol. I: Power Series - Integration - Conformal Mapping - Location
(.) with this linear approximation of gain (ii); gain (-) and gain
of Zeros, John Wiley & Sons, New-York, 1974.
samples (*) (iii), respectively phase (-) and phase approximations
[10] A. Papoulis, Signal analysis, McGraw-Hill, 1977.
(.) (iv) for signals whose parameters are affected by real random
[11] Lacrimioara Buzan, Phase Approximation in Linear and
perturbations
Logaritmic Frequency Domain, M.S. Thesis, Technical University
of Cluj-Napoca, 2004.
[12] Lacrimioara Buzan, Corneliu Rusu, Radu Ciprian Bilcu, Pauli
7. CONCLUSIONS Kuosmanen, Phase Approximation in Linear and Logarithmic
Frequency Domain, Proceedings of the First International
Symposium on Control, Communications and Signal Processing,
We have presented two methods for phase Hammamet, Tunisia, 21-24 March 2004, pp.709-712, ISBN: 0-
approximations: non-compact gain technique for 7803-8380-X
linear frequency domain and the approach based on [13] L. A. Zadeh and C. A. Desoer, Linear System Theory,
McGraw-Hill, New York, 1969.
logarithmic sampling of gain for logarithmic
frequency domain, using signals affected by random
perturbations. A comparison of the behavior of these
algorithms, considering signals affected by
perturbations, respectively signals that are not
affected by perturbations, was also presented.
Unlike the non-compact gain technique where we
need only the gain samples, the logarithmic sampling
of gain requests for two parameters: which
describes the frequency sampling, and k used to
provide a satisfactory approximation of the integral in
(7). There is a trade-off between and k [7].
Indeed, as k increases, has to tend to one more
rapidly. For this reason we have considered = 2m ( q ) ,
with m(q ) = 22 q for q = 1, ,9 . The gain samples
for non-compact gain technique were available by
sampling with the frequency interval
[ 0.01,10] .
From the experimental results we can conclude that
best achievements can be obtained using the linear
frequency domain approximation, unless when the
number of gain samples is low, where the logarithmic
sampling of gain is superior, but we can also see that
both methods for phase approximation behave well
when we have considered as test signals, signals
affected by random perturbations.

101
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Reconstruction Methods for Missing Portions in Signals

Lucian Stanciu1, Leonard Banu
Abstract On the storage medium the information can N
be lost from a portion of a signal. The missing or
disturbed portion of the audio signal is replaced by a
x n' = h
k =1
'
k xnk , (5)
weighting average of the forward and backward
The backward extrapolation equation is
extrapolated signals. A new weighting function is
N

h
proposed for the signal reconstruction method. A
method for the detection of the impulsive noise, using x n'' = ''
k xn+k (6)
two thresholds, is presented. k =1
Keywords: extrapolation, disturbance, reconstruction The impulse response vectors h and h are
obtained by Burg method [1]. The forward and
I. INTRODUCTION backward extrapolated signal is given by
xn = wn xn' + (1 wn )xn'' . (7)
There are many situations when the long portions
in the recorded audio signals are removed or affected where wn is a weighting function
by the impulsive noise. A method for the restoration 1 1
1 (2u )a, un n ns
based on the separation of autoregressive processes is 2 n 2 and un = .
proposed. The corrupted samples are replaced by a wn =
1
(22u )a, u > 1 ne ns
weighted average of the signals extrapolated from 2
n n
2
areas preceding and corrupted area. For the detection An other proposed weighting function is:
of the impulsive noise a method with two thresholds
is used. wn1 = 2u n 3 3u n 2 + 1 (8)
This function is obtained using the polynomial:
II. RESTORATION METHOD FOR LONG g ( x) = ax 4 + bx 3 + cx 2 + dx + e
PORTIONS IN AUDIO SIGNAL
with the conditions: g (0) = 1, g (1) = 0, g ' (0) = 1,
The impulse response function is obtained from g ' (1) = 0 . The solution is : a = 0 , b = 2 , c = 3 ,
M = 2 N samples of the known signal by using the
d = 0 and e = 1 .
equation:
In Fig.1 the original and the bilateral extrapolated
signal are given for 1000 extrapolated samples using
Xh'= x (1)
N=1000 coefficients of extrapolation.
where The general weighting functions wn, 1-wn, for
h' = [h1 ' , h2 ' ,....., hN ']T , (2) a=3, and wn1, 1-wn1 are illustrated in Fig. 2.
and The relation for signal to noise ratio is:
W 1
x = [ x N +1 , x N + 2 ,......, x2 N ]T . (3)
x ' 2
n
The matrix X contains the shifted samples of the RSZ = 10 lg W 1 n=0 (9)
signal:
xN xN 1 xN 2 L x1 (x n xn' ) 2
n =0
x xN xN 1 L x2 where xn is the original signal and the term x n x n'
X = N +1 . (4)
M M M M represents the error of extrapolation. The mean square

x L xN
2 N 1 x2 N 2 x2 N 3 error (MSE) between the estimate and the desired
signal is given by:
The one-step forward extrapolation equation is:

1
University Politehnica of Bucharest, Department of Telelcommunications,
Str. Iuliu Maniu 1-3, Bucharest, Phone: 0214024815, e-mail: lucians@comm.pub.ro

102
 W 1
1
MSE =
2W (x
n =0
n xn' ) 2 (10)

In Fig. 3 the values of RSZ and MSE are plotted

as functions of number of impulse response
coefficients N, by using comparatively the weighting
functions wn and wn1. The original and the bilateral
extrapolated signal are given for 1000 extrapolated
samples. The length of the impulse response is varied
from 100 to 2000.
The spectral distortion is used to evaluate the
quality of the restoration and is defined as:
N sd
1
(10 log X (k )
2 2
M sd [dB ] = [ 10 log X (k ) ]1 / 2
N sd k =1 a)
(11)
where Nsd is the length of the discrete Fourier
transforms of the reference and the processed signals,
respectively.

b)
Fig.3. RSZ and MSE as functions of number of impulse response
coefficients N, by using comparatively the weighting functions wn
and wn1 [a) signal 1; b) signal 2)]

The illustrations from Fig. 5 justifies the choice

of the parameter a.
Fig.1. Original signal and bilateral extrapolated signal for N=1000 Fig. 6 presents the parameters RSZ, MSE as
functions of bilateral extrapolation length. The quality
is very high at the beginning and at the end of the
extrapolated section.

Fig.2. Weighting functions wn , 1- wn for a=3 and wn1, 1- wn1.

a)
Fig. 4 illustrates the variation of Msd as function
of number of impulse response coefficients N, for the Fig. 4. (to be continued)
signal 1 [a)] and signal 2 [b)]. For the signal 2 we
have better restoration results.

103
 b)
Fig.4. Msd as function of number of impulse response coefficients Fig.6. RSZ and MSE as functions of extrapolation length
N, for the signal 1 [a)] and the signal 2 [b)]. measured for last 500 extrapolated samples, for a bidirectional
extrapolation
III. CANCELLATION OF THE IMPULSIVE NOISE e(k ) with the amplitude greater than the threshold
which is given by the relation:
For the classical method, the audio signal affected
by perturbations, y (k ) , is segmented in frames of N = K e (14)
samples. Every frame is modeled as an autoregressive where e is the estimated value of the variance for the
process (AR) of order p: signal e(k ) . The affected samples will be replaced by
y (k ) = x(k ) + d (k ), an interpolation algorithm.
where x(k) is affected by the additive impulsive noise,
d(k), and
p
x(k ) = a ( j ) x ( k j ) + e( k ) ,
j =1
k = p, N 1. (12)

Fig. 7. a) Corrupted signal; b) Input signal; c) Restored signal.

This method of detection can be improved by

using two thresholds, by unifying adjacent
perturbations and by changing the detection threshold
Fig.5. The choice of the parameter a.
in the reprocessing of the signal in a frame.
The two thresholds are: the detection threshold
In the relation (12) a ( j ) are the model parameters
(the initial threshold) D and the location threshold L
and e(k ) is the input signal associated with x(k ) . The which are in the relation:
linear prediction cant model correctly signals with L = bD , 0 < b < 1 (15)
very rapid variations and so the detection of the The threshold L is used to localize the perturbations,
impulsive noise will be made (Fig. 7). Its necessary as is shown in Fig. 8.
to estimate the parameters of the autoregressive model To reject the wrong time interval for the small
of the signal and then to pass the perturbated signal amplitude noise perturbations, the location threshold
y(n) through the error filter of the forward prediction is expressed as a function of the current iteration
with the transfer function: number:
p
Li = bi D , i = 1,2,L, imax (16)
I ( z) = 1 a( j ) z
j =1
j
(13)
where
i 1
The samples of the signal y(n) which f
corresponds to the samples of the signal bi = r b1 (17)

104
In the relations (14) and (15) r is the reduction factor, real-time applications. The information in the
f is a parameter that controls the reduction speed of b preceding and in the following data sections can be
and i max is the maximum number of iterations to pass used to recover the lost information. This recovering
to the next step (Fig. 9). The choice of the parameters is not perfect. The parameters RSZ, MSE, Msd and its
imax , f and r is based on the experimental proposed graphical representations, for different
situations, give the appreciation of this recovering.
observations. The final judgement is determined by the human ear,
With this method more iterations could be because audio signals do not exactly satisfy the
necessary to detect all the noise pulses from a frame requirements of being fully perdictable, and therefore
(especially for the small amplitude noise pulses). This they cannot be perfectly extrapolated.
algorithm contains also stop criterions for the
processing of a frame. Tab. 1: Comparative measurements between the results of the
The Table 1 give comparative measurements modified (MOD) and conventional (CONV) methods
between the results of the modified (MOD) and
conventional (CONV) methods. The parameters were Undetected pulses Wrong detections
the same for all the six signals. [%] [%]
MOD CONV MOD CONV
Signal 1.56 16.41 3.26 1.17
1
Signal 2.1 20.39 1.56 0.68
2
Signal 1.56 10.71 1.83 0.87
3
Signal 4.21 21.52 2.40 0.93
4
Signal 1.63 14.86 6.65 3.54
5
Signal 1.85 12.11 8 4.45
6

Fig. 8. a) Original signal affected by impulsive noise; b) Detection

and location thresholds in the input signal.

Fig. 10. a) Audio signal affected by a perturbation; b) Restored

audio signal.

Fig.9. Reduction of the location threshold value as a function of REFERENCES

iteration number, with the parameter f.
This method has a good sensitivity for the
parameter variations of the signal and the noise.
For the modified method the next parameters
have been used: N=1024, p=40, K=5, b1=0.5, r=0.5,
n=3, f=3, imax=7.
IV. CONCLUSIONS
This paper presented methods for the correction
of the disturbances in audio signals and for the
estimation and removing for long pulses from old
recordings. These methods can be implemented in
[1] A. Mateescu, S. Ciochin, N. Dumitriu, A. erbnescu, L.
Stanciu, Prelucrarea numeric a semnalelor, Ed. Tehnic,
Bucureti, 1997.
[2] P. A. A. Esquef, L. W. Biscainho, P. S. R. Diniz, F. P.
Freeland, A Double Threshold Approach to Impulsive Noise
Detection in Audio Signals, EUSIPCO sept. 2000 Tampere.
[3] P. A. A. Esquef, L. W. Biscainho, V. Valimaki, An efficient
Algorithm for the Restoration of Audio Signals Corrupted with
Low-Frequency Pulses, J. Audio Engeneering Society, Vol.51, No.
6, June 2003.
[4] I. Kauppinen, J. Kauppinen, Reconstruction Method for
Missing or Damaged Long Portions in Audio Signal, J. Audio
Engeneering Society, Vol.50, No. 7/8, July/August 2002.

105
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Information Theoretic Approach of Filterbanks

Performing Energy Compaction
Daniela Trniceriu1, Valeriu Munteanu1
Abstract We propose a way to get optimal filterbanks, subband coders that use uniform scalar quantizers in
which maximize the coding gain. The optimization is each channel, as Fig. 1 shows. The distortion measure
made over the filters, so that the mean - squared error of is the mean - squared reconstruction error. The
the reconstructed signal is minimized. The filters result optimal filters result functions of input signal
to be functions of input signal statistics. We also show
that for the particular case when the decimation factor
statistics. For the special case when M=2, we found
M=2, the maximization of the coding gain is equivalent that the optimal filters are those that perform energy
to the design of an optimum energy compaction filter. compaction.
Keywords: coding gain, energy compaction, filterbanks.
II. PRELIMINARIES
I. INTRODUCTION
For the maximally decimated subband coder (identical
In recent years there has been a considerable interest decimation ratio M in all subbands) shown in Fig. 1,
in studying filterbanks from various points of view, the analysis and synthesis filters are denoted by
due to their applications in speech, image processing H i (e j ) and Gi (e j ) , i = 0, M 1 , respectively.
and communications [1] [2]. The main problem The blocks denoted by Q represent the quantizers in
addressed by the researches consists in designing each channel.
optimal filters to attain maximum coding gain and The corresponding polyphase representation is shown
energy compaction, when quantizers are present [3].
Given a fixed budget of R bits for the subband in Fig. 2, where R(e j ) and E (e j ) are the
quantizers, the goal is to optimize both the analysis polyphase matrices [1].
and synthesis filters and to choose a subband bit
allocation strategy, such that the average variance of
the error e[n] at the output of the filterbank is
minimized. We consider the M-band uniform subband
coder shown in Fig. 1.

Fig. 2. Polyphase representation of filterbank in Fig.1.

The filter bank is said to be a biorthogonal or perfect

reconstruction (P.R.) one, if R(e j ) E (e j ) = I .
It is said to be orthonormal or paraunitary, if E (e j )
Fig. 1. Maximally decimated subband coder
is unitary for all . In the orthonormal case, the P.R.
condition is [1]
In [3] there are derived some necessary conditions for
optimality. In general, the optimum solution for filters Gi (e j ) = H i* (e j ) (1)
is different from the contiguous staking, which is the The conditions for biorthogonality and orthonormality
traditional subband split. For large number of are
subbands, the coding gain is not significantly different
from the optimum one, but for a small M, there could H k (e j )G m (e j ) = ( k m) (2)
be a significant difference. In this paper we are M
interested to minimize the distortion measure of
and

Facultatea de Electronic i Telecomunicaii, Bd. Carol I, Nr. 11, 213737, Iasi, 6600, e-mail tarniced@etc.tuiasi.ro

106
 H k ( e j ) H m ( e j ) = ( k m) ( 3) direct x2
M G SBC ( M ) = (7)
SBC M 1 2
1/ M

respectively, where F ( e j ) is the Fourier

xi

M i =0
transform of f [Mn] . The orthonormality condition
This result is valid for orthonormal filterbanks, when
implies in particular that each filter H i (e j ) satisfies
2 M 1
the Nyquist(M) constraint H i (e j ) = 1 , i. e.
M i =0
2
xi = M x2 (8)

M 1

H [e
2
j ( 2k / M )
] = M for all . (4) For fixed input power spectrum density S x (e j ) , the
k =0
variance xi2 depends only on the analysis filters
An aliasfree(M) filter is defined to be one whose H i ( e j ) .
output can be decimated by a factor of M, without The subband coder is said to be optimal, if the coding
aliasing. gain is maximized. This is equivalent to minimizing
the product of subband variances.
A. Basic Model for Signal and Quantization Noise
The input x[n] to the subband coder is assumed to be C. Total Decorrelation of Subbands
real valued and wide sense stationary (WSS) with This property implies
zero mean and power spectral density S x (e j ) . The
total bit budget is R bits per sample. The quantizer in { }
E vi [n]v k* [m] = 0 , for i k and for all m,n. (9)
each channel is scalar and uniform and it has allocated
a budget of Ri bits, so that E{} stands for the statistical mean. This condition is
necessary and sufficient for optimality in orthogonal
M 1
1 transform coding, but not for orthonormal subband
R=
M R
i =0
i (5) coders.

D. Majorization Property
We assume that the quantization noise qi [n] is Let S k (e j ) denote the power spectrum density of
additive and independent of the signal, and the noise 2
source in different channels are wide sense stationary the k-th decimated subband signal v k [n] and vk , its
and zero mean. This model does not require each variance. Assume that the subbands have been
q i [n] white or uncorrelated with the others (for the numbered such that
case of biorthogonal filterbanks, the white,
uncorrelated noise model is required in coding gain v20 v21 ... vM
2
1 (10)
derivation). This is the standard model used at high bit
rates. Each quantizer is assumed to have the variance
of the form
{
The set of subband power spectra S k (e j ) has the }
majorization property if

qi2 = c 2 2 Ri vi2 (6)

S 0 (e j ) S1 (e j )... S M 1 (e j ) (11)

B. Coding Gain Total decorrelation and majorization properties are

The quantity in (5) which is the average bit rate is necessary and sufficient for optimality of an M-band
assumed to be fixed. The coding gain is defined as the orthonormal filter bank.
ratio between the mean square value of the direct
2
quantization error ( direct = qPCM ) with the same bit E. Compaction Filter
rate R and the mean square value, , of the Fig.3 shows a branch of an M channel analysis
reconstructed error x[n] x[n] . bank, for which y2 = v2 .

direct
G(M ) = (7)

Fig. 3
Using the standard noise model, an expression for the
coding gain GSBC(M) of the orthonormal subband The optimum energy compaction problem is to
coder in Fig. 1 is derived in [1]: maximize the variance

107
 solution. Given an arbitrary paraunitary matrix
2
2 d U (e j ) , we can always write E opt (e j ) as:
H (e
j
y2 = ) S x ( e j ) (12)
2
0
E opt (e j ) = F (e j )U (e j ) (16)
2
subject to the constraint that H (e j ) is Nyquist
where U(e j ) is unitary and therefore, nonsingular
(M).
x[n] is assumed to be zero mean WSS input, having matrix and F (e j ) is a diagonal matrix. Due to
relation (16), the filter bank in Fig. 2 can be redrawn
the power spectrum density S x (e j ) .
as in Fig. 4. Since U(e j ) is unitary, the errors
The compaction gain is given by
e[n] = x[n] x[n] and e y [ n] = y[n] y[ n] have the
y2 same mean square value.
Gcomp = (13)
x2

F. Principal Component Filter Banks

The principal - component filterbanks [5], [6] satisfy
two fundamental properties, that is total decorrelation
(rel.9) and spectral majorization (rel. 11). The analysis
polyphase matrix of principal - component filter
banks is the matrix of ordered eigenvectors of
Fig.4. Equivalent representation of the polyphase matrix as a
S x ( e j ) . cascade of principal component filterbank and remainder filters

III. CODING GAIN MAXIMIZATION Now we can design F (e j ) to be optimal for its input
y[n] , i. e., it minimizes the mean square value of
In order to maximize the coding gain in rel (7), we
have to minimize the variance of the reconstruction e y [n] . If U (e j ) is optimal orthonormal, we can
error. In biorthogonal filter banks, according to Fig. 1, always split the design of the optimal biorthogonal
it can be writen as:
E(e j ) into design the optimal orthonormal system

1
M 1 U (e j ) which is a principal - component filterbank
E{ x[n] x[n]
2
= }=
M and design the biorthogonal F(e j ) for y[n] .
n=0
M 1 In Fig. 5a one branch of a filter bank is considered
1 2 d
c2
2
G i ( e j )
2 Ri
vi2 = and P(e j ) is choosen to satisfy
M 0 2
i =0 (14)
M 1
1 2 R
2 d P(e j ) = F 1 (e j )
c2
2
(17)
j j
H i (e ) S x (e )
M i =0 0 2
2 2 d In Fig. 5b the quantizer is modeled using the model

0
G i ( e j )
2 described in Section II.

Considering the inequality between the arithmetic and

the geometric means, the minimum value of the
variance of the reconstruction error is

M 1 2 d

2
= c 2 2 R H i ( e j ) S x ( e j ) Fig. 5
2
i =0
0
(15)
1/ M Let e y [ n] = y[n] y[ n] be the reconstruction error. Its
2 d
2

j
Gi (e ) variance is
0 2
2
2 2 d
P (e
By minimizing the product in the right hand side over e2y = E{ e y [n] } = q2 j
) =
the analysis/synthesis filters, they result signal 2
0
adapted. (18)
2 2
For the filter bank in Fig. 2, let d2 2d
S P (e
2 R j j j
j j c2 yy (e ) F (e ) )
E(e ) = E opt (e ) be an optimal biorthogonal 2 2
0 0

108
Taking into account rel. (17) and making use of the
Chauchy Schwartz inequality, the previous relation
satisfies the following inequality
2 gain (see rel.13).
d
e2y
0
S yy (e j )
2
with equality only when
F ( e j ) = P * ( e j )
where is an arbitrary nonzero constant. Since
P(e j ) = F 1 (e j ) , must be of the form = K 2
and
F (e j ) = KS yy
1 / 4
(e j )
This choice of F (e j ) depends on the input signal
statistics and assures the minimum value for the
variance of the reconstruction error and therefore, the
maximum coding gain.
In general, the optimal F (e j ) and its inverse may
not be realizable filters. They may be replaced with
causal, stable approximations, which result in
suboptimum coding gain.
Considering the spectral flatness measure of the input
signal defined as [2]
2
d
exp{ ln S x (e j )
2
} 2
x2 = 0
x2
we observe that the coding gain is large for small
values of x2 , that is, for a peaky spectrum and
small for values of x2 closed to unity, that is for a
relatively flat spectrum.
In general, for fixed number of subbabds M,
biorthogonal filter banks provide better gain than
orthonormal filter banks. But, for a special shape of
power spectrum density of the input signal, when it
consists of contiguos constant frequency segments of
length 2 / M , the performances of biorthogonal and
orthonormal filter banks are the same. This is due to
the fact that when we use brickwall filter banks with
contiguos staking, the subband signal xi [n] has a
constant power spectrum density in the passband.
Therefore, the signals {v k [n]} satisfy the
decorrelation as well as the majorization properties
and the filter bank is optimal orthonormal.
IV. THE TWO CHANNEL CASE
The conditions to maximize the coding gain are not
the same as those to get the maximum compaction
(19) For the M=2 case, the maximization of the coding
gain is equivalent to the design of an optimum
compaction filter [3].
The coding gain is
(20) x2
G SBC (2) = (23)
x20 x21
But, due to the orthonormality, x20 + x21 = 2 x2 , so
that, relation (23) becomes
(21) x2
G SBC (2) = (24)
x20 (2 x2 x20 )
or, in terms of the compaction gain
1
G SBC ( 2) = (25)
Gcomp (2)(2 Gcomp (2))
The denominator in relation (24) is minimized by the
choice of H 0 ( e j ) subject to filterbank
orthonormality condition, that is equivalent to
Nyquist(2) condition
2
H 0 ( e j ) + H 0 ( e j ) = 2 (26)
(22)
If H 0 (e j ) is designed to be the optimum energy
compaction filter for the given input power spectrum
density S x (e j ) , that is, its output variance x20 is
maximized under the Nyquist constraint, the other
filters H 1 (e j ) , G0 (e j ) and G1 (e j ) are chosen
as in relation (1) to get perfect reconstruction.
REFERENCES
[1] P. P. Vaidyanathan, Multirate Systems and Filter Banks.
Englewood Cliffs, NJ: Prentice Hall, 1993.
[2] Jayant N. S. and Noll P., Digital Coding of Waveforms.
Englewood Cliffs, NJ: Prentice-Hall, 1984.
[3] P.P. Vaidyanathan, Theory of optimal orthonormal
subband coders, IEEE Trans. Signal Processing, vol. 46,
pp. 1528 1543, June 1998.
[4] M. Unser, On optimality if ideal filters for pyramid and
wavelet bases for signal decomposition, IEEE Trans.
Signal Processing , vol. 41, pp. 3591 3596, Dec. 1993.
[5] M. K. Tsatsanis, G. B. Giannakis, Principal component
filter banks for optimal multiresolution analysis, IEEE
Trans. Signal Processing, vol. 43, pp. 1766 1777, Aug.
1995.
[6] A. Kirac, P.P. Vaidyanathan, Theory and design of
optimum FIR compaction filters, IEEE Trans. Signal
Processing, vol. 46, pp. 903 919, Apr. 1998.
109
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49 (63), Fascicola 2, 2004
Denoising Over-Sampled Signals
Andr Quinquis1, Alexandru Isar2, IEEE Members and Dorina Isar2
Abstract - This paper presents a new denoising method for
over-sampled constant within intervals signals corrupted by
additive noise. The novelty of this paper is a special MAP
filter, called composed bishrink. A complete statistical
analysis of this filter is reported. Some simulations are
presented. The results obtained are compared with the
results of other denoising methods and with other state-of-
the art filtering techniques.
Keywords: wavelets, denoising, soft thresholding, bishrink.
I. INTRODUCTION
In recent years, the techniques that use multiscale and
local transform-based algorithms have become
popular in noise filtering applications. In particular
the use of non-linear filters in the DCT domain was
studied, [1]. In this paper, we consider local transform
based denoising. We propose such an algorithm,
based on the discrete wavelet transform, (DWT).
Section II deals with the local DWT-based denoising.
In section III, a statistical analysis of the new
denoising method is presented. The use of local filters
in the DWT domain is described in the following
section. In section V, numerical simulation results are
presented and discussed. The last section is dedicated
to some concluding remarks.
II. LOCAL DWT-BASED DENOISING
The following model of the observed signal corrupted
by additive noise is considered in this paper:
x [k ] = s [k ] + n [k ]
(1)
where s and n represent the useful part and the noise.
The problem is to estimate s starting from x. The
noise is usually considered to be an uncorrelated with
s, stationary random process, with a null mean and a
variance n 2 . To estimate the signal s, Donoho, [2],
proposed the following method:
1. The Discrete Wavelet Transform (DWT) of the
signal x is computed. The result is the signal yi = y +
ny. The noise ny converges asymptotically to a
Gaussian white one, with the same variance, [3].
2) Communications Department, Politehnica University
Timisoara, e-mail: isar@hermes.ee.utt.ro
110
2. A non-linear filtering is applied in the wavelet
domain:
where t is a threshold. This system is called soft
( )
sgn { yi [k ]} yi [k ] t , yi [k ] > t
y0 [k ] =
( 2)
0, if not
thresholding filter. Because the noise ny is Gaussian, if
t>3n, the probability P(ny>t) is very little (the rule of
3 sigmas). So the noise is quasi entirely suppressed.
This is the reason why the signal y0 is a denoised
version of the signal yi. This is a non-linear adaptive
filter whose statistic analysis was presented in [4].
The adaptability is due to the selection of the
threshold value in function of the noise power.
3. Taking the inverse DWT (IDWT) of the signal y0,
the denoised version of the signal s, x0, is obtained.
The principal disadvantage of the already described
denoising method is due to the fact that it is based
only on the estimation of the noise variance (the
useful part of the input signal is ignored) and on
hypotheses confirmed only asymptotically. This is the
reason why in the following another denoising
strategy, based on the use of a Maximum a Posteriori,
MAP, filter, will be described.
III. A STATISTICAL ANALYSIS OF THE DWT
The probability density function, (pdf), of the wavelet
coefficients at the mth scale, (after m iterations),
k
x Dm (k being equal with 1 for detail coefficients and
with 2 for approximation coefficients) is given by the
following relation:
N (k ) M0
f k (a) = ...
x Dm
r1 = 1 q2 = 1
(3)
M0
f d ( k , r1 , q2 ..., qm , a )
qm = 1
where:
 f d ( k , r1 , q2 ..., qm , a ) = k = k + k (9)
x Dm s Dm n Dm
= G ( k , r1 , q2 ..., qm ) (4) If the input noise is a zero mean white Gaussian, the
correlation of its wavelet coefficients becomes:
f x ( G ( k , r1 , q2 ..., qm ) a )
and:
k
n Dm
[ n1 ] = n2 [ n1 ] (10)

G ( k , r1 , q2 ..., qm ) = At any scale the noise in the wavelets domain is also

white, having the same variance. So, a single
1 estimation of its variance, for example using the detail
= (5)
m coefficients obtained after the first iteration, is
F ( k , r1 ) mo [ ql ] sufficient. For any type of input noise, when m tends
l =2 to infinity, the relation (8) becomes:
where:
k [ n1 , p1 ] = x ( 0 ) [ n1 p1 ] (11)
m [ r ] k = 2 n D
F ( k , r1 ) = 0 1 (6)
m1 [ r1 ] k = 1
So, asymptotically, the noise in the wavelets domain
becomes white. Unfortunately this is also only an
M 0 represents the length of the impulse response asymptotic result. Combining this result with the
result obtained after the pdf analysis, it can be
m0 (of the low-pass filter used in the computation of
observed that after a given number of iterations,
the DWT) and M1 represents the length of m1 (the Nu2 , the noise in the wavelet domain is white and
high-pass filter used in the computation of the DWT) Gaussian. The mean of the wavelet coefficients is:
and the number of the convolutions in the first group
0, k =1
{ }
from the relation (3) is given by:
M , k = 2
E k
x Dm [ n1, p1 ] = m (12)
N (k ) = 0 (7) 2 2 x , k = 2
M1 , k = 1 In practice the number of iterations of the DWT is
In conformity with (3), the pdf of the wavelet high. The length of the approximation coefficients
coefficients is a sequence of convolutions. Hence, the sequence obtained after the last iteration is small. This
random variable representing the wavelet coefficients is the reason why this sequence is not filtered in
can be written like a sum of independent random practice. The variance of the wavelet coefficients is
variables. So, the central limit theorem can be applied. given by:
This is the reason why the pdf of the wavelet
k 2
coefficients tends asymptotically to a Gaussian, when 2 k = E x Dm =
the number of convolutions in (3) tends to infinity. x Dm
This number depends on the mother wavelets used (13)
1
{ }
2
and on the number of iterations of the DWT. For = m
k
x ( 2 u) (u ) du
mother wavelets with a long support, this number 2
increases very fast. The slower convergence is The correlation of the DWT of the useful part of the
obtained for the Haar mother wavelets, which has the input signal, s, is given by:
shorter support. This is the case analyzed in this
paper. The filter used in the DWT domain must be k
s Dm
[ n1 ] = 2m s 2m n1 (14)
constructed having in mind that after a number of
iterations the distribution of the wavelet coefficients its mean by:
can be considered Gaussian. The problem is to 0, k =1
establish this number, Nu1 . Another problem is to E { k
s Dm }
[ n1 ] = m
2 2 , k = 2
(15)
find the wavelet coefficients distribution law for the s
first iterations (before to reach the Gaussian law). The
and its variance by:
k
correlation of the wavelet coefficients x Dm is given
2 k = 2m s2 (16)
by: s Dm

[ n1, p1 ] = E x Dmk [ n1 ] ( x Dmk [ p1 ])

* So, the variances of the detail wavelet coefficients
k = sequences of the useful component of the input signal
x Dm increases when the iteration index increases.

( ( ))
1
m
x 2 w+2 p 2 (8) IV. MAP FILTERS EXPLOITING THE
2 p = INTERSCALE DEPENDENCY OF THE DETAIL
2
COEFFICIENTS
{ }(
2
e
jw( n1 p1 )
k w+2 p 2 ) dw
In conformity with (8), there is an important
Because the signals s and n are not correlated it can be correlation between a wavelet coefficient at a given
written: scale and the same coefficient situated in the same

111
position at the next scale (named the parent of the 2 mm
considered coefficient). This correlation can be l = 1 2 (24)
exploited to construct adaptive filters acting at a given and:
scale and using for the estimation of their parameters g, g > 0
information obtained at the next scale, [5]. Using the ( g )+ = (25)
parent and child wavelet coefficient of the input signal 0, if not
it is possible to estimate the child coefficients of the and for the models in (20) and (22) the solution of the
DWT of the useful part of the input signal, with the maximization problem in (19) is:
1m m
2
aid of a bishrink filter, [5]. Let 1 yi be the considered 1l
y= 1 yi (26)
m m 2
detail coefficient and 2 yi its parent. The statistical 1 2 m
+n
parameters of the child coefficients can be determined So, the input-output relations of the composed
using their parent coefficients and the neighbor child bishrink filter are (23) and (26). The noise variance is
coefficients, located in a window with a length of 3, estimated using the details obtained after the first
centered on the current child coefficient. It can be m m
written: iteration and the variances 1 and 2 are estimated
yi = y + n y (17) in moving windows centered on the current child and
parent coefficients. First the means are estimated in
where: each window and second the variances. But, applying
( ) ( )
y i = 1 yi , 2 yi ; y = 1 y , 2 y ; n y = 1n y , 2 n y (18) ( ) the relation (16), a different estimation of the local
variance of the child coefficients can be obtained:
The MAP estimation of y, realized using the m
2
observation y i , is given by: 1n
d = (27)

{( )}
2
y ( y i ) = arg max ln fn y ( y i y ) f y ( y ) (19) To profit of these two estimations of the local
y variances, obtained at two successive scales, it can be
In the following, we will consider that the DWT of written:
the noise is distributed following a zero mean m
2
Gaussian: 1m
m +
( 1ny ) +( 2ny ) 1m
2 2 2
= (28)
2
( )
fn y n y =
1
2 n
e 2 n2
(20) This estimation will be used in (24) and (26),
m
substituting 1 , for the input-output relations of the
Concerning the model of the DWT of the useful composed bishrink filter.
component, in the case of the composed bishrink
filter, for the first Nu2 iterations, a Laplace V. SIMULATION RESULTS
distribution will be considered (like in the case of the
bishrink filter, [5]): A useful input signal constant within intervals was
considered. This is a data sequence, specific for the
( 1 y ) +( 2 y )
3 2 2
3 communication in the base-band. This sequence has a
fy ( y ) = e (21) number of 16384 symbols, each having b=128
2 samples. A portion of this signal is represented in
and for the other iterations, a Gaussian distribution figure 1. Taking into account the waveform of this
will be considered: signal, the Haar mother wavelets must be used. The
( 1 y ) +( 2 y )
2 2 DWT was computed on blocks, each having a length
of 4096 samples. The maximal number of iterations
1
fy ( y ) = e 21 2 (22) (equal with 12) was used for the computation of each
DWT. For the implementation of the composed
2 1 2
bishrink, a value of Nu2 =8, was used.
(like in the case of the Wiener filter, [6]). For the
models in (20) and (21) the solution of the
maximization problem in (19) is:
2
3mn
( yi ) + ( yi )
2 2
1 2

l
1m + 1 Figure 1. The waveform of the useful component of the
y= yi (23) input signal.
( 1 yi ) + ( 2 yi )
2 2

In the following figure is represented the dependency

where: between the output and input SNRs for different

112
 Figure 2. The dependence of the output SNR of the input
SNR.
denoising methods. The filter used in the wavelets
domain gives the difference. It can be observed that
these dependencies are linear. The curves describing
the bishrink filter and the composed bishrink filter are
superposed. These filters give the better results. These
are superiors to the results obtained using the soft-
thresholding filter. The poor results are obtained using
the Wiener filter. Finally, a comparison between the
use of an adapted filter and a denoising system, into a
communication application is discussed. Each of the
two systems are connected at the output of a
communication channel, that adds a zero mean white
noise to the data sequence, which beginning is
represented in figure 1. The first system is a filter
adapted to a rectangle, having a duration equal with b.
Six experiments are made, with different noise
variances. At the output of the investigated system
(adapted filter or denoising system), an ideal sampling
system is connected. Three hypotheses, concerning
the synchronization, are used. The first hypothesis
supposes a perfect synchronization. The second
hypothesis accepts a little loss of synchronization (the
sampling moments are delayed with 3 b / 8 ) and the
third hypothesis accepts a more important loss of
synchronization (the sampling moments are delayed
with b / 2 ). The output of the sampling system is
connected to the input of a comparator. The output of
this comparator represents the output of the simulated
receiving unit. The denoising system uses a soft
thresholding filter when the input SNR is inferior to
16.65 dB and a composed bishrink when the input
SNR is superior to 16.65 dB. The better result is
obtained with the adapted filter with perfect
synchronization (ad.filt.perf.sync, the continuous line
in figure 3). For the other hypotheses, an analysis,
taking into account the value of the input SNR must
be made. The synchronization losses do not affect the
performances of the denoising system
(denois.perf.syn, the dashed line in figure 3,
denois.part.sync.1 and denois.part.sync. 2) but affect
the performances of the adapted filter. In the second
hypothesis, the denoising system is better than the
113
Figure 3. A comparison between the use of an adapted filter
and a denoising system, in a base-band communication
application.
adapted filter (ad.filt.part.sync 1, the dash dot line in
figure 3) for input SNRs superior to 10 dB. Also, in
the third hypothesis, the denoising system is superior
to the adapted filter (ad.filt.part.sync 2, the dotted line
in figure 3), for input SNRs superior to 10.47 dB.
Practically in this case the adapted filter cannot be
used.
VI. CONCLUSION
In this paper is proposed a new denoising method
based on the use of the composed bishrink filter in the
wavelets domain. This method takes into account also
the statistics of the useful part of the input signal. That
makes that this method to perform better than the
denoising method using the soft thresholding filter for
input SNRs superior to 10 dB. It can be used in
communications, replacing the adapted filter, when
the synchronization is difficult.
REFERENCES
[1] K. O. Egiazarian, V. P. Melnik, V. V. Lukin, J. T.
Astola, Local transform-based denoising of radar image
processing, Nonlinear Image Processing and Pattern
Analysis XII, Edward R. Dougherty, Jaakko T. Astola,
Editors, Proceedings of SPIE vol. 4304, 2001.
[2] D. L. Donoho, "De-noising by Soft Thresholding",
Technical Report no.409, Stanford University,
December 1992.
[3] M. Borda, D. Isar, "Whitening with wavelets",
Proceedings of ECCTD'97 Conference, Budapest,
August 1997.
[4] A. Isar, A. Cubichi, Miranda Naforni, Algorithmes et
techniques de compression, Editura Orizonturi
universitare, Timisoara, 2002.
[5] L. Sendur and I. W. Selesnick, Bivariate shrinkage
functions for wavelet-based denoising exploiting
interscale dependency, IEEE Trans. on Signal
Processing. 50(11): 2744-2756, November 2002.
[6] H. Zhang, A. Nostratinia, R. O. Wells Jr., Image
Denoising via Wavelet-Domain Spatially Adaptive FIR
Wiener Filtering, IEEE ICASP, Istanbul, June 2000, vol.
5, 2179-2182.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Invisible watermarking for Copyright Protection
M. A. Matin1
Abstract Due to the rapid expansion of the Internet and
the overall development of digital technologies, millions of
users, who are scattered all over the world, are able to use a
vast number of multimedia products. Every participant in
this process wants to assert their rights, which are given by
their role in the business string. Naturally, solutions to
digital copyright protection are required urgently to tackle
the problem of unauthorized copying and distribution. The
aim of this paper is concerned with inserting copyright
information into host image. In this paper, discrete cosine
transform (DCT) domain watermarking technique for
copyright protection of still digital images is analyzed. The
DCT is applied in blocks of 8 8 pixels as in the JPEG
algorithm. The watermark can encode information to track
illegal misuses concerned with the protection of copyright
information contained in digital images.
I. INTRODUCTION
In general, digital images and digital video-streams
can be easily copied one way or another. Even
though such copying may violate copyright laws, it is
widespread. The ease with which electronic images
may be copied without significant loss of content
contributes to illegal copying. One of the goals of
digital watermarking is authentication for copyright
protection. To prove the ownership of an image, a
perceptually invisible pattern (a watermark) is
embedded into the image and ideally stays in the
image as long as the image is recognizable.
II. REQUIREMENTS OF WATERMARKING
Digital watermarking, particularly digital image
watermarking, has several conflicting requirements.
The three most important requirements are
perceptibility robustness, and capacity[1]. For
example: a very robust watermark can be obtained by
highly modifying the host data for each bit of the
watermark by increasing the watermark strength.
However, this large modification will be perceptible.
As a second example, increasing the number of
embedded bits increases the capacity but decreases the
robustness. Therefore, the maximum amount of
modification that can be acceptable for the quality of
the media and robustness are the two determining
1
Department of Computer Science and Engineering, BRAC University,
Bangladesh, e-mail: rumel120@yahoo.com
114
factors for the maximum amount of watermark bits that
can be stored in a data object.
III. BUILDING WATERMARKING
It consists of two parts:- The first part is concerned
with insertion strategy i.e. where in the host signal shall
we place the information?. The second one is watermark
structure -how shall we place the additional information
into the signal?. It is often necessary to utilize Human
Visual System (HVS) models for adaptively embedding
the watermark. This can reduce the impacts of
modifications on image quality or for the same visual
quality a much stronger watermark can be embedded.
The human eye is sensitive to the following
characteristics of image-contrast, frequency, luminance
sensitivity, edges and texture area[2]. One can combine
the above four properties to construct a perceptual mask
which determines the amount of modification permitted
on each image cover data (pixels, transform coefficients)
value. Using perceptual masks, energy can be added
locally in places where the human eye cant notice it.
This increases robustness and hence capacity
IV. WATERMARK EMBEDDING APPROACH
There are two general approaches to embedding a digital
watermark. One approach is to transform the host image
into its frequency domain representation and embed the
watermark data therein. The second is to directly treat
the spatial domain data of the host image to embed the
watermark. Bruyndonckx et al. in [3] proposed a spatial
domain scheme for copyright labeling of digital images
based on pixel region classification.
The advantage of spatial techniques is that they can
be easily applied to any image, regardless of subsequent
processing (whether they survive this processing
however is a different matter entirely). A possible
disadvantage of spatial techniques is they do not allow
for the exploitation of this subsequent processing in
order to increase the robustness of the watermark.
In addition to this, adaptive watermarking
techniques are a bit more difficult in the spatial domain.
Both the robustness and quality of the watermark could
be improved if the properties of the cover image could
similarly be exploited. For example, it is generally
preferable to hide watermarking information in noisy
regions and edges of images, rather then in smoother
regions. The benefit is two-fold: degradation in smoother
regions of an image is more noticeable to the HVS, and
secondly becomes a prime target for lossy compression
schemes.
Taking these aspects into consideration, working in a
frequency domain of some sort becomes very attractive.
Frequency domain watermarking was introduced by Cox
et al.[4]. Coxs approach uses spread spectrum
communication techniques to embed a bit in the image.
However, it needs the original image to decode the
watermark and Smith et al.[10] refer to these approaches
(when the original image is needed in the decoding
process) as of limited interest because of their narrow
range of practical applications. The classic and still the
most popular domain for image processing is that of
Discrete-Cosine-Transform, or DCT. Koch et al.[5]
reported an efficient DCT domain watermarking
techniques resisting to JPEG compression. But our
proposed approach is robust also against attacks such as
filtering, cropping, Scaling and geometric rotation.
The DCT allows an image to be broken up into
different frequency bands, making it much easier to
embed watermarking information into the middle
frequency bands of an image. The middle frequency
bands are chosen such that they avoid the most visual
important parts of the image (low frequencies) without
over-exposing themselves to removal through
compression and noise attacks (high frequencies) [6].
V. FREQUENCY DOMAIN TECHNIQUE
One such technique utilizes the comparison of
middle-band DCT coefficients to encode a single bit into
a 88 DCT block. We first divide the NxN image into
(N/8)*(N/8) = N2/64 non overlapping 8x8 blocks; then
take DCT on each block and embed the watermark
middle-band DCT coefficients 8x8 Discrete Cosine
Transform (DCT) is defined as:
I(u,v)=
m(u ) n(v) 7 7 ( 2 k + 1) u
. X ( k , l ). cos( ) cos(
2 2 k = o l = 0 16
and 8x8 Inverse Discrete Cosine Transform (IDCT)
is defined as:
X(k, l)=
7 7 m(u ) n(v) ( 2 k + 1) u
. l ( u , v ) cos( ) cos(
u =0 v =0 2 2 16
where k,l,u,v {0,1,2,3,4,5,6,7} and
1 1
m(u)= for u=0 and m(u)=1 for u >0,m(v)=
2 2
for v=0 and m(v)=1 for v>0
DCT and IDCT are linear transformations and all
DCT coefficients are real. Any image block can be
represented as a superposition of scaled DCT
transformed images scaled with DCT coefficients.
A. Selection of DCT coefficient
The low frequency components of an image are
perceptually the more significant ones and any
modification on them deteriorates the image fidelity.
Therefore, watermarking shouldnt be applied on low
frequency components. On the other hand, the high
frequency components are the ones, which are usually
less significant in terms of fidelity. As a consequence,
compression techniques utilize this property and
suppress the high frequency components first to reduce
the size of images. Therefore, the watermarking
techniques that modify high frequency coefficients
cannot be robust carriers of watermark. This leaves us
with the choice of mid frequency coefficients.
B. DCT based techniques
One such technique utilizes the comparison of
middle-band (FM ) DCT coefficients to encode a single
bit into a DCT block. Suppose two locations Bi(u1,v1)
and Bi(u2,v2 ) are chosen from the FM region for
comparison. Rather then arbitrarily choosing these
locations, extra robustness to compression can be
achieved if we base the choice of coefficients on the
recommended JPEG quantization shown below in table
2. If two locations are chosen such that they have
identical quantization values, we can feel confident that
any scaling of one coefficient will scale the other by the
same factor preserving their relative size.
Table 1 Definition of DCT regions
FL
FM
( 2 l + 1) v
)
16
( 2 l + 1) v In table 1, FL is used to denote the lowest frequency
)
16
components of the block, while FH is used to denote the
higher frequency components. FM is chosen as the
embedding region.
115
 Table 2 - Quantization values used in JPEG The watermarking procedure can be made somewhat
compression scheme [7] more adaptive by slightly altering the embedding process
0 0 7 to the method shown in equation 2.
16 11 10 16 24 40 51 61 I * (1 + k * W x , y ) In FM region
I W x, y = x, y -
12 12 14 19 26 58 60 55 Ix, y In FL and FH region

14 13 16 24 40 57 69 56 (2)

14 17 22 29 51 87 80 62 This slight modification scales the strength of the

18 22 37 56 68 109 103 77
24 35 55 64 81 104 113 92
49 64 78 87 103 121 120 101
72 92 95 98 112 100 103 99
Based on the table, we can observe that coefficients
(4,1) and (3,2), or (1,2) and (3,0) would make suitable
candidates for comparison, as their quantization values
are equal. Say Bi denotes the 8x8 DCT block and two
locations Bi(u1,v1) & Bi(u2,v2) are chosen from FM
region. The DCT block will encode a 1 if Bi(u1,v1) >
Bi(u2,v2); otherwise it will encode a 0. The coefficients
are then swapped if the relative size of each coefficient
does not agree with the bit that is to be encoded [7].
The swapping of such coefficients should not alter the
watermarked image significantly, as it is generally
believed that DCT coefficients of middle frequencies
have similar magnitudes. The robustness of the
watermark can be improved by introducing a watermark
strength constant k, such that Bi(u1,v1) - Bi(u2,v2) > k.
Coefficients that do not meet this criteria are modified
using random noise to satisfy the relation. Increasing k
thus reduces the chance of detection errors at the expense
of additional image degradation [7].
Another possible technique is to embed a PN(Pseudo
random noise) sequence W into the middle frequencies
of the DCT block. We can modulate a given DCT block
x,y using the equation (1) shown below.
I + k * Wx, y In FM region
IW x, y = x, y ------(1)
Ix, y In FL and FH region
Where I x , y is the original image and k is the watermark
strength.
For each 8x8 block x,y of the image, the DCT for the
block is first calculated. In that block, the middle
frequency components FM are added to the PN sequence
W, multiplied by a gain factor k. Coefficients in the low
and middle frequencies are copied over to the
transformed image unaffected. Each block is then
inverse-transformed to give us our final watermarked
image IW [8].
watermarking based on the size of the particular
coefficients being used. Larger ks can thus be used for
coefficients of higher magnitudein effect
strengthening the watermark in regions that can afford it
and weakening it in those that cannot [8].
For detection, the image is broken up into those
same 8x8 blocks, and a DCT performed. The same PN
sequence is then compared to the middle frequency
values of the transformed block. If the correlation
between the sequences exceeds some threshold T, a 1
is detected for that block; otherwise a 0 is detected.
Again k denotes the strength of the watermarking, where
increasing k raises the robustness of the watermark at the
expense of quality [8].
C. Proposed approach
Researchers can compare different algorithms and
see how a method can be improved or whether a newly
added feature actually improves the reliability of the
whole method [9].
In section 4 Building watermarking we discussed
about the watermark structure. The most straight-forward
approach would be to embed watermark (text strings)
into an image by allowing an image to directly carry
information such as author, title, dateand so forth. The
drawback however to this approach is that ASCII text in
a way can be considered to be a form of LZW (Lempel
Ziv-Welch) compression, where each letter being
represented with a certain pattern of bits. By
compressing the watermark-object before insertion,
robustness suffers.
Due to the nature of ASCII codes, a single bit error
due to an attack can entirely change the meaning of that
character, and thus the message. It would be quite easy
for even a simple task such as JPEG compression to
reduce a copyright string to a random collection of
characters. The properties of the HVS (Human visual
system) can easily be exploited in recognition of a
degraded watermark.
In this work the host image is divided into 4096
blocks of size 8x8. The binary watermark with a size of
2050 pixel is embedded into the image. The algorithm
works on selected 1000 of 8x8 DCT Coefficient blocks
and the coefficients of the same quantization value is
taken for comparison and are encoded such that (4,1) >
(3,2) when watermark bit is 0 and that (4,1) < (3,2) when
watermark bit is 1, and the two values are adjusted such

116
that their difference >= k. Finally the block is
transformed back into spatial domain.
For detection, the watermarked image is broken up
into those same 8x8 blocks, and a DCT is performed The
same PN sequence is then compared to the middle
frequency values of the transformed block.

Figure 1(a): Original images

VI. RESULTS AND DISCUSSION

The experiment involved evaluating the reliability

of extracted watermark and demonstrating the copyright
effectiveness of the proposed approach. In this work,
five kinds of manipulations are considered- filtering,
lossy JPEG compression, cropping, scaling and rotation.
The experiments were performed on monochrome
images with a size of 512512 pixels. Figure 2(a) shows
three images that were used:- airplane, Lena, bird and Figure 1 (b): watermarked images
were selected to represent three kinds of images - those
containing large smooth areas, containing both smooth PSNR=38.3 dB PSNR=34.1dB PSNR=34.7dB
and detailed areas, and with large amount of details. wPSNR=40.2dB wPSNR=35.6dB
wPSNR=35.4dB

PSNR (Peak signal to noise ratio) is calculated using the

equation 3 to give us a rough approximation of the The ever- popular miss November (Lena) image is
quality of the embedded image in the experiments. used as a reference image. From the difference
between original and watermarked image of Lena, the
max( x) 2 error is visible. The error is most significant at black
PSNR = 10 log10 --------------(3) hair. At the receiver site, the watermark is extracted
x x2 from the transmitted image and compared with the
original watermark (Copyright) to perform the
Where x is the image under test and x is the original copyright protection.
image.
In the above equation the PSNR penalizes the
visibility of noise (watermark) in all regions of the image
in the same way. However, due to phenomena of contrast
masking the visibility of noise in flat regions is higher
than that in textures and edges.

Therefore, a simple approach to adapt the classical

PSNR for watermarking applications consists in the
introduction of different weights for the perceptually
different regions oppositely to the PSNR where all
regions are treated with the same weight. Originally this Figure 2:(a) Low pass filter (b) Recovered watermark
idea was presented by Netravali and Haskell [11] with
application to image compression. Application to
watermarking quality evaluation was reported in [12]
using the NVF (noise visibility function) as a weighting
matrix:
max( x) 2
wPSNR = 10 log10
x x2 NVF
max( x) 2
= 10 log10 ------------(4)
NVF ( x x)2
Figure 2: (a) Median filter (b) Recovered watermark

Figure 2(a) and 4(a) shows a low pass and median

filtered watermarked image using a 3x3 filter mask
consisting of 0.9 intensity values. The median filtered
image is more blurred than the low pass filtered image

117
(which is blurred also compared to the original host
image). The reconstructed watermark is also still better
in median filter.
Figure 4:(a) Index-100jpg (b) Index-25 jpeg
The above figure shows watermarked image compressed
using lossy index-100 JPEG and index-25 JPEG
compression. The index ranges from 0 to 100, where 0 is
the best compression and 100 is the best quality. The
reconstructed watermark is a good reproduction in our
experiment.
Figure 5: (a) Cropping (b) Recovered watermark
Figure 5(a) shows a cropped watermarked image
cropped with a mask of size 340x425 pixels. The
reconstructed watermark is still recognizable.
Figure 6:(a) Rotation 2 degree (b) Recovered watermarked
Geometric transforms are one of the most difficult
conditions for a watermarking technique to deal with
embedding domain. This can be chosen both by shifting
or rotating invariance such as Cartesian or Polar DCT;
however these domains are typically resistant to only a
specific geometric distortion.
The only difference between the rotated image and
the cropped image is the bilinear interpolation used to
realign the pixels after it is rotated back to its original
118
alignment. The bilinear interpolation can be
approximated as an averaging filter.
Figure 7 (a): Scaling (b) Recovered watermarked
The scaling experiment was done by scaling the
watermarked image down to one quarter of its original
size (256x256) and rescaled back to 512x512 using
bilinear interpolation The algorithm requires the pixels
in the watermarked image to be in the corresponding
location as the original host image in order to extract the
watermark correctly.
VII. CONCLUSION
In this Paper the message is invisibly embedded into the
source image. A verification key, which is stored and
known only to the author, is produced in the embedding
step and used in the verification process to extract the
embedded message inserted in the host. Here some
attacks such as low pass filtering, median filtering,
lossy JPEG compression, cropping, rotation and Scaling
has been done on watermarked image to destroy the
copyright information but it is still recoverable and
recognizable of the owner.
REFERENCES
[1].Cox, J., M. L. Miller and J. A. Bloom, Digital Watermarking,
Morgan Kauffman Publishers, ISBN 1-55860-714-5, 2002.
[2]. Swanson, M. D., B. Zhu and A. H. Tewfik, Robust data hiding for
images, 7th IEEE Digital Signal Processing Workshop, pages 37-40,
1996.
[3].O. Bruyndonckx, J.J. Quisquater, and B. Macq, Spatial method for
copyright labeling of digital images, in Proc. IEEE Workshop
Nonlinear Signal and Image Processing, Halkidiki, Greece, June 1995.
[4].I.J. Cox, J. Kilian, T. Leighton and T. Shamoon, Secure Spread
Spectrum Watermarking for Multimedia, Technical Report 95-10,
NEC Research Institute.
[5] E. Koch and J. Zhao, Toward robust and hidden image copyright
labeling, in Proc.Workshop Nonlinear Signal and Image Processing,
Marmaros, Greece, June 1995.
[6] Podilchuk, C. I. and W. Zeng, Perceptual watermarking of still
images, ElectronicProceedings of the IEEE Signal Processing Society
1997 Workshop on Multimedia Signal Processing, Princeton, New
Jersey, June 1997.
[7] N.F. Johnson, S.C. Katezenbeisser, A Survey of Steganographic
Techniques in Information Techniques for Steganography and Digital
Watermarking, S.C. Katzenbeisser et al., Eds. Northwood, MA: Artec
House, Dec. 1999, pp 43-75
[8] G. Langelaar, I. Setyawan, R.L. Lagendijk, Watermarking Digital
Image and Video Data, in IEEE Signal Processing Magazine, Vol 17,
pp 20-43, September 2000
[9] F.A.P. Petitcolas, Watermarking Schemes Evaluation , in IEEE
Signal Processing Magazine, Vol 17, pp 58-64, September 2000
[10] J. Smith and B. Comiskey, Modulation and information hiding in
images, , in Proc. First International Workshop on Information
Hiding, Lecture Notes on Computer Science, Cambridge, UK, pp. 207-
226, June 1996.
[11] A. Netravali, B. Haskell, Digital Pictures Representation and
Compression, Plenum Press, New York, 1988.[12] S.
Voloshynovskiy, S. Pereira, A. Herrigel, N. Baumgartner, T. Pun,
A generalized watermark attack based on stochastic watermark
estimation and perceptual remodulation, in: P.W. Wong, E.J.
Delp (Eds.), IS&T/SPIE's 12th Annual Symposium, Electronic
Imaging 2000: Security and Watermarking of Multimedia Content
II, SPIE Proceedings, Vol. 3971, San Jose, CA, USA, 23} 28,
January 2000.

119
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Multiple Embedding Using Semi-Fragile Watermarks for

Medical Images
Dominic Osborne and Derek Abbott1, Matthew Sorell and Derek Rogers2

Digital medical images require that information
quality cannot be comphromised. With increasing
demands to communicate these types of images
over wireless systems, the need to verify image
integrity is mandatory. This paper develops a
technique using embedded watermarking to verify
that important detail has not been comphromised
as a result of incidental degradation or unexpected
compression.
Keywords: Semi-Fragile Watermarking, Authentication,
Medical Images.
these types of images. This can be extended to any
image with a critically important region. As this type
of watermarking specifically survives JPEG
compression, transmission of a small image file is
possible without sacrificing image quality in the ROI
where no watermark is placed. This scheme can be
applied in one of two ways illustrated in Fig. 1.

1. INTRODUCTION

Increasingly medical images are acquired, stored and

transmitted digitally. This is especially the case for
digital images that are used in radiology [1]. As these
types of images are typically of large size,
compression allows for the cost of storage to be
reduced and the speed of transmission to be increased.
Fig. 1. Two possible wireless image scenarios. The first wireless
Although the cost of transmission bandwidth is link involves dual encoding of the ROI and ROB resulting in a
decreasing there remains a need of authentication for larger file size and improved ROI quality. The second transmission
these types of images and provision for compression involves the complete lossy JPEG encoding of the entire image or
so that feasible transmission is possible. The use of a any near-lossless pixel encoding methodology, such as GIF or
JPEG-2000.
simple system that provides some compatibility with
current image compression standards is essential as
The first operates externally to the JPEG standard and
complex compression schemes are expensive to
allows for the entire image to be compressed to
develop and deploy [2] and allow for very limited
specified level. This results in good bit rate
usage. This paper presents a technique that can be
performance consistent with typical levels of JPEG
used to verify the integrity of medical images prior to
compression at the extent of some degradation of the
any diagnosis that is made after transmission over a
ROI. The other scheme operates by embedding a
wireless link through the use of semi-fragile
watermark is an identical way, but encoding the DCT
watermarking, which is robust to JPEG [3]
coefficients directly into a hybrid-coded JPEG-like
compression. This has provided a level of assurance
file. This technique results in minimal degradation
that important detail is present and has not been lost
and near-lossless encoding of the ROI at the expense
as a result of incidental degradation over a noisy
of a larger file size and slightly inferior bit-rate
channel. The type of image that will be used to
performance. This must function within the
highlight this type of detail includes a spiral hairline
framework of the JPEG standard rather than operating
fracture, which is a classic example of this type of
as an external system.
diagnostic detailed information. A special subset of
authentication watermarking is implemented around
the ROI into the ROB to provide authentication of

1
Centre for Biomedical Engineering (CBME)
The University of Adelaide, SA 5005, Australia e-mail dosborne@eleceng.adelaide.edu.au, dabbott@eleceng.adelaide.edu.au
2
School of Electrical and Electronic Engineering
The University of Adelaide, SA 5005, Australia, e-mail matthew.sorell@adelaide.edu.au, derek.rogers@unisa.edu.au

120
 2. PROBLEM STATEMENT
Adding small amounts of noise to corrupt the
bitstream of an image file that has been channel-coded
does not usually affect the importance of the
diagnostic features present in the image after
transmission has taken place. Incidental distortions
that are not corrected through channel decoding [4]
may slightly distort the file structure of the
compressed image file without any noticeable change
to perceptual quality. This could involve a loss of
diagnostic feature information, which for medical
images is detrimental as detailed density information
is mandatory. Hence it is critical to authenticate image
quality prior to any diagnosis that is made [5]. A
classic example of this type of feature information is
shown in the Infant's Fracture of Fig. 2.
Fig. 2. Non-displaced hairline fracture from the leg of an infant,
which is often invisible on initial radiographs. If this type of image
was transmitted from a hospital to a mobile hand-held device there
would be an immediate need to evaluate image quality as a result of
the possible loss of image transform coefficients in the ROI or
unexpected levels of compression that might degrade feature
content in this region.
To maintain as much detail as possible, digital
medical images are typically stored without loss of
information using lossless compression schemes. The
long term digital storage or mobile transmission of
such images is prohibitive without the use of lossy
image compression to reduce the image file sizes. As
a typical example a mammogram may be digitized at
20482048 pixels at 16 Bits Per Pixel (bpp), leading
to a file which is over 15 megabytes in size [6]. The
use of lossless formats is widely accepted because no
image information is discarded and data is
interchangeable from one format to another. This
leads to a different representation of the image file,
but guarantees consistent visual appearance and
diagnostic quality of the image. For widespread
usage, lossy compression involves the use of JPEG
standards. The most common of these which is
implemented in most hardware is Baseline JPEG. This
involves performing the Discrete Cosine Transform
(DCT) [7] on 88 image pixels to create micro-
blocks, quantization of these coefficients and entropy
coding of the result. Contrary to excellent
developments in lossy image compression, these types
121
of schemes are viewed with suspicion by many
members of the scientific and medical community
who believe that image alteration may lead to loss of
diagnostic or scientific value.
3. PREVIOUS WORK IN ROI WATERMARKING
The concept of ROI watermarking was first proposed
By A. Wakatani [8] who placed signature information
into the ROB. A progressively compressed version of
a signature image is used and the most significant
information is embedded into the region closest to the
ROI. This method allows for the signature image to be
detected with moderate quality from a clipped version
of the image that included the ROI. This system was
intended for use over web-based medical image
database systems with primary focus placed on
ensuring copyright and intellectual property
protection. The ROI area in the original image is
specified prior to compressing the signature image
using a progressive encoding algorithm to generate a
bitstream. This allows for increasing visual detail as
the extracted bitstream is followed. The payload is
embedded into pixels around the ROI in a spiral way
as depicted in Fig. 3
Fig. 3. Since the ROI is the most critical aspect of a medical image,
it may be clipped to include the ROI. The signature image is
compressed using Embedded Zerotree Wavelet (EZW) coding so
that the whole image can be reproduced with average quality and
the entire signature image can be retrieved. The quality of the
resulting signature image is directly correlated with the length of
the bitstream extracted.
Another recent ROI watermarking scheme is proposed
by Lie et al [9], which is designed to operate within
the framework of the JPEG-2000 standard, targeting
ROI compressed images. A dual watermarking
scheme is proposed in which critical image content is
to be authenticated. Two different types of
watermarks are used, one being fragile and the other
robust. The embedding process for the robust
watermark takes place at differing resolution layers to
ensure that malicious changes are detected and
provides flexibility in determining the extent of
alteration to discriminate intended attacks from
unintended ones. In order to accurately detect which
areas have been altered, the first watermark W1 which
is sensitive and fragile is hidden in the ROI. The
second watermark, W2 is composed of features of
mid-frequency wavelet sub-bands and is robustly
watermarked into the ROB using features from the
ROI as the signature.
The robust watermark proposed is designed to survive
after acceptable levels of low-pass filtering and JPEG-
2000 compression and not to survive malicious
attacks. This signature is based on wavelet coefficient
properties of the ROI, where features are extracted
based on absolute differences between corresponding
coefficients in the LH3 and HH3 subbands on 8 8
blocks. Similarly in this work a signature is based on
the absolute differences between corresponding
coefficients in adjacent DCT tiles from inside the
ROI, which have been uncorrelated as part of the
signature extraction process. It is mentioned in [9],
that the procedure degrades the ROB significantly,
however this is not a primary concern as the ROB
area is typically encoded at a low quality and gains
minimal attention from users. The main focus of the
works by [9] and [8] is copyright protection and
assurance that malicious attacks on the embedded
watermarks are prevented. Our focus is primarily
concerned with integrity verification as images are to
be transmitted in error-prone lossy transmission
channels, such as those encountered in mobile phone
telephony to degradation experienced in Wireless
Local Area Networks (LANs.) The most useful
contribution in our work is assurance of ROI image
content integrity after image files are subject to
incidental degradation in these environments. This is
made possible with extraction of DCT signature
coefficients from the ROI and embedding multiply in
the ROB.
4. WATERMARKING TECHNIQUE USED
If the signature information is lumped and localized
within the ROB it is possible to authenticate and
verify the diagnostic integrity of such images. A
simple method to multiply watermark involves
embedding in the same shape of the ROI in the eight
regions surrounding the ROI or fewer regions if the
space in the ROB is unavailable. A visual impression
of this method is shown in Fig. 4
Fig. 4. Multiple embedding in the ROB: The algorithm embeds a
signature in the eight regions surrounding the ROI or in fewer
regions if space is unavailable. Watermarking takes place
following the direction of the arrows.
122
Multiple embedding can give the receiver additional
confidence in the unlikely event that both a watermark
and signature are corrupted in an identical way and
the watermark is falsely detected as authentic. It may
also be of benefit if one watermark is corrupted.
Semi-fragile (or robust) watermarking is specifically
designed to withstand application specific
transformation operations, such as lossy compression
and geometric distortions, but is designed to be
corrupted as a result of undesirable alterations
including malicious manipulations and incidental
degradation over mobile links which may or may not
be perceptible to the receiver. Semi-fragile robust
signature embedding ensures that the watermark
survives JPEG compression or slight degradation up
to a point where the value of the work is lost. Because
ROI compression has been successfully subjectively
evaluated in ROB of diagnostic medical images [10],
the radiologist can have greater confidence that the
diagnostic value of the image has not been
comphromised.
The basis of singular semi-fragile watermark
extraction and embedding was initially developed by
C. Y. Lin [11]. Standard lossy image compression
systems involves converting an image into some
transform domain, such as wavelet or block DCT
domain and quantizing the coefficients in order to
reduce their entropy. Coefficients are quantized to a
level proportional to how easy it is to perceive
changes in them and the property of quantization of
coefficients is exploited to remove redundancy in the
image. Let x q be the result of quantizing x to an
integral multiple of a quantization step size, q.
x
x q = q + 0.5 (1)
q
Consider s to be a real valued scalar quantity and q1
and q2 as quantization step sizes with q2 q1 , then:
((s q1 ) q2 ) q1 = s q1 (2)
If s is quantized to an even multiple of the larger step
q1 and then by a smaller step q2 the effect of the
second quantization can be reversed. The watermark
should survive as long as the quantization that is
performed during compression uses smaller step sizes.
The watermark embedding and extraction procedure
is designed to survive typical levels of JPEG
compression, where images are quantized in the block
DCT domain. The quantization step size for each
coefficient depends on its frequency. These step sizes
are obtained by multiplying the transform coefficients
by a predefined quantization table, which is scaled by
a constant. A signature is extracted from the low
frequency terms of the micro-blocks of the ROI and
embedded into the high frequency terms of the ROB
as a semi-fragile watermark, which is illustrated on a
block level in Fig. 5.
Fig. 5. Spatial representation of the bits used for the signature (left)
from the ROI and the bits used for the watermark (right)
corresponding to the peripheral regions. Matching shades indicate
where the payload bits go.
Low-frequency coefficients are used as the payload as
they represent the most important picture information
that cannot be degraded through incidental losses.
High frequency coefficients can be used for the
embedding in the peripheral regions, as these areas
are diagnostically less important and will be degraded
through compression. A signature for the image is
extracted by converting the medical image into its 88
block DCT representation and grouping microblocks
of the image into pseudo-random pairs according to a
specified seed. For each pair of DCT blocks, 8
corresponding low frequency coefficients are
compared to obtain 8 bits of the binary signature.
Consider two blocks that have been grouped Ca and
Cb, then:
0 : Ca [i, j ] < Cb [i, j ]
Signature =
1 : Ca [i, j ] Cb [i, j ]
where i and j are the coordinates of a low frequency
coefficient from Fig. 5. A hundred randomly selected
grayscale images were tested for their Peak Signal to
Noise Ratio (PSNR) after using embedding took
place, as seen in Fig. 6.
Fig. 6. Deterioration of image quality with increasing embedding
strength with resulting distortion quantified with the Peak Signal to
Noise Ratio (PSNR). This is to be expected and is unavoidable as
robust watermarking is used, which is nearly always perceptible to
the observer. The tradeoff here is the greater the embedding
strength, the more compression the image can survive at the
expense of a perceptually degraded image.
The greater the embedding strength, the more
compression the image can survive and the more
perceptible the watermark will be. This is not a
problem as removal of the watermark can be
performed easily at the receiving end.
5. SYSTEMATIC IMPLEMENTATION
The main sub-systems used in the systematic design
of the ROI semi-fragile watermarking scheme
designed to work within the framework of JPEG are
illustrated in Fig. 7.
(3)
Fig. 7. Dual encoding scheme designed to work within the
framework of the JPEG standard. A ROI is specified and copied
from an image that then undergoes a block-based DCT and
quantization to minimize the number of non-zero coefficients for
the purposes of high compression resulting in improved bit-rate
performance. The compression level is specified by the user as a
quantization table multiplier. Sub-systems standard to JPEG are
shaded in grey.
The image undergoes a block-based DCT specified by
a tile or block size, which is typically 88 pixels.
These coefficients are rounded and quantized and
Huffman [12] encoded. Those areas not shaded in
grey include operations within the framework of the
standard that can be used for more accurate ROI
integrity verification than the system that operates
externally to JPEG. The ROI in its transform
representation replaces the same region in the full
image whose coefficients have been quantized. This
ensures that the ROI is stored near-losslessly while
the ROB is compressed using lossy JPEG
compression and contains at least one watermark.
6. ROBUSTNESS TESTING
Survival of JPEG compression is one of the primary
requirements of the ROI watermarking scheme. This
is mandatory if operation external to JPEG is
required, where the pixel-based image compression
method is treated as part of the communication
123
channel. The watermarked image is permitted to
undergo types of lossless compression, which will not
degrade the image pixels or lossy JPEG, which can be
applied up to a threshold specified by the user by the
embedding strength. Robustness to varying levels of
JPEG compression took place on 100 grayscale
images of arbitrary types and varying resolutions from
256256 to 12801280 pixels. The ROI was specified
to occupy a sufficiently small area at the center of
each image so that 8 watermarks could be embedded
around this region. Results are shown in Fig. 8.
Fig. 8. Testing the robustness of 100 images with the semi-fragile
ROI watermarking scheme designed to withstand JPEG
compression. As expected, the system fails the authentication test
consistently after each of the three watermark embedding levels.
The results of this test demonstrate that the ROI
watermarking scheme survives JPEG compression
levels up to and exceeding the watermark embedding
strength used on 90% of the images. This is shown to
be consistent for three typical JPEG compression
levels. These results are almost identical to those
obtained in [13], in which the performance of a
similar watermarking method was tested where a
signature was extracted from an image and a singular
watermark embedded in the same region. As the
scheme developed involves embedding a signature
into the same coefficients in 8 8 DCT transform
blocks, it is expected to survive similar levels of
compression resulting in correlating sets of results.
Approximately 10% of the images do not survive
JPEG compression for quantization levels exceeding
the embedding strength. This problem can easily be
rectified by setting the embedding strength slightly
above the level of required JPEG compression.
7. BIT-RATE PERFORMANCE OF HYBRID
CODING SCHEME
If the image is sufficiently robustly watermarked and
converted to spatial domain for complete JPEG
compression, the resulting file size is directly related
to the level of compression. This results in a bit-rate
124
performance that is identical to JPEG. If degradation
of the ROI through JPEG quantization is not
permissible and hybrid coding is preferred as
illustrated in the flow diagram of Fig.7, the bulk of the
bit budget will be stored in the ROI. This is because
quantization does not take place in this region and all
ROI transform coefficients must be encoded, which
are typically non-zero. As the ROB can undergo
compression through quantization, the majority of
coefficients will be zero. This will result in a file size
that is dependent on the size of the selected ROI. The
larger this region is, the more near-lossless
compression is required, the larger the file size. For a
typical fracture or tumor, the area of the ROI does not
usually extend beyond 20% of the entire image. This
is also verified in work in [14] and [10], where ROI
Maxshift JPEG-2000 compression was utilized to
compress these types of medical images. Strom et al
[6] also validated the effectiveness of combined lossy
and lossless JPEG compression with these types of
ROI sizes. It was shown through extensive subjective
testing that the diagnostic value of the medical image
did not degrade for very low bit rate coding. These
approaches reinforce that the ROI is exactly the area
where all diagnostic information is located. Bit-rate
performance was evaluated in Fig. 9 with and without
the use of watermarking and with sizes of ROI
varying from complete lossy compression, where the
peripheral regions were the entire image to the
extreme of having the entire image encoded near-
losslessly as a ROI.
Fig. 9. Bit-rate performance which can be compared with entire
lossy JPEG compression, where the area of the ROI is zero to entire
lossless compression in which the percentage of the ROI is 100.
The most practically applicable areas of these curves
include those areas up to and around having a 20%
area devoted to the ROI. Within this area of the curve
the use of one or more embedding regions does not
significantly affect the size of the medical image file.
It would appear that each embedding region in Fig. 8
increases the file size by approximately 0.1 bpp,
which without watermarking results in a compression
level of 2 bpp. The increase in file size is insignificant
in comparison with complete near-lossless JPEG
encoding that provides minimal compression of 5 bpp
of the original grayscale image. After baseline JPEG
compression, which corresponds to using a
quantization multiplier of 1, it is typical for the
resulting quantized 88 DCT coefficient ROB blocks
to be reduced from 64 to an average of 2 coefficients.
This confirms the majority of the bit-budget is
devoted to the ROI, which makes the compression
scheme useful for low bandwidth mobile
communication. This does assume that the ROI
chosen is relatively small in comparison to the rest of
the image.
8. CONCLUSION
The use of watermarking can be used to verify the
integrity of digital images. A method is developed
which is designed to be used on images that have
critically important regions. The scheme is designed
to perform multiple watermarking around the ROI
containing features from this region. This technique
can provide assurance that this region has not been
corrupted as a result of incidental degradation
resulting from a wireless link or that the ROI has not
exceeded a compression level threshold. As
watermarked images can be compressed, a smaller file
size is achievable. This facilitates verification of ROI
integrity as well as wireless communication. The
method used allows the user to evaluate quality of this
region in a received image without the need of a
reference image. This is most useful for transmitted
medical images where high levels of quality assurance
are mandatory prior to making any diagnosis. The
technique used can be systematically designed in two
ways, one which fits in a modular way into the JPEG
standard resulting in minimal changes and hybridly
coding the ROI and the ROB resulting in superior
ROI quality. Alternatively this method can operate
extraneously to the standard providing greater
compliance with improved bit-rate performance. This
results in a degraded ROI when lossy JPEG is used on
the watermarked image pixels or improved picture
quality if lossless picture encoding techniques are
used. This method could also be used to monitor
image or video quality for quality control systems or
benchmarking image/video processing systems and
algorithms. A limitation is that authentication based
on watermarking cannot replace classical
cryptographic authentication protocols that protect
communication channels. Embedded robust
watermarking for ROI integrity verification can allow
for compression and provision of image integrity.
This can be most useful for medical images that must
be transmitted quickly in a wireless environment.
125
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[1] D. Osborne, D. Abbott, R. Coutts and D. Rogers, An Overview
of Wavelets for Image Processing for Wireless Communications,
SPIE Smart Structures, Devices and Systems, vol 4935, 2002, pp
427-435.
[2] D. Clunie, Lossless Compression of Grayscale Medical Images
Effectiveness of Traditional and State of the Art Approaches
[3] G. K. Wallace The JPEG Still Picture Compression Standard,
Communications of the ACM 34, 4th April, 1991, pp 30-45.
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Importance Watermarking for Image Verification in Radiology,
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Transform, IEEE Transactions on Computer Theory, vol c-23,
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Using a Compressed Signature Image, Proceedings of the 35th
Hawaii International Conference on System Sciences. IEEE
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[9] W. N. Lie, T. L. Hsu and G. S. Lin, Verification of Image
Content Integrity by Using Dual Watermarking on Wavelets
Domain, IEEE International Conference on Image Processing
(ICIP), vol 2, 2003, pp 487-490.
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Coding in Medical Imaging Applications, Proceedings of the 2nd
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 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
A Wavelet-Based Watermarking for Still Images
Corina Nafornita1
Abstract We present a robust watermarking method
for still images, which uses the similarity of the discrete
wavelet transform (DWT) and the human visual system
(HVS). In order to make the mark imperceptible, the
lowest frequencies are left unmodified, and the rest of
the coefficients from the other sub-bands are spatially
selected using an adaptive threshold. We test the
robustness of the mark against different types of attacks,
thus evaluating the robustness of the method proposed
herein. We compare our performances with another
frequency-based watermarking method.
Keywords: watermarking, copyright protection, discrete
wavelet transform
I. INTRODUCTION
In the last decade we have been witnesses to an
explosion in the use and distribution of digital
multimedia data. PCs with Internet connections have
made the distribution, both legal and illegal, of data
and applications much easier and faster [1].
Since ancient times, there have been ways of
establishing the identity of the owner of an object in
case of dispute which range from simple inscribing
the name of the owner on the object to embedding the
owners seal in the object (like a tattoo on the head of
slave) [2]. In the digital world, though, more
sophisticated means are required to ensure the same,
since copying and reproducing works of others has
become extremely easy and the reproduced work
generally spreads at the speed of light across the
globe.
While encryption is a solution to protect the data
transmitted from seller to buyer, watermarking has
been proposed as a solution to ensure the copyright
protection.
Digital watermarks can be used to identify the works
as belonging to a company or individual. Watermarks
encrypt the information as an imperceptible signal,
which is added to the data in such a way that it is
always retained [3].
Common types of multimedia data are image, video,
audio data. Our paper concentrates on the
watermarking for still images, although the same
principles can be applied to both video and audio data.
To be effective in the protection of the ownership of
intellectual property, the watermark should be [2, 4]:
1
Politehnica University of Timisoara, Communications Dept.
Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail corina@etc.utt.ro
126
1. difficult/impossible to remove, at least
without visibly degrading the original image,
2. robust against image modifications that are
common to typical image-processing
applications (e.g. scaling, dithering,
cropping, compression),
3. imperceptible to the human visual system
(HVS),
4. detectable with or without the original signal
informed decoder and blind decoder,
respectively,
5. resistant against the ownership deadlock
known as the IBM attack, appears whenever
in the same data there are several watermarks
claiming the same copyright. A solution is
proposed by Craver et al in [5]: invertible
and quasi-invertible watermarking schemes.
Current watermarking techniques for multimedia data
developed in literature are spatial/time domain
methods [14] and frequency domain methods [11-13].
Another possible classification is spread-spectrum
(SS) techniques [6] and non-SS techniques, such as
the QIM developed by Chen et al [7]. Our method
embeds the watermark in the wavelet domain, and
uses the characteristics of the human visual system by
selecting the coefficients from each subband with a
thresholding scheme.
The paper is organized as follows. Section II
describes the proposed method. In Section III we
present the simulation results and some attacks.
Finally we give some concluding remarks.
II. PROPOSED METHOD
The discrete wavelet transform (DWT) decomposes
the image into a high-high (HH), high-low (HL), and
low-high (LH) subband for each resolution level, and
a low-low (LL) subband for the coarsest resolution
level. The LL band is also known as the
approximation subimage because it contains most of
the information from the image. The HL, LH, HH
subbands are the detail subimages containing the
horizontal, vertical and diagonal details. The details of
the image such as edges and textures are confined into
the HH, LH, and HL subbands of the DWT of the
image. We take into account the fact that the HVS is
not sensitive to small changes in high frequencies of
the image, but is rather sensitive to changes affecting
the smooth parts of the image, that is, the coarsest
resolution level of the image. Therefore, we place the
mark into the wavelet domain, specifically, into the
HH, LH, and HL subbands, selecting only part of
these coefficients, leaving the LL subband
unmodified.
A. Insertion procedure
Let X be the original gray-level image and the
watermark W a pseudo random sequence, with binary
() { }
values: w i 1,1 and length N w . The basic
steps for embedding the mark are:
(a) Wavelet decomposition of the original image by L
levels to obtain a multiresolution decomposition:
Y = DWT ( X )
{ }
= LL , HL , LH , HH , HL
x
L
x
L
x
L
x
L
x
L 1 ,..., HH
(b) Compute threshold for each subband
Let the approximation coefficients be c m, n and ( )
the detail coefficients from the resolution level j and
sub band s be d s , j ( m, n ) , where s {h, v, d }
and j {1,..., L} . The threshold is computed as
follows
Ts , j = q j max{d s , j (m, n )}
m,n
where q j is a level-dependant variable.
(c) Embed watermark
For each subband, if the detail coefficient is higher or
equal to the above computed threshold, embed the
watermark using
d sw, j ( m, n ) = d s , j ( m, n ) 1 + w ( m, n ) , (6)
where is a parameter that controls the level of the
watermark.
(d) Compute the IDWT from these new coefficients
w
We obtain the watermarked image X .
It is obvious that the higher the strength of the
mark and the lower the variables q j are, the more
robust yet visible the watermark will be.
B. Extraction procedure
The extraction process requires the original image, or
at least some significant vector extracted from the
DWT of the cover work, specifically, the detail
coefficients with a value above the computed
threshold.
To extract the mark from the watermarked possibly
distorted work, X
w
, we make use of the wavelet
coefficients d s , j (m, n ) , that should contain a
watermark bit:
d s , j (m, n ) d s , j (m, n )
w (m, n ) = sgn , (7)
d s , j (m, n )
A random guess is made for the watermark bit in the
location (m, n ) if d s , j (m, n ) = d s , j (m, n ) or if
d s , j (m, n ) = 0 .
If the mark has been embedded in different locations
several times, the most common bit value is assigned
for the recovered watermark bit.
We make use of the correlation coefficient to compare
the original and the extracted mark:
w(n )w (n )
Nw
c(w, w ) = n =1
(8)
w (n ) w (n )
Nw 2 Nw 2
1
x
n =1 n =1
where ( ) [ ]
c w, w 1,1 . If the correlation
coefficient is above a specified threshold, the
watermark is positively detected in the image.
III. SIMULATION RESULTS
We performed simulations using several images Lena,
Boat, Barbara, Peppers, all with size 256 x 256 (Fig.
(5) 1). The watermark was a binary pseudo-random
sequence with N w = 256 . The Daubechies 10pt
wavelet was used to produce the wavelet coefficients.
In all tests we used the following parameters: the
number of resolution levels L = 3, the strength of the
watermark = 0.1 , and the level-dependent
variables q1 = 0.06 , q2 = 0.04 and q3 = 0.02 .
We extract the watermark in two ways (Fig 1):
- from all levels, using a majority rule,
(detector NC1)
- from the coarsest level only (since the lowest
frequencies are not so affected by common
signal distortions). (detector NC2)
We investigate the effect of common signal
distortions (median filtering, JPEG compression,
AWGN) on the correlation coefficient between the
original and the recovered mark. We compare the
performances of our method with the results obtained
using the method proposed by Cox in [6]. The
watermark used was bipolar and its length was for a
better comparison, 256 bits. Also, the number of
repetitions of the mark was the same in both cases.
The watermarked images using our method were not
significantly distorted from the originals, whereas for
the method presented by Cox et al the difference was
clearly visible. The following table shows the values
127
of PSNR for each watermarked image, as a measure
of the distortions introduced by the watermark:
PSNR, proposed PSNR, Cox et al
method method
Lenna 45.39 dB 27.19 dB
Boat 44.35 dB 25.35 dB
Barbara 44.18 dB 26.44 dB
Peppers 45.55 dB 25.75 dB
We present for each image the detector response as a
function of the filter size M, compression ratio and
signal-to-noise ratio, in case of median filtering, JPEG
compression and additive white noise, respectively.
The detector response was computed as a mean value
of 32 responses for 32 uncorrelated watermarks (Fig.
2-5).
The plots marked with the o and + symbols are the Consiliul National al Cercetarii Stiintifice din
results from the proposed method, with the detector
NC1 and NC2 respectively, while the remaining plots
are from the method proposed in [6].
Setting the threshold value in the detection process at
0.5 we have the followings.
Median filtering attack:
For all watermarked images, except Boat, the attack
by median filtering with filter size larger than M=3
leads to a correlation smaller than 0.5. In fact, only
the detector NC2 allows filtering with filter size M=3.
For Boat watermarked image, not even the NC2
detector is successfully used in finding the mark.
JPEG compression:
For Lenna, the correlation is smaller than 0.5 at a
compression rate of 16 (detector NC2 and Cox) and
10 (NC1), respectively.
For Boat and Barbara, the correlation is smaller than
0.5 at a compression rate of 13 for NC2, 10 for Cox
and 7 for NC1.
For Peppers, the compression rate values for which
the correlation is smaller than 0.5 is 15 (NC2, Cox)
and 8 (NC1).
AWGN attack:
For Lenna and Peppers, the detector response in the
Cox et al method is above 0.5 at a signal-to-noise
ratio of 5 dB, having a considerably better
performance than detector NC1 (12 dB) and NC2 (15
dB).
For Boat and Barbara, the detector values are
approximately the same for each method: 3 dB (Cox),
around 14 dB (NC2) and 7 dB (NC1).
IV. REMARKS
We proposed a robust wavelet-based watermarking
method that embeds the mark in coefficients selected
in such a manner that the visible impact on a human
128
observer isnt very high. By embedding the
watermark bits into the edges and textures of the
image we make use of the human visual system. One
can see that both methods, proposed in [6] and ours
are image-dependant. Apparently, the Cox method is
superior for AWGN attack, comparable with the NC2
detector in the case of JPEG compression, and inferior
for median filtering. However if we take into account
the fact visibility of the mark, an essential aspect of a
watermarking system, it is possible that our methods,
with the two proposed detectors (NC1 and NC2) to be
considered comparable or better than the Cox method
in the given situation.
Future work will concentrate into the study of coding
the watermark bits for a better performance.
ACKNOWLEDGEMENT
This work was supported by a grant from the
Invatamantul Superior, Romania, cod CNCSIS 47
TD.
REFERENCES
[1] G.Voyatzis, I. Pitas, Problems and Challenges in Multimedia
Networking and Content Protection, TICSP Series No. 3, Editor
Iaakko Astola, March 1999.
[2] I. Cox, M. Miller, J. Bloom, Digital Watermarking, Morgan
Kaufmann Publishers, 2002.
[3] A. Sequeira, D. Kundur, Communications and Information
Theory in Watermarking: A Survey, Multimedia Systems and
Applications IV, A. G. Tescher, B. Vasudev, and V. M. Bove, eds.,
Proc. SPIE (vol. 4518), pp. 216-227, Denver, Colorado, August
2001.
[4] M. Borda, I. Nafornita, Digital Watermarking Principles and
Applications, Proc. Of Int. Conf. Communications 2004, pp.41-54.
[5] S. Craver, N. Memon, B. Yeo, M. Yeung, Resolving Rightful
Ownerships with Invisible Watermarking Techniques: Limitations,
Attacks, and Implications, IEEE Journal On Selected Areas In
Communications, Vol. 16, No. 4, May 1998.
[6] I. Cox, J. Killian, T. Leighton, T. Shamoon, Secure Spread
Spectrum Watermarking for Multimedia, IEEE Transaction On
Image Processing, 6, 12, pp.1673-1687, 1997.
[7] B. Chen, G. W. Wornell, Quantization Index Modulation: A
Class of Provably Good Methods for Digital Watermarking and
Information Embedding, IEEE Trans. On Information Theory,
Vol. 47, No. 4, May 2001.
[11] D. Kundur, D. Hatzinakos, Diversity and Attack
Characterization for Improved Robust Watermarking, IEEE
Transactions on Signal Processing, Vol. 49, No. 10, pp. 2383-2396.
[12] C. Nafornita, A. Isar, Digital Watermarking of Still Images
using the Discrete Wavelet Transform, Buletinul tiinific al UPT,
Tom 48(62), Fascicola 1, 2003, pp. 73-78.
[13] C. Nafornita, M. Borda, A. Kane, A Wavelet-Based Digital
Watermarking using Subband-Adaptive Thresholding for Still
Images, microCAD 2004 International Scientific Conference,
University of Miskolc, 18-19 March 2004, pp.87-92.
[14] N. Nikolaidis, I. Pitas, Robust Image Watermarking in the
Spatial Domain, Signal Processing, Vol. 66, No. 3, pp. 385-403,
1998.
 (a) (b)

(c) (d)
Fig. 1: Original images used for simulations: Lenna (a), Boat (b), Barbara (c) and Peppers (d).

129

(a)
(b)
(c)
Fig. 2: Detector response to attacks against watermarked Lena:
median filtering (a), JPEG compression (b), AWGN (c). The
plots marked with the o and + symbols are the results from the
proposed method, with the detector NC1 and NC2 respectively,
while the remaining plots are from the method proposed in [6].
(a)
(b)
(c)
Fig. 3: Detector response to attacks against watermarked Boat
(median filtering, JPEG compression, AWGN). The plots marked
with the o and + symbols are the results from the proposed
method, with the detector NC1 and NC2 respectively, while the
remaining plots are from the method proposed in [6].
130

(a)
(b)
(c)
Fig 4: Detector response to attacks against watermarked Barbara:
median filtering (a), JPEG compression (b), AWGN (c). The
plots marked with the o and + symbols are the results from the
proposed method, with the detector NC1 and NC2 respectively,
while the remaining plots are from the method proposed in [6].
(a)
(b)
(c)
Fig 5: Detector response to attacks against watermarked Peppers:
median filtering (a), JPEG compression (b), AWGN (c). The
plots marked with the o and + symbols are the results from the
proposed method, with the detector NC1 and NC2 respectively,
while the remaining plots are from the method proposed in [6].
131
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

The statistical behaviour of the chaotic signals: application
to cryptography
Adriana Vlad1,2, Adrian Luca1
Abstract The paper is devoted to the analysis of the Section II presents the investigations for the first order
statistical behaviour of chaotic signals having in view statistical description (the probability distribution
their suitability for the cryptografic applications. The function), [3]-[8]. Some results are displayed in
investigation is illustrated using the chaotic signals Table 1 and Fig. 2. With this purpose we considered
provided by the logistic function. The procedure of
investigation combines information theory notions with the (0; 1) interval (the chaotic signal values)
statistical inferences based on one or two data sets. The consisting of Q = 27 non-overlapping equal length
following statistical tools are considered: probability intervals.
estimations with multiple confidence intervals, test on
probability and test on equality between probabilities.
The type II statistical error plays a special role in the
design of the experimental data size.
Key words: chaotic signals, ergodicity, statistical
inferences on probability, type II statistical error, noisy
information channel.

I. INTRODUCTION

The chaotic behaviour [1], [2] has been noticed for

many dynamic systems in discrete time. For such a Fig. 1. A sample of the random process which models the
system, the x k state at time k depends on the chaotic system; R = 3.9 and x 0 = 0.113 .

previous state, x k +1 = f ( x k ) . Several types of

f functions can be considered in practice. Our paper
considers the logistic function:
x k +1 = Rx k (1 x k )
where the R parameter belongs to the (0;4) interval
and the x k value belongs to the (0; 1) interval. The
chaotic behaviour is met when R 3.5699456 .
Notice that all the numerical results presented in our
paper correspond to the logistic function; however,
the investigation can be applied to any chaotic
systems.
Let us consider the equation (1). Fig. 1. shows x k as a
function of k {0,1,...,200} , corresponding
x 0 = 0.113 and to R = 3.9 . When considering the
same R = 3.9 value but different x0 initials values, a
set of different curves are obtained. We consider each
of these curves as a sample of a random process.
Hence, the sample space is the (0;1) interval in which
x 0 is randomly selected [3], [8].
Section III investigates the conditional probabilities
p( I j / I i ) , i.e. the probability that a trajectory
(chosen randomly from the ensemble) passes through
Ij interval at k 2 iteration on the condition that at k1
(1)
iteration the same trajectory passes through Ii interval.
A problem was whether we can speak or not about the
statistical independence of the two discrete random
variables sampled at k1 and k 2 iterations.
Section IV is devoted to the applications of chaotic
systems to the cryptographic field.
II. FIRST ORDER STATISTICAL
DESCRIPTION
to Be the random sequence obtained from the chaotic
signal when, instead of the continuous random
variable sampled at each k iteration, we consider a
discrete random variable with Q values. The Q
discrete values are assigned to the Q non-overlapping
equal length intervals covering the (0; 1) interval of
values taken by the chaotic signal. We illustrate the
procedure of investigation for Q = 27 intervals. We

1
Faculty of Electronics and Telecommunications, POLITEHNICA University of Bucharest
2
The Research Institute for Artificial Intelligence, Romanian Academy
corresponding address: adriana_vlad@yahoo.com

132
started the study with Q = 27 intervals having in view texts.
some immediate applications in enciphering natural
Table 1. First order statistical description of the random process modelling the chaotic systems.
k = 200 k = 300 k = 500 k = 2000
p r p r p r p r
I1 0.0000; 0.0370 0 0 0 0 0 0 0 0
I2 0.0370; 0.0741 0 0 0 0 0 0 0 0
I3 0.0741; 0.1111 0.05000 0.08543 0.05380 0.08219 0.05493 0.08130 0.05160 0.0840
I4 0.1111; 0.1481 0.03980 0.09626 0.04330 0.09212 0.04450 0.09082 0.04590 0.08935
I5 0.1481; 0.1852 0.03500 0.10291 0.03540 0.10231 0.03590 0.10156 0.03510 0.10276
I6 0.1852; 0.2222 0.02810 0.11526 0.02770 0.11612 0.03030 0.11087 0.02450 0.12367
I7 0.2222; 0.2593 0.02500 0.12116 0.02320 0.12717 0.02600 0.11996 0.02290 0.12802
I8 0.2593; 0.2963 0.02020 0.13650 0.02240 0.12948 0.02200 0.13067 0.02250 0.12918
I9 0.2963; 0.3333 0.02170 0.13160 0.02190 0.13098 0.02140 0.13253 0.02090 0.13414
I10 0.3333; 0.3704 0.07290 0.06989 0.06990 0.07149 0.07000 0.0750 0.07110 0.07084
I11 0.3704; 0.4074 0.04290 0.09257 0.04380 0.09155 0.03910 0.09716 0.04250 0.09303
I12 0.4074; 0.4444 0.04990 0.08552 0.04750 0.08776 0.04840 0.08690 0.04800 0.08728
I13 0.4444; 0.4814 0.02500 0.12240 0.02800 0.11164 0.02740 0.11677 0.02410 0.12472
I14 0.4815; 0.5185 0.02670 0.11833 0.02990 0.11164 0.03000 0.11145 0.02900 0.11341
I15 0.5185; 0.5556 0.02420 0.12445 0.02650 0.11870 0.02500 0.12240 0.02720 0.11721
I16 0.5556; 0.5926 0.02380 0.12552 0.02310 0.12745 0.02200 0.13067 0.02330 0.12689
I17 0.5926; 0.6296 0.02350 0.12634 0.02400 0.12498 0.02360 0.12606 0.02830 0.11484
I18 0.6296; 0.6667 0.02580 0.12043 0.02310 0.12745 0.02350 0.12634 0.02650 0.11879
I19 0.6667; 0.7037 0.02620 0.11949 0.02460 0.12341 0.02310 0.12745 0.02290 0.12802
I20 0.7037; 0.7407 0.02680 0.11810 0.02430 0.12419 0.02360 0.12606 0.02300 0.12774
I21 0.7407; 0.7778 0.02665 0.11879 0.02370 0.12579 0.02470 0.12326 0.02680 0.11810
I22 0.7778; 0.8148 0.02940 0.11261 0.02630 0.11925 0.02885 0.11443 0.02870 0.11402
I23 0.8148; 0.8519 0.03260 0.10676 0.03080 0.10994 0.02850 0.11443 0.03000 0.11144
I24 0.8519; 0.8889 0.06710 0.07308 0.07170 0.07052 0.06680 0.07325 0.06360 0.07520
I25 0.8889; 0.9259 0.07270 0.06999 0.07040 0.07122 0.06780 0.07267 0.07210 0.07031
I26 0.9259; 0.9630 0.10590 0.05690 0.11020 0.05569 0.11000 0.05575 0.11090 0.05549
I27 0.9630; 1.0000 0.07780 0.06747 0.07450 0.06908 0.08300 0.06514 0.07860 0.06710

Fig. 2. Histograms for the frequencies distribution of Table 1: Q = 27 intervals, at k = 200 (left) and at
k = 500 (right); on the vertical axis the occurrences of the intervals.

133
We started the investigation with verifying the first
order stationarity of the chaotic signal by considering
the Q discret intervals. This implies a comparative
study of the discrete random variables sampled at
different iterations.
We determine the probablity that at the k iteration
the chaotic signal passes through a certain Ii interval
(chosen from the Q possible intervals). Then, we try
to verify if this probability depends or not on the
k iteration (the sampling time) while preserving the
same Ii interval. For the statistical inferences used the
experimental data should comply with the i.i.d. model
(i.e. observations coming up from independent and
identically distributed random variables). Moreover,
for the statistical tests we compared independent data
sets.
The first experimental results are presented in Table 1
where we considered four different iterations:
k = 200 , k = 300 , k = 500 , k = 2000 and
N = 10000 trajectories for each sampling time. Note
that for each k sampling time we used N = 10000
different trajectories, generated by different initial
conditions (randomly chosen from (0; 1) interval)).
Hence, for the four iterations k = 200 , k = 300 ,
k = 500 , k = 2000 we had at our disposal four
independed i.i.d. data sets (that means we generated
40000 different trajectories of the chaotic signal).
For example for the I10 = (0.3333; 0.3704) interval,
at k = 200 , the estimated value of the p probability
that the chaotic signal passes through I10 is
p = m / N = 0.07290 (m is the occurrence number of
the investigated interval in the considered i.i.d. data
set).
Each time we experimentally checked-up the de
Moivre-Laplace conditions in the form
Np (1 p ) 14 , [4]-[7]. As a consequence we can
say that the p true (unknown) probability lies inside
the ( p * (1 r ); p * (1 + r )) = (0.06780; 0.07799)
interval computed with 1 = 0.95 statistical
confidence level;
r = z / 2 * p (1 p ) / N = 0.06989 is the relative
experimental error, where z / 2 = 1.96 is the
/ 2 point value corresponding to the standard
Gaussian law (of 0 mean and 1 variance).
For the same I10 = (0.3333; 0.3704) interval, but at
k = 500 iteration, the estimated value is p = 0.07000 ,
the 95% confidence interval is (0.06499; 0.07500)
and the relative experimental error r = 0.0750 . It
can be noticed that the two confidence intervals for
the probability overlap; this brings some evidence in
the favor of the stationarity assumption.
The fact that the confidence intervals overlap
encouraged us to a more detailed investigation.
134
Thus, we continued the study with applying the test
on the equality between two probabilities (see
Appendix ).
We succesively compared the two data sets (one for
k = 200 and another one for k = 500 ) for each Ii
interval ( i = 1 27 and j = 1 27 ) in Table 1.
Table 2 presents the results only for I10, I12, I18 and I22
intervals. All the four tests were passed; the test
values and the decisions are shown in Table 2. As a
conclusion, the stationarity assumption is again
sustained.
Table 2. Experimental values for the test on the
equality between two probabilities
I10 I12 I18 I22
Test: T1
z 0.0819 1.2248 1.2910 1.4523
H 0 / H1 H0 H0 H0 H0
2*10-11 9.8*10-6 0.1532 0.0328
Because all tests are passed, the probability of type
II statistical error (that means H 0 accepted, although
the two compared probabilities are not equal) is
important. It was computed according to (4) (see
Appendix).
Fig. 3 shows the values as a function of p1 .
There are three plots for values: = 0.1 ,
= 0.15 , = 0.2 , N1 = N 2 = N = 10000 , = 0.05 .
Table 2 presents the values for the corresponding
intervals, when = 0.20 . For = 0.10 , values
are much larger. For a better accuracy (low values for
while < 0.15 ) we need to resume the experiment
generating much more trajectories of the chaotic
signal.
Fig. 3. The type II error size for the test on equality between
probabilities. On the horizontal - p1 probablity ; on vertical -
values. The curves corresponding to:
= 0.1 - +, = 0.15 - and = 0.2 -
 Fig. 4. Frequency distribution representation: the histograms if we discretize in Q=6 intervals at k = 200
(left) and at k = 500 (right); on the vertical axis the occurrences of the intervals.

Fig. 2 shows histograms coresponding to the probability of the test was the temporal value for the
frequency distribution from Table 1. Instead of the corresponding Ii interval in Table 3.
relative frequencies of the intervals, the histograms
are constructed on the basis of intervals occurences. Table 3. Temporal description
The study was resumed for Q=6 intervals. This x0 = 0.31 x0 = 0.456 x 0 = 0.758
number of Q=6 intervals could be of some interest in
the cryptographic field when two iterations are I10 0.0725 0.0711 0.0708
simultaneously considered and assigned to an I12 0.0481 0.0487 0.0483
alphabet character of the natural language (for
exemple letters, punctuation marks). Fig. 4 presents I18 0.0214 0.0219 0.0213
histograms for the random process modelling the
I22 0.0253 0.0273 0.0234
chaotic signal.
For a temporal description we generated several
individual trajectories of the chaotic signal for Thus, the null hypothesis H 0 has the form
L = 10000 iterations. We measured how many times H 0 : p = p 0 where p 0 denotes the temporal
the investigated trajectory (randomly chosen from the
ensamble) passes through a certain interval of values; probability obtained for a certain trajectory.
~ the occurrence number. The relative occurrence We successively applied this test (Table 4) for I10, I12,
be m
I18 and I22 interval considering the i.i.d. data sets
number ~ p=m ~ / L was computed for each I interval of
i
obtained at k = 500 . The theoretical p 0 probabilities
values (Ii is the same from Table 1 where we
are those from Table 3 and the trajectory with initial
discretized the (0; 1) interval in Q = 27 non-
condition x 0 = 0.31 . All the tests were passed, thus
overlapping intervals of equal length).
sustaining again the ergodicity assumption.
Another issue was if ~ p=m ~ / L (the temporal relative
frequency of the investigated interval) lies inside of Table 4. Test of probability
the confidence interval for the probability Test I10 I12 I18 I22
corresponding to the same Ii investigated interval (at
k iteration). p0 0.0725 0.0481 0.0214 0.0253
As an illustration we used three curves with initial 0.9641 0.1402 0.8983 0.7642
z
condition: x 0 = 0.31 , x0 = 0.456 si x 0 = 0.758 and
four investigated intervals: I10, I12, I18 and I22 (see H0/H1 H0 H0 H0 H0
Table 3). We computed the temporal relative
frequency ~ p=m ~ / L of the investigated interval for
III. SECOND ORDER STATISTICAL
each trajectory. For example for the trajectory with DESCRIPTION
x 0 = 0.31 the temporal relative frequency
corresponding to I10 interval is ~
p = 0.0725 . Here, we again consider the (0; 1) interval of values of
Looking at Table 1, the 95% confidence interval for the chaotic signal discretized in Q non-overlapping
the probability assigned to I10 interval at k = 500 was intervals of equal length. For the second order
(0.06499; 0.07500) . We can see that ~ p lies inside statistical description we shall consider
this confidence interval for the probability. simultaneously two iterations ( k1 and k 2 ).
We resumed this type of investigation for each Ii This leads to the noisy information channel shown in
interval and several trajectories; all the numerical Fig. 5. Fig. 5 illustrates our procedure of
results sustained the ergodicity assumption of the first investigation considering Q = 6 intervals. As a
order distribution function. consequence, the input space X = {x1 ,..., xi ,..., x 6 }
We continued the verify this type of ergodicity by corresponds to the Ii interval at k1 iteration and the
using a test of probability [4], [6], [7]. In this test the
i.i.d. data sets is the same we used in Table 1 for a { }
output space Y = y1 ,..., y j ,..., y 6 corresponds to the
fixed k (the considered iteration) and the theoretical Ii interval at the k 2 = k1 + k iteration.

135
 Table 6. Noise matrix estimation (proportions)
X Y p(yj/xi)
y1 y2 y3 y4 y5 y6
x6 I6
I1 y1 0.1020 0.0977 0.2239 0.1011 0.1236 0.3518
x1
x5 I5
x2 0.1229 0.1066 0.2051 0.1192 0.1048 0.3415
xi Ii
x3 0.1184 0.1055 0.2129 0.1144 0.1100 0.3388
p( y j / xi ) Ij yj
x4 0.1020 0.1067 0.2229 0.1199 0.1265 0.3220

x5 0.1108 0.1092 0.2024 0.1117 0.1209 0.3451

x1 I1 I1 y1 0.1142 0.1128 0.1939 0.1053 0.1205 0.3533
x6
k1 k 2 = k1 + k p(yj) 0.1157 0.1107 0.2010 0.1059 0.1191 0.3476
Fig. 5. The channel diagram (transition graph)

Table 7. The mutual information of the information

The problem is whether the random discrete variables channel
sampled at two iteration are statistically independent k1 k2 I ( X ;Y )
or not; in the affirmative case, for what k = k 2 k1 k = 10 200 210 0.016035
distance we can think of independence.
With this purpose we verify if the following relation 300 310 0.013822
p( y j / x1 ) = ... = p ( y j / x 6 ) = p ( y j ) is valid or not.
k = 30 200 230 0.001164
p( y j ) is the probability that the chaotic signal passes
300 330 0.002171
through Ij interval at k 2 iteration and p( y j / xi ) is
k = 50 200 250 0.001642
the probability that a trajectory (chosen randomly
from the ensemble) passes through Ij interval at k 2 300 350 0.001812
iteration on the condition that at k1 iteration the same
trajectory passes through Ii interval. This time the mutual information was computed
For a quick decision concerning their independence, considering a large number of trajectories:
we computed the mutual information, (2). N = 50000 . We can notice that when we can speak
about independence the k distance is larger than for
I ( X ; Y ) = H ( X ) H ( X / Y ) = H (Y ) H (Y / X ) (2) Q = 6 (Table 7).

The Table 6 shows the conditional probabilities Table 8. The mutual information of the channel
p ( y j / xi ) for k1 = 200 and k 2 = k1 + 50 = 250 . k1 k2 I ( X ;Y )
The mutual information corresponding to Table 6 is k = 10 300 310 0.136066
very low: I ( X ; Y ) = 0.001642 . This suggest the
independence between the input and the output (also k = 20 300 320 0.011214
revealed in Table 6 by the equality between k = 50 300 350 0.008182
probabilities estimates p( y j / xi ) p( xi ) , i = 1 6
and j = 1 6 ). k = 500 300 800 0.008017
We resumed this procedure of verifying the statistical k = 1000 300 1300 0.007521
independence. Table 7 shows some results that
indicate a k = k 2 k1 distance for which we can
speak about independence; this happens for k 30 . We also computed the conditional probability and the
The investigation was carried out on N = 10000 mutual information I ( X ; Y ) for different k1 and
trajectories. k = k 2 k1 . All results sustain the second order
This procedure based on the noisy information stationarity for the discrete random process assigned
channel assigned to discretized chaotic signal was to the chaotic signal.
further resumed for Q = 27 intervals. Some results
are presented in Table 8. IV. CONCLUSION AND OPEN PROBLEMS

136
This paper suggests how to obtain from the chaotic
signal a stationary discrete information source (and
according to case practically zero-memory) having the
same symbols as a printed natural language. For
example we can generate an information source with
Q = 27 symbols that may correspond to printed
Romanian (the alphabet whitout blank and
punctuation marks), where we omit some very low
frequency characters. The message generated by this
information source (provided by the chaotic signal)
may be a key in a various enciphering methods.
An immediate example that can be further used in
different variants is to make a summation modulo
Q (successively for each character) between the
plaintext and the key. On the basis of the entropy
(redundancy) of the information source corresponding
to the key and also using some knowledge about the
entropy of natural language (the plaintext) we can
evaluate the performance of the cipher.
ACKNOWLEDGEMENT
The autors wish to thank to Professor Jean-Pierre
Barbot, the head of the ECS research group from
ENSEA, Cergy, France where this study was started,
for his assistance and support.
APPENDIX
In the appendix we briefly present the test on the
equality between two probabilities:
Be there two samples each complying with i.i.d.
statistical model with the sample
N1 = N 2 = N = 10000 . Denoting by m1 the number
of successes of the event in the first data sample, the
probability estimate is p 1 = m1 / N1 . Similarly, in the
second data sample, the probability estimate is
p 2 = m 2 / N 2 . The two statistical hypotheses (null
hypotheses H 0 /alternative hypotheses H 1 ) are:
H 0 : p1 = p 2 and H 1 : p1 p 2 . We have to verify
whether the two estimates p 1 and p 2 derive from the
same theoretical probability. We apply the test based
on the z test value defined in (3):
depends on the p1 and p 2 = p1 (1 ) value for fixed
, N1 and N 2 . It is denoted by ( p1 , p 2 ) and is
computed according to equation (4):
*z / 2
1 (x ( p1 p2 ))2
( p1, p2 ) = 2
exp(
2 2
)dx
*z / 2 2
(4)
where:
= p(1 p )(1 / N1 + 1 / N 2 ) ;
= p1 (1 p1 ) / N1 + p 2 (1 p 2 ) / N 2 .
REFERENCES
[1] Al. Serbanescu, Aplicatii ale sistemelor haotice in comunicatii,
Editura Academiei Tehnice Militare, Bucuresti, 2004.
[2] M.S. Baptista, Cryptography with chaos, Physics Letters, A
240, (1998), pp. 50-54.
[3] Al. Spataru, Fondaments de la theorie de la transmission de
linformation, Presse Polytechniques Romandes, 1987.
[4] J. Devore, Probability and Statistics for Engineering and the
Sciences, second edition, Brooks/Cole Publishing Company,
Monterey, California, 1987.
[5] A. Vlad, B.Badea, Metode statistice in prelucrarea datelor.
Estimarea statistica, Editura Paideia, Bucuresti, 2002.
[6] A. Vlad, B. Badea, M. Mitrea, Metode Statistice in Prelucrarea
Informatie. Compendiu si Aplicatii, Editura Metropol, Bucresti,
1999.
[7] V. Craiu, Verificarea Ipotezelor Statistice, Editura Didactica si
Pedagogica, Bucuresti, 1972.
size [8] A. Vlad, M. Ferecatu, M. Mitrea, Teoria transmisiunii
informatiei II: elemente teoretice ilustrate in MathCAD, Ed.
Paideia, Bucuresti 2002.

z = ( p 1 p 2 ) / p1 (1 p1 ) / N1 + p 2 (1 p 2 ) / N 2 ,
where p1 = p 2 = p (m1 + m 2 ) /( N1 + N 2 ) (3)

If z z / 2 (where the z / 2 is / 2 point value

corresponding to the standard Gaussian law of 0 mean
and 1 variance) then we shall consider that the two
probabilities are equal. Otherwise, i.e. when
z > z / 2 , we reject the equality hypothesis at an
significance level.
Type II error means not to reject H 0 although it is
false. This happens when the test value passes the test,
however p1 p 2 . The probability of this situation

137
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Considerations and Results in Multimedia and DVB

Application Development on Philips Nexperia Platform
Radu Arsinte1, Ciprian Ilioaei2
Abstract This paper presents some experiments
regarding applications development on high
performance media processors included in Philips
Nexperia Family. The PNX1302 dedicated DVB-T kit
used has some limitations. Our work succeeded to
overcome these limitations and to make possible a
general-purpose use of this kit. For exemplification some
typical applications, important both for multimedia and
DVB are analyzed: MPEG2 video stream decoding and
MP3 audio decoding being the most popular. These
original implementations are compared (in speed,
memory requirements, and costs) with Philips Nexperia
Library.
Keywords: Multimedia, Media Processors, DSP, DVB
I. INTRODUCTION
Modern multimedia embedded applications is present
in different forms in our life. DVB set-top boxes,
DVD players, satellite receivers are few examples of
these kind of well-known products.
Implementing embedded multimedia applications is
possible only by using high performance processors.
Using general PC based platforms for development is
possible, but the goals to achieve lowest cost, lowest-
consumption products are possible only by using so
called Media Processors.
Such processors are present in the offer of many large
semiconductor companies. Some examples are given
in [1], [2].
Philips, a recognized pioneer in video-audio
many video and audio data formats in real time,
PNX1300s are well suited for a broad range of
applications such as Internet appliances, Web-cams,
smart display pads, video and screen phones, PVR,
videoconferencing, video editing, video security,
Internet radios, DVD, wireless LAN, and digital TV
sets and set-top boxes. They also support applications
in a JavaTM virtual machine environment.
Supported by the comprehensive TriMediaTM SDE
software development environment, PNX1300s are
comparable in ease of programmability to general-
purpose processors. The SDE enables multimedia
application development entirely in the C and C++
languages.
Our work was intended to make an exploration of the
Trimedia (Nexperia) and integrate this technology
into a general multimedia system development. Our
previous work in DVB technology was rather
theoretical [3], [4], and this occasion, to use a high
performance processor has offered the opportunity to
start real-time embedded multimedia
implementations.
II. DEVELOPMENT SYSTEM-DESCRIPTION
The system used for present development was initially
designed for straight DVB-T applications.
The block diagram is presented in fig.1.
Connector CI MS interface JTAG interface

technology is involved in development of a high-

performance, low-cost media processors, Nexperia Tuner
Decodor
canal DVB-T
PNX1300 Series which delivers up to 200 MHz of TDA10046
FLASH
power to a variety of multimedia applications. Video out
PNX1300 Series processors achieve over seven Video encoder
SAA7129
Processor PNX1302
billion operations per second in applications requiring Audio out

real-time processing of video, audio, graphics, and Audio encoder RAM

communications datastreams.
PNX1300 processors are ideal building blocks for
devices required to process several types of PCI slot Keyboard Boot Receiver IR
multimedia datastreams simultaneously, including the Interface EEPROM

latest standards such as MPEG-4, MPEG-2, H.263, Keyboard

MP3, and Dolby Digital. With ample computational

power available to capture, compress, and decompress Fig.1. Block diagram of the DVB-T Nexperia board

1
Facultatea de Electronic i Telecomunicaii Cluj-Napoca, Departamentul Comunicaii, Str. G.Baritiu 26-28,
e-mail radu.arsinte@com.utcluj.ro
2
Tedelco SRL, Calea Turzii 42, Cluj-Napoca

138
 Few details regarding this block schematic.
1. Processor PNX1302
- offers data processing capabilities
2. Tuner
- RF processing of incoming TV signal
- re-encodes the audio/video information into RF
channel
3. Channel decoder (TDA10046)
- COFDM demodulation
- outputs TS (Transport Stream) to Nexperia
4. CI Connector (Common Interface)
- Links the receiver module with CI (Conditional
Access) module
- Transmits a scrambled signal and receives the
descrambled signal
5. Video encoder
- transforms video output stream (VO) of PNX1302 in
CVBS PAL/SECAM/NTSC
- video data transfer is performed using ITU656
standard
6. Audio encoder
- Converts audio I2S in analog audio
7. PCI slot
- standard PCI interface used to add (interface) of
compatible devices
8. Flash memory
- stores the executable program
9. RAM memory
- temporary stores data and settings
10. MS (Micro Stick) interface
- used to store data (removable peripheral)
11. Keyboard Interface
- used to read local keys
12. JTAG interface
- used in debugging processes
Most of the components are common for a
microprocessor system. The core is the Nexperia
processor with features adapted to real-time
processing of audio-video data.
All TriMedia (Nexperia) processors consist of an
internal 32-bit high-speed data bus, which is
connected to external SDRAM. Attached to this
highway are chip internal DMA interface blocks, the
VLIW CPU, and coprocessor blocks.
A heavily simplified diagram is presented in Fig.2 [5].
Fig.2. Block diagram of the Nexperia processor
139
III. SOFTWARE DEVELOPMENT IN NEXPERIA
ENVIRONMENT
A professional application is constructed within the
framework of software architecture optimized for
streaming multimedia data. This framework allows
software modules to be developed independently
because it clearly defines the interface between these
components. A programmer can easily integrate
diverse modules as they connect in a common way.
This software architecture is known as the TriMedia
Streaming Software Architecture (TSSA) [6]. Several
dozen TSSA components are now available, and they
are used extensively in the design of the complete
product. TSSA uses a data driven design. The RTOS
provides a foundation that allows the system to be
factored into independent tasks that communicate
using queues and semaphores. A given task will sleep
until data is available, process the data, send it along,
and sleep again.
The architecture of a TSSA application is given in
Fig.2. [5]
The priority-based scheduler of the RTOS kernel
handles scheduling. Priorities are generally set using a
rate-monotonic rule. A priority-based scheduler is
chosen over a deadline scheduler because of its
predictable behavior in overloaded conditions. High
priority tasks continue to meet their deadlines, while
low priority tasks are deferred.
TSSA has several features. Some may be useful for a
given application, some may not. TSSA brings in all
aspects of the TriMedia software architecture. It
describes a method of constructing and connecting
autonomous, task-based components that
stream data between them.
TSSA provides a framework for components, whether
streaming or not, that includes:
- A standard Application Programmer Interface (API)
- Common data formats (as defined in header files
that are contained on the CD)
Fig 3. TSSA architecture for Nexperia applications [5]
Unfortunately this way to develop applications
depends of libraries delivered by Philips and third
parties. For educational purposes an application could
be developed in more traditional way in C or C++,
without a streaming architecture.
Software support for Nexperia family has a main
component IADK (Integrated Advanced Development
kit). IADK contains the libraries of all the 4.AAC decoder/player: this application decodes and
components needed in applications and the NDK plays the encoded AAC files
(Nexperia Development Kit). We had also software Characteristics:
support for the stand-alone systems (SAS). - program was tested on Philips_ATV1 board
The environment has the following folders: - using non-streaming architecture
-audio: the libraries for the audio software - limitation given by RAMDISK size
components like Audio Digitizer, Audio Renderer or - doesnt work in real time(at this moment)
MP3 Decoder. - compiled with nohost option for our board
-video: libraries for the video components like Video
Digitizer, Video Renderer, Mpeg decoder etc. 5.AAC encoder :this application encodes .wav and
-tssa: libraries for some components that make some .pcm files to AAC format
actions like File Reader or Copy IO Characteristics:
-mdm: libraries for Transport Stream Demux and - program was tested on Philips_ATV1 and works
Programme Stream Demux components satisfactory
-net :libraries for HTTP network communication - is using non-streaming architecture
support and for RPC sockets - compiled with nohost option for our board
-build: the directory where we built components file
libraries and the applications for our board. 6.AVI decoder :this application decodes
-sas: contains the SAS environment support. uncompressed AVI files and displays the images
Characteristics:
IV. RESULTS - program was tested on Trimedia Zapper board
- uses non-streaming architecture
The development system was used to test some - limitation given by RAMDISK size
original applications useful in laboratory works and - the program works fine for the small AVIs
demonstrations. Most applications avoid the copyright - compiled with nohost option for our board
problems using original implementations for
algorithms and code sequences. This was a 7.MPEG-2 video decoder: this application implements
requirement of the project, making the applications the MPEG-2 decoding algorithm (IDCT, Huffman,
independent of IADK, which costs about 10.000$ for etc) and displays the images
the smallest configuration. It is necessary to have only Characteristics:
NDK, which has an affordable cost. - program was tested on Philips_ATV1 board and
Here are some of the applications and brief results of it works
tests. - almost reaches real-time(93-95%)
- using non-streaming architecture
1.PCM Player: this application plays PCM files - compiled with no-host option for our board
(*.pcm), which contain audio PCM sequences
Characteristics: 8.Image processing: some standard operations like
- program was tested binarization, edge detection were implemented over a
- using non-streaming architecture picture.
- has a video interface, coming from YUV files
- limitation given by the RAMDISK size Every application has a complex structure, and a
- compiled with nohost option for our board thoroughly implementation. For exemplification we
are presenting some details of the MP3 encoder
2.YUV Player: this application displays images that implementation.
are read from the .y, .u and .v files
Characteristics: Digital
Audio
Bit or
Quantized
Bitstream
Encoded
Filter Samples Bitstream
- program was tested on DVB T board and works Input
Bank Noise Formatting

according to specifications Allocation

- using non-streaming architecture

- no memory limitation
- compiled with nohost option for our board
Signal to
Psychoacoustic Mask Ratio

3.MP3 Player: this application plays mp3 files up to Model

128kb bitrate
Characteristics:
- program was tested on DVB-T board
- doesnt works in real time(for the moment)
- limitation given by the RAMDISK size
- compiled with nohost option for our board
Fig.4. Block diagram of the MP3 encoder
This simple diagram (fig.4), extracted from MPEG
standard [7] results in a lot of tables and associated
files, and presenting them is far beyond the space
allocated for this article.

140
The source files for the encoder have at least 4000
program lines (including tables).
After compilation this results in an .out file of about
10MB. This could look a large file, but the file
includes the source file (MP3), the input and output
buffers. This is a small amount of the RAM available
(32MB) in the DVB-T system used in our
experiments.
V. CONCLUSION AND FUTURE WORK
Our activity brought us the following achievements:
1. Extended work with the compiler and with other
Trimedia tools;
2. Unterstanding how makefiles work;
3. Creating the executables (*.out) for some specific
applications ;
4. Simulating those executable files with tmsim ;
5. Building the support for all given platforms:
Foxbox (ATV), DVE, Trimedia Zapper ;
6.Generating the library files (*.a) for all the software
Trimedia components that came with the two support
CDs.
Using low level and complexity software tools offers
the possibility to obtain results comparable with the
TSSA architecture, in small applications or
educational environments.
Obviously, the total efficiency is lower than a
professional application, using libraries optimized
many times, but for most applications efficiency is
less important than simplicity and affordability.
141
Future work will be concentrated on program
optimization, and extension of the application base.
For the moment the goal remains educational
application development, but some applications could
be ported on embedded processors from Nexperia
Family (PNX85xx series) for commercial usage.
REFERENCES
[1] Texas Instruments - TMS320DM641/TMS320DM640 Video/
Imaging Fixed-Point Digital Signal Processors Data Manual,
June 2003 .
[2] ST Microelectronics - STi5518 Single-Chip SET-TOP BOX
Decoder with MP3 and Hard Disk Drive Support data sheet -
2001
[3] Radu Arsinte, Ciprian Ilioaei - Some Aspects of Testing Process
for Transport Streams in Digital Video Broadcasting Acta
Technica Napocensis, Electronics and Telecommunications, vol.44,
Number 1, 2004
[4] Radu Arsinte - A Low Cost Transport Stream (TS) Generator
Used in Digital Video Broadcasting Equipment Measurements
Proceedings of AQTR 2004 (THETA 14) - 2004 IEEE-TTTC-
International Conference on Automation, Quality and
Testing,Robotics May 13-15, 2004, Cluj-Napoca, Romania
[5] Chuck Peplinski, Torsten Fink - A Digital Television Receiver
Constructed Using A Media Processor- Philips Semiconductor
1999
[6] *** - TriMedia Programmers Reference Manual (SDE 2.0)
1999, Philips Semiconductors
[7] *** - International Organisation for Standardisation - ISO/IEC
JTC1/SC29/WG11 Coding of Moving Pictures and Associated
Audio - ISO/IEC JTC1/SC29/WG11 - November / 1994
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Shape Similarity Measure For k Nearest-Neighbor

Queries
Irina G. Mocanu1
Abstract One of the content based image retrieval
techniques is the shape based technique which allows
users to ask for objects similar in shape to a query
object. In this paper, a region-based approach to shape
representation and similarity measure is presented. The
proposed algorithm is based on the grid descriptor
method. Its performance is compared with the grid
descriptor method in case of the k nearest-neighbor
queries. The performance is tested using a database of
synthetic shapes. Experimental results show that the
proposed method performs favorably compared with the
grid based method in case of the k-nearest neighbor
queries.
Keywords: shape representation, shape similarity
measure, image retrieval, grid descriptor
I. INTRODUCTION
With the explosive growth of multimedia
applications, the ability to index or retrieve
multimedia objects in an efficient way became an
increasingly active area. A major data type stored and
managed by these applications is represented by two
dimensional objects. Objects contain many features,
like color, texture, shape. Shape is an important low-
level image feature. The shape representation of
objects can be used for indexing, retrieval and as a
similarity measure.
A good shape representation and similarity
measurement for recognition and retrieval purposes
should have the following two important properties
[1]: (i)each shape should have a unique
representation, invariant to translation, rotation, and
scale; (ii)similar shapes should have similar
representations so that retrieval can be based on
distances among shape representations.
There are generally two types of shape descriptors [2]:
contour-based shape descriptors and region-based
shape descriptors. Contour-based shape descriptors
exploit only boundary information, they cannot
capture shape interior content and these methods
cannot deal with disjoint shapes. In contrast, in
region-based techniques all the pixels within a shape
region are taken into account to obtain the shape
representation. These techniques can describe disjoint
shapes.
The paper presents a new method for shape similarity
measure based on the grid descriptor method. The
grid method is a region based method [1]. The
proposed algorithm outperforms the grid descriptor
method in case of k nearest-neighbor queries.
The rest of the paper is organized as following:
section II presents the grid descriptor method; section
III describes the new technique for shape
representation and similarity based on the grid
descriptor method; section IV shows the results of the
retrieval experiments; section V concludes the paper.
II. THE GRID DESCRIPTOR METHOD
The grid-based method attracts interest for its
simplicity in representation. In this method, a grid
space is overlaid over the shape [1], [3]. The grid
space consists of fixed size square cells. In Fig. 1 the
shape is mapped on to a grid of fixed cell size in a
manner that the shape is justified to the top left
corner. The grid is then scanned from left to right and
top to bottom. 1 is assigned to the cells of the grid
partially or wholly covered by the shape and 0 to the
cells outside of the shape boundary, which gives us a
sequence of numbers which can be used for shape
representation.
Fig. 1 Mapping a shape to a grid.
The sequence of 1s and 0s for the shape boundary is
defined the binary number for the shape boundary.
The binary numbers obtained for the shape boundaries
in Fig. 1(a) and Fig. 1(b) are 001111000 011111111
111111111 111111111 111110011 001100011 and
001100000 011100000 111100000 111100000
011111100 000111000 respectively. The difference
between the two shapes can be calculated as the
number of cells in the grids that are covered by one
shape and not the other and hence the sum of 1s in
the result of the exclusive-or of the two binary

1
Universitatea Politehnica Bucuresti - Facultatea de Automatica si Calculatoare, Sectia Calculatoare
Spl. Independentei Nr. 313, 060032, Bucuresti, e-mail irinam@cs.pub..ro

142
numbers [1]. In the above case the difference between
the shapes is 27 by XOR operation on the two sets.
Hence, in grid method two objects are similar in
shape, if and only if the difference between their
binary representations is less than a prespecified
threshold, and they have similar eccentricities.
However, it must be noted that the binary number
obtained for the same shape with a different
orientation in space or with a different scale will be
different. However, it must be noted that the binary
number obtained for the same shape with a different
orientation in space or with a different scale will be
different. The criteria for invariance of indices is not
met and hence it is required to normalize the shape to
achieve scale, rotation and translation invariance. The
normalization process involves three steps: (i) shape
boundaries are normalized for rotation, (ii) they are
normalized for scale, (iii) they are normalized for
translation. The principals steps of computing grid
descriptor are: (i) binary image, (ii) major axis, (iii)
rotation normalization, (iv) scale normalization, (v)
translation normalization, (vi) scan grid cells.
The following definitions are needed to perform the
normalization process [3]:
Major axis: is the straight line segment
joining the two points on the boundary
farthest away from each other (in case of
more than one, select any one);
Minor axis: is perpendicular to the major axis
and of such length that a rectangle with sides
parallel to major and minor axes that just
encloses the boundary can be formes using
the lengths of the major and minor axis;
Basic rectangle: the above rectangle formed
with major and minor axis as its two sides.
A shape after rotation will have a different binary
number. This is because rotation changes the spatial
relationships between the grids and the shape. This
problem can be solved by normalizing the shape for
rotation. The purpose of rotation normalization is to
place shape regions in a unique common orientation.
Hence the shape region is rotated such that its major
axis is parallel to the x-axis. There are still two
possibilities as shown in Fig. 2 b) and d), caused by
1800 rotation.
Fig. 2. a) a shape before rotation normalization b), c), d), e) the
shape after rotation normalization
143
Further, two more orientation are possible due to the
vertical and horizontal flips of the original region, like
in Fig. 2 c) and e).
Both the shape size and the grid size affect the binary
number derived for a boundary. This problem is
handled by choosing a fixed length of the major axis
(the standardized major axis) and then scaling the
shape in a manner that the major axis of the shape
equals the standardized major axis. Scaling
normalization is thus achieved by scaling along the
major-axis so that the major axis of the shape
becomes equal to the length of the standardized major
axis. The shape is scaled along the minor axis
proportionally in order to maintain the perceptual
similarity of the shape, like in Fig. 3 a), b) and
respectively in Fig. 3 c).
Fig. 3. a) and b) two similar shapes before scale normalization. c)
the shapes after scale normalization
To improve the efficiency of this method, another
shape feature, eccentricity was used [1]. Eccentricity
of shape is the ratio of the major axis to the minor
axis. Therefore, for two objects to be similar, their
sequences of numbers and their eccentricity values
should be similar [1]:
a) If two normalized shapes have the same
basic rectangle, the distance between them is
equal to the number of positions having
different values in their corresponding binary
sequences;
b) If two normalized shapes have very different
basic rectangles (i.e., they have very different
minor axis lengths), there is no need to
calculate their similarity, because the shapes
are very different. The difference threshold
between minor axes depends on applications
and cell size. Normally, if the lengths of the
minor axes of two shapes differ by more than
3 cells, these two shapes are considered quite
different;
c) If two normalized shapes have slightly
different basic rectangles, it is still possible
these two shapes to be perceptually similar.
It is added 0s at the end of the index of the
shape with shorter minor axis, so that the
extended index is of the same length as that
of the other shape. The distance between
these two shapes is calculated as in the first
case a).
The grid descriptor algorithm is applicable for
contour-based shape and it assumes shape boundary
coordinates have been known. This algorithm is
extended into describing region-based shape. The
main extension to the grid method is the method of
finding the major axis and region interpolation after
scale rotation.
In the case of region shape, boundary information is
not known. That is not practical to find the major axis
of a region shape by traversing all the points in the
shape region, the computation would be O(N2), where
N is the number of pixels in the shape region.
Therefore, the major axis for a region shape is found
by searching the outer border point pairs on the shape
boundary in a number of directions (e.g. 360
directions). The algorithm for calculating the major
axis involves three major steps: (i) find the bounding
box of the shape; (ii) find the pair of boundary points
in a number of directions; (iii) find the two points at
the furthest distance in the found boundary points.
The algorithm is described bellow:
1. Find the bounding box of the shape;
2. Start from a line segment d0 which passes
through the shape center, trace from the two
end points towards the center along the line
segment. If a shape point is found, it is a
boundary point. For every tracing, two
boundary points are found;
3. Increase the angle of d0 by an increment of
2 /n (n is the number of directions to trace),
repeat step 2;
4. Repeat step 3 until boundary points at all
orientations are found;
5. Find the two points p1, p2 with the furthest
distance in the above boundary points, then
p1p2 is the major axis.
III. A NEW SHAPE SIMILARITY MEASURE
Consider the following five shapes A, B, C, D, E, F
with similar eccentricities (7/4, 7/4, 7/6, 7/4, 7/4) from
Fig. 4. If we notate with d(x, y), the distance between
shapes x and y conform with the grid method, then
d(A, B) = 4, d(A, C)=4, d(A, D)=3, d(A, E)=3.
Hence, shapes D and E will be more similar with
shape A than shapes B and C, which is not
conforming to human intuition.
Fig. 4. Five shapes with their eccentricity values a) shape A,
b)shape B, c)shape C, d)shape D, e)shape E
144
If we want to extract all the images which are
similarly with the shape from a query, this method of
calculating distances between shapes doesnt influnce
the result. In the knearest-neigbor query [4], the users
query is specified by a vector and an integer k. The k
objects whose distances from the query vector are the
smallest are retrieved. Using the grid method for
evaluating the k nearest-neighbor query, the results
will not be conforming to human intuition. For
example, if we want to perform a k nearest-neighbor
query to extract shapes similar with shape A from
shapes of Fig. 4, for k=2, the result will be shapes D
and E. Shape B is the most similar with shape A,
conform with human intuition. To obtain this, a new
method to calculate shape differences was proposed.
Instead of calculating the difference between two
shapes like in grid method, we associated a weight to
each cell in the grid that are covered by one shape and
not the other,. At a first stage, this weight is chosen to
be inverse proportionally with the number of cells
neighbors which are covered by the two shapes. The
difference between two shapes will be the sum of
these associated weights. For example d(A, B)= 5-
2+5-2+5-2+5-2=8, d(A, C) =8-3+8-3+5-0+5-0=20,
d(A, D) = 8-4+8-5+8-7=8 and d(A, E) = 8-4+8-6+8-
7=7. In this case the shapes D and E are more similar
to shape A than shape C, too. But this is not sufficient.
We must differentiate between a shapes peek
produced by noise (shapes B and C) and a hole in a
shape (shapes D and E). The cells neighbors
considered for determining the associated weight of
the respective cell may form a continuous sequence
(like in case of shape B and C) or not (like in case of
shapes D and E). At a second stage, the weight
associated with a cell will be multiply with a
predefined factor in case that the considered cells
neighbors will not form a continuous sequence. For
example if =2, (A, D) = (8-4)* +(8-5)* +8-
7= 15 i d(A, E) = (8-4)* +(8-6)* +8-7=13 and
d(A, B) = 8, d(A, C) = 20. Therefore, using this
method for calculating distances between shapes, for a
k nearest-neighbor queries with k=2, the result consist
of shape B amd E, which is more similar with human
intuition than grid descriptor method.
IV. RETRIEVAL EXPERIMENTS
To test the proposed method, a retrieval framework,
on a database with synthetic shapes was implemented.
The performance has been evaluated using precision
and recall [1]. Precision is defined as the ratio of the
number of similar shapes retrieved to the total number
of shapes retrieved. Recall is defined as the ratio of
the number of similar shapes retrieved to the total
number of similar shapes in the whole database.
Precision indicates accuracy of the retrieval and recall
indicates the robustness of the retrieval performance.
For each query object the relevant items
in the database are the object shapes which are
perceptually similar to the query object shape. The
database used consists of approximate 3,000
polygons. The average precision and recall of the V. CONCLUSIONS
shapes used as k nearest-neighbor queries is given
in Fig. 5. This paper presents a new shape similarity method
based on the grid descriptor representation. and
1.2
retrieval method based on the shapes contour
1
which has a better retrieval performance compared
0.8 to the distance histogram method. The method is
Precision

0.6 invariant to translation, scale and rotation. The

0.4 distance histogram method discards spatial
0.2
information to obtain rotation invariant. In the
0
0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
proposed method, the radii together with the edges
Recall directions associated with them are used for shape
grid descriptor method proposed method representation.

Fig. 5. Average retrieval performance of the two methods for k REFERENCES

nearest-neighbor queries
For k nearest-neighbor queries, the precision and
recall of the proposed method is similar with the
grid method, but the shapes retrieved using this
method will be conform to human intuition more
than in the case of the grid method.
[1] Guojun Lu - Multimedia Database Management Systems,
Artech House, 1999
[2] Mayatham H. Safar, Cyrus Shahabi - Shape Analysis and
Retrieval of Multimedia Objects, Kluwer Academic Publishers,
2003
[3] Atul Sajjanhar, Guojun Lu - A Grid Based Shape Indexing
and Retrieval Method, Australian Computer Journal, Vol. 29,
nr. 4, pag. 131-140, 1997
[4] V. S. Subrahmanian Principles of Multimedia Database
System, Morgan Kaufmann Publishers, 1997

145
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Shape Representation and Retrieval Using Centroid Radii

and Turning Angle
Irina G. Mocanu1
Abstract - Among all issues related to Content Based In the studies which have conducted to this paper,
Image Retrieval systems, retrieving images based on the distance histogram method [11] was used. In
shapes is an important one. In the paper a contour- distance histogram method, shapes are represented
based approach to shape representation and using the distances from the centroid to its
similarity measure is presented. The shape
representation is based on centroid radii and turning
boundary (radii). The shapes will be compared by
angle. The proposed algorithm is invariant to computing their histograms using the radii. The
translation, scale and rotation. The effectiveness of method is invariant to translation, scaling and
algorithm in the content-based retrieval of shapes is rotation and does not care about the starting point
illustrated using a database of synthetic shapes. The for obtaining the radii. On the other hand, the
results of the experiments show the competitiveness of method has a drawback: in some cases, for
the algorithm. different shapes, their distance histograms will be
Keywords: shape representation, shape similarity the same. This it is happening because the method
measure, image retrieval, distance histograms, uses only the radii and does not care about the
turning angle
spatial information.
The method described in this paper is a contour
I. INTRODUCTION
based method and represents a shape by its radii
and direction of the edges to eliminate the
Interest in the potential of digital images has
drawback of the distance histograms method.
increased enormously over the last few years. The
The rest of the paper is organized as following:
internet collection of images has become very
section II presents the distance histogram method;
large. However, the process of locating a desired
section III describes a new technique for shape
image in a large and dynamic collection emerges as
representation and similarity based on turning
a challenging problem. Hence, the image database
angle and centroid radii; section IV shows the
query problem is becoming widely recognized and
results of the retrieval experiments; section V
the search for solutions is an increasingly active
concludes the paper.
area.
Problems with traditional methods of image
II. THE DISTANCE HISTOGRAM
indexing have led to the rise of interest in
METHOD
techniques for retrieving images on the basis of
automatically derived features such as color,
This method, proposed in [11], is based on a
texture and shape. Shape of objects contained in an
histogram computed from the distances between
image is an important image feature. For retrieval
the centroid of the shape and its boundary.
based on shapes, images must be segmented into
Accordingly to this method, the main steps for
individual objects using certain methods, possibly a
representing a shape are:
semiautomatic method [12]. After that, the basic
1. Calculate the centroid of a shape, knowing its
issue of shape-based image retrieval is shape
polygon which approximates the boundary of the
representation and similarity measurement between
shape;
shape representations.
The calculation of the (x, y) centroid coordinates of
Shape matching has been approached in a number
a shape is based on Greens theorem in plane. First,
of ways like Fourier descriptors [2], curvature scale
it is necessary to calculate the area of the polygon.
space [5], turning angle [4], centroid radii [14],
Given a polygon and its vertices (xi, yi), i=0,1,,n,
distance histograms [11], Zernike moments [3],
x0=xn and y0=yn, the area of a polygon in plane is
geometric moments [1] and grid descriptions [7].
obtained using the formula [9]:

1
Universitatea Politehnica Bucuresti Facultatea de Automatica si Calculatoare, Sectia Calculatoare,
Spl. Independentei Nr. 313, 060032 Bucuresti, email: irinam@cs.pub.ro

146

1 n 1
A = x i y i +1 x i +1 y i
2 i =0
The area computed by (1) is a signed value, where
a negative sign indicates that the vertices are in
clockwise order and a positive sign indicates that
the vertices are in counter clockwise order.
The centroid coordinates are:
_
x _ y distances will be in [0, 1]. And because the sample
x= ,y =
A A
where
1 n 1
(x i+1 + x i )(x i y i+1 x i+1 y i )
x =
6 i =0
and
1 n 1
( y i+1 + y i )(x i y i+1 x i+1 y i )
y =
6 i =0
2. Select a set of sample points in the boundary of
the polygon, and calculate the distances between
the sample points and the centroid of the shape;
The number of sample points is variable. It can be
changed for different situations. But the sample
points are not uniformly distributed around the
boundary. Each edge has assigned a number of
sample points proportionally to its length. For
example, if the length of an edge is Li, the
perimeter of the shape is L, and the total number of
sample points is N, then the number of sample
points in that edge Ni will be:
L
Ni = i N
L For the two different shapes in Fig. 1, applying the
These points are evenly spread on the edge.
The sample points and the centroid of the shape are
used to calculate the distances. For example, given
a sample point si = (xi, yi) and the centroid c = (xc,
yc), the distance between them is:
d(s i , c) = ( x i x c ) 2 + ( y c y i ) 2
3. Construct a distance histogram based on the
distances obtained at step 2;
the range of all distances [0, Dmax] is separated into
several ranges, i. e. R ranges. Then, the ranges of
distances can be represented by: [0, Dmax/R],
(Dmax/R, 2 Dmax/R], (2 Dmax/R, 3 Dmax/R], , ((R-a)
Dmax/R, Dmax] and the distance histogram can be
represented as: D: (d0, d1, d2, , dR-1), where di,
i [0, R-1] is the number of distances belong to
this distance range.
(1)
4. Normalize the distance histogram;
Two similar shapes at different scales will have
different values of distances calculated at step 2. To
make this representation invariant to scale these
distances must be normalized for scaling. The
process of distance normalization consists of
dividing the value of all distances by the value of
the maximum distance. After that the value of all
(2) points are chosen based on the length of the edge,
evenly spread on it, two similar shapes with
different sizes will generate the same normalized
distances. Therefore the method is invariant to
scale after normalization.
This approach is invariant to translation because
(3) the distance set will not change after translating the
shape. The sample points are chosen in each edge
proportionally with the edges length and they are
spread evenly on it. The location of the sample
points will not change after rotating the shape.
Therefore this method is invariant to rotation
(4) because the distance set will not change after
rotating.
After representing the shapes by distance
histograms, the similarity among them can be
calculated by Euclidean distance between their
distance histograms. For example, for two shapes
with the distance histograms D1: (d10, d11, d12, ,
d1R-1) and D2: (d20, d21, d22, , d2R-1) the distance
between them is:
R 1
d(D1, D2) = (d1
i =0
i d2i ) 2 (7)
III. A NEW SHAPE REPRESENTATION
(5)
distance histogram method, their histograms will
be similar as it is shown in Fig. 2. Therefore, using
the distance histogram method the two shapes will
be similar, although they look different. This is
because the distance histogram method discards
spatial information.
(6)
Fig. 1. Two different shapes with similar histograms
To consider spatial representation, the method
proposed in this paper represents a shape using the
radii and directions of the edges. The directions of
the edges are represented by the turning angles of
each edge. Turning angle [4] is defined as the angle
147
formed with a reference axis by the counter-
clockwise tangent to the boundary of a shape which
goes from a boundary point of a shape to the next
one. In Fig. 3 the edge directions of the shape are
represented by their turning angles, where the
reference axis is considered to be the x axis.
Therefore, a shape will be represented by its edge
directions, each edge direction having associated a
list of corresponding radii.
Fig. 2. The histograms of shapes from Fig. 1
Fig. 3. A shape and turning angles of each edge
The main steps of the proposed method are:
1. determine the edges directions of the shape;
2. calculate the centroid of the shape like in
distance histogram method equation (2);
3. select a set of sample points in the boundary
of the polygon and calculate the distances
between the sample points and the centroid of
the shape, like in distance histogram method
equation (5);
4. construct a list of edges directions such that
each entry in this list will have associated all
the radii corresponding to that direction.
This representation is invariant to translation, but it
is not invariant to scale. For making it invariant to
scale, like in distance histogram method, all the
distances between the centroid and the chosen
sample points will be normalized. The
normalization of the radii consists of dividing the
value of all distances by the value of the maximum
distance. The representation is not invariant to
rotation. For a shape rotated with an angle , all
the edges directions will increase or decrease with
the same angle .
To see if two shapes are similar using this method,
first the lists of directions must be compared, and if
these directions are similar the radii associated with
them must be also compared.
148
Representing the shapes as above, one shape can
have more edge directions that the other. But it
cannot say that the shapes are different because the
shapes can be affected by noise. To make this
supposition, first it is necessary a preprocessing
stage to approximate the boundary of a shape. The
boundary approximation is a process that
eliminates insignificant shape features and reduces
the number of data points.
This preprocessing stage is used to reduce the
influence of noise and to simplify the shapes by
removing irrelevant features and keeping only
which are relevant. The boundary of the shape is
analyzed in a number of evolution steps [8]. On
every evolution step, a pair of consecutive line
segments s1, s2 is substituted with a single line
segment joining the endpoints of s1 and s2. The key
property of this evolution is the order of the
substitution. The substitution is done according to a
relevance measure K given by:
(s1 , s 2 )l(s1 )l(s 2 )
K (s1 , s 2 ) = (8)
l(s1 ) + l(s 2 )
where ( s1 , s 2 ) is the turn angle at the common
vertex of segments s1, s2, and l is the length
function normalized with respect to the total length
of a polygonal curve. The evolution algorithm
assumes that vertices which are surrounded by
segments with high value of K(s1, s2) are important
while those with a low value are not. The segments
correspond to noise are small segment pairs which
result in small values of the relevance measure K.
Thus these segment pairs are removed in an early
stage of the evolution process. The vertices of the
simplified contour are also vertices of the original
contour.
After that, the boundary of the shape will be
without noise. Now for two shapes having different
numbers of edges directions it can be said that the
shapes are not similar and this method does not
compute the distance between them.
The similarity between two shapes will be
calculated in two steps:
Compare the edges directions. For two shapes A
and B with lists of edges directions (dA1, dA2, ,
dAn) and (dB1, dB2,, dBn) they may be similar if
their directions correspond. This means that the
directions are the same, or one list of directions is
rotated with an angle . For testing this, the
differences between each pair of angles need to be
computed. If all these n differences are
approximately the same (less than a predefined
threshold) the shapes may be similar. But when
constructing the lists of edges directions it is not
known which edge is the first one. Therefore to see
if the directions of the two shapes are the same, the
differences between angles of one list of directions
and circularly shift of the other list are computed.
Doing this, the method will be invariant to rotation
and does not care about the starting point. If an
equality of all these differences is found, then the
shapes may be similar, otherwise the shapes are not
similar. If an equality between the two lists of
directions exists, suppose the order of these lists of
directions is (dA1, dA2, , dAn) and (dB1,
dB2,, dBn).
Compare the radii associated with the lists of
edges directions obtained in step 1. For example,
it must compare the radii corresponding to
direction dA1 with the radii corresponding to
direction dB1, and so on. In fact it must be
compared the lists of radii. To compare these
values it is sufficient to compare the standard
deviation of the radii associated with each
direction. The distance between the two shapes will
be the Euclidean distance between the standard
deviations of radii corresponding to each direction.
Consider that the standard deviations of radii
corresponding to each direction are (stdA1, stdA2,
, stdAn) and (stdB1, stdB2, , stdBn). Then
the distance between the two shapes will be:
n
(stdA' stdB' )
d(A, B) =
i =1
IV. RETRIEVAL EXPERIMENTS
To test the retrieval performance of the proposed
method compared with the distance histogram
method, a retrieval framework has been
implemented on a database with synthetic shapes.
The performance has been evaluated using
precision and recall [6].
Precision is defined as the ratio of the number of
similar shapes retrieved to the total number of
shapes retrieved. Recall is defined as the ratio of
the number of similar shapes retrieved to the total
number of similar shapes in the whole database.
Precision indicates accuracy of the retrieval and
recall indicates the robustness of the retrieval
performance.
The database used consists of approximate 3,000
polygons. The average precision and recall of the
shapes used as queries is given in Fig. 4.
1.2
1
0.8
Precision
0.6
0.4
0.2
0 [12] Zhang, D. S. and G. Lu: An Integrated Approach to Shape
0.1 0.2 0.3 0.4
Distance Histogram
Fig. 4. Average retrieval performance of the two methods
The method proposed in this paper outperforms the
distance histogram method by precision and recall.
This method by the preprocessing stage for
boundary approximation has a supplementary step
compared to distance histogram method. But this
stage eliminates the noise and reduces the number
of edges, making easier to obtain the sample points.
V. CONCLUSIONS
This paper presents a new shape representation and
retrieval method based on the shapes contour
which has a better retrieval performance compared
to the distance histogram method. The method is
invariant to translation, scale and rotation. The
distance histogram method discards spatial
information to obtain rotation invariant. In the
proposed method, the radii together with the edges
directions associated with them are used for shape
representation.
In the presented method a preprocessing step for
reducing noise is necessary. However this step
reduces the number of edges for a shape and
computing the sample points from the shapes
boundary is easier.
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Based Image Retrieval. The Fifth Asian Conference on
Computer Vision (2002) 652-657
Proposed Method
[13] G. Iannizzotto and L. Vita: A New Shape Distance for
Content Based Image Retrieval -
jada1.unime.it/~ianni/publications/mmm96.pdf
[14] K-L. Tan and L.F.Thiang: Retrieving similar shapes
effectively and efficiently Multimedia Tools and Applications.
Kluwer Academic Publishers (2001)
149
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Considerations about the design requirements
for analog anti-aliasing filters
Adrian Stoica, Constantin I. Vizitiu, Ioan Nicolaescu, Lucian Anton1
Abstract - This is an overview on theoretical aspects of anti-
aliasing filters, which are used together with high speed
sampling Analog to Digital Converters (ADCs) in many
applications. The main purpose is to define and to identify the
particularities for analog filters designed for baseband and
undersampling applications, in order to relieve the
contributions of filtering on ADC dynamic range
improvements.
Keywords: sampling process, Analog to Digital Conversion
and Filtering, Data Acquisition Systems.
I. INTRODUCTION
For the ADCs used in data acquisition systems, in
measurement instruments, and signal processing
applications, are specified the time-domain
performances (full scale, resolution, slew rate,
accuracy, precision, integral or differential non-
linearity, time conversion per bit etc.). In addition, the
applications of ADCs in digital communications and
high definition television require comprehensive
frequency-domain specifications (gain and bandwidth,
signal to noise ratioSNR, sampling rate, effective
number of bits-ENOB, spurious free dynamic range
SFDR etc.).
An important aspect of using ADCs in digital
communications systems is the understanding of the
sampling process and the possible causes of
distortions, which limit the system performances.
The sampling process can be discussed from either the
time or frequency domain or both. In the digital
communications applications has more results the
frequency-domain analysis, and for this reason we
assume that is the right way to study the basic aspects
of high speed sampling process.
II. THE ALIASES SIGNALS
The unwanted signals for ADCs, which result in the
signal sampling process, are named images
frequencies or alias signals.
The anti-aliasing filter is a good choice to assure an
unambiguously high speed sampling process into the
ADCs, in order to remove the unwanted signals,
which have frequencies outside of the desired Nyquist
zone of sampler.
There are two types of sampling:
Baseband sampling, for signals without
modulation;
Undersampling, for signals with modulation
and carrier frequency.
For baseband sampler we consider a single harmonic
signal, with frequency noted f a , sampled at a
frequency noted f s by ideal Dirac pulses. It must to
respect the Shannon condition f s 2 f a . But in the
frequency domain it can be identified the images
frequencies of the original signal, around every
multiple of frequency f s like in fig. 1, and these
frequencies are given by the relation:
fikm = k f s f a ;
(1)
fikp = k f s + f a ; k = 1, 2, 3, ...
f i1m f i1 p f i2 m f i2 p
fa fs 2 fs 3 fs 2 2 fs
Zone 1 Zone 2 Zone 3 Zone 4
Fig. 1 The Nyquist zones for ideal sampler and the images frequencies
The first Nyquist zone (Zone 1), defined like the basic
bandwidth (baseband), has the left limit on zero
frequency and the right limit on first half of sampler
frequency.
In the frequency domain there is an infinite number of
Nyquist zones, each of them having bandwidth f s 2 .
If the analog signal frequency increases up to f s 2
value, the alias signal with frequency fi1m decrease
direct to f s 2 value, and the frequency fi1 p increase
up to 3 f s 2 value. It can obtain f a = f s 2 = fi1m as
a critical case for correct sampling process.

1
Communications and Electronic Systems Department, Military Technical Academy, Bucharest, Address: 81-83, Bd. George
Cosbuc, Sector 5, Bucuresti, Romania, Tel. 0213354660 / 0197, e-mail: astoica@mta.ro, vic@mta.ro

150
If f a still increase, and f a f s 2 , theoretically the filter may be programmable by software to optimize
fi1m = f s f a component falls inside the first Nyquist the circuit respond and the filtering characteristics.
For practical signal processing the ideal sampler can
zone, and certainly this gives an unwanted signal at be obtained with an ADC followed by a FFT
the output of ADC. This case is similar to the analog processor, which only provide an output in the
mixing process and the alias signal fi1m = f s f a is frequency interval [0, f s 2] , even for useful signal
like an intermediary frequency for radioreceiver. or alias one. For this reason it is necessary to use the
For undersampling we consider the signal bandwidth filtering ahead the sampler circuit in order to remove
Ba = f 2 f1 = 2 f , which is symmetrical around the frequency components which are outside the first
carrier frequency, noted f cr . In fig. 2 is shown a Nyquist zone whose aliased components fall inside it.
The anti-aliasing analog filter is a good choice to
signal in the third Nyquist zone, centered around a
assure an unambiguously high speed sampling process
carrier frequency.
into the ADCs both for baseband sampling and for
Attenuation ML f cr MH undersampling.
a( f ) 2 f s f1 f1 f2 Generally, for analog filter circuits, the
3 fs f2
transition bandwidth is smaller if more poles are used
Band of in the filter design. In these conditions it is necessary
Image Image Image Image
DR

interest
to use filters with high complexity, and for most of
f ADC producers the Elliptic filters with more than 10
fs
poles are a popular choice. The Bessel filters or the
NZ = 1 NZ = 2 NZ = 3 NZ = 4 NZ = 5 Chebyshev filters (with the ripple error under 1 dB),
0 0.5 1 1.5 2 2.5
both with minimum 8 poles, represent another options
Fig. 2 The aliases signals for undersampling process
to implement the analog anti-aliasing filter. The
In this case the signal bandwidth must be Ba < f s , Butterworth filters, which achieve 6dB attenuation per
octave for each pole of filter transfer characteristic (or
and f = f 2 f cr = f cr f1 given from the Nyquist 20 dB per decade), is the most usual type of filter used
criteria with the centered carrier frequency. in low cost practical applications.
For the baseband sampled signals are used the low-
III. THE ANTI-ALIASING FILTERS pass analog anti-aliasing filter (LP-AAF), but for the
undersampled signals must use the band-pass analog
A generally block diagram for a data conversion anti-aliasing filter (BP-AAF).
system is shown in the fig. 3, with the analog filtering
at input and digital filtering of outputs in order to IV. THE DESIGN RECQUIREMENTS
reject all unwanted signals.

Analog Input Digital Outputs

The main characteristics of LP-AAF are the
followings: the pass band, the attenuation value in the
ANALOG ANALOG (Programmable) pass band ( aPass , ideally 0 dB), the attenuation value
IMPUT LOW PASS ADC DIGITAL
INTERFACE FILTER FILTER in the stop band ( a Stop , usually more than 60 dB), the
Fig. 3 The block diagram for an analog-to-digital conversion system dynamic range DR ( DR = aPass aStop ), the cut-off

In the concrete applications it can be used only the frequency (the corner frequency), simply noted f a ,
analog filtering, or the digital filtering (Fast Fourier which is equal to the highest frequency in analog
Transform FFT, Finite or Infinite Impulse Response input signal spectrum, and the transition band, which
FIR or IIR), or both in very sophisticated systems. is the interval [ f a , ( f s f a )] .
There are some differences between the filtering in Only the frequencies from transition band have the
the analog domain and this filtering process in the aliases signals in the band pass [0, f a ] , but the
digital domain.
aliases components have the levels under the limit of
The analog filtering is more suitable for high
dynamic range, regarding fig. 4.
frequency, and can remove the extraneous noise
The band pass of LP-AAF is lower than the width of
spikes which can saturate the sample-and-hold
first Nyquist zone, and in these conditions the aliased
amplifier (SHA) at the input of ADC, and the alias
components between f a and f s 2 are not interest
signals which can produce distortion in the conversion
process. and do not limit the desired dynamic range for ADC.
The digital filtering has a proper action after analog- The dynamic range of LP-AAF is chosen based on the
to-digital conversion and for this reason cant reduce requirements for signal fidelity and ADC resolution.
the influence of analog noise in the conversion Usually the stop band attenuation is between 60 dB
process, but it can remove the noise injected in ADC and 80 dB, and the transition band is the interval
parts during the conversion. In addition, the digital [0.45 f s , 0.6 f s ] .

151
 1
Attenuation fc = f s . (4)
a( f ) fa fs fa 2DR [ dB ]
20n
aPass
2 10 1

For the Butterworth filter with DR= 80 dB and n = 5

Alias
DR signals Ideal shape on obtain f c = 0.16 f s 2 .
Real characteristic If the analog signal is imposed, it must to select
f c f a , and for the signal-to-noise ratio (SNR) which
aStop Transition
Band f is also imposed a SNR = DR , choosing n 4 , it
fs 2 fs can be calculated the minimum sampling frequency
Pass Band Stop Band for ADC:
2 DR [ dB ]
Fig. 4 The main parameters for LP-AAF 20n
f s 2 f a 10 1 , (5)
Choosing a higher rate of sampling it can reduce the
requirements for filter sharpness and therefore the For N bits ADC with Gauss noise, the minimum SNR
number of poles for filter characteristic can be reduce value is given by the next relation:
(and the filter complexity too). This aspect is
RMS ADC signal
illustrated in fig. 5, which shows how the increasing SNR = = 2 N 1 6 , (6)
of the sampling frequency decreases the sloping rate RMSGauss noise
of filter characteristic, while maintaining the analog
corner frequency and the dynamic range requirements. This equation is valid if the Gauss noise is measured
over the entire first Nyquist zone, from DC to f s 2 .
a( f ) a( f ) It can be expressible in decibels by relation:
fa fs fa fa fs fa

SNR = [ 6.02 N + 1.76] dB . (7)
DR <

f f If the bandwidth of analog signal, Ba = f , is less

0.5 f s fs 0 .5 f s fs than f s 2 , than the SNR increase because the
Fig. 5 The increasing sampling frequency reduces the order of LP-AAF quantization noise within the signal bandwidth is
smaller. In these situations the SNR must be calculate
The transfer function for Butterworth filter is given by with the relation:
relation:
G f
Y (s) = n 1
, (2) SNR = 6.02 N + 1.76 + 10 log s dB . (8)
a0 s + a1s + ... + an 1s 2 + an s + 1 2 f
n

where G is the desired gain for filter circuit, usually For ADC with 8, 12, or 16 bits it can obtain the
G = 1 , and the coefficients a0 , a1 , an 1 , an are given in desired SNR at 50 dB, 74 dB, or 98dB, which mean
reference [1], [2]. the desired dynamic range DR at 60 dB, 80 dB, or
The amplitude response in dB is given by relation: 100dB. If Ba is six times less than f s 2 , the
correction term in (8) increase SNR with 7.8 dB.
G In reference [1] are presented some interesting results
a = 20 log H ( j ) = 20 log n
, (3)
2 of LP-AAF simulation with Microchip FilterLab
1+ software, and the possible dynamic range with usually

c filters order is presented in table I.

Table I The dynamic range with various LP-AAF
where the cut off frequency is f c = c 2 f a , and Filter Dynamic range / Maximum attenuation
n is the order of filter, the number of poles for the order amax [dB]
transfer function. n Butterworth Bessel Chebyshev
In these conditions we have a single equation and two 4 80 66 90
unknown variables: n and c . 5 100 79 117
Usually the design process is started with the 6 120 92 142
attenuation a DR , and with n 4 . If the sampling 7 140 104 169
frequency is imposed in the digital system, it must to
calculate the corner frequency of low pass filter from The implementation of analog anti-aliasing filter may
the relation (3) as well: use the active cells with op amp, like successive
Salen-Key low pass filters or Multiple Feedback low
pass filters (MFB cells). These solutions are most

152
popular and offer a good satisfaction for ADCs which Flash (Parallel) ADC;
process signals in the first Nyquist zone. Successive approximation ADC;
Sigma-Delta ADC;
Pipeline ADC;
Bit per stage (Serial) ADC.
Based on our study and applications result a
recommendation for pipeline ADC in various
applications, because the producers offer numerous
variants of these ADCs, and their price isnt so high..

V. CONCLUDING REMARKS

The analog filtering can be a critical part of a data

conversion system, because the anti-aliasing filters
remove the ambiguity in the data conversion process
of ADCs.
A good design for this filtering part is a warranty for
Fig. 6 The Sallen-Key double pole low pass cell the good functionality of digital systems.
There are a lot of dynamic characteristics (key
For the ADCs suitable for digital communications specifications) for all high speed ADCs as well:
applications it must to allow dynamic performances power supply, maximum power dissipation, input
into a higher order Nyquist zones. In these cases it is range, input impedance, signal to noise and distortion
necessary to use ADCs with the dynamic range given ratio (SINAD), effective number of bits (ENOB),
by the stopband attenuation of the BP-AAF, analog bandwidth, which can be specified FPBW (full
recommended for undersampling process. power bandwidth) or SSBW (small signal bandwidth).
In order to allow the placement of the carrier Some of them are implied by filtering features and
frequency in the middle of the NZ order Nyquist zone these characteristics must be analyzed in the context
of undersampling process it must to select the of the desired application with ADCs.
sampling frequency based on the relation:
4 f cr REFERENCES
fs = , NZ = 1, 2,3,... . (9)
2 NZ 1
The NZ order must be an integer, and it is normally [1] B.C. Backer, Anti-Aliasing, Analog Filters for Data Acquisition
chosen as large as possible while is satisfied the Systems, AN699, http://www.microchip.com.
[2] D.E. Johnson, J.L. Hilburn, Rapid Practical Designs of Active Filters,
Nyquist condition f s > 2 f . John Wiley and Sons, New York, 1975.
For example, we consider an analog FM signal with [3] W. Kester, High speed sampling and high speed ADC, EDN,
bandwidth Ba = 8 MHz , centered around carrier USA, 1994.
[4] http://www.maxim-ic.com site of MAXIM Company
frequency. If we assume only M L = M H = 1MHz [5] http://www.st.com site of STMicroelectronics from Europe
(Geneva).
margins for band of interest in the Nyquist zone, then
it must use a minimum sampling frequency
f s = 10 Msps (Mega samples per second).
The desired NZ order can be calculated as the lowest
integer value, which results from the next relation:
4 f cr f s
NZ = . (10)
2 fs
By calculus on obtain NZ = [ 23.5] = 23 , and from
relation (9) result f s = 10, 666 Msps .
For the filter the attenuation in the stop band is
aStop SNR given by relation (8), and the width of
transition bands is smaller than 2 M L or 2 M H .
In reference [2] are recommended some variants for
active and passive circuits which can be used to
realize the band pass filters. Still remain a problem the
availability of high speed op. amp. with the maximum
frequency over 300 MHz.
Now the fabricants of IC offer five types of ADCs
architecture for high resolution (more than 8 bits) and
high speed conversion (more than 100 Ksps):

153
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Software Setting Telephone Links Using ATMEL

Microcontrollers in Time Switching Network PABX
Petru Duma1

Abstract The work briefly presents the resources of a
low capacity telephone exchange with temporal switching,
using PCM encoding for the conversation signal, then the
stages for performing local, outgoing and incoming
telephone links. Based on these stages, macro-states graphs
were designed to model the processes that perform
telephone links and then used to write the command
software using the assembly language for ATMEL family
microcontrollers.
Keywords: macro-states graph, local telephone link,
outgoing telephone link, incoming telephone link,
command software.
I. INTRODUCTION
The process of establishing a telephone connection
through a low capacity electronic telephone exchange
(PABX) is a complexe sequential process. A graph of
states must be used in order to describe it. The
transitions between states are induced by certain events,
which are a generalisation of the input signal, meaning
that they can be input signals and/or the result of system
internal processing.
It is practically proven that the graph describing a
software real-time telephone connection, using periodic
interrupt operation, includes very many states.
Moreover, using this description, complex problems
emerge while implementing the services.
Analysing the steps of performing local, outgoing and
incoming telephone links, it results that certain steps are
repeated several times, having different initial or final
data.
Based on these facts, there were defined macro-states,
which perform the steps of establishing a telephone link,
such as: receiving the pulse selection information,
determining the type of the call, checking the selection
information, establishing (interrupting) the connection,
transmitting (interrupting) a tonality, a vocal message
etc.
The macro-states are designed so that they can be called
from any program sequence and they return to any
program sequence also.
This last facility allows a very simple succession of
macro-states and a simple development of telephone
services.
II. THE RESSOURCES OF THE LOW CAPACITY
ELECTRONIC TELEPHONE EXCHANGE WITH
TEMPORAL SWITCHING
The general structure of a low capacity electronic
telephone exchange is presented in figure 1, where the
notes have the following meanings:
I - interface to the external medium;
CIA - individual subscriber circuits;
CJ - junction circuits;
RC - temporal connection network;
MT - tones and calling signal machine;
C - microcontroller;
UC - command unit;
PC - personal computer.
Fig. 1.
The external medium includes the subscribers' lines
(LAb) through which the subscribers' telephone sets are
connected to the exchange, and the junction lines (LJ)
through which the exchanges themselves are connected.
This interface performs the following functions:
monitoring the lines, receiving and transmitting the
selection information, supply, adapting, galvanic
separation, protection, testing and processing the
conversation signal which consists of sense separation,
filtering, sampling and holding, analogic-digital and
digital-analogic conversion. These last functions are
implemented with TCM29C13N integrated PCM
codecs.
The interface to the external medium consists of the

154
individual subscriber circuits and the junction circuits.
CIA performs the interfacing of the subscribers lines to
the exchange and the following functions, from the point
of view of the command software:
- detects the apparition of a call from a subscriber when
closing the DC loop;
- receives the pulse selection information by converting
the line pulses in TTL-leveled voltage pulses;
- transmits the calling signal to the subscribers.
CJ performs the interfacing of the urban telephone
exchange lines to the low capacity exchange and the
following functions, from the point of view of the
command software:
- detects the apparition of a call from a distant exchange;
- receives the DTMF selection information from a
subscriber connected to a distant urban exchange for
automatic processing the incoming calls;
- transmits the calling signal to the distant exchange;
- transmits the pulse and DTMF selection information to
a distant urban exchange for automatic processing the
outgoing calls.
For the reception and the transmission of the tone
selection information, specialized integrated circuits
(M8880) are used as they are performing these functions
with a minimal need for supplementary hardware.
The temporal connection network has no internal
blocking and performs the switch of any channel from
any input PCM line in any channel in any output PCM
line.
The network is implemented using digital integrated
temporal switch MT8980 which features 8 PCM input
and 8 PCM output lines, each consisting of 32 channels,
so that the switch has a capacity of 256 x 256 channels.
The tones machine sends towards the subscribers all the
signals required for the exchange's operation (dialing
tone, busy, reverse call, unexisting number, warning,
false call etc.), through the exchange's connection
network, as well as the calling signal. MT can also send
to the subscribers various information vocal signals.
The digital switch which implements connection
network includes, from the point of view of the
command software, a control register, a data memory
(256x8 bits) and a command memory (256x11 bits)
which includes a low section (256x8 bits) and a high
section (256x3 bits). By programming the control
register, the data and command memory can be written
and read and, therefore, any link between any two
subscribers can be performed, as well as any tone or
vocal message can be transmitted to any subscriber.
The distributed command unit of the low capacity
electronic telephone exchange is implemented using
several ATMEL family microcontrollers hierarchized on
two levels of priority.
On the lower level are the peripheral microcontrollers
that control the individual subscriber circuits, the
junction circuits, the temporal connection network, the
tone machine and perform a large amount of simple but
critical in time command and control operations. On the
upper level is placed a main processor that performs the
connection settings and telephone services. The user has
access to the command unit through a personal computer
155
that communicates serially with the main processor.
III. STEPS OF PERFORMING A
LOCAL TELEPHONE LINK
Performing a local telephone link between subscribers A
and B (assuming that A is the calling subscriber while B
is the called subscriber) includes the following steps:
- in the first moment, subscriber A is standing by;
- subscriber A picks up the receiver in order to initiate a
telephone conversation; the DC loop is closed, which
performs the subscriber's A call to the exchange;
- after this call, the exchange connects the calling
subscriber to the command unit, so that the selection
information can be received;
- subscriber A receives the dial tone which constitues an
invitation to the transmission of the pulse or tone
selection information;
- the selection information transmission is initiated;
- the dial tone transmission is interrupted;
- the first digit of the selection information is received
and processed;
- after receiving the first digit, the type of telephone link
is determined (local, outgoing or service);
- if the first digit received is different from 9 and 0, it is
a local telephone link and the reception of the other
digits in the selection information is continued;
- after receivig the selection information, the number is
verified, which means to determinate whether the called
subscriber exists, if the connection between the two
subscribers can be performed and if the called
subscriber is free;
- if the called subscriber's number does not exist, then an
unexisting number tone is transmitted towards the
calling subscriber;
- if the telephone connection cannot be performed, then
a busy exchange tone is transmitted towards the calling
subscriber;
- if the called subscriber is busy, then the busy tone is
transmitted towards the calling subscriber;
- if the number transmitted exists, the connection can be
performed and the called subscriber is free then the
reverse call tone is transmitted towards the calling
subscriber and the calling signal is transmitted towards
the called subscriber;
- if, after a certain time interval, the called subscriber
answers, then the transmission of the reverse call tone
and the calling signal is interrupted;
- afterwards, the connection between the two subscribers
is commanded;
- after performig the connection, the subscribers start
their conversation, and the system awaits for one of the
subscribers to hang up;
- after the conversation is finished (one of the
subscribers hangs up) the connection between the
subscribers is interrupted;
- the system returns to the initial status.
From each step of this process, the selection information
for a certain telephone service can be received and also
each subscriber can hang up, and the process returns to
the initial status.

IV. THE MACRO-STATES GRAPH FOR THE
PROCESS OF ESTABLISHING A LOCAL
TELEPHONE CONNECTION
In fig.2 is shown the macro-states graph for the process
of establishing a local telephone connection through a
low capacity electronic telephone exchange, using the
steps described above.
Fig. 2.
The macro-states were defined as follows:
M0 - macro-state for receiving the pulse and tone
selection information;
M1 - macro-state for determining the type of call (local,
156
outgoing, service);
M2 - macro-state for processing the selection
information;
M3 - macro-state for establishing the connection
between the two subscribers;
M4 - macro-state for interrupting the connection
between the two subcribers which were in conversation;
M5 - macro-state for transmitting to a subscriber a tone
signal or the calling signal;
M6 - macro-state for interrupting the transmission of a
tone signal or of the callig signal;
M7 - macro-state for waiting the standing-by calling
subscriber;
M8 - macro-state for waiting the calling subscriber
which picked up the receiver has to initiate an action;
M9 - macro-state for waiting the calling subscriber
which has picked up the receiver has to initiate an action
and the standing-by called subscriber to answer the call;
MA - macro-state for conversation; it is a waiting macro-
state while the two suscribers have the receivers picked
up;
MB - macro-state for waiting the interruption of the
telephone connection.
The events are represented on the graph's arches and
have the following meanings:
a - subscriber A has picked up the receiver;
p - specific processings for the current macro-state;
i - the reception of the selection information;
t - time counting;
c1 - first digit received;
ni - unexisting number;
l - the connection can be performed;
b - subscriber B has picked up the receiver;
AE - outgoing call;
S - telephone service;
AL - local call.
Using the macro-states and the events defined above, the
process of performing a telephone connection through a
low capacity electronic telephone exchange can be
followed on the graph. Every macro-state includes
severals states and the evolution between these is
depicted also by a graph of stages.
V. STEPS OF PERFORMING AN OUTGOING
TELEPHONE LINK
Performing an outgoing telephone link between
subscribers A and B (assuming that A is the calling
subscriber connected to the low capacity exchange while
B is the called subscriber of the distant urban high
capacity exchange) includes the following steps:
- the first eight steps are the same as in the situation
when a local link is performed;
- if the first digit received is 0, it is an outgoing
telephone link;
- the first free junction line is located and engaged;
- if no junction line is free, the calling subscriber is sent
the busy tone;
- if there is a free junction line, the conversation link is
established between the calling subscriber and the
junction line;
- the calling subscriber receives a dialling tone from the
distant urban exchange;
- the pulse selection information is received from the
calling subscriber and it is DTMF transmitted to the
exchange;
- the state of the calling subsriber is monitored;
- if the calling subscriber hangs up, the conversation link
is interrupted and the junction line is released.
From each step of this process, the selection information
for a certain telephone service can be received and the
calling subscriber can hang up, and the process returns
to the initial status.

VI. THE MACRO-STATES GRAPH FOR THE

PROCESS OF ESTABLISHING A OUTGOING
TELEPHONE CONNECTION

In fig.3 is shown the macro-states graph for the process

of establishing an outgoing telephone connection.
Besides the macro-states M0 - MB defined in previous
paragraph, a few more macro-states were defined as
follows:
MC - macro-state for locating a free junction line;
MD - macro-state for occupying the junction line;
ME - macro-state for transmitting a digit to the distant
exchange.
The events are represented on the graph's arches and
have the meanings mentioned in paragraph 4 with
additional two:
J - status of the junction line;
c - digit of the DTMF selection information.
Using the macro-states and the events defined above, the
process of performing an outgoing telephone connection
can be followed on the graph.

VII. STEPS OF PERFORMING AN INCOMING

TELEPHONE LINK

Performing an incoming telephone link between

subscribers A and B (assuming that A is the calling
subscriber connected to the distant urban high capacity
exchange, while B is the called subscriber of the low
capacity exchange) includes the following steps:
- the calling subscriber A picks up the receiver, receives
the dialling tone and transmits the telephone number for
calling the low capacity exchange on the junction line;
- if the called junction line is free, the urban exchange
sends the reverse call tone to subscriber A and calling
signal on the junction line. Fig. 3.
These steps are performed by the high capacity urban
exchange. - the beginning of the DTMF selection information
- the junction line is free and awaits an incoming call; transmission is awaited;
- when the call is received, the junction is engaged and - if during a pre-set time interval (e.g. 20 seconds) the
from the urban exchange, the transmission of the reverse transmission of the tone selection information does not
tone and of the calling signal is interrupted; start, then the dialling tone transmission is canceled and
the subscriber A is connected to a duty subscriber or to
- the low capacity exchange transmmits from the the exchange's operator, which performs the link
tone machine to subscriber A a vocal required;
information message (e.g. "You called the - if the DTMF selection information reception started,
Technical University. Please dial the interior the dialling tone transmission is cancelled and the
number desired."), then the dialling tone; selection information is fully received and then

157
processed;
- if the called subscriber's number does not exist, then an
unexisting number tone is transmitted towards the
calling subscriber;
- if the telephone connection cannot be performed, then
an busy exchange tone is transmitted towards the calling
subscriber;
- if the called subscriber is busy, then the busy tone is
transmitted towards the calling subscriber;
- if the number transmitted exists, the connection can be
performed and the called subscriber is free then the
reverse call tone is transmitted towards the calling
subscriber and the calling signal is transmitted towards
the called subscriber;
- if the subscriber B does not answer in a specified time
interval (e.g. 40 seconds) the transmission of the reverse
call tone and the calling signal is interrupted and, after a
vocal warning message, the junction is released;
- if, after a certain time interval, the called subscriber
answers, then the transmission of the reverse call tone
and the calling signal is interrupted;
- afterwards, the connection between the two subscribers
is initiated;
- after performig the connection, the subscribers start
their conversation, and the system awaits for one of the
subscriber B to hang up;
- after the conversation is finished (subscriber B hangs
up) the connection between the subscribers is
interrupted;
- the system returns to the initial status.
During this process, the calling subscriber cannot access
any service of the low capacity exchange, while the
called subscriber can only access the telephone services
after the conversation starts.

VIII. THE MACRO-STATES GRAPH FOR THE

PROCESS OF ESTABLISHING AN INCOMING
TELEPHONE CONNECTION

In fig.4 is shown the macro-states graph for the process

of establishing an incoming telephone connection in low
capacity electronic telephone exchange.
Besides the macro-states M0 - MB defined in previous
paragraph, a few more macro-states were defined as
follows:
MF - macro-state for monitoring the status of the
junction line and for engaging it;
MG - macro-state for transmitting an informational vocal
message;
MH - macro-state for awaiting the calling subscriber to
transmit the DTMF selection information;
MI - macro-state for receiving the DTMF selection
information through the junction;
MJ - macro-state for awaiting the called subscriber to
answer;
MK - macro-state for conversation between the two
subscribers;
ML - macro-state for conversation between the duty
subscriber or the operator and the calling subscriber;
MM - macro-state for releasing the junction line.
Fig. 4.
The events are represented on the graph's arches and
have the meanings mentioned in paragraph 4 with an
additional one:
Op - the duty subscriber or the operator has picked up
the phone.
Using the macro-states and the events defined above, the
process of performing an incoming telephone
connection can be followed on the graph.
IX. CONCLUSIONS
Based on the macro-state graphs that model the
processes for setting telephone links (local, outgoing
and incoming) and on the method of software
implementation of multiprocess systems, the author

158
designed the command program of the main processor [2]. GRINSEC: La commutation electronique, Ed. Eyrolles, Paris,
1988.
that controls a low capacity telephone exchange with
[3]. Hintz J.K., Tabak D., Microcontrollers. Arhitecture,
temporal switching, using PCM encoding for the Implementation and Programming, McGaw Hill, 1993.
conversation signal. [4]. Radulescu T.: Telecomunicatii, Media Publishing, Bucureti,
There were also written the command programs of the 1994.
peripheral processes for the individual subscriber [5]. XXX: ATMEL Microcontroller Family, Data Book, 1990.
[6]. XXX: MITEL, Data Book, 1992.
circuits, the junction circuits, the temporal switching [7]. XXX: Telecom Design Solutions. Teltone Component Data Book,
network and the tone machine. 1990.
The command programs, written in the assembly [8]. XXX: TEXAS INSTRUMENTS, Data Book, 1992.
language of ATMEL family microcontrollers, require an
amount of around 1-2 Kb of program memory for each
peripheral process and of 8 Kb for the main process of
setting telephone links.
The use of macro-states in the manner described above
allows easily re-sequencing them for implementing also
the telephone services.
REFERENCES

[1]. Borcoci E.: Sisteme de comutaie digitale. Ed. Vega, Bucureti,

1994.

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TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Variable Step Size Affine Projection Adaptive Algorithm

Implementation
Sorin Zoican .1

Abstract - The paper presents a new variable step-
size adaptive algorithm for affine projection (VSS-
AP) with a faster convergence and lower
misadjustment. The VSS-AP algorithm is compared
with the LMS algorithm. The possibility of real time
implementation of this algorithm is investigated,
using a DSP microcomputer (ADSP 21161 Analog
Devices).
plant filter of length M, ui - the (1 x M) input
vector, Ui = [ ui ui-1 ui-K+1]T the (K x M)
input matrix ( T denote transpose), di = [ d(i) d(i-1)
d(i-K+1)]T a K x 1 vector , ei the residual
error (K x 1) vector.
The VSS-AP algorithm is the following:

Keywords: variable step-size adaptive algorithm, d(i) = ui w0 (1)

digital signal processor (DSP)
I. INTRODUCTION
The performance of LMS or NLMS algorithms
are drastically deteriorate if the input signal is
colored input, not a white noise one. Many
techniques were proposed in the literature in order
to overcome this. The main idea in all of these
techniques is to control the adaptation parameter
(the step size) but the performance (in terms of
convergence speed and final residual error)
depends of the estimation of how far the filter is
from optimal performance.
If this estimation take into consideration only the
input vector data (that is the current sample and
some delayed input sample) the performance will
be not very good, of course depends of the input
signal characteristics. On the other hand if the
estimation of the filter performance uses an input
matrix the performance will be better even the
input signal is colored.
In the second case the computational effort
will be significantly increased and the round off
errors can compromise the algorithm convergence.
II. THE VARIABLE STEP-SIZE
ALGORITHM
wi = wi-1 + (i) U*i (U*iUi)-1 ei (2)
ei = di Ui wi-1 (3)
The criterion used to control the algorithm
performance is the closing to the optimal
performance in terms of minimum mean square
deviation (MED) given by E {||w0 wi ||2}, where
E{.} represents the expectation operator.
Let pi = U*i (U*iUi)-1 Ui wi-1 the affine
projection of wi-1 onto the range space of matrix U*i
.
The minimum mean square deviation ca
be written using the expression (2) and by
minimizing the MED with respect of step size
parameter, , the following formulae will be
obtained for the step size parameter [1]:
(i) = max .[ || pi || 2 / || pi || 2 + C] (4)
where C is a small positive constant, is positive
constant close to 1 and || . || denote the norm
operator.
The parameter pi can be estimated by time
averaging as in equation (5) [1]:
pi = pi + (1-) U*i (U*iUi)-1 ei (5)

The VSS-AP algorithm take into

consideration K input vectors.
We note: d(i) desired signal, w0 the
coefficients of the finite impulse response (FIR)

1
POLITEHNICA University of Bucharest, Electronic and Telecommunications Faculty
Telecommunication Department, sorin@elcom.pub.ro

160
 III. THE IMPLEMENTATION AND Interval timer
MAIN RESULTS On-Chip SRAM (1 Mbit)
SDRAM Controller for glueless interface to
The new algorithm was implemented on an SDRAMs
evaluation kit with DSP processor (ADSP21161 External port that supports:
EZ-KIT). Interfacing to off-chip memory peripherals
The ADSP-21161 SHARC DSP is the first Glueless multiprocessing support for six ADSP-
low-cost derivative of the ADSP-21160 featuring 21161N SHARCs
Analog Devices Super Harvard Architecture. Like Host port read/write of IOP registers
other SHARCs, the ADSP-21161 is a 32-bit DMA controller
processor that is optimized for high performance Four serial ports
DSP applications. The ADSP-21161N includes a Two link ports
100 MHz core, a dual-ported on-chip SRAM, an SPI-compatible interface
integrated I/O processor with multiprocessing JTAG test access port
support, and multiple internal busses to eliminate 12 General Purpose I/O Pins
I/O bottlenecks.
The ADSP-21161 offers a Single-Instruction- The main characteristics of the ADSP21161
Multiple-Data (SIMD) architecture. Using two are presented below:
computational units the ADSP-21161 can double High performance 32-bit DSP
cycle performance on a range of DSP algorithms. applications in audio, medical, military,
Figure 1 shows a block diagram of the ADSP- wireless communications, graphics,
21161, illustrating the following architectural imaging, motor-control, and telephony
features: Super Harvard Architecture-four
Two processing elements, each made up of an independent buses for dual data fetch,
ALU, Multiplier, Shifter and Data Register File instruction fetch, and nonintrusive, zero-
Data Address Generators (DAG1, DAG2) overhead I/O
Program sequencer with instruction cache
PM and DM buses capable of supporting four 32-
bit data transfers between memory and the core
every core processor cycle

Figure 1. The block diagram of the ADSP-21161

Single-Instruction-Multiple-Data (SIMD) each with a multiplier, ALU, shifter, and

computational architecture- two 32-bit register file
IEEE floating-point computation units, Serial ports offer I 2 S support via 8
programmable and simultaneous receive

161
 and transmit pins, which supports up to
16 transmit or 16 receive channels of
audio
Integrated peripheralsintegrated I/O
processor, 1 Mbit on-chip dual-ported
SRAM,SDRAM controller, glueless
multiprocessing features, and I/O ports
(serial, link, external bus, SPI, & JTAG)
ADSP-21161N supports 32-bit fixed, 32-
bit float, and 40-bit floating point
formats.
100 MHz (10 ns) core instruction rate
Single-cycle instruction execution,
including SIMD operations in both
computational units
600 MFLOPS peak and 400 MFLOPs
sustained performance
1 Mbit on-chip dual-ported SRAM (0.5
Mbit block 0, 0.5 Mbit block 1) for
independent access by core processor and
DMA
400 million fixed-point multiply and
accumulation operations (MACs)
sustained performance
Dual Data Address Generators (DAGs)
with modulo and bit-reverse addressing
Zero-overhead looping with single-cycle
loop setup, providing efficient program
sequencing
IEEE 1149.1 JTAG standard test access
port and on-chip emulation
Single Instruction Multiple Data (SIMD)
architecture provides:
1. Two computational processing
elements
2. Concurrent execution--Each
processing element executes the
same instruction, but operates on
different data
Parallelism in busses and computational
units allows:
1. Single-cycle execution (with or
without SIMD) of: a multiply
operation, an ALU operation, a
dual memory read or write, and
an instruction fetch
2. Transfers between memory and
core at up to four 32-bit floating-
or fixed-point words per cycle,
sustained 1.6 Gigabytes/second
bandwidth
3. Accelerated FFT butterfly
computation through a multiply
with add and subtract
DMA Controller supports:
1. 14 zero-overhead DMA
channels for transfers between
ADSP-21161N internal memory
and external memory, external
peripherals, host processor,
serial ports, link ports or Serial
162
Peripheral Interface (SPI)
interface
2. 64-bit background DMA
transfers at core clock speed, in
parallel with full-speed
processor execution
3. 800 Mbytes/s transfer rate over
IOP bus
4. Host processor interface to 8-,
16- and 32-bit microprocessors,
the host can directly read/write
ADSP-21161 IOP registers.
32-bit (or up to 48-bit) wide synchronous
External Port provides glueless connection
to asynchronous, SBSRAM and SDRAM
external memories
Memory interface supports programmable
wait state generation and wait mode for
off-chip memory
24-bit address, 32-bit data bus. 16
additional data lines via multiplexed link
port data pins allow complete 48-bit wide
data bus for single-cycle external
instruction execution
32-48, 16-48, 8-48 execution packing for
executing instruction directly from 32-bit,
16-bit, or 8-bit wide external memories
32-48, 16-48, 8-48, 32-32/64, 16-32/64,
8-32/64, data packing for DMA transfers
directly from 32-bit, 16-bit, or 8-bit wide
external memories to and from internal
32-, 48-,or 64-bit internal memory
SDRAM Controller for glueless interface
to low cost external memory
Extended external memory banks (64 M-
words) for SDRAM accesses
Multiprocessing support provides:
glueless connection for scalable DSP
multiprocessing architecture and
distributed on-chip bus arbitration for
parallel bus connect of up to six ADSP-
21161s, global memory and a host
Two 8-bit wide link ports for point-to-
point connectivity and array
multiprocessing be-tween ADSP-21161
Serial Ports provide:four 50 Mbit/s
synchronous serial ports with
companding hardware, 8 bi-directional
serial data pins, configurable as either a
transmitter or receiver, TDM support for
T1 and E1 interfaces, and 128 TDM
channel support
The VSS-AP algorithm was implemented
and compared with the LMS algorithm under the
following circumstances: the input signal was
chosen as a colored signal (that is, filtering a white,
zero-mean, Gaussian random sequence through a
second order IIR filter), K=4, M=17, = 0,995, C
= 10-4, max = 1.0 .
 The main results, illustrated in figure 2, reasonable. Using an adequate technique, such as
show a significantly performance improved for the switching buffers, a real time implementation is
VSS-AP ( both for convergence speed and realizable.
misadjustment). The execution time is quite

residual error - LMS residual error - VSS

0.6 0.5

0.4
0.4
0.3

0.2 0.2

0.1
0
0

-0.2 -0.1

-0.2
-0.4
-0.3
-0.6
-0.4

-0.8 -0.5
0 100 200 300 400 500 600 700 800 900 1000 0 100 200 300 400 500 600 700 800 900 1000

Figure 4. The main results (K=4)

residual error - VSS (K =4) residual error - VSS (K =7)

0.8 0.5

0.6 0.4

0.3
0.4

0.2
0.2
0.1
0
0

-0.2
-0.1

-0.4 -0.2

-0.6 -0.3
0 100 200 300 400 500 600 700 800 900 1000 0 100 200 300 400 500 600 700 800 900 1000

163
 residual error - VSS (K =8) residual error - VSS (K =10)
0.5 0.6

0.4 0.5

0.4
0.3

0.3
0.2
0.2
0.1
0.1
0
0

-0.1
-0.1

-0.2 -0.2

-0.3 -0.3
0 100 200 300 400 500 600 700 800 900 1000 0 100 200 300 400 500 600 700 800 900 1000

Figure 3. The VSS adaptive filter performance for various values of parameter K

If the dimension of the input matrix is we use a Gauss-Jordan inversion algorithm

increased then the performance of the adaptive witch was implemented in C language.
filter is better. For K large enough the The execution time (in processor cycles) is
performance is quite similar to RLS algorithm. illustrated for several values of the parameter
In order to evaluate the inverse of the matrix U K in figure 4.

VSS-AP adaptive filter computational effort

180000
Number of processor cycles

152393
160000
140000
120000
100000 87419
80000
60000
40000 17447
20000 1382 4289
0
1 2 4 8 10
K

Figure 4. The computational effort for VSS-AP implementation

The VSS-AP implementation takes an This algorithm woks very well for ill-
execution time between and 0.2 ms and 16 ms conditioned input signal (e.g. speech signal).
witch is a quite reasonable execution time if
the switching buffer technique is involved. V. REFERENCES
This techniques requires as a timing condition
that the acquisition time for the whole signal
window (about 20 ms) must be greater than [1] Shin, H.C, et. al, Variable Step Size NLMS
total execution time for signal processing in and Affine Projection Algorithms , IEEE Signal
the current window. Processing Letters, vol.11, no. 2, february 2004,
pp. 132-135
[2] S. Haykin, Adaptive Filter Theory, 3rd ed.,
IV. CONCLUSIONS Prentice Hall, 1996
[3] ADSP2116x Hardware Reference Manual,
A new adaptive algorithm was presented. Analog Devices 2000
The performance of the new algorithm is better [3] ADSP2116x Software Manual, Analog Devices
than classical algorithm (similar with RLS 2000
algorithm) but the computational effort is
smaller than RLS.

164
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TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

QRD-LSL Algorithm Suitable for Implementation on

D.S.P.
Andrei Alexandru Enescu, Constantin Paleologu, Silviu Ciochin1
Abstract This paper deals with modifications brought
to the QRD-LSL algorithm presented in [5], in order to
implement it on Digital Signal Processor (DSP). It is
necessary to choose a powerful processor, with a parallel
architecture that allows several instructions to be
executed simultaneously. This is the main reason for
choosing Motorola SC140 processor. In addition, we
have structured the algorithm in a way to allow a high
complexity algorithm to be run in real-time in a specific
application. The paper presents the main features of this
DSP and then makes comparisons between the original
algorithm, described in [1] and the improved version
with low complexity.
Keywords: adaptive lattice algorithm, least squares,
digital signal processor
Another issue is the fixed-point representation.
Since fractional representation is used (i.e. for a
given number of bits, B, the range for any
fractional variable is [-1;1-2-B]), care must be
taken when arithmetic operations are made, in
order to avoid overflow. An overview on the
dynamics of the algorithm variables will prove
that scaling is needed in the implementation
process, because some variables are greater than
unit value.
Moreover, this paper presents the
implementation tricks used to simplify the
algorithm in order to implement it on a digital
signal processor.

I. INTRODUCTION
II. AN OVERVIEW OF SC140 ARCHITECTURE
It has been shown in [2], [3] that QRD-LSL algorithm
has great performances when used in echo canceller As we have chosen to implement the algorithm on
configuration. However, in the original version, the SC140, it is necessary to describe the features of this
complexity is quite large, making the algorithm processor first. The specific features of this
impossible to be implemented in real-time. In [1], a architecture, described in [6] are the following:
modified version of the algorithm is presented. A
comparison between the two algorithms is presented
High level abstraction of the Application
Software
in Table 1:
Applications development in C language
Hardware supported integer and
Table 1.
fractional data
Adaptive
algorithm QRD-LSL MQRD-LSL Scalable performance
multiplications 25M+11 22M+10 4 ALUs (Arithmetic logic Units) and 2
AAUs (Address Arithmetic Units)
divisions 4M+2 4M+2 4 MMACS (million multiply and
additions/ subtractions 8M+3 8M+3 accumulate operations per second) for each
square-root operations 4M+2 0 megahertz of clock frequency
High Code Density for Minimized Cost
It is to be noticed that the square root operations 16-bit wide instruction encoding
require a large computing time, as they must be
approximated by another technique, e. g. Taylor The core important features are:
series. Either way, the computing time increases
significantly and the application area becomes
Up to 10 RISC MIPS for each megahertz
of clock frequency
restricted due to a reduced sampling frequency. Thus,
the main goal is to minimize the number of A true (16*16) + 4040-bit MAC unit
instructions within the implemented algorithm. in each ALU

1
Facultatea de Electronic i Telecomunicaii, Catedra de
Telecomunicaii Bd. I. Maniu 1-3, Bucureti, e-mail: {aenescu, pale, silviu}@comm.pub.ro

165
 A true 40-bit parallel barrel shifter in
each ALU
16 x 40-bit data registers for fractional
and integer data operand storage
16 x 32-bit address registers (8 can be
used as 32-bit base address registers)
4 address offset registers and 4 modulo
address registers
Unified data and program memory space
(Harvard architecture)
Byte addressable data memory
Fig 1. Block Diagram of SC140 Core
In Fig. 1, we present the SC140 core, including:
- DALU (Data Arithmetic Logic unit)
- A register file of 16 x 40-bit
registers
- 4 parallel ALUs (each one
containing a MAC unit and a
BFU- bit-field unit)
- 8 data bus shifter/limiters
- AGU (Address Generation Unit)
- 2 AAUs, accessing 16 address
registers, 4 offset registers, 4
modulo registers
III. AN ANALYZE OF QRD-LSL IN DSP
IMPLEMENTATION CONTEXT
In [1], a version of QRD-LSL algorithm is
presented, after eliminating complex operations,
such as square roots. The algorithm is presented in
Table 2:
Table 2.
1. Initialization
For order m = 1, 2,K , M
166
However, the main feature that we have already
mentioned is the C compiler and the ability to convert
C source code into assembly code. The complexity of
QRD-LSL algorithm is quite large and therefore the
need for flexibility is important, since programming in
C code is much easier than implementing the
algorithm direct in assembly code. The C compiler
supports ANSI C standard and also intrinsic functions
for ITU/ETSI primitives. Assembly code integration is
also possible, which optimizes supplementary the
code.
The block diagram of SC140 core, as presented in [2],
is described in Fig. 1.
f , m 1 (0) = b, m 1 (0) = 0 , pm (0) = 0
Bm 1 (0) = , Fm 1 (0) =
// is a small positive constant
end
2. Computations
For time n compute
f ,0 (n) = b ,0 (n) = x(n)
// x(n) is the input at time n
0 ( n) = d ( n)
// d (n) is the desired response at time n
0 ( n) = 1
b ,0 (n 1) = 1 , f ,0 (n 1) = 1
For order m = 1, 2,K , M
Bm 1 ( n 1) = Bm 1 (n 2) +
2
+ b , m 1 (n 1) b , m 1 (n 1)
Bm 1 (n 2)
cb , m 1 (n 1) =
Bm 1 (n 1)
b*, m 1 (n 1)
sb , m 1 (n 1) = b , m 1 (n 1)
Bm 1 (n 1)
 f , m (n) = f , m 1 (n) b , m 1 (n 1) *f , m 1 (n 1) At the price of losing from finite precision, some of
the bits used in quantization can be used for
scaling, regarded in binary arithmetic as a simply
f ,m1 (n) = cb,m1 (n 1) f , m1 (n 1) + sb,m1 (n 1) f , m1 (nright-shifting
* *
) by the same number of bits.
m (n 1) = cb, m 1 (n 1) m 1 (n 1) 450
2
Fm 1 (n) = Fm 1 (n 1) + f , m 1 ( n 1) f , m 1 ( n) 400

350

Fm 1 (n 1) 300
c f , m 1 (n) =
Fm 1 ( n) 250

f*, m 1 (n) 200

s f , m 1 (n) = f , m 1 (n 1) 150
Fm 1 (n)
100
b , m (n) = b , m 1 (n 1) f , m 1 (n) b*, m 1 (n 1)
50

b*, m 1 (n) = c f , m 1 (n) b*, m 1 (n 1) + 0

0 1000 2000 3000 4000 5000 6000 7000 8000 9000
+ s f , m 1 (n) b , m 1 (n 1)
Fig. 2. Evolution of backward cost functions for the
m +1 (n) = m (n) b , m (n) pm* (n 1) 64 cells of the structure
pm* ( n) = cb , m 1 (n 1) pm* (n 1) + sb , m 1 ( n 1) m (n)
Then, the formula used for actualization of this cost
function becomes:
b, m (n 1) = cb, m 1 (n 1) b, m 1 (n 1)
f , m (n) = c f , m 1 (n) f , m 1 (n 1) B m 1 (n 1) = B m 1 (n 2) +
end 2 (1)
2 + b, m 1 (n 1) b, m 1 (n 1) 2 b
BM (n) = BM ( n 1) + b , M (n) b , M ( n)
BM (n 1) In (1):
cb , M (n) =
BM ( n)
b*, M (n) B( n) = B(n)2 b (2)
sb , M (n) = b , M (n)
BM ( n)
Then, the variables that also depend on B(n) are
M +1 (n) = M (n) b , M (n) pM* (n 1) actualized as follows:
pM* (n) = cb , M ( n) pM* ( n 1) + sb , M (n) M ( n)
M +1 (n) = cb, M (n) M (n) B m 1 (n 2)
c b , m 1 (n 1) = (3)
eM +1 (n) = M +1 (n) M +1 (n) B m 1 (n 1)
end
*
b, m 1 (n 1)
s b , m 1 (n 1) = b , m 1 (n 1) 2 b (4)
B m 1 (n 1)
We focus on the echo cancelling configuration,
since it is demonstrated in [5] that this algorithm
proves good performances in double-talk The same modifications stand also for the forward
configuration. prediction part with the proper index replacements.
Because the standard input signals, the learning If the number of bits used in scaling is properly
curve and other technical details can be found in chosen, then there is no overflow.
[5], we only show the dynamic range of some Even though an asymptotic limit for the cost
variables during convergence process. Assuming a functions has not yet been found, an upper bound
css_st standard signal at the far end and a sinusoid for them can be deduced. Let us denote that:
of normalized frequency 0.1 at the near end and
considering an echo path with a length of the lm = lim Bm (n) (5)
n
impulse response of 64, we present the evolution of
the cost functions on both forward and backward From (1), assuming that the last term is smaller
prediction branches. than 1 in a correct fractional representation, we get:
It can be easily seen that, during convergence, the lm l m + 1
samples increase to a very large value, much
greater than 1. This observation is valid, especially
and therefore:
for the first cell, as we can see from Fig. 2. Also it
is to be noticed that a similar evolution have the
samples for the forward prediction cost function.

167
 1 [ J m 1 , m , m , em , m 1 , pm 1 ] =
lm (6)
1 = prediction( J m 1 , m 1 , m 1 , m 1 ,
em 1 , m 1 , pm 1 )
As shown in [3], a small residual error in an echo
2
cancelling configuration is achieved with a RLS aux = J m 1 + m 1 m 1 2 b
adaptive algorithm by setting the forgetting factor
as closer to 1 as possible. All the same, if we set J m 1
c m 1 =
to the maximum possible represented number on aux
short format of 16 bits, i.e. 1-2-15, then lm is limited J m 1 = aux
by 215. The number of bits used for scaling should m = c m 1 m 1
be log2 (lm)=15, which is unacceptable, since it is
exactly the precision used for a short format m 1
s m 1 = 2 b
variable. Thus, a trade-off is required between echo J m 1
cancellers theoretical performances and the
precision used for cost functions. m = m 1 m 1 m 1
In order to test echo cancellers performances, the m = cm 1 m 1 + s m 1 m 1
algorithm has been implemented using Code if flag
Warror C Compiler for SC140. A simulation on the
evaluation board was run, with a sinusoid of a em = em 1 m 1 p m 1
normalized frequency 0.05 as near-end signal and a p m 1 = c m 1 p m 1 + s m 1 em 1
scaling of 10 bits. The signals are described in Fig. end
3:

Fig. 3. Far-end signal. Near-end signal. Output
signal. Residual error.
IV. OPTIMIZING TECHNIQUES USED FOR
QRD-LSL
In this paragraph, we evaluate the number of cycles
needed by the algorithm per iteration. The goal is
to minimize this number, in order to lower the
computational time per iteration under the
sampling time of the CODEC.
If we take advantage of the fact that the structure is
symmetrical, because of the similarities between
the forward prediction structure and the backward
prediction structure, then we can use two identical
blocks for each lattice cell, thus we can call twice a
function in C language during one iteration.
A behavioral description of the block is given in
Table 3.
Table 3
Fig. 5. Block diagram for one cell prediction
The filtering part is included in backward
prediction part and is performed if a flag is set.
This flag is set before the backward prediction and
reset before the forward prediction. Then, iteration
is described in Table 4.
Table 4.
= 1
for m = 0 , M
flag = 1; // backward prediction
[ Bm 1 , b , m , f , m , em , f , m 1 , pm 1 ] =
= prediction( Bm 1 , b , m 1 , m 1 , f , m 1 ,
em 1 , f , m 1 , pm 1 )
flag = 0 ; // forward prediction
[ Fm 1 , f , m , b , m , em , b, m 1 , pm 1 ] =
= prediction( Fm 1 , f , m 1 , f , m 1 , m 1 ,
em 1 , b , m 1 , pm 1 )
= b

168
end
Another optimization technique, accomplished
using this procedure is that all the transformations
are made in-place, regardless of the iteration (i.e.
moment of time), saving a large amount of
memory. Choosing an appropriate level of
optimization from the C compiler, Code Warrior
(0-3), makes further optimization. As well, the
proper use of intrinsic functions from C compiler
can further reduce the number of cycles.
A very good approximation on computing time per
iteration shows that it is proportional to adaptive
filters order:
tc M (7)
We shall refer to from now on as proportionality
constant. During implementation on StarCore, the
evolution of this constant was most relevant and it
is described in Fig. 6.
The evolution of proportionality constant
1000
900
800
700
600
500
400
300
200
100
0
1 2 3 4 5 6 7 8
Steps
Fig. 6. Evolution of proportionality constant
In Fig. 6, on X axis, optimization steps are
represented in time. The optimization process also
included taking advantages of the parallel
architecture and can be found in [2].
We describe the stages from Fig. 2:
1-3. Intrinsec optimizations from procedure
prediction (use auxiliary value for J, rearrange the
cos factor computation)
4. Modularization by using the procedure
prediction
5-6. Levels 0-1 of optimization
7. Use in-place transformation
8-9. Levels 2-3 of optimization
V. REMARKS
A modified version of QRD-LSL algorithm,
suitable for implementation on DSP has been
presented, along with the description of the
architecture of the StarCore, the DSP used for
implementation. Because this algorithm is known
to be complex and because it needs to run in real
time, it is necessary to furthermore reduce the
number of cycles. The next step is to write special
assembly routines within the program, replacing
the complex operations.
REFERENCES
[1] C. Paleologu, S. Ciochina, A. A. Enescu, Modified
versions of QRD-LSL Adaptive Algorithm with Lower
Computational Complexity, Rev. Roum. Sci. Techn.
Electrotechn. et Energ., vol. 46, no.3, 2001.
[2] C. Paleologu, S. Ciochin, A.A. Enescu, A Network Echo
Canceller Based on a SRF QRD-LSL Adaptive Algorithm
Implemented on Motorola StarCore SC140 DSP, IEEE
Int. Conf. ICT 2004, Fortaleza, Brasil, 2004
[3] S. Ciochina, C. Paleologu, On the Performances of
QRD-LSL Adaptive Algorithm in Echo Cancelling
Configuration, Proc. IEEE ICT 2001, Bucharest,
Romania, vol.1, 2001, pp. 563-567.
[4] S. Haykin, Adaptive Filter Theory, Third Edition, Prentice
Hall International, Inc. Englewood Cliffs, 1996.
[5] C. Paleologu, S. Ciochina, A. Enescu, A Simplified
QRD-LSL Algorithm in Echo Cancelling Configuration,
Proc. IEEE ICT 2002, Beijing, China, vol.1, 2001, pp.
563-567.
[6] SC140 DSP Core Reference Manual, Revised 1, 6/2000
[7] St. Gay, An Efficient, Fast Converging Adaptive Filter
for Network Echo Cancellation, Proc. Asilomar, Pacific
Grove, CA, Nov. 1998, pp 394-398.
[8] Ph. Regalia, Numerical Stability Properties of a QR-
Based Fast Least Squares Algorithm, IEEE Trans. Signal
Processing, vol. 41, no. 6, June 1993, pp 2096-2109.
[9] M. Hartenek, R. W. Stewart, J. G. McWhirter, I.K.
Proudler, Algorithmic Engineering Applied to the QR-
RLS Adaptive Algorithms, Proc. 4th International
Conference on Math. Signal Proc., Warwick, U.K. 1996.
[10] Regalia P., Bellanger G. On the Duality Between Fast QR
Methods and Lattice Methods in Least Squares Adaptive
Filtering, IEEE Trans. Signal Processing, vol. 39, no. 8,
April 1991, pp. 879-891.
[11] Ciochin S., Negrescu C., Adaptive Systems, Ed. Tehnic,
9 Bucharest, 1999.
[12] Liu J. "A Novel Adaptation Scheme in the NLMS
Algorithm for Echo Cancellation", IEEE Signal Processing
Letters, vol. 8, no. 1, January 2001, pp. 20-22.
[13] J.G. Proakis, C. M. Rader, F. Ling, C. L. Nikias, Advanced
Digital Signal Processing Algorithms, Macmillan
Publishing Company, 1992.
[14] W.M. Gentleman, Least Squares Computations by Givens
Transformations without Square-Roots, J. Inst. Math. Its
Appl., vol. 12, 1973, pp. 329-336.
[15] ITU-T Recommendation G.168, Digital Network Echo
Cancellers, 2000, Draft 3.
[16] ITU-T Recommendation G.711, pulse code modulation
(PCM) of voice frequencies, CCITT-Blue Book, Volume
III, Fasc. III. 4, pp. 175-184.
[17] A. Andronache, C. Anghel, S. Pop, A Novel Adaptation
Scheme in the NLMS Algorithm for Digital Network
Echo Canceller Implemented on Motorola StarCore
SC140, Int. Conference COMM 2002, Dec. 2002,
Bucharest, Romania
169
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

An improved LMS algorithm for single and doubletalk

echo canceller implemented on Motorola DSP SC140
Daniel Silion, Dorin Panaitopol, Mircea Sorin Rusu1

Abstract - In this paper an improved single and C. Echo cancelling method

doubletalk echo canceller system is proposed in
conformity with the ITU-T recommendation G.168,
implemented on a Motorola SC140 digital signal
processor. The performance of the adaptive system by
comparison between convergence and error attenuation
of different kinds of LMS (Least Mean Square)
algorithms was studied. MatLab simulations results led
to the conclusion that the GNGD algorithm satisfies the
conditions presented above.
Keywords: echo canceller, LMS, GNGD.
Digital network echo cancellers are designed to
eliminate echo reflected from the user and to allow
successful transmission of voiceband data and fax.
Echo cancellers are devices placed in the 4-wire
portion of a circuit (which may be an individual
circuit path or a path carrying a multiplexed signal)
and are used for reducing the echo by subtracting an
estimated echo from the signal produced by hybrid
circuit.

I. INTRODUCTION

A. The apparition of the Echo

In telecommunication networks, the combination of

reflections from network components such as 2- to 4-
wire converters, together with the signal processing
and the transmission delay produce echo. Echo has a
major effect on voice quality in telecommunication
networks.
Fig.2. A simplified model of an Echo Canceller

In Fig. 2 it is represented a simplified model of an

Fig.1. Echo apparition
B. Problems involved
Users may encounter difficulty in talking or listening
over a telephone connection caused by the echo. It
may also affect the transmission of voiceband data,
fax and text over phone lines.
echo canceller, where x(n) is the input signal vector
(far-end signal) for the time instant n, ec(n) is the
echo, z (n) is the near-end sequence ( z (n) = 0 in case
of single-talk), d (n) is a combination of the echo and
the speech of the near-end talker, y (n) is the adaptive
filters output, w(n) is the filter coefficient (weight)
vector and e(n) is a combination of the reduced echo
and z (n) .
II. ADAPTIVE ALGORITHMS FOR ECHO
CANCELLATION
The adaptive algorithm is used to adjust filters
coefficients in order to minimize the echo, so in case

1
Faculty of Electronics, Telecommunications and Information Technology - University Politehnica of Bucharest
1-3 Iuliu Maniu Bd., Bucharest, Romania
e-mail: danny_silion@yahoo.com, dpanaitopol@yahoo.com, sorin252000@yahoo.com

170
of singletalk is desirable that d (n) y (n) = 0 and in from below by the performance of the NLMS,
case of doubletalk d ( n) y ( n) = z (n) 0 . whereas it converges in environments where NLMS
diverges. The GNGD is shown to be robust to
significant variations of initial values of its
A. LMS (Least mean square algorithm) parameters. Simulations in the prediction setting
support the analysis.
The least mean square algorithm is a simple, yet most The proposed GNGD algorithm is described by:
frequently used, algorithm for adaptive finite-impulse
response (FIR) filters. It is described by the following:
y (n) = x T (n) w(n) (5)
y (n) = x (n) w(n)
T
(1)
e( n) = d ( n) y ( n) (6)
e( n ) = d ( n ) y ( n ) (2)
w(n + 1) = w(n) + (n)e(n) x(n) (7)
w( n + 1) = w( n) + e( n) x( n) (3)

( n) = (8)
x ( n) 2 + ( n)
2
The parameter is the step size (learning rate) that
defines how fast the algorithm is converging.
Ideally, we want an algorithm for which the speed of e(n)e(n 1) x T (n) x(n 1)
convergence is fast and the steady-state (n) = (n 1) (9)
( x(n 1) 2 + ( n 1)) 2
2

misadjustment is small when operating in a stationary

environment, whereas in a nonstationary environment
the algorithm should change the learning rate where the (.) T is the vector transpose operator and
according to the dynamics of the input signal, so as to || . || 2 denotes the Euclidean norm.
achieve as good a performance as possible.
Because of this, the normalized LMS (NLMS)
algorithm has been introduced. D. Variable Step-Size NLMS

B. NLMS (normalized LMS) The step-size governs the rate of convergence and
the steadystate excess mean-square error. To meet the
The step size of NLMS was found to conflicting requirements of fast convergence and low
be (n) = x(n) 2 , 0 < < 2 , where || . || 2 denotes misadjusment, the step-size needs to be controlled. In
2

standard LMS, various variants for controlling the

the Euclidean norm. Equation used for the step-size have been proposed. The performance of
actualization of the filters coefficients is described these schemes is determined by how accurately they
below: can estimate how far the filter is from optimal
performance. In this paper, we have discussed three
kinds of VSS-NLMS:
w( n + 1) = w(n) + e( n ) x ( n ) (4)
+ x ( n)
2

VSS-LMS:
In theory, value = 1 provides the fastest (n) = (n 1) + e 2 (n) (10)
convergence, whereas in practice, the step size of the
NLMS algorithm needs to be considerably smaller. RVS-LMS:
To preserve stability for close-to-zero input vectors, (n) = (n 1) + p 2 (n) (11)
the optimal NLMS learning rate is usually modified
as x(n)
2
( x(n) 2 + ) , where is a small
2
p(n) = p( n 1) + (1 )e(n)e(n 1) (12)
2

positive constant.
VS-NLMS:
C. GNGD (generalized normalized gradient descent x2 (n)
algorithm) ( n) = ( n) (13)
e2 (n) +
The GNGD represents an extension of the normalized
least mean square (NLMS) algorithm by means of an x2 (n) = 1 x2 (n 1) + (1 1 ) x 2 (n) (14)
additional gradient adaptive term in the denominator
of the learning rate of NLMS. This way, GNGD e2 (n) = 2 e2 (n 1) + (1 2 )e 2 (n) (15)
adapts its learning rate according to the dynamics of
the input signal, with the additional adaptive term
e 2 ( n)
compensating for the simplifications in the derivation (n) = (n 1) + (1 ) (16)
of NLMS. The performance of GNGD is bounded x ( n) +
2

171
To preserve stability for close-to-zero input vectors,
two very small positive constants and were
introduced as a necessity in the experiments.

III. REQUIREMENTS OF G168

RECOMMENDATION

Experiments were performed with ITU-G168 echo

paths. The tests use special signals such as noise,
tones, group 3 facsimile signals, and a subset of the
composite source signals (CSS) consisting of the
band-limited CSS with speech like power density
spectrum and the band-limited CSS for double talk.
The CSS emulates the characteristics of the speech,
and its use as a test signal improves the ability to Fig.3. Comparison between GNGD, NLMS and RVLMS algorithms
measure echo canceller performance for speech
signals. In experiments were used the signals css_st,
for far-end talker and css_dt, for near-end talker.

IV. SIMULATION RESULTS

Using MatLab simulations, different algorithms were

compared by the point of view of convergence speed
and echo attenuation. The echo signal was obtained
using ITU-G168 echo paths, especially B1 path with
N=64 elements.
The algorithms were studied in optimum conditions,
the parameters for each algorithm were chosen by
means of faster convergence and better attenuation.
In experiments, for NLMS and GNGD, value = 1
provides the fastest convergence, whereas in practice, Fig.4. Comparison between GNGD and VSNLMS algorithm
the step size of the NLMS algorithm needs to be
considerably smaller. Whatever, the interest is to find
a optimum algorithm by comparisons in the same
conditions and environment.
The tests for GNGD were made using = 1 and
= 0.9 .
Parameters used for Variable Step-Size NLMS
algorithms are shown in Table 1.

Table 1

VSS-LMS RVS-LMS VS-NLMS

max = 1 = 0.021 1 = 0.03
min = 0.02 = 0.995 2 = 0.93
= 0.995 = 0.99 = 0.90
= 0.015 = 0.013 Fig.5. Comparison between GNGD and VSSLMS algorithm

As it is shown in figures presented below, some
algorithms are proved to have a lower speed of
convergence and others a lower attenuation of the
echo, than GNGD.
In Fig. 6 are presented the test signal (CSS_ST), the
echo signal and the error signal (the minimized echo)
for GNGD algorithm. For single-talk case, z (n) = 0 ,
so in this simulation, CSS_DT is not used. In this
case, d ( n) = ec( n) .

172
 e(n)e(n 1) xT (n) x(n 1)
(n) = (n 1) (9)
( N x2 (n 1) + (n 1)) 2

where x2 (n) = (1 ) x2 (n 1) + x 2 (n) (17)

As it can be seen below, GNGD behaves better than

NLMS or the other studied algorithms even in the
case of double-talk. In Fig. 8 were compared the echo
attenuations in both cases.

Fig.6. Signals for GNGD algorithm single-talk

The analyse was also made in the case of doubletalk,

were z (n) 0 . In this case, d ( n) = z (n) + ec(n) .

Fig.8. Comparison between GNGD and NLMS algorithm

double-talk

V. IMPLEMENTATION ON DSP

The real time algorithm implementation was tested on

an Application Development System equipped with
SC140 DSP and A/D, D/A codec at 8kHz sampling
frequency. The program was optimized for the
improved SC140s architecture (4 ALUs, dual data
memory transfer) to reduce time execution cycles.
The code implementation was performed using
MatLab reference code. The results using fixed-point
C specific instructions were similar to MatLab
simulations.

VI. CONCLUSION

Fig.7. Signals for GNGD algorithm double-talk

The error vector is used as a criterion to determine
how close the adaptive filter is to optimum
performance. The algorithms show improved system
In case of double-talk, the equations used for the performance.
actualization of the filters coefficients for NLMS (4), A generalized normalized gradient descent algorithm
and for GNGD (7), (8), (9) were modified into: for
linear adaptive filters has been proposed for an echo
NLMS: cancellation system. It has been derived as an
extension of the normalized least square algorithm
w(n + 1) = w(n) + e( n ) x ( n ) (4)
+ N x2 (n) where the learning rate comprises an additional
GNGD: adaptive factor, which stabilizes NLMS and makes
GNGD suitable for filtering of nonlinear and
( n) = (8) nonstationary signals. Unlike the previously proposed
N x2 (n) + (n)
gradient adaptive step size algorithms, GNGD has

173
been shown to be robust to the initialization of its
parameters.
VII. ACKNOWLEDGMENT
The authors wish to thank Freescale Semiconductor -
Romania for their support to accomplish this study.
Implementations of algorithms presented above were
performed using software and equipment donated by
Freescale Semiconductor - Motorola Romania to our
laboratory.
174
REFERENCES
[1] ITU-T Recommendation G.168, Digital Network Echo
Cancellers, 2000.
[2] ITU-T Recommendation P.56, Objective measurements of
active speech level.
[3] Danilo P. Mandic, A Generalized Normalized Gradient
Descent Algorithm, IEEE Signal Processing Letters, Vol. 11,
No. 2, February 2004.
[4] Hyun-Chool Shin, Ali H. Sayed, Woo-Jin Song, Variable
Step-Size NLMS and Affine Projection Algorithms, IEEE
Signal Processing Letters, Vol. 11, No. 2, February 2004.
[5] Silviu Ciochina, Cristian Negrescu, Adaptive systems,
Editura Tehnica, Bucharest, 1999.
[6] Jianfeng Liu, A Novel Adaptation Scheme in the NLMS
Algorithm for Echo Cancellation, IEEE Signal Processing
Letters, Vol. 8, No. 1, Jan 2001.
[7] Tomas Gnsler, Steven L. Gay, M. Mohan Sondhi, Jacob
Benesty, Double-Talk Robust Fast Converging Algorithms for
Network Echo Cancellation, IEEE Transaction on Speech and
Audio Processing, Vol. 8, No. 6, November 2000.
[8] SC140 DSP Core Reference Manual, Revision 3, November
2001.
[9] SC100C Compiler Users Manual, Revision 2, November
2001.
[10] StarCore SC140 Application Development Tutorial, Revision
0, March 2003.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Precision Electronic Driver for Pneumatic Engines

Marius Otesteanu1, Daniel Criste1
Abstract. - The paper presents high precision electronic
solutions for propulsion and steerage an air-cushion
industrial transporter. Compared to the usual on / off
control solution for electromechanical interfaces, the
proposed adaptive PWM control solution allows up to
ten times error reduction in pneumatic motors use. The
software algorithm implementation and the
microcontroller embedded system are presented.
Keywords: pneumatic systems, propulsion control,
steerage control, intelligent control, embedded systems,
pulse-width modulation
I. INTRODUCTION TO AIR CUSHION TRANSPORT
The air cushion transporter is an industrial vehicle
used to move, usually indoor, different loads, from
tens of kg to hundreds of tones. It offers the
possibility to move large charges and to position them
with high accuracy, on any layout [1].
Using air cushion transporters in assembly lines, for
moving cars, buses, tramcars, railway wagons or
airplanes, the area of the production facility can be
reduced several times, with investment advantages. In
order to efficiently use the limited space in an
assembly hall, the movement must be precisely
controlled, in speed, forward or backward, in
direction, angle to right or to left, etc.
Because of the wide charge range, of the different
movement and positioning necessities and accuracy,
of the need to fit particular configurations, the
transporter control must be accurate, adaptive and
flexible.
For air cushion transport, the surface must be
horizontal, smooth and continuous. The transporter
uses at least 4 air cushions, for stability and at least 2
active wheels, for propulsion and steerage (figure 1).
When stopped, the air cushions have low pressure, so
the transporter lies on wheels, with all weight. Before
moving, the air cushions are pressured, so the
transporter is elevated fractions of millimeters from
the surface, because of the air wave.
Almost all the weight is taken over by the air
cushions, but a small percentage of the weight is still
pressing the wheels. This is necessary for good
wheels surface contact, in order to move the
transporter, with propulsion and steerage control.
For propulsion, forward or backward, at least the
front wheel is driven. For steerage, with a defined
angle to left or right, usually only the front wheel is
driven.
For each wheel, one motor is necessary for propulsion
(rotation in a vertical plane) and a second motor is
necessary for steerage (rotation with an angle in
horizontal plane). So, the basic structure, of the
transporter, with 2 wheels, needs 4 driving motors.
Theoretically, the motors can be electric, hydraulic or
pneumatic. Practically, the pneumatic motors are
preferred because all pneumatic system components
already exist: air compressor, flexible pipes to the
cushions, electro valves, pneumatic regulators and
others. So the all pneumatic solution results cheaper
than any hybrid solution.
AIR CUSHION
PROPULSION AND
STEERAGE WHEEL
Fig. 1 Basic configuration of air-cushion transporters.
II. CONTROL SIGNALS FOR A TRANSPORTER
For best performance in controlling the transporter,
additional functions are useful. In order to minimize
the operations area, when transporting huge charges,
the radius of the curve can be reduced, by steering
both wheels, with opposite angles. The smallest circle
radius is obtained for 45 angle, corresponding to the
circle function. Another important movement is the
lateral parking, parallel to a wall. This is the cross
function, obtained by rotating, both wheels, with 90.
In this mode, both wheels are driven for propulsion.
The circle and cross control signals are obtained from
switches, so they are, always, digital signals.

1
Politehnica University of Timisoara, Bv. V. Parvan 2, 300223 Timisoara, Romania
marius.otesteanu@etc.utt.ro

175
The forward and backward propulsion control signals
and the left and right steerage control signals are
obtained from two joysticks, which can be either
digital, or proportional resulting either digital or
analog signals. The type of signals used, digital or
analog, is determined by the particular application.
III. PNEUMATIC SYSTEMS BEHAVIOUR
The pneumatic motors used in transporters are
characterized by inertia. Compared to precise electric
motors, pneumatic motors, because of the inherent air
compression, have a slow response to impulse
commands. Thats why digital control can only be
used in applications where small trajectory or
positioning errors are accepted. For higher accuracy
driving, the proportional control is recommended.
Precision driving, in forward backward propulsion,
means quick start, relative high speed motion and stop
at a definite marker. Usually, positioning errors of 5
10 mm are obtained with pneumatically driven
wheels.
For steerage, the propulsion wheel is left right
driven with a second pneumatic engine. Precision
steering means forward motion (0), cross motion
(90) or any definite angle. Usually, angular errors of
3 5 are obtained.
With the traditional on / off driving pneumatic
engines, because of the air compression, higher
accuracy cannot be obtained. The inertia results very
high, especially with hundreds of tones charge.
Alternatively, PWM driving is not proper for moving
mechanical devices, as electrovalves.
Because the pneumatic mechanisms used depend on
the pressure of the compressed air, on the compressed
airs behavior in the pneumatic circuits and, not less
meaningful, on the control mode of the admission and
evacuation of the air, all those factors have to be taken
into account in order to develop a proper control
solution.
The first approach was oriented on the pressure of the
air inserted in the pneumatic circuits. It was
determined that for high pressures of 5 6 Bar, the
mechanism has a very good behavior, excepting the
zone close to the target position, where the movement
continues with the same high speed. When the target
position is assumed reached and the decision to stop
the air admission is taken, due to the high air pressure,
a very long relaxation process appears until reaching
the normal atmospheric pressure. During this
relaxation process, the mechanism continues to move
after the target position is reached, producing a high
positioning error, E/2.
Lower air pressures of 2.5 3 Bar were tested, but
with such pressures, the answer of the pneumatic
equipment is very poor, and it is not possible to move
the mechanism to the required position within the
maximum delay accepted for such operations (5 s).
From this point of view, the solution would be to use
a pneumatic device to control the air admission and/or
176
evacuation. Unfortunately, such devices offer poor
positioning precision at high costs.
Another approach was oriented on the possibility to
modify the air circuits by introducing some variable
volume elements. This would result in reducing the
speed and therefore the mechanisms inertia in the
positioning zone. By these means, the precision may
be substantially improved, but the main disadvantage
is the necessity of a control equipment which should
decide the moment and time period when the
admission volume is modified. A possible solution
would be the use of proportional electrovalves, also
known as boosters, but this solution implies very high
costs, and is only used in systems with very high
performance requirements.
The third, and last approach, was considered the most
adequate for the solution to be implemented. This
approach tries to find a compromise between the
problems appeared in the previous approaches, by
controlling the air pressure, and the air admission in
the equipment, obtaining a quasi-variable pressure,
which can be controlled between some limits, enough
to obtain steerage angular errors as low as 0.25.
IV. NECESSITY OF FEEDBACK CONTROL
The traditional on / off driving pneumatic equipments,
even controlled by electronic command modules,
following continuously the parameter indicating the
current position of the mechanism (figure 2) have
great inertia and the positioning error is also great. In
this case the system may even oscillate, because it is
not able to stop within the required error window.
Electronic Pneumatic
control Command driven
system mechanism
Control value
Fig. 2 Pneumatic equipment driven by electronic control system.
In order to stop the mechanism with a very small
error, close to the target position, it is necessary to
reduce the movement speed of the mechanism
dynamically and almost linearly, by controlling the air
pressure in the pneumatic circuit to a value which
ensures the equivalent of a mechanical braking until
the target position. Figure 3 emphasis the possibility
of instability when high pressure (high force and
rotation speed) is used, associated with narrow error
window (for high accuracy), e/2.
Position
5-6 bar 2-3 bar 0-0.1 bar
initial Intelligent control
ON/OFF control
+E/2
+e/2
target
-e/2
-E/2
Time
Fig. 3 Behavior of pneumatic systems near the accepted error window.
 V. CONTROL ALGORITHM
Because there are no pressure sensors to allow the
electronic system to monitor this parameter, the only
way to get information from the pneumatic-
mechanical equipment is through angular position
transducers. These transducers provide information on
the mechanical position of the pneumatic driven
equipment, by means of electrical signals. Using this
data, an intelligent electronic system is able to
calculate the distance between the current position
and the target position and to determine the necessary
time and position for the appropriate braking, in order
to stop the rotation on the target position. The
electronic system has to modify the air pressure in the
pneumatic circuit, according to the computed values.
The solution to implement the control above is to
command the electrovalves, which control the air
admission in the pneumatic circuit, by variable width
electrical pulses, as represented in figure 4.
Position
p1
PWM
p2
control
p3
+e/2
target
Control
signal
Time
Fig. 4 PWM command of electrovalves.
The intelligent control algorithm is implemented in
two steps:
in the first step, the distance to the target position
is computed and a decision is taken whether this
distance is long or short; if the distance is long,
the air admission is continuously controlled until,
considering the rotation speed and the distance to
the target position, the braking decision is taken;
in the second step, it is assumed that the distance
to the target position is short and the braking is
started by controlling the air admission with
pulses (PWM).
When the distance to target is small from the
beginning (for very small deviations control), the
continuous command of air admission is skipped and
the admission is directly PWM controlled. The first
pulses have larger width to ensure the necessary
pressure in the installation to start the movement of
the mechanism. Then, after the mechanism starts
moving, the pulse width decreases, as the mechanism
gets closer to the target position.
Using variable width pulses, the air admission in the
pneumatic equipment is kept under control, and the
pulse width change determines the decreasing of the
mechanism rotation speed. The method grants a good
flexibility, because if further changes are required,
they will affect only the electronic part, especially the
software, which allows easy parameters changing,
necessary for efficient control, and of some values
which depend on the operating modes.
177
This solution offers a good compromise between
positioning precision and equipment costs, the
pneumatic and mechanical equipment being the same,
just the proposed electronic control module having to
be added.
VI. HARDWARE IMPLEMENTATION
Considering the possibility to integrate on a low cost
single chip multiple modules as processor, memory,
A/D and D/A converters, counters, comparators,
PWM modules and so on, a microcontroller is
indicated for the implementation of the solution.
Position
10-bit
Transducers
ADC
Pneumatic
driven engine Microcontroller
Electrovalves
Turning speed I/O I/O
P P
Direction
O O
R R
Propulsion
T T
User interface
- position command
- parameters setting
Fig. 5 Microcontroller based electronic control system.
An important feature is the A/D resolution. The
steerage implies wheel rotation up to the range of
45 plus a cross rotation of 90, resulting a full
steerage range of 135 ( 45 90). Using a 10-bit
A/D converter, with 1024 quantization steps, a
resolution of 0.13 can be achieved, which is
necessary to control errors as low as 0.25. The
Microchip family of microcontrollers PIC16F87x
complies with all the above requirements and besides
offers some additional peripherals which allow the
implementation of serial communication through
RS232, RS485 and I2C interfaces, for interaction with
other similar devices or with intelligent terminals like
PCs [2].
VII. SOFTWARE IMPLEMENTATION
In the software implementation, the next two
parameters have to be optimized:
the minimum duration of a control pulse which
can open the electrovalve (referred also minimum
response time) and
the maximum frequency, at which the
electrovalve can work, when a fixed frequency
PWM control signal is used.
The usual working frequencies of electrovalves,
according to the producers, are limited at 33 Hz. The
minimum response time was experimentally
determined at about 12 ms.
Those limitations are caused mainly by the moving
mechanical parts, which require relatively long time
delays to start / stop moving. Therefore, for efficient
control of the electrovalves, the minimum pulse width
was set at 15 ms, and the working frequency at 25 Hz
(i.e. 40 ms period). Further practical tests proved that
only when the pulse width changes with at least 5 ms,
a visible effect on the movement speed can be
observed.
Starting from the above considerations, an intelligent
control algorithm was implemented to control the
admission electovalves. The feedback is based on the
information obtained from the transducer, placed on
the rotating mechanism. The signal provided at the
control output has a frequency of 25 Hz and a width,
adjustable in steps of 5 ms, according to the
movement speed and the distance to the target
position. By decreasing the pulse width, while the
position of the mechanism gets closer to the target
position, the movement speed is decreased and, at the
target position, the mechanism can be stopped within
the required precision window of 0.25. The main
purpose of the braking is to minimize the mechanical
inertia at the moment, when the air admission in the
pneumatic installation is stopped.
The software implementation of the PWM algorithm
is running on the microcontroller in real-time, because
all the signals have a slow time variation. Therefore a
5 ms period was chosen for the systems clock
(diagram in figure 6), enough to allow the
microcontroller to perform all the computations
required by the control algorithm for up to 3 control
loops and other computations for processing
additional external information.
NO
5ms timer
interrupt? Command
outputs
YES
Decide pulse
DAC
width
Sample position
Compute Estimate remaining
current error distance
Fig. 6 Flow diagram of the software implementation.
This 5 ms time solves also the problem of precise
change of the pulse width steps. So, by implementing
a counter which counts eight 5 ms time slots, a 40 ms
period (25 Hz) is obtained for the control signal. It
must be taken into account that the input signal must
be active in the first 3 time slots (15 ms the
minimum pulse width), and if it is decided that a
longer pulse is required, more active time slots will be
added. If the decision requires the direct command,
then all the 8 time slots will be active. The
information provided by the position transducer is
sampled each 5 ms and if it is assumed that the
mechanism reached the required position, the control
signal is disabled, whether the PWM cycle is
completed or not.
The algorithm relies on the principle of minimizing
the error between the reference position (target
position) and the current position of the mechanism
(indicated by the transducer). The pulse width is
modified as the mechanism gets closer to the target
position. The change of the pulse width depends also
on some adjustable parameters, which can be set by
an operator:
braking shape,
braking duration,
error window.
For large error window, the movement is stable, but
the final error might be too large. For narrow error
window, if the braking system is not precise enough,
oscillations occur, which can become dangerous for
charges of tones. The set of parameters has to be
carefully chosen from a wide range of possibilities
(with 3 variables) in order to achieve the best
performance.
These parameters must take into account the working
conditions and the external factors which may
influence the working parameters of the pneumatic
equipment. Various working modes may be chosen,
from abrupt braking shape in a small time window
with up to 2 steps for the pulse width, to slow braking
with 5 steps for the pulse width, involving a longer
braking time, as represented in figure 7.
100%
87.5%
75%
62.5%
Command 50%
steps 37.5%
STOP
PWM
signal
Ts 5 ms 40 ms Tmin 15 ms
Tpwm
Legend:
Ts sampling period
Timer
... ... Tpwm PWM signal period
interrupts Tmin minimum pulse width
Fig. 7 Pulse widths for 5-steps braking.
These working modes are specific for situations when
the moving system encounters various degrees of
resistance in achieving the target position. If an
undesired slow movement of the mechanism is
detected, before getting close to the target position,
the algorithm decides to increase the pulse width of
the control signal. By this means, the general pressure
in the pneumatic driving system increases, resulting
an increased speed of the positioning mechanism and
avoiding the possible blocking situations due to
insufficient pressure.
178
 VIII. CONCLUSIONS

The paper presents the results of the applied research

and development of an embedded system designed to
improve the accuracy of positioning and steerage air-
cushion industrial transporters [3]. The main target
was to avoid oscillations for the straight forward
movement. To this end, the angular error was reduced
as low as 0.25, by using an adaptive steerage
control algorithm, based on low frequency PWM.
The system and the algorithm can be used for any
pneumatic motor in order to precisely control the
rotation phase.

ACKNOLEDGMENT

This paper presents results obtained in a research and

development grant, supported by CNCSIS Romania
and DELU GmbH, Germany, producer of air cushion
transport systems.

REFERENCES

[1] Michels, H., Handelman, H.: Luftgleitkissen

unterstutzen die flexibile Montage. Fordern und
Heben 47, 1997, Nr. 6, 46-48
[2] Microchip Technology Inc. PIC 16F87X Data
Sheet, 28/40-pin 8-bit CMOS FLASH
Microcontrollers, 2001
[3] Otesteanu, M.: Embedded system for air-cushion
transporter control, IFAK Workshop on
Programmable Devices and Systems, Krakow,
Poland, 2004

179
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Hardware Simulation and Debugging For Microchip

RISC Microcontrollers
Aurel Gontean, Mircea Bbi, Roxana Jibleanu1
Abstract This paper introduces the possibility of using
simultaneously both the Microchip MPLAB IDE and the
Lab Center Proteus simulation software. Differences
between the simulated results and the real world
behavior are emphasized in terms of timing and
initialization. Source code debugging (stepping and free
run modes), assembling and compiling within Proteus
are discussed.
Keywords: simulation, microcontroller, debug
I. INTRODUCTION
Programming embedded systems usually rely on
Integrated Development Environment software, such
as Microchips MPLAB IDE (Fig.1). The source code
is edited, simulated and eventually programmed in the
microcontroller using the same application - a step
ahead compared with the command line DOS based
versions of earlier assemblers.

Fig. 1. MPLAB IDE - A Typical Integrated Development System.

1
Facultatea de Electronic i Telecomunicaii, Departamentul
Electronic Aplicat, Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail {aurel.gontean; mircea.babaita; roxana.jibleanu}@etc.utt.ro

180
Although evolved, modern IDE software is not
hardware oriented it is impossible to simulate
continuous voltage values, currents, and transistors
not to mention LEDs, LCDs, buzzers or motors. There
are areas not covered at all by the IDE software
RS232, I2C, or SPI communication, or difficult to use
PWM, soft delays. Therefore there is a high demand
for an appropriate simulation tool in both design and
teaching embedded systems.
II. PROTEUS SIMULATOR
Proteus software offered by Labcenter Electronics is a
solution allowing for mixed analog and digital
simulation, along with models for Microchip
midrange microcontrollers.
Fig.2. Using Fourier analyses to evaluate the output filtered signal.
Fig.3. PWM Simulation Results.
181
A software PWM generator is almost impossible to be
evaluated under MPLAB control. Once built, the hex
file is loaded in the microcontroller (Fig. 2) and the
dynamical behavior may be analyzed with virtual
instruments (Fig. 3).
The combination of the two software packages allows
the user to design most or even all of the schematics
and to edit, debug and verify the software with little
or even no need for prototyping, shortening the
overall design time.
A. Hardware interaction
Beyond pure software simulation, Proteus offers also
hardware interaction. A microcontroller model with a
digital sounder may work together in order to generate
audible sounds. Fig. 4 demonstrates a typical case
where Proteus VSM is running a simulation of a PIC
program which generates audio tones in real time. The
sounder model picks up the transitions on port A,
RA3, and converts them into a 44 kHz data stream
which is sent to the sound card. On a Pentium II or
better PC, the simulated PIC will run fast enough to
generate audio tones in real time.
Fig.4. Using the Logic Analyzer.
B. Using displays
Fig. 5. Using Digit Multiplexing.
A variety of displays is a key feature of the simulator.
Fig. 5 introduces the 7 digit display multiplexing
schematic, where the real behavior is fairly closed
with the simulated one.
Another useful feature is the LCD simulation for the
standard LM032 display (fig.6). The simulation
supports both 4 bit and 8 bit modes, while busy flag
can be read on D7, exactly like in the normal use.

C. Serial transmission

Serial transmission may be accomplished either with

the virtual terminal (Fig. 7), or using the hardware
interaction implemented in the Compim device. The
virtual terminal allows reading and transmitting data
using standard TTL levels, while the Compim device
reads and writes an existing Com port in the PC.
XON/XOF and hardware control may be used and
framing error is signalized through proper indicators. Fig.6. The LM032 LCD simulation.

Fig.7. RS-232 Transmission.

III. LIMITATIONS AND ERRORS programmable warnings for stack under -or overflow,
and incorrect jumping computation.
MPLAB IDE is a powerful tool which is continuously
improved by Microchip. Some peripherals are not
supported at all (such as the USART), and others are
only partial simulated (the ADC conversion does not
change the ADRESH and ADRESL registers). There
are differences in timing Timer1 has a constant 4 s
error compared with the real behavior.
Proteus simulator has also some errors, for example
the 74LS148 model is wrong the I 0 input does not
trigger the EO nor the GS lines, as it should (Fig.
8). Another drawback is that only single file code can
be source code debugged there is no possibility to
link files within Proteus. However difficult items such
Fig. 8. There is no reaction for the I0 input line stimulus.
as banking are accurate simulated there are

182
 IV. COMBINED ACTION
For educational and design purposes, the best result is
accomplished working using both packages. Changes
in MPLAB source code are reflected in the .hex file
and loaded in background in the Proteus model each
time a new simulation is started (there is no need to
restart the simulator). Working with an ICSP
programmer is a low cost debug solution, suitable for
students; while an ICD2 debugger is a professional
approach at a moderate price, with the advantage of
stepping through the code, breakpoints, real timing,
the Proteus simulation may replace it for most cases.
V. CONCLUSIONS
The combination of the two software packages allows
the user to design most or even all of the schematics
183
and to edit, debug and verify the software with little
or even no need for prototyping, shortening the
overall design time.
For teaching purposes, the overall performance is
excellent, the learning curve being dramatically
reduced.
REFERENCES
[1] http://www.microchip.com.
[2] http://www.labcenter.com.
[3] * * * MPLAB IDE Simulator, Editor. Users guide,
Microchip, 2001.
[4] * * * MPLAB ICD2Debugger. Users guide, Microchip,
2002.
[5] A. Gontean, The PIC16F84A Microcontroller, Editura
Orizonturi Universitare, 2004 (in Romanian).
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

General Purpose PIC16F84A Based Development Board

Mircea Bbi, Aurel Gontean, Roxana Jibleanu1
Abstract The goal of this paper is to present the main EPIC-1 board capabilities [3] and expands the main
features of a PIC16F84A microcontroller based features of the microcontroller, providing:
development board, intended for teaching students - 8 analog inputs;
microcontrollers and embedded systems. - 8 digital inputs and 8 digital outputs;
Keywords: microcontroller, development board
- 1 isolated digital input and 1 isolated digital
output;
I. INTRODUCTION
- 1 general purpose output with relay;
- display interfaces for a LCD 16x2 display, 2
PIC16F84A is a high performance RISC controller
digits and 8 LEDs;
operating up to 20 MHz, offering 1024 words of
- 4 buttons input, a buzzer;
program memory, 68 bytes of RAM and 64 bytes of
- a standard RS232 interface;
EPROM. The package provides only 18 pins, and
- an ICSP programming port for In Circuit
only 13 may be used for input/output, making any
Programming.
extension suitable. The proposed board enhances the

Fig. 1. PIC16F84A development board.

The board is compatible with both Microchips
MPLAB IDE and JDM programmer, thus enabling the
student with means for editing, simulating,
programming and testing his own routines. Being
FLASH based, the PIC16F84A microcontroller may
be erased and program without removal from the
socket via the JDM programmer and ICSP port. The
board comes with specific demos highlighting the
hardware characteristics of the system (usage of shift
registers for input and output, multiplexing digits,
building and external bus line) and also advanced
software topics (interrupt driven RS232 routines,
LCD interfacing, SPI communication with ADC).
All the hardware and software involved with this
board was developed in Timisoara, at the Electronics
and Telecommunication Faculty.

1
Facultatea de Electronic i Telecomunicaii, Departamentul
Electronic Aplicat, Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail {mircea.babaita; aurel.gontean; roxana.jibleanu}@etc.utt.ro

184
 II. BOARD DESCRIPTION

Due to the small number of I/O pins, shift registers

are used to add more inputs and outputs. There are
two such registers, one is input which increases the
number of inputs to 8 more, and one output which
increases the number of outputs also to 8 more. They
are connected with a microcontroller via four lines
found on the port A.
Several displays are used for program monitoring:
eight LED diodes which can be moved to output of
shift register on port B pins, but that can also be
removed completely by taking all eight jumpers off;
two seven segment digits in multiplex mode that can
successfully satisfy some numeric applications. Fig. 3. LCD usage.
Multiplexing is achieved with two transistors found
on RA0 and RA1 lines; LCD display is very useful
because it provides a lot of information to a
programmer, and its lowered cost promises it will
become an integral part off development systems of
the future. Using a potentiometer the contrast can be
adjusted.
Beeper on pin RA3 for sound signals can also serve as
a means of indication. RA3 pin is shared by several
components so it is necessary to choose which one of
them is to be used.
One of the most important elements pf the board is the Fig. 4. Connecting an optocoupler.
Microchips 8-Channel 12-Bit A/D Converters with
SPI interface [2] with an external reference of 5V.
The ADC interface is serial type so it takes three pins
for communication with the microcontroller.
RS232 communication is accomplished using a
standard MAX232 transceiver. RB0/INT line is used
as an Rx pin due to the interrupt feature of the pin,
while RA3 is used as the Tx pin (PIC16F84A has no
USART, so all communication is done be software).

III. RESULTS

Fig. 5. Multiplexed digit display.

Fig. 6. Serial transmission communication.

Several applications have been developed and tested

with the board. They include LCD display routines
(fig. 2 and 3), interfacing an optocoupler (fig. 4),
Fig. 2. Connecting an external LCD to the development board.

185
displaying with the multiplexed digits (fig. 5), SPI
interfacing for the shift registers and ADC, complete
RS232 communication software.
IV. CONCLUSIONS
The novel approach in this development board is the
possibility of in circuit programming the
microcontroller, using the ultra-low cost JDM
programmer. Various displays possibilities and
flexible architecture for the input switches make the
debug process an easy task. Although the PIC16F84A
186
has limited peripheral capabilities, with the extensions
provided buy the board the student or the designer
may experiment, debug and verify specific projects.
REFERENCES
[1] * * * MCP3204/MCP3208. 2.7V 4-Channel/8-Channel 12-Bit
A/D Converters with SPI, Microchip, 2001.
[2] A. Gontean, The PIC16F84A Microcontroller, Editura
Orizonturi Universitare, 2004 (in Romanian).
[3] * * *, EPIC-1 User Guide, Mikroelektronika, 2001.
[4] http://www.microchip.com.
[5] http://www.mikroelektronika.co.yu.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Development System Equipped with AT89S8252
Microcontroller
Petru Duma1, Luminia Scripcariu2
Abstract The work describes the hardware structure of a
development system based on AT89S8252 microcontroller.
This system is used for testing and checking various user
applications based on systems equipped with ATMEL
family microcontrollers. The monitor program of the
system and its commands are also presented.
Keywords: development system, ATMEL microcontroller,
monitor program commands.
I. INTRODUCTION
Systems equipped with microcontrollers are classified
depending on their purpose as dedicated systems and
development systems.
Dedicated microcontroller-based systems are used only
in the specific applications for which they were
designed, have a reduced volume of hardware and the
command software is included in the internal program
memory of the microcontroller.
Development microcontroller-based systems are used
for educational purposes, i.e. initiating the users in the
hardware and software technique of microcontroller-
based applications, but also for checking and testing
both programs and hardware structures designed for
various applications.
The hardware volume of these systems is larger, while
the monitor program must allow viewing and/or editing
resources contents, writing user programs, executing
them on segments, etc.
II. DEVELOPMENT SYSTEM EQUIPPED WITH
AT89S8252 MICROCONTROLLER
AT89S8252 microcontroller includes on a single chip
the following resources:
- a microprocessor optimized for command and control
applications;
- an 8 Kbytes in-system reprogrammable downloadable
FLASH program memory, which can be used up to
1,000 writing/erasure cycles;
- an interface for serial program loading;
- a three-level internal program memory access blocking
system;
- a 256 bytes RAM data memory;
- a 2 Kbytes EEPROM data memory, which can be used
187
up to 100,000 writing/erasure cycles;
- a memory space for special function register (SFR);
- four 8-bit parallel ports (P0, P1, P2, P3);
- three 16-bit counters (T0, T1, T2);
- programmable Watchdog timer (WDT);
- a unit for the serial asynchronous data communications
(UART);
- a serial peripheral interface (SPI);
- maskable interrupt system, with six sources on two
priority levels;
- low-power idle and power-down modes;
- clock oscillator operating on frequencies up to 24 MHz
(33 MHz).
The development system equipped with AT89S8252
microcontroller has the structure shown in fig.1.
The clock generator is internal and requires connecting
at pins XTAL1, XTAL2 a quartz crystal of 11.0592
MHz and two 30 pF capacitors.
The automatic power-on initialization of the
microcontroller is performed by the external RC group
(10 k, 10 F).
The manual initialization of the microcontroller can be
made using the K switch.
The monitor program of the development system is
placed in the internal FLASH program memory of the
microcontroller which requires /EA pin to be connected
at +5V.
In order to run applications with the development
system, they must be stored in an external non-volatile
memory. The author chose to use a 32 Kb EPROM
memory chip (27256), connected as in fig.2 and
addressed in 0000 7FFFH memory space, by
connecting BA15 to /CE and signals /PSEN to /OE.
In this case (/EA= + 5V) the program memory of the
development system consists of the 8 Kb of
microcontroller internal FLASH memory, addressed in
the memory area of 0000 1FFFH, and of 24 Kb of
external EPROM memory, in the memory area between
2000H 7FFFH. The solution is useful for development
systems with ATMEL microcontrollers that dont have
internal program memory, case that requires pin /EA to
be connected to Gnd.
 Fig. 1.
The microcontroller has separate command signals for
accessing the program memory (/PSEN- addresses 64
Kbytes of program memory) and the data memory (/RD,
/WR also address 64 Kbytes of data memory). In order
to execute user programs stored in the data memory, the
program memory space and data memory space must be
joined.
Fig. 2.
The SRAM memory chip (55257), connected as in fig.3,
has also a 32 Kbytes capacity and it is addressed in
8000H FFFFH memory space, by connecting line
BA15 to /CE and the signal /PSEN./RD./WR to /OE.
Fig. 3.
When addressing an external memory, P2 port generates
the high part of the address bus, while P0 port generates
the low part of the address bus multiplexed with the data
bus. In the first stage of any access cycle to the external
memory, the low part of the address bus is delivered on
the P0 port lines, while during the other stages, these
lines are used by data.
Using an external latch (74373) on the falling edge of
188
the signal ALE, the lower part of the address bus is
demultiplexed by the data bus (fig.4).
Fig. 4.
Using external memory in the structure of the
microcontroller-based system determines the loss of P0,
P2 and partly P3, which become data, addresses and
commands busses.
The communication of the development system with a
serial console / personal computer requires the use of the
serial asynchronous interface (UART) and of a counter
(T1 or T2) for setting the serial communication rate.
These resources of the microcontroller are indispensable
for numerous applications and therefore, several
peripheral circuits must be connected to the systems
structure in order to compensate.
The peripherals used are: INTEL 8255, INTEL 8253
and INTEL 8251, connected as in fig.5.
The INTEL 8255 is a parallel programmable interface
with 3 parallel ports of 8 bits each and compensate ports
P0, P2 and P3.
The INTEL 8253 is a programmable counter/timer
interface with three 16 bits counters and compensates
the counter used for setting the serial communication
rate.
The INTEL 8251 is a programmable serial interface,
used for transmitting and receiving serial synchronous
and asynchronous data (UART) and compensates the
microcontrollers serial interface.
These peripherals are assumed as data memory locations
and are addressed using 8205 decoder.
 Fig. 5.
III. THE MONITOR PROGRAM COMMANDS
After connecting the development system to the DC
power supply, the microcontroller starts to execute the
monitor program, which consists, at the beginning, of a
sequence of initializations, then the consoles display is
cleared and a monitor program launching message is
displayed. Afterwards, a loop is started, during which
the console keyboard, used for the transmission of the
commands towards the microcontroller-based system, is
tested.
When a user command is issued, the monitor program
receives the command and its parameters, tests the
syntax then allows its execution. After performing the
users command, the loop is resumed, awaiting a new
command.
Once the microcontroller-based system powered-on, its
specific process is working continuously, executing
either the monitor program or commands or user
programs issued by the user through the monitor
program.
The general syntax of a command is:
P1 _ P2 _ P3 _ ..... _ Pk .
where: is the name of the command starting with a
letter; P1, P2, P3, ..., Pk are the commands parameters,
consisting of data or addresses.
Between the parameters, a separator must be used,
consisting of BLANK ( _ ) or COMMA ( , ), any
command being concluded with terminator RETURN
(.). Any command of the monitor program can be
abandoned at any time if O key is pressed.
The syntaxes and the effects of the commands accepted
by the monitor program are described below, in
alphabetical order.
A .
Displays on the console the list of the commands
accepted by the monitor program.
B .
Sets the serial asynchronous communication rate for the
UART interface (19,200 bits/s, 9,600 bits/s, 4800 bits/s
2,400 bits/s, 1,200 bits/s, 600 bits/s, 300 bits/s, 150
bits/s).
189
CA P1 _ P2 .
Performs arithmetic operations: addition (P1+P2),
subtraction (P1-P2), multiplication (P1*P2), and
division (P1:P2), between parameters P1 and P2.
CL P1 _P2 .
Performs logic operations: AND, OR, XOR between
parameters P1 and P2.
DC P1 _ P2 .
Displays on the console the user program in machine
code, stored in the program memory from address P1 to
address P2. Each console row displays the address and
the 1 up to 3 instruction byte. The user program is
written in machine code using substitute command or is
loaded through the serial asynchronous interface.
DF P1 .
Displays on the console the user program listing
(maximum 26 28 Kb), assembled on a personal
computer. This program is stored in the external RAM
memory starting from address P1, using the serial
asynchronous interface.
DH .
Displays on the console the internal data memory area
of the microcontroller, indirectly addressable between
addresses 80H and FFH.
DL .
Displays on the console the internal microcontroller data
memory area, directly and indirectly addressable
between addresses 00H and 7FH; in the memory space
between 00H-1FH are placed the four memory banks of
eight general registers each, while the memory area
20H-2FH is also addressable on bit level.
DP P1 .
DD P1 .
DE P1 .
Displays on the console the contents of a program
memory area (DP command), external data memory area
(DD command) and internal EEPROM data memory
area, respectively, (DE command) from address P1 until
address P1+80H. Each row displays the address and the
contents of 16 memory bytes in hexadecimal.
DP P1 _ P2 .
DD P1 _ P2 .
DE P1 _ P2 .
Displays on the console the contents of a program
memory area (DP command), external data memory area
(DD command), and internal EEPROM data memory
area, respectively (DE command) from address P1 until
address P2.
DP command displays any internal program memory
area within the memory space between 0000 1FFFH
and any external program memory area within the
memory space between 2000H FFFFH, if /EA= + 5V,
or even any external program memory area whatsoever
if /EA=Gnd.
DE command validates the access to the internal
EEPROM data memory, then displays the specified area
and, in the end, re-blocks the access to this area.
DS .
Displays on the console the internal memory zone of the
microcontroller, addressable directly between the
addresses 80H and FFH; this memory zone consists of
the special function registers (SFR) area.
DT P1 _ P2 .
Displays on the console the characters corresponding to
ASCII codes stored in the external data memory from
address P1 to address P2.
The console display can be suspended if key A is
pressed, resumed if key C is pressed, or aborted if key
O is pressed.
E P1 _ P2 _ P3 _... .
Extracts break-points in the user program from
addresses P1, P2, P3, ...
FD P1 _ P2 _ P3 .
FE P1 _ P2 _ P3 .
FI P1 _ P2 _ P3 .
Fills the external RAM data memory (FD command),
the internal EEPROM data memory (FE command) and
the internal RAM data memory, respectively (FI
command) from address P1 to P2 with P3 data value.
G P1 _ P2 _ P3 _... .
Determines the execution of a user program from
address P1 until one of the addresses of the break-points
P2, P3, ... . After executing the user program segment,
the break-points assigned by this command are removed.
I P1 _ P2 _ P3 ... .
Introduces break-points in the user program at the
addresses P1, P2, P3...
LC .
Loads from a personal computer through the serial
interface, a machine code user program into the external
RAM program memory and runs the program. The
address used for loading the user program, the length of
the program, the execution address and the break-point
addresses are transmitted using the header of the file that
includes the user program.
190
LP P1 .
Loads from a personal computer through the serial
interface, the user program listing file (max. 26-28 Kb)
into the external RAM memory starting from address
P1. The machine code user program is extracted from
this file.
MD P1 _ P2 _ P3 .
ME P1 _ P2 _ P3 .
MI P1 _ P2 _ P3 .
Transfers the contents of the external data memory area
(MD command), internal EEPROM data memory area
(ME command), and internal data memory area,
respectively (MI command) from address P1 to address
P2 into the memory area starting at address P3 .
N .
Clears the console display.
P .
Authorizes the password access of the user to the SFR
peripheral registers.
R P1 _ P2 .
Receives a hexadecimal data block through the serial
interface, from a computer or another development
system and stores it in the external RAM data memory
from address P1 to address P2 .
SD P1 _ ... .
Displays and/or substitutes the contents of an external
data memory area that starts at address P1. The
command displays every byte of any external data
memory area, while the substitution is performed only in
RAM external data memory locations.
For every substitution commands, the use of BLANK
separator displays and/or substitutes the contents of the
following memory location, while the use of COMMA
separator displays and/or substitutes the contents of the
previous memory location. The command is closed with
( . ) terminator.
SE P1 _ ... .
Displays and/or substitutes the contents of an internal
microcontroller EEPROM data memory area that starts
at address P1. When the command is issued, the access
to the internal EEPROM data memory is validated, then
the memory locations are displayed and/or substituted,
and in the end, the access to this memory is blocked.
SI P1 _ ... .
Displays and/or substitutes the contents of an internal
microcontroller RAM data memory area that starts at
address P1.
SP P1 _ ... .
Displays and/or substitutes the contents of a program
memory area that starts at address P1. The command
displays every byte of any internal program memory
area within the space between 0000H and 1FFFH and of
any external program memory area within the space
between 2000H and FFFFH if /EA= + 5V and,
respectively, any external program memory area if
/EA=Gnd. The substitution is performed only in the
RAM external program memory locations.
SS P1 _ ... .
Displays and/or substitutes the contents of an internal
microcontroller data memory area destined to the special
function registers (SFR) that starts at address P1. Certain
microcontroller peripheral circuits provide password-
protected access.
ST P1 _ ... .
Substitutes the contents of an external RAM data
memory area with the ASCII codes of the characters
issued by the user, starting from address P1. The
command is closed when CTRL+P key is pressed.
T P1 _ P2 .
The command executes the user program from address
P1 and performs P2 instructions.
W P1 _ P2 .
Transmits a hexadecimal data bloc from the external
data memory from address P1 to address P2 towards a
computer or another development system through a
serial interface.
X .
Displays the names and the contents of the user registers
of the AT89S8252 microcontroller. The systems
console displays the following registers:
R0, R1, R2, R3, R4, R5, R6, R7 (bank 0)
R0, R1, R2, R3, R4, R5, R6, R7 (bank 1)
R0, R1, R2, R3, R4, R5, R6, R7 (bank 2)
R0, R1, R2, R3, R4, R5, R6, R7 (bank 3)
ACC, B, PSW, SP, DPTR0, DPTR1, PC
P0, P1, P2, P3
TCON, TMOD, TH0, TL0, TH1, TL1
T2CON, T2MOD, RCAP2H, RCAP2L, TH2, TL2
WMCON
IE, IP
PCON
SCON, SBUF
SPCR, SPSR, SPDR
XA .
Displays the names and the contents of the data and
control registers of the serial asynchronous interface
used for UART data transmission and reception, then
the names of the control flags and their binary values.
SCON, SBUF
SM0, SM1, SM2, REN, TB8, RB8, TI, RI, SMOD
XB .
Displays the names and the contents of the basic
registers of the microcontroller.
ACC, B, PSW, SP, DPTR0, DPTR1, PC
XC .
Displays the name and the contents of the low power
consumption mode register, then the names of the
control flags and their binary values.
PCON
PD, IDL
XE .
Displays the name and the contents of the command and
control register for access to the internal data EPROM
memory, then the names of the control flags and their
binary values.
191
WMCON
EEMEN, EEMWE, DPS, RDY/BSY
XF .
Displays the names and the contents of the flag register,
then the names of the flags and their binary values.
PSW
C, AC, F0, RS1, RS0, OV, PSW.1, P
XI .
Displays the names and the contents of the registers
used for maskable interrupt system validation and for
setting the priority level of an interrupt source, then the
names of the flags and their binary values.
IE, IP
EA, ET2, ES, ET1, EX1, ET0, EX0, PT2, PS, PT1,
PX1, PT0, PX0, IE1, IT1, IE0, IT0
XP .
Displays the names and the contents of the parallel
input-output ports on byte and bit levels:
P0, P1, P2, P3
P0.0, P0.1, P0.2, P0.3, P0.4, P0.5, P0.6, P0.7
P1.0, P1.1, P1.2, P1.3, P1.4, P1.5, P1.6, P1.7
P2.0, P2.1, P2.2, P2.3, P2.4, P2.5, P2.6, P2.7
P3.0, P3.1, P3.2, P3.3, P3.4, P3.5, P3.6, P3.7
XS .
Displays the names and the contents of the control,
status and data registers of the peripheral serial interface
SPI, then the names of the control flags and their binary
values.
SPCR, SPSR, SPDR
SPIE, SPE, DORD, MSTR, CPOL, CPMA, SPR1,
SPR0, SPIF, WCOL
XT0 .
Displays the names and the contents of the data and
control registers of counter T0, then the names of the
control flags and their binary values.
TCON, TMOD, TH0, TL0, T0
GATE0, C/T0, M10, M00, TR0, TF0
XT1 .
Displays the names and the contents of the data and
control registers of counter T1, then the names of the
control flags and their binary values.
TCON, TMOD, TH1, TL1, T1
GATE1, C/T1, M11, M01, TR1, TF1
XT2 .
Displays the names and the contents of the data and
control registers of counter T2, then the names of the
control flags and their binary values.
T2CON, T2MOD, RCAP2H, RCAP2L, TH2, TL2,
RCAP2, T2
TF2, EXF2, RCLK, TCLK, EXEN2, TR2, C/T2,
CP/RL2, T2OE, DCEN
XTW .
Displays the name and the contents of the control
register of the watchdog timer, then the names of the
control flags and their binary values.
WMCON
WDTEN, WDTRST, PS2, PS1, PS0
Y .
Determinates the software initialization of AT89S8252
microcontroller.
 IV. CONCLUSIONS REFERENCES

[1]. Cpn O., Proiectarea cu microcalculatoare integrate, Ed.

The development system equipped with AT89S8252 has Dacia, Cluj Napoca, 1992.
a minimal structure, it was built practically by the author [2]. Hintz J.K., Tabak D., Microcontrollers. Arhitecture,
and represents a useful instrument in testing and Implementation and Programming, McGaw Hill, 1993.
checking user applications, designed for [3]. Lance A. Leventhal, Programmation en langage assembleur, Ed.
Radio, Paris, 1989.
microcontrollers from ATMEL family. [4]. Peatmann B.J., Design with Microcontrollers, McGraw Hill, 1998.
Using microcontroller systems for these applications [5]. Somnea D., Vldu T., Programarea n Assembler, Ed. Tehnic,
offers the following advantages: small volume, high Bucureti, 1992.
reliability, low power consumption, minimal costs, etc. [6]. XXX ATMEL, Family Microcontroller, Data Book, 1998.
[7]. XXX INTEL, Data Book, 1988.
The monitor program written by the author for any [8]. XXX CATALYST SEMICONDUCTOR, Data Book, 1989.
microcontroller in ATMEL family offers various [9]. XXX Texas Instruments, Data Book, 1992.
possibilities and implements very useful features. The
author implemented a command set including 52
monitor program commands, the command set being
expandable with user-written commands.
The system communicates with a personal computer,
which allows editing and assembling user programs. The
real-time operation was tested and verified on the
microcontroller. The monitor program is stored in an
area of some 8 Kb of program memory.

192
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Digital Video Broadcasting Terrestrial Modulator

Implemented on Motorola MSC8101 DSP
Daniel Victora Ene1, Radu Mihnea Udrea2, Ionu Pirnog2, Cristina Sucholotiuc2
Abstract This paper presents implementation aspects
of a DVB-T modulator on MSC8101 Motorola DSP. It
gives a general description of the DVB-T system, but it
also identifies all the main processes applied to the data
stream: starting with transport multiplex adaptation
and energy dispersal, outer coding and outer
interleaving, inner coding and inner interleaving,
mapping and signal modulation, OFDM transmission.
Keywords: DVB-T modulator.
I. INTRODUCTION
Digital Video Broadcasting (DVB) is a term that is
generally used to describe digital television and data
broadcasting services that comply with the DVB
standard. The potential advantages of digital
television broadcasting over conventional analogue
broadcasting are numerous and well known. For the
broadcaster, digital technology offers significantly
improved operational flexibility, providing the means
for completely new services that go beyond the scope
of conventional television programs. Important
benefits concern the broadcasting infrastructure, with
better integration with the digital studios and playout
centers and, thanks to digital compression, more
efficient use of the broadcast spectrum. The viewer
become an active participant in the broadcasting
process, having generally improved video and audio
quality, improved program and service choice, better
navigational aids to facilitate the choice between the
various services on offer and greater control over
content delivery.
In fact, there is no single DVB standard, but rather a
collection of standards, technical recommendations
and guidelines (DVB-S and DVB-C for satellite and
cable). The established MPEG-2 standard was
adopted in DVB for the source coding of audio and
video information and for multiplexing a number of
source data streams and ancillary information into a
single data stream suitable for transmission. DVB-T
standard defines a system performing the adaptation
of the baseband TV signals from the output of the
MPEG-2 transport multiplexer to the terrestrial
channel characteristics. Actually, DVB technology
allows the broadcasting of data containers, in which
all kinds of digital data can be transmitted:
compressed image, sound or data. Comparing to other
data broadcasting systems, a key difference in DVB is
that the different data elements within the container
can carry independent timing information. This allows
audio information to be synchronized with video
information in the receiver, even if the video and
audio information does not arrive at the receiver at
exactly the same time.
The DVB standard also provides flexibility. For
example, a 38 Mbit/s data container could hold eight
standard definition television (SDTV) programs, four
enhanced definition television programs (EDTV) or
one high definition television (HDTV) program or
550 ISDN channels, all with associated multi-channel
audio and ancillary data services.
This paper presents implementation aspects of a
DVB-T modulator on MSC8101 Motorola DSP. It
gives a general description of the DVB-T system, but
it also identifies all the main processes applied to the
data stream: starting with transport multiplex
adaptation and energy dispersal, outer coding and
outer interleaving, inner coding and inner
interleaving, mapping and signal modulation, OFDM
transmission.
Tests were performed both using Matlab and
CodeWarrior C interface for MSC8101 processor in
conformity with ETSI EN 300 744 standard (Digital
Video Broadcasting framing structure, channel coding
and modulation for digital terrestrial television).
II. DVB-T MODULATOR SYSTEM
The DVB-T system addresses the terrestrial
broadcasting of MPEG-2 coded TV signals. Therefore
an appropriate adaptation of the digital coded
transport stream to the different terrestrial channel
characteristics is necessary. These requirements result
in a flexible transmission system that uses a multi-
carrier modulation, the so called Orthogonal
Frequency Division Multiplex (OFDM) technique,
combined with a powerful concatenated error
correction coding (Coded Orthogonal Frequency
Division Multiplex, COFDM).

1
Freescale Semiconductor Romania, tirbei Vod 26-28, phone 3052449, e-mail: ENED001@freescale.com
2
Politehnica University of Bucharest Faculty of Electronics and Telecommunications, Iuliu Maniu 1-3,
e-mail: mihnea@comm.pub.ro, ionut_pirnog2001@yahoo.com, cristina_sucholotiuc@yahoo.co.uk

193
 Fig. 1. Transmitter block diagram
In the modulator system (Fig. 1), the following
processes shall be applied to the data stream: transport
multiplex adaptation and randomization for energy
dispersal, outer coding (Reed-Solomon code), outer
interleaving (convolutional interleaving), inner coding
(punctured convolutional code), inner interleaving,
mapping and modulation, Orthogonal Frequency
Division Multiplexing (OFDM) transmission.
Since the system is designed for digital terrestrial
television services to operate within the existing VHF
and UHF (Very and Ultra-High Frequency) spectrum
allocation for analogue transmissions, it is required
that the system provides sufficient protection against
high levels of Co-Cannel Interference emanating from
existing PAL/SECAM/NTSC services.
It is also a requirement that the system allows the
maximum spectrum efficiency when used within the
VHF and UHF bands; this can be achieved by using
Single Frequency Networks (SFN) operation.
In fact, two modes of operation are defined in the
OFDM technique with two options in the number of
carriers: a "2K mode" and an "8K mode". The "2K
mode" is suitable for single transmitter operation and
for small SFN networks with limited transmitter
distances. The "8K mode" can be used both for single
transmitter operation and for small and large SFN
networks.
As far as bandwidth requirements are concerned, the
preferred channel spacing is 8 MHz, but if desired, 7
MHz or 6 MHz spacing is also possible by scaling
down all system parameters.
The error correction can be separated in two blocks:
the outer coding and outer interleaving that are
common to the Satellite and Cable Baseline
Specifications (DVB-S and DVB-C) and the inner
coding is common to Satellite Baseline Specifications.
The use of inner interleaving is specific to the DVB-T
system.
To accommodate different transmission rates, in
addition to five code rates, three types of non-
differential modulation schemes can be selected:
QPSK, 16-QAM and 64-QAM. The 16-QAM and 64-
QAM can also be used in combination with uniform
or non-uniform mapping rules and thus input data
streams can be separated in a low and a high priority
data stream with different error protection for
hierarchical transmission purposes. These two
bitstreams are mapped into the signal constellation by
the Mapper and Modulator. This feature allows the
simultaneous broadcasting of different programmes
with different error protection and coverage areas.
194
III. IMPLEMENTATION ASPECTS
A. Motorola MSC8101 DSP Performances
The family of the MSC8101 processor implements a
new model of instructions execution called VLES
(Variable Length Execution Set), which allows in
general the usage of more parallels addressing and
computing units, during the same cycle.
The next lines present the most important features of
the present processor as following:
- up to 10 MIPS (Million Instructions Per Second)
for every Mhz frequency;
- 4 ALU (Arithmetic Logic Unit) which include
dedicated circuits for addition, multiplication and
bit operating units;
- in every ALU there a presented MAC (Multiply
and ACcumulate) and shifter units.
Concerning the registers, the MSC8101 processor
presents next features:
- 16 registers of 40 bits length used for integers and
fractional operations;
- 16 register for addressing of 32 bits length, from
which 8 bits can be used for generating base
addresses in buffers;
- 4 offset registers for addressing and 4 registers for
circular addressing.
Some other features of this DSP are about the
presence of the orthogonal instruction set coded on 16
bits; the possibility of executing up to 6 instructions in
one cycle. Another important feature of the MSC8101
is that it has a CMOS logic which allows reduced
power consumption.
B. Transmitter blocks implementation
DSP implementation consists on processing each
MPEG-2 transport packet of 188 bytes. Outer coding
is formed by a RS encoder (204,188) followed by a 12
stages interleaver. Inner coding is a convolutional
code of rate . The output from convolutional
encoder can be optionally punctured. Accordingly, the
overall convolutional code rate is 1/2, 2/3, 3/4, 5/6 or
7/8. Interleaving is performed here both bit-wise and
symbol-wise. Former is done depending on the
symbol rate (bits/symbol), based on typical
permutation operators. The symbol-wise interleaver
maps the bit words onto the active subcarriers. Every
carrier is modulated by a modulation symbol. QPSK,
16-QAM and 64-QAM are used as modulation
methods, e.g. 2, 4 or 6 bits per modulation symbol.
The bits are assigned to the particular points in the
phase space according to the so called Gray-code
mapping. The advantage of this mapping is the fact
that closest constellation points differ only in one bit.
Each frame is formed by 68 OFDM symbols. Four
frames constitute one super-frame. Each symbol is
formed by 6817 samples in 8K mode and 1705
subcarriers in 2K mode. The symbol is formed by
applying a IFFT operator to the 8K/2K samples
resulted as the data samples, intercalated with pilot
samples and zero subcarriers. A computational
efficient radix 4 IFFT algorithm was implemented to
be used both in 2K or 8K modes in order to achieve
time requirements for a real-time processing.
The transport multiplex adaptation and
randomization block consists of a shift register of 16
cells that randomizes input data and derandomizes
received data. Randomization is done for the 187
bytes within each transport packet. The sync byte
from the beginning of each packet is not randomized.
The scrambling operation consists in a logical XOR
function between input data and a scrambled
sequence. We used for performing the data processing
two buffer of 1504 bytes length. Since the scrambling
is not applied to the sync byte, we modified the
content of the scrambled sequence for bytes having
the index 0, 188, 2*188 and so on with 0x00.
Each transport packet of 188 bytes enters a Reed-
Solomon encoder (204,188) in order to become an
error protected packet. The Reed-Solomon code
allows to correct up to 8 random erroneous bytes in a
received word of 204 bytes. An alternative RS
(255,239) may be approached, by adding 51 zeros at
the beginning of the information word, discarded after
RS coding procedure. The software implementation is
about computing of 16 parity bytes and appending
them to the end of one bloc data of 188 bytes length.
In order to perform this operation which is based on
the input bloc data of 188 bytes we used according to
the standard an a-logarithmic table and a table which
describes the coding polynomial.
The outer interleaving represents the next data
processing bloc in the entire chain of the data
computing. Due to the working way of this
interleaver, for a software implementation we
simulate its functionality basing only on the circular
addressing capabilities of the DSP.
The next module of the data processing chain is about
the inner coding. This bloc works together with a
puncturing which describes the way of how the output
data from a base inner convolutional code of rate 1/2
are passing for obtaining other coding rates. The
system allows punctured rates of 2/3, 3/4, 5/6 or 7/8
in addition to the mother code of rate 1/2.
The algorithm used for implementing the base inner
coding is based on a logical XOR performed between
the content of different delay registers. Due to the fact
that this block can work with different rate we take
195
Fig. 2. Convolutional encoder
into account for the output buffer the maximum length
which is about 1632 elements. This number of bits is
obtained from the next relation:
1
Numberbits = 204 bytes 8bits / octet (1)
rate
where rate is .
Inner interleaving is performed here bit-wise followed
by symbol interleaving. Former is done depending on
the symbol rate (bits/symbol), based on typical
permutation operators. The symbol interleaver maps
the bit words onto the active subcarriers.
First interleaving, which is performed at the bit level,
requires several interleaving schemes. We used for the
data storage up to 6 tables. Also we have to keep in
mind that, according to the standard, the input data is
split in 2, 4 or 6 pieces (as the modulation is QPSK,
16-QAM, or 64-QAM) and after that on each sub-
stream it was applied the interleaving algorithm.
Next block is also about the interleaving which is
performed at the symbol level. Regarding the
implementation in C we used for this two output
buffers each one having different length, according to
the modulator operation in 2K symbols or 8K
symbols. One symbol here is represented by data pairs
obtained at the output of the bit interleaver. The new
indexes of the symbols are obtained by using an
permutation table which contain an pseudo-aleator
sequence.
The symbol mapping represents the next module and
is about assigning to each symbol a complex number,
based on the signal constellation corresponding to the
QPSK, 16-QAM and respectively 64-QAM. Fig. 3
presents for instance the signal constellation for the
QPSK modulation scheme.
Fig. 3. Signal Constellation for QPSK Modulation
Concerning the implementation, one thing must be
underlined here. In the theory the value of c is 1, but
here this value was considerate lower than 1. This is
about the fact that the medium energy for 4 signals is:

2 2 2 2
1 + j + 1 j + 1 + j + 1 j 8
E{c c*} = = = 2 (2)
4 4

Due to the fact that the standard states to have a

medium energy of 1 and due to the limitations
concerning the number representation the value of c is
c=0.530330. In a similar way it was processed for the
others signal constellations.
Some information for the control of the transmission
as continual pilots, scattered pilots and TPS pilots
must be inserted in the data streams before the Inverse
Fast Fourier Transform (IFFT). The implementation
2K: execution time 80 700 cycles out of 79 200
of the IFFT that was performed is based on the
Motorola benchmarks algorithm. This is about of two
algorithms one of them, radix 2, used for input vector
length of N=2M and the other on, radix 4, used for
vector lengths of N=4M.
Each frame is formed by 68 OFDM symbols. Four
frames constitute one super-frame. Each symbol is
formed by 6817 samples in 8K mode and 1705
subcarriers in 2K mode. The symbol is formed by
applying a IFFT operator to the 8K/2K samples
resulted as the data samples, intercalated with pilot
samples and zero subcarriers. After IFFT, we pad a
cyclic prefix (1/4, 1/16, 1/32, 1/64).

IV. PERFORMANCES

Regarding the performance of the implemented
modulator it must be underlined first of all that there
was needed to be processed a great amount of data.
That is way the cycles burned for performing all the
operation contained in the transmission chain have big
values for both operating mode of 2K or 8K.
With all of these, it may be said that this modulator
work in real time, meaning that it was possible to
obtain a number of cycles close to the limit but lower
that this. Next figure presents these results together
with the cycle burned for each group of functions.
In fig. 3 we denoted with F1...F6 next 6 group of
functions:
F1 Iner Coding Function;
F2 1st Interleaving and Convolutional Encode
F3 Bit Interleaving Function
F4 - Symbol Interleaving Function
F5 Symbol Mapping Function
F6 Inverse Fast Fourier Transformer Function
V. CONCLUSIONS
8K: execution time 348 500 cycles out of 384 000
Fig. 3. Cycles burned for the modulator 2K mode
(right) and 8K mode (left)
The target of the cycles count were achieved on the
one hand thanks to the MSC8101 capabilities and on
the other hand thanks to the optimization technique
used in C such as: loop merging, split computation
and so on. Even if the implemented solution for the
DVB-T modulator, work in real time, due to the fact
that we are at the time limit the future development of
this work will consists in developing the big cycles
burning module in assembling language.
VI. ACKNOWLEDGMENT
Implementations of algorithms presented above were
performed using development software and DSP
boards donated by Freescale Semiconductor Romania
to our University.

The implementation of the DVB-T modulator REFERENCES

represent a complex work due to the fact that on the
one hand there are a lot of operation that must be
performed and on the other hand due to the great
amount of data that have to be processed.
Taking into account that an implementation intends to
have a code which is running in a real time, the work
becomes more difficult.
[1] ETSI EN 300 744 V1.4.1 Digital Video Broadcasting (DVB);
Framing structure, channel coding and modulation for digital
terrestrial television, Ian. 2001
[2] Motorola SC140 DSP Core Reference Manual rev 2
[3] S. L. Linfoot, A Comparison of 64-QAM and 16-QAM DVB-T
under Long Echo Delay Multipath Conditions, IEEE Transactions
on Consumer Electronics, Volume 49(4), November 2003, pp. 978-
982

196
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Telephone interface for Remote Control Systems with
network capabilities
Tiberiu Ionica1, Cornel Balint2
Abstract This paper presents the design and
development on a telephone interface based on M8880
DTMF circuits and vocal memory ISD33120, controlled
with an 80C552 microcontroller. The interface is
connected like slave in a RS-485 multipoint network that
use balanced lines. The master initializes interface, send
commands and read data and status information.
The voice generator circuit is used for two functions.
First, after recognition of telephone call, send a vocal
message indicating the valid operation. Second, for
transmission, after a initiated call, send to called station
the information like vocal messages.
Keywords: telephone interface, DTMF, voice memory,
call progress, microcontroller.
I. INTRODUCTION
Many private consortia and companies have been
writing specification for residential networks to
provide a communication infrastructure for home
systems and connections to external networks [1].
This paper extends and improves the telephone
interface presented in [2], with network capabilities
and vocal generated messages.
When using standard telephone set and Public
Telephone Network for remote control is very useful
for a user to send the information like vocal messages
instead DTMF tones.
First, for reception, after recognition of telephone
call, the interface is connected to line and wait for a
set of DTMF tones which can be used like commands.
It this situation, is not acknowledge (confirmation) for
received or executed commands. The new proposed
interface sends a vocal message for the user about the
allowed options, solicits the user to push only some
buttons and confirms reception of code (DTMF tone)
and execution of specific command.
Second, for transmission, after a initiated call, the
interface presented in [2] send only DTMF tones
which must be decoded. The proposed interface sends
vocal messages according to the produced events,
directly accessible to a human user.
The interface can be connected, like a slave
module, to a serial multipoint bus using RS 485
1
Department of Automation and Industrial Informatics, University Politehnica of Timisoara Blvd.V. Parvan 2, 300223 Timisoara,
Romania, Phone: +40256-403245, E-mail: tionica@aut.utt.ro
2
Department of Communications, University Politehnica of Timisoara, Blvd. V. Parvan 2, 300223 Timisoara, Romania, Phone:
+40256-403310, E-mail: cbalint@etc.utt.ro
197
standard. In this case, the functionality of the interface
is made by a master module, which sends commands
and receives data based on communication protocol.
The schematic diagram of the proposed home
automation system is depicted in figure 1.
RS RS 232/ RS 485
PC
RS 485
Master 232
Convertor
Phone Other
interface
(slave)
slave
Fig. 1. Telephone interface in home automation system
The core of the proposed system is a personal
computer that can run a wide range of automation
applications. A RS 232/RS 485 converter allows
connection of more slave module each of them
specialized for an automation function or interface
with automation equipment. The phone interface
system is used for two basic functions.
First, the system can recognize a telephone call
and answer with a vocal message. The caller can
access the system with a password (DTMF code) and
the system receive DTMF signals which can be used
to remote control for a range of home appliances or
send vocal messages in order to inform the caller
about system status. The system can also record a
short message.
Second, the system can initiates a call (in DTMF
or pulse dialing) to a specified number following the
call progress tones, and to send a vocal message to the
called station to indicate an event produced in system
Using the network capabilities the entire home
automation system can access these two basic
functions of the phone line interface.
II. THE PHONE LINE INTERFACE
The schematic diagram of phone line interface
with voice memory and network capabilities is
presented in figure 2.
 Multipoint bus
RS 485
Interface
RS 485
Phone line
Hybrid
(2/4 wire)
Speaker
Fig. 2. Phone line interface
A. Phone Line Interface
The structure of the interface is based of the
interface presented in [2], based on a matching
transformer with 1:1 transform ratio and 600 ohm
impedance, connected to the line via a normal open
contact of a relay. In the transformer secondary an
active hybrid circuit is used in order to separate the
received and transmitted signals. A ring signal
detector isolated throug an optocoupler (not figured in
fig. 2) was provided in order to detect the incoming
calls. For DTMF tone decoding and generating and
for call progress tones decoding a high performance
DTMF transceiver M8880 was used.
The functions of M8880 transceiver consist of
receiver with internal high gain amplifier and a
DTMF generator for all 16 standard tone pairs with
low distortion and high accuracy. In addition, call
progress mode can be selected allowing the detection
of various tones that identify the progress of a
telephone call on the network [3].
B. Voice Memory
As voice memory was used a single-chip voice
record/ playback devices (ISD 33120) [4].
The speech samples are stored directly into on-
chip nonvolatile memory without the digitalization
and compression operations usually associated with
other storage solution. Direct analog storage provides
a true and natural sounding reproduction of voice. The
device retains the message for up to 100 years without
power. The recording operations can be repeated
typically for more that 100.000 times. The ISD 33120
chip is also ideal for playback applications, where
single or multiple message playbacks are controlled
by the SPI interface.
The analog input of ISD 33120 memory is
connected to phone line through hybrid circuit or to a
microphone trough an appropriate amplifier. The
input signal is selected by an analog CMOS switch
driven by the microcontroller. The audio output signal
from ISD 33120 memory can be connected to phone
198
TxD / RxD
Microcontroller
(80C522)
DTMF Transceiver
M8880
SPI
interface
Speech memory
ISD 33120
Microphone
line or to a control speaker, by an analog CMOS
switch. (For simplicity, the analog CMOS switches
are not represented in fig. 2).
C. Microprocessor Interface
The hardware interface between M8880 and
microcontroller need for nine signals: 4 bit
bidirectional data bus and 5 control signals, including
read/write control and clock [2], [5].
Additionally signals from the microcontroller
ports are used in order to drive the relays and read the
incoming call signal status [2].
A four wire serial peripheral interface (SPI) is
provided for ISD 33120 control and addressing
functions [4]. The ISD 33120 is configured to operate
as peripheral slave device with a microcontroller-
based SPI bus interface, that provides read/write
access to all internal registers. The device has a serial
data input (MOSI Master Out Slave In) and a serial
data output (MISO Master In Slave Out), providing
data transfer between slave device and master
microcontroller, using a serial clock (SCLK) line in
order to synchronize data transfers. An interrupt
signal (/INT) and internal read-only status register are
provided for hand-shake operations. The INT signal,
activated at the end of messge, can be used for
compose more sofisticated message by concatenating
independent words.
III. NETWORK CAPABILITIES
A. Network Structure
To connect the telephone interface to a home
network in a Home Automation System need to add
network capabilities.
The telephone interface and other types of
modules are connected, like slaves, to RS-485 bus that
can be extended through the automated house. An RS-
485 to RS-232 converter provides the communication
with a standard PC that runs the Home Automation
control software. Master-slave connection is P2 enable transmission and number of
implemented according to fig. 1. retransmission
The master initializes the interface and send Enable transmission ( b7 )
DTMF codes which is transmitted in line. Also, Number of retransmission at no dial tone (0
master read DTMF codes received from telephone 7) (b4 b6)
line and decoded by M8880 circuit and status Number of retransmission at line busy (1
information indicating the progress of a call on 15) (b0 b3)
telephone network. P3 waiting times
The authors build a network structure using the waiting time at line busy (1 15sec.) (b4
Open System Interconnection of ISO Reference b7)
model (OSI-ISO model) [6] adapted for field bus waiting time at ring back tone (1 15sec.)
systems, which use only layers 1, 2 and 7. In this (b0 b3)
model, each layer implements a specified protocol and P4 number of digits for telephone call
provides services to the layer above and also uses P5 list of the digits for the number in ASCII format
services from the layer below. 2. Master read the code (DTMF tone) received by
1. Physical layer defines signal voltages and slave from the telephone line:
physical connections for sending bits across
a physical transmission media. We adopt RS CMD1 0
485 standard for multipoint systems that use
balanced lines.
2. Data link layer handles transmission of data 3. Master sends the code, which will be transmitted by
packets between station of the network, slave on the telephone line:
checking for errors, control internal data flow
and access control on channel. We CMD2 N Codes (N bytes)
implement this layer entirely in software.
3. The application layer contains the specific 4. Master read the status byte, which identifies the
functions for the functionality of each progress of an initiated telephone call:
module.
The interface can be connected, like a slave CMD3 0
module, to a serial multipoint bus using RS 485
standard. In this case, the functionality of the interface At the data link layer the software adds to previous
is made by a master module, which sends commands messages an address byte and calculates a CRC-16
and receives data based on communication protocol. sum for errors detection, building a packet will be
transmit on the bus:
B. Communication Protocol
ADR Previous Master commands CRC-16
Access to the bus is controlled by the master/slave
technique, which is the simple and therefore efficient The addressed slave (telephone interface) receives
bus protocol. the packet (at data link layer), evaluates the CRC-16
Communication is based on the principle where a field, and, if no errors are detected, sends the
master sends a request and the addressed slave returns extracted message to application layer.
an immediate response. Requests can be a read or a The application executes commands and prepares
write type. an answer for the master with the following structure:
The command send by the master at application
layer contain one command code, a number of DATA LENGTH DATA
parameters that follows and the parameters list. CMD STATUS
(0 N) (1 N)

CMD DATA LENGTH (0 N) DATA (N bytes) where:

CMD received command, returned to the master
In this application the master sends four STATUS status of the command execution. This
commands types: byte is updates by booth layers of the slave
1. Initialization of the slave (CMD_INIT) DATA LENGTH number of data bytes, which
follow
CMD_INIT 5 P1 P2 P3 P4 P5 Similar to master side, the slave data link layer
adds to message the slave address and a CRC-16 sum.
where: For each received commands the slave responds in a
P1 reception parameters specific way:
Enable reception (b7) 1.
Reception timeout (1 15 sec.) (b3 b6) CMD_INIT STATUS DATA LENGTH (= 0)
The number of call reception (1 - 7) (b0 b2)

199
 2. Send remote commands in DTMF signaling
CMD1 STATUS
DATA LENGTH CODE (e.g. light on/off)
(= 1) RECEIVED Receive a vocal messages that informs about
3. state of home system
DATA LENGTH Records vocal message or play a previous
CMD2 STATUS
(= 0) stored messages (answer machine functions)
Stop the call and disconnect from line
4. To avoid unauthorized access to remote system,
DATA LENGTH the interface can accept commands only after the user
CMD3 STATUS STATE
(= 1) provides an access code (DTMF). In same sense was
implemented and tested a function (provided by the
where STATE can take the following value : public exchange) that identifies the calling number
EV_OK telephone call successfully finished (CLIP Call Line Identity Presentation) that allows
EV_NOT_ANSWER the called station not answer system to accept commands only from a list of
EV_TONE_NOT_OK no dial tone possible calling numbers.
EV_REV_CALL_BUSY busy tone
B. Remote Data or Message Transmission
C. CRC Generation
In case of a system event, the program
A multi-byte CRC error checking protocol should automatically sends a call to one pre-programmed
be used on all data transmissions between the master numbers from a list of numbers (e.g.: owner - mobile,
and slave nodes of an RS-485 communications police, fire brigade etc.).
system. After connected to line, the interface uses call
The sending system will calculate a CRC and progress information in order to complete the call. If
append it to the message. The receiving system will dial tone is present, the program dials the number and
calculate a new CRC based on the entire message, analyzes ring-back tone. If called station answers, the
including the appended CRC bytes. If the CRC information about produced event is transmitted like
calculated is not equal to zero, then an error occurred vocal message.
in the transmission and all data should be ignored.
The CRC-16, 16 bits polynomial used is V. CONCLUSION
X16+X15+X2+1, and will detect all single/double
errors, all errors with odd bits and all burst errors The phone line interface with network capabilities
shorten than 16 bits [6]. is a necessary connection for automated home system
with external networks. The user can send orders from
IV. FUNCTIONAL DESCRIPTION remote location using DTMF codes poll the state of
system or receive information about any event
The application program is written in C language produced in home using prerecorded vocal messages.
and tested using a development system with 80C552 An other way to communicate with external
microcontroller (DSM) [2]. networks is an Internet connection that is presented by
The DSM allows the user to develop application authors like a future improvement.
programs (in C or assembly language) with full access
to the resources of microcontroller and system. REFERENCES
At application level the program is implemented in
many independent modules. [1] K. Wacks, Home Systems Standards: Achievements and
For the new introduced voice memory functions Challenges, IEEE Communications Magazine, April 2002.
[2] T. Ionic, C. Balint, Telephone Interface for Home Automation
was implemented following library functions: Systems, Buletinul Stiintific al Universitatii Politehnica din
Rec_Mic recording a message from microphone, Timisoara, Periodica Politechnica, Transactions On Automatic
to a specified address. Control And Computer Science Vol.48 (62), 2003.
Rec_Line recording a message from phone line [3] G. Niculescu, L. Ioan, Tehnici si Sisteme de Comutatie, Ed.
Matrix Rom, Bucuresti, 2000.
to a specified address. [4] http://www.winbond-usa.com/products/isd_products/
Play_Mes play a recorded message from a chipcorder/datasheets/33120/33120.pdf
specified address. [5] *** M8880 Data Sheet, www.teltone.com.
[6] A. S. Tanenbaum, Retele de calculatoare, Ed. Byblos,
Bucuresti, 2003
A. Telephone Call Analysis and Data Reception

For an incoming call on line, after receiving the

programmed number of calls signals, the interface is
coupled to line. The software sends vocal messages,
which informs the user for options allowed by system.
The options implemented are:

200
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Prepayment Gas Meter A New Trends in Natural Gas

Metering Technology
Monica Sabina Crainic1
Abstract II THE LEVELS OF PREPAYMENT
Natural gas is a non-regenerable energy source. For this METERING SYSTEM
motive she must be managed properly to protect it for
future generation. Proper management of natural gas Electronic prepayment metering system operates on
reserves requires submetering. Submetering of natural
gas consumption and revenue collection is traditionally
three level. At the lowest level, are the gas meters,
accomplished using diaphragm gas meter. To resolve which are installed at the consumers home. The next
some problem of revenue collection new technologies level is the credit sales point which are placed at the
liken prepayment metering is implemented. In this utilitys office or appointed agents. The consumers
context we at AEM Luxten Lighting Co produce who need credits (who need natural gas) come to
prepayment gas meter credit sales point.
Upon his request and need certain amount of credit
Keywords: flowmeters, diaphragm gas meter, natural that is
gas submetering, prepayment device Upon his request and need certain amount of
credit that is equal to certain amount of natural gas is
I INTRODUCTION
loaded in his card and the billing is made by the
officer against the cash money. In this respect, the
Revenue collection is one of the core activities
processes like meter reading, billing after
of any utility inclusive natural gas distribution
consumption and difficulties in collecting natural gas
company. This has traditionally been accomplished
costs will be removed. The smart card has a security
using conventional credit meters like diaphragm gas
passworth inside and each smart card has its own
meter, with regular meter reading, extension of
passworth which means each card is different from
credit to customers and normal credit collection
each other. When consumer comes to the credit sales
mechanisms.
point with his smart card to purchase credit, the card
This process is costly, with numerous inherent
is checked in PC. In this check, it is controlled which
problems for both utility and customers. To solve
municipality, consumer and meter the card belongs
some of these problems new technologies like
to. According, to the data obtained from the card and
prepayment metering is implemented which offer
user, the card is accepted or refused and the billing is
benefits to both parties.
made with reference to the name of the consumer. If
Prepayment metering in its simplest form
the data obtained from the check of the card doesnt
refers to the payment of utilities prior to the use of
comply with that obtained from the consumer, the
the utility until such time as the credit has expired.
card will not be loaded. At the top level is the credit
The concept of prepayment metering is not a
selling points control center (central PC) which is
novel concept having first been introduced in the
necessary to ensure a common data base for reporting
form of coin gas meters in the United Kingdom [1-2].
as well as to provide for total management,
This concept was refined in the 1980s through the
administration, financial and engineering control.
use of electronic or numeric transfer of the credit and
other information.

1
AEM Luxten Lighting Company SA, Gas and Water Meter Research Department, 26 Calea Buziasului 300693
Timisoara, Romania, Tel: 4-0256-222200, Fax 4-0256-490928, e-mail sales@aem.ro

201
 III GAS METER WITH SMART CARD
The gas meter with smart card comprises
mechanical gas meter with pulse output, valve group,
electronic hardware, battery group electronic
software and smart card.
3.1 MECHANICAL GAS METER WITH PULSE
OUTPUT
These are volumetric dry, diaphragms gas
meters [3-4] meant for measuring domestic natural
gases consumption. They comply with OIML R6,
R31 and to SR 6681-98 provisions. Some technical
characteristics are presented in Table 1
Their cases are cupped steel bodies with
electrostatic spray paint with epoxipolyesteric
powder. The rotation of gear is transferred via a
magnetic coupling.
The gas meter with pulse output converts the
data obtained from mechanical meter into an
electrical signal by means of a reed switch group that
is activated by a permanent magnet. It makes a
sensitive reading and gives 2 pulses per one liter.
3.2 VALVE GROUP
consuming energy. So that it saves the energy in the
battery. The valve is able to evaluate the
opening/closing data sent from the electronic
hardware. Other performance of the pressure pulse
valve are presented in table 2.
3.3 ELECTRONIC HARDWARE
This unit controls the mechanical meter and
the valve and establishes the communication of the
system with the gas administration or distribution
company by means of smart card. The electronic
hardware is a very unique one and all components are
gathered in one PBC. The socket switch locks on
them are used in the circuit to ensure the rigidity of
the system and facilitating the assembly process. The
electronic components used are chosen with a great
care and of the best quality.
3.4 DISPLAY
Thanks to the Alphanumeric LCD used, the
characters can be read easily. Without any difficulty
and hindering the eye look. When the display is
active, its power consumption is so lowly by the help
of display driver.

The valve used in system is a gas valve 3.5 BATTERY GROUP

actuator bistable action [6] which can be customized
to meet various flow control applications. Battery group supply energy to the
electronic hardware and to the valve. The life of the
Table 1 battery C Lithium size is 10 years. Battery which as a
voltage of 3,6 V is kept in their own battery
G 1,6 G 2,5 G 4 protections. The battery connected with the locked
Cyclic volume V (dm3) 1,2 1,2 1,2 sockets belonging to PBC provide a speedy and an
Maximum flow Qmax (m3/h) 2,5 4 6 easy replacement with the new ones. The battery is
Minimum flow Qmin (m3/h) 0,016 0,025 0,040 located in the meter apart from the meter and the
Maximum pressure Pm (bar) 0,5 valve group such that it is possible for the gas
Environmental and gas - 20 +50 C administration officers to replace the battery without
temperature range dismantling the seal of gas distribution company. The
One pulse value (on request) 0,002 m3 charge of the battery is checked by the electronic
Counter range 99999,999 hardware continuously. I(f it is detected that voltage
Connections G1 or G1 of battery is lower than the working voltage or the
Maximum admissible errors 3% for battery is out of service because of any reason, the
QminQ<Qmax electronic hardware record all the data and shuts
1,5% for down the valve.
0,1QminQQmax
Weight (kg) 2,5 3.6 ELECTRONIC SOFTWARE
Overall dimensions (mm) 243 228 172
Thanks to the special software developed,
The operating voltage of the valve is 1,9 V minimum the continous and secure data communication
ambient temperature of 20 C (relay drive circuit). between the meter and the valve group and the
The power consumption of the valve is of the lowest synchronization of the smart card with the central PC
ones to ensures the longest life of the battery. The has been established.
valve is actuated within 20 ms either in operating and The datas that carried by smart card are as
closing. Once the valve is closed or opened it keeps follows:
its position until it is activated again without the name of the natural gas administration,

202
 Table 2 [6] the type of consumer,
the number of consumer,
PERFORMANCE
Shrouded coil Intrinsically safe potted the number of meter,
construction and dual the information about the credits,
zener circuit the information about the spare credits, that will
Coil winding resistance be determined by the administration,
ELECTRICAL

Operating pulse (flat top)*
Diaphragm /spindle stroke
Overtravel (shut-off)**
MECHANICAL
7,0 5% at 20 C the charge condition of the battery,
(with actuator seat
mounted uppermost)
the last data at which the credits has been loaded
to the meter,
100 ms minimum
duration or capacitor the information about the valve malfunctioning,
discharge pulse 2,5 V- the information about the meter malfunctioning
peak and
Energy typically 60m the information about the consumption.
Joule The datas and messages that maintain the
7,0 mm nominal. Shut system to be controled are: the alarm messages about
off against flow valve and meter malfunctioning, electronic hardware
pressure and valve battery charge information and a message
1,5 mm maximum/ 500 about meter out of credit.

gF minimum to seal The main control unit checks if there is

gas at 30 mBar warns the consumer by displaying the message on its
Flow rate** maximum pressure display. The electronic controls the amount of natural
Leakage 0 to 6 m3/h gas that passes through the meter to and decreases the
2,0 l/h maximum number of credits as a function of the consumption.
Minimum life** (when shut-off) In case the meter is out of credit and the valve is
20 years/ closed, the consumer inserts his card into the meter to
Weight 7500 operations open the valve and run the spare credit. The
300 g consumer can also observe on meters display the
Ambient temperature - 25 C to + 55 C necessary information that he is consuming the spare
ENVIRONMENTAL

range credit. In a certain period of time, if a certain amount

Gas humidity 3 % RH from 20 C of consumption is exceeded by the consumer that
to + 10 C belong to the first tariff, the meter switches to the
60 % RH from + 10 C second tariff automatically. When this period of time
to + 50 C passes, the meter switches it self to the first tariff
again. The information about the tariff and level of
consumption can be loaded in the card with three
different levels. In case of inquiry from the
Diaphragm/Seat Moulded in gas
consumer, the meter can make itself out of service for
approved nitrile
certain period of time (in case of holidays or absence
Seat to suit orifice inlet
of the family for a long period of time). If no gas
port of meter; 24 42
flow is detected by the meter in 10 days for example,
CONSTRUCTION

Spindle connection
Switch
Materials
* Typical value. Other combinations are possible to
suit various and applications
** Typical values. Other flow rates and working
pressure possible up to 10 psi
mm diameter range
the valve closes itself automatically for the safety
Spindle to diaphragm
reasons. The valve can be opened and meter can be
spring; ball joint or
put in service again by using a card. This card is used
clipped
by administration officer. The duration 10 days can
Indication for end of
be adjusted up to request.
stroke closure; no
Checking the level of battery is made by the
volts reed switch
electronic hardware continuously. In case the battery
circuit
level is lower than the level it should be, all the datas
Compatible with
are stored in the EEPROM memory. By means of the
natural and
smart card the information about the battery levels
manufactured gas
are carried to the Credit Sales Point letting the
administration be informed too. In case of
malfunctioning of the valve a message is displayed.
The meters goes on metering the consumption and
recording the consumption higher than the amount of

203
credit as the consumers debt. In the case of the meter
box is opened by the unauthorized people or
interfered deliberately, the electronic control unit
terminates the gas flow by shutting off the valve and
records the date in smart card. After the maintenance
is performed by the administration, the system is
update by the authorization card
The time and the data adjustment of the
meter is made by means of a card. The real time
clock placed in the electronic hardware can be
programmed to detect the working hours, working
days the month and even the number of days in
February adding one day to the year every 4 years.
If the consumer uses the spare credit and if
the meter goes out of credit not in the working hours,
valve will not close itself until the new working days
starts letting the gas being consumed by lending
credit. When the consumer goes to Credit Sales Point
to purchase credit, the landed credits will be
decreased from the number of new credits loaded.
3.7 SMART CARD
The cards used in control system conforms
with the ISO7816-2 and they have secure memory. If
a wrong card is inserted into the meter, the main
control unit identifies that card and warns the
consumer displaying a message. The names of the
cards used in the system are:
The consumer card by which the data transfer
data displaying and credit purchasing processes
can be executed.
The authorization card which executes all the
operations that the consumer card does with the
exception of the credit loading. In necessary, one
can change all the datas and reload them with
this card.
Redundant Credit Card by which in case the
consumer inquires his redundant credits he
makes back loading by using this card taking
finished credit loaded in the meter. After having
finished this operation, he delivers this card to
the Credit sales Point and can take his money
back.
Switching off Card which is used by gas
administration and executes the meter switching
off process by shutting off the valve.
Reset Card which is used to reset the datas of
the microprocessor in the electronic control unit.
Time Card which adjusts the data and the time in
the electronic control unit.
IV CONCLUSIONS
Pre-payment metering is not merely a choice
of a different brand of meters. Pre-payment metering
replaces not only the classical gas meter but also the
204
billing system, the reading of meters and the
administration of the revenue collection. Pre-payment
benefits both utilities and consumers. Utilities benefit
because payment is received on average 45 days
early than with a conventional billing system. This is
not, however the only advantage. Pre-payment
metering offers improved customer service, no meter
readers required, eliminate of bad debts,
disconnection and reconnection fees, ensure a hand
control and eliminate inaccurate meter reading.
The typical user of pre-payment gas
metering system is a member of lower income groups
in the population [7-8]. He will appreciate the fact
that he now has direct control over his budget and
often his acceptance of pre-payment is much higher,
because there is a direct link between the money, he
spends and the value he gets. Also pre-payment
metering system require no cost for
disconnection/reconnection and no waiting
reconnection and offer ability to payback debts. To
implement a pre-payment metering system, means a
change of mind set, a change in the way to revenue
collection is managed, a change in IT procedures, a
change in customer service, a change in metering and
a change consumer behavior. Because pre-payment
gas meter is much more expensive than a
conventional meter to be able to reap the benefits as
expressed above, all parties need to be into the
system and appreciate the benefits they themselves
will receive.
REFERENCES
[1] *** Improvements in or relating to coin-feed meters GB
Patent no 191505193 from 30 March 1916
[2] *** Improvements in prepayment gas meters GB Patent no
191403216 from 8 July 1915
[3] *** Omega Transactions: Technical Reference Series vol. 4
Flow & Level Measurement 2001
[4] Dane Enrich A guide to metering technologies ASHRAE
Journal October 2001 p.33
[5] *** Catalogue AEM Luxten Lighting Company SA
[6] *** Pressure pulse valve actuator BLP Components Ltd
Catalogue http://www.blpcomp.com
[7] *** Fuel Pouerty: Low Income, Prepayment Meters and
Social Obligations Center for Management under Regulation
University of Warwick and Center for Competition and Regulation
University of East England March 2001
[8] Roger D Colton Prepayment meters and the Low-Income
Utility Consumer Fischer, Sheehan & Colton Public Finance and
General Economics October 1998
 Buletinul tiinific al Universitii Politehnica din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Currents and voltages measurements using aquisition

board DAQ 6024 E
Simona MORARU1, Daniela FAUR2, Brandusa PANTELIMON2,
Catalin VOINA2, Andreea COSAC1

Abstract - The program who is the subject of this paper the block diagram (actually the source code this one
develops many possibilities in order to facilitate reading contains the corresponding instructions, constants,
and analysing electrical quantities (voltage, current) for functions and pointers from front panel). Flowing
analogy and digital signals. Using acquisition boards for data is determined in block diagram using links
analogical or digital data from various tranducers,
signals can be analysed or conditioning and
represented by lines between icons [3].
measurements instruments can be created or simulated The hardware-software system used contains the
(virtual instrumentation). following components and programmes:
Keywords: acquisition board, rezistive inductive load, - Toshiba Laptop
one phase rectifier - Windows XP
- LabVIEW 6i, Measurement & Automation
I. INTRODUCTION DAQ Card - 6024E (National Instruments) for
PCMCIA adapters.
In order to increase the works productivity, data
acquisition and data analyse system are used in II. THEORETICAL CONSIDERATIONS
automations systems. One of them is LabVIEW
System [1-5]. In order to do the aquisitions we will use the
In the following lines we will present some of its next theoretical considerations. The resistive voltage
possibilities and performances. We will especially divider is made with coiled resistors and it could
describe data acquisition using DAQ 6024 E 5 6
reach a high level precision ( 10 ...10 ) or it is
acquisition board.
realized with metal resistors and it has a low precision
LabVIEW represents a graphical alternative to 2 3
the conventional programming design for ( 10 ...10 ), but good enough for analogical and
instrumentation It is equipped with all necessary tools digital instrumentation. The divider is used for D.C.
for testing the measurement systems. LabVIEW is a or low frequency voltage measurement.
graphical developed environment designed in order to
create flexible and scalable test, to measure and to
control more rapidly the applications, at a minimal
price. The fastness of this program is high, due to the
introduction of an intuitive graphical interface.
LAbVIEW uses a generally graphical language
for programming called G, containing wide
libraries with proper functions. The LabVIEW
programs are called virtual instruments and are made
from two parts, distributed in two windows:
-the front panel (containing the necessary elements Fig.1 Voltage divider
for interactive operations and the display of the
results) The input, known measure is the D.C. voltage
1
Electrical Engineering Research Institute, Bucharest, Romania,
U 1 and the output measure is the D.C. voltage U 2 .
simona492273@yahoo.com If the divider works no load, it will result an output
2
University Polytechnic Bucharest, Bucharest, Romania, voltage:
dfaur@electro.masuri.pub.ro

205
 R2
U 2 = R2 I = U 1 (1)
R1 + R2

and the divided factor is:

U2 R2 1
D= = = (2)
U 1 R1 + R2 R
1+ 1
R2 Fig.3 Electrical diagram for one phase rectifier
The shunt is an input current-voltage and the output voltage for a resistive load
convertor. It is used for currents measurements in
D.C. circuits. The used D.C. shunt is made from Sampling Theorem (Shannon, 1949): any
manganin and it is included into devices in case the signal in continuous time, with a limited spectrum,
currents are less than 20-30A or it is external, as a can be represented without loosing information
separate piece, for a 1000A current. through a sample series of the original signal, or in
other words, through a discrete signal.
The data aquisition systems principal
components are sampled circuits, the memorized ones
and the analogical-numerical convertors. The
numerical and analogical signals caused by
processing can be used to memorize and give back
the information or to command the execution
elements (motors, relay), which control the physical
processes.
To operate with discrete amplitude signals
Fig. 2 The shunt
means a special attention. The result might be often a
sum of quantification noises, with a statistical
The shunt resistance RS is defined between the characterization factors and consequences.
voltage terminals. We can write the following
equations:
IV. DAP APPLICATION
I = I S + I T
(3) In DAP application (the programme is named
RI S = RI R Data_Aquisition_Programme) we will read,
memorize and compute analogical and digital signals,
The shunt establishes the next factor between the particular currents and voltages, in order to analyse
the behaviour of certain system in stationery or
output measure I R and the input one I :
permanent mode.
The programme allow to operator to record data
I R +R R simultaneously, on maximum 16 analogical channels
n= = S =1+ . (4)
IR RS RS (ACH) and 8 digital channels (DIO). The aquisition
board DAQ Card-6024E doesnt admit data reading
synchronization if the aquisition is made
We obtain the computed relation for the shunt
simultaneously for analogical signals in ACH socket,
resistance:
respectively digital signals in DIO socket. Because of
this reason we use both signals type, analogical and
R digital, in ACH socket. We are interesting in data
RS = . (5)
n 1 reading synchronization for at least one digital
channel.
The rectifier has a converting function for the We can use also the special DIO sockets when
electrical energy form A.C. into d.c. His working is the object isnt the perfect synchronization between
depending on the load type, connected at its output. different channels. In this case, a late of
This dependence is shown in a very simple diagram, approximatively 1 second might appear between
see fig.3. The diode should be an ideal one ( u D = 0 signals.
The aquisition board DAQ 6024E can operate
for working state; i D = 0 for blocking state). with a maximum analogical scan rate of 200000
scans/second, meaning a maximum scan rate for each

206
channel of 200000/16 = 12500 scans/second.
Considering that a scan rate of 1000 readings/second
is equal with a millisecond data reading, we can
affirm that the technological possibilities of this board
are properly [4-5].
DAQ 6024E board allows signal aquisition
between 10Vd.c. limits. Its obviously that we need
an intermediate electronic board to adapt the real
acquired signals to the specified interval (10Vd.c.),
with suitable scan factors. The board admits the
independently scanning for each channel.
The two applications windows are described in
fig. 4 and fig. 5. In DAP programme we used many
specific functions:
- each channel has his own configuration;
- START/STOP for the acquisition, controlled
by the user or from one digital channel command
(this allows to display data with a seted number of
seconds before/after 0/1 passing on that very
channel) ;
- many possibilities for changing the parameters
(scaling factors for each channel, delay factors on the
OY axis for each channel, zoom on OX axis, the
memory size used by the programme, the scan rate,
the channels number for reading and displaying, the
cursor for reading the exact acquisitioned values) ;
- the mean for diagrams (we use the arithmetical
mean, with a setted number of points);
- the tangent is computed with the help of two
selected points coordinates, for the data to be
analysed in every particular mode.
zoo m
Fig. 4. The Front Panel
207
Fig. 5. Bloc Diagram
Data are displayed in two different ways:
- in real time (one second constantly updated);
- historically (displaying the entire interval
ordered by the user).
Data reading is made as long as the programme
goes on. One digital channel can order START and/or
STOP recording, having the possibility to extend and
set a gap in seconds or milliseconds before START
and after STOP, equal or different periods.
The advantage appear when we want to record a
transitory phenomenon about whom we dont know
exactly the moment it will be happen. Data reading is
permanently, but data recording has a controlled
start/stop, given by the operator or presetted.
This is the way to avoid a useless loading of
memory or even an overcharge of hardware system.
The diagrams allow to simultaneously displaying
all channels for reading or only a few of them after
selection.
The utility of this programme is the possibility to
use it for tracing and visualization of electrical
quantities, any deviation from the normal behaviour
is unliked and it must be eliminated without any
delay. (Example: hydroelectric power
stations, power stations the entire national circuit of
electrical and thermical energy).
The possibility of change the principal
parameters, which interfere in the acquisition and in
the recording, and also the filtering of the data are
very important programme performances. We can
print all acquired diagrams.
In Fig. 6a we choose to show an 35 seconds
recording, with a 1000 readings/second scan rate, for
the Excitation System supply voltage (SRAT), in case
of changing it with the Backup Supply (AAR).
Through a 10 data meaning we will obtain the Fig. 6b
diagram.
 52 IV. CONCLUSIONS
50

48
The numerical computing techniques are
limitated from the maximum frequency for analogical
Amplitudine (V)

46
input signals and also from numerical computed
44 speed point of view.
42 In an application these limitations are
depending on the data acquisition system
40
characteristics, on the work speed of the numerical
38 computing systems and on the numerical computing
36 algorithms complexity.There is applications in which
SELECTIE 0 5000 10000 15000 20000 25000 30000 35000
CANAL Timp (citiri/sec) a real time data computing is demanded, meaning that
49 the computing algorithms are correlated with the data
48 access speed. Because of the time axis discretization
47
the analogical signals become discrete. The signal
becomes discrete if we also divide the OY axis. One
Amplitudine (V)

46
condition for the signal to be a good approximation
45
for the analogical one is that the sampling frequency
44
must be big enough reported to the maximum
43
frequency from the sampled signal spectrum
42 (Shannon Theorem). The most insignificant bit signal
41 level must be small enough (the scales must be small
40
15000 20000 30000
on OX and also on OY).We are interested in these
0 5000 10000 25000 35000
MEDIERE Timp (citiri/sec) requirements because the final purpose of this
research paper is to simulate an industrial process, in
Fig. 6 a) Original Data; b) Mean Data. the aim to know it better, to control and to predict it.
The useful signal, representing the physical
Another example of aquisition is the voltage phenomenon or systems behaviour, is mixed with
wave for the current and the voltage obtained at the perturbations, at aquisition and through the
output of a one phase rectifier with an inductive- transmition channel. The discretization introduces a
resistive (RL) load. noise too. The perturbations and noises are
continuous time phenomena, like the useful signals.
Between them is a subjective difference, the
specialists point of view. Because of the high
mathematical level, it is hard to analyse and to
separate them.
The virtual instrumentation utilization
advantages are in the decreasing expenses with new
instruments (the system acquisitioning price, the
expenses with the development and the maintenance)
and increasing performances (flexibility,
reutilization, and reconfiguration).Low prices and
Fig.9 The dropping voltage in case of resistive-inductive load high performances are the desired qualities customers
for one phase rectifier.
expect from their delivers.

REFERENCES

[1] Maier, V., Maier, C. LabVIEW n Calitatea

Energiei Electrice, Ed. Albastr, Cluj Napoca,
2002
[2] Cottet, F., Ciobanu, O. - Bazele Programrii
n LabVIEW, Ed. MatrixRom, Bucharest, 1998
[3] LabVIEW User Manual National
Instruments, January, 1998
Fig. 10 The dropping voltage in case of resistive load and [4] Stanomir, D. Semnale i Sisteme
a D.C. voltage supply for one phase rectifier. Analogice, Ed. Politehnica, Bucharest, 2002
[5] Ghinea, M. - Procesarea Digital a
Semnalelor, Ed. Tritonic, Bucharest, 1997

208
 Buletinul tiinific al Universitii Politehnica din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Considerations regarding the factors involved in
calibration of DC voltage standard calibrator
Livia Dragomir1, Ion Sandu1, Brndua Pantelimon2
Abstract - The estimation of measurement uncertainty is
a compulsive task of each metrological laboratory which
has implemented the quality system. This paper describes
the factors which can influence the process of calibration
of a DC voltage standard calibrator, the method for
calculating the true value of the measured and the
estimation of the measurement uncertainty.
Keywords: Standard calibrator, calibration, true value,
measurement uncertainty.
I. INTRODUCTION
The implementation of a quality system represents one
of the main requirement which must be satisfied if the
recognition in conformity with SR EN ISO 17025 [1]
of metrological laboratory is desired. The recognition
of the quality system by an accreditation organization
demonstrates that the laboratory meets the standards of
quality and technical competency imposed by SR EN
ISO 17025. The accreditation is based upon this
agreed standard with specified measurement
parameters and uncertainties. The estimation of the
measurement uncertainties and the assurance of the
traceability to SI of all the standards used are
compulsive requirements for the laboratory mentioned
above.
The article refers to calibration [2] of calibrators, on
the functions of generating DC voltage within the
(0 1000) V range.
Digital multimeter and meter calibrators are multiple
use instruments found in most calibration and
standards laboratories.
Digital multimeter and meter calibrators provide the
functions needed to calibrate DMMs and other meters.
As DMMs have grown in functionality, metrology
laboratories found that an ensemble of single function
calibrators was required to calibrate the DMMs. The
newest models are multifunction calibrators that
provide all or nearly all of the functions needed to
calibrate most DMMs.
A typical multifunction calibrator provides one or
more quantities: direct and alternative current and
voltage, electrical resistance stimulus.
1
National Institute of Metrology, Bucharest, Romania,
dragomir@inm.ro and sandu@inm.ro
2
University Politechnic Bucharest, Bucharest, Romania,
bpante@electro.masuri.pub.ro
209
Voltage calibrators, direct current and resistance
calibrators are apparatus providing voltages, direct
current and electrical resistances of scalable or direct
variable known and fixed values, at the output
terminals.
Single function calibrators can be used to complement
the multifunction calibrator to provide additional
functions, such as capacitance, inductance, and
frequency stimulus as well as thermocouple and
resistance temperature detection emulation.
Calibrators are designed for calibration and
verification of apparatus measuring voltage and
current (analogical and digital voltmeters and
ampermeters) or electrical resistance (generally, digital
multimeters).
From the point of view of displaying the generated
value, calibrators may:
- display the generated value on decades and levels
- display the generated value on a digital display
II. CALIBRATION
A. Technical operating conditions
Reference conditions:
- environment temperature (23 2) OC, unless
technical specification provides otherwise
- relative humidity (50 15) %, unless technical
specification provides otherwise
If reference conditions are not complied with,
calibration is stopped.
The generation error of calibrators in reference
conditions should not exceed the error provided in the
technical specification of the apparatus. For some
calibrators, technical book provides the expanded
uncertainties; in such case uncertainties determined are
compared with the ones in the specification.
The value generated by the calibrator, when all
decades indicate zero, or when display indicates zero,
should not exceed the value provided by the technical
specification.
B. Calibration method C. Determining the value of the measurand

Methods used for the calibration [5] are: The true values of the calibrators are determined in
- substitution the reference conditions.
- direct comparison In the absence of other indications, the zero
- indirect comparison adjustment is made on the smallest interval (if
The selection of method is made according to the possible).
accuracy of the calibrator to be calibrated and the Calibrator will be calibrated to the best accuracy,
measured quantity. corresponding to long-term stability.
For the calibration of calibrators on the function to We use standards ensuring:
generate electrical direct voltage, we can use two - stable values of generated quantities
methods: - required resolution
- substitution, for calibrators with high accuracy - measurement intervals to cover
- direct comparison, for calibrator with lower generation intervals of the calibrators to
accuracy. be calibrated
For calibration of high-accuracy calibrators on the - required accuracy/uncertainty
function to generate direct voltage, the multiple- Determining the value of the measurand and data
function calibrator, is used as standard, and the digital processing, for the calibration of calibrators on the
multimeter with 8 digits or 7 digits is used as functions of generating direct voltage using the
measuring mean. substitution method.
The assembly method is shown in Fig. 1. The calibrator generates direct voltages for each
measuring point; they can be read on the digital
multimeter display.
We conduct n readings for each point, in conditions of
+ + repeatability. We calculate the average X X .
Standard Calibrator to For the same measuring points, standard calibrator
calibrator be measured
_ _ generated voltages measured with the same
multimeter. We conduct n readings for each point, in
repeatability conditions. We calculate the average X E .
We determine the true value [2] according to formula:
K
X = XN + ( X X - X OX +XTX ) ( X E - X OE -XE - XDE
+ XTE) + XR+ XX (1)
Digital
multimeter + where:
XN the nominal value displayed on the
_ calibrator to be calibrated
XX average value for n direct voltages
values generated by the calibrator
Fig. 1 Assembly in case of substitution and read on a digital multimeter
with resolution appropriate for
Most often, the calibration process may include three measuring
approaches of the calibration, depending on the X OX average value for n zero direct
customers requests and level of information:
voltages generated by the calibrator
a) calibration before adjustment, in case we
and read on the same digital
notice errors more significant than the
multimeter having the same
measurement errors provided in the technical
resolution
specification of the apparatus to be calibrated
-adjustment (manual or using a software) XTX correction of calibrator due to
-calibration after adjustment environment temperature
b) calibration XE average value for n direct voltages
c) adjustment, when the Customer knows that generated by the standard calibrator
the apparatus must be adjusted and informs the read on the same digital multimeter,
laboratory thereof with resolution appropriate to
- calibration measuring
If errors are more significant than the ones admitted X OE average value for n zero direct
by the technical book, the calibrator will be adjusted, voltages generated by standard
if possible, and the errors determined prior and after calibrator and read on the same
adjustment will be specified in the calibration digital multimeter having the same
certificate. resolution

210
 XE the correction of the value of the the value in the calibration certificate of the standard
standard (the values in the used.
calibration certificate)
Uncertainty of element XE.
XDE correction due to time drift of the
standard The value of the measurement uncertainty U(XE)
XTE correction of standard calibrator provided in the calibration certificate of the standard
due to environment temperature calibrator,.
XR correction due to variations of the e) Correction due to drift in time of standard
supply network XDE is taken into account depending on the data
XX correction due to instability of the provided by the history of the standard used.
value displayed by the digital
Uncertainty of element XDE.
multimeter
In formula (1), the first bracket represents the basis Based on the history of the standard, a time drift a is
for the calculation of the value indicated by the evaluated. The associated uncertainty is calculated as:
calibrator to be calibrated.
Each element in formula (1) has a certain value and a
related uncertainty [3, 4]. a
XDE) = (6)
a) average value X X , for direct voltage 3
generated by the calibrator, for n readings made on
the digital multimeter, in conditions of repeatability, f) Correction of standard calibrator, due to
is calculated according to the formula: environment temperature XTE, is applied only when
measurements are conducted at a temperature
n differing from the reference temperature and the
X Xi technical book provides temperature correction
XX = i =1 (2) coefficients cTE, for standard calibrator
n

a = cTE (T-T0 ) = XTE (7)

Uncertainty of element XX

is calculated: Uncertainty of element XTE.

Applicable only in case the measurement temperature
n
differs from the reference temperature. In such case,
(Xi =1
Xi X X )2 we measure temperature T to which we have noticed a
u( X X ) = (3) variation a.
n(n 1)

b) values and uncertainties for terms X OX , a

u(XTE) = (8)
X OE are calculated with formulas similar to formulas 3
(2) and (3).
g) Correction of calibrator, due to
c) average value X E for n readings environment temperature XTX , is applied only when
conducted with the same digital multimeter, when measurements are conducted at a temperature
voltage values (delivered by standard calibrator) differing from the reference temperature and the
values are calculated by formula technical book provides temperature correction
coefficients cTX. The value of a is calculated with a
n formula similar to formula (6).
X Ei
Uncertainty of element XTX.
XE = i =1 (4)
n
Applicable only in case the measurement temperature
differs from the reference temperature. In such case,
Uncertainty of element XE
we measure temperature T to which we have noticed a
is calculated: variation a.

n
a
(X Ei X E )2
u(XTX) = (9)
u( X E ) = i =1
n(n 1)
(5) 3

h) Correction due to variations of the supply

d) The correction of the value of the standard network XR. No correction is applicable, thus
XE - the difference between the nominal value and

211
 XR= 0. Table 1

Quan Estimate Standard Probability Sen- Uncer-

Uncertainty of element XR . -tity uncer- distribution sibi- tainty
tainty lity contri-
We estimate the modification of the multimeter coe- bution
indication ( a) when there are instabilities in the xi u(xi) fici-
supply network. The uncertainty is estimated: Xi ent ui(y)
ci

a XN 1V 0 0
u(XR) = (10)
6 0.9999740 V 0.086 V Normal 1 0.086 V
XX

i) Correction due to instability displayed by

X OX -0.0086 mV 0.086 V Normal 1 0.086 V
the multimeter to be calibrated XX. Generally, no
correction is applicable, thus XX= 0.
Uncertainty of element XX . XTX 0 0 Rectangular 1 0

We estimate the modification of the multimeter XE -0.9999923V 0.011V Normal 1 0.011V

indication ( a) due to its own instability (depends on
the apparatus resolution). The uncertainty is 0.0045 mV 0.011V Normal 1 0.011V
X OE
estimated:

a XE -0.0000008V 0.6V Normal 1 0.6V

u(XX) = (11)
3 XDE 0 2.8V Rectangular 1 2.8V

The composed standard uncertainty is calculated: XTE 0 0 Rectangular 1 0

XR 0 0.24V Triangular 1 0.24V

uc(X) = u2 (X i ) (12)
X 0.999 976 8 V 2.9 V

The expanded standard uncertainty (for k = 2) III. CONCLUSIONS

associated to the value of measurand is:
To calculate the true value of the measurand and to
U = k uc(X) (13) estimate the measurement uncertainty represent two
essential requirements for traceability assurance.
The standard uncertainty: The influence of each of the quantities mentioned in
this paper is different from a measurement method to
uc(X) = u ( X ) = 2.9V
2
i
another.
Thus, for each method of measurement the uncertainty
has to be calculate in a particular way.
The expanded uncertainty [3, 4] was calculated using According with SR EN ISO 17025:2001 authorized or
a coverage factor k = 2, which correspond to a accredited metrological laboratories have their own
confidence level of 95%: calibration procedures which include how to estimate
the uncertainty for the measurement method used.
U = k uC = 2 2.9 =6 V.
REFERENCES
The reported results are: X = 0.999 977 V,
U = 6 V [1] SR EN ISO 17025:2001, Cerine generale pentru competena
laboratoarelor de ncercri i etalonri
[2] SR EN ISO 13251:1996, Vocabular internaional de termeni
An example for estimation of standard uncertainty for fundamentali i generali n metrologie
multifunction calibrators for 1 V, is shown in table 1 [3] EA 4/02, Expression of the uncertainty of measurement in
calibration
[4] SR 13434/1999, Ghid pentru evaluarea i exprimarea
incertitudinii de msurare
[5] Iliescu, C., Golovanov, C., Szabo, W., Szekely, I., Barbulescu, D.,
Msurri electrice i electronice, Editura Didactic i Pedagogic,
Bucureti, 1983

212
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Polarization Method for Electrical Current Measurement
Viorica SIMION1, Brandua PANTELIMON1, Costin STEFANESCU2

Abstract This paper presents a polarization method bring about his anisotropy, meaning that the refractive
for electrical current measurement using optical fiber. index is depending on the polarization plane of light.
Classical polarization method has the advantage of being The effects used are: electrical optical effect (Kerr,
simple but for getting a good performance the optical Pockels) or magneto-optical effect (Faraday).
components have to be carefully assembled due to their
sensitivity to disturbance (temperature, vibration). In II. THE FARADAY EFFECT
the field of power systems the most common sensors are
those used for the measurement of electrical quantities The current measuring is based on Faraday effect in a
and to oversee the electrical equipment under voltage. single-mode fiber, which is wrapped around a
Many times these sensors are located in hardly
conductor in a closed loop [2]. The light waves
accessible places (vacuum spaces, with oil, gas, etc.) or
connected at a higher potential than earth potential. polarization direction is rotated by the magnetically
fields is parallel to the light waves path (fig. 1).
Keywords: Polarization Method, Current Measurement,
Optical Fiber, Optical Fiber Sensor, Faraday Effect The Faraday effect is not reciprocal, meaning it
doesnt vary with the light propagation distance, while
I. INTRODUCTION optical activity is a reciprocal phenomenon. This
solution led to decrease sensitivity to temperature
Regularly, for the measurement of electrical currents variations. It must be avoided to establish a fixed
are used devices and sensors, which allow galvanic point in the fiber and the twist sense changing must be
separation from the conductor where the current to done carefully.
measure passes, like current transformers or sensors From a macroscopic point of view, the Faraday effect
based on Hall or Faraday effects. Other important appears like a rotation of the polarization plane of a
requirement is the immunity to the electromagnetic polarized light. This light is propagating in optical
perturbations encountered in electrical power stations. fiber that is the magnetic field. This rotation of the
The research of new sensors called unconventional polarization plane, due to Faraday effect is
it was quickly oriented to optical measurement proportional to the magnetic field circulation ds along
methods and especially for using Faraday effect). This the luminous trajectory (path) according to the
method was first investigated using a glass bar, expression:
developed later due to the apparition of optical fibers.
Optical fibers sensors can perform all these F = V H (s )ds (1)
requirements because galvanic separation and L
electromagnetic interferences are inherent properties where:
of optical fibers [1]. S curvilinear abscissa along the luminous
In the field of power systems the most common trajectory;
sensors are those used for the measurement of L - total length of luminous trajectory;
electrical quantities and to oversee the electrical V Verdet constant (a property of propagation
equipment under voltage. Many times these sensors
medium).
are located in hardly accessible places (vacuum
spaces, with oil, gas, etc.) or connected at a higher
potential than earth potential
The measurement of electrical quantities (current,
voltage) using optical fibers is based on the
polarization modulation. This phenomenon supposes
that the quantity to measure produce modifications of
the birefringence at the optical fiber level. The
birefringence of the active medium can be realized to
_____________________________________________________________
1
Faculty of Electrical Engineering, Bucharest Politehnica University, 313, Splaiul Independentei Street,
phone. +4021-4029258, email: vsimion@electro.masuri.pub.ro
2
Faculty of Computer Science, Bucharest Politehnica University, 313, Splaiul Independentei Street, phone: +4021-4029351,
Email: cstefan@cs.pub.ro
213
 IV. THE ANSWER OF OPTICAL FIBER
CURRENT SENSOR

The Jones vector is used for characterization the

answer of optical fiber sensor and we write this vector
like a matrix product for each optical component:
J = P R F J 0 (2)
Fig. 1 Measure principle where:
J 0 the Jones matrix for linear polarized light;
III. THE ELETRO-OPTICAL SYSTEM
R F - the matrix of optical fiber core;
In this paper is presented one method through whom P - the matrix of polarizer.
the magnetic field has been created namely: When the light is linear polarized the Jones vector J 0
- realization of a coil around the optical fiber; is:
The used optical fiber is a silica monomode fiber
E cos
with low birefringence LoBi. The fiber is the most J 0 = x 0 = I 0 (3)
important part of the sensor. The use of a multi-mode E y0 sin

fiber is not recommended, as a linear state of The Jones matrix of optical fiber core was written
polarization entering a fiber will be quickly destroyed, when the linear birefringence is 0.
therefore a single-mode fiber will be used instead.
Even in a single mode fiber there is a loss of cos F - sin F
R F = (4)
polarization of the light caused by anisotropy effects sin F cos F
(linear intrinsic and extrinsic refractions). The matrix of polarizer is:
This type of fiber allows the transmission of a linear
1 0
polarized light beam maintain the polarization plane P = J 0 = P0 J 0 (5)
position. The apparition of a magnetic field produces 0 0
a rotation of the polarization plane (the essence of the So, the Jones vector was completed and the luminous
measurement method). The electro-optical source is a intensity is:
laser diode having the wavelength corresponding to I = I 0 cos 2 [( ) + F ] (6)
the maximum intensity p = 662 nm. Its properties
The answer of optical fiber current sensor has a
are: the threshold current intensity I th = 22 mA, the maximum sensibility when we analyze the light at
intensity for the optimal current I op = 23 mA, the = / 4 . For this function point, the luminous
intensity is:
optical power output P0 = 1 mW. The Faraday effect
I
being dispersive along the wave length we have to use I = 0 (1 sin 2 F ) (7)
2
a monochrome source which has a good coupling
The sensibility of the both signals (alternating and
efficiency with the single-mode fiber. The wavelength
direct signals) is:
is chosen as short as possible because Verdet constant
I I
is proportional to 1 / 12 . The coupling between the S = 1 2 = sin 2 F (8)
laser diode and the fiber optic is a sensitive part of the I1 + I 2
system. The electronic power supply for the laser
diode realized an adjustment for the current intensity V. EVALUATION OF LOSSES IN SYSTEM
throw the laser diode. It has a block for a slowly start
to eliminate the eventually over growths that can We have identified the parameters that produce the
appear at connection (this over growths can destroy system losses. These are: reflection of light, fiber
the diode). optic intrinsic attenuation and the brush-up of both
The ORIEL analyzer is used for analyzing the light ends of the fiber [6]. The transmittances are:
beam outlet from the sensor. It is positioned between 2
the optical fiber connector and photodiode. It has a n n2
Trefl = 1 1 (9)
rotator calibrate with 0,10 0 resolution and the properly n1 + n2
analyzer. where: n1 and n 2 are the refractive indices in the
Also, it has been used: one ammeter for alternating core and cladding.
current ( I n = 5 A , c = 0,5 ), one inductor coil, one The attenuation transmittance is:
l A
rheostat ( 30 / 5 A ) that allow a fine adjustment of 4
current intensity and the autotransformer that can Tat = 10 10 (10)
assure an adjusting voltage for the serial circuit to The total transmittance for the system without
induce the measuring current. polarizer is:

214
 Ttot = Trefl Tat T fin Tld fo (11)
The total transmittance when using a light analyzer in
system is:
TP tot = T pol Ttotala (12)

In this case the losses are: the emission and the

reception reflection and the attenuation losses.

VI. EXPERIMENTAL DETERMINATIONS

The two realized systems allowed measuring a

current intensity between 0 5 A (small interval,
easy to be done). That signifies that the method
sensibility is very good, especially because this sensor
is used generally, for hundreds or thousand amperes
domains.
Before the beginning of the measures the polarizer
characteristic had been traced and using the smallest
squares method we can identify his function:
U pol ( ) = 0,9074 + 19,012 cos ( 9,96 ) [mV] (13) coefficient for P = 99,73% , n = 6 ); the related error
2
Fig. 2 The analyzer response
Depending on the measured value U it has been
measured:
- the medium value of the measurement for each value
of the current; the experimental squared deviation
(rectified) ; the trust interval (the absolute error of
fidelity), where t=1 (t is the Student repartition

corresponding to the studied domains; the regression

After that, the minimal and the maximal point of this
function, based on the theoretic model (we used the
characteristic was identified. The minimal point is a
smallest squares method) :
conventional 0, and its position depending to the
connector position. If the last one is moving, the U (I ) = U 0 + S I [mV] (16)
minimal point is changed. After these two points were
determined it was identified the half of the where: U 0 = 3.432 [mV]
transmission point (fig. 2). This corresponds to a 450 - the medium value of the angle corresponding to the
angle, after cos2 law. Any rotation of the polarization medium voltage and to the angle determined on the
plane caused by an electric current produce a linear theoretic value; the absolute true error; the related true
variation of transmitted luminous intensity error; - the total error.
The rotation of the polarization plane depending on We observed the values for the total errors. The total
the electric current intensity to measure is given by: error obtained when we measured the direct current is
(t ) = V Lb H = V N i (t ) (14) 2.89% and for the alternating current is 1.89% . This
values are almost the related true errors.
where:
The solution to increase the useful signal and in fact
V the Verdet constant ( 24 10 5 degrees/A ); meaning the sensibility are:
Lb the length of the optical path in the coil ( 0,2 m ); - increasing the current intensity through the laser
N the turn numbers of the coil (1200 ); diode;
According the second equation we have obtained - the increasing the photo detector sensibility;
the relation between the output voltage of the - the extension of the measurement domain for the
measurement system U out and the value of the current intensity as well as the magnetic field
intensity;
rotation angle of the polarization plane.
u (t ) = 0,9074 + 19,012 cos 2 ( (t ) 9,96 ) [mV] (15)
Using the optical fiber current sensor it have
measured some the values of the direct and alternating
currents. The experimental determination had been
repeated six times to estimate the trust interval
(repeatability).

Fig. 3 The transfer function for the direct current

215
 will increase, meaning that the rapport signal vs. noise
will be better.
We could use a Wollaston prism instead the analyzer,
allowing the compensation of certain influence
quantities, but this solution is expensive.
These types of optical current sensor have the small
dimensions and it offered the immunity of the
electromagnetic perturbations and the galvanic
isolation.

REFERENCES
[1] Royer, P., Capteur de courant et traitements associes,
SEE Limoges, 14-15 oct 1995;
Fig. 4 The transfer function for the alternating current
[1] Dakin, J., Culshaw, B., Optical Fiber Sensors: Principles
and Components, Artech House, Boston, London, vol. I,
1988;
[2] Chai, Y.,Handbook of Fiber Optics. Theory and
Applications Academic Press, Inc., San Diego, California,
1980;
[3] www.oxford-electronics.com;
[4] Stanciu, M., Senzori cu fibre optice, Ed. Secorex, Bucureti,
2001;
[5] Ionescu, A., Traductoare pentru automatizri industriale,
Ed. Tehnic, Vol. I, Bucureti, 1985;
[6] Simion, V., Pantelimon, B., tefnescu, C., Aspecte privind
pierderile din sistemele de msurare cu fibre optice,
Proceedings of International Metrology Conference, (ISBN:
973-99385-5-8), vol.3, pag. 855-858, Bucharest, Romania,
September 18-20, 2001;

Fig. 4 The error characteristic for the measurement

direct current

Fig. 4 The error characteristic for the measurement

alternanting current

VII. CONCLUSION
The current sensor described above works employing
the Faraday effect. The major difficulty in the
construction of such a sensor consists of using a
classical detection configuration.
The used solution attests the stability and the good
linearity of the method and allows the projection for a
very large scale of current by a good choosing of the
inductor. Our solution confers a good linearity and
that will permit the design for a large current domain.
When the currents have large values, the sensibility

216
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

General considerations regarding the measurement and

the harmonization of the national system of standards in
the field of the public domain of measuring electric
energy, according to the requirements of the European
and international standardization organizations
Elvira Buzac1, Ionel Urdea Marcus2, Costin Cepic3

Abstract: The paper presents the efforts made
in order to accurately measure electrical energy,
n activity belonging to the public domain.
The electric energy meters are instruments submitted
to the mandatory metrological control of the state,
in order to assure the consumer protection in this
field. Some matters of concern for the state regarding
the transposing of European and international norms
in view of the accession of our country to the European
Union are also presented.
Key words: static meter
commercial transactions, work safety and
environment protection, (), are subjected to the
mandatory metrological control of the state.
The manufacturers of measuring instruments in our
country such as AEM and Luxten Lighting from
Timisoara, Electromagnetica from Bucharest, are
interested in providing quality products, that meet the
requirements of international standards, and especially
of the European ones, in order to comply with the
rigours of the European Union.

II. WAYS TO PROVIDE CONSUMMER

I. INTRODUCTION PROTECTION IN RELATIONSHIP WITH THE
MEASUREMENT OF ELECTRICAL ENERGY
The importance of the accurate measurement of
electrical energy used by low energy consumers The issuing in the latest years of a legislation
(household consumers and small companies) regarding the quality of services and the consumer
increased in the latest years as a result of a new protection led to steps towards reaching these
legislation, aiming at providing consumer protection objectives undertaken by the state bodies that are
in this field. responsible with the enforcement of these laws.
The Ordinance of the Government (OG) No. 20/1992 In order to protect the consumer, the metrological
regarding metrology, approved and amended by Law control of the state is exerted upon the instruments in
No. 11/1994, with other subsequent amendments, several ways, as for example [1]:
stipulates in Chapter 1, Art. 3, as follows: in order to
protect natural and legal persons against the harmful a) pattern approvals;
effects of incorrect or fake measurements in the public b) calibrations;
domain activities, such as trade relations and c) initial metrological verifications;
d) periodic metrological verifications;

1
Romanian Bureau of Legal Metrology, National Institute of Metrology, Electrical Measurements Laboratory, sos. Vitan Birzesti
No. 11, Bucharest, Romania, e-mail: buzac@inm.ro
2
Romanian Bureau of Legal Metrology, National Institute of Metrology, Interdisciplinary Metrology, sos. Vitan Birzesti No. 11,
Bucharest, Romania, e-mail: urdea@inm.ro
3
Politechnica University Bucharest, Faculty of Electrotechnics, Measurement Apparatuses and Converters Department,
sp. Independentei No. 331, Bucharest, Romania, e-mail: costin@electro.masuri.pub.ro

217
 e) metrological verifications after the repair or IEC 13251:1996:
modification of an instrument; International vocabulary of basic and general
f) metrological surveillance of measuring terms in metrology
instruments.
and specifies the metrological and technical
According to the provisions of OG No. 20/1992, the conditions for the following types of control: pattern
Romanian Bureau of Legal Metrology (BRML), a approval, initial verification, verification after repair
specialised body of the state central public and periodic verification. The norm applies to
administration, responsible with the co-ordination of induction and static single phase and three-phase
the metrology activities in Romania, identifies the active electric energy meters of accuracy classes 0.2;
measuring instruments used in the public domain and 0.5; 1 and 2, used in AC networks.
nominates them in the Official List of measuring
instruments submitted to the mandatory metrological III. MODERN INSTRUMENTS FOR THE
control of the state. MEASUREMENT OF THE ELECTRICAL
This list is published, and periodically updated, in the ENERGY
Official Journal of Romania.
Among the measuring instruments submitted to the There is a variety of instruments that may be used to
mandatory metrological control of the state, there are: measure electric energy. Beside the classical
watt-hour meters for active electric energy, whose
- measuring rules and tapes;
operation is based on the electromagnetic induction
- meters for cold water up to DN 800 and hot water
principle, a wide variety of electronic meters has been
up to DN 400;
developed in the latest years at a rapidly increasing
- electronic converters for gas volume;
rate, along with the development of microelectronics,
- programmable clocks for watt-hour meters;
based on the following measurement principles:
- load cells;
- single phase active/reactive electric energy
- the double amplitude and duration modulation
meters;
principle;
- gas analysers;
- the Hall multiplier principle;
- medical monitoring equipment for patients, etc.
- the thermoelectric multiplier principle.
Using its specialised departments, BRML establishes
The rapid modernization of the electric energy meters
the appropriate metrological control mechanisms,
is also a result of the manufacturers efforts to meet
applicable to each type of measuring instrument, as
the current requirements of the consumers regarding
well as the maximum permissible interval between
the technical, structural and metrological
two subsequent metrological verifications [1].
characteristics of these measuring instruments used in
The measurement of electric energy belongs to the
the public domain.
public domain and, therefore, electric energy meters
Due to the fact that these measuring instruments are
are instruments submitted to the mandatory
involved in the commercial transactions between the
metrological control of the state.
suppliers and the consumers, they are submitted to
The metrological assessment of these instruments is
the mandatory control of the state and, therefore, they
carried out according to the Legal Metrology Norm
are part of the regulated area, with specific aspects
(NML) No. 5-02-97, currently in force.
related to the consumer protection that are to be dealt
This norm has been prepared in compliance with the
with.
international regulations applicable to this field,
Based on its role and competence and accumulated
namely:
experience, at the National Institute of Metrology,
within the AC Measurements workgroup of its
IEC 60687:1992:
Electrical Measurements Laboratory, a large number
Alternating current static watt-hour meters for
of electric energy meters for low energy consumers
active energy (classes 0.2 S and 0.5 S)
were assessed, within their type testing for the pattern
approval certificates granted by the Pattern Approval
IEC 60387:1992:
Department (SAM) of BRML, as well as within their
Symbols for alternating current electricity meters
initial verification, their subsequent periodic
verification and their verification after repair.
IEC 60521:1988:
The active electric energy meters for household and
Class 0.5, 1 and 2 alternating-current watt-hour
industrial consumers are tested in metrology
meters
laboratories where all the tests required by the
specialized norms in force are carried out in order to
IEC 61036:1996:
assess the meters.
Alternating current static watt-hour meters for
A new issue that has to be currently dealt with is the
active energy (classes 1 and 2)
quality of the electric energy.

218
Tests such as:
- impulse voltage tests for circuits and between the
circuits;
- tests for electromagnetic compatibility (EMC);
- tests of immunity of electrostatic discharges;
- tests of immunity to electromagnetic HF fields;
- fast transient burst test;
- radio interference measurement;
stipulated in the European norms are now also
stipulated in the Romanian standards, and are meant
to lead to a correct assessment of the behaviour of the
electric energy measuring instruments.
The importance of accurate measurement of electric
energy used by small energy consumers tends to
increase.
It is now possible to collect the data regarding the
energy used by each customer directly into the
memory of a central computer, instead of sending
each month an employee of the electricity distributing
company to read the meter installed at each individual
consumer.
The development of the static energy meters should
also be mentioned, as these enable the consumer to
pay for the electric energy using a card (similar to the
card used to pay for the subway fare).
The measurement accuracy, the low own energy
consumption, the possibility to programme hourly
tariffs, the possibility to be connected to a computer
via a built-in interface and to store information, the
reduced volume and weight, as well as other
advantages provided by the electronic meters, makes
it possible to successfully use these instruments to
measure electric energy.
IV. PRESENT CONCERNS REGARDING THE
DEVELOPMENT AND HARMONIZATION OF
THE NATIONAL SISTEM OF STANDARDS IN
COMPLIANCE WITH THE REQUIREMENTS OF
THE EUROPEAN AND INTERNATIONAL
STANDARDIZATION ORGANIZATIONS
Within the efforts of our country to accede to the
European Union, the state institutions are organising
their activities accordingly, in order to reach this goal.
In this context, adopting European standards that may
interest various branches of the Romanian industry
represents a permanent preoccupation the Romanian
Standardization Association (ASRO). Thus, in 2003,
ASRO and INM collaborated within the CALIST
2003 Programme (quality and standardization) in
carrying out the project "Adopting European
Standards as Romanian Standards for various
branches of Electrotechnics."
The technical committee TC 164 of ASRO, supported
by INM, whose main topic is "Equipment for
measuring electric energy and load control," is
involved in translating and adopting European norms
as Romanian standards. Thus, in the framework of the
219
CALIST 3 Programme, nine European norms were
translated and adopted as Romanian standards within
TC 164, such as:
IEC 61358:1996:
Acceptance inspection for direct connected
alternating current static watt-hour meters for
active energy (classes 1 and 2)
IEC 62053-31:2001:
Electricity metering equipment (a.c.)
Particular requirements
Part 3: Pulse output devices for electromechanical
and electronic meters (two wires only).
IEC 62053-61:2001:
Electricity metering equipment (a.c.)
Particular requirements,
Part 31:Power consumption and voltage
requirements
IEC 62056-2:2001:
Electricity metering
Particular requirements
Data exchange for meter reading, tariff and load
control
Part 21: Direct local data exchange
Also to be mentioned is the fact that the international
norms on which NML 5-02-97 was based, as
mentioned above, subsequently became European
norms that have already been adopted as Romanian
standards as well.
V. CONCLUSIONS
Romania is very much interested in producing high
quality goods and services that comply both with
international and with European standards. Providing
consumer protection in all the fields of the public
domain besides being a constant topic for mass media
proved to be a beneficial activity not only for the
consumers, but also for the producers of goods and
services.
BRML is constantly concerned with organizing
metrology in Romania on modern principles, as well
as with the maintenance and development of the
national system of measurement standards, as a basis
for uniform and correct measurements. When
approving investment projects, BRML aims at
purchasing highly accurate measuring equipment, that
would support the quality systems in compliance with
the ISO/IEC 17025: 1999 implemented in all its
subordinate laboratories.
The ISO/IEC 17025: 1999 standard, "General
requirements for the competence of testing and
calibration laboratories," which, was adopted in 2000,
without any changes, as an European standard was
translated and adopted in 2001 as a Romanian
standard.
All these actions, that were carried out and are still REFERENCES
going on at the institutional level of metrology, as [1] *** Ordinance of the Government No. 20/1992 regarding
well as the continuous efforts of the manufacturers to Metrology.
improve the quality of their products intended for use
[2] *** SR 13251/1996, International vocabulary of Basic and
in the public domain, together with the programmes General Metrological Terms.
initiated by the Romanian government, in order to
harmonise the Romanian laws and regulations with
the EU legislation, are all aiming at facilitating the
accession to the European Union of our country.

220
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Measuring current-voltage (i-v) characteristics

for non-linear electrical devices
Valentin Dogaru Ulieru1, Costin Cepisca2, Horia Andrei3, Adela Husu4,
Cristina Dogaru Ulieru5, Traian Ivanovici6
Abstract Measurement and automation technology was
changed once the National Instruments company introduced
the concept of virtual instruments, using the LabVIEW
(Laboratory Virtual Instrument Engineering Workbench)
graphical programming. In order to fully describe the
behavior of nonlinear electrical devices such as
incandescent lamps, photovoltaic cells, ecc., the
manufacturer will usually provide a set of one or more
current-voltage characteristic curves. In spite of this
widespread availability of electrical devices data from the
manufacturer, regarding electrical devices we may need to
measure the current-voltage characteristics for a particular
device in the laboratory.
Keywords: virtual instrument, data acquisition, signal
conditioning, curve fitting
I. INTRODUCTION
Data acquisition uses a combination of PC-based
measurement hardware and software to provide a
flexible, user-defined measurement system. If the
system changes, you often can reuse the virtual
instrument components without purchasing additional
hardware or software. The fundamental task of all
measurement systems is the measurement and/or
generation of real-world physical signals.
Measurement devices help you acquire, analyze, and
present the measurements you take. Virtual
instruments represent a visualization of complex
measurement systems on a standard personal
computer in the form of a virtual user interface.
II.DATA ACQUISITION
The term PC-based data acquisition refers to the
acquisition of data both by means of PC-based plug-in
cards, designed as classical multifunction cards, and
external box systems (SCXI, VXI-DAQ, parallel
port). A multifunction card is a PC plug-in card
handling analog input and output capabilities, digital
input/output ports, counters and timers. PC-based
instruments combine the advantages of integration
into industrial standard PC hosts and bus systems
(PCI, VXI, PCMCIA, etc.), offering the flexibility of
stand-alone devices, for example, multimeters,
oscilloscopes, functional generators, analyzers.
The analog input specifications give you information
on both the capabilities and the accuracy of the DAQ
product:
number of channels the number of analog
channel inputs is specified for both single-ended
and differential inputs for devices with both input
types.
sampling rate this parameter determines how
often conversions can take place. A faster
sampling rate acquires more data in a given time
and can therefore often form a better
representation of the original signal. Data can be
sampled simultaneously with multiple converters,
or it can be multiplexed, where the analog-to-
digital converter (ADC) samples one channel,
switches to the next channel, samples it, switches
to the next channel, and so on. Multiplexing is a
common technique for measuring several signals
with a single ADC.
resolution the number of bits that the ADC uses
to represent the analog signal is the resolution.

1
Assoc. Prof.dr.eng. Faculty of Electrical Engineering, University Valahia Targoviste, Romania18-24, Bdul. Unirii, Targoviste,
Dambovita, Romania, Phone/fax: +40-245-217 683; email: dogaru@valahia.ro
2
Prof.dr.eng. Faculty of Electrical Engineering, University Politehnica Bucharest, Romania, 313, Splaiul Independentei, 77 206,
Phone/fax: +40-21-410 04 00/ +40-1-410 43 55; email: costin@electro.masuri.pub.ro
3
Prof.dr.eng. Faculty of Electrical Engineering, University Valahia Targoviste, Romania, 18-24, Bdul. Unirii, Targoviste, Dambovita,
Romania, Phone/fax: +40-245-217 683; email: handrei@valahia.ro
4
As..drd.eng. Faculty of Electrical Engineering, University Valahia Targoviste, Romania18-24, Bdul. Unirii, Targoviste, Dambovita,
Romania, Phone/fax: +40-245-217 683; email:ahusu@valahia.ro
5
stud. Faculty of Electrical Power Engineering, University Politehnica Bucharest, Romania, 313, Splaiul Independentei
6
Eng. Faculty of Electrical Engineering, University Valahia Targoviste, Romania18-24, Bdul. Unirii, Targoviste, Dambovita, Romania

221
 range range refers to the minimum and
maximum voltage levels that the ADC can
quantize
Before you begin developing measurement
applications, you must install and configure the
measurement hardware. The software drivers need the
hardware configuration information to program the
hardware properly. When measuring a physical
phenomena, a transducer must convert this
phenomena into a measurable electrical signal.
Common types of signal conditioning include
amplification, linearization, transducer excitation and
isolation. Some signal conditioning can be performed
in the software in the Data Acquisition function
palette.
Fig. 1. Data acquisition subpalette
Analog input acquisitions use grounded and floating
signal sources.
Fig. 2. Signal source reference configuration
Grounded signal sources have voltage signals that are
referenced to a system ground, such as earth or a
building ground. Grounded signal sources share a
common ground with the DAQ board. Floating signal
sources contain a signal that is not connected to an
absolute reference. Some common examples of
floating signals are batteries, thermocouples,
transformers, PV cells.
A measurement system can be placed in one of three
categories: differential, referenced single-ended,
nonreferenced single-ended. In a differential
measurement system, you do not need to connect
either input to a fixed reference. DAQ devices with
instrumentation amplifiers can be configured as
differential measurement systems. An ideal
differential measurement system, reads only the
potential difference between its two terminals inputs.
A referenced single-ended measurement system
measures a signal with respect to building ground.
DAQ devices often use a nonreferenced single-ended
measurement system, wich is a variation of the
referenced single-ended measurement system. In these
case, all measuremnts are made with respect to a
common reference, because all of the input signals are
already grounded (AISENSE is the common reference
for taking measurements and all signals in the system
share this common reference. AIGND is the system
ground).
LabVIEW for Windows installs a configuration utility
for establishing all board and channel configuration
parameters. This utility is known as the Measurement
& Automation Explorer MAX. After installing a
DAQ board in computer, MAX utility reads the
information the Device Manager records in the
Windows registry and assigns a logical device number
to each DAQ board. The configuration of channels for
this application is presented in figure 1.
Fig. 3. Channels configuration
III. EXPERIMENTAL RESULTS
The semiconductor diode is a device that will conduct
current in one direction only. A bias refers to the
application of an external voltage to a semiconductor.
There are two ways a P-N junction can be biased:
A forward bias results in current flow through the
diode (diode conducts). To forward bias a diode, a
positive voltage is applied to the anode lead
222
 (which connects to P-Type material) and the where q is the electronic charge and is equal to
negative voltage is applied to the cathode lead 1.602 x 10-19 coulombs, k is the Boltzmann
(which connects to N-Type material). constant with a value of 1.381 x 10-23 J/K and T is
A reverse bias results in no current flow through the temperature in Kelvin.
the diode (diode blocks). A diode is reverse biased The zener diode uses a p-n junction in reverse bias to
when the anode lead is made negative and the make use of the zener effect, which is a breakdown
cathode lead is made positive. phenomenon which holds the voltage close to a
The P-N Junction region has three important constant value called the zener voltage.
characteristics:
1) The junction is region itself has no charge carriers
and is known as a depletion region.
2) The junction (depletion) region has a physical
thickness that varies with the applied voltage. A
forward bias decreases the thickness of the depletion
region; a reverse bias increases the thickness of the
depletion region.
3) There is a voltage, or potential hill, associated with
the junction. Approximately 0.3 of a volt is required
to forward bias a germanium diode; 0.5 to 0.7 of a Fig. 5. Zener diode
volt is required to forward bias a silicon diode.
The social strong involvement of the energy systems
Semiconductor diodes are made by joining two
and the complex boundaries between these system
different types of semiconductor materials in a special
and all other technical systems, or the environment,
way so that when a proper polarity voltage is applied,
have imposed the development of some researches on
electrons readily pass through one material to the
unconventional process of producing electric energy.
other. However, if the voltage is reversed, there is
Nowadays there are many unconventional methods
very minimal electron flow.
for obtaining electric energy, based on much or less
In other words, a semiconductor diode allows current
studied physical or chemical phenomena. Photovoltaic
to pass through when in forward bias, and blocks
systems convert sunlight energy into electric energy
current when in reverse bias. They also have
and they are characterized by modularity, functional
properties or characteristics that enable them to
autonomy and long function period.
perform many different electronic functions.
To ensure the accuracy of the measurement, the
operating parameters of the photovoltaic system and
the configuration of the acquisition system are taken
into account and have imposed the signal conditioning
and the setting of the signal source, of the field and of
the channels.

Fig. 4. Semiconductor diode - IV characteristic

The general characteristics of a semiconductor diode

can be defined by the ideal diode equation for forward
and reverse bias regions which is given as Fig. 6. PV module current-voltage characteristics
qkVTD
I D = I 0 e 1
As we have to determine the operation characteristics
(1) of the photovoltaic panels and the panel arrays, the
application allows to measure the values of current
and voltage, to simultaneously trace characteristics

223
(current-voltage, power-voltage, power-charge photovoltaic panels. After shutting off the experiment,
resistance), to present the measured parameters the investigators apply appropriate curve fitting
(during the data acquisition) in tables, continuous techniques to determine the functional relationship
acquisition, to save data into files for future I=f(U) manifest in their data. Curve fitting represent a
processing. technique for extracting a set of curve parameters or
Building integrated photovoltaics, the integration of coefficients from the data set to obtain a functional
photovoltaic cells into one or more of the exterior description of the data set. LabVIEW provides built-in
surfaces of the building envelope, represents a small VIs that perform a least-squares fit of data to
but growing photovoltaic application. In order for commonly used equations including a strainght line,
building owners, designers, and architects to make an exponential curve and a mth order polynomial. As
informed economic decisions regarding the use of an illustrative example of how these curve-fitting VIs
building integrated photovoltaics, accurate predictive function, see the image shown in fig.8.
tools and performance data are needed. At Valahia
University of Targoviste, enclosed in the ICOP
DEMO 4080-90 European research program, a
photovoltaic system has been realized, with an
installed power of 10 kWp, composed by 66
OPTISOL SFM 72 Bx photovoltaic modules made by
Pilkington Solar International and 24 ST 40 modules
produced by Siemens. These modules are connected
to Sunny Boy invertors.

Fig. 8. Curve - fitting

IV. CONCLUSIONS

In this paper, a LabVIEW graphical program is

designed to measure the current and voltage in dc
circuit, for ploting characteristic curves for non-linear
devices. LabVIEW is the most powerful measurement
and control language available to execute the control
algorithms and to present the results is a user-friendly
format.

REFERENCES
[1] R. Bishop, Learning with LabVIEW 6i Pretince Hall PTR,
Upper Saddle River, New Jersey, 07458, 2001
[2] J. Essick, Advanced LabVIEW labs, Pretince Hall PTR,
Upper Saddle River, New Jersey, 07458, 1999
[3] R. Jamal, LabVIEW applications and solutions, Pretince Hall
PTR, Upper Saddle River, New Jersey, 07458, 1999
[4] V. Maier, LabVIEW in calitatea energiei electrice, Ed,
Albasta, Cluj-Napoca, 2001
[5] *** National Instruments - Data Acquisition Basics Manual
[6]*** National Instruments Measurements Manual

Fig. 7. PV array (5 modules) characteristics

current - voltage
power - resistance charge
power - voltage

The Sunny Boy Control device and the Sunny Data

Control data acquisition soft-ware are used to monitor
the operating parameters of the unit. By using the
acquisition system and the software, there can be
determined the operation characteristics of

224
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Accurate Sinewaves Implemented With a 16-bit Fixed-

Point Digital Signal Processor
Daniel Belega 1

Abstract In this paper the performances of two
methods in the implementation of a sinewave with a
16-bit fixed-point digital signal processor (DSP)
TMS320C5x are evaluated and compared. The methods
used are the recurrence formula method and the look-
up-table (LUT) method. By each method the sinewaves
are generated by simulation and with the TMS320C5x
Starter Kit (DSK) board. The performances of the
methods are appreciated by means of the dynamic
parameters of the sinewaves implemented: signal-to-
noise and distortion ratio (SINAD) and total harmonic
distortion (THD). The accuracy of the frequency of the
sinewaves implemented is also determinate.
Keywords: implementation of the sinewave, digital
signal processor, accurate estimate of the sinewave
dynamic parameters.
Moreover, the accuracy of the frequency of the
sinewave implemented is determinate.
II. THE METHODS USED FOR IMPLEMENTING
THE SINEWAVES
A discrete-time sinewave can be implemented by the
following two methods:
A. The recurrence formula method
We consider a continual-time sinewave x(t)
characterized by amplitude A and frequency fin. After
the sampling with frequency fs (fs > 2 fin) the discrete-
time sinewave x[n] is obtained. x[n] is given by

f
x[n] = A sin 2 in n , n = 0, 1, 2, K (1)
I. INTRODUCTION f s
The sinewaves are the easiest to generate in practice at
Based upon (1), after some simple algebra, the
the frequencies with adequate accuracy. From these
following recurrence relationship is achieved
reasons the sinewaves are used in many applications,
such as communication, instrumentation and control.
x[n] = 2ax[n 1] x[n 2] n = 2, 3, 4, K (2)
The sinewaves can be implemented with analog or
digital circuitry. Due to its advantages concerning the
accuracy, speed and the easiest in establishing the f
parameters the microprocessors and the digital signal where: a = cos 2 in ;
fs
processors (DSP) are mostly employed for
implementing the digital sinewave generators [1]-[3]. x[0] = 0 (from (1));
The objective of this paper is the determination and f
x[1] = A sin 2 in (from (1)).
the comparison of the performances of two methods fs
in the implementation of a sinewave with a 16-bit
Thus, from (2) it follows that the discrete-time
fixed-point DSP-TMS320C5x. The methods are ones
sinewave x[n] can be implemented by means of a
of the most used in the implementation of a digital
recurrence formula.
sinewave: the recurrence formula method [1], [4] and
the look-up-table (LUT) method [3].
B. The look-up-table (LUT) method
Using each method the sinewaves are generated by
If N samples of the x[n] (n = 0, 1, 2, , N-1) are
simulation and by means of the TMS320C5x Starter
acquired, then the relationship between the
Kit (DSK) board. The accuracy of the sinewave
frequencies fin and fs is given by
implemented was determinate by its dynamic
parameters: signal-to-noise and distortion ratio
(SINAD) and total harmonic distortion (THD). f in J +
= (3)
fs N

1
Dept. of Measurements and Optical Electronics,
Faculty of Electronics and Telecommunications,
e-mail: daniel.belega@etc.utt.ro

225
 generated with 15.625 kHz sampling frequency. The
where J is the number of sinewave cycles (J is an sampling frequency of the acquisition process was
integer) and 0 < 1. equal to 15.625 kHz. Were acquired N = 1024
For = 0 the sampling process is coherent with the samples.
sinewave and (3) represents the coherent sampling
relationship between the frequencies fin and fs. In this
case, based upon (1), x[n] is periodically with the
period N, i.e. x[n] = x[n + N]. So, x[n] is completely
represented by its first N samples (n = 0, 1, 2,, N-1).
The LUT method consists in the following two steps:
step 1: The first N samples of x[n] (n = 0, 1, 2,,
N -1) are stored in a memory (table).
step 2: The sinewave is generated by stepping the
table, wrapping around at the end of the table
whenever n N, i.e. the following sequence is
obtained

x[n modulo N], n = 0, 1, 2, (4)

where n modulo N is the remainder of the division n/N (a)

when this quotient is computed as an integer.
For 0 the sampling process is noncoherent with
the sinewave. In this case for implementing the
sinewave by LUT method an algorithm is proposed.
This algorithm consists in the following steps:
step 1: A memory with a capacity K samples is
considered. The values kfin/fs (k = 0, 1, 2,, K) are
computed. The k value for which kfin/fs is the nearest
closely to an integer number is determinate. This
value k = M (M K) and q = round(Mfin/fs), where
round(b) is the most closely integer to b.
~
step 2: The frequency f in = qf s / M is computed.
~
step 3: The first M samples of ~ x [n] = Asin(2 f n/f ),
in s
n = 0, 1, , M-1, are stored in the memory.
step 4: The sinewave ~ x [n] is generated by stepping (b)

through the table, wrapping around at the end of the

table whenever n N, i.e. the following sequence is
obtained
~
x [n modulo M], n = 0, 1, 2, (5)

x [n] are very small if K

The errors between x[n] and ~
is high (K 4096 samples).

III. SIMULATION RESULTS

By each method the sinewaves are simulated using the

simulator sim5x.exe given by Texas Instruments [5].
The samples of the sinewaves were obtained by
means of programs written in C.
The dynamic performances SINAD and THD of the
sinewaves were determinate by the algorithm
proposed in [6]. The frequencies of the sinewaves
were estimated by the interpolated fast Fourier
transform (interpolated FFT) algorithm [7]. The
Blackman-Harris 191dB window [8] was employed
for estimating the dynamic performances and the
frequencies of the sinewaves.
Fig. 1 shows the performances of the sinewaves
implemented by the recurrence formula method at
some frequencies. The samples of the sinewaves were
226
(c)
Fig. 1. The performances of the sinewaves implemented by the
recurrence formula method.
In Fig. 2 are presented the performances of the
sinewaves generated by the LUT method with the
algorithm proposed at the same frequencies as for the
sinewaves implemented by the recurrence formula
method. The samples of the sinewaves were generated
with 15.625 kHz sampling frequency. The memory
capacity was K = 7 K samples. The sampling
frequency of the acquisition process was equal to
15.625 kHz and the number of samples acquired was
N = 1024.
IV. EXPERIMENTAL RESULTS

The experimental results were carried out by means of

the TMS320C5X DSK board [9]. The TMS320C5X
DSK is a low-cost, simple, stand-alone application
board equipped with the TMS320C50 DSP. DSK
contains an analog interface circuit (AIC)-TLC32040,
which provides the necessary conversion between the
analogue and digital domain. For this purpose it
incorporates a band-pass antialiasing input filter, a 14-
bit analog-to-digital converter (ADC), a serial port by
which AIC communicate with TMS320C50, a 14-bit
digital-to-analog converter (DAC) and a low-pass
output reconstruction filter. DSK is connected to a PC
via a RS232 interface.
(a) The acquisition system was also a TMS320C5X
board. The experimental setup is presented in Fig. 3.

Sinewave generator

RS232 OUT

IN

DSK5X board #1
PC #1

Acquisition
(b)

RS232 OUT

IN

DSK5X board #2
PC #2

Fig. 3. The experimental setup.

The same algorithms as in simulation estimated the

dynamic performances and the frequencies of the
sinewaves.
Fig. 4 shows the performances of the sinewaves
implemented by the recurrence formula method at the
(c) same frequencies as in simulation. The samples of the
sinewaves were generated with 15.625 kHz sampling
Fig. 2. The performances of the sinewaves implemented by the frequency. There were used two sampling frequencies
LUT method with the algorithm proposed. fs = 7.95 kHz and fs = 15.625 kHz and were acquired
N = 1024 samples.
From Figs. 1 and 2 it follows that the performances of
the sinewaves implemented by the LUT method with
the algorithm proposed are much higher than the ones
obtained by using the recurrence formula method.
These results are achieved because the samples of the
sinewaves are obtained in the case of using the
recurrence formula method, in comparison with the
ones obtained by using the LUT method, after some
operations with fixed-point arithmetic, which are
affected by errors. However, the recurrence formula
method leads, also, to very accurate results.
227
 (a)
(b)
(c)
Fig. 4. The performances of the sinewaves implemented by the
recurrence formula method at two sampling frequency of the
acquisition process:
fs = 7.95 kHz (star) and fs = 15.625 kHz (circle).
Fig. 5 presents the performances of the sinewaves
generated by the LUT method with the algorithm
proposed at the same frequencies as in simulation.
There were used, also, two sampling frequencies of
the acquisition process fs = 7.95 kHz and
fs = 15.625kHz and were acquired N = 1024 samples.
228
(a)
(b)
(c)
Fig. 5. The performances of the sinewaves implemented by the
LUT method at two sampling frequency of the acquisition process:
fs = 7.95 kHz (star) and fs = 15.625 kHz (circle).
By comparison the performances SINAD and THD of
the sinewaves implemented with TMS320C5X DSK
board at the acquisition sampling frequency
fs = 15.625 kHz with the ones obtained from
simulated sinewaves it follows that the performances
are severely degraded because of the poor
performances of the low-pass output reconstruction
filter of the AIC [9]. Due to the behavior of the AIC
output filter the dynamic performances of the
sinewaves implemented by both methods are very
close. However, the dynamic performances of the
sinewaves are, in many cases, superior to the ones
obtained with the analog sinewave oscillators.
From Figs. 4(c) and 5(c) it follows that the sinewaves
frequencies are more accurate at small frequencies
when the LUT method with the proposed algorithm is
employed than when the recurrence formula method is
used.
Another important conclusion drawn from Figs. 4 and
5 is that the performances of the sinewaves are not
affected by the value of the sampling frequency of the
acquisition process.
V. CONCLUSION
This paper was focused on the evaluation and the
comparison of the performances of the sinewaves
implemented with a 16-bit fixed point DSP-
TMS320C5X by two methods. These methods were
ones of the most used in the implementation of a
digital sinewave: the recurrence formula method and
the LUT method. In the case in which the acquisition
sampling process is noncoherent with the sinewave
for implementing the sinewave by LUT method an
algorithm is proposed.
The simulation results indicate that the performances
of the sinewaves implemented by the LUT method
with the algorithm proposed are much higher than the
ones obtained by using the recurrence formula
method. The experimental results were carried out by
means of the TMS320C5X DSK board.
Unfortunately, the dynamic performances of the
sinewaves implemented by this board are severely
degraded by comparison with the simulated ones
because of the poor performances of the low-pass
output filter of the AIC of the DSK board. However,
the dynamic performances obtained are, in many
cases, superior than the ones provides by the analog
sinewave oscillators. From the experimental results, it
follows that the sinewaves frequencies are more
accurate, at smaller frequencies, when the LUT
method with the proposed algorithm is used.
229
ACKNOWLEDGEMENT
The results reported here were partially obtained in
the framework of the CNCSIS Grant AT 32940
number 1, dedicated to the analog-to-digital
converters dynamic testing in multi-tone mode.
RFERENCES
[1] Texas Instruments, TMS320C5X DSK, Applications
Guide, 1997.
[2] Burr-Brown, Intelligent Instrumentation, The Handbook of
Personal Computer Instrumentation, Data Acquisition,
Test, Measurement & Control, 7th Edition, 1994.
[3] D. Garcia, Precision Digital Sine-Wave Generation with
the TMS32010, Application Report, Texas Instruments,
1989.
[4] D. Belega, Placa de dezvoltare TMS320C5X DSK in
aplicatii, Editura Politehnica Timisora, 2002.
[5] A. Iacovliev, Structuri numerice de prelucrare. Lucrri de
laborator, Centru de multiplicare Universitatea
Politehnica Timioara, 1995.
[6] Benetazzo, L., Narduzzi, C, Offelli, C., and Petri, D.,
A/D Converter Performance Analysis by a Frequency-
Domain Approach, IEEE Trans. Instrum. Meas., vol. 41,
pp. 834-839, Feb. 1992.
[7] C. Offelli, D. Petri, Interpolation Techniques for Real-
Time Multifrequency Waveforms Analysis, IEEE Trans.
Instrum. Meas., vol. 39, pp. 106 - 111, Feb. 1990.
[8] O. M. Solomon, The Use of DFT Windows in Signal-to-
Noise Ratio and Harmonic Distortion Computations,
IEEE Trans. Instrum. Meas., Vol. 43, No. 2, pp. 194-199,
April 1994.
[9] Texas Instruments, TMS320C5X DSK, Users Guide,
1997.
.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
On Frequency Measurement by using Zero Crossings
Septimiu Mischie1
Abstract The paper presents the problem of frequency
measurement of periodical signals by using the counting
of zero crossings of those signals. Theoretical principles
are used in order to implement one frequency meter
instrument. Because this operation needs digitizing of
the measured signal, and it is desired as the proposed
instrument to measure a wide range of frequencies with
a minimum resolution, o method for aliasing detecting is
presented. Thus, if the signal frequency becomes larger
than one half the sampling frequency, the user is
informed, and the sampling frequency is increased.
Keywords: zero crossings, sampling frequency, aliasing
I. INTRODUCTION
For a periodic waveform, the frequency f and the
period T are related by the following operation
f=1/T.
Usually, frequency or period measurement of periodic
waveform is achieved by means of counter-timer
instruments [1], [2], [3] by counting the pulses from a
time interval. The hardware of these instruments is
not simple. The measurement time can have for
instance, values from 1 ms to 10 s [3], depending on
desired resolution. One cheap solution is using the
data acquisition boards because these devices can
implement very many functions, by using of proper
software. Thus, one possibility is represented by using
counter-timer on-board circuits, but is further
necessary a trigger circuit in order to transform the
input signal (generally an analog signal) in a square
TTL signal. In this paper is proposed a solution which
first achieves digitizing of the measured signal, and
then by using the obtaining samples, the zero
crossings from an observer time are counted. Based
on number of zero crossings, the frequency is
computed.
The zero crossings are frequently used on frequency
measurement [4], [5]. Namely, one half the period of
the signal is computed as the number of sampling
intervals between two contiguous zero crossings. For
this method is necessary a high number of data
samples in one signal period in order as the accuracy
to be acceptable. This requirement limits the
maximum frequency of the measured signal, but has
the advantage of a small measurement time.
1
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail septimiu.mischie@etc.utt.ro
In the proposed solution, the maximum frequency to
be measured is limited only by sampling theorem, i.e.
at from sampling frequency. The sampling
frequency is automatically selected, in order to obtain
the minimum resolution. The measurement time is
higher than that from [4] and [5], but its value is not
critical because the proposed method is used to an
instrument similarly with that from [3]. However, the
main idea from this paper is a method for detecting
the phenomenon of aliasing. This requirement is
necessary because each measurement range has a
different sampling frequency, and if the signal
frequency becomes higher than one half the sampling
frequency, the obtaining result will be false. Thus, if
this event happened, the measurement process will be
stopped, and then will be resumed with another
sampling frequency, which is higher than precedent
(1) one. This method can be applied, so will be seen in
the next section, only if at the beginning of the
measurement process the aliasing is not present. This
idea has applications also in other fields where the
frequency of acquisition signal is variable and the
sampling frequency must be a little bit higher then
two-fold this frequency.
The paper is organized as follows. Section 2
presents the theoretical principles that are used in
order to implement the presented methods. Section 3
presents the practical achievement of proposed
solutions, and section 4 presents experimental results
that has been obtained.
II. THE FREQUENCY MEASUREMENT
USING THE COUNTING OF ZERO
CROSSINGS
Let a signal s(t) as in fig.1, having the frequency
f=1/T. This signal is sampled by sampling frequency
fs=1/Ts, and then is digitized by means of an analog to
digital converter.
The period T can be obtained by
t
T= m , (2)
Np
where tm is the measurement time and Np is the
number of periods. Np is not mandatory to be an
integer. Because
230
 t m = ( N 1)Ts ,
where N is the number of acquired data samples, and
using equations (1) and (2), it follows that the
frequency is
N p fs
f = . (4)
N 1
Fig.1 A periodical waveform
The number of periods Np can be computed based
on number of zero crossings Nz, if there are two zero
crossings per signal period. Thus, from fig.2 it can be
observed that at two zero crossings correspond
minimum 1/2 periods and maximum 3/2 periods, or
Np=Nz/2.
Fig.2 Data samples with two zero crossings
Equation (5) gives the number of periods with a
quantization error of 1/2 periods.
Finally, using (4) and (5) the frequency can be
computed by
N z fs
f = . (6)
2( N 1)
The relative accuracy in the measurement of
frequency f according to (4) is
(3) 0.5 1
f = + fs = + fs , (7)
Np Nz
where fs represents the relative accuracy of sampling
frequency and depends on the manner of practical
implementation of the sampling process (usually, the
on-board counter-timer circuit give the clock for this
process). It follows that the single way to minimize f
is represented by choosing a higher value for Np, or
equivalently, for Nz. For a given frequency, Np is
direct proportional with N (see equation (4)).
Also, based on (6) it follows that the frequency
resolution (also the minimum measurable frequency)
is
fs
f = . (8)
2( N 1)
If f is divided by f given by (6) is obtained the
amount 1/Nz, that is, the first term of f. It follows that,
the minimizing of f by Nz is equivalent with the
minimizing of frequency resolution.
Thus, it can be say that the two parameters of this
method of frequency measurement are fs and N. The
choosing of these parameters must be done based on
the next requirements:
1. The maximum value that can be measured is fs /2.
2. The resolution given by (8) must be as minimum
as possible.
3. The measurement time given by (3) must be also
as minimum as possible.
It is evidently that the requirements 2. and 3. one
contradicts each other, because f=1/(2tm). Hence, a
(5) compromise must be done in order to choose these
parameters. Thus, one possibility is that sampling
frequency to be maximum possible, and number of
samples variable, in order to modify the resolution,
having very high values for low resolutions.. This
possibility has the drawback that the processor will be
very busy with the reading of data samples, or if the
samples are written in a table, this needs a large
amount of memory. Therefore, the proposed solution
is as the sampling frequency to be as minimum
possible function of signal frequency (that is a little
amount higher than twofold of signal frequency), and
the number of samples to be also variable, but the
values will be less than in first possibility.
The using of this method on frequency
measurement, needs as previously presented, the
verifying of the sampling theorem. If this theorem is
satisfied, i.e. f fs/2, the number of zero crossings is
from (6)
2( N 1) f
Nz = , (9)
fs
and if this theorem is not satisfied or the aliasing is
present, i.e. f > fs/2, the number of zero crossings
which are detected is
231
 2( N 1)
N za = kf s f , (10)
fs
where k is a positive integer, and its value is that the
expression |kfsf | to be minimum. It can be shown
that for k = 0, equation (10) is identically with (9).
Thus, for a correct measurement, namely for f<fs/2,
when N is constant, the increasing of f is equivalent
with the increasing of number of zero crossings, and
the decreasing of f is equivalent with the decreasing of
number of zero crossings (Nz in equation (9) ). If f
Fig.3 Variation of zero crossings function of sampling frequency
For this purpose, first in fig.3 is presented the
variation of number of zero crossings function on
sampling frequency for a constant number of samples,
if also the signal frequency is constant.. Thus, two
statements result from this figure:
1. If sampling theorem is satisfied, fs 2f, the
number of zero crossing decreases as sampling
frequency increases.
2. If sampling theorem is not satisfied, fs<2f, the
number of zero crossings can increase or can decrease
as sampling frequency increases.
From these statements, results that the aliasing
phenomenon can be not detected.
However, if sampling theorem is satisfied, and then
the signal frequency increases, that f>fs/2 but f<fs, in
this case number of zero crossing increases as
sampling frequency increases, in opposite with above
first statement. In conclusion, the aliasing can be not
232
increases and become higher than fs/2, the number of
zero crossings results from equation (10) with k = 1
and will decrease. Thus follows the false conclusion
that the frequency would decrease. It is specify that in
all presented cases, the sampling frequency is
constant.
In order to avoid this false measurement, it is
desired the detecting of the aliasing phenomenon, that
is the case when f > fs/2. One way to do this is by
making the data acquisition with two different
sampling frequencies.
detected, but the transition from aliasing free to
aliasing can be detected.
In order to materialize above statements, one
proposes the following idea.
On each measurement, further on current data
acquisition with sampling frequency denoted as fs1
and corresponding number of zero crossings Nz1, it
makes an additional data acquisition with fs2 = kfs1,
where k is higher than 1, followed by the computing
of the new value of the number of zero crossings,
denoted as Nz2. In fig.4 are presented the variations of
number of zero crossings function on signal
frequency, for both values of sampling frequency, fs1
and fs2. Also, the number of samples N is constant.
From fig.4 it can be seen that, if f<fs1/2, then Nz2 <Nz1.
This is the normal case. If f > fs1/2, i.e. the aliasing
occurs, the inequality Nz2 <Nz1 is satisfied, but only for
a little interval, namely for f (fs1/2, fc). f=fc (denoted
as critical frequency) represents the point for which
the two curves one intersects. If f > fc, one obtains that
Nz2 >Nz1, this being the element which signals that
aliasing was occured.
It is noted however that this idea for aliasing
detecting can be applied only if the measurement
process begin in a case for which the aliasing not
occur and the increasing of frequency to be that its
value to be not greater than sampling frequency.
In order to computes the frequency fc beginning the
aliasing is detected, the expressions of both zero
crossings, for f > fs/2 must be equals, as next is
presented.
2( N 1)( f s1 f c ) 2( N 1) f c
= (11)
f s1 f s2
It results that
f s1 f s 2
fc = (12)
f s1 + f s 2
Thus, the proposed methods were implemented on a
data acquisition board of type National Instruments
PCI 6023.
Mainly, this algorithm does the following:
-the acquisition of data samples begin at maximum
value of sampling frequency, in accordance with
features of the used data acquisition board; if it is
necessary, the sampling frequency is decreased until a
approximate value of frequency can be computed;
- based on previously computed frequency, the
optimum sampling frequency and the optimum
number of samples are choose;
-the permanent frequency measurement is achieved:
the frequency is computed and displayed, and it is
checked if the aliasing occurs; if it there are, an
message is displayed and the sampling frequency is
increased to the next value.
In order to achieve the measurement ranges, it is
necessary the settling of the sampling frequency (i.e.
the highest limit of range is one half the sampling
frequency). Further, the number of samples must also
be settled because, so was pointed, these two amounts
together establish the frequency resolution. Next, in
the table 1, the sampling frequency, the resolution, the
number of samples and the measurement time for
each range are presented. The values correspond to
equations (8) and (3).

Table 1
Range fs Hz Resolution N tm

0-250 500 Hz 0.1 Hz 2501 5s

Hz
0-1000 2000 0.1 Hz 10001 5s
Hz Hz
0-2500 5000 1 Hz 2501 0.5
Fig.4 Variation of zero crossings function on signal frequency for Hz Hz s
two sampling frequencies 0-10 20000 1 Hz 10001 0.5
kHz Hz s
In the interval for which f (fs1/2, fc), the aliasing 0-25 50000 1 Hz 25001 0.5
phenomenon can be not detected and therefore this kHz Hz s
interval must be as little possible. 0-50 100000 10 Hz 5001 0.05
In order to achieve this requirements, it is kHz Hz s
necessary as fs2 to be as near as fs1, or the coefficient k 0-100 200000 10 Hz 10001 0.05
to be a little amount higher than 1. There are however kHz Hz s
two restrictions regarding on choosing of fs2. 0-250 500000 10 Hz 25001 0.05
1. The value must be sufficient high in order to kHz Hz s
allow the variation of number of zero crossings with
least 1 unit. It can be seen that for first two ranges, the
2. The desired increasing of fs2 comparative with fs1 measurement time has high values. If is not necessary
must be possible, accordingly of features of data a resolution of 0.1 Hz, the measurement time can be
acquisition board (the minimum possible variation of decreased. It is pointed out that in order to check if
sampling frequency). there is aliasing, it is necessary one more time
interval, which is less than the measurement time, as
III. THE IMPLEMENTED ALGORITHM how will be seen. In this time interval the signal
frequency must keep constant.
Ideas that have been presented in previous section The implemented algorithm contains the next steps:
allow the achievement of an algorithm that has the 1. Acquire N1 = 501 data samples with maximum
goal on frequency measurement with the minimum sampling possible, fs= fsmax (fsmax=500000 Hz).
resolution. 2. Compute the number of zero crossings Nz. If Nz>
Nt go to 4. Nt is a thereshold value that allows

233
computing the frequency with an acceptable accuracy IV. EXPERIMENTAL RESULTS
(Np=50).
3. Acquire N1 data samples with fs/2. Go to 2. The presented algorithm was experimental tested in
4. Compute the frequency f using (6). Set fs= 2f. order to achieve the desired frequency meter, and the
5. Choose the sampling frequency function of fs: the obtained results will be presented in this section.
less value from table 1 which is higher than fs. Denote Thus, a HM 8130 function generator was used in
this value by fs1. Also, function on selected sampling order to apply the signal to be measured. The type of
frequency, from table 1 results the number of samples waveform was mainly sine, but also square, triangle
N. or sawtooth were used. This generator displays the
6. It is make the current frequency measurement, value of generated frequency and will be used as
with the sampling frequency to fs1. reference for comparison with the implemented
The next two steps are executed continuously. instrument.
6.1. Compute Nz1 at fs1 for N samples. Compute Two categories of experiments have been achieved.
frequency f using (6) and display its value. First, the accuracy of measurement was verified, for
6.2. Compute Nz2 at fs2=kfs1 for N/10 samples. If all ranges of the instrument. Second, the manner of
Nz2>Nz1/10 the aliasing occurs; display the message detecting the aliasing was tested.
Aliasing and utter a sound; set fs1 to the next value Thus, in order to verify the precision of the
accordingly tab.1. instrument, different frequencies were settled to the
In first three steps, the frequency is computed with HM 8130 function generator. For each frequency, a
an accuracy of 2 percent, because the thereshold Nt number of 50 measurements were made with the
has been imposed as 50. The value of 501 for number implemented instrument and the obtained results were
of data samples allows as the sampling frequency to stored.
be decreased to 1/20 from signal frequency. In table 2, for each frequency from HM 8130 are
In order as the needed time for aliasing detecting to presented the distribution of the obtained values fm
be as low as possible, in the second data acquisition (the value and the number of achievements).
only N/10 samples has been acquired.
If is necessary, for instance when the measurement Table 2
signal is changed, the step.6 can be executed only f (from HM8130) fm
once, and then the algorithm can be resumed 239.30 Hz 239.3 Hz 239.4 Hz
beginning with step.1, but in this mode the needed 49 1
time is higher than that for executing of step.6. 729.00 Hz 729.0 Hz 729.1 Hz
Based on previous presented two restrictions, the 43 7
constant k has value of 1.05. 2207.0 Hz 2207 Hz 2208 Hz
Also, next is presented the way for determination of 48 2
number of zero crossings, where e(i) represents the 8554.0 Hz 8554 Hz 8555 Hz
sample at moment i and Nz represent the number of 45 5
zero crossings: 23930 Hz 23930 Hz 23931 Hz
For i=2,,N
38 12
If e(i-1)e(i)<0 or e(i-1)=0, Nz=Nz+1.
41040 Hz 41040 Hz 41050 Hz
It is specify that before of determination the number
47 3
of zero crossings, the dc component of signal is
removed by using the adaptive LMS algorithm [7]. 91670 Hz 91670 Hz 91680 Hz
This algorithm was implemented by a program in C 40 10
language. For this purpose, the low level functions 239.28 kHz 239.28 kHz 239.29 kHz
from NI-DAQ software was used in BorlandC 5.0. 36 14
The algorithm is an off-line real time algorithm (see
[6]), because it processes a group of N data samples, From the results from table 2 it can be seen that the
that previous are stored in the memory. This way was quantization error from numbering of periods brings
imposed because the functions from NI-DAQ give about the frequency resolution. For all frequency, the
groups with a pre-established number of data samples, number of achievements of the results without error is
which can be processed only when the acquisition is higher than that of the results affected by quantization
ready. For this reason, the measurement time is a little error.
amount higher then that from table 1, namely with the In order to verify the detection of aliasing, on each
required time for computing the number of zero measurement range, the frequency of measured signal
crossings and frequency. However this increasing is was modified beginning from values a little amount
visible only for last three ranges, thus the new value less than one half the sampling frequency, to values a
of the measurement time for these ranges is about of little amount higher than one half the sampling
0.07s. frequency, and then the value for which the
instrument signals aliasing was kept in mind. This
value has been compared with the theoretical value,
which is obtained using equation (12).

234
 Thus, in table 3 these data that allow the detection
of aliasing are presented. fcexp represents the
experimental value which has been obtained for
signaling of aliasing.

Table 3
fs1 fc fcexp
500 Hz 256.09 Hz 256.3 Hz
2000 Hz 1024.39 Hz 1025.5 Hz
5000 Hz 2560.97 Hz 2564.3 Hz
20000 Hz 10243.9 Hz 10210 Hz
50000 Hz 25609.7 Hz 25643 Hz
100000 Hz 51219.5 Hz 51.405 kHz
200000 Hz 102439 Hz 102.05 kHz
500000 Hz 256097 Hz 257.15 kHz

Based on data from table 3 it can be seen that the

theoretical idea for aliasing detecting was acceptable
verified by experiments.

V. REMARKS

The paper presents an instrument for frequency

measurement, which is achieved with a data
acquisition board, based on zero crossings counting.
The theoretical contributions regarding to aliasing
detecting are presented. An algorithm that allows the
measurement of frequency and switch among ranges
automatically when the aliasing is detected, is also
presented. Finally, the experimental results were
found in good agreement with the theoretical
considerations.

REFERENCES

[1] M. Irshid, W. Shahab, B. El-Asir, A Simple Programmable

Frequency Meter for Low Frequencies with Known Nominal
Values, IEEE Transactions on Instrumentation and Measurement,
vol. 40, No. 3, pp. 462-466, August 1991.
[2] M. Prokin, DMA Transfer Method for Wide-Range Speed and
Frequency Measurement, IEEE Transactions on Instrumentation
and Measurement, vol. 42, No. 4, pp. 842-846, August 1993.
[3] ***, Programmable Universal Counter HM 8122, User Manual,
Hameg GmbH, Frankfurt, Germany
[4] G. DAntona, A. Ferrero, R. Ottoboni, A Real-Time
Instantaneous Frequency Estimator for Rotating Magnetic Islands
in a Tokamak Thermonuclear Plasma, IEEE Transactions on
Instrumentation and Measurement, vol. 44, No. 3, pp. 725-728,
June 1995.
[5] V. Friedman, A Zero Crossing Algorithm for the Estimation of
the Frequency of a Single Sinusoid in White Noise, IEEE
Transactions on Instrumentation and Measurement, vol. 44, No. 3,
pp. 1565-1569, June 1993.
[6] P. Arpaia, F. Avallone, A. Baccigalupi, C. De Capa, Real-Time
Algorithms for Active Power Measurement on PWM-Based
Electric Drivers, IEEE Transactions on Instrumentation and
Measurement, vol. 45, No. 2, pp. 462-466, April 1996.

235
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

The development of a new system to measure Camber and

Toe using stereo cameras
Liviu Toma Jr.1, Fangwu Shu1, Alimpie Ignea2, Werner Neddermeyer1, Michael Schnell1
Abstract This paper presents a new measurement
system used in car industry. The goal is to adjust the
position of the front wheels of a car with respect to the
axle where the wheels are mounted. For that, it is
required to measure two angles. They are known in the
technical literature as Camber and Toe. Our new idea
was to use three stereo sensors and structured light to
solve this problem. In our approach, we present two
methods for measuring these two angles. The obtained
accuracy is 0.085 degrees (5 minutes).
Keywords: car industry, wheel alignment, axle, Camber,
Toe, stereo sensor, structure light
I. INTRODUCTION
II. DEFINITION OF CAMBER AND TOE
In this section we define and then explain the two
angles we want to measure with our vision system.
Camber is the inward or outward tilt of the wheel
measured from top to bottom, reference [12]. This
angle is adjusted to prevent excessive tire
deterioration and to enhance straight ahead stability. It
is measured in degrees and has several methods of
adjustment. In figure 1, one can understand better the
definition of this angle. In this figure, there are
presented three possible situations for this angle:
positive Camber, negative Camber and zero Camber.

The wheel alignment problem is an important task and

concerns all car producers. There were developed a
lot of measurement systems to be used for solving this
problem. At the beginning there were produced
systems based only on mechanical methods. The
disadvantage of these methods was the time for
measuring which was too long. Also, the accuracy of
the measured results was influenced by the errors of
the tire surfaces. The second step was to build
measurement systems, which use both mechanical and
optical methods for measuring. In this category we
have systems based on laser technology and systems,
which use cameras. The first ones have the
disadvantage that they are very expansive. For the
second type the accuracy is the task that must be
improved. It is also important to know that the
systems, which use cameras, can be based on the
multi-camera concept or stereo camera concept. As
one can see from the title of this paper, the method Fig. 1. Definition of Camber
developed by us is making use of the stereo concept.
The reminder of this paper is organized as follows. In The angle formed by a horizontal line through the
section 2 we present details about the definition of plane of one wheel versus a perpendicular line to the
Camber and Toe. Section 3 presents the stereo sensor centerline is called the individual toe, reference [12].
and the device used to project structured light. The This is the most critical tire angle. When a horizontal
main section of this paper is section 4, where we line is drawn through the plane of each wheel, and
present the measurement system developed by us. The they intersect in front of the wheels, this is called toe-
analysis of the measured results is presented in section in or positive toe. When they intersect behind the
5. Section 6 concludes. wheels, this is called toe-out or negative toe. In figure

1
University of Applied Sciences, Informatics Department, Neidenburgerstr. 43, D-45877, Gelsenkirchen, Germany,
e-mail toma@informatik.fh.ge.de, shu@informatik.fh-ge.de, werner.neddermeyer@informatik.fh-ge.de
2
Facultatea de Electronic i Telecomunicaii, Departamentul de Masurari si Electronica Optica,
Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail alimpie.ignea@etc.utt.ro

236
2, one can understand better the definition of this
angle.
Fig. 2. Definition of Toe
III. STEREO SENSOR AND LIGHT PROJECTOR
In this section we present some important aspects
about the stereo sensor and the light projector we
used, in order to be able to describe our system, in
section 4, shortly and efficiently.
The stereo sensor, which was used in our
measurement system, is the result of three years of
work. One can understand all our steps concerning the
building, the calibration and the verification of the
stereo sensor by consulting from the reference list the
following: [7], [8], [9], [10] and [11]. With this sensor
one can measure the 3D position of a certain point
with an absolute accuracy of 0.1 mm. The distance
between the point to be measured and the stereo
sensor can be adjusted between 150 mm and 500 mm
without any influence to the absolute accuracy. The
measurement area is defined by a square with a side
of 100 mm.
Fig. 3. Stereo sensor and light projector
In figure 3, one can see the stereo sensor. It is
composed from two cameras mounted in a parallel
configuration, reference [8]. After the calibration of
237
the stereo sensor is realized, we are able to measure
3D coordinates of different points. These coordinates
are measured with respect to the stereo sensor frame.
The stereo sensor frame is defined in the calibration
procedure and remains fixed to the sensor after the
calibration, reference [11].
The light projector can be also seen in figure 3, just
below the cameras. Our goal was to be able to create
on the tire surface, using light, some marks, which
could be further measured with the stereo sensor.
In figure 4, one can see the shape of the structure light
created by the light projector on the tire surface.
3
Fig. 4. Structured light projected on the tire surface
There are two possibilities to make use of this
structured light. First one is to use as marks the
intersections between the light and different forms
existing on the tire surface. As one can see in figure 4,
in this category are included points 1 and 3. The
second one is to use as marks the crosses defined by
the structured light itself on the tire surface. To this
category belongs point 2.
IV. DESCRIPTION OF THE MEASUREMENT
SYSTEM
A. Presentation of the system
As we said at the beginning of this paper the goal is to
build a vision system for measuring Camber and Toe.
Having now the explanations presented in section 2
and 3, we are able to define a mathematical model in
order to reach this goal.
First of all we define a coordinate frame for the
wheel. We call this, the wheel frame. The origin of
this frame is situated in the middle of the tire. Axe z is
perpendicular to the tire so that the plane determined
by axes x and y is parallel to the tire. Axe x is
horizontal. One can see all these details in figure 7.
With these notations, Camber is determined by
measuring the rotation of the wheel frame around x
axe and Toe is given by the measured value of the
rotation of the wheel frame around y axe.
In this moment we know what we have to measure so
the next problem, which must be solved, is how we
have to measure. In order to explain way we built our
measurement system in the way we did, it is necessary
to present here some details about the measurement
procedure. A detailed description of the measurement
procedure will be presented in part C of this section.
The angle information we need is obtained by
knowing the orientation of the tire plane (the plane
defined by axes x and y) relative to a reference plane.
So, the task is to measure this tire plane. It is known
that a plane is determined by at least three points,
which are not all situated on the same line. Starting
from the plane definition we decided to use three
stereo sensors placed on a circle at equal relative
distances between them, as one can see in figure 5.
1200
1200
1200
of the stereo sensors frames with respect to the
reference frame. We denoted with SR the reference
frame situated in the middle of the plate and with SS1,
SS2 and SS3 the stereo sensors frames. In figure 6, it is
drown only one sensor frame, because the situation is
similar for the other two. The mathematical
explanation, which follows for one sensor, will be
applied in the same way for the other two sensors.
With T(SR-SSi) we denoted the transformation from the
reference frame to one sensor frame.
The calibration plate, we used, has 121 points and we
know very precisely their position with respect to the
reference frame. We denote the coordinates of one
point from this plate with xR, yR and zR. The same
point will be measured with the stereo sensor and we
obtain the coordinates xSi, ySi, and zSi. According to the
reference [1], between these coordinates we have the
following relation:

(xR yR z R 1)T = SS R T (xSi

Si
y Si z Si 1)T . (1)

Using more than four points for each sensor we obtain

an over determinate system of nonlinear equations.
According to the references [4], [5] and [6] such
systems are solved in two steps. First step, we make
the system linear and second step we use least square
Fig. 5. Description of the measurement system
methods to find the solution of the system.
B. Calibration procedure
C. Measurement procedure
We explained in section 3 that in the calibration
procedure of the stereo sensor is defined a stereo There are two different methods of measuring. Until
sensor frame, reference [11]. It means, the coordinates now, we have implemented in practice only one
of the points measured with a calibrated stereo sensor method and obtained test results, which are presented
are given relative to its defined stereo sensor frame. in section 5. The second method is described shortly
The three stereo sensors, which are fixed on a rigid at the end of this section.
plate, as one can see in figure 5, are first calibrated The method, which we have implemented, is based on
(see section 3). This means, each one has its own identifying marks of type noted with 3, as one can see
frame. The next step is to find the relative position in figure 4. This method used the fact that on the
and orientation of these three frames with respect to a surface of the tire there are several profiles, which
reference frame. This is in fact the calibration modify the shape of the structured light projected on
procedure of our measurement system. it. Because these shape modifications are very small
we had to develop image processing algorithms to
provide us enough and accurate information. We have
used sub-pixel accuracy and segmentation methods
y according to the references [2], [3] and [11].
y Our goal is to identify points, which are in the same
plane and of course this plane must be parallel to the
x tire plane. As one can see in figure 7, there are some
SSi SR profiles having circle shape on the surface of the
x wheel. The big advantage for these circles is that they
z define, at least theoretically, each one a plane, which
z is parallel to the tire plane.
The structured light allows us to take for each stereo
sensor maximum three points per circle. This way, we
T(SR-SSi) can use maximum nine points to compute the plane
where the circle is situated. Using the best-fit method
we eliminate from these points those, which have big
Fig. 6. Calibration of the measurement system errors and finally, compute a plane parallel with the
In figure 6, one can see the calibration plate we have tire plane. Having this plane, we can compute the
used in order to compute the position and orientation values for Camber and Toe.

238
The second method is based on identifying marks of In figure 9, one can see the distribution of the errors
type noted with 2, as one can see in figure 4. The idea for Toe. They are situated between -0.38 and 4.86
is to use the light crosses for identifying which pixel minutes.
from the image obtain with one camera of the sensor
corresponds to a certain pixel from the image obtained 5.00

with the other camera. This way, we can measure the d

e i
4.00

3D coordinates for a lot of points belonging to the l n

t 3.00
light lines projected on the tire. With this information a m
2.00
the next step is to calculate the tire plane and its i
a n

orientation relative to a reference plane. n u

g t
1.00

l e 0.00
e -3.000 -2.000 -1.000 0.000 1.000 2.000 3.000
Profile 1 y
-1.00
ang le in d eg r ee

Profile 2 Fig. 9. Distribution of the errors for Toe

Rot x = Toe
x
VI. CONCLUSIONS
Rot x = Camber
We have succeeded to build a vision system using
XY- Plane = Tire Plane simple methods and cheap components with a good
z
accuracy. These first results obtained in the
measurement procedure confirm us the fact that our
vision system could be further developed and
Profile 3 improved. Using the second method, improving the
quality of the structured light and developing better
Fig. 7. Explanations for the measurement procedure image processing algorithms and mathematical
algorithms we will be able to reach the accuracy of
V. ANALYSIS OF THE MEASURED RESULTS 0.1 minutes with our vision system.

To test our system we use a special device having a REFERENCES

wheel and the possibility to adjust it at different
[1] Richard P. Paul, Robot Manipulators: Mathematics,
angles between 3 and 3 degrees for both Camber and Programming and Control, The MIT Press Cambridge,
Toe. Before we start a normal measurement the wheel Massachusetts and London, 1981, pp. 1-63
is fixed so that, the special device indicates 0 for both [2] V. Gui, D. Lacrama, D. Pescaru, Prelucrarea Imaginilor,
Camber and Toe. For this position, we make a zero Editura Politehnica, Timisoara, 1999
[3] J. R. Parker, Algorithms for image processing and computer
measurement. It means all the measurements, which vision , Copyright 1997 by John Wiley & Sons, Inc.
follow to this zero measurement, are done relative to [4] St. Manusar Metode numerice in rezolvarea ecuatiilor
this zero wheel frame. neliniare, Editura Technica, Bucuresti, 1981
For the diagrams, which follow we have measured ten [5] Pavel Naslau, Metode numerice, Editura Politehnica,
Timisoara 1999
different orientations of the wheel, but sometimes [6] Numerical Recipes in C: The Art of Scientific Computing
keeping one angle fixed. On the horizontal scale we (ISBN 0-521-43108-5), Copyright 1988-1992 by Cambridge
have represented the real value of the angle in University Press, pp.59-71
degrees. On the vertical scale we have represented the [7] L. Toma Jr., A. Ignea, W. Neddermeyer, A Comparison
Between Two Camera Calibration Methods, Buletinul tiinific al
difference between the measured value and the real Universitii "Politehnica" din Timioara, Tomul 45(59), 2000
value of the angles. The unit used for this difference is Fascicola 1, Vol. 2, pp. 222-227
the minute. [8] L. Toma Jr., F. Shu, A. Ignea, W. Neddermeyer, The
In figure 8, one can see the distribution of the errors analysis of the errors for a stereo sensor built in two
configurations, Buletinul tiinific al Universitii "Politehnica"
for Camber. They are situated between 0.38 and 4.06 din Timioara, Tomul 47(61), 2002 Fascicola 1-2, Vol. 2, pp. 174-
minutes. 178
[9] W. Neddermeyer, A. Ignea, L. Toma Jr., Stereo Vision
d 5.00 Sensor for Industrial Applications, SCI2002 Orlando - Florida,
e i
4.00
USA, Proceedings, Volume IX, pp. 157-161
l n
t
[10] W. Neddermeyer, A. Ignea, L. Toma Jr., F. Shu A new
a m 3.00 solution for matching the left and the right image models with high
i
accuracy, SCI2003 Orlando - Florida, USA, Proceedings, Volume
a n 2.00
n u X, pp. 145-149
g t 1.00 [11] L. Toma Jr., F. Shu, W. Neddermeyer, A. Ignea, Stereo
l e
e
Vision Sensor for 3D Measurements A complete solution to
0.00
-2.500 -1.500 -0.500 0.500 1.500 2.500 produce, calibrate and verify the accuracy of the measurements
ang le in d eg r ee results, ICINCO2004 - Setubal, Portugal, Proceedings, Volume 2,
pp. 410-416
[12] http://www.allwheelalignment.com/alignment.htm
Fig. 8. Distribution of the errors for Camber

239
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

A Correlation Analysis
of Measured CO-Concentration Signals
Sabin C. Ionel1
Abstract Ideal low-pass filtering can be used as a 600 cycles/hour. The temporal length covered by each
useful pre-processing method in order to eliminate the signal is 5h50' . The concentrations are measured in
measuring noise from signals representing CO-
concentration. Thereafter, the correlation coefficient [ mg / m3 N ]. Such signal pairs were measured, by
proves to be a suitable tool in a comparative analysis of day and by night, in several parks and crossroads of
the data. If the signals are measured simultaneously and Timioara city.
in the same location, using different instruments, the
Signals CDH1,2,3 Signals CDS1,2,3
correlation analysis allows a comparison of the 20 10
measuring systems. Applied to signals measured with a
single instrument, the correlation analysis can put into
evidence diurnal repeatability of CO-concentrations. 10 5
Keywords: CO-concentration, ideal filtering, correlation
coefficient. 0 0
1000 2000 3000 1000 2000 3000
20 20

I. INTRODUCTION
10 10
CO-concentration signals have a nonstationary,
random character. Consequently, certifying the 0 0
1000 2000 3000 1000 2000 3000
validity of CO-concentration data delivered by 40 40
different measuring devices can be a difficult task.
In this paper, a method allowing comparative 20 20
analysis of CO-concentrations measured with two
fully different instruments is presented. Thus, the 0 0
classic HORIBA instrument measures the CO- 1000 2000 3000 1000 2000 3000
concentrations locally, in a certain point, while the Sample Number Sample Number
Siemens-HAWK analyser delivers CO-concentrations Fig. 1. Typical signals measured with Horiba (CDH*) and with the
Siemens-Hawk (CDS*) instrument, respectively.
spatially averaged over a distance of 10 100m .
Obviously, the two instruments do not measure the All concentrations are measured in [ mg / m3 N ].
same quantity, but a comparison of the measured data
is still meaningful since both signals are representing
the pollution level in the same location [1], [2], [3]. III. IDEAL FILTERING

Signal conditioning is an unavoidable step in any

II. THE MEASURED SIGNALS
Fig.1 shows three signal pairs measured in a crossroad
(C), by day (D) with the Horiba instrument (H)
respectively using the Siemens-Hawk (S) analyzer.
From top to bottom, the signal pairs represented in
fig.1 are: (CDH1-CDS1), (CDH2-CDS2) and (CDH3-
CDS3). The signal pairs were measured on
consecutive days, in the same hourly interval. Each
CO-concentration signal contains 3500 samples
achieved at a sampling period of 6 seconds,
corresponding to a sampling frequency of (1/6) Hz, or
signal processing application [4]. Ideal filtering
can be chosen as an useful pre-processing
method. An ideal low-pass filter has the
frequency characteristic:
1 for f f c
H( f ) = (1)
0 for f > f c
Using direct and reverse FFT technique, one can
easily implement ideal low-pass and high-pass filters.
The cut-off frequency f c can be established on
empirical basis. The CO-concentration signals

1
Facultatea de Electronic i Telecomunicaii, Departamentul Electronic Aplicat
Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail: sabin.ionel@etc.utt.ro

240
presented in fig. 1 were filtered using f c = 2 cycles / measured with the Horiba and Siemens devices,
hour. The low-pass (LP) part of the measured signals respectively. Correlation coefficients of the LP part of
can be seen in fig. 2. The rapid fluctuations of the the air pollution signals represent also a good
initial signals, including measuring noise, were measure of the diurnal repetition of the CO-
elliminated through low-pass filtering. concentration level. This correlation is meaningful
since the pair of signals were measured on
Signals CDH*-low Signals CDS*-low consecutive days or nights, starting at the same hour.
10 10

8
5
IV. CORRELATION ANALYSIS
6
0
1000 2000 3000 1000 2000 3000 The measured CO-concentration samples can be seen
10 10 as realizations of some random variables. The
correlation coefficient of two random variables
8 5 x[n] and y[n] , defined by [5].
6 0 C
1000 2000 3000 1000 2000 3000 rxy = (2)
12 20 x y
where C and x , y are the covariance respectively
10 10
the variances of x[n] and y[n] , can be used as a
8 0 comparison tool between two or several data series.
1000 2000 3000 1000 2000 3000
Since -1 rxy 1 , through comparative analysis one
Sample Number Sample Number
can distinguish three different cases:
Fig. 2. LP signal pairs CDH*-low and CDS*-low,
in [ mg / m3 N ]. positive correlated data for 0.33 < rxy 1;

uncorrelated data for - 0.33 rxy 0.33 (3)
In the same manner, using ideal high-pass

filtering, one can obtain the high-pass (HP) part of the negative correlated for - 1 rxy 0.33;
measured signals. Thus, the signals CDH*-high and
CDS*-high represented in fig. 3, contain only For example, the correlation coefficients matrix for
frequencies greater then 2 cycles / hour. the six LP signals represented in fig. 2 are shown in
Table 1. Numbering the LP signals according to Table
Signals CDH*-high Signals CDS*-high
20 2 2, the correlation coefficient rij = r ji characterizes the
0 0 resemblance between signals i and j .

-20 -2
1000 2000 3000 1000 2000 3000 Table 1
20 5 1.0000 -0.0774 0.0448 -0.2851 -0.0144 -0.2620
-0.0774 1.0000 -0.4630 0.6906 -0.5757 0.7735
0 0
0.0448 -0.4630 1.0000 -0.1601 0.1942 -0.5015
-0.2851 0.6906 -0.1601 1.0000 -0.4156 0.5531
-20 -5
1000 2000 3000 1000 2000 3000 -0.0144 -0.5757 0.1942 -0.4156 1.0000 -0.6787
50 10 -0.2620 0.7735 -0.5015 0.5531 -0.6787 1.0000

0 0
-50
1000 2000 3000
-10
1000 2000 3000
Sample Number Sample Number
Fig. 3. HP signal pairs CDH*-high and CDS*-high
in [ mg / m3 N ].
The HP part of the signals can be analysed in
order to evaluate the measuring noise, to optimise the
sampling period, the cut-off frequency etc. However,
in the following, we are interested only in the LP part
of CO-concentration signals. Using the correlation
coefficients between the LP part of the signals
measured under similar conditions (the same place
and time interval for one signal pair; however,
different days for different signal pairs), one can
compare the tendencies of the CO-concentrations
Table 2
Signal number i, j LP Signal
1 CDH1-low
2 CDS1-low
3 CDH2-low
4 CDS2-low
5 CDH3-low
6 CDS3-low
For an easier interpretation, the correlation matrix is
graphically represented in fig. 4. Obviously, the
signals measured with Horiba instrument are not
correlated (absence of diurnal repetition). On the
contrary, the signals measured with the Siemens
instruments are highly and positive correlated,
denoting a diurnal repetition of CO-concentration.

241
 Correlation Coefficients for the LP Signals, CDH*-low and CDS*-low Correlation Coefficients for the LP Signals, PDH*-low and PDS*-low
1 1
1 1
S S
D 0 DFig.0 6. A graphical representation of the correlation coefficients
C; P;
1 1 for the signals PD**-low
H H
D -1 D -1
C 1 2 3 4 5 6 P 1 2 3 4 5 6
1 1
2 2
S S
D 0 D 0
C; P;
2 2
H H
D -1 D -1
C P 1 2 3 4 5 6
1 2 3 4 5 6
1 1
3 3
S S
D 0 D 0
P;
C;
3
3 H
H D -1
D -1 P 1 2 3 4 5 6
C 1 2 3 4 5 6
Channels: 1,3,5 ---> PDH1,2,3-law; 2,4,6 ---> PDS1,2,3-law
Channels: 1,3,5 ---> CDH1,2,3-law; 2,4,6 ---> CDS1,2,3-law
Fig. 6. A graphical representation of the correlation coefficients
Fig. 4. A graphical representation of the correlation coefficients
for the signals PD**-low
for the signals CD**-low

Correlation Coefficients for the LP Signals, PNH*-low and PNS*-low

A similar representation of the correlation 1
coefficients matrix for signals CN**-low obtained 1
from data measured in a crossroad (C) by night (N), S
N 0.5
can be seen in fig. 5. P;
1
H
Correlation Coefficients for the LP Signals, CNH*-low and CNS*-low N 0
1 P 1 2 3 4 5 6
1 1
S
N 0
C; 2
1
S
H N 0.5
N -1 P;
C 1 2 3 4 5 6 2
1 H
N 0
2 P 1 2 3 4 5 6
S
N 0 1
C;
2 3
H S
N -1 N 0.5
C 1 2 3 4 5 6 P;
1 3
3 H
S N 0
N 0
C;
P 1 2 3 4 5 6
3 Channels: 1,3,5 ---> PNH1,2,3-law; 2,4,6 ---> PNS1,2,3-law
H
N -1
C 1 2 3 4 5 6
Channels: 1,3,5 ---> CNH1,2,3-law; 2,4,6 ---> CNS1,2,3-law Fig. 7. A graphical representation of the correlation coefficients
for the signals PN**-low
Fig. 5. A graphical representation of the correlation coefficients
for the signals CN**-low
which is weakly but still positive correlated with
the other signals.
As shown in fig. 5, there is no correlation between the
signals CNH*-low. The uncorrelatedness of these
signals proves the absence of diurnal repetition of the V. CONCLUSIONS
concentrations measured with Horiba instrument. The
signal CNS1-low is negative correlated with all other The pair of signals measured in a place with low air-
sequences. However, the signals CNS2 and CNS3 are pollution level (park), are, with some exceptions,
highly positive correlated, denoting good diurnal positively correlated. In such environments, the
repetition of the pollution level measured with the measured signals put into evidence diurnal repetition
Siemens-Hawk analyser. of the concentrations levels.
In fig. 6 the correlation coefficients between the The signals measured with Horiba instrument in a
signals PD**-low are presented. The coefficients have crossroad (high polluted medium) are practically not
moderate positive values with a single exception: the correlated, denoting absence of diurnal periodicity. In
signal PDS3 (number 6) follows a decreasing such environment, the CO-concentrations measured
tendency, in contradiction with the other five with the Siemens devices, show diurnal repetition.
sequences. This cold be explained by the fact that the
Fig. 7 shows the correlation coefficients concentrations measured by the Siemens instruments
computed for the signals PN**-low. All coefficients are average values, i.e. signals in which the low
are positive. One can remark very good correlation frequencies (denoting diurnal periodicity) are
between all signals. An exception is the signal PNH3 accentuated.

242
 REFERENCES
[1] * * * ROSE (Remote Optical Sensing Evaluation) EU Project
G6RD/CT/2000/00434, (2001-2004).
[2] Bisorca D., Ionel Ioana, Popescu F., Ionel S., Ungureanu C.,
Air Quality investigation by means of remote sensing, with
application to CO thermodynamic measurements in the city of
Timioara, 13-th International Conference on Thermal
Engineering and Thermogrametry (THERMO), Budapest, 18-20
June 2003, pp.274-279.
[3] Ionel Ioana, Ionel S., Beurteilung von Luftqualitt mittels
optischen Fernmessystemen, in Vergleich zu der ND-IR Methode,
VDI Optische Technologien, 4. Konferenz ber Optische
Analysenmesstechnik in Industrie und Umwelt, 7-8 Oktober 2004,
Dsseldorf.
[4].Hoffmann J., Ionel S., Signakonditionierung mit Wavelet-
Techniken, Horizonte, Nr.24, Juli, 2004, pp.16-19.
[5] A. Papoulis, Probability, Random Variables and Stochastic
Processes, Third Edition, McGraw-Hill, Inc., New York, 1991.

Acknowledgement
The research has been carried out in the frame of the
ROSE Project, contract NB GR6D-CT2000-00434,
founded by the European Commissions Competitive
and Sustainable Growth Program.

243
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Analysis of biomolecular sequences through

spectral based methods
erban Mereu 1
Abstract In this paper we apply a few computational location n [1]. For example, the string ACCTG has
and visual tools, specific to digital signal processing, to N = 5 , u A [0] = 1 , uT [0] = 0 , u C [2] = 1 and
the analysis of biomolecular sequences. In particular, we
prove that color spectrograms can help in visually u C [3] = 0 .
identifying protein coding areas of the DNA strand and The four binary indicator sequences uniquely
provide, in the form of local "texture", significant determine the character string corresponding to a
information about biomolecular sequences, thus DNA segment. For each n, three of the four sequences
facilitating the understanding of local nature, structure take the value of 0 and one takes the value of 1. They
and function. We also show that the magnitude of a
properly defined function in the spectral domain can be
are a redundant, linearly dependent set of sequences
a predictor for the existence of protein coding regions in because, for all n,
DNA sequences.
Keywords: DNA spectrograms, frequency-domain u A [ n] + u T [ n] + u C [ n ] + u G [ n ] = 1 . (2)
analysis, genome analysis.
A. Discrete Fourier Transform
I. INTRODUCTION The Discrete Fourier Transform (DFT) of a sequence
x[n] , of length N, is itself another sequence X [k ] , of
The mathematical treatment of macromolecular
the same length N:
biological sequences corresponding to chains of
nucleotides or amino acids is usually done by 2 k
N 1 j
considering such sequences to be strings of characters X [k ] = x[n] e N
n
, k = 0,1,..., N 1 (3)
like A, T, C and G. If, however, we assign a n =0
numerical value to each of these abstract characters,
then such sequences become numerical and amenable The sequence X [k ] provides a measure of the
to digital signal processing. In this paper, we
frequency content at frequency k, which
demonstrate that digital signal processing of
N
numerical biomolecular sequences can provide a set corresponds to an underlying period of samples,
of novel and useful tools in the field of k
bioinformatics. where the maximum frequency (period 2) corresponds
In the case of DNA segments, assume that we N
to k = , assuming that N is even.
assign the number a to the character A, the number t 2
to the character T, the number c to the character Using the definition in (3), the resulting
C, and the number g to the character G. In sequences U A [k ] , U T [k ] , U C [k ] and U G [k ] are the
general, a, t, c and g can be complex numbers. DFTs of the binary indicator sequences u A [n] , uT [n] ,
The numerical sequence resulting from a
character string of length N can be written as: u C [n] and u G [n] , respectively.
If we assign numerical values a, t, c, and g, then
x[n] = au A [n] + tuT [n] + cu C [n] + gu G [n] , from (1) and (3) it follows that:
(1)
n = 0,1,2,..., N 1
X [k ] = aU A [ k ] + tU T [ k ] + cU C [ k ] + gU G [k ] ,
(4)
where u A [n] , uT [n] , u C [n] and u G [n] are the binary k = 0,1,..., N 1
indicator sequences, which take on the value of either
1 or 0 at location n, depending on whether the In the case of pure DNA character strings (i.e.,
corresponding character exists or not, respectively, at without assigning numerical values), the sequences

1
Facultatea de Electronic i Telecomunicaii, Catedra de Telecomunicaii
Bd. Carol I nr.11, 700506, Iai, e-mail: smereuta@etc.tuiasi.ro

244
U A [k ] , U T [k ] , U C [k ] and U G [k ] provide a four-
dimensional representation of the frequency
spectrum of the character string. The quantity:
2 2 2 2
S[k ] = U A [k ] + U T [k ] + U C [k ] + U G [k ]
can be used as a measure of the total power spectral
content of the DNA character string at frequency k.
From (2) and (4) it follows that:
0, k 0
U A [k ] + U T [k ] + U C [k ] + U G [k ] = . (6)
N , k = 0
Therefore, we can reduce the dimensionality of
the frequency spectrum representation from four to
three, reflecting the same property of the binary
indicator sequences [2]. One way to do this is simply
to ignore one of the four frequency components. If we
wish to reduce the dimensionality in a manner that is
symmetric with respect to all four components, we
may adopt the technique [3], in which three numerical
sequences x r , x g , and xb are defined from the
corresponding coefficients ( a r , t r , c r , g r ), ( a g ,
t g , c g , g g ), ( ab , t b , cb , g b ), in the following way:
the four three-dimensional vectors have magnitude
equal to 1 and point to the four directions from the
center to the vertices of a regular tetrahedron,
corresponding to the four DNA bases.
For example, we can choose:
2 2 1
( a r , a g , ab ) = (0,0,1) , ( t r , t g , t b ) = ( ,0, ) ,
3 3
2 6 1
( c r , c g , cb ) = ( , , ) , ( g r , g g , g b ) =
3 3 3 C. elegans (GenBank Accession number NC 000967 -
2 6 1
= ( , , ) , resulting in
3 3 3
2
x r [ n] = (2uT [n] u C [ n] u G [n])
3
6
x g [ n] = (u C [n] u G [n]) , (7)
3
1
x b [ n] = (3u A [n] uT [n] u C [n] u G [n])
3
from which we can find three DFTs: X r [k ] , X g [k ]
and X b [ k ]
B. Short Time Fourier Transform
Instead of evaluating the DFT of a full-length
sequence, we have the option of evaluating the DFTs
of several of its subsequences. This strategy makes
sense particularly in the case of long sequences
consisting of several segments with different
characteristics.
For example, we may apply a sliding window
of length L to a sequence of length N, where N > L ,
resulting in a sequence of DFTs. Each of these
DFTs provides a localized measure of the frequency
content, and is an example of a location-dependent
(5)
Fourier transform, known as the short-time Fourier
transform (STFT).
II. DNA SPECTROGRAMS
The display of the magnitude of the STFT is called a
spectrogram and it has long been used in the analysis
of speech signals. The appearance of the spectrogram
visually provides significant information to a trained
observer about the local nature of the sound signal.
Similarly, we can use spectrograms to visually
provide information about the localized frequency
content of biomolecular sequences. In this case, to
maximize the information content of the spectrogram,
we can use color-coding to display the magnitudes of
all individual sequences U A [k ] , U T [k ] , U C [k ] and
U G [k ] simultaneously, rather than merely displaying
the overall magnitude as given in equation (5).
In this case, it is preferable to reduce the
dimensionality from four to three (retaining all
information content) using equations (7). We can then
create one colored spectrogram combining all three
spectrograms of the corresponding magnitudes by
color-coding red for X r [k ] , green for X g [k ] , and
blue for X b [k ] .
For example, Fig. 1 shows the spectrogram using
DFTs of length N = 60. The data come from a DNA
stretch of 4000 nucleotides from chromosome III of
http://www.ncbi.nlm.nih.gov/entrez).
Fig. 1. Color spectrogram of a DNA stretch.
The vertical axis corresponds to the frequencies
k from 1 to 30, while the horizontal axis shows the
relative nucleotide locations, starting from nucleotide
858001. The genomic annotations establish that the
DNA stretch contains three regions (C. elegans
telomere-like hexamer repeats) at relative locations
(953-1066), (1668-1727), and (1807-2028) [4]. These
three regions are well depicted as bars of high-
intensity values corresponding to the particular
245
frequency k = 10 (because hexamers -period 6-
N
correspond to = 10 ). Furthermore, the frequencies
6
k = 6 (corresponding to a periodicity of 10) and its
multiples, appear to play a prominent role in the
whole region of the 4000 nucleotides.
For comparison purposes, Fig. 2 shows the
texture of a spectrogram coming from a sample of
totally random DNA, i.e., in which each type of
nucleotide appears with probability 0.25 and
independent of the other nucleotides.

Fig. 3. Plot of the spectrum of a coding DNA region,

demonstrating peak at frequency k = N / 3.

REFERENCES
[1] R. Voss, Evolution of long-range fractal correlations and 1/f
noise in DNA base sequences, Physical Review Letters, vol.
68(25), p. 3805-3808, 1992.
[2] W. Li, T.G. Marr, K. Kaneko, Understanding long-range
correlations in DNA sequences, Physica D., vol. 75, p. 392-416,
1994.
Fig. 2. Color spectrogram of totally random DNA.
[3] B.D. Silverman, R. Linsker, A measure of DNA periodicity,
Journal of Theoretical Biology, vol. 118, p. 295-300, 1986.
[4] http://www.ncbi.nlm.nih.gov/entrez
III. PROTEIN CODING DNA REGIONS [5] J.-M. Claverie, Computational methods for the identification of
genes in vertebrate genomic sequences, Human Molecular
Genetics, vol. 6(10), p. 1735-1744, 1997.
Protein synthesis is governed by the genetic code [6] J.W. Fickett, Recognition of protein coding regions in DNA
which maps each of the 64 possible triplets (codons) sequences, Nucleic Acids Research, vol. 10, p. 5303-5318, 1982.
of DNA characters into one of the 20 possible amino [7] B. Alberts, D. Bray, A. Johnson, J. Lewis, M. Raff, K. Roberts,
P. Walter, Essential Cell Biology, New York, Garland Publishing,
acids (or into a punctuation mark, like a stop codon, 1998.
signaling termination of protein synthesis).
One of the most relevant and yet unsolved
problems in bioinformatics is to accurately and
automatically annotate sequences by identifying such
regions using gene prediction [5], [6]. It is clear [7]
that the total number of nucleotides in the protein
coding area of a gene will be a multiple of three.
N
The frequency k = is of particular
3
importance for protein coding DNA regions because it
corresponds to a period of three samples, equal to the
length of each codon (triplet of nucleotides).
We now show how frequency-domain analysis of
DNA sequences can be a powerful tool for
specifically identifying protein coding regions in
DNA sequences. In Fig. 3 we have plotted the
sequence S [k ] , as defined in (5), for a coding region
of length N = 1320 inside the genome of the baker's
yeast (formally known as S. cerevisiae),
demonstrating a peak at frequency k = 440 ( = N 3 ).
This peak confirms the genetic findings reported for
S. cerevisiae [4].

246
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

K - complex Detection using the Continuous Wavelet

Transform
Ctlin I. Dumitrescu1
Abstract The wide variety of waveform in
EEG signals and the high non-stationary nature of many
of them is one of the main difficulties to develop
automatic detection system for them. In sleep stage
classification a relevant transient wave is the K-complex.
This paper comprehend the developing of two
algorithms in order to achieve an automatic K-complex
detection from EEG raw data. These algorithms are
based on a time-frequency analysis and two time-
frequency techniques, the Short Time Fourier
Transform (STFT) and the Continuous Wavelet
Transform (CWT), are tested in order to find out which
one is the best for our purpose, being of two wavelet
functions to measure the capability of them to detect K-
complex and to choose one to be employed in the
algorithms. The first algorithm is based on the energy
distribution of the CWT detecting the spectral
component of the K-complex. The second algorithm is
focused on the morphology of the K-complex waveform
after the CWT. Evaluating the algorithms results reveals
that a false K-complex detection is as important as real
K-complex detection.
Keywords: wavelet, k-complexe, STFT.
INTRODUCTION
Since the discovery of the
Electroencephalogram (EEG) by the German
psychiatrist Hans Berger in 1924 extensive studies
about electrical activity of the human brain have been
carried out. One of these studies correspond to sleep
stage classification. In the last twenty years several
researches and significance advances have been made
in the field of automatic sleep stage classification
since it is one of the diagnostic tools needed for
assessment of a number of sleep disorders. Automatic
sleep analysis is based on the detection of various
waveforms in the EEG and other bioelectric signals,
and inferring different sleep stages from the detection
of these waveforms. However, the strong non-
stationarity nature (transient phenomena) of EEG
signals has represented one of the main difficulties in
the developing of reliable systems for sleep
classification.
A non-stationary signal is defined as a short
time event whose frequency content vary in time. A
traditional analysis technique, for this kind of signals,
that provide an image of the frequency contents of a
signal as a function of the time is the time-frequency
analysis. Several methods or time-frequency
distributions can be used, for example the
spectrogram (Short Time Fourier Transform) which
calculate the power spectrum of the investigated
signal seen through a time windows function that slide
along the time axis. In this work we will concentrated
in another time-frequency distribution, the Continuous
Wavelet Transform (CWT). The CWT can be seen as
an operator that takes a signal and produces a function
depending of two variables: time and scale. In this
way the CWT is able to provide information of
features corresponding to the signal that are
dependent on the scale used. The scale-dependent
structure is strongly linked with the frequency content
of the signal giving to the CWT a great potential for
detecting and identifying signals with exotic spectral
features like transients behavior.
Detection of transient signals in
Electroencephalograms has been a subject of research
for several years. In sleep EEG one of the most
relevant transient signals is the K-complex. In
literature we have found a sort of methods and
algorithms for detection of K-complexes using Neural
Networks, feature based approach, independent
component analysis, adaptive filters, statistics
methods among others. In order to introduce the
reader in the K-complex detection field, we will give
a brief explanation about some studies which have
been carried out in this field.
In this report we will try to probe whether or
no using wavelet transform we can improve detection
of K-complexes. At the beginning of the last century
the Haar transform gave the first step in the wavelet
career, but this transform was not very used until early
eighties, when geophysicians, theorical physicians
and mathematicians developed a solid theory for
Wavelet. Since then, Wavelet has been used in several
applications, like signal processing, data compress,
time-frequency analysis, multirresolution analysis,
statistics, vibrations and many others.

1
graduate degree of Doctor of Electronics Engineering, University Politehnica of Bucharest.
Address: 1, Polizu St. sector 1, 011061, Bucharest, Romania, phone 0722-539019, mail catalindumi@ yahoo.com

247
 In the last fifteen years wavelet has been
widely used in EEG analysis as much as epilepsy and
Alzheimer diagnosis as sleep stage classification.
The main of this work is to extract
information from sleep EEG raw data about the
presence of K-complexes. We decided to work in the
time-frequency domain instead of either pure time
domain or pure frequency domain as previous works
in this field. In order to implement a time-frequency
analysis the Continuous Wavelet Transform will be
employed because it has been probe to be an efficient
tool in extraction of transient characteristics from a
collection of raw data. Therefore, the problem
statement of this work is to build and evaluate a K-
complex detection system using the wavelet transform
and, posteriorly, evaluate the algorithm performance
trying to find out possible important faults that may
affect the system.
1. Relevant Theory
This chapter will try to cover all the necessary
theoretical background in order to give the reader a
better approach to the sleep stage classification and
time-frequency analysis using wavelet transform. It
begins with the basic concepts of sleep classification
and a brief description of the bioelectrical signal
involved, particularly the electroencephalogram
(EEG). Then, an explanation of the relevant EEG
waveforms is given. As a first step toward a process
of EEG transient signal detection, the Fourier
Transform and the Short Time Fourier Transform are
explained. Finally, a review of the definition and basic
proprieties of the Continuous Wavelet Transform,
with the corresponding example and reason of why
this Transform will be used for time-frequency
analysis are given.
1.1 Sleep Analysis
Sleep analysis is a medical tool of vital
importance for the diagnosis and treatment of several
kinds of sleep disturbance and psychiatric or
neurological disorders. Today, a typical study of sleep
includes records of the muscle tone (EMG), of the eye
movements (EOG) and of the cerebral activity (EEG)
although depending on the clinical purpose other
physiological parameters like respiration, heart rate,
blood pressure, body temperature, hormonal
secretions are used. On the basis of such recordings a
certain number of sleep stage are distinguished by
criteria that have been standardized by general by
general agreement [Rechtschaffen and Kales, 1968].
1.2 Electroencephalogram (EEG)
The electroencephalogram (EEG) is a
bioelectrical signal that reflects electrical activity
emitted by neurons within the brain. This electric
recording from the brain activity show continuous
time-varying voltage oscillations with typical
amplitudes from 10 to 500 V and a frequency range
of from 0.5 to 40 Hz.
248
The study of EEGs has a long and fruitful history,
and I knew that due to my time and equipment
constraints I could not tackle the general problem of
EEG interpretation, so I restricted myself to a narrow
scope just to get a feel for the problem. The question I
posed for myself was this: is it possible to devise a
computer program which will analyze an EEG signal
and detect a particular waveform pattern? Sleep in
humans can be divided into two major categories:
Rapid Eye Movement (REM) sleep, and non- REM
(abbreviated NREM) sleep. REM sleep is
characterized by coordinated, darting movements of
the eyes as if scanning a scene, and is most correlated
with dreaming. NREM sleep on the other hand is
distinguished by its lack of eye activity. NREM sleep
is subdivided into four stages (Figure 1) with stage 1
being the lightest stage of sleep, sometimes
experienced by nighttime drivers who suddenly
realize theyve been driving for a few seconds in the
wrong lane, and stage 4 being the deepest stage of
sleep, characterized by total muscle paralysis and
insensitivity to external stimuli. The different stages
of sleep are distinguished from each other by the
predominant EEG waveforms at a given time in the
recording (Figure 2). Thus stage 1 is characterized by
so-called theta waves (between 4 and 7.75 Hz), stage
2 is composed of sleep spindles (14-15Hz) and K-
complexes, and stages 3 and 4 are composed of
primarily delta activity (mainly 4Hz). In the waking
adult, alpha activity is characterized by waves
between 8 and
13Hz, and beta rhythm is characterized by waves
greater than 15 Hz. I chose to study stage 2 of NREM
sleep, because the K complex can be easily
distinguished from the spindle signals, and because
data for stage 2 was already available.
Figure 1: Stage of sleep during the course of the night
Figure 2: EEG waveforms in various stages of sleep,
for young and elderly subjects.
K-complexes are relative large wave with a
duration that should exceed 500 milliseconds. In sleep
analysis, the scoring of stage 2 is evidenced by the
presence of one or more. This EEG waveform have a
well-outlined negative sharp wave, immediately
followed by a positive component. Before and after a
K-complex there is a period of low amplitude which
is useful to distinguish the K-complex from Delta
activity [Bankman 1992], [Didier 1994].
1.3 Time-Frequency Analysis
Short-Time Fourier Transform (STFT)
The STFT is a time-frequency tool that consists
of a Fourier transform with a sliding time window.
The time localization of frequency components is
obtained by suitably pre-windowing the input signal.
energy calculated and second before and after the K-
complex. The first criterion, about the frequency
range, was settled using the LabView Based on
literature [Mallat, 1998], [Kaiser, 1994], [Polikar,
1996] the most used wavelets for time-frequency
analysis have been Mexican Hat and Morlet wavelet.
Consequently, these two wavelet were chosen for
further analysis. The Mexican hat function is the
t2
second derivative of the Gaussian function e 2
and
is:
1 t2
2

The STFT is defined as follows:

= 4
(1 t )e
2 2
(4)
M 1
3
S x [n, k ] = x[m]w[m n]W
m =0
km
M (1)
The Morlet function is a complex wavelet. The
wavelet transform of a real signal with this complex
2
wavelet is plotted in modulus-phase form, however, in
j this work just the real part will be used. Morlet
where, W N = e N
, j= 1 , x is the input wavelet is:
signal, w is the analysis window, k is the frequency t2

offset, and m is the time delay [Qian 1996]. =e 2
e j 5t (5)
being its real part as:
Continuous Wavelet Transform (CWT)
t2
It is defined as the sum over all the time of the
signal multiplied by scaled, shifted versions of the Re[ ] = e 2
cos(5t ) (6)
wavelet function g.. Given a finite energy signal x(t)
and a normalized sampling period , Ts = 1 we can
present a discrete wavelet analysis of the sampled
sequence x[n] = x(t ) t = nTs , n Z as follows:
Table 1. Scale range and its corresponding pseudo-frequency range
for both Mexican hat and Morket wavelet.

(t )dt = 0

L2 ( R ) (2)
After determine which wavelet use, the next step
The discrete synthesis operation can be presented as was to settle the location in time of the K-complex
follows: within its respective 10 seconds epoch signal and its
respective time duration T. The K-complex interval T
1 t b
, f (a,b) = , f (a,b) = f (t) is the value which must be equal or greater that 0.5
*
CWT dt (3)
a a seconds and equal or lower than 1.5 seconds (see fig.

3).
where, l , k (a, b) = f , a ,b (t ) [Oppenheim and Posteriorly, the CWT was computed and from the
Shafer, 1989]. absolute values of the obtained coefficients matrix,
the highest value in amplitude and its respective
2. Methods and Implementation frequency value were looked assuming that this
2.1 Wavelet Selection frequency is the corresponding spectral component of
In order to choose the wavelet that will be the K-complex. The wavelet coefficients
employed in the K-complex detection algorithm, corresponding only to this spectral component will be
criteria based on how the wavelet spreads the signal called line of frequency. Consequently, using the
energy in time was developed. Thus, the chosen signal extracted from this line of frequency, as it is
criteria were based on two main points: depicted in the right illustration on figure 4.
1. The K-complex frequency range is from 0.5
Hz to 3.5 Hz.
2. A K-complex has to have a notorious
amplitude difference between the K-complex
energy and the energy registered and second Figure 3. K-complex time period T.
before the K-complex and one second after
it. This criterion tries to make the distinction
between a K-complex and the burst of delta
activity.
Based on these criteria, the best wavelet for the
detection algorithm will be that which give the biggest
difference the energy of the K-complex and the

249

Figure 4. Continuous wavelet transform (absolute value) of the K-
complex shown where the maximum amplitude correspond to the
pseudo frequency content of the K-complex.
2.2 Algorithm Design
As K-complex are transient phenomena from
EEG an algorithm will be developed in order to
achieve an automatic detection of these transient
signals. The algorithm will be based on time-
frequency analysis searching the manner of how
quantifies the energy distribution of K-complex in the
time-frequency plane. To develop this algorithm the
CWT will be employed because this tool has
demonstrated a good performance in transient
detection and feature extraction in several previous
works [Bailey, 1998], [Schiff, 1994]. Employing
some of the same parameters used in the wavelet
selection process, the design of this K-complex
detection algorithm will be based on the Energy
Distribution of the K-complex in the time-frequency
plane using the CWT. The wavelet employed in this
algorithm will be the Mexican Hat wavelet function.
As in the wavelet selection procedure, the
frequency criterion was based on theory assuming that
a K-complex has a frequency range between 0.5 and
3.5 Hz. The pseudo-frequency range obtained was
splitted into 17 pseudo-frequency values which were
used to calculate the CWT. The scale and pseudo-
frequency range are in table 2. The number selected to
split the pseudo-frequency range was established
basically in order to obtain an acceptable resolution in
the time-frequency representation, without
compromises the time performance of the algorithm.
Table 2. Scale to frequency transformation using the Mexican
Hat wavelet.
The energy distribution criteria were carried out
taking a 10 seconds epoch signal with a single clear
K-complex and computing the energy value the
frequency line belonging to the highest value found in
the CWT matrix of that signal. As we defined in the
wavelet selection criteria, the pseudo-frequency line
250
corresponding to the highest absolute value in the
CWT matrix, will be the K-complex spectral
component. This was probed by comparing the
Fourier transform of the original signal with the
Fourier transform of the frequency line corresponding
to the maximum value found in the CWT matrix. As
is illustrated in figure 5 we can see that the CWT
pseudo-frequency line obtained, the energy per on
second was computed having a result of ten energy
value per epoch. To calculate the energy per one
second E, intervals of 200 samples were taken
(because the original signal is sampled at 200 Hz, 1
second contain 200 samples) computing the energy
as:
200 2
E = si , si = i location sample (7)
i =1
Using the K-complex database an attempt to find
a common behavior of the energy in the presence of a
K-complex was tried.
Figure 5. Left top: K-complex in a ten seconds epoch from
EEG; Right top: CWT for scale 57.00 that correspond to the
pseudo-frequency of 0.88 Hz, it can be seen how the wavelet try to
assimilate the shape of the K-complex. From this signal the energy
value was computed; Left bottom: Fourier transform of the K-
complex, the highest, amplitude correspond to 0.88 Hz; Right
bottom Fourier transform of the CWT pseudo-frequency line.
Figure 6. Energy distribution
3. Results and Discussion
After finish the experimental test, the algorithm
performance was tested using the entire eight hours
EEG signal (channel 4 signal corresponding to the
record position Fp2-M1). Before start the test, a new
visual selection of K-complex was made. In this
classification we scored 235 K-complex along the
entire night. Before run the algorithm through the
entire night EEG signal, the obtained results were not
as satisfactory as we expect. A total number of 955
event were detected as k-complex. From the 235
previously identified K-complexes, a number of 179
K-complex were detected and 56 were not detected.
Therefore, based on this results a total number of 776
false K-complexes were classified as K-complexes by
the algorithm. The summarized results can be seen in
Table 3 and in Figure 7 and 8.
Table 3. Results of the algorithm performance
Figure 7. Pie chart plot that shows the percentage distribution
of table 3 (discrimination between K-complexes and other transient
signal).
Figure 8. Pie chart plot that describe the percentage
distribution achieved in the detection of real K-complexes only.
The algorithm performance has a good capacity
to exclude false K-complexes, but the main idea of
obtaining a good K-complex detection algorithm, and
at the same time, trying to minimize the number of
criteria used for the detection was to much restrictive
in the criteria number.
4. Conclusion
In this report we tried to cover the necessary
theoretical and practical topics in order to develop
different algorithms based on the Continuous Wavelet
Transform for K-complex detection on EEG signals.
A description of the sleep stage classification, Fourier
Transform, Short Time Fourier Transform and
Continuous Wavelet Transform was given. The STFT
and the CWT are two different tools with the same
aim: time-frequency analysis. Are their performance
are also different. Therefore, when time-frequency
analysis is required, we should be very careful about
the features of the signal to analyze, since for some
signals the STFT could be more appropriate than the
CWT and also in the other direction. For example in
signals with no transient content and a limited band
width, the STFT has a good performance and the
computation time is not large, but when there are
transient signals involved, the CWT becomes
necessary, and the computation time increase. Two
wavelets function were tested with the purpose to
obtain a quantitative description about how these two
different wavelets, Mexican hat and Morlet, are
capable to achieve a good K-complex detection taken
251
in account the morphology, frequency content, time
duration and power spectrum of the K-complex. From
this test, the most important conclusion we could
extract was that the wavelet capability in the detection
of K-complex has a strong dependence on the wavelet
waveform. Since the waveform of the wavelet has
probed to be an importance parameter for transient
signal detection we would like to left this field open
for further analysis based on other different wavelet
depending on the application they will be used. The
way to use the CWT was a precise bandpass filter
we could obtain a very narrow frequency band or only
one pseudo-frequency line without big distortion in
the signal shape.
We achieved a very good separation of
frequencies in a range 0.5 3.5 Hz (17 frequency
lines) and very good signal suppression in the exterior
from this frequency range. This feature of CWT was
implemented in both algorithms to detect K-complex
signals and was achieved a good results to detect
them. To know the real capacity of the algorithms to
detect K-complex, they were tested using a single
channel from eight hours EEG signal. From the
indices specificity, sensitivity and validity we
obtained very different results. The performance of
the algorithm based on the energy distribution was
relatively poor to make a good discrimination
between real K-complexes and false K-complexes.
The lack of enough criteria for K-complex detection
could be the answer of this poor performance. During
our experience we realized that the decision regarding
the detection of a K-complex may need to be
corroboration by a single consideration that we did
not take in account. This consideration is concerning
to the vicinity of sleep spindles and K-complexes.
Another interesting point to mention was the fact that
detection of K-complexes was based on the research
of only real K-complexes since from the results
obtained we realized that a more difficult task to carry
out would be the develop of accurate criteria in order
to achieve a better recognition between Delta activity
and K-complex. When looking in the false K-
complexes detected as K-complexes we realized that
is possible to find real K-complexes in this set of
signals. Almost all these signals are out of stage two,
and some of them just in the edge of a particular stage
two. This makes to use think that we found real K-
complexes in these signals, and a deeper investigation
should be made on this field. One possible reason for
this problem is that we only looked for K-complexes
in stage 2, since we did not find one single reference
about the existence of K-complexes out of stage 2.
Another reason is a possible not proper stage
classification. Even when all signals in question were
real K-complexes, the performance of the algorithm
will not be good enough, therefore, a criterion for
make the difference between K-complexes and Delta
waves is highly necessary in order to improve the
validity of the algorithms.
 REFERENCES

[Bailey et al. 1998] Bailey, T. C., Sapatinas, Powell (1998).

Signal detection in underwater sound using wavelets.
Journal of the American Statistical Association, 93: 73-83.
[Bankman 1992] Bankman L. N. (1992). Feature-based
detection of the K-complex wave in the human
electroencephalogram using neural networks. IEEE Trans.
On Biomedical Engineering, 39(12):1305-1310.
[Didier 1994] Didier, H. (1994). Comparison of detection
methods : application to K-complex detection in in sleep
EEG. Proceeding of the 16th Annual International
Conference of the IEEE, 2:11218-1219.
[Kaiser, 1994] Kaiser, G. (1994). A Friendly Guide to
Wavelets. Birkhauser. ISBN: 0-8176-3711-7.
[Mallat, 1998] Mallat, s. (1998). A Wavelet Tour of Signal
Processing. Academic Press. ISBN: 0-12-466606-X.
[Oppenheim and Shafer, 1989] Oppenheim, A. and
Shafer, R. (1989). Discret time signal processing. Pretince
Hall. ISBN:0-13-216292-X.
[Policar, 1996] Polikar, R (1996). The wavelet tutorial.
URL:http//engineering.rowan.edu/%7polikar/wavelets/wtut
orial.html
[Qian 1996] Qian, S. (1996). Joint Time-Frequency
Analysis, Methods and application. Pretince Hall. ISBN:0-
13-254384-2.
[Rechtschaffen and Kales, 1968] Rechtschaffen, A. and
Kales A. (1968). A manual of standardized terminology,
techniques and scoring system for sleep stage of human
subjects. Technical reports, Washinton, DC, Public Service,
US Gov. Printing Office.
[Schiff 1994] Schiff, S T, Unser A. (1994). Fast wavelet
transformation of EEG. Electroencephalography and
Clinical Neurophysiology, 91(6):442-455.

252
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

PC-based system for automated calibration

of a digital voltmeter
Dan Stoiciu, Mihaela Lascu1
Abstract The paper presents a system built around a
PC and the application in LabVIEW for automated
calibration of a digital voltmeter against a high accuracy
voltmeter. The devices communicate within the system
through standard interfaces RS 232 and IEEE 488.
Keywords: automated calibration, standard interfaces,
virtual instrument.
I. INTRODUCTION
Calibration of a digital voltmeter (DVM) must be
done against an etalon that can be either a voltage
calibrator or a high accuracy digital voltmeter. In the
second case a voltage source is also needed. In both
cases several voltage values have to be measured and
compared with those indicated by the etalon. These
operations can easily be done automatically by using
an appropriate experimental setup that consists of a
PC, a programmable power supply, the DVM to be
calibrated and a high accuracy voltmeter.
II. DESCRIPTION OF THE SYSTEM
The system that has been developed for the calibration
of a DVM consists of a PC, a programmable power
supply, the DVM to be calibrated and a high accuracy
voltmeter (fig. 1).
An application (virtual instrument - VI) has been
developed in order to control all the devices and to
accomplish the required tasks. The inputs of this VI
are: the voltage range of the DVM to be calibrated, its
number of digits, the formula given in its specification
for calculating the maximum permissible error, the
number of points (voltage values) and the number of
times the measurements should be repeated at each
point in order to obtain an averaged result. The tasks
to be accomplished by the system are described in the
following:
Step 1 The VI calculates the first voltage value and
sends it to the programmable power supply.
Step 2 After a convenient delay the readings of the
two voltmeters are sent to the PC. Step 2 is repeated
the required number of times, and the readings are
averaged. The averaged results are output in a table,
and the error is calculated and compared to the
maximum permissible error.
Steps 1 and 2 are repeated for the required number of
points.
In the end, the operator sees a table with the averaged
readings of the two voltmeters, the actual error and
the maximum permissible error. Additionally, an
array of LEDs indicates whether and at which point
the error exceeds the limit.
The front panel of the VI is shown in fig. 2.

RS232
DVM DVM (standard)
PC
Power supply

IEEE 488

Fig.1. System schematics.

The power supply and the high accuracy voltmeter are

connected to the PC via the IEEE 488 parallel
interface [1]. For the DVM to be calibrated a 41/2 digit
Voltcraft 4650 CR with built-in RS 232 serial Fig. 2. Front panel of the VI.
interface was considered.
1
Facultatea de Electronic i Telecomunicaii, Departamentul de Msurri i Electronic Optic,
Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail dan.stoiciu@etc.utt.ro, mihaela.lascu@etc.utt.ro

253
A part of the block diagram is shown in fig. 3. The block diagrams of the main VI and of the subVIs
have been supplied with error handling blocks. These
blocks are not necessary if the system operates
properly, but if the application stops, it is very
difficult to find out why. The general structure of
such a block is presented in fig. 4.

Fig. 4. Error handling block.

III. EXPERIMENTAL RESULTS

Fig. 5 presents the front panel of the main VI after

Fig. 3. Part of the block diagram of the main VI. running the application.

A hierarchical approach has been considered,

consisting of a main VI, subVIs, subVIs of the subVIs
and so on. The main VI consists of a two-frame
sequence structure that implements steps 1 and 2
described above.
In step 1 a voltage value is calculated and sent to the
programmable power supply by calling the
hm8142.vi. This is a subVI that manages the
communication with the power supply via the IEEE
488 parallel interface [1].
Step 2 consists of a three-frame sequence structure:
i. a 3-second delay to allow the DVMs indications
to settle
ii. the value measured by the high accuracy
voltmeter is input to the PC
iii. the value measured by the DVM to be calibrated
is input to the PC.
Frames ii and iii are then repeated the desired number
of times (for obtaining averaged measurements). The Fig. 5. Front panel of the main VI
averaged results are then written into the main table of after running the application.
the front panel. Steps 1 and 2 are repeated for the
required number of points. A printed report can be easily generated.
In frames ii and iii two other subVIs are called:
hp3455.vi and voltcraft.VI, respectively. These IV. CONCLUSION
subVIs manage the communication between the PC
and the two voltmeters, by means of parallel and The VI presented in this paper eases very much the
serial interfaces, respectively. job of the human operator. It also gives way to
In the main VI the maximum permissible error for implementing an e-learning application, namely
each voltage value to be measured is also calculated conducting the experiment from a remote location [3].
and written into the main output table on the front
panel. The actual error is obtained by subtracting the REFERENCES
two measured voltage values after averaging. The [1] S. Mischie, Interfee pentru instrumentaie programabil.
actual error is also written in the main table on the Standarde i aplicaii, Ed. Politehnica, Timioara, 2004.
front panel. If the actual error is less than the [2] http://www.ni.com
maximum permissible error, the corresponding LED [3] D. Stoiciu, C. Dughir, A web-based teaching tool for
laboratory classes, Symposium on Electronics and Telecommuni-
on the front panel will be powered on. A non powered cations, Timisoara, 2004
LED indicates the voltage value for which the
maximum permissible error has been exceeded.

254
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

The measurement of dc magnetic field

Alimpie Ignea1, Adrian Mihiu2
Abstract The measurement of dc magnetic field,
especially for low intensities, returns in actuality.
Beyond the importance of measuring the terrestrial
magnetic field with his variations, a large amount of new
standards concerning generally the electromagnetic
compatibility impose as well the measurement of the dc
magnetic field. For example, in [1] the dc magnetic field
is introduced as the reference magnitude, and in [2] is
the reference level established for the magnetic field
concerning the surrounding environment, which assures
the staff protection.
The paper presents a measurement method of the
magnetic field induction based on the Frster probe and
the obtained experimental results.
Keywords: measurement of dc magnetic field, Frster
probe, nonlinear circuit
I. INTRODUCTION
The dc magnetic field represents a particular
form of the electromagnetic field, which can be
emphasized nearby the permanent magnets or circuits,
where there is a dc current flow through, and which
exercise forces and moments upon neighborhood
bodies. The ideal medium concerning the magnetic
field is the vacuum. Between the magnetic field
intensity and the magnetic induction it is possible to
write the following relationship:
B = 0 H (1)
Where: 0 = 4 10 7 is a universal constant factor
of proportionality representing the magnetic
permeability of the vacuum.
The measurement of the magnetic field represents a
delicate task, because of the field produced by the
temporary or permanent magnetization at the
interacting bodies level, taking into account the
material characteristics. This is the reason, why it is
difficult to realize such probes for picking up
information concerning this kind of field.
Another problem that appears during the
measurement process is the existence of the magnetic
terrestrial field, which is, as well, a dc field. That
means, when measuring the dc magnetic field, the
1
Facultatea de Electronic i Telecomunicaii, Departamentul de Masurari si Electronica Optica,
Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail alimpie.ignea@etc.utt.ro
2
Facultatea de Electronic i Telecomunicaii, Departamentul de Masurari si Electronica Optica,
Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail adrian.mihaiut@etc.utt.ro
magnetic terrestrial field value in the measuring point
appears like a systematic error related to the measured
value. For correcting this error is it necessary to know
previously the value of this field, thats why it is
necessary measuring it.
In literature [3], there are some measurements
methods presented, one of them is the measurement
using the Frster probe. As a principle, this type of
transducer uses the specific behavior property of
ferromagnetic materials taking into accounts a
simultaneous magnetization generated by ac and dc
fields. In the situation of introducing a magnetized
ferromagnetic element in an ac magnetic field,
overlapped by a dc magnetic field, this produces a
changement of the magnetization behavior on an
asymmetric hysteresis cycle, leading into the
generation of the harmonic component second order,
having a level proportional with the ac signal
amplitude and the dc component magnitude of the
resulting magnetic field.
II. MEASUREMENT METHOD PRINCIPLE
It is much easier to obtain the second order
harmonic component, when the transducer is with two
parallel cores differential variant - having the
principle schema presented in fig.1; on each of the
two cores we have the same numbers of windings,
N1 N2
* *
N1 N2
* *
Fig.1. Frster Probe
N1 and N2, respectively N2 and N2. Windings
N1 and N2 named, as well as, supplied windings
are connected in opposition, so that in every moment

255
the generated ac fluxes generated by these to be equal
and of contrary signs.
The measurement windings N1and N2 are
phased series connection; with such a configuration,
the odd harmonics will be reciprocally canceled, and
the even will add together, which is enhancing the
signal to noise ratio, also doubling the probe
sensitivity.
If we consider that the dependency between the
magnetic induction (for a saturated core, fig. 2) and
the magnetic field intensity is:
B = a1 H a 3 H 3
I determinated by the following formula:
3
B = a1H a3 H
H Where: Pa is the transformer apparent power, K
Fig. 2. Hysterezis characteristic.
Where a1 and a2 are parameters with a1 > a 3 and the
intensity of magnetic field H comes from a constant
field H0 and a variable field generated by a sinusoidal
current I sin t .
It results that in the two inductors we will have
the following magnetic inductions:
B1 = a1 (H 0 + N 1 I sin t ) a 3 (H 0 + N 1 I sin t )3
B 2 = a1 (H 0 N 1 I sin t ) a 3 (H 0 N 1 I sin t )3
Because the secondary windings are series in
phase, at their inputs, we will obtain the induced
voltage having following form:
d ( )
u= = S [ a1 (N 1 N 2 N 1N 2 )I cos t +
dt
+ 3a 3 H 02 (N 1 N 2 N 1N 2 )I cos t + (4)
( 2 2
)2
+ 6a 3 H 0 N 1 N 2 + N 1 N 2 I sin t cos t +
( 3
)
+ 3a 3 N 1 N 2 N 1 N 2 I sin 2 t cos t
3 3
] presented in fig. 3. The signal generator SG, delivers a
Where S is the probe surface.
If the primary and respectively the secondary
windings are identical (N 1 = N 1 = N 1 ) and
(N 2 = N 2 = N 2 ) the obtained output voltage
becomes:
u = 6a 3 SN 12 N 2 I 2 H 0 sin 2t
It is worth to make some observations
concerning relationship (5).
1. The non-symmetries between the winding lead
to the fundamental and third order harmonic
component appearance that can be eliminated using a
band pass filter centered on the second harmonic.
2. The sensitivity of the method is rising with the
number of turns belonging to the two inductors,
exciting ca signal frequency and amplitude and
obviously depends on the used core, taking into
account the section S and the coefficient a3, that
characterizes the core saturation.
(2) 3. It is known from the electrical transformer
theory [4] that the core section can be saturated if it is
Pa
S<K (6)
f
constant depending on the construction core material,
and f frequency.
If for a first approximation we are considering
that the apparent power is proportional to I2 and
taking into account relationship (6), it results that the
obtained output voltage will be:
U = ka 3 f N 12 N 2 I 3 H 0 (7)
Where k is a constant.
Finally, it results that for having a raised
sensitivity it is necessary to raise:
a) the excitation current or
b) the turns number of the windings, especially
the primary winding (otherwise this helps in obtaining
(3)
a saturated core). The frequency increasement
contributes a little in sensitivity increasement and
produces as well the leakage in magnetic material
increasement.
III. THE MAGNETIC INDUCTION
MEASUREMENT PROBE CALIBRATION
Following the upper conclusions, a Frster probe
with permaloy core, having a section of
approximately 3 mm2, has been realised.
Measurement schematic of a dc magnetic field is
B
SG V
C
Probe
Fig.3 Measuring schematic with selective voltmeter
(5)
sinusoidal signal with fix or adjustable frequency and
is connected to the Frster probe through the capacitor
C. It is important that the input probe signal is purely
256
sinusoidal, without distortions or dc component, sensitivity stay near constant because the loss in
because the presence of this dc component could lead
U2[mV] car frecv car frecv
to an additionally magnetization of the core and
therefore to the appearance of measurement errors.
The selective microvoltmeter V, has at the signal 200
input a band-pass filter with a rejection factor with
respect to the central frequency, that can be adjusted 150
two level, 25 dB respectively 40 dB, concerning the
desired selectivity extraction of the measured signal. 100
The input voltmeter signal can be measured directly
or passed through the filter that can be adjusted for the 50
desired frequency. As well, at the output of the f[Hz]
0
voltmeter, an oscilloscope can be connected for the
visualization of the measured signal. 0 500 1000 1500 2000 2500
In this schematic, the selective voltmeter Fig. 4. The dependency between the output voltage and
frequency is adjusted to a double frequency in respect the frequency.
to the generator frequency, choosing in this way the
magnetic core is grown
measurement of the second order harmonic
In fig. 5 is represented the output voltage
component.
dependency with respect to the excitation current.
The measurement schematic calibration has been
done using the Helmholtz inductor; the Frster probe f=1500 f=1500
has been placed in the middle of the Helmholtz 150
inductors, with an east-west orientation, so that the dc
terrestrial magnetic field that operates upon the probe
to be canceled. The Helmholz inductors are supplied 100
by a known dc current, a constant magnetic field will
be generated, and the voltmeter tuned on the
frequency second order harmonic component, will 50
indicate a proportional value of the measured I[mA]
magnetic field.
In the calibration experiment have been used: 0
The magnetic field generator (Helmholtz 0 50 100
inductors), which has a constant kB = 596 T/A
known with an error = 0,5%. Fig. 5. The dependency between the output voltage and
Digital multimeter type M3650D, accuracy class the excitation current.
0,5 1 digit
Selective nanovoltmeter Unipan, type 237, It is possible to observe that around 50 mA, the
accuracy class 1,5. curve is a third order characteristic and when the
After the calibration the dc magnetic field current is greater than 50mA the curve becomes
measurement constant has been obtained: saturated.
The explanation of this effect consists of the
kB I fact, that at high magnetic field, the core is saturated
kc = = 0,476 T/mV (8)
U and the hysterezis curve can be considered like a
breaked line. In this case, for a sinusoidal magnetic
The measurement uncertainty will be equal with [5]: field, the magnetic induction becomes trapezium.
For a trapezium signal (fig.6), the second
I k k
2 2
cl I k I cl U
2 harmonic component has the amplitude [6]:
B = B B + B A + B2 V =
U 100 3 U 100 3 U 100 3 (9)
sin (2tc / T0 )
A T0
U2 = (10)
k I 1 2 2 tc
= B 2B + cl A2 + clV2 = 0,005 T/mV
U 100 3
Where, A is the amplitude of the trapezium signal, T0
That corresponds to an approximately deviation the period of the signal and tc rising time.
of 1%. If we have tcT0, the sinusoidal function is
In fig. 4 is represented the dependency between approximated with its argument and the expression
the output voltage and the frequency for a supply (10) becomes:
current and an external constant induction, revealing
the sensitivity dependency with respect to the square A
root frequency, but if the frequency is large, the U2 = (11)

257
 We can see that the second harmonic
amplitude is independent of period and rising time.
A especially for low intensities, based on the Frster
tc T0 t We observe that the frequency of the
Fig. 6. Trapezium signal
Now, we suppose that we have two
complementary trapezium signals, one of them
positive and the second one negative and a delay
of T0/2 between them, the first with an amplitude of
A+ and the second with an amplitude of A-.
In this case, the amplitude of the second
harmonic component will be:
2 GHz), www.acero.ro,
U 2 = (12)
As a conclusion it can be said that the
sinusoidal field applied to the Frster probe is large
enough and the output voltage is dependently only
on the continuous component.
258
IV. CONCLUSIONS
The measurement of a dc magnetic field,
probe is a very sensitive method.
To obtain a large sensitivity and small errors it
is necessary that the core of the probe is saturated.
sinusoidal magnetic field has to be of some kHz,
because at larger frequencies, the loss of the
magnetic material is bigger.
Referencies
[1] ***, EN 62052-11: General requirements test
and test conditions for electricity metering
equipment (AC),
[2] ***, 1999/519/EC, Recomandarea Consiliului
din 12 iulie 1999 privind reducerea expunerii
publicului la cmpuri electromagnetice(0-300
[3] Wiener, U., Masurari electrice, vol II, Ed.
Tehnica, 1969,
[4] Richter, R., Transformatorul electric, Ed.
Tehnic, Bucureti, 1960,
[5] Ignea, A., Stoiciu, D., Msurri electronice,
senzori i traductoare, Ed. Politehnica, Timioara,
2003,
[6] Ignea, A., Introducere n compatibilitatea
electromagnetic, Ed. de Vest, Timioara, 1998.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
The simulation of the effect of the geometry of the
Rectangular Double Barrier structure about the
transmission and reflection coefficient
Ana Rosu-Niculescu1, Teodor Petrescu2
Abstract - This paper is focused on the simulation of the
effect of the geometry of the quantum Rectangular
Double Barrier (RDB) structures about the transmission
and reflection coefficient of them, in view of obtained
electronics devices with good transmission
performances. The theoretical base of this study is
finding out in the Generalized Impedance Method
(GIM) and the practical simulation is realized using
PHP and JAVA scripts.
This simulation shows that, the transmission
performances of these devices can by easy modified by
the changing of the height and thickness sizes of the
potential barriers and quantum well.
Keywords: simulation, Rectangular Double Barrier
structures, transmission coefficient, reflection
coefficient, electronics devices
I. INTRODUCTION
Since the demonstration of the resonant tunneling in
semiconductor heterostructure, the Double Barrier
Resonant Tunneling (DBRT) structure has become a
generic device, stimulating a large number of
theoretical and experimental works [1-6]. The study
of the resonant states and the power transmission
coefficient across such potential barriers is important
both from a fundamental and a practical point of view.
Several methods have been reported in literature. The
Generalized Impedance Method (GIM) [1] and the
Complex Valued Equivalent Circuit Method
(CVECM) [3] treat the quantum effect devices by
using the analogy between the Schrdinger equation
and the equation voltage-current in the transmission
lines theory.
In this paper, the theoretical base is finding out in the
Generalized Impedance Method, witch is used to
study the transmission and reflection coefficient of the
GaAs-AlxGa1-xAs RDB structures with different
geometry sizes. This theoretical study is completed
with a practical simulation realized using PHP and
JAVA scripts, in which are calculated and graphic
represented the transmission coefficient, T, and the
reflection coefficient, , of the devices. The
parameters of the RDB structures (the thickness and
1
ORANGE S.A., Departamentul Transmisiuni, Bucure~ti, e-mail ana.rosu@orange.ro
2
Facultatea de Electronica si Telecomunicatii, Dept. Telecomunicatii, Bucuresti, e-mail teodor.petrescu@munde.ro
259
the height barrier) are introduced using a HTML fail.
In this way, you can obtain the quantum structure with
the wished transmission performances. This spectral
simulation is an interactive Java applet, in witch you
can set parameters, obtained from PHP calculation
script.
Making experiments with this simulator you can
predict the geometry of the RDB witch leads of the
structures with grate power transmission coefficient
and small reflections. The simulation shows that the
better choose is to use a structure with the thickness of
the potential barriers equal with them, the thickness of
the quantum well much grater of them and the small
height of structure. In this case, the T coefficient 1,
and the coefficient is approximated null.
Our results show that, the transmission performances
of these devices can by easy modified by the changing
of the height and thickness of the potential barriers
and quantum well.
II. THEORY
A. The Generalized Impedance Method applied
in Rectangular Double Barrier structures
To study the properties of the transmission in the
GaAs-AlxGa1-xAs quantum structures, we used the
GIM method to calculate the reflection and
transmission coefficients.
Let us assume that an electron with the energy E is
incident on the potential barrier
(Fig. 1).
V2
Incident electron
Reflected electron
Region 1 Region 2
V1
0 x
Fig. 1. A potential barrier with an incident electron
The complex conjugate of the wave functions can be
written as
 1* ( x ) = A1+ (e 1 x + e 1x ), x<0,
2* ( x ) = A2+ e 2 x , x > 0,
where
i = j (2mi / = 2 )( E Vi ) ,
mi and Vi (i=1,2) are the propagation constant,
the effective mass and the potential,
respectively, for the ith region; is the wave
amplitude reflection coefficient.
Obs.: For to obtained a better accurate of the
calculation, the effective mass mi is assumed to be the
different in both materials, depending of the Al
concentration, X, from the structure.
Based on this observation, mi is calculated thru:
mb = (0.0667 + 0.083 X ) m0 (4)
m w = 0.0667 m0
where
m0 is the effective mass of the free electron;
mw - the effective mass of the electron in the
quantum well;
mb - the effective mass of the electron in the
potential barrier.
Vi function of Al concentration, X, is:
V b = 0.57 (1.155 X + 0.37 X 2 ) . (6)
Using equation (1) we obtain relation (7):
(
i ( x) = j (= / mi ) (d i /d x) = Ai + e i x e i x w
ith
Z 0 i = j mi / i = . (8)
Let us now write the equation for voltage (U) and
current (I) used in the transmission lines with
generalized distributed impedance. These are:
U ( x ) = I + Z 0 (e x + e + x ) , (9)
+ x + x
I ( x ) = I (e e ),
where
Zl Z 0
= (11)
Zl + Z 0
is the voltage reflection coefficient, Zl and Z 0 are the
load and the characteristic impedances of the
transmission line, respectively. If we compare the
equations for i* and i and the corresponding
expressions for U and I for a transmission line we see
that these are analogous equations. Thus, we can
regard Z0 as the characteristic impedance of a region.
Also, the ratio of i* and i (analogous to the ratio of
the voltage and current), will define, at any plane x,
the quantum mechanical wave impedance
Z i ( x) = * i ( x) / i ( x) (12)
Thus, the input value of impedance, Zi = Z (l ) ,
may be expressed in terms of the load impedance
Zl = Z (0) [1] as:
Z l cosh( l ) + Z 0 sinh( l )
Zi = Z0
Z 0 cosh( l ) + Z l sinh( l )
(1) The method can be used for the arbitrary potential
barrier structures, so that we consider a
(2) symmetrical double barrier structure with a
rectangular quantum well (RDB), as shown in
figure 2:
(3)
GaAs AlAs GaAs AlAs GaAs
1 2 3 4 5
E2 Vb
E1
a b c d
l2 l3 l4
(5)
Fig. 2. Schematics of the resonant tunneling effect and resonant
transmission obtained for E = En, when electrons tunnel
resonantly into the nth bound state of the well
The characteristic impedance for the rightmost section
serves as the load and, because the region 5 is
theoretically infinite, Z l = Z 0,5 . Having the load
impedance at point d, the input impedance Zc at the
point c is calculated using relation (13), with
Z 0 = Z 0, 4 and l = l 4 . With Zc as the load at point c,
) the input impedance at point b is computed and this
process is repeated until the point a is reached. The
reflection coefficient is calculated using the equation
(11) with Za as the load impedance and Z 0,1 as the
characteristic impedance. The transmission coefficient
T (E ) is given by:
2
T ( E ) = 1 ( E ) . (14)
(10)
We have used this method in the PHP script from the
simulator of paper, for to obtained the better RDB
quantum structures which lead at electronic devices
with good transmission performances. Also, for to
realize a complete study, we have utilized the Smith
chart by using an interactive Java applet, in witch you
can set parameters, obtained from PHP calculation
script. The theory and the mode of work of this
interactive Java applet will be presented in the next
subsection.
B. The description of the interactive Java
applet
In any transmission system, a source sends energy to a
load, such as an antenna. Ideally, we design the
transmission network such that the characteristic
impedances of the source, the transmission line and
the load are all identical. Unfortunately, many real-
world situations prevent the match from being perfect.
For example, we might want an antenna (the load) to
be useful over a broad range of frequencies. But the
(13) characteristic impedance of an antenna is unlikely to
260
stay constant with frequency, especially if the
frequency span is great.
When the transmission line impedance does not match
that of the load, part of the transmitted waveform is
reflected back towards the source. The reflected wave,
which varies in phase and magnitude, adds to the
incident (transmitted) wave and the sum is called a
Standing Wave. The reflected wave causes the
amplitude to vary as a function of position along the
transmission line. The Standing Wave Ratio (SWR),
which is the ratio between the maximum and
minimum amplitudes of the total waveform, will in
this case be greater than one. If there is no reflected
wave, i.e., if the impedance match is perfect, the
amplitude of the total waveform (incident plus
reflected wave) will be the constant, regardless of
where we measure it along the transmission line. The
result is a SWR of 1. SWR = 1 indicates maximum
power transfer to the load. SWR can be inferred by
measuring the reflection coefficient of the circuit. The
network analyzer is a tool that enables us to do just
that. If we know the reflection coefficient, we can
determine the characteristic impedance of the load by
using a Smith Chart. The Smith Chart has circles of
constant resistance and arcs of constant reactance. The
relationship between reflection coefficient and
characteristic impedance is shown in the diagram (fig.
3).
Fig. 3. Position using the pointer device (mouse)
The Smith Chart can help us translate the reflection
coefficient into impedance. First, we calculated the
reflection coefficient using the PHP script. Place the
reflection coefficient, by using either the mouse or the
drop-down input boxes, at the desired value (real +
imaginary) on the Smith Chart. Hit the Play button
(triangle), and the program will display
(fig. 4) a circle with a radius equal to the reflection
coefficient magnitude (constant VSWR circle). Notice
that if you move the reflection coefficient anywhere
on this circle, you can see from the waveform at the
left that the SWR is the same, only its phase changes.
(Phase values are not shown around the chart in this
program; however, the phase is calculated and shown
at the left side of the screen.)
261
F
ig. 4. Drawing the VSWR circle.
In general, only the horizontal line (diameter) is
labeled with (normalized) resistance values and only
the unit (outer) circle is labeled with (normalized)
reactance values. To read the desired values, it is
necessary to follow the appropriate circle of constant
resistance to the diameter line, and to follow the
appropriate arc of constant reactance to the unit circle.
Hit Play again, and the program will display the
constant-resistance circle and the constant-reactance
arc for you (fig. 5). The actual values are calculated
and shown at the left side of the screen.
Fig. 5. Drawing the constant R circle and the constant X
arc
Experiment with the simulator and you will see that
you can predict the geometry size of the symmetric
RDB structure witch leads of the electronic devices
with grate power transmission coefficient and small
reflection.
III. THE SIMULATION DESCRIPTION AND
RESULTS
Since the our final goal is to obtain electronics
devices, with good transmission performances, from
RDB structures, this present simulator helps us to
calculate and graphic represent the transmission
coefficient and the reflection coefficient on the Smith
Chart. The mode of work of the simulator is present
step by step in following.
First, using a HTML file, we can introduce the
geometry size of RDB. Having these input dates, the
next step is to determine the transmission coefficient
and the reflection coefficient of the chosen structure.
These coefficients are obtained from PHP script. The
last step is the graphic representation of the
transmission coefficient function of the incident
electron energy (using another PHP script) and the
representation of the reflection coefficient on the
Smith Chart by utilizing the interactive JAVA applet
witch receives the dates from the first PHP script.
By realize more experiments with this simulator we
can anticipate the RDB geometry witch leads to
electronic devices with wishing transmission
performances.
Thus, we can observe that, if we chose an asymmetric
RDB structure (l2 = 20,
l3 = 100, l4 = 35 and X = 0.45 =>
Vb = 338.96 meV) it will obtained a devices with Fig.7. Reflection coefficient for the asymetric RDB structure
small transmission coefficient (T < 0.7
- fig. 6) and with grate reflection ( Ox axe of the
Smith Chart fig.7).

Fig.8. Transmission coefficient function of the incident electron

energy, T (E), for the symmetric RDB structure; Erez = 292 meV, T
Fig.6. Transmission coefficient function of the incident = 0.99942
electron energy, T (E), for the asymetric RDB structure; Erez =
287 meV

The better choose is to use a symmetric RDB, for

example, a structure with the thickness of the
potential barriers equal with them (l2 = l4 = 20) and
the thickness of the quantum well much grater of
them (l3 = 100). In this case, the T coefficient 1
(fig. 8), and the center of the VSWR circle is near the
center of Smith Chart (fig. 9).
From figure 8, we can see that using thus symmetrical
structures it obtained filters with good transmission
performances, witch permit to pass only the electrons
with energy, E = Erez. The asymmetric RDB structures
havent this function (fig. 6), being disadvantage from
this point of view. At this conclusion arrived thru the
Fig.9. Reflection coefficient for the symmetric RDB structure
studies from another papers [7-9], too. Thus, this
simulator proves that we can obtain electronics
In this moment, the Data Base contains 40 records
devices, with wished transmission performances, from
(table 1), but it can be up gradated and interrogated
RDB structures by simple changing of the geometry
using another PHP scripts. The task of the
size of them.
interrogation PHP script is to permit an easy
Using the analogue PHP script, we can create a Data
identification of the symmetric RDB structures witch
Base in MySQL, in witch it stockade the values of the
lead at the electronic devices (filters in THz range)
transmission coefficients and resonant frequencies of
with wished transmission performances.
the submilimetric filters obtained with symmetric
Using table 1, in figure 10 it represent the
RDB structures, for different height and thickness of
dependences: T (Vb), Frez (Vb), T (Gb), Frez (Gb), for to
the potential barriers, keeping constant the thickness
help us to identification the better geometry structure
of the quantum well
for our applications.
(l3 = 100).

262
Table 1
Thickness Height of Resonant Transmission
of the potential frequency coefficient
potential barriers [THz]
barriers [meV]
[]
10 193.12 5.803 0.99999
10 297.08 30.225 0.99971
10 313.7 74.718 0.99994
10 305.37 30.467 0.99999
10 322.08 74.718 0.99999
10 330.5 74.959 0.99995
10 338.96 74.959 0.99999
10 103.49 4.594 0.99937
10 147.55 28.049 0.99996
10 232.26 29.258 0.99987
11 305.37 73.75 0.99999
12 338.96 73.75 0.99999
13 297.08 30.225 0.99983
13 313.7 72.783 0.99986
13 103.49 4.836 0.99892
14 170.15 27.807 0.99987
15 272.46 29.742 0.99977 Fig.10. The dependences T (Vb), Frez (Vb), T (Gb), Frez (Gb), for
15 313.7 71.574 0.99994 symmetric RDB structure with Gw = 100
15 322.08 71.816 0.99999
15 103.49 5.078 0.99963 IV. CONCLUSIONS
16 297.08 30.225 0.99998
In conclusion, making more experiments with this
17 338.96 30.951 0.99981 simulator you can predict the geometry of the RDB
17 216.48 28.533 0.99997 witch leads of the electronic devices with wishing
17 82.038 4.8361 0.99731 transmission performances. In this way, our present
18 103.49 24.906 0.99823 study clear proved that, for to obtained the circuits
20 272.46 29.742 0.99884 with good transmission power coefficient, it is
20 103.49 24.906 0.99999 necessary to use a symmetric RDB structure with the
thickness of the potential barriers equal with them and
23 272.46 29.742 0.99273
the thickness of the quantum well much grater of
24 103.49 24.18 0.9998 them. Thus, the transmission performances of the
25 193.12 27.565 0.99635 electronic devices can by easy modified by the
26 103.49 23.938 0.99993 changing the geometry size of the RDB structures.
27 118.01 5.8033 0.99998 Much more, for a easy identification of the geometry
28 216.48 28.049 0.99892 structure with leads at the wishing performances,
29 103.49 23.697 0.9995 using MySQL, in this paper we created a Data Base,
in witch it stockade the values of the transmission
32 67.944 4.836 0.99611 coefficients and resonant frequencies of the
32 67.944 4.836 0.99611 submilimetric filters obtained with symmetric RDB
34 103.49 22.971 0.99866 structures, for different height and thickness of the
35 67.944 4.836 0.99982 potential barriers. This Data Base can be anytime up
38 67.944 4.836 0.99761 gradated and interrogated using PHP scripts.
41 103.49 22.487 0.9995 So, our paper present an original and modern
simulator, realized with PHP scripts and interactive
JAVA applet (two language programs very much used
in Internet, in the last time) witch permit us to
determinate the transmission performances of the
electronic devices obtained from RDB quantum
structures.

263
 REFERENCES

[1] A.N. Khondker, M.R. Khan, A.F.M. Anwar, Transmission

Line Analogy of Resonance Tunneling Phenomena: The
Generalized Impedance Concept, J. Apply. Phyis, vol.63, 1988, pp.
.5191.
[2] R. Tsu and L. Esaki, Appl. Phys. Lett. 22, 1973, pp. 562.
[3] N. Ohtani, N. Nagai, M. Suzuki and N. Miki, Electronics and
Communications in Japan, 74, 1991, pp. 11.
[4] D. Lippens and O. Vanbesien, Microvawe Opt. Technol. Lett. 2,
1989, pp. 233.
[5] T. Wei, S. Stapleton and O. Berolo, IEEE Trans. Electron
Devices 42, 1995, pp. 1378.
[6] N. Ohtani, N. Nagai, M. Suzuki and N. Miki, A Study of a
Resonant Condition of Symmetric Triple-Barrier Structures by
Using Circuit Theory, Electronics and Communications in Japan,
Part 2, 74, 1991, pp. 276.
[7] A.Niculescu, T. Petrescu, Resonant Tunneling in GaAs-AlAs
Double Barriers, Proc. of the 22-th Ed. of Int Semic. Conf.,
Sinaia, 1999, vol.1, pp. 135.
[8] C.Weisbuch, B. Vinter, Quantum Semiconductors-
Fundamentals and Applications, Academic Press, New York,
1991.
[9] M.I. Lepa, J.J.M. Kwaspen, T.G. van de Roer, Microwave
Analysis of Double Barrier Resonant Tunneling Diodes,
International Semiconductor Conference, Sinaia, 20thEdition, 1997,
vol 2, pp. .341.

264
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

MEMS based broadband phase shifters

Stefan Simion 1
Abstract A phase shifter based on an array of MEMS distance g + t d from the CPW central line, which,
is designed and then analyzed by computer simulation.
It is shown that for a stable bridge position, the
under the bridge, has the width W ( g is the air gap
maximum ratio of the equivalent capacitance cannot under the bridge and t d is the thickness of the
exceed 1.2. For a microwave signal applied to the bridge, dielectric layer deposited on the CPW central line,
it is also shown that due to the very small time response
of the MEMS, the equivalent capacitance depends on the
having the dielectric constant, r,d ). When a voltage,
DC voltage only. For this type of phase shifter, if the V , is applied on the metal bridge, the electrostatic
maximum operating frequency is much lower than the force, Fe , changes the distance between the two
Bragg frequency, the phase shift varies linearly versus
the frequency, so broadband applicatons such as pulse metal plates of the MEMS, from g + t d to
processing are possible. g + t d z . From the mechanical point of view, the
Keywords: MEMS, phase shifter.
following formula may be used for the spring
I. INTRODUCTION constant, k [5]:
1
W W
3
During the last years, distributed circuits such as k = k0 1 1 + 1
switches, phase shifters, BPSK modulators (see for lb lb
example [1], [2], [3], [4]) have been integrated by
using MEMSs (micro-electromechanical systems).
Due to the small MEMS series resistance, these W 32 Ewtb3 8(1 )wbtb
where k0 = + , E is the
circuits have very low insertion loss, being an lb l3 l
important advantage for millimeter-wave frequency b b
range. On the other hand, the MEMS device has a
high time response to the operating applied voltage, Youngs modulus, is the Poissons ratio and is
compare to the Schottky-varactor diodes, devices also the residual stress. The dynamic equation of motion
used for these applications. The circuits based on an for the bridge, which may be solved in order to find
array of MEMSs may have a low insertion loss, but the position of the bridge along the z -axis, is (see for
also a large frequency bandwidth, due to their example [6]):
distributed structure.
d 2z dz 0 wbWV 2
In this paper, a design procedure for the phase shifters m + b +kz = , where m
dt 2 dt
2
based on an array of MEMSs loading a CPW t
2 g + d z
transmission line, is presented in detail. The MEMS r ,d

analysis is performed for DC voltage as well as for RF
signal, showing that the device has a linear behaviour
is the mass of the bridge and b is the damping
in the microwave frequency range. The MEMS phase
shifter is analysed for sinusoidal and square input coefficient. This dynamic equation has been solved by
signals, wherefrom the broadband characteristics of using a time domain method [7]. Applying this
these types of circuits are put into evidence. method, for l b = 300 m, wb = 60 m, t b = 1 m,
g = 2.5 m, t d = 0.3 m, r ,d = 7.5 and for two
II. MEMS ANALYSIS values of the bridge width ( W = 140 m and
W = 90 m), g z may be computed and graphically
The MEMS cross section is given in Fig. 1, where lb
represented as in Fig. 2 a. It is important to observe
and t b are the length and the thickness of the metal
bridge, respectively. In this paper, the width of the
metal bridge is wb . The bridge is suspended at a

1
Academia Tehnica Militara, George Cosbuc 81-83, 75275, Bucharest, Romania, e-mail: simions@mta.ro

265
 1.40
1.35
1.30
1.25 W=140m

1.20

Fig. 1. The MEMS cross section.
that g z has a very sharp variation for the applied
voltage around the pull-down value. Then, for values
of the applied voltage close to the pull-down voltage,
the position of the bridge is not stable. In this case, for
MEMS applications such as phase shifters, in order to
control the phase shift value, the applied voltage must
not exceed ~25V, for W = 140 m and ~30V, for
W = 90 m. It is not difficult to show that the
normalized equivalent MEMS capacitance, is:
Cs 1 possible to be given by the DC voltage only. To prove
= ,
Co
1
z this, for the MEMS geometrical dimensions given
zo
where:
t wW
z o = g + d and Co = C s V =0 0
r t
g+ d
r
is the MEMS capacitance for V = 0 (the minimum
capacitance value). The normalized capacitance,
C s / C o , versus z / z o is shown in Fig. 2 b, for the
same data as for Fig. 2 a. Taking into account the
maximum value for the applied voltage given above,
from Fig. 2 b, it is drawn the conclusion that the
maximum capacitance ratio is ~1.2, a similar
estimation as in [8].
Cs/Co
1.15
1.10
W=90m
1.05
1.00
0 5 10 15 20 25 30 35 40
Voltage [V]
(b)
Fig. 2. The distance between the bridge and the dielectric layer, (a)
and the normalized capacitance (b), versus the applied voltage.
The MEMS is not a fast device. If a sinusoidal signal
is applied on the MEMS bridge, arround a DC
voltage, then the bridge position cannot follows the
fast variation of the signal. Therefore, if the signal
frequency is high enough, the bridge position is
above, the simulation results concerning the MEMS
behaviour to the RF signal are shown in Fig. 3, for
W = 140 m and in Fig. 4, for W = 90 m, where the
amplitude of the sinusoidal signal, Vampl , is equal to
10V and the DC voltages are equal to 25V and 15V,
respectively. The RF frequency is 10KHz, 30KHz and
200KHz. From these figures, it is observed that for a
microwave signal applied on the bridge, the MEMS
equivalent capacitance depends on the DC voltage
only, even if this is a nonlinear capacitance, due to the
small time response of this kind of device. As a result,
even for high power microwave signal, the MEMS
may be seen as a linear device. This observation is
important for a phase shifter based on a MEMSs
array, because the phase shift will depend only by the
apllied DC voltage, without other small signal
constrains.

2.5 III. PHASE SHIFTERS DESIGN AND ANALYSIS

W=90m The phase shifter consists of an array of MEMSs,

2.0
periodically loading a CPW transmission line. If the
length of the CPWs which connect two consecutive
1.5 MEMSs, l , is small enough, a lumped equivalent
g-z [m]

circuit may be used [7]. In this paper, Rl , Ll , Cl

1.0 and Gl are the lumped values of the CPW equivalent
W=140m
circuit, which depend on the CPW distributed
0.5 parameters ( R , L , C and G ). Also, R s , Ls and
C s will be used as notations for the equivalent
0.0
0 5 10 15 20 25 30 35 40 45 resistance, inductance and capacitance of the MEMS,
Voltage [V] respectively. The formulas used for the procedure
(a) design of the switch may be derived from this
equivalent circuit. If the maximum operating
frequency, f max , is lower enough compare to the
resonance frequency of the MEMS, then the influence

266
of the MEMS series inductance may be neglected
1
( Ls << ). Also, because the losses due to the
C s
CPW transmission lines and due to the MEMSs are
small, so Rl 0 , Gl 0 and R s 0 . Therefore,
taking into account that the circuit consists of n
identical cells, the Bragg frequency and the input
impedance of the circuit are given by:
1 Ll
fb = and Z in ,
Ll (C + Cl )
s
C s + Cl

where, the expression for Z in is true only if

2
f max / f b2 << 1 . For the CPW characteristic
Ll
impedance, Z c , the formula Z c = , may be
Cl
used. For the circuit design, a few constraints of the (c)
input data must be respected: Z in = 50 and the
value for the characteristic impedance of the CPW Fig. 3. The distance between the bridge and the dielectric layer,
transmission line, Z c , must be higher in order for the MEMS operating at VDC = 25 V, Vampl = 10 V,
to W=140m and f = 10 KHz (a), 30 KHz (b) and 200 KHz (c).

(a) (a)

(b)
(b)

267

(c)
Fig. 4. The distance between the bridge and the dielectric layer,
for the MEMS operating at VDC = 15 V, Vampl = 10 V, W=90m
and f = 10 KHz (a), 30 KHz (b) and 200 KHz (c).
decrease the CPW electrical length, (so to decrease
the length of the circuit), but, on the other hand, it
must be lower in order to reduce the CPW losses,
( Z c must be chosen to minimize ). For
Z in = 50 , f max / f b = 0.15 , r = 11.9 (the
substrate dielectric constant for silicon), t = 1 m (the
thickness of the CPW gold metallization),
w + 2 s = 300 m ( w and s are the width and the slot
of the CPW between MEMSs), is minimized if
Z c is 60 - 80 [7]. In this paper, Z c =75.
Combining the expression for f b , Z in and Z c , they
are obtained, Ll = Z in / ( f b ) , (
Cl = Z in / f b Z c2
[ ]
and C s = (Z c / Z in ) 1 C l . The electrical length
2
of the CPW transmission line which connects two
consecutive MEMSs may be obtained with
= 2f max Cl Ll and then, by using a commercial
software, the CPW length, l , it is easily computed.
The distances g and t d are usually imposed by the
technological constraints. Assuming that the values
for g , t d and r ,d are known, the formula for w W
g r ,d + t d
is: wW = C o . For n cells, the maximum
o r ,d
phase shift introduced by the circuit, at the maximum
operating frequency, may be computed as
2f max n Ll ( C s + Cl ) . This formula for may
be used to compute the number of cells, n , if the
value for is imposed.
Two phase shifters have been designed, the first one
for W = 140m and the second one for W = 90m.
The design formulas introduced above have been
applied for a maximum operating frequencies, f max ,
of 17GHz for the first circuit and 25GHz for the
second one, in the both cases f b / f max =0.15. For
the two circuits, they were obtained =11.46deg,
w = 50 m, s = 125m, while the CPW lengths and
the CPW equivalent inductance and capacitance,
between two consecutive MEMSs, are l =223m,
Ll = 0.141nH, Cl = 25fF, for the first circuit and
l = 150 m, Ll = 0.094nH, Cl = 16.7fF, for the
second one. Imposing W = 140 m and W = 90 m
(geometrical dimensions for the MEMSs analyzed in
section II), for the both circuit, wb = 60 m. Also,
Cs =31.3fF and Cs =20.9fF, corresponding to
VDC =25V and 15V, respectively. Taking into
account the others geometrical and electrical MEMS
parameters (given in section II), it is obtained
Ls = 50.75 pH. The MEMS series resistance depends
on the frequency. For these values and n = 38, the
maximum phase shift is 385deg, for the both
phase shifters, computed at f max = 10GHz for the
first circuit and f max = 15GHz, for the second one.
The two phase shifters have been numerically
analyzed, in order to obtain de magnitude and the
phase for S 21 and also the magnitude for S11 (the
CPW losses effect are included). The results are
shown in Fig. 5, wherefrom, for the phase shift, it is
observed a good agreement between the analytical
and simulated values (see Fig.5 a). Also, a return loss
better than 30dB (see Fig. 5b), has been obtained for
the both circuits, while the insertion loss is smaller
than 0.5dB for the first circuit and 0.4dB for the
second one, up to the maximum operating
frequencies. The second phase shifter is lossless
compare to the first one because is shorter (the two
) circuits have the same number of cells).
The phase shift introduced by the circuit may be also
analyzed in the time domain. Fig. 6 shows the results
for the second phase shifters and two values of the DC
voltage, VDC , 10V and 30V. The frequency of the
input signal is 15GHz. The delay time is different in
the two cases because the MEMS equivalent
capacitance depends on the DC voltage. From this
figure, the time delay introduced by the circuit is
~74ps for VDC =10V and ~79ps for VDC =30V, these
results being in good agreement (5% error) with those
obtained by using formula n Ll (Cs + Cl ) . The 5ps
difference between the two time delay values at
15GHz means a phase shift difference of 27deg. In
some applications which ask for a larger value of the
phase shift difference, the number of cells must be
increased or/and to minimize the influence of the
CPW equivalent capacitance.
In order to evaluate the broadband characteristic of
this phase shifter, the next simulation has been
performed for a square pulse applied to the input of
268
the second circuit. For V DC =10V and pulse width of
10ps, 30ps, 50ps, the output waveforms presented in
Fig. 7 show that if the pulse width decreases, for a
given Bragg frequency, the dispersive characteristic
of the circuit contributes to the pulse shape distorsion.
This is because for short pulses, the condition
f b / f max =0.15, which assures the less dispersive
character of the circuit, is not true for the spectral
components having important amplitudes above
f max 25GHz. From Fig. 7, it is observed that for
pulse width greater than 30ps, the dispersive character
of the circuit may be neglected. For an input pulse
width of 50ps, the delay time introduced by the circuit (b)
is the same as for a sinusoidal wave (see Fig. 6), for
V DC =10V as well as for V DC =30V (see Fig. 8),
showing the broadband characteristic of the circuit.

IV. CONCLUSIONS

In this paper, a pahase shifter based on an array of

MEMSs has been designed and analyzed. It has been
shown that in order to use MEMSs in these
applications, the maximum ratio of the equivalent
capacitance cannot exceed 1.2, for a stable bridge
position. Also, due to the very small time response of
the MEMS, if a microwave signal is applied to the
bridge, the bridge position as well as the equivalent
(c)
capacitance depend on the DC voltage value only.
The analysis results obtained for the MEMSs array Fig. 5. The numerical results obtained for the phase shift, return
phase shifter show that if the maximum operating loss and insertion loss, for the first (1) and the second (2)
frequency is much lower than the Bragg frequency, phase shifter.
then the phase shift varies linearly versus the
frequency. Therefore, the delay time is practically
constant up to this maximum operating frequency, so
the circuit may be also used as delay circuit for
pulses.

(a)

(a)

(b)

Fig. 6. The input (a) and the output (b) waveforms for the second
phase shifter, for VDC =10V and VDC =30V (the input signal
has the frequency equal to 15GHz).

269
Fig. 7. The output waveforms for pulses of different widths, applied
to the input of the second phase shifter ( VDC =10V).

Fig. 8. The output waveforms for a pulse of 50ps width, applied to

the input of the second phase shifter operating
to VDC =10V and VDC =30V.
REFERENCES

[1] J. Rizk, G..-L. Tan, J. B. Muldavin, G. M. Rebeiz, High-

isolation W-band MEMS switches, IEEE Microwave and Guided
Wave Letters, vol. 11, no. 1, Jan. 2001, pp. 10-12.
[2] J. B. Muldavin, G. M. Rebeiz, Inline capacitive and DC-
contact MEMS shunt switches, IEEE Microwave and Guided
Wave Letters, vol.11, no.8, August 2001,
pp. 334-336.
[3] J. S. Hayden, G. M. Rebeiz, Low-loss cascadable MEMS
distributed X-band phase shifters, IEEE Microwave and Guided
Wave Letters, vol. 10, no. 4, April 2000, pp. 142-144.
[4] N. S. Barker, G. M. Rebeiz, Distributed MEMS transmission-
line BPSK modulator, IEEE Microwave and Guided Wave Letters,
vol. 10, no. 5, May 2000, pp.198-200.
[5] S. Simion, Modeling and design aspects of the MEMS switch,
Proc. of the International Semiconductor Conference, 2003, pp.
125-128.
[6] J. B. Muldavin, G. M. Rebeiz, Nonlinear electro-mechanical
modeling of MEMS switches, Proc. of the IEEE Digest, 2001.
[7] S. Simion, I. Sima, Design of MEMS array phase shifters,
Proc. of Communications 2004, vol.1, pp. 329-334.
[8] N. S. Barker, G.M. Rebeiz, Optimization of distributed MEMS
phase shifters, Proc. of IEEE Digest, 1999, pp. 1-4.

270
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Seria ELECTRONIC i TELECOMUNICAII
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Tom 49(63), Fascicola 2, 2004

Optical coherent and incoherent systems frequency

analyze in Cartesian coordinate
Toadere Florin 1

I i = h(u , v ) I o ( , )dd
Abstract-Our goal in this paper is to make a study about 2
optical coherent and incoherent systems frequency (4)
response. We begin from the definition of an optical
system then we define transfer function for coherent and AMPLITUDE TRANSFER FUNCTIONS TYPICAL
incoherent systems. We find the response of this system
to a step indices stimulus, then we generalize for an
FOR COHERENT CASE.
image and finally we make a comparison between two
different systems. We define input and output frequency spectrum
Go ( f x , f y )
INTRODUCTION (5)
= U o ( x, y ) exp{ j 2 ( f x u + f y v)}dvdu
Generally speaking optical systems can be seen as a
Gi ( f x , f y )
black box with an input and an output, at the input we
(6)
have object plane and at the output image plane which = U i ( x, y ) exp{ j 2 ( f x u + f y v)}dvdu
is obtained from convolution between object image
and transfer function of optical systems (the black box We define transfer function
can contain one ore more optical elements). Here we Hi ( fx , f y )
will have a diffraction element. (7)
U i (u , v) = h(u ; v )U o ( , )d d (1) = h(u , v) exp{ j 2 ( f x u + f y v)}dvdu
We apply convolution to (3) and we obtain:
h(u , v; , ) optical system impulse response Gi ( f x , f y ) = H ( f x , f y )Go ( f x , f y ) (8)
but system response to optical impulse is Fourier This is the relation between image and object plane in
transform (Fraunhoffer diffraction) of diffraction frequency.
element aperture. But transfer function is Fourier Transform of impulse
h(u , v) = response system. Then we will have;
A 2 (2) H ( fx , f y ) =
z i P( x, y) exp{ j z (ux + vy )}dxdy
A 2
P ( x, y ) exp{ j (ux + vy )}dxdy}
i
F{
Next we will try to calculate h(u,v) for coherent and zi zi
incoherent case.
What do we understand by coherent and incoherent = ( A zi ) P( zi f x , zi f y ) (9)
illumination? If we put A zi =1 then
Coherent illumination is made by lasers.
Incoherent illumination is made by diffuse source like H ( f x , f y ) = P( zi f x , zi f y ) (10)
sun or gaze lamp. As a conclusion for coherent illumination Amplitude
For coherent illumination the system is described by Transfer Function is the aperture trough which the
amplitude convolution equation. light passes and the diffraction is made. For a square
U i (u , v) = h(u ; v )U o ( , )d d (3) aperture we will have:
x y
For incoherent illumination the system is described by P(x,y)= rect ( )rect ( )
intensity convolution equation.
2w 2w

1
Facultatea de Elcetronica si Telecomunicati, Departamentul Bazele
Electronicii, str. Baritu nr.26, Cluj Napoca , tflorin@bel.utcluj.ro

271
 Next we will study optical coherent systems response
The transfer function will be: to a step indices stimulus for a square aperture. We
zi f x zi f x will study 2D and 3D case Fig. 1
H ( f x , f y ) = rect ( )rect ( )
2w 2w

Fig. 1. first line present 2D case; second line present 3D case; first column present step indices stimulus;
second line present square aperture; third line present response

OPTICAL TRANSFER FUNCTION TYPICAL FOR We define transfer function

INCOHERENT CASE H( fx, fy )

h(uv) exp[ j 2 ( f x u + f y v)]dudv

2
We define normalized frequency spectrum I i and I o :
=

Go ( f x , f y ) 2
h(u , v) dudv

=
I o (u , v) exp[ j 2 ( f x u + f y v)]dudv
We apply convolution to (4) and we obtain:
(13)

I o (u , v)dudv Gi ( f x , f y ) = H ( f x , f y )Go ( f x , f y ) (14)

(11)
H( f x , f y ) optical transfer function
Gi ( f x , f y )
Optical transfer function and optical amplitude

=
I (u, v) exp[ j 2 ( f u + f
i x y v)]dudv function on their definition imply function h (optical
system impulse response) so there is a relation
I (u, v)dudv
i
between this two function. Optical transfer function is
the normalized autocorrelation of amplitude transfer
(12) function.

272
 z i f x z i f x
P( x + 2 , y + 2
)
H( fx, fy )=
P( x, y)dxdy
z i f x z i f x
P( x 2 , y 2
)
dxdy (15)
P( x, y)dxdy
This represent area of superposition of two apertures
z i f x z i f x
of the same shape one at , the other at
2 2
z i f x z i f x Fig. 2 H( f x , f y ) for incoherent case.
, divided at total area of the two
2 2 When this area is normalized with total area 4w2
apertures as in Fig. 2 have
Mathematical relation of common area is: fx fy
(2 w z f x )(2 w z f y ) H ( f x , f y ) = tri ( )tri( )
A( f x , f y ) = i i 2 f0 2 f0
0 w
f0 = cutoff frequency for coherent case
2w 2w zi
fx ; fy
zi zi Next we will study optical incoherent systems
response to a step indices stimulus for a triangular
aperture. We will study 2D and 3D case Fig. 3.

Fig. 3. first line present 2D case; second line present 3D case; first column present step indices stimulus; second
line present triangle aperture; third line present response .

273
CONCLUSION
Comparing response in Fig. 1 and Fig. 3 we see a
great difference between optical coherent and
incoherent system response for a step indices. So for
optical coherent systems we have a response with
oscillation at the end (Gibb phenomenon) and a phase
difference from the axe of symmetry. Optical
incoherent systems do not have oscillation at the end
and phase difference. Finally to have a clear view we
will put an image instead of step indices stimulus and
well see how acts in the two cases. Fig.4

Fig. 4. first line present coherent case; second line present incoherent case; first column present input image;
second line present square aperture and triangle aperture; third line present output image.

REFERENCES
1. A. Papoulis: System and transform with application in
optics McGraw Hill, New York 1968.
2. J.W.Goodman: Introduction to Fourier optics McGraw
Hill, New York 1968.
3. E. Hecht: A. Jajac: Optics Addison Wesley, Reading,
MA 1974.
4. M. Born, E. Wolf: Principle of Optics Pergamon New
York 1964.
5. Keigo Izuka Engineering Optics Springer, Verlang
Berlin, Hailderberg 1985
6. J.D.Gaskill Linear System Fourier transform and Optics
John Wiley and Sons New York 1978
7. L.Boas Mathematical Methods In the Physical Science
Wiley New York 1966
8. B.E.A Saleh, M.C.Teich Fundamentals of Photonics
John Wiley New York, NY, 1991

274
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Cartesian coordinate optical filter analyses

Toadere Florin 1

Abstract-Our goal in this paper is to develop a study

about how Cartesian coordinate optical filter acts in
frequency. We use coherent illumination and we begin
from the amplitude transfer function which is the optical
transfer function of filter in this case of illumination. We
define different type of filter LPF, HPF and BPF in
Cartesian coordinate 2D and 3D then we pass an image
trough this filter and we see the filter effect on final
image function of filter dimension which we used. For
this we use an algorithm in three steps: we compute
Fourier transform of an image, then we multiply this
with filter transfer function then we make inverse Fig. 2 optical filter
Fourier transform on result.
RESULTS OF SIMULATION
INTRODUCTION
In the case of coherent illumination which we use in
To develop our study we shall recall what is an optical
this paper the amplitude transfer function (see bibl.9)
filter?
is the filter transfer function. By coherent illumination
An optical filter is a 4f telecentric system; this is an
here we understand parallel light (Fraunhofer
aligned system consisting of an input image, first lens,
diffraction).
Fourier plane, second lens and output image like in
the Fig. 1
Frequency domain method

Let g(x,y) be an image formed by the convolution of

an image f(x,y) and a linear, position invariant
operator h(x,y) so we have g(x,y)=h(x,y)*f(x,y); then
in the frequency domain we will have:
G(u,v)=H(u,v)F(u,v);
H(u,v) is optical transfer function and G, H, F are
Fourier transforms of g, h, f.
Our goal after we compute F(u,v) is to select H(u,v)
1
so that the desired image g(x,y)=F [H(u,v)F(u,v)] to
Fig.1 4f system have some new feature of f(x,y).
So in Cartesian coordinate we will define LPF, HPF
The aim of lens in such a system is as fallow: lens and BPF.
have property of making Fourier transform in optics.
If in the Fourier plane we put an aperture (let say for LPF Low pass filter
the beginning an arbitrary aperture) then we will have
an optical filter like in the Fig. 3 where we can see the A low pass filter has a transfer function like this:
difference between filtered image and original image:
1 if D(u , v) D0
H(u,v)=
0 if D (u , v) > D0

1
Facultatea de Electronica si Comunicatii, Catedra Bazele
Electronicii, str.Baritu nr.26, Cluj Napoca, tflorin@bel.utcluj.ro
D 0 is a specific nonnegative quantity and D(u,v) is

275
the distance from H(u,v) to the origin of frequency
plane. The point of transition between 1 to 0 is called
the cutoff frequency; here cutoff frequency is D 0 .
We have the LPF 2D representation in Fig. 3 and 3D
representation in Fig. 4
LPF have the next characteristics: high frequency
reduction but they make blur, noise reduction because
noise is installed at high frequency.
Next we will study how LPF acts for different cutoff
frequency for an image with 256x256 dimensions we
will use the next cutoff dimension: 128x128, 80x80,
and 40x40. The result is illustrated Fig. 5

Fig. 3 filter transfer function 2D

Fig. 4 filter transfer function 3D

Fig. 5 The graphics present the same image filtered with filters having different cutoff; as the cutoff frequency
decrease we will have more blur on the image and oscillation. Oscillations around image are caused by
transitory regime passing from 1 to 0.

HPF high pass filter

276
A HPF has inverse properties like LPF, and is defined representation in Fig. 7
like HPF=1-LPF which we can write like this: HPF has the next characteristics: block low pass
frequency and let to pass high frequency. As effect it
0 if D (u , v) D0 enhances the noise, reduces basic characteristics of an
H(u,v)= image and is used for edge detections. Next we will
1 if D(u , v) > D0 study how HPF acts for different cutoff frequency for
an image with 256x256 dimensions we will use the
D 0 is cutoff frequency
next cutoff dimension: 128x128, 80x80, and 40x40.
The result is illustrated in Fig. 8

Fig. 6 filter transfer function 2D

Fig.7 filter transfer function 3D

with the 2D representation in Fig. 6 and 3D

Fig. 8 the graphics present the same image filtered with filters having different cutoff; as the cutoff frequency
increase we will have edge detection more pronounced on the image and oscillation. Oscillations around image
are caused by transitory regime passing from 1 to 0.
BPF high pass filter

A BPF is defined like difference between two LPF

277
with different cutoff frequency BPF=H1(u,v)-H2(u,v) It acts like a median filter and has the characteristics
1 if D(u , v) D01 that we can reduce noise without blurring image and
H(u,v)= we can cheep edge characteristics. Next we will study
0 if D (u , v) > D01 how BPF acts for different cutoff frequency for an
image with 256x256 dimensions we will use three
windows with dimensions: 40x220, 100x160, 40x220
1 if D(u , v) D02 for first LPF and other three different windows with
H(u,v)=
0 if D(u , v) > D02 dimensions 80x180, 100x160, 125x135 for second
filter as in Fig. 11
with the 2D representation in Fig. 9 and 3D
representation in Fig. 10

Fig. 9 filter transfer function 2D Fig.10 filter transfer function 3D

Fig. 11 the graphics present the same image filtered with filters having different dimensions. It keeps specific
characteristics of LPF and HPF and is a mediator filter between this two filter

278
CONCLUSION

This paper tries to distinguish with concrete example

basic characteristics of LPF, HPF and BPF in
Cartesian coordinate. These filters are ideals because
are defined like 1 and 0 but they play an important
role because they describe how an image is filtered in
low, high and medium spatial frequency domain.
Because ideal character of this filter we observe
oscillation around image, oscillation specific
transitory regime. There are other optical filters like
Gaussian, Hamming, and Butterworth which have a
better transitory regime. Optical filter in Cartesian
coordinate are very important in Optical Fourier
signal and image processing.

REFERENCES

1. A. Papoulis: System and transform with application in

optics McGraw Hill, New York 1968.
2. J.W.Goodman: Introduction to Fourier optics McGraw
Hill, New York 1968.
3. E. Hecht: A. Jajac: Optics Addison Wesley, Reading,
MA 1974.
4. M. Born, E. Wolf: Principle of Optics Pergamon New
York 1964.
5. Keigo Izuka Engineering Optics Springer Verlag Berlin,
Heidelberg 1985.
6. J.D.Gaskill Linear systems, Fourier transform and
Optics John Wiley and Sons New York 1978.
7. M.L.Boas Mathematichal Methods In the Physical
Science Wiley New York 1966
8. R.C.Gonzales, R.E.Wods Digital Image
Processing Adison Wesley 1992
9. Toadere Florin Optical coherent and incoherent
systems frequency analyze in Cartesian coordinate
ECT 2004 Buletinul Universitatii Politehnica Timisora
Seria Electronica si Teleomunicatii, Tom 49(63),
Fascicula 1-2, 2004
10. B.E.A Saleh, M.C.Teich Fundamentals of Photonics
John Wiley New York, NY, 1991

279
 Buletinul tiinific al Universitii "Politehnica" din Timioara
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TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Space and frequency analyze of an LSI optical system with

different input signals
Toadere Florin1

Abstract-Our goal in this paper is to develop a calculus Knowing the impulse system response we can
algorithm beginning from Optical Fourier Transform calculate in space with convolution relation between
and then to generalize to optical linear shift invariant
systems. We begin from the premise that an optical
input and output signals:
system can be seen like a black box with an input and an
output. Our system property analyze can be made in
y( u1 , u 2 ) = x(v , v
1 2 )h(u1 v1 ; u 2 v 2 )dv1 dv 2
space with the help of convolution and in frequency by (4)
multiplying specters, these operation are If in (4) we apply Fourier Transform then we will
complementarily. Optical system also can be seen from have Y( 1 , 2 ) =X( 1 , 2 ) H( 1 , 2 ) were:
the perspective of components which is made. So in our
example we choose components which can made Optical X( 1 , 2 ) =

x(u , u
Fourier Transform: a square aperture and a convergent
lens. 1 2 ) exp[ j (1u1 + 2 u 2 )]du1 du 2
input spectrum (5)
INTRODUCTION
H( 1 , 2 ) =
We consider an optical system which is characterized
by input signal x( u1 , u 2 ) and output signal
x(u , u
1 2 ) exp[ j (1u1 + 2 u 2 )]du1 du 2
frequency response (6)
y( u1 , u 2 ) . We will find a relation between input In conclusion convolution in space is:
signal and output signal by the means of system y( u1 , u 2 ) =f( u1 , u 2 ) *g( u1 , u 2 ) (7)
impulse response and transfer function. In optics we And equivalently in spatial frequency by the means of
utilize linear shift invariant systems. By linear optical Fourier transform we have:
Y( 1 , 2 ) =X( 1 , 2 ) H( 1 , 2 )
systems we understand that the system has a linear
(8)
relation between input signal and output signals by the
relation: In optical linear systems (7) and (8) can be
generalized:
F[a x1 ( u1 , u 2 ) +b x 2 ( u1 , u 2 ) ]=
y( u1 , u 2 ) =f( u1 , u 2 ) *g( u1 , u 2 ) **j( u1 , u 2 ) (9)
aF[a x1 ( u1 , u 2 ) ]+bF[ x 2 ( u1 , u 2 ) ] (1)
Y( 1 , 2 ) =X( 1 , 2 ) H( 1 , 2 ) ...J( 1 , 2 )
If at the input we apply a 2D optic impulse
(10)
(u1 1 , u 2 2 ) then output image will be In conformity with result in eq.9 and eq.10 we will
called system response to optical impulse with the make a simulation of an optical system made by a
notation h (u1 , u 2 ; 1 2 ) so we will have the square aperture and a convergent lens.
relation:
FIRST EXAMPLE
F[ (u1 1, u 2 2 ) ]=h (u1 , u 2 ; 1 2 ) (2)
The system is linear shift invariant if system response In table 1 we will se mathematical relation in parallel
to optical impulse is independent of impulse input for the two situations. We have as input signal a
position or more clearly an input impulse variation harmonically plane wave. Results are presented in
produce the same variation at the output. Fig. 1
F[ (u1 1, u 2 2 ) ]= h(u1 1, u 2 2 ) (3)

1
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Electronicii, str. Baritu nr.26, Cluj Napoca , tflorin@bel.utcluj.ro

280
 Spatial domain analyses Frequency domain analyses
Input signal: A(x,y)=sin(x)sin(y) ( ) 2 [ (1 1 ) (1 2 )]
A( 1 , 2 ) =
[ ( 2 2 ) ( 2 + 2 )]
Aperture transfer B(x,y)=rect(x)rect(y) B( 1 , 2 ) =sinc( 1 )sinc( 2 )
function
convolution C(x,y)=A(x,y)*B(x,y) C( 1 , 2 ) =A( 1 , 2 ) B( 1 , 2 )
Lens transfer x2 y2 1 2 22
function D(x,y)=pi/2exp(- ) D( 1 , 2 ) =exp(- )
2 2 2 2
convolution E(x,y)=C(x,y)*D(x,y) E( 1 , 2 ) =C( 1 , 2 ) D( 1 , 2 )
Table 1

Fig. 1 The first column present analyses in space domain for the case considered in Table 1, on the x and y axes
we have spatial dimensions. The second column present frequency domain analyses on x and y axes we have

281
spatial frequency dimension.
SECOND EXAMPLE This means that we have at the input 16 harmonics
plan waves and equivalent spectrum consisting in 64
We will make a more complex analysis with the help points. The rest of the calculus is in conformity with
of an optical binary signal, made by the next optical situation in Table 1
Fourier string:
A(x,y)=[sin(x)+sin(3x)/3+sin(5x)/5+sin(7x)/7]
[sin(y)+sin(y)/3+sin(5y)/5+sin(7y)/7]

Fig. 2 The first column present analyses in space domain, on the x and y axes we have spatial dimensions. The
second column present frequency domain analyses on x and y axes we have spatial frequency dimension.

THE THERD EXAMPLE

harmonically plane waves of diverse spatial frequency
We have as input signal an image. It is know that an and the spectrum will be a sum of points with random
image can be decomposed in a very large number of distributions in spatial frequency domain as in Fig. 3

282
Fig. 3 The first column present analyses in space domain, on the x and y axes we have spatial dimensions. The
second column present frequency domain analyses on x and y axes we have spatial frequency dimension.

CONCLUSION but produce noise. In frequency analyze we observe

In this paper we try to develop a calculus algorithm
for an optical linear shift invariant system, made by
optical component which can make optical Fourier
transform. We try to have a global view of the process
both in space and frequency. In spatial domain
analyze we observe that after we convolve an image
with an aperture the image change space position,
without distortion. Convergent lens reduces signals
that input signal spectrum is made by 4 points then 64
points and finally from a very large number. This
spectrum is multiplied with transfer function and as
result we have the same number of sample but other
amplitude and function of their spatial position lens
let to pass only that component that intersect lens
(low frequency components).

283
 REFERENCES

1. A. Papoulis: System and transform with application in

optics McGraw Hill, New York 1968.
2. J.W.Goodman: Introduction to Fourier optics McGraw
Hill, New York 1968.
3. E. Hecht: A. Jajac: Optics Addison Wesley, Reading,
MA 1974.
4. M. Born, E. Wolf: Principle of Optics Pergamon New
York 1964.
5. Keigo Izuka Engineering Optics Springer Verlag Berlin,
Heidelberg 1985.
6. J.D.Gaskill Linear systems, Fourier transform and
Optics John Wiley and Sons New York 1978.
7. M.L.Boas Mathematichal Methods In the Physical
Science Wiley New York 1966
8. R.C.Gonzales, R.E.Wods Digital Image
Processing Adison Wesley 1992
9. B.E.A Saleh, M.C.Teich Fundamentals of Photonics
John Wiley New York, NY, 1991

284
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Design of small-size planar filters using FDTD and wavelet

analysis
Marian G. Banciu1, George Lojewski2, Liviu Nedelcu1, Drago Gheu1, Nicolae Militaru2,
Diana Brnaru2, Teodor Petrescu2
Abstract Compact planar resonators and filters were
r
designed by using a Finite-Difference Time-Domain
shown in Fig. 1. Every electric field E component
surrounded by four circulating magnetic field
method. The accuracy of the FDTD method was r r
increased by extending the Brengers perfectly matched H components, and
r Hevery
component is
layer to non-homogeneous media. By applying the
wavelet analysis and signal estimation techniques, the surrounded by four circulating E components.
total simulation time was reduced by 5 times. The
devices were developed on a substrate with 10.8
dielectric constant and 0.635 mm height. Each resonator
occupied down to 32% of the surface of a folded half-
wavelength resonator. Two-pole and four-pole cross-
coupled filters were developed for 900 MHz frequency
band applications.
Keywords: FDTD, microwave devices, planar
resonators, microstrip filters

I. INTRODUCTION

Compact cost-effective devices are strongly required

by the modern wireless communications systems.
When devices with distributed parameters are
designed at rather low frequencies of mobile
communications such as 900 MHz, the size reduction
is a major requirement. Transmission line models are Fig. 1. The FDTD cell
not sufficiently accurate when analyzing structures
with discontinuities such as bends, junctions, open- Let us consider the curl equation
ends, etc. In this paper, an accurate improved FDTD
method is proposed for compact filters. E x 1 H z H y
(1)
= E x
t y z
II. DESIGN METHOD

The 3-D Finite-Difference Time-Domain (FDTD) A field component such as E xn +1 / 2 (i, j , k ) at time step
method [1] was employed due to its accuracy and n+1/2 is not calculated directly in FDTD scheme, but
versatility. The method selects from all four Maxwell it may be approximated by the arithmetic average
equations the two curl equations and solves them between the components calculated at time steps n
numerically by finite differences in time-domain. The and n+1. The final reult is an explicit expression of
grid points where the electric and magnetic fields are the component at time step n+1 when the components
calculated alternate in space forming the FDTD cell as at previous time steps are already known.

(i, j, k )t t n +1 2 1 n +1 2 1 n +1 2 1 n +1 2 1
1 H z i, j + , k H z i, j , k H y i, j , k + H y i , j , k
2 (i, j, k ) (i, j, k ) 2 2 2 2
E
n +1
(i, j, k ) = E x (i, j, k ) + (i, j, k )t
n
(2)
x
(i, j, k )t y z
1+ 1+
2 (i, j, k ) 2 (i, j, k )

1
National Institute of Materials Physics, Atomistilor 105 bis, Bucharest-Mgurele, e-mail gbanciu@infim.ro
2
Politehnica University of Bucharest, Faculty of Electronics Telecommunications and Information Technology, Iuliu Maniu 1-3,
061071, Sector 6, Bucharest, e-mail george.lojewski@munde.pub.ro

285
The devices were designed on a 0.635 mm height
Rogers substrate with 10.8 dielectric constant. The
FDTD grid was x = y = 0.15 mm and
z = 0.127 mm. The time step was chosen of
t = 0.27 ps, in order to satisfy the stability criterion
Courant-Friedrichs-Levy. The field excitation was
chosen as a Gaussian pulse
( t T0 ) 2
Ez (t ) = Ez 0 (t )e T2
(3)
where T = 38 t and T0 = 4 T.
III. THE NON-HOMOGENEOUS PERFECTLY
MATCHED LAYER
The accuracy of the FDTD method depends very
much on the quality of the absorbing boundary
conditions. The perfectly matched layer (PML) idea
is to use a lossy material to match the incident waves.
However, for an isotropic lossy material, the match
occurs only at normally incidence, therefore such a
material has only a limited application for absorbing
boundary condition [2]. A PML should match waves
of arbitrary incidence, polarization, and frequency.
The Brengers innovation consits in a derivation of a
split-field formulation of Maxwells equations;
namely, each vector field component is split into two
orthogonal components. The PML technique
decomposes each field projections in two, and the
wave incident to PML is attenuated via electric ()
and magnetic (*) conductivity. The extremely small
reflection is satisfied by the impedance matching
condition perpendicularly to the PML layer.
In order to analyze the new microstrip devices, a Non-
Homogeneous Perfectly Matched Layer (NH-PML)
was developed. For such a non-homogeneous
boundary, the values of electric () and magnetic (*)
conductivities should satisfy the impedance matching
condition.
*
j j = const. , (4)
2f 2f
where is the magnetic permeability, is the electric
permittivity and f is the frequency. The conductivities
increase with the depth into the PML. The efficiency
of the NH-PML increases with its thickness. For a
given thickness best profile for the conductivity is the
geometric series profile such as
0 r ( g 1) l
l* = g (5)
2 ln g
where l = 0, 1, 2 represents the grid point index
inside the NH-PML, 0 is the electrical conductivity
at the PML interface and g is the geometric
286
progression ratio. The absorption profile of the non-
homogeneous PML was established after empirical
investigations. Satisfactory results were found for
0 = 1 mS, g = 2.3 for a layer thickness of 13 cell.
IV. FDTD SIGNAL PROCESSING
The long computation time is a major drawback of the
FDTD method. The FDTD signal was processed in
order to reduce the number of iterations. The wavelet
analysis is based on the possibility of developing a
signal f(t) in wavelet packets as in relation
f (t ) = c j 0 (k ) j 0,k (t ) + d j (k ) j ,k (t ) (6)
k k j= j0
where the functions j 0,k (t ) = 2 j0 / 2 (2 j0 t k ) and
,
j ,k (t ) = 2 2 j0 / 2 (2 j t k ) , (t) is the mother
j /2
wavelet and (t ) is the scaling function.
The wavelet analysis rejects the noise caused by the
finite precision; it also clears the effects of the FDTD
signals of too high frequency, which cannot be
accurately computed. An accurate FDTD simulation
of the propagation of signals of too high frequency,
greater than ~30 GHz in our case, is unnecessary and
it would involve a very fine mesh and extensive
computer resources.
The time step required by the stability criterion is too
small and a conventional Fast Fourier Transform
would provide a low-resolution signal in frequency
domain. Therefore, the FDTD signal is not only de-
noised but also de-sampled. The first nmin points of the
FDTD signal, are used as a training set to the signal
estimation technique. In Fig. 2, nmin = 100 and the
estimated signal y4(t) is a close approximation of the
FDTD signal y3(t). The total FDTD number of
iteration time was reduced to a fifth of the initial
number.
0.05
Signals y3(n) and y4(n)
y3
y4
0
-0.05
100 200 300 400 500 600
Time step n
Fig. 2. The comparison between the the FDTD signal y3(t) and the
estimated signal y4(t)
 V. FILTER DESIGN Two-pole filter

0
Resonators and filters were developed on the substrate
with the characteristics mentioned above. In the first -10

|S21| and |S11| (dB)

approximation, the novel compact resonators were
-20
described by using the transmission line theory as
modified stepped-impedance resonators (SIR) [5]. -30
However, due to the discontinuities effects, the final
geometry for a given resonance frequency was -40

established by FDTD analysis.

-50
For an accurate filter design, the variation of the
coupling coefficients with the relative position of the -60
resonators was investigated by using FDTD. The
200 400 600 800 1000 1200
square shape of the proposed resonators allows a Frequency (MHz)
variety of electric, magnetic and mixed couplings
between resonators. As an example, the dependence
of the magnetic coupling coefficient with the coupling Fig. 4. Two-pole filter response using compact resonators. The
gap s is e is presented in Fig. 3. solid line and the dashed line represent the measured and simulated
response, respectively, for the structure shown in the a) insert. The
dotted line represents the simulated response of the filter shown in
0.08 the insert b).

0.07

0.06
Coupling coeficient

0.05

0.04

0.03
(a)
(b)
0.02

0.01

0 Fig. 5. Four-pole cross-coupled resonator using compact resonators

0 0.5 1 1.5 2 2.5 3 3.5 4
Coupling gap s [mm]
of different type.

Four-pole filter response

Fig. 3. The dependence of the magnetic coupling coefficient versus
the coupling gap s 0

The proposed designs allow direct couplings with the -10 S21 measured
external circuit. The variation of the external quality
S11 and S21 (dB)

S21 FDTD
S21 narrow band model
factor Qext with the coupling line position was also -20
S11 measured
analyzed by employing the FDTD method.
-30
Several two-pole filters were developed by employing
different couplings between resonators. For the two- -40
pole filters in Fig. 4 the coupling between resonators
is the mixed. Each resonator in the insert of Fig. 3 has -50

only a 9.52 mm size, thus the proposed resonator

-60
occupies only 32% of the surface area of an open loop 500 600 700 800 900 1000 1100
resonator designed for the same frequency on the Frequency (MHz)

same substrate. Fig. 6. Measured versus simulated responses of the filters in Fig. 5.
Despite the fact that filter shown in the insert (b) of
Fig. 3 exhibits two transmission nulls at each side of An example of a cross-coupled filter developed with
the pass-band, it is very hard to control the positions new compact planar resonators is shown in Fig. 5.
of these nulls. For an improved filter response, it was Each resonator is 10.64 mm in size. This filter is
already shown that a full control of these nulls can be characterized by a coupling matrix
provided by filters with cross-coupled resonators [3].
The extra negative couplings result in a sharper filter 0 0.0360 0 0.0081
roll-off for an increased filter selectivity. The newly 0.0360 0 0.0259 0 , (7)
developed square planar resonator can be effectively M =
employed for cross-coupled filter design [4, 5]. 0 0.0259 0 0.0360

0.0081 0 0.0360 0

287
and an external quality factor Qext= 17.86. As it is
shown in Fig. 6, the measured response of the four-
pole cross-coupled filter follows closely the simulated
response.
VI. CONCLUSIONS
A 3-D FDTD method was developed in order to
accurately design compact resonators and filters. A
non-homogeneous perfectly matched layer (NH-PML)
with a geometric series profile for the electric and
magnetic absorption was developed as absorbing
boundary conditions. The FDTD signal processing by
using wavelet packets and signal estimation
techniques resulted in a reduction by up to five times
of the computation time.
The FDTD method was successfully applied to small-
size filter design. The proposed resonators occupy
down to 32% of the surface area of a folded half-
wavelength resonator designed on the same substrate
for the same frequency.
288
Two-pole and four-pole cross-coupled filters were
developed for 900 MHz wireless systems such as
GSM and GPRS for the 900 MHz. However, the same
design technique can be applied for devices of the 3G
communications standards. The devices are cost-
effective; they do not require via-holes and any
additional lumped elements.
REFERENCES
[1] A. Taflove (Editor), Advances in Computational
Electrodynamics The Finite Difference Time-Domain method,
Norwood, MA, Artech House, 1998
[2] J.-P. Brenger, Perfectly Matched Layer for the FDTD
Solution of Wave-Structure Interaction Problem, IEEE Trans.
Antennas and Propag., vol. AP-44, 1996, pp. 110-117
[3] G. Lojewski, Computer Aided Design of Some Pass-Band
Microstrip Compact Filters of Quasi-Elliptic Type (in Romanian),
Telecomunicaii, No. 2, 2003, pp. 43-51
[4] M. G. Banciu, G. Lojewski, T. Petrescu, A. Ioachim,
L. Nedelcu, R. Cacoveanu, N. Militaru, D. Brinaru,
D. Ghimpeteanu, Small-Size Filters with Improved Characteristics
for Wireless Communications, Proceedings of the 35th Scientific
METRA Symposium, Bucharest, May 27-28, 2004, pp. 437-440
[5] M. G. Banciu, G. Lojewski, Proceedings of the International
Conference, COMMUNICATIONS 2004, Bucharest, June 3-5,
2004, p. 341-346, 2004
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Suggestions for Availability Improvement of Optical Cables

Ivan Rados1,Tanja Sunaric2, Pero Turalija3
Abstract - This article analyzes suggestions for availability II. ON AVAILABILITY IN GENERAL
improvement of optical cables. There are specially analyze two
different suggestions: decrease numbers of failures respectively Availability A of some system in the time frame is defined as a
increase mean time to failure and decrease mean time of repair ratio of time during which the system is functional in relation to
optical cable. The date, which used in this article, are results of
attended of failure rate during lasting several years exploitation of
the total operational time [2], i.e. its probable that the system is
optical cables in SDH network HT d.o.o. Mostar. Based on the functional in some time frame.
data about failure rates, unavailability and mean down times of MTTF
A= (1)
optical cables are made the suggestions for availability MTTF + MTTR
improvement.
where, MTTF ( Mean Time To Failure) is mean time till the
Key words: failure rate, availability, mean down time
failure occurs and MTTR (Mean Time To Repair) mean time of
repair.
I. INTRODUCTION MTTF = 1 / (2)
where, is the failure rate defined as the number of failures per
The introduction of new services and the need of high time unit.
quantity of data transmission require the high capacity 1 FIT (Failure in Time) =1 failure per 10 9 hours
transmission systems (SDH, WDM). One of the important n
elements of the transmission system is transmission media - in = (3)
this article it is optical cable. The failures - interrupts the MT
communication between great number of users - are making where, n is the number of failures over monitoring time, M the
great losses for network operators [1]. Therefore, the availability length of installed cables in km and T monitoring period in
performances of optical network greatly depend on availability hours.
of optical cables. For the entire optical network, comprising L kilometers of
The HT d.o.o. Mostar began whit installation and using of the cables, mean time to failure (MTTF) is obtained as follows:
optical cable as a transmission media in spring 1994. Ever since 1
MTTFnetwork = (4)
the optical cables have become the main transmission media at cable
all network levels. The date about failure rate, which is L[km]
km
analyzed, refereed to the two periods: from spring 1994. to the Unavailability U is probability complementary to availability
May 01, 1999 - we did not application suggestions for [3], i.e.
availability improvement of optical cable- and from May 01,
MTTR
1999 to May 01, 2001 - we application one of the suggestions U = 1 A = = MTTR (5)
for availability improvement of optical cable. We will explain MTTF + MTTR
after in the article how we decided which suggestions we
application on our network. Based on collected data we are In reporting about system/network performances, unavailability
analyzed the availability performances on which based the U is often expressed as MDT (Mean Down Time) in minutes per
availability improvement is achieved. year [3], i.e.

MDT = 365 60 24U [min/ year ] (6)

1 2 3
Croatian Telecommunication d.o.o. Mostar, Transmission Department
Kneza Branimira bb, 88000 Mostar, Bosnia and Herzegovina
e-mail: ivan.rados@ht.ba, tanja.sunaric@ht.ba, pero.turalija@ht.ba

289
 III. FAILURE ANALYSIS AND AVAILABILITY
CALCULATION
In order to calculate the availability of optical cables, data on
failures and time to repair of optical cables are used. The
collected data referred to the period from spring 1994 to May
01, 1999 and period from May 01, 1999 to May 01, 2001. All
cables are installed sub-surface, in polyethylene pipes and above
them a warning tape was installed as a supplementary way of
protection. Fibers of all cables are standard, single-mode with a
diameter of 9/125m for the use of 1310 nm and 1550 nm
wavelengths.
According to the collected data the main cause of most
failures is outside interference (86,48%), where digging
participates in 72.97% of cases [3]. The vehicle owing to the
improper depth of installed cable causes two failures (5.40%)
and the fire causes three failures (8.10%). Four failures
(10.80%) are the consequence of the planned works by the HT
d.o.o. Mostar. These failures lasted relatively a short time
because of previously well-done preparations.
From the point of view of the optical failures availability we
distinguish [4]:
- failures which break individual fibers in the cable, if there is
no automatic protection, the operator has manually to direct
the traffic to the correct fibers or to repair the faulty once,
- failures which simultaneously break all fibers in the cable,
unless the network is ring, the operator has to repair all
fibers in the cable with no regard to the existence of some
protection mechanisms (optical modules). According to the
collected data in 100% of causes happens the break of all
fibers, either caused by digging or by fire or by vehicle.
Generally, two measures to repair are being used [2]:
- temporary repair time,
- permanent repair time.
Temporary repair time is the time needed for service restoration
after the failure.
This time to repair includes:
- time needed to report the failure to the maintenance team
and their arrival to the telecommunication center,
- time needed for the preparation of splicing material (cable)
and vehicles,
- way to the failure location,
- laying of the new piece of cable (if needed) and its splicing,
- final measurements.

Table 1 Monitoring period, length and number of failures of optical cables

Monitoring period Length (km) Number of failures

spring 1994 May 01, 1999 795 23
May 01, 1999 May 01, 2001 456 14
spring 1994 May 01, 2001 1251 37

Table 2 Failure causes of optical cables until May 01, 2001

Failure causes No. of repor. failures Relativ. % failures

Digging 27 72.97
Installat./workman errors 1 2.70
Defective connector 0 0.00
Fire 3 8.10
Vehicle 2 5.40
Fibers arrangement 0 0.00
Cable displacement 4 10.80
TOTAL: 37 100.00

According to the experience, the time needed to report to the
maintenance team and their arrival to the telecommunication
center is less than an hour. As there is no data on exact distance
from the maintenance centers to the failure location, the time
needed to arrive to the failure location is no special analyzes.
But when the distance from the maintenance team center to the
failure location is short, the influence of the arrival time to the
failure location in relation to the total time to repair is
insignificant.
The greatest influence on the time to repair has the type of
failure, for example: difficult access to damaged cable, necessity
for digging and installing the new piece of cable, cable capacity

290
and splicing of fibers of different manufacturers, unfavorable
weather conditions.
If only two fibers on one cable are actively used, regard to the
availability there are exist two cases:
- repair of active fibers wherewith the system becomes
available,
fibers in the cable (or in more cables) is taken for the calculation
of the time to repair.
Permanent repair time includes, in addition, final storage of
new splicing closures, final construction works and final
protection of a new cable segment.
In this article the temporary repair time is used as mean time to
repair for the availability calculation owing to its influence on
availability.
Until May 01, 1999 HT d.o.o. Mostar had only one team with
three members for maintains of optical cables. Two members
out of three do the splicing and the one do finally
measurements. The maintenance team had only one splicer and
one OTDR, what practically mean that they be able do splicing
only on one side of optical cable (if is necessity for installing the
new piece of cable). It was 60 % of the exact documentation of
installing optical cables. Average time for repair during this
period was 15,70 hours.
From May 01, 1999 HT d.o.o. Mostar took precautions with aim
to decrease average time to repair of optical cable (detail
explanation in chapter 4). The results of this precaution were
decrease average time to repair on 13,43 hours.
That we on the best way see influence to decrease average time
to repair on availability of optical cables in this chapter we are
presume that no decrease its that mean we are use for
calculation average time to repair 15,70 hours.
- repair of all fibers in the cable.
On the area covered by telecommunication network of HPT
d.o.o. Mostar actively exist more transmission systems via the
single mode cable, so the time needed to repair all
Unavailability of optical cables per km is obtained as a
product of failure rate per km of cable and the mean time to
repair as shown in Table 3.
For the unavailability calculation of the optical cable besides
the mean time to repair and failures rate per km it is necessary to
know the failure rates of the splices on the fiber and failure rate
of connectors on the optical distribution frame. Data on failure
rates of splices (30 FIT) and connectors (100 FIT) are taken
from the [5] and [6]. The total length of the cable stage consists
of delivered cable from factory with an avarage length of 4 km.
According to this length the number of splices is calculated as:
Length of cable / 4.
The number of connectors on the optical distribution frame is 2
for the average stage length which is used in this calculation.
The result analysis in Table 4 shows that unavailability
increase almost linearly to the cable length and depends on
failure rate and mean time to repair of optical cables.
For SDH network HT d.o.o. Mostar, mean time to failure
(MTTF) is obtained as follows:
1 h
MTTFnetwork =
460.61 1251 failures
= 1735 h 73 days

Table 3 Failure rate (), unavailability (U) and mean down time (MDT) calculated for optical cables

(FIT/km) U x 10-5 MDT (min/year)

460.61 0.72 3.80

Table 4 Failure rate (-total), unavailability (U) and mean down time (MDT) calculated for different optical link lengths

Length No. of No. of -total MTTR U x10-5 MDT

(km) splices connect. ( FIT ) (h) (min/year)
20 5 2 9562.20 15.70 15.01 78.89
40 10 4 19124.40 15.70 30.02 157.78

Table 5 Failure rate (-total), unavailability (U) and mean down time (MDT) calculated for different optical link lengths
(n=32 failures)

Length No. of No. of -total MTTR U x10-5 MDT

(km) splices connect. ( FIT ) (h) (min/year)
20 5 2 8317.40 15.70 13.06 68.64
40 10 4 16634.80 15.70 26.12 137.28

291
IV. SUGGESTIONS FOR AVAILABILITY IMPROVEMENT
From the availability expression (1) can be seen that the
availability depends on the mean time between failures and the
mean time to repair of optical cables. Availability improvement
can be obtained by increasing the mean time to failures and
decreasing the mean time to repair of optical cables [4].
A. Increasing of mean time to failure
The increase of mean time to failures, relatively, the decrease
of the number of failures can be achieved by preventive
protection of optical cables against digging and by using the
surveillance system for preventive maintenance. As most
failures on the optical cables are caused by digging it is
necessary to attract special attention to it. Although most
countries have laws for preventive protection of underground
cables there are still unsatisfactorily defined punishments (fees)
for their infringements (digging without previous consent). The
law must have to most rigid punishments (invitation prior to
digging). While digging belong to the category of
instantaneous breaks, the others belong to the category
preventive because they are caused by complete loss of cable
characteristics owing to the outside interference.
In our country still no have law about preventive protection of
underground cables, what is a possible conclude on the base of
number of failure during both monitoring periods, most of the
failures were caused digging without previous consent and
transgressor has no adequate punishment. That we are show
influence mean time to failure on availability we will suppose
that we already have the legal regulations regarding the
protection of underground cables during the failure monitoring
period and that, through change of that law, the mean time to
failures increased from 73 to 84 days, which means that the
number of failures decreased from 37 to 32, representing a
decrease of about 13 %, and that the mean time to repair
remained same, i.e. 15.70 hours. As the failure rate is just
proportional to the number of failures, so, decreasing the
number of failures also decreases the failure rate by 13.5 % or to
398.37 FIT.
As seen in the Table 5, the decrease of the number of failures
resulted in the availability improvement, relatively, the decrease
of MDT, for example, for d=20 km from 78.89 to 68.64
min/year or by 12.99%. Surveillance system for preventive
maintenance can foresee the possible failure location using the
metal protective layer on the cable as a sensor. Surveillance
system alarms when the entirety of the outer sheath or the
splicing point is being breaked, indicating that potential failure
should be removed. As optical cables installed within the HPT
d.o.o. Mostar have been exploited a short time (the first one
about seven years), there were no deterioration as yet of the
292
cable characteristics caused by the outside interference. For the
preventive failure protection against outside interference (long-
term exploitation) it will be necessary to install the surveillance
system in order to foresee a failure. Costs for installation of such
a system would be slight compared to with failure losses on the
cable.
B. Decreasing of mean time to repair
From the unavailability expression can be seen that
availability depends on the mean time to repair. In order to
decrease the MTTR it is essential to have a maintenance plan
which should contain the following components [4]:
- exact documentation,
- maintenance team,
- training,
- equipment,
- plan of action,
- practice,
- continued process of improvement.
Exact documentation on optical cables is one of the most
significant components for diminishing the MTTR. It includes:
cable traces, number of failures in cable, fibers attenuation,
splicing points, cable lengths, trace marking and outer-metal
shield condition. Additional 34% of documentation made during
period from May 01, 1999 to May 01, 2001, that mean than till
now we made documentation for 94% of installing optical cable.
Also it procured one mobile computer for frequent modification
and bring up to date. Only two persons have access to that base
and they are responsible for its processing. Besides the exact
data base for diminishing the MTTR it is necessary to exactly
know where are the tools and material needed to repair, as well
as the key to the entrance of the building. Plan of action contains
instructions on who is calling whom and when, as well as the
numbers of fixed and mobile telephones. Now, maintenance
team has seven members: five on the failure location and two at
terminals (one in each). Four members out of five do the
splicing (two teams of two members) and the fifth have the
radio connections with the members at terminals.
Maintenance staff has to know to use the splicers and measuring
equipment. Owing to the ever-improving measuring and
connecting equipment for different types of cables, regular
training of the maintenance team is very important because each
improvement which leads to diminishing the time to repair
increases the availability of optical cables. In HT d.o.o. Mostar
they have training two times a year at least in order to acquire
new knowledge. Training in the field is more purposeful
measure than the classroom teaching. Well planned and sudden
exercise is the best way of the emergency staff training (one per
year). The aim of each exercise is to achieve better results each
time.
 Quantity and kind of equipment depend on geographical
spreading, network size, and the number of skilled staff. If
network is too large and geographically spread there must be
more maintenance teams. As network of HT d.o.o. Mostar no
geographically spread, for the now is sufficient one maintenance
team of optical cable. Our maintenance team has one
reflectometer (OTDR) for measuring at 1310 and 1550 nm, two
splicers with tools (cutter, air, screwdrivers...), optical power
meter, voltmeter and the car.
Using above mentioned suggestions, the MTTR of the cable is
obtained to 13.43 hours. The availability would also be
improved considerably, as shown in Table 6. In the concrete, the
MTTR decrease of 14.45 % results in the availability
improvement of 14.46 % or to, the decrease MDT from 78.89 to
67.48 min/year.
Every greatest availability improvement would be achieved
by the simultaneous decrease of the number of failures (32) and
the decrease of the mean time to repair of the cables (13.43
hours), as shown in following table. In the concrete, mean down
time of failure is decrease for 25.57%, or to decrease from 78.89
to 58.71 min/year.

Table 6 Failure rate (-total), unavailability (U) and mean down time (MDT) calculated for different optical link lengths
(MTTR=13.43 h)

Length No. of No. of -total MTTR U x10-5 MDT

(km) splices connect. ( FIT ) (h) (min/year)
20 5 2 9562.20 13.43 12.84 67.48
40 10 4 19124.40 13.43 25.68 134.97

Table 7 Failure rate (-total), unavailability (U) and mean down time (MDT) calculated for different optical link lengths
(n=32 failures, MTTR=13.43 h)

Length No. of No. of -total MTTR U x10-5 MDT

(km) splices connect. ( FIT ) (h) (min/year)
20 5 2 8317.40 13.43 11.17 58.71
40 10 4 16634.80 13.43 22.34 117.42

V. CONCLUSION REFERENCES

Data on failures and time to repair, which are analyzed in this
article, refer to the 7 years exploitation of optical cables within
the HT d.o.o. Mostar transmission network. The analysis show
that the most frequent cause of the optical cables break is
digging (72.97%) and regardless to the break cause there has
been breaks of all fibers in the cable. The analysis of temporary
repair time shows that it mostly depends on the type of failure
and cable capacity. Availability improvement of optical cables
can be achieved by increasing of mean time to failure,
relatively, decreasing the number of failures as the most
frequent case of break, and decreasing the mean time to repair.
The law on underground cables protection and monitoring
system for preventive maintenance would be the cause of
decreasing the number of failures. The mean time to repair cable
is decreased considerably by using the plan of maintenance. An
approximate unavailability can be used to evaluate availability
of different structures.
[1] C. Coltro, Evolution of Transport Network Architectures, Alcatel
Telecommunication Review, 1st Quarter 1997, pp.10-18,
[2] I. Jurdana and B. Mikac, An Availability Analysis of Optical Cables,
Proceedings WAON`98, pp. 153-160,
[3] T. H. Victor, Update on Interim Results of Fiber Optic System Field
Failure Analysis, Bellcore, New Jersey,
[4] I. Rados, Availability analysis of Synchronous Digital Hierarchy
Network, Master thesis, University of Zagreb, 2000, (Croatian).
[5] D. Gardan, Availability analysis of the fibre optic local loop,
Optical Access Networks, EFOC & N, pp. 28-32, 1994.
[6] P. N. Woolnough and N. E. Andersen, FITL System Recommendations
Final Report, Deliverable FIRST 1.0.11, pp. 1-42, 1 May 1996.

293
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Simulations of impulse response for diffuse indoor

wireless channels
Mihai Telescu1, Laura Ghisa2, Pascal Besnard3, Adrian Mihaescu4
Abstract Calculating channel impulse response is means that the incidence angle is never taken into
important in all forms of communications. In the case of account. In other words not only is the original source
diffuse infrared wireless networks channel response Lambertian but so are all the secondary sources the
varies abruptly with changes in receiver/emitter position walls or furniture. This isnt usually a big
so the development of tools capable of estimating this
response in any circumstances is even more important.
approximation since many materials act as purely
Nowadays John Barrys spatial discretisation algorithm diffusive reflectors. However it is to see what happens
is the most frequently used method of calculus [1]. In if we do take in account specular radiation as well
this paper we show a way to improve the algorithms (paragraph IV).
speed and a way to account for the influence of both
diffuse and specular radiation.
Keywords: diffuse infrared, indoor free space II. THE ALGORITHM
communications, wireless channel, pulse response
We suppose that a source S emits a unit pulse at time
I. INTRODUCTION 0 inside a rectangular room where we have one
receiver R. In a first approach we suppose that the
Radio wireless dominates the market environment is purely diffusive, governed by
nowadays with a variety of systems such as WiFi, Lamberts law:
Bluetooth, UWB. However optical infrared, especially
indoor, remains very practical in certain applications, n +1
either as an alternative to radio or as a complement. R S ( ) = P cos n ( )
Optical wireless users enjoy better privacy since, 2 S (1)
contrary to microwaves, light does not pass through [ / 2, / 2]
walls. Optical security is the only medical issue that
these networks pose, unlike microwave, which rise
more and more concern. Electromagnetic Part of the light arriving on the receiver will
compatibility is no longer a problem, making indoor come directly from the source (if a direct pass exists).
infrared communications viable even in such This is the 0 order impulse response.
environments as hospitals.
Research is led nowadays to increase bit rate n +1 n
h0(t; S, R) cos ()d rect( / FOV) (t R/ c) (2)
in optical wireless networks and to find multiple 2
access solutions.
In 1993 John Barry published a method of The rest will be reflected on the walls of the
calculating indoor channel impulse response by room so that the total impulse response will be:
dividing the total surface of the room in elementary
surfaces and calculating the respective k order
response for each of these surfaces as a function of the
k-1 responses of all surfaces. Because of the recursive
h(t ; S , R ) = h k (t; S , R) (3)
k =0
calculus the result is slow and several optimizations where each term hk represents the pulse response
have been brought since its invention [2]. While given by the light arriving on the receiver after a
developing a link simulator we have brought our own number of k bounces.
contribution (chapter III).
To this day impulse response simulations have
used purely diffusive environments, which basically
1,2,4
Facultatea de Electronic i Telecomunicaii, Dept.Comunicaii
Bd. V. Prvan Nr. 2, 300223 Timioara, mihaescu@etc.utt.ro
3
ENSSAT, Lannion, France

294
 h ( k ) (t ; S , R ) =
= h ( 0) (t ; S , {r , n, FOV , dr 2 }) h ( k 1) (t ; {r , n,1}, R )
S The enormous volume of calculus renders
(4)
But, following John Barrys method, if we
consider the walls discrete, made up of N indivisible
elementary surfaces the function becomes:
h ( k ) (t ; S , R ) h (0) (t; S , i ) h (k 1) (t; i , R) (5)
and more explicitly:
n +1
h( k ) (t; S , R)
2 But we can gain more in terms of speed if wee
N
r cosn ( ) cos( )
( d2
rect( / FOV ) (6)
i =1
h ( k 1)
(t d / c;{r, n,1}, R)A)
where A is the area of the receiver and the
rectangular function is defined above.
Figure 1 Source Receiver configuration
A program has been written implementing this
algorithm. It calculates hk for a given number of
parameters. In the figure bellow we show our results
for the standard 5/5/3 room used by most authors with
the typical parameters [1]. A k=3 maximum reflection
order was taken into consideration.
Figure 2 Standard room pulse response
III. OPTIMIZATIONS
spatial numerical algorithms slow even on powerful
machines so any amelioration in speed is more than
welcome. We have considered the two possible
solutions proposed in [1]: a direct implementation of
(4) and an implementation using look up tables. The
first is more suitable for inferior k where high
resolution is needed. The second works better for the
superior reflection orders where a high spatial
resolution (high N) is needed.
Certain aspects are to be considered during the
implementation. Computing the vectors absolute
value not by using the Euclidian distance should do
for example calculating the distance between a
source-receiver pair, which is needed to calculate the
delay.
consider physical reality carefully. Light undergoing k
bounces between point A and point B will always
travel slower than light undergoing k-1 bounces under
the same points A and B. Therefore the first non-zero
sample that appears when calculating hk will appear
after the first non-zero sample hk-1. It therefore
becomes useless to calculate a certain number of
points as shown in figure 2.
h(0)
h(1)
h(2)
h(3)
It is useless to calculate the value of
these points, the will most certainly
equal 0
Figure 3 Optimized solution
IV. RESULTATS USING PHONGS LAW
As we have shown in Chapter I we have considered a
purely diffusive reflection model as used in previous
publications. In reality however a small part of the
total radiation is specular. It is quite hard to model the
complete phenomenon taking into account both
specular and diffusive radiation.
In image processing, however, Bui Tong Phong
proposed in 1975 an empirical reflection model,
which served to generate computer images, this
seemed more plausible to the human eye [4]. The
model is realized by taking into account the specular
radiation that becomes significant when dealing with
smooth surfaces.
295
 A variant of Phongs law was also used by Yang
PHONG
and Lu to calculate infrared illumination diagrams in a
room, [4].

Echelle linaire
We have used a similar variant to compute
impulse response. The Lamberts law (1) is replaced
by Phongs law:
1
RS (, ) = [rd cos() (1 rd ) cosm ( )]

[ / 2, / 2] (8)
[ / 2, / 2]
where rd represents a coefficient indicating the
percentage of diffuse radiation and m is a parameter s *10
-11

describing specular radiation. For rd = 1 Phongs law b)

becomes a first order Lamberts law Figure 3 Lamberts law results (a) vs. Phongs law
The equations we obtain for h0 and hk are: results (b)

Wall Material Reflection 1st Phong 2nd Phong

1
h0 (t; S , R) [rd cos( ) (1 rd ) cosm ( )]d coefficient coefficient coefficient
(9) Ceiling Wall 0.184 1 1
paint
rect( / FOV) (t d / c) Floor Wooden 0.128 0.6 6
boards
West Brown 0.0884 0.5 2.8
shelves
East, Glass 0.0625 0.001 13
h ( k ) (t ; S , R ) North,
South
1 r [rd cos( ) (1 rd ) cos m ( )] cos( ) Table 1
d2
(10)

rect ( / FOV ) h ( k 1) (t d / c; {r , n,1}, R)A V. CONCLUSIONS

Wireless infrared communications can replace
wireless radio, sometimes with better results and
lower costs. It is however important to observe that in
In figure 3 we show the differences the results the case of non-directive indoor networks only a very
obtained when using Phongs and Lamberts law. small fraction of the total radiation emitted by the
The parameters of the source receiver and channel are source actually arrives on the receiver. For every Watt
listed in table 1. emitted less than a Watt is received in most cases.
This fact and combinded with strict eye safety power
LAMBERT limitations are in fact the main problem.
Under these circumstances it is even more
Echelle linaire

important to conduct thorough research to a optimized

model of the transmission channel.

REFERENCES
[1] J.R. BARRY, J.M. Khan & all Simulations of Multipath
Impulse response for Indoor Optical Channels
[2] Y. A. Alquad, Mohsen Kaverhad, MIMO Characterisation
of Indoor Wireless Optical Link Using a Diffuse-Transmission
Configuration IEEE Transactions on Communications vol
-11
51 No 9 September 2003
s *10
[3] Bui Tong Phong Illumination for Computer Generated
a) Pictures Communications ACM 1975
[4] H. Yang, C. Lu Infrared Wireless LAN using Multiple
Optical Sources IEEE proceedings 2000

296
 Buletinul tiinific al Universitii "Politehnica" din Timioara
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TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Experimental characterization of impulse response for

optical indoor wireless channels
Laura Ghisa1, Mihai Telescu2, Pascal Besnard3, Adrian Mihaescu4
Abstract Indoor infrared wireless communications can different shapes and sizes, is described in [2].
offer in certain applications a valuable alternative to Moreover computer estimation of impulse response
wireless radio. Its main advantages being: low cost, becomes difficult when important features like the
medical safety and unlicensed spectral band. The most inclusion in the model of the effect of office furniture
suitable infrared band seems to be 1550nm because of
the low noise and optical safety, even if sensible
and people is needed [3].
receptors at this wavelength are still hard to find at the
edge of current technology. In this paper we show an
experimental set-up that allows us estimate the impulse
response function at the 1550nm wavelength for indoor
transmissions using a reduced-model of the actual
indoor environment.
Keywords: diffuse infrared, indoor free space optics,
wireless channel, impulse response

I. INTRODUCTION

Infrared (IR) is a medium extremely suitable for

short-range flexible indoor communications such as
ad-hoc wireless local area network [1]. More, high
quality wireless access to information networks and
computing resources by users of portable computing
and communication devices is achieved via infrared
wireless links with low delay, high data rates, and
reliable performance. All of these need an accurate
characterization of the channel and understanding of
the performance limits and design issues for wireless
optical links. Indoor wireless IR channels are based
in one of the following emitter-to-receiver
configuration, fig.1: (i) line of sight. (LOS); (ii) quasi-
diffuse (Non-LOS & Directed or Hybrid); and diffuse
(Non-LOS & Non-Directed). The operation of LOS
channel relies on a free path between emitter and
receiver, which have to be pointed to each other. In
quasi-diffuse channels, emitter and receiver have to be
directed to the same surface, that reflects part of the
optical beam toward the detector. In diffuse systems
emitter and receiver have wide radiation and
collecting patterns and the optical and optical signal
may go into receiver after multiple reflections on
surfaces that surround the communication cell. A
standard algorithm, implemented to estimate the
impulse response and frequency response of the
infrared channels in empty indoor environments of
1,2,4
Fac. de Electronic i Telecomunicaii, Dept.Comunicaii
Bd. V. Prvan Nr. 2, 300223 Timioara, mihaescu@etc.utt.ro
3
ENSSAT, Lannion, France
Fig. 1 Different indoor wireless IR channels
II. CHANNEL MODEL. METHODS TO GET
CHANNEL IMPULSE RESPONSE.
Usually for optical wireless communications is used
an intensity modulation with direct detection (IM/DD)
system. In this case the propagation environment can
be replaced by an equivalent filter. The transmitted
waveform is the instantaneous optical power of the
infrared emitter. The received waveform is the
instantaneous current in the receiving photodetector,
which is proportional to the integral over the
photodetector surface of the total instantaneous
optical power at each location. The channel can be
modeled as a baseband linear system as indicated in
Fig .2 and equation 1.
Optical power Photocurrent
Rh(t) RH(f)
X(t) Y(t)
Signal-independent
noise n(t)
Fig. 2. Modeling link as a base-band linear, time-
invariant system

297
 x 2 y 2 z 2 x 2 ( R) y 2 ( R) z 2 ( R)
R= = = = = = =
y (t ) = x(t ) Rh(t ) + n(t ) (1) x1 y1 z1 x1 ( R ) y1 ( R) z1 ( R )
Where R is the detector responsivity, n (t) is signal- x 2 (S ) y 2 (S ) z 2 (S )
independent additive noise and h(t) is the channel = = = (3)
impulse response, [4]. x1 ( S ) y1 ( S ) z1 ( S )
The channel can be described in terms of frequency A2
R2 = ( 4)
response: A1

h(t )e
j 2ft
H( f ) = dt (2) 1.5

which is the Fourier transform of h(t).

h0[arbitrary units]
1
Channel response can be determined experimentally
using three methods:
1. The direct method
0.5
A short impulse is emitted and the response is

room
RMC
measured directly. Although this method is
successfully employed in low frequency applications,
for example in acoustics. It is hard to use in infrared. 00 1 2 3 4 5 6 7 8 9 10
2. The spread spectrum method t[ns]
A pseudo-random binary sequence is transmitted and Fig.3. The impulse response for the direct link
the received signal is cross-correlated with the input
sequence to yield the impulse response. For the multi path components a correction is needed:
3. The frequency sweep method R scales the time axes and the function amplitude is
Signals are transmitted by sweeping through a finite
scaled by R . In Fig.4 we show the high orders
range of frequencies and the complex channel
response captured for the entire frequency range. The impulse response for R=2.
broader the range of frequencies considered, the
closer is the approximation of the measurements to 0.1

the channel transfer function. 0.09

This is the method we chose to implement. 0.08

0.07
RMC
h1[arbitrary units]

0.06
III. MESUREMENT OF THE IMPULSE
0.05
RESPONSE FOR A REDUCED SIZE MODEL
0.04

0.03
Here we show that the impulse response function of
0.02
an actual room can be estimated once we measured
the impulse response function of a reduced model
0.01
room
0
0 10 20 30 40 50 60 70
channel (RMC). This is our most important theoretical
result. It offers a high degree of flexibility to our t/R[normed
experiment allowing us to test real environment time]
conditions in a laboratory. It allows us to anticipate
technological advances in infrared 1550nm receivers Fig.4. The impulse response for the reflected links
by performing experiments with existent low-cost
photodiodes. The measurement resolution
In order to estimate the response of a room we must If we reduce the room dimensions we reduce both
build our model keeping certain proportions. For this spread distance and time, so we need a wider band for
we used a link simulator of impulse response for the measurement system.
diffuse indoor optical wireless channels, [2]. c
f max = (5)
Let x1, y1, z1 be the dimmensions of the room, x1(R), d
y1(R), z1(R), and x1(S), y1(S), z1(S) the coordonates of d c
the reciever and of the source respectivlly. We will x = = (6)
3 3 f max
have x2, y2, z2, x2(R), y2(R), z2(R), x2(S), y2(S), z2(S)
for the model. A1 is the size of the reciever used in We find the relationship between the spread distance
the real room and A2 is the size of the reciever used (d) and the band with of the measurement system
in the model. (fmax) in equation (5), where c is the speed of light.
We have empirically shown that if the relations (3) Consequently the spatial resolution (x) of the model
and (4) exist between the model and the real room is given by equation (6).
than the impulse response given by the direct optical For example in a 1mx1mx0.60m model for 1ns
path care of identical amplitude and differ only by a temporal resolution one needs 30cm of spread
time offset, as shown in Fig. 3. distance, which corresponds to a spatial resolution of

298
17.3 cm and needs 1 GHz of the band with of the V. RESULTATS
measurement system. Other values are shown in Table
1.It's obviously that a smaller temporal of resolution is Here we present the measured impulse response for a
demanded than a greater measurement frequency is direct link in a 1mx1mx0.60m reduced model room.
needed. The distance between the emitter and the receiver is
62 cm. The measurement frequency is 2 GHz which
Table 1 assures a temporal resolution of 0.5 ns and a spatial
Measurement Temporal Spatial resolution resolution of 8.66 cm (see table 1).
frequency resolution [cm]
0.1
[GHz] [ns]
1 1 17.3
0.08
2 0.5 8.66

h0 max 0.102544
2.5 0.75 13 0.06
10 0.1 1.73
0.04
tr= 2.003ns
IV. EXPERIMENTAL SET-UP 0.02

A sketch of the experimental set-up is shown in 0 20 40 t[ns]

Fig.5. The system can measure by means of the
spectral method the impulse response of all types of
channels presented in figure 1, in all kind of rooms or
environments of different size and shapes, empty or
filled with office furniture and people. The key of the
experiment relies on a reduced model channel (RMC)
of the real environmental channel to be tested. Here
we demonstrate that between the two channels there is
a scale factor relationship. The RF output a HP 8753A
vector analyzer is swiped from 300kHz to 3GHz and
than applied to an external optical modulator of a
1550nm laser diode optical signal in order to provide
after splitting and detection, the reference R(f) and the
output of the RMC A(f) signals. The impulse
response is obtained by inverse Fourier transform of
the frequency characteristics of the reduced model
channel: H(f)=A(f)/R(f). Compared to [4] the
advantage of RMC is obviously: a measuring dynamic
of 60-70 dB available using optical amplification and
common 1550 nm emitters and receptors is sufficient
to estimate superior impulse response bounds (k=2.3),
which is not the case for real environments.
Vector analyzer
GPIB
RF R A B
PD-PIN
LD Modulator Reduced model
CHANNEL
20
Splitter
80
Optical Amplifier Diffuser
Fig. 6. The measured impulse response for the direct
link
The measurement result is illustrated in Fig.6. A
Dirac form impulse response is obtained at
tr=2.003ns which is in good agreement with the
calculated freeway propagation time between the
emitter and the receiver.
VI. CONCLUSIONS
We have presented a method and an experimental set-
up for measuring the impulse response for indoor
infrared channels. All kind of real indoor
environments can be measure on a reduced model
channel. We have shown that between the real
channel and the model exists a mathematical
relationship.
REFERENCES
[1] IEEE Wireless Communications-Special issue on Optical
Wireless Communication-April 2003
[2] M.Telescu, L. Ghisa, P. Besnard, A.Mihaescu - "Simulation of
impulse response for diffuse indoor optical wireless channels"-to be
presented to ETC'2004, Timisoara, Oct.2004
[3] M. Abtahi, H. Hashemi - "Simulation of indoor propagation
channel at infrared frequencies in furnished office environements",
1995-IEEE, Vol.0-7803-3002-1, p.306
[4] J. M. Kahn, J. Kreuse, J. B. Carruters Experimental
characterization of non-directed indoor infrared channels, IEEE
APD
Trans. Comm., 43,2/3/4,1995,p.1613

Fig. 5. Experimental set-up

299
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

A discrete model for reference source noise in indirect

frequency synthesis
Teodoru Emil, Demeter tefan11

Abstract-Indirect frequency synthesis preserves its

actuality and is still offering good possibilities to achieve 0 (s) K 0 K F(s)
performing systems. Practical problems refer to = = S (s) (1)
i (s) K 0 K F(s)
obtaining competitive acquisition times, under an s+
intense traffic and noisy environment. The paper is N
focusing on significant aspects such as transfer
functions, signal/noise ratio. This paper presents a brief where:
description of the theoretical principle of these systems, i(s) Laplace image of input phase variation due to
especially those concerning aspects of the noise content
in output signal due to reference source. A Matlab
the perturbation;
support for solutions of the applications and simulation 0(s) Laplace image of the output phase variation
is also used. due to i(s);
Keywords: transfer function, reference source noise. 0(s) output signal with perturbations;
K0[rad/s/V]- the VCO gain;
I. FREQUENCY CHARACTERISTIC OF THE K[V/rad] the gain of the phase detector in phase
REFERENCE regime ;
F(s)- Laplace transform of the loop filter transfer
If we are to analyse the behaviour of an numerical function;
frequency synthesizer when the reference oscillator N the integer divider ratio of the programmable
or voltage controlled oscillator signals are frequency divider of the loop.
or phase modulated by an useful or noise signal, we For a type 2 second order loop the function (1)
must determine frequency and phase characteristics becomes:
versus the useful or noising signal.
Consider an indirect frequency synthesizer like in
0i (s) 2 n s + 2n
fig.1, with frequency variable reference phase i(s). = N 2 (2)
0 (s) s + 2 n s + 2n

where:
- damping factor of the loop;
n natural pulsation of the undamped system.
With the substitution s = jm in (2) we have the
response of the reference oscillator at a noise
component of frequency m:
0 i ( j m ) 2j n m + 2n
= N (3)
0 ( j m ) 2m + 2n m + 2n
The transfer function of the reference oscillator noise
is:

1
Academia Fortelor Terestre Sibiu, Catedra de Stiinte tehnice
Bd. Revolutiei 1-3, e-mail: eteodoru@actrus.ro

300
 II. EQUATIONS 2
T T
m 2 = n + 2 n +1
If some elements (phase detector, frequency dividers) 2 2
2
of the system are digitally ones, a digital approach is T (9)
possible, so that the noise transfer function is more m 1 = 2 + n
2
useful with z transform: 2
m 0 = n T 2 n T
+1
0 i ( z ) 2
n z + n 1z + n 0 2 2

= 2 (4)
i ( z ) m 2 z 2 + m 1 z + m 0
respectively:
The equation with finite differences for the input
i(n) and output 0i(n) samples is: T
2
n 2 = n T + n N
2
1
0i ( n ) = [ n 2 i ( n ) + n 1 i ( n 1) + ( n T ) N
2
m2 n 1 = (10)
1 2
+ n 0 i ( n 2 )] [ m 1 0 i ( n 1) + (5) T
2
m2 n 0 = n + n N
+ m 0 0i ( n 2) 2

With bilinear transform method the transfer function The relations (9) and (10) allow the obtaining of the
in z is: noise levels of the output versus the reference noise,
using Simulink models.
0i (z)
=
i (z)
2 2 2
a2 + a1 T + a0 T z2 + 2a2 + a0 T z + a2 a1 T + a0 T
2 4 2 2 4 III. DIAGRAMS
=
2 2 2
b2 + b1 T + b0 T z2 + 2b2 + b0 T z + b2 b1 T + b0 T We realize two situations:
2 4 2 2 4
a) = 0,707; N = 1000; n T = 2 0,1 (11)
(6) generating the following sets of coefficients m and n:

m 2 = 1,045
where:
m1 = 1,998 (12)
m = 0,957
a 2 = 0 0

a1 = 2n N (7) n 2 = 45,409
2
a 0 = n n 1 = 1,974 (13)
n = 43,445
and: 0
b2 = 1

b1 = 2n (8) The Simulink model is given in fig. 2, and the
2 diagrams of the input/output noise in fig. 3,4, in time
b0 = n and frequency domains, respectively.
The noise is simulated with a gaussian noise generator
T is the period of the reference signal. and we search to observe the response of the system
Making the identification between the relations (6),(7) regarding this signal. So we use both a spectral
and (8), the coefficients m and n are: analyser and a oscilloscope to the output.

301
 Periodogram

10^(-5)
Generator zgomot
gaussian Periodogram 1
Constant Product1 Periodogram1

1 1
Scope1
0.956 z z
-43.435
Product3 Unit Delay Unit Delay1
1/m2
Product4
n0
Sum

1.974

n1 Product2
Periodogram

45.409
Periodogram Frequency
Vector Scope1
n2 Product5

-1.998

m1

Scope2
Product6 Sum2

0.956 z

1/m22 Product7 Unit Delay3

Product8

Unit Delay4

1
0.957
z
m0

Fig.2. The Simulink model for =0,707; N=1000; nT=20,01.

Fig.4. Noise diagrams of the system

output in time and frequency domains.

b) = 0,707; N = 1000; n T = 2 0,02 (14)

The sets for m and n:

m 2 = 1,543

m1 = 1,803 (15)
m = 0,654
0

n 2 = 542,917

n 1 = 197,392 (16)
n = 345,525
0
Fig.3. Noise diagrams of the system
input in time and frequency domain. The Simulink model and time and frequency diagrams
are in fig. 5, 6, and 7, respectively.

302
 Periodogram

10^(-5)
Generator zgomot
gaussian Periodogram 1
Constant Product1 Periodogram1

1 1
Scope1
0.91513 z z
-84.857
Product3 Unit Delay Unit Delay1
1/m2
Product4
n0
Sum

7.88

n1 Product2
Periodogram

92.74384
Periodogram Frequency
Vector Scope1
n2 Product5

-1.9960

m1

Scope2
Product6 Sum2

0.91513 z

1/m22 Product7 Unit Delay3

Product8

Unit Delay4

1
0.91514
z
m0

Fig.5. The Simulink model for =0,707; N=1000; nT=20,02.

Fig.7. Noise diagram of the system

output in frequency domain.

IV.CONCLUSIONS

Based on the relations (9) and (10) we calculated the

coefficients mk, nk ( k = o,1,2) of the noise transfer
function. These values were introduced in the finite
differences equation (5) and this equation was then
represented in a Simulink model. We use the value
0,707 for the damping factor (allowing an optimal
dinamic response), a value of 1000 for the dividing
ratio N of the programmable divider of the loop and
the normate frequency n T = 2 0,1 in case a) and
20,02 in case b). The system was tested with
Gaussian distributed random input signal.
Analysing the diagrams we can see that the
synthesizer has the behaviour of a low-pass filter in
Fig 6. Noise diagrams of the system
the loop in regard to the input noise. The
input in time and frequency domains. characteristic depends on damping factor of the

303
system and normate frequency nT. The slow phase
variations are transmitted to the output.
The synthesizer works like a phase tracking system
with gain of N. Because the reference source noise
has a contribution in the spectral density of the output
noise in [6]:

2n s + 2n
Si () N 2 (17)
s 2 + 2n s + s 2

where Si() is the reference source noise spectral

density, the conclusion is that despite of the small
value, this noise appears multiplied by N2. If the
dividing ratio N is variable (to obtain all the
frequencies in the work range) then the lowest
frequencies will produce the lowest reference noise
effect.

REFERENCES

[1] Egan,W.F.- Frequency synthesis by phase lock,

2nd ed., John Wiley & Sons, 2000.
[2] Rhode, U.L. - Microwave and wireless
synthesizers, John Wiley & Sons, 1997.
[3] Scarlatescu, M.- Emitatoare radio,p.a II-a, Ed.
Academiei Tehnice Militare, Bucuresti, 1996.
[4] Mateescu, A., Ciochina, S., Dumitriu, N.,
Serbanescu, A., Stanciu, L. Prelucrarea
numerica a semnalelor, Ed. Tehnica, Bucuresti,
1997.
[5] Dragu,I., Iosif, I.M. - Prelucrarea numerica a
semnalelor discrete in timp, Editura Militara,
Bucuresti, 1985.
[6] Bechet, P. .- Cercetari privind dezvoltarea
structurilor de sintetizoare de frecventa pentru
aplicatii speciale, Teza de doctorat, UTCluj-
Napoca, 2001.

304
 Buletinul tiinific al Universitii "Politehnica" din Timioara
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TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
A Physical Laboratory for Smart Transducers Education
Virgil Tiponut1
Abstract In this paper, the development of a physical
laboratory for practical activities in the field of smart
transducers is presented. The themes of the laboratory
works are close related to the curricula of the Smart
Transducers course and are focused on the functionality
and the structure of the most used type of smart
transducers, the transmission and processing of
acquired data. The paper will highlight that by carefully
choosing the hardware and the software components, the
cost of the implementation can be kept low without
scarifying the educational purposes.
Keywords: sensors, actuators, smart transducers,
educational tool.
I.. INTRODUCTION
Many universities are now offering courses and
laboratory works in the field of sensors/actuators, in
order to improve the knowledge of the graduates in
the field. However, teaching students on sensorics in a
laboratory, or training technical staff, may be a
difficult task. Different approaches to this problem
has been used in order to overcome some difficulties
that arise.
The Virtual Laboratory is the simplest solution with
some other advantages: easy of access for technical
community, no constrains on the topics to be studied,
users could teach themselves by selecting their
particular route of their interest, etc. [1]. The main
drawback of virtual laboratory is the limited level of
education, no practical experience being offered to the
users.
Distance Access Laboratory is another educational
tool, used for acquiring practical skills in different
areas of techniques [2] [3]. Now the users can access
the lab experiments from anywhere at any time and
follow them at their own peace. Distance Access
Laboratory solution allow the users to be much closer
to the physical phenomena for an affordable cost of
the investment.
The conventional laboratory works, in despite of their
disadvantages: limited number of participants at a
time, no remote access, high level of investment, are
the most effective educational tool when the practical
skills are a primary goal.
In this paper, the problem of the implementation of a
physical laboratory will be approached. This
laboratory has only educational purposes, and is
1
Facultatea de Electronic i Telecomunicaii, Departamentul Electronica Aplicata
Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail: virgil.tiponut@etc.utt.ro
305
entirely devoted to practical activities with
undergraduates students in the field of sensors and
actuators.
There are different strategies of teaching sensorics,
depending of the personal experience of teaching
staff, the existing curriculum of the branch of
electronic engineering and the most important
targeted audience, who are, here, the students.
In some cases, the educational interests are focused on
sensor technologies, sensor structure and sensing
effects [1]; in other cases, the general structure and
applications of sensors/actuators in different areas of
industry are studied (industrial measurement, fields of
automatic, robotics ,etc.). It is important to point out
that in the first case, it is quite difficult or even
impossible to implement laboratory works on that
topics.
In the present paper, the new approach of teaching
sensors/actuators is to include in the planned
educational material all important aspects necessary to
a graduate, to successful integrate of a transducer into
applications in a networked environment. According
to this strategy, the following steps are necessary in
order to develop low cost/high efficiency laboratory
works on sensors/actuators:
Starting from the schematic structure of a
smart transducer (ST), we will establish first
the most important aspects covered by Smart
Transducer course;
Based upon the content of the course, the
topics of the laboratory works will be
developed then;
In the next step, the necessary hardware and
software resources will be chosen, in order to
meet all requirements of the task and to keep
down the cost;
Finally, laboratory works for each type of
ST have to be developed.
In this way, a good balance between the cost of the
implementation and the educational efficiency of the
laboratory works can be obtained.
Please observe that we have used in the above the
term Smart Transducer. This is a general term
introduced by IEEE1451 Standard; transducer
includes both sensors and actuators. A Smart
Transducer is A transducer that provides functions
beyond those necessary for generating a correct
representation of a sensed or controlled quantity. This
functionality typically simplifies the integration of the
transducer into applications in a networked
environment. A Smart Sensor is A sensor version of
a smart transducer [4].
We highlighted the above mentioned terms because
they are important entities in IEEE1451 Standard, and
the proposed content of sensors/actuators course and
the corresponding laboratory works will be developed
around this standard.
II. PLANNED EDUCATIONAL MATERIAL
The IEEE1451 is a family of standards for connecting
ST (sensors and actuators) to networks. These
standards will enable network-capable but network-
independent plug-and-play transducers for use in
embedded products, distributed data acquisition and
control systems, and networked controllers.
The key elements of a ST, according to the IEEE1451
Standard, are depicted in Fig.1 [4] [5].
Fig.1. Schematic diagram of an IEEE1451 ST
A ST includes a Smart Transducer Interface Module
(STIM) and a Network Capable Application Processor
(NCAP). The field break between STIM and NCAP is
a standardized Transducer-Independent Interface
(TII).
Note that the transducers themselves are considered
part of the STIM. Actually, in order to provide the
critical self-identification features (stored in a
Transducer Electronic Data Sheet TEDS), the
transducers must be inseparable from the STIM block
during normal use. A STIM includes also, signal
conditioning and conversion circuitry, a TEDS, and
necessary logic circuits to implement the digital
interface and protocols to communicate with a NCAP.
Other several critical elements are also defined [6]:
the various formats for TEDS, the transducer
functional type (sensor, actuator, buffered sensor,
event sensor, etc.), and a general-purpose calibration
and correction engine.
Communication between the STIM and NCAP is via a
TII. The TII is build around synchronous serial
306
communications, based on the SPI (Serial Peripheral
Interface) protocol. The NCAP usually initiates a
measurements or action by means of triggering the
STIM, and the STIM responds with an
acknowledgement once the requested function is
completed.
After this short review of the most important elements
of the IEEE1451 Standard, we are capable now to
develop a curricula for a Smart Transducer course.
The planned material could be highlighted from the
following six important aspects that are the six main
chapters of this course.
Smart Transducer Structure and Caracteris-
tics: Smart Transducer Interface Module
(STIM), Network Capable Application pro-
cessor (NCAP), Transducer Independent In-
terface (TII), Electronic Data Sheet (EDS),
signal conditioning circuitry, A/D and D/A
converters, digital processors;
Distance Access of Smart Transducers: wired
and wireless busses (Radio Frequency,
Mobile Phone and Infrared support), Internet
access;
Sensor Technologies: semiconductor techno-
logies, polymer films, thin and thick-films,
Microelectromechanical Systems (MEMS)
technology;
Sensors (sensing effect, sensor structures,
characteristics):force/moment, visual, tactile,
ultrasonic, global positioning system (GPS),
biosensors;
Actuators and micro actuators;
Application of Smart Transducers: industrial
applications, robotics including mobile and
personal robots, biomedical applications.
Please observe that not all of the above questions are
directly related to the Smart Transducer course (for
example A/D and D/A conversions, digital
processors) but all these knowledge are necessary in
order to successfully implement a smart transducer
application.
Having in mind the proposed curricula, we are
proposing the following list of subjects for Smart
Transducers laboratory works:
Implementation of a ST (implementation of
STIM, including TEDS;
Implementations of a NCAP ( TII only);
Implementation of a NCAP (the bus side):
wired buses ( RS-232, RS-485 and Ethernet
buses) and wireless (Radio Frequency,
Mobile Phone and Infrared support);
Force/Moment ST;
Visual ST;
Tactile ST;
Ultrasonic ST;
Global Positioning System (GPS);
Experiments with muscle wires.
For actual implementation of the laboratory works, we
have to choose now appropriate hardware and
software resources. The most convenient solution for
STIM implementation is to use highly integrated
circuits, containing on the same chip signal
conditioning circuitry, A/D and D/A converters,
digital I/O, and some control logic(a microcontroller
core with memory resources and some peripherals).
Moreover, in some cases (ultrasonic, tactile ST) a
high speed A/D converter is needed, while in other
cases (force/moment transducer) high resolution
converters are necessary. The most recommended
circuits, which meet these requirements, are the
family of microconverters developed and
manufactured by Analog Devices [6]. These circuits
have been design according to the philosophy system
on chip and ready to be used for STIM
implementation [7].
The software applications are an important task in the
development process of the laboratory works. In order
to successfully complete this step, appropriate
software development tools have to be used.
In universities, MATLAB and LabVIEW are the
most preferred software tools for simulations, data
acquisition and control, data analysis and data
presentation. LabVIEW, a graphical programming
language, is a easy to use and a very efficiently tool,
especially when a Graphical User Interface (GUI)
have to be developed; on the other hand, MATLAB is
recommended for process simulation and data
presentation.
In our application, where each laboratory work has its
own GUI, we will prefer to use LabVIEW for
software development.
The whole system includes, as hardware resources, a
number of modules, each of them having a well
defined functionality. Up to date, the following
modules have been developed, build around
microconverters from Analog Devices:
STIM module for fast processes (12 bit
resolution and a sampling rate of 400
KSPS);
STIM module for industrial processes (24 bit
sigma-delta converter and 1.37 KSPS);
NCAP module with RS-232 interface;
NCAP module with RS-485 interface;
NCAP module with Mobile Phone interface.
Other NCAP modules ( having ETHERNET, Radio
Frequency and Infrared support) are under
development.
The STIM and NCAP modules are designed to be
capable to interconnect each other and with
sensors/actuators. As an example, the top view of
STIM module for fast processes is presented in Fig.2.
In some cases (ultrasonic smart transducer,
experiments with muscle wires) additional circuits,
for signal conditioning, are placed nearby
sensors/actuators.
307
Fig.2. A top view of the STIM module for fast
processes.
In this way, we can easily implement the hardware
resources for all of the above mentioned laboratory
works, using only a limited number of modules. This
strategy keeps down the cost of the whole system
without sacrificing the educational content.
III. EXAMPLE OF LABORATORY WORK
In order to evaluate the efficiency of the proposed
strategy, in this chapter will be presented, as an
example, a laboratory work on a Smart Transducer
Implementation. Actually, the purpose of the practical
activity is to study the structure, behavior and some
characteristics of the hardware/software components
included in a ST. A force/moment sensor will be
connected to the input of the STIM for demonstration.
The structure of the smart sensor is depicted in Fig. 3.
RS-232
F/M STIM NCAP
Fig. 3. The schematic diagram of the
Force/Moment Smart Transducer.
It includes the well-known elements: a force/moment
sensor connected to a STIM and a NCAP. A host
computer controls the activity of the system and
processes the acquired data. For the purpose of
simplicity, NCAP is interconnected to the host
computer using a simple RS-232 bus.
As we already known, STIM element is build around
a microconverter chip (ADuC824) developed by
Analog Devices. The schematic diagram of this circuit
is presented in more detalied in [8].
This highly integrated circuit incorporates all
provisions necessary for a STIM implementation:
conditioning circuitry, two independent 24 bit sigma-
delta A/D converters with on-chip digital filtering, a
data flash memory for TEDS implementation and a
8052 compatible microcontroller system. It has been
proved that STIM equipped with ADuC824
microconverter chip is simple, low cost and has
excellent performances, being the most appropiate
solution in this application.
In the present laboratory work NCAP play a minor
roll, because STIM can be actually directly connected
to the serial interface of the personal computer
(UART within microconverter should be used for this
purpose). However, NCAP has been included in the
structure of the smart sensor in order to maintain the
compatibility with IEEE1451 Standard and to allow
the students to experiment the TII between STIM and
NCAP.
NCAP is implemented on a 8052 microcontrolller
system, having a RS-232 compatible serial interface,
which allow the interconnection with the host
computer.
Some digital I/O lines of Port 0 are used for TII
implementation, as is represented in Fig. 4. The
corresponding lines from the STIM belong to the
P3.2/nINT0 NTRIG P1.0
P3.4/T NACK P1.1
S nSS NIOE P1.2 N
T Fig. 5. A snapshot of the GUI implemented
C
I SCLOC DCLK P1.3 A on the host computer.
M MOSI DOUT P1. P
MIS DIN P1.5
NINT nINT1
. In the present paper, a topics for Smart Transducers
Fig. 4. The STIM and NCAP connections
Serial SPI interface. NINT is the only TII signal line
that the STIM is permitted to assert at will. The other
STIM outputs are not really under the full control of
the STIM. NTRIG is used by the NCAP to control the
timing of when the new data is taken (or is act upon in
the case of an actuator). The NCAP is the SPI
communications master and always controls data
clock DCLK. Other than that, the NCAP controls all
communications and all message exchanges are
originated by NCAP. A more detailed presentation of
the handshaking between STIM and NCAP, which
has been used to implement here the communication
protocol, is given in [7].
The software resources of the laboratory work
includes the following main components:
a small foot print real time operating system
(RTOS) and some software applications,
resident on the NCAP system;
software applications for data acquisition and
process control, resident on the STIM;
software application running on the host
computer, which controls the functionality of
the whole system; at the same time, this
application assure the man-machine
interface, using an appropriate GUI .
The software applications can be easily accessed by
the students in order to change/improve them. The
programs loaded on STIM and NCAP are developed
in assembling and C languages respectively. The Keil
uVision2 IDE development tool has been used for this
purpose. The RS-232 serial interface, included on
308
both STIM and NCAP, allow in a simple manner to
load the modified/improved versions of the software
applications, using appropriate loaders.
The GUI is a very flexible application, which allow
the user to send control strings to the NCAP and to
visualize the acquired data. The analogical signals are
presented on a waveform graph while the digital I/O
signals are displayed on array of LEDs.
A snapshot of the developed GUI is shown in Fig. 5.
V. CONCLUSIONS
laboratory works has been proposed, centered around
the latest standard in the field, IEEE1451 Standard. It
has been proved that by using appropriate hardware
and software resources, an important number of
laboratory works, with a high educational content, can
be developed, while the cost is kept down. The
proposed educational laboratory will significantly
improve the knowledge of graduates on the new era of
plug-and-play transducers.
RFERENCES
[1] G. Harsanyi, I. Lepsenyi, P. Gordon, P. Bojta, G. Ballun, and Z.
Illyefalvi-Vitez, SensEdu-An Internet Course for Teaching
Sensorics, IEEE Sensors Journal, vol. 2, no. 1, February 2002, pp.
34-40.
[2] N. Swamy, O. Kuljaca, and F.L. Lewis, Internet-Based
Educational Control Systems Lab Using NetMeeting, IEEE
Transactions on Education, vol. 45, no. 2, May 2002, pp.145-151.
[3] K. K. Tan, T. H. Lee, and C. Y. Soh, Internet-Based
Monitoring of Distributed Control Systems- An Undergraduate
Experiment, IEEE Transactions on Education, vol. 45, no. 2, May
2002, pp.128-134.
[4] IEEE 1451.1, Draft Standard for a Smart Transducer Interface
for Sensors and Actuators Network Capable Application
Processor (NCAP) Information Model, Institute of electrical and
Electronics Engineers, Inc., New York, 1966.
[5] IEEE 1451.2, Draft Standard for a Smart Transducer Interface
for Sensors and Actuators Transducer to Microprocessor
Communication Protocols and Transducer Electronic Data Sheet
(TEDS) Formats. Institute of electrical and Electronics Engineers,
Inc., New York, 1966.
[6 ] http//www.analog.com, Analog Microcontrollers.
[7] B. OMara, P. Conway, Designing an IEEE-1451.2 Compliant
Transducer, Sensors Online,2001.
[8] * * * MicroConverter, Dual-Channel, 16-/24-Bit ADCs
with Embedded FLASH MCU. ADuC824 Data Sheets, Analog
Devices.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
On Radio Spectrum Measurements With The ESVB
Rohde & Schwarz Test Receiver
Gabriel SRBU1, Dorel AIORDCHIOAIE2
Abstract The learning process need good examples
in order to be efficient. This paper present two
experiments, useful in didactical process of Radio
communication laboratory, based on the R&S ESVB
radio test receiver. Starting from the R&S scan.exe
software we developed a LabView based software
application in order to control and communicate with
the ESVB equipment.
Keywords: telecommunications, radio spectrum
scanning, Test Receiver, remote control, LabView
application.
1. INTRODUCTION
With the 2004-year our department has a set of
Rohde & Schwarz equipments, mainly for radio-
communication tests and measurements, composed
of ESVB-22 test receiver, a signal generator and a
digital radio tester.
In this work we present the results accumulated in
testing the ESVB-22 radio test receiver equipment
in order to better understand the specific
measurements functions and to develop new
software that match the specific behaviour of a fine
radio instrument in the radio communication
laboratory.
The work describes first the main features of the
used equipment, including the remote and the
manual operation, wiring with external devices and
other specific functions, as in section 2. Next, in the
section 3, two applications are presented,
concerning the radio spectrum scanning and the
retrieving and storing of generated data. The
explanations include the basic hardware
configuration, the software configuration reflected
in logical diagrams and parameters set-up. Section
4 presents the measurements results. Finally,
Section 5 presents the conclusions of the
considered experiments, which polarize our future
work.
1,2
Dunrea de Jos Galai University Electrical and Electronic
Engineering Faculty Electronics and Telecommunications
Department Radiocommunication and Signal Processing
Laboratory Domneasc 47, Galai 800008, Romania Phone:
++4-0236+460182; 461353 Emails: {gabriel.sirbu,
dorel.aiordachioaie}@ugal.ro
309
2. DESCRIPTION OF THE EQUIPMENT
R&S ESVB-22 Test Receiver features the
bandwidths and signal weighting facilities required
for terrestrial digital video (DVB-T) as well as for
audio broadcasting (DAB). In conjunction with its
high measurement rate it can be used in mobile and
stationary coverage measurements. Also, it is
suitable for measuring signal and interfering field
strengths; it includes all functions of EMI
(electromagnetic interference) Test Receiver.
The few main features that outline the ESVB are
[1]:
Precision field-strength measurements
using test antennas, providing high
measurement accuracy with typical error
of 0.5 dB.
Radio frequency range 20MHz to 2050
MHz.
RMS and average detector for all test
bandwidths.
Manual operation or control by internal
processor or external computer.
Automatic overload detection.
Average, RMS, peak and quasi-peak
detectors operating in parallel.
RF attenuator, switcheable in 10 dB steps
in the range 0dB to 120 dB
The ESVB is equipped with a steep-sided 1,5 MHz
channel filter (SAW type) for use in DAB
networks. For DVB-T applications it is fitted with
an 8 MHz IF filter (SAW) of high selectivity for
adjacent-channel operation. The test receiver is
including also an I/Q test demodulator with a
bandwidth of 0.75 MHz (DAB) and 4 MHz (DVB-
T).
All these hardware is powered by an internal
powerful processor system, which consists of [1]:
- Macros for automatic and
semiautomatic test runs.
- Automatic level calibration.
- Automatic consideration of
frequency-dependent transducer
factors.
- Full programmability of all
internal functions via IEC/IEE
bus.
 - High-speed measurement with
external triggering.
In order to make further signal evaluation and for
driving or feeding additional devices, the ESVB
features the following interfaces [1]:
Coding and supply socket (ANTENNA
CODE) for active antennas and for coding
of transducer factor.
74.7 MHz IF output for connecting a
spectrum analyzer.
10.7 MHz IF output for evaluating the IF Figure 1 IEEE 488.2 serial link used in both applications.
signal e.g. with an oscilloscope.
Controlled in phase and quadrature signal With this configuration all data obtained with the
output for evaluating signals of any Test Receiver is transferred along an HP serial
modulation. cable to a computer. Via this bus almost all
Envelope detector output (VIDEO instrument settings can be effectuated,
OUTPUT) for evaluating the rectified IF measurements triggered and test results read in by
signal e. g. with an oscilloscope the PC.
User interface with: Basically, both software applications have the same
- 6 TTL ports for controlling logical structure, shown in the Figure 2.
external devices
- Input for external trigger signals GPIB initialization,
- Outputs for analog display start
voltage with and without meter communication
simulation with ESVB
- RS-232 interface for firmware
updates by reprogramming the
built-in flash EPROMs by means
of an IBM-compatible PC
- Parallel interface (PRINTER Setup the ESVB: start
INTERFACE) for connecting a freq., stop freq.,
printer incremental freq.
IEC/IEE-bus interface.
Connector for MF2-compatible keyboard
for text entry.
Output for internal oven-controlled crystal
reference frequency (10 MHz). Start
Battery input for independent powering scans

3. THE EXPERIMENT APPLICATIONS SETUP

In our work we considered two different software

Acquire
applications. The first one, ROHDE & SCHWARZ While not data +
SCAN.EXE, is a software application made under end of format the
QuickBasic. The second one is made under spectrum resulting
LabView environment and we developed this data data
software application.
In order to setup the two experiments, we have
made an IEEE 488.2 serial link between PC and
ESVB and a RF coaxial cable link between ESVB
and test antenna.
Stop scans
The basic hardware structure of the experiment is
shown in the Figure 1.
Figure 2. The common logical structure.
The LABVIEW based application follow the same
logical structure. In this application there is no
representation of the scanned spectrum. The
resulting data is only provided for further
processing, it is not formatted.

310
In the Figure 3 are shown the elements of the
LabView application, which represent the main
program.
Figure 3 The main program of the LabView application.
There are also two subroutines, which are appealed
by the main program. They are represented in
Figure 4.
Figure 4 The structure of the LabView application
The GPIB serial interface is initialised in
subroutine in Figure 5. This subroutine also
initialises the starting parameters of ESVB, and the
SRQ (service request) of the GPIB, which is
needed in asynchronous transfer between ESVB
and PC along IEEE serial bus.
Figure 5 The initialisation subroutine.
In order to further process the final obtained data,
this must be concatenated, and then transformed
from binary to ASCII. We present in Figure 6 the
responsible subroutine with this job of
311
concatenation. Finally, the concatenated data is
converted into ASCII string (in the main program),
and it can be further processed (not implemented).
Figure 6 The concatenation (reading long data) subroutine.
4. RESULTS OF THE MEASUREMENTS
The QuickBasic application SCAN.EXE generates
two outputs: one is a graphic representation and the
other is an ASCII file. We present in the figures 7
and 8 the two outputs, the graphic and the file,
respective.
Figure 7 The 20MHz 2050MHz graphical representation of
scanned radio spectrum
Figure 8 Partial results from the ASCII representation of
scanned radio spectrum, stored into a file.
File lines consist of data records made up by the
following parts:
- First column: Frequency in Hz;
- Second column: Level in dB with
0.01dB resolution;
 - Third column: Status word,
which is 0 for error free.

5. CONCLUSIONS

The test receiver is a complex and powerful tool

that combines the highest test requirements with
fully remote controlled features. We presented in
this paper a software tool developed by us in the
LabView environment for educational purpose, and
a ROHDE & SCHWARZ tool developed under
QuickBasic. Both applications are running, they are
obtaining the data from the receiver, and only the
Rohde&Schwarz application is, for now, capable of
writing files or drawing graphics. Thus, the next
step of the work is to solve the data interpretation
aspect and to add more and complex options in the
remote mode functioning of R&S ESVB Test
Receiver. The results are useful in our effort of
understanding the measurements functions and to
develop new software algorithms for other complex
radio-communication and radio-measurement
applications, for educational and
telecommunications market targets.

ACKNOWLEDGEMENTS

The people of Electrical and Electronics

Engineering Faculty gratefully acknowledge the
Rohde & Schwarz Company for the donation of the
radio communication equipments. Special thanks to
Professor Viorel Mnzu for assistance and support
in developing the communications facilities on our
faculty.

REFERENCES

1. Rohde & Schwarz , GmbH, http://www.rohde-

schwarz.de/

312
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
On GSM Mobile Phone Measurements With The CTS-65
Rohde & Schwarz Digital Radio Tester
Gabriel SRBU1, Dorel AIORDCHIOAIE1
Abstract The learning process need good examples
in order to be efficient. This paper present an
experiment, useful in didactical process of Radio
communication laboratory, based on the R&S CTS
digital radio tester. The digital radio tester seems to
be very flexible and useful for simple applications like
identification of the parameters of the mobile phones,
but also in other complex experiments.
Keywords: Mobile Radio Measurements, Digital
Radio Testers, Radiocommunications.
1. INTRODUCTION
With the 2004 year our department has a set of
Rohde & Schwarz equipments, mainly for radio-
communication tests and measurements, composed
of CTS-65 (Digital Radio Tester), a signal
generator and a receiver tester.
This work describes an example of using the CTS-
65 digital radio tester, as result of our effort in
understanding the measurements functions and to
develop new software algorithms for other complex
radio-communication and radio-measurement
applications.
The work describes first the main features of the
used equipment, as in the section 2. Next, in the
section 3, the setup of the experiment is described
with the structure of the measurement system, logic
diagram and parameters setup. Section 4 presents
the results of a mobile phone test measurements.
Finally, the conclusions and next steps to follow are
considered and presented.
2. DESCRIPTION OF THE EQUIPMENT
R&S CTS-65 has both manual and remote
operation for GSM 850/900/1800/1900 mobiles and
DECT-phones behaviour analysis. Digital Radio
Tester - CTS is an extremely compact, modular yet
1
Dunrea de Jos Galai University Electrical and Electronic
Engineering Faculty Electronics and Telecommunications
Department Radiocommunication and Signal Processing
Laboratory Domneasc 47, Galai 800008, Romania Phone:
++4-0236+460182; 461353 Emails: {gabriel.sirbu,
dorel.aiordachioaie}@ugal.ro
313
powerful measuring instrument. It combines great
ease of operation and the necessary test depth for
use in all service areas for mobile and cordless
phones. Both the newcomer and the service
specialist will be able to conveniently carry out fast
automatic functional tests as well as complex and
comprehensive manual measurements down to
component level. The CTS can be fully controlled
via six softkeys and one hardkey. Maximum
operating convenience is obtained by connecting an
external PC keyboard. In addition to the local
display, an external monitor can alternatively be
connected via the VGA interface. All features are
also under Remote Control available, [1]. CTS
simulate a GSM base station and can completely
control the mobile for testing purposes.
For remote control we have used a R&S software
tool called CTS-Go, which is an Windows
application made by Rohde & Schwarz .
The GSM measurements can be found in 5
different modes of operation [2]:
1. The manual test mode, for exact fault
location.
2. The auto test mode, for versatile diagnose.
3. The quick test mode, for go / no go
diagnose.
4. The module test mode, which includes RF
measurements without signalling, like
burst analysis, mobile RF generator, and
narrowband spectrum monitor.
5. The remote controlled mode, with an
associated PC software application.
GSM Signalling Parameters are [1]:
1. Synchronization of mobile phone with
base station (which is simulated by CTS).
2. Location update.
3. Call establishment (mobile originated/
mobile terminated).
4. Call release (mobile originated/ mobile
terminated)
5. Power Control.
6. Measurement of transmitter power.
7. Handover (channel change).
8. Dual band handover.
9. GPRS signalling (attach / detach).
 10. All coding schemes CS1- 4 Off ->On
the mobile
GSM measurement functions are [1]:
Power versus time (Power time
template).
Timing error.
Echo test (voice test, includes also
testing of loudspeaker and Make a
microphone). call from
Function test of mobile's keypad mobile
through display of dialled number
Display of
o IMSI (international mobile
subscriber identity)
o IMEI (international mobile
equipment identity)
Sensitivity Handshake
o Bit error rate BER and mobile with
RBER CTS
o RxLev, RxQual and BLER
Phase and frequency error

Also, the DECT measurement capabilities are [1]:

1. Synchronization of DUT with the CTS.
2. Call set-up. Execute all programmed
3. Call release. tests and trace an
4. Echo test. analysis report
5. Detection and display of REPI (FP).
6. Normal transmit power (NTP).
7. Power ramp versus time.
8. Modulation characteristics versus time.
9. Frequency offset.
10. Maximum modulation deviation.
11. Frequency drift. Release the call
12. Timing (jitter, packet delay). from mobile
13. Bit error rate (BER), frame error rate
(FER). Figure 1- the experiment operations diagram

3. SET-UP OF THE GSM MOBILE ANALYSIS The structure and the components of the experiment
EXPERIMENT are shown in Figure 2.
The experiment we have made consists in a RS-232
upload/download link between a PC and CTS, with
a serial cable DTE <-> DTE. For the analyse
purpose a Motorola mobile was used, with a 50
ohms RF coaxial cable link to the CTS. The
operations diagram is shown in Figure 1:

Make physical
connections
PC <->CTS +
Figure 2 The structure and the components of the experiment:
CTS <-> mobile desktop computer, CTS-65 and a mobile phone

4. RESULTS OF THE MEASUREMENTS

Start the
CTS-Go In the remote mode, the CTS communicate with the
application PC and changes obtained data and control
commands, which are reflected in CTS_Go
application screen. Figures 3 and 4 illustrate the
make a call from mobile and execute all

314
programmed tests and trace an analysis report
moments obtained in a simple test experiment
concerning a Motorola mobile phone.

Figure 3 Identification of mobile phone, Motorola type

Figure 5 The final report imported in Microsoft Excel

Figure 4 Executing one of the programmed tests on CTS-go

application.

All the measures made by the CTS equipment can

be optionally traced on a printer or saved in a file
on PC. The file can be either saved into a particular
format, or can be text formatted, saved and
imported in programs like Microsoft Word or
Microsoft Excel. Thus, the final traced report can
be easily stored and further processed. In the
figures 5 and 6 we have illustrated two examples of
traced reports, one imported in Microsoft Excel and
the other in the original format, respectively.

Figure 6 The original format report : 7 of 34 programmed tests

failed on tested mobile

315
 5. CONCLUSIONS

The digital radio tester seems to be very flexible

and useful for simple applications like identification
of the parameters of the mobile phones, but also in
other complex experiments. If some-thing should
be critical we can say that the graphical information
able on the built in display is not yet possible to
export on a remote computer connected via RS-232
for further processing objectives.

The next step of the work is to implement a

communication tool between CTS and PC in order
to grab the images from CTS built in display, and to
save and further process them. We also expect to be
able to control much better the functions of the
R&S CTS 65 testing equipment.

ACKNOWLEDGEMENTS

The people of Electrical and Electronics

Engineering Faculty gratefully acknowledge the
Rohde & Schwarz Company for the donation of the
radio communication equipments. Special thanks to
Professor Viorel Mnzu for assistance and support
in developing the communications facilities on our
faculty.

REFERENCES

1. Rohde & Schwarz , GmbH, http://www.rsd.de/

2. Gottfried Holzmann, New measurement

functions in Digital Radio Tester CTS55,
Application Note of Rohde & Schwarz , 1997.

316
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Scattering parameters of symmetrical networks

Aldo De Sabata1 Ladislau Matekovits2
Abstract Following a recent publication, teaching of II. EXAMPLES OF SCATTERING PARAMETERS
the scattering parameters is further investigated in this CALCULATION
paper. It is shown that complete calculations, sometimes
considered too complicated and therefore avoided, can The first example of a symmetrical, resistive network
be alleviated and made transparent, by exploiting the
symmetry of the electrical networks involved.
is reproduced from [1] in Fig. 1.

Keywords: scattering parameters, electrical engineering 2

education, microwaves
r r
I. INTRODUCTION
r r
In a recent paper [1], the author identifies and corrects 1
two deficiencies in the way scattering parameters are
taught in microwaves courses: an insufficient number r r
of calculation examples in textbooks, and the lack of
an adequate physical interpretation for the results. For 3
illustration purposes, he presents three "carefully Fig. 1: A symmetric resistive network.
selected examples", of which the first one is
calculated by hand, and, for the other two, circuit- and All values for the resistances are normalized to the
microwave-oriented software packages are used reference resistance (same value for all the 3 ports,
(PSpice and Microwave Office respectively). It turns even if for symmetry exploitation it is enough to have
out that these examples are also ideal for illustrating the same reference impedance only at ports 2 and 3).
the simplifications to hand calculations when the In order to calculate S11, S21, and S31, the network
topological symmetry of certain circuits is exploited, must be completed with the elements drawn with
and therefore, complete solutions for the S parameters dashed lines in Fig. 2 (a). In [1] the scattering
are presented here. parameters are obtained using wye-to-delta
Hand calculations are important for understanding the transformations. However, a simpler way is to note
behavior of circuits, which can influence beneficially that, due to symmetry, points A and B in Fig. 2 (a) are
system design, the interpretation of results, or finding at the same potential. Consequently, they can be
errors when simulation software is used. Real life electrically connected without altering the voltages
circuits are generally described by complicated and currents at ports 1, 2, and 3. The equivalent
algebraic expressions; therefore every method for network is represented in Fig. 2 (b), from which there
alleviating their derivation is valuable. results by inspection
All the examples deal with reciprocal networks,
therefore the scattering matrix is symmetric
r r 1 11r + 4
S ij = S ji , i j . However, for illustration purposes, rin ,1 = r + || r + + = , (1)
2 2 2 8
calculations will be performed for all the scattering
and
parameters, and reciprocity will be used for
verification.
rin ,1 1 11r 4
S11 = = . (2)
rin ,1 + 1 11r + 12

1
Facultatea de Electronica si Telecomunicatii, Departamentul de Masurari si Electronica Optica, Bd. V. Parvan Nr. 2, 300223
Timisoara, Romania, e-mail aldo.desabata@etc.utt.ro
2
Politecnico di Torino, Dipartimento di Elettronica, C.so Duca degli Abruzzi, 24, I-10129 Torino, Italy, e-mail
ladislau.matekovits@polito.it

317
 v
be considered: first, the "even" case, e g 2 = e g 3 = ,
2
A
(for which "+" will be used), and second, the "odd"
2

v2 A 2 i2 1 E
r r 1
1 r
1 r + eg2
r r rin,2
eg1 +
-
rin, r r 1 r D r
- C
3 1 v1 r r
B
v3
1 v3 F
(a)
B 3 i3 1 + eg3
(a) -
1 1 r r
v3+
2i+
eg1 +
rin,1 r/2 r/2
- 1/2 +
v/2
r/2 r/2
-
1 r r
1/2 v2=v3
1
(b) (b)

Fig. 2: a) The network in Fig. 1 with additional

components for the computation of the reflection i- -v3-
coefficient S11 at port 1; b) equivalent circuit due to
symmetries. 1 +
r r v/2
Also due to symmetry, and considering the -
1 r r
computation formula (9) in [1],

1 1
2
2v 2 8
S 21 = S 31 = 2 = = .
e g1 r r 1 11r + 12
1 + r + || r + +
2 2 2
(3)
For the rest of the parameters, the network in Fig. 3
(a) can be used. In order to apply the definition, for
calculating S22, S12, and S32 it must be
imposed e g 3 = 0 , while for S33, S13, and S23 one must
choose e g 2 = 0 . Of course, in this case S12 = S13 due
to symmetry, and the fact that S 23 = S 32 results not
only from symmetry, but also from reciprocity.
Advantage will be taken of symmetry for avoiding
again wye-to-delta transformations. Inspiration comes
from the even and odd modes on coupled transmission
lines [2], or common mode and differential mode in
differential amplifiers [3]. The following method is
also used in [4] and tackled in [5]. Two situations will
v1-
(c)
Fig. 3: a) The network in Fig. 1 with additional
components for the computation of the Si2, Si3
coefficients; b) equivalent circuit, "even" excitation; c)
equivalent circuit, "odd" excitation.
v
case, e g 2 = e g 3 = (denoted by "-"). In the even
2
case, points A and B in Fig. 3 (a) are at the same
potential, and can be electrically connected, and
similarly points E and F. The resulting network is
represented in Fig. 3 (b). In the odd mode, points D
and C in Fig. 3 (a) are at the ground potential, and the
network schematic can be redrawn as in Fig. 3 (c).
By inspection, it can be written

318
v1+ =
v 1
=
4v
, v1 = 0 , z 2 ' z1
|| + z 2 '
2 1 r r 11r + 12 z 2 2 ,
+ || + r + r + 1 v 2 = e g1 in
2 2 2 1 + z in z1 z 2 ' z1
+ || + z 2 '
r r 2 2 2
|| + r + r + 1
v v 2 2 v 4 z2 '
v 3+ = + = , v3 = v 2 .
2 2 1 r r 2 11r + 12 z1
+ || + r + r + 1 + z2 '
2 2 2 2
r

v v r
v 3 = 1 2 = ,
2 r 2 r+2
1+
2
v2
v/2 4v z2 1
2i + = = ,
1 r r 11r + 12
+ || + r + r + 1 2
2 2 2 z1 z1
A v3
v 1 1 3
2 v
i = = . z2
r r+2 +
1+ eg1 zin,1 z1
2 z2 z1 B
- 1
4
Due to linearity, for e g 2 = v and e g 3 = 0 , the values
z2 1
for the electrical quantities defined in Fig. 3 (a) can be
obtained from v1 = v1+ + v1 , v 3 = v 3+ + v 3 ,
v
i 2 = i + + i , and rin , 2 = 1 . Finally, the following (a)
i2
scattering parameters result:
1 z1/2 z1/2
1
rin , 2 1 11r + 8r 8
2
S 22 = = ; (4) eg1 +
rin , 2 + 1 11r 2 + 34r + 24 zin,1 z2 z2'/2 v2 z2' v3
-
2v1 8
S12 = = ; (5)
v 11r + 18
2v 18r + 16
S 32 = 3 = . (6) (b)
v (11r + 12)(r + 2)
Fig. 4: a) Symmetric four-port reactive circuit with additional
The second example is the network represented with components for S parameters computation (dashed line); b)
solid line in Fig. 4(a). The circuit contains the equivalent circuit.
normalized impedances z1 and z2, which are
respectively an inductance and a capacitance in [1].
Now, the formulas for the scattering parameters [1]
Due to symmetry, it is sufficient to find the S
can be applied:
parameters at only one of the four ports. The elements
it must be completed with in order to calculate Si1,
z in ,1 1
i=14 are drawn with dashed line in Fig. 4 (a). The S11 = =
same symmetry allows for the electrical connection of z in ,1 + 1
points A and B, and the equivalent network ( z1 z 2 z1 2 z 2 )( z1 + 3z 2 + z1 z 2 )
represented in Fig. 4 (b) is obtained, where z 2 ' = 1 || z 2 . = + (7)
( z1 z 2 + z1 + 2 z 2 )( z1 + 4 z 2 + z1 z 2 )
There results at a glance
z 2 ( z 2 1)
z z ' z +
z in ,1 = z 2 || 1 + 2 || 1 + z 2 ' , ( z 2 + 1)( z1 + 4 z 2 + z1 z 2 )
2 2 2
2v 2 2 z 22
S 21 = = , (8)
e g1 (1 + z 2 )( z1 + 4 z 2 + z1 z 2 )

2v 3 2z2
S 31 = = S 21 . (9)
e g1 z1 + 2 z 2 + z1 z 2

319
A general purpose, non circuit- or microwave-
oriented software package, such as Matlab, can be
used for plotting the insertion loss, return loss or 2 i2 1
zc, , L
isolation for various values of the elements in the
circuit. 1
1
v2 + e
In figs. 9 and 10 of [1], the S parameters seem to be r g2
rin,2
evaluated at a few frequency points and the response -
+ zc, , L v3
in the whole frequency range is obtained by an eg1 rin,1
interpolation. Usually, this approach is used when - i3 1
determination of the response in one point requires 3 + e
2' g3
long computation time. -
The proposed method can be used both for the
determination of the response of the circuit at few 1'
sampling points followed by an interpolation scheme,
or to compute the response in the whole range with a 3'
relative large number of points. (a)
The computation of the S11, S12, S13 was implemented
in a Matlab script. Its run on a Pentium III with
1GHz clock required 0.01 s for 1000 sampling 1
1 zc/2, , L
points.
As a third example, a symmetric Wilkinson power eg1 + 1/2 v2= v3
divider [1]is schematically represented with solid line zin,1
-
in Fig. 5 (a); on the same figure, with dashed lines
(b)
the circuits needed for the calculation of the S
parameters, are also represented. All the impedances zc/2, , L 2i2+=2i3+ 1/2
are normalized to a given reference impedance, here 1
considered identical for the three ports. The v +
1+ 1 v2+= v3+ eg2
transmission lines in the circuit are supposed
=eg3
identical, with (normalized) characteristic impedance 1' - =v/2
zc, propagation constant = + j and length L. (c)
For the calculation of S11, S21, and S31, one must
take e g 2 = e g 3 = 0 . Due to symmetry, the voltages zc, , L i2-=-i3- 1
over the lines are equal at equally spaced points from 1 eg2
the generator; consequently, there is no current flow v1-=0 r/2 v2-=- v3- + =-eg3
through the resistance r, and the two transmission =v/2
1' -
lines can be conceptually connected in parallel, as in
Fig. 5 (b). The result is a transmission line with the (d)
same propagation constant, but with a characteristic
impedance zc/2.
(Indeed, consider a standard equivalent circuit of a Fig. 5. (a) The Wilkinson power divider (solid line) and the
differential length of transmission line, [2, p. 86]. components needed for the calculation of the scattering
Conceptually connecting two such circuits in parameters (dashed line); (b) equivalent circuit for the
calculation of the scattering parameters at port 1; (c) "even"
parallel, node by node, in the case of identical,
excitation; (d) "odd" excitation.
corresponding voltages and currents, reduces by a
factor of two the line resistance and the line z in ,1 1 ( z c2 2) sinh(L) z c cosh(L)
S11 = = . (11)
inductance, and doubles the line capacitance and the z in ,1 + 1 ( z c2 + 2) sinh(L) + 3 z c cosh(L)
line conductance. The substitution of these values in
the expressions of the characteristic impedance and
the propagation constant [2, p. 88] yields the above
In the following, it will be needed the use of an
stated result.)
expression for the voltage on a transmission line of
The normalized input impedance in Fig. 5 (b) is
normalized characteristic impedance zc, propagation
constant , and length L, terminated on a normalized
1 zc
+ tanh(L) load impedance zL, and connected to a generator eg, of
zc 2 2
z in ,1 = , (10) normalized internal impedance zg. Such an expression
2 zc 1
+ tanh(L) can be found in most textbooks on transmission lines;
2 2 the variant adopted here is

and, consequently

320
 z c e ( L x ) + L e ( L + x ) zc
v( x) = e g , (12) v 2 (1 + L + )e L
z g + zc 1 L g e 2L v1+ = ,
2 1 z c 1 L + g + e 2 L
+
z L zc 2 2
where x is the distance to the load, and L = , zc
z L + zc
v 2 1 + L + e 2 L v
z g zc v 2+ = v 3+ = .
g = 2 1 z c 1 L + g + e 2L 2
are the Fresnel reflection coefficient to +
z g + zc 2 2
the load and to the generator respectively.
The application of (12) at the load end for the Finally, from Fig. 5 (c) there results directly
transmission line (x=0) in Fig. 5 (b) gives i2+ = i3+ = v 2+ = v3+ .
In the odd case, by taking into account Thvnin's
zc theorem applied to the generator, the reflection
e g1 L
v 2 = v3 = 2 (1 + L )e r
z 1 L g e 2L
. (13) 1 || 1
1+ c coefficients are L = 1 , g = 2 , and
2 r
1 || + 1
2
1 zc consequently, from (12) and Fig. 6 (d)

v1 = 0 ,
In (13) the reflection coefficients are L = 2 2 ,
1 zc r
+
2 2 v 2 zc 1 e 2 L v
v2 = v3 = 2 L
,
z 2 r r
1 + 1 || + z c 1 + e 2
1 c g

and g = 2 . There follows 2 2

zc i2 = i3 = v2 = v3 .
1+
2 As shown above, in connection to the second
example, the electrical diagram needed for the
2v 2 2 zc calculation of the scattering parameters relative to
S 21 = S 31 = = . port 2 is the superposition of the situations
e g1 (2 + z c ) sinh(L) + 3z c cosh(L)
2
represented in Fig. 5 (c) and (d). For
(14) 2v1 2(v1+ + v1 )
example S12 = = ; performing the
The scattering parameters relative to ports 2 and 3 are v v
respectively identical. The equivalent diagram for the calculations leads to the right member of (14), as
v S12 = S 21 due to reciprocity. Symmetry and
even mode ( e g1 = 0, e g 2 = e g 3 = in Fig. 6 (a)) is
2 reciprocity also imply S13 = S12 = S 31 . New
represented in Fig. 6 (c), and the equivalent diagram expressions will be obtained for S 32 = S 23 ,
v and S 22 = S 33 .
for the odd mode ( e g1 = 0, e g 2 = e g 3 = ) is
2 In the first case
represented in Fig. 6 (d). In the first case, the two
transmission lines have been connected together point 2v 3 2( v 3 + + v 3 )
by point, and in the second case, it has been taken into S 32 = = =
account the fact that port 1 appears short-circuited v v
(terminals 1 and 1' are at the ground potential), due to 2 cosh(L) + z c sinh(L)
= zc (15)
symmetry. (2 + z c2 ) sinh(L) + 3z c cosh(L)
In the even case, the reflection coefficients at the load
rz c (r + 1)(1 e 2L )
z .
1 c r + ( r + 2) z c r + 1 e 2L
and the generator are L+ = 2 and
zc
1+ For S22 the input impedance at port 2 is needed:
2
v + v 2 + v
1 zc z in , 2 = 2+ . There results
i2 + + i2
g + = 2 2 respectively. The application of (12)
1 zc
+
2 2
at the load end and generator end yields

321
 z in , 2 1 2( v 2 + + v 2 ) REFERENCES
S 22 = = 1+ =
z in , 2 + 1 v [1] M. N. O. Sadiku, "Deficiencies in the Way Scattering
2 cosh(L) + z c sinh(L)
Parameters Are Taught", IEEE Transactions on Education, Vol. 46,
= zc + (16) No. 3, pp. 399-404, August 2003.
(2 + z c2 ) sinh(L) + 3z c cosh(L)
[2] R. Collin, Foundations for Microwave Engineering, 2nd ed.
rz c (r + 1)(1 e 2L ) New York: McGraw-Hill, 1992.
+ 1.
r + (r + 2) z c r + 1 e 2L [3] P. L. Gray, C. L. Searle, Electronic Principles. Physics, Models,
and Circuits, New York: John Wiley & Sons, 1969.
A Matlab script was implemented again for the
[4] R. Badoudal, Ch. Martin, S. Jacquet, Les Micro-ondes, Vol., 1,
computation of the S11, S22, and S23 parameters. Its run Paris: Masson, 1993.
on a the same Pentium III with 1GHz clock required
less than 0.1 s for 5000 sampling points. [5] D. M. Sazonov, A. N. Gridin, B. A. Mishustin, Microwave
Circuits, Mir, Moscow, 1982.

III. CONCLUSION

In this work, some educational applications of

electrical networks symmetry for scattering
parameters calculation was illustrated, using both
lumped elements networks and networks containing
transmission lines. Reflection and transmission
coefficients have been calculated in closed form, in
order to alleviate the interpretation of the results, and
highlight the effect of various parameters.
Comparisons between results obtained by the
proposed approach and those in [1], where they have
been computed numerically, with dedicated software,
are in good agreement,.

322
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
A Management Service for Student Examination Results
with Nomadic Access
Marcel Cremene1, Iulian Bena, Ligia Chira, Clin Loghin
Abstract The paper subject is related to the fields of e-
learning and distance learning. We are proposing an
electronic classbook with nomadic access and SMS
notification. The idea is to integrate a result
management system with a publishing and a notification
system in order to facilitate the management of
examination results and students presence. In order to
build such a system, a distributed architecture is
proposed. For the implementation, several software
technologies are used: J2ME, jSMSEngine, WAP-WML,
VXML. Our proposal is an example of mobile and fixed
terminals utilization in the educational field.
Keywords: electronic classbook, nomadic access, mobile
devices.
I. INTRODUCTION
During the latest years we have witnessed a large
number of mobile technologies developing very fast.
This implies the existence of a large variety of mobile
phones, PDAs, and also a large variety of
communication solutions such as GPRS, Bluetooth,
WLAN and others. Users are nomadic by nature and
they frequently need the same service in different
places at different moments. We call this nomadic
access.
One of the potential fields for nomadic services is
education. Lately, the interest for e-learning and
distance learning is increasing. Mobile devices have
their own place in this field. These new technologies
can help the actors in the educational field to
communicate much better; actors like: pupils,
students, teachers, professors and even parents.
In this paper we are interested in developing a
nomadic access system that will help teachers manage
the examination of the students, and the students to
find out their results by using fixed and especially
mobile devices.
This paper is organized as follows: the next
section presents some existent solutions related to our
subject; the third section presents our architecture and
its components; the fourth section presents the
implementation and some screenshots and the last one
concludes the paper. We consider that the main
contribution of the paper is the integration between e-
1
Universitatea Tehnica Cluj-Napoca, Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii, str. Baritiu, Nr. 26, Cluj-Napoca, cremene@com.utcluj.ro
323
learning, m-learning and the nomadic access and it is
materialized in the proposed architecture.
II. EXISTENT APPROACHES AND SOLUTIONS
THAT ARE RELATED TO OUR SUBJECT
In this section we intend to make a short overview of
the research fields related to our subject. These fields
are: educational portals, e-learning, m-learning (m
from mobile) and nomadic access architectures.
A. Educational portals
In the latest years numerous educational portals can
be seen on the market. The goal of an education portal
is to offer an integrated environment dedicated to the
educational field. For instance, in Savoie, France, an
education portal [3] was implemented. It provides a
virtual environment for students, professors and
administrative staff, where they can use individual
tools such as: e-mail, news, agenda, address book, and
the most important, a virtual desk where each user can
manage different objects as documents, images, etc. It
also provides groupware tools: forum, chat, and a
virtual team desk very useful for team projects.
Anyway, this portal is available only on the Web and
does not enable nomadic access.
B. E-learning and m-learning platforms
In Blended Learning, Mobility, and Retention:
Supporting First Year University Students with
Appropriate Technology, [2], Andy Stone illustrates
how an SMS-based notification system may be used
for the first-year students to update their schedule and
how and where to collect marked coursework from
the department office for the first time.
A very interesting approach is the one of Jo
Colley and Geoff Stead that proposes a collaborative
mobile board. The mobile user can access its content
through the web and fill it, from the mobile terminal,
with short and multimedia messages (SMS and
MMS), [5].
The idea of multiple accesses to learning
information is present in the paper A System for
Adaptive Platform-Independent Mobile Learning,
[7]. Here the SMS-based interaction is supplemented C. Architecture
by a voice-based one, using the H323 Gateway and an
Interactive Voice Response. The proposed system architecture is described in
figure 1. It contains several components:
C. Nomadic access architectures
The nomadic access architecture is an architecture that
allows user access to a service from different
terminals and through different networks. The most
important problem to solve here is the integration of
different constrains and technologies.
Our attention was focused on user interface
integration solutions because we want to offer access
for the same service using different user interfaces.
The idea is to have a unique description for the user
interface that can be automatically transformed into
platform specific user interfaces. Several languages
have been developed to allow a platform independent
description of the user interface: UIML (User
Interface Markup Language) [8], XIML (eXtended
Interface Markup Language) [1], SunML (Simple
Unified Natural Markup Language) [4] and others.
These languages allow us to describe the user
interface in a platform independent manner. Several
renderers create a concrete terminal dependent user
interface.
III. OUR PROPOSAL
A. Objective
As we have seen in the previous section, there are
several field specific solutions for educational
services, mobile learning services and nomadic access
services. In our case we need something that
integrates all this solutions. In this paper we intend to
propose an integrated solution for educational
nomadic services, in particular we want to develop a
system that will help the evaluation and publishing of
students grades. For high school or gymnasium it is
possible to involve the parents by sending them
instant notifications about the pupils activity.
Professors
B. Functions interface
Basically, this system must perform the following
functions:
- For the teachers or professors: mark absences
and observations for the laboratory courses,
edit the examination results.
- For the pupils or students: check up the
notification results and the number of
accumulated absences.
- For the parents: check the pupils grades and
absences and be instantly notified about the
absences
The system will integrate all this functions, will check
the allowed absence number, will send a notification
to the parent who wants to know whether the pupil is
at school.
324
- The rendering components: HTML, WML,
VoiceXML. A renderer [4] transforms a
UIML or SunML user interface description
into a concrete user interface. This gives us
the possibility to have only one interface
description for several platforms and we are
using it for the publishing of the examination
results.
- The J2ME client is a MIDlet, dedicated to
absences and observation management. The
professor/ teacher is the one who uses it. The
connection can be wired or wireless.
- The SMS Notify is an interface to the GSM
that allows the system manager to send
notifications to the parents. The parent must
subscribe to this service first.
- The Database contains basically all the
persistent information related to the users:
user account information, examination
results, absences, etc. Different users are
authenticated using this database, and they
have access to services based on their status.
- The User Interface description contains the
unique description for the examination
results and absences publishing service. This
fact reduces the design effort for all the
required interfaces.
J2ME
Client
SMS
Parents Students
interface interface
HTML WML Voice
Renderer Renderer Renderer
SMS
Notify
System
Database Manager
(Mysql, User
Interface
Excel)
description
Figure 1. The overall system architecture
 - The System Manager links all the other using the HTTP protocol. This client
components and contains the business logic. application allows the professor to mark the
It uses the user interface description and the absences and to introduce comments for each
renderers in order to interact with the user student. The user interface is based on the
according to his particular interface. At this java.microedition.lcdui package. A
level we have the application algorithm and screenshot of this client is shown in figure 2.
the task (user machine dialog) description
- The SMS Notify component is based on
and succession. The manager implementation
JSMSEngine package. This component needs
can be based on a workflow engine for
a physical interface that is a mobile phone or
instance.
a model connected by a serial interface with
the machine that runs the SMS Notify
IV. IMPLEMENTATION
component.
A. Technologies
- The Database is partially based on MySQL
Several technologies were used in order to build this and partially on Excel. It was easier to use
prototype: MySQL in order to store information such as
username, password, status and absence
- J2ME: represents a highly optimized Java
number. Figure 3 shows a screenshot of the
runtime environment, which specifically
MySQL database interface. The advantage of
addresses the vast consumer space, which
using Excel was the compatibility with the
covers the range of extremely tiny
existent list used by the professors and it is
commodities such as smart cards, phones or
also easy to configure the algorithm to
a pager all the way up to the set-top box, an
compute the final result for a student. A
appliance almost as powerful as a computer.
dedicated connector is used in order to
- Java Servlet: a Java technology that offers a connect Java with Excel.
fast, powerful, portable environment for
- The User Interface description is a UIML
creating dynamic content for all the XML
file. This represents the user menus and they
based technologies as HTML, WML,
will be transformed into different concrete
VoiceXML.
implementations by the renderers.
- JSMSEngine: is an API package, written in
- The System Manager is a complex Java
Java, which allows sending or receiving the
application that implements the systems
SMS messages from PC, by using a mobile
workflow. For instance, the first user action
phone or a GSM modem.
is the authentication then he should see the
- UIML: is an XML language for defining user main menu, the functions and so on.
interfaces. It can be used to define buttons,
C. Screenshots
menus, lists and other controls that allow a
program to function in a graphical interface. In figure 2 we show the mobile user interface that
It also defines actions to take when certain allows the professor to mark the absences.
events take place.
- kXML-RPC: is a RPC (Remote Procedure
Call) middleware implementation for the
mobile phone. The advantage of using this
communication protocol is the high
abstraction level. Aversion for J2ME mobile
phones exists. This protocol encodes the
method calls using the XML syntax and send
this messages over the HTTP protocol.
B. Prototype description components and use
cases
A prototype was implemented using the technologies
described above. The architecture components were
implemented as follows:
- The HTML, WML and voice rendering
components are UIML renderers based on
LiquidUI [8] software distribution
- The J2ME client is a small MIDlet
application with a simple user interface that Figure 2. The mobile client
communicates with the System Manager

325
The mobile client was tested with the J2ME Wireless
Toolkit simulation environment. The client size is
about 56Kbytes and the kXML-RPC connector takes
about 24Kbytes.
The communication can be based on GPRS,
WLAN, IrDA, Bluetooth or even cable. Anyway,
because of the price, it is not recommended to use
GPRS.
a solution may be to try to automatically generate the
platform dependent elements.
Another problem is the ergonomics of the
automatically generated user interface. Anyway this is
another research direction and we intended just to
reuse this existent technology in order to prove our
architecture.
In figure 3 we show the MySQL database interface.

Figure 3. The MySQL database interface

In figure 4 we show an UIML code sequence from our The records correspond to the absences and
application. observation statistics. The grades are stored as one
Excel file for the reasons explained in the section IV.
<?xml version="1.0"?>
<!DOCTYPE uiml PUBLIC
"-//Harmonia//DTD UIML 2.0 V. CONCLUSIONS
Draft//EN""UIML2_0g.dtd">
<uiml>
<interface>
<structure> This paper has presented an integrated system that
<part id="Main" class="Wml"> allows actors involved in an educational process -
<part id="Welcome" class="Card"> professors, students, pupils and parents - to manage
<style> the student results and absences. We consider this
<property name="title">Bine ati venit
!</property> proposal a contribution to the e-learning and m-
</style> learning fields.
<part id="newP" class="Paragraph">
<part id="titletxt" class="Text"> We are focusing on the nomadic access for the
<style> educational services because, as we have seen in
<property name="content">Ati
accesat pagina catalog al sectiei de
chapter II, this field is less approached in literature.
comunicatii al unversitatii tehnice din cluj The proposed system is available not only on the Web
napoca</property> but also on mobile devices and even on very simple
</style> terminals like fixed phones. This can be done by using
</part>
</part>
an automatically generated user interface.
</part> A prototype was implemented in order to test our
proposition. Numerous software technologies were
Figure 4. UIML code sequence.
used because the nomadic access implies extended
platform diversity.
Even if UIML wants to be platform independent it has The implementation has also some limitations
platform dependent elements. This is a drawback and that come basically from the user interface automatic

326
generation, which is not optimised, and, as we have
seen, the platform independence is not always
complete.

REFERENCES
[1] A. Puerta, J. Einstein, XIML: A Common Representation for
Interaction Data, In proceedings of the International Coference on
Inteligent User Interfaces, IUI 2002, pp. 214 225, ISBN: 1-
58113-382-0, San Francisco, USA, 2002.
[2] A. Stone, Blended Learning, Mobility, and Retention:
Supporting First Year University Students with Appropriate
Technology, In the Proceedins of the third European Conference
on Mobile Learning, MLEARN 2004, Rome, Italy, 2004, ISBN 1
85338 855 6.
[3] C. Martel, La modlisation des activits conjointes: rles,
places et positions des participants, PhD Thesis, University of
Savoie, September 1998.
[4] J. Fierstone., A-M. Pinna-Dery, M. Riveill, Architecture
logicielle pour l'adaptation et la composition d'ihm - mise en
oeuvre avec le langage sunml. Technical Report I3S/RR-2003-02-
FR, Laboratoire I3S, Universit de Nice, ESSI - BP145 - F-06903
Sophia Antipolis, Januay 2003.
[5] J. Colley, G. Stead, Mobile learning = collaboration, In the
Proceedins of the third European Conference on Mobile Learning,
MLEARN 2004, Rome, Italy, 2004, ISBN 1 85338 855 6.
[6] M. F. Vaida. JAVA 2 Enterprise Edition (J2EE) Aplicatii
multimedia, Ed. Microinformatica, Cluj, 2002
[7] N. Capuano, M. Gaeta, S. Miranda, L. Pappacena, A System
for Adaptive Platform-Independent Mobile Learning, In the
Proceedins of the third European Conference on Mobile Learning,
MLEARN 2004, Rome, Italy, 2004, ISBN 1 85338 855 6.
[8] UIML 3.0 Specifications: http://www.uiml.org/specs/uiml3/,
February 2002.

327
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

A new approach of op-amp amplifier biasing

Gabriel Oltean 1, Emilia Sipos 1, Ioana Oltean 2

Abstract A new method to deal with the biasing

problem of op-amp amplifier is presented. The purpose
is to provide for the students a structured method that II. THE NEW METHOD
highlights the line of reasoning to solve the problem of
amplifying a signal with no d.c. component, using an op- The desired translated VTC looks like the one in
amp operated from a single power supply. The main Fig.2.
idea is: the signal source should see a VTC that allows
both positive and negative swing for the signal. The vO [V]
paper presents the general method and its application VPS
for a non-inverting and for an inverting amplifier.
Keywords: VTC, bias voltage, equivalent circuit VO=VPS/2
I. INTRODUCTION
0 vI [V]
Although in most of the op-amp amplifier
Fig.2. The translated VTC
applications the supply is differential, in some cases
exist only one power supply (e.g. auto equipments). Presentation of the method is much easier to explain
The voltage transfer characteristic-VTC of an op-amp on some block schemes. The initial amplifier with
amplifier operated from a single (positive) power VTC given in Fig.1 can be represented by the block
supply is the one given in Fig.1. As all of us know,
with this VTC we cannot directly amplify a vi(t)
signal with a zero dc level. The solution is to bias the +VPS
circuit to operate around a point near the middle of the
VTC. In the electronic circuit books [1], [2], [3] is
presented only the final solution (applying a dc ~
voltage VI to obtaining at the output the dc voltage VO vi vI Av vO
in the middle of VTC). The reasoning used to reach
that final solution is well hidden for the students. Due
to this lack of information, the students are not able to
solve the same problem for a new circuit or in another Fig.3. The block scheme of a single
similar situation. supply amplifier
Our idea is to present a structured method to deal with scheme in Fig.3.
the amplification matter in an op-amp amplifier
operated from a single power supply. The method is The amplifier see to its input a voltage with only the
based on the following idea: the signal source should variable component vi, so the output voltage will have
see a different VTC, so that for a zero input voltage only a variable component too. The amplifier
the output should be placed in the middle of the equations are:
output voltage range.
vO [V] vI = vi (1)
VPS v O = v I Av (2)
vO = vi Av (3)
VO=VPS/2 vo = vi Av (4)

0 vI [V] Comparing the Fig.1 and Fig.2 one can observe that
VI the VTC in the second figure results by horizontally
Fig.1. The VTC of a single supply op- translation of the initial VTC with an appropriate dc
amp amplifier voltage, lets name it VBIAS. This translation is made
1
Technical University of Cluj Napoca,, Str. C. Daicoviciu 15, 400020, e-mail Gabriel.Oltean@bel.utcluj.ro
2
Electrotechnical School Group Edmond Nicolau, Cluj-Napoca
328
on the x axis and is obtained by series connection of a
VPS
+VPS
R
VBIAS
A B A C1 B
~

vi vI vI Av vO vi R

Fig.5. The circuit necessary to obtain the biasing

Fig.4. The block scheme to translate the VTC
with VBIAS and it plays the role of a voltage source,
VBIAS source to the input of the amplifier, as it is resulting the equivalent scheme in Fig. 6.
presented in [4]. The VBIAS value should be in
accordance to the value of the dc voltage needed at VBIAS
the output. Therefore, the value of VBIAS will be A B
choused such a way to receive for the output voltage
a value equal with half of the supply voltage VPS. The
block scheme to obtain the translated VTC (Fig.2) is vi
given in Fig.4.
The amplifier see now to its input a voltage vI equal Fig.6. The equivalent circuit for the circuit from Fig.5
with the sum between the bias voltage VBIAS and the
variable input voltage vi. The output voltage vO will The final block scheme that solves the problem of
also have a dc and a variable component. amplifying a zero dc level signal in an amplifier
operated from a single power supply is given in Fig.
vI = VBIAS + vi (5) 7.
vO = vI Av = VBIAS Av + vo Av (6)
+VPS
Because the dc output voltage must be at the middle
of the VTC results that we should have: R
A C1 B
VBIAS Av = VPS / 2 (7)
vi R vI Av vO
V /2
VBIAS = PS (8)
Av

So, beside the small signal gain also results a dc gain. Fig.7. The final block scheme of the biased amplifier
To improve the operation of the amplifier (reduce the
effects of finite input dc offset voltage) it is often a
III. METHOD ILLUSTRATIONS
great idea to roll off the gain to unity at dc,
especially if the amplifier has large voltage gain [3].
A. Non-inverting amplifier
This can be done by using a capacitor in the negative
feedback path of the amplifier. This capacitor should
The circuit with the VTC presented in Fig.1
be placed so that it can be able to play a double role:
corresponds to a non-inverting amplifier that looks
first, to set the desired a.c. gain (Av) (by its short-
like the one in Fig. 8. To be able to use this circuit to
circuit equivalence in the small signal regime) and
amplify the vi voltage we apply all the steps presented
second, to set a unitary d.c. gain (by its open-circuit
in the previous paragraph.
equivalence in the d.c. regime).
7 +VPS
The voltage output equation of the amplifier became:
+
vO = VBIAS + vo Av (9) 6
vO
R1
2
To avoid the use of an extra dc voltage source, VBIAS -
vi 1K 4
can be obtained by a resistive voltage divider from
VPS to the ground. For equal resistances the value of R2
VBIAS results VBIAS = VPS/2. To prevent the dc current
to flow through the input signal source we use a 10K
capacitor to insert to the input the VBIAS dc voltage - Fig.8. The non-inverting amplifier
see Fig. 5. In the steady state regime (after the ending
of the transient regime) this capacitor is charged up

329
First we translate the VTC using the VBIAS voltage The equations in the circuit are:
series connected at the input of the amplifier. The
VBIAS voltage is obtained through the voltage divider R2 R
from the power supply voltage VPS. The series v O = (v in + V BIAS )(1 + ) 2 V BIAS (10)
R1 R1
connection is realized by the C1 capacitor, resulting
the intermediary circuit depicted in Fig. 9. v O = v in Av + V BIAS (11)
R
4 Av = 1 + 2 (12)
12Vdc VPS R1
R3
1k
C1 0 The VTC and the waveforms of the input voltage with
3 zero dc level vin(t) and the final vO(t) obtained after
+
10u 6 vO
simulation are presented in Fig. 12.
R4
2
1k
0
R1 R2
vin 0
1k 10k

(a)
0 0

Fig.9. The non-inverting amplifier after connecting

the bias voltage

To roll off the gain to unity at dc, we introduce

another C2 capacitor on negative feedback (Fig. 10).
7

12Vdc (b)
VPS
R3
1k
C1 0
3
+
4
10u 6
vO
R4
2
1k -
0
R1 R2
vin 0 Fig.12. The simulation results
1k 10k (a) VTC; (b) waveforms
C2
10u B. Inverting amplifier
0 0

Fig. 10. The final circuit for non-inverting amplifier In Fig. 13 is presented the circuit and in Fig. 14 the
VTC of an inverting amplifier.

The equivalent circuit that shows how the VBIAS

voltage appears in the circuit is the one in Fig.11. The 7 +VPS
equivalence is made for the steady state regime, where 3
+
the capacitors can be 7replaced with voltage sources. vO
6
R1 0
4 VPS 2
12Vdc -
4 0
VBIAS=6Vdc
0 vi R2
3 +
6 vO
0
2
- Fig.13. The inverting amplifier
vin R1 R2
0
1k 10k According with our previously presented method the
VBIAS=6Vdc VTC should be horizontally translated with VBIAS.
Unlike the non-inverting amplifier, in this case only
0 0
the C2 capacitor is necessary on the negative feedback
Fig.11. The equivalent circuit for non-inverting amplifiers

330
 vO [V]
The simulation results are presented in Fig. 17: the
VPS VTC in Fig 17.(a) and the waveform of the input
voltage vi(t) and the final voltage vO(t) in Fig 17.(b).
VO=VPS/2 The simulation results showed that our method is
correct.
0 vI [V] 12V
VI
Fig.14. The VTC of the inverting amplifier
8V

path of the amplifier, because the input is applied at (a)

the non-inverting input of the op-amp. The capacitor 4V
realized both the series connection of the VBIAS with
input voltage and the unitary dc gain. The final circuit
is the one in the Fig. 15. 0V
-5.0V 0V 5.0V
+VPS V(VO)
7 V_Vin
R3

VBIAS 10V
3
+
vO (b)
6
C2 R1 5V
4 vi 2
-

R4 0 0V
R2
0s 2.5ms 5.0ms
0 V(VO) V(Vin:+)
10k
Time
Fig.15. The final circuit for inverting amplifier
Fig.17. The simulation results for inverting amplifier
(a) VTC; (b) waveforms
The equivalent circuit that shows how the VBIAS
voltage appears in the circuit is the one in Fig.16. The
equivalence is made for the steady state regime, where IV. CONCLUSIONS
the capacitor can be replaced with a d.c. voltage
source and also the voltage divider can be replaced We presented an intuitive method to deal with the
with a d.c. voltage source. amplification of a zero d.c. level variable signal using
an op-amp operated from a signal power supply. The
examples presented here (for a non-inverting and for
12Vdc VPS an inverting op-amp amplifier) demonstrates the
usefulness of the method, especially for the students
in the struggle with electronics. This method can
7
save classroom instruction time and help the students
3 5 0 to understand and easily solve this kind of problem.
+
VBIAS=6Vdc 6 vO This method can be further use to try new way to bias
R1
2
some amplifiers in a desired operating point, even
- transistor amplifiers.
vin 1k
VBIAS=6Vdc 4 0
R2 REFERENCES
0 0 10k
[1]. Sedra, A.S., Smith, K.C., Microelectronic
Fig.16. The equivalent circuit for inverting amplifier Circuits, Holt, Rinehart and Winston, Inc, 1987
[2]. Oltean, G., Dispozitive si circuite electronice.
The equation of the amplifier is:
Dispozitive electronice, Risoprint, Cluj-Napoca, 2003
[3]. Horowitz, P, Hill, W., The Art of Electronics,
R2 R
vO = (vin + VBIAS ) + (1 + 2 )VBIAS (13) Cambridge University Press, 1997
R1 R1 [4]. Oltean, G., Gordan, Mihaela, Oltean, Ioana, A
R2 new method to deduce the voltage transfer
v O = v in + V BIAS (14) characteristic for some two-port network, Acta
R1
Tehnica Napocensis Electronics and
R2 Telecommunications, Vol. 40, Nr. 2, 2000, pp 17-20.
Av = (15)
R1

331
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
A Toolkit for Internet Based Distance Laboratory
Development
Sebastian V. Tiponut1
Abstract This paper describes the design and
implementation of a distance laboratory, suitable to be
used in a large range of experiments with proper
customization. Using the well-known client-server
paradigm, this design tries to accomplish three goals:
simplicity of design, future extensibility and ease of use.
This implementation differs from other existing distance
laboratory projects in the following way: Firstly, it can
be interfaced with a wide range of experiments.
Secondly, the core of this system has been designed using
XML for a greater flexibility. Finally, it was written to
allow further development and enhancements.
Keywords: distance learning, distance laboratory,
Internet, education
I. INTRODUCTION
Experimentation is the cornerstone of the modern
science, a way to convince others of the correctness of
a particular scientific theory or hypothesis. Even if
experimentation is of such importance, one may doubt
the reasons for which students are reenacting
experiments with already well known results. To
enhance the learning techniques, provide more insight
and consolidate the knowledge on a particular topic
the best way is to let the students do the experiment
themselves. Most of the time, theory is too dry and
academical abstracts are just not enough to provide a
thorough understanding of a certain matter. Thus, real
life situations are ideal to supplement the theory,
teaching the student how to interpret and correlate
experimental results with a particular hypothesis. It is
very important that, besides the experiment goals, the
art of experimental designs be communicated to the
student through the laboratory experience.
Distance laboratories are a new appearance in the
field of distance learning. They are meant to provide a
hands-on approach, allowing experiments to be
carried out from remote locations rather than
requiring the personal presence of the human
operator. As with any new technique, many
advantages and drawbacks have already been spotted.
To be successful and to prove itself as a viable
alternative to classic experiments, distance
laboratories must offer more benefits than
1
Inginer, S.C. Spidernet S.R.L. Calea Bogdanesti 6A
Timioara, e-mail seba.tiponut@spidernet.co.ro
332
deficiencies. Distance laboratories do not require
synchronization between the instructor and the
students; experiments can be accessed whenever the
student needs them. Safety issues are also eliminated
because dangerous equipment is not physically
handled and unsafe experiments are performed
remotely. Distance laboratories can also nullify the
effects of space and time allowing access to
experiments from a particle accelerator, nuclear
research center or to obtain weather or seismographic
data. Among the limitations that a distant laboratory
may suffer are the lack of control of the experiment
due to poor design of the user interface, need of
technology standards for software and hardware and
the intrinsic technical difficulties in designing such a
complex system.
Different approaches to designing distance
laboratories have been taken. National Instruments
[12], the company which is producing the Labview
software has designed a toolkit which, in conjunction
with Labview software, can be used to create user
interfaces meant to control distant experiments [2],
[3], [4]. A few distance laboratories have already been
deployed using this toolkit: Cyberlab from Stanford
University, the Virtual Laser Laboratory from
Dahlouise Laboratory or the Lecture Enhancement
from Swiss Federal Institute of Technology. Another
example, presented in [1], uses Java applets to create
a graphical interface and custom client-server design
to connect the interface to the experiment. [5]
describes the remote access of a chemistry experiment
using a Web interface to collect data, display and
analyze the results.
This paper introduces a new design for such a
distance laboratory. Instead of concentrating on
implementations, we choose to detail the parts
involved in this system, the communication protocol
and the interfaces between them. This will help others
to build similar laboratories - using other
programming languages, on different platforms and
targeting a broad range of experiments. Our approach
boasts the benefits of a standardized system which can
undergo several implementations while keeping
information exchangeability and compatibility.
II. GENERAL PRESENTATION OF THE SYSTEM
The proposed distance laboratory system consists of
seven elements which are interacting using a set of
predefined, well-known, interfaces. The core of this
system is a client-server structure. The client is
represented by a graphical user interface which
communicates with the server using predefined
network routines; its function is to display collected
data. To mediate the concurrent access to the physical
experiment of a large number of clients, the server
part has been designed. It comprises of networking
routines, decision code and a user-written driver
whose purpose is to interface the server with various
data acquisition systems which collect data from and
control the experiment.
To make the graphical interface more adaptable to
different types of experiments, a predefined, custom
set of widgets has been designed. This widget set -
which is supposed to be extended and even
standardized - contains from simple primitives like
buttons, frames to complex objects in form of gauges,
2D or 3D displays, sliders etc. In order to facilitate the
construction of various graphical interfaces in a very
simple and platform independent manner, the
graphical interface is described in XML. An XML
parser is then building the interface based on this file,
instantiating the widgets from a library. One can
associate an experiment with an XML description of a
graphical interface, customizing the latter to suit
experiment's needs.
One server can host multiple experiments, possibly
different experiments. Because of that they must be
controlled using distinct graphical interfaces. Thus a
need arises to associate a particular experiment with
the correspondent XML description of the graphical
interface. This association was done using a very
simple HTML-like page where, modeled after the
World Wide Web. The path of the graphical interface
description is associated with a hyperlink which, if
clicked, will start the downloading process of that
particular XML description. It also supports some
primitive form of text formatting. A special browser is
used to display the page downloaded from a separate
server. The user can click on the hyperlink, have the
XML description downloaded and then the graphical
interface built from it. Once the graphical interface is
generated the experiment may start. A general view of
the system is depicted in Fig.1.
As stated in the Introduction, this design emphasizes
portability and standardization. Thus an
implementation must comply with the specified
protocols, file formats and interfaces. There are two
communication protocols: the first one is mediating
333
the communication between the graphical user
interface and the experiment; the second is a very
simple file transfer protocol used to retrieve the page
containing the list of the available experiments. The
former protocol has double purpose: it transmits the
commands issued from the graphical interface and it
transmits back the data from the server.
Fig.1 General view of the system
The file format for the page listing the experiments
along with a short description is an XML page which
contains tags which specifies the location of the
graphical interface description on the server. It
specifies for each experiment the experiment
identification and an arbitrarily long description of the
experiment itself.
At the core of creating the user interface stays the
XML file containing its descriptions. It is composed
of tags, each tag describing a widget along with its
characteristics. Some widgets rely upon the presence
of others; thus, the XML parser is doing a check to
assure no mistakes have been made during file
editing. The library of widgets content will be
described later on, in the section dedicated to
implementation.
The physical experiment is connected to the server via
a data acquisition system and a driver which is
mediating between the two. Because the data
acquisition system can be connected to the computer
using a serial, parallel, I2C or other type of
connection, the driver must be designed to handle
these particularities. Another issue the driver must
solve is the communication protocol with the data
acquisition system.
III. PRESENTING AN IMPLEMENTATION
The author has chosen to implement a distant
laboratory using the design presented above. The
programming language in which the whole distant
laboratory was written is Python. It has been selected
because it is a very powerful object oriented
programming language, supporting well designed
XML and general purpose libraries. Configuration
files, written in XML (for ease of parsing), have been
designed for all the components.
The heart of the system is the parser which is
generating the graphical interface by interpreting an
XML file containing the interfaces' description, as
presented above. This parser is a standalone program
which is usually called within the browser once an
XML file with the interface description has been
downloaded; it can also be ran from console, passing
the file to be parsed as its first argument. A minimal
set of widgets have been designed to allow the
construction of a graphical interface suitable to be
used for basic experiments. Among the widgets
included in this set are:
a graphical XY display; one can customize
the number of ticks and the start and stop
values separately for each of the two axes;
a sliding bar which can be used to input
values; the start/stop values and the
increment are customizable;
an amplifier which multiplies the values
outputted by the sliding bar with a certain,
configurable, factor;
buttons which can be used for a variety of
operations;
frames, used to group the other widgets
together.
The server which is serving the user interface XML
descriptions is called the cache_manager. The server
which is used by the user interface to mediate the
communication with the hardware is called the
gateway. Both servers have XML configuration file.
IV. THE PILOT EXPERIMENT
This is a simulated experiment which is supposed to
test the distant laboratory whose implementation was
presented above. A piece of software is meant to
substitute the data acquisition system and the physical
experiment. This is simulating the characteristic of the
diode which can be obtained by applying an
increasingly high voltage between the anode and the
cathode of the silicon diode. The use inputs different
voltage values which will be sent to server for
evaluation. The simulation plugin will compute the
associated current value and it will reply with pair of
coordinates which represents a new point on the
graphic.
After the graphic has been drawn it can be saved for
later use. Saving a graphic is also saving the
environment (the number of ticks and the values from
both axis and the measurements units).
V. DISCUSSION
This paper presents the design of a distant laboratory
along with a possible implementation. One of the
334
goals the author had in mind when conceiving the
structure of this system was to make it versatile for a
wide range of experiments. The widget library can be
extended with new types of widgets which can be
used to broaden the types of experiments for which
this model can be used.
Another goal which was targeted was to make the
system as portable as possible. That is why it was
designed as an open project the interfaces and the
protocols between the components are presented in
detailed so that one can implement a compatible
system or just a part of it. While the system could
have been designed platform independent and
distributed as a software bundle we rather wish to
emphasize the benefits of a known protocol and an
open design schematic.
Last but not least, the system was thought to be open
to improvements and modifications while maintaining
backward compatibility. This makes this design very
different from existing implementation of distance
laboratories which are - the vastness majority - close
source and very hard to develop and to integrate.
The implementations decisions are also making this
distance laboratory different from other distance
laboratories. It is using XML as its core for both
creating a graphical interface for a given experiment,
communication between components and
configuration files. The servers are designed such way
that they can mediate the access of a large mass of
users to the available experiments. Using the Python
language for all the components offers some degree of
portability. The system was conceived to be used on a
Unix machine but with minimal changes it can be ran
on other platforms as well.
REFERENCES
[1] T. Chang, D. Hunt, Web-Based Distance Experiments: Design
and Implementation' 2000 International Conference on
Engineering Education, Taiwan.
[2] J. Henry, LabVIEW Applications in Engineering Labs, ASEE
Conference Anaheim, California, June, 1995.
[3] J. Henry, Controls Laboratory Teaching via The World Wide
Web, ASEE Paper, June 1996.
[4] J. Henry, Internet Laboratory Server in Engineering Systems
Laboratory, Conference at M.I.T., June, 1997.
[5] H. Shen, Z. Xu, V. Kristiansen, O. Strom, M. Shur, Conducting
Laboratory Experiments over the Internet, IEEE Transaction on
Education, Vol. 42, pp. 180-185, August, 1999.
[6] R. Berntzen, J. Strandman, T. Fjeldly, Advanced Solutions for
Performing Real Experiments over the Internet, 2001 International
Conference on Engineering Education, Norway.
[7] B. Aktan, et. al. Distance Learning Applied to Control
Engineering Laboratories, IEEE Transactions on Education, v37,
No. 3, pp. 320-326.
[8] D. Knight, S. DeWeerth, A Distance Learning Laboratory for
Engineering Education
[9] J. A. del Alamo, et. al. Educational Experiments with an
Online Microelectronics Characterization Laboratory
[10] E. Bobkov, Y. Sheynin, The Web-based Technology in
Laboratories for Distance Learning and Training
[11] J. C. Piower, et. al. Web-Based Educational Experiments
[12] http://www.ni.com/academic/live_experiments.htm
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
A Complete Laboratory on Evolutionary Electronics
Rustem Popa1
Abstract Evolutionary design might be a promising
option to the conventional design of electronic circuits.
Each project is assembled from a number of component
parts and then is tested in the frame of an evolutionary
algorithm. We have presented in this paper some
evolutionary experiments of digital and analog electronic
circuits design, both by simulating evolution in software
and by true evolution in hardware. These experiments
are conducted with our students in the laboratory of
Evolutionary Systems.
Keywords: genetic algorithms, evolvable hardware,
electronic circuits, evolutionary computation
I. INTRODUCTION
Evolutionary systems are designed by the means of
evolutionary computation. These designs are
evolved by a process of natural selection, like in the
living matter. The mechanism of evolution is entirely
blind and has no particular object other than
survivability. The survivability of the organism can be
seen as a process of assembling a larger system from a
number of component parts and then testing the
organism in the environment in which it finds itself.
The concept of assemble-and-test together with an
evolutionary algorithm can explore the entire design
space because of the absence of imposed rules of
design. In this way, in electronics, evolutionary design
generates new unexpected and usually useful
electronic circuits.
The building of new electronic circuits by
evolutionary computation has been created the
concept of Evolvable Hardware (EHW). The usual
design process is in a top-down way and begins with a
precise specification. EHW is applicable even when
no hardware specification is known before. Its
implementation is determined through a genetic
learning in a bottom-up way. A Genetic Algorithm
(GA) is intended to mimic Darwinian evolution. A
population of solutions, called chromosomes, is
maintained, and goes through a series of generations.
For each new generation, some of the existing
chromosomes survive, while others are created by a
type of reproduction and mutation from a set of
parents. EHW combine knowledge of both GA and
electronic circuits design to evolve new circuits.
1
Universitatea Dunrea de Jos din Galai, Departamentul
Electronic i Telecomunicaii, Str. Domneasc Nr. 111,
800201, Galai, e-mail: Rustem.Popa@ugal.ro
Research in EHW can be divided into intrinsic
evolution, which refers to an evolutionary process in
which each circuit is built in electronic hardware and
tested, and extrinsic evolution, that uses a model of
the hardware and evaluates it by simulation in
software.
We are convinced that very soon, as reprogrammable
integrated circuits will become larger and larger, and
the design techniques will be improved, EHW will be
dominant in electronics, and the electronic engineer
must be ready for this future evolution. It is true that,
for the time being, the complexity of evolved circuits
is so far small. The main problem is the representation
of the circuit in chromosomes, because complex
circuits need a great number of architecture bits,
which directly influences the GA search space.
We have prepared some experiments of extrinsic
evolution in digital and analog circuits, and for
solving of some wellknown optimization problems,
like the generation of test vectors in digital circuits,
the finding of the global minima in a multimodal
function, or the solving of the Traveling Salesman
Problem (TSP). We have also prepared two
experiments of intrinsec EHW in digital circuits by
using common digital CMOS circuits. The first one is
a test platform with controlled switches for
experiments with simple building blocks made-up
from few transistors. The second platform consists of
CMOS switches, some simple logic gates, and three
JK flip-flops for experiments with registers and
counters. Finally, a real Xilinx XCR3064 CoolRunner
CPLD mounted on a XCRP board may be used to
implement some extrinsic EHW circuits.
The remaining sections of the paper are organised as
follows: Section II describes in more detail the genetic
learning component of the EHW. Section III shows
some examples of digital circuits designed by
software with a GA and then implemented in a 64
macrocell Xilinx CPLD. Section IV shows some
examples of simple analog and digital circuits
implemented by evolution on real hardware
configurable boards. Finally, Section V provides the
conclusions and future work.
335
 Fig. 1. A GUI for the search of a test vector which
points a stuck-at 0 fault in the marked node
II. GENETIC LEARNING IN EHW
All the developed algorithms are based on GAs, an
adaptive searching technique for solving optimisation
problems based on the mechanics of natural genetics
and natural selection. The success of the application
of GAs to an optimisation problem depends on the
representation of chromosomes, fitness function,
method of crossover, mutation operation, and on the
diverse information from the chromosomes. When the
diversity is lost before the global optimum solution is
found, the performance of GAs deteriorates and their
solution processes converge prematurely. Moreover,
the mutation operation is important. While the
mutation operation adds new information to a
chromosome, it can also destroy useful information
held in the chromosome.
In GAs the search is conducted using information of a
population of candidate solutions, called
chromosomes, so that the chance of the search being
settled in a local optimum can be significantly
reduced. Four essential components need to be
designed in applying a GA for an optimisation
problem: chromosomes representation, crossover
operator, mutation operator and fitness function.
In a reconfigurable circuit, each bit of a chromosome
represents usually the state of a programmable switch.
The entire chromosome represents the state of all
switches, that is a complete circuit, which may be
good or bad, according with his fitness. The initial
population of chromosomes (bit strings) is generated
randomly. All these potential solutions are evaluated
using a fitness function. In our case, for a single
boolean function, fitness is the ratio between the
number of the correct values of the function and the
number of all possible values (which is 2 , if the
n
boolean function has n input variables). A well-
designed circuit will be obtained only when the value
of fitness is 100%. An approximately value of the
fitness is unacceptable here.
The next step is selection and reproduction. For each
individual, a number of copies are made, proportional
336
Fig. 2. A GUI for the generation of an optimum
set of test vectors by hibridation of a GA
to its fitness, while keeping the population size
constant. The least fit individuals are deleted. This is
the survival of the fittest part of the GA.
The next step is crossover, where individuals are
chosen two at a time, as parents. They are converted
into two new individuals, called offsprings, by
exchanging parts of their structure. Thus, each
offspring inherits a combination of features from both
parents. We have obtained the best results with one
point crossover, with a probability of 80%. This
operator may be used more times on different selected
pairs of chromosomes in a generation.
The next step is mutation. A small change is made to
each resultant offspring, with a small probability.
After mutation is performed on an individual, it no
longer has just the combination of features inherited
from its two parents, but also incorporates the
additional change caused by mutation. This ensures
that the algorithm can explore new features that may
not yet be in the population. It makes the entire search
space reachable despite the finite population size. The
whole process is repeated for several generations, and,
if the best chromosome in population will have the
fitness of 100%, then this bit string represents a good
solution for our function.
III. EXPERIMENTS WITH EXTRINSIC EHW
The first set of experiments show the generation of
complexity with very simple rules in unidimensional
and bidimensional Cellular Automata (CA), and the
solving of some complex NP-problems (the finding of
the global minima in a multimodal function, or the
solving of the TSP) with GAs. These experiments
have been ample described in [7].
Another set of experiments have been prepared for the
purpose of automated generation of test vectors in
digital circuits. If we want to generate a test to detect
a stuck-at 0 fault in the marked node of the circuit
represented in the Fig.1., the required vector is
1111111111000000000, a combination of bits nearly
impossible to find using a random approach ([6]). As
 x1
x2
x1
f
x3
Fig. 3. The circuit achieved by evolution for the
boolean function from the equation (1)
we can see in the Graphic User Interface (GUI) from
the Fig.1, the GA used to solve this problem has
found the correct solution in 40 generations. The
algorithm uses a population of 32 chromosomes and a
mutation rate of 3%. Fitness was calculated as the
sum of (1 if fault is excited or 0 otherwise) + (fraction
of inputs in AND gate set to 1) + (fraction of inputs in
OR gate set to 0). The maximum value of the fitness
defined in this way is 3.
By using the GUI from the Fig.2., we can solve the
Fault Coverage Code Generation Problem for a more
complex combinational logic. The problem consists in
finding of a given number of test vectors that
maximizes the fault coverage of the circuit. We have
chosen two ways of hibridation of the standard GA:
by using the inductive search, like in the Fig.2., or by
using the simulated annealing algorithm. The example
from the Fig.2. shows that only 6 test vectors could
cover more than 75% from the total number of stuck-
at 0 faults in the circuit. All these GUIs (and also
those from the first set of experiments) have been
developed in Matlab 5.3.
A. The Implementation of a Boolean Function
We have considered a boolean function represented in
a minimal disjunctive form by using a Karnaugh map:
f = x1 x 2 x3 + x1 x3 + x 2 x3 (1)
This representation has a cost of 7 gates and 13
inputs, including inverters. By applying some
switching-algebra theorems our function may be
written in the next form:
f = x3 x1 x 2 (2)
Now, the cost of implementation is of only 3 gates
and 5 inputs. Unfortunately, there is no algorithm to
find this convenient form of the function, only the
heuristics and experience of the human designer.
We have tried to find another representation of this
function by evolutionary design. We have used the
Fig. 4. The evolution of the fitness across 50
generations
idea given in [2]. Each combinational circuit is
represented as a rectangular array of logic gates. Each
of these gates has two inputs and one output, and the
logic operator may be selected from a list. At the
beginning of the search, all the gates from the matrix
are disposable to implement a functional circuit. Once
a functional solution appears, then the fitness function
is modified such that any valid designs produced are
rewarded for each gate which is replaced by a simple
wire. The algorithm tries to find the circuit with the
maximum number of gates replaced by wires that
performs the function required.
The chromosome defines the connection in the
network between the primary inputs and primary
outputs. We have used a network of 4 gates, a
population of 32 chromosomes, 10 of them beeing
changed each generation, a single point 100%
crossover and 5% rate mutation. A feasible solution
has been obtained in less than 50 generations, as we
can see in the Fig.3. and in the Fig.4. The cost is given
now by 3 inverting gates and 6 inputs (one of the
gates in the network is useless), and this solution has
the minimum delay time between any input and the
output of the circuit, in a gate level implementation.
B. The Implementation of a Finite State Machine
The Finite State Machine (FSM) represented in the
Fig.5. is a sequence detector with one-input, one-
output and 6-internal states. When the input sequence
011 occurs, the output becomes 1 and remains on this
logic value until sequence 011 occur again. In this
case, the output returns to 0, and maintain this value,
until a new sequence 011 appears.
Initially a GA has been used to find optimal state
assignment. The chromosome represents the FSM as a
list of states. The initial population is generated
randomly. The goal of the GA is to extract the
optimum state assignment, which requires the least
number of logic gates. For that reason the number of
2-inputs AND/OR logic gates are used to define the
fitness function. The optimum state assignment is
given in the Fig.5. A more detailed description of this
problem is presented in [1].
337
 1

X
S/Y S0/0
0
X
0 S1/0
1 0

S2/0
1 D Q Y
1 2
Q
1 S3/1
S0: 000 0 D Q
S1: 010 1
S2: 001 0 S4/1 Q
S3: 100 1 0
S4: 110 D Q
S5: 101 S5/1 0
CLK Q
Fig. 5. A sequence detector described as state
transition graph and GA state assignment Fig. 6. Evolved optimal circuit solution of the
sequence detector

Then, the extrinsic EHW has been used to find the

functional design of combinational parts of the
sequence detector. We have used the same method
presented in the subsection A and in [2].
The equations of the evolved optimal combinational
circuit are the following:
The software we have used is Xilinx Integrated
Software Environment (ISE) 6.1i, a complete CAD
environment for implementation of complex digital
circuits. We have generated the source file of the new
project (schematic diagram or VHDL) and have
obtained the fitter report and the timing report.

D 2 = Q 2 Q0 + x Q 2 + x Q 2 Q0 (3) We have implemented the boolean function from the

D1 = x
D0 = x Q1
y = Q2
The schematic diagram of the circuit is given in the
Fig.6. A bad state assignment may conduct to much
more complex equations for the combinational circuit
of the FSM.
C. Some experiments with Xilinx XCR3064XL CPLD
subsection A on the basis of equations (1) and (2) and
the circuit from the Fig.3. We have obtained the same
(4)
results, so we can assume that our software finds an
optimal way in connecting the hardware resources of
(5) the circuit, even if the function is not done in a
minimal form. The circuit has used a single macrocell
(6) from the maximum number of 64 (that is 1/64), only
two product terms from the maximum number of 224
(that is 2/224), and only 3 function block inputs from
the total number of 160 (that is 3/160). The pad to pad
delay is 6 ns, and the total delay of the circuit is not
more than this value. For more complicated functions,
evolutionary design may offer a better fitting of
circuit resources (a less number of product terms).

The circuit XCR3064XL, is a Xilinx CPLD with 64
macrocells and 1500 usable gates, providing low-
power and very high speed, and beeing in-system
programmable through JTAG IEEE 1149.1 Interface.
Unfortunately, this circuit has only 1000 erase/
programming cycles guaranteed, so it can not be used
with intrinsic EHW.
This programmable circuit is mounted on a board,
called Digilab XCRP, delivered by Digilent, Inc. This
low cost platform can be used to implement a wide
variety of digital circuits. The programming pins of
the circuit are directly connected to the parallel port
pins of the computer.
In sequential circuits, the optimal state assignment is
crucial. The sequence detector from the subsection B,
implemented with the equations 3,4,5 and 6, has used
only 3/64 macrocells, 3/224 product terms, and 3/160
function block inputs. The same circuit, implemented
with a non optimal state assignment has used 4/64
macrocells, 9/224 product terms, and 4/160 function
block inputs. Even the combinational time delay is
different for these circuits (4.7ns in the first case and
7,2ns in the second case). Its true that the main
differences in the complexity of these circuits are
given by the state assignment, but it seems that
evolutionary design is more efficient even for the
combinational part of a FSM.

338
 1 0 1 R sw
OUT
1 1
1 1
+5V GND IN
Fig. 7. Instantiation of the NOT gate on the
evolvable testbed
D. The Implementation of Analog Circuits
Analog circuit synthesis entails the creation of both
the topology and the sizing (numerical values) of all
of the circuits components. The difficulty of this
problem is well known and the first auspicious
approach, based on genetic programming, was
presented in [3]. Another method of automatically
generating analog circuit designs based on a parallel
GA and a set of circuit constructing primitives is
presented in [5].
Both methods need a huge computation power (few
days on a parallel computer with 64 processors in the
first case, or a network of workstations in the second
case). We have only verified through PSpice
simulation some of the circuits presented in [3] and
[5]. Interesting is the fact that not all given circuits
have been successful in our simulations.
IV. EXPERIMENTS WITH INTRINSIC EHW
We have prepared two set of experiments on intrinsic
evolvable hardware of digital circuits by using
common digital CMOS circuits.
A. A Test Platform for Intrinsic Analogic EHW
The first one is a test platform designed specifically
for simple experiments into intrinsic hardware
evolution. Based on an idea from [4], this testbed is in
fact a matrix of analogue switches, connected to some
plug-in boards, which contain the desired building-
blocks for experimentation. In [4] is described a great
motherboard with 12 integrated circuits (IC)
CD22M3494, each of them beeing a matrix of 16 8
analogue switches. These ICs are very expensive, so
we have built a much smaller board with only 20 ICs
4066, each of them having only 4 analogue switches,
that is a total number of 80 programmable switches.
As a starting point for experimentation, bipolar
transistors were used as the evolutionary building-
block, and the task was to evolve a NOT gate. The
digital input to the testbed is provided by a computer
via a digital input/output board, and the output is
connected to an A/D converter on the board.
339
R sw
+5V
R sw R sw
IN
OUT
R sw
R sw
GND
Fig. 8. The equivalent circuit diagram for the NOT
evolved gate
We have used a standard GA, with a population of 50
chromosomes, 18 bit each of them, with a single point
crossover, proportional selection and elitism. The
mutation rate was 5%. The evolved circuit with a
single transistor is shown in the Fig.8. The
corresponding chromosome that build up the state of
the analogue switches represented in the Fig.7. is
100100010001001010.
The on-switches resistances might be about 50 for
CD22M3494 circuit, or about 300 for 4066 IC. The
circuit from the Fig.8. conforms to the NOT function
in that its output corresponding to 0 input is of slightly
higher voltage than that corresponding to a 1 input,
however this difference (only 1V) is too small to be of
any practical use. If we repeat evolution, with much
more switches, we can see that these additional
switches are placed in parallel, reducing the combined
resistance from the emiter. This configuration will
give a good voltage swing and the circuit will become
an inverter with a NPN transistor, like we know.
B. A Test Platform for Intrinsic Digital EHW
The second platform consists of CMOS switches,
some AND gates and three JK flip-flops for
experiments with registers and counters. The most
suitable way to connect each data input of a flip-flop
to a lot of different signals from the circuit, is by
using CMOS analog multiplexers/demultiplexers.
The schematic diagram from the Fig.9. shows the
building block used to design this board. Each data
input of a JK flip-flop has an 8-channel analog
multiplexer 4051. The first two inputs in the
multiplexer are constants 1 and 0. The next 3 inputs
are the direct or inverse outputs of the flip-flops,
selected by 2-channel 4053 multiplexers. Finally, the
last three inputs in the multiplexer produce AND
functions between any two different flip-flop outputs.
A building-block uses 12 bits, so the length of a
chromosome is 36 bits. We have used a standard GA
with a population of 100 chromosomes, with a single
point crossover, the mutation rate of 1%, proportional
selection and elitism.
 Q2 1 Q
Q2 0
Q1
Q1 MUX
Q0
Q0
4053 4027 Q
i
J Q
4051
K Q
CLK
4081
CLK
Fig. 9. The building-block for intrinsic digital EHW
An example of an evolved counter with 5 states is
presented in the Fig.10. We have used all the three
building-blocks from the board, including an AND
gate.
V. CONCLUSIONS
Evolutionary design is in fact a creative machine for
new designs and may be useful for electronic
engineers. The experiments presented here display the
generation of complexity with very simple rules, and
the solving of complex NP-problems with simple
GAs. GAs may be useful for automated generation of
test vectors and for synthesis of digital and analog
circuits. Analog circuit synthesis usually needs more
powerful computers, but in the near future this
impediment will be certainly avoided.
Continued research on simple, even though
unimpressive circuits, is a major factor on the
development of EHW. Analysis of such circuits is
far from impractical, and is likely to contribute to the
understanding of the properties that evolution can and
cannot exploit, and why ([4]).
In intrinsic EHW experiments, students have the
entire control over the architecture or type of basic
configurable element, and this may be an advantage
over a board equiped with a FPGA device. Evolution
may be able to exploit a different system of
interconnections or architecture. But is understandable
that more complex circuits might be evolved in a
FPGA board and these new experiments are a goal for
the future.
Future research must be done in this area. Firstly it is
important to find a better representation of the circuit
in chromosomes, because complex functions need a
great number of architecture bits, which directly
340
Q2
1 J
K Q
CLK
J Q Q1
K Q
CLK
J Q Q0
K Q
CLK
CLK
Fig. 10. An example of an evolved counter with 5 states
influences the GA search space. EHW successfully
succeeds only when fitness reaches 100% and in huge
search spaces this condition may be not always
possible. This is the main reason that for the time
being the complexity of evolved circuits is so far
small.
ACKNOWLEDGMENT
The author would like to thank the Xilinx, Inc. for
their academic donation, which consists in Xilinx
Integrated Software Environment (ISE) 6.1i software
and the Digilab XCRP circuit board provided by
Digilent, Inc.
REFERENCES
[1] B. Ali, A. E. A. Almaini, T. Kalganova, Evolutionary
Algorithms and Their Use in the Design of Sequential Logic
Circuits, Genetic Programming and Evolvable Machines, nr. 5,
2004, pp. 11-29.
[2] C. C. Coello, A. D. Christiansen, A. H. Aguirre, Use of
Evolutionary Techniques to Automate the Design of Combinational
Circuits, International Journal of Smart Engineering System
Design, nr. 4, 2000, pp. 299-314.
[3] J. R. Koza, F. H. Bennett III, D. Andre, M. A. Keane, F.
Dunlap, Automated Synthesis of Analog Electrical Circuits by
Means of Genetic Programming, IEEE Transactions on
Evolutionary Computation, vol. 1, nr. 2, 1997, pp. 109-127.
[4] P. Layzell, The Evolvable Motherboard A Test Platform for
the Research of Intrinsic Hardware Evolution, Technical Report
CSRP 479, January 1998, School of Cognitive and Computing
Sciences, University of Sussex, UK
[5] J. D. Lohn, S. P. Colombano, A Circuit Representation
Technique for Automated Circuit Design, IEEE Transactions on
Evolutionary Computation, vol. 3, nr. 3, 1999, pp. 205-219.
[6] P. Mazumder, E. M. Rudnick, Genetic Algorithms for VLSI
Design, Layout & Test Automation, Prentice Hall PTR, 1999.
[7] R. Popa, M. Iliev, V. Nicolau, Evolutionary Systems for
Electronic Engineers, Proc. of the Scientific Conf. with Int.
Participation INTER-ING 2003, 6-7 November 2003, Tg. Mure,
Romania, vol. II, pp. 187-192.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
IeL Com an integrated module for
communication in E-learning
B.Orza1, M. Gvan1, A. Vlad1, A. Olah1, A.Vlaicu1
Abstract IeL Com is a module of the
integrated environment for educational activities
management IeL. This module, developed by us,
permits students, professors, tutors and
administrators to communicate each other using a
chat-whiteboard application. The application
permits a group of persons to draw together, on a
shared table, using the client application that runs
on a computer connected in a network. Application
puts at command users a set of graphic and textual
objects that could be used to realize graphic
objects. Participants can communicate through
text box (chat) using a dedicated zone in the
window.
Users can be students and tutors involved in the
virtual university managed by IeL environment.
There is necessary the authentication of the users,
this will have access only to those chat rooms in
which he is involved. He can also have more than
one chat or whiteboard open window, one for each
user involved in the session.
IeL Com is a client-server application developed
using the C# technology. Due the utilization of the
technology specialized for the development
distributed applications (.Net Remoting), that
permits appeal of the object methods located on
server as these objects find out on local client, we
could use OOP (Object Oriented Programming)
concepts and architectures like polymorphism,
listener concept or view-control architecture.
Keywords: e-learning, remoting, whiteboard, .net
I. INTRODUCTION
The application will allow a group of people to draw
together, on a joint whiteboard, each of the members
of the group being in front of a pc connected to the
network. The application allows the user to draw
using different graphic objects and different ways of
manipulating them. The participants can communicate
by writing text on a special design part of the screen.
1
Technical University of Cluj Napoca, 26-28 G. Baritiu street, Cluj Napoca,
0264-401309, fax. 0264-591689, Bogdan.Orza@com.utcluj.ro
341
II. TEHNOLOGY
.NET Remoting is for web services, what ASP was to
CGI programming. .Net gives us a large array of tools
and facilities, e.g. allows the work with objects that
keep their state after being called. It also allows
different transfer mechanisms (HTTP and TCP),
coding mechanisms (SOAP) and security mechanisms
(IIS, SSL).
One of the major advantages of this technology is the
easy way in witch one can create distributed
applications. There are no intermediary steps in the
case of compiling the proxy/stub like in the case of
Java RMI. One doesnt have to define interfaces in
special programming languages like in the case of
CORBA and DCOM. By changing only a word in the
configuration file we have the possibility to select the
coding format used starting from the binary format to
the SOAP format.
A. EXTENDED ARCHITECTURE
The technology offers to the developers and
administrators a great variety of protocols and
formats. Every time a client application gets a
reference to an object on the server, the object will be
represented through a proxy, thus masking the
destination object. The methods of the object will be
called through the proxy. Every time a call will reach
the proxy, the call will be converted into a message,
and the message will pass through numerous layers.
The message will be passed to the serialization layer
that will convert it into a special format (SOAP,
binary). The serialized message will then reach the
transport channel, where it will be transferred to the
server through a protocol like HTTP or TCP. On the
server side, the message cross inversely the formatting
layer, the serialized message being brought to the
original form and then sent to the dispatcher. Finally
the dispatcher calls the method of the object and sends
back the answer through the same layers.
By changing only a few parameters in the
configuration file, we can change through different
types of layer implementations without writing any mechanism and reference counter. In this case the
code writing. This way an application that uses TCP server receives the messages from the client at
can be very easily modified so that it will use HTTP predefined intervals of time. When it stops receiving
as a transport channel, thus having a better scalability. messages, the server will release the resources.

In the Internet era, we still dont know too much of

the clients at the other end of the line, we cannot relay
Proxy
Object on the possibility to create a direct TCP connection
Dispatcher
on the between the clients and server. The user may be
server behind a firewall which allows only HTTP traffic to
pass through. The same router can block pings sent by
Formator Formator the server to the user. Taking into account all this, the
.Net Remoting object life span management can be
customized for every application. First to an object
Transp. will be assigned a certain life span, and at every client
Transp.
channel call that life span will be increased. Also a so called
channel
sponsor register to the object on the server may exist.
The sponsor is contacted before the life span expires,
Fig. 1. Simplified architecture of .NET and if exists the object life span will be increased.
Remoting

B. INTERFACE DEFINITION III. APPLICATION CLASSES

Many distributed systems like DCE/RPC, RMI and
J2EE, need to manually create the so called
proxy/stub objects. The proxy encapsulates the
connection to the remote object and sends calls to the
object on the server. In many of those systems
(CORBA, DCE/RPC and DCOM) the source code
that generates these objects must be written in IDL
(Interface Definition Language) and precompiled in
order to generate the header files for some
programming languages.
In contrast with this traditional approach, .Net
Remoting uses a generic proxy for all this kind of
objects. This is possible because .Net was conceived
from the beginning as a distributed applications
platform, this facilities being added lately to the other
technologies.
C. DATA SERIALIZATION
Iel Com is a client server application. We will further
describe the server side architecture and the
architecture of the client-server/server-client
communication module.
Because of the use of a specialized technology for
calling the remote objects (.Net Remoting), we wore
able to use OOP specific concepts and architectures
like polymorphism, listeners, or model view controller
architecture.
The next figure shows the simplified architecture of
the classes used for this application as the way they
communicate with the IeL platform.
IeL Com
Server
Client
ServerModelListner
C o n n e c tio n C o n L o g in

All the frameworks used for distributed applications

C o n n e c tio n
W C lie n tM o d e l
U s e r in te r f a c e

support the automatic coding of objects in any of the

D a ta P r o v id e r
W S erverM od el
i

following formats binary, XML or SOAP. The .Net Remoting

problem arises when we want to transfer a copy from HTTP
the server to the client, but COM+ doesnt offer this
facility as Java RMI and EJB do. In this case we use
L o g in
O g in

ActiveX objects for the transfer, but their use means

sending a big amount of data through the network.
ClientModelListener
In .Net it is sufficient to mark the object with the
Serializable attribute or to implement the Iserializable
interface and the platform will take care of the rest. IEL HTTP Request IeL
We also may transfer data via XML. IEL Client ServerWeb Data Provider Data Base
WEB Browser (JSP, servlet)
HTTP Response
D. OBJECT LIFE SPAN MANAGEMENT
There are three ways to control the life span of objects
in distributed applications. The first one consists of a
Fig. 2. Simplified architecture of the
connection (e.g. a TCP connection between the client
application
and the server). When this connection is closed, the
object/objects on the server will be destroyed. DCOM
uses another method; it combines the pinging

342
Classes WClientModel and WServerModel
correspond to the models from Model View
Controller, WConnection to the controller,
WhiteboardGui and ChatGui to the interface.
IV. COMMUNICATION MECHANISM BETWEEN
THE CLIENT AND THE SERVER
A. COMMUNICATION PROTOCOL
In the first phase the client sends to the server an
authentication request. The server takes the request
and interrogates the database. If the name and
password are correct, the server will return to the
client a connection object that contains some
information for the user (the user type, the name, the
database id etc.). If the authentication is not made the
server will return NULL, this is interpreted by the
client interface by displaying an error message. If the
authentication is made the client can communicate
with the server by sending and receiving WObject
objects or derivate objects form WObject. The
communication is made between users subscribed to a
certain course, they send objects to the server, and the
server sends those objects to the rest of the users on
the same course. At the server side and client side the
objects are stockated, thus allowing operations like
save and undo, if this will be done only at the server
side the traffic will be much higher if one would need
to save or undo.
For instance if we want to obtain information
regarding the number of students on every course, the
number of online users at a certain moment of time,
the courses that are running now, the client calls the
getUserInfs (WIntMess mess) method of the
WConnection object. By doing this the user does not
interrogates directly the database, but communicates
with it through objects on the server, thus increasing
the application level of security.
We use WIntMess objects to manage the application,
although the objects are derived from WObject (to
assure transparency at the transport layer) they will be
not stored in the lists of the corresponding course on
the server and the client, but they will be used only to
transmit data regarding different events that are
occurring, like: a user enters or exits, someone logs on
another machine, someone makes an undo, etc.). In all
this cases the client or the server will create such
objects, set their desired message attribute and they
will send them in the network to let know the users of
the events described earlier.
At the client side, the objects received from the server
will be sent to the graphic interfaces of the chat and
whiteboard in order to be displayed. For this we have
used the listener pattern, which means that we add to
a list all the WClientModeListner objects
corresponding to the graphical interface and then we
run through the list every time we receive a new
343
object that will be finally sent to the graphical
interface.
B. CLASSES AND OBJECTS USED FOR
COMMUNICATION
The communication between the client and the server
is made using the WObject objects. From this generic
type we have derived all the other objects which are
used for drawing or text display.
WObject
WLine WEllipse WIntMess .....
Fig. 3. Inheritance relationship between the
objects used for drawing
So at the client we will display drawings in graphical
format not as bitmaps. The advantage of this
representation is that we keep the semantic content of
the drawings, and thus one can very easily modify or
change the drawings. The use of this solution is based
on the concept of polymorphism, so in the network we
transmit only on object type, WObject, which can be
instances of other objects derived from WObject
(WEllipse, WLine, WIntMess, etc.) that will be
recognized and treated as needed at the moment they
reach the client graphical interface.
To transfer objects through the network we can use
the following: value transfer or reference transfer. The
value object transfer supposes the serialization of the
objects, including also the referential objects from the
class, in a persistent form from which they will be
reconstructed at reception. An object can be serialized
if it's marked with the [Serializable] attribute or if it
implements the ISerializable interface. After the
serialization we will have an XML document that will
be sent to the server which will interpret and remake
the original object.
It is important to outline that transfer by value does
not imply the existence of remote objects. All the
object methods will be locally executed in the same
context as calling one. This means that the compiled
classes need to be available also at the client side.
Although objects, that are derived from the
MarshalByRefObjects, will not imply that.
When an object that needs to be transferred by value
has a reference to another object, this last one has to
be derivate from MarchalByRefObjects or need to be
marked with the [Serializable] attribute.
The other types of objects are the ones that run on the
server and allow the client to call their methods. It is
mandatory that these objects inherit the
MarshalByRefObjects class. Instead of transferring a
value that points to such an object, in the network it
will be transferred only one type of objects: ObjRef,
objects contain the name of the server/ip and an
identifier, indicating uniquely an object on the server.
In the case of IelCom we use both transfer types. The
graphical objects and text are serialized in the XML
format and they are sent from the client to the server
where they are stored on the corresponding server list
and then depending on the destination they have they
are sent to the appropriate client. Every object has two
addresses. The first one is the user name of the user
(this name is unique in the data base) and the second
one is a combination between identifiers representing
the address of the client and the address of the
graphical interface to which the object is sent. All the
other identifiers are attributes of WObject class, in
such a way that all the derived classes will inherit
them. WLogin Class makes the authentication and
creates the connection object (WConnection) for each
client. This class inherits the MarshalByRefObjects,
so the transfer is made through the interface. Such a
method is WConnection doLogin (String user, String
pass, String type) which checks if the user is in the
database and if the answer is affirmative it will create
the connection object.
Client 1
Whiteboard GUI
doLogin() ChatGUI
Server
whiteboard
sendData() Client 2
getData() Whiteboard GUI
Login
ChatGUI
Connection 1
doLogin()
Connection n sendData() Client n
getData() Whiteboard GUI
ChatGUI
Fig. 4. The logging mechanism
So all the clients will communicate with the server
through their communication object, thus the
application is more secure because the client does not
have direct access to any data on the server, this
makes possible the implementation of a more
complex security system that doesnt exist at this at
this point in our application. The connections are
object uniquely identified by two attributes: the client
user name and a number generated by the server.
The connections are instances of WConnection class
having two important methods: sendData (WObject
obj) that allows transfer of objects from the client to
the server, and addNewObj (WObject obj) through
which a client receives objects from the server.
C. CLIENT AND SERVER SIDE OBJECT
STORING
344
The object storing is made on the server side but also
on the client side. We chose this method because we
wanted to keep a low traffic when we want to save the
drawings. The storing is made in lists of ArrayList
type, where we can store WObject objects and
WObject objects. The user subscribed at the virtual
university can open a chat and/or a white board
session for every course that is running, and he can
communicate with any other user through a private
chat.
For each of these communication methods there is a
list where the text and objects of the drawings are
stored. The server keeps the corresponding models of
every opened course with active users; the client
keeps only the ones where the user is active at that
moment of time.
To distribute the objects from the server to the
destination clients it was implemented an algorithm
based in the listener concept. This concept presume
that once a connection is made, it is added on a list on
the server (WServerModel) and at the moment an
object appears on the server, the list will be read and it
will be sent to the appropriate connections.
From here through a thread the objects will be taken
by an object corresponding to the WClientModel
client model and the temporary list will be emptied.
The following code sample is an example of the way
objects are distributed on connections through the
function wakeUpListners (WObject obj).
The listener concept is also used in the case of
graphical interfaces (whiteboard and chat) which are
registered as listeners to WClientModel. So the
objects that come will be redirected depending on the
destination application (the text for chat and the
graphical objects for the whiteboard).
The WClientModel class as the WServerModel class
implements the Singleton pattern, in such a way that
there will be only one instance of every class on the
whole application. To obtain the private class
constructor is declared, such that we cannot instantiate
the class outside. First is declared a static attribute of
the class type, the value of this attribute will be set
initially to null. It will be declared also a static
method that will initiate the attribute just once. The
uniqueness of the object corresponding to the models
is very important because it is necessary to be able to
obtain references to them from different points of the
application, more then this they offer flexibility
because we can add new modules without doing
important changes of the application.
So any graphical interface, or any other module that
needs the user identification data, can access them
through the reference to the WClientModel provided
by the method getInstance().
For a better management of the application we have
defined the WIntMess class. Through objects of that
class the messages are sent and received from the
server. This objects are not stored on the server, they
are used just for the management of the application.
There are two attributes of the class: query and
response. When the client wants to obtain some
information from the server, for instance the number
of student in a course, he will not point directly the
data base because that can cause security problems.
The solution is the creation of a WIntMess object with
the query attribute set with the proper message, which
is then sent to the server. The server will take the
object and analyze the request and after that it will set
the response attribute of the object with the object that
holds the information desired by the client. The object
will be then sent on the connection that came from.
This type of object is also use to signal if a client
connects on another machine, if he left the
application, or if he wants to create a new drawing.
An interesting advantage of the application is that of
undo. Although for stand-alone application this is
quite a simple thing to implement, in the case of
distributed applications this arise some problems. The
first problem appears when we want to establish the
way we want to make the undo. There are at least to
possibilities: the first one is that the user is able to
make undo only to the objects that he created, but this
contrast the principle of shared whiteboard, because
the users must be able to modify also the work of
others. The second possibility is that the user can
make undo on all the objects on the drawing, this is
also what we choose for our implementation. So it
was created a WIntMess object that sends the undo
message to the server every time a user hits the Undo
button from the graphical interface. This message
reach the server, the server will update the model
using the updateModels (WIntMess mess) method,
which will eliminate the last added object. The
message is also sent on the connections corresponding
to the online users through the wakeUpListeners
(WObject obj).
V. CONCLUSIONS
Taking into account the continuous growing of the use
of computers in the academic environment, such an
application (chat and whiteboard) is a very useful tool
that can be successfully used for distance education.
At the moment the application doesnt need a large
bandwidth for transferring the information from the
client to the server, so it can be used even with poor
internet connections (e.g. dial-up).
IeL Com is part of the integrated environment for
distance education IeL, being a synchronous
communication solution for students and teachers,
administrators and tutors, creating a virtual space
where the teachers or the tutors can teach their
courses and the students can ask questions and receive
their answers in real time.
345
The application was developed using .NetRemoting
technology, because it is a good compromise between
the bandwidth needed to communicate and the ease of
implementation. Although it requires a larger
bandwidth than the use of sockets, .NetRemoting
provides the programmer an advanced
implementation environment, which abstracts the
transport layer from the OSI model, allowing the
transmission of objects through the network and the
calling of remote methods. In the case of
.NetRemoting as in the case of Java RMI, due to
actual security demands, the application configuring
process is hard enough, because often there are added
new security levels from one version of the frame
work to the other, thus the need of adding new
information in the configuration files.
The use of these technologies allows an easier
implementation of the object oriented programming
concepts (polymorphism, inheritance, etc.), it adds
scalability plus to applications so that one can add
more easily new modules and facilities. This is also
the case of IeL Com, the developer can add new
graphical objects, deriving the appropriate classes
belonging to WObject, without worrying about the
transmission through the network.
Bibliography
[1] I. Rammer, Advanced .Net Remoting (C# edition), APress
2002.
[2] A. Turtschi, C# .Net, Syngress Publishing, Inc.
[3] A. Vlaicu, V. Dobrot, S. Iacob, Tehnologii multimedia,
sisteme, reele i aplicaii , UTCN.
[4] B. Orza, M. Givan, S. Cristea, A. Vlaicu, "IEL 2 - an
Integrated Solution for Management, Evaluation and
Communication in E-Learning" , International Conference
Advanced tools for E-learning in the Environmental Education, 12-
13 February, Napoli, Italy.
[5] B. Orza, M. Givan, S. Cristea, A. Vlaicu, "Integrated solution
for management, evaluation and communication in distance
education systems ", Optimization Of Electrical And Electronic
Equipment Optim 04, May 20-22, 2004, Brasov, Romania
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Low-cost electronic board improves electronics laboratory

efficiency
Sabin Ionel, Marllene Dnei1

Abstract A low-cost electronic board (Laborplatine)
connected to a PC can be used to simulate the
facilities of a digital oscilloscope, function generator,
voltmeter etc. Using this board in order to achieve
real signals from electronic circuits, one can organize
frontal and/or guided experiments which improve
efficiency of electronics laboratory This can be
important in the context of a reduced teaching
program for disciplines like Electric Circuits or
Fundamentals of Electronics.
Keywords: electronic board, electronics laboratory,
frontal experiments
I. INTRODUCTION
The basic idea in developing the electronic board
presented in this paper was to offer a cheap
solution, so that students interested in electric and
electronic circuits can do their own experiments at
home, as individual study [1]. However, a future
reduced program according to the Bologna process
determines an imperative efficiency improvement
of all teaching activities, including practical
training. In this paper the utilization of the
electronic board in teaching electric and electronic
circuits through frontal experiments is suggested.
II. THE ELECTRONIC BOARD
The electronic board presented in fig.1 was
developed at the University of Applied Sciences
Karlsruhe, Germany [1], [2]. The size of the board is
160mm x 100mm and it can be connected for data
transmission to the serial port of the PC. The
minimal requirements for the computer are: CPU
frequency greater than 200MHz, free memory more
than 20MB, 17 CRT monitor or 15 Notebook with
1024 x 768 pixels, Windows 95, 98, XP or NT.

Fig. 1 The electronic board (Laborplatine)

1
Facultatea de Electronic i Telecomunicaii, Departamentul Electronic Aplicat,
Bd. V. Prvan Nr. 2, 300223 Timioara, sabin.ionel@etc.utt.ro; marlene.daneti@etc.utt.ro

346
 Fig. 2 A representation of the electronic board on the PC monitor
The reduced price (100) of the electronic board
implies certain constraints. For example, the sampling
frequency of the two input signals is only 2 MHz.
This is, however, more than enough for experiments
on basic electric and electronic circuits. The supply
voltages for the electronic board are +12V (150mA)
and -12V (60mA). The software for the PC (written in
the HP-VEE language) as well as the latest update can
be downloaded from internet [2].
III. THE IMPLEMENTED DEVICES
The electronic board itself can deliver a dual supply
voltage for the circuit under experimentation: (0 to 12)
V and (-0 to -12) V.
A DC-voltmeter can measure constant voltages
in ranges from 2V to 20V. A DC-ampmeter is also
implemented for constant currents in ranges from
60A to 200mA.
The implemented function generator (quartz
controlled PLL synthesizer) delivers, via BNC,
usual signals (sinus, rectangle, triangle and positive
or negative sawtooth) as well as customized
waveforms (programmable function).
Certainly, the most important instrument
implemented on the electronic board is the digital
oscilloscope. The impedance of the two Y1, Y2 inputs
(via BNC) is 1M||13pF and the bandwidth for each
channel is 2MHz (-3dB). Both inputs have DC, GND
and AC (time constant 0.1 sec) facilities. One can choose
the sensitivity in the following steps: 10, 20, 50, 100,
200, 500mV/Unit and 1, 2, 5V/Unit. Important functions
like external triggering (BNC output), FFT and signal
averaging are also provided. Fig. 2 shows a
representation of the electronic board on the PC monitor.
IV. FRONTAL AND GUIDED EXPERIMENTS
Frontal experiments can be easily carried on if the
computers in laboratory are connected in a local
347
area network. Combining problem solving (specific
seminary activity) with PSpice simulation and
frontal experiments using the electronic board, one
can assure better understanding of theoretic and
practical issues regarding electric and electronic
circuits. This approach can accelerate the learning
and save time. Groups of two or three students
guided by the laboratory assistant in their work
with the electronic board will also realize the
importance of cooperation and teamwork. The
feedback from the students is also important, since
they can raise interesting problems. Our experience
shows that students are better motivated to do
simulations and experiments using the PC as main
tool, than working with several different measuring
instruments. Certainly, the frontal and guided
experiments under the control of the laboratory
assistant must be continued with individual
assessments and hands-on exercises developed by
the students in their own free time.
V. CONCLUSIONS
The presented electronic board is a cheap and
suitable solution for experiment-based teaching and
learning of electric and electronic circuits. It can be
utilized not only for individual study, but also in
organized laboratory and classroom activities.
Especially, frontal and guided experiments based
on this electronic board can improve teaching
efficiency, in comparison with classical laboratory
training utilizing expensive measuring instruments.
REFERENCE
[1] R. Koblitz, Neue Laborplatine, fh-magazin WS2003/2004,
24 Jahrgang, Nr.48, pp.72.
[2] http://www: /fbeit.fh-karlsruhe.de/laborplatine/
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

A web-based teaching tool for laboratory classes

Dan Stoiciu, Ciprian Dughir1
Abstract The paper presents a solution that the
authors have used for implementing a virtual laboratory
to allow students to conduct an experiment from a
remote PC, as part of an e-learning process.
Keywords: web-based teaching tool, e-learning, virtual
laboratory.
I. INTRODUCTION
With the rapid growth of Internet and number of PC
users, web-based tools for teaching purposes become
more and more present in teaching activities [1], [2],
[3]. The paper presents an application that can be run
from a remote PC by a student such as to conduct an
experiment from a remote location, without the need
of a physical presence in the classroom. The
application is intended to allow each student of a
group of students to individually conduct the
experiment, as part of the implementation of an
e-learning process.
II. DESCRIPTION OF THE APPLICATION
One important part of an e-learning system is the one
that plans the execution of the experiment by the
student(s). A reliable system is therefore needed. This
system should also be able to monitor the execution
such as no interference or overlapping between
several users occur. The application that has been
developed consists of two parts: the practical
experiment and the planning and handling part. The
experiment chosen to be conducted is the automated
calibration of a digital voltmeter [4]. The system that
has been developed for this goal consists of a PC, a
programmable power supply, the DVM to be
calibrated and a high accuracy voltmeter (fig. 1).
The power supply and the high accuracy voltmeter are
connected to the PC via the IEEE 488 parallel
interface. For the DVM to be calibrated a 41/2 digit
Voltcraft 4650 CR with built-in RS 232 serial
interface was considered. An application (or virtual
instrument - VI) has been developed in order to
control all the devices and to accomplish the required
tasks. The inputs of this VI are: the voltage range of
the DVM to be calibrated, its number of digits, the
formula given in its specification for calculating the
maximum permissible error, the number of points
(voltage values) and the number of times the
measurements should be repeated at each point in
order to obtain an averaged result. The tasks to be
accomplished by the system are described in the
following:
Step 1 The VI calculates the first voltage value and
sends it to the programmable power supply.
Step 2 After a convenient delay the readings of the
two voltmeters are sent to the PC. Step 2 is repeated
the required number of times, and the readings are
averaged. The averaged results are output in a table,
and the error is calculated and compared to the
maximum permissible error. Steps 1 and 2 are
repeated for the required number of points. In the end,
the operator sees a table with the averaged readings of
the two voltmeters, the actual error and the maximum
permissible error. Additionally, an array of LEDs
indicates whether and at which point the error exceeds
the limit. The front panel of the VI is shown in fig. 2.

RS232
DVM DVM (standard)
PC
Power supply

IEEE 488

Fig.1. System schematics.

Fig. 2. Front panel of the VI.
1
Facultatea de Electronic i Telecomunicaii, Departamentul de Msurri i Electronic Optic
Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail dan.stoiciu@etc.utt.ro, ciprian.dughir@etc.utt.ro

348
The planning and handling part of the application has the professor, by e-mail, a complete report concerning
been developed in HTML and PHP, and the data base the experiment carried out.
for the data about the users has been implemented in
MySQL. When the user accesses the e-learning site, a
window appears and asks the user for identification
(fig. 3).

Fig. 5. Time frame reservation page.

Fig. 3. Identification page.

If the user is not yet in the data base he or she can

create a personal account. The following data must be
filled in: the name of the user, a name for the new
account, and a password that has to be reentered (fig.
4).

Fig. 6. Page for connecting to the virtual laboratory.

III. CONCLUSION

The paper illustrates a way in which the Internet is

Fig. 4. Page for creating a new account.
The program checks the data and provides a dialog
with the user (user name too short, improper password
or non identical passwords). Once the account
created, the user is redirected to the identification
page. After being identified, the user is shown a page
where he or she can choose a time frame for running
the experiment (fig. 5). The time required for the
experiment does not normally exceed 20 minutes.
Therefore, in the application, 30-minute time slots
have been provided.
In the selected time frame the user can access the
e-learning site where the page in fig. 6 will be
displayed. The user can choose to be connected to the
virtual laboratory, where the front panel of the VI
shown in fig. 2 will be displayed.
The user can remotely launch the application and after
running it the front panel of the VI on the remote PC
changes its appearance and provides the user the
results of the experiment. The user can easily print a
report with these results. He or she can also submit to
used to implement an e-learning procedure. It offers
the advantage of a relative flexibility by offering the
students a wide choice of time frames for conducting
the experiment. The drawback, as for all virtual reality
issues, is the loss of practical feeling (the students do
not have a touch and feel of the real instruments and
of the experimental setup).
REFERENCES
[1] R. C. Clark, R. E. Mayer, E-Learning and the Science of
Instruction : Proven Guidelines for Consumers and Designers of
Multimedia Learning, Pfeiffer, 2002.
[2] M. J. Rosenberg, E-Learning: Strategies for Delivering
Knowledge in the Digital Age, McGraw-Hill, 2000
[3] W. Horton, Katherine Horton, E-learning Tools and
Technologies : A consumer's guide for trainers, teachers,
educators, and instructional designers, John Wiley & Sons, 2003
[4] D. Stoiciu, M. Lascu, PC-based system for automated
calibration of a digital voltmeter, Symposium on Electronics and
telecommunications, Timisoara, 2004

349
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Monitoring Network Activities in Web Based Training

Courses
Camelia Popescu1, Adrian Rp1, Paul Svasta1

Abstract - Because it is very important to have a proper understanding of the educational process of each stu-
image of all activities dealing with the field of electronic dent. [2]
packaging, the aim of the current paper is to present a As part of our efforts to increase the quality of educa-
model of how to monitor educational and research activi- tion in Electronic Packaging, our team has developed
ties in an Integrated CAD Environment. The system
described in this paper is a client/server application, has
a flexible software platform which consists of two
an easy to use interface, and a secure database. Such a operational programs: a server program and a client
system is a necessity in an academic environment be- program and the client program has three modules: for
cause is a tool for a better evaluation, for making statis- monitoring laboratory activities, applications opened
tics with the most accessed sites, for avoiding troubles on workstations and web traffic.
made by students who speculate systems or networks
weak points. II. GENERAL ASPECTS OF THE SOFTWARE
PLATFORM
Keywords: monitoring application, client/server arhitec-
ture
Why would be necessary such a monitoring instru-
ment? A few reasons could be: the pursuit to under-
I. INTRODUCTION
stand the assimilated information, the improvement of
student evaluation system, research activity not to be
The problems of electronic packaging may cover a
disturbed by other unwanted activities (file deletion,
large area and represents "The engineering discipline
file coping, database altering, etc.).
that combines the engineering and manufacturing
technologies required to convert an electrical circuit
The monitoring is not a restrictive instrument for the
into a manufactured assembly. These include at least
activity or the research direction chosen by students; it
electrical, mechanical and material design and many
is an early correction and rectifying instrument of
functions such as engineering, manufacturing and
eventual errors in the training process.
quality control." [1]
The software application MA-ICADE (Monitoring
Shaping the students for designing and manufacturing
Activities Integrated CAD Environment) is a cli-
of electronic packaging involves an interdisciplinary
ent/server application which can be accessed and used
and multidisciplinary approach. In this context, there
from school laboratories. A client is a computer that
are at least two important aspects: one of them is the
queries, through a message, another computer, named
teaching and the other one is the evaluation of the ac-
server, for information or services. The server,
quired knowledge. These two activities concern all the
through another message, delivers the information or
students involved in educational activities, especially
services to the client, the whole operation being trans-
in CAD-CAM-CAE because they define the high per-
parent to the user.
formance in human resources. All this implies persons
responsible for education with a lot of qualities as
Programming languages as Delphi, PHP (recursive
high qualification, creativity, patience and talent. Be-
acronym for "PHP: Hypertext Preprocessor") and
yond this, sometimes the educational process involves
HTML and also database free tools as MySQL and
a great amount of work routine, which could lead to
Apache Server were used in order to develop MA-
professional one. Many of these problems could be
ICADE Software. In order to monitor the web traffic
avoided by using a Computer Added Instruction
was used SQUINT a free application from the Inter-
(CAI). Through such a software application the com-
net because it offers daily, weekly and monthly re-
puter can take over a large part of
ports of the Internet activity in the network.
the routine and consequently teachers could afford to
spend more time for creativity and for a better
Fig.1 presents a logical diagram of the MA-ICADE
SOFTWARE APPLICATION that emphasizes the
1
University Politehnica of Bucharest.

350
main parts of the application and the relationships workstation is on, or if the server program is on. If
among module. not, an error message will be post on the professor
display.

This program has been done exclusively in Delphi.

IV.CLIENT PROGRAM

The program opens with a splash form which presents

the name of the application, the version, the owner and
the logo.

Fig.1 Logical diagram of the MA-ICADE SOFTWARE

APPLICATION

The server program runs on the students workstations

and the client program runs on the professors work-
stations.
Fig.3 Splash Form

Each module from the client program has his benefits.

Certain problems, created by ingenious students which
speculate errors, can be avoided.
The MLA (Monitoring Laboratory Activities) module
is useful for a better evaluation of the students. Long
term evaluation applies, because the punctual one does
not reflect the acquired level of knowledge.
The AOW (Applications Opened on Workstations)
module permits to see what applications are used at a
specific moment of time.
The access to this program is granted after a success-
ful login. Different operations (switching between
modules, saving information on the database etc) can
be performed by authorized persons.
Besides the friendly interface, the client program has
also a help module who guides the user.

The WT (Web Traffic) module permit restricted ac-

cess to some web pages and also to make statistics
with the most accessed sites. It can be done a cache
with the most accessed web pages in order to manage Fig.4 The help system of the application
rapidly the informations.

III. SERVER PROGRAM To better understand the way in which this program
works, we have to look at the functional diagram of
For the tests, the server program was installed on 16 the client program.
workstations. It starts with the operating system and
runs in the background. Students can see an icon in
the task-bar and they know their activity is monitored.

Fig.2 The icon of server program

The program receives requests from the client through

HTTP protocol and sends screen shots and lists with
all applications there are opened on the workstation.
Before the request, the client program verifies if the

351
 Fig.5 Functional diagram of the client program

Another important feature of the client program is the

possibility to have database management. This man-
agement consists of several functions, which can be
used only by authorized persons as administrator,
teachers and lecturers. Database management is about
searching, adding, deleting and modifying any of the
stored items. Fig.6 presents how a teacher is added,
modified or deleted from the database.

Fig.8 The applications database

Fig.6 Administrators page

Fig.7 Teachers administrative page Fig.9 A screen shot from a workstation

This program had been done using Delphi, PHP and
HTML: contains Delphi written code for laboratory
activity and local stations supervision, contains PHP
written code for database administration, contains
HTML written code for the applications help file.
A. MLA MODULE
The students knowledge evaluation is one of the most
important activities of the didactic process. The MLA
(Monitoring Laboratory Activities) module is particu-
larly useful for teachers and lecturers during this activ-
ity, as it helps, prevent time losses, subjectivism dan-
ger and occurrence of routine and redundancy, during
the students' knowledge evaluation.
B. AOW MODULE
Another functionality of the client program is referring
to the applications opened on each workstation. It
shows application lists opened in 16 list boxes. It has a
popup menu for saving the application list. Not Con-
nected or Error messages are printed. If students are
working with application which are not designated for
training activities they will be seen and in the future
the teacher will have the possibility to close the un-
necessary applications. The friendly interface permits
the switching between modules. Buttons from the pre-
vious windows can be accessed as well as other func-
tionalities of the application

MLA module allows the instructors to simultaneously

watch, on a single monitor, all the 16 station monitors
on which the students work. Before attempting to take
screenshots from a station, the program verifies if that
station is on, through a ICMP package (Internet Con-
trol and Monitoring Protocol). It shows error messages
when it is the case and it allows saving the captured Fig.10 A list with opened application
images. Each print screen of students activities can be
saved in a secure database and the professor can add
notes in a table, regarding this activity.

352
 C. WT MODULE
For monitoring web traffic, it was chosen the Squint
program. It presents daily, monthly and yearly reports
of Internet activity. The following information is re-
corded: the number of downloaded bytes, the time on-
line, the longest session of a user and a table with the
time of Internet access. It presents a complete image
of the accessed sites as well as links to those pages,
time and downloaded pages
The application is supported by any Windows version.
The activity can be monitored from any computer in
the network which has the client program installed.
The screen resolution is unimportant.
REFERENCES
[1] Charles A. Harper and Martin B. Miller - "Electronic Packaging
Microelectronics and Interconnection Dictionary", McGraw-
Hill, 1993, page 67
[2] Glaser R. - Programmiertes Lernen und
Unterrichstechnologie, Franz Cornelsen Verlog 1971.
[3] A. Drumea, P. Svasta, V.D. Ene, Some Aspects of Educational
Environment in Packaging, XXV International Conference IMAPS
Poland 2001, Rzeszow Polanczyk, 26-29 September 2001, pp.
31-36
[4] Paul Svasta, Virgil Golumbeanu, Ciprian Ionescu, Norocel Dra-
go Codreanu, Daniel Leonescu, Marian Vldescu, Dan Tudor Vuza
- Training Program in Electronic Passive Components Education,
The 52nd Electronic Components and Technology Conference, May
28-31, 2002, San Diego, California USA, pp. 780-786
[5] PHP Documentation, www.php.net/docs.php
[6] Mysql Documentation, www.mysql.com/documentation

Fig.11 Web pages accessed from 192.168.1.9

V. COURSE OF DEVELOPMENT

In the future modification of the monitoring system

and modification of the application as it can monitor
other network structures will be done. It will also be
implemented the possibility to send messages on sta-
tions. In order to develop a friendly application the
help system will be modified.

VI. CONCLUSIONS

The main target of the MA-ICADE SOFTWARE

APPLICATION is to support the teachers in evalua-
tion and feedback processes. A proper statistic for test
results is improving the educational activity.

As part of the actions that must be performed in the

designing and manufacturing of an electronic package,
Computer Added Design (CAD), Computer Added
Engineering (CAE) and Computer Added Manufac-
tures (CAM) are specific types of activities which
need this kind of software. With MA-ICADE
SOFTWARE APPLICATION an important amount of
time can be saved.

Regarding the complexity of MA-ICADE, a very

friendly interface is proved, to make it as simple as
possible to use. A very important aspect of the MA-
ICADE SOFTWARE APPLICATION is that it allows
the possibility of feedback, which facilitates the tailor-
ing of the educational process, and especially of the
evaluation activity, on the students particular needs,
leading to significantly improved results for both stu-
dents and didactic personnel.

353
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Mobile Accident Warning System

-The LoRD-
Dan-Marius Dobrea1, Nicolae Cleju2, Andrei-Tudor Sechelea3, Alexandru Banar4
Abstract - The Location Retrieval Device (LoRD) is a
powerful, versatile and accurate warning system.
Designed to be integrated on a car, it instantly reacts in
case of an accident, providing the rescue units with
extremely useful information about the coordinates,
speed of the car, the structural integrity and
temperature of the cockpit. The project is based on two-
way SMS messaging between the main components: the
mobile LoRD modules, that send the information using a
mobile phone, and the base station that receives these
messages, and also manages and configures the modules.
Keywords: emergency, GSM, GPS, SMS, location
retrival
To instantly notify a central base station when an
accident took place,
To provide the coordinates of the location, so the
emergency unit can reach it in minimum time,
To gather other crucial information from the site
of the accident: the temperature (in order to
detect a possible fire) and the structural integrity
of the car,
To continue to periodically send as much data as
possible after the accident took place, providing a
real-time evolution of the situation.

I. INTRODUCTION II. SYSTEM OVERVIEW

One of the major problems with respect to the design II.1. The way it works
and management of any emergency system is the
ability to quickly handle any distress call. The faster The overall system is composed of mobile LoRD
the better is the main principle that governs the (Location Retrieval Device) units located in cars and
functioning of any emergency service. Only a quick a central base station that receives all the emergency
response to fires, car crashes or accidents can notifications. They communicate by means of SMS
diminish the risks of human injuries or fatalities. (Short Message Service) messages. The functioning
According to the US Bureau of Transportation of the system is presented schematically in Fig.1.
Statistics, car crashes are an important issue of public
safety around the world, with a number of 42,815
fatalities and 2,925,758 injured people in 2002, in the
US [1]. The victims chances of survival can be
greatly improved with an accurate response, implying
that the paramedic units should be notified as soon as
possible about the precise location where an accident
took place. The lack of precision means wasting
extremely precious time.

We set to develop a system that reduces and almost

Fig. 1. The block scheme (simplified) of the entire system
completely excludes wasted time in a distress call. It
helps rescue teams locate the victims easier and faster.
Our systems main goals are:

1
Technical University "Gh. Asachi", Faculty of Electronics and Telecommunications, Iasi, Romania, e-mail: mdobrea@etc.tuiasi.ro
2
Student, Technical University "Gh. Asachi", Faculty of Electronics and Telecommunications, Iasi, Romania, e-mail: nikcleju@gmail.com
3
Student, Technical University "Gh. Asachi", Faculty of Electronics and Telecommunications, Iasi, Romania, e-mail: astek@personal.ro
4
Student, Technical University "Gh. Asachi", Faculty of Electronics and Telecommunications, Iasi, Romania, e-mail: alex584@apropo.ro

354
 Format of the Configuration Message
Security Code Time Interval (x 10 seconds) Temperature Limit
Format of the Emergency Message
Security code Car structural integrity Longitude
Fig. 2. Format of the most important messages
Before using any LoRD module, it must be remotely
configured by the central base station. This is done by
sending a message to it that contains the necessary
information: the phone number to which the LoRD
will reply, the time interval between two consecutive
messages and the temperature limit which will trigger
the thermal alarm. This process is allowed only if one
specific jumper is set. After a successful
configuration, unsetting this jumper causes the LoRD
to reject any other configuration attempt.
In normal mode, the LoRD module continuously
monitors for a possible abnormal situation. The events
that trigger the alarm are the car security systems
(opening of the airbags, car alarm, or panic button) or
the detection of a fire, when the temperature exceeds
the previously programmed limit. Therefore the
LoRD gathers information about the car status and
location and uses the attached mobile phone to send it
through a SMS to the base station. The message
contains the coordinates of the car, obtained with a
GPS Receiver, the temperature, in order to detect a
possible fire, and information about the extent of the
cockpit damage, Fig.2. If it is not damaged, the LoRD
continues to send messages periodically, allowing the
base station to monitor the evolution of the situation.
There is also a remote control mode, as an alternative
function of the LoRD. The base station may send a
request to receive alert messages, although no
accident was reported. This can be useful, for
example, in tracking down the LoRD in scenarios like
mountain rescues, fire-fighting or retrieving stolen
vehicles. The base station sends a START message,
which will enable the LoRD to send back notifying
messages. The analogue command STOP will
disable any message sending.
The central station runs the software that manages all
the incoming messages. It is developed in the
LabWindowsTM /CVI version 5.5 integrated
environment and it uses a dedicated database server to
securely store the information. Upon receiving an
emergency message, the software reads it from the
attached mobile phone and presents it to the human
operator.
# Latitude # Speed # Temperature # Error State
II.2. Security and reliability features
As the system handles critical data, the performance
requirements are very high, and we had to consider
reliability and security issues from the early stages of
the project.
The module performs numerous self-tests to ensure
the reliability of the information supplied. Any error
detected is reported in the SMS sent to the base. Any
phone error is visually signaled to the driver.
To ensure a prompt reaction in case of an accident the
alarm signals and the entire serial communication are
processed through the microcontrollers system of
interrupts. The alarm signals have the highest priority,
so as soon as a collision occurs the LoRD begins to
send the emergency messages.
The SMS messages by which the LoRD modules and
the base station communicate with each other contain
a certain security code, computed from the senders
own number. By checking this security code against
the senders phone number, as it is reported by the
phone network, the receiving device is able to detect
and reject fake messages.
The whole system is designed to be infrastructure-
independent in order to have the highest level of
generality and to be easy extendable. The
communication process is independent of the type of
phone network, because all of the GSM, CDMA and
TDMA technologies support SMS technology. The
LoRD module, the mobile phone, and the GPS
Receiver have their own power supplies, which are
switched on only in case of a crash; the rest of the
time they are powered by the car battery.
Using the dedicated MySQL database server adds to
the overall independence and security. The server can
reside on a remote host (MS Windows, Unix etc.), and
communicate through SSL secure TCP/IP protocols.
We decided upon a database server because its
security, platform-independence and the capacity to
handle and store data with minimal risks of failure
make it an excellent choice.
355
 III. HARDWARE IMPLEMENTATION OF THE maximum 10 s with the frequency given by the
LORD MODULE formula:

The LoRD was designed around an AtmelTM n

microcontroller of the 8051 family. We decided upon
this family as it is commonly used and has proven
very reliable since its development in the 80s.
There are three main interconnecting subunits: the
AT89C4051 microcontroller, the BU4066B Quad
Bilateral Switch and the MAX232N dual RS-232
Transmitter/Receiver.
The AT89C4051 microcontroller is a low voltage,
high performance, 8-bit CMOS microcomputer with
4K bytes of Flash programmable and erasable read-
only memory PEROM. Because of its useful built-in
features, high flexibility and extremely high
reliability, it was the best cost-effective solution we
could find for our LoRD module, fully meeting our
requirements.
The MAX232N features two pairs of drivers and
receivers for communication through the RS-232
serial interface. It transmits and receives data from
two different devices that communicate through the
serial interface: the mobile phone and the GPS
receiver.
f = az (1)
60
where a is the number of cylinders, z is a constant
with the value of 0,5 and n is the rpm (revolutions per
minute) number, ranging, for an average car, from
900 to 7500. These spikes would generate system
activation if the RC filter wouldnt clear the line.
Solving for the worst scenario (n = 7500) we get a
minimum 4 ms period between the parasite spikes.
This value is used to calculate the required values for
the resistor and the capacitor, ensuring good filtering
without distorting the waveforms in normal
functioning.
Finally, six switches directly connected to some of the
microcontroller inputs are to be placed on the car
cockpit key components. This makes it easy to assess
the amount of damage caused by an accident because
the message sent by the LoRD also contains the status
of the switches, damaged ones appearing as closed
contacts. So by sending their state of these switches,
the rescue teams immediately know the full extent of
the accident and are able to swiftly act accordingly.

The BU4066B is a multiplexing circuit, which

handles data from the MAX232N, deciding on which
of the two devices (phone or GPS) sends and receives
data from the microcontroller.

The LoRD also features a LM7805 voltage converter

circuit. The LM7805 is powered either by a 12V-car
battery or by a 9V-accumulator, providing that the car
battery is damaged in the accident. The voltage output
is regulated at the +5V level required by the digital
circuits on the LoRD module.

A 2-wire I2C bus interface is integrated in the LoRD,

ensuring the communication between the
microcontroller and the DS1629 digital thermometer
[2] . The DS1629 provides 9-bit temperature readings,
indicating the temperature of the device. It works as a
temperature-to-digital converter in the range of -
550C to +1250C (-670F to 2570F). The circuit also
features a thermal alarm, which sets a particular pin to
low state, consequently activating an interrupt request
in the microcontroller, whenever the temperature of
the device exceeds the programmed over-temperature
limit stored in a special function register (TH).

An external connector is interfaced to the car security

systems (airbag opening command, car alarm and
panic button) so that the system is activated whenever
an unwanted event takes place. For this particular
connection a RC filter is necessary to eliminate the
noises that appear due to the car motor. A four-stroke
gasoline engine generates parasitic spikes of Fig. 3. Main program organization

356
 IV. THE SOFTWARE OF THE LORD MODULE
IV.1. Overview
The microcontroller software was developed in
assembly language, as it is faster than a similar
program developed in a C environment. The program
flow is represented in Fig.3.
Upon startup, the LoRD runs an initialization routine,
and then it normally remains in a loop where it
performs various self-tests and component-tests.
Whenever an external interrupt is activated, the
hardware interrupt routines are executed and they set
an acknowledgement semaphore. The software
verifies the status of these acknowledge flags and if
an alarm is indicated it runs the message sending
routine.
IV.2. The Initialization Routine
The initialization routine starts with a self-test it
erases all the locations in the internal memory and
verifies if they are valid. Then it enables internal
timers, interrupts, serial communication and initializes
external devices.
The AT89C4051 microcontroller has two timers. One
of them, TIMER 0, is used to set the time interval
between two consecutive messages, and the other one,
TIMER 1, must set the default 9600 bps baud rate
used by the microcontroller to communicate through
the serial port.
To ensure the priority of alarm operations, we used all
of the microcontrollers interrupts in the following
way:
Two external interrupts: they have the highest
priority [3], and the activation of any of them will
determine the microcontroller to send alarm
messages.
Two timer interrupts [3]: they increment some
time counters each time the interrupts are
activated;
The serial interrupt: it has the lowest priority and
it is activated each time a character must be sent
or received [3].
IV.3. Testing Routines
While it remains in a continuous loop, the
microcontroller performs periodical tests to check the
system integrity:
It checks for the presence of a mobile phone by
sending the Hayes AT command and verifying
that the answer received was the expected OK
(see section IV.5. Serial communication
routines). This test is the most important of all,
because without the phone the LoRD cannot send
any emergency message. That is why any phone
357
error is visually signaled using a LED.
It verifies the temperature sensor by writing a
certain bit pattern to a location in its memory.
Receiving acknowledge bits assures of the proper
functioning of the I2C bus, and the correct
retrieval of the pattern validates the sensor circuit
itself.
It verifies the presence of the GPS Receiver by
polling commands [4] that ensure the device is
connected and working properly, as it must have
acquired at least three satellites in order to be able
to function. In case the Receiver acquires no
signal, it returns null instead of the coordinates,
in which case an error flag is set.
This continuous testing allows the fastest response in
case of an emergency. Whenever an external interrupt
is activated, signaling an emergency, the LoRD does
not waste any more time on tests. The current state of
the device is already determined, so it can start
sending the message as soon as possible.
IV.4. Processing Hardware Interrupts
There are two causes that will determine the
microcontroller to enter the alarm state: either a signal
from the car interface (the opening of the airbags or
other alarm systems) activates External Interrupt 0, or
the temperature exceeds a critical level, activating the
External Interrupt 1.
We used the entire interrupt system in a way that
enables the LoRD to react as promptly as possible to
these situations.
We had to analyze how to obtain the fastest response
to an emergency, still doing all the necessary tests in
order not to compromise the reliability degree of the
provided information. (i.e. the LoRD should detect
and report any temperature sensor error, because the
base station should know that the temperature
readings may be incorrect). The following issue,
which cannot be avoided, is very important: if the
interrupt is requested during any serial
communication, the communication process must be
allowed to finish. Sending the emergency SMS
requires serial communication with the phone, so
simply cutting off the previous communication may
issue scrambled commands to the phone or the GPS
receiver. Our solution is presented in Fig.4.
When an interrupt takes place during a test that
involves serial communication with a device, the
interrupt routine is executed but it only sets an
acknowledge semaphore. The software is therefore
allowed do terminate the current communication
process. After every test, the program checks these
semaphores and, if it determines that an interrupt took
place, runs the routine which sends the emergency
message. This way, the maximum possible delay
between signaling an emergency and the actual start
 Fig. 4. Processing the interrupt requests

of the message sending routine is the maximum time
spent in a testing subroutine.
The other alternative means doing all the tests after
the interrupt request, to ensure the reliability of the
information. This implies that the delay until the
sending of the message would be the sum of all tests
durations. This clearly increases the response time and
hence the risks of failure during a crash.
IV.5. Serial Communication Routines
The communication with the mobile phone is based
on the assumption that the phone supports the Hayes
commands. These are ASCII commands that are
transmitted over a RS232 connection and allow using
the phones capabilities. We were interested mostly in
the ability to send and receive SMS messages, but
also to read phone numbers stored in the SIM card.
Sending and reading a SMS message also implies
converting the text to and from a special format, the
PDU (Protocol Description Unit) format, which the
phone uses.
messages received from the LoRD units. It also
allows easy configuration of the new units and
reconfiguration of the existing ones. The software
controls one mobile phone attached to the serial port
of the PC, which is used to send and receive
messages. Created in the LabWindowsTM /CVI
version 5.5 integrated environment, the software
displays information about received messages, while
in the background it handles receiving and checking
the arrived messages, extracting critical information
from them and inserting it into a database. The human
operator is asked to assign the best rescue unit
available to answer the distress call.
The database server that stores and manages the
information is the MySQL 4.0.18 server. The software
communicates with the server using the MySQL C
API, sending SQL commands for retrieving or
inserting data.

The communication process with the GPS Receiver is

carried out much in the same way as with the phone.
In fact, it is even simpler to implement. The Receiver
responds to various control NMEA (National Marine
Electronics Association) sentences by returning
strings of characters containing the required data.
These are filtered and the necessary information is
extracted.

IV.6. Temperature Sensor Routines

The message transmitted to the base station also
contains a temperature reading provided by the
DS1629 circuit. The command to read the temperature
and the temperature returned by the DS1629 are
transmitted through a two-wire serial bus the I2C
bus.
V. THE BASE STATION
V.1. Overview
The base station runs the software that monitors
Fig. 5. Main application interface
The main window of the application, shown in Fig.5,
displays everything the human operator needs to see
in a simple and clear manner. It shows the received
messages and their evolution, the rescue teams and
their status, and easily allows assigning one or more
rescue teams to one of the distress messages. It lets
the operator configure and reconfigure LoRD units
and also use the remote control mode by sending
START or STOP command messages.

358

V.2. Communication with the phone
This is accomplished using the Hayes commands
that the phone supports. On startup, the software
opens the serial port of the PC and checks if a phone
is present. If so, it attempts to do a necessary
configuration of the phone. This essentially tells it to
send a short notifying line of text over the serial
connection whenever a new SMS message is received.
Then the software enters an idle state, when it
permanently checks the phones status while
monitoring both the serial port for notification alerts
and the interface for the human operators actions.
When the phone receives a SMS from the network, it
sends a notifying text over the serial connection. Then
the software obtains the message after a series of
stages, which include decoding it from the PDU
format and checking the security bytes.
To send a SMS, the software uses the same process
mentioned above, only reversing the stages. When the
human operator wants to configure / reconfigure a
LoRD unit or to send a start / stop command to it, the
software sends the necessary message to the unit,
containing the configuration options or the command.
V.3. Communication with the database server
To ensure proper storing and handling of the critical
emergency information, as well as fast access and
sorting, we decided to use a dedicated database server.
Using an existing database server ensures a very low
rate of possible failures, as its strong and weak spots
are already known and the server is used by many
other applications and web services.
Another issue we were interested in was the security
of handling and storing the data. The server uses
encrypted passwords and has a powerful and versatile
security scheme, as it can restrict access depending on
the user, the host the user logs from and the user
privileges. It also has support for SSL secure TCP/IP
connections, although in our project we do not use
them. In conclusion, the security functions available
are sufficient even for an industrial implementation
with very strict security requirements.
We used the available C API to interact with the
server. The API provides C functions that allow
access to everything the server is capable of.
359
VI. SUMMARY
In this paper we present a general overview of the
system we have developed, as well as provide detailed
information about its functioning parameters.
The LoRD is designed to meet all requirements an
emergency system has. Providing important, real-time
data about the site of any kind of accident, the
warning system we built instantly notifies the rescue
units of the extent of the danger someone could be in.
It also has many built-in security and user-friendly
capabilities.
Our system was built to prove the fundamental
concept of our project: gathering remote information
and sending it to a base station the instant an accident
occurs. Further developments, like integrating more
different types of sensors car integrity indicators,
smoke detectors etc. may be carried out. In the end,
it is even possible to fully embed the LoRD module
into the car computer system using phone and GPS
OEM modules.
Amateur hikers, mountain rescue units, firefighters on
a mission, all of them can benefit from using our
LoRD module without many modifications. Traffic
jams and chain collisions can also be detected early
and partially avoided. Furthermore, the remote
activation mode supports tracing applications like
retrieving stolen vehicles, detecting cars exceeding
speed limits using the facilities of the GPS, or
tracking people who may be lost. The social
usefulness of the LoRD system is based on the large
number of different applications that call for such an
emergency management system, being implied by the
potential to minimize injury extent and to save human
lives.
REFERENCES
[1] Web Site of The US National Bureau of Transportation
http://www.bts.gov/publications/national_transportation_st
atistics/2003/html/table_02_01.html (road safety statistics)
[2] DS1629 Digital Thermometer Data Sheet, Maxim
Semiconductors,
http://www.maxim-ic.com
[3] 80C51- Based 8-Bit Microcontrollers, Phillips
Semiconductors
[4] B. Hofmann-Wellenhof, H. Lichtenegger, J. Collins, GPS
Theory and Practice, Springer WienNewYork, 1992
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
An improved MIMO-OFDM channel estimator in the
tracking phase
Andrei Alexandru Enescu, Silviu Ciochin1
Abstract MIMO (Multiple Input Multiple Output)
channels provide high theoretical capacity, today
intensively exploited in modern communication systems,
where they employ space diversity or/and provide a
spectral efficiency increase. The performances of such
communication systems however greatly depend on the
channel estimator accuracy [1]. Keeping in account the
possible fast channel variation, just a test sequence
based periodic evaluation may not be sufficient. Hence,
it is necessary to design a channel estimator for the
tracking phase in which the channel coefficients are
updated during the data frames.
Keywords: MIMO channel, channel estimator, OFDM
I. INTRODUCTION
The OFDM (Orthogonal Frequency Division
Multiplexing) systems are known to provide good
spectral efficiency along with mitigating the effects of
frequency-selective fading by transmitting the useful
information over separate subchannels. Most of the
communication standards state that the same power is
transmitted over the subcarriers, without any
supplementary power control. At the channel output,
some of the subcarriers will be more affected by
fading than others and thus, after equalization the
subcarriers will feature different values of signal-to-
noise ratio (SNR).
This is the main reason for which space diversity has
been used for the past years in order to combat the
effect of fading in different forms.
The most common and attractive method because of
its simplicity is space-time coding, which uses a
certain way to introduce redundancy, somehow as in
the linear binary channel codes.
The MIMO channel can be used in order to increase
the spectral efficiency in spatial multiplexing systems
context, when different information is sent over
different antennas (layers).
In both cases, the channel estimator has the same
structure and it is initially based on training symbols,
either preambles (first one or two OFDM symbols are
known at both Tx and Rx) or pilots (a number of
known subcarriers modulator per each OFDM
symbol). In [2] and [3], some classic channel
1
Facultatea de Electronic i Telecomunicaii, Catedra de
Telecomunicaii Bd. Iuliu Maniu Nr. 1-3, Bucureti, e-mail {aenescu,silviu}@comm.pub.ro
360
estimators in the initial training phase are presented.
When dealing with mobile environments though,
some fast updating channel estimators are used, with a
high accuracy.
The main advantage of an OFDM system is the
reduced complexity of the equalizer or channel
estimator.
This paper is organized as follows. We present the
time and the frequency models for the MIMO-OFDM
channel and describe a time domain estimator, which
is presented in [2], for the training and the tracking
phases. In the fourth section, we present an improved
estimation method, sustained by simulation results
shown in Section V.
II. TIME DOMAIN AND FREQUENCY DOMAIN
CHARACTERIZATION OF MIMO-OFDM
CHANNEL
If no intersymbol interference occurs, i.e. the cyclic
prefix length is assumed to be greater than the channel
impulse response length and any other
synchronization errors are neglected (carrier offset,
sampling clock offset, phase noise etc.), the input-
output relation of the MIMO channel may be directly
expressed in time domain as a cyclic convolution
product:
Nt
ri ( n, l ) = hij ( n, l ) t j ( n, l ) + ni ( n, l ) (1)
j =1
where
ri(n,l) represents the received sample for symbol
n, at receive antenna i, at lth time instant
tj(n,l) represents the transmitted sample for
symbol n, at receive antenna i, at lth time instant,
hi,j(n,l) is the propagation path impulse response
lth tap, sampled at time n, from antenna j to
antenna i,
ni(n,l) is the noise impinged on receive antenna i,
on symbol n and time l.
Nt is the number of transmit antennas and Nr the Let us define the vectors
number of receive antennas.
ri ( n ) = Ri ( n, 0 ) L Ri ( n, K 1)
T

The convolution product may be explicited as

ni ( n ) = N i ( n, 0 ) L N i ( n, K 1)
T
(5)

( )t ( n, l p )
K 1
hij ( n, l ) t j ( n, l ) = hij n, p h ij ( n ) = hij ( n, 0 ) L hij ( n, K 0 1)
T

K0 j K0
p =0

(2) and an extended channel taps vector of length N t K 0

where K0 represents the channel impulse
h i ( n ) = hTi ,1 ( n ) , hTi ,2 ( n ) ,L , hTi , Nt ( n )
T
response maximum length. (6)
In frequency domani, after the FFT at the
receiver, we obtain where T denotes the transpose operator.
The equation (3) may be then written as
Nt
Ri ( n, k ) = H ij ( n, k ) T j ( n, k ) + N i ( n, k ) (3)
j =1
ri ( n ) = D ( n ) hi ( n ) + ni ( n ) (7)

where where
Ri(n,k) represents the received sample on
subcarrier k, for symbol n, at receive antenna i, d 0,1 ( n ) L d 0, Nt ( n )
Tj(n,k) represents the transmitted sample on
D (n) = M O M (8)
subcarrier k, for symbol n, at transmit antenna j, d
Hi,j(n,k) are the propagation path transfer K 1,1 ( n ) L d K 1, Nt ( n )
function, sampled at time n, normalized
frequency k, from antenna j to antenna i, and
Ni(n,k) is the noise impinged on receive antenna
i, on symbol n and subcarrier k. WK ( l ) = WKl 0 WKl 1 L WK (
l K0 1)
(9)
Nt is the number of transmit antennas and Nr the
number of receive antennas.
In (3), k takes values from 0 to K-1, where K is with
the total number of subcarriers.
Obviously, the signals from (3) are strictly dl , q ( n ) = Tq ( n, l ) WK ( l ) , l = 0,K , K 1, q = 1,K , N t (10)
related through a Discrete Fourier
Transformation with the samples from (1). j
2
In (9), we denoted by WK = e the Nth root of the
T j ( n, k ) = DFT {t j ( n, l )} ( k )
K

unit value.
Ri ( n, k ) = DFT {ri ( n, l )} ( k ) (4) The problem of estimating the channel taps can be
H ij ( n, k ) = DFT {hij ( n, l )} ( k )
equivalently formulated to finding the optimum set of
taps h i ( n ) , which minimizes the least squares
The channel estimator must thus compute a total
(LS) cost function
number of KN t N r values of channel transfer
function coefficients, or a number of K 0 N t N r 2
J i ( n ) = ri ( n ) D ( n ) h i ( n ) (11)
taps of the channel impulse response in time
domain.
Based on the above relation, the paper develops From [2], we get the solution of the problem,
an improved version of a least squares based given by the normal matrix equation
time-domain channel estimator, that makes use
of a conveniently chosen set of training symbols, Q ( n ) h i ( n ) = pi ( n ) (12)
by extending the estimation procedure in the
tracking phase, when no training symbols are where we denoted
available.
Q ( n ) = DH ( n ) D ( n ) (13)
III. TIME-DOMAIN CHANNEL ESTIMATOR

Training phase pi ( n ) = DH ( n ) ri ( n ) (14)

361

1 Nt
(23)
H
is the Hermitian operator (T*). h i( r ) ( n ) = p(i ) ( n ) Q r , j ( n ) h i( j ) ( n 1) , r = 1, N
r

K t
j =1
The complexity of the estimator derived in (12) j l
may be further reduced when choosing an One should notice that the matrices p, Q are not yet
appropriate modulation scheme for the training known at the current iteration, since they depend on
symbols [2] i.e. current transmitted symbols, according to (13), (14).
The transmitted symbols will be detected using the
previous channel taps from the equation (3),
T j (n, k ) = T1 (n, k )WK ( j 1) k , N , j = 2, N t (15) expressed by matrix operands

K r% ( n, k ) = H
% ( n, k ) t% ( n, k ) + n% ( n, k ) (24)
= (16)
Nt
in which we denoted
where takes the integer part of the number.
r% ( n, k ) = R0 ( n, k ) L RNr 1 ( n, k )
T
(25)
(12) assumes a constant modulus constellation.
In this case,
t% ( n, k ) = T0 ( n, k ) L TNt 1 ( n, k )
T
(26)
Q ( n ) = KI N t K 0 (17)
H 0,0 ( n, k ) L H 0, Nt ( n, k )
and the estimated solution becomes
H ( n, k ) = M O M (27)
H ( n, k ) L H
1 Nr ,0 N r , Nt ( n, k )
h i (n) = p i (n) (18)
K The decision for t% ( n, k ) may be taken in a
similar LS matrix equation solving, where fast
Tracking phase
algorithms (e.g. QR decomposition) may be
The tracking phase updating relation is used.
t% ( n, k ) = H % ( n, k ) 1 r% ( n, k )
% H ( n, k ) H (28)
Nt

Q ( n ) h ( n ) = p( ) ( n ) ,
r, j
( j)
i i
r
r = 1, N t (19)
j =1 After solving (28), we compute all the matrices p, Q
and solve (23).
If the symbols are of constant modulus, without In Fig. 1, we present the estimator block diagram in
generally fulfilling the constraint (15) then training phase, without explicitly showing the
decision block [5].
Q j , j ( n ) = KI K 0 (20)
IV. IMPROVED CHANNEL ESTIMATOR
and
Our idea arises from the natural observation that (16)
is a linear system that may be iteratively solved, once
h i( r ) ( n ) =
1 (r) Nt
p i ( n ) Q r , j ( n ) h i( j ) ( n ) , r = 1, N (21) having a primary estimated solution set. Suppose that
K we solve the linear system at a rate Tsys and that the
t
j =1
j l OFDM symbol period is Tsym. Then the number of
affordable iterations will be
If when computing pi( r ) ( n ) and Q r , j ( n ) we use
the detected appropriate symbols Tj(n,k) and if Tsym
I = (29)
we assume that the channel varies slowly enough Tsys
as to fulfill
Obviously, this number can be increased up to
hi( j ) ( n ) hi( j ) ( n 1) (22) any value in a pipeline based structure. However,
the price to be payed is reflected in the
then complexity increasing with a factor equal to I.

Table 1 summarizes the training and the tracking

phases along with the proposed modifications.

362
 Fig. 1. Channel estimator in the training phase

Table 1 One can see that the transmitted symbols may be

estimated after each iteration, which, however, will
- training phase slow down the estimation process.
compute h (i j ) ( 0 )
- tracking phase
n=1,2, V. SIMULATIONS
i=1:I
- detect t% j ( n, k ) if i=1 The channel estimator was tested in multiple
scenarios. The instrument that measures the
- compute i hi( j ) ( n ) based on performances will be MSE (Mean Squared Error),
defined in time domain as
detected t% j ( n, k ) and previously
estimated channel vector
1 Nr Nt 2
hi ( n ) or I hi( ) ( n 1)
i 1 ( j ) j E ( n) =
K 0 Nt N r
h ( ) ( n ) h( ) ( n )
i =1 j =1
i
j
i
j
(30)
- optional: recompute t% j ( n, k )

363
The channel model is the one used in [5], model
3.2.1 and 3.2.2, with TRMS/Ts=2, the Doppler offset
fd=1kHz.
We intend to study the variation of MSE with
respect to the number of iterations.

Case 1. Nt=2, Nr=1, QPSK modulation, SNR=30dB

(Fig. 2)

Fig. 3. Time evolution of MSE[dB] (64-QAM (2-

1), SNR=30dB)
The mathematical explanation resides in the fact
that 16-QAM does not fulfill the constant modulus
condition imposed over the constellation. One may
consider freezing the training estimator for the rest
of the frame.

Case 3. Nt=2, Nr=1, QPSK modulation, SNR=10dB

Fig. 2. Time evolution of MSE[dB] (QPSK (2-1),
(Fig. 5)
SNR=30dB)
In each case, the first estimate is the most accurate,
as it is expected, since it is based on a priori known
symbols. Starting with the second symbol, MSE
increases and remains on an approximate constant
value, depending on the assumed channel model.
If the first estimate is used along the first frame, the
MSE will increase along the burst.
With respect to the case when only one iteration is
performed, a 17dB gain is obtained when using
I=2. Another 10dB are gained if I=3.
If no update is performed, we lose 15dB with
respect to the case when I=1.
An intuitive remark shows us that the tracking
phase estimation performances may be brought as
close as possible to the training phase performances Fig. 4. Time evolution of MSE[dB] (QPSK (2-1),
if the number of iterations is sufficiently large. For SNR=10dB)
instance, the difference between the training and
the tracking phase is of 38dB for I=1, 21dB for I=2 Once the signal to noise ratio decreases, the MSE
and 11dB for I=3. differences between the estimators become smaller
and the MSE dynamic range narrowens. For
Case 2. Nt=2, Nr=1, 64-QAM modulation, instance, when I=2 we obtain almost the same MSE
SNR=30dB (Fig. 3) as in the case of I=3.

The MSE obtained in the training phase is

comparable to that from QPSK case.
However, in the tracking phase, not only that the
performances are much worse than in QPSK case,
but increasing the number of iterations has negative
effects on MSE. Once the signal to noise ratio
decreases, the MSE differences between the
estimators become smaller and the MSE
dynamic range narrowens. For instance, when
I=2 we obtain almost the same MSE as in the
case of I=3.
VI. CONCLUSIONS AND FUTURE WORKS
This paper presents a new improved MIMO-
OFDM channel estimator for the tracking phase, in
time-domain. The original estimator is based in the
training phase on a least-squares optimization. The
tracking phase makes use of a primary estimate of
the transmitted symbol and thus reestimates the
channel vector. The procedure may be reiterated
more times during a block period. The mean
squared error between the estimated channel vector
and the ideal channel vector is significantly

364
improved, especially when the signal to noise ratio
is high. Theoretically, when the number of
iterations becomes larger, MSE in tracking phase
approaches MSE in the training phase.
The estimator itself features poor performances
when using non-constant modulus constellations
(QAM-type).
The next step in our study will be simulating the
algorithm in finite precision and elaborating an
efficient digital structure with low speed and low
area.

REFERENCES
[1] R. Narasimhan, Performance of Diversity Schemes for
OFDM Systems with Frequency Offset, Phase Noise and
Channel Estimation Errors, IEEE Transactions on
Communications, vol. 50, no. 10, Oct. 2002
[2] Ye Li, Nambirajan Seshadri, Sirikiat Ariyavisitakul, Channel
Estimation for OFDM Systems with Transmitter Diversity in
Mobile Wireless Channels, IEEE Journal on Selected Areas in
Communications, vol.17, no.3, Mar. 1999
[3] S. Coleri, M. Ergen, A. Puri, A. Bahai, A Study of Channel
Estimation in OFDM Systems, Globecom 01, Vol. 1, pp 136-140
[4] W. Bai, C. He, L. Jiang, H. Zhu , Blind Channel Estimation
in MIMO-OFDM Systems, IEEE Transactions on Broadcasting,
vol. 48, no. 3, Sep. 2002
[5] A.A. Enescu, Space-time coded OFDM communication
systems, B. Sc. Thesis, UPB, June 2003
[6] W. Bai, C. He, L. Jiang, H. Zhu , Blind Channel Estimation
in MIMO-OFDM Systems, IEEE Transactions on Broadcasting,
vol. 48, no. 3, Sep. 2002
[7] S. Zhou, G. Giannakis, Finite Alphabet Based Channel
Estimation for OFDM and Related Multicarrier Systems, IEEE
Transactions on Communications, vol. 49, no. 8, Aug. 2001
[8] Ye Li, Leonard Cimini, Nelson Sollenberger, Robust
Channel Estimation for OFDM Systems with Rapid Dispersive
Fading Channels, IEEE Transactions on Communications,
vol.44, no.9, Jul. 1998
[9] IEEE 802.16 Broadband Wireless Access Working Group,
Channel Models for Wireless Fixed Applications

365
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Performance Evaluation of a Multiple Access DCSK
System under a Noisy Multiuser Environment
Dorin ANDREI1, Clin VLDEANU2, Alexandru ERBNESCU1
Abstract - In this paper a multiple-access
technique for use with DCSK (MA-DCSK) under a
noisy condition is proposed and analyzed. In order to
evaluate the performance of the system, a simple one
dimensional iterative map is used to generate the
chaotic signals for all users. As would be expected, the
proposed scheme achieves similar error probabilities
for all users and the error performance degrades as
the number of users increases. Bit-error rates for
different number of users and computer simulations
are performed to verify the results.
I. INTRODUCTION
Chaotic signals are characterized by their
sensitive dependence on initial conditions as well
as random-like behavior. Moreover, their
continuous broadband power spectrum feature
renders them useful in encoding information in
communications. Typically, in a chaos-based
digital communication system, digital symbols are
mapped to nonperiodic chaotic basis functions. For
instance, in chaos-shift-keying (CSK), different
symbols are mapped to different chaotic attractors
which are produced either by a dynamical system
for different values of a bifurcation parameter or by
a set of completely different dynamical systems
[1]. A coherent correlation CSK receiver is then
required at the receiving end to decode the signals.
Noncoherent detection is also possible provided the
signals generated by the different attractors have
different attributes, such as mean of the absolute
value, variance and standard deviation.
The optimal decision level of the threshold
detector will depend on the signal-to-noise ratio in
general, although specific examples with noise-
invariant threshold can be designed for CSK.
1
Military Technical Academy, Bucharest
2
Politehnica University of Bucharest
366
As in other communication systems, its performance
increases with the symbol energy or the signal-to-
noise ratio. To overcome the threshold level shift
problem, differential CSK (DCSK) is proposed. The
advantage of the DCSK over CSK is that the
threshold is always set at zero and is independent of
the noise effect.
In conventional communication systems, the
allocated spectrum is shared by a number of users.
Multiple-access techniques such as frequency-division
multiple access (FDMA), time-division multiple
access (TDMA) and code-division multiple access
(CDMA) are commonly used. Similar to CDMA,
CSK/DCSK spreads the spectrum of the data signal
over a much larger bandwidth as compared to FDMA
and TDMA. As a result, multiple access becomes an
essential feature for practical implementation of the
system.
In this paper, a multiple-access technique for use
with DCSK (MA-DCSK) under a noisy condition is
proposed and analyzed. The proposed scheme gives
equal average data rates of all users. As in a single-
user DCSK system, each bit duration is always
divided into two time slots for all users. To minimize
the correlation between signals, the frame periods and
the arrangements of the reference and sample
waveforms of all users are different. In order to
evaluate the performance of the system, a simple one-
dimensional iterative map is used to generate the
chaotic signals for all users. As would be expected,
the proposed scheme achieves similar error
probabilities for all users and the error performance
degrades as the number of users increases. However,
we show that when the correlation between samples
of the same/different chaotic signals is low, achieved
by using a large spreading factor, a low bit-error rate
(BER) can be achieved. Section 2 describes the
DCSK modulation method. The multiple-access
technique is discussed in Section 3. Results found by
the analytical method are compared with simulation in
Section 4.
 II. REVIEW OF DCSK
DCSK was first proposed by Kolumban et al.
[1]. By using a chaotic carrier to spread the digital
signal over a large bandwidth, the spread signal
possesses some of the advantages of spread
spectrum communications such as mitigation of
multipath fading and low probability of detection.
In DCSK, each bit duration is first divided into
two equal time slots and every transmitted symbol
is represented by a pair of chaotic signal samples
sent in the two slots. The first sample serves as the
reference (reference sample) while the second one
carries the data (data sample). If a +1 is to be
transmitted, the data sample will be identical to the
reference sample, and if a -1 is to be transmitted,
an inverted version of the reference sample will be
used as the data sample. Assume the system is
discrete and starts at k = 0 . Let 2 be the
spreading factor, defined as the number of time
units occupied by a binary symbol, where is an
integer. Fig. 1 shows a typical transmitted
waveform, for a spreading factor of 10. At the
receiving end, the reference sample and the
corresponding data sample are correlated.
Depending on whether the output is larger or
smaller than the threshold zero, a +1 or -1 is
decoded. Fig. 2 shows the output waveform of the
correlator, which is sampled at multiplies of 2
time units and Fig.3 shows the performance of the
DCSK in the presence of the noise.
Fig.1. A typical DCSK signal (spreading factor = 10)
Fig.2. Output of the correlator and the decoded symbols
Fig.3. BER versus Eb / N 0 in a DCSK system
(spreading factor = 100)
III. ANALYSIS OF MA-DCSK SYSTEM
In a MA-DCSK system, to avoid excessive
interference, and hence, misdetection, the separation
between the reference and data samples must be
different for different users. A multiple access scheme
has been proposed by Lau et al [2,3] where the
separation between the reference and data samples
differs for different users, as illustrated in Fig. 4. For
all users, each bit duration is first divided into 2 slots.
For user i , 2i consecutive slots are collected to form
a frame. Hence, the slot duration (half of bit duration)
is the same for all users but the frame periods are
different for different users. In each frame of user i ,
the first i slots (slots 1 to i ) will be used to transmit
i sets of reference samples while the remaining i
slots (slots i + 1 to 2i ) are used to transmit i sets of
data samples. If a binary symbol +1 is to be
transmitted in slot i + 1 , the samples in slot 1 are
repeated in slot i + 1 . If a -1 is to be sent, an
inverted version of the samples in slot 1 will be
transmitted in slot i + 1 . Similarly, in slot i + 2 , the
same or inverted copy of the samples in slot 2 is sent,
and so on. As a result, the reference and data samples
of user i will be separated by i slots. Therefore,
within a frame of length i bit periods (or 2i time
slots), i bits of information will be sent. The data
rates of all users are the same.
Fig. 5 shows a multiple access DCSK
communication system in a discrete-time mode. In the
transmitter of the i th user, a chaotic map is used to
{ }
generate a chaotic signal xk( )
i
with zero mean. The
chaotic maps for different users are different in
general. Assume that chaotic samples are sent in
each slot (spreading factor = 2 ). Consider the
transmitted signal of user i during the m th time slot,
i.e., for time k = ( m 1) + 1, ( m 1) + 2,..., m .
Denote the output of the transmitter by yk(i ) . If the slot
is a reference sample slot, yk(i ) = xk(i ) . If the slot
corresponds to a data sample slot sending a binary
symbol +1, yk(i ) = xk(i) i . Otherwise, if the slot
367
 Fig.4. Transmission scheme for the MA-DCSK communication system

corresponds to a data sample slot sending a binary l

N
N
symbol -1, yk(i ) = xk(i) i . Thus, for the m th time
= a( ) x( )
k = ( l 1) +1 u =1
l + k al(+v )j xk( v+) j v + k + j
u u
k
v =1
u

slot of user i , we define am( i ) as (5)
+1 if yk(i ) = xk(i) l where i = 0 ( i = 1, 2,..., N ) for reference samples,
am( i ) = (1) and i = i for data samples. The decoding of the
(i ) (i )
1 if yk = xk l
symbol corresponding to this pair of time slots,
where l = 0 for reference samples and l = i for
denoted by l( j ) , can be done according to the
data samples. Therefore, the transmitted signal of
user i during the m th time slot can be represented following simple rule
by +1 if yl( j ) > 0
l( j ) = . (6)
yk(i ) = am( i ) xk(i) l ; ( j)
1 if yl < 0
k = ( m 1) + 1, ( m 1) + 2,..., m (2) In [2] several assumptions are made for
derivation of BER. With these assumptions
The overall transmitted signal at time k, denoted by (chaotic sequences from different generators are
yk , equals independent of one another, all chaotic sequences
have zero mean) and if is assumed that all users
yk = i =1 yk(i ) .
N
(3) transmit with equal average power, the BER for the
Assuming the channel is AWGN, the j th user can be written as [2,3]
received signal at time k , denoted by rk , is simply
1 2 ( j ) 2 ( N 2 1)
( j)
given by BERDCSK = erfc + +
2
rk = yk + k (4)
where denotes the AWGN with zero mean and
1

E
1 2
variance N 0 / 2 . For each user, the signal received E 2

+4 N b + 2 b
(7)
during a reference sample slot will correlate with N0 N 0

the signal at the corresponding data sample slot.
Depending on whether the output is larger or where Eb denotes the average bit energy,
smaller than the threshold, a +1or -1 is
Eb = 2 Ps , Ps = E ( xk ) and
2
decoded. Such a correlator-based DCSK receiver is
shown, also, in Fig. 5. Note that the sampling
switch only operates during the second half of each

var xk

( j) 2
( )
frame and the threshold detector produces a ( j) = 2
(8)
Ps
decoded symbol after each slot during that period
of time. Clearly, the performance of the MA-
Consider the j th user and the received signal DCSK system is affected by
1. spreading factor 2 ;

{( x ) } for a given P ;
during the l th time slot. Suppose the slot 2
( j)
corresponds to a reference-sample slot for the j th 2. variance of k s

user, i.e., al( j ) = +1 . The reference samples in this 3. number of users N;

slot will then correlate with the received samples j 4. noise power.
slots later, i.e., in the ( l + j ) th slot. The output of
Thus, for a fixed Eb / N 0 , we may improve the
the correlator is
l
BER for the j th user by making one or more of
N N
yl( j ) = sk(u ) + k sk( v+) j + k + j the following adjustments.
k = ( l 1) +1 u =1 v =1

368
 1. Change the spreading factor 2 until the
optimal BER is obtained;
( x( ) )
2
2. Minimize the variance of k
j
for a
fixed Ps ;
3. Reduce the number of users N .

IV. SIMULATIONS, RESULTS AND

DISCUSSIONS

The quadratic map (x k +1 = 1 2 xk2 ) and cubic

map ( xk +1 = 4 xk3 3xk ) are used in our simulations.
The number of users in the system is assigned up to
5 and different initial conditions are assigned to
different users to generate the chaotic signals.
In Fig. 6 are shown the results for a 3-user and
a 5-user MA-DCSK system in which all chaotic
sequences are generated from the cubic map and a
spreading factor of 200 ( = 100 ) is used. It is
observed that BERs increase (degrade) as the
number of users increases for a given Eb / N 0 . This
is apparently due to the increasing inter-user Fig. 7. BER versus spreading factor under different Eb / N 0
interference. a) 3-user system; b) 5-user system

V. CONCLUSIONS

In this paper, a multiple-access technique for

use with DCSK under a noisy condition is
analyzed. The access scheme of different users has
been described and the corresponding noncoherent
receiver has also been designed to decode the
signals.
The MA-DCSK system has been studied with
the assumption that the time slots are synchronized
among all participating users. We expect that the
interference between users will not vary too much
even when the time slots are not synchronized.
Finally, the performance of the system should
be further investigated over a multipath and fading
channel. These will be left to future publications.

REFERENCES

[1] G. Kolumban,M.P. Kennedy and L.O. Chua,

Fig. 6. BER versus Eb / N 0 under different spreading factor
a) 3-user system; b) 5-user system
Finally, assuming the quadratic map is used,
we plot the BER against the spreading factor under
different Eb / N 0 for a 3-user and a 5 user system.
Fig. 7 shows that when the spreading factor
increases initially, the BER improves. After the
spreading factor has reached an optimum value,
increasing the spreading factor further will
deteriorate the BER.
The role of synchronization in digital
communications using chaos Part II: Chaotic
modulation and chaotic synchronization, IEEE
Transactions on Circuits and Systems, vol. 45, No.
11, November 1998.
[2] F.C.M. Lau, M.M.Yip, C.K.Tse and S.F.Hau,
A Multiple-Access Technique for Differential
Chaos-Shift Keying, IEEE Transactions on
Circuits and Systems, vol. 49, No. 1, January 2002.
[3] F.C.M.Lau and C.K.Tse, Chaos-Based Digital
Communication Systems, Springer-Verlag, 2003.
[4] D. Andrei,C. Vladeanu, A multiple access
DCSK system, International Conference
Communications 2004, 3-5 June 2004, Bucharest.

369
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
A Low Complexity Decision Feedback Equalization for Sparse
Wireless Channels
Adrian Florin PUN, erban Georgic OBREJA *
AbstractIn this paper, a low complexity
decision feedback equalizer (DFE) appropriate for
channels with long and sparse impulse response (IR) is
studied. Such channels are encountered in many high-
speed wireless communications applications. It is shown
that, in cases of sparse channels, the feedforward and
feedback (FB) filters of the DFE have a particular
structure, which can be exploited to derive efficient
implementations of the DFE, provided that the time delays
of the channel IR multipath components are known. This
latter task is accomplished by other technique, which
estimates the time delays based on the form of the channel
input-output cross-correlation sequence in the frequency
domain. A distinct feature of the resulting DFE is that the
involved FB filter consists of a reduced number of active
taps that implies some computational savings than
conventional DFE. The resulting DFE also exhibits,
improved tracking capabilities and faster convergence as
compared with the conventional DFE, that implies a
shorter training sequence . Moreover, the new algorithm
has a simple form and its steady-state performance is
almost identical to that of the conventional DFE.
Index Terms Adaptive equalizers, decision
feedback equalizers (DFEs), multipath channels.
I. INTRODUCTION
In many wireless communication systems the
involved multipath channels exhibit a long time
dispersion, and delay spreads of up to 40 s are often
encountered. A typical application of this is high
definition television (HDTV) signal terrestrial
transmission, where the involved channels consist of a
few non-negligible echoes, some of which may have
quite large time delays with respect to the main signal
(see for instance the HDTV test channels reported in
several ATSC documents and summarized in [1]). If the
information signal is transmitted at high symbol rates
through such a dispersive channel, then the introduced
intersymbol interference (ISI) has a span of several tens
up to hundreds of symbol intervals. This in turn implies
that quite long adaptive equalizers are required at the
receivers end in order to reduce effectively the ISI
component of the received signal. Note that the
situation is even more demanding whenever the channel
frequency response exhibits deep nulls.
The adaptive decision feedback equalizer
(DFE) has been widely accepted as an effective
* Politehnica University of Bucharest e-mail: adi@radio.pub.ro, serban@radio.pub.ro
370
technique for reducing ISI [2], [3]. Moreover, it has
been shown that the DFE structure is particularly
suitable for multipath channels, since most part of ISI is
due to the long postcursor portion of the impulse
response (IR). Recall that an important feature of the
DFE is that the postcursor ISI is almost perfectly
cancelled by the feedback (FB) filter, provided of
course that the previous decisions are correct. Since the
postcursor ISI is cancelled by the FB filter, a relatively
shorter feedforward (FF) filter is adequate to reduce the
remaining ISI. Moreover since noise is involved only in
the output of the FF filter, the DFE exhibits less noise
enhancement effects as compared with linear equalizers.
In high-speed wireless applications, of the type
described above, the implementation of a DFE
algorithm becomes a difficult task for two main reasons.
First, due to the small intersymbol interval, the time
available for real-time computations is very limited.
Second, due to the long span of the introduced ISI, the
DFE must have a large number of taps, which implies
heavy computational load per iteration.
During the last decade there have been many
efforts in different directions toward developing
efficient implementations of the DFE. As such
directions, we mention IIR methods, block adaptive
implementations, efficient algebraic solutions, modified
DFE schemes, etc. [4][12]. As mentioned above, in the
applications of interest, the involved multipath channel
has a discrete sparse form. Efficient DFE schemes
which exploit the sparseness of the channel IR have
been derived in [13][15].
In this paper, a new DFE algorithm,
appropriate for sparse multipath channels is studied,
proposed in [19]. The algorithm consists of two steps.
In the first step, the time delays of the multipath
components are estimated in a novel way by properly
exploiting the channel IR form [16]. In the second step,
the DFE is applied, with the FB filter having a
significantly reduced number of taps. These taps are
selected so as to act only on time positions associated
with the estimated time delays of the involved multipath
components. A distinct feature of the novel approach
followed in this paper is that the required channel
parameters are the locations of the multipath
components. This is opposed to most of the existing
works [14], [15], in which the whole channel IR has to
be initially estimated. Moreover, the relation between
the active FB tap positions and the echo time delays is
determined by investigating the special structure of the represents the phase shift due to free space propagation
FF and FB filters in cases of sparse channels. of the i-th multipath component, plus any additional
The main advantages of this algorithm with phase shifts which are encountered in the channel. If the
respect to the conventional DFE are its lower channel IR is assumed to be invariant within a small-
complexity, faster convergence, and improved tracking scale time interval, then (1) can be simplified to
capabilities. It is important to note that due to the faster
convergence the proposed algorithm requires a shorter hc ( ) = i e ji ( i ) (2)
training sequence as compared with classical DFE, thus, i

it offers an additional saving in bandwidth. Note that its The overall channel IR, including the combined
overall complexity is of the order of the number of
multipath components and hence it is, in practice,
transmitter and receiver filters response, say , p ( )
several times lower as compared with the conventional can be written as

h ( ) = i e ji p ( i )
DFE.
(3)
The paper is outlined as follows. In Section II,
i
the multipath channel is described and the problem is
formulated. In Section III, the proposed efficient As mentioned in the introduction, in this paper, we deal
method for estimating and tracking the time delays is with sparse multipath channels having a relatively long
presented. The new DFE algorithm is developed in IR. Due to the sparseness of the multipath channel IR
Section IV and relevant computational issues are and the form of the pulse shaping function p ( ) ,
discussed. In Section V, the new algorithm is tested and
some indicative experimental results are provided. which decreases rapidly, the overall symbol spaced
Section VI concludes the work. channel IR remains sparse and can be expressed as
L
h ( nT ) = hnl ( nT nlT ) (4)
II. PROBLEM FORMULATION l =0

where L+1 is the number of the dominant IR

In this section, we first formulate the problem components appearing at the symbol spaced time
of information transmission through a multipath
channel, and we recall the conventional and well- instants, hnl is the complex amplitude of the l-th
studied DFE structure which is our starting point for the component, and nlT its respective delay, with T being
derivation of the new equalization technique. The
the symbol period. Delay n0 corresponds to the main
notation used throughout the paper is as follows. x ( n )
signal3 (n0=0 ), while the remaining ones correspond
denotes a scalar sample or symbol of sequence { x} at
either to causal ( n0 > 0 ) or to anticausal ( n0 < 0 )
time n , ci ( n ) is the coefficient of filter c at time n,
components. The symbol spaced IR spans k1 precursor
and X ( k ) is the k-th frequency bin of the discrete and k2 postcursor symbols, respectively. That is the
symbol spaced channel IR can be written in the vector
Fourier transform (DFT) of a sequence related to { x} .
form
Finally, vectors and matrices are denoted as lower case
bold roman type and as upper case slanted, respectively. h = h k1 L h0 L hk2
T
(5)

A. Baseband Multipath Channel From the total of the ( k1 + k2 + 1) IR coefficients

The multipath channel is encountered in only L+1 are assumed to be non-negligible, located at
almost all wireless communication systems, however, the nl positions.
its particular form is highly dependent on the specific
system and the application environment (i.e., bit rate,
modulation type, carrier frequency, transmitterreceiver B. Conventional DFE
separation and the around topography, cell type - if it is Taking into account (4), the sampled output of
for a cellular system - and the receivers motion within the multipath channel can be written as follows:
the cell, etc.). L
In general, the baseband IR of a multipath x ( n ) = hnl d ( n nl ) + w ( n ) (6)
channel with discrete components is written as [3] l =0

hc ( t , ) = i ( t , ) e ( c i i ) ( i ) (1)
j 2 f ( t ) + ( t , )
where {d } is an independent identically distributed
(i.i.d.) symbol sequence with variance d and {w} is
i 2

where i ( t , ) , i (t ) are the real amplitudes and zero-mean complex white Gaussian noise uncorrelated
excess delays, respectively, of the multipath component with the input sequence, with variance w2 . Note that
at time t. The phase term 2 f c i ( t ) + i ( t , ) symbol period T has been omitted for reasons of

371
simplicity. Obviously, { x} suffers from intersymbol an appropriate partitioning of both channel input and
output sequences and is described below.
interference due to the presence of undesired multipath
Let us first formulate the following 2N-DFT
components and in most cases equalization is necessary
for reliable reception. sequences for k = 0,1,..., 2 N 1
As mentioned in the introduction, the DFE N + p 1 2
jmk
structure is particularly suitable for equalizing multipath
channels. The LMS-based adaptive DFE is given by the
D (k ) =
m= p
d (n + m) e 2N
(12)
following set of equations:
2 N 1 2
jmk
X (k ) = x ( n + m) e
0 N

ck ( n ) x ( n k ) + bk ( n ) d% ( n k )
2N
d ( n ) = (13)
m =0
k = M +1 k =1
(7) where p is assumed to be an overestimated value of the
noncausal size of the channel IR (i.e p > k1 ). The
d% ( n ) = f {d ( n )} (8)
same is presumed for the quantity N-p as far as the size
e ( n ) = d ( n ) d% ( n ) (9)
of the causal part of the channel IR is concerned. If
these facts hold true, the method which is described
ck ( n + 1) = ck ( n ) 2 c x* ( n k ) e ( n ) below detects the positions of all precursor and
(10) postcursor components. Note that X ( k ) in (13) is
k = M + 1,..., 0
based on a 2N-length output sequence, while D ( k ) in
bk ( n + 1) = bk ( n ) 2 d% * ( n k ) e ( n )
b
(12) results from an N-length input sequence padded
(11) with zeros. As it will become evident from the
k = 0,..., N subsequent derivation, this is done in order for all
samples of the cross-correlation sequence to be equally
where { x} and {u%} denote the equalizers input and weighted. Indeed, if we consider the expected value of
decision sequences, respectively, ck are the the product of the above sequences, we obtain
coefficients of the Mlength FF filter, and bk are the E { X ( k ) D* ( k )} =
coefficients of the N-length FB filter ( N is taken at least 2 N 1 N + p 1 2
j (i m)k
equal to the channel span [10]). f {} stands for the = E { x ( n + i ) d * ( n + m )} e 2N

i =0 m= p
decision device function, , are the step sizes and
c b
(14)
* denotes complex conjugation. It is assumed that a
training sequence of appropriate length is available where E {} denotes the expectation operator. If we
ensuring convergence of the equalizer. That is the now substitute(6) to (14), we get
equalizer operates initially in a training mode and then
L
E { X ( k ) D* ( k )} = hnl
switches to a decision directed mode. In the following
sections, first, a frequency domain procedure is
l =0
proposed for detecting the time delays of the multipath

components of the channel IR. Then, a new efficient 2 N 1 N + p 1 j (i m)k
DFE structure is derived, which takes advantage of the E {d ( n + i nl ) d ( n + m )} e * N

special properties of the multipath channel. i =0 m= p
(15)
Since p is larger than the noncausal part of the
III. ESTIMATION OF THE ECHO DELAYS
channel IR, it is easily shown that for every l the indices
of d and d* in (15) are identical for N combinations of m
A well-established nonparametric procedure
for estimating the time delays of the multipath and i (with m = i nl ). Therefore, due to the i.i.d.
components is based on a proper cross-correlation of property of the input sequence (15) is written as
the input symbols with the corresponding channel
L
E { X ( k ) D* ( k )} = N d2 hn e
output samples. In a time domain implementation, the jnl k
N
(16)
estimation of the cross-correlation sequence for N lags l
l =0
requires O ( N ) operations per sample. It is shown
that, an appropriate frequency domain expression of the for k = 0,1,..., 2 N 1 . That is, we end up with a sum
cross-correlation sequence can be viewed as a sum of nl
complex harmonics, with the unknown time delays of complex harmonics at normalized frequencies .
interpreted as frequencies. Thus, to estimate the time
2N
delays, we suggest an FFT-based scheme of complexity Applying the 2N - IDFT to the resulting sequence, the

log ( N ) per sample. The proposed scheme stems from

372
locations nl of the multipath components are Determination of Dominant Components: We
determined at the nonnegligible points of the IDFT. see from (17) that samples RN of { x} and {d } are
Obviously, in a practical situation, time used to compute C
( R)
( k ) . The L+1 IDFT points of
averaging is used instead of E {} in order to
DX
(17) having the highest amplitude are then chosen as the
implement (16). In cases where the channel is assumed desired locations. The number L of the dominant
stationary, the above procedure can be done once during undesired can be computed by setting a threshold and
the training phase and then the obtained time delays can select the locations of the IDFT points of (17) having
be used in the algorithm as described in the next amplitudes which exceed this threshold.
section. Of course, in most situations in practice, the
channel exhibits variations and, thus, the required time
delays have to be tracked continuously. During IV. THE NEW METHOD
tracking, the frequency domain expression of the cross-
correlation sequence is formed using the decisions In the proposed algorithm, we focus our
provided by the equalizer (which operates in a decision attention to the demanding FB part and reduce the
directed mode). computational load by properly selecting O(L) number
Exponentially fading memory is imposed on of taps out of N taps. The main idea behind the
the estimation procedure by including a forgetting derivation of the algorithm is that due to the channel
factor in the frequency domain expression of the sparseness, the FB filter also possesses a specific sparse
cross-correlation sequence as follows: form. After exploiting its sparse form, the FB filter is
R 1 built so as to act only to a restricted set of tap positions.
CDXR = Rr ( k ) Dr* ( k )
( ) ( R 1 r ) N As a result, the algorithm offers significant
(17)
r =0
computational savings while its steady-state error
performance is similar to that of the conventional DFE.
where 0 < 1 and
N + p 1 2 A. Derivation of the Algorithm
jmk
Dr ( k ) =
m= p
N + p m 1
d ( n + rN + m ) e 2N
(18)
It is well known [2], [4] that in the minimum
mean-squared error (MMSE) DFE, the FF and FB
2 N 1 2 coefficients can be expressed in terms of the channel IR
jki
Xr (k ) = x ( n + rN + i ) e
i =0
2N coefficients. Indeed, based on the assumption that
previously detected symbols are correct, the
minimization of the mean-squared error (MSE)
for k = 0,1,..., 2 N 1 . Note that if factor were
E{ e (n) } leads to the following set of equations for
2
included only in (17), then the exponential weighting
would be applied on a block-by-block basis, thus the FF filter c M and the FB filter b N
affecting the tracking capabilities of the new
algorithms. However, additionally including in (18)
1
2
is equivalent to applying an exponential window in the c M = H1H1H + w2 I M H1e M + k1 (20)
time-domain sample-by-sample computation of the d
cross-correlation lags. When a new N-length block of
H H c
input and output samples is available, CDX ( k ) is
( R)
bN = 2 M (21)
updated as 0( N k2 )1
CDXR +1 = N CDXR + RR ( k ) DR* ( k ) where ( )
( ) ( ) H
stands for the conjugate transpose
(19)
k = 0,1,..., 2 N 1 operation, IM is the M M identity matrix,
e M + k1 = [ 0 ... 0 1] M ( k1 + M ) ,
T
and the
Recall that quantity C
( R)
( k ) can be
DX
M k2 matrices are given as shown in (22) and (23),
interpreted as a sum of complex harmonics with
unknown frequencies and complex amplitudes. Indeed,
as can be easily seen by inspecting (16), the frequency
bin k corresponds to the sequence index while the time
delay nl corresponds to the unknown frequency.

373
 h k1 h k1 +1 L h1 h0 h1 L L hM 2 hM 1
0 h L h2 h1 h0 L L hM 3 hM 2
k1

H1 [ H11 | H12 ] =
M M M M M M M M M M
(22)
0 0 L h k1 h k1 +1 L L L L hM k1
M M M M M M M M M M

0 0 L 0 0 L h k1 L h1 h0

hM hM +1 L hk2 0 L 0 we easily derive the following results concerning

h h L L LL L zeroth-, first-, and second-order terms of c M ,
H 2 = M 1 M
M
(23) respectively.
M M M M M M
There exists a zeroth-order contribution, equal to
h 1 h 2 L L LL hk2
h0*1 , to the last element of c M .
It is shown in [20], that for the special class of For each ni < 0 , there is a first-order contribution,
channels we consider, and for medium or high SNRs,
the solution of (20) can be approximated very closely
* 2 *
equal to h0 hni , to the ( M + ni ) th element of
by the solution of the much more simple set of c M . This is obvious if we see that Fe M is in fact the
equations. last column of F.
H c = e M
H
(24) For each combination of ni , n j with
12 M
ni + n j < 0 there is a second-order contribution, equal
e M = [ 0 ... 0 1] . Obviously, to solve the
T
where
above system the condition that the M M matrix
* 3 *
(
to h0 hni hn j , to the M + ni + n j th element of .
*
)
H
H12 is nonsingular is required. c M This is shown by forming the product of F with its
last column Fe M and taking into account the positions
1) Solution of H12c M
H
= e M : From (22), H12
H
of the nonzero elements of the channel IR.
can be written as

H12H = h0*I M + F = h0* ( I M + h0*1F ) (25) In conclusion, vector c M can be expressed up

to second-order approximation as follows:
H
where F results from H12 after removing its main
c M h0*1e MM h0* 2 hn*i e(M i ) +
( ) M +n
diagonal, i.e., ni < 0
(28)
0 h L 0 hn*j e(M
M + ni + n j )
h
* *
1 L h k1 +h * 3 *
h* 0 L h*
L 0
0
ni + n j < 0
ni

1 k1 +1

F= M M M M M h*k1 (26) where e M = [ 0 L 0 1 0 L 0]

(k ) T
of length M has
M M M M M M
* * one in its k-th position. Following the analysis of the
hM* 2 hM* 3 L L 0 h*1
Appendix, we deduce that the FF filter c M , computed
hM 1 hM 2 L L h1* 0
exactly via (20), can also be expressed by (28) with a
very high degree of precision.
By assuming that F < h0 , where stands for any
2) The FB filter b N : The form of the FB
matrix norm, we can express c M by means of a Taylor
filter b N can be now obtained by combining (21),
series expansion as follows :
(23), and (28). More specifically, from (23) and (28)
c M = ( H12 ) eM
H 1 and the result of the Appendix, we get
(27)
H 2H c M h0*1h 2M h0* 2 hn*i h(2
( ) M + ni )
h0*1 ( I M h0*1F + h0* 2 F 2 ) e M +
ni < 0
(29)
+ h0* 2 hn*i hn*i h(2
where up to second-order terms have been kept in the M + ni + n j )
expansion. Due to the sparseness of the channel IR and
ni + n j < 0
the form of matrix F , c M can be directly expressed in
(k )
terms of the nonzero coefficients of the channel IR. where h 2 stands for the k-th column of H 2 . Since
H

More specifically, from (27) and the definition of e M ,

for each l with nl > 0 , hnl is the nl-th element of the
*

374
 H
last column of H 2 and matrix H 2H has a Toeplitz
form, we deduce from (21) and (29) that the FB filter
possesses approximately the following structure.
1) There are first-order (primary) nonzero taps at the
positions nl where nl > 0 is a position of a causal
component in the channel IR.
2) For each primary tap at nl > 0 , there are second-
order nonzero taps at the positions nl + ni > 0 , where
ni < 0 are positions of the anticausal components in the
channel IR.
3) For each primary nonzero tap at nl > 0 , there are
third-order terms located at nl + ni + n j > 0 , where
ni , n j is any combination of component locations
with ni + n j < 0 .
Thus, it turns out that the FB filter has a
sparse form and, hence, can be restricted to act to the
above positions only. In case strong echoes are not
present in the channel IR, a second-order
approximation of the FB filter [points 1) and 2) above]
seems to be sufficient for the proposed algorithm to
achieve a performance similar to that of the
conventional DFE. However, when there are strong
components in the channel IR (especially strong
precursor components), a higher number of taps should
be considered for the FB filter as dictated by point 3).
This results in a slight increase of the computational
complexity of the proposed algorithm. In any case, the
FB filter comprises a small number of taps and the
novel sparse equalizer offers considerable
Real
Table I
Comparison in terms of numbers of complex
multiplications
Conventional
2M + 2 N
DFE - LMS
SDFE-2 2 M + 3 log 2 ( N ) + 2 L1 ( L2 + 1) + 5
SDFE-3 2 M + 3 log 2 ( N ) + 2 L1 ( L2 + S + 1) + 5
V. SIMULATION RESULTS
The low complexity DFE algorithms have
been tested for different sparse channels (including
measured microwave channels ) and various noise
specifications. Their performance has been evaluated
for time invariant channels and also for slow time
varying channels. Some simulation result are described
below.
Fig. 1 shows a typical terrestrial HDTV channel IR.
The channel IR is the convolution of test channel D of
[17] with a square-root raised cosine filter with 11.5%
rolloff. Note the presence of four postcursor
components, including a strong far echo, and one
precursor component of relatively low magnitude. The
input to the channel is a 16-quadrature amplitude
modulation (QAM) sequence, while complex white
Gaussian noise is added to the channel output, resulting
in an SNR of 25 dB.
0.5
0

computational savings compared with the conventional

DFE. -0.5

B. Complexity Issues
The main feature of the algorithm described in -1
0 20 40 60 80 100 120 140
Section IV-A is that instead of a long FB filter, it uses a
small number of nonzero FB taps. As a result, it is
expected that its computational load will be equally 0.3

reduced compared with the conventional DFE structure.

0.2
In Table I, the computational complexity (expressed in
Im aginary

number of complex multiplications per sample) of the

0.1
studied algorithm is compared with that of the
conventional DFE, under the assumption that N is a
0
power of two. Both cases of a second [SDFE-(2)] and a
third [SDFE-(3)] order approximation of the FB filter
-0.1
are considered as analyzed in Section IV-A2. In Table 0 20 40 60 80 100 120 140
I, L1 , L2 correspond to the number of detected causal
and noncausal multipath components, respectively. S Fig.1. Multipath channel impuls response
stands for the number of pairs of locations ni , nj for
which ni + n j < 0 . It can be verified that both In Fig. 2, the performance of low complexity
variations of the new DFE have significantly lower technique (SDFE) is compared with that of the
computational complexity compared with that of the conventional DFE, for channel IR of Fig.1. In the
conventional DFE. This is so because the complexity of estimation of echo delays algorithm, a threshold of 0,02
the conventional algorithm depends linearly on N, has been set for selecting the multipath components
while the complexity of the proposed algorithm with important amplitudes. The number of training
depends on log 2 ( N ) . symbols is large enough to compare the convergence

375
and steady-state error performance. The red curve
1
correspond to the conventional DFE and the blue and 10
black curves-to the SDFE with second- and third-order DFE-LMS
approximation of the FF and FB filter, respectively. We SDFE(2)
see that both SDFEs have greater steady-state 0
SDFE(3)
10
performance and faster convergence than conventional

Mean Square Error (dB)

DFE with the same length of FF and FB. Also SDFE(3)
converge to steady-state faster than SDFE(2) and -1
depends on the estimation of echo delay algorithm 10
convergence. Figure 3. depictes constellation of
received signal (a), and equalized signal employed
-2
conventional DFE (b), SDFE-2 (c) and SDFE-3 (d) 10
after achieving convergence.

-3
10

-4
10
0 2000 4000 6000 8000 10000 12000
Number of QAM-16 symbols
Fig.2. Convergence and steady-state MSE curves of DFE-
LMS and SDFEs for the channel IR of Fig.1.

Fig.3. Constelation of received signal (a), equalized signal

DFE-LMS (b), SDFE-2 (c), SDFE-3 (d).

In order to investigate the tracking ability of
the new algorithm in a time-varying environment, we
consider the following scenario. After 7500 iterations,
the phases of all postcursor components of the channel
of Fig. 1 start continuously rotating, while their
amplitudes are kept fixed. The phase rotation step is
(0.1/360) rad per iteration. In Fig.4. we see that the new
algorithm tracks the change in the environment and as a
result the misajustment error is small, lower than in the
case of conventional DFE. The convergence and track

376
 capability of SDFE depend on the convergence of [6] P. Crespo and M. Honig, Pole-zero decision feedback
equalization with a rapidly converging adaptive IIR algorithm,
estimation of echo delay algorithm.
IEEE J. Select.Areas Commun., vol. 9, pp. 817829, Aug. 1991.
[7] N. Al-Dhahir and J. M. Cioffi, Stable pole-zero modeling of long
1 FIRfilters with application to the MMSE-DFE, IEEE Trans.
10
Commun., vol.45, pp. 508513, May 1997.
SFDE-3
[8] K. Berberidis, T. A. Rontogiannis, and S. Theodoridis, Efficient
DFE
block implementation of the DFE, IEEE Signal Processing
0
10 Lett., vol. 5, pp.129131, June 1998.
[9] K. Berberidis and P. Karaivazoglou, A block adaptive DFE in
the frequency domain based on tentative decisions, in Proc.
EUSIPCO,Rodos, Sept. 1998.
Mean Square Error

-1
10
-2
10
-3
10
-4
10
0 0.5 1 1.5
Number of 16-QAM symbols
x 10
Fig.4. MSE curve under slow time-varying CIR
VI. CONCLUSION
In this paper, the sparse form encountered in
many high-speed wireless communications channels is
properly exploited in two ways. First, the time delays of
the dominant multipath components are efficiently
estimated and continuously updated using a frequency
domain approach. Second, an approximate form of the
FB filter of the MMSE-DFE is derived. Based on the
above, a new efficient DFE algorithm is proposed,
whose FB filter comprises a reduced number of active
taps. Depending on the multipath channel conditions,
there may exist a tradeoff between the number of FB
taps used and the performance of the proposed
algorithm. In general, the new algorithm offers
considerable computational savings, faster
convergence, and acceptable tracking capabilities while
exhibiting almost identical steady-state performance in
most practical cases, as compared with the conventional
DFE. The features of the new algorithm have been
confirmed through extensive simulation tests.
[10] N. Al-Dhahir and J. M. Cioffi, Fast computation of channel-
estimate based equalizers in packet data transmission, IEEE
Trans. Signal Processing,vol. 43, pp. 24622473, Nov. 1995.
[11] S. Ariyavisitakul and L. J. Greenstein, Reduced-complexity
equalization techniques for broadband wireless channels, IEEE
J. Select. AreasCommun., vol. 15, pp. 515, Jan. 1997.
[12] R. Gupta, Kiran, and E. Lee, Computationally efficient version
of the decision feedback equalizer, in Proc. IEEE ICASSP,
Seattle, 1999, pp.12571260.
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selectable decision feedback equalization, IEEE Trans.
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2 decision feedback equalization for multipath channels with
4
large delay spreads,IEEE Trans. Commun., vol. 47, pp. 927
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separable least squares approach, in Proc. 14th Digital Signal
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377
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Terestrial Digital Video Broadcasting (DVB-T). System

performances simulation
erban Georgic Obreja1, Adrian Florin Pun1

Abstract: This paper analizes the performance of a
Terrestrial Digital Television system. The performances
are estimated using a Matlab Simulink model for the
DVB-T system.
Keywords: DVB_T system, terrestrial broadcasting,
OFDM, QAM modulation, bit error rate, channel coding
I. INTRODUCTION
DVB-T is the standard for Digital Television
Terrestrial Broadcasting defined for Europe. The
DVB family standards allows for digital video and
audio broadcasting as well as transport of multimedia
services. For terrestrial broadcasting the system was
designed to operate within the existing UHF spectrum
allocated for analogue television. The system was
developed for 8MHz channels but it can be
reconfigured also for 7or 6MHz channels. The net bit
rate available in the 8MHz channel ranges between 4
and 32 Mbit/s, depending of channel coding
parameters, modulation type and guard interval.
The Coded Orthogonal Frequency Division
Multiplexing modulation system was chosen, being
suitable for the multipath propagation environment of
terrestrial radio channels. The system uses a large
number of carriers per channel allowing the reduction
of symbol rate over one carrier. In this way the
symbol interval is increased and a better protection to
multipath propagation is obtained. The OFDM may
operate with two modes: 8k FFT mode and 2k FFT
mode. The system can select between different levels
of QAM modulation and different inner code rates
and also allows two level hierarchical channel coding
and modulation. Moreover, a guard interval with
selectable width separates the transmitted symbols,
which allows the system to support different network
configuration: 8k mode for large single frequency
networks and 2k mode for small or mobile networks
II. DVB-T System
Figure (1) shows block diagram of a DVB-T
transmitter. The input data are divided into groups of
188 bytes, which are scrambled and coded by an outer
shortened Reed-Solomon code (204,188,t=8). This
code can correct up to eight erroneous bytes in a
frame of 204 bytes. The coded bits are interleaved by
a convolutional interleaver that interleaves byte- wise
with a depth of 12 bytes and then again coded by a
rate 1/2, constraint length 7 convolutional code with
generator polynomials (171,133 octal). The rate of
this latter code can be increased by puncturing to
2/3,3/4,5/6, or 7/8. The convolutionally encoded bits
are interleaved by an inner interleaver mapped onto
QPSK, 16QAM, or 64QAM symbols.
To obtain reference amplitude and phase to
perform coherent QAM demodulation, pilot
subcarriers are transmitted. For the 8k mode, in
each symbol there are 768 pilots, so 6,048
subcarriers remain for data. The 2k mode has
192 pilots and 1,512 data subcarriers. The
position of the pilots varies from symbol to
symbol with a pattern that repeats after four
OFDM symbols. The pilots allow receiver to
estimate the channel both in frequency as well as
in time, which is important as for mobile receivers
there can be significant channel changes within a
few OFDM symbols.
Outer Outer Inner Inner
Coder Interleaver Coding Interleaver
D/A
Frame OFDM Guard
Adaptation interval Front end
Pilot &
TPS
Fig. 1 DVB-T transmission system block diagram
Terrestrial DVB use OFDM with two possible
modes, using 1,705 and 6817 subcarriers,
respectively [1]. These modes are known as 2k
and 8k modes, respectively, as these are the size
of FFT/IFFT needed to generate and demodulate
all subcarriers.. Basically, the 2k system is a

1
Politehnica University of Bucharest,Electronics and Telecommunications Faculty, Telecommunications Departament,
Bd. Iuliu Maniu Nr. 1-3, 061071,Bucuresti, e-mail serban@radio.pub.ro

378
simplified version which require an FFT/IFFT Table2
that is only a quarter of the size that is needed for 2K mode
the 8k system. Because the guard time is also Guard 1/4 1/8 1/16 1/32
four times smaller, the 2k system can handle less Interval
Duration of 2048xT
delay spread and less propagation delay symbol part 224s
difference among transmitters within a single TU
frequency network but is less sensitive to the Duration of 2512xT 256xT 128xT 64xT
Doppler effect. The FFT interval duration for the guard 56 s 28 s 14 s 7 s
8k system is 896s while the guard time can interval
Symbol 2560xT 2304xT 2176xT 2112xT
have four different values from 28 to 224 s. The
duration 280 s 252s 238 s 231 s
corresponding values for the 2k system are four TS=+TU
times smaller.
The transmitted signal is organized in frames. Each
frame has duration of TF, and consists of 68 OFDM The emitted signal is described by the following
symbols. Four frames constitute one super-frame. expression:
Each symbol is constituted by a set of K = 6 817
carriers in the 8K mode and K = 1 705 carriers in the
67 K max
2K mode and transmitted with a duration TS. It is s (t ) Re e j 2f ct c m,l ,k m,l ,k (t )
composed of two parts: a useful part with duration TU m =0 l = 0 k = K min
and a guard interval with duration . The guard
interval consists in a cyclic continuation of the useful where
part, TU, and is inserted before it. Four values of guard
intervals may be used according to table 5. The m,l ,k (t ) =
symbols in an OFDM frame are numbered from 0 to k'
67. All symbols contain data and reference j 2 TU (t lTS 68mTS )
information. = e pTS < t < ( p + 1)TS
0 else
Since the OFDM signal comprises many separately-
modulated carriers, each symbol can in turn be
considered to be divided into cells, each k - the carrier number
corresponding to the modulation carried on one l OFDM symbol number
carrier during one symbol.
m transmission frame number
In addition to the transmitted data an OFDM frame
K number of transmitted carriers
contains:
- Scattered pilot cells; TS symbol duration
- Continual pilot carriers; TU inverse of the carrier spacing
- TPS carriers. guard interval
The pilots can be used for frame synchronization, fc central frequency
frequency synchronization, time synchronization, (K + K min )
k ' = k max
channel estimation, transmission mode identification 2
and can also be used to follow the phase noise. cm,l,k complex symbol
The carriers are indexed by k [Kmin; Kmax] and The apparent complexity of these equations can be
determined by Kmin = 0 andKmax = 1 704 in 2K simplified if it is noted that the waveform emitted
mode and 6 816 in 8K mode respectively. The spacing during each transmitted symbol period depends solely
between adjacent carriers is 1/TU while the spacing on the K complex values cm,l,k which define the
between carriers Kmin and Kmax are determined by complex amplitude of the K active carriers for that
(K-1)/TU. The numerical values for the OFDM period. Each symbol can thus be considered in
parameters for the 8K and 2K modes are given in isolation; for example, the signal for the period from
tables 1 and 2. t= 0 to t=TS is given by:
Table 1
8K mode
K max
s (t ) Ree j 2f ct c 0,0,k e j 2k (t ) / TU
'
Guard 1/4 1/8 1/16 1/32
Interval k = K min
Duration 8192xT
of symbol 896s
part TU There is a clear resemblance between this and the
Duration 2048xT 1024xT 512xT 256xT inverse Discrete Fourier Transform (IDFT):
of guard 224 s 112 s 56 s 28 s 1 N 1
interval xn = X q e j 2nq / N
N q =0
Symbol 10240xT 9216xT 8704xT 8448xT
duration 1120 s 1008s 952 s 925 s Since various efficient Fast Fourier Transform
TS=+TU algorithms exist to perform the DFT and its inverse, it

379
is a convenient form of implementation to use the
inverse FFT (IFFT) in a DVB-T modulator to
generate N samples xn corresponding to the useful
part, TU long, of each symbol. The guard interval is
added by taking copies of the last N/ TU . of these
samples and appending them in front. This process is
then repeated for each symbol in turn, producing a
continuous stream of samples which constitute a
complex baseband representation of the DVB-T
signal. A subsequent up-conversion process then
gives the real signal s(t) centered on the frequency fc
The OFDM symbols constitute a juxtaposition of
equally-spaced orthogonal carriers. The amplitudes
and phases of the data cell carriers are varying symbol
by symbol according to the mapping process. The
power spectral density Pk (f) of each carrier is given
by the following expression:
We have 1704 carriers with only 1512 useful carriers.
The symbol period is Ts=280s. In every symbol we
have 1512 useful carriers and 4 bits per carrier (one
16QAM symbol). We have a convolutional code with
rate and a Red Solomon code with the ratio
204/188. It results that the useful information rate is
rD=4/2*188/204*1512*106/280=9,9529 Mbii/s.
In a superframe (4 frames of 68 symbols each) there is
an integer number of Red Solomon block (204) so
there is no need for bit stuffing.
The spectrum of the simulated 2k OFDM signal is
given in figure 3 and time representation in figure 4.

sin ( f f k )TS
2
Pk ( f ) =
( f f k )TS

The overall power spectral density of the modulated

data cell carriers is the sum of the power spectral
densities of all these carriers. A theoretical DVB
transmission signal spectrum is illustrated in figure 2
(for 8 MHz channels). Because the OFDM symbol
duration is larger than the inverse of the carrier
spacing, the main lobe of the power spectral density
of each carrier is narrower than twice the carrier
spacing. Therefore the spectral density is not constant Fig. 3 The simulated OFDM signal spectrum
within the nominal bandwidth of 7,608 259 MHz for
the 8K mode

Fig. 2 The theoretical OFDM signal spectrum

III SIMULATED SYSTEM

We have implemented in Simulink the DVBT
transmitter and receiver system. We didnt include in
our model the synchronization block. We used the
following parameters for DVBT system: code rate
for the convolutional coder, /Tu=1/4, 16 QAM
modulation and 2k mode for OFDM signal. For radio
transmission we used an AWGN channel model
combined with an multipath Rayleigh fading channel.
Fig. 4 The OFDM signal time representation
We can see that the that we have a relatively large
peak to average power ratio which brings
disadvantages like increased complexity of the A/D
and D/A converters and reduced efficiency of the RF
amplifier.
In figure 5 it is presented the received 16 QAM
constellation

380

Fig. 5 The received 16 QAM constelation
We have simulated the system behavior for different
S/N ratios. The resulted BER is presented in the
diagram in the figure 6.
Fig. 6 BER(S/N)
V. CONCLUSIONS
We have implemented in Simulink an DVBT
communications system and we have analyzed its
performance for 2k mode and 16 QAM modulation.
We intend to complete the model by implementing
also the 8k mode, and the 64QAM and QPSK
modulations.
We intend to adapt this system for the DVBH (DVB
for mobile communications) standard. The DVB
system performance study permits the
REFERENCES
[1] ETS 300 744, "Digital broadcasting systems for
television, sound and data services; framing structure,
channel coding, and modulation for digital terrestrial
television,. European Telecommunication Standard,
Doc. 300 744, 1997.
[2] R. V. Nee and R. Prasad, OFDM Wireless
Multimedia Communications, Norwood, MA: Artech
House, 2000.
381
[3] J. A. C. Bingham, "Multi-carrier modulation
for data transmission: An idea whose time has
come", IEEE Communications Magazine, vol.28, no.
5, pp. 5-14, May 1990.
[4] A. V. Oppenheim and R. W. Schafer,
Discrete-Time Signal Processing, Englewood Cliffs,
NJ: Prentice Hall, 1989
[5] Yiyan Wu, William Y. Zou, Orthogonal
Frequency Division Multiplexing: A Multi-Carrier
Modulation Scheme, IEEE Transaction on
Consumer Electronics, Vol. 41,No. 3, August 1995,
pp. 392 - 399
[6] William Y. Zou, Yiyan Wu, COFDM: An
Overview, IEEE Transactions onBroadcasting, Vol.
41, No. 1, March 1995, pp. 1 8
[7] R. R. Mosier and R. G. Clabaugh, Kineplex, a
bandwidth-efficient binary transmission system,
AIEE Transactions, Vol. 76, January 1958,
[8] Robert Chang, Synthesis of Band-Limited
Orthogonal Signals for Multichannel Data
Transmission, The Bell System Technical Journal,
December 1966, pp. 1775 -1796
[9] Robert Chang, Orthogonal frequency division
multiplexing, US. Patent 3,488445, filed November
14, 1966, issued January 6, 1970
[10] S. B. Weinstein, Paul M. Ebert, Data
Transmission by Frequency-Division Multiplexing
Using the Discrete Fourier Transform, IEEE
Transactions on Communication Technology, Vol.
COM-19, No. 5, October 1971, pp. 628 - 634
[11] Carl Magnus Frodigh, Perols Leif Mikael
Gudmundson, Adaptive channel allocation in a
frequency division multiplexed system, US. Patent
5,726,978, June 22, 1995, Issued: March 10, 1998
[12] S. OLeary, F. Ryan, B. Wynne, and C. Gilliam,
Interactive digital terrestrial televisionThe wireless
return channel and the EU sponsored WITNESS
project, IEEE Trans. Broadc., vol. 47, pp. 160163,
June 2001.
[13] R. Mhiri, D. Masse, and D. Schafhuber,
Synchronization for a DVBT receiver in presence of
co-channel interference. IEEE PIMRC-02, (Lisbon,
Portugal), Sept. 2002. submitted.
[14] P. Hoeher, S. Kaiser, and P. Robertson, Two-
dimensional pilotsymbol-aided channel estimation by
Wiener filtering, in Proc. IEEE ICASSP-97,
(Munich, Germany), pp. 18451848, April 1997.
[15] Y. Li, Pilot-symbol-aided channel estimation for
OFDM in wireless systems, IEEE Trans. Veh.
Technol., vol. 49, pp. 12071215, July 2000.
[16] M. Sandell, Design and Analysis of Estimators
for Multicarrier Modulation and Ultrasonic Imaging.
PhD thesis, Lulea University of Technology, Lulea,
Sweden, 1996.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Implementation of an OFDM Synchronizer

Ciprian Coma, Dnu Burdia, Doru Chiper1
Abstract The development of faster signal processing
components and technologies recently induced an
increased interest in Multicarrier or Orthogonal
Frequency Division Multiplexing (OFDM); this is a
multiplexing technique which converts a frequency-
selective fading channel into several nearly flat-fading
channels and combats the intersymbol interferences
(ISI) caused by multipath propagation. One of the
problems of the OFDM technique is the synchronization
of the receiver, so this paper will focus on how
synchronization can be achieved and implemented.
Keywords: OFDM, synchronization, timing offset,
carrier frequency offset.
I. INTRODUCTION
OFDM is used in some wireless communications
applications, like WLAN, being included in standards
as IEEE 802.11 (USA), ARIB MMAC (Japan),
HIPERLAN/2 (ETSI BRAN Europe). Also OFDM is
employed for Digital Audio Broadcast (DAB)
applications and known as Digital Multitone (DMT)
for broadband wireline communication systems,
namely high-bit-rate / asymmetric digital subscribers
line (HDSL / ADSL) [1]. Advantages of OFDM are
that it is bandwidth efficient and that it is rather
insensitive to frequency selective fading and timing
offset. The most important disadvantage though is that
OFDM is sensitive to carrier frequency offset (CFO).
II. SYSTEM MODEL
An OFDM symbol can be constructed as suggested in
Fig. 1 [2], [6]. First, the data to be transmitted is
mapped to a complex value Xk in the frequency
domain, according to a QAM signal constellation.
Second, the IDFT is calculated, normally using an
IFFT algorithm, to get a complex time domain OFDM
symbol
1 N 1
xn = IFFT { X k } = X k
e N
N k =0
where N is the number of subcarriers. To make
OFDM more robust against multipath and timing
offset, each symbol is extended with a guard interval
or a cyclic prefix (CP). The CP is constructed by
copying the last Ng samples of the OFDM symbol (Tes
being the sampling period) at the beginning of it.
1
Gh. Asachi Technical University of Iai, Telecommunications Department, 11 Carol I Blvd., Iai, 700506, Romania, e-mail:
ccomsa@etc.tuiasi.ro
Thus, the OFDM symbol transmitted is
x N g x N g +1 ...... xN 2 xN 1 . Finally, the time
domain signals are D-A converted, mixed with a
carrier, filtered and transmitted through the air. In the
receiver, the opposite operations are performed using
A-D conversion and DFT calculation. Since QAM
uses coherent detection, the receiver has to estimate
the phase to be able to successfully recover the
information that was sent. Coherent detection means
that all the signal alternatives are known in the
receiver. It also has to compensate for different
distorsion caused by the channel.
In order to give a mathematical description of an
OFDM system we assume a system with N
subcarriers, a bandwidth of B Hz and an OFDM
symbol length of TS seconds, of which TCP is the
length of the cyclic prefix. The spacing between
subcarriers is given by (2).
1 N
T= = = TS TCP (2)
f B
Fig. 2 illustrates the baseband OFDM model
mathematically described bellow [1], [3], [4]. Every
nth OFDM symbol of the transmission stream can be
written as a set of modulated carriers transmitted in
parallel. Relations (3) express the waveforms used in
modulation.
1
e j 2 f k (t TCP ) , t [ 0, TS )
k (t ) = TS TCP , where
0 , otherwise
N 1 1
f k = fC + k , k = 0,..., N 1 , for passband or (3)
2 T
k
f k = , k = 0,..., N 1 , for baseband echivalent
T
Note that nonzero term of k (t ) has the period
[TCP , TS ) and k (t ) has a common part (4).
N
) , for t [ 0, TCP )
2 kn
j
k (t ) = k (t + (4)
(1) B
If d n,0 ,..., d n, N 1 denotes the complex symbols,
obtained by QAM mapping of the input data stream,
the nth OFDM symbol sn (t ) is expressed by (5) and
the infinite sequence of OFDM symbols transmitted is
obtained by juxtaposition of the individual ones.

382
 N 1 Considering the channel to be fixed over the OFDM
s (t ) = sn (t ) = dk ,nk (t nTS ) (5) symbol interval, denoting it by ch( ) and taking into
n = n = k = 0
account the orthogonality condition expressed by (9),
we obtain after some mathematical operations the
output data, given by (10).
T
TCP l (t ) k (t ) = (k l )
*
(9)

ek = hk d k + nk , where
B
TCP j 2 k
hk = ch( ) e N d and (10)
0
TS
nk = n(TS t ) k* (t ) dt
TCP

By sampling the low-pass equivalent signal of (3) and

(5) at a rate N times higher than the subcarrier
symbol rate 1/T, we can obtain the discrete model of
the baseband OFDM system, where the
Fig. 1. The OFDM system model. modulation/demodulation with waves / can be
replaced with iDFT/DFT (or practically with
Assuming the impulse response ch( ; t ) of the IFFT/FFT) and the channel model with discrete-time
physical channel (possibly time variant) is restricted convolution, like in Fig. 1.
to the length of cyclic prefix [ 0, TCP ) , the
III. SYNCHRONIZATION PROBLEM
received signal becomes (6), where n(t ) is the
complex, additive and white Gaussian (AWGN) It is important to know what happens when the
channel noise. received signal is not synchronized in time and
r (t ) = ( ch * s ) (t ) =
TCP
ch( ; t )s (t ) + n(t ) (6) frequency and how to counteract this effects. The
0 synchronization must address a few issues. First, the
The filter from the receiver is matched to the last part receiver has to estimate the symbol boundaries and
[TCP , TS ) of the transmitter waveform (7), the CP the optimal timing instants that minimize the effects
being this way effectively removed in the receiver. of inter-carrier and inter-symbol interferences.
Since the cyclic prefix contains the ISI, the sample Second, the receiver has to estimate and correct for
output from the receiver filter bank contains no ISI. the carrier frequency offset of the received signal,
Also, we can ignore the time index n when calculating because the subcarriers are orthogonal only if the
the sampled output at the kth matched filter (8). receiver and the transmitter use the same frequencies.
Further, the phase information must be recovered if
* (T t ) , t [ 0, TS TCP )
k (t ) = k S (7) coherent demodulation is employed [8], [9], [13].
0 , otherwise The most challenging is the timing and frequency
synchronization, which is considered in this paper,
ek = (r * k )(t ) t =T = r (t ) k (TS t ) dt (8) neglecting the sampling frequency offset and the
S
phase tracking. The timing offset is the difference
between the estimated timing instant and the correct
IFFT timing instant. The timing offset 0 smaller than the
d n ,0 sn ,0 (t )
CP length can be seen as if each complex OFDM
k 0 (t ) 2 k 0
j
symbol is multiplied by e NTes , where k is the
d n, N 1 sn , N 1 (t ) ch( ; t ) subchannel number. The multiplication causes a
subchannel dependent rotation in the complex plane.
N 1 (t ) n(t ) As long as the timing offset is small enough this
TS rotation can be estimated and corrected in the channel
FFT estimator or in the phase tracker. Because of the
en ,0 rn (t ) differences between the oscillators in the transmitter
rn (t ) and receiver, Doppler shift, etc. each path in the
0 (t )
k received signal is affected by a carrier frequency
rn (t ) offset (CFO) f , which has the same effect as if each
en, N 1
sample n is multiplied by e j 2 nfTes , causing a time
N 1 (t ) varying rotation in the complex plane. Since a
Fig. 2. Mathematical OFDM system model. coherent system has to calculate and compensate the
phase continuously, for example by using the known

383
pilot subcarriers, the rotation can be estimated and
compensated for [11], [12], [14].
IV. SYNCHRONIZER IMPLEMENTATION
In order to estimate the CFO, in this paper a time
domain maximum likelihood (ML) estimation is
considered, using a preamble with short and long
training symbols as in the IEEE802.11 standard,
where, to help the receiver accomplish synchronized
reception, a known data sequence, the preamble, is
transmitted at the beginning of each packet. The first
half of the preamble consists of ten identical short
symbols. Each short symbol consists of 16 samples.
The second half of the preamble consists of two
identical long symbols, each 64 samples long,
preceded by a 32-sample CP. The symbols are
designed so that the correlation between two
subsequent samples is minimal.
To estimate the CFO, first the correlation between the
first and the last Ng samples has to be calculated,
where Ng is the number of the samples in the CP.
N g 1
R= r ( k ) r* ( k + N ) (11)
k =0
The estimate of the CFO is found by finding the angle
of this correlation.
1 REFERENCES
CFO = arg R (12)
2
The preamble is designed to aid the CFO estimation.
Each short training symbol can be seen as a CP to the
other short symbols, making it easy to calculate an
average. The same is the case for the long training
symbols.
Even after the CFO has been corrected, in reality,
there will always exist a small residual frequency
offset (RFO). This RFO can be used to track the CFO
by measuring the phase rotation between two OFDM
symbols. The most natural way to compensate for
CFO would be to simply feed back the estimated CFO
to the oscillator in the receiver. The compensation
can, however, be done digitally by multiplying each
incoming sample by e j 2 nfTes .
Also, by comparing continuously the correlation for
the short training symbols and the received energy
and taking into account that they should be the same
when the received symbol contains the short training
symbols, the timing estimation can be accomplish.
This method gives a packet detection solution, but a
further fine timing estimation can later be found by
correlating the incoming symbols with the known
long symbols and then try to find the maximizing
instant [7].
In Fig. 3 the structure of CFO estimator and corrector
is presented. At the start of the packet, the incoming
samples pass through rotator unchanged and into the
correlator [7]. The output from the correlator is then
transferred into the angle calculator, which computes
the argument of the complex signal. The mean is then
calculated and finally a CFO estimate is found. The
first estimate is then used to decrease the CFO before
384
the new estimate is calculated using long training
symbols.
Fig. 3. The structure of the frequency estimator and corrector.
V. CONCLUSIONS
The method presented for timing and frequency
synchronization is based on measuring the phase
difference between the cyclic prefix and the symbol
during the short and long training symbols in the
packet preample. The OFDM system has been
implemented using Matlab and Simulink [5], [10],
while the CFO estimator and corrector was
implemented and integrated in the main Matlab model
as a HDL synthesizable model [6].
The simulations show acceptable performance in an
AWGN (Additive White Gaussian Noise) channel. At
an SNR of 25 dB the mean remaining CFO after the
correction was 0.2% of the distance between two
subcarriers. In a multipath environment expected
performance degradation was observed.
[1] Coma C. R., Bogdan I., System Level Design of Baseband
OFDM for Wireless LAN, International Symposium on
Signals, Circuits and Systems, July 10-11, Iai, 2003,
Proceedings, pp. 313-316
[2] Hazy L., Initial Channel Estimation and Frame
Synchronization in OFDM Systems for Frequency Selective
Channels, http://www.sce.carleton.ca/ ~hazyl/MEng/, 1997
[3] Intini A. L., Orthogonal Frequency Division Multiplexing for
Wireless Networks, Santa Clara University of California, 2000
[4] Lawrey E. Ph., Adaptive Techniques for Multiuser OFDM,
Thesis submitted in December 2001 for the degree of PhD,
James Cook University, Australia
[5] http://www.mathworks.com/matlabcentral/fileexchange/
[6] Olsson M., A Rapid Prototyping of an IEEE802.11a
Synchronizer, Linkping University, E-press, Student Thesis
ISY 3290 2002, www.ep.liu.se/exjobb/isy/2002/3290.
[7] Olsson M., Implementation of an IEEE802.11a Synchronizer,
Proc. Of Swedish SoC Conf., Eskilstuna, April 8-9, 2003.
[8] Ramasami Vijaya Chandran, Orthogonal Frequency Division
Multiplexing, University of Kansas, USA, May 2001
[9] Richard van Nee and Ramjee Prasad, OFDM for Wireless
Multimedia Communications, Artech House, Boston, 2000
[10] Thorpe Ch., OFDM Wireless Simulation Using Simulik,
International DSP Conference, Stuttgart, May, 2001
[11] Van de Beek J. J., Sandell M., Brjesson P. O., ML Estimation
of Timing and Frequency Offset in Multicarrier Systems,
Research Report TULEA 1996:09, Division of Signal
Processing, Lulea University of Technology, 1996.
[12] Van de Beek J. J., Syncronzation and Cannel Estimation in
OFDM Systems, PhD Tesis, Division of Sygnal Processing,
Lulea University of Tecnology, 1998
[13] Van Nee R. and Prasad R., OFDM for Wireless Multimedia
Communications, Artech House Publishers, Boston, 2000
[14] Yang B., Letaief K. B., Cheng R. S., Cao Z., Timing Recovery
for OFDM Transmission, IEEE Journal on Selected Areas In
Communications, Vol. 18, No. 11, November 2000.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Windowing Techniques for OFDM Systems

Ciprian Coma, Florin Beldianu, Paul Cotae1
Abstract Orthogonal Frequency Division Multiplexing OFDM symbol by an window means the spectrum is
(OFDM) is a technique which converts a frequency- going to be a convolution of the window function
selective fading channel into several nearly flat-fading with a set of impulses at the subcarrier frequencies.
channels and combats the intersymbol interferences Applying filtering, a convolution is done in the time
(ISI) caused by multipath propagation. One of the
problems of the OFDM technique is the inter-carrier
domain and the OFDM spectrum is multiplied by the
interference (ICI) due to frequency offset and phase frequency response of the filter. When using filtering,
error. To reduce the ICI, pulse shaping and windowing extracare has to be taken not to introduce too much
techniques can be applied. To obtain an optimal rippling effects on the envelope of the OFDM
windowing function, some ICI minimization criteria, as symbols. Also, digital filtering implementation is
Lagrange, can be used or a numerical algorithm specific much more complex than windowing [13].
for adaptive filtering can be employed. This paper analysis the performance enhancements
Keywords: OFDM, windowing, pulse shaping, inter- expressed in ICI power brought by the use of some
carrier interference, raised-cosine pulse. types of pulses in pulse shaping and windowing
functions and compares the results, looking for an
I. INTRODUCTION optimal windowing function. In order to obtain this,
some ICI minimization criteria, as Lagrange, can be
Orthogonal Frequency Division Multiplexing used or a numerical algorithm specific for adaptive
(OFDM) is a multiplexing technique which converts a filtering can be employed.
frequency-selective fading channel into several nearly
flat-fading channels and combats the intersymbol II. SYSTEM MODEL
interferences (ISI) caused by multipath propagation.
That is why it is used in some wireless In order to give a mathematical description [4], [5] of
communications applications, like WLAN, being an OFDM system we assume a system with N
included in standards as IEEE 802.11 (USA), ARIB
subcarriers, a bandwidth of B Hz and an OFDM
MMAC (Japan), HIPERLAN/2 (ETSI BRAN
symbol length of TS seconds, of which TCP is the
Europe). Also OFDM is employed for Digital Audio
Broadcast (DAB) applications and known as Digital length of the cyclic prefix. The spacing between
Multitone (DMT) for broadband wireline subcarriers is given by (1), as shown in Fig. 1.
communication systems, namely high-bit-rate / 1 N
T= = = TS TCP (1)
asymmetric digital subscribers line (HDSL / ADSL). f B
Advantages of OFDM are that it is bandwidth
efficient and that it is rather insensitive to frequency
selective fading and timing offset. The most important f
N=6 subcarriers
disadvantage though is that OFDM is sensitive to
carrier frequency offset (CFO). Hence, OFDM is
sensitive to frequency offset which leads to
intercarrier interferences (ICI) and hence performance
degradation. This kind of performance degradation
and techniques for estimating the frequency offset
was discussed in some references [9], [10], [13]. The
use of pulse shaping to reduce the sensitivity of
OFDM systems to frequency offset is examined [1]-
[3], [10]-[12], while another new windowing
functions for OFDM are reported [7]. Instead of
windowing, conventional filtering techniques may be f
used to reduce the out-of-band spectrum. Windowing
and filtering are dual techniques: multiplying an Fig. 1. Subcarriers of an OFDM system.

1
Gh. Asachi Technical University of Iai, Telecommunications Department, 11 Carol I Blvd., Iai, 700506, Romania, e-mail:
ccomsa@etc.tuiasi.ro

385
Fig. 2 illustrates the baseband OFDM model Also, we can ignore the time index n when calculating
mathematically described bellow [4], [5]. Every nth the sampled output at the kth matched filter (7).
OFDM symbol of the transmission stream can be
* (T t ) , t [ 0, TS TCP )
written as a set of modulated carriers transmitted in k (t ) = k S (6)
parallel. Relations (2) express the waveforms used in 0 , otherwise
modulation.
ek = ( r * k )(t ) t =T = r (t ) k (TS t ) dt (7)
1
e j 2 f k (t TCP ) , t [ 0, TS ) S
k (t ) = TS TCP , where Considering the channel to be fixed over the OFDM

0 , otherwise symbol interval, denoting it by ch( ) and taking into
N 1 1 account the orthogonality condition expressed by (8),
f k = fC + k , k = 0,..., N 1 , for passband or (2)
2 T we obtain after some mathematical operations the
k
f k = , k = 0,..., N 1 , for baseband echivalent output data, given by (9).
T T
TCP l (t ) k (t ) = (k l )
*
Note that nonzero term of k (t ) has the period (8)

[TCP , TS ) and k (t ) has a common part (3). ek = hk d k + nk , where

N B
k (t ) = k (t + ) , for t [ 0, TCP ) (3) hk =
TCP
ch( ) e
j 2 k
N d and (9)
B 0
TS
nk = n(TS t ) k* (t ) dt
IFFT TCP
d n ,0 sn ,0 (t )
By sampling the low-pass equivalent signal of (2) and
k 0 (t ) (4) at a rate N times higher than the subcarrier
symbol rate 1/T, we can obtain the discrete model of
d n, N 1 sn , N 1 (t ) ch( ; t ) the baseband OFDM system, where the
modulation/demodulation with waves / can be
N 1 (t ) n(t ) replaced with iDFT/DFT (or practically with
TS IFFT/FFT) and the channel model with discrete-time
FFT convolution.
en ,0 rn (t ) An OFDM symbol can be constructed as follows [4],
rn (t ) [6]. First, the data to be transmitted is mapped to a
0 (t )
k complex value Xk in the frequency domain, according
rn (t ) to a QAM signal constellation. Second, the IDFT is
en, N 1 calculated, normally using an IFFT algorithm, to get a
N 1 (t ) complex time domain OFDM symbol
2 kn
N 1 j
1
Fig. 2. Baseband OFDM system model. xn = IFFT { X k } = Xke N
(10)
N k =0

If d n,0 ,..., d n, N 1 denotes the complex symbols,
obtained by QAM mapping of the input data stream,
the nth OFDM symbol sn (t ) is expressed by (4) and
the infinite sequence of OFDM symbols transmitted is
obtained by juxtaposition of the individual ones.
N 1
s (t ) = sn (t ) = dk ,nk (t nTS )
n = n = k = 0
Assuming the impulse response ch( ; t ) of the
physical channel (possibly time variant) is restricted
to the length of cyclic prefix [ 0, TCP ) , the
received signal becomes (5), where n(t ) is the
complex, additive and white Gaussian (AWGN)
channel noise.
r (t ) = ( ch * s ) (t ) =
TCP
ch( ; t )s (t
0
The filter from the receiver is matched to the last part
[TCP , TS ) of the transmitter waveform (6), the CP
being this way effectively removed in the receiver.
Since the cyclic prefix contains the ISI, the sample
output from the receiver filter bank contains no ISI.
where N is the number of subcarriers. To make
OFDM more robust against multipath and timing
offset, each symbol is extended with a cyclic prefix
(CP). The CP is constructed by copying the last Ng
samples of the OFDM symbol (Tes being the sampling
period) at the beginning of it. So, the OFDM symbol
(4) transmitted is x N g x N g +1 ...... xN 2 xN 1 . Finally,
the time domain signals are D-A converted, mixed
with a carrier, filtered and transmitted through the air.
In the receiver, the opposite operations are performed
using A-D conversion and DFT calculation.
III. PULSE SHAPING AND WINDOWING
The complex envelope of one N-subcarrier OFDM
) + n(t ) (5) block with pulse-shaping is expressed as (11), where
j = 1 , fc is the carrier frequency, fk is the
subcarrier frequency of the k-th subcarrier, p ( t ) is
the time-limited pulse shaping function and
ak,k=0,1,,N-1 is a complex-valued data symbol
transmitted on the k-th subcarrier.

386
 N 1 Fig. 3. Frequency functions Pr ( f ) , Prc ( f ) and Pbtrc ( f ) .
x ( t ) = e j 2 fc t ak p ( t ) e j 2 f k t (11)
Another family of pulses has recently been reported
k =0
[11], than are intersymbol interference-free. It is about
Equation (12) expresses the condition that the Fourier
conjugate root pulses, which are not linear phase,
transform of the pulse p ( t ) should have spectral whereas the root raised-cosine (RC) pulses are linear
nulls at the frequencies 1 T , 2 T , to ensure phase. The first-order conjugate-root pulse phase is
subcarrier orthogonality [8]. piecewise linear, while the fourth-order pulse phase is
not piecewise linear. First-order conjugate-pulses are
j 2 ( f f )t 1 , k = m
p ( t ) e k m dt = 0 , k m (12) expressible in the time domain in closed-form (17),
while fourth-order pulses have more complicated

k m forms. Both first-order conjugate-pulses and fourth-
fk fm = (13) order pulses together with root raised-cosine pulses
T are plotted in the time domain for = 0.35 in Fig. 4.
There are considered here three time-limited-pulses,
which are Pr ( f ) , Prc ( f ) and Pbtrc ( f ) denoting the t t
sin cos
rectangular pulse, the raised-cosine pulse (in the time- p (t ) = Tes Tes (17)
domain) and the better than raised cosine (BTRC) t
pulse (in the time-domain), defined as (14), (15) and ( 2 t + T es )
Tes
(16), where is the roll-off factor and 0 1
[12]. When = 0 both raised-cosine and the BTRC
pulse coalesce into the rectangular pulse. The Fourier
transforms are denoted by Pr ( f ) , Prc ( f ) and
Pbtrc ( f ) respectively. Fig. 3 shows the frequency
functions of these pulses for = 0.2 and = 1 .
1 , T t T
pr ( t ) = T 2 2 (14)
0 , otherwise
1 T (1 )
, 0 t
T 2
1 T (1 ) T (1 ) T (1 + )
prc = 1 + cos
T
t , t (15)
2T 2 2 2

0 , otherwise

1 T (1 )
, 0 t
T 2
2 ln 2 T (1 )
1 e T
t
2
T (1 ) T
T , t
pbtrc ( t ) = 2 2 (16) Fig. 4. Time-domain plots of the RC and conjugate-root pulses for
2 ln 2 T (1+ )
t

T (1 + ) = 0.35
1 1 e T 2 ,
T
t
T 2 2

IV. PERFORMANCES OF PULSE SHAPING

0 , otherwise
Frequency offset, f ( f 0 ) , and phase error, ,
are introduced during transmission because channel
distortion or receiver crystal oscillator inaccuracy.
The received signal after multiplication by
e(
j 2 ( f c +f )t + )
becomes (18).
N 1
r (t ) = e (
j 2ft + )
ak p ( t ) e j 2 fk t (18)
k =0

r (t ) e
j 2 f m t
a$ m = dt =


= am e j p ( t ) e dt +
j 2ft
(19)

N 1
+e j ak p ( t ) e j 2 ( f k f m +f )t
dt
k =0
k m

387
The m-th subchanel correlation demodulator, thus, types of pulses and it concluded that the employment
gives the decision variable for transmitted symbol am of the better than raised-cosine pulse rather than the
raised-cosine pulse gives a substantial improvement in
in (19), where the first term contains the desired
signal component, and the second term is the ICI. the reduction of ICI caused by frequency offset in an
OFDM system. However, further work has to be done
Combining (13) with (19) gives (20), where P ( f ) is in comparison those pulse shapes with another
the Fourier transform of p ( t ) . Hence, the power of windowing functions, reported to the distortion
the desired signal is (21) and the ICI power is (22). introduced and to the implementation complexity.
N 1 Also, another pulse shapes may be found using a
mk
a$ m = am e j P ( f ) + e j ak P f (20) method of parametric construction of ISI-free pulses,
k =0 T based on some optimization criteria (as Lagrange
k m
method) of some properties of interest, like ICI.
2
P ( f )
2
m = am (21)
REFERENCES
N 1 N 1
k m nm
ICI
m
= ak an* P
T
+ f P
T
+ f (22)

[1] Beaulieu N. C. and Cheng J., Precise error rate analysis of
k =0 n =0 bandwidth efficient BPSK in Nakagami fading and co-channel
k m nm
interference, IEEE Trans. Communications., vol. 52, pp. 149
The ICI power depends not only on the desired 158, Jan. 2004.
symbol location, m, and the transmitted symbol [2] Beaulieu N. C. and Damen M. O., A parametric construction of
sequence, but also on the pulse-shaping function and ISI-free pulses, Proc. of 8th Canadian Workshop on Information
Theory, 2003, pp. 121124
the number of subcarriers. However, (22) gives the [3] Beaulieu N. C., Tan Ch. C., Damen M. O., A Better Than
average ICI power, averaged across different Nyquist Pulse, IEEE Communications Letters, Vol. 5, No. 9,
sequences as September 2001
N 1 2 [4] Coma C. R., Bogdan I., System Level Design of Baseband
k m
ICI
m
= P
T
+ f

(23) OFDM for Wireless LAN, International Symposium on Signals,
Circuits and Systems, July 10-11, Iai, 2003, Proceedings, pp.
k =0
k m 313-316
[5] Edfors O., Sandell M., Van de Beek J. J., An Introduction to
For the same value of the BTRC pulse outperforms Orthogonal Frequency-Division Multiplexing, 1996
the others, including raised cosine (RC) pulse, as [6] Intini A. L., Orthogonal Frequency Division Multiplexing for
shown in Fig. 5 for = 1 [12]. This interesting Wireless Networks, Santa Clara University of California, 2000
[7] Lawrey E. Ph., Adaptive Techniques for Multiuser OFDM,
behavior occurs despite the fact that the tails of Thesis submitted in December 2001 for the degree of PhD,
Pbtrc ( f ) and Prc ( f ) decay as f 2 and f 3 , James Cook University, Australia
[8] Proakis J. G., Salehi M., Communication Systems Engineering -
respectively. Second Edition, Prentice Hall, 2002
[9] Ramasami Vijaya Chandran, Orthogonal Frequency Division
Multiplexing, University of Kansas, USA, May 2001
[10]Sun Yi, Bandwidth-Efficient Wireless OFDM, IEEE Journal on
Selected Areas in Communications, vol. 19, no. 11, Nov. 2001
[11]Tan Ch. C., Beaulieu N. C., Transmission Properties of
Conjugate-Root Pulses, IEEE Transactions on
Communications, Vol. 52, No. 4, April 2004
[12]Tan P., Beaulieu N. C., Reduced ICI in OFDM Systems Using
the Better Than Raised-Cosine Pulse, IEEE Communications
Letters, Vol. 8, No. 3, March 2004
[13]Van Nee R. and Prasad R., OFDM for Wireless Multimedia
Communications, Artech House Publishers, Boston, 2000

Fig. 5. The ICI power for different pulse shaping functions.

One can also consider the comparative performances

of different pulses in terms of the average signal
power to average ICI power ratio, denoted SIR. It will
be shown that the BTRC pulse also outperforms the
other pulses considered.

V. CONCLUSIONS

This paper analyzed the performance enhancements

expressed in ICI power brought by the use of some

388
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

BER Performances of a Differential OFDM System in

Fading Channels
Marius Oltean1, Eugen Mrza2, Miranda Naforni3
Abstract In an OFDM system, various modulation
methods can be used in order to encode the binary
information. If a differential phase modulation scheme is
chosen, data can be encoded in the relative phase of
consecutive symbols in each subchannel or in the relative
phase of symbols in the adjacent subchannels. The two
methods exhibit two essentially different behaviors in
fading conditions.
In this paper, we shall investigate the BER
performances of both modulation types. The
performance evaluation is based on computer
simulation. We will consider a multipath fading channel,
as met in mobile communication systems.
Keywords: OFDM, differential, fading
I. INTRODUCTION
OFDM (Orthogonal Frequency Division
Multiplexing) is one of the most promising
modulation techniques that were proposed for being
used in the 4th generation wireless systems. In a
typical mobile radio channel the transmitted signal is
subjected to multipath fading which generally exhibits
time selectivity (commonly referred to as Doppler
effect) and frequency selectivity [1], [2]. The
influence of the ISI (Inter-Symbol Interference) can
be reduced by increasing the duration of the
transmitted symbol. Using OFDM, the high-rate data
sequence to be transmitted is split into a large number
of lower speed symbol streams, each of them
modulating a different carrier. The carrier spacing is
selected such that all carriers used are orthogonal each
other over a symbol interval. As it is well known, the
orthogonal signals can be separated at receiver by
correlation techniques. In addition, a cyclic prefix (a
copy of the last several samples of an OFDM
frame) is inserted at the beginning of each OFDM
symbol, in order to counteract the inherent time-
dispersive nature of the channel, preventing two or
more symbols to interfere each other [3], thus
inducing ISI. The cyclic prefix gives also an
appearance of periodicity or circularity to the
1
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail marius.oltean@etc.utt.ro
2
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail eugen.marza@etc.utt.ro
3
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail miranda.nafornita@etc.utt.ro
signal, facilitating the equalization process to the
receiver.
The lengthening of the symbol duration,
introduced in order to combat the frequency-
selectivity is however limited by the time-variant
nature of the channel that generates the Doppler
effect. Larger the symbol duration, higher the
probability that the channel parameters vary during
the transmission of an OFDM frame giving rise to
frequency offsets of the carriers, thus destroying their
orthogonality and generating inter-carrier interference
(ICI).
The transmitter and receiver for OFDM can be
efficiently implemented using Fast Fourier Transform
(FFT), a rapid mathematical algorithm of processing
Discrete Fourier Transform (DFT).
The data symbols that modulate multiple
orthogonal carriers in OFDM are obtained using a
classical digital modulation scheme. Various
modulation methods could be employed such as
BPSK, QPSK (also with their differential form) and
QAM with several different signal constellations.
If a differential phase modulation is chosen (the
particular case of an OFDM-DBPSK system) there
are two options to perform it. Thus, data can be
encoded in the relative phase of consecutive symbols
in each subchannel (corresponding samples in
adjacent OFDM symbols, or frames), obtaining an
inter-frame differential modulation. On the other
hand, data can be encoded in the relative phase of
symbols in adjacent subchannels (consecutive
samples of an OFDM symbol), achieving an in-frame
differential modulation. The two methods exhibit two
essential different behaviors in fading conditions.
In this paper we realize a performance
comparison of the two methods, rarely reported in the
literature, focusing on the differences between them.
The performance evaluation of both methods is based
on computer simulation. We will consider a multipath
fading channel, as met in mobile communications
systems.

389
 caused by the multipath fading is indicated by the
II. SYSTEM DESCRIPTION AND FADING Doppler spectrum. In the second case, the channel
CHANNEL MODEL multipath intensity profile and the length of the
OFDM symbol indicate the phase change rate due to
In OFDM, the available bandwidth is partitioned fading conditions. Therefore, the two methods exhibit
into N subchannels. The desired high-rate symbol essentially different behaviors although both encode
stream is achieved by simultaneously transmitting N the data differentially. Another major difference
slower rate substreams using N orthogonal between the two methods (that gives them their
subcarriers. The binary data to be transmitted is specific name) is that the consecutive OFDM symbols
differentially encoded using a DBPSK modulation are interconnected through differential encoding in the
scheme, obtaining a sequence of complex data first case (inter-frame differential modulation), while
symbols (fig. 1). no successive OFDM symbols connection is realized
by the second method (in-frame differential
modulation), where the differential encoding is
DBPSK si , n Cyclic prefix
IFFT performed on the samples belonging to the same
Binary Modulation insertion frame (or to the same future OFDM symbol), as
Information illustrated in the fig. 2b.
Channel

After the differential encoding of the binary
s i,n Cyclic prefix message using one of the two methods presented
Decision FFT removal above, the sequence si,n is obtained, si,n denoting the n-
Estimated
Information TS th symbol of the i-th frame,
k
N where (0 n N 1, < i < ) . The n-th carrier is
Fig.1: The OFDM transceiver model
modulated by the samples {s i,n , < i < } and the
There are two possibilities to perform differential
modulation in the presented OFDM scheme. Data can modulated carriers (orthogonal one-another) are
be encoded in the relative phase of adjacent symbols added together to form the OFDM symbol to be
in each subchannel (correspondent samples in two transmitted. In a practical implementation, the N
consecutive OFDM symbols) or in the relative phase samples of the OFDM symbol corresponding to the i-
of samples transmitted in adjacent subchannels, that is th frame are generated by processing {si,n }stream
consecutive samples of an OFDM symbol (see fig. 2). using the fast implementation of the Inverse Discrete
Since the IFFT block accepts N parallel samples to its Fourier Transform (IDFT) (see fig. 1). In order to
entry, the whole difference of the two methods can be combat the inter-symbol and inter-carrier interference
thought as follows: if the phase modulation is introduced by the frequency selectivity and the time
separately achieved on each of the N parallel streams selectivity of the radio channel, each OFDM symbol
that constitute the entry to the IFFT block, then we are is preceded by a cyclic prefix of L samples. The
in the case of the first presented modulation type, cyclic prefix is a circular extension of the time
namely an inter-frame modulation is performed (see domain samples, being obtained by copying the last L
fig. 2a). If the modulation is made on the serial samples of the OFDM symbol in the front of it. The i-
stream, prior to the parallel conversion required by th transmitted symbol (including the prefix) contains
IFFT, then an in-frame modulation is chosen, since N N+L time domain samples, of which the m-th sample
consecutive serial samples will simultaneously is given by the equation below:
modulate N orthogonal carriers, forming an OFDM
nm
symbol (fig 2b). E S N 1 j 2
g i ( m) = si ,n e N
, m = L,..., N 1
Serial stream, N + L n =0
consecutive frames S/P (1)
IFFT
conversion
Correspondent Assuming the data symbols are statistically
samples OFDM symbol
a) Relative phase independent and having a unit average energy, the
coding transmitted average energy per symbol equals ES. The
transmitted signals can be expressed in complex form
I as:
S/P F
conversion
F s(t)= p(t iTS ) g i (t ) (2)
T i =
Relative phase
coding
b) where gi(t) represents the analogical waveform
OFDM symbol
corresponding to the OFDM symbol, obtained after a
Fig.2: (a) Inter-frame modulation (b) In-frame modulation DAC conversion of the sequence {gi(m)},
Both methods have an irreducible error rate because m=0,1,,N-1. p(t) is the pulse-shaping waveform of
of the random change of the relative phase, caused by each symbol, defined as:
the fading channel. In the first method the distortion

390
 1, for t t s power of the two multipath components are
p(t ) = (3) considered for channel simulation. The BER
0, otherwise
computation was averaged over 20000 transmitted
OFDM symbols. Neither channel coding nor further
equalization to the receiver were considered at this
TS =+ts stands for the total duration of an OFDM stage. A comparison of the two methods is made,
symbol, composed by the cyclic prefix period () and studying the influence of the block length N, of the
by the observation period (ts). The fading channel channel multipath delay spread and of the Doppler
(assuming Rice conditions) can be modeled as a 3-ray shift introduced by the time-variant character of the
tapped delay line with one line-of-sight (LOS) path channel on the BER performance in both in-frame and
and two multipath components. If h(t,) denotes the inter-frame DBPSK-OFDM system. We emphasize
channel impulse response at time t-, it can be the essential different behaviors of the two methods
expressed as: with respect to the parameters presented above.
The BER performances of the DBPSK-OFDM
h(t , ) = 2 PS ( ) + P1 a1 (t ) ( 1 ) + system in a Rayleigh fading channel, as a function of
(4) the normalized delay of the second multipath 2/T is
P2 a 2 (t ) ( 2 )
illustrated in the figures 3,4.

where PS is the power of LOS signals, P1 and P2 are

the powers of multipath replicas. The channel impulse
response above can be viewed as a summation of the
LOS deterministic signal and two attenuated (a1(t) and
a2(t) are independent time-varying complex Gaussian
random processes with maximum Doppler shift fd,
accounting for these attenuations) and delayed
replicas (1 and 2 represent the delays of the two
multipath components). An important parameter
characterizing the Rician fading channel is the Rice
factor, defined as the ratio of the deterministic LOS
component power Ps and the multipath components
power Pm=P1+P2, i.e. K=Ps/Pm. As a special case, the
channel is AWGN when K, while Rayleigh Fig. 3: BER performances of inter-frame and in-frame
fading conditions are met for K=0. modulation in a Rayleigh fading channel, SNR=40dB,
The received signal can be written as [4]: P1/P2=0dB, N=32
In the figure 3, one can observe that the inter-frame

modulation is significantly more sensitive to the
r (t ) = s (t ) h(t , )d + n(t ) (5) Doppler shift than the in-frame modulation. Thus, at
0
the two normalized Doppler shifts (fD*TS) taken into
account, the performance of the in-frame DBPSK
where n(t) is a complex Gaussian noise and h(t,) is
system is almost identical, especially for an important
the impulse response of the multipath fading channel
multipath delay of the channel. On the other hand,
at the time t-. At the receiver, after doing FFT to the inter-frame modulation performs significantly better
signal, the output of each m-th subchannel can be for low values of the maximum Doppler shift, proving
obtained as:

t S + iTS
1 j 2f ( t iTS )
rm ,i =
tS
{ s (t )h(t , )d + n(t )} e D
dt
iTS 0

(6)
Finally, the differential detector decides what symbol
was transmitted.

III. SIMULATION RESULTS AND DISCUSSIONS

The BER performance of an OFDM system with

both DBPSK in-frame and inter-frame modulation
was studied by the means of computer simulation. To
simplify the computer implementation, the cyclic
prefix duration is considered to be equal to the serial
Fig. 4: BER performances of inter-frame and in-frame modulation in
symbol duration, i.e. T= for all the simulations. a Rayleigh fading channel, SNR=40dB, P1/P2=0dB,
Two-ray Rayleigh fading conditions, with equal N=32

391
a sensitivity of this method to the time-variant channel
character. The same observation becomes more
obvious regarding the figure 4, where another
difference between the two Doppler shifts is taken
into account.
The BER performance of the inter-frame
DBPSK-OFDM system for three different Doppler
shifts is plotted in the figure 5, in order to stress the
effectiveness of this parameter. It is shown that the
maximum Doppler shift has a significant influence on
the BER, especially when the delay of the second
multipath is small. For large delays of the second
multipath the main amount of errors is brought by the
ISI introduced by the multipath components, which
confirms the conclusion in [5], respectively in [4] for
an in-frame DBPSK modulation.
IV. ABOUT REFERENCES
References should be numbered in a simple form [1],
[2], [3], and quoted accordingly [1]. References are
not allowed in footnotes. It is recommended to
mention all authors; et al. should be used only for
more than 6 authors.
Table 1
Parameter Value Unit
I 2.4 A
U 10.0 V
Fig. 5: BER performances of inter-frame DBPSK modulation in a
Rayleigh fading channel, SNR=40dB, P1/P2=0dB, N=32
In the figure 6, we plotted the BER performance of
both inter-frame and in-frame OFDM-DBPSK
systems with respect to the average SNR per bit and
the parameter used is still the normalized maximum
Doppler shift (fD*TS).
Fig. 6: BER performances of inter-frame and in-frame DBPSK
modulation in a Rayleigh fading channel, 2/T=1, P1/P2=0dB, N=32
The two inner curves correspond to in-frame
modulation, showing the little sensitivity of the
392
method to the variation of the maximum Doppler shift
parameter comparing to the inter-frame modulation
(whose performance is indicated by the two outer
curves). If at low maximum Doppler shift inter-frame
modulation performs better, once the value of this
parameter grows, the in-frame modulation method
becomes more efficient. It can also be observed that at
significant Doppler shifts, the two methods exhibits a
very poor improvement of BER performance, with
respect to SNR, especially for the inter-frame
modulation.
The influence of the block length N on the BER
performance in both modulation types is illustrated in
the figure 7, where maximum Doppler shift is
considered to be constant. As stressed in [6], the in-
frame OFDM-DBPSK system significantly improves
its performance when the block (or, equivalently, the
OFDM symbol) length increases, considering the
same multipath delay spread of the channel impulse
response. On the other hand, it turns out that the BER
performance of inter-frame modulation is almost
identical for the different values chosen for the
parameter N. In the three simulated situations
(N=16,32,64), the system performs to within 1-2dB
spread of the results. It can be asserted that the
performance obtained using this modulation type is
very little sensitive to the OFDM symbol duration.
Fig. 7: BER performances of inter-frame and in-frame
DBPSK modulation in a Rayleigh fading channel, 2/T=1,
P1/P2=0dB, fD=0.0001
Figure 8 shows the BER performance evaluated
over a range of normalized delay spreads of the
channel impulse response. Unlike the previous
evaluation, inter-frame modulation improves its
performance significantly when the OFDM symbol
length N increases. The same expected pattern is
exhibited by the in-frame modulation method. A
comparative analysis illustrates that in-frame
modulation is slightly more sensitive to the parameter
N than the inter-frame modulation, when the
performance is evaluated against the normalized delay
of the second multipath. At small values for the delay
of the second reflected path (comparable with the
serial symbol duration) no correlation between the
frame length and the BER performance can be
asserted, since the system performs better for N=16
than for N=64.
Fig. 8: BER performances of inter-frame and in-frame DBPSK
modulation in a Rayleigh fading channel, SNR=40dB, P1/P2=0dB,
fD=0.0001
In the figure 9, the effect of multipath delay
spread on the both methods is studied. The maximum
Doppler shift was kept constant while the BER
performance against signal-to-noise (SNR) ratio was
plotted for two different values of the normalized
delay of the second multipath component.
Considering a small value of the mentioned
parameter, the in-frame OFDM-DBPSK system
performs to about 18dB better than the inter-frame
system. To lie in such a situation, a correspondent
value was chosen for the normalized Doppler shift
(fD*TS=0.025). When a three times bigger value was
considered for the normalized delay of the second
multipath component inter-frame modulation
performed better, despite the significant value of the
Doppler shift. One can conclude that in-frame
modulation is more sensitive to the multipath delay
spread introduced by the inherent dispersive nature of
the radio channel.
Fig. 9: BER performances of inter-frame and in-frame DBPSK
modulation in a Rayleigh fading channel, P1/P2=0dB,
fD*TS=0.025, N=32
393
IV. CONCLUSIONS
In this paper we have studied the BER performance of
an OFDM-DBPSK system with two distinct phase
modulation types. The principles of both in-frame and
inter-frame modulation in an OFDM transmission
scheme were briefly exposed, accentuating on their
differences. The essential different behavior in
multipath fading conditions was emphasized by
means of computer simulation. The inter-frame
modulation system, while generally performing better
has though shown to be more sensitive at the variation
of the Doppler shift parameter. The in-frame
modulation method allows significant performance
improvement by increasing the data-block length. The
multipath delay spread degrades the BER
performance of both studied modulation types. Even
if at delay spreads that significantly exceed the cyclic
prefix duration the performance is only slightly
improved increasing the signal-to-noise ratio, the
inter-frame modulation proved to be more resistant to
the inter-symbol interference introduced by the
multipath delayed components.
REFERENCES
[1] B. Sklar, Rayleigh Fading Channels in Mobile Digital
Communication Systems- Part I: Characterization, IEEE Commun.
Mag., July 1997;
[2] B. Sklar, Rayleigh Fading Channels in Mobile Digital
Communication Systems- Part II: Mitigation, IEEE Commun.
Mag., July 1997;
[3] J.A.C. Bingham, Multicarrier Modulation for Data
Transmission, An Idea Whose Time Has Come, IEEE
Communication Magazine, Vol. 31, No. 5, May 1990
[4] Lu, J. , Tjhung, T.T., Adachi, F., BER performance of OFDM
system in frequency-selective Rician fading with diversity
reception, available online:
http://www.cwc.nus.edu.sg/~cwcpub/zfiles/ict97_2.pdf.
[5]Lupea, E., Bianu, M., Slgean, M., Oltean, M., Naforni, M.,
BER Performance of Frequency Selective Channels with Cyclic
Prefix Based Equalizers, Buletinul tiinific al Universitii
Politehnica Timioara, Tom 47 (61), Fascicola 1-2, 2002
[6]Chini, A., Analysis and Simulation of Multicarrier Modulation in
Frequency Selective Fading Channels ,Ph. D. Thesis, 1994, Chapter
3
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Optimal Chaotic Asynchronous DS-CDMA

Communications over Frequency-Nonselective Rician
Fading Channels
Clin Vldeanu1, Radu Lucaciu2, Dorin Andrei3
Abstract The use of some chaotic sequences is a new
approach for spread-spectrum systems, ofering better
BER (Bit Error Rate) performances and system capacity
than conventional sequences. This paper presents some
optimal sets of chaotic sequences for the A-DS-CDMA
(Asynchronous Direct-Sequence Code-Division Multiple
Access) system under the SGA (Standard Gaussian
Approximation) condition. The A-DS-CDMA
communication over frequency-nonselective Rician
fading channel with AWGN (Additive White Gaussian
Noise) is considered.
Keywords: A-DS-CDMA systems, BER, SGA,
Chebyshev chaotic sequences, Rician fading channel.
I. INTRODUCTION
In A-DS-CDMA system using the single user
matched-filter receiver the average BER performances
depend mainly on the correlation properties of the
spreading sequences assigned to the users. Hence, the
problem of guaranteeing minimum performances to
each user is equivalent to choosing spreading
sequences with respect to an adequate correlation
criterion.
Most DS-CDMA systems presented to date have used
binary PN sequences, including Gold sequences and
Kasami sequences, which prove some quasi-
orthogonality correlation properties. These sequences
present crosscorrelation values that depend on the
generator polynomial degree [1].
A new family of spreading sequences is represented
by chaotic sequences generated from the orbits of
some dynamical discrete systems. These sequences
present noise-like features that make them good for
DS-CDMA systems. A single system, described by its
discrete chaotic map, can generate a very large
number of distinct chaotic sequences, each sequence
being uniquely specified by its initial value [2]. This
dependency on the initial state and the nonlinear
character of the discrete map make the DS-CDMA
system using these sequences more secure.
It was shown that sequences generated by chaotic
Chebyshev polynomial maps are exact thus mixing
and ergodic, and show good correlation properties [6],
[7], and [9].
This paper is organized as follows. The second
paragraph is presenting the BER estimation for the A-
DS-CDMA system for optimal spreading sequences
and perfectly random (white) spreading sequences,
assuming a frequency non-selective fading channel
with AWGN noise. The third part of the paper
describes the design method for optimal sets of
chaotic sequences for the A-DS-CDMA system based
on the ergodicity of the dynamical systems with
Chebyshev polynomial maps. The fourth part presents
some simulation results compared to the theoretical
average values for both optimal and white sequences.
Finally, some conclusions are taken.
II. OPTIMAL BER COMPUTATION FOR A-DS-
CDMA SYSTEM AND RICIAN NONSELECTIVE
FADING CHANNEL
According to [3], [4], [4] and [9] the overall (non-
faded) interference variance for the desired ith user
from all other users in an A-DS-CDMA system can be
computed as:
PT 2 K
(i ) =
2
A 3
rk ,i , (1)
12 N k =1
k i
where P represents the common signal power, T is the
users data signal symbol duration, N is the spreading
factor, which is also identical to the period of the
spreading sequences, N = T / TC , TC is the chip
interval duration, rk ,i is the interference term

1
Politehnica University of Bucharest, Electronics and Telecommunications Faculty, 1-3,
Iuliu Maniu Bvd., Bucharest, Romania; Phone: +4021 4024765; E-mail: calin@comm.pub.ro
2
Technical University of Timisoara, Electronics and Telecommunications Faculty, 2,
Vasile Parvan Bvd., Timisoara, Romania; Phone: +40256 403325; E-mail: lucaciu@etc.utt.ro
3
Military Technical Academy, 81-83, George Cobuc Bvd., Bucharest, Romania;
Phone: +4021 3354665/ext. 0308; E-mail: dandrei@mta.ro

394
corresponding to the interfering user k, and K is the
total number of users.
The interference term rk ,i is written in terms of the
autocorrelation function C k as [3], [4]:
N 1
rk ,i = 2C k (0)C i (0) + 4 C k (l )Ci (l ) +
l =1
N 1
+ [C k (l )Ci (l + 1) + Ci (l )C k (l + 1)]
l =0
It is known that when perfectly random sequences
(white noise-like sequences) are employed the overall
interference variance from (1) can be written as [3],
[6]:
PT 2 (K 1)
A2 , white (i ) = , (3)
6N
where the normalization Ci (0) = 1 was considered.
According to [6] and [7] assigning the same
interference variance to every user can attain the
lower bound of average BER for all the users. This
can be done with C k (l ) = C i (l ) = C (l ) , for all k,i, and
l. With the normalization Ci (0) = 1 , the solution that
minimizes the interference power in (1), considering
the expression (2) for each term in the sum, leading to
minimum BER under the SGA assumption is given
by:
r l N r N l
C k (l ) = ( 1)l , l = 0,1, 2, ... , N 1, k (4)
r N r N
where r = 2 3 .
Note that when l << N , C k (l ) ( r )l , which decays
exponentially with alternate sign. By introducing
relation (4) into (2), the minimum interference power
is obtained for user i:
PT 2 (K 1) 3 r 2 N r 2 N
A2 , optimum (i ) =
12 N r 2 N + r 2 N 2
For large values of N, the second term in (5) rapidly
decreases to 1. It is important to note that the second
term in (5) is very close to 1 even for N=5 (it differs
from 1, starting with the sixth decimal value). With
this approximation the minimum interference variance
is:
PT 2 (K 1) 3
A2 , optimum (i ) = , (6)
12 N
Comparing the optimum case with the case when
white sequences are employed, the first one offers an
increase in the system BER performances which
increases the number of users accommodated for the
same mean BER, by
K optimum
1.1547 , (7)
K white
for large numbers of users. It is obvious from (7) that
the optimum case increases the number of users by
, (2)
more than 15% than the white sequences case, for the
same mean BER.
The output signal of a Rician nonselective fading
channel is the sum of a non-faded version of the input
signal (specular component) and a non-delayed
Rayleigh faded version of the input signal (scatter
component). All communications links are assumed to
fade independently. We also assume that all users
have the same faded power ratio 2 .
Under the SGA assumption, the average BER for any
user i over the Rician fading channel is given by [6],
[7], [8]:
P
T
BERA (i ) = Q 2
, (8)
( )
2
2 PT + 1 + 2 2 (i ) + 2
A n
4
where A2 (i ) is the overall (non-faded) interference
power for the desired ith user from all other users in
N T
an A-DS-CDMA, n2 = 0 is the variance for the
4
additive Gaussian noise with two-sided PSD (Power
Spectral Density) N0/2 [3], [5], and the numerator is
the is the useful component (the desired contribution
from any user i). The Q-function is given by
1
2
Q( z ) = ey
2
dy .
z 2
The theoretical BER for both optimum and white
sequences is presented in Fig. 1 for the following
system parameters values: K {3, 5, 7, 9}, N =31 and
2 = 0.1 .
, (5)
III. OPTIMAL SETS OF CHAOTIC SEQUENCES
FOR THE A-DS-CDMA SYSTEM
A second-order time-averaged statistic of the
spreading sequences is needed for the sequence
design and performance analysis. According to the
ergodic theory the autocorrelation function of a
sequence generated by a measure-preserving and
ergodic transformation can be estimated statistically
[6].
An example of ergodic transformation is the nth
degree Chebyshev polynomial defined by:
Tn (x ) = cos(n arccos(x )) , (9)

395
 where x takes values from the interval [-1, 1]. It was
shown that the Chebyshev polynomials of degree 1 N N 1
n 2 are mixing and thus ergodic and they have an C (0 ) = P 2 (xi ) P (x ) (x )dx =
2
N i =1
invariant measure. 1
(11)
Performance of asynchronous DSCDMA (N=31), over a fading AWGN channel, 2 = 0.1
=
2
1 r 1 ( r 2 N not )= A
( )
-1
10
2 1 r2

and the normalized autocorrelation function of such a

-2
sequence is given by
10

C (l ) 1 1 N N
R
E A
=
A N l
P(xi )P(xi+l )
B i =1

( )
Th. wh. fad. BER K=3 1
-3
1
P(x )P T p l (x ) (x )dx =
10

Th. wh. fad. BER K=5
Th. wh. fad. BER K=7 (12)
Th. wh. fad. BER K=9 A 1
(r ) , for finite l
Th. opt. fad. BER K=3
lN
r N l
( 1)l
Th. opt. fad. BER K=5

(r )
Th. opt. fad. BER K=7
N
10
-4 Th. opt. fad. BER K=9 r N

0 5 10 15 20 25
Eb/N0(dB)
It is easy to note that the average value of the left-side
Fig.1. The theoretical BER performances for optimum
sequences and white sequences A-DS-CDMA system over C (l )
frequency-nonselective Rician fading channel with AWGN.
term, E x0 , is equal to the right-side term in
A
The common faded power ratio is = 0.1 .
2

(12), even for finite N if the initial condition x0 is

The dynamical systems given by Chebyshev
polynomial maps are from a special class of
dynamical systems with Lebesque spectrum [6]. The
Lebesgue spectrum systems denoted as (x ) are
associated with a special set of orthonormal basis
{ }
functions f , j (x ) , j F for Hilbert space L2 ,
where labels the classes and j labels the functions
within each class. These particular basis functions
f , j (x ) have an important property such that,
f , j o = f , j +1 , , j F . This property states
that all other basis functions in the same class can be
generated from one basis function by using
compositions with powers of the dynamical system
(x ) .
For the particular case of Chebyshev polynomial
maps, we consider the pth degree polynomial map,
i.e., (x ) = T p (x ) , where p 2 is prime. The
associated basis functions for L2 {[ 1,1]} are also
Chebyshev polynomials {Ti (x )}i=0 .
Let us consider the polynomial function P(x ) in the
Hilbert space L2 {[ 1,1]} with the following
expression:
N
P(x ) = ( r ) jT p (x ),
j x [ 1, 1]
j =1
By using the ergodic theory the average of P 2 (x ) for
a sequence generated by a Chebyshev transformation
T p (x ) is given by
selected randomly according to the corresponding
invariant measure.
Considering the same value for the parameter
r = 2 3 as in (4) the output sequences
{y1 , y2 ,..., y N } generated by
1
yi = P(xi ), xi +1 = T p (xi ) (13)
A
are the optimal spreading sequences for the A-DS-
CDMA system. Hence, each user is assigned a
different spreading sequence generated by the same
Chebyshev map but having a different initial
condition or generated by a different-degree
Chebyshev map.
IV. SIMULATION RESULTS
The A-DS-CDMA system using optimal Chebyshev
polynomial maps of degree p = 3 and Gold
sequences generated by primitive polynomials of
degree n = 5 , having the period N = 2 n 1 = 31 , was
considered.
The estimated average BER was evaluated for K = 10
users and the energy-per-bit to noise DSP ratio
Eb / N 0 taking values from 0 to 30 dB. The common
(10) faded power ratio is taking the value 2 = 0.1 .
The resulting BER as a function of the ratio Eb / N 0
is depicted in Fig. 2.
The A-DS-CDMA system capacity is also an
important parameter to measure. The average BER
was estimated considering several values for the
number of users K{12, , 30} having the same

396
 value of the energy-per-bit to noise DSP ratio DS-CDMA system using optimal sequences offers a
Eb/N0=18dB. The resulting BER values as a function capacity increase of about 15% than when white
of the number of users K is depicted in Fig. 3. sequences or Gold codes are used.
The sensitive dependency of chaotic maps on the
Performance of asynchronous DSCDMA (N=63), over a fading AWGN channel, 2 = 0.1 initial condition offers both a greater number of
-1
10
available sequences and security increase.
The simulation results show that optimal Chebyshev
sequences are better than Gold sequences in terms of
the average BER per user, which is consistent with the
analytical result presented Section II and depicted in
-2 Fig. 1.
10
There are some differences between the simulation
R
E and analytical results given the fact that Gold
B
sequences are not perfectly white, Chebyshev
sequences are in fact pseudo-optimal, and the SGA
approximation is not quite valid for a small number of
10
-3 Th. wh. fad. BER K=10 users.
Th. opt. fad. BER K=10
Sim. Gold fad. BER K=10
Sim. Cheby. fad. BER K=10 REFERENCES
0 5 10 15 20 25 30
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CDMA and Wideband CDMA Cellular Networks. IEEE
Fig.2. Theoretical and simulated BER for A-DS-CDMA Communications Magazine, no.9, pp. 48-54, September 1998.
system over frequency-nonselective Rician fading channel [2] O. Feely, Nonlinear Dynamics of Discrete-Time Electronic
with AWGN, using Chebyshev and Gold sequences (N=63, Systems, IEEE Circuits and Systems Society Newsletter, vol. 11,
K=10). The common faded power ratio is = 0.1 . 2 no. 1, pp. 1-12, March 2000.
[3] M. B. Pursley, Performance Evaluation for Phase-Coded
Spread-Spectrum Multiple Access Communication Part I: System
Analysis, IEEE Transactions on Communications, vol. COM-25,
Performance of asynchronous DSCDMA (N=63), over a fading AWGN channel, 2 = 0.1 no. 8, pp. 795-799, August 1977.
[4] M. B. Pursley, D. V. Sarwate, Performance Evaluation for
Phase-Coded Spread-Spectrum Multiple Access Communication
Part II: Code Sequence Analysis, IEEE Transactions on
-2 Communications, vol. COM-25, no. 8, pp. 800-803, August 1977.
10
[5] T. S. Rappaport, Wireless Communications Principles and
Practice, Prentice-Hall, 1996.
[6] C.-C. Chen, K. Yao, K. Umeno, E. Biglieri, Design of Spread-
Spectrum Sequences Using Chaotic Dynamical Systems and
Ergodic Theory, IEEE Transactions on Circuits and Systems I:
R
E
B
Fundamental Theory and Applications, vol. 48, no. 9, pp. 1110-
1114, September 2001.
[7] G. Mazzini, R. Rovatti, G. Setti, Interference Minimization by
Autocorrelation Shaping in Asynchronous DS-CDMA Systems:
Chaos-Based Spreading is Nearly Optimal, IEE Electronics
Th. wh. fad. BER EbN0 = 18 dB Letters, vol. 35, no. 13, pp. 1054-1055, June 1999.
Th. opt. fad. BER EbN0 = 18 dB
Sim. Gold fad. BER EbN0 = 18 dB
[8] E. Geraniotis, Direct-Sequence Spread-Spectrum Multiple-
Sim. Cheby. fad. BER EbN0 = 18 dB Access Communications over Nonselective and Frequency-
-3
10 Selective Rician Channels, IEEE Transactions on
12 14 16 18 20 22 24 26 28 30
Communications, vol. COM-34, no. 8, pp. 756-764, August 1986.
K (no. of users)
[9] C. Vladeanu, D. Andrei, BER Performance Evaluation for
Asynchronous DS-CDMA System using Optimal Chaotic
Fig.3. Theoretical and simulated BER for A-DS-CDMA Spreading Sequences, IEEE International Conference
system over frequency-nonselective Rician fading channel COMMUNICATIONS 2004, Proceedings, vol. 1, pag. 153-159,
with AWGN, using Chebyshev and Gold sequences (N=63, Bucharest, Romania, 3-4 June 2004.
K{12, , 30}, Eb/N0=18dB). The common faded power ratio
is = 0.1 .
2

V. CONCLUSIONS

A family of optimal spreading sequences for the A-

DS-CDMA system with the SGA approximation
hypothesis was introduced for minimizing the average
BER. The generation method for the optimal
Chebyshev maps and their correlation properties were
also presented.
The BER performance of the A-DS-CDMA system
was estimated assuming a frequency non-selective
fading channel with AWGN noise. An asynchronous

397
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Noise impulse generation with convenient characteristics

in time and frequency domain
Ioan I. Duma

Abstract - In the paper "Impulse generation with Telekom networks were performed. The authors
appropriate amplitude, length, inter-arrival, and abandoned the Henkel-Kesler (HK) model which was
spectral characteristics," by I.Mann et.al., [1] the proven to be more realistic for the modeling practice
authors present what the guest editors believe is the in favour of Weibull distribution in order to facilitate
most recent statistical model of nonstationary impulse the use of the results of Tough and Ward [3] on
noise.
The Henkel/Kessler (HK) model discussed in that paper
random noise generation with prescribed amplitude
proved to be a good fit for all measured impulse noise and spectral characteristics. In the following, we shall
voltage amplitude distributions collected in both the apply the Tough and Ward method to the HK model.
networks of Deutsche Telekom (DT) and British
Telecom (BT). Nevertheless, in order to facilitate the use II. STATISTIC MODELING FOR THE
of the results of Tough and Ward [3] on random noise PROBABILITY DENSITY OF IMPULSES
generation with prescribed amplitude and spectral AMPLITUDE
characteristics, a Weibull type density was investigated
as a possible alternative since it simplfies an
approximate realization of the stochastically varying The original model known as HK (Henkel- Kessler)
spectral properties. The authors recognized that HK model was proposed in [2]
model is a better fit than the Weibull density and can be Probability density function of the impulses
considerd as more realistic while suggesting that in amplitudes is given by:
further studies the Tough- Ward method for the DT
data sets will be finalized. u
This paper proposes a suitable method for simulating ( ) 1/ 5
1
impulse noise with Henkel/Kessler amplitude probability f i (u ) = e u0 , u0>0 (1)
240u 0
density function and impulse length according a stable
probability density function.
It tells that this is a probability density symmetric to
Keywords- impulse noise, nonstationary noise, xDSL, - the origin.
stable distribution. It can be demonstrated that the transformation of
1
u5
variable y = leads to a gamma symmetric
I. INTRODUCTION u0
1 y
distribution = y 4e .
Telecommunication companies and equipment 2(5)
manufacturers are interested in modeling the impulse The model was proven to be appropriate for the
noise that is disturbing the xDSL systems. I. Mann et information gathered from both networks (BT and
al. [1] present a statistical model considered to be the DT). Nevertheless, in order to facilitate the use of the
most recent nonstationary noise impulse model. A results of Tough and Ward on noise generation, a
method for the simulation of noise impulses with symmetric Weibull probability density was used
given amplitude, length and spectral density
characteristics is proposed. 1 b y
Impulse noise is considered to be one of the main P( y ) = 12 b y e (2)
causes of signals degradation in xDSL systems. That
is why companies are interested in modeling this The parameters for theWeibull and HK models in BT
noise. A noise impulse model must describe, in a and DT network measurements are given in table 1:
statistical sense, both time domain and frequency
domain impulses properties.
In Mann & Henkel model, the parameters are chosen
according to the empirically obtained statistics when
measurements in the British Telecom and Deutsche

398
Table 1
Weibull HK 1 y 1
Il = P(( )1 / 5 ,5) + (7)
a b u0 2 u0 2
BT(CP) 0,263 4,77 9,12 V
DT(CP) 0,486 44,4 23,23 nV The right side integral is known
DT(CO) 0,216 12,47 30,67 nV
1 x 1
Ir = erf ( )+ (8)
CP Customer Premises 2 2 2
CO- Central Office
u

2 t 2
In order to test the xDSL systems, synthetic impulses where erf (u ) = e dt is the error function.
are generated using the Tough-Ward method that 0
combines the amplitude probability density function The memoryless nonlinear transform (MNLT)
with the correlation function model to produce y = g (x ) can be numerically obtained in Matlab from
impulses with appropriate time domain and frequency
domain properties. the incomplete gamma function gammainc(y,a) and
First of all, this method assumes to find a memoryless the reverse of the error function erfinv(w). The result
nonlinear transform (MNLT) that maps between a is a symmetric with respect to the origin
zero-mean, unit variance Gaussian probability density function y = g (x ) (Fig 1 left)
and the required probability density function. This is In the following the correlation function of the
then used to calculate the relationship between process y is evaluated. This can be expressed in the
correlation coefficients of the two processes. Once form:
this relationship is found, then it is possible to impose
n =
a correlation onto the input Gaussian sequence of 1 RG (t )
given length by filtering with a FIR having a spectrum (0), (t ) =
2

n=0 2 n n!
that corresponds to the input correlation function.
The Gaussian filtered sequence is fed to the 2
x2 x
memoryless nonlinear transform in order to generate exp( ) H n g ( x)dx
impulse with given amplitude and spectral density
2 2
characteristics.
(9)
To find the memoryless nonlinear
transform y = g ( x ) , the cumulative distribution where H n are the Hermite polynomials of nth degree.
functions for the normal pdf and for the required pdf Once we have evaluated the integrals
are equated

x2 x

) H n g ( x)dx
u t2
( ) 1/ 5
exp( (10)
2
1 1
e u0 du = e 2 dt (4)
2 2
240u 0
y x
we have a power series representation of the mapping
Left side integral can be calculated: between the correlation functions of the input
Gaussian and the output non-Gaussian processes. This

( )
1 4 series are rapidly convergent.
Il = v + 4v 3 + 12v 2 + 24v + 24 e v (5) The integral (10) is numerically evaluated and the
48 resulting polynomial is used to generate a lookup
table relating the input and output correlation
1/ 5
y coefficients.
where v = . So far, we have established a readily evaluable and
u0 invertible mapping between the correlation functions
For the purpose of this paper it is recommended to use of the input Gaussian and the output non-Gaussian
the incomplete gamma function given by: processes related by the nonlinear
transformation y = g (x ) .
x
Using this we can tailor the correlation properties of

1
P ( x, a ) = t a 1e t dt , x R, a R+ (6)
(a ) the input Gaussian process through the methods
0 described in [1].

Starting from (6) it is easy to obtain the left side

integral:

399
 6
x 10
6 1

4
0.5

2
0
y 0 Ry

-0.5
-2

-1
-4

-6 -1.5
-4 -2 0 2 4 -1 -0.5 0 0.5 1
x Rg
Fig. 1. The mapping between the input and the output correlation functions under the MNTL for HK density

Frequency
-3
x 10
3

2.5
III. A NEW MODEL FOR LENGTH
DISTRIBUTION 2

The approach for modelling the probability density of 1.5

impulse duration in [2] is left unchanged to be a sum
of two log-normal forms. 1

1 t 0.5
ln 2
1 2 s 12 t1
f l (t ) = B e
2 s1t 0
0 500 1000 1500
(11)
1 t 2



time
2 ln
1 2s 2 t2
+ (1 B) e
2 s 2 t
The stable law is a direct generalization of Gaussian
distribution and in fact includes the Gaussian as a
The typical parameters of the model are given in limiting case.
table 2 The main difference between the non-Gaussian stable
distribution and the Gaussian distribution is that the
Table 2 tails of the stable density function decay less rapidly
B S1 T1 S2 T2 than the Gaussian density function. This characteristic
(s) (s) of the stable distribution is one of the main reasons
BT(CP) 0,45 1,25 1,3 21,5 129 why the stable distribution is suitable for modeling
DT(CP) 1 1,15 18 - - signals and noise of impulsive nature.
DT(CO) 0,25 0,75 8 1,0 125 The stable distribution is very flexible as a modeling
tool in that it is determined by four parameters: 1) the
In this paper we propose an alternative model for location parameter a 2) the scale parameter b, also
length probability density function, a stable called dispersion, 3) the index of skewness and 4)
distribution the characteristic exponent. For more information
about the stable distribution, we refer the reader to
0 x<0 appendix.
b 2 3
pb ( x) = b
2x x 2
x>0
2x e IV. CONCLUSIONS

We presented the Tough-Ward procedure in the case
of Henkel Kessler model for the amplitude probability
density function, an unsolved problem in June 2002
when the paper [1] was published. A new model for
length distribution is proposed. This is an -stable
distribution with = 0.5 .

400
 APPENDIX and F has the Pearson density

STABLE DISTRIBUTIONS 0 x<0

b 2 3
pb ( x) = b
2x x 2
x>0 (A.5)
A distribution function (d. f.) is said to be stable if for e
2x
every n 2 there exist constants a n and bn > 0 such
that
and
b
F ( x) = F (bn x + a n ) .
*n
(A.1) Fb ( x) = 2(1 N ( )) , x>0 (A.6)
x
where
F *n is the n-fold convolution of F with itself.
The corresponding characteristic function (c.f.) x

e
f := E[e itX ] is also called stable. A d.f. F is stable 1 y2
N ( x) = 2
dy (A.7)
if and only if (iff) for every collection X 0 , X 1 , ..., X n 2

of n+1 mutually independent random variables with a

common distribution (i.i.d r.v.), X k ~ F , 0 < k n,
(iii) If = 2 then f is normal.
n 2 , there exist constants a n and bn >0 such that Using (A.3)) one obtains
Proposition 3 Let Fk = S ( k , k ; a k , bk ), k=1,2, be
X0 ~
1
( X 1 + + X n an ) (A.2) stable such that 1 = 2 = . Then F = F1 * F2 is
bn also stable and we have
F = S ( , ( 1b1 + b2 2 ) /(b1 + b2 ) ; a 1 + a 2 , b1 + b)
Every stable d. f. belongs to its own domain of
attraction [4]. Proposition 4 Let F = S ( , ; a, b) be stable. Then
the following assertions are true
Proposition 1 [4], [5] Only the norming constants (i) F1 ( x) := F ( x + ) , >0 is also stable. We
bn = n1 / are possible. have F = S ( , ; a1 , b1 ) where

Proposition 2 [5] A non-degenerate d.f. F with c. f. f b1 = b / and
is stable iff there exit real constants , , a and b a
if 1
with 0 < 2 , 1 and b >0, such that
a1 = (A.8)
a ( 2 / ) ln
if = 1


ln f (t ) = iat b t [1 + i sgn t. (t )] (A.3)
(ii) F is symetric with respect to x0 iff
where a = x0 and either = 2 or = 0
(iii) F is one-sided d.f. iff < 1 and = 1 . In

tan if 1 this case a=sup {x : F ( x) = 0} if = 1 and
(t ) = 2 2 a=inf {x : F ( x) = 1} if = 1
ln t if = 1
(iv) If 1 < 1 then the probability density
function p ( x, , , a, b) decreases as
The parameter is the characteristic exponent and 1
const. x for x . If = 1 and < 1
is the skewness parameter. The parameters a and b
in (A.3) are respectively location and scale for x a 0 , and if = 1 , 1 for
parameters. We will denote a stable d. f. F by x , p ( x, , , a , b ) decreases
F = S ( , ; a, b) . exponentially.
Example (i) For F = S (1,0 ; 0,1) we obtain < if 0 r < If = 2 the
r
(v) EX
t
f (t ) = e so that F is the Cauchy d. f. stable distribution is Gaussian and
r
EX < for all r 0 .
(ii) For F = S (1 / 2,1 ; 0, b) we have Thus, if < 1 stable laws have inverse power (i.e.
algebric) tails. This proves that the tails of stable laws
ln f b (t ) = b t [1 i sgn t ] (A.4) are much thiker than those of the Gaussian

401
distribution. An important consequence of (iv) is the (at points of continuity) where G is a proper
nonexistence of the second order moment (except for distribution not concentrated at a single point.Then:
the case = 2 ).
(i) There exists a function L that varies slowly
Proposition 5 Let X be an -stable random variable. at infinity and a constant with 0 < < 1
If 0 < < 2 , then such that
(ii)
EX
p
= if p x L( x)
1 F ( x) (A.15)
(1 )
and
(iii) Conversely if F is of the form ( A.15) it is
p
EX < , if 0 p < . possible to choose a n such that
nL(a n )
1 (A.16)
If = 2 , then a n
and in this case (A.14) holds with G = G
p
EX < for all p 0 . (from the theory of regular variation a positive
function L defined on (0, ) varies slowly (at ) if
All non-Gaussian stable distributions have infinite L( sx) s
variance. for all x > 0 , 1 ).
L( s )
Let X 1 , ..., X n be a collection of i.i.d. r.v. and X ( n )
the largest among them. If the X j have the stable REFERENCES
density (A.4) then
[1] I. Mann, S. McLaghlin, W. Henkel, R. Kirkby,and T. Kessler, "

{ }
Impuse Generation with appropriate amplitude length inter-arrival,
2 / x
P n 2 X ( n ) x e b (A.9) and spectral characteristics," IEEE J. Select. Areas Commun., vol
20 pp. 901-911, June 2002
[2] W. Henkel, T. Kessler and H.Y. Chang, Coded 64-CAP
Proof: If a limit distribution G exists we have ADSL in an impulse noise environment - -Modeling of impulse
noise and first simulation results, IEEE J. Select. Areas Commun,
F n (n 2 x) G ( x) at all points of continuity. Passing vol 13, no.9, 1995, pp. 1622-1633.
[3] R. J. A. Tough , K. D. Ward, " The correlation properties of
to logarithms we get gamma and other non-Gaussian processes generated by
memoryless nonlinear transformation," J. Phys. D: Appl. Phys. vol
n[1 F ( n 2 x)] log G ( x) (A.10) 32 pp.3075-3084, Dec. 1999.
[4] W. Feller, An Intoduction to Probability Theory, vol II, New
We have for n York: Wiley, 1965.
[5] H.- J. Rossberg, B. Jesiak and G. Siegel Anallytic Methods of
Probability Theory, Berlin, Akademie-Verlag, 1985
b
n 2 N 1 b 2 = log G ( x)

xn
2 x

(A.11)

Propositon 6 For fixed 0 < < 1 the function

( s) = e s is the Laplace transform of a
distribution G with the following properties:
G is stable

1
x [1 G ( x)] x (A.12)
(1 )

x 0
e x G ( x) 0 . (A.13)

Proposition 7 Suppose that F is a d.f. concentrated

on (0, ) such that

F *n ( a n x ) G ( x ) (A.14)

402
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

UWB communications systems based on orthogonal

waveforms set
Florin Crciun(1), Cosmin Mateescu(1), Octavian Fratu(1), Simona Halunga (1)
Abstract1: Ultra wideband (UWB) signals are a new form Considering that transmitted pulse is
of time hopping spread spectrum (TH-SS) signals with

tx (t ) = ( )d ,
pulse position modulation (PPM). Typically an UWB is the received pulse will be
defined as any signal in which the 3 dB bandwidth of the
signal is at least 25 percent of its center frequency. UWB t
signals are using previously allocated RF bands, by hiding A (t ) + n(t ) , where the constant A and are the
signals under the noise floor. attenuation and, respectively, propagation delay
Keywords: spread spectrum, time hopping, ultra-
experienced by the signal. The noise n(t ) is modeled as
wideband modulation,
AWGN with two-sided power density N 0 2 Watts/Hz.
I.INTRODUCTION
In this paper we consider that a UWB pulse is
Recent development in the area of wireless modulated by the second derivate of a Gaussian
communications systems indicates that ultra wideband 2
(UWB) technology is an attractive solution for short- t
function exp 2 properly scaled. In this case
range multiple-access communications due to a number tn
of attractive characteristics.
the transmitted pulse is:
The UWB signal is obtained using the impulse radio
technique, obtained by combining the Pulse Position t
2
Modulation (PPM) with Time Hopping Spread tx (t ) = t exp 2 (1)
Spectrum (TH-SS) technology. Impulse radio tn

communicates using baseband impulses of very short
duration, typically on the order of a nanosecond, thereby and the received pulse is :
spreading the energy of the radio signal very thinly from
near d.c to a few gigahertz. Even when those pulses are
applied to appropriate designed antennas, they t
2 t
2

propagates with distortion. The antennas behave as = 1 4 exp 2 (2)

tn tn
filters, and, even when propagation occurs over free
space, a differentiation of the impulse is produces when
the wave is radiated. In this paper, the combined effects where t n = 0.4472 was a optimum value for a measured
of the channel and antenna are modeled as a waveform. Using those values, the pulse duration is
differentiation operation. Hence, the received pulse is
the derivate of the transmitted pulse.
T = 2 ns. The normalized correlation function of the
impulse (t ) is determined by :

1
(1)
University POLITEHNICA of Bucharest,
( ) =
E (t ) (t )d > -1 (3)
Electronics, Telecommunications and Information Theory Faculty,
Telecommunications Department,
1-3 Iuliu Maniu Blvd, 061071,Bucharest 6, Romania where:
Phone: (4021) 402 4996; Fax: (4021) 410 2379
ofratu@elcom.pub.ro, shalunga@elcom.pub.ro

403
 + The bandwidth for (t ) is determined by :

2
E = ( ) d (4) ( ft )2 ( ft )2
F ( f ) = 2tn n
exp n (6)
2
2
is the energy of the signal.
1 2
The final relation is given by: which has a maximum at f = = 1.7842 GHz .
tn
2
4 2
4 2
( ) = 1 4 + exp (5)
tn 3 tn tn

transmitted pulse received pulse autocorrelation signal bandwidth for the elementary pulse
0.1 0.25

0.08 1 1

0.06 0.8 0.8 0.2

0.6 0.6
0.04
0.4 0.4
0.02 y 0.15
g
r
e e 0.2 e 0.2 e
d d n d
ut 0 ut e ut
i i d i
pl pl 0 e
zil 0 n
g
m -0.02 m a a
a a m 0.1
-0.2 mr -0.2
o
-0.04 n
-0.4 -0.4
-0.06 -0.6 0.05
-0.6
-0.08 -0.8 -0.8

-0.1 -1 -1 0
0 1 2 3 4 0 1 2 3 4 0 1 2 3 4 0 2 4 6
time [ns] time [ns] time [ns] frequency [GHz]

Figure 1. The waveforms for transmitted pulse, received pulse, autocorrelation signal and the bandwidth for the elementary pulse

- Tf is the frame interval;

- ck( ) is the spreading code used by the th
II. TIME HOPPING SIGNALS
user;
When operates is a densely populated radio - Tc is the spreading code chip interval
environment, the impulse radio interferes with other - dk
( ) is the th user data sequence.
[ k / Ns ]
narrowband radio signals that co-exists in the same
frequency range. The design of the UWB signal has to
be properly done, such that those narrowband signals The signal transmitted by the -th user consists of a
will be affected as little as possible. This requirement large number of monocycle waveforms shifted to
impose the use of spread spectrum technique. different time instants; the transmitted pulse tx (t ) is
referred to as a monocycle.
The simplest method to spread the spectrum of those
ultra wide bandwidth low-duty-cycle pulse trains is time
hopping, with data modulation at the rate of many
pulses per data symbol.
A pulse train of the form (t kT f ) consists of
k =0
monocycle pulses spaced T f seconds apart in time.
A typical hopping format with pulse position data
modulation (PPM) is given by: The frame time or pulse repetition time T f typically
may be a hundred to a thousand time the monocycle
wide. The result is a signal with a very low duty
x ( ) (t ) = (t kT f ck( )Tc dk( ) ) (7) cycle.
[ k / Ns ]
k =0
where: In figure 2 is illustrated an uniform spaced
- the superscript represent a particular user; monocycle pulse train and its power spectrum. It can

404
be easity seen that the frequency response of this vulnerable to occasional catastrophic collisions in
equally spaced pulse trains include both continuous and which a large numbers of pulses from two signals are
discrete spectral lines at regular intervals, so multiple- received at the same time instant.
access signals composed of uniform spaced pulses are
Uniform spaced pulse train
1

0.5
e
d
ut
i
pl
m 0
a

-0.5
0 20 40 60 80 100 120 140 160 180 200
time [ns]

Power Spectrum
0

-10
]
B
d[
r -20
e
w
o
P
-30

-40
0 0.5 1 1.5 2 2.5 3
Frequency [GHz]

Figure 2. Uniform spaced pulse trains and its power spectrum

To eliminate the catastrophic collisions that may multiple-access interference in many situation can be
occur in multiple access systems, each link, (indexed modeled as a Gaussian random process.
by ) is assigned to a distinct pulse shift pastern
{c } which is referred refer to as a time hopping
( )
k
Because the hopping code is periodic with period N p , the
code. These pseudorandom codes are periodic with

the period N p , i.e., c

( )
k + iN p = c , ()i, k . Each
( )
k
waveform (t kT f ck( )Tc ) is periodic also, with the
k =0
code element is an integer in the range period T p = N p T f .
0 c k( ) < N h . The time hopping code provides
therefore an additional time shift to each pulse, in the One effect of the hopping code is that it reduces the power
pulse trains with the k-th monocycle undergoing an spectral density from the line spectral density ( 1 T f apart)
added shift of c k( ) Tc . The added time shifts caused of uniformly spaced pulse train with finer line spacing
by code are discrete times between 0 and N hTc 1 T p apart, as we can see in figure 3.
seconds.
Comparing figures 2 and 3, one can observe that, when the
We further assume that N h Tc T f and hence the bearer impulses are not randomized the power spectrum is
dominated by spectral lines whereas, when the
ratio N h Tc T f indicates the fraction of the frame
randomization codes are used, it reduces the spectrum
time T f over which time-hopping is allowed. If lines, and the power spectrum is predominately
continuous.
N h Tc is to small, then catastrophic collision remain
a significant possibility. Conversely, with a large
enough value of N h Tc and well designed codes, the

405
 Randomised pulse train
1

0.5
e
d
ut
i
pl
m 0
a

-0.5
0 20 40 60 80 100 120 140 160 180 200
time [ns]

Power Spectrum
0

-10
]
B
d[
r -20
e
w
o
P
-30

-40
0 0.5 1 1.5 2 2.5 3
Frequency [GHz]

Figure 3. Randomised pulse train and its power spectrum

r (t ) = A (1) X (1) (t (1) ) + ntot (t ) , t m (9)

The data sequence { }
d i( ) of user is an M-ary
where
(1)
m, d m

( )
(1 d m M ) symbol stream that convey the m = [mN s T f + (1) , (m + 1) N s T f + (1) ] (10)
information offered by the in some form to digital data.
This information is transported over the radio channel and
Nu
using the above pulse train, by modifying the pulse
positions. It should be noted that the pulse time delay ntot (t ) = A( ) x ( ) (t ( ) ) + n(t ) (11)
introduced due to modulation is relatively small =2
compared with the time delay resulting from pulse
spreading with the spreading code (pulse When the receiver is perfectly synchronized to the first
randomization), so the effects of pulse position user signal (e.g. having learned the value of (1) ), the
modulation on the power spectrum are insignificant. receiver is able to determine the sequence { m } of time
The received signal can be modeled as: intervals, with interval m containing the waveform
(1)
Nu representing data symbol d m . In this case the detection
r (t ) = A ( ) x ( ) (t ( ) ) + n(t ) (8) problem reduces to coherent detection of M equally-
=1 energy, equally-likely signals in the presence of
where multipleaccess interference in addition with AWGN,
- A( ) is the attenuation of user s signal over the and therefore the optimal receiver is a complicated
radio channel, structure that takes advantage of all of the receivers
knowledge regarding the characteristics of multiple-
- ( ) represents time asynchronism between the access interference.
clocks of the user s transmitter and receiver, and
- n(t) represents non-multiple-access interference Due to the complexity of the analysis, the multi-user
modeled as AWGN. detector will not be considered here. Instead we will
assume that ntot (t ) is a zero-mean Gaussian random
If the receiver wants to demodulate a particular user
signal (lets say user 1), representing the m-th data process. Hence, the detection problem becomes
(1) (1) coherent detection of M equal-energy, equally-like
symbol d m , where d m is one of M equally-likely signals in presence of a mean-zero Gaussian
symbols, then the received signal r(t) is: interference in addition to AWGN.

406
In this paper we consider data been carried by if AWGN any TOR > T will perform identically, so we
orthogonal signals. choose TOR = 2T .
To detect all the M signals we will need to correlate the
III. ORTHOGONAL SIGNALS input signal with all the M reference signals. In order to
design the receiver for the OR PPM signals we
Orthogonal signals (OR) represents a particular case of consider:
PPM TH-SS signals, for which the data is given by :
x(t ) = S j (t (1) C 01 (t (1) ) + n(t ) (15)
dik = [(k + i 1) mod M ]TOR (12)
The construction of orthogonal signals is therefore where S j is one of the signals in (14) ,
given by : ( m +1) N s 1

Si (t ) =
N s 1

( )
t kT f [(k + i 1) mod M ]TOR , i = 1,2,..., M (13)
( )
Cm (t ) = Tc ck( ) p(t kT f ) (16)
k = mN s
k =0
where TOR > T . In this paper a simulation has been and n(t) is AWGN. In this case, each of the M channel
correlation output can be written :
performed for Nu = 1 , and therefore the time hopping
{ } and the delay
N sT f
ck( ) (1)
x(t )S i (t
(1)
sequence have no effect in the yi = C 01 (t (1) ))dt
correlation properties of the PPM signals, and they were 0 (17)
omitted in this analysis. Ns 1 M 1
= k =0 q=0 q,[(k +i1) mod M ] z(k , q)
For the OR PPM signals the normalized correlation where:
coefficients are given by :
kT f + (1) + ck(1)Tc + ( q +1)TOR
1, if i = j
i, j =
x(t ) (t kT f ck(1)Tc qTOR )dt
(14) (1)
z (k , q) =
0, if i j
kT f + (1) + ck(1)Tc + qTOR
hence the normalized correlation matrix AOR is a
(18)
M M identity matrix.
and q,q ' is the Kronecker delta. From the expression
For a fixed impulse waveform (t ) , N s and T f , the for y i , i = 1,2,...M it is clear that the receiver neerds
orthogonal signal depends only on TOR . In the presence only one corelator and M store and sum circuits. This is
illustrated in figure 4.
y1
Store and Sum
kT f + (1) + ck(1)Tc + ( q +1)TOR
r(t)

kT f + (1) + ck(1)Tc + qTOR
z(k,q) y2
Store and Sum

(t (1) kT f c k(1)Tc qTOR )

q = 1,2,..., M
Template
generator yM
Store and Sum

(t (1) kT f ck(1)Tc )

Code delay c k(1) Code

generator

(t (1) kT f ) (1)

Frame Clock (1) mod T f Link Selector

Figure 4. Receiver block diagram for the reception of the first users signals

407
In figures 5 and 6 can be observed the benefits of using radio modulation using orthogonal signals is potentially
block waveform modulation. By using values for M able to support thousands of users.
higher than 2, it is possible to increase the number of
users for a fixed probability of error as it is shown in
figure 5. REFERENCES:
0
[1] R.A. Scholtz Multiple Access with Time-Hopping Impulse
Modulation MILCOM93, Bedford, MA, Oct. 11-14, 1993, pp. 447-
-5 450.
[2] F. Ramirez-Mireles, R. A. Scholtz, Multiple access with time
hopping and block waveform PPM modulation, ICC Conference, pp.
-10 775-779, June 1998.
e
)
rr [3] Vicenzo Lottici, Aldo DAndrea, Umbro Mengali Chanel
b
P(
g
Estimation for Ultra Wideband Communications, IEEE Journal, vol.
ol
0
1 -15 20, no. 9, December 2002.
[4] Moe Z. Win, Robert A. Scholtz Impulse Radio: How it works
IEEE Communication Letters, vol. 2, nr. 2, Februarie 1998.
-20 M=2
[5] Paul Hansell, Selcuk Kirtay Ultra Wideband Compatibility
M=4
M=6
Final Report to the Radiocommunications Agency, January 2002
M=8
M=16
-25
0 100 200 300 400 500 600 700 800 900 1000
Nu

Figure 5.The base 10 logarithm of probability of bit error for

orthogonal PPM as function of Nu user for different values of
M,using Rb=9.6 Kbps.

If we use a high bit data rate and lower bit error

probability we obtain he curves shown in figure 6.
0

-5

-10
)
rr
e
b
P (
g
ol
0
1 -15

-20 M=2
M=4
M=6
M=8
M=16
-25
0 50 100 150 200 250 300 350 400 450 500
Nu

Figure 6. The base 10 logarithm of the probability of bit error for

orthogonal PPM signals as function of Nu for different values of M,
using Rb=1048 Kbps.

V. CONCLUSION
In this paper we have shown that for applications
requiring high data rate (1024 Kbps) combined with low
probability of bit error (10-8 ), impulse radio modulation
using orthogonal signals is potentially able to support
hundred of users.

Similarly, for applications requiring low data rate (9.6

Kbps) and moderate probability of error (10-4) impulse

408
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Mobility Concept for Wireless ATM networks
Marius Moise1
Abstract In this paper, the Wireless ATM rerouting
procedures are analyzed and categorised, based on a
standard network topology, derived from the Wireless
ATM reference model. A new operational concept for a
mobile ATM network called Mobile Network
Architecture based on Virtual Paths (MNAVP), in which
the network nodes are connected to each other via
preestablished permanent virtual path connections with
fixed capacity assignments, is being proposed and
described. Finally, the handover hysteresis concept is
introduced and a hysteresis gain is defined and
calculated as the factor by which the handover rate is
reduced through the use of the hysteresis.
Keywords: WATM, handover, routing, hysteresis.
I. INTRODUCTION
Over the last few years, one of the major
commercial successes in the telecommunications
world has been the widespread diffusion of cellular
mobile telephone services, whose provision relies on
sophisticated algorithms implemented by state-of-the-
art dedicated computer equipment. Lately, the
challenge resides in upgrading the service offer to
mobile users to include high-speed data
communication services. A natural approach in this
direction is to adopt the Asynchronous Transfer Mode
(ATM) in the wireless environment, resulting in the
so-called wireless ATM (WATM) network. However,
ATM was developed for fixed networks and mobility
management functionality had to be added to the
traditional set of capabilities. Mobile or Wireless
ATM consists of two major components: the radio
access part which deals with the extension of ATM
services over a wireless medium and the mobile ATM
part wich addresses the issue of enhancing ATM for
the support of terminal and service mobility in the
fixed portion of the WATM network. Wireless ATM
started as a technology designed to be used for LAN
or fixed wireless access sollutions, where low mobility
constraints are encountered. Further research projects
and standardisation activities coordinated by the ATM
Forum demonstrated the feasibility of broadband radio
access networks based on ATM technology, which
1
Ph.D. Student, Politehnica University of Timisoara, Faculty of Electronics and Telecommunications Engineering
Private: Pilsenseestr.5, 82229 Seefeld, Germany
Phone: +49-179-2960468, Fax: +49-179-332960468, E-Mail: marius_moise@yahoo.com
409
can offer full-scale mobility together with all the range
of ATM service capabilities existent also in the fixed
ATM networks [8].
Mobility management has two distinct
components: location management dealing with the
correspondence between the subscribers data and his
current location and handover management which
controls the dynamic re-routing and transfer of
connection for the terminals crossing cell boundaries.
The frequency-domain supposed to be used for
Wireless ATM, situated in the Ghz range, will imply
the existence of small size cells, to cope also with the
increased demands regarding system capacity. This
will lead, in conjunction with a higher terminal
mobility, to a very large number of handover of virtual
connections. Furthermore, smaller cells have tighter
delay constraints, as the overlapping distances of the
cells are smaller. The more complex handover
procedure has higher requirements regarding radio
ressource management functions for the air interface
paired with network signalling and control functions
for handover control, Quality of Service (QoS)
management and rerouting of the connection to the
new network access point. Exactly these rerouting
procedures are the subject of this paper. A new
operational concept based on Virtual Paths is
introduced and the principles of handover hysteresis
are analysed using a discrete Markov chain model.
II. WIRELESS ATM NETWORK ARCHITECTURE
A various number of reference architectures can be
taken into account when we talk about Wireless ATM,
ranging from simple mobile terminals to complex
systems containing mobile ATM switches built in in
ships, planes or satellites. One standard reference
scenario contains a broadband wireless access system
providing unrestricted roaming capabilities within a
certain area of continuous radio coverage (Fig.1). The
base stations (Radio Access Point, RAP) are of
picocellular size and implement the physical transport
medium, multiple access control, data link control and
basic radio ressource management capabilities. The
RAP does not necessarily have to provide ATM-based
physical transport, it could use as well any other
access technology, as for example CDMA, also
because the error detection and correction capability
of the ATM stack is typically low, since it was
designed for a reliable network. For this paper, we
assume though the existence of a ATM radio interface
capable of transmitting ATM cells over the wireless
medium. Special Mobile ATM switches (MAS) are
positioned at the border of an ATM network,
supporting end-system mobility by possesing the
necessary extensions in the signalling and control
planes to provide functions for mobility management
and also connection handover.
All the RAPs associated with a particular MAS
form a so called zone of continuous coverage.
Terminal mobility inside a certain zone and the
handover associated with it (intra-zone handover) is
handled locally by the MAS itself. Neighbouring
zones with uninterrrupted radio coverage can form an
area in which, at any time, a RAP can be found to
hand a connection over to while the terminal is
moving without restrictions. The size of such an area
is not limited, it could take the size of the entire
network.
ACS
ATM Switch
MAS MAS MAS
RAP
Zone 1
Zone 3
Zone 2 Intra-zone
handover Inter-area handover
Inter-zone handover
Fig. 1: Architecture of the Wireless ATM network
It is not mandatory that all the switches are able of
supporting end-system mobility, therefore we
introduce an hierarchically superior instance, called
Area Communication Server (ACS), providing
mobility control for a specific area. The ACS
represents a mobility supporting ATM switch in
charge of processing the protocoll requests in case of a
inter-zone handover. It also serves as anchor point
(AP) for the active connections of the terminals inside
this area. By using the ACS, the impact of the end-
system mobility on the network can be significantly
reduced, because there is no need anymore for
mobility specific functionality outside the ACS area.
The disadvantage consists in the fact that connectivity
cannot be guarateed for terminals leaving this area.
A consequence of the high mobility of the
terminals is the requirement of a permanent
reestablishment of the virtual connection, in order to
reach their current point of access to the network. This
implies, beyond signaling and handover control, a
process of rerouting of the connection in the ATM
network. QoS control based on requirements coming
from the connection itself has to be provided in order
to ensure the lossless and in sequence delivery of the
ATM cells during the handover process.
III. CONNECTION REROUTING IN WATM
NETWORKS
We can categorise the approaches for connection
rerouting in four basic categories: full reestablishment,
connection extension, incremental reestablishment and
multicast reestablishment. They are schematically
presented in Fig.2 and Fig.3, showing the connection
phases during two handover steps for the different
basic methods.
The most simple method is the complete
reestablishment of the connection. For each change of
a RAP coverage area, due to terminal mobility, a
completely new VC connection is being set up
between the mobile terminal and its peer. This can be
done in absence of any defined handover control
functions, only by the interaction of the two end
systems. The major disadvantages consist in the very
long duration of the procedure and the complete
interruption of the service. Quite opposite to this
procedure, the connection extension handover keeps
the impact on a local scale.
Full Connection
MAS reestablishment extension
MAS MAS MAS MAS MAS MAS
Zone 4
Handover 1 Handover 2 Handover 1
Handover 2
Fig. 2: Connection reestablishment and connection extension
Each handover is only prolounging the connection
from the old RAP to the new RAP, by this achieving a
very high speed, with the cost of a high routing
inefficiency, due to the fact that no rerouting of the
connection is performed. Loops can easily occur if the
terminal is moving back and forth in a limited area,
between only few neighbouring cells. This method has
to be combined with a routing optimisation algorithm,
otherwise ressources are waisted. An example
illustrating this scheme is presented in [9].
The multicast concept is also dealing with
inefficiencies regarding the utilisation of network
ressources. The multicast tree is established at
connection setup time and can remain static or be
dinamically updated for the duration of the connection
410
[2],[16]. All routes leading to the RAPs which will be
presumly used by the mobile terminal during the
connection are precalculated and assigned as a
complete set to the connection at call setup time. This
leads to a very fast handover procedure with a
minimum of signalling load because no extra routing
and call processing is necessary due to the preparation
work done.
Incremental
reestablishment
MAS MAS MAS MAS MAS
Handover 1 Handover 2 Handover 1
Handover 2
Fig. 3: Incremental reestablishment and multicast extension
Last but not least, the incremental reestablishment
represents a more complex and therefore efficient
scheme during which a rerouting decision is made for
each individual handover. This decision affects only a
portion of the connection, namely the one between the
new RAP and some Cross Over Switch (COS) inside
the current ACS area [13]. The high efficiency and
handover speed are due to the calculation of the
optimal path to the new destination RAP for each
handover. The probability of reusing the longest part
of the connection is quite high, which enables a fast
handover, without going through the loop of routing
decision.
IV. MOBILE NETWORK ARCHITECTURE
BASED ON VIRTUAL PATHS
Basically, there are two alternatives to operate the
mobile ATM network as described in Fig.1: either VP-
based or VC-based. We choose a VP-based mode,
defining a concept called the Mobile Network
Architecture based on Virtual Path (MNAVP). In the
MNAVP, the MAS are net-worked with their
corresponding ACS over the fixed ATM network via
preestablished permanent Virtual Path Connections, as
shown in Fig.4. The VPs of this architecture have
fixed capacity assignments defining a virtual mobile
network topology over the fixed ATM infrastructure.
In this VP-based network, all intermediate ATM
switches between ACS and MAS are only performing
VP-switching (cross-connect functionality). Two
VPCs carried on the same physical link are not
statistically multiplexed. Inside a single VPC
statistical multiplexing is being applied.
This virtual networking approach has several
advantages. First of all, the preestablished VP
topology eliminates the need for complex call routing
functions and switching table updates along the VC-
route, which facilitates fast handover connection
setup. Second, call admission control decisions only
have to be taken in the switches terminating the virtual
path connections (MAS and ACS), again reducing
connection setup complexity. Further, the
establishment of a virtual mobile network is ideally
suited for QoS-management and QoS-guarantees in a
multi-operator fixed network environment. The
heterogenous nature of multiservice WATM virtual
Multicast
extension
connections with a broad variety of QoS-constraints
and requirements is paid attention to by separating
traffic with different QoS-characteristics onto different
VPs as proposed for fixed ATM networks [4], i.e.
connections with similar QoS-requirements are
MAS aggregated in one VP-subnetwork. Connections
carrying multirate services can be aggregated in single
VPCs, as long as they can be statistically multiplexed.
Different types of services, e.g. CBR and VBR, which
reduce the statistical multiplexing gain when
transported together within a VPC, are separated onto
different, parallel VP subnetworks.
ACS ACS
Physical VC-tree Logical VP-tree
topology topology
MAS MAS MAS
MAS MAS MAS
Fig. 4: MNAVP topology
The obvious advantages of the MVPA concept are,
however, achieved at the cost of losing bundling gain
and a somewhat less efficient statistical multiplexing
on the physical links, resulting in a reduced utilization
of physical resources. The VP-based virtual topology
networking concept in the fixed ATM subsystem is
peered by VPC operated RAP-links (Fig.4), where
again VPCs are used for multiservice traffic
management between the MAS and the radio access
points. With this two-staged approach in the MNAVP
design, handovers can be handled in a partly
distributed fashion, i.e. intra-zone handovers are
handled locally by the MAS, whereas the ACS is
involved only in inter-zone handovers. To further keep
part of the handover processing within a zone, the
zone can be extended virtually by the use of VPs
connecting RAPs of neighbouring zones to a MAS so
that actually in the MNAVP virtual network the zones
overlap to some extent (Fig.4, Fig.6). By that, the
number of inter-zone handovers can be reduced, and
the MAS has to do most of the work in handover call
processing.
Consequently, a VC connection in a wireless ATM
network consists of two different segments (Fig. 5):
411
 the fixed segment from the ACS into the fixed
network with the ACS operating as a Cross Over Anchor Domain A Anchor Domain B

Switch (COS). This segment ist established at call ACS ACS ACS ACS
setup time and doesnt change during the lifetime of a
call.
the mobile segment, which follows a
MAS MAS
predetermined VP-route in the mobile network from MAS MAS MAS

the COS to the current RAP. This segment has to be MAS MAS MAS
rerouted during a connections lifetime due to user
mobility. Anchor Domain A Anchor Domain B

Virtual Connection Circuit VCC

Single Homing Dual Homing
Mobile segment Fixed segment

Fig. 6: VP based Dual Homing

MAS COS
MT RAP
Fig. 5: Fixed and mobile segment of a VC connection
The most complex handover situation occurs,
when the mobile moves to a zone belonging to a
different ACS. This inter-zone handover situation
generates the highest call processing load. The ACS
responsible for the connection during call setup
remains the anchoring point (COS) for that
connection, switching between the its fixed and
mobile segment. The mobile segment has to be
extended from that COS to the new RAP via the
second ACS. Therefore, each ACS establishes VPCs
with its neighbouring ACS, reserved for carrying the
inter-ACS handover connections. For best support of
wide area mobility, this inter-ACS backbone
network could have a full mesh topology. The
MNAVP connection anchoring mechanism can
produce non-optimal routes, which could be optimized
by allowing the anchoring point COS to change during
the lifetime of a connection. To reduce the number of
inter-zone handover, each MAS can be virtually
linked to more than one ACS. At connection setup
time, one ACS has to be chosen as the connections
COS. This decision can be based on load-balancing
considerations as well as mobility prediction analysis
in order to minimize the number of inter-zone
handover.
At call setup, the connection is switched through
the ACS belonging to the actual physical domain. In
the situation of a inter-zone handover, when the MT is
crossing physical domain boundaries, it still remains
in the anchor domain, due to the logical structure of
the network. A rerouting of the connexion over a new
ACS is necessary only in the case of a repeated inter-
zone handover. This applies also for the movement in
the backward direction. By thus, a routing hysteresis
which prevents the frequent occurence of handover in
case of limited geografical mobility, is introduced.
The hysteresis helps preserving one of the significant
advantages of the MNAVP-based rerouting procedure,
namely that inside an anchor domain, the mobile
segment between ACS / COS and MAS consists of
only one VC segment, which would be lost in case of
handover to another domain.
The analysis of the hysteresis procedure is made
based on an discrete Markov chaining model. Lets
consider, for the start, the case without hysteresis,
shown in Fig. 7 and 8.
Physical border
2 5 8
1 10
3 9
6
Anchor domain 4 7 Anchor domain
A B

V. HANDOVER HYSTHERESIS IN MNAVP Fig. 7: Handover between neighbouring domains without hysteresis

p1,2 p4,5 p7,8

The advantage of the logical network of MNAVP p2,4 p5,7 p8,10
p1,1 2 p4,4 5 p7,7 8 p10,10
relies in the flexibility offered for the case of an p3,2 p6,5
p9,8
1 p2,3 4 p5,6 7 p8,9 10
unexpected high handover traffic between two p3,1 p6,4 p9,7
3 6 9
neighboured zones, called also anchor domains.
p4,3 p7,6 p10,9
This can be implemented by means of a VPC-based
Dual Homing of the MAS at the ACS. One MAS is Fig. 8: Discrete Markov model for the handover succesion without
hysteresis
connected to two or more ACS via VP connection,
increasing in this way the size of the anchor domain
This model describes the states of the different
which leads in the end to an partial overlapping of
types of handover from Fig.7. The total sum of
those domains.
transition probabilities for one state is for all i states
correspondingly

412
 p1,1 1 p3,1 p16,1

1 1 1
p2,3 1 p 4,3 p2,3
n

=1
p4,5 1 1 p6,5

i ,j
(1)

1 1 1


p5,6 1 p7,6 p5,6
j =1
p7,8 1

1 1
P =
H p9,9 1 p10,9 p8,9

1
The equation (2) describes the matrix of transition-


p11,10 p12,10 1


p11,10 1 p12,10 1
states
p12,14 p13,13 1

1 1 1

p14,13 1 p15,13 1

p1,1 p3,1 p15,16 0

p p3,2 (2)
1,2
p2,3 p4,3

p2,4 p4,4 p6,4 Now, the effect of the handover hysteresis can be

PH =
p4,5 p6,5
quantified. The hysteresis gain can be defined as being
p5,6 p7,6 the factor by which the frequency of occurance of an
p5,7 p7,7 p9,7
inter-zone handover between anchor domains can be
p7,8 p9,8
reduced.
p8,9 p10,9
p8,10 p10,10
p5 + p6 (7)
GH =
p 3b + p 8a
To be able to calculate the probability of
occurrence of an handover, the state-probabilities of
the Markov-chain are needed Physical border

PH p = p (3) Anchor domain

A
After homogenizing the equation system and 2a
converting the matrix, we obtain the equation (4) 1a 5a Anchor domain
3a 6a B

4a 7a 8a
p1,1 1 p3,1
(4)
1 1
p2,3 1 p4,3
3b
p4,5 p6,4
5b 8b
1 1
PH = p5,6 1

p7,6 6b 10b
p7,8 p9,7 9b

1 1 4b 7b

p8,9 1 p10,9
Physical border
0

Fig.9: Handover between neighbouring domains with hysteresis

and with the condition p e = 1 for the sum of the state p1,2
p2,4
p4,5
p5,7 p7,8
p1,1 2a p4,4 p6,5 5a p7,7
probabilities, the equation is transformed in
1a p3,2 4a p5,6 7a 8a
p2,3 p6,4 p8,9
n 3a 6a

i =1
pi = 1 (5) p3,1
p4,3
p15,13
p7,6
p12,10
p8,11

p16,2 p13,12 p10,9

p15,15 5b p12,12 p11,10 8b
p14,13
The probabilities of assuming the states 5 and 6 p16,1 3b 4b 7b p10,11 10b
(corresponding to the inter-domain handover cases in 6b
p13,14 p11,12 9b p9,9
p15,16
Fig.7) are in fact the probabilities of occurrence of this p14,15 p12,14 p9,11
handover type. Fig.10: Discrete Markov model for the handover succesion with
hysteresis
In the case of a handover hysteresis, the handover
sequence depends upon the network nodes involved,
The border conditions are considered symmetrical.
as shown in Fig.9 and 10. The Markov model has to
Lets consider the random chosen variables from
be extended with the corresponding states and the
interzone-handover are in this case reprezented by the Table 1, which describe the local movement in terms
of the probability that the MT does not leave, during
states 8a and 3b. Similar to before, we obtain the
its movement, the region it belongs to. The parameter
equation (6) below
of the model with hysteresis described in Fig.8 can be
determined by inserting the probabilities from Table 1
at the zone borders and, at the same time, respecting
condition (1).

413
Table 1: Parameter for the numerical example of handover
hysteresis.
Parameter Value
p1,1 , p10,10 0,8
p4 , 4 , p 7 , 7 0,1
p2 ,3 , p8,9 0,3
GH
25
20
15
10
5 [2]
0 p
5,6
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9
Fig.11: Hysteresis gain
The main parameter which describes the hysteresis
gain is the probability of occurrence of consecutive
inter-zone handover. If this probability equals zero,
the MT crosses the border, in one direction, only once.
If it is high, the the movement area is very restricted
and the frequency of the border crossing high. Fig.11
shows this dependency and it can be observed that
even in the case of a non-local movement, when
p 5,6 = 0 , the probability of handover-occurrence is
reduced to almost one third. This is based on the fact
that, due to the hysteresis, one intra-zone handover at
the border domain (handover 4 and 7 in Fig.7) is not
immmediately followed by an inter-zone handover,
because, as shown in Fig.9, in this case of restricted
mobility area, only inter-zone handover are performed
(handover 4a and 7b).
CONCLUSIONS
In this paper a Wireless ATM reference
architecture has been presented, consisting of a
wireless access system coupled with the support of
mobile end-systems within the ATM network. The
network is subdivided in several areas served by one
area server ACS, areas which are in fact handover
domains providing uninterrupted radio coverage and
full mobility support. After a brief description of the
rerouting schemes proposed already, a new
networking concept called MNAVP has been
proposed, establishing a virtual mobile subnetwork
over the fixed ATM infrastructure, based on
preestablished virtual paths with fixed capacity
assignments. This network design minimizes
414
signalling and call admission load, maintaining a low
handover connection setup latency and facilitating
high handover rates. The concept of handover
hysteresis is being introduced, with the benefit of
decreasing the number of inter-zone handover, for
terminals showing limited geografical mobility. The
frequency of occurrence of this cost-intensive
handover type can be decreased with beneficial effects
on the ressource budget, and this is demonstrated
using a Markov decision process. A hysteresis gain is
defined and calculated as the factor by which the
handover rate is being reduced through the use of the
hysteresis.
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Local and Global Handovers for Mobility Management in Wireless
ATM Networks, IEEE Personal Communications Magazine, vol.
4, no. 5, October 1997, pp. 16-24.
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Networks. Greenwich: Manning, 1996.
[12] Mitts, H.; Hansn, H.; Immonen, J.; Veikkolainen, S.:
Lossless handover for wireless ATM, MONET Mobile Networks
and Applications Volume 1 (1996), no. 3, pp. 299-312.
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wireless ATM networks, MONET Mobile Networks and
Applications Volume 1 (1996), no. 3, pp. 313-334.
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Conference Proceedings European Personal Mobile
Communications Conference EPMCC97, Bonn, 30. Sept. 2. Okt.
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Interface and Network Protocols of the Mobile Broadband System,
IEEE Personal Communications Magazine, August 1996.
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architecture and protocols to support efficient handoffs over an
ATM / B-ISDN personal communications network, MONET
Mobile Networks and Applications Volume 1 (1996), no. 2, pp. 123-
139.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Lossless handover scheme for Mobile ATM networks
Marius Moise1
Abstract The standardized approach to Mobile ATM
network handover is a hard backward or forward
handover scheme with no guarantees for the integrity of
the data stream. These handover functions are detailed
in this paper and a more performant virtual connection
handover protocol, using in-slot signalling techniques in
order to facilitate lossless handover, is introduced. The
QoS aspects related to this scheme are discussed and its
OAM implementation is presented.
Keywords: WATM, handover, QoS, OAM.
I. INTRODUCTION
The broadband service requirements for future
generations of mobile communications can be
accomodated also by high bandwith ATM networks.
However, ATM was developed for fixed networks and
mobility management functionality had to be added to
the traditional set of capabilities. Mobile or Wireless
ATM consists of two major components: the radio
access part which deals with the extension of ATM
services over a wireless medium and the mobile ATM
part wich addresses the issue of enhancing ATM for
the support of terminal and service mobility in the
fixed portion of the WATM network. Wireless ATM
started as a technology designed to be used for LAN
or fixed wireless access sollutions, where low mobility
constraints are encountered. Further research projects
and standardisation activities coordinated by the ATM
Forum demonstrated the feasibility of broadband radio
access networks based on ATM technology, which
can offer full-scale mobility together with all the range
of ATM service capabilities existent also in the fixed
ATM networks [8].
The frequency-domain supposed to be used for
Mobile ATM, situated in the Ghz range, will imply the
existence of small size cells, and this will result, in
conjunction with a higher terminal mobility, to a very
large number of handover of virtual connections. The
more complex handover procedure has higher
requirements regarding radio ressource management
functions for the air interface paired with network
signalling and control functions for handover control,
Quality of Service (QoS) management and rerouting
1
Ph.D. Student, Politehnica University of Timisoara, Faculty of Electronics and Telecommunications Engineering
Private: Pilsenseestr.5, 82229 Seefeld, Germany
Phone: +49-179-2960468, Fax: +49-179-332960468, E-Mail: marius_moise@yahoo.com
415
of the connection to the new network access point.
The standardized approach to Wireless ATM network
handover is a hard backward or forward handover
scheme with no guarantees for the integrity of the data
stream [4]. In order to cope with the requirements of
future multimedia applications, more complex
solutions, which are also including options for QoS
control, are needed. These handover functions of a
Mobile ATM network are detailed in this paper and a
more performant handover protocol, using in-slot
signalling techniques in order to facilitate lossless
handover, is presented.
II. MOBILE ATM NETWORK ARCHITECTURE
One standard reference scenario for Mobile ATM
contains a broadband wireless access system
providing unrestricted roaming capabilities within a
certain area of continuous radio coverage (Fig.1). The
base stations (Radio Access Point, RAP) are of
picocellular size and implement the physical transport
medium, multiple access control, data link control and
basic radio ressource management capabilities. The
RAP does not necessarily have to provide ATM-based
physical transport, it could use as well any other
access technology, as for example CDMA, also
because the error detection and correction capability
of the ATM stack is tipically low, since it was
designed for a reliable network. For this paper, we
assume though the existance of a ATM radio interface
capable of transmitting ATM cells over the wireless
medium. Special Mobile ATM switches (MAS) are
positioned at the border of an ATM network,
supporting end-system mobility by possesing the
necessary extensions in the signalling and control
planes to provide functions for mobility management
and also connection handover.
All the RAPs associated with a particular MAS
form a so called zone of continuous coverage.
Terminal mobility inside a certain zone and the
handover associated with it (intra-zone handover) is
handled locally by the MAS itself. Neighbouring
zones with uninterrrupted radio coverage can form an
area in which, at any time, a RAP can be found to
hand a connection over to, while the terminal is
moving without restrictions. The size of such an area
is not limited, it could take the size of the entire
network.
ACS
ATM Switch
MAS MAS MAS
RAP
Zone 1
Zone 3
Zone 2 Intra-zone
handover Inter-area handover
Inter-zone handover
Fig. 1: Architecture of the Wireless ATM network
It is not mandatory that all the switches should be
able of supporting end-system mobility, therefore we
introduce an hierarchically superior instance, called
Area Communication Server (ACS), providing
mobility control for a specific area. The ACS
represents a mobility supporting ATM switch in
charge of processing the protocoll requests in case of a
inter-zone handover. It also serves as anchor point
(AP) for the active connections of the terminals inside
this area. By using the ACS, the impact of the end-
system mobility on the network can be significantly
reduced, because there is no need anymore for
mobility specific functionality outside the ACS area.
The disadvantage consists in the fact that connectivity
cannot be guaranteed for terminals leaving this area.
A consequence of the high mobility of the
terminals is the requirement of a permanent
reestablishment of the virtual connection, in order to
reach their current point of access to the network. This
implies, beyond signaling and handover control, a
process of rerouting of the connection in the ATM
network. QoS control based on requirements coming
from the connection itself has to be provided in order
to ensure the lossless and in-sequence delivery of the
ATM cells during the handover process.
III. HANDOVER FUNCTIONS OF A MOBILE
ATM NETWORK
Several different handover protocols have been
described in the literature [1][2][3][5][7][8][15][16]
[17]. Based on the number of simultaneously active
radio connections, one can distinguish between two
main streams in the handover techniqes: the hard
handover and the soft handover. In the case of the hard
handover, it always exists only one active radio
connection and the handover control flow can be
directed either across the current RAPs air interface
(backward handover) or across the target RAPs air
interface (forward handover). The soft handover tries
to eliminate the disadvantage of the hard handover,
consisting of the interruption of the data stream during
the connection switchover, by establishing and
activating a second radio connection to the target
RAP. The areas covered by the Mobile ATM
handover techniques so far are presented in Fig. 2.
with losses
(WATM CS1)
Backward
lo s s le s s
Hard
with losses
MAS
(WATM CS1)
Forward
lo s s le s s
Handover
Backward
Zone 4
Soft
Forward
(impossible)
Fig. 2: Mobile ATM handover schemes
Until now, no WATM handover mechanisms which
allow forward handover and, at the same time,
maintain cell loss QoS guarantees, have been
proposed. Nevertheless, future Mobile ATM systems
will demand the flexibility and robustness in handover
control as well as an increase in QoS.
Soft handover is considered to be the handover
sollution of future wireless network systems, the so
called 4G or Next Generation Mobile Systems. During
a soft handover process, the MT is able to
communicate simultaneously with both RAPs.
Therefore, each connection posesses two active
mobile segments between the MT and the COS. There
are several known methods, developed for the fixed
ATM networks, able to establish cell synchronicity
between two different paths [9],[10],[17]. Cell
synchronicity is mandatory for a adaptive and
latency-free switching between the two paths,
achieving by this a quality gain (macrodiversity). The
periodical in-slot signalling procedure upon which the
lossless handover scheme proposed in this article is
based, is also suited to synchronise the two paths, as it
has been described in the Alignment Server Method
[9],[10].
IV. A LOSSLESS HANDOVER SCHEME
During the handover of virtual connection there
are some situations when errors occur [17]. These
errors cause on the downlink the loss of in transit cells
due to forced handover decisions and they can also
cause on the uplink disruption of ATM cell sequence
at the handover Cross Over Switch (COS) due to a
transfer delay mismatch between the old and the new
path (Fig.3).
The result of the cell loss is a degradation of the
QoS of a virtual circuit connection by affecting the
416
data stream integrity. Several methods, based on an in-
band signalling approach with two-way handshake, for
preventing cell loss and disruption have been
described in the literature [15],[18],[19]. Their
advantage consists in their relative simplicity which
easies the implementation. Their main disadvantage
comes up in the case of a single signaling cell loss or
in emergency handover situations, when the mobile
looses the connection to the old cell before the
signalling cell reaches him.
MAS
h Fig. 4: Principle of Sync-Tag-Handover STH
P at RAP 1
ld
O Downlink loss of
in-transit cells
COS
Uplink loss of cell
sequence Handover
Ne
w
Pa
th
MAS
RAP 2
Fig. 3: VC-Handover, error scenarios
The procedure proposed in this paper called Sync-
Tag-Handover (STH) is dealing with these aspects,
being therefore more robust. The way this algorithm
works is shown in Fig.4: the COS inserts periodically
into the downlink ATM cell stream in-band signalling
cells containing a tag, carrying sequential numbering
information. The mobile terminal sends these tags
back as an acknowledgment receipt for the correctly
received segments of the downlink cell stream. At
transmission, each cell stream segment is also copied
into a retransmission buffer inside the COS and
deleted out of this buffer when the receipt was
received. At handover, the mobile terminal is sending
an End-of-Line signal in the uplink direction, is
performing the handover and resyncing with the new
RAP by sending it the number of the last tag correctly
received on the downlink from the old RAP. By this,
the COS knows that it has to start retransmitting the
content of the buffer, starting with the tag-number just
mentioned, to the new RAP. The STH scheme is
designed to flexibly support both handover variants:
backward handover and forward handover. This
handover scheme enables the mobile station to
instantaneously detach from its current RAP and hand
over its connection to the target RAP at any time. As
during backward handover the handover signalling is
exchanged with the old RAP, this instantaneous
detach facility of STH is of no particular use in this
case. But it can be very successfully exploited to
provide zero cell loss forward handover. In the
forward handover case, the control message flow will
be routed across the new RAP after the MT
successfully attached itself. Because of the periodic
insertion of sequence numbers and buffering of not yet
417
confirmed cell stream segments in the COS, there is
no need to perform an in-slot signalling handshake
before the handover to enforce zero cell loss.
MT RAP old COS RAP new MT
Handover Initiation Signaling
The MT can abruptly detach from the failing links
RAP, while at the same time maintaining state
information to facilitate a zero cell loss handover. Cell
sequence information and retransmission buffer are
concurrently kept up to date and can be readily
evaluated for cell stream re-synchronization after the
successful attach at the new RAP and the rerouting
decision in the ACS/COS. Should the connections
QoS contract require zero cell loss during handover,
the buffering mechanism will allow ATM cells
otherwise lost (i.e. at most those not yet acknowledged
by an uplink tag) to be forwarded by the COS to the
new RAP. The mobile terminal will then use the
stored numbering information to resynchronize the
ATM cell stream just as in the backward handover
case.
In a WATM system, soft handover macro diversity
calls for ATM cell level synchronization of two
communication paths between MT and ACS/COS.
Each path carries the same user data stream, but
displays possibly different delay properties. Therefore,
synchronisation between the two cell streams is
mandatory and once this state has been reached, it is
possible to dynamically select the best path, on a per
cell stream segment or even on a per cell basis.
Synchronization and dynamic path selection takes
place in the ACS/COS for the uplink direction and in
the MT for the downlink direction. STH as basic
handover protocol is prepared to support cell level
synchronization for this dualcast situation with
minimal extensions. The in-slot signaling mechanism
already provides means for synchronizing the two
communication paths over the different RAPs.
Whereas this synchronization is used to protect cell
sequence and prevent cell duplication in the hard
handover case, it can be directly applied to
synchronize two continuous ATM cell streams during
the soft handover phase. Once a soft handover
situation has been detected and path diversity has been
established by activating a second radio link to the
handover candidate RAP, the COS will start
dualcasting the downlink cell stream on both paths.
Using the in-slot numbering information, the MT is
able to synchronize these two paths and to
dynamically select one path on a per cell stream
segment basis. The in-slot tag completing the selected reception of the receipt tag, which makes the fill-up
cell stream segment is then looped back on both level of the buffer at least equal to twice the product
uplink diversity paths. This enables the COS to also bandwith-delay, to which the number of tags,
activate its dynamic path selection algorithm. Once generated during this time interval, has to be added.
the radio conditions are sufficiently stable to guarantee The maximum level can be calculated with the
reliable communication via the new RAP, dualcasting formula
can be terminated and the handover completed. Soft 2 t oldd
handover is highly resource intensive in terms radio C RB = RC (2 t oldd + T ) + (1)
frequency spectrum and cell transport bandwidth. On
T
the other hand, STH based soft handover does not rely in which RC represents the average cell rate of the
on cell segment retransmission and therefore does not connection. The result represents at the same time, in
require any retransmission buffering space in the case of a handover, the maximum number of cells
COS. The only buffering space required is a small which have to be retransmitted to the new base station
path synchronization buffer in MT and COS. RAPnew. The optimal tag-interval can be calculated
Moreover, soft handover provides the best QoS out of the equation
performance of all handover alternatives. The old and
C RB 2 t oldd
the new path are synchronized on the ATM cell level, = RC =0 (2)
facilitating path selection and therefore handover T ( T )
switching/connection rerouting in realtime without out of which we obtain the best tag-interval as being
intro-ducing any delay or disruption into the cell
2 t oldd
stream. It is therefore the most attractive handover Topt = (3)
alternative for real time services which require an RC
outstanding cell loss performance during handover. During the retransmission of the cells stored already in
the buffer, the new ones, coming from the fixed
V. BUFFER ADMINISTRATION segment of the connection have to be queued first, in
order to be sent segmentwise afterwards on the new
The central element of the STH scheme is the buffer mobile path. The buffer can be emptied only if the
storing copies of the ATM cells sent in downlink reading rate is higher than the average cell rate of the
direction. As soon as the in slot signalling function
has been activated for a connection, a first tag is connection R EB > RC . In this case, the time interval
introduced in the cell stream and, at the same time, the in which the content of the buffer is retransmitted is
first copy is stored in the buffer. The procedure is C
continued until the moment in time when the second t EB = RB (4)
R EB
tag sent is received from the MT, confirming the
reception of the first segment of the cell stream. Buffer
max
MT C RB
COS RAP new

old t DCOS
t Dd
tD M T
new
t Dd
tD H O
C RB

R dT +1
C

Fig. 5: Downlink propagation delays for the backward HO

HO

This can be calculated as being equal to 2 t oldd + T on

the timescale presented in Fig.6, in which t oldd is the t
old new
2 t Dd dT t DEB 2 t Dd
propagation delay on the old downlink until the
handover permission is received from COS (see Fig.5) Fig. 6: Buffermanagement for STH
and T represents the time interval between two
As stated before, the new cells fill up the buffer,
consecutive tags. This segment and the tag belonging
during retransmission, with the rate RC , until the tags
to it will be deleted from the buffer, as having been
completely received by the MT. The whole process is can be again received in uplink direction. In Fig.6 is
executed periodically and the repetition period equals also presented the worst case in which the MT
T . At the same time, a new segment is stored into the performs the physical handover shortly after the
reception of the last tag in uplink direction, with the
buffer and the old one is deleted from it upon

418
result, that all the cells forwarded from the buffer did
not yet reach the MT. They are now retransmitted with
the increased rate R EB > RC towards MT and
segmentwise confirmed. The buffer is emptied step by
step and the system regains a normal, stable state. The
maximum fill-up level of the buffer can be calculated
out of the formula below:
RC 2 t new
max
C RB = C RB (1 + ) + 2 RC t new + d (5)
T
d
R EB
VI. OAM IMPLEMENTATION OF THE STH
SCHEME
Obviously, the proposed algorithm relies on the
standard ATM OAM functionality [7],[14]. Basically,
the in-band signaling mechanism used in order to
protect data corresponds directly to the OAM
principle, in which connection specific management
and operations information are transmitted inside the
user-data stream. The delay measurement mechanism
used in the STH method for dynamically adapting the
retransmission buffer size is an adaptation of the VCC
OAM cell loopback. The loopback capability of the
ATM-OAM enables the dynamic insertion of both the
intermediated connection points and endpoints and to
be transmitted back by a third, remote point.
This remote point is in the case of STH the MT,
which sends back on the mobile segment the
information generated by the COS. A separate counter
is used to determin the number of user data cells
received on the corresponding VC after the last
received numbering tag. This information is then used
to resynchronize the data stream after the handover.
After the reception of a certain numbering tag sent
back by the endpoint, the COS discards the content of
the buffer belonging to the corresponding connection,
in other words, the cells stored in it prior to the
transmission of the SAI signal. This assures that the
buffer contains only copies of user data cells either
not yet received or not yet acknowledged by the MT.
The SAI process is continuous, in a loop, and is
reinitialized after every successful handover attempt.
Its main purpose is the management and the content-
refresh of the STH retransmission buffer. After the
validation of a handover attempt and the establishment
of the connection between the COS and the new RAP,
the buffer content is transmitted to the new RAP.
Once the MT received the permission for
handover, it has to finish the transmission of user data
cells on the old connection, this being done with the
help of the MDI signal. This OAM cell marks the last
user data cell transmitted in the uplink direction via
the old RAP. Upon receiving the MDI cell, the COS
ceases transmitting on the old downlink and updates
its routing tables to switch over the connection to the
new route via the new RAP and connect the
connections fixed segment to the new mobile
segment. After that, the old mobile segment is shut
419
down. The user data cells contained in the
retransmission buffer are sent to the new RAP
followed by the normal cell stream received from the
fixed segment. Handover completion is signalled by a
HCI OAM cell which has also the role of triggering
the queueing of user data cells following this signal,
until the post handover resynchronisation is initialized
by the MT at the new RAP. The buffer in the new
RAP is activated upon receipt of HCI and starts
storing cells. The MT itself starts the transmission on
the new link, after completing the pysical part of the
handover, by sending also a HCI signal containing the
numbering tag of the last SAI correctly received on
the old downlink together with the value of the
synchronisation counter. In a similar way, the RAP
discards the appropriate number of cells from the
buffer and starts regular transmission in both up- and
downlink direction. A final HCI signal sent on the
uplink by the MT is flagging successful handover
completion to the COS. The described mechanism
protects connections with high requirements regarding
QoS from cell loss or cell sequence mismatch. The
signalling flow used by the STH handover scheme is
similar to the OAM F5 flow. In addition to the
described handover procedure, new signaling
functions are needed for proper handover triggering
and setup or shutdown of the mobile segment.
VC-Segment for handover
COS
MT
RAP
SAI
HCI
MDI
HCI
HCI
OAM flow of the F5 handover segment
Fig. 7: OAM signals of the STH method
Due to the very small cell size, handover are
occuring more and more frequently and they should be
seen as a normal process of continuous improvement
of the radio link rather than an emergency situation.
The aim of every handover scheme should be
therefore to reduce the impact to the connection in
terms of QoS to a minimum. To achieve this goal, the
OAM functionality for handover should be in charge
of in-time recognition of a handover situation and
fast switching of the active connections to the new
acces point RAP, if possible before radio link failure.
This is different than standard ATM fault situations,
where a certain delay responding to alarm indications
is provided in the ATM layer waiting for lower layer
protection mechanism to be activated and at the same
time a short disruption of service in the order of
several hundred milliseconds is regarded as acceptable
due to the singular nature of ATM VPC/VCC faults.
Therefore, the standard OAM fault management is not
suited to handle handover situations, a distinct
handover management being necessary. The
implementation of this can still be realized together
with the ATM OAM functions because, as seen
before, the special requirements are mainly in the field
of the performance of the handover OAM functions.
Table 1: OAM signals for STH handover management
Signal Use for
First cell sent to the new RAP
Handover Complete
for resynchronisation after
Indication (HCI)
successful physical handover
Mobile Detach Indication Last cell sent to the old RAP
(MDI) before handover
Synchronisation and Returned OAM signal
Acknowledgement containing the cell numbering
Information (SAI) information tag
In the particular case of STH handover, an OAM
cell stream per mobile VCC is required, meaning that
the handover management cells are belonging to an F5
type of OAM flows. This in-band signalling handover
management stream is at the same time a segment
oriented stream, having the endpoints at the mobile
terminal MT, access point RAP and respectively,
COS. The OAM signals are shown in Tab.1 and the
message flow for the acknowledgement of received
numbering information and the synchronisation
process during handover is displayed in Fig. 7.
The signal having the most frequent occurrence is
SAI, due to the fact that it is transporting a numbering
label in order to exchange synchronisation points in
the user data stream between COS and MT. The SAI
cells are the one that are inserted periodically into the
user data stream of a VC by the COS. Those cells
contain the sequentially numbered tag defined in the
STH handover scheme. The user data cells arriving
after SAI are copied by the OAM module to the STH
retransmission buffer. The SAI cells have to be
returned to the COS by the endpoint of the mobile
segment of the VC which is the mobile terminal MT.
When receiving an SAI signal, MT would reset the
synchronisation counter, save the numbering tag
received with the SAI signal and send back the SAI
cell to the originating point.
CONCLUSIONS
Based on a mobile ATM reference architecture,
consisting of a wireless access system paired with the
support of mobile end-systems within the ATM
network, and after detailing the handover functions
related to it, a new handover procedure called Sync
Tag Handover STH is being proposed and discussed
in this paper. To justify the need of a lossless
handover scheme, the potential error scenarios are
presented and also the way in which the new
procedure overcomes them. The tag insertion
mechanism is explained in detail together with the
420
administration of the buffer storing copies of the ATM
cells sent in downlink direction on the mobile
segment. Finally, the OAM implementation of the
procedure is presented, underlying the advantages of
this scheme in terms of maintaining QoS and easy
implementation in existing ATM OAM.
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Interface and Network Protocols of the Mobile Broadband System,
IEEE Personal Communications Magazine, August 1996.
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Design Rules for Lightweight Short-Range Wireless

Networks
Axel Sikora1
Abstract The proliferation of mobile computing the control of medium access control (MAC)
devices including laptops, personal digital assistants functionality is eased, due to the inherent
(PDAs), and wearable computers has created an detection of other stations activity.
enormous demand for wireless personal area networks
(WPANs). WPANs originally enabled convenient
the networks may become interconnected.
interconnection of devices around an individual person Cluster and mesh topologies may be
or computer. From this starting-point, a broad variety of implemented, where some stations additionally
new wireless appliances has been developed, allowing provide routing and relaying for the network.
proximal devices to share information and resources. applications become interoperable, as the same
Major fields of application for these wireless short-range data objects are used.
networks are industrial, scientific, and medical (ISM),
but also consumer electronics and smart home
If many applications use the same technology, the
appliances. Many of these applications are very cost-
sensitive, however depend on a high degree of quantity of required chips is severely increased.
interoperability thanks to standardization. This This enables mass production at the silicon
contribution deals with concrete design guidelines to foundries, leading to low cost.
combine these two challenges for IEEE802.15.4 [3] and This enables early scaling of wireless circuitry
ZigBee [5] networks. in order to use newest process technologies.
This again reduces cost in production, but also
Keywords: WPAN, Short-Range Wireless Network, allows the reduction of power consumption.
Design Rules, IEEE802.15.4, ZigBee Monolithic integration becomes profitable only
for high volumes. It allows further decrease of
I. INTRODUCTION cost and power consumption, and of form
factor.
Short-range wireless connectivity is a convenient add- The number of silicon foundries will be
on for many applications, as they can be controlled increased, allowing better choice for system
remotely. However, up to now, mostly proprietary designers and second sourcing.
point-to-point connectivity was offered for closed
systems. This is true for many commercial systems,
If the number of designs is big enough, tools and
e.g. remote control in home and industrial automation,
libraries will be supported. This may concern
and for scientific research, e.g. [4]. With the
network planning and analyses, as well as
upcoming definitions of IEEE802.15.4 [3] and ZigBee
programming tools and libraries.
[5], there is the big chance to use only one network.
Standardization promises huge advantages:
Design houses and consultants invest only in
standardized solutions to address the largest
It allows the use of networks independant of the
possible market.
application. Up to date, there is still a huge
number of networking technologies which are
Security can never be achieved by scrutinity but
dedicated to applications. This holds true for the
by using open solutions being developed and
many wired fieldbus and industrial Ethernet
discussed by the community.
protocols, but also for the various proprietary
wireless protocols, being used in 433 MHz,
868 MHz, and 2,4 GHz-band. If all applications
use a common network technology,

1
Department of Information Technology, University of Cooperative Education Loerrach,
Hangstrasse 46-50, D79539 Loerrach, Germany, e-mail: sikora@ba-loerrach.de

421
However, there are caveats connected with was postponed several times, giving room to the
standardization. argument that standards impede short time to
market.
It leads to additional overhead in the systems, if
functionality has to be implemented just for III. LEVELS OF STANDARD
conformitys sake. CONFORMANCE

It may lead to a longer time-to-market as the

process of standardization implies reconciliation A. Modularity within the standards
and compromises of different market-players.
There are two directions of modularity envisaged in
the IEEE802.15.4 and ZigBee standards. The
In some cases, not the best solutions are
horizontal modularity describes different classes of
standardized, but those most acceptable for all
devices, shown in Fig. 2. This differentiation was
parties in the standardization bodies.
included to allow the optimum design of low-cost
applications with as little overhead as possible. The
Standardized components can be interconnected.
functionality of the different devices and the
Apart from the huge benefit of internets,
constraints of the different device classes is described
interdependency of systems is increased.
in Table 1.

II. WPAN STANDARDS IEEE802.15.4 AND Application e.g. e.g. ...

ZIGBEE Device Type (1) Light Sensor Lighting
Controller

The history of the new standards IEEE802.15.4 [3]

and ZigBee [5] begins in the second half of the ZigBee Logical ZigBee ZigBee ZigBee
Device Type (2) Coordinator Router End
nineties with the discussion of HomeRF Lite. The Device
formerly monolithic approach was then differentiated
into two modules, which are shown in Fig. 1. ZigBee Physical FFD RFD
Device Type (3) Full Function Device Reduced Function
Device
System developer

Applications / Profiles
Application Application Application (1) Distinguishes the type of device from an end-user perspective
Object 1 Object 2 Object 3 Specified in Profiles
(2) Distinguishes the Physical Device Types deployed in a specific
ZigBee Applications Framework API ZigBee network
ZigBee Application Support Sub-Layer (3) Distinguishes type of ZigBee hardware - Based on 802.15.4 RFD
and FFD definitions
ZigBee General
ZigBee Alliance

Operational ZigBee Security Fig. 2: Device Classes in IEEE802.15.4 and ZigBee

ZigBee AFME
Framework Toolbox
(ZB GOF)
Device Classes Characteristics
Pairing/Binding
Addressing Reduced limited to star topology
NLDE-SAP NLME-SAP Function cannot become a network coordinator
ZigBee Network Layer ZigBee Network Layer Device talks only to a network coordinator
Data Entity (ZB NLDE) Management Entity (ZB NLME) any topology
Full Function
MCPS-SAP MLME-SAP capable to become a network coordinator
Device
MAC Common Part MAC Layer Management may talk to any other device
limited to star topology
IEEE802.15.4

Sublayer (MCPS) Entity (MLME; MAC PIB)

PD-SAP PLME-SAP End Node cannot become a network coordinator
PHY Layer Management
talks only to a network coordinator
PHY layer
Entity (PLME; PHY PIB) may route traffic within the network, but may
RF-SAP Routing Device not capable to talk to next networking
hierarchy
Fig. 1. Protocol Stack of IEEE802.15.4 and ZigBee [5] may transfer traffic to next networking
Gateway
hierarchy, but may not be capable to route
Device
traffic within the network
IEEE802.15.4 describes Wireless Medium Access
Control (MAC) and Physical Layer (PHY) Table 1: Characteris of device classes in IEEE802.15.4 and ZigBee
Specifications for Low-Rate Wireless Personal
Area Networks (LR-WPANs). It was approved 12 As the standards follow a layered communication
May 2003 by the IEEE-SA Standards Board. model, vertical modularity allows the implementation
ZigBee specifies network layer, security toolbox of the separate layers. The features and the constraints
and application profile. It is due to be ratified by of these solutions is described in Table 2.
ZigBee Alliance within this year. This schedule

422
 Level of
Characteristics
Standardization
Devices use the cheap and low-power RF
Layer 1:
chips with proprietary L2-protocols
Devices use the IEEE802.15.4-library for
Layer 2: medium access, but use proprietary L3-
protocols
Devices use the ZigBee network
Layer 3: funktionality with own application
protocols
Layer 7: Devices use ZigBee application profiles
Table 2: Characteristics of standardized devices with vertical
modularity
B. Necessity of compromises
Albeit this modular approach first implementations of
the described standards show, that the complexity
even of the smallest system is much higher than
originally anticipated. This holds true especially for
the size of program memory, which currently seems
much too large to fit into an 8-bit MCU with 32 or
64kByte of flash memory together with an application
of reasonable size. As memory footprint continues to
be of major importance to allow lowest cost and
power consumption, the necessity arises to
compromise the complex standards. This is
unfortunately caused by the fact that the modularity
of the above described IEEE802.15.4 is still too
coarse-grained for real life products. As this holds
true for software-based products, the situation clearly
is different for hardware-based solutions. However,
IEEE802.15.4 was defined with a software
implementation in mind, which may complicate
hardware design. Currently, no developments for full
MAC functionality are observed. Up to date, only
partial hardware-accelators are available, e.g.
AES-128 encryption and decryption [1].
It has to be clearly stated that the author is a supporter
of standards. However, the ideas of scientific research
can not be directly implemented in real-life products.
Therefore, this contribution describes rules for the
bottom-line of light versions of the standards, that
reasonably support coexistence and interoperability.
This seems to be ever more important as there are
already various approaches for light versions simple-
MAC-implementations which do not follow these
basic rules.
C. Requirements for light versions
Basic requirements
To propose trade-off to a standard is a dangerous
activity as this may call the whole standard into
question. Therefore, these trade-offs shall follow strict
rules. These rules are now described. In this chapter,
the overall rules are listed, where in the next chapter
the appropriate IEEE802.15.4 extensions are
discussed.
423
Rule 1: Lightweight devices shall not disturb
standard devices.
Rule 2: Standard devices shall understand
messages from lightweight devices.
Rule 3: Lightweight devices shall ignore messages
not included in the lightweigt standard and shall
not be obstracted.
Rule 4: All routines in the lightweight devices
covering parts of the full standard shall comply
with the format and the behavior. This is essential
for a smooth migration path to future
enhancements.
Lightweight MAC protocol
Based on the above, the bottom-line functionality of a
lightweight MAC protocol is described:
Rule 1: All devices shall follow the 802.15.4
frame format, so that all other standard-compliant
devices may understand the messages of the light
devices. This clearly does not impose major
overhead on these devices, as the IEEE-frame
format allows a minimum size of headers:
6 Bytes PHY header are compulsory. Out of
those, 4 Bytes are for synchronization
purposes, which cannot be omitted in any other
non-standard systems.
5 Bytes MAC header are minimum, when
working without any addresses. Out of those, 2
Bytes are for Frame Check Sequence, which
also should not be omitted in any system.
However, it does not seem to be necessary that
light-devices understand full-blown systems. This
approach can be observed in many other
networking standards, e.g. in CAN-standard [2],
where systems with extended 29-Bit long
addresses (V2.0B compliant) may be intermixed
with older systems with their 11-Bit long
addresses.
Rule 2: All devices shall understand 802.15.4
beacon frames. However, it does not seem to be
necessary to implement all options. IEEE802.15.4
enables reliable networking with an enhanced
processing for orphaned devices with many
options that blow up the memory footprint. In
simple network topologies without enhanced real-
time requirements, the same reliability can be
achieved with the use of watchdog timers and re-
transmission of association requests.
Rule 3: All devices shall support CSMA/CA-
medium access as defined in IEEE802.15.4 non-
slotted access. This is a major retrenchment as the
conformity to slotted access sets high
requirements on the real-time capability of
devices. The slotted medium access is
 synchronized with the beacons that define REFERENCES
contention access periods (CAP) and contention
[1] http://www.chipcon.com/files/CC2420_Data_Sheet_1_2.pdf
free periods (CFP). However, if the guaranteed
[2] Road vehicles -- Controller area network (CAN) ISO
time slots (GTS) in the CFP are not kept free, as Standard 11898; http://www.iso.org
some light devices do not run a time-based access [3] http://www.ieee802.org/15/pub/SG4a.html
scheme, this has a destructing impact on the [4] http://webs.cs.berkeley.edu/
[5] http://www.zigbee.org.
quality of service in the slotted network.
Therefore, it seems to be necessary, that a device
not supporting slotted access detects a beacon
with GTS definition, should leave this channel by
selecting another channel (rule 4).

Rule 4: All devices shall support a dynamic

channel selection (DCS) to ease coexistence as
much as possible. Unfortunately, DCS is described
only in ZigBee standard. Nevertheless, it seems to
be indispensable that light devices with no support
for slotted access leave the channel as fast as
possible.

D. Light ZigBee protocol?

As the ZigBee standard is not yet ratified, it is still too

early to desribe possible lightweight ZigBee rules.
However, it seems to be understood by the ZigBee
alliance that fine-grained vertical modularity is of
major importance for the market success. This can be
illustrated with routing. Routing normally is a
functionality that may be done with a limited program
code, but with higher consumption of data tables. For
the ZigBee-standard, it is currently envisaged to run
three routing levels:
Non-routing devices (end nodes).
Minimum routing nodes (RN-), that have no
routing table and engage in limited route
discovery.
Full routing nodes (RN+), that have routing tables
and engage in route discovery to fill it.
The same holds true for security solutions. However,
it is highly questionable, if this approach is useful.
Encryption algorithms, e.g. AES-128, call for
hardware implementations, especially when the
host-processor is a low-frequency 8-Bit-MCU. For
these hardware accelerators, a uniform solution
with long encryption keys may mean less cost than
a modular approach with different key lengths.
Short encryption keys, i.e. 64 or even 32 Bits for
symmetric encryption, do not provide security.
This is especially the case for future-proof systems
that shall be operation for decades.

IV. CONCLUSIONS

It is reasonably possible to design lightweight

wireless short-range devices already today which have
good a migration path, minimum impact on full
standard-compliant networks and additionally offer
lowest cost and power-consumption. This article
described the most important rules the design of those
lightweight systems.

424
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Performance Evaluation of Single-Carrier Broadband

Transmission with Frequency Domain Equalization
Marius Oltean1, Andy Vesa2, Eugen Marza3
Abstract One of the most challenging problems in
high-rate data transmissions is to combat the inter-
symbol interference introduced by the multipath
propagation. Since adaptive equalization in time domain
proved oftentimes to be a too expansive solution in terms
of computational complexity and implementation costs,
some different equalization approaches must be
considered. In this paper we shall investigate the
performance of two recent-years findings in this
direction, i.e. OFDM (Orthogonal Frequency Division
Multiplexing) and SC-FDE (Single Carrier with
Frequency Domain Equalization).
Keywords: OFDM, SC-FDE, equalization, wireless
I. INTRODUCTION
Wireless transmission systems are increasingly
required to provide high-rate data transmission with
high bandwidth efficiency. In radio-channel
communications, the transmitted signal is subjected to
multipath fading caused by phenomenon that affect
the radio-wave propagation: reflection, diffraction and
scattering. As a result of these phenomenon, the
transmitted signal arrives at the receiver following
different paths, with different strengths and delays.
Moreover, oftentimes in mobile radio
communications, the receivers mobility and the
motion of the objects situated between transmitter and
receiver determine the temporal variation of the
multiple radio propagation paths. Consequently the
radio channel is far from an ideal data transmission
medium, exhibiting both frequency selectivity and
time-variant characteristics.
Multipath propagation causes time dispersion of
the signal. Thus, the energy corresponding to a
symbol overlaps with the energy corresponding to the
subsequent symbols, the inter-symbol interference
(ISI) arising. One of the most challenging issues in
radio communication is to overcome this
phenomenon, which, moreover, especially impairs the
transmission when high-data rates (short duration of
the information symbols, respectively) are demanded.
The classical approach in order to mitigate this
problem is the use of adaptive equalization techniques
at the receiver, but there are practical difficulties in
operating this equalization in real-time conditions at
several Mb/s and with compact, low-cost hardware
[1]. In fact, the different types of time-domain
equalizer structures, e.g. the MLSE (Maximum
Likelihood Sequence Estimator), the linear filter
equalizer and the DFE (Decision Feedback Equalizer)
all have the common problem that the computational
complexity grows at least quadratically with the
desired bit rate, becoming totally unattractive [2].
Habitually, these equalizers are used in classical
single-carrier schemes, when the data symbols
(amplitude and/or phase modulated pulses which will
in turn modulate a sinusoidal carrier) are sequentially
transmitted one-by-one, the frequency spectrum of
each symbol allowed to occupy the entire available
bandwidth.
In this paper we will study two of the recent years
approaches that counteract the ISI phenomenon in a
distinct manner: OFDM (Orthogonal Frequency
Division Multiplexing) and SC-FDE (Single-Carrier
with Frequency Domain Equalization). OFDM splits
the high-rate data stream into a number of lower
speed substreams, each of them modulating a different
carrier. The multiple subchannels that carry the data
do not interfere each other since the correspondent
subcarriers are all orthogonal one-another. Modern
Fast Fourier Transform (FFT) processing techniques
are used to generate and demodulate the signal.
OFDM appears to offer a better
performance/complexity trade-off than conventional
SC modulation with time domain equalization, since
the equalization complexity merely grows slightly
than linearly with the bit rate [3].
Because the transmitted OFDM symbol is a sum
of a large number of slowly modulated subcarriers, it
has a high peak-to-average power ratio, even if a
constant envelope modulation (such as BPSK or
QPSK) is performed on each subchannel. This

1
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail marius.oltean@etc.utt.ro
2
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail andy.vesa@etc.utt.ro
3
Facultatea de Electronic i Telecomunicaii, Departamentul
Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara, e-mail eugen.marza@etc.utt.ro

425
generates difficulties regarding the RF power [Xk]
Signal [xk ] CP [xcp k ]
amplifiers that must be used in a practical IFFT Channel
mapping insertion
implementation. The transmitter power amplifier

requires an increased power back-off for obtaining a [ycp k ]
[Xk ] Channel [Yk ] [yk] CP
wide range of linearity in order to faithfully reproduce invert
FFT
extraction
all the peaks of the signal envelope, which will
significantly increase the cost of this component.
Detection
SC-FDE is an approach that counteracts this
inconvenience, keeping a low complexity of the
whole equalization process, similar to that of OFDM. Fig.1: Block diagram of an OFDM system
The data are transmitted using a single carrier with a
classical modulation scheme, which will eliminate the The data are first grouped in blocks of M bits.
problem of an expensive peak-to-average power ratio. Each m bits are then mapped to one of 2m complex
The whole complexity of the equalization is valued symbols Xk, k=0,1,,N-1 using a digital
distributed at the receiver and is still based on FFT modulation scheme.
processing. The basic mathematics of this approach The N time domain signal samples forming
are practically the same as in OFDM, as we will se in an OFDM symbol (x0, x1, ..., xN-1) are obtained
the next section. A frequency domain - linear or through IFFT processing according to (1):
adaptive - equalization must be performed in order to
counteract the inherent time-dispersive nature of the 2
N 1 jk n
radio channel. x[n] = X[k] e N , n = 0,1,..., N 1 (1)
k =0
In the next section of the paper we will study the
equalization concepts proposed by the two
approaches, which moreover could efficiently be used Afterwards, L-1 cyclic prefix samples are
simultaneously in a dual-mode. The BER added in front of the signal sequence, that becomes:
performance of both methods is investigated in the [xcp]=(xN-L+1, xN-L+2,...,x0,..., xN-1). This signal is
section III by means of computer simulation. The analog converted and modulates a RF carrier, before
relevant conclusions regarding the two methods are being transmitted in the channel. If we consider the
outlined in the final section. equivalent baseband discrete model of the channel as
a FIR filter of order L, then the Z-domain channel
II. OFDM AND SC-FDE: BASIC CONCEPTS response is given by:

A. OFDM
OFDM represents an optimized version of
the multicarrier modulation techniques. The finding of
this approach was to replace a single-carrier serial
transmission at a high data rate with a number of
slower parallel data streams that will simultaneously
modulate orthogonal carriers. By creating N parallel
substreams, the bandwidth of the modulation symbol
is reduced by the same factor, or, equivalently, the
duration of the modulation symbol will be N times
higher. The lengthening of the transmitted symbol
will significantly reduce its sensitivity to ISI. The N
parallel transmission channels do not interfere each
other, since their correspondent subcarriers are
orthogonals. This is in fact the basic idea that lies
behind OFDM. The generation of the multiple carriers
is done by performing Inverse Fast Fourier Transform
(IFFT) processing at the transmitter. To the receiver,
data are recovered using FFT processing, which
extracts the subcarriers. In addition, a cyclic prefix is
inserted in front of each symbol, in order to prevent
two consecutive blocks to interfere because of the
time-dispersive channel character and, furthermore, to
facilitate the equalization process to the receiver. In
the figure 1, the block diagram of an OFDM system is
shown.
L 1
H (z) = h[n ] z n (2)
n =0
Since we must assume our channel to be time
variant, its impulse response will depend on the time
at which the impulse is applied. We shall assume,
however, that the channel impulse response is static
(will not change) for the duration of an IFFT frame.
The equivalent baseband signal at the
channel output can be obtained by the well known
operation of convolution:
y cp [n ] = x cp [n ] * h[n ] (3)
Discarding the L-1 CP samples from the
received sequence, the remaining (useful) signal can
be expressed as:
y[n]=x[n]h[n] (4)
where denotes the circular convolution operator.
The relation above is in fact the main reason that lies
behind the use a cyclic prefix. Its insertion will
transform the convolution between the data sequence
and the channel impulse response into a circular
convolution [4], which preserves the temporal support
of the signal, thus avoiding the interference of two
successive OFDM symbols due to the time dispersive

426
nature of the channel. If the cyclic prefix duration
spans more than the multipath delay spread of the
channel, the interference from the previous
transmitted blocks is totally eliminated through this
operation of CP insertion/extraction. Furthermore, the
equalization process is facilitated at the receiver, a
simple channel inversion allowing theoretically a
perfect data recovering, as we will see next.
Since x[n]=IFFT{X[k]} and taking into account
the FFT demodulator, the received symbols Y[k] can
be expressed as:
Y[k]=FFT{IFF T{X[k]}h[n]}
But, the FFT of a circular convolution of two
discrete time signals yields a spectral multiplication:
Y[k ] = FFT{IFFT{X[ k ]}} FFT{h[n ]} = X[k ] H[ k ],
k = 0,1,..., N 1
where H[k] represents the sampled frequency
response of the equivalent discrete channel,
corresponding to the frequencies k=k(2/N). The
crucial consequence of the relation above is that the
modulation symbol X[k] could be recovered at the
receiver by a simple pointwise division operation,
commonly referred to as a one-tap frequency domain
equalizer. Thus, the CP theoretically eliminates both
IBI (each block preserves its temporal support) and
inter-carrier interference (ICI) (each serial symbol
received on the k-th carrier will depend only on the
corresponding k-th carrier transmitted symbol, not
being affected by the adjacent carriers).
B. SC-FDE
SC-FDE is an alternative equalization approach,
which eliminates some of the OFDM disadvantages
(especially the high peak-to-average power ratio),
while keeping approximately the same low
complexity. The block-diagram of a SC-FDE system
is illustrated in the figure 2.
Signal [xk ] CP [xcp k ]
Channel
mapping insertion
[Xk ] Channel [Yk ] [yk] CP
IFFT FFT
invert extraction
[ x k ] Detection
Fig.2: Block diagram of a SC-FDE system
The data are still transmitted in blocks of N
samples, but this time the transmission is a classical
serial single-carrier transmission. The data are
encoded using a digital modulation scheme, obtaining
the sequence [xk], where k=0,,N-1. A cyclic prefix
is added in front of each data block. If the cyclic
prefix duration is longer than the multipath delay
spread of the channel, the interference introduced by
temporal dispersion of the previous transmitted block
is totally absorbed by the circular extension, which is
discarded to the receiver. Moreover, since the
equalization process is still based on FFT processing,
the appearance of periodicity that cyclic prefix
confers to the signal will facilitate this process. The
mathematics of this method are essentially the same
as for OFDM with the difference that in SC-FDE, all
the equalization complexity is allocated to the
receiver. The data block [yk] arrived to the receiver is
(5) first FFT-processed, then the influence of the
frequency-selective channel impulse response is
eliminated by an simple channel inversion operation.
In adaptive SC-FDE, the adaptation of FDE transfer
function can be done using least mean square (LMS),
root least square (RLS), or least-square minimization
methods. An inverse FFT returns the equalized signal
in the time domain prior to the detection of data
(6) symbols. In terms of complexity, for channels with
severe delay spread, SC-FDE is simpler than time
domain equalization, because equalization is
performed on a data block at a time, using a
computationally efficient FFT algorithm.
Furthermore, as proposed in [2], OFDM and SC-
FDE could simultaneously be used with high
efficiency in a transmission modem. It is easy to
notice from the presented block-diagrams that the two
types of systems mainly differ in the placement of
IFFT operation. A system in which a radio modem
can be configured to work in both OFDM and SC-
FDE mode could be simply implemented by
switching the IFFT block between transmitter and
receiver. As observed in [2] since SC-FDE
concentrates all the complexity to the receiver, such a
system could be appropriated for an uplink, thus the
most complex part of the communication issues being
solved by the base station. Using OFDM in downlink
will reduce the complexity of the processing that must
be done by the mobile station. Such an arrangement
has two obvious advantages: concentrates the main
amount of processing in the base station and reduces
the power consumption of the mobile station that uses
a single-carrier mode for transmission and an OFDM
mode for reception.
III. EXPERIMENTAL RESULTS
[ycp k ]
BER performances of both OFDM and SC-FDE
systems were studied by means of computer
simulation. Both Rice (Line-of-Sight) and two-ray
Rayleigh (Non-Line-of-Sight) conditions with perfect
channel knowledge were taken into account in order
to simulate the signal propagation in the radio
channel. As a parameter of the simulation, the relative
power of the two multipath components denoted by P1
and P2 in a Rayleigh fading channel was considered.
The Rice factor K, defined as the ratio between the
power of line-of-sight deterministic signal and the
power of the multipath components is also modified
during the simulations. The fading was modeled as
427
quasi-static that is it remains unchanged during the
transmission of a data block. For the simulated coded
transmissions, a BCH encoder with rate 1/2 was
implemented. This block-code can correct 6 bits in a
block of length N=64, as was taken in all the
simulations. The binary information is mapped using
two classical constant-envelope modulation schemes:
QPSK and DBPSK respectively.
In the figure 3, a comparison of BER
performances of OFDM and SC-FDE methods is
illustrated. One can observe that for low values of
SNR, OFDM performs generally better than SC-FDE,
especially for coded transmission. When SNR
exceeds the critical range of 12dB to 15dB, SC-FDE
becomes more reliable, mainly when uncoded data
blocks were transmitted. This confirms that, while
desirable, the coding is not strictly imposed in single
carrier method, unlike in OFDM where the coding is
mandatory in order to combat the high amount of
errors on the carriers attenuated by the frequency-
selective channel response.
Fig. 3: A comparison of OFDM and SC-FDE with
QPSK modulation, Rayleigh fading channel, P1=P2
In the figures 4 and 5 the influence of the Rice
factor (K) and of the relative power of the second
multipath (P2) on BER performance of both systems
is aimed. It is obvious that the two systems behave
Fig. 4: The influence of P2 and K parameters on
BER performance of an QPSK-OFDM system
428
almost identically in respect to these parameters. BER
performance slightly improves when a LOS
component is introduced, or when the relative power
of the second multipath decreases. The effectiveness
of the last parameter becomes clear especially for high
SNR values, when the spread of the results is within
about 1dB (for SC-FDE) and 4dB (for OFDM). Thus,
the OFDM system proved to be slightly more
sensitive to the considered parameters than the other
system.
Fig. 5: The influence of P2 and K parameters on
BER performance of a SC-FDE with QPSK system
The influence of the cyclic prefix length for the
performance of both studied systems is shown in the
figure 6. As mentioned previously, if the cyclic prefix
Fig. 6: The influence of ndelay parameter on BER
performances of both OFDM-QPSK and SC-FDE
QPSK systems
spans more than the channel impulse response
duration, the interference introduced by the previous
transmitted blocks is entirely eliminated [5]. In order
to study the influence of cyclic prefix duration, the
parameter ndelay is defined as the ratio of the
multipath delay spread of the channel and the cyclic
prefix duration. If in the past simulations we avoided
this issue by taking a unit value for ndelay, this time
we also study the case when the multipath delay
spread of the channel spans two times the duration of
the cyclic prefix. It is obviously regarding the figure 6
that SC-FDE is generally more sensitive than OFDM
to a cyclic prefix that is insufficient in order account the modulation method used. As expected,
counteract the time-dispersive nature of the channel. DBPSK generally performs better than QPSK.
Though, one can notice that for high values of SNR,
SC-FDE overcomes OFDM.
All simulations were repeated using a differential
modulation scheme, namely DBPSK. In the figure 7,
BER performance of both systems with and without
encoding is illustrated.

Fig. 9: The influence of P2 and K parameters on BER

performance of a SC-FDE with DBPSK system

Fig. 7: A comparison of OFDM and SC-FDE with

DBPSK modulation, Rayleigh fading channel,
P1=P2
The figure 7 confirms the conclusions previously
inferred for QPSK modulation. The critical range
where SC-FDE becomes more reliable moves 1-2dB
to the left when compared to the first case.
The figures 8 and 9 emphasize the influence of K
and P2 parameters on both presented methods. The
same insignificant spread of the results like in case of
QPSK modulation can be observed.

Fig. 10: The influence of ndelay parameter on BER

performances of both OFDM-DBPSK and SC-FDE
DBPSK systems

Fig. 8: The influence of P2 and K parameters on

BER performance of a DBPSK-OFDM system

In the figure 10 the influence of the cyclic prefix

length is studied. The conclusions that can be drawn
slightly differ from those observed in the case of
QPSK modulation. SC-FDE drastically improves its
performance at high SNR values, despite the
insufficient duration of the cyclic extension.
In the figures 11 and 12, a comparison of the
performances of both systems is done, taking into
Fig. 11: A comparison of DBPSK and QPSK
modulations when used in an OFDM system
There are though some refinements that make the
difference between SC-FDE and OFDM. While in
both QPSK and DBPSK used with OFDM the coding
gain is approximately the same (within 9dB to 10dB),

429
SC-FDE seems to not improve in the same manner its
performances. Thus, the coding gain of SC-FDE
system with QPSK modulation is significantly inferior
to the same system used with DBPSK modulation.
Furthermore, the coded SC-FDE systems, as
previously seen, performs generally worse than coded
OFDM, observation confirmed by the performance
curves in the figure 12. Thus, the coding gain in this
case covers a range of values between approximately
4dB and 8dB.
REFERENCES
[1] D. Matic, OFDM as a possible modu lation technique
for multimedia app lica tions in the range o f mm waves ,
available on-line at:
w ww .u bicom. tudelf t.nl/M MC/D ocs/introOFDM .pdf
[2] M. Huemer, A. Koppler, L. Reindl, R. Weigel, A Review of
Cyclically Extended Single Carrier Transmission with Frequency
Domain Equalization for Broadband Wireless Transmission,
European Transactions on Communications (ETT), Vol. 14, No. 4,
pp. 329-341, July/August 2003
[3] D.Falconer, S.L. Ariyavisitakul, A. Benyamin-Seeyar, B.
Eidson, Frequency Domain Equalization for Single-Carrier
Broadband Wireless Systems, IEEE Communications Magazine,
pp. 58-66, April 2002
[4] Chini, A., Analysis and Simulation of Multicarrier Modulation
in Frequency Selective Fading Channels ,Ph. D. Thesis, 1994,
Chapter 3
[5] Werner Henkel, Georg Taubck, Per lding, The cyclic prefix
of OFDM/DMT an analysis, 2002 International Zrich Seminar
on Broadband Communications, February 19-21 Zrich,
Switzerland

Fig. 12: A comparison of DBPSK and QPSK

modulations when used in a SC-FDE system

IV. CONCLUSIONS

In this paper, a comparison of two of the recent

years equalization approaches, OFDM and SC-FDE,
was performed. The basic mathematics of these
equalizations methods were studied, and it is shown
that they are essentially the same. Furthermore, the
two methods offer approximately the same conditions
in terms of computational complexity and signal
processing involved. Though, when compared to
OFDM, SC-FDE seems to provide some advantages
regarding its practical implementation. The BER
performance of both systems in similar fading
conditions is investigated in the section III by means
of computer simulation. Both Rayleig and Rice fading
conditions were taken into account for modeling of
the radio channel. The performances of the two
systems are essentially the same in the considered
conditions. There are yet some differences that could
be observed. While uncoded SC-FDE generally
performs better at important values of SNR, when
coded, OFDM becomes more reliable. If a cyclic
prefix that cannot entirely combat the time dispersion
of the signal introduced by the channel is used,
OFDM performs generally better than SC-FDE. Both
systems slightly improve their performances when a
LOS component appears or when the power of the
delayed multipath signal decreases.

430
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS

Tom 49(63), Fascicola 2, 2004

Some aspects about frequency hopping radio networks

Paul Bechet1, Stefan Demeter2, Radu Mitran3, Simona Miclaus4
Abstract As part of the management activity specific wall spectrum, so that some spectral power will be
for modern frequency hopping radio systems, the radiated into the adjacent channels (figure 1).
interference studies include the Adjacent Channel
Interference (ACI) level and the Co-Channel
Interference (CCI) level. The paper is focused on the
study of these interference levels and draws some
conclusions regarding the use of FH radio systems. We
will compare the use of radio hopping systems in two
representative operating modes: fixed secure frequency
and frequency hopping. The final aim of the paper is to
improve the electromagnetic compatibility in a complex
radio network, by offering a guide to facilitate the
assignment and allocation of the radio resources.
Keywords: frequency hopping, interference,
electromagnetic compatibility.

I. INTRODUCTION
The term battlefield spectrum management refers to
managing electromagnetic spectrum resources in
order to support telecommunications (including
weapon systems) and electronic warfare (EW)
requirements [1]. This type of management includes
allocating and assigning generated frequency
resources and the distribution of the variables for
frequency hopping (FH) radio systems. The items of
the management include frequencies, TSK
(Transmission Security Keys) variables, net
identifiers, COMSEC (Communications Security)
variables, and time. However, equipment parameters
impose some constraints on the distribution schemes
for TSK, net identifiers, and frequency allocation
schemes of the hop sets.
Radio frequency interference is inherent in all
wireless systems and is one of the most significant
design parameters of cellular and other mobile radio
systems.
This paper investigates the power efficiency
performance of frequency hopping radio systems
operating in an adjacent channel interference (ACI)
environment.
Adjacent Channel Interference (ACI) is caused by
modulation, filter and radio design imperfections.
Transmitted signal is not band-limited to a brick
Fig. 1. Adjacent channel interference
Interference power caused by the first upper and
lower interfering signals is designated as ACI-1 and
the one caused by second signals is designated as
ACI-2.
It is important to know the values of the spacing
between channel (W) and bandwidth of the receiver
filter. In several system specifications the receiver
filter is assumed to be a brick wall filter [2].
The integrated (total) ACI power can be expressed as
[2]:
S y ( f ) H ( f w)
2
df
ACI = A (WTb ) = (1)
S y ( f ) H ( f ) df
2
where: Sy(f) is the power signal density (PSD) of the
signal; H(f) is the receive filter transfer function; w =
WTb is the normalised carrier spacing between
adjacent channel.
The method used to assess the ACI levels is a
practical one and is based on measuring this
parameter in two representative operating modes: the
fixed secure frequency mode and frequency hopping
mode.
The instrument we used is a R&S FSH3 Spectral
Analyzer [3]. Comparing the results we can provide

1
coala de Aplicaie pentru Transmisiuni, Informatic i Rzboi Electronic,
Departamentul de Cercetare, Bd. Vasile Milea 2-3, Sibiu, e-mail Paul.Bechet@personal.ro
2
Academia Forelor Terestre, Str. Revoluiei 3-5, Sibiu, e-mail demeter@actrus.ro
3
Centrul 196 Rzboi Electronic, Buzu
4
Academia Forelor Terestre, Str. Revoluiei 3-5, Sibiu

431
an efficient allocation of the resources for frequency
hopping radio systems.
II. RESULTS AND DISCUSSION
The results are representative for a simple tactical
network where the power of the frequency hopping
systems is uniform distributed in the coverage area.
Fixed secure operation mode is one specific for
frequency hopping systems due to the spectral
performance in the critical cases of propagation like
urban or mountain areas. The levels of ACI are
presented in figure 2 and more details are shown in
table 1.
hopping mode. Using a different frequency position in
the narrow hopping band, ACI level values can be
observed for a variable operation frequency of the
equipment. Due to the limited spectral analyzer
capability we are focused only to the results between
two frequencies of the successive hopps. The values
of these frequencies are 35 MHz and 36 MHz. For a
hop speed of 100 hops/s, it is necessary to set the
acquisition time of the analyzer to 20 ms. The ACI
levels are presented in figures 3, 4 and figure 5. More
details are shown in table 2.

Fig. 3. The ACI levels in the Hopping Frequency Mode

at a frequency distance of 500kHz and 1.5MHz.

Fig. 2. The ACI levels in the Fixed Secure Frequency Mode

Table 1
Frequency Attenuation
distance (dB)
(kHz)
100 58.5
300 69
500 72.8
526 60.8
600 73.9
626 56.6
700 75.2 Fig. 4. The ACI levels in the Hopping Frequency Mode
780 56.3 at a frequency distance of 800kHz and 1.8MHz.
800 76.3
1040 56.4
1100 77.9
1400 78
1600 78.4

We can observe maximum ACI levels for several

frequency distances. The frequency positions of these
levels remain constant for variable value of the
hopping system operation frequency. The spectrum
manager must avoid these frequencies.
A secure operation mode, specific for a jamming
environment, is the frequency hopping one. The
equipment capabilities provide narrow or wide band
Fig. 5. The ACI levels in the Hopping Frequency Mode
hopping modes. Our interest is only to estimate the at a frequency distance of 1.5MHz and 2.5MHz.
ACI level, so it is recommended to use the narrow

432
 Bar frequency band

Table 2
Frequency Attenuation 2 MHz 2 MHz

distance (dB)
(kHz)
200 0.3
Fixed secure frequency Hopping frequency Fixed secure frequency
300 0.9 band
500 4.4
700 6.2 Fig. 6. An analysis of two radio networks, one
operating in the fixed secure mode and the second
800 8.9 operating in the frequency hopping mode
900 14.5
1100 18.1 List of frequency hopping network
1200 20.9
f1 f2 f3 f4 fn
1300 25.4
1400 29.3
1500 34 2 MHz 2 MHz
1600 37.9
1700 43
2000 50
Fixed secure frequency network
Comparing with fixed secure mode, one can observe
increased values of the ACI levels. An acceptable 40
Fig. 7. An analysis of two radio networks, one
dB ACI level requirement imposes a minimum of 1.7 operating in the fixed secure mode and second
MHz frequency distance. This means considerable operating in a list of frequencies hopping mode
constraints in the process of managing the radio
resources.
the case of inserting the frequency value of the fixed
III. CONCLUSION secure network.
The situation that we considered in this paper is valid
In this paper we investigate the ACI level for only in the case of uniform power of the frequency
frequency hopping systems in two representative hopping equipment.
operating modes: fixed secure frequency and
frequency hopping.
Fixed secure mode provides good spectral efficiency REFERENCES
but is easy to detect by the enemy.
Frequency hopping mode is less power efficient but [1] *** - FM 11-32 , Combat Radio Operations, Headquarters
provides ECCM (Electronic Counter Counter Department of the Army Washington, DC, 1999.
Measures) communications. [2] K. Feher, Wireless Digital Communications: Modulation &
Spread spectrum Applications, Prentice-Hall, New Jersey, 2000.
In the process of resources planning it is [3] *** - Operating Manual: Handheld Spectrum Analyzer R&S
recommended to combine the advantages of both FSH3, 2001.
operation modes, so we expect to have, in same area,
complex radio networks.
A simplified situation is presented in figure 6 where Acknowledgements:
the spectrum manager must allocate the frequency Present work was supported by a CNCSIS Grant / 2004 from the
resource between two radio networks, one operating Romanian Ministery of Education and Research to the Military
on a fixed secure frequency and second one operating Application School for Communications, Informatics and
Electronic War, in Sibiu.
in the frequency hopping mode.
A minimum 2 MHz frequency distance, upper or
lower, is requested between the frequency resources
of the radio networks, if a 40-dB ACI level
performance is considered.
Another situation is the one represented in figure 7,
where the frequency hopping radio network is a
particular one, operating with a list of frequencies. A
minimum of 4 MHz frequency distance is
recommended between two consecutive frequencies
of hopping list. This requirement is necessary only in

433
 Buletinul tiinific al Universitii "Politehnica" din Timioara
Seria ELECTRONIC i TELECOMUNICAII
TRANSACTIONS on ELECTRONICS and COMMUNICATIONS
Tom 49(63), Fascicola 2, 2004
Recognition of OFDM modulations : approach based on
high-order time-frequency methods
Marius Salagean1, Cornel Ioana2, Andre Quinquis3
Abstract Nowadays a deep interest for processing
of different types of digital modulations (FSK,
PSK, QAM and OFDM) exists. This paper studies
a technique for discrimination of the OFDM
modulation. The approach is based on the high-
order time-frequency representation and filtering
process with orthogonal filters.
Keywords: OFDM, time-frequency, orthogonal filters.
I. INTRODUCTION
The identification of the digital modulation type of a
signal has found applications in many areas, including
electronic warfare, surveillance and threat analysis.
Recognition of communication signals became an
independent discipline in the electronic warfare. The
goal is to intercept, analyze, classify and, eventually,
understand the message carried out by the
communication signals.
In this paper a potential method is proposed to
classify different types of numeric modulations. In
section II the characteristics of the PSK, QAM, FSK,
and OFDM modulations are described. In section III
the time-frequency representations based on
polynomial phase signal processing are presented.
The new method for recognition of the OFDM
modulations is illustrated in section IV. Furthermore,
in section V some simulation results are depicted.
Section VI will close this communication.
II. DIGITAL MODULATIONS: PSK, QAM,
FSK, OFDM
The class of PSK (Phase Shift Keying) modulations is
widely used in numeric satellite TV broadcasting,
being very robust to perturbations. The general
equation of PSK is given by:
mPSK (t ) = (t kT )cos(2f t +
k
c k ) (1)
1
Facultatea de Electronic i Telecomunicaii, Departamentul Comunicaii Bd. V. Prvan Nr. 2, 300223 Timioara,
e-mail marius.salagean@etc.utt.ro
2
ENSIETA, Departement E3I2, 2 rue Francois Verny, 29806, Brest - France, e-mail ioanaco@ensieta.fr
3
ENSIETA, Departement E3I2, 2 rue Francois Verny, 29806, Brest - France, e-mail quinquis@ensieta.fr
434
where the symbol k is 0 + (2m + 1) with 0 m
M
(M-1) (the common value of 0 is 0).
The QAM (Quadrature Amplitude Modulation)
modulation is defined as:
mQAM (t ) = k
k cos(2f c t + k ) (2)
where:
b
k = a k2 + bk2 et k = arctan k (3)
ak
It is clear that QAM signals contain a phase as well as
an amplitude modulation. Compared to PSK signals,
one distinction in QAM signals is that it does not have
constant amplitude. This type of modulation is used in
high-speed modems.
The FSK (Frequency Shift Keying) modulation can be
considered as a non-linear frequency modulation. It
can be represented as:
m FSK (t ) = cos(2f c t + (t )) (4)
where (t) depends on the integral of modulator
signal. Thus, the frequency varies with the message
and the information is carried out by the instantaneous
frequency.
In comparison with the modulations described
previously, which are single-carrier modulation
techniques, the OFDM (Orthogonal Frequency
Division Multiplex) modulation is a multiple-carrier
technique. OFDM is a method that allows to transmit
high data rates over extremely hostile channels at a
comparable low complexity.
The OFDM spread spectrum technique distributes the
data over a large number of carriers that are spread
regularly over a frequency band. This spacing
provides the "orthogonality" in this technique
resulting in a high spectral efficiency. Thus, the
complex signal is represented by:
M 1 N 1
mOFDM (t ) = X
l =0 k =0
k e j 2f k t e j 2f c t p(t lT ) (5)
 Using the ml-HIM concept (relation (6)), Barbarossa
where Xk represents the k-th information symbol,
and al [3] introduced the Product HAF: the ml-HAFs
fk=f0+kf stands for the k-th subcarrier and p(t)
computed, via relation (7), for different lag sets:
represents the pulse shaping function. N and M are the
total number of subcarriers and total number of
(l ) (l )
transmitted blocks respectively.
OFDM has become very popular as it is used in such
T = K 1 ;
K 1
= { i }i = 1, K 1 (12)
l = 1, L
major applications as digital subscriber lines (DSL)
and digital audio broadcasting (DAB). are multiplied, obtaining also a more robust method
and a cross-terms free representation:
III. HIGH-ORDER TIME-FREQUENCY
METHODS K 1 ( l )
i
L
( l)

PHAF ( ; T ) = mlHAFK s ; K 1
i =1
, (13)
It is well known that there is no transformation from K 1
i
l =1 (1)
the Cohen's class that can produce the complete
i =1
concentration along the instantaneous frequency (IF)
when this one is a nonlinear function of time. Still, the effect of error propagation remains a serious
Therefore, different high order distributions have been limitation of the PHAF when we try to estimate a
developed in order to better match the non-linear deeply non-linear IF laws (underwater transitory
time-frequency behavior of the analyzed signal. On signals, digital modulations, etc). Therefore, in [5] is
the other hand, the polynomial phase signal (PPS) proposed a new procedure for polynomial order
constitutes a good model in a variety of applications, compensation, the WarpCom method, based on
e.g. radar imagery, mobile communication systems, unitary transform phase reducing [4].
etc. Let consider a signal modeled by a Kth order PPS
As it was illustrated in [1], [2], the classical HAF (relation (8)). Using a modern version of the HAF
algorithms present some limitations, related to the (PHAF operator or the approach proposed in [3]), we
noise robustness, the cross-terms presence and the obtain an accurate estimate of the Kth order
effect of the error propagation. In order to solve the polynomial coefficient, denoted by a$ K . With this
first two aspects, the multi-lag HAF (ml-HAF) value, we construct the following time axis warping
concept has been initially proposed in [2]. In fact, the function:
ml-HAF is based on the generalization of the high t
1/ K

t w( K ) = w K (t ) = (14)
wK
order instantaneous moment HIM [2]: w K : t
a K

MK[x(t); K-1] = MK1[x(t +K1); K-2]MK* 1[x(t K1); K-2] (6)
The effect of the associated unitary operator U (14) on
where i = ( 1 , 2 ,..., i ) . Applying the FFT to the PPS is depicted by:
1/ K
K

[ ]
(1), we obtain the ml-HAF of the signal x(t) :
K 1
(K ) m
(UK y )(tw(K ) ) = A~ exp jaK t


exp j am tw

a K m =0


m lH A F K [ s ; , ] = H IM K s ( t ) ; e j t d t (7)

(15)
[ ]
Assuming a PPS model for the analyzed ~ K 1 m
a
= A e xp j am t w(K ) exp j K t
signal, i.e.: m = 0 a K
14 442444 3 14 4 244 3
K SPP oforder (K 1)th residual
s ( t ) = A exp j ( t ) = A exp j ak t k (8)
k =0
1 / K 1
the main property of HIM is that, the Kth order where ~ 1 t
(16). Since all the terms
A=A
HIM is reduced to a harmonic with amplitude K a K a K

~
A2 , frequency ~ and phase :
k 2
in (16) are known and non-random, the induced
amplitude modulation can be compensated, for
M k [ y (t ); ] = A 2
K 1
( ~
exp j ~ k t + k ) (9) example, trough an amplitude weighting using the
~ = k! K 1a (10). inverse of relation (16).
where k k Therefore, the result of the warping transform of a Kth
Based on these results, Porat [1] has proposed an order PPS consists in a (K-1)th order PPS for the new
algorithm, which estimates sequentially the temporal variable t w(K ) . The (K-1)th order PHAF of
coefficients {ak}. At each step, using a spectral
analysis method, we estimate the spectral peak and, this signal, with respect to t w(K ) , peaks to a frequency
using the HAF, we compute an estimation value ( a k ) location related, via relation (10), to the aK-1
of ak. With this value, the effect of the phase term of coefficient. Once aK-1 is estimated, we construct the
the higher order is removed: (K-1)th order unitary operator UK-1:

{ }
1 /( K 1)
t (K )
s (k 1) (t ) = s (k ) (t ) exp ja k t k (11) w K -1 : t w(K ) K
w -1
( )
t w(K 1) = w K -1 t w( K ) = w
a K 1


(17)

435

which removes the (K-1)th order component. The
process is iterated (see figure 1) until all polynomial
coefficients are estimated.
y(m)(t)
PHAF-based {am }m=0, K
estimation method
Construction of axis
Phase term removing
warping operator
1/ m
(m) Um (m 1) = t w(m)
( ) ( )
y m 1 Uy m tw tw
a
m batteries are composed of 2k, k=0, 1,2 filters. Thus,
Fig. 1. Polynomial coefficient estimation based on unitary
transform phase reducing
IV. METHOD FOR RECOGNITION OF OFDM
MODULATION
For separation of modulation OFDM the method
proposed is based on the filtering process with an
orthogonal filter-bank and a high-order time-
frequency representation. The time-frequency space
of an OFDM signal is very complex due to numerous
carrier frequencies included in the signal. The main
idea is to recover and to locate these carrier
frequencies in the frequency band of interest.
Therefore, this technique is based as follows:
1. Determination of the frequency band of
interest in the time-frequency plane of the
signal using a detection method based on
cumulates of order 4 [6],
2. Construction of an orthogonal filter-bank in
order to perform a sub-bands filtering
operation in the frequency band of interest.
These filter batteries are composed of 2k,
k=0,N filters considering that each bank of
filters introduces a level of decomposition,
3. Stationarization and Extraction of the signal
using the WarpCom time-frequency
representation [5] for each filtered signal
obtained at step 2,
4. Calculation of the frequency marginal and
the standard deviation {i,k}, where i is the
decomposition level and k is the position of
the filter in the filter-bank, for the normal
distribution (Pdf - Probability density
function) corresponding to each stationary
signal obtained in step 3,
5. Construction of a decomposition tree and
location of the minima {i,k} for each branch
of the decomposition tree.
Steps 1 and 2 are considered to be a filtering process
and steps 3 and 4 a time-frequency characterization
process. The types of the filters implied are :
Chebyshev Type II, Morlet, Sinc. Other types of
filters could be taken into account.
V. RESULTS
At first to test the efficiency of the proposed method,
the signal analyzed is of type OFDM obtained by the
sum of four QPSK (Quaternary Phase Shift Keying)
quasi-analytic signals. The spacing of the carrier
frequencies is constant and the normalized values are
spread in the interval (00.5): 0.2, 0.224, 0.248,
0.273. In this case, the frequency band of interest is in
range of 0.2 to 0.3. The orthogonal filter-banks
considered are Chebyshev Type II filters. These filter
the decomposition tree will have two decompositions
levels: first, composed by the output of the battery
with 2 filters and the second composed by the output
of the battery with 4 filters.
In this case the number of carrier frequencies and
their spectral localization is known. Thus the design
of the filter batteries is progressive : with 2 filters and
finally with 4 filters. The second bank of filters is
constructed so that each filter is centered on each
carrier. Normally, taking into account the procedure
described above in step 5, the carriers (the minima for
each branch) in the second level of decomposition of
the tree are retrieved. In other words, the detection of
the carrier frequency is accurate. This situation is
illustrated in figure 2. The minima are represented
with encircled stars.
Fig. 2. Decomposition tree for an OFDM signal with 4 carriers: 0.2,
0.224, 0.248, 0.273, in normalized frequency
Because of the possible errors that could appear in the
method proposed, its possible to detect not all the 4
carriers in the last level of the tree: only 3, 2, 1 or
none, in the worse case. They could appear in the
upper levels.
Consequently, we apply this algorithm taking into
account the simulation method of Monte Carlo for
N=50 signals described above. The results obtained
are presented in table 1.
Table 1
4 carriers 3 carriers 2 carriers
detected detected detected
35 times 14 times 1 times
The results show good performances: from 50 times,
the proposed method have detected all the carriers 35
times in the second level of decomposition (the
situation depicted in figure 2), 14 times the algorithm
436
have picked up 3 carriers and 1 time only 2 carriers.
The efficiency of the technique is very dependent on
the construction of the batteries of filters and the band
of interest covered by these batteries. The
performances could be improved with a battery of
filters more centered on the carrier frequencies as well
as worse results are obtained with a battery of filters
that are not precisely centered on the carriers.
The next signal analyzed is of type QPSK with a
normalized frequency of 0.248. The batteries used are
the same as before In this case, the localization of the
minima in the decomposition tree has to be different.
Normally, on the second level of decomposition it has
to detect only one carrier.
The results using the simulation method of Monte
Carlo for N=50 such signals are presented in table 2.
Table 2
4 carriers 3 carriers 2 carriers 1 carrier
detected detected detected detected
0 times 0 times 17 times 27 times
From the results obtained is evident the different
structure of the decomposition tree, being possible to
discriminate the OFDM modulation from others types
of numeric modulation.
Next, the signals analyzed are real communication
numeric signals : an OFDM signal issued from a
multi-path channel and an FSK signal. The first signal
has the following caracteritistics: 250 kHz sampling
frequency, 32 carrier frequencies (base carrier located
at 12.5 kHz), signal-to-noise (SNR) 5 db, 3201
samples and 9 symbols. The algorithm generates with
the following parameters : band of interest (1041
kHz), 5 levels of decomposition (with the batteries
composed of 2k, k=0, 1,2,3,4,5 filters of type
Chebyshev Type II). The decomposition tree resulted
is presented in figure 3.
Fig. 4. Decomposition tree for a real FSK signal
We can remark that for the FSK signal, is not needed
many levels of decompositions : the carrier can be
detected in the upper levels as shown in figure 4.
VI. CONCLUSIONS
In this paper, a new method for the discrimination of
the OFDM modulation was proposed. The principle is
to filter the carrier frequencies in order to obtain, via
WarpCom time frequency-method, the stationarized
version of the signal. Furthermore, the corresponding
standard deviation of the frequency is involved for the
construction of the decomposition tree. A further
research direction will be needed to improve and
study the factors that affect the effectiveness of the
method.
REFERENCES
[1] B. Porat, Digital Processing of Random Signals, Pretince
Hall, 1993.
[2] Y. Wang and G. Zhou, On the use of high order ambiguity
functions for multicomponent polynomial phase signals, IEEE
Trans. on Signal Processing, vol. 65, pp. 283-296, 1998.
[3] S. Barbarossa, A. Scaglione and G.B. Giannakis, Product
High-Order Ambiguity Function for Multicomonent Polynomial-
Phase Signal Modeling, IEEE Transactions on Signal Processing,
vol. 46, No. 3, 1998.
[4] R. Baraniuk, Unitary Equivalence: a new twist on signal
processing, IEEE Trans. on Signal Processing, vol. 43, no. 10,
October, 1995.
[5] C. Ioana, A. Quinquis, "High-Order Warped Based Ambiguity
Function", IEEE-EURASIP Workshop on Non-Linear Signal and
Image Processing, Grado, Italy, 2003.
[6] C. Ioana, A. Quinquis, "Transient signal detection using
Overcomplete Wavelet Transform and high order statistics",
International Conference on Acoustic, Speech and Signal
Processing -ICASSP 2003, Hong Kong, April, 2003.

Fig. 3. Decomposition tree for a real OFDM signal

The second real signal is a FSK signal with the

following caracteritistics : 250 kHz sampling
frequency, single-carrier frequency located at 25 kHz,
signal-to-noise (SNR) 0 db, 2000 samples and 200
symbols. The algorithm parameters are : band of
interest (2.947.1 kHz), 5 levels of decomposition
(with the batteries composed of 2k, k=0, 1,2,3,4,5
filters of type Chebyshev Type II). The result is
showed in figure 4.

437
438
 Index of Authors

Abbott Derek, fasc. 2, p. 120;

Aghion Cristian, fasc. 1, p. 29;
Aiordachioaie Dorel, fasc. 1, p. 106, fasc. 2, p. 309, fasc. 2, p. 313;
Alexa Dimitrie, fasc. 1, p. 7, fasc. 1, p. 10, fasc. 1, p. 15, fasc. 1, p. 23, fasc. 1, p. 102;
Andrei Dorin, fasc. 2, p. 366, fasc. 2, p. 394;
Andrei Horia, fasc. 2, p. 221;
Anton Lucian, fasc. 1, p. 333, fasc. 2, p. 150;
Arsinte Radu, fasc. 2, p. 138;
Avram Adrian, fasc. 1, p. 94;
Bak Lszl, fasc. 1, p. 209;
Balint Cornel, fasc. 1, p. 283, fasc. 1, p. 287 l, fasc. 2, p. 197;
Balta Horia, fasc. 2, p. 33, fasc. 2, p. 38;
Banar Alexandru, fasc. 2, p. 354;
Banciu Marian G. , fasc. 2, p. 285;
Banu Leonard, fasc. 2, p. 102;
Barbu Tudor, fasc. 1, p. 291, fasc. 1, p. 328;
Bbi Mircea, fasc. 1, p. 59, fasc. 1, p. 89, fasc. 2, p. 180, fasc. 2, p. 184;
Blan Ionu, fasc. 2, p. 65;
Bechet Paul, fasc. 2, p. 431;
Beldianu Florin, fasc. 2, p. 385;
Beldianu Spiridon Florin, fasc. 1, p. 324;
Belega Daniel, fasc. 2, p. 225;
Bena Iulian, fasc. 2, p. 323;
Besnard Pascal, fasc. 2, p. 294, , fasc. 2, p. 297;
Blaga Tudor, fasc. 1, p. 383, fasc. 1, p. 388;
Borbs Pl Istvn, fasc. 1, p. 372;
Borda Monica, fasc. 1, p. 337 fasc. 2, p. 50;
Bota Vasile, fasc. 2, p. 7, fasc. 2, p. 13;
Botoca Corina, fasc. 1, p. 226, fasc. 2, p. 91;
Brassai Tihamr Sndor, fasc. 1, p. 209, fasc. 1, p. 214;
Briciu Marian, fasc. 1, p. 272;
Brnaru Diana, fasc. 2, p. 285;
Buchman Attila, fasc. 1, p. 232, fasc. 1, p.236;
Budura Georgeta, fasc. 1, p. 226, fasc. 2, p. 91;
Burdia Dnu, fasc. 1, p. 159, fasc. 1, p. 173, fasc. 1, p. 178, fasc. 2, p. 382;
Burileanu Corneliu, fasc. 1, p. 305;
Buzac Elvira, fasc. 2, p. 217;
Cmpeanu Andrei, fasc. 1, p. 164;
Carean R., fasc. 1, p. 311;
Crstea Horia, fasc. 1, p. 94;
Cruntu Alexandru, fasc. 1, p. 262;
Cepisca Costin, fasc. 2, p. 217, fasc. 2, p. 221;
Chen V., fasc. 1, p. 199;
Chioncel Cristian, fasc. 2, p. 87;
Chioreanu A., fasc. 1, p. 311;
Chiper Doru Florin, fasc. 1, p. 112, fasc. 1, p.117, fasc. 1, p. 266;
Chiper Doru, fasc. 2, p. 382;
Chiper R., fasc. 1, p. 7;
Chira Ligia, fasc. 2, p. 323;
Ciochin Silviu, fasc. 1, p. 297, fasc. 2, p. 69, fasc. 2, p. 165, fasc. 2, p. 360;
Ciugudean Mircea, fasc. 1, p. 142;
Crlugea Mihaela, fasc. 1, p. 136, fasc. 1, p.153;

439
Cleju Nicolae, fasc. 2, p. 354;
Codreanu N.C., fasc. 1, p. 98;
Coma Ciprian, fasc. 1, p. 117, fasc. 2, p. 382, fasc. 2, p.385;
Coma Ciprian-Romeo, fasc. 1, p.178;
Constantinescu Mihai, fasc. 1, p. 399;
Copaciu Flavius, fasc. 1, p. 388;
Cosac Andreea, fasc. 2, p. 205;
Costin Mihaela, fasc. 1, p. 291, fasc. 1, p.328;
Cotae Paul, fasc. 2, p. 385;
Crciun Florin, fasc. 2, p. 403;
Crainic Monica Sabina, fasc. 2, p. 201;
Crciun Adrian Virgil, fasc. 1, p. 40, fasc. 1, p.131;
Cremene Marcel, fasc. 2, p. 323;
Cre Clin, fasc. 1, p. 20;
Criste Daniel, fasc. 2, p. 175;
Croitoru Victor, fasc. 1, p. 399;
Dvid Lszl, fasc. 1, p. 209;
Dnei Marllene, fasc. 2, p. 346;
De Sabata Aldo, fasc. 2, p. 317;
Demeter Stefan, fasc. 2, p. 431;
Dimitriu Bogdan, fasc. 1, p. 159, fasc. 1, p.173;
Dobrea Dan-Marius, fasc. 2, p. 354;
Dobrot Virgil, fasc. 1, p. 367, fasc. 1, p.383, fasc. 1, p. 388;
Dogaru Ulieru Cristina, fasc. 2, p. 221;
Dogaru Ulieru Valentin, fasc. 2, p. 221;
Dragomir Ioan-Virgil, fasc. 1, p. 367;
Dragomir Livia, fasc. 2, p. 209;
Dughir Ciprian, fasc. 2, p. 348;
Duma Ioan I. , fasc. 2, p. 398;
Duma Petrut, fasc. 2, p. 55, fasc. 2, p. 154, fasc. 2, p. 187;
Dumitrescu Ctlin I. , fasc. 2, p. 247;
Emil Teodoru, fasc. 2, p. 300;
Ene Daniel Victora, fasc. 2, p. 193;
Enescu Andrei Alexandru, fasc. 2, p. 69, fasc. 2, p. 165, fasc. 2, p. 360;
Faur Daniela, fasc. 2, p. 205;
Fazakas Albert, fasc. 1, p. 136, fasc. 1, p. 168;
Frca Cristian, fasc. 1, p. 20;
Fericean Gabriel G. , fasc. 2, p. 50;
Fetil Lelia, fasc. 1, p. 136, fasc. 1, p. 168;
Feteanu Gigi, fasc. 1, p. 249;
Florea M., fasc. 1, p. 10, fasc. 1, p. 15;
Fratu Octavian, fasc. 2, p. 19, fasc. 2, p. 403;
Gal Jnos, fasc. 1, p. 164, fasc. 2, p. 87;
Galatchi Dan, fasc. 1, p. 358, fasc. 1, p. 362;
Gasparel Aida-Virginia, fasc. 1, p. 367;
Gavrincea tefan, fasc. 1, p. 184;
Gheu Drago, fasc. 2, p. 285;
Ghisa Laura, fasc. 2, p. 294, fasc. 2, p. 297;
Giurgiu Luminia, fasc. 1, p. 249;
Gvan M. , fasc. 2, p. 341;
Gontean Aurel, fasc. 2, p. 180, fasc. 2, p. 184;
Goras Tecla., fasc. 1, p. 7, fasc. 1, p. 102;
Gora Liviu, fasc. 1, p. 102;
Grama Lcrimioara, fasc. 2, p. 96;
Grama Gheorghe, fasc. 1, p. 34;
Groza Robert, fasc. 1, p. 136, fasc. 1, p. 168;

440
Gui Vasile, fasc. 1, p. 321;
Halunga Simona, fasc. 2, p. 19, fasc. 2, p. 403;
Hintea Sorin, fasc. 1, p. 136;
Hurgoi Florin, fasc. 1, p. 20;
Husu Adela, fasc. 2, p. 221;
Ignea Alimpie, fasc. 2, p. 236, fasc. 2, p. 255;
Ilioaei Ciprian, fasc. 2, p. 138;
Ioana Cornel, fasc. 2, p. 434;
Ionacu Cristian, fasc. 1, p. 159, fasc. 1, p. 173, fasc. 1, p. 178;
Ionel Sabin C. , fasc. 2, p. 240, fasc. 2, p. 346;
Ionescu C., fasc. 1, p. 98;
Ionica Tiberiu, fasc. 2, p. 197;
Iosif Florin D., fasc. 1, p. 333;
Isar Alexandru, fasc. 2, p. 110;
Isar Dorina, fasc. 2, p. 110;
Ivanovici Traian, fasc. 2, p. 221;
Jibleanu Roxana, fasc. 2, p. 180, fasc. 2, p. 184;
Jipa Rzvan, fasc. 1, p. 147;
Jurca Lucian, fasc. 1, p. 193;
Keller Guenter, fasc. 1, p. 63, fasc. 1, p. 85;
Kovaci Maria, fasc. 2, p. 33, fasc. 2, p. 38;
Krneta Radojka, fasc. 1, p. 47;
Laitinen Jyrki, fasc. 1, p. 321;
Lascu Dan, fasc. 1, p. 63, fasc. 1, p. 85;
Lascu Mihaela, fasc. 2, p. 253;
Lazar Gabriel, fasc. 1, p. 383;
Lazr A., fasc. 1, p. 10, fasc. 1, p. 15;
Lazr Luminia-Camelia, fasc. 1, p. 10, fasc. 1, p. 15;
Lzrescu Dan, fasc. 1, p. 253;
Lzrescu Vasile, fasc. 1, p. 253;
Levinthal Adam, fasc. 1, p. 147;
Loghin Clin, fasc. 2, p. 323;
Lojewski George, fasc. 2, p. 285;
Luca Adrian, fasc. 2, p. 132;
Lucaciu Radu, fasc. 2, p. 394;
Lungu erban, fasc. 1, p. 73, fasc. 1, p. 153;
Lupu Eugen, fasc. 1, p. 275, fasc. 1, p. 279;
Mailat Adrian, fasc. 1, p. 131;
Malutan Raul E., fasc. 1, p. 337;
Marchegay Philippe, fasc. 1, p. 142;
Marcus Ionel Urdea, fasc. 2, p. 217;
Mrza Eugen, fasc. 2, p. 389, fasc. 2, p. 425;
Mateescu Cosmin, fasc. 2, p. 403;
Matekovits Ladislau, fasc. 2, p. 317;
Matin Mohammad Abdul, fasc. 2, p. 114;
Mrnescu Valentin, fasc. 1, p. 189, fasc. 1, p. 193;
Mereu erban, fasc. 2, p. 244;
Mic Daniel, fasc. 1, p. 184, fasc. 1, p. 236;
Miclaus Simona, fasc. 2, p. 431;
Miclu Nicolae, fasc. 1, p. 226, fasc. 1, p. 256;
Micul Emil, fasc. 1, p. 184;
Mihaescu Adrian, fasc. 2, p. 294, fasc. 2, p. 297;
Mihiu Adrian, fasc. 2, p. 255;
Milenkovic Sanja, fasc. 1, p. 47;
Militaru Nicolae, fasc. 2, p. 285;
Mischie Septimiu, fasc. 2, p. 230;

441
Mitran Radu, fasc. 2, p. 431;
Moca Vasile V., fasc. 1, p. 279;
Mocanu Irina G. , fasc. 2, p. 142, fasc. 2, p. 146;
Moise Marius, fasc. 2, p. 409, fasc. 2, p. 415;
Moldoveanu Alexandru, fasc. 1, p. 399;
Moraru Bogdan, fasc. 1, p. 388;
Moraru Simona, fasc. 2, p. 205;
Moussaoui A., fasc. 1, p. 199;
Munteanu Doru P., fasc. 1, p. 301;
Munteanu Valeriu, fasc. 2, p. 106;
Naforni Corina, fasc. 1, p. 164, fasc. 2, p. 126;
Naforni Miranda, fasc. 1, p. 372, fasc. 1, p. 377, fasc. 2, p. 389;
Nstase Ana, fasc. 2, p. 13;
Neagoe Victor-Emil, fasc. 1, p. 343, fasc. 1, p. 348;
Neddermeyer Werner, fasc. 2, p. 236;
Nedelcu Liviu, fasc. 2, p. 285;
Negoiescu Dan, fasc. 1, p. 79;
Nicolaescu Ioan, fasc. 2, p. 150;
Nicula Dan, fasc. 1, p. 147;
Oancea Eugeniu, fasc. 1, p. 301;
Obreja erban Georgic, fasc. 2, p. 370, fasc. 2, p. 378;
Olah A., fasc. 2, p.341;
Oltean Gabriel, fasc. 1, p. 220, fasc. 2, p. 328;
Oltean Ioana, fasc. 2, p. 328;
Oltean Marius, fasc. 2, p. 389, fasc. 2, p. 425;
Oniga tefan, fasc. 1, p. 184, fasc. 1, p. 232, fasc. 1, p. 236;
Orza B., fasc. 1, p. 311, fasc. 2, p. 341;
Osborne Dominic, fasc. 2, p. 120;
Oteteanu Marius, fasc. 1, p. 393, fasc. 2, p. 175;
Palade Tudor, fasc. 1, p. 352;
Paleologu Constantin, fasc. 2, p. 165, fasc. 2, p. 69;
Panaitopol Dorin, fasc. 2, p. 170;
Pan Gheorghe, fasc. 1, p. 40, 131;
Pantelimon Brandusa, fasc. 2, p. 205, fasc. 2, p. 209, fasc. 2, p. 213;
Partheniu Cezar, fasc. 2, p. 75;
Paca Sorin, fasc. 1, p. 53;
Pun Adrian Florin, fasc. 2, p. 370, fasc. 2, p. 378;
Perioar Lucian Andrei, fasc. 2, p. 28, fasc. 2, p. 60;
Petrescu Teodor, fasc. 2, p. 259, fasc. 2, p.285;
Petreu Dorin, fasc. 1, p. 20;
Pirnog Ionu, fasc. 2, p. 193;
Pletea I.V., fasc. 1, p. 7, 23;
Polgar Zsolt, fasc. 2, p. 7, fasc. 2, p. 13;
Pop Ovidiu Aurel, fasc. 1, p. 73;
Pop Petre G., fasc. 1, p. 275, fasc. 1, p. 279;
Popa Cosmin, fasc. 1, p. 122, fasc. 1, p. 126;
Popa Gheorghe Daniel, fasc. 1, p. 393;
Popa Rzvan, fasc. 2, p. 25;
Popa Rustem, fasc. 2, p. 335;
Popescu Camelia, fasc. 1, p. 98, fasc. 2, p. 350;
Popescu Sorin, fasc. 2, p. 44;
Popescu Victor, fasc. 1, p. 153;
Popescu Viorel, fasc. 1, p. 53, fasc. 1, p. 59, fasc. 1, p. 69, fasc. 1, p. 85;
Popescu Vladimir, fasc. 1, p. 305;
Popovici Adrian, fasc. 1, p. 59, fasc. 1, p. 89;
Pradel Gilbert, fasc. 1, p. 241;

442
Preda Radu O., fasc. 1, p. 315;
Puschita Emanuel, fasc. 1, p. 352;
Quinquis Andr, fasc. 2, p. 110, fasc. 2, p. 434;
Rados Ivan, fasc. 2, p. 289;
Rangu Marius, fasc. 1, p. 94;
Rp Adrian, fasc. 1, p. 98, fasc. 2, p. 350;
Rdescu Radu, fasc. 2, p. 25, fasc. 2, p. 65;
Rogers Derek, fasc. 2, p. 120;
Romanca Mihai, fasc. 1, p. 131;
Ropot Armand-Dragos, fasc. 1, p. 343;
Rosu-Niculescu Ana, fasc. 2, p. 259;
Rusu Mircea Sorin, fasc. 2, p. 170;
Sadi Francois, fasc. 1, p. 241;
Salagean Marius, fasc. 2, p. 434;
Sandu Ion, fasc. 2, p. 209;
Schnell Michael, fasc. 2, p. 236;
Scripcariu Luminita, fasc. 2, p. 55, fasc. 2, p. 187;
Sechelea Andrei-Tudor, fasc. 2, p. 354;
Serafin Petru, fasc. 1, p. 407, fasc. 1, p. 412;
Shu Fangwu, fasc. 2, p. 236;
Sikora Axel, fasc. 2, p. 421;
Silion Daniel, fasc. 2, p. 170;
Simion Stefan, fasc. 2, p. 265;
Simion Viorica, fasc. 2, p. 213;
Simon Csaba, fasc. 1, p. 372, fasc. 1, p. 377;
Simu Dan, fasc. 1, p. 44;
Sipos Emilia, fasc. 2, p. 328;
Sirbu Gabriel, fasc. 2, p. 309, fasc. 2, p. 313;
Srbu A., fasc. 1, p. 7;
Sorell Matthew, fasc. 2, p. 120;
Stanciu Lucian, fasc. 2, p. 102;
Stefanescu Costin, fasc. 2, p. 213;
Stoian Rodica, fasc. 2, p. 28, fasc. 2, p. 60;
Stoica Adrian, fasc. 1, p. 333, fasc. 2, p. 150;
Stoiciu Dan, fasc. 2, p. 253, fasc. 2, p. 348;
Sucholotiuc Cristina, fasc. 2, p. 193;
Sunaric Tanja, fasc. 2, p. 289;
Svasta Paul, fasc. 1, p. 98, fasc. 2, p. 350;
Szkely Iuliu, fasc. 1, p. 214;
Szopos Erwin, fasc. 2, p. 81;
chiop Adrian, fasc. 1, p. 69;
elaru Clin, fasc. 1, p. 352;
erbanescu Alexandru, fasc. 2, p. 366;
tefan Demeter, fasc. 1, p. 249, fasc. 2, p. 300;
Trniceriu Daniela, fasc. 2, p. 106;
Telescu Mihai, fasc. 2, p. 294, fasc. 2, p. 297;
Teodorescu T.D., fasc. 1, p. 266;
Teodorescu Tiberiu, fasc. 1, p. 112;
Tigaeru Liviu, fasc. 1, p. 29;
Tiponut Sebastian V. , fasc. 2, p. 332;
Tiponu Virgil, fasc. 1, p. 236, fasc. 2, p. 305;
Toadere Florin, fasc. 2, p. 275, fasc. 2, p. 271, fasc. 2, p. 280;
Toderean Gavril, fasc. 1, p. 262;
Tofilescu Pompilian, fasc. 1, p. 34;
Toma Liviu Jr. , fasc. 2, p. 236;
Toma Norbert, fasc. 2, p. 81;

443
Tome Marin, fasc. 1, p. 53;
Tordai Botond, fasc. 1, p. 377;
Trip Nistor Daniel, fasc. 1, p. 69;
Tudos Ana Maria, fasc. 1, p. 328;
Tulbure Traian, fasc. 1, p. 147;
Turalija Pero, fasc. 2, p. 289;
igeru Liviu, fasc. 1, p. 205;
opa Marina, fasc. 2, p. 81;
Udrea Radu Mihnea, fasc. 1, p. 297, fasc. 1, p. 315, fasc. 2, p. 193;
Ungureanu Mihaela, fasc. 1, p. 253;
Ursaru Ovidiu, fasc. 1, p. 205, fasc. 1, p. 29;
Varga Mihaly, fasc. 2, p. 7;
Vesa Andy, fasc. 2, p. 425;
Vilda Pedro Gmez, fasc. 1, p. 337;
Vizireanu Drago Nicolae, fasc. 1, p. 297, fasc. 1, p. 315;
Vizitiu Constantin I. , fasc. 1, p. 333 , fasc. 2, p. 150;
Vlad A., fasc. 2, p. 341;
Vlad Adriana, fasc. 2, p. 132;
Vladescu Clin, fasc. 2, p. 366;
Vlaicu Aurel, fasc. 1, p. 311, fasc. 2, p. 341;
Vldeanu Clin, fasc. 2, p. 394;
Voina Catalin, fasc. 2, p. 205;
Zinca Daniel, fasc. 1, p. 367, fasc. 1, p. 383;
Zoican Roxana, fasc. 1, p. 358, fasc. 1, p. 362;
Zoican Sorin, fasc. 2, p. 160.

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