Proceedings

2009 International Conference

on Computer Engineering and Technology

ICCET 2009

Volume I
 Proceedings

2009 International Conference

on Computer Engineering and Technology

Volume I

January 22 - 24, 2009

Singapore

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Jianhong Zhou and Xiaoxiao Zhou

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 2009 International Conference
on Computer Engineering
and Technology

ICCET 2009
Table of Contents
Volume - 1
Preface - Volume 1 ...........................................................................................................................................xiv
ICCET 2009 Committee Members - Volume 1..................................................................................xvi
ICCET 2009 Organizing Committees - Volume 1..........................................................................xvii

Session 1
Overlapping Non-dedicated Clusters Architecture .................................................................................................3
Martin Štava and Pavel Tvrdík
To Determine the Weight in a Weighted Sum Method for Domain-Specific
Keyword Extraction ..................................................................................................................................................11
Wenshuo Liu and Wenxin Li
Flow-based Description of Conceptual and Design Levels ................................................................................16
Sabah Al-Fedaghi
A Method of Query over Encrypted Data in Database ........................................................................................23
Lianzhong Liu and Jingfen Gai
Traversing Model Design Based on Strong-Association Rule for Web
Application Vulnerability Detection .......................................................................................................................28
Zhenyu Qi, Jing Xu, Dawei Gong, and He Tian
Attribute-Based Relative Ranking of Robot for Task Assignment ....................................................................32
B.B. Choudhury, B.B. Biswal, and R.N. Mahapatra
A Subjective Trust Model Based on Two-Dimensional Measurement .............................................................37
Chaowen Chang, Chen Liu, and Yuqiao Wang
A Genetic Algorithm Approach for Optimum Operator Assignment in CMS ................................................42
Ali Azadeh, Hamrah Kor, and Seyed-Morteza Hatefi

v
Dynamic Adaption in Composite Web Services Using Expiration Times .......................................................47
Xiaohao Yu, Xueshan Luo, Honghui Chen, and Dan Hu
An Emotional Intelligent E-learning System Based on Mobile Agent
Technology .................................................................................................................................................................51
Zhiliang Wang, Xiangjie Qiao, and Yinggang Xie
Audio Watermarking for DRM Based on Chaotic Map ......................................................................................55
B. Lei and I.Y. Soon
Walking Modeling Based on Motion Functions ...................................................................................................60
Hao Zhang and Zhijing Liu
Preprocessing and Feature Preparation in Chinese Web Page Classification ..................................................64
Weitong Huang, Luxiong Xu, and Yanmin Liu
High Performance Grid Computing and Security through Load Balancing .....................................................68
V. Sugavanan and V. Prasanna Venkatesh
Research of the Synthesis Control of Force and Position in Electro-Hydraulic
Servo System ..............................................................................................................................................................73
Yadong Meng, Changchun Li, Hao Yan, and Xiaodong Liu

Session 2
Features Selection Using Fuzzy ESVDF for Data Dimensionality Reduction ................................................81
Safaa Zaman and Fakhri Karray
PDC: Propagation Delay Control Strategy for Restricted Floating Sensor
Networks .....................................................................................................................................................................88
Xiaodong Liu
Fast and High Quality Temporal Transcoding Architecture in the DCT
Domain for Adaptive Video Content Delivery .....................................................................................................91
Vinay Chander, Aravind Reddy, Shriprakash Gaurav, Nishant Khanwalkar,
Manish Kakhani, and Shashikala Tapaswi
Electricity Demand Forecasting Based on Feedforward Neural Network
Training by a Novel Hybrid Evolutionary Algorithm ..........................................................................................98
Wenyu Zhang, Yuanyuan Wang, Jianzhou Wang, and Jinzhao Liang
Investigation on the Behaviour of New Type Airbag ........................................................................................103
Hu Lin, Liu Ping, and Huang Jing
Performance Evaluation of PNtMS: A Portable Network Traffic Monitoring
System on Embedded Linux Platform ..................................................................................................................108
Mostafijur Rahman, Zahereel Ishwar Abdul Khalib, and R.B. Ahmad
PB-GPCT: A Platform-Based Configuration Tool .............................................................................................114
Huiqiang Yan, Runhua Tan, Kangyun Shi, and Fei Lu
A Feasibility Study on Hyperblock-based Aggressive Speculative Execution
Model .........................................................................................................................................................................119
Ming Cong, Hong An, Yongqing Ren, Canming Zhao, and Jun Zhang

vi
Parallel Method for Discovering Frequent Itemsets Using Weighted Tree
Approach ...................................................................................................................................................................124
Preetham Kumar and Ananthanarayana V S
Optimized Design and Implementation of Three-Phase PLL Based on FPGA .............................................129
Yuan Huimei, Sun Hao, and Song Yu
Research on the Data Storage and Access Model in Distributed Environment .............................................134
Wuling Ren and Pan Zhou
An Effective Classification Model for Cancer Diagnosis Using Micro Array
Gene Expression Data .............................................................................................................................................137
V. Saravanan and R. Mallika
Study and Experiment of Blast Furnace Measurement and Control System
Based on Virtual Instrument ..................................................................................................................................142
Shufen Li and Zhihua Liu
A New Optimization Scheme for Resource Allocation in OFDMA Based
WiMAX Systems .....................................................................................................................................................145
Arijit Ukil, Jaydip Sen, and Debasish Bera
An Integration of CoTraining and Affinity Propagation for PU Text
Classification ............................................................................................................................................................150
Na Luo, Fuyu Yuan, and Wanli Zuo

Session 3
Ergonomic Evaluation of Small-screen Leading Displays on the Visual
Performance of Chinese Users ...............................................................................................................................157
Yu-Hung Chien and Chien-Cheng Yen
Curvature-Based Feature Extraction Method for 3D Model Retrieval ...........................................................161
Yujie Liu, Xiaolan Yao, and Zongmin Li
A New Method for Vertical Handoff between WLANs and UMTS
in Boundary Conditions ..........................................................................................................................................166
Majid Fouladian, Faramarz Hendessi, Alireza Shafieinejad, Morteza Rahimi,
and Mahdi M. Bayat
Research on Secure Key Techniques of Trustworthy Distributed System .....................................................172
Ming He, Aiqun Hu, and Hangping Qiu
WebELS: A Multimedia E-learning Platform for Non-broadband Users .......................................................177
Zheng He, Jingxia Yue, and Haruki Ueno
Implementation and Improvement Based on Shear-Warp Volume Rendering
Algorithm ..................................................................................................................................................................182
Li Guo and Xie Mei
Conferencing, Paging, Voice Mailing via Asterisk EPBX ................................................................................186
Ale Imran and Mohammed A. Qadeer
A New Mind Evolutionary Algorithm Based on Information Entropy ...........................................................191
Yuxia Qiu and Keming Xie

vii
An Encapsulation Structure and Description Specification for Application
Level Software Components ..................................................................................................................................195
Jin Guojie and Yin Baolin
Fault Detection and Diagnosis of Continuous Process Based on Multiblock
Principal Component Analysis ..............................................................................................................................200
Libo Bie and Xiangdong Wang
Strong Thread Migration in Heterogeneous Environment ................................................................................205
Khandakar Entenam Unayes Ahmed, Md. Al-mamun Shohag, Tamim Shahriar,
Md. Khalad Hasan, and Md. Mashud Rana
A DSP-based Active Power Filter for Three-phase Power Distribution Systems .........................................210
Ping Wei, Zhixiong Zhan, and Houquan Chen
Access Control Scheme for Workflow .................................................................................................................215
Lijun Gao, Lu Zhang, and Lei Xu
A Mathematical Model of Interference between RFID and Bluetooth in Fading
Channel ......................................................................................................................................................................218
Junjie Chen, Jianqiu Zeng, and Yuchen Zhou
Optimization Strategy for SSVEP-Based BCI in Spelling Program Application ..........................................223
Indar Sugiarto, Brendan Allison, and Axel Gräser

Session 4
A Novel Method for the Web Page Segmentation and Identification .............................................................229
Jing Wang and Zhijing Liu
Disturbance Observer-Based Variable Structure Control on the Working
Attitude Balance Mechanism of Underwater Robot ..........................................................................................232
Min Li and Heping Liu
Adaptive OFDM Vs Single Carrier Modulation with Frequency Domain
Equalization ..............................................................................................................................................................238
Inderjeet Kaur, Kamal Thakur, M. Kulkarni, Daya Gupta, and Prabhjyot Arora
A Bivariate C1 Cubic Spline Space on Wang's Refinement .............................................................................243
Huan-Wen Liu and Wei-Ping Lu
Fast Shape Matching Using a Hybrid Model ......................................................................................................247
Gang Xu and Wenxian Yang
A Multi-objective Genetic Algorithm for Optimization of Cellular
Manufacturing System ............................................................................................................................................252
H. Kor, H. Iranmanesh, H. Haleh, and S.M. Hatefi
A Formal Mapping between Program Slicing and Z Specifications ...............................................................257
Fangjun Wu
Modified Class-Incremental Generalized Discriminant Analysis ....................................................................262
Yunhui He
Controlling Free Riders in Peer to Peer Networks by Intelligent Mining .......................................................267
Ganesh Kumar. M, Arun Ram. K, and Ananya. A.R

viii
Servo System Modeling and DSP Code Autogeneration Technology
for Open-CNC ..........................................................................................................................................................272
Shukun Cao, Heng Zhang, Li Song, Changsheng Ai, and Xiangbo Ze
Extending Matching Model for Semantic Web Services ...................................................................................276
Alireza Zohali, Kamran Zamanifar, and Naser Nematbakhsh
Sound Absorption Measurement of Acoustical Material and Structure Using
the Echo-Pulse Method ...........................................................................................................................................281
Liang Sun, Hong Hou, Liying Dong, and Fangrong Wan
Parallel Design of Cross Search Algorithm in Motion Estimation ..................................................................286
Fan Zhang
Influences of DSS Environments and Models on Current Business Decision
and Knowledge Management ................................................................................................................................291
Md. Fazle Munim and Fatima Binte Zia
A Method for Transforming Workflow Processes to CSS ................................................................................295
Jing Xiao, Guo-qing Wu, and Shu Chen

Session 5
An Empirical Approach of Delta Hedging in GARCH Model .........................................................................303
Qian Chen and Chengzhe Bai
Multi-objective Parameter Optimization Technology for Business Process
Based on Genetic Algorithm ..................................................................................................................................308
Bo Wang, Li Zhang, and Yawei Tian
Analysis and Design of an Access Control Model Based on Credibility ........................................................312
Chaowen Chang, Yuqiao Wang, and Chen Liu
Modeling of Rainfall Prediction over Myanmar Using Polynomial Regression ...........................................316
Wint Thida Zaw and Thinn Thu Naing
New Similarity Measure for Restricted Floating Sensor Networks .................................................................321
Yuan Feng, Xiaodong Liu, and Xiangqian Ding
3D Mesh Skeleton Extraction Based on Feature Points ....................................................................................326
Faming Gong and Cui Kang
Pairings Based Designated Verifier Signature Scheme for Three-Party
Communication Environment ................................................................................................................................330
Han-Yu Lin and Tzong-Sun Wu
A Novel Shared Path Protection Scheme for Reliability Guaranteed
Connection ................................................................................................................................................................334
Jijun Zhao, Weiwei Bian, Lirong Wang, and Sujian Wang
Generalized Program Slicing Applied to Z Specifications ................................................................................338
Fangjun Wu
PC Based Weight Scale System with Load Cell for Product Inspection ........................................................343
Anton Satria Prabuwono, Habibullah Akbar, and Wendi Usino

ix
Short-Term Electricity Price Forecast Based on Improved Fractal Theory ....................................................347
Herui Cui and Li Yang
BBS Sentiment Classification Based on Word Polarity ....................................................................................352
Shen Jie, Fan Xin, Shen Wen, and Ding Quan-Xun
Applying eMM in a 3D Approach to e-Learning Quality Improvement ........................................................357
Kattiya Tawsopar and Kittima Mekhabunchakij
Research on Automobile Driving State Real-Time Monitoring System Based
on ARM .....................................................................................................................................................................361
Hongjiang He and Yamin Zhang
Information Security Risk Assessment and Pointed Reporting: Scalable
Approach ...................................................................................................................................................................365
D.S. Bhilare, A.K. Ramani, and Sanjay Tanwani
An Extended Algorithm to Enhance the Performance of the Gridbus Broker
with Data Restoring Technique .............................................................................................................................371
Abu Awal Md. Shoeb, Altaf Hussain, Md. Abu Naser Bikas, and Md. Khalad Hasan

Session 6
Prediction of Ship Pitching Based on Support Vector Machines .....................................................................379
Li-hong Sun and Ji-hong Shen
The Methods of Improving the Manufacturing Resource Planning (MRP II)
in ERP ........................................................................................................................................................................383
Wenchao Jiang and Jingti Han
A New Model for Classifying Inputs and Outputs and Evaluating the DMUs
Efficiency in DEA Based on Cobb-Douglas Production Function ..................................................................390
S.M. Hatefi, F. Jolai, H. Kor, and H. Iranmanesh
The Analysis and Improvement of the Price Forecast Model Based on Fractal
Theory ........................................................................................................................................................................395
Herui Cui and Li Yang
A Flash-Based Mobile Learning System for Learning English as Second
Language ...................................................................................................................................................................400
Firouz B. Anaraki
Recognition of Trade Barrier Based on General RBF Neural Network ..........................................................405
Yu Zhao, Miaomiao Yang, and Chunjie Qi
An Object-Oriented Product Data Management .................................................................................................409
Fan Wang and Li Zhou
Study of 802.11 Network Performance and Wireless Multicasting .................................................................414
Biju Issac
A Novel Approach for Face Recognition Based on Supervised Locality
Preserving Projection and Maximum Margin Criterion ....................................................................................419
Jun Kong, Shuyan Wang, Jianzhong Wang, Lintian Ma, Baowei Fu, and Yinghua Lu

x
Association Rules Mining Based on Simulated Annealing Immune
Programming Algorithm .........................................................................................................................................424
Yongqiang Zhang and Shuyang Bu
Processing Power Estimation of Simple Wireless Sensor Network Nodes
by Power Macro-modeling .....................................................................................................................................428
M. Rafiee, M.B. Ghaznavi-Goushchi, and B. Seyfe
A Fault-Tolerant Strategy for Multicasting in MPLS Networks ......................................................................432
Weili Huang and Hongyan Guo
A Novel Content-based Information Hiding Scheme ........................................................................................436
Jun Kong, Hongru Jia, Xiaolu Li, and Zhi Qi
Ambi Graph: Modeling Ambient Intelligent System .........................................................................................441
K. Chandrasekaran, I.R. Ramya, and R. Syama

Session 7
Research on Grid-based Short-term Traffic Flow Forecast Technology ........................................................449
Wang Xinying, Juan Zhicai, Liu Xin, and Mei Fang
A Nios II Based English Speech Training System for Hearing-Impaired
Children .....................................................................................................................................................................452
Ningfeng Huang, Haining Wu, and Yinchen Song
A New DEA Model for Classification Intermediate Measures and Evaluating
Supply Chain and its Members ..............................................................................................................................457
S.M. Hatefi, F. Jolai, H. Iranmanesh, and H. Kor
A Novel Binary Code Based Projector-Camera System Registration Method ..............................................462
Jiang Duan and Jack Tumblin
Non-temporal Mutliple Silhouettes in Hidden Markov Model for View
Independent Posture Recognition ..........................................................................................................................466
Yunli Lee and Keechul Jung
Classification of Quaternary [21s+1,3] Optimal Self-orthogonal Codes ........................................................471
Xuejun Zhao, Ruihu Li, and Yingjie Lei
Performance Analysis of Large Receive Offload in a Xen Virtualized System ...........................................475
Hitoshi Oi and Fumio Nakajima
An Improved Genetic Algorithm Based on Fixed Point Theory for Function
Optimization .............................................................................................................................................................481
Jingjun Zhang, Yuzhen Dong, Ruizhen Gao, and Yanmin Shang
Example-Based Regularization Deployed to Face Hallucination ....................................................................485
Hong Zhao, Yao Lu, Zhengang Zhai, and Gang Yang
An Ensemble Approach for Semantic Assessment of Summary Writings .....................................................490
Yulan He, Siu Cheung Hui, and Tho Thanh Quan
A Fast Reassembly Methodology for Polygon Fragment ..................................................................................495
Gang Xu and Yi Xian

xi
A Data Mining Approach to Modeling the Behaviors of Telecom Clients ....................................................500
Xiaodong Liu
Simulating Fuzzy Manufacturing System: Case Study ......................................................................................505
A. Azadeh, S.M. Hatefi, and H. Kor
Research of INS Simulation Technique Based on UnderWater Vehicle Motion
Model .........................................................................................................................................................................510
Jian-hua Cheng, Yu-shen Li, and Jun-yu Shi
Modeling and Simulation of Wireless Sensor Network (WSN) with SpecC
and SystemC .............................................................................................................................................................515
M. Rafiee, M.B. Ghaznavi-Ghoushchi, S. Kheiri, and B. Seyfe

Session 8
Sub-micron Parameter Scaling for Analog Design Using Neural Networks ..................................................523
A.A. Bagheri-Soulla and M.B. Ghaznavi-Ghoushchi
An Improved Genetic Algorithm Based on Fixed Point Theory for Function
Optimization .............................................................................................................................................................527
Jingjun Zhang, Yuzhen Dong, Ruizhen Gao, and Yanmin Shang
P2DHMM: A Novel Web Object Information Extraction Model ....................................................................531
Jing Wang and Zhijing Liu
An Efficient Multi-Patterns Parameterized String Matching Algorithm
with Super Alphabet ................................................................................................................................................536
Rajesh Prasad and Suneeta Agarwal
Research on Modeling Method of Virtual Enterprise in Uncertain
Environments ............................................................................................................................................................541
Jihai Zhang
Design of Intrusion Detection System Based on a New Pattern Matching
Algorithm ..................................................................................................................................................................545
Hu Zhang
To Construct Implicit Link Structure by Using Frequent Sequence Miner
(FS-Miner) ................................................................................................................................................................549
May Thu Aung and Khin Nwe Ni Tun
Recognition of Eye States in Real Time Video ...................................................................................................554
Lei Yunqi, Yuan Meiling, Song Xiaobing, Liu Xiuxia, and Ouyang Jiangfan
Performance Analysis of Postprocessing Algorithm and Implementation
on ARM7TDMI .......................................................................................................................................................560
Manoj Gupta, B.K. Kaushik, and Laxmi Chand
NURBS Interpolation Method with Feedrate Correction in 3-axis CNC System .........................................565
Liangji Chen and Huiying Li
Implementation Technique of Unrestricted LL Action Grammar ....................................................................569
Jing Zhang and Ying Jin

xii
Improving BER Using RD Code for Spectral Amplitude Coding Optical
CDMA Network .......................................................................................................................................................573
Hilal Adnan Fadhil, S.A. Aljunid, and R. Badlishah Ahmad
USS-TDMA: Self-stabilizing TDMA Algorithm for Underwater Wireless
Sensor Network ........................................................................................................................................................578
Zhongwen Guo, Zhengbao Li, and Feng Hong
Mathematical Document Retrieval for Problem Solving ..................................................................................583
Sidath Harshanath Samarasinghe and Siu Cheung Hui
Lossless Data Hiding Scheme Based on Adjacent Pixel Difference ...............................................................588
Zhuo Li, Xiaoping Chen, Xuezeng Pan, and Xianting Zeng

Author Index - Volume 1 .............................................................................................................................593

xiii
 Preface
Dear Distinguished Delegates and Guests,

The Organizing Committee warmly welcomes our distinguished delegates and guests to the
International Conference on Computer Engineering and Technology 2009 (ICCET 2009), held on
January 22 - 24, 2009 in Singapore.

ICCET 2009, ICACC 2009 and ICECS 2008 are sponsored by International Association of Computer
Science and Information Technology (IACSIT) and Singapore Institute of Electronics (SIE), and the
accepted papers of ICECS 2008 have been included in the ICCET proceeding as a special session. If
you have attended a conference sponsored by IACSIT before, you are aware that the conferences
together report the results of research efforts in a broad range of computer science. These conferences
are aimed at discussing with all of you the wide range of problems encountered in present and future
high technologies. The ICCET 2009, ICACC 2009 and ICECS 2008 are organized to gather members
of our international community of computer and control scientists so that researchers from around the
world can present their leading-edge work, expanding our community’s knowledge and insight into
the significant challenges currently being addressed in that research. The conference Program
Committee is itself quite diverse and truly international, with membership from the Americas, Europe,
Asia, Africa and Oceania.

This proceeding records the fully refereed papers presented at the conference. The main conference
themes and tracks are Computer Engineering and Technology. The conference aims to bring together
researchers, scientists, engineers, and practitioners to exchange and share their experiences, new ideas,
and research results about all aspects of the main conference themes and tracks and discuss the
practical challenges encountered and the solutions adopted. The main goal of these events is to
provide international scientific forums for exchange of new ideas in a number of fields that interact
in-depth through discussions with their peers from around the world. Both inward research; core areas
of computer control and outward research; multi-disciplinary, inter-disciplinary, and applications will
be covered during these events.

The conference has solicited and gathered technical research submissions related to all aspects of
major conference themes and tracks. All the submitted papers in the proceeding have been peer
reviewed by the reviewers drawn from the scientific committee, external reviewers and editorial board
depending on the subject matter of the paper. Reviewing and initial selection were undertaken
electronically. After the rigorous peer-review process, the submitted papers were selected on the basis
of originality, significance, and clarity for the purpose of the conference. The selected papers and
additional late-breaking contributions to be presented as lectures will make an exiting technical
program. The conference program is extremely rich, featuring high-impact presentations.

xiv
The high quality of the program – guaranteed by the presence of an unparalleled number of
internationally recognized top experts – can be assessed when reading the contents of the program.
The conference will therefore be a unique event, where attendees will be able to appreciate the latest
results in their field of expertise, and to acquire additional knowledge in other fields. The program has
been structured to favor interactions among attendees coming from many diverse horizons,
scientifically, geographically, from academia and from industry. Included in this will to favor
interactions are social events at prestigious sites.

We would like to thank the program chairs, organization staff, and the members of the program
committees for their work. Thanks also go to Ms. Lisa O'Conner, CPS Production Editor, Conference
Publishing Services (CPS), IEEE Computer Society, for her wonderful editorial service to this
proceeding.

We are grateful to all those who have contributed to the success of ICCET 2009. We hope that all
participants and other interested readers benefit scientifically from the proceedings and also find it
stimulating in the process. Finally, we would like to wish you success in your technical presentations
and social networking.

We hope you have a unique, rewarding and enjoyable week at ICCET 2009, ICACC 2009 and ICECS
2008 in Singapore.

With our warmest regards,

Yi Xie
January 22 - 24, 2009
Singapore

xv
 ICCET 2009 Committee Members
V.Saravanan, Karunya University, India
Gunter Glenda A., University of Central Florida, USA
Wen-Tsao Pan, Jinwen University of Science and Technology, China (Taiwan)
Gopalakrishnan Kasthurirangan, Iowa State University, USA
Anupam Shukla, Indian Institute of Information Technology, India
Wei Guo, Tianjin University, China
Mahanti Prabhat Kumar, University of New Brunswick, Canada
Hrudaya Ku Tripathy, Institute of Advanced Computer and Research, India
Narasimhan V. Lakshmi, University of Newcastle, Australia
Jinlong Wang, Qingdao Technological University, China
Amrita Saha, West Bengal University of Technology, India
Yi Xie, Cagayan State University, Philippines
Sevaux Marc, University of South-Brittany, France
Amir Masoud Rahmani, Islamiz Azad University, Iran
Yang Laurence T., St. Francis Xavier University, Canada
Lau Bee Theng, Swinburne University of Technology Sarawak, Malaysia
Poramate Manoonpong, University of Gottingen, Germany
Tahseen A. Jilani, University of Karachi, Pakistan
Qian Chen, Columbia University, USA
Zhihong Xiao, Zhejiang Wanli University, China

xvi
 ICCET 2009
Organizing Committees
Honor Chairs
R. C. Eberhart, Purdue University, USA
A. Kandel, University of South Florida, USA
J.D. Pinter, Dalhousie University, Canada

Conference Chairs
S.R. Bhadra Chaudhuri, Bengal Engineering and Science University, India
Jianhong Zhou, Sichuan University
Xiaoxiao Zhou, Nanyang Technological University, Singapore

Conference Steering Committee

Yi Xie, Cagayan State University, Philippines
Hoang Huu Hanh, Hue University, Vietnam
Kamaruzaman Jusoff, Yale University, USA

Program Committee Chairs

S.M. Aqil Burney, University of Karachi, Pakistan
Nazir Ahmad Zafar, University of Central Punjab, Pakistan
Nashat Mansour, Lebanese American University, Lebanon

Publicity Chairs
Basim Alhadidi, Al Balqa’ Applied University, Jordan
M. Aqeel Iqbal, Foundation University, Pakistan
Brian B.C. Shinn, Chungbuk National University, Korea

xvii
 International Conference
on Computer Engineering and Technology

Session 1
 2009 International Conference on Computer Engineering and Technology

Overlapping Non-Dedicated Clusters Architecture

Martin Šťava and Pavel Tvrdı́k

Department of Computer Science and Engineering
Czech Technical University in Prague
Prague, Czech Republic
{stavam2,tvrdik}@fel.cvut.cz

Abstract—Non-dedicated computer clusters promise
more efficient resource utilization than conventional
dedicated clusters. Existing non-dedicated clustering so-
lutions either expect trust among participating users, or
they do not take into account a possibility of running
multiple independent clusters on a same set of comput-
ers.
In this paper, we argue how an ability to run multiple
independent clusters without requiring trust among
participating users can be capitalized to increase user
experience and thus attract more users to participate
in the cluster. A generic extension of non-dedicated
clusters that satisfies these requirements is defined and
a feasibility of one particular extension is demonstrated
on our implementation.
I. I NTRODUCTION
Clusters build from commodity computers are a popular
computational platform. They are used as a cost-effective
alternative to expensive supercomputers [1], [2], as a scal-
able high available solution for commercial applications [3]
as well as load leveling clusters for ordinary day-to-day use
[4], [5].
A concept of traditional dedicated clusters was extended
by several existing projects [5]–[8] to support utilization
of non-dedicated computers, usually standard workstations.
These projects rely on users offering their idle computers to
participate in a cluster.
Methods of attracting users to participate vary. Some
projects offer only a good feeling from the participation. In
other environments, like university laboratories, the partici-
pation may be enforced by a system administrator. The most
interesting method used, however, seems to be a reciprocal
offer of cluster computing power to volunteering users. In
this case, the users are granted a computing power of the
cluster proportional to the power they have given to the
cluster.
The reciprocal computing power trading, however, needs
a well suited cluster architecture. In case there is just a single
instance of non-dedicated cluster running, the volunteers
cannot use the earned cluster computing power directly from
their machine, instead they first need to login to the cluster
and perform their resource demanding computations there.
Such a scenario is clearly not well suited for the volunteering
users. For example, a user who is about to perform a parallel
compilation on his machine would like to use his granted
cluster time, but he can not, since he can perform the cluster
operations only from the cluster itself, not from his machine.
On the other hand, if there is support for coexistence of
multiple clusters, the machines of volunteering users can
form their own clusters and the users can use the granted
cluster time transparently from their machines.
Second important aspect for attracting users to participate
in non-dedicated clusters is a trust relationship. Users offer-
ing their computers as non-dedicated computing resources
should not be required to fully trust the cluster administrators
and neither should the administrators be required to trust the
users. Any such a trust requirement complicates forming and
expansion of the cluster.
In this paper, we first briefly review the most important
existing architectures. Then we present a relaxation of ex-
isting architecture concepts and argue abouts its advantage
over existing systems. A feasibility of the architecture is
demonstrated for one particular case on our research clus-
tering solution called Clondike [8].
II. S COPE
We primarily focus on clusters attempting to provide a
single system image (SSI) illusion at the operating system
level. This is a reasonable limitation, since the motivation for
developing SSI clusters is, similarly as in our architecture,
in improving user experience.
III. E XISTING ARCHITECTURES
The most common form of clustering are dedicated clus-
ters. In these clusters, all machines are fully dedicated to the
cluster, they all share user account-space, process space, and
file system. Kerrighed [9] or OpenSSI [10] are well known
examples of such clusters.
An extension of dedicated clusters are non-dedicated clus-
ters [5]–[8]. These clusters consist of one or more dedicated
machines, forming a core of the cluster, and any number of
non-dedicated machines. The non-dedicated machines can
join or leave cluster at any time, but they do not fully belong
to the cluster even at the time they are joined. This separation

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DOI 10.1109/ICCET.2009.66
is often achieved by running the cluster code inside virtual
machines running on the non-dedicated machines.
An interesting alternative to standard architectures is rep-
resented by the openMosix [4]/Mosix2 [11] projects. Cluster
machines do not share common file-system, but they are still
assumed to share the user account-space. Users of a cluster
are not provided standard SSI features, but rather a different Figure 1. Dedicated cluster. All nodes belong to a single cluster.
SSI depending on a machine they are logged in.
Another significant architecture are multi-clusters. This is
not really an architecture of a single cluster, but the term
refers to clusters interconnected together. Multi-clusters are
gaining popularity during last years as a next logical step
towards an idealized grid solution. Examples of projects
supporting the multi-cluster architecture are Mosix2 [11] or
LSF [12].
Figure 2. Non dedicated cluster. The upper left node is forming a core of
non-dedicated cluster and is using 2 other nodes as non-dedicated nodes.
IV. P ROPOSED ARCHITECTURE

In this section, we define our envisioned architecture.

A basic building block of the proposed cluster can be
either a single machine or a dedicated SSI cluster. There
can be mixed environments, where some blocks are single
machines and some are clusters. We will refer to these blocks
uniformly as nodes in both cases, distinguishing explicitly
between clusters and single machines where required.
The nodes can be connected in an arbitrary way, but the
key factor is that each node forms a core of its own SSI
cluster, using the other nodes as non-dedicated blocks. By
forming a core we mean that it defines its account space,
file system, and process space.
Every node can possibly form its own administrative
domain. In addition, there should be no requirement for
trust among participating nodes. These two attributes imply
a need for strong security model of the architecture imple-
mentation.
As implied by the definition, every node can use the
others as its non-dedicated nodes. When some node is used
as a non-dedicated block, its own view of SSI still exists.
Moreover, its SSI view is fully separated from the SSI view
of the node that is using this node as its non-dedicated node.
Nothing prevents 2 nodes to interact with each other, using
each other as its non-dedicated node.
A node can be used as a non-dedicated node by more
than one node. In that case, all SSI views participated by
the node, including the local view, should be separated from
each other.
Because of the nature of the proposed solution, we will
refer to it in the paper as an overlapping non-dedicated
cluster (ONDC).
illustrate schematically the difference among the three types
of clusters.
Mosix with its architecture is very close to the ONDC
with all nodes as single machines, but its architecture seems
to be driven more by technical aspects than by an intentional
design. The biggest problem with Mosix is that it requires
a full trust among all participating nodes (and it assumes
either shared user account space or a consensus on mapping
of user ids).
Multi-clusters are as well a special case of ONDC.
Similarly as Mosix, current projects require trust among the
participating clusters, which may be a limiting factor in real
deployments.
The existing grid solutions are close to ONDC especially
when ONDC is used for a large scale deployment in a dis-
tributed area. The main difference is that the grid solutions
are primarily targeted only on large scale deployments. In
contrast, ONDC can be useful as well for a local resource
sharing. User having a few machines at home would likely
not use any of grid solutions to interconnect them, but
an ONDC cluster may be a good candidate for that.
The concept of overlapping clusters is similar to virtual
organizations mechanisms (VO) [13], [14]. Modern grid
solutions, like XtreemOS [15], are often based on VOs. The
main difference between VO and ONDC concept is that the
virtual organizations are designed for a mutual cooperation

V. C OMPARISON WITH OTHER ARCHITECTURES

Non-dedicated clusters are an extension of dedicated

clusters. The ONDC is an extension of the non-dedicated Figure 3. ONDC. All nodes are using the other nodes as non-dedicated
clusters, thus it is a super-set of both. Figures 1, 2, and 3 nodes. All clusters are isolated from each other.

4
agreement and some degree of trust, while our solution does
not require any relation among cluster users. ONDC is,
indeed, based on the assumption that the users does not know
each other and have a very limited trust among themselves.
VI. A DVANTAGES OF THE ONDC OVER OTHER
ARCHITECTURES
The key advantage of the ONDC architecture is an unique
combination of a system without trust requirements and
ability to form a separate cluster from each participating
node. This combination of features has a high potential for
attracting users to participate in the cluster.
By relaxing the trust requirement, users can easily join
the cluster. This idea was already leveraged by successful
projects like BOINC [1]. If there are trust requirements,
users generally have to undergo some registration and pos-
sibly a (mutual) trust review process, which itself may be
sufficient to deter users from joining.
By allowing a coexistence of multiple independent clus-
ters we enable a natural user rewarding mechanism, where
the users can get back the resources offered to the cluster
by using the other nodes.
The ability of each node to form its own cluster (and
hence export its file system) is another factor contributing
to easy expansion of a cluster. Any user needs just a common
ONDC code and does not need to install or configure
anything specific for the other clusters. The cluster nodes
can immediately use his machine as a non-dedicated node
and the user’s node can as well immediately use all other
machines.
Another advantage of allowing each node to form its
own cluster is a natural option of coexistence of different
installations of clusters (even with conflicting versions of
software installed on them) on the same physical hardware.
An important architectural advantage of the ONDC ar-
chitecture is a better fault-tolerance with respect to stan-
dard non-dedicated clusters. Fault-tolerance in standard non-
dedicated clusters relies on fault-tolerance mechanisms of its
dedicated core. When the core fails, the whole cluster fails.
In ONDC, when some core fails, the cluster formed by this
core stops to work, but all other clusters are still functional
(they just possibly loose some processes running on the
crashed node). Clearly, this does not increase fault-tolerance
of any cluster in the ONDC. But the non-dedicated clusters
are generally based on the idea of utilizing idle machines
and the ONDC allows continuous utilization of those idle
machines even in presence of some cluster failures.
VII. U SE CASES
In order to better illustrate the architecture, we will
describe a few possible use-cases of ONDC in this section.
These are just an illustrative examples. All of the mentioned
examples can coexist and cooperate as a single instance of
ONDC.
5
The smaller scale example can be a university computer
laboratory. Currently, if the clustering is to be used in this
environment, there is either a single cluster shared by all
users (enforcing the same environment for them), or there
is some non-transparent job scheduling system, where the
users can send their jobs to be processed. With ONDC, each
computer can form its own cluster, using resources of any
other computer which is not being used at the moment. In
addition, if any user brings his own laptop, he can simply
plug it to the network, and start using the other computers as
non-dedicated nodes of a cluster based on his laptop (and of
course, the computers in the laboratory can use his laptop,
if it is idle). He does not need any administrator privileges
for the computers in the network.
Another use case of ONDC are the multi-clusters. There
is a clear demand for such computing platforms, underlined
by the existence of commercial solutions like Mosix2 or
LSF. The existing solutions could benefit especially from
the security research of the ONDC architecture, as this is
directly applicable to them.
In a largest scale, the ONDC architecture can be used as
a world-wide cluster, similar to the SETI [16] or BOINC [1]
projects. For such a project, a standard non-dedicated cluster
can be sufficient. ONDC based solutions could possibly
attract larger user base, since users can be rewarded by their
offered time with the proportional cluster computing power.
In addition, with ONDC not only single volunteer computers
can be connected, but whole clusters and multi-clusters can
be connected and offer their spare computing power. As
long as they are already taking use of (compatible) ONDC
infrastructure, it would be a simple configuration task to let
the participation in the wider cluster build on ONDC.
VIII. R ELATIONSHIP WITH MULTI CORE COMPUTING
Any project being developed should plan for the future as
well. The ONDC architecture targets on using commodity
computers. It is always hard to predict future hardware
evolution, but commonly believed future development in the
industry is that the commodity computers are going to have
more and more cores, while the cores themselves will not
become much faster.
This is, indeed, a perfect match for ONDC architectures.
There are 2 main factors. First, the developers of software
will have to focus more on the parallelization of common
programs. The cpu utilization patterns we can expect are as
follows. Most of the cores would be utilized for a limited
time when cpu intensive parallelized tasks are running. For
the rest of the time most of the cores would be idle. Actually,
such patterns are already seen now with current machines.
For example, in IT companies with frequent application
compilations or in graphical studios where rendering rep-
resents the most CPU intensive operations.
ONDC can contribute a lot to such environments. Properly
parallelized applications can use other machines computing
power at the time of high CPU demand. Assuming that
the limited high cpu demand periods do not always overlap
for all machines, there can be often some spare resources
available in the network (of course, the biggest problem
here can be support for distributed shared memory and I/O
bounded tasks, but this is more technical question and is out
of the scope of this paper).
The second factor contributing to non-dedicated clusters
generally is the observation that some programs are not
easily parallelizable (either due to the nature of algorithms or
due to a prohibiting complexity of such a parallelization).
Users who work mostly with such programs would have
some of their cores idle for most of the time. Machines
of these users are good candidates to participate as non-
dedicated nodes in any non-dedicated clustering solution.
Clearly, more research is needed in this area, especially
to measure impact of such a sharing, since the usage of
another core is not for free (cache conflicts, bus contention,
power consumption, etc.) and the cpu is not the only resource
consumed by a running application (memory usage, network,
disk I/O, etc.).
IX. I MPLEMENTATION
The proposed ONDC architecture is quite generic and
there may be many implementations. The existing projects
closest to the ONDC architecture are multi-clusters and
Mosix. They fail, however, to address the trust less nodes
cooperation requirements. To verify our ideas, we have
started development of our own implementation of the archi-
tecture. In this section we will briefly describe our system
and its technical background, changes required to meet the
ONDC and how we address 2 key topics - scheduling and
security/trust handling.
A. Clondike
The original idea of Clondike [8], [17] was to implement
a first non-dedicated clustering solution based on the Linux
operating system. Clondike is still in an experimental re-
search phase, but it already supports most of the require-
ments on such systems.
Clusters based on Clondike consist of one dedicated node,
called core node and a number of non-dedicated nodes called
detached nodes. This is a typical setup of any non-dedicated
cluster, although some may contain a whole dedicated cluster
as their core.
A key feature of Clondike is a support for both preemptive
and non-preemptive process migration based on process
checkpointing. With the migration support, it is possible to
utilize detached nodes that would sit idle otherwise. The
usage of detached nodes is continuously monitored by the
core node and if there is some opportunity to migrate a
core node local process to idle detached node, the migration
mechanisms are used.
6
B. Technical background
The implementation consists of 3 parts. The lowest level
is a kernel patch, that is kept as minimal as possible so that
upgrades to new kernels are not unduly complicated. The
patch consist mostly of a simple hooks for second part of the
system that are a kernel modules. These modules implement
most of the lowest level functionality required for process
migration and actually the process migration support itself.
This is technically a most complicated part and a description
of this implementation is out of the scope of the paper. The
implementation details can be found in [17].
Finally, there is a userspace part of the system. It makes
use of the kernel part of the system, interacting with it via
a special control file system (exporting cluster specific data
about processes), system signals (for requesting migration)
and as well using a standard linux kernel netlink sockets
when a bidirectional interaction is required (for example
non-preemptive scheduling decisions on process execution).
The userspace part performs all tasks that do not need to be
directly in kernel, like scheduling, monitoring or information
distribution. From a practical point of view (coding and
debugging) it is a big advantage to put as much functionality
as possible to userspace.
C. Changes required to support ONDC
Clondike was since the beginning designed to allow
cooperation of untrusted parties, so this functionality did
not require any modification to match ONDC requirements.
In order to allow overlapping clusters, an extension to
standard Clondike system was required. The system needs
to allow coexistence of a core node and a detached node
on a single physical machine. This way, it can act as a
core of its own cluster, while still offering its resources to
another cluster. Second related extension is an ability to act
as a detached node of multiple independent clusters. This
is a natural requirement, if we have for example cluster of
3 nodes (Alfa, Beta, and Gamma) each of them forming
its cluster, we would like them to use all other nodes as
detached nodes. Therefore Alfa will use Beta and Gamma
as detached nodes, while Beta will use Alfa and Gamma as
detached nodes. Gamma is then acting as a detached node
for 2 independent clusters (and as a core node of its own
cluster).
D. Scheduling
As outlined in previous sections, the main advantage of
overlapping clusters support is a possibility of a scheduling
based on reciprocal computing power exchange. Economy
inspired market based schedulers [18], [19] studied in a stan-
dard non-dedicated Clondike environment [20] seems to be
a promising candidate for our goals. Mosix multi cluster
solution has some support for overlapping clusters and the
market based schedulers are one of the scheduling options
used [21].
 Market based schedulers seems to be a better candidate for
ONDC than for standard non-dedicated clusters. In standard
non-dedicated clusters with market based schedulers, the
situation is a bit cumbersome. The user offers his machine
to the cluster and gets some credit for that. He can then
use this credit for execution of his tasks, but not directly
on his machine (since his machine acts as a non-dedicated
node only. In non-overlapping cluster case, the tasks started
locally on detached nodes are not clustered, they are in fact
isolated from the cluster environment as much as possible).
He must login to the cluster core and he can use his credit
only there. In ONDC, he can use his credit transparently
from his machine, since that machine forms a cluster on its
own.
To illustrate the difference between non-dedicated cluster
and ONDC we will use an example of 2 nodes cluster,
with nodes Alfa and Beta. Let Alfa be a core node of non-
dedicated cluster and Beta a detached node. When somebody
runs a calculation on Alfa that will use resources on Beta,
the owner of Beta will get a credit to run a comparatively
expensive calculation on cluster formed by Alfa. To use his
credit, he needs to login to Alfa and execute some process
there (since Beta is only a detached node). In contrast, in
ONDC case both machines can be a core node. So when the
owner of Beta has credit to execute something on Alfa, he
can execute a local process and that could be transparently
migrated to Alfa and use the credit there.
Despite the apparent suitability, the market based schedul-
ing strategies were not yet tested in ONDC version of
Clondike system and it is a future research topic to do
so. Since the scheduler in Clondike resides completely
in userspace, it is much easier task to implement a new
scheduler than in other similar systems.
A simpler and more straightforward scheduling strategy
was used in the existing prototype. Each cluster in the
ONDC version of Clondike has its own scheduler running
on the cluster core node. This scheduler is tracking load and
a count of cluster tasks running on associated detached nodes
and it is trying to level the load of these nodes (including
the core node itself, which is preferred for some tasks). The
scheduling decision is performed only at the process start
time, so only non preemptive migration is used. To honor
the local user priority over cluster users, the node with some
active local tasks does not accept any cluster jobs, even if
some scheduler requests such a migration (so, if machine
Alfa is running a local user’s job and a scheduler on machine
Beta requests a job migration on Alfa, he is refused as long
as the local tasks are running). Similarly, if the machine has
too many remote jobs running on itself, it does not accept
any new migration requests.
The scheduling algorithm performed by each node is as
follows:
1) If local cluster running tasks
7
count is lower then threshold,
do not emigrate task.
2) Else find the least loaded remote
node, that is bellow its accepting
threshold and try to emigrate there
3) If no remote node is found,
the task is kept running locally.
This strategy is very simple and has many problems in
a real life, none the less it served well for system testing
as will be demonstrated in the Performance section. It make
use of the cluster overlapping nature and allows reciprocal
computing power sharing, but unlike the market strategies,
it does not have any fairness guarantees. Obviously, it is
usable only in a smallest scale and only for scheduling
of sufficiently independent processes (i.e. not for collection
of closely cooperating highly dependent processes like for
example a MPI application. Such processes may need some
co-scheduling [22] techniques to perform well.)
E. Security and trust
The security functionality and trust management of origi-
nal Clondike system is directly applicable in ONDC environ-
ment. In this section we will briefly review the mechanisms
used, details can be found in [23], [24].
Since non dedicated clusters span over administrative
domain boundaries and they are potentially used in an
untrusted network environment, the line security must be
ensured. Clondike is currently relying on establishing IPSEC
channels between nodes to provide transparent link level
security.
By the ONDC definition, there should be no trust require-
ment between node owners. From the security point of view,
it implies two main possible classes of attacks. First, an
owner of machine forming cluster core node can try to send
malicious code to remote machine to get access to that node.
Second, the owner of machine acting as a detached node can
read anything from memory of cluster processes running on
his machine. Moreover, he can alter those processes memory
and code.
The first attack is technically easy to prevent. The pro-
cesses of cluster users run with the least possible privileges
on the detached nodes and thus the detached node is pro-
tected against these processes.
There is no reliable way how to prevent second type
of attacks, where detached node owner reads or modifies
cluster processes running on his machine. The owner of
the detached node has superuser privileges so he can do
basically everything. In some special cases the results of
remote execution can be quickly verified on the core node,
but this is not always the case.
Our approach to this issue is based on deferring the
security critical decisions to the cluster users. Each user
can specify what programs can run on what machine. In
 Table 1
addition, the user can specify the files from his file system PARAMETERS OF THE TEST MACHINES . M EMORY IS IN G IGABYTES ,
that can be access from other nodes. All nodes for which BUILD TIME IN FORMAT MINUTES : SECONDS , AND THE SEQUENTIAL
TIME IS IN SECONDS .
no specific rules are specified are considered of a same trust
level, we refer to them as anonymous nodes. The anonymous Name Cores Mem. Build time Seq. time
Alfa 2 4 2:13 6
nodes can as well participate in cluster, the users can use Beta 2 2 3:28 7
them for executing of processes whose results can be either Gamma 2 2 3:46 8
easily verified, or for processes that perform operations with Delta 1 1 6:43 11
non-sensitive data.
To illustrate the decisions the user can make, we will use
example with 3 nodes Alfa, Beta, and Gamma. A user on stigmata. If any of the visited nodes does not have required
Alfa can trust owner of Beta, but trust less (or perhaps do not privilege, the process is terminated.
even know about) machine Gamma. The user would specify
that any process can be migrated to Beta and the Beta can X. P ERFORMACE
access any file on his file system. The user on Alfa has as
well some process that performs some resource demanding,
but easily verifiable operations (NP-complete calculations
are one possible example). He can then specify that this
process can be executed on any node. The process may need
to write result somewhere, so the user may need to give a
write access to some restricted part of the file system to any
node, so that the result can be written.
The user defined restriction has to be obeyed by the
scheduler. The scheduler, however, cannot always know
which files are going to be accessed by a process being
migrated to a detached node. So the processes on the
detached nodes must be monitored for a file system access
requests and when a violating access is detected an action
must be taken. The action can be either rollback to previous Figure 4. Graph with times when a single compilation was started from
process checkpoint, taken before migration to the detached the machine Delta
node, or a process termination. Migration back to core node
is not an option, since the process may be already altered
by the owner of detached node.
The last problem in the user defined restrictions enforce-
ment is a direct result of preemptive migration support.
Thanks to the migration capabilities, the process can visit
many nodes during its life time. An owner of any of visited
detached nodes can alter the process being executed. This
means, the file system access violation checks must not
consider only the node from which they are being performed,
but all nodes that were visited during the process execution.
The problem can be illustrated again on our 3 nodes
cluster example. When a non-sensitive process is migrated to
untrusted node Gamma, the owner can modify the process.
For example, he can change it to delete all accessible files. If
the process gets to execution of these command on Gamma, Figure 5. Graph with times when a single compilation was started from
it will be eventually terminated due to access violation. In the machine Alfa
case the process migrates back to Alfa before executing the
deletion code, and the process visit history is not consulted, There is a vast amount of cases that could be measured.
it would be allowed to execute and all accessible files would We have decided to demonstrate one common possible use
be deleted. case - the parallel compilation. This is a good representative
A solution to this (and related) problem is discussed in of a harder class of problems, as there is relatively big
[24] and a stigmata mechanism is proposed. Each process overhead due to a high communication-to-computation ratio.
is marked by a stigma of all nodes visited and any sensitive Moreover, the applications used were not designed to run in
operation is consulted against user defined rules and all a cluster environment.

8

Figure 6. Graph showing overheads of runs corresponding to Figures 5 and
4 both with and without IPSEC. The overheads are expressed in percents.
Figure 7. Graph with times when each machine simultaneously started
one compilation
In our demonstration, we use standard unmodified gnu
tools [25] like make, gcc, etc. The application being
compiled is a Linux kernel, which is a sufficiently large
application to benefit from cluster parallelization.
Clondike does not have all security mechanisms imple-
mented yet, but the performance most demanding part of
the security, the IPSec [26] based channel security, was used
and therefore the performance figures are representative.
Tests were performed on a realistic platform, using 4
heterogeneous computers. All test machines have x86 64bit
architecture and are interconnected with a standard 100Mbps
Ethernet. Table 1 lists all key characteristics of the machines
used for testing. It is generally hard to compare machines
performance, so rather than meaningless frequency numbers,
the table captures the time it takes to build the kernel on each
of the machines. Another important value is a sequential
portion of the build time - this includes mainly the final
linking.
Each test was performed 10 times and the presented values
represent the minimum time achieved. It is not purpose
9
of this paper to do a detailed analysis of variance (time
variance was in range of few seconds), worst case scenarios,
etc. In cases when multiple concurrent compilations were
measured, the choosen set of results was one with a shortest
time of the compilation on the slowest machine.
The first set of performed tests demonstrates a standard
non-dedicated cluster functionality of Clondike. Figure 4
shows compilation times of a single compilation started
from the slowest machine (Delta). Each group of bars shows
compilation times for different cluster configurations. In
each bar group, there are 3 running times: the best time
achieved in the cluster, the best time achieved in a cluster
with IPSEC, and a theoretical minimum time. The first 2
times are clear. The theoretical optimal time is calculated as
follows:
X
St + ((Ts − St )/(Ts )) ∗ (1/( 1/Ti )),
i∈setof participatingnodes
(1)
where St denotes the sequential part of the compilation,
Ts denotes the time of compilation on the node that started
it, and Ti denotes the sequential compilation time on node i.
The set of participating nodes includes the node that started
the compilation.
The theoretical minimal time accounts for the sequential
time of the calculation, in order to isolate inherent par-
allelization limitations (due to Amdahl’s law) from inef-
ficiencies of the system itself. Ratio of achieved times to
theoretical minimum times calculated this way reflects the
overhead of the system (there is still some inherent limitation
due to network transfers, but this cannot be so easily cleaned
up).
There are a few noteworthy observations regarding the
graph on Figure 4. First, the measured times are very good,
even though the overhead is increasing with the number of
nodes, as can be seen on Figure 6. The increasing overhead
is due to inefficiencies of current experimental scheduler
that cannot effectively use all machines especially in the
end of calculation. The second important observation is
that the overhead due to security (represented by IPSEC)
is apparent, but still quite small (15% in the worst case).
Quite low security overhead is due to the fact that IPSEC
was configured to run the AES encryption, which is very
fast on 64bit platforms.
In contrast to Figure 4, Figure 5 captures the results of
a single compilation started from the fastest machine (Alfa).
The important observation here is that the compilation has
lower overhead both with and without the IPSEC. The key
factor for the lower overhead is the fact that less percentage
of work is sent to other machines. In addition, based on the
runtime observation of the system behavior while compiling,
it seems that the scheduler can more effectively use other
machines while running on Alfa. This is, however, an area
that requires more research in future.
 The second set of tests captures an ONDC-specific use
case in which all machines start a compilation at the same
time. This is a use-case, that cannot be performed on any
standard non-dedicated cluster, since only core machines can
use the others in that case. In ONDC, each machine should
first do only its compilation and offer resources only after it
is done with that. Results of this test can be seen on Figure 7:
each machine completes its work in about the same or better
time than in non-clustered case (the slightly worse times in
some cases are only due to random time variations).
Probably the most important number is the time of a
compilation on the slowest machine (Delta), since it shows
the total time spent since the start of the compilations till
the end of the last. If we divide this number by 4, we get
an average build time of each of the kernels in this case.
In numbers, it would be around 58 seconds for a non-
secured compilation. The best non-secured time measured
when starting from Alfa was 1 minute and 2 seconds, so
we can see that this case is even more resource utilization
efficient.
There can be many other test combinations even for
this simple use case, but we believe the presented results
demonstrate clearly enough that parallelization of a ordinary
programs can be achieved with acceptable overheads.
XI. C ONCLUSIONS
In this paper, we have defined an extension of existing
clustering concepts. We have discussed its relationship with
other architectures and its advantages. We have argued why
we believe that the architecture is going to be even more
interesting in the future. To verify our ideas, we have
demonstrated feasibility of one possible use case of the
architecture.
ACKNOWLEDGEMENTS
This research has been supported by the Czech Grant
Agency GACR under grant No. 102/06/0943 and by the
research program MSMT 6840770014.
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 2009 International Conference on Computer Engineering and Technology

To Determine the Weight in a Weighted Sum Method for Domain-Specific Keyword

Extraction

Wenshuo Liu Wenxin Li

Key Laboratory of Machine Perception
Peking University
Beijing Supertool Internet Technology Co.Ltd
Beijing, China
{lwshuo, lwx}@pku.edu.cn

Abstract—Keyword extraction has been a very traditional
topic in Natural Language Processing. However, most
methods have been too complicated and slow to be applied
in real applications, for example in web-based system. This
paper proposes an approach which will complete some
preparing works focusing on exploring the linguistic
characteristics of a specific domain. This part can be
completed once and for all and thus reduce the burden in
the real extraction process. It is a weighted sum method and
the preparing work focus on finding out the weight. Once we
have the weight, the extraction can be completed by
addition, multiplication and sort, which are quite simple for
modern computer. Experimental results show the
effectiveness of the proposed approach.
I. INTRODUCTION
Keyword extraction is the process of extracting a few
salient words from a certain text and using the words to
summarize the content. This task has been widely studied
for a long time in the natural language processing
communities, because it is important for many text
applications, such as document retrieval. Domain-specific
keyword extraction came into sight when researchers
found out fully exploiting domain-specific information
can greatly improve the performance of this task.
Traditional methods focused on efficient algorithms
to improve the performance of the task of keyword
extraction. This paper presents a method emphasizing on
doing enough and effective preparing works, in order to
simplify the real extraction process. The difference is
illustrated in Figure 1. While in traditional methods,
every step must be performed for every document, the
weight extraction part in my work is finished once and for
all. In my work, keyword extraction involves assigning
scores to each candidate words considering various
features, sort the candidates according to the score and
choose the few top ones. Four different features are used:
TF×IDF, the part-of- speech (PoS) tag, relative position
of first occurrence and chi-square statistics. Some
experiments show that, the weighted sum of the feature
vector can be a good choice for the score, as long as we
have a proper weight vector. Different domains have
different characteristics in the usage of word, so a certain
weight vector working well in one domain might be
totally ineffective in another domain. The weight vector is
mainly the domain-specific information we here need to
explore. This article is primarily about constructing a
model to learn the weight vector. The model is very much
like a perceptron.

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DOI 10.1109/ICCET.2009.136
 II. RELATED WORK
Probabilistic methods and machine learning have
been widely used in the task of keyword extraction. Peter
D. Turney (1999) developed the system GenEx. The
system exploited genetic algorithm and used to be the
state of the art. Eibe Frank, Gordon W. Paynter, Ian
H.Witten, Carl Gutwin, and Craig G. Nevill-Manning
(1999) described a simple procedure, called KEA, which
was based on Naïve Bayes. KEA was proved to be
equally effective compared with GenEx, and even
outperformed when fully exploiting domain-specific
information. Hulth (2004a) and Hulth (2004c) presented
approach using supervised machine learning. Their
approaches constructed the prediction models from texts
with manually assigned keywords. Graph-based
algorithms have also been explored. Xiaojun Wan,
Jianwu Yang and Jianguo Xiao (2007) proposed an
iterative reinforcement approach to simultaneously
finishing the task of keywords extraction and document
summarization. Their approach fully exploited the
sentence-to-sentence, word-to-word, and sentence-to-
word relationship.
III. DATA REPRESENTATION
The input document will be first split up to get
separate terms. However, the terms themselves are
useless. It is their attributes that matter. In this article, I
choose four attributes, as mentioned above, to form a
feature vector for each candidate keywords. Web page
articles, which are classified into suitable domains, are
used for training and testing. Those articles all have
manually assigned keywords for the model to learn. For
any document, each candidate words will be represented
as a four-dimension feature vectors. Words which are so
common that they have no differentiating ability, such as
“的”, have been stored in a stop list and removed during
pre-processing.
A. TF×IDF
TF×IDF combines term frequency (TF) and inverse
document frequency (IDF). It is designed to measure how
specific a term T is to a certain document D:
12
TF×IDF (T, D) = P[term in D is T] × log P[T in a
Document]. (1)
The TF is measured by counting the times that term
T occurs in document D, and the IDF by counting the
number of documents in the corpus in a specific domain.
B. POS
When inspecting manually assigned keywords, the
vast majority turns out to be nouns. But there are still
differences between different domains. For example, in
entertainment news, keywords might always be people’s
names, which are nouns. But in sports field, verbs are also
quite important. We count the occurrences for each kind
of PoS tag as manually assigned keywords in the whole
corpus and then divide by the total number of keywords.
For example, when we consider noun:
PoS(noun)=
manually assigned keywords which are noun
manually assigned keywords
(2)
The results are numbers between 0 and 1 and they
indicate which kinds of words are more likely to be
keywords in the target domain.
C. Relative Position of First Occurrence
Not only the occurrence, but also the location of the
terms is important. Terms occurring in, for example,
headlines and in sentences at certain positions, such as in
the first sentence of paragraphs, are shown to contain
more relevant terms. This feature is calculated as the
number of words that precede its first appearance, and
then divided by the document’s length. For example,
considering term T in document D:
RPFO(T,D) =
the position of first appearance
the length of the document
(3)
The result is a number between 0 and 1 and indicates
the proportion of the document preceding the term’s first
appearance.
D. Chi-Square Statistic use the result to update the weight by adding the result to
For term T and Domain D, chi-square statistic is it. For example, when we consider the feature TF×IDF:
defined as: E (keywords 'TF × IDF )
β TF×IDF =
CHI (T , D) = (n11 × n22 − n12 × n21 ) ∑ TF × IDF (5)
(n11 + n12 + n21 + n22 ) (4) E (non − keywords ' TF × IDF )
× −
(n11 + n12 )(n21 + n22 )(n11 + n21 )(n12 + n22 ) ∑ TF × IDF
In the equation, n11 indicates the times that T occurs After similar calculation we have a vector β
in domain D, n21 indicates the times that terms other than as ( β TF×IDF， β PoS， β FirstOccurence， β Chi ) for update.
T occur in domain D, n22 indicates the times that terms
other than T occur in domains other than D, n22 indicates So after the nth article, we update the weight vector
the times that T occurs in domains other than D. The chi- ω by: ωn+1 = ωn +β
square statistic is used to test the dependence between a
term and a domain. The higher this value is, the more (6)
dependent term T is on the domain D. If n11×n22-n12×n21 > Thus with every article, we examine the difference
0, then term T is positively relevant to domain D, and if between keywords and non-keywords, and update the
n11×n22-n12×n21 < 0, then term T is negatively relevant to weight vector according to the difference. And finally we
domain D. have a weight vector.

IV. TRAINNING MODEL V. EXPERIMENTS AND EVALUATION

As the title suggested, it is a weighted sum method
and finally in the real extraction task all have to be done
are multiplication, addition and sort. So far we have
talked about how candidate keywords are generated and
represented. In order to get the weighted sum of the four-
dimension feature vector, we still miss the weight vector.
We need weights because the four features have different
discriminating ability. Apparently the more the feature
can discriminate between keywords and non-keywords,
the higher weight it should be assigned. We can find the
weight manually and actually that is the inspiration of this
work. But doing it manually is too time-consuming if we
try to determine weight vectors for many domains. We
use some learning ideas from perceptron.
All the articles in the training corpus are with
manually assigned keywords. The weight vector is
initially set to all zero, namely (0, 0, 0, 0). In every
article, first on each dimension we examine the average
value of keywords and non-keywords respectively,
calculate the difference between them, and divide the
results by the sum of all candidate keywords. Then we
We scraped 2563 web pages from
http://tech.mop.com/. All texts are about information
technology (IT). 1563 of them were used for training and
1000 for testing. The whole text and the keywords
manually assigned in the meta
keywords tag were extracted. We used the Chinese lexical
analysis system ICTCLAS (Institute of Computing
Technology, Chinese Lexical Analysis System) to
complete word segmentation and PoS tagging. This
system is developed by Chinese Academy of Science.
The weight vector we get for this IT domain is
(66.59, 22.76, 8.250, 4.202). On average, 6.8 keywords
are manually assigned per text. In the following
experiments we choose 7 words with the highest scores as
the keywords. The extracted 7 words are compared with
the manually assigned keywords. The results are
presented in Table 1, which shows: precision(P);
recall(R); and F-measure(F) .
P refers to the proportion of automatically selected
keywords which are also manually assigned. R refers to

13
the proportion of manually assigned keywords selected by the weighted sums, which are used to rank the candidate
this method. keywords, are not fully exploited. If used properly, they
If we denote: might well benefit text categorization, document retrieval,
ASMA = the number of terms both automatically and other natural language processing tasks
selected and manually assigned
AS = the number of terms automatically selected
ACKNOWLEDGMENT
MA = the number of terms manually assigned
Then P and R are defined as: I’d like to thank Minghui Wu, Xuan Zhao, Hao Xu,
ASMA Songtao Chi and Chaoxu Zhang from Beijing Supertool
P= (7)
AS Internet Technology Co.Ltd for all the help and
R is defined as: suggestions they have provided.
ASMA I also want to thank Chun-Tsung Endowment Fund for
R= (8)
MA giving me a chance to take part in real research.
F-measure combines precision and recall. It is usually I would especially like to take the opportunity to thank
used as a standard information retrieval metric: professor Von-Wun Soo, who has been so kind and given
2× P× R me a lot of valuable instructions while I was in National
F= (9)
P+R Tsing Hua University.
After observation, we found out that the keywords REFERENCE
manually assigned are not all reliable, so we manually [1] Xiaojun Wan, Jianwu Yang and Jianguo Xiao. 2007.
refined 200 documents. 100 of them are used for training Towards an iterative reinforcement approach for
simultaneous document summarization and keyword
and others for testing. Table 1compared the performance extraction. In proceedings of the 45th Annual Meeting of
the Association of Computational Linguistics (ACL’07). pp.
of our method on different data set. The second 552-559. Prague, June 2007
experiment shows a significant increase. This method [2] Anette Hulth and Beáta B. Megyesi. 2006. A study on
automatically extracted keywords in text categorization. In
relies heavily on the data. As long as the data is reliable, the proceedings of the 21st International Conference on
Computational Linguistics and 44th Annual Meeting of the
this method can perform quite well. Association of Computational Linguistics. pp 537-544.
Sydney, July 2006
VI. CONCLUSION AND FUTURE WORKS [3] Anette Hulth. 2004a. Enhancing linguistically oriented
automatic keyword extraction. In the proceedings of the
In this article, we explored a new method on domain- Human Language Technology conference/North American
Chapter of the Association for Computational Linguistics
specific keyword extraction. This method focuses on Annual Meeting (HLT/NAACL 2004). Boston, May 2004
doing enough and effective preparing works to explore [4] Anette Hulth. 2004b. Reducing false positives by expert
combination in automatic keyword indexing. In: Nicolov,
the linguistic characteristics of a specific domain, and N., Botcheva, K., Angelova, G., and Mitkov, R., (eds.),
Recent Advances in Natural Language Processing (III),
thus simplifies the real extraction task. And experiments Current Issues in Linguistic Theory (CILT), pp. 367–376.
John Benjamins.
show
[5] Anette Hulth. 2004c. Combining Machine Learning and
Natural Language Processing for Automatic Keyword
TABLE I. THE PERFORMANCE Extraction. (Ph.D. thesis.) Department of Computer and
Systems Sciences, Stockholm University.
P R F [6] Anette Hulth. 2003. Improved automatic keyword
extraction given more linguistic knowledge. In proceedings
raw data 0.449 0.554 0.496 of the Conference on Empirical Methods in Natural
Language Processing (EMNLP’03), pp.216-223. Sapporo,
refined data 0.644 0.732 0.685 July 2003.
[7] Takashi Tomokiyo and Matthew Hurst. 2003. A language
model approach to keyphrase extraction. In proceedings of
that it did lead to a better performance. However, there the 2003 Conference on Empirical Methods on Natural
are still much to improve. We still can not make sure the Language Processing (EMNLP’03). Sapporo, July 2003.

weight vector we get is the optimum solution. Moreover,

14
[8] Li sujian, Wang Houfeng, Yu shiwen and Xin Chengsheng.
2003. News Oriented automatic Chinese keyword
indexing, Sighan workshop ACL’03, Sapporo, July 2003.
[9] Peter D. Turney. 2000. Learning algorithms for keyphrase
extraction. J. Information Retrieval. pp.303-336. NRC
44105
[10] Eibe Frank, Gordon W. Paynter, Ian H.Witten, Carl
Gutwin, and Craig G. Nevill-Manning. 1999. Domain-
specific keyphrase extraction. In the proceedings of the
16th International Joint Conference on Artificial
Intelligence (IJCAI’99), pp. 668-673.
[11] Peter D. Turney. 1999. Learning to extract keyphrase from
text. Technical Report ERB-1057, National Research
Council, Institute for Information Technology, 1999

15
 2009 International Conference on Computer Engineering and Technology
Flow-based Description of Conceptual and Design Levels
Sabah Al-Fedaghi
Computer Engineering Department
Kuwait University
Kuwait
sabah@alfedaghi.com
Abstract—Building an Information System involves a first
phase of conceptual modeling of the “real world domain,” then
a second phase of design of the software system. For
describing the software system, object-oriented methods and
languages (e.g., UML) are typically used. For the system
analysis process that produces the conceptual description,
object-oriented techniques or their semantic extensions are
used. This paper concentrates on UML as the most widely used
language for these purposes. It proposes an alternative to
extending UML, through development of a conceptual
modeling methodology based on the notion of flow. This flow
model provides a conceptually uniform applied description
that is appropriate for the system analysis phase.
Keywords-conceptual modeling; UML; software system
analysis
I. INTRODUCTION
Understanding and communication processes in the
business and organizational domain are an essential aspect of
development of information systems. Conceptual modeling
is a technique that represents the “real world domain” and is
used in this understanding and communication. The resultant
model serves as a guide for the subsequent information
systems design phase that involves description of the
software system under development. This later phase
typically utilizes object-oriented techniques to model the
software system. In this paper, UML is used as an example
of an object-oriented modeling language and methodology.
An information system (IS) represents a real business
establishment. It should reflect the reality of an organization
and its operations. Consequently, building IS begins by
drawing a picture of the business establishment as part of a
real-world domain. A conceptual model is a picture
describing this real world domain independent of any aspects
of information technology. On the other hand, the IS design
phase is concerned with describing the information system
through design models intended to describe the software
system. UML is an important tool in both of these processes.
According to Bell [4], “production of UML diagrams, as
opposed to the engineering content behind them, is the single
most important activity in the software-development life
cycle.” Researchers have examined and some proposed
extending the use of object-oriented software design
languages such as UML (e.g., Evermann and Wand [10];
Dussart et al. [7]; Opdahl and Henderson-Sellers [12]).
According to Evermann and Wand [9], “UML is suitable for
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DOI 10.1109/ICCET.2009.130
conceptual modelling but the modeller must take special care
not to confuse software aspects with aspects of the real world
being modelled.”
The problem with this approach is “that such languages
[e.g., UML] possess no real-world business or organizational
meaning, i.e., it is unclear what the constructs of such
languages mean in terms of the business” [9]. The Object-
oriented IS design domain deals with objects and attributes,
while the real-world domain deals with things and properties.
It is known that UML lacks constructs to facilitate rich
conceptual modeling. Scott W. Ambler observed that
“although the UML is in fact quite robust, the reality is that it
isn’t sufficient for your modeling needs” [3]. He then
suggested using “UML as a base collection of modeling
techniques which you then supplement with other techniques
to meet your project’s unique needs” [3].
There are many proposed UML extensions to enhance its
capabilities through incorporating semantic features for
conceptual modelling. However, UML has been developed
with the specific intention of designing and describing
software. In contrast, conceptual modeling languages are
intended specifically for describing the application domain.
Works that extend UML for use in conceptual modelling
claim that UML can be used in both systems analysis and
system design; thus it eases the transition between these two
stages. This paper argues that it is not a matter of lacking
appropriate conceptual structures, but rather that UML is
inherently an unsuitable tool for use in conceptual modeling.
Furthermore, the paper argues that enhancing UML to
incorporate the semantics necessary for conceptual modeling
will not lead to achieving the desired results. Instead we
present a new conceptual model called the flow model (FM).
We claim that FM provides conceptually a “better”
description. FM has not yet been formalized; however, its
features suggest that it is also appropriate for the system
design phase. These claims are illustrated in Fig. 1.
Strongly claimed Weakly claimed
Conceptual model (real world domain descriptions)
FM FM
UML Semantically FM + (Needs further
extended UML research)
UML
Information system design and software modeling
Figure 1. The claims in this paper.
16

II. THE FLOW MODEL
The flow model (FM) was first introduced by Al-Fedaghi
[2] and has been used since then in several applications such
as information requirements [1]. Even though this section is
simply a review the basic model, it includes new illustrations
of the model.
A flow model is a uniform method to represent things
that “flow,” i.e., are exchanged, processed, created,
transferred, and communicated. “Things that flow” include
information, materials (e.g., manufacturing), money, etc. To
simplify the review of FM, we introduce flow in terms of
information flow.
Information goes through a sequence of states as it moves
through stages of its lifecycle, as follows:
1. Information is received (i.e., it arrives at a new sphere,
like passengers arriving at an airport).
2. Information is processed (i.e., it is subjected to some
type of process, e.g., compressed, translated, mined).
3. Information is disclosed/released (i.e., it is designated
as released information, ready to move outside the current
sphere, like passengers ready to depart from airports).
4. Information is transferred (disclosed) to another
sphere, e.g., from a customer’s sphere to a retailer’s sphere).
5. Information is created (i.e., it is generated as a new
piece of information using different methods such as data
mining).
6. Information is used (i.e., it is utilized in some action,
analogous to police rushing to a criminal’s hideout after
receiving an informant’s tip). Using information indicates
exiting the information flow to enter another type of flow
such as actions. We call this point a gateway in the flow.
7. Information is stored. Thus, it remains in a stable state
without change until it is brought back to the stream of flow
again.
8. Information is destroyed.
The first five states of information form the main stages of
the stream of flow, as illustrated in Fig. 2.
When information is stored, it is in a sub-stage because it
occurs at different stages: when information is created
(stored created information), processed (stored processed
information), or received (stored received/row information).
The five-stage scheme can be applied to individuals and to
organizations. It is reusable because a copy of it is assigned
to each agent. Suppose that a small organization has a
manager and a secretary. Additionally, it has two
departments with two employees in each department.
Therefore, it comprises nine information schemes, one for
the organization at large, one for the manager, one for the
secretary, one for department 1, and so forth.
The five information states are the only possible
“existence” patterns in the stream of information. To follow
the information as it moves along different paths, we can
start at any point in the stream. Suppose that information
enters the processing stage, where it is subjected to some
process. The following are ultimate possibilities for the
information:
17
1. It is stored.
2. It is destroyed.
3. It is disclosed and transferred to another sphere.
4. It is processed in such a way that it generates implied
information (e.g., a is the father of b and b is the father of c
generates the information that a is the grandfather of c).
5. It is processed in such a way that it generates new
information (e.g., comparing certain statistics generates the
information that Smith is a risk).
6. It is used to generate some action (e.g., upon decoding
or processing the information, the FBI sends its agents to
arrest the spy who wrote the encoded message).
In the uses substage, information is not a patient. The
patient is a term that refers to the thing that receives the
action.
The storage and uses/actions sub-stages (called
gateways) can be found in any of the five stages; however, in
the release and transfer stages, information is not usually
subject to these sub-stages, so we apply these sub-stages only
to the receiving, processing, and creation stages without loss
of generality. To illustrate the “gateway” sub-stage, consider
that a person in Barcelona uses the Internet to ask a person in
New York whether it is raining in New York. First, the query
flows to the receiving stage of the person in New York. The
New Yorker’s reception of the query triggers an action such
as physical movement to look outside to check whether it is
currently raining. The gateway in his/her information system
transforms the information flow to an action flow.
III. UML VS. FM
To illustrate the advantages of FM-based conceptual
modeling, we compare UML and FM description. The
discussion will include some important features of FM.
Fig. 3 shows a typical example of an UML graph [13].
Created
Communicated
Processed
Disclosed
Received
Figure 2. States of information.
Shop
Sell product
Customer
Customer Vender
Figure 3. Typical UML graph.
The figure includes two use cases: Sell product and Shop
Customer support. In UML, the two actors seem to be Customer Disclosure Vender
directly connected to the interior of Shop. They are
connected to the shop indirectly through having the two use Creation Creation
cases in the Shop. Conceptually, this in an ambiguous Creation
connection. Processing
Processing
The Shop is an active entity in the interaction. The usual Processing
Sell product
situation is that every thing that is received, processed,
created, released, and communicated by the two use cases is
first received, processed, created, released, and Receiving Cust. support Receiving
communicated by the Shop. The “Shop” is different from
“Sell products” and “Customer support.” It may be argued Disclosure Disclosure
Receiving
that “Sell products” is that part of the Shop that interacts
with the customers. Still, in the general case, the “global
sphere” of the Shop may or may not allow this direct Communica- Communication
Communica-
interaction. tion tion
The point here is that the semantics of the “lines”
connecting the Shop to Customers and Vendors is
ambiguous. Additionally, the kind of connection is not clear. Figure 4. FM description that corresponds to figure 3.
Apparently, the “Sell product–Customer support” connection
indicates flow of information (e.g., order, response to order),
and/or money. The vendor-“Sell product–Shop” connection Example: Imagine the following scenario:
may also denote these flows. There is apparent materials - The customer creates an order for a product.
(products) flow in customer-“Customer support” and vendor- - The “sell product” module in the shop receives the
““Customer support.” If we replace shop with “Market,” order and processes it.
then the semantics of the connections represent everything: - The “sell order” module sends the request to another
eggs, money, personnel, materials, etc. processing module, “customer support,” to deliver the
The actors in the UML description of Fig. 3 are modeled product to the customer.
as black boxes without interiors. The shop’s description - The “customer support” module sends a request to the
seems to be incomplete. Consider the following hypothetical vendor to deliver the product to the customer.
situations: - The vendor receives the request, creates the product,
1. A system that registers only information. For example, and delivers it to the customer.
a time clock hat reads information on ID card and registers Fig. 5 shows the resultant flows in the scenario.
the ID.
2. A system that reads information and processes it. For
Shop’s informational
example, a time clock that scans employees’ fingerprints
Customer’s 5 Vender’s
then searches in a fingerprint database to allow entrance. information Disclosure information
3. A system that reads data, processes it, then produces sphere sphere
new information. For example, a machine that reads health Creation
data of a person, compares it with tables to decide whether
the person is fit for a certain job or not. Processing
Processing
These three situations are represented in UML by an
actor, a use case. Nevertheless, each may necessitate Creation Sell product 7
different requirement that should be understood at this level 1 3 4
by all participants. For example, generating personal Receiving
Customer. support
information may be governed by privacy rules.
Fig. 4 shows an FM description that corresponds to Fig. Disclosure
3. In FM, Shop and the actors are spheres. Spheres can Receiving
receive, process, create, release, and communicate (transport) Communica- 2
these flowing things. “Sell product” and “Customer support” tion Communication 6
are processes in the Shop. The “flowing things” are
information, products, and money. Each of these flows is
represented by its own flow in the actual conceptual. Customer’s Receiving Vender’s physical
physical 9 sphere Manufacturing
However, Fig. 4 is drawn in a very general way to illustrate sphere
different directions of flow. The following example shows Transport Transportation Releasing
specific types of flows. 8

Figure 5. Example FM with two types of flow.

18
In Fig. 5, there are two types of flows: information flow
(orders and requests) and physical products flow. At circle 1
in the figure the customer creates an order. It flows from the
creation stage to the disclosure stage, then to the
communication stage. The order crosses to the shop’s
informational stage at circle 2. It flows to “sell order” in the
shop’s processing stage (circle 3).
The figure shows the remaining three spheres where the
creation stage is omitted because it is unused. The “sell
order” module sends to the “customer support” module to
initiate a request to deliver the product (circle 4). The
disclosure stage causes the creation of delivery request that is
to be released to the vendor (circle 5). The delivery request
flows to the vendor’s informational sphere (circle 6). Then, it
flows to its processing stage, which triggers (circle 7) actual
actions in the vendor’s physical sphere. Notice that the
dotted arrow indicates change in the type of flow, from
informational flow to actions flow. The triggered flow does
not necessarily start from the communication/transporting
stage. In this last sphere the product is created
(manufactured) and transported to the customer (circle 8).
Finally, the customer’s transporting action (e.g., going to the
post office) allows him/her to receive the product.
Such a scenario is understood by the customer, the shop
manager, analysts, and software developers. Even though the
FM model appears to be an “extensive” description in
comparison with the UML description, the FM
representation is simple because of its repeated application of
the five-stages schema and the uniform application of a
single flow in each sphere. We claim that the FM description
is a “better” (i.e., more complete, more uniform) conceptual
tool than UML. Our weaker claim is that FM can be
complemented with some UML structures such as procedural
control flow inside the modules “sell product” and “customer
support.”
IV. COMPARISONS OF SOME FEATURES
FM differs from UML in its basic structure and
components. Further research may lead to inject some UML
constructs into FM. This section compares some of their
features.
A. Actors and Spheres
In UML, an actor is an outsider who interacts with the
system. “Outside” here refers to controllability (the system
cannot control the outsider). However, interaction between
the system and actors embeds controllability of interaction
from either side. Controllability in UML refers to the interior
processes of an outsider that the system cannot control.
It is observed that “ad hoc mixing of abstract notation (2-
D ovals, boxes), stick figures and 3-D elements make UML
appear jarring. More effort could have been made to
construct a uniform and aesthetically pleasing
representation” (Reference.com, 2008). In FM there is no
necessity for differentiation between actors and the system.
Actors and the systems are simply spheres of information
that interact (hence communicate) with each other. All are
“outside,” but interact with each other. This is in line with
Occam's razor, where one should not make more
19
assumptions than the minimum needed. The system acts on
actors and actors act on the systems; thus from the functional
point of view they are all interacting entities. Roles (a term
used extensively in UML-related works) are different kinds
of interactions between two spheres. When two systems
interact with each others, then both are actors.
Distinguishing between the system and the actors is
suitable for computer developers. In computer science, there
are—most of the time—two sides: computer and
applications. This is not a central concern in FM. Secondly,
the notion of actors is introduced for lack of a common
notion that covers systems, human beings, and other
information spheres. The five stages schema in FM provides
such a general notion. UML mixes information spheres with
such notions as events. In FM, events are products of
spheres. No spheres means no events.
The only thing that distinguishes “the system” from other
actors is that the system is the central actor. Every actor
interacts with the system. Interactions among actors are
usually minimal and related to their interactions with the
system. FM is more general; it represents any type of
interaction topology. This is especially important for a
network environment. In FM the basic concept is sphere,
regardless whether spheres (e.g., actors) are in the interior or
exterior of any sphere (e.g., system). In UML we may also
introduce artificial actors such as those representing common
behavior shared by a number of different actors.
Conceptually, this makes the concept of actors very
confusing. Actors in this case are abstractions of something
common to outsiders. In FM, these are, again, spheres just
like any other spheres.
The FM approach is suitable for describing
generalizations (e.g., specialization, inheritance, or sub-
classing) as structural relationships among actors. In FM,
there is a uniform modeling of interiors and exteriors of
spheres. An information sphere (as an example of a sphere)
<includes> information spheres regardless whether they are a
system’s information spheres or not. Similarly, UML
associations are associations among spheres regardless
whether they are systems or not. This conceptual
generalization eliminates a great deal of UML technicalities
but also allows more expressive power, since we can now
model the interior flow (stages) of all assemblies.
Additionally, the flow of information may eliminate the
specification of certain kinds of associations among UML
actors. For example, "navigable association" that indicates
one actor controlling or "owning" the interactions can be
reflected in FM by the flow of information. Simply, a sphere
does not “turn on” until the flow reaches it (it receives
information). This happens when the controlling sphere
allows the flow of information to the controlled sphere. The
direction of arrows in FM has one meaning: flow of
information from one sphere to another. We will not discuss
these details, but the sweeping generalization of notions
accompanied by zooming in on interiors of spheres promises
abundant modeling benefits. Notice that FM allows
interaction between different spheres. In UML, these
interactions cannot be represented in the same diagram as the
system.
 It is important to notice that use cases are not (internal)
design tools, rather they are “communication,” meaning they
Administrator User specify requirements and features as they are commonly
understood by participants including users, managers, and
Server Browser developers.
In UML, Use Cases are supposed to be top-level service
Start/Stop Set Target that the system provides to its actors. Internal behavior
Browser URL within the system should not appear in the top system level.
Server The status of common Use Cases (those used by parts of the
Serve Page Serve Page system and actors) is not clear. In FM, it is easy to identify
Use Cases of outside spheres through identifying where they
cross the sphere’s boundary. Here we achieve simplification
Figure 6. The interactions between a user, a web browser, and in addition to preserving identification of Use Cases.
the server it contacts (From [6]).
V. UML WITH SEMANTICS
Instead of UML, this section discusses some proposals to
enrich UML with semantics. There are many extensions of
Example: Suppose we want to diagram the interactions UML in this direction; consequently, to achieve some depth
between a user, a Web browser, and the server it contacts. in making comparisons with FM methodology, we will
Fig. 6 shows the resultant description. concentrate on works that constrain UML constructs in order
UML does not allow drawing interaction lines between to align it with conceptual modeling.
the actors because it is not clear what the interactions mean. Evermann and Wand [9] proposed “to extend the use of
Thus, at the least we would draw two diagrams. Fig. 7 shows object-oriented design languages [such as UML] into
different FM information spheres and possible flow of conceptual modeling for information system analysis.” The
information among them. proposed semantics are used to derive modeling rules and
B. Use cases guidelines on use of object oriented languages for business
domain modeling. Fig. 8 shows an example of UML class
Use Case Diagrams are descriptions of the relationships diagram (from Evermann and Wand [9]). It depicts a
between a group of Use Cases and participants (e.g., actors situation typically found in object-oriented models.
who interact with the system) in the process. The Diagrams Clearly, the description is not suitable for conceptual
tell what the system should specify and facilitate requirement models. It is loaded with technical consideration. More
specifications and communication among users, customer, importantly, it employs heterogeneous notions. Consider the
and developers. Use Cases can <<include>> Use Cases. edge between Customer and order. It is supposed to stand for
a relationship between them. What type of relationship can
exist between a customer (a person or corporation) and
Browser “order”? Apparently, “order” means a request to purchase
Processing something. Ontologically, there is nothing in the “real world
Processing domain” called “order.” It is an act of some agent. Ordering,
requesting, objecting, demanding, expressing, signaling, etc.
Start/Stop Set Target Receive have a similar ontological nature: acts of an agent.
URL Page
Customer
Serve Page Order 1 Name
dateReceived Address
Server is Prepaid
creditRating():string
number:String
Receiving price:Money
Disclosure Disclosure
Personal
Communication Coperate Customer customer
Line items 1 ContactName
Communication Receiving CreditCard#
creditRating
Order Line creditLimit Customer
Quantity:integer Remind()
Price:money billForMonth(Integer) Sales rep
Communication Communication IsSatistied:boolean
1 Product 0..1 Employee
Disclosure Receiving
Disclosure Receiving
Processing Figure 8. Example UML class diagram without ontological
Processing semantics (from [11]; used by [9])
Creation User Administrato
Creation

Figure 7. Interactions among several infospheres

20
 What is the relationship between an agent and its acts?
The answer is “acts flow out of agents,” in the sense that
agents create, process, receive, transfer, and communicate
acts. So is having the order stand side by side with its actor
(e.g., creator, processor) conceptually disturbing? The best
we can do is to explain the edge between Customer and
Order as actor/actee correspondence. “Order” is a type of
“things that flow.”
As mentioned previously, in UML, an actor is an outsider
who interacts with the system.
In FM, orders and products are “things that flow,” as shown
in Fig. 9. A property is defined in FM as something that
flows in entities. Notice that each entity in FM has multiple
spheres, just as a human being has several “systems”:
digestive system, musculoskeletal system, nervous system,
etc. Food does not flow in the nervous system, and obtaining
oxygen is not the business of the digestive system. Similarly,
a business information sphere—here, we do not mean
computer information system denoted previously as IS—is
different from its physical sphere, which is different from its
money sphere, etc.
In Fig. 9, the arrows (except in the product sphere)
represent the flow of orders. The black boxes represent the
customers. According to our understanding of the case, the
corporate customer’s sphere contains the employee’s sphere.
Hence, orders are created in the corporate sphere either by
the corporation itself or by some of its employees. Orders are
also created for individual customers. The flow from these
sources of orders are received by the company (circles 1 and
2). We assume that the company has two units: one to
control the queue of orders (the Order Line box) and one to
deliver products (the Product box).
The company divert the flow to the Order Line sphere
(circle 3), which in turn triggers (circle 4) the Product sphere
to release the product (circle 5). The FM description can
include a great deal more details; however, we tried to stick
to what is expressed by the UML model. The dotted arrow in
the figure denotes a change in the type of flow from “flow of
orders” to “flow of products.” To simplify the figure, the
unused internal arrows have been removed.

1 Company 2 Receiving
Receiving
Transporting
Processing
Transporting

Transporting
Processing Receiving
Releasing
Releasing
Releasing
Creation
Processing
Creation Personal customer
Creation
Corporate customer
3

Receiving Transporting 4 Creation

Employee Releasing
Transporting

Receiving
Processing Receiving
Releasing
Order Releasing
Releasing Line
Processing Processing

Creation Creation Transporting Product

5

Figure 9. FM description of the system in Figure 8.

21
 International Conference on Conceptual Modeling,
We claim that the FM model as described in Fig. 9 is a Yokohama, Japan, Nov. 27-30, 2001.
conceptualization that is suitable for both systems analysts [11] Fowler M., and Kendall, S. (2000). UML distilled: A brief
guide to the standard object-oriented modelling language.
and software developers. The flow notion is familiar in Addison-Wesley, Reading, MA
analysis, where many flow-based models are used, such as [12] Opdahl, A., and Henderson-Sellers, B. (2002). Ontological
the circular flow model used by economists and the supply evaluation of the UML using the Bunge-Wand-Weber model.
chains used in operation research. It is also suitable for Software and Systems Modeling, 1(1), 43-67.
software developers, where the control flow is a basic [13] Reference.com (accessed March, 2008), Unified Modeling
concept in computer science. Language,
http://www.reference.com/browse/wiki/Unified_Modeling_La
VI. CONCLUSION nguage
This paper has examined the problem of building a
conceptual model of “real world domain” as part of the
process of developing a business information system. For
this purpose object-oriented techniques are utilized, typically
based on UML as an object-oriented modeling language. In
addition, extensions of UML are proposed for conceptual
modeling. While such efforts are commendable, this should
not hinder the researchers from seeking fundamentally
different approaches. The flow model introduced in this
paper is drastically different in nature from UML. However,
there is a possibility that it can complement and be
complemented by UML in developing new methodologies
for conceptual modeling.

REFERENCES

[1] Al-Fedaghi, S. “Software Engineering Interpretation of

Information Processing Regulations”, IEEE 32nd Annual
International Computer Software and Applications
Conference (IEEE COMPSAC 2008), Turku, Finland, July
28 - August 1, 2008.
[2] Al-Fedaghi, S. “Some aspects of personal information
theory,” 7th Annual IEEE Information Assurance Workshop,
United States Military Academy, West Point, NY, 2006.
[3] Ambler S. W. (2007). Be Realistic About the UML: It's
Simply Not Sufficient
http://www.agilemodeling.com/essays/realisticUML.htm
[4] Bell, A. E. (2004.). Death by UML Fever, Communication of
the ACM, Vol. 2, No. 1 – March.
[5] Bunge, M. A. (1977) Ontology I: the furniture of the world,
Vol. 3. D. Reidel Publishing Company, Dordrecht, Holland.
[6] Carnegie Mellon University (accessed March 2008), UML
Use Case Diagrams: Tips and FAQ, accessed March 2008.
http://www.andrew.cmu.edu/course/90-754/umlucdfaq.html
[7] Dussart, A., Aubert, B. A., and Patry, M. (2002). An
evaluation of inter-organizational workow modeling
formalisms. Working paper, Ecole des Haute Etudes
Commercials Montreal.
[8] Evermann, J. (2005). Thinking Ontologically {Conceptual
versus Design Models in UML, In: Rosemann, M. and Green,
P. (eds.) Ontologies and Business Analysis. Idea Group
Publishing, 82-104.
[9] Evermann, J., and Wand, Y. (2005). Ontology based object-
oriented domain modelling: fundamental concepts,
Requirements Eng 10: 146–160.
http://www.mcs.vuw.ac.nz/~jevermann/EvermannWandRE05
.pdf
[10] Evermann, J. and Wand, Y. (2001). Towards ontologically
based semantics for UML constructs. In H. Kunii, S. Jajodia,
and A. Solvberg, editors, Proceedings of the 20th

22
 2009 International Conference on Computer Engineering and Technology

A Method of Query over Encrypted Data in Database

Lianzhong Liu and Jingfen Gai

Key Laboratory of Beijing Network Technology, School of Computer Science and Engineering
Beijing University of Aeronautics and Astronautics
Beijing, 100083 China
lz_liu@buaa.edu.cn, gaijingfen0313@163.com

Abstract—The encryption mechanism is an effective way
to protect the sensitive data in database from various
attacks. When the data is encrypted, the query
performance degrades greatly. How to query encrypted
data efficiently becomes a challenge. In this paper, a
scheme to support query over encrypted data is proposed.
Firstly, we extend two-phase framework to complete
query, constructing different indexes for different data
types. A new method to construct Bucket index for
numeric data is proposed and bloom filter compression
algorithm is used for the character string, which will be
saved in database as a numeric data. We analyzed the
false positive probability of bloom filter to get optimal
parameters. Experiment results show the performance of
our scheme has improved.
Keywords-query; encrypted data; Database; security
I. INTRODUCTION
Database is the most important part of information
System, enterprise databases hosts much important data,
while they are threatened by inside and outside attacks.
Database encryption, as an active protect mechanism, is
an effective way to protect data integrality and privacy.
When data is encrypted, how to query encrypted
data becomes important and challenging. Many
researches adopt two-phase query method, avoiding
full table decryption; in which constructing index for
encrypted data is the key to improve performance. We
construct different indexes to support different
computations for different data types in two phase
query, the first of which returns coarse result by using
index, and the other of which decrypts the encrypted
result and does a refined query.
We perform range query of numeric by constructing
a bucket index, and complete fuzzy query of characters
effectively by using the bloom filter as the encrypted
index. Firstly, we use triple to express a character
string as a set, build bloom filter on it according to the
real situation, then convert the bloom filter into a
numeric data, since a character string’s match is usually
not as quick as the match of a numeric data [2] . We
analyze the relation of the parameters that affect the
false positive probability. The experimental results
shows that the performance of the queries in our
scheme has been improved compared with the pairs
coding method and the traditional method.
II. RELATED WORK
H. Hacigumus et al. introduced DAS model [3] , in
which database is provided as a service to the client.
Then in [4] they proposed two-phase query, in which
the query is completed on the client side and the server
side together, and they have proposed bucket index,
which support the range query for the numeric data.
And they combined privacy homomorphism technique
to support arithmetic computation [5] . Bijit Hore
further optimized the bucket index method on how to
partition the bucket to get the trade between the
security and query performance [6] . The methods
based on index is supported by DBMS, and focused on
the query performance at the cost of storage space.
There are also some researches on the fuzzy query
of character string. Zhengfei Wang proposed pairs
coding function to support fuzzy query over the
encrypted character data [7] [8] . Pairing coding
method encodes every adjacent two characters in
sequence and converted original string directly to
another characteristic string by a hash function. This
method can’t deal with singe character, and could
perform badly for big character string. Paper [9] had
proposed characteristics matrix to express string and
the matrix will also be compressed into a binary string
as index. Every character string need a matrix size of
259x256, it is large and will lead to much computation;
in addition, the length of index has come to more than
hundred bits, which is not suitable for storage in
database.
III. THE ARCHITECTURE OF ENCRYPTED DATABASE
A. Encrypt storage model
R( A1 ,..., Am ,..., An )
We use to represent a database
relation model, and we classify the data into sensitive
and not-sensitive. Our encryption scheme base on
column-level, the column-level encryption minimizes
the number of bytes encrypted, with flexible choice of
encrypted fields and good efficiency especially for only
a few fields sensitive. A compromise solution between

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 23

DOI 10.1109/ICCET.2009.173
performance and security can be achieved by only
encrypting the sensitive fields [1] .
In order to fasten the query of encrypted data, we
got encrypted column and index column for every
sensitive column data. We build different types of
index to support different computations. Encrypted
column is used for equation query, which is another
virtue of column level encryption. We build bucket
index to support range query, mainly for numeric data
and date data. And we construct index using bloom
filter for character data to support fuzzy query.
Corresponding encryption relation can be expressed as
E
( E
R S A1 ,..., Am ,..., An , A1S ,... ArS ,... AmC (1)
where the column with upper E stands for the
encryption column, the column with upper S stands for
the bucket index column, and C stands for the index
column for character string. They are expressed as
follows:
AiE = Encryt ( Ai ), AiS = Sindex( Ai ), AiC = Cindex( Ai ) (2)
Where we assume Encrypt() is the encrypt function,
Sindex() is the bucketIndex() function and Cindex() is
the function generating the character index.
B. The process of encrypted data
Our scheme extends two-phase query framework [4]
as shown in Fig.1, an Encryption/Decryption Engine
layer is added between application and DBMS. Every
time an application puts forward a query, the
application interface on its own side accepts it and
transfers it to the E/D Engine. The application interface
is the only thing we provide to applications like setting
a client proxy, in contrast, we have an application
listener for listening request come from every
application interface on the E/D engine side like setting
a server proxy. The communication between
application and the E/D engine depends on interact of
Store Module
(with encryption)
Insert
Security
Application
interface
Dictionary
Coarse Querying on
Select
Application/
Web Browser
Querying Translator
Querying
Result
Querying
condition
Querying Result Filter
（ With decryption ）
Temp Query Result
Encryption/Decryption Engine
Figure 1. Architecture of storage and query of encrypted data
the two proxies. Store Module or Querying Translator
deal with it according to SQL type, using metadata in
Security Dictionary to translate user SQL.
The Store Module will turn the insert SQL into a
corresponding one with data encrypted. The query
Translator will split an original query over unencrypted
relations into: (1) a corresponding query over encrypted
relations to run on DBMS, (2) a client-query for post-
processing results of the server-query, which will be
sent to QRF (Querying Result Filter) as a querying
condition. QRF takes the encrypted result as input, and
then computes every record by the query condition,
) decryption function will be called when encrypted data
involved, output of process is exact query result set,
and then QRF sends the result to application through
the secured communication. In fact, the essential of two
phase query is generation and use of index.
IV. CONSTRUCT BUCKET INDEX FOR NUMERIC DATA
The construction of index should follow two
principles, firstly the index filters false record
efficiently, and secondly it should be safe enough not
to leak the true value. Moreover there are many data
types in DBMS, and there will be many types of
queries. For example, character data need equation and
fuzzy query, numeric data needs equation and range
query, while date data needs query with “between…and”. It
is not possible an index supporting all computations, so
we built different types of indexes according to the data
type.
Different from the bucket index in DAS [4] , we
construct order-preserving bucket index for numeric
data by a series computations, without storing and
accessing the metadata about partition information,
which also enhance the data security. The computations
are as follows:
Step 1: we divide every data with a Seed, in order
to get the quotient and residue, and we got the digit
To Store Encrypted data number of a numeric data. So we have IB as the initial
bucket number.
v−r
IB = digitNum(v) || (3)
seed
Encrypted data Step 2: In order to enhance the security of the
bucket number, we have the IB blurred. We take a
DBMS
monotone increasing function (F(x)) to transform IB, in
this way we have final bucket index, and the true value
Encrypted
data
Plain
text
is obscured and safe on the condition that F(x) is secure.
It is proved easily that a larger v will have a larger
IB, also a larger F(IB), this characteristic will support
range query for numeric data, see details in [4] . Seed
and F(x) are the only ones we need to protect, which
also decides the query efficiency.
24
 V. CONSTRUCT INDEX FOR CHARACTER STRING
Large quantities of string data are used in many
applications and fuzzy query is used frequently. How to
execute a query efficiently over the encrypted string
data are the emphases of this paper.
A. Triple for character string
In this section, we construct a triple to present a
character string, which can express the characters in the
string and the relations of the characters. It is defined
as follows:
Definition 1 Triple for a character string: For a
string s=’ c1 c2 ….. cn’, we have w presenting the set of
word split by blank, comma and full stop characters. u
presents the set of characters in s. r is used to describe
the relationship of adjoining characters. So the triple is
comprised of these three, At last we have t = w, u , r .
B. Bloom filter on the triple
A Bloom filter is a simple randomized data
structure that answers membership query with no false
negative and a small false positive probability [11] , It is
space-efficient, using m-bits array for representing a set
S={s1,s2,…,sn}. Initially, all bits in the bit array are set
to 0. A bloom filter uses k independent random hash
functions h1,…,hk with range {0,…,m-1}. For each
element s ∈ S , the bits hi(s) are set to 1 for 1 ≤ i ≤ k . To
check if an element x is in S, is to check whether all
hi(x) are set to 1, if not, then clearly x is not a member
of S, if all hi(x) are set to 1, we think that x is in S with
false positive probability [12] .
String s= ”a1a2…an”
String[] w={w1 …wn}
String[] r={r1,r2 …rn} char[] u={a1,a2 …ar}
s1 s2 si Sn
Si
.
h1 h2 hk
m-bits array
0 m
Figure 2. The bloom filter built on triple
We build bloom filter index on the triple according
to the real situation, which is shown in Fig 2. Every
element in the u and r will be encoded into multiple
positions of the binary string with the tag of ‘1’, then
25
the character string will be mapped to a m-bits array
comprised of ‘0’ or ‘1’, which will then be stored in the
database as index in the form of numeric. In this way, a
fuzzy query over character data equivalent is a
membership query over the bloom filter only by simple
& operation of numeric data, not full-table decryption
any more.
C. Analysis of the false positive probability for bloom
filter
The key of our method is to minimize the false
records returned, which depends on false positive
probability of the bloom filter, while the false positive
probability depends on multiple factors. We need to
analyze these relations. The probability of a false
positive for an element is
k
⎛ ⎛ 1⎞ ⎞
kn k
f= (1 − p ) = ⎜1 − ⎜1 − ⎟ ⎟ ≈ ⎛⎜1 − e m ⎞⎟
k − kn
(4)
⎜ ⎝ m⎠ ⎟ ⎝ ⎠
⎝ ⎠
where a large m and a small n both have a large f, and f
( )
is smallest when k = (ln 2) ⋅ m [12] .
n
When a query condition is “LIKE %matchStr%”,
the probability of a not-matched record returned is
f NUMBER( matchStr ) , where NUMBER(matchStr) stands for
the number of elements that matchStr has. So we have
the larger matchStr is, the smaller of a false record
returned probability gets.
In practice, k must be an integer, and smaller k
might be preferred since they reduce the amount of
computation necessary. When k=1, we will see that it
becomes pairs coding function, we can say pairs coding
is a special case of bloom filter. By the conclusion
above, pairs coding function has
⎛ 1⎞
n
−n
f = 1 − p = 1 − ⎜1 − ⎟ ≈ 1 − e m , k = 1 (5)
⎝ m⎠
( )
The more (ln 2) ⋅ m is close to 1, the better of the
n
false positive probability is, where n is the number of
elements, and m is the total bits size of the index. So if
n is large, a larger m will be needed, or else the false
positive probability will be high.
D. Algorithm of index based on bloom filter
The bloom filter algorithm we used is as described
in Fig. 3 where we use MD5 to get k hash functions,
and then convert all the hash positions to numeric as
index.
Now we can convert the LIKE query over encrypted
relation to the query over the index attribute. Assume
the condition for a query is “WHERE Ai like
‘%matchStr%”, we first get the index value of
‘matchStr’, then the condition changed to WHERE AiS
&value=value. Take ORDERS relation as an example,
we make index for the sensitive column is o_comment,
 Fig. 4 is the query over ORDERS, and the sensitive
column is O_COMMENT. We compare the method of
pairs coding and bloom filter when k=1, which shows
the improvement of our method. Fig. 5 is query about
PART, and the sensitive column is P_CONTAINER.
Also we compare our method to pairs coding method
extended by different k and traditional method, the
result shows the improvement of the bloom filter based
index is obvious.
k=1,m=31 bloom filter
1.2
pairs coding
fullTable Decryption
y 1
t
i
l
i
b
a 0.8
b
o
r
p 0.6
y
r
e
u
q 0.4
e
s
l
a
f 0.2

0
1 2 3 4
order.comment like %String%

Figure 4. Comparison of bloom filter, pairs coding and

Figure 3. The bloom filter algorithm traditional methods

let len=2,m=31,k=4. When a query is ‘o_comment like bloom filter,k=2

‘%unusual%’’, we get the index value is 480497654 comparsion between extended pairs coding,k=3
with the same parameters. Then a converted query different methods extended pairs coding,k=2
condition is pairs coding,k=1
1
fulltable decrypt
bitand(e_comment, 480497654)= 480497654 (6) 0.9
y
t 0.8
i
VI. THE EXPERIMENT AND THE ANALYSIS l
i 0.7
b
a
b 0.6
We leave the efficiency of bucket index out of o
r
p 0.5
account of this paper. Purpose of our experiments is to y
r
e 0.4
test the performance of fuzzy query among our method, u
q 0.3
pairs coding and the traditional method. According to e
s 0.2
l
a
TPC-H standard, we use the tool of Benchmark Factory f 0.1
for Database to make our database, we take scale=1, 0
we use table ORDERS and PART as experimental data 1 2part.container
3 like4 %string%5 6

source. The encryption algorithm is DES, length of the
Key is 128; programming language is Java; the
environments are windows XP, P4 2.66 CPU, 1G RAM
and database is Oracle 10g.
We have the following definition of false query
：
probability:
Definition 2 We assume the number of tuples
returned in the first phase of the query is n1, and the
number of tuples satisfying the original query is
n2, n1 ≥ n2 , then the false query probability is:
n1 − n 2
fq = (7)
n1
By the way fq is different from f, f is the probability
of an element not in the set being taken as a member,
and fq is the proportion of get the wrong records, while
fq can reflect f. We get fq by executing the fuzzy query
over encrypted relation, and got n1, n2 recorded.
Figure 5. Comparison of different index methods
VII. CONCLUSION
We proposed a query scheme to query encrypted
data efficiently by using different types of index for
different data, and then converted the query over the
sensitive attribute into the query over the index
attribute in two phase query. We construct a bucket
index without saving the metadata of every partition,
which enhance the security of the sensitive data. We
focus on the fuzzy query over encrypted character
string. Firstly, we use a triple to denote a string, using
bloom filter to check the existence of elements of the
Triple. Thirdly, we analyzed the parameters of bloom
filter to get the minimal false positive probability.
Finally, we evaluated the performance and efficiency of
our two phase query. Compared with conventional

26
queries and the pairs coding method, the experimental
results show the performance improved.
ACKNOWLEDGMENT
This work is partly supported by the National
High-Tech Research and Development Plan of China
under grant No. 2005AA113040 and the Co-Funding
Project of Beijing Municipal Education Commission
under Grant No. JD100060630.
REFERENCES
[1] S. Sesay, Z. Yang, J. Chen and D. Xu. “A secure Database
Encryption Scheme”. Consumer Communications and
Networking Conference (CCNC), 2005, pp. 49-53.
[2] Y. Zhang, W. Li and X. Niu. “A Method of Bucket Index over
Encrypted Character Data in Database”. Intelligent Information
Hiding and Multimedia Signal Processing, 2007, pp. 186-189.
[3] H. Hacigumus, B. Iyer and S. Mehrotra. “Providing Database
as a Service”. In the Proceedings of the International
Conference on Data Engineering (ICDE), 2002, pp. 29-38.
[4] H. Hacigumus, B. Iyer, C. Li and S. Mehrotra, “Executing SQL
over encrypted data in the database service provider model,” In
ACM SIGMOD Conference, 2002, pp. 216-227.
27
[5] H. Hacigumus, B. Iyer, and S. Mehrotra. “Efficient execution
of aggregation queries over encrypted relational databases”. In
the proceedings of Database Systems for Advanced
Applications (DASFAA), 2004, pp. 125-136.
[6] B. Hore, S. Mehrotra and G. Tsudik. “A Privacy-Preserving
Index for Range Queries”. In Proceedings of the 30th VLDB
Conference, 2004, pp. 720–731.
[7] Z. Wang, J. Dai, W. Wang and B.L. Shi. “Fast Query over
Encrypted Character Data in Database”. Communications In
Information and Systems, 2004, pp.289-300.
[8] Z.F. Wang, W. Wang and B.L. Shi. “Storage and Query over
Encrypted Character and Numerical Data in Database”. In
Proceedings of the 2005 The Fifth International Conference on
Computer and Information Technology, 2005, pp. 591-595.
[9] H. Zhu, J. Cheng and R. Jin. “Execution Query over Encrypted
Character Strings in Databases,” Frontier of Computer Science
and Technology, 2007, pp. 90-97.
[10] Y. Ohtaki. “Partial Disclosure of Searchable Encrypted Data
with Support for Boolean Queries”, Availability, Reliability
and Security (ARES08), 2008, pp. 1083-1090.
[11] J. Bruck, J. Gao and A. Jiang. “Weighted Bloom Filter”.
Information Theory, pp. 2304-2308, 2006.
[12] M. Mitzenmacher. “Compressed Bloom Filter”. IEEE/ ACM
Transactions on Networking, vol. 10, no. 5, 2002, pp. 604-612.
 2009 International Conference on Computer Engineering and Technology
Traversing Model Design Based on Strong-association Rule for Web Application
Vulnerability Detection
QI Zhenyu1, XU Jing1, GONG Dawei1, TIAN He1
Institute of machine intelligence, Nankai University
TianJin, China, 300071
qi_zhenyu@hotmail.com; xujing@nankai.edu.cn; flyingdonkeycn@msn.com; fyonatian@hotmail.com
Abstract—With more important function in information
society, software dependability has been in higher demand.
Web application vulnerability has become one of the biggest
threats for software security. Detecting and solving
vulnerability is the effective way to enhance software
dependability. Most active method traverses all web links
and interactive units in traversing step, which is easy to
cause low efficiency and no pertinence. This paper focuses
on characteristic of web application, especially web pages
and presents one traversing model based on high-related
rule, which make detection more efficient. From hacker’s
view, this model applies the HITS algorithm to generating a
series of pages which may be used by hackers as attacking.
Due to ignoring content in web pages for HITS algorithm,
we adapt improved Apriori algorithm to get optimized
frequency set, on basis of which we deduce high-related rule
between properties of interactive unit and way of attacking.
Finally, this paper presents the traversing model for web
application vulnerability detection.
Keywords-software dependability; Web Vulnerability; HITS
algorithm; Apriori algorithm
I. INTRODUCTION
Web application has been applied into many kinds of
field. More popular web application becomes, more
vulnerability can be used as attacking. But the speed of
vulnerability detection development is slower obviously.
Therefore, active detection method simulating hacker
becomes the development trend. Can we find out attack
point from so much information as quickly as hacker in
the web pages? The answer is yes. Traversing problem in
vulnerability detection is presented on basis of this
consideration.
Web application not only consists of web pages, but
also hyper links、active links、active interactive units,
which contain lots of logical relations. These relations is
the right thing hackers look for, because logical relation
means that there could be data stream between pages.
Web pages which are browsed and interacted most
frequently are the pages which interest hackers most.
HITS is the classical algorithm based on hyperlinks. It
distinguishes the importance of links according to weight.
But pagerank based on weight has nothing to do with
initial parameter selection [6], which will be
demonstrated in section 2. Therefore, these pages getting
high weight by HITS algorithm are the place easy to
generate vulnerability. They provide basis for effective
detection but bring unavoidable cursoriness.
Although the result of HITS is good, there are still
some problems. Because this algorithm ignores the
relation between content of web pages and way of
attaching. Therefore, this model adapt association rule
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.79
algorithm to get some high-related rule by analyzing
transaction data stored in database. In section 3 we will
discuss association rule.
Finally, this paper presents one traversing model,
which provides strong support for simulating attack
model and then give some method to traverse web page
aiming to detect web application vulnerability.
II. OVERVIEW OF HITS ALGORITHM
When one page links to another page, it means that
there exists some logical relation between them. Some
links from different pages but to one page could display
the importance of this page. So lots of link information
can be used as an important resource to find content
correlation.
HITS algorithm mines the link structure of the Web
and discovers the thematically related Web communities
[1, 2] that consist of ‘authorities’ and ‘hubs’. Authorities
are the central Web pages in the context of particular
query topics. For a wide range of topics, the strongest
authorities consciously do not link to one another. Thus,
they can only be connected by an intermediate layer of
relatively anonymous hub pages, which link in a
correlated way to a thematically related set of authorities.
These two types of Web pages are extracted by iteration
that consists of following two operations.
∑y ∑x
xp = q ， yp = q (1)
q→ p p →q
For a page p, the weight of Xp is updated to be the sum
of Yp over all pages q that link to p: where the notation
q→p indicates that q links to p. In a strictly dual fashion,
the weight of Yp is updated to be to the sum of Xp.
Therefore, authorities and hubs exhibit what could be
called mutually reinforcing relationships: a good hub
points to many good authorities, and a good authority is
pointed to by many good hubs?
HITS algorithm gives every page two weight:
authorities weight and hub weight. Finally it outputs a
series of pages with highest weight. Kleinberg [3]
demonstrates that this algorithm convergence. Other
papers [4], [5] also have description of this algorithm.
Firstly, HITS gets start set which consists of pages
through initial URL and then generate base set on basis of
it. The base set includes pages pointed by start set and
others pointing to start set. The rest of algorithm mainly
works in the base set. We define Xp and Yp as
non-negative integers for every page in the base set. If the
value of Xp is big, the page will be considered a good
authority; if the value of Yp is big, the page will be
considered a good hub.
We can mark pages by integers like {1,2,…,n} and
define their n×n Adjacency Matrix. If Pi points to Pj, then
28
the value in matrix of (i, j) is 1, else 0. Similarly, we
define authority weight and hub weight as x=(x1, x2, …,
xn), y=(y1, y2, …, yn), then the matrix form of formula (1)
is:
x ← AT y , y ← Ax (2)
Stretching out formula (2) we can get:
x ← AT y ← AT Ax ← ( AT A ) x ,
y ← Ax ← AAT y ← ( AAT ) y
Therefore, vector x, y can be deduced by formula (3).
According to Linear Algebra theory, Iterative sequences
will convergence at Eigenvector by standardization. In
other words, hub weight and authority weight are intrinsic
property of page. So it has nothing to do with initial
vector and selected parameter [6].
III. ASSOCIATION RULE EXTRACTION IN WEB
APPLICATION
In client side web page is actually HTML document,
which can be got and read by user easily. When we
collect all HTML documents of one web application
displaying in the client side, all of them constitute
document database. HTML document is actually general
text plus special tag. These tags are keywords for
analyzing HTML document. We can find out some
association rules in web pages. This type of analysis
collects keywords displaying together and then find their
relation.
Admittedly, keywords in HTML should be collected;
some keywords in webpage document also should be paid
attention such as username, password, search, upload,
download etc. Because there is association between these
text and interactive form corresponding to some
keywords. Since main source where hackers get
information is information in webpage, it is important to
find relationship between web application page and
hacker attack.
Association analysis deals with etyma and then
removes useless words. After pre-processing, we could
find a series of keywords. Finally, association rule [7-9]
could be used as extracting high-related rule. In database,
every document could be viewed as a transaction, and
keywords in document could be viewed as items. So
database could be displayed as follows:
{document_id, set_of_keywords}
A. Association Rule
Association rules mining can he stated as follows: Let
I = (i1 , i2 ,..., im ) be a set of items.
Let D = (T1 , T2 ,..., T j ,...., Tn ) , the task-relevant data, be
a set of transactions in a database, where each transaction
T j (j=1，2，⋯ ，n) such that T j ⊆ I . Each transaction
is assigned an identifier, called TID. Let A he a set of
items, a transaction T is said to contain A if and only
if A ⊆ I . An association rule is an implication of the
form A → B where A ⊆ I ， B ⊆ I and
A ∩ B = ∅ . The rule A → B holds in the transaction
set D with support s, where s is the percentage of
transactions in D that contain A B (i.e., both A and B).
This is taken to be the probability P(A B). The rule has
confidence c in the transaction set D if c is the percentage
of transactions in D containing A that also contain B. This
is taken to be the conditional probability, P (B|A). That is,
confidence(A->B)= P (B|A)=support(A B) ／
support(A)=c , support(A->B)=P(A B)=s.
The popular association rules Mining is to mine strong
association rules that satisfy the user-specified both
minimum support threshold and confidence threshold.
(3) That is, minconfidence and minsupport.
If support(X)≥minsupport, X is frequent itemsets.
Frequent k-itemsets is always marked as LK. If
support(A->B)≥minsupport and
confidence(A->B)≥minconfidence ， A->B is strong
correlation. Several Theorems are introduced as follows:
(1)If A⊆B，support(A)≥support(B).
(2)If A⊆B and A is non-frequent itemset，then B is
non-frequent itemset.
(3) If A⊆B and B is frequent itemset，then A is
frequent itemset.
B. Apriori algorithm
Apriori algorithm is an influential algorithm for mining
frequent itemsets for Boolean association rules, employs
an iterative approach known as a Level-wise search,
where k-itemsets are used to explore (k+l)-itemsets. First
the set of frequent 1-itemsets is found. This set is denoted
L1,then L1 is used to find L2, the set of frequent
2-itemsets, which is used to find L3, and so on, until no
more frequent k-itemsets can be found. The finding of
each Lk requires one full scan of the database. The
problem of mining association rules can be decomposed
into two major steps: 1) Find out all frequent itemsets; 2)
Use the frequent itemsets to generate the strong rules.
Once all frequent itemsets from transactions in a database
D have been found, it is straightforward to generate
strong association rules from them, where strong
association rules satisfy both minimum support and
minimum confidence. In the meantime, these association
rules should satisfy the minimum weighted
interestingness.
C. Improved Apriori algorithm API
Based on the Apriori algorithm, an improved algorithm
for association rule mining named APL is proposed.
Based on a strategy of transaction database fuzzy pruning,
APL reduces the size of the database and improves the
efficiency of the algorithm. It doesn't like the Apriori
which fixes a new candidate after each complete database
scanning. It can add new candidate at any start point. It
dynamically values the support of all the counted itemsets.
If all the subsets of an itemset are frequent, the itemsets
will be as a new candidate. Based on the definition of the
frequent itemset, if itemset I is not frequent, namely P (I)
< minsupport, if we add item A into I, the result itemsets
(I U A) can not be more frequent than I, so I U A is not
frequent.
Theorem 1: Any item set in Lk must be the super set of
a certain item set in Lk-1.
Theorem 2: If a transaction does not contain any item
set in Lk-1, then deleting this transaction will not affect
the calculation of Lj (j > k)
The conclusion of Theorem 1 and Theorem 2 is that a
29
transaction contains no item set in Lk-1, that is to say
when it is 0, the transaction can be deleted from the
database. Regarding the "0" here, can it be further
changed? If the 0 can be replaced by a bigger number,
then it will be faster to prune the database obviously.
Based on the thinking above, we extend theorem 1 and
theorem 2, and put forward more theorems below:
Theorem 3: Any item set in Lk must be the super set of
k certain item sets in Lk-1.
Theorem 4: If a transaction contains less than k item
sets in Lk-1, then deleting this transaction will not affect
the calculation of Lj (j > k)
Theorem 3 and 4 guaranteed this conclusion: during
every time calculating the support of Lk, if a transaction
contains less than k+1 item sets in Lk, mark this
transaction a delete and consider no more in the scans
later. Compared with theorem 1 and 2, evidently theorem
3 and 4 can delete more invalid transactions.
The improved algorithm, APL deletes more invalid
transactions, further reduces the records which will be
scanned next time, and improves the efficiency of the
algorithm.
D. Strong association in Web Application Vulnerability
Detection
Algorithm: Use the mediate results of database
scanning last time to prune the database. It deletes more
transactions, prunes the database with larger scale, and
further reduces the time needed by the next scan.
Input: Transaction database D, minimum support
threshold minsupport
Output: All the frequent item sets in the database D.
L1= {large 1-itemsets}
return L1
for(k=2; Lk-1≠null; k++){
Ck=Apriori-gen (Lk-1);
//the item set that new generated
Forall transations t D {
If (Ldelete=0) {
Lt=subset (Lk, t); //the candidate item set
contained in t
Forall candidates l Lt
l.count++;
If (|Lt |<k+l) {
t.delete=1; //make a delete mark}}}
Lk={l k| l.count > minsupport }
return L = kLk; //return all the frequent item sets
According to the output of API, it is easy to find out
that hackers attack web application vulnerability when
interactive. We can deduce strong relation between
interactive properties and attack type, which provides
support for traversing on purpose. So this paper presents
one traversing model more efficient from hacker’s view.
IV. TRAVERSING MODEL BASED ON STRONG
ASSOCIATION RULE
From hackers’ view, it is their final purpose to find
weakness in web application. They will look for useful
information continuously and then do attack on purpose
by this information.
As for web application, hackers can not browse source
code directly. They do interaction by browser as general
users. So they get information just by analyzing HTML
30
document. It is fundamental for detection to collect more
information that hackers are interested in. As figure1 this
paper designs one traversing model based on strong
association.
HITS algorithm outputs a series of high-weight pages
by inputting initial URL. Through response from server,
these URL requests get pages in the form of document.
Each document is stored in database, which provides data
for analysis. By association rule algorithm we can get a
frequent itemsets. But its performance is not good enough.
So we combine APL and feedback from database to
enhance performance. Through several cycle of
optimizing, it is quicker and better. Finally, this strong
association will be used to direct vulnerability design in
simulating attack.
V. DATA IN WEBPAGE
In HTML document there are links, filename, input
field、hidden field、selection field of form. We locate
these data mainly by HTML tags. For example, find
<href> to locate links and find <form> to locate several
kinds of field in forms. In all kinds of data, the most
important ones are dynamic forms and links.
Content embedded in Tag <href> should be found to
get web pages, such as <a
href=http://news.sina.com.cn/20080110/n33543.shtml
class=red>alcate bell</a>. What need to pay attention is
that URL should be indeed part of web application.
Traversing depth and breath could be set in advance. We
must avoid traversing same page repeatedly, which could
lead to a dead cycle.
Dynamic form is to generate dynamic web pages by
sending request to server. All kinds of CGI program are
responsible for responding dynamic form. So input and
output of CGI program is the object we concern most.
Content in the dynamic form is actually Input box
embedded in the tag <form>.
Typical form is listed as follows:
<FORM name=" myform”
action=http://myWEB/a.cgi method=" post”>
<input type=hidden name=site value='com'
length-10>
<input type=text name=ABC Value="abc"
length=10>
<input type=hidden name=chatlogin
value='in'>
<input type=hidden name=product
value='mail'>
<input type=submit name=sumbmission
value=“submit”>
</FORM>
Dynamic form can be located by finding keywords
<form> and then we get the properties of form by some
keywords like method, post, get, text, hidden, name,
value, length. If there are dynamic forms in response,
corresponding request must include these forms. They
could be used by predictable technology based on
experiment, which conclude programmer’s tradition、rules
in IT field. It is meaningful to combine experience of test
export to web application traverse.
VI. CONCLUSION AND FURTHER WORK
This paper presents one traversing model for Web
application vulnerability detection, which improve the
efficiency in the traversing process. The model gets a set
of web pages by HITS algorithm. This set provides
dataset for Apriori algorithm. Strong association between
interactive properties and attack types could be got and
then support the simulating attack model.
Admittedly, it is necessary to design a reasonable way
to store data which was got from page documents. How to
search easily, how to take less space and how to avoid
redundancy is our future work.
AKNOWLEDGMENT
The research work here is sponsored by Tianjin
Science and Technology Committee under contract
08ZCKFGX01100 and 06YFJMJC0003.
REFERENCES
[1] Kleinberg J, Lawrence S. The Structure of the Web. Science,
2001, 294: 1849-1850
[2] Flake G W, Lawrence S, Giles C L. Efficient Densification of
Web communities. In Proc. of the Sixth International Conference on
Knowledge Discovery and Data Mining (ACM SIGKDD-2000),
2004:150-160
[3] Kleinberg J. Authoritative Sources in a Hyperlinked Environment.
Figure 1.
Proc. of 9th ACMSIAM Symposium on Discrete Algorithms. Also
Appeared as IBM Research Report RJ 10076, 1997-05
[4] Chakrabarti S, Dom B E, Gibson D, et al. Mining the Link
Structure of the World Wide Web. IEEE Computer, 1999, 32(8)
[5] Bharat K, Henzinger M R. Improved Algorithms for Topic
Distillation in a Hyperlinked Environment. In Proceedings of the
ACM-SIGIR, 1998
[6] Kleinberg J. Authoritative Sources in a Hyperlinked Environment.
Proc. of 9th ACMSIAM Symposium on Discrete Algorithms. Also
Appeared as IBM Research Report RJ 10076, 1997-05
[7] R Agrawal ，T Imielinski，A Swami．Mining Association Rules
between Sets of Items in large Database[C]．In：Proceedings of the ACM
SlGMOD Conference on Management of Data．1993：2O7—216
[8] Agrawal R, Srikant R. Fast algorithms for mining association
rules in large database[R]. Technical Report FJ9893, San Jose, CA: IBM
Almaden Research Center, 1994.
[9] Agrawal R, Srikant R. Fast algorithms for mining association
rules [A]. In: Proc of 20th Int Conf Very Large Databases (VLDB’94)
[C]. CA: [s.n.], 1994. 487-499
Traversing Model
31
 2009 International Conference on Computer Engineering and Technology
Attribute-based Relative Ranking of Robot for Task Assignment
B.B.Choudhury B.B.Biswal
Department of Mechanical Engineering Department of Mechanical Engineering
NIT Rourkela, Orissa, India NIT Rourkela, Orissa, India
e-mail:bbcnit@gmail.com e-mail:bibhuti.biswal@gmail.com
Abstract-Availability of large number of robot configurations
has made the robot workcell designers think over the issue of
selecting the most suitable one for a given set of operations.
The process of selection of the appropriate kind of robot must
consider the various attributes of the robot manipulator in
conjunction with the requirement of the various operations
for accomplishing the task. The present work is an attempt to
develop a systematic procedure for selection of robot based on
an integrated model encompassing the manipulator attributes
and manipulator requirements. The developed procedure can
advantageously be used to standardize the robot selection
process with view to perform a set of intended tasks. The work
is also aimed at creating an exhaustive list of attributes and
classifying them into different distinct categories. The coding
scheme for the attributes and the relative ranking of the
manipulators are illustrated with example.
Keywords-Multirobot,Relative ranking , Attributes.
I. INTRODUCTION
Recent developments in information technology and
engineering sciences have been the main reason for the
increased utilization of robots in a variety of advanced
manufacturing facilities. Robots with vastly different
specifications are available for a wide range of applications.
The selection of robots to suit a particular application and
production environment from among the large number
available in the market has become a difficult task. Today’s
market provides variety of robot manipulators having
different configurations and capabilities.The capability of a
robot manipulator can be assessed through parameters like
number of joints, type of joints, joint placement, link
lengths and shapes, workspace, manipulability, ease and
speed of operation, etc. The speed of operation significantly
depends on the complexities of the kinematic and dynamic
equations and their computations. Hence, in order to select
a suitable robot, both aspects of kinematics and dynamics
should be looked into. Selecting the right kind of robot for
an application is not easy. In addition, just meeting the
customer requirements can be a challenge. The addition of
system integration in workcell design processes may further
complicate the picture. However, the parameters that
determine the capability of the robot is heterogeneous in
natrure and therefore, formulating an integrated model with
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 32
DOI 10.1109/ICCET.2009.205
R.N.Mahapatra
Department of Mechanical Engineering
SIT Bhubaneswar, Orissa, India
e-mail:rabindra@silicon.ac.in
all these becomes a difficult task at times. Fortunately, a
number of tools and resources are becoming available to
help designers select the most suitable robot for a new
application. However, none of these solutions can take care
of all the demands and constraints of a user specific robotic
workcell design. Eventually the designers must use the
available information and make their own decisions. In this
paper, we propose a new mathematical based methodology
for robot selection to help designers identify feasible
robots, and then outline the most appropriate cases for
smoothing robot selection process. It deals with the issues
of using past experiences or cases to understand, plan for,
or learn from novel situations. The results of this study will
help robot workcell designers to develop a more efficient
and effective method to select robots for robot applications.
Offodile et al. [1] developed a coding and classification
system which was used to store robot characteristics in a
database, and then selected a robot using economic
modeling. Liang and Wang [2] proposed a robot selection
algorithm by combining the concepts of fuzzy set theory
and hierarchical structure analysis. The algorithm was used
to aggregate decision makers’ fuzzy assessments about
robot selection attributes weightings, and to obtain fuzzy
suitability indices. Rao and Padmanabhan [3] proposed a
methodology based on digraph and matrix methods for
evaluation of alternative industrial robots. A robot selection
index was proposed that evaluates and ranks robots for a
given industrial application. The index was obtained from a
robot selection attributes function, in turn obtained from the
robot selection attributes digraph. The digraph was
developed based on robot selection attributes and their
relative importance for the application considered. The
authors suggested step by step procedure for evaluation of a
robot selection index. Zhao and Yashuhiro [4] introduced a
genetic algorithm (GA) for an optimal selection and work
station assignment problem for a computer-integrated
manufacturing (CIM) system. Boubekri et al. [5] developed
an expert system for industrial robot selection considering
functional, organizational and economical factors in the
selection process. The use of DEA for robot selection has
been addressed by Khouja [6]. Huang and Ghandforoush
[7] stated the procedure to evaluate and select the robot
depending on the investment, budget requirements and
comparing the suppliers of the robots.But they had assumed
that the user knows which robot to buy and the question
was from whom to buy.
 II. MANIPULATOR ATTRIBUTES can be coded as per the parameters coding scheme
With the growth in robot applications, large numbers of mentioned in Table II.
robots are available for various manufacturing applications. TABLE II. THE CODING SCHEME
Proper identification of manipulator attributes is critically
important when comparing various alternative robots. Parameter Parameter coding scheme Code
General 1 2 3 0 0
6
Sometimes it may be possible to arrive at a rational choice Physical 4 5 6 7 8 9 10 4 0 0 0 0 0 6
without formal application of some quantitative or semi- Performance 11 12 13 14 15 16 17 0 5 0 0 5 0 1
quantitative methodology by mere articulation of what 18 0
attributes are important in the context of particular Structure 19 20 7 0
alternatives under consideration. However, in most cases the Application 21 0
user needs to be assisted in identifying the robot attributes Sophistication 22 23 0 0
Control 24 25 26 27 28 29 3 0 0 1 0 0
logically. A robot manipulator can be specified by a number Availability 30 31 0 0
of quantitative attributes such as payload capacity, Task 32 33 34 35 4 4 8 5
repeatability, horizontal reach, etc. However some attributes
such as built quality, after sales service etc. cannot be TABLE III. IDENTIFICATION CODE
expressed quantitatively. These may be expressed by a rate on Sl. Attribute Information Code
the scale of (say 1-10). There are some attributes which are 1. Price range 0
informative in nature, such as type of drive, the coordinate 2. Type of robot 0
system, which may be denoted by some number whose 3. Arm geometry Any one 6
numerical value will have no significance .There are also 4. Type of actuators Any one 4
attributes of which the quantification is not available and 5. Weight of the robot 0
6. Size of the robot 0
needs to be done by some mathematical model and analysis. 7. Type of grippers supported 0
For instance reliability can be expressed in terms of Mean 8. Number of axes 0
Time Between Failure (MTBF) or Mean Time To Repair 9. Space requirements of the 0
(MTTR) methods. Attributes like life expectancy may be 10. Maximum reach 1000 mm 4
estimated through experimentation, if not mentioned by the 11. Types of end effectors 0
manufacturer. 12. Payload of the robot 5kg 5
13. Workspace 0
TABLE I. MAJOR ATTRIBUTES FOR MANIPULATOR AND TASK 14. Stroke 0
15. Velocity 50 deg/sec 5
Attribute type Parameter 16. Accuracy 0
17. Repeatability ±0.02mm 1
18. Resolution 0
General Price range, Type of robot and Arm geometry
19. Degree of freedom 2 7
Physical Type of actuators, Weight of the robot,
20. Type of joints 0
Size of the robot, Type of grippers
21. Working environment 0
supported, Number of axes ,Space
22. Maintainability 0
requirements of the robot, Maximum
23. Safety features - 0
reach
24. Control of robotic joints Any one 3
Performance Workspace,Payload of the robot, Stroke, 25. Gripper control - 0
Maximum end effector speed, Accuracy, 26. Sensors 0
Repeatability, Resolution 27. Programming method Any one 1
28. Number of input channels - 0
Structure/architecture Degree of freedom (DOF) availability, Type of 29. Number of output channels - 0
joints 30. Down time 0
31. Reliability 0
Application Working environment 32. Space 4
33. Time 4
Sophistication Maintainability and Safety features 34. DOF required 8
35. Force 5
Control/feedback system Control of robotic joints, Gripper control,
Sensors, Programming method, Number of
input and output channels of the controller The identification code in Table III specifies the attribute
information with the allotted code in the respective cells. The
Availability/reliability Downtime and Reliability identification of various pertinent attributes and their
values, rates and estimates help the user for create a
Task Space, Time, DOF, and Force
database. These data individually or collectively help the
user to select the most suitable robot for a task that he
intends to perform. Although the number of joints, axes,
and degree of freedom (DOF) are usually considered as
The manipulator attributes are found out based on its
basic parameters, the sequence of joints and their respective
broad area as general parameters, physical parameters,
orientations and arrangements are left untouched. The
performance based, etc. are given in Tabel. I. The attributes
robots with same number of joints and joint sequence but

33
with different joint orientations will have different
performance characteristics. This architecture gets
representation in the formulation of the problem for the
present work. The ease of operation, termed as
manipulability, can be quantified as manipulability measure
and can be used as an attribute. The main attributes have
been broken down to sub-attributes and sub-sub-attributes
so that the robot manipulator can be identified in very
precise and detailed manner. The information supplied by
the manufacturer to the user is not sufficient and it is
required to be more elaborate. The ‘0’ represents that the
information relating to the particular cell is not available.
This coding scheme can be used as it is for the visual
comparison between two robots up to certain extent. It
allows faster comparison in various formats.
III. THE ROBOT SELECTION PROCESS
The robot selection criteria should include the key
parameters such as; degrees of freedom, pay load, swept
area, maximum reach, maximum speed, cost, and
repeatability. Based on different robot applications, some of
the robot selection criteria can be ignored and others may
become critically important. Recent advances in robotics
technology allow robotic workcell engineers to design and
implement more complex applications than ever before.
Further, some of these considerations also make the robot
selection process more complicated. There will be few
attributes, which will have direct effect on the selection
procedure. These attributes may be set aside as 'pertinent
attributes' as necessitated by the particular application and/or
the user. The threshold values to these 'pertinent attributes'
may be assigned by obtaining information from the user and
the group of experts. Henceforth the selection procedure
focuses solely on the pertinent attributes leaving out the
rest. On the basis of the threshold values of the pertinent
attributes, a shortlist of robots is obtained. This is achieved by
scanning the database for those attributes, one at a time, to
eliminate the robot alternatives, which have one or more of
these attribute values that fall short of the minimum required
(threshold) values. To facilitate this search procedure an
identification system has been made for all the robots in the
database. The robot selection system can be divided into
four activities; i) operation requirements and data library of
robots, ii) coding scheme, iii) selection of attributes, and iv)
ranking of robots. The first activity mainly consists of
listing the requirements of robots and the desired
operations. The robot selection attributes are identified for
the given application and the robots are shortlisted on the
basis of the identified attributes satisfying the requirements.
A mini-database is thus formed which comprises these
satisfying solutions i.e., alternatives which have all attributes
satisfying the acceptable levels of aspiration. The problem is
now one of finding out the optimum or best out of these
satisfying solutions. The selection procedure, therefore, needs
to rank these solutions in order of merit. The first step is to
represent all the information available from the database
about these satisfying solutions in the matrix form. Such a
matrix is called as decision matrix, ‘D’. Each row of this
matrix is allocated to one candidate robot and each column
34
to one attribute under consideration. Therefore an element
dij of the decision matrix ‘D’, gives the value of jth
attribute in the row (non-normalized) form and units,
for the i th robot.
The normalized specification matrix, ‘N’, is then
formed from the decision matrix, ‘D’. Normalization is
used to bring the data within particular range. Moreover,
it provides the dimensionless magnitudes. This
phenomenon is used to calculate the normalized
specification matrix. The normalized specification matrix
has the magnitudes of all the attributes of the robots on
the common scale of 0 to 1. It is a value, which indicates
the standing of that particular attribute magnitude when
compared to the whole range of the magnitudes for all
candidate robots. An element n ij of the normalized
matrix ‘N’ can be calculated as;
d ij 1/ 2
nij =
⎛ m ⎞
⎜∑
⎜ d 2⎟
ij ⎟
⎝ i =1 ⎠
where, d ij is an element of the decision matrix, ‘D’.
The next step is to obtain information from the user
or the group of experts on the relative importance of
one attribute with respect to another. Values of all such
comparisons are stored in a matrix called as weight
matrix, ‘W’, where wij contains the relative importance of
ith attribute over the jth attribute. The value obtained
from the weight matrix is applied to the normalized
specifications since all attributes have different
importance while selecting the robot for particular
application. The matrix, which combines the relative
weights and normalized specification of the candidates,
is weighted normalized matrix, ‘V’. It gives the true
comparable values of the attributes. This can be
obtained as follows:
⎡ w 1n 1,1 w 2 n 1,2 … w n n 1,n ⎤ ⎡ v1,1 v1, 2 v1,n ⎤
⎢w n … ⎥ ⎢v ⎥
V=⎢ ⎥ = ⎢ 2,1 ⎥
1 2,1
⎢ ⎥ ⎢ ⎥
⎢ ⎥ ⎢ ⎥
⎢⎣ w 1n m,1 w 2 n m,2 … w n n m,n ⎥⎦ ⎢⎣ v1,1 v1,1 v1,1 ⎥⎦
V. ILLUSTRATIVE EXAMPLE
The example considers a task of pick-and-place, to be carried
out by a suitable robot where it is expected to avoid some
obstacles. The minimum requirement for this application is
tabulated in Table IV. The objective values of the robot
selection attributes, which are given in Table V. are of
different dimensions and units.Hence they are normalized.
In order to demonstrate and validate the methodology of the
proposed method five robots with different configurations
and capabilities are considered.
 ⎡ MR DF PL VL AG AT CM RT RP SC TE DF1 FR ⎤
⎢ 13 2 6 1 3 1 6 0 .5 1 1 6 13 2 ⎥⎥
⎢
TABLE IV. MINIMUM REQUIREMENT OF A ROBOT ⎢ 12 3 5 2 4 2 5 1 1 .5 2 5 12 3 ⎥
⎢ ⎥
⎢ 11 4 4 3 5 3 4 1.5 2 3 4 11 4 ⎥
Sl Parameter Values ⎢ 10 5 3 4 6 4 3 2 2.5 4 3 10 5 ⎥
⎢ ⎥
⎢ 9 6 2 5 7 5 2 2 .5 3 5 2 9 6 ⎥
W =⎢ 8 7 1 6 8 6 1 3 3 .5 6 1 8 7 ⎥
1 Load capacity minimum 5 kg. ⎢ ⎥
⎢ 7 8 1 6 9 6 1 3 3 .5 6 1 7 8 ⎥
2 Repeatability 0.02 mm ⎢ 6 9 2 5 10 5 2 2 .5 3 5 2 6 9 ⎥
3 Velocity at least 50 deg/sec ⎢ ⎥
⎢ 5 10 3 4 11 4 3 2 2.5 4 3 5 10 ⎥
4 Types of drives any one ⎢ ⎥
⎢ 4 11 4 3 12 3 4 1.5 2 3 4 4 11 ⎥
6 Degree of freedom at least 2 ⎢ 3 12 5 2 13 2 5 1 1 .5 2 5 3 12 ⎥
7 Arm geometry any one ⎢ ⎥
⎣ 2 13 6 1 14 1 6 0 .5 1 1 6 2 13 ⎦
8 Control mode any one
9 Robot programming any one
Step 3: Calculation of the normalized specification
matrix.
⎡0 .844 0 . 713 0 .877 0 .796 0 . 876 0 .344 0 .691 0 .979 0 . 674 0 . 259 0 .339 0 .447 0 .447 ⎤
⎢ 0 .428 0 .475 0 .438 0 .437 0 .388 0 .241 0 .461 0 .195 0 .505 0 .287 0 .398 0 .447 0 .447 ⎥
⎢ ⎥
N = ⎢ 0 .171 0 .356 00 .146 0 . 331 0 . 174 0 . 803 0 .345 0 .039 0 .337 0 .323 0 .435 0 .447 0 .447 ⎥
⎢ ⎥
⎢ 0 .171 0 .285 0 .109 0 .199 0 . 174 0 .241 0 .276 0 .019 0 .337 0 . 571 0 .406 0 .447 0 .447 ⎥
TABLE V. CRITERIA FOR ROBOT SELECTION ⎣⎢ 0 .155 0 . 237 0 .072 0 .159 0 .145 0 .344 0 .345 0 .019 0 . 252 0 .647 0 . 61 0 .447 0 .447 ⎦⎥

Step 4: Calculation of normalized value (N.V.),

∑
Criteria Robot-1 Robot-2 Robot-3 Robot-4 Robot-5
Maximum Reach(MR) 1000 2000 5000 5000 5500 ranking factor ( σ =W * N), and total score σ.
DOF(DF) 2 3 4 5 6
Payload(PL) 5 10 30 40 60 Step 5: Calculating the average of the ranking factors
Velocity(VL) 50 90 120 200 250 of all the robots
Arm geometry(AG) 4 9 20 20 24
Actuator(AT) 7 10 3 10 7 TABLE VI. RANKING FACTOR WITH ONE SET OF WEIGHTAGE
Control mode(CM) 4 6 8 10 8 OF ROBOT-1
Repeatability(RT) 0.02 0.1 0.5 1.0 1.0
Robot programming(RP) 3 4 6 6 8 Parameter Value N.V W Ranking factor(σ) ∑σ
Space(SC)* 0.5 0.45 0.4 0.3 0.2 MR 1000 0.844 13 10.972
Time(TE)* 0.359 0.3 0.28 0.3 0.2
DOF(DF1)* 3 3 3 3 3 DF 2 0.713 2 1.462
Force (FR)* 5 5 5 5 5 PL 5 0.877 6 5.262
VL 50 0.796 1 0.796
AG 4 0.876 3 2.628

35.7715
*These values pertain to task-1 of the fifteen tasks AT 7 0.344 1 0.344
actually considered for the problem. However only CM 4 0.691 6 4.146
RT 0.02 0.979 0.5 0.4895
one task has been considered for calculation due to
RP 3 0.674 1 0.674
page constraints. SC 0.5 0.259 1 0.259
The normalized values of all these parameters are TE 0.359 0.339 6 2.034
taken to form the decision matrix. The procedure for the DF1 3 0.447 13 5.811
selection of the robot is as follows FR 5 0.447 2 0.894

The calculation of the total ranking factor, ∑ σ, for one set

:Step 1: Formation of decision matrix, ‘D’. weights in robot-1 is presented in Table VI. The
⎡ MR DF PL VL AG AT CM RT RP SC TE DF1 FR ⎤
⎢1000 2 5 50 4 7 4 50 3 2 2.78 3 5 ⎥⎥
calculations of ranking factors are made for the other robots
⎢
⎢2000 3 10 90 9 10 6 10 4 2.22 3.33 3 5 ⎥
with a total of 12 different sets of weights. The values of
D =⎢ ⎥ these ranking factors for all the robots are given Table VII.
⎢5000 4 30 120 20 3 8 2 6 2.5 3.57 3 5 ⎥
⎢5000 5 40 200 20 10 10 1 6 3.33 3.33 3 5 ⎥
⎢ ⎥ V.RESULTS AND DISCUSSION
⎣⎢5500 6 60 250 24 7 8 1 8 5 5 3 5 ⎦⎥

The robots are arranged in order of their ranking factor

based on the significant attributes chosen keeping the
application of the robots in view. However, before a final
Step 2: Formation of weight matrix, ‘W’. decision is taken to select a new robot, the factors such as
economic considerations, availability, management constraints
and corporate policies, etc. may be considered. These factors
were not previously considered in coding and evaluation.
According to the results obtained and the analysis thereby,
Robot-5 and Robot-4 has the highest ranking factors should
be recommended as the best robot alternative. The 1st and

35
2nd ranked robots have the highest figures amongst all the
robots. As a result of the application of both numerical and
qualitative inputs and outputs, two robot alternatives are
found to be more efficient compared to other candidates. In
order to discriminate between these two robot alternatives
the ranking factor should be looked at. The ranking curves
of robots are shown in Fig. 1. The average values of the
ranking factor are presented in Fig. 2.
TABLE VII. VALUES OF TOTAL RANKING FACTOR
Robot Value of ∑ σ with different set of weights.
Robot-1 35.7355, 36.799, 37.8625, 38.926, 39.9895, 41.053,
41.798, 42.2245, 42.651, 43.0775, 43.504,
Robot-2 23.7325, 24.1855, 24.6385, 25.0915, 25.5445,
25.9975, 26.4325, 26.8495, 27.2665, 27.6835,
28.1005, 28.5175
Robot-3 17.5315, 18.6095, 19.6875, 20.7655, 21.8435,
22.9215, 23.2805, 22.9205, 22.5605, 22.2005,
21.8405, 21.4805
Robot-4 16.1235, 16.8095, 17.4955, 18.1815, 18.8675,
19.5535, 19.8415, 19.7315, 19.6215, 19.5215,
19.4015, 19.2915
Robot-5 17.2025, 17.688, 18.1735, 18.659, 19.1445, 19.63,
19.857, 19.8255, 19.794, 19.7625, 19.731, 19.6995

TABLE VIII. SCORES OF ROBOT

Sl. No Robot Average Relative Relative
Ranking Factor Ranking Rating
1 Robot-1(R-1) 40.6293 4 Low
2 Robot-2(R-2) 26.17 3 Medium
3 Robot-3(R-3) 21.304 2 Medium
4 Robot-4(R-4) 18.703 1 High
Figure 1. Ranking curves of robots 5 Robot-5(R-4) 18.703 1 High

The present work is aimed at developng a generalized

Figure 2. Comparison of robots
Although ranking of robots on the basis of the
manipulators parameters alone has been attempted by some
previous researchers, ranking of the robots in view of
performing a given set of tasks is a novel attept. The
present work considers practcal aspects and takes and takes
experimental data to form an integrated model for relative
ranking of the available candidate robots. On the basis of
the ranking factors the robots are rated as ‘Low’, ‘Medium’
and ‘High’ and in relation to the group of the robots under
consideration and are shown in Table VIII.The procedure
provides a coding system for robots depicting the various
attributes. It recognizes the need for, and processes the
information about, relative importance of attributes for a
given application without which inter-attribute comparison
is not possible. Essentially the present work contributes to
developing a methodology based on matrix methods which
helps in selection of a suitable robot from among a large
number of available alternative robots
tool to combine manipulator attributes and task
requirements in a comprehensive manner for relative
ranking of the manipulators.In the intial phase of the
formulation ,35 attributes of the robots are identified and
consciously coded to take care of the characterstics of a
robot manipulator precisely. The methodology developed
through this work can be applied to any similar any similar
set-up .This is sure to help the designers and users in
selecting the robots correctly for the intended application.
REFERENCES
[1] O.F.Offodile ,P.K.Lambert and R.A.Dudek. “Development of a
computer aided robot selection procedure (CARSP)”. Int J Prod Res 1987;
25:PP.1109–12.
[2]A.M.Layek , L.J.Resare. “Algorithm based decision support system for
the concerted selection of equipment in machining/assembly cells”,
International Journal of Production Research, 2000, 38:PP.323–339.
[3]R.V.Rao , K.K.Padmanabhan, “Selection, identification and
comparison of industrial robots using digraph and matrix methods”,
Robotics and Computer Integrated Manufacturing,2006, 22:P.373–383.
[4] L.Zhao ,T. Yashuhiro, “Genetic algorithm for robot selection and work
station assignment problem” Computers & Industrial Engineering
,1996,31:PP.599–602.
[5]N. Boubekri , M.Sahoui and C. Lakrib, “Development of an expert
system for industrial robot selection”. Comput Ind Eng ,1991; 20:PP.119–
27.
[6] M.Khouja, and D.E. Booth, “Fuzzy clustering procedure for evaluation
and selection of industrial robots”. Journal of Manufacturing Systems,
1995, 14, PP.244- 251.
[7] P.Y. Huang, P. Ghandforoush, “Robotics procedures given for
evaluating selecting robots”, Industrial Engineering 16 , 1984,
pp. 44–48.

36
 2009 International Conference on Computer Engineering and Technology
A Subjective Trust Model based on two-dimensional measurement*
Chang Chaowen1,2, Liu Chen2, Wang Yuqiao2
1. School of Electronic and Information Engineering, Xi’an Jiaotong University, Xi'an, China
2. Institute of Electronic Technology, Information Engineering University, Zhengzhou, China
ccw@xdja.com
Abstract
Trust models like Beth, Jøsang and EigenRep
evaluated trust degree by the history of interactions or
the reputations, and some other learners brought fuzzy
theory, cloud model into the researches of trust
management. All of these models based on the
trustee’s outer information. However it is more
reasonable to get the evaluations by both the outer
information and the inner attributes. Aiming this issue,
this paper gets the trustee’s inner attributes through
the remote attestation mechanism under the trusted
computing specification of TCG, and then combines
the history of interactions, and designs a trust model
based on two-dimensions (attributes, experiences) in
order to make the evaluation of the trust degree more
flexible and reliable. The experiments show that this
two-dimensions-based model can not only avoid the
trust accumulation spoofing better but also calculate
the initial trust value more effectively.
1. Introduction
M.Blaze brought forward the concept of trust
management and the corresponding trust model system
PolicyMaker[1] and KeyNote[2] for the first time in
1996. It is very difficult to give a standard definition
because of the complexity of trust relationship and the
variety of application requirement. At present the study
of trust management models can be divided into two
parts: Credential-based models and Evidence-based
models[3]. In the Credential-based models, e.g.
PolicyMaker[1], KeyNote[2], REFEREE[4], the essence
of trust is using a precise and reasonable method to
describe or deal with the complicated relationships;
whereas the Evidence-based models consider
subjectivity, inaccuracy and evolvement adequately. In
the open environment, the Evidence-based models are
able to reflect the variety of the environment.
There are some classic Evidence-based models, e.g.
Beth model, Jøsang model and EigenRep model. Beth
*
This work was supported by National High Technology Research and Development Program of China (863 Program, No.2007AA01Z479)
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.86
model [5] brought in the concept of experiences to
describe and measure the trust relationship. Beth
model classifies the trust into direct trust and
recommendation trust. Jøsang proposed a trust
management model based on subjective logic [6][7], and
brought in the concept of evidence space and opinion
space to describe the trust relationship and measure it.
This model bases on the Beta distribution function of
expression for posteriori probability estimates of
binary events, gives the pcdf (probability, certainty
density, functions) which is characterized by the
number of positive events and negative events that are
observed. EigenRep model [8] is a classic global trust
model at present. The kernel of it’s is that if entity
someone wants to know global reputation values of
any entity K, it needs to get the trust information of K
from entity J who has interacted with K.
What all of these typical models considered is the
outer information of the trustee, it’ll result in some
problems [9]: (1) Using events probability to describe
and measure trust relationship will result in confusion
between the subjectivity of trust and the random
probability. Maybe it is not very reasonable. (2) Most
of these calculate arithmetical average value to
integrate the trust degree from different
recommendation paths. It is hard to eliminate the
impact of malicious recommendation. (3) Although
these models have calculating formula about trust,
most of them don’t have an effective way to get the
initial trust degree.
The literature [10] proposed a new method based on
fuzzy theory according to the third party’s trust values
of recommendation so that the assess of users trust
values become more flexibility and more reliable. The
literature [11] defined the assurance trust and global
reputation, described the trust relationship effectively,
assured the anonymity of entities and the secrecy and
integrity of trust value and prevents aiming cheat and
combining cheat. The literature [12] proposed a
formalism method for subjective trust using cloud
37
model, and a qualitative reasoning mechanism of trust
cloud to enable trust-based decisions.However, these
ameliorations merely consider the outer information as
the same as classic models instead of the inner
attributes of the trustee.
As we know, for everything, there are close
relevancy between outer information and inner
attributes, outer information is the phenomena and
inner attributes is the essential, based on which, we
adopt the remote attestation mechanism under the
trusted computing specification of Trusted Computing
Group(TCG) [13] to get the inner characteristic of
trustee.
2. Get inner attributes base on TCG
remote attestation
TCG offers a specification for the trusted computing
and security technology definition and development of
industrialized criterion [13]. The Trusted Platform
Module (TPM) [14][15] is the foundation stone of trusted
computing. In the trusted node with TPM, any module
must be measured before it obtain the control right,
and the measured values should be preserved in the
Platform Configuration Register (PCR) to prevent
them from being tampered.
Integrity measurement [16][17] is a very important
function of trusted computing. Based on TPM, the
measurement values in PCRs cannot be tampered and
written in advance. When a attestation begins, trustee
convey the PCR values, Stored Measurement Log
(SML) and other correlative information to remote
trustor by a trusted way [18], the trustor then can
estimate the security state of the trustee according as
the integrity report, thereby determines whether
interact with it. The specification of TCG has defined
the protocol for the remote attestation [14], which offers
us an effective method to get the inner information of
trustee. Based on the remote attestation protocol, we
can know the main application attributes’ present state
is trusted or not, thereby we can evaluate the trust
degree at a bran-new aspect.
However, the platform configurations may not
reflect the actual state of the platform sometimes [19][20].
For example, when software updates, the configuration
will be changed and attested unsuccessfully, but the
platform state is still secure and trusted. What is more,
even though platform state attests successfully, the
interaction between the two parties may also get fail
because of the trustee’s low enthusiasm to offer some
services to trustor. So it is also not integrated and
objective to evaluate the trust degree only by the TPM-
based attestation.
38
3. A trust model based on two-dimensional
measurement
As above, present models emphasize particularly on
only one factor, the outer factor or inner factor. So it is
hard to avoid being unilateral. Combining the remote
attestation mechanism under the trusted computing
specification of TCG with the interaction experience
between the two parties, estimating trust degree of the
trustee synthetically can make the measured results
more accurate.
3.1 Definition and Properties of Trust
1) Roles in Trust Relationship
Trustor: The proposer of the interacting services. It
is also the verifying party.
Trustee: The provider of the interacting services. It
is also the verified party. In our model system, the
trustee must set the TPM chip.
2) Definitions
Definition 1: Trust is to believe others, is a cognitive
process that trustor believe the trustee has the ability to
meet its requests through judging the inner attributes
and the outer behaviors of trustee. This definition
emphasize that although trust is a subjective behavior,
it must base on the objective.
Definition 2: Trust degree is the quantitative
denotation of trust grade. This paper has used
subjective logic-based model for reference, used the
ternary aggregate {b,d,u} to describe the trust
relationship. The aggregate is in the situation:
B + d + u = 1, { b ,d ,u }∈[0, 1]3
b is the trust degree, d is the distrust degree and u is the
incertitude degree. All of them are continuous values.
Definition 3: Inner attributes trust degree is the trust
degree of trustee’s inner attributes, here it means the
trust grade of integrity of the platform configurations.
Definition 4: behavior trust degree is the trust degree
of trustee’s outer behaviors; here it means the trust
grade of interactions history with other nodes or the
recommendation of others and so on.
Definition 5: integrated trust degree is the mean value
of the inner attributes trust degree and the behavior
trust degree by weight.
3) Properties of Trust
According to the definitions above, trust is a
dialectic unification of subjective and objective, and it
should have some properties:
Property 1: Uncertainty means it is hard to describe
trust in an accurate way; there is no absolute trust or
absolute distrust. Thereby uncertainty can also be
called relativity.
Property 2: Asymmetry means trust is one-way, A
trusts B doesn’t mean that B trusts A too.
 Property 3: Incomplete transitivity means A trusts B Behavior trust degree describes the trusted situation
and B trusts C, but A could not trust C completely even of the trustee’s outer behaviors. We use the theory of
though B recommends it. Some other literatures think evidence space in Jøsang model [21][22] for reference,
trust has the transitivity, thus it is hard to avoid the regard the events that have been validated
malicious recommendation effectively. In this paper, unsuccessfully as the negative events and the events
we suppose that recommendation is only a reference that have been validated successfully as the positive
but not the unattached way to calculate trust degree. events. Assume r denotes the number of the positive
events and s denotes the number of the negative events,
3.2 Descriptions of Trust Relationship we can calculate the behavior trust degree TO = {bO, dO,
1) Inner attributes trust degree evaluation algorithm uO} by the formula:
(1) Through the TCG protocol, trustor gets the r
configurations integrity information which is needed in bo =
trustee’s platform, the information contains PCR r + s +1
values, SML and so on. s
do =
(2) Validate the configurations integrity of trustee r + s +1 (3-3)
platform, and get the number f of the PCR values
1
which have been validated unsuccessfully in uo =
PCR0 ,PCR1 , … , PCRn. Assume that the n + 1 - f r + s +1
PCR values which have been validated successfully 3) Integrated trust degree evaluation algorithm
have the same trusted situation, the ternary aggregate is Integrated trust degree is the mean value of the inner
{bS, dS, uS}. Here, bS denotes the possibility that the attributes trust degree and the behavior trust degree by
module has not been impacted by the malicious code weight. Assume WI is the weight of inner attributes
(modules may impact each other due to non-isolation). trust degree and WO is the weight of behavior trust
The f PCR values which have been validated degree, WI +WO=1, we can calculate the integrated trust
unsuccessfully have the same trusted situation too; the degree T = {b,d,u} by the formula:
ternary aggregate is {bF ,dF, uF}. Here, dF denotes the b = WIbI + WObO
possibility that a single PCR value which has been d = WIdI + WOdO (3-4)
validated unsuccessfully may destroy the security of
the system (That a PCR value is validated u = WIuI + WOuO
unsuccessfully doesn’t mean the system is threatened 4) the Initial Trust Value of Integrated trust degree
predicatively, e.g. when a software updates can result In the initial state, there were no interactions between
in the fail validation but it is hurtles). We can calculate the two parties, the parameters r = 0, s = 0, the
the inner attributes trust degree TI = {bI,, dI , uI} by a behavior trust degree is TO = {0,0,1}, and the
simple formula: integrated trust degree T = {b,d,u} is:
n +1− f f b = WIbI
bI = bS + bF d = WIdI
n +1 n +1 (3-5)
n +1− f f u = WIuI + WO
dI = dS + dF From this formula, we can see that the initial trust
n +1 n +1 (3-1) value can be obtained by the inner attributes trust
n +1− f f degree, i.e. validating the configurations integrity of
uI = uS + uF trustee’s platform.
n +1 n +1
In the situation that trust neither the trust degree nor
3.3 Evaluation of Trust
distrust degree doesn’t attenuate, both uS and uF is 0.
Trustor sets the threshold T0 ={b0,d0,u0}, the set of
The formula is:
threshold bases on the context and the subjective
n +1− f f
bI = bS + bF factors of the trustor, we don’t discuss how to set the
n +1 n +1 threshold in this paper. The evaluation of trust contains
n +1− f f three conditions as follows:
dI = dS + dF Condition 1: integrated trust degree satisfies
n +1 n +1 (3-2) T(b) ≥ T 0(b0 ) ; T(d) < T 0(d 0 ) ; T(u) < T 0(u 0 )
uI = 0 Then trustor regards trustee trusted, the
2) Behavior trust degree evaluation algorithm interactions are allowed.

39
Condition 2: integrated trust degree satisfies
T(b) < T 0(b0 ) ; T(d) ≥ T 0(d 0 ) ; T(u) < T 0(u 0 )
Then trustor regards trustee distrusted, the
interactions are un-allowed.
Condition 3: integrated trust degree satisfies
T(b) < T 0(b0 ) ; T(d) < T 0(d 0 ) ; T(u) ≥ T 0(u 0 )
Then trustor sends requests to validate the
configurations integrity of trustee’s platform.
4. Emulation experiment and analysis
4.1 Experiment 1: Avoid Malicious Accumulation of
Trust Degree
The calculating formula of the subjective logic-based
model is as formula 3-3. There is a problem in this
method: Assume that trustee offered services honestly
at the first stage, so that the trust degree could
accumulate to a very high grade; then it offered one
malicious service suddenly, the trustor would have not
the least guard and suffer a serious loss. We call this
problem as malicious accumulation of trust degree.
The subjective trust model based on assurance [11] has
proposed the concept of attenuation element that can
decrease the impact about the above problem.
Attenuation element β ∈ {0,1} , β m denotes the β
value at the m time. The smaller this value is, the more
obviously the trust degree attenuates, and the faster the
interacting experience is forgotten. The trust degree is
influenced mainly by the latest interactions. The
variety of β can also reflect the stability of the
interactions, if they are instable, β value will decrease.
At the beginning, we can suppose that every
interacting evidence will influence the trust degree
equally, i.e. β1 = 1 . Then every time the interacting
result changes, β decreases a certain value ∂ until to
a certain lower limit. The assurance-based trust model
can make the trustor who suffered loss adjusts its states,
but it cannot avoid the cheat from happening. The trust
model based on two-dimensions proposed in this paper
can avoid the cheat effectively. We will contrast this
model with the subjective logic-based model and the
assurance-based trust model to illuminate its advantage
in anti-cheat aspect.
At the beginning of this experiment, the
configurations of trustee’s platform are all right. After
the two parties interacted successfully for some times,
malicious code destroyed the integrity of the platform
but trustee concealed it deliberately and continued to
offer the service. Parameters are set as follows:
The number of PCR values which are applied by the
trustor is 10, i.e. n+1=10; PCR values are all validated
40
successfully when configurations are all right, f=0,
{bS,dS,uS}={1,0,0}; After running of the malicious
code, the integrity has been destroyed, the number of
PCR values which are validated unsuccessfully is 2,
f=2 ， {bS,dS,uS}={0.4,0.6,0}, {bF,dF,uF}={0.2,0.8,0};
In initial time, the number of positive events r=5 and
the number of negative events s=5; there are 40 times
of interactions between the two parties totally, the
former 30 times are successful and the later 10 times
are unsuccessful, the malicious code run to destroy the
integrity of the platform at the 15th time. WI=0.8,
WO=0.2, the threshold T0={0.5,0.3,0.2}. Fig.1 has
described the trend of trust degree in our model, the
Jøsang model and the assurance-based model in which
Fig.1 the Trend of Trust Degree
The analysis of the result: In the former 30 times, all
interactions are successful, the trust degree in the
Jøsang model will accumulate to a high level gradually.
Until the first time that interaction is fail, the trust
degree will play down. The trust degree in the
assurance-based model descends much faster than the
Jøsang model when interactions begin to fail, but it
doesn’t avoid the threat from happening. The model
based on two-dimensions could detect the startup of
the malicious code, the trust degree is at a high level at
the beginning, and descends to a quite low level
immediately when the malicious code runs, so that
some loss can be avoided.
4.2 Experiment 2: Set Initial Trust Value
Assume that trustee is the malicious node who has
destroyed the integrity of the platform. Parameters are
set as follows: The number of PCR values which are
applied by the trustor is 10, i.e. n+1=10; the number of
PCR values which are validated unsuccessfully is 2,
f=2; the number of positive events r=0 and the number
of negative events s=0; {bS,dS,uS}={0.4,0.6,0},
{bF,dF,uF}={0.2,0.8,0}, WI=0.8, WO=0.2; the
threshold T0={0.5,0.3,0.2}.
The node repeats to join into the network 5 times
independently. Using the formula of our model, the
initial values are all {0.36, 0.64, 0}, and the
mechanism of random number get 5 distinct initial
values.
Tab.1 the Initial Trust Value for 5 Random Numbers
Seq. 1 2 3 4 5 Model for Large Scale Distributed Environment. Journal of
Initial 0.4, 0.9, 0.8, 0.2, 0.6,
values 0.5, 0.1, 0.1, 0.7, 0.2,
0.1 0 0.1 0.1 0.2
The analysis of the result: There will be a very large
possibility (the probability in this experiment is 3/5) to
result in that trustor trusts the malicious node and
interacts with it by the mechanism of random number.
5. Conclusions
In the open networks, the Evidence-based trust
models consider subjectivity, inaccuracy and
evolvement adequately. The classic Evidence-based
trust models considered trustee’s outer information, so
that the essential characteristics of trustee cannot be
reflected directly and exactly. This paper designs a
trust model based on two-dimensions to make the
evaluation of the trust degree more flexible and
reliable. The experiments show that this two-
dimensions-based model can not only avoid the trust
accumulation spoofing better but also calculate the
initial trust value more effectively. However, the
improved model does not consider the
recommendation trust, so it is needed to be studied in
the future works.
6. References
[1]M.Blaze,J.Feigenbaum,J.Lacy,Decentralized Trust Man-
agement, Proceedings of the 1996 IEEE Symposium on
Security and Privacy,IEEE Computer Society,Washington,
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[2] Jøsang A ． A Logic for Uncertain Probabilities[J].
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[3] Yuan Shi-jin.The Research on Key Technologies of Trust
Management. Doctor Degree Paper for Fudan University.
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[4] Chu Y-H ， Feigenbaum J ， La Macchia B et
al ． REFEREE ： trust management for Web applications
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[5] Beth T, Borcherding M, Klein B. Valuation of trust in
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[6] Jøsang A, Knapskog S J. A Metric of Trusted Systems,
Global IT Security[M]. Wien: Austruan Computer Society,
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[7] Jøsang A. Trust-based Decision Making for Electronic
Transactions [EB/OL]. Proceedings of the 4th Nordic
Workshop on Secure Computer System(NORDSEC’99),
http://security.dstc.edu.au/staff/ajosang/paper.html, 1999.
[8] S Kamvar, Schlosser MT. EigenRep: Reputation
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123-134.
[9] Li Xiao-yong, Gui Xiao-lin. Research on Dynamic Trust
Software,Vol.18, No.6, June 2007,pp.1510-1521. (in Chinese)
[10]Zhang Yan-qun, Zhang Chen. Model of trust values
assess based on fuzzy theory. Computer Engineering and
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[11]Gao Cheng-shi, Fu Jiang-liu, Shen Chang-xiang, Liu Yi.
Subjective Trust Model Based on Assurance in Open
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[12]Meng Xiang-yi, Zhang Guang-wei, Liu Chang-yu, Kang
Jian-chu, Li He-song. Research on Subjective Trust
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[13]TCG Best Practices Committee. Design, Implementation,
and Usage Principles for TPM-Based Platforms Version 1.0.
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[14]Trusted Computing Group(TCG). TPM Specification
v1.2 Design Principles. [EB/OL]. https://www.trusted
computinggroup.org/specs/TPM/, 2005.
[15] Trusted Computing Group(TCG) ． Trusted platform
module protection profile，July 2004．
[16] Reiner Sailer,Xiaolan Zhang,Trent Jaeger, Leendert Van
Doorn. Design and Implementation of a TCG-Based Integrity
Measurement Architecture[A]. In:Thirteenth Usenix Security
Symposium[C]. San Diego, CA : USENIX press, 2004:223–
238.
[17]Xinwen Zhang, Michael J.Covington, Songqing Chen
and RaviSandbu. SecureBus:Towards Application-
Transparent Trusted Computing with Mandatory Access
Control[A]. In: ASIACC'07[C],Singapore:ACM Press,2007:
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[18]Trusted Computing Group.TCG Software Stack(TSS)
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 2009 International Conference on Computer Engineering and Technology

A Genetic Algorithm Approach for Optimum

Operator Assignment in CMS

Ali Azadeh, Hamrah Kor, Seyed-Morteza Hatefi

Department of Industrial Engineering and Center of Excellence for Intelligent Base Experimental Mechanics and
Department of Engineering Optimization Research, College of Engineering, University of Tehran, Iran
Aazadeh@ut.ac.ir,hkor@ut.ac,ir,s_m_hatefi@yahoo.com

Abstract—This paper presents a decision making approach
based on a hybrid GA for determining the most efficient
number of operators and the efficient measurement of
operator assignment in cellular manufacturing system (CMS).
The objective is to determine the labor assignment in CMS
environment with the optimum performance. We use The GA
for getting near optimum ranking of the alternative with
accordance to fitness function. Also, the GA approach is
performed by employing the number of operator, average lead
time of demand, average waiting time of demand, number of
completed parts, operator utilization and average machine
utilization as attributes, and Entropy method for determining
the weight of attributes. Furthermore, values of the attributes
procured by means of simulation.
Keywords - Genetic Algorithm, simulation, decision making,
Entropy method, Visual SLAM, CMS . Introduction
I. INTRODUCTION
Cellular manufacturing systems are typically designed as
dual resource constraint (DRC) systems, where the number
of operators is less than the total number of machines in the
system. The productive capacity of DRC systems is
determined by the combination of machine and labor
resources. Jobs waiting to be processed may be delayed
because of the non-availability of a machine or an operator
or both. This fact makes the assignment of operators to
machines an important factor for determining the
performance of cellular manufacturing (CM) systems and
therefore, the development of a multifunctional workforce a
critical element in the design and operation of CM
systems.[5]
II. METHODOLOGY
This paper presents a GA approach for select the
optimum operator allocation (for further information about
GA method see [4]). First, we generate most scenarios and
run them in the Visual SLAM software (for to obtain the
extinct answer, we consider number of run equal 30) [7].
Then, by GA solve the problem and specify the best
scenario. In the section fourth, we show empirical
illustration.
For solving MADM problems, it is generally necessary to
know the relative importance of each criterion. It is usually
given as a set of weights, which are normalized, and which
add up to one. The importance coefficients in the MADM
methods refer to intrinsic “weight”. Some works deserve
mention because they include information concerning the
methods that have been developed for assessing the weights
in a MADM problem. The entropy method is the method
used for assessing the weight in a given problem because,
with this method, the decision matrix for a set of candidate
scenarios contains a certain amount of information. In other
words, the entropy method works based on a predefined
decision matrix. Since there is, in scenario selection
problems, direct access to the values of the decision matrix,
the entropy method is the appropriate method. Entropy, in
information theory, is a criterion for the amount of
uncertainty, represented by a discreet probability
distribution, in which there is agreement that a broad
distribution represents more uncertainty than does a sharply
packed one. The Entropy idea is particularly useful for
investigating contrasts between sets of data.
Figure 1 presents the proposed simulation and multi
attribute approach for optimum operator assignment.
We begin by defining the system and its components. The
system is a set of permanent and temporary entities taking
into consideration the entities attributes and relationships
between them. This set is directed to achieve a specified
objective. Permanent entities, such as machines and
manpower in a manufacturing system, are named as a
server. Temporary entities are incorporated into the system
represented within the simulation model and they pass and
then leave the system. The attributes of each entity are
considered as an entity characteristic, which is required for
identifying the temporary entities.
Examples of entity characteristic are arrival time, part
number, and processing time of the system with each
temporary entity [8].

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 42

DOI 10.1109/ICCET.2009.211
 Data Collection
Input data
Define scenarios
Generation of outputs data
by computer simulation
GA for operator
assignment
Final assignment: By utilizing GA
Figure. 1: The overview of the integrated GA Simulation approach
III. PROBLEM DEFINITION
Manned cells are a very flexible system that can adapt to
changes in the customer’s demand or changes quite easily
and rapidly in the product design. The cells described in
this study are designed for flexibility, not line balancing.
The walking multi-functional operators permit rapid
rebalancing in a U-shaped. The considered cell has eight
stations and can be operated by one or more operators,
depending on the required output for the cell. The times for
the operations at the stations do not have to be balanced.
The balance is achieved by having the operators walk from
station to station. The sum of operation times for each
operator is approximately equal. In other words, any
division of stations that achieves balance between the
operators is acceptable. Operators perform scenario
movements in cell. Once a production batch size arrives to
the cell, it is divided to transfer batch sizes. Transfer batch
size is the transfer quantity of intra-cell movements of
parts. The ability to quickly rebalance the cell to obtain
changes in the output of the cell can be demonstrated by the
developed simulation model. The existing manned cell
example for a case model is presented in Fig. 2. [1]
43
D: Decoupler Station:
Operator
Operator movement when out of work
Operator movement with parts
Figure. 2: The existing manned cell example for the case model.
Alternatives consisted of reducing the number of
operators in the cell is as follows in details:
1. eight operators (one operator for each machine)
2. seven operators (one operator to only two machines
and one operator for each of others)
3. six operator (one by one operator to two by two
machines and one operator for each of others)
4. five operators (one by one operator to two by two
machines and one operator for each of others)
5. four operators (one by one operator to two by two
machines and one operator for each of others)
6. six operators (one by one operator to three
machines and one operator for each of others)
7. four operators (one by one operator to three
machines and one operator for each of others)
8. three operators (one by one operator to three
machines and one operator to only two machines)
9. five operators (one operator to four machines and
one operator for each of others)
10. three operators (one operator to four machines
and one by one operator to two by two machines)
11. three operator(one operator to four machines and
one operator to three machines and one operator to
one machine)
12. two operator(one by one operator to four by four
machines)
In simulation experiments, when the machines are assigned
to the operators, the cycle time of the bottleneck resource is
chosen close as possible as to the cycle time of the operator.
The developed model includes the some assumptions and
constraints as follows:
ƒ The self-balancing nature of the labor assignment
accounts for differences in operator efficiency.
ƒ The machines have no downtime for the simulated
time.
ƒ The time for the operators to move between machines
is assumed to be zero. The machines are all close to each
other.
ƒ The sum of the multi function operation times for each
operator is approximately equal
ƒ There isn’t any buffer for the station work.
As mentioned, outputs collected from simulation model are
the average lead time of demand, the average of waiting
time of demand, average operator and machine utilization
and number of completed parts per annum. The results of
the simulation experiment are used to compare the
efficiency of the alternatives. Each labor assignment
scenario considers 3 shifts which 1, 2 or 3 shift per day.
Moreover, a flexible simulation model is built by Visual
SLAM which incorporates all 36 scenarios for quick
response and results. Furthermore, ability to quickly
rebalance the cell to obtain changes in output of cell can be
demonstrated by simulation model.
So in the developed model different demand levels and part
types have taken into consideration. System performance is
monitored for different workforce levels and shift by means
of simulation. In simulation experiments, each of the
received demand’s parts has a special type and level. The
types of parts that cell can product and levels of demand
within cell determined as two and three in experiment. The
time processing of jobs for each of the station works is
related to the part type. The objective of scenarios consists
of reducing the number of operators in the cell is observing
how the operation is distributed among the operators. After
deletion of transient state, the 36 scenarios are executed for
2000 hours (250 working days, each day composed of 3
shifts, each shift consist of 8 hours of operation). Each
scenario also replicated 30 runs to ensure reasonable
estimates of means of all outputs could be obtained.
The Table I shows Output of the simulation model.
IV. APPLICATION OF GA MODEL
36 scenarios are selected as the core of our study. The
main structure of GA in this study is formed based on the
assumption which describes that the best scenario among
them could be a scenario with the indices in which each is
the maximum value of its possible values. Therefore, the
scenario 37 a scenario with the best possible attribute,
called our goal in the problem, which illustrates the
maximum present abilities in operator assignment. To
achieve the appropriate rank (array), every possible array,
44
composed 36 scenarios, is considered as a 64-bit
chromosome. Then, according with the sequence in the
chromosome, the total distance among the first scenario
which can be scenario from 1 to 36 and our goal (scenario
37), the second scenario with the first and the next with
upper ranked scenario are calculated respectively. The total
distance mentioned above is a dependent variable to
scenarios’ positions in the array. Consequently, in each
chromosome we can find a new value for total distance.
Undoubtedly, the best sequence of the scenarios is an array
which has the minimum total distance with high internal
cohesion among its scenarios. In fact, our fitness function is
a multivariate combination that its most prominent
components are total distance and variance. The above
concepts of genetics are achieved through a set of well-
defined steps as follows:
Step 1: Normalize the index vectors. The six attributes must
be normalized and have same order to be used in GA.
Indicesܺଵ , ܺଶ and ܺଷ have opposite order than the rest of
the indices.
Step 2: Standardize the indices ܺଵ – ܺ଺ .They are
standardized through predefined mean and standard
deviation for each index.
Step 3: Define the production module. This module is
defined to create and manipulate the 50-individual
population by filling it with randomly generated
individuals. Each individual is defined by 64-bit string.
Step 4: Define recombination module which enlists four
sections:
• Tournament selection operator chooses individuals with
probability 80% from the population,
This is considered as a popular type of selection methods in
GA. The basic concept in tournament is that the best string
in the population will win both its tournaments, while the
worst will never win, and thus never be selected. However,
in this study, the other kinds of selection methods named
sigma scaling and rank selection are considered in order to
best determine method.
• Uniform crossover operator which combines bits from the
selected parents with the probability 85%.
• Mutation operator consists of making (usually small)
alterations to the values of one or more
Genes in a chromosome
• Regeneration operator which is used to create 100-
individual generations.
Step 5: Define evaluation module:
The fitness function to determine the goodness of each
individual based on the objectives is defined by total
distance and variance that can be shown by:
 VI. SIGNIFICANCE
We have inspired the GA approach from the TOPSIS
methodology. As we know the TOPSIS methodology is
based on minimum distance from best scenarios and
maximum distance from the worst alternative. In this GA
approach our goal is to obtain total ranking in all scenarios
that their fitness function is the least distance between
scenarios. We are not looking for the best and the worst
In mentioned formula, ݀ ்௢௧௔௟ indicates total distance solution, but also the solution between these two point are
between adjacent scenarios [9]. important for us, this is the difference this approach and
The evaluating operator assesses the ability of each current methods. As you can see in the following Table our
chromosome to satisfy the objective. Therefore, our GA method satisfies this goal.
motivation to obtain the best array in this problem is to
minimize the fitness function mentioned above. After VII. ANALYSIS OF VARIANCE (ANOVA)
producing 1000 generations we reach the best fitness
ANOVA is used to evaluate the effects of the optimum
function value, 295.362, related to the chromosome which
operator assignment in CMS model. DMU’s efficiencies
can be shown by the sequence, at Table I.
are considered for GA model in comparison with DEA and
V. RESULTS PCA methods. First, it is tested whether efficiencies have
the same behavior in GA ( ߬ͳ ), DEA ( ߬ʹ ), PCA ( ߬͵ )
According to results, the scenario 12-2 that one by one models. Furthermore, it is tested whether the null
operator to four bye four machines with two shifts for day,
hypothesis Ho: ߤଵ ൌ ߤଶ ൌ ߤଷ is accepted. It is concluded
assigned to the system is the most efficient. The second
that the three treatments differ at ߙ ൌ ͲǤͲͷ . Furthermore,
best scenario is the scenario 12-1 that similar to scenari12-2
the least significant difference (LSD) method is used to
but the shift per day is one. The third scenario is the 9-1
compare the pairs of treatment meansߤଵ ǡ ߤଶ ܽ݊݀ߤଷ . That is
that five operators (one operator to four machines and one
Ho:ߤ௜ ൌ ߤ௝ for all i് ݆. The results of LSD revealed that at
operator for each of others) with the one shift per day.
ߙ ൌ ͲǤͲͷ ߤଵ ൐ ߤଶ ǡ ߤଷ and ߤଶ ൐ ߤଷ and hence treatment 1
(GA) produces a significantly greater efficiencies than
TABLE I. GA SOLUTION other treatments. The advantage of the
GA model with respect to efficiencies is shown in Table
II, III.
scenario rank Scenario rank
31 1 8 19
TABLE II. ANOVA
35 2 27 20
25 3 30 21 Sum of Adj Mean
Source DF F P
34 4 24 22 Squares Square
28 5 33 23 block 3 112.0
243.972 81.324 0.0
10 6 20 24 8
treatment 35 80.65 2.3044 3.18 0.0
26 7 11 25
error 105 76.18 0.7256
19 8 4 26
13 9 17 27 Total 143 400.8147
22 10 21 28
7 11 15 29 TABLE III. MULTIPLE COMPARISONS

23 12 5 30 Mean
(I) (J) 95% Confidence
Difference Std. Error Sig.
36 13 12 31 VAR VAR Interval
(I-J)
32 14 2 32 Lower Upper Lower Upper Lower
16 15 18 33 Bound Bound Bound Bound Bound
29 16 9 34 GA DEA 2.40335(*) .24957 .000 1.9099 2.8968
14 17 3 35 PCA 3.38752(*) .24957 .000 2.8941 3.8809
1 18 6 36 DEA GA -2.40335(*) .24957 .000 -2.8968 -1.9099
Score 295.362 PCA .98417(*) .24957 .000 .4907 1.4776
PCA GA -3.38752(*) .24957 .000 -3.8809 -2.8941
DEA -.98417(*) .24957 .000 -1.4776 -.4907
* The mean difference is significant at the .05 level.

45

VIII. CONCLUSION
In this study, we used GA, as a powerful method, to rank
the operator assignment problem based on the attribute
discussed in this paper. As shown in Table I, the two
Scenarios, 31 and 6 are determined as the best and the
worst respectively. Furthermore, GA approach able to
ranking the alternative by near optimum fitness function
.the GA method based on the minimum distance between
order of the scenarios, in fact, GA determine the best
solution by ranking with minimum distance. In this study,
for us, determining the best order of scenarios important
rather than just best and worst solution, in fact, this is the
main difference with the previous research [1, 2, 3, and 6].
For future effort proposed the hybrid simulation which
GA and simulation software operate simultaneous and
select useful scenario.
REFERENCES
[1] A.Azadeh, M. Anvari, “Implementation of Multivariate
Methods as Decision Making Models for Optimization of
Operator Allocation by Computer Simulation in CMS,” In
proceeding of the 2006 summer computer simulation
conference,2006,Calgary, Canada
46
[2] V.I. Cesani, H.J. Steudel, “A study of labour assignment
flexibility in cellular manufacturing systems”, Computers
& Industrial Engineering, vol.48, no.3, pp. 571–591. 2005
[3] T.Ertay, D. Ruan ,” Data envelopment analysis based
decision model for optimal operator allocation in CMS”,
European Journal of Operational Research, vol. 164, pp. 800–
810, 2005.
[4] S.E.Haupt, R.L.Haupt, ”Practical genetic algorithm”,
second edition, John Wiley, 1998
[5] D. J. Johnson, U. Wemmerlov, “On the relative
performance of functional and cellular layouts—an analysis
of the model-based comparative studies literature”,
Production and Operations Management, vol.5, no.4, 309–
334, 1996.
[6] M. Khouja, “The use of data envelopment analysis for
technology selection”, Computers and Industrial
Engineering, vol. 28, no. 2, pp. 123–132, 1995
[7] A. B. Pritsker, “Introduction to simulation and SLAM II”,
Fourth Edition, John Wiley and system publishing
Corporation, 1995.
[8] O. Shanian, O. Savadogo, “TOPSIS multiple-criteria
decision support analysis for material selection of metallic
bipolar plates for polymer electrolyte fuel cell”, Journal of
Power Sources, vol. 159, pp. 1095–1104, 2006
[9] V. Ebrahimipour, A. Azadeh, K. Rezaie, K. Suzuki,” A
GA–PCA approach for power sector performance ranking
based on machine productivity” , Applied Mathematics and
Computation, vol. 186, pp. 1205–1215, 2007.
 2009 International Conference on Computer Engineering and Technology
Dynamic Adaption in Composite Web Services Using Expiration Times
YU Xiaohao, LUO Xueshan, CHEN Honghui
Department of Information System and Management
National University of Defense Technology
Changsha 410073, China
yxhtgxx@yahoo.com.cn
Abstract—A key challenge in composite web processes is that
the quality of participating services changes during the life
time of the process. It is necessary to find an available
equivalent service to substitute the failed component which has
been made unavailable. In this paper different from the
previous method based on recompositing the web process, we
present an approach of discovering the “same skill” web
services to replace the expired one by using the service
expiration times.
Keywords-Web service adaption; Service matching;
Expiration times
I. INTRODUCTION
In volatile environments[2] where the Quality of Service
(QoS) items such as deadlines, quality of products, and cost
of services may change frequently, the composite web
process may become invalidate if it not update with the
changes. As an example, consider an information fusion
application. There are a number of web services (WS) for
giving information about a certain target. To complete a task
we may design a web process using web service composite
method [5,7,10]. When the web process executes, the
previous selected web service may not satisfy the designer’s
requirement because of the unstable communication link,
target change or service unavailable. A straightforward
approach to solve the problem is to try a different process for
achieving the same goal. It is time consuming and ignores
the fact that different service providers may provide “same
skill” [11] web services. In these cases, we suppose that the
failed web services may be substituted by a new one
identified in UDDI registry center. In this context, we
considere the “task” as a capability offered by a web service.
The replacement service should have the “same skill” with
the failed one i.e. to have same functionality and QoS.
By using the insight that service providers are often able
to guarantee that the QoS items will persist for some amount
of time [2,4], the paper presents a process adaption
framework and algorithm that can support dynamic service
selection of a failed task.
II. RELATED WORK
Recently, researchers are increasingly turning their
attention to managing processes in volatile environments. Au
et al. [2] provide a T-correct algorithm to ensure the web
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.146
HU Dan
Airforce Engineering University,
Xi’an Shaanxi 710077, China
process to be unchanged for T times. The web process will
be recomposed after T times. It is time consuming and may
lead to frequent unnecessary computations. In [4], the value
of changed information with expiration times (VOC) and
Query-Cost are used to judge if it is deserved to choose a
new plan when the service parameter changes. This method
can assure to select the best plan and reduce the computation
times, but it also need to recompose web services when
compute the expected cost of following the optimal policy,
π ∗ .In [3],the authors use pre-defined event-condition-
action rules as a workflow adaptation strategy. When a
change occurs in the environment, the event will be triggered
to adapt the workflow. While this method proposes a usually
used basis for performing contingency actions and has been
implement in many BPEL execution engine, this method also
has limit that in unpredictable conditions or complex
workflows it can’t list all possible actions and preconditions
that may arise. [11] offers an architecture of Relevance-
Driven Exception Resolution but there is no algorithm of
how to select the “same skills” services. Doshi et al. [12]
offers a solution by using Bayesian learning. However, the
Bayesian learning method does not have any advantage in
the time consuming of updating the parameters, and the
approach may bring on plan recomputations that can’t bring
about any change to the failed Web process.
III. DEALING WITH WEB PROCESS USING EXPIRATION
TIMES
Existing approach for Web Service Composition
formulate the problem in different ways, depending mainly
on how to select the optimal web services to meet the user’
requirements. However, because the characteristics of the
service providers who participate in a web process may
change during the life-cycle of a process [11], these web
process systems are currently ill-suited for dealing with
exceptions effectively. To address the changes of web
services, [2, 4] use the insight that service providers are often
able to guarantee that their reliability rates and other quality-
t
of-service parameters will remain fixed for some time exp ,
after which they may vary, While this is similar in concept to
t
our approach. WS providers may define exp in a WS-
Agreement document. In this paper, instead of finding the
optimal policy and calculate the query-cost, we take an
47
architecture and an algorithm to find “same skill” services in
order to replace the one that expired in the composite process
flow [8].
A. Definition
We assume that Task1 is already functioning in BPEL
execution. If Task1 expires then we will choose Task2 to
replace it. To find “same skill” service Task2, we define two
matching rules as follows:
(1)Input/output matching: Inputs and outputs are
functional attributes of a web service. Note that descriptions
of inputs and outputs go beyond the specifications employed
in a WSDL specification, since the WSDL is machine-
oriented types, whereas the input/output matching includes
higher-level semantics. To formally express these semantics,
domain ontologies or do-main taxonomies can be employed
[6,9]. In the web service replacement, the operation names
can different but the input and output must semantic
equivalent. Motivated by functional match and current
concept similarity techniques [1], we set out from the
following two rules to define a Input/Output semantic match.
I I
In these two rules, task 1 and task 2 are the input set of Task1
O O
and Task2, task 1 and task 2 are the output set of Task1 and
Task2. These two rules which make the Input set of Task1
equal or contain Task2’s and also make the Output set of
Task2 equal or contain Task1’s semantically to ensure that
Task2 adapt to the web process context of the failed web
service Task1.
Rule 1:
for each i ' ∈ I task2 there exist an i ∈ I task 1 and i ' ⊇ i ;
here i ' ⊇ i indicate that i = i ' or i is the subclass of i '
Rule 2:
for each o ∈ Otask1 there exist an o ' ∈ O task2 and o ⊇ o ' ;
here o ⊇ o ' indicate that o = o' or o ' is the subclass of o
(2)QoS matching: There is a set of metrics that measure
the Quality of Service. These kinds of QoS properties have
been emphasized on previous research works [5], such as
availability, successful execution rate, execution cost,
reputation, frequency. Two providers of same task
supporting the same generic Input and Output may have
different values of QoS. Consider two web services that are
functional equivalent, i.e. they have the same Input/Output
attributes. Also, suppose Task1 which has response
availability A1 and cost C1 has already functioned in BPEL
execution, then when Task1 is failed to complete, if the
following condition is true Task2 can replace Task1 in the
web process:
(Input/Output matching) AND (A2 ≥ A1) AND (C1≥C2)
B. Adaption Framework
Fig.1 shows the proposed service selection framework
which is designed to support the dynamic adaption in
composite web services using expiration times. In the Figure,
the framework is composed of 5 modules: Service
Monitoring Module (SM), Failed diagnosing Module (FD),
Failed Recovery Module (FR), Composition service
Management Module (CSM) and Execution Engine Module
48
(EE). When a web service expired and the BPEL
orchestrator decides to resolve it by finding and replacing the
task that stopped the web process, the sequence of process
performed are presented below step by step. In step1 when
SM module detects a BPEL execution exception, it sends a
signal/message tagged by the failed component id to EE
module to stop the execution. In step 2 FD module analyses
causations of the exception and then send the failed service’s
characters (Input and Output parameter, QoS and so on) to
the CSM module which will update the database of
replacement policy. Here the replacement policy can be pre-
defined policies or “default”[11] (all attribute values of the
replacement task need not only to be the Input/Output
matching but also QoS “better” than the corresponding
attributes of the failed task), “minimize-cost”, “maximize-
reputation” and so forth. To keep the “default” not changing
with time, it will be updated only when a component failed at
the first time. In step 3 the CSM module returns the
replacement web service characters. Using this message in
step 4, the FR module decides the execution path and selects
the required web service in UDDI to replace the failed one.
The EE module executes the new web process in step 5.
Fig. 1: Service Replacement Framework
C. Algorithm
In Fig. 2, we show the algorithm for adapting the web
process in volatile environment. When there is an exception,
The algorithm will select the “same skill” web service to
replace the expired one (line 4-17). Notice that a web service
might expire while query the “same skill” web service, we
must anticipate this and add those services in advance to the
set, ES (in line 14). In line 11, the algorithm invokes the
∗
procedure which finds the service s that can assure T times
fixed to replace the expired one. The parameter T is
*
designated by the police. In line 12, the t[s ] is set zero,
∗
because the service s is the newly selected service. In line
t
14, the times query is added to the other executing services,
during which other services may expire. If there is no
expired services and the web process runs successfully, each
tresponse
service’s execution times will be added by (In line
tquery
18-20). In the algorithm the times is the times spend to
 t recomposition with the component increasing. That is
find the “same skill” web service and the times response is
the times spend during the web process successfully
executing.
Fig. 2: Service Replacement algorithm
IV. EXPERIMENT
In this section we simulated our algorithm using a web
process scenerio with several web service components in
Fig. 3. For each component we randomly created some
relevant web services by using the OWLS_TC V26. We
change some web service description profiles such as giving
each service an expiration times randomly from 1s to 20s,
building the ontology relationship between web services’
interface description. In our experiment, we randomly
selected a web service for each component as an
initialization. We compare the successful executing times
with the method based on process recomposition.
Fig. 3: Experiment Scenerio
The result is shown in Fig. 4(a). It can be seen that the
average execution times of the web process using our
approach are longer than the method based on process
49
because the recomposition of a web process will consume
more and more time when the component number of a web
process increases. However the recomposition method may
be more efficiency, when there is no satisfied web service for
the failed component. Under this condition we can avoid the
time of finding the failed one by modify the Policy.
In Fig. 4(b), we compare the runtimes taken in effectively
executing the web process when we increase the average
web services expiration times. Notice that the successful
execution times of a web process increase with the expiration
times increasing. That is because the shorter are the web
services’ expiration times the longer times that spend on
finding replacements or recompositing a web process.
(a) The components change
(b) The expiration times change
Fig. 4: Successful execution times of a web process using different method
In the experiment we can find that out approach is
excellent in increasing the execution times of a web process
when there are replacements of the failed web services.
V. CONCLUSION AND FUTURE WORK
In this paper we deal with dynamic adaption of web
process in volatile environments. To reuse the available web
services in a web process and increase it’s execution times,
we propose an intelligently web process adapting method by
finding the failed services’ replacement. First we define the
two conditions that must be satisfied when we select the
“same skill” web services. Then we present the process and
algorithm of our approach to deal with the expired web
service.
Our future work is to abstract the Business Process
Language (BPEL) to make the process language support the
dynamic adaption of the web process. We will also study the
storage strategy of “same skill” web services to accelerate
the searching times in UDDI registry center.
 REFERENCES
[1] A.. Fernandez, S. Ossowski: Exploiting Organisation Information for
Service Coordination in Multiagent Systems: Proc. Of 7th int. Confon
Autonomous Agents and Multiagent System ,May,12-16., 2008. 257-
264(AAMAS 2008).
[2] T. C. Au, U. Kuter, and D. S. Nau. Web service composition with
volatile information. In International Semantic Web Conference, 52-
66, 2005.
[3] R. Muller, U. Geriner, and E. Rahm. Agentwork: a workflow system
supporting rule-based workflow adaption. Journal of Data and
Knowledge Engineering., 51(2):223-256, 2004.
[4] J. Harney, P. Doshi. Speeding up Adaption of Web Service
Compositions Using Expiration Times. International World Wide
Web Conference Committee May 8-12, 2007. 1023-1032(WWW
2007).
[5] Liangzhao Zeng, Boualem Benatallah. Quality Driven Web Service
Composition. International World Wide Web Conference Committee.
May 20-24, 2003(WWW 2003).
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[6] K. Sivashanmugam, K. Verma, A. Sheth,.: Adding semantics to web
services standards. Proceedings of the 1st International Conference on
Web Services (2003)
[7] A. Charfi, M. Mezini,.: Aspect-Oriented Web Service Composition
with AO4BPEL. Proceedings of the European Conference on Web
Services (2004)
[8] P. Wohed, M. Dumas,.: Analusis of Web Services Composition
Language: the Case of BPEL4WS,200-215. Spriner-Verlag Berlin
Herdelberg.
[9] OWL services Coalition: OWL-S: Semantic markup for web
services(2004) OWL-S White Paper
http://www.daml.org/services.owl-s/1.1/owl-s.pdf.
[10] D. Nau, T.C. Au, O. Ilghami, W. Murdock, D. Wu,: SHOP@: An
HTN planning system. JAIR 20(2003)379-404.
[11] Kareliotis Christos, Vassilakis Costas, Georgiadis Panayiotis,:
Toward Dynamic Relevance-Driven Exception Resolution in
Composite Web Services.
[12] P. Doshi, R. Goodwin, R. Akkiraju, and K. Verma.: Dynamic
workflow composition using markov decision process. Journal of
Web Services Research(JWSR), 2(1):1-17,2005.
 2009 International Conference on Computer Engineering and Technology

An Emotional Intelligent E-learning System Based on

Mobile Agent Technology

Zhiliang Wang, Xiangjie Qiao, Yinggang Xie

School of Information Engineering
University of Science & Technology Beijing
Beijing, China
wzl@ies.ustb.edu.cn, qxj7711@163.com, yinggangxie@163.com

Abstract—The emergence of the concept “Affective
Computing” makes the computer’s intelligence no longer be a
pure cognitive one. Current e-learning systems cannot instruct
students effectively since they do not consider the emotional
state in the context of instruction. In this paper, we construct
an emotional intelligent e-learning system based on mobile
agent technology. A dimensional model is put forward to
recognize and analyze the student’s emotion state and a virtual
teacher’s avatar is offered to regulate student’s learning
psychology with consideration of teaching style based on his
personality trait. A “man-to-man” learning environment is
built to simulate the traditional classroom’s pedagogy in the
system.
Keywords-emotional intelligent; two-dimension model;
mobile agent; virtual teacher
I. INTRODUCTION
Learning is one of the cognitive process affected by one’s
emotional state [3]. While Current E-learning system is
referred to as cognitive education. Though Intelligent
Tutoring Systems (ITS) can provide individualized
instruction by being able to adapt to the knowledge, learning
abilities and needs of each individual student, they are still
not as effective as one-on-one human tutoring. In a word,
nowadays learning systems are lack of emotional interaction
in the context of instruction. They cannot simulate the
traditional classroom scenario: The teacher and students can
face to each other in the class. As Wu Qinglin [4] said, an
educational psychologist, an effective individualized learning
system should be not only intelligent but also emotional. So
E-learning systems should have emotional intelligence to
meet the need. In our project, we will take the student’s
emotional state into account by analyzing his learning
psychology or motivation. Researchers in neurosciences and
psychology have found that emotions are widely related to
cognition, they exert influences in various behavioral and
cognitive process, such as attention, long-term memorizing,
decision-making, etc[6][7]. Rozell and Gardner’s [8] study
pointed out that when people have negative attitudes towards
computers, their self-efficacy toward using them reduces,
which then reduces their chances of performing computer-
related tasks well compared to those with positive attitudes
towards computers. This research also emphasized that
individuals with more positive affect exert more effort on
computer-related tasks. The Artificial Psychology theory put
forward by Professor Zhiliang Wang [1] proposes imitating
human psychology activities with artificial machines
(computers, objective function algorithm) by means of
information science. It offers the theoretical and modeling
basis for realizing the emotional intelligence in e-learning
systems. In this project, we construct a virtual teacher agent
to communicate with students in the process of learning. The
virtual teacher can respond to student’s current state to
regulate their emotion to optimum by words or expressions.
Thus we can establish a harmonious environment for human
and computer’s interaction, with an emphasis upon “man-to-
man” instruction.
Moreover, to process largely emotional computation on
the web, system would occupy the bandwidth frequently.
Therefore we shall consider the cost of the network, the
whole system’s basic architecture and technological support.
Traditional paradigms for building distributed applications
are RPC and most recently its object cousins RMI and
CORBA. For this class of paradigms, functionality of
applications is partitioned among participating nodes.
Different participants use message-passing to coordinate
distributed computation. Computation itself is partitioned,
participants exchange intermediate results and other
synchronization information. While for Mobile Agent
paradigm, computation is migrated toward resources. This
will largely reduce the data flow in the Internet. It also has its
particular characteristics of Mobility, Autonomous,
Personalized and Adaptive to diminish network’s load,
conquer the network’s latency, adapt to a new environment
dynamically, and improve the system’s robustness and fault
tolerance. In this paper, we try to experiment with such
paradigm---Aglet. Aglet is the shorthand for agent plus
applet. It provides us an infrastructure for building
distributed applications based on mobile agent technology.
It’s easy for us to implement the information’s gathering and
retrieval by utilizing its mobility property.
II. THE ARCHITECTURE BASED ON MOBILE AGENT
The system’s architecture is as shown in Figure1. Next
we will introduce every server and component’s
functionality and role in the system.

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DOI 10.1109/ICCET.2009.17
 III. STUDENT’S LEARNING PSYCHOLOGY MODEL
We often try to estimate a person’s internal state by
observing facial expressions, a visible change in heartbeat,
voice inflections and even eye and other body movements.
To interact most effectively, it is often useful for us to try to
gain insight into “invisible” human emotions and thoughts
by interpreting these non-verbal signals. Teachers use these
non-verbal signals to make judgments about the state of
students to improve the accuracy and effectiveness of
interactions. This non-verbal communication is completely
lost when people communicate with computers [11].
In our research, the emotion of a student is mainly
recognized by facial expression captured by a camera. By
Figure 1. System’s architecture analyzing the facial expression, we can get the student’s
learning psychology. Here, we utilize a two-dimension
The E-learning Service Center Server (ESCS) is in model to describe a student’s emotion. That is interest level
charge of the general management of the system. It will and attention level.
initialize the entire environment (e.g. Classroom Server, Interest level is mainly depended on the distance between
Student Server’s Creation), monitor other servers’ activity, student and computer. General speaking, if the student is
provide registration service and cooperate with other agent interested in the instruction he will sit more nearer the
systems. computer to care the learning material. By computing the
When a student enters our system, the system will firstly student’s face area obtained by the camera, the interest level
provide three services in a web page for him to choose as he can be achieved [13] (See Equation (1)). Here, the Ei is the
likes. One is “Class”, which is provided for students to have
classes. The second is “Question and Answer” and the third x is the face area, the xF max is the
Interest level, the
is “Online Test”. These three functions are all implemented
based on mobile agent technology. maximum face area can be detected, and the x F min is the
When students want to have classes with our system, the minimum face area can be detected.
information about the courses and teachers will be offered. If ⎧ 0 x ≤ xF min
the student is the first time to use our system, he should
⎪ 1
⎪⎪⎛ x − xF min ⎞2
firstly register his information and wait for the system’s
verification. After the student passes the verification, he can
Ei = ⎨⎜ ⎟ x ≤x≤x
start his class with a virtual teacher say hello to him. At the ⎜ x − x ⎟ F min F max (1)
same time, the ESCS will send Student Server Management ⎪⎝ F max F min ⎠
Agent (SSMA) to create the Student Agent (SA) and manage ⎪ 1 xF max ≤ x
other agents in Student Agent Server. Each student will ⎪⎩
match one SA respectively. On the other hand, the ESEC If the student pays much attention to the instruction, his
will create Teacher Agent and send it to the Teacher Agent pupil will become much bigger than usual. Attention level is
Server. And the Classroom Server will receive the Mobile to detect whether the student is learning carefully enough
Student Agent’s request to dispatch a Virtual Teacher Agent [13] (see Equation (2)). Here, the E a is the Attention level,
(VTA) to begin a class. Information Gathering Agent (IGA)
will start to gather pedagogical information and provide the x is the space between eyelids, the pi is current
useful pedagogical tactics for VTA through data mining
mechanism. Here the VTA shall do four things in the context interest level, the xe max is the maximum of averaged pupil
of instruction. First, Listening to the student’s changing state size and the xe min is the minimum of averaged eyelids’
especially emotional state. Second, analyzing the student’s
emotional and cognitive data to decide next action. Third, distance.
Regulating the student’s state by doing some activities such ⎧ 0 x ≤ xe min
as words expression or facial expression. Fourth, Receiving ⎪ 1
student’s question and return the answer to the student. VTA ⎪ ⎛ x ⎞ 2
tell the ESEC to create a Mobile Query Agent (MQA) and ⎪⎪⎜ p − xe min ⎟
dispatch it to every campus aglet server that has registered in Ea = ⎨⎜ i ⎟ xe min ≤ x ≤ xe max (2)
the ESEC and the query result will then be fetched and ⎜ xe max − xe min ⎟
arranged well to the students.
⎪⎜ ⎟
⎪⎝ ⎠
⎪ 1 xe max ≤ x
⎪⎩
Finally we get the general learning psychology of the
student by Equation (3). Here, the C is the student’s

52
cognitive evaluation value, α, β, γ is the weight of Ei , E a of one emotion x changes to another emotion y. On the
other hand, the change of teaching style can be obtained by
and C respectively.
(6). For each dimension of personality according to the
P = αEi + β Ea + γC 2 (α + β + γ = 1) OCEAN model, θ ep defines a weight of personality factor
2 2
(3)
p for emotion e. For example, θ joyO is the weight of
IV. VIRTUAL TEACHER’S EMOTION MODEL personality factor O for the joy emotion. And Wi denotes
Positive affects are fundamental in cognitive the percentage of each dimension of personality. Finally we
organization and thought processes; they also play an
important role to improve creativity and flexibility in can get the new emotion esn +1 by equation (7).
problem solving [8]. Reciprocally, negative affects can
block thought processes [9]. It is an illusion thinking that TABLE I. TEACHING STYLE VS. OCEAN
learning environments that don't consider motivational and OCEAN Teaching Style
emotional factors are adequate. Since there is not a real Openness Active
teacher or a real classroom in an e-learning environment, we Conscientiousness Responsible
Extraversion Energetic
construct a teacher’s avatar as a virtual teacher to analyze Agreeableness Easy-going
student’s emotion state and give proper regulation to adjust Neuroticism Rigorous
their negative emotion.
The virtual teacher’s emotional model contains three
modules: Sense Module (SM), Thinking Module (TM) and ⎡ p xx p xy p xz ⎤
⎢ ⎥
Behavior Module (BM). p s = ⎢ p yz p yy p yz ⎥ (4)

STB =< SM , TM , BM > ⎢ p zx p zy p zz ⎥⎦

⎣
⎡ p xx p xy p xz ⎤
The virtual teacher will listen to the student’s learning ⎢ ⎥
state through VTA by Sense Module. Once the SM senses Δes = e p s = [x s , y s , z s ]⎢ p yz
n
s p yy p yz ⎥ (5)
the changing it will inform the TM to analyze the student’s ⎢ p zx p zy p zz ⎥⎦
state and decide how to regulate the student’s state by doing ⎣
some activities like words or facial expression. Thinking ⎡θ aa θ ab θ ac ⎤
Δ p = [Wa , Wb , Wc ]⎢⎢θ ba θ bb θ bc ⎥⎥
Module is also a control module, it is made up of knowledge
and personality. Knowledge is just as the regulation rule; it (6)
depends on the psychological and pedagogical theory and ⎢⎣θ ca θ cb θ cc ⎥⎦
experience. As every person has his own personality, we
take the virtual teacher’s teaching style as his personal trait. esn+1 = esn + Δes + Δp = esn + esn ps + Δp = [xl , yl , zl ] (7)
For example, some teachers are rigorous while some are
easy-going with his students. And the words and phrases
will be different according to their teaching style as well.
For instance, when the teacher wants to give the student
suggestion, rigorous teacher may express as “You
should…”, “let’s…”, while the easy-going teacher will say
“I suggest that you…”, “Perhaps you would like to…” or
“Maybe you could…” etc.
The change of the virtual teacher’s emotional state
depends on two factors: external stimuli and teaching style.
External stimuli are the student’s new emotional states.
Meanwhile, we adopt the OCEAN model [12] to map the
teaching style (see Table 1). To effectively regulate the
student’s state, three couple (six) emotions are employed in
the model: joy and angry, pride and sad, hope and
disappointment (see Figure 2). The virtual teacher’s avatar Figure 2. Various emotions for the virtual teacher
is revised from the Ken Perlin’s applet program
(http://mrl.nyu.edu/~perlin/). Suppose the probability of V. TESTBED
changing Emotion is Ps (see (4)), and then the change of the
We have constructed a prototype and developed some
emotion will be (see (5)). The p xy means the probability agent modules to test the effectiveness of the architecture.

53
Figure 3 (a) shows the classroom server’s management
interface and (b) is the student’s learning window. The
virtual teacher will respond to the student’s emotion state
through verbal or non-verbal expressions. Students can ask
to the teacher whenever they are confused with the
instruction and answer the questions forwarded by the virtual
teacher as well.
VI. CONCLUSION AND FUTURE WORK
In this paper, we have constructed an emotional
intelligent e-learning system based on mobile agent
technology. The system’s functionalities include: (1)
Students can quickly find the course-related information and
the data share is well implemented; (2) Students can
communicate with the real teacher or the virtual teacher
emotionally; (3) Students can do online testing to test
themselves independently. In the future work, we will revise
the details of our system and find some school students to
test the system’s performance.
ACKNOWLEDGMENT
This paper is supported by the National Natural Science
Foundation of China Grant #60573059 and Natural Science
Foundation of Beijing Grant #KZ200810028016.

(a) Classroom server’s management window REFERENCES

[1] Wang Zhiliang. Artificial Psychology-a most Accessible Science
Research to Human Brain. Journal of University of Science and
Technology Beijing, 22 (5), 478-481, (2000).
[2] Picard, R. W.. Affective Computing. MA: MIT, 1997
[3] D. Goleman. Emotional Intelligence. New York: Bantam Books,
1995.
[4] Wu Qinglin. Pedagogical Psychology---to Pedagogue. Shanghai:
East China Normal University Press, 2003.
[5] E.J. Rozell, W.L. Gardner. Cognitive, motivation, and affective
processes associated with computer-relatedperformance: a path
analysis. The Journal of Computers in Human Behavior,16 (2000),
199–222, 2000.
[6] Damasio, A. (1994) Descartes Error – Emotion, Reason and the
Human Brain. Putnam Press, NY.
[7] Frasson, C. Using cognitive Agents for Building Pedagogical
(b) Student’s learning window Strategies in a Multistrategic Intelligent Tutoring System. Bayonne:
Deuxième journée Acteurs, Agents et Apprentissage, 1998.
Figure 3. System’s interface
[8] Isen, A. M. Positive Affect and Decision Making. Handbook of
Emotions, 2000.
In addition, we have carried out the incremental testing
[9] Idzihowski, C. Baddeley, A. Fear and performance in novice
to test the system’s scalability. Table 2 shows the time of parachutists. Ergonomics, 30, 1463-1474, 1987.
feedback when query all the sub agent systems. [10] Reed, G. F. Obsessional cognition: performance on two numerical
tasks. British Journal of Psychiatry, 130, 184-185, 1977.
TABLE II. TESTING RESULT [11] Sarrafzadeh, A. Hosseini, H.G. Fan, C. Overmyer, S.P. Facial
subsystem’s time for obtaining the time for obtaining all expression analysis for estimating learner’s emotional state in
number first data data intelligent tutoring systems. Proceedings of the 3rd IEEE International
Conference on Advanced Learning Technologies,2003, pages 336-
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337.
3 13.7s 17.6s
4 14.2s 21.3s [12] P.T. Costa and R.R. McCrae. Normal personality assessment in
5 14.5s 22.5s clinical practice: The NEO personality inventory. Psychological
Assessment, (4): 5-13,1992.
[13] Yinggang Xie, Zhiliang Wang, Lijuan Wang, An Ping, Xiangjie
Qiao. Affective Recognition Based on Approach-Withdrawal and
Careness, ISAI’062006-08.

54
 2009 International Conference on Computer Engineering and Technology
Audio Watermarking for DRM based on Chaotic Map
B. Lei
School of Electrical and Electronic Engineering,
Nanyang Technological University,
Singapore, 639798
leib0001@ntu.edu.sg
Abstract—This paper focuses mainly on using chaos
encryption to protect AVS audio efficiently. A novel
digital watermarking approach is introduced for audio
copyright protection. A meaningful gray image embedded
into digital compressed audio data is researched by
quantizing MDCT coefficients (using integer lifting
MDCT) of audio samples based on chaotic map. The
proposed watermarking algorithm can extract the
watermarking image without the help from the original
digital audio signal. The experimental results show that
the proposed digital watermarking approach is robust to
the audio degradations and distortions such as noise
adding, compression, low pass filtering, resampling, and
re-quantization. The digital watermarking scheme is able
to provide a feasible and effective audio copyright
protection.
Keywords-component; formatting; style; styling; insert
(key words)
I. INTRODUCTION
Digital audio watermarking embeds inaudible
information into digital audio data for the purposes of
copyright protection, ownership verification, covert
communication, and/or auxiliary data carrying. With
the widespread infusion of digital technologies and the
ensuing ease of digital content transport over the
Internet, Digital Rights Management (DRM) of
multimedia data have therefore become of a critical
concern. Audio Video coding Standard (AVS) is
China's second generation source coding\decoding
standard with fully Intellectual Properties. As the sixth
part of AVS standard, AVS DRM aims to offer the
universal and open interoperable standard for various
DRM requirements in digital media industry. Audio
content protection also plays an important role in many
digital media applications.
Digital watermarking technique has received a great
deal of attention recently in industry and academic
community, but currently digital watermarking
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.196
I. Y. Soon
School of Electrical and Electronic Engineering,
Nanyang Technological University,
Singapore, 639798
schemes mainly focus on image and video copyright
protection [1-3]. However, audio watermarking
methods have, to our knowledge, not been studied
much. Early work on audio watermarking embedding
achieved inaudibility by placing watermarking signals
in perceptually insignificant regions. One popular
choice was the higher frequency region [4]. In some
systems, the Fourier transform magnitude coefficients
are replaced with the watermarking sequence. Some
previous works on MP3 are always based on the
frequency-selective method that selects different
frequency coefficients to be encrypted. For a review of
the early watermarking schemes and main
requirements of a watermarking scheme, the reader
may consult [5].
The majority of watermarking schemes proposed to
date use watermarking generated from pseudorandom
number sequences. As a single seed will reproduce the
same sequence of numbers each time the generating
function is iterated. Pseudorandom sequences have an
advantage in that they can be easily generated and
recreated. Chaotic functions have been used to
generate watermarking sequences [6-8]. Similar to the
pseudorandom number sequence, a single seed (along
with an initial value) will always reproduce the same
sequence.
Another trend in digital audio watermarking is to
combine watermarking embedding with the
compression or modulation process. The integration
could minimize unfavorable mutual interference
between watermarking and compression, especially
preventing the watermarking from being removed by
compression. In one scheme, watermarking embedding
is performed during vector quantization. The
watermarking is embedded by changing the selected
code vector or the distortion weighting factor used in
the searching process. Although the audio
watermarking methods described above have their own
features and properties, they share some common
problems as follows:
55
 (l) The amount of hidden information is small. For on the mapping. The chaotic sequence has the
example, some of them can only embed pseudo- character which is sensitive to initial values and
random bit sequence or binary image. random alike. The simplest chaotic sequence is one
(2) The detection procedure needs the original dimensional logistic map which is unimodal and
digital audio signals. defined as xk 1 P xk (1  xk ), xk  (0,1) , where
(3) The robustness and invisibility are not so good, 0  P d 4 is bifurcation parameter. In practice, when
for the Human Auditory System (HAS) is not taken
into account adequately.
P is restricted within narrow limits, i.e. 3.57  P d 4 ,
(4) Most of transform domain digital audio xk is in chaotic state, which can be used to select
watermarkings have round-off error problems, because embedding points randomly.
they utilize the floating point calculation.
In this paper, we introduce a new adaptive digital
audio watermarking algorithm based on the III. BINARY IMAGE WATERMARKING EMBEDDING
quantization of MDCT coefficients (Integer Lifting AND EXTRACTION
MDCT). The features of the proposed algorithm are as
follows:
(1) Embedded watermarking as meaningful gray A. System Model
image.
(2) Blind detection without resorting to the original There are three ways to embed the binary image as
digital audio signal or any other side information. watermarking to audio A:
(3) Integer lifting MDCT to solve round-off error A 'i Ai  D xi (1)
problems. A 'i Ai (1  D xi ) (2)
This paper proceeds as follows. Section 2 describes
the proposed watermarking generation. In Section 3, A 'i Ai e axi (3)
the watermarking embedding and detection procedures Where Ai is the audio transform coefficient, and a
are provided. Section 4 shows the experimental results
and discussions. Conclusions are given in Section 5. is embedding stretch factor. We adopt equation (2) as it
can be adaptable to Ai change. Our proposed
embedding and extraction scheme can be seen in
II. GENERATION OF WATERMARKING WITH
Figure 1.
CHAOTIC MAPS
Original AVS Compressed Watermarked AVS
audio encoder bitstream audio decoder
The majority of watermarking generation schemes
proposed to date use a pseudorandom number
Scrambling with Watermarking Watermarking Extracted
generator to create a watermarking sequence which is chaotic sequence embedding
key
extraction watermark

embedded in the cover work. A chaotic function is

unpredictable, indecomposable and yet contains Figure 1. Proposed embedding and extraction scheme
regularity which is sensitive to initial conditions. The
motivation for using a chaotic function to generate a
watermarking is that a single variable seeding the
chaotic function will always result in the same output B. Pre-processing of Image Watermarking
(mapping) when certain constraints or initial conditions
are placed on the mapping. The use of chaotic If the watermarking is visually differentiable
functions for the generation of watermarking has been M 1 u M 2 binary image which can be represented as:
previously proposed [9-11], for example, the Bernoulli W {w(i, j ), 0 d i  M 1 , 0  j  M 2 } (4)
Map, Skew Tent Map and also logistic Map. In this
study, the logistic map was selected, as it is a well- where w(i, j )  {0,1} . Because W is a two-dimension
behaved chaotic function which has been extensively binary image, we should do some dimension reduction
studied. operation if we want to embed it into audio Ai . The
The watermarking is generated using deterministic
two-dimension image is converted to one-dimension
chaotic maps. The primary advantage is that it is
sequence with equation (5).
possible to investigate the spectral properties of the
V {v(k ) w(i, j ), 0 d i  M 1 , 0 d j  M 2 , k i * M 2  j}
resulting watermarking. A single variable, seeding the
chaotic function will generate the same output (map) (5)
when certain constraints or initial conditions are placed

56
To eliminate the similarity in the neighboring elements
and enhance the watermarking robustness, we adopt
the linear recursive shifter to sort all the elements in
the random sequences.
V p sort (V ) {v p (k ) v(k '), 0  k , k '  M1 M 2 } (6)
After sorting, the k ' elements in V will shift to K
elements in the v p . To further improve the
watermarking anti-attack ability, we modulate the
watermarking sequence by chaotic signal and generate
chaotic sequence {r (l )} with secret seed K, then use
{r (l )} to modulate {v p (k )} and get {s(l )} with m
modulation factor seen as follows.
s (l ) v p ( «¬1/ m »¼ k ) † m(1)
v p ( k ), r (1)  {0,1}(0 d k  mM 1 M 2 )
(7)
Chaotic modulation has the advantages of high anti-
interference ability, low-power spectral density and
higher confidentiality. But it is based on the increase of
the signal capacity to improve the robustness.
C. Watermarking Embedding
There is original audio signal with L sampling data
and M 1 u M 2 binary image. In this paper, we employ
MDCT transform to embedded watermarking W. The
detailed embedding process can be shown as follows:
(1) Segmentation: Segment M 1 u M 2 data with
length N from A and denoted as:
Ae { Ae (k ), 0 d k  ( M 1 u M 2 )}  (8)
Where Ae (k ) is expressed as:
Ae (k ) {a (k u N  i ), 0 d i  N } (9)
(2) Watermarking dimension reduction: Because the
image is two-dimension, it should be converted to one
dimension shown in equation (9) before it is embedded
into audio signal.
V {v(k ) w(m1 , m2 ),
(10)
0 d m1  M 1 , 0 d m2  M 2 , k m1 u M 2  m2 }
After the dimension reduction, w(m1 , m2 ) in the image
W can be expressed with v (k).
(3) Pseudo random sequence sorting: To eliminate
the correlation in the neighboring elements, we adopt
the linear recursive shift register to sort all the elements
in V to improve the robustness.
V p sort (V ) {v p (k ) v(k '), 0 d k , k '  ( M 1 u M 2 )}
(11)
(4) Apply MDCT: Do the MDCT to all the audio
data
De (k ) MDCT ( Ae (k ), 0 d k  ( M 1 u M 2 ) 2)
57
Where De ( k ) {De (k )( m), 0 d m  N } ˈ De (k )( m) is
the kth audio data.
(5) Choose the intermediate frequency coefficient:
We choose the median coefficients in the MDCT
domain to satisfy the digital audio signal.
(6) Embedding watermarking: Adaptively modify
the medium coefficient De (k )(mw ) by sorting to
embed v p (k ) .
­° De (k )(i)(1  av p (k )), mk  i  m(k  1)
De '(k )(i ) ®
°̄ De (k )(i)otherwise 13)
where m is modulation factor, a is stretching factor
which can be changed to control the watermarking
embedding intensity. If a is too small, the embedding
signal is too small to be perceptible, which is weak in
anti-interference. If a is too large, it can reduce the use
value of the signal and cause the hearing distortion,
however, it is robust to the attack. The value can be
chosen according to the application.
(7) Inverse MDCT: Do the inverse MDCT of D 'e
A 'e (k ) IMDCT ( D 'e (k )), 0 d k  ( M 1 u M 2 ) (14)
(8) Replace A 'e with Ae to get the final
watermarking signal.
D. Watermarking Extraction
The watermarking extraction algorithm is expressed
as follows:
(1) Segmentation: Segment digital audio signal Ae .
(2) MDCT: Do the MDCT of Ae and Ase .
(3) Extract watermarking:
1
s '(l ) ( Dse (k )(l )  De (k )(l )), mk d l  m(k  1)
a u De (k )(l )
(15)
(4) Generate the chaotic sequence{r (l )} with key K,
and then do the Exclusive OR operation to
get {vsp '(l )} :
vsp '(l ) s '(l ) † r (l )
(16)
(5) Reconstruct {vsp '(k )} from {vsp '(l )} , if the sum
of the m segments is more than > m / 2@  1 , then
watermarking {vsp '(k )} is 1, otherwise, it is 0.
(6) Inverse sorting: The one-dimension sequence
can be obtained by inverse sorting which is expressed
as:
 Vs {vs ( k ) vsp ( k '), 0 d k , k '  M 1 M 2 }
(17)
The watermarking is increased to 2-D dimension,
therefore, the finally extracted watermarking image is
shown in equation (18).
Ws {ws (i, j ) vs (k ), 0 d i  M1 ,0 d j  M 2 , k i * M 2  j}
(18)
IV. EXPERIMENT AND RESULTS
A. Watermarking Security Test
The security test is shown in Figure 2. With right
key, seed=0.38, watermarking can be extracted
correctly. However, if the seed=0.3800001, the
watermarking can not be recovered correctly. With
only 0.0001 differences, the images are quite different
which means we can not extract the watermarking
without correct key.
Figure 2. Watermarking security test ( Left: extracted image with
correct key, right: extracted image without correct key)
B. Watermarking Attack Experiment and Analysis
extracted results are displayed in Figure 3. The PSNR
after embedding is 40.23dB. The original and
embedded watermarking audio is very similar visually
and aurally which also can be seen from the
waveforms.
water
mark
water water
mark mark
Figure 3. Watermarking embedding and extraction comparison
To test the robustness of the watermarking, the
embedding audio watermarking is under a variety of
attacks, such as re-sampling, low pass filtering,
clipping, noise and AVS compression attack. The
watermarking is extracted from the audio signal and
computed the similarity of the original and embedded
audio watermarking. Without any processing, the
normalized correlation coefficient of the extracted
watermarking is 1. The attacks are realized as follows:

To remove the effect of observer’s experience,

experiment and device conditions, and other objective
and subjective effect, PSNR and normalized coefficient
are adopted to objectively assess the original and a b c d
extracted watermarking.
PSNR˖
max w '2 ( n)
PSNR ( w, w ') 10 log10 (19)
1
N
¦ (w '(n)  w(n))2 e f g
Figure 4. Extracted watermarkings after attacks (a. original
Normalized Coefficient (NC) ˖
watermarking. b. re-sampling attack, re-quantization attack, d.
M1 M 2
adding noise attack, e. low pass filter attack, f. clipping attack, and
¦¦ w(i, j )w (i, j )
i 1 j 1
s g. AVS compression attack)
NC ( w, ws ) (20)
M1 M2
(1) AVS attack: The audio bitstream goes through
¦ w (i, j ) ¦ w
i 1
2

j 1
s
2
(i, j ) the encoder and decoder. The compression ratio is 10:1
(2) Clipping: Cut 10% of the original audio.
In this paper, we use 22.05 KHz sampling rate, 8bit (3) Low pass filter: Using Butterworth filter, cutoff
quantization resolution and the length of the mono
frequency is 22.05 kHz, the order is 6.
channel audio signal is 30s. The audio signal contains
(4) Median filter: 4 order median filter.
the speech, popular, country and classic music. A
(5) Gaussian noise: Embedding Gaussian noise
64×64 binary image after chaotic encryption was
embedded to the original audio. The initial value of whose mean is 0 and variance is 0.01.
chaotic sequence is 0.8 and P 2 . The embedding and (6) Re-sampling: Up-sample the original audio to

58
22.05 kHz, then down-sample to 11. 025 kHz
(7) Re-quantization: Quantize the audio frequency
from 16-bit to 8-bit, and then quantize it back to 16-bit.
After various attacks, the extracted watermarkings
are described in Figure 4. we can see that, the
algorithm can resist some attacks and maintain
robustness. It can basically meet the application
requirements for AVS DRM and copyright protection.
Moreover, it is secure and easy to implement.
V. CONCLUSIONS
In this paper, AVS audio watermarking embedding,
extraction and detection for DRM is realized based on
chaotic map in the MDCT domain. In the MDCT
domain, the watermarking intensity and capacity can be
modified adaptively. The chaotic key is introduced into
the standard AVS audio bitstream to improve
robustness. The experiment shows that the proposed
algorithm and scheme can resist some attacks while
retain good quality. This scheme can be applied to the
AVS DRM for copyright protection.
REFERENCES
[1] C.L, Podichuk and E.J., Delp., “Digital watermarking:
Algorithms and applications”. IEEE Signal Processing
Magazine, vol.14, no.4, pp. 33-46, 2001.
59
[2] J., Ruanaidh., S.,Pereira, “A secure robust digital image
watermarking”, Electronic imaging: processing, printing and
publishing in colour, 1998.
[3] R.,Venkatesan, M., Jakubowski, “Image watermarking with
better resilience”. In: Proceeds ICIP, 2000.
[4] M., Barni, F., Bartolini, and A., Piva, “ Improved wavelet-
based watermarking through pixel-wise masking”, IEEE Trans
Image Process, vol. 10, pp.783–91, 2001.
[5] Identification and protection of multimedia information.
Special issue on Proceedings of the IEEE 87, 1999.
[6] A. Tefas, A., Nikolaidis A N., Nikolaidis, V., Solachidis, S.,
Tsekeridou, and I., Pitas, “Markov chaotic sequences for
correlation based watermarking schemes”, Choas, Solitons &
Fractals vol. 17, pp.567–573, 2003.
[7] RL., Devaney, “A first course in chaotic dynamical systems –
theory and experiment”. Cambridge, MA: Perseus Books;
1992.
[8] S., Tsekeridou, V., Solachidis, N., Nikolaidis, A., Nikolaidis,
A., Tefas, and I., Pitas, “Bernoulli shift generated
watermarkings: theoretic investigation”,In: Proceedings of
IEEE international conference on acoustics, speech and signal
processing, pp. 1989–1992,2001.
[9] A., Nikolaidis A and I., Pitas, “Comparison of different chaotic
maps with application to image watermarking”, In: Proceedings
of IEEE international symposium on circuits and systems,
pp.509–512, 2001.
[10] A., Mooney and JG., Keating, “Noisy optical detection of
chaos-based watermarkings” in: Proceedings of SPIE Photonics
North vol.5579, pp.341–350, 2004.
[11] A., Mooney amd JG. Keating, “The impact of the theoretical
properties of the logistic function on the generation of optically
detectable watermarkings”, In: Proceedings SPIE, Technology
for Optical Countermeasures, Optics/Photonics in Defence and
Security, pp. 120–129, 2004.
 2009 International Conference on Computer Engineering and Technology
Walking Modeling Based on Motion Functions
Hao Zhang
School of Computer Science and Technology
Xidian University
Xi’an, P. R. China
zhanghao@xidian.edu.cn
Abstract—A modeling method of motion-function is presented
to represent the motion in detection of video images within the
physical and motional characters of body. It concentrates on
the characters of video surveillance, with the examples of
walking videos, and results in walking functions by extracting
and dealing with the data from the experiments. On this basis,
more experiments are required to validate all coefficients in
functions, and induce final expressions of gait function and
modeling.
Keywords-computer vision; video surveillance; motion
recognition; motion-function; biometrics
I. INTRODUCTION
The vision analysis [1] of body motion in computer
vision consists of detection, tracking, identification and
action understanding in the sequences of video images.
Because of the demands for sensitive scenarios, including
banks, airports, customs and so on, it is an important
research domain to identify body movement without
surveillance in long distance.
There are three main methods [2] of action identification,
for example, based-on template, probability-network and
syntactic technique. Many scholars, such as Murase and
Sakai [3], Shutler [4], Johnson and Bobick [5], Brand [6,7],
Oliver [8], Kitani [9], Liang Wang [10] and so on, devoted
themselves into this domain and presented many identifying
methods. With the characters of video surveillance [11],
there are many factors existing in environmental restriction
in application of identifying methods based-on gait, so that it
does not meet the applicable demands of video surveillance.
As a result, on the basis of Long Ye’s idea [12], we utilize
the consecutive characters of video information with the
physical law, and present a method of motion-function to
analyze body motion. Then we analyze the data of body
motions and induce and validate the functions furthermore.
The rest of the paper is organized as follows. Section 2
introduces the theoretic model on biometrics. Section 3
describes modeling and functions in motion information.
Section 4 discusses the experimental results. Conclusive
remarks are addressed in the final section, Section 5.
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.12
Zhijing Liu
School of Computer Science and Technology
Xidian University
Xi’an, P. R. China
liuzhijing@vip.163.com
II. THEORETIC MODEL
The physical characters of human and motion are utilized
in this model, and three functions between body biometrics
and time, which are the proportion of step length to human
stature and time, human silhouette and time, the coordinates
of human centroid and time, are constructed. Then body
motions can be detected in term of these factors, such as
walking, running, jumping, creeping etc.
A. The Relationship between the Proportion of Step
Length to Human Stature and Time
First of all, because of the restriction on physical factors,
the step length is certain to be restricted by human stature.
Videlicet, step length and stature accord with the proportion,
and it is universal to body motion. Next, while walking,
human’s arms and legs move periodically. It is universally
that body motion is periodic.
B. The Relationship between Human Silhouette and
Time
As video information represents 3D in the form of 2D, it
is difficult to extract human model of 3D directly. Human
silhouette of 2D can be extracted directly, consequently, the
states of body motion are determined by the variations of
rectangles’ areas, and the variations of body motion are
detected in consequence.
C. The Relationship between the Coordinates of Human
Centroid and Time
Body motion in video information is equal to mechanism
motion of a body in physics. In physics, the centroid of body
is an important reference point of mechanism motion in
study. Its definition is a system of particles made of N ones,
whose mass center is the centroid of system. In the study of
body motions, the centroid is the reference point and
described the physical parameters, such as orientation,
velocity etc, which reflect the state of motion, so that the
goal of detection in body motion is achieved.
III. MODELING AND FUNCTIONING
As shown in Fig. 1, two walking videos of 90° and 45°
are selected, which are the angles between recording
orientation and body motion one. For short, they are labeled
as 90° video and 45° one respectively. On the basis of these,
60
some experiments are implemented. There are several steps
as follows: detecting body motion, extracting human
silhouette with background subtraction, labeling with
rectangle, as shown in Fig. 1, finding the data of length (l),
width (w) and center coordinates (x, y) of the rectangle. By
formula, the areas of rectangles are calculated, and then
motion cycles are analyzed in order to induce the expressions
of walking function.

Figure 2. The data and fitting curves on the

proportion of step length to stature

The fitting curve of 90°video:

2π π
f = 0.15 ⋅ sin( ⋅ (t − − 0.06)) + 0.38 (2)
0.7 2
(a) 90°video
The fitting curve of 45°video:

2π
f = 0.04 ⋅ sin( ⋅ (t − 0.325)) ⋅ exp(0.5 ⋅ t ) + 0.36 (3)
0.68

B. The Relationship between Human Silhouette and

Time
In video surveillance, due to the restrictions of recorders
and algorithm, the exact silhouette of human is not extracted
directly and immediately. As a result, the areas (s) of
rectangles on human silhouette are computed by Eq. 4, and
(b) 45°video are utilized to represent equivalently. On the basis of data
extracted, the proportions (g) of areas of rectangles between
Figure 1. The walking videos of 90°and 45° adjacent frames in video images are found by Eq. 5, as
shown in Fig. 3.
A. The Relationship between the Proportion of Step
Length to Stature and Time s(i ) = l (i ) ⋅ h(i ),1 ≤ i ≤ tn , (4)
By physical characters of human, the proportion of
walking racer’s step length to stature in competition is
si +1
between 0.6 and 0.7 [13], by means of statistical data of g (i ) = ,1 ≤ i ≤ tn − 1， (5)
videos, it concludes that the proportion of step length and si
stature is between 0.5 and 0.6. The proportions (f) of step where tn is time of video.
length to stature are calculated by Eq. 1. The data are fitted by means of the proportion of areas of
rectangles to time, as shown in Fig. 3, and the expressions
w(i ) were induced as follows:
f (i ) = ,1 ≤ i ≤ tn , (1) The fitting curve of 90°video:
l (i )
2π
g=0. 35 ⋅ si n( ⋅ (t − 1.4)) + 1.02 (6)
where tn is time of video. 0.7
The discrete data are represented and fitted with time (t), The fitting curve of 45°video:
as shown in Fig. 2. Therefore, the expressions of the 2π
proportions of step length to stature in videos of 90° and 45° g = 0.25 ⋅ sin( ⋅ (t − 0.78)) − 0.17 ⋅ exp(−0.628 ⋅ x) + 1.1 (7)
0.68
are induced, as shown in Eq. 2 and 3.

61
 Figure 3. The data and fitting curve on the proportion of the
rectangle areas for the adjacent frames
C. The Relationship between the Coordinates of Human
Centroid and Time
In mechanism motion, human centroid is used to describe
physical parameters of body motion, as its consequence, a
method [14] to calculate the centroid is presented and
applied to the representation of silhouette characters [10]. In
video surveillance, because of the recording angles and so
on, it is not easy to calculate the centroid by extracting data.
Consequently, the center of rectangle is used to replace the
centroids equivalently. In addition, by means of extracting
data, Euler distance is used as follow:
d = ( x − x0 ) 2 + ( y − y0 ) 2
where (x0, y0) is an initial pixel of video to represent the
distance of body motion. The data are extracted, as shown in
Fig. 4.
By means of the relationship between distance (d) and
time (t), the data were fitted, as shown in Fig. 4. As a result,
the expressions were induced, as shown in Eq. 9 and 10.
The fitting curve of 90°video:
d = 77.38 ⋅ t + 1.354
The fitting curve of 45°video:
d = 74.48 ⋅ t − 7.224
TABLE I. THE EXPRESSIONS OF WALKING FUNCTIONS
Video Type The Fitting Curves of 90°Video
2π
f f = A1 ⋅ sin( ⋅ (t − P1 )) + B1
T T
2π
g g=A2 ⋅ si n( ⋅ (t − P2 )) + B2
T T
d d = A3 ⋅ t + B3
Figure 4. The data and fitting curves on the
distance of adjacent rectangle centers
IV. EXPERIMENTAL RESULTS
The experimental results above conclude three arrays of
expressions on walking function in 90° and 45° videos, as
shown in Table 1.
With the experiments on testing set, the relative errors of
main parameters of walking functions, such as period (T),
amplitude (A), phase (P), and offset (B, C, D), were less than
10%, so the expressions were proved to be valid.
The expressions and results above prove that human
actions, angles, periods and velocity are determined by
variation of the relevant parameters.
V. CONCLUSION
(8)
The identification of body motion is one of main and
latest orientation on study. Based on the laws of mechanism
motion, the characteristic data of human walking are
extracted from videos. These data are fitted by the characters
of video surveillance to find the expressions of walking
function. Meanwhile, the validity of expressions is proved by
experimental results. The expressions of human walking
derived from the experiments are with the characters of high
efficiency in computation. Consequently, it will be applied to
video surveillance in order to detect and affirm abnormal
(9) actions.
On basis of the experiments, the functions of human
walking are studied at large. In contrast, other simply ways
of actions, such as running, jumping, and complex ones
which consist of many simply ones, remain to be studied and
(10) presented further.
The Fitting Curves of 45°Video
2π
f = A1' ⋅ sin( '
⋅ (t − P1' )) ⋅ exp( B1' ⋅ t ) + C1'
2π
g = A2' ⋅ sin( '
⋅ (t − P2' )) + B2' ⋅ exp(C2' ⋅ x) + D2'
d = A3' ⋅ t + B3'
62
 ACKNOWLEDGMENT
The research was supported in part by the Ministry of
Education of the People’s Republic of China and Research
Project (Guangdong Province. No. 2006D90704017). The
authors would also like to thank the anonymous reviewers
for their valuable comments that resulted in an improved
manuscript.
REFERENCES
[1] WANG Liang, HU Wei-Ming, TAN Tie-Niu. A Survey of Visual
Analysis of Human Motion. Chinese Journal of Computers, vol. 25,
no. 3, pp. 225-237, Mar. 2002.
[2] DU You-tian, CHEN Feng, XU Wen-li, Li Yong-bin. A Survey on
the Vision-Based Human Motion Recognition. Chinese Journal of
Electronics, vol. 35, no. 1, pp. 84-90, Jan. 2007.
[3] Murase H, Sakai R. Moving object recognition in eigenspace
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[5] Johnson A, Bobick A. A multi-view method for gait recognition using
static body parameters. In Proc International Conference on Audio-
and Video-based Biometric Person Authentication, 2001, pp. 301-
311.
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[6] Brand M. Understanding manipulation in video. In Proc Int Conf
Autom Face Gest Recogn, 1996, pp. 94-99.
[7] Brand M, Oliver N, Pentland A. Coupled hidden markov models for
complex action recognition. In Proc IEEE Comput Soc Conf Comput
Vision Pattern Recognit, 1997, pp. 994-999.
[8] Oliver N M, Rosario B, Pentland A P. Bayesian computer vision
system for modeling human interactions. IEEE Trans Pattern Anal
Mach Intell, vol. 22, no. 8, pp. 831 – 843, Aug. 2000.
[9] Kitani K M, Sato Y, Sugimoto A. Deleted interpolation using a
hierarchical Bayesian grammar network for recognizing human
activity. In Proc IEEE workshop on VS PETS, 2005, pp. 239-246.
[10] WANG Liang, HU Wei-Ming, TAN Tie-Niu. Gait-Based Human
Identification. Chinese Journal of Computers, vol. 26, no. 3, pp. 353-
360, Mar. 2003.
[11] Weiming Hu, Tieniu Tan, Liang Wang, Steve Maybank. A Survey on
Visual Surveillance of Object Motion and Behaviors. IEEE Trans
Syst Man Cybern Pt C Appl Rev, vol. 34, no. 3, pp. 334-352, Aug.
2004.
[12] YE Long, FU Jie, ZHANG Qin, LIU Jian-bo, WANG Hui. Gait
Tracking Based on Motion Functional Modeling. Journal of
Communication University of China (Science and Technology), vol.
14, no. 3, pp. 13-16, Sept. 2007.
[13] XIONG Xi-Bei, YAO Guo-Qiang, XU Shu-Kui. Basic Text of Track
and Field. Beijing Sport University Press, Beijing, 1997.
[14] FAN Yi-fang, ZOU Liang-chou, YUAN Zhi-run. Research on the
Locating of Image Centroid of Human Body Movements——Locating
the Human Body Centroid. Journal of Guangzhou Institute of
Physical Education, vol. 24, no. 3, pp. 35-37, 2004.
 2009 International Conference on Computer Engineering and Technology

Preprocessing and Feature Preparation in Chinese Web Page Classification

Weitong Huang Luxiong Xu

Department of Computer Science and Technology Fujian Normal University, Fuqing Brunch
Tsinghua University Fuqing, Fujian,China
Beijing,China,100084 xlx@fjnu.edu.cn
huangwt@tsinghua.edu.cn

Yanmin Liu
Department of Computer Technology and Applications
Qinghai University
Xining, Qinghai,810016
lyanmin@qhu.edu.cn

Abstract—A detailed design and implementation of a Chinese Chinese word segmentation, stopword removal, stemming,
web-page classification system is described in this paper, and and vocabulary selection.
some methods on Chinese web-page preprocessing and feature
preparation are proposed. Experimental results on a Chinese
web-page dataset show that methoss we designed can improve
the performance from 75.82% to 81.88%.

Keywords- Text classification, Chinese web-page

preprocessing, Feature preparation

I. INTRODUCTION
With the rapid development of Internet, the amount of
web-pages has been increasing dramatically. While
providing all-embracing information, the large collections of
web-pages bring people great challenge: how to find what
they need. In order to organize and utilize information
effectively, people hope to classify web-pages according to
their contents. Web-page classification is widely applied in
the field of vertical search, personalized search, etc. Web- Figure 1. A Chinese Web-page Classification System
page classification has been a hotspot in the research domain
of text mining and web mining. In this paper, we will
introduce a Chinese web-page classification system we 1) HTML Parsing
implemented. The system design and our proposed methods The purpose of HTML Parsing is to remove html code
will be described in section 2. In Section 3, the experimental that is irrelevant and extract text. We save the parsing result
results on a Chinese web-page dataset are shown as well as as files of XML format for further preprocessing and feature
some discussions. Finally, we conclude our work in Section extracting. The procedure of HTML Parsing is as follows:
4. Firstly, remove html source code embedded in tags such
as “style”, “script”, and “applet”. Reserve the attributes of
II. SYSTEM DESIGN AND IMPLEMENTATION
“meta” and “a” and discard attributes of other tags. The
A. System Architecture reason is that the attributes of tags is not much helpful to
classification task except those of “meta” and “a”.
There are three parts in our Chinese web-page Secondly, we add user-defined tags for text in a web-
classification system: web-page preprocessing, feature page, because no special tags are prepared for text in
preparation, and web-page classification. The system standard HTML. The rule is: the “plain text”, i.e. the non-
architecture is illustrated in Figure 1. hyperlink text is identified by “text” tag, while the hyperlink
B. Web-page Preprocessing text is identified by “anchor” tag. Following the above rule,
a HTML web-page can be represented as a HTML tag tree,
In the whole data mining task, 60% of the work is data
whose leaf nodes are plain text identified by “text” tag or
preprocessing. There are six procedures in web-page
hyperlink text identified by “anchor”.
preprocessing: HTML parsing, English lexical analysis,

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 64

DOI 10.1109/ICCET.2009.72
 Usually the page layout is controlled by “table” tag,
while the tables along the two sides, the top and the bottom
are often noises such as navigation bar and ads. Once there
is a HTML tag tree described above, some algorithms [1]
[2] can be applied to remove noises and extract content text.
In this paper, we adopt a more simple and effective method
to remove noises. Through experiments (see section 3.2), we
discover that in a content page (not a hub page), the
hyperlink text is usually used for navigation or
advertisement. So it is effective to remove noises to only
discard hyperlink text.
2) English Lexical Analysis and Chinese Word
Segmentation
In English lexical analysis, spaces are usually used for
separators, while there are no separators between words in
Chinese text. Therefore, Chinese lexical analysis is a
prerequisite to Chinese information processing. As we
know, there are two open source Chinese word
segmentation projects. One is ChineseSegmenter [5] which
works with a version of the maximal matching algorithm.
When looking for words, it attempts to match the longest
word possible. This simple algorithm is surprisingly
effective. Institute of Computing Technology in China
developed a Chinese lexical analysis system ICTCLAS [6]
using an approach based on multi-layer HMM.
ICTCLAS includes word segmentation, Part-Of-
Speech tagging and unknown words recognition.
Because of ICTCLAS’s higher precision, we adopt it as the
Chinese word segmentation module in our classification
system.
3) Stopword Removal and Stemming
Words with high frequency are not only helpless to
distinguish documents but also increase the dimension of
feature space. Such words are called stopword. We maintain
a stopword list containing 580 English words and 637
Chinese words, which is used for stopword removal. In
English, many words have variations. Using stemming [7],
words having the same stem can be considered as to be
equal.
4) Vocabulary Selection
A sentence in natural language is made up of words of
different parts of speech, such as nouns, pronouns, articles,
verbs, adjectives, adverbs, prepositions, and conjunctions.
Among them, nouns and verbs express more semantic
meaning. So it is a feasible way to choose only nouns and
verbs as feature words in order to reduce feature space,
remove noises and improve classification quality. We use
ICTCLAS[6] to extract nouns and verbs from a sentence,
and the experiments are in section 3.3.
C. Feature Extraction and Weighting
HTML web-pages are semi-structured text. A web-page
consists of two parts: <head> part and <body> part. The
former includes “meta” and “title” which are the summary
description of the whole page, and the latter is the actual
content text that we can see visually. The content text in the
65
<body> part includes two types of text. One is plain text
which is content related, and the other is hyperlink text.
Assigning different weight to different parts can improve
the quality of classification. Our experiments are in section
3.2.
D. Feature Selection
One difficulty in text classification is the high dimension
of feature space. The task of “feature selection” is to remove
less important words from the feature space. Many feature
selection methods have been studied, for example,
document frequency (DF), information gain (IG), mutual
information (MI), and x'-test (CHI). [3][4] compared the
above four methods. The result shows that IG and CHI gain
the best performance; DF has comparable performance with
IG and CHI, and it has the advantage of simplicity and low
computation complexity; MI is the worst.
We adopt document frequency selection in our system. The
assumption of this method is: the features (words) with low
document frequency have small influence on the
classification accuracy.
We use the revised formula below to compute the
document frequency of word wt .
freq ( wt , Ci )
freq( wt ) = ∑
Ci ∈C num(Ci )
freq ( wt , Ci ) is the document frequency of wt in
class Ci ; num(Ci ) is the number of documents in class Ci .
E. Classifier
1) Naïve Bayes Classifier
The Naïve Bayes Classifier is a simple but effective text
classification algorithm. It performs very well in practice
with high accuracy. Given a document and a classifier, we
determine the probability P (c j | d i ) that the document
belongs in class c j by Bayes’ rule and the naïve Bayes
assumption:
|d i |
P(c j | di ) ∝ P(c j ) P(di | c j ) ∝ P(c j )∏ P( wdi ,k | c j )
k =1
P (c j ) indicates the document frequency of class c j
relative to all other classes; P ( wt | c j ) indicates the
frequency that the classifier expects word wt to occur in
documents in class c j ; wdi ,k denotes the k th word in
document di .
1 + ∑ d ∈D N ( wt , di ) P(c j | di )
P( wt | c j ) = i
| V | + ∑ s =1 ∑ d ∈D N ( ws , di ) P(c j | di )
|V |
i
 V is the vocabulary; D is the set of training Section 2.2.1 has defined plain text, i.e. the non-
hyperlink text in web-pages, which are always correlated
documents; N ( wt , d i ) is the count of the number of times with the topic of the web-pages and contain few noises. We
word wt occurs in document di ; P (c j | d i ) ∈ {0,1} extract only the plain text in every web-page as a feature
extraction scheme, and compare it with the one extracting the
1 + ∑ d ∈D P(c j | di ) full text. Figure 2 shows the experimental results. When the
P (c j ) = i
amount of feature words varies from 1,000 to 10,000, the
|C |+| D| average Micro-F1 of full-text-scheme is 75.82%, while
| C | is the number of classes. | D | is the number of plain-text-scheme gets to 78.93%. The results shows that the
scheme of extracting only plain text as features can improve
training documents. the classification quality by 3.11%, by effectively removing
2) Evaluation Measure the noises in the web-pages.
We use the standard measures to evaluate the
performance of our classification system, i.e. precision,
recall and F1-measure[8].

III. EXPERIMENTS
A. Chinese Web-page Dataset
We download 6745 Chinese web-pages from
http://www.sohu.com/ as our training set, and 2814 web-
pages from http://www.sina.com.cn/ as our test set. These
web-pages are distributed among 8 categories. Table 1
shows the amount of documents in both training and test set
in each category. We have excuses to choose training set Figure 3. Classification Results of Different Header-Weighting Schemes
and test set from two different sources. Our former work
show that the performance on the dataset from only one data The header of a web-page, which is concise and exact,
source is surprisingly high (usually above 95%), but that is reflects how the page author highly generalized the content
not the real world. We decide to use the test set of a on the web-page. Properly raising the header weight in the
different source to evaluate the effect of training. feature space can improve the classification quality. We
have done an experiment to get the ratio of the header
TABLE I. CHINESE WEB-PAGE DATASET weight to the body text weight: Choose 4,000 feature words
by document frequency method, and compare the different
Category Name Training Set Test Set (from
classification quality when the ratio of the header weight to
(from sohu) sina)
Auto 841 351 the body text weight varies from 2:1 to 10:1.
Business 630 263 C. Vocabulary Selection
Entertainment 1254 523
Health 784 327
House 736 307
IT 1050 438
Sports 841 351
Women 609 254
Sum 6745 2814
B. Feature Extraction and Weight Distribution

Figure 4. Classification Results using vocabularies of different parts of

speech

The results, illustrated in figure 3, shows Micro-F1 is

Figure 2. Classification Results of Different Feature Extraction Schemes
79.75% when header and body text share the same weight,
and Micro-F1 rises up as the header weight is raised until
gets to a maximum when the ratio of the header weight to
the body text weight is 5:1. Therefore, we let the ratio of the
header weight to the body text weight to be 5:1. Figure 2

66
shows the experimental results of feature extraction after We are satisfied with our improvement, because many cases
raising the header weight. The figure shows the average cannot be assigned to only one category absolutely. For
Micro-F1 goes up from 78.93% to 82.00% thanks to example, web-pages about women’s health information can
increasing the header weight. be classified into the category “Health” as well as the
We do the research on different candidate vocabularies category “Women”, but this is the real world.
including all words, nouns and verbs. The results are
illustrated in figure 4. The classification result of using only IV. CONCLUSION
nouns and verbs is clearly better than that of using all In this paper, a series of web-page preprocessing and
words, which means nouns and verbs are enough to reflect feature preparation methods are proposed. Through
the content of a web-page and can eliminate the noises experiments, we give the following conclusions: extracting
caused by pronouns, adjectives, quantifiers, adverbs, and only “plain text” can eliminate noises in web-pages
consequently improve the classification quality. effectively; both raising the header weight and choosing
D. Final Experimental Results only nouns and verbs as candidate features can improve the
classification quality. On our Chinese web-page dataset, the
Considering the experimental results comparisons and proposed methods improve the measurement Micro-F1
analysis mentioned above, we set down a final web-pages greatly, from 75.82% to 81.88%, compared to full-text
classification scheme: extract the plain text in web-pages, method. In the future, we will enrich our Chinese web-page
raise the ratio of the header weight to the body text weight dataset and do experiments on larger and more datasets.
to 5:1, choose nouns and verbs as features, select 4,000
feature words using document frequency method, and use
Naïve Bayes classifier to train and test the Chinese web- REFERENCES
page dataset. The precision and recall of each category is [1] Lawrence Kai Shih and David R. Karger. Using URLs and Table
illustrated in Table 2. Micro-F1 is 81.88% when we use the Layout for Web Classification Tasks. In Proceedings of WWW’04,
preprocessing and feature preparation methods mentioned New York, New York, USA, 2004
above. In comparison, Micro-F1 is only 75.82%, showed in [2] Chengjie Sun, Yi Guan. A Statistical Approach for Content
Extraction from Web Page. Journal of Chinese Information
figure 2, if we extract the full text and don’t use special Processing, 2004, 18(5): 17~22.
feature preparation methods. [3] Yiming Yang, and Jan O. Pedersen. A Comparative Study on Feature
Selection in Text Categorization. In Proceedings of ICML, Nashville,
TABLE II. CLASSIFICATION RESULTS OF EACH CATEGORY Tennessee, USA, 1997
Category Name Precision Recall [4] Songwei Shan, Shicong Feng, and Xiaoming Li. A Comparative
Auto 87.69% 83.19% Study on Several Typical Feature Selection Methods for Chinese
Business 87.72% 76.05% Web Page Categorization. Computer Engineering and Applications,
2003, 39(22): 146~148.
Entertainment 85.89% 95.41%
Health 64.39% 69.11% [5] Chinese Segmenter:
House 84.56% 82.08% http://www.mandarintools.com/segmenter.html
IT 70.54% 90.18% [6] ICTCLAS:
Sports 97.90% 92.88% http://www.nlp.org.cn/project/project.php?proj_id=6
Women 87.69% 44.88% [7] Porter Stemming Algorithm:
Micro-F1 81.88% http://www.tartarus.org/martin/PorterStemmer/
[8] C.J.van Rijsbergen. Information Retrieval. Butterworth, London,
1979, 173~176

67
 2009 International Conference on Computer Engineering and Technology
HIGH PERFORMANCE GRID COMPUTING AND SECURITY THROUGH
LOAD BALANCING
V.Prasanna Venkatesh
Final year
Department Of Information Technology
St.Joseph’s College Of Engineering,
Chennai, India
E-mail: vprasanna.1310@gmail.com
Abstract
It has always been a great deal to balance
both security and performance. But here
we have presented a scenario in which a
local scheduler of a networked system can
schedule its jobs on and across other
systems in the network securely. There by
creating an environment in which no
system is idle when even two systems are
communicating. This clubs both
distributed computing and load sharing.
Thus high performance is assured.
When we look onto the other side of the
coin of such networked systems, security
concerns arise. These issues are handled
by employing new level of managerial
systems called Authenticated Distribution
Managers (ADMs) for managing the co-
operating systems. ADM does the work of
job division and integration based on the
security implications of its Authenticator.
Any system can select any other system as
an ADM and transfer its security policies
to it. Thus the security is not compromised
even if there is load distribution across
various systems. Thus we have presented
an overview of a high performance
network without security being
compromised.
Keywords:
Security, High Performance, Load
sharing, balancing, Authenticated
Distribution Managers (ADM), job
division, Security Authentication.
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.77
V.Sugavanan
Final year
Department Of Information Technology
St.Joseph’s College Of Engineering
Chennai, India
E-mail: sugavanan.v@gmail.com
Introduction
The topic chosen here for discussion,
elaborates on the basic concepts of grid
computing and load balancing and their
advantages, and advocates the
development of a new technique, a
technique which doesn’t try to eliminate
the short comings of either of the
concepts, but involves the combination of
the advantages of both of these concepts,
and try to throw light on the tremendous
potential that this technique possesses for
providing high performance output in the
distributed network, along with
enhancement in resource utilization and
optimization.
The discussion proceeds by explaining
about a grid system and discusses the load
sharing ideas of the grid, and emphasize
on its advantage of high resource
utilization.
The section following the discussion about
grid computing explains about the
advanced methods of load balancing,
which provides various security
implementations and protocols that
provide highly secure operations.
The new technique that we discuss later
on, combines these two advantages, that is,
it detects a new opening in the security
implementations of load balancing by
distributing them through a grid system,
which allows faster implementation of
68
these security measures and in an efficient explored as much as the other areas of
manner. distributed systems. The obvious
Thus, in our new technique, we try to drawback of such kinds of new systems
implement security in grid computing by would be the security implementations.
using the advanced security provided by For this reason, we look for systems that
load balancing systems in a grid system have well established security
for producing high performance, high implementations that can be performed
resource utilization and high-speed faster through a grid system.
systems.

Grid computing:

A grid is defined as a group of nodes that

are connected to each other through a
number of connecting paths for JAVA C PYTHON
CLIENT CLIENT CLIENT,etc
communication. Grid computing allows a
cluster of loosely coupled computers to
perform large tasks or tasks that generally
consume more resources and time than is
feasible for a single system. This is a type
of parallel computing technology in which
the computers involved, called “nodes”,
are almost completely independent of each
MANAGEMENT INTERFACE
other in terms of resources, timing and
synchronization aspects. These nodes are
more heterogeneous than those present in Security,
other distributed systems. The nodes authentication and
involved in the process are voluntary, and other protocol
implementation
the interaction and mediation between the
nodes of the grid are performed through a
JAVA C PYTHON
management interface. All transactions in SERVICE SERVICE SERVICE,etc
the grid are performed through the
management interface.
Advantage:
A grid computing system as discussed
above provides sharing of a task that a
single node is unable to perform. Hence, it
utilizes its available resources to provide Load Balancing:
enhanced performance. this is the main
feature of grid computing, that is, high Load balancing is the process in which the
performance through enhanced resource entire workload on a system can be
utilization. distributed among the nodes of a network
so that resource utilization and time
Disadvantage: management can be enhanced in the
Grid computing is a new concept and the network.
implications in this field have not been

69
This technique is very useful in providing
optimization of the resources of the
network, where in, the server manages the
distribution of the workload among the
nodes of the network. The server that
performs this task is called the “Load
Balancer”.
The main features of a load balancer are:
Security:
The load balancer provides security of
data that needs to be shared between
systems at the time of load balancing.
These security aspects involve the
implementation of available security
protocols like:
1. The SSL (Secure Socket
Layer) protocol for
TLS(Transport Layer
Security) ,
2. Protection against DDoS
(Distributed Denial of Service) and
3. Security through firewalls.
Authentication:
Authentication is required in load
balancing when the load is distributed to
many nodes in the network. It is necessary
that each of the client systems are
authenticated nodes of the network so that
data transfer can be performed.
Thus, authentication of the client
systems is an important aspect of the
discussion.
Reliability:
Load balancing techniques provide
reliability through redundancy, which is
indispensable in any network that
implements high security.
Redundancy is required also for
authentication services and in the
implementation of the basic feature of
distributed systems, namely, parallelism.
70
Advantage:
The notable feature of the load balancing
concept, from the above discussion, is the
highly advanced security implementations
through various security and
authentication protocols.
This feature can as well be combined with
the grid concept to enhance its security
implementations.
Disadvantage:
The drawback of this concept lies in the
fact that implementation of so many
security features on each and every node
in a grid system results in enormous time
consumption. This will ultimately cause
the main feature of grid systems to suffer
attrition, namely, higher performance by
providing faster outputs.
It is thus unfeasible to implement the
security of a load balancing system to each
and every node of the grid system every
time two or more nodes are
communicating.
The approach discussed below provides a
simple, feasible solution to the above
situation and aims at solving it by
combining the above two concepts into a
single system.
Approach:
To attain this goal, we primarily explore
the inherent parallelism principles of most
complex problems, through the prism of
their immanent natural distributed
parallelism, and to find effective and
efficient methods of implementation of
security aspects required for the
distributed environment to provide
reliability to the entire network.
We now implement the usage of the
security aspects of a load balancing
system, by integrating them in a set of
loosely coupled system in a grid. This is
performed by using the concept of
 distribution of security implementation
throughout a grid, where, every node has
access to the grid only after authentication
have been performed by the respective
management agent.
Hence, in a grid system, since every node
is authenticated, it is possible to distribute
the security aspects of one node in the
system by implementing them through
other nodes in the grid, whose states are
monitored by a specially designated
system in the grid.
Such a system in the grid is called the
Authenticated Distribution Manager
(ADM).
Communicating
system
A primary node designates the
Authenticated Distribution Manager. The
ADM can be chosen from a set of idle
systems present in the grid. Once the
primary node obtains its ADM, it hands
over its security systems and protocol
implementations to the ADM. The ADM
then identifies other idle nodes in the grid
and distributes to each of these nodes, the
load of security implementation for the
Primary node.
In this way, the load of security
implementation for one node in a grid can
be balanced between all the idle nodes
present in the grid, thus reducing the time
requirements for the security
implementation and also providing
optimization of resources in the grid.

ADM
Primary node
Distribution of
security
implementation of
primary
node, by ADM.
Idle nodes in the grid

Role of the ADM: Advantages:

In a grid system, when a node is in The advantages of this new technique are
communication with a system, which may listed as follows:
or may not be part of the grid, it needs to 1. High security to grid systems.
implement its security standards and 2. High performance to grid systems
protocols. This node, which requires through faster security
security implementation, is called the “ implementations.
Primary node ”. 3. Improved efficiency in grid
computing.

71
 4. Utilizing resources of a grid system
for security implementation.
5. Improved resource optimization by
utilizing idle nodes in a grid.
Key References :
Scope for future research:
1. Asser.N.Tantawi, Don Towsley,
The possibility of certain drawbacks in the Optimal static load balancing in
new approach discussed can be in the distributed computer systems, in
implementation of distribution algorithms the Journal of ACM, Volume 32,
in the ADM. issue 2, pages 445-465.
Constraints may occur in situations where 2. Parvin Asadzadeh, Rajkumar
none of the nodes in the grid system might Buyya, Chun Ling Kei, Deepa
be idle. In such situations, it might be Nayar, and Srikumar Venugopal,
necessary to maintain at least one node Global Grids and Software
called the “master node”, that will Toolkits, Technical Report,
pretend to be always idle in a grid, and GRIDS-TR-2004-5, Grid
perform only the functions of the ADM. Computing and Distributed
Any further possibilities in the above- Systems Laboratory, University of
discussed technique, is available for Melbourne, Australia, July 1, 2004.
research and further areas are to be
explored in future.

Conclusion and future vision:

Thus, we enable the implementation of

high performance in the grid system by
balancing the load of security
implementation, thereby reducing the time
requirement for a task in the grid, and also
implement the security of the grid system,
using load balancing between the idle
nodes of the grid. This in turn allows
resource optimization.
The idea discussed above allows
integration of the two major futuristic and
fast developing techniques used in
distributed environments, namely, grid
computing and load balancing, that
perform nullification of each others’
drawbacks by implementing one technique
within another’s domain, that not only
integrates their advantages, but also
provides tremendous scope for
development of security as well as
performance of any distributed system.

72
 2009 International Conference on Computer Engineering and Technology
Research of The Synthesis Control of Force and Position in Electro-Hydraulic
Servo System
Yadong Meng, Changchun Li, Hao Yan, Xiaodong Liu
School of Mechanical, Electronic and Control Engineering, Beijing Jiaotong University, Beijing,
100044, China
06116282@bjtu.edu.cn
Abstract
The synthesis control of force and position in
electro-hydraulic servo system has been applied to
some special occasions. Analyzed a typical system
which adopted this kind of control algorithm ，
Applying unchanged structure theory, the compensated
transfer function has been deduced, the compensated
state-space model is presented. Considering the
synthesis control should meet the requirement to force
and position at the same time , introduced the linear
quadratic performance fonctionelle, the optimized
control formula of the synthesis control of force and
position on total state feedback has been deduced . The
simulation results proved that the compensated state-
space model has more superior performance.
1. Introduction
The electro-hydraulic servo system aim at the force
control or the position control independently is familiar.
In some special occasions, the force control and the
position control in one system need to be run
simultaneously, they always disturb each other, but
their performance (given curves) should be met
together. This kind of control task has been widely
studied in robotic field [6,7], but few works have been
done in electro-hydraulic servo system.
In some existing electro-hydraulic servo systems,
we faced with the demand of force control
accompanied by position control, but we considered it
as a special case. For example, The load simulator used
commonly in aerospace field, it is a typical force
application system disturbed by position interference
signal, the position interference signal is a random
signal uncontrolled, The purpose of each control
algorithm applied on this system is to eliminate the
position interference, so, The equivalent input of
position interference signal to system is zero. We can
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.35
control force application accurately. Literature [1]
discusses all kinds of force application system with
influence in detail. The other one system is position
control system with force feedback, The force
feedback signal influence on position control, but the
position control is the final task of system, We regulate
system according the force feedback value to ensure
position control[2-4], or to keep the force unchanged [5] ,
In such a system, the force control is auxiliary of the
position control.
2. System description and modeling
2.1. System description
Fig.1 shows a typical synthesis control system,
which contained two proportion valves, two symmetry
hydraulic cylinders, an elastic load, a position sensor,
and two pressure sensors. The left side subsystem is
force application control system; the right side
subsystem is position control system, an elastic load
between two cylinder rods. The time-position curve of
the right side cylinder rod is given, the time-force
application curve of the left side cylinder rod is given
simultaneously, and there is no relation between two
given curves. The value measured by force sensor is
the elastic force of spring, the driving force of the left
side force application control system is worked out
from pressure difference multiplied by efficient area.
The system parameters as follows: F 0 is the given
driving force, F1 is the practical driving force, D 0 is
the given cylinder rod position, D 1 , D 2 are practical
position of the two cylinder rods respectively, W 1 1 , W 2 1
are input signal amplifiers, W 1 2 , W 2 2 are transfer
functions of two amplifiers respectively, W 13 , W 23 are
transfer functions of two proportion valves
respectively, W 1 4 is coefficient of force-electro
transform, W 2 4 is coefficient of position-electro
73
transform, I 1 , I 2 are electric current to two proportion B1V = W13 I 1 (7)
valves respectively, m 1 , m 2 are total mass of two B2V = W23 I 2 (8)
cylinder rods respectively, A 1 , A 2 are efficient area of
I1 = W12 ( F0W11 − F1W14 ) (9)
two cylinders respectively, V 1 , V 2 are overall volume
of pipeline and cylinder of two cylinders respectively, I2 = W22 ( DW
0 21 − D2W24 )
(10)
K 1 B , K 2 B are flow coefficient of two proportion The elastic force of load spring:
valves respectively, K 1 p , K 2 p are flow-pressure Fs = K s ( D1 − D 2 ) (11)
coefficient of two proportion valves respectively,
C 1 t , C 2 t are total leakage coefficient of two cylinders 2.3. The system transfer function
respectively, β e is the bulk elastic module of
hydraulic oil, K s is the elasticity coefficient of spring, The coefficient of feedback W14 and W24 are relative
F s is the elastic force of elastic load. to the operation of system feedback signal , We need to
study system open circuit transfer function at first ,
D0 This two coefficients is set to zero while deducing
W21 system open circuit transfer function .
F0 For the left side force application subsystem , It have
two input and single output , The two input signals are
W11 W22
: The given driving force of the left side cylinder F0 ;
I1
I2 −
The practical position of the right side cylinder D 2 .
W12
W24 The single output signal is: The practical driving force
D1
− of the left side cylinder. For the right side position
control subsystem , It have two input and single output
, The two input signal are : The given cylinder rod
D2 position of the right side system D 0 ; The elastic force
W14
of elastic load Fs . The single output signal is: The
Fig.1 System circuit practical position of right side cylinder rod D 2 .
According to equations (1)(3)(5)(7)(9) , We can
2.2. The system fundamental equations
deduce the transfer function relative to input signal F0
This is a typical synthesis control system, It has two and D 2 , output signal F1 .
input signals: the given driving force and the given F1 ( s ) =
position, two output signals: practical driving force and
⎛ m1 2 ⎞ (12)
practical position. According to the basic working ⎜ s + 1 ⎟ W 1 2W 1 3 K 1 B A1W 11 ⋅ F0 − A12 s ⋅ D 2
theory of electro-hydraulic servo system[1,8], The load ⎝ Ks ⎠
flow equations of the two proportion valves as follows: m 1V1 3 m 1 ⎛ V ⎞ A2
s + K 1t s 2 + ⎜ 1 + 1 ⎟ s + K 1t
Q 1 L = K 1 B B 1V − K 1 p P1 L (1) 4βeK s Ks ⎝ 4βe K s⎠
Q2 L = K 2 B B2V − K 2 p P2 L (2) Where K 1t = K 1 p + C 1t is total flow-pressure
Where Q1 L , Q 2 L are load flow of two cylinders coefficient of left side force application subsystem.
respectively , B1V , B2V are opening of the two According to equations (2)(4)(6)(8)(10) , We can
deduce the transfer function relative to input signal D 0
proportion valve cores respectively , P1 L , P2 L are load
pressure of the two proportion valves respectively . and Fs , output signal D 2 .
The load flow equations of the two cylinders: D2 (s ) =
V (3) ⎛ V2 ⎞
Q1 L = A1 sD1 + C1t P1 L + 1 sP1 L
4βe ( A2 K 2 BW 23W 22W 21 ) D 0 − ⎜ s + K 2 t ⎟ Fs (13)
⎝ 4βe ⎠
V2 (4)
Q2 L = A2 sD2 + C 2 t P2 L + sP2 L m 2V 2 3 ⎛K V ⎞
4βe s + m 2 K 2 t s 2 + ⎜ s 2 + A22 ⎟ s + K 2 t K s
4βe ⎝ 4βe ⎠
The movement equations of two cylinder rods:
Where K = K + C is the total flow-pressure
m1 s 2 D1 = F1 + K s ( D 2 − D1 ) (5) 2t 2p 2t

coefficient of right side position control subsystem.

m2 s 2 D2 = A2 P2 L + Fs (6)
The transfer function equations (12)(13) do not
The equations relative to feedback and gain: contain the feedback elements, they are open circuit

74
transfer function. The system control block diagram ⎛ V ⎞⎛ 1 ⎞ (15)
can be draw out according to the system foundation T2 = ⎜ 2 s + K 2 t ⎟⎜ ⎟
equations, Fig.2. ⎝ 4βe ⎠⎝ W23 K 2 B ⎠
The system structure has changed after
compensation. The force application control system
3. Construct the compensated state-space changed into a SISO system which input signal is F 0
model and output signal is F1 . The position control system
3.1. Compensated by unchanged structure changed into a SISO system which input signal is D 0
theory and output signal is D 2 . The transfer function
equations (12) (13) change to:
The literature [1] discussed unchanged structure ⎛ ⎞
theory in detail, and the structure compensation has ⎜ W ⋅W W K A ⎟ (16)
F1 = ⎜ 11 12 13 1 B 1 ⎟ ⋅ F0
been done to the electro-hydraulic position control ⎜ V1 ⎟
⎜ s + K 1t ⎟
system and the electro-hydraulic force application ⎝ 4 βe ⎠
system respectively. Applying suitable structure W 21 ⋅ W 22W 23 K 2 B
D2 = ⋅ D0 (17)
compensation can improve the system response speed, A2 s
eliminate strong outer interference. In this design, Two
structure compensation elements have been applied
simultaneously, in order to eliminate the cross 3.2. The compensated state-space model
influence between force application subsystem and
position control subsystem. The blocks connected to The state-space equation is the foundation of
dashed line in Fig.2 are the structure compensation modern control theory. For the multi input and multi
blocks. output system, the state-space model is more suitable.
According to equation (16), set the given driving
m1s − force F 0 = u 1 , the practical driving force F1 = x 1 ,
D2 leads to the following equation:
Ks s s
m1 s 2 + K s m1 s 2 + K s
x1 = a11 x1 + b11u 1 (18)
D1 4 β e K 1t 4 β eW11 ⋅ W12W13 K 1 B A1
A1 Where a 1 1 = − b11 =
V1 V1
W13 K1B A12 According to equation (17), set the given cylinder
B1V
− 1 rod position D 0 = u 2 , the practical cylinder rod
F1
W12 W13 K1B A1 V1
s + K1t position D 2 = x 2 , leads to the following equation:
4βe
F0 x 2 = b22 u 2 (19)
W11
W 2 1W 2 2 W 2 3 K
D0
W21
Fs
1 A2
− Where b 2 2 =
A2
2B

m2 s2 A2
P2L From the equations (18)(19), we can deduce the
⎛ V2 ⎞⎛ 1 ⎞ following state-space equations:
⎜ s + K2t ⎟⎜ ⎟ V2 K
⎝ 4βe ⎠⎝ W23 K2 B ⎠
s + 2t x = Ax + Bu
4β e A2 A2 (20)
− y = Cx
B2V K 2 B D2
W22 W23 1 s Where
A2 ⎡x ⎤ ⎡F ⎤ ⎡a 0⎤ ⎡b 0⎤
x = ⎢ 1⎥ = ⎢ 1 ⎥ , A = ⎢ 11 , B = ⎢ 11 ⎥
Fig.2 Frame chart of the system ⎣ x2 ⎦ ⎣ D2 ⎦ ⎣0 0 ⎥⎦ ⎣ 0 b22 ⎦
The input signal of the compensation element in the ⎡u ⎤ ⎡F ⎤ ⎡1 0⎤
, u = ⎢ ⎥ = ⎢ ⎥ , C=⎢
1 0
force application system is the left side cylinder rod
⎣ u 2 ⎦ ⎣ D0 ⎦ ⎣0 1⎥⎦
velocity D 1 , The compensated transfer function is : The equation (20) is the compensated state-space
A1 (14) model of the synthesis control of force and position
T1 =
W 13 K 1B control system, this model is based on two
The input signal of the compensation element in the compensated elements added in system, the additional
position control system is right side cylinder load compensated elements are relative to the accuracy of
pressure P2 L , The compensated transfer function is : the compensated model.

75

4. Realization of The linear quadratic
optimized control
4.1. Application of the linear quadratic
optimized control
The control algorithm which is presented by the
linear quadratic optimized control is the function of
system state variables, the closed loop optimized
control by state feedback can be obtained.
Because the synthesis control of force and position
in electro-hydraulic servo system had two input control
variables, at the same time, asked two output variables
meet the requirement, Introduced the performance
fonctionelle is convenience. Due to the state variables
change at any moment, we change equation (20) as
follow:
x ( t ) = Ax ( t ) + Bu ( t ) (21)
y ( t ) = Cx ( t )
The initial condition of the system is:
x (t0 ) = x0 (22)
The quadratic optimized control problem as follows :
Set the demand output is z ( t ) , the output error is
e (t ) = z (t ) − y (t ) (23)
The performance fonctionelle is
1 tf
J = ∫ ⎡⎣ eT ( t ) Q1e ( t ) + u T ( t ) Q2 u ( t ) ⎤⎦ dt (24)
2 t0
Where Q 1 , Q 2 are positively definite matrixes.
The explanation are: in time segment ⎡⎣ t 0 , t f ⎤⎦ ,
looking for the optimized control u ( t ) , make the
performance fonctionelle reach minimum value . The
explanation on the synthesis control of force and
position control are: In certain time segment, make the
practical driving force and the practical position accord
with the given driving force and the given position , at
same time , The consumed power should be less as far
as possible .
4.2. The realization of the optimized control
The above optimized control problem belongs to
multi tracker problem in the optimized control. The
real substance is to keep the lesser error, so to reach the
optimized control of consumed power and error.
Solving the equations should introduce Riccati
equation, the solution as follow[9] :
The optimized control:
u * ( t ) = − Q 2− 1 B T P x ( t ) + Q 2− 1 B T g (25)
Where P, g are satisfied with the follows equation:
− PA − AT P + PBQ2−1 BT P − C T Q1C = 0 (26)
76
−1
g ≈ ⎡⎣ P B Q 2− 1 B T − A T ⎤⎦ C T Q1 z (27)
The state curve on optimized control submits to the
follow equation:
x ( t ) = ⎡ A − B Q 2− 1 B T P ⎤ x ( t ) + B Q 2− 1 B T g
⎣ ⎦
(28)
The optimized control u * ( t ) contained state
variable x ( t ) , This optimized control formula make
use of all state information of system , construct the
closed loop feedback control on base of all state
variable , and this formula is achieved easily in electro-
hydraulic servo system , we can measure the pressure
in the two sides of cylinder , The driving force of
cylinder can be worked out , The other variables can be
measured easily .
5. The simulation of system
In this design, we adopted physical model
simulation, the simulation parameters as follows: the
given driving force curve is a standard sinusoid, the
period is 2s, the amplitude is 10N, the minimum is
45N, the maximum is 55N. The given position curve is
a standard sinusoid too, the period is 1s, the amplitude
is 10mm, the minimum is -5mm, the maximum is
5mm.
The signals of given force and given position are
standard sinusoid signals, but the periods are different,
the coupling influence is seriously. Keep the other
conditions no change, the simulation executed on non
compensated model and compensated model.
Practical force (N)
Given force (N)
Time (s)
Fig.3 The given force and practical force before
compensation
The Fig.3 shows the curves of given driving force
and practical driving force on base of non compensated
model , Fig.4 shows the curves of given driving force
and practical driving force on base of compensated
model .The Fig.5 shows the curves of given position
and practical position on base of non compensated
model, Fig.6 shows the curves of given position and
practical position on base of compensated model.
Compared the Fig.3 and Fig.4, we can deduce that
the error reduce after model compensation, and the
error of force application system disturbed by position
influence near by the minimum force value is
restrained. Compared the Fig.5 and Fig.6 , The
practical position curve is more close to given position
curve , The speed of system response increase, the
position error decrease.
Practical force (N)
compensated state-space model is a simplified double
input and double output system., introduced the linear
quadratic performance fonctionelle, the optimized
control formula of the synthesis control of force and
position on total state feedback have been deduced,
The simulation result indicated, utilized compensated
state-space model, The control performance of system

have been improved obviously.

Given force (N)

References
[1] Changnian LIU, Optimization design theory of hydraulic
servo system, Beijing: Metallurgical industry press, 1989, 08,
pp. 229-282.
Time (s)
Fig.4 The given force and practical force after [2] Yongjian FENG, “The Self-adaptation Control Model for
compensation Eliminating Surplus Force Caused by Motion of Loading
System”, MACHINE TOOL & HYDRAULICS, no.4, pp.53-
55, 2002.
Practical position (mm)
Given position (mm)

[3] Xinmin WANG, Weiguo LIU, “ Neural-network Internal

Feedback Control for Electro-hydraulic Servo Loading ”,
ACTA AERONAUTICA ET ASTRONAUTICA SINICA, vol.
28, no. 3, pp. 690-694, 2007.

[4] Liwen WANG, Yuhua LI, “ A Study of Suppress

Strategy for Extra Force on Control Loading Hydraulic Servo
System ” , MACHINE TOOL & HYDRAULICS , vol.36, no.1,
Time (s) pp. 87-89, 2008.
Fig.5 The given position and practical position
[5] Changchun LI, Yadong MENG, Xiao-dong LIU,
before compensation XinZHOU, “Motion Synchronization of Electro-hydraulic
Servo System”, ACTA ARMAMENTII, vol.28, no.6, pp. 765-
768, 2007.
Practical position (mm)
Given position (mm)

[6] Yun ZHANG , Zhi LIU, “ Neural network robotic control

for a reduced order position/force model ” , CONTROL
THEORY & APPLICATIONS , vol. 24, no. 4, pp. 541-545,
2007.

[7] Xianlun WANG, Yuxia CUI, Jing HUANG, “ Intelligent

Force/Position Control of Robotic Manipulators Based on
Fuzzy Logic ”, JOURNAL OF SIMULATION, vol. 19, no.11,
Time (s) pp. 2467-2471, 2007.
Fig.6 The given position and practical position after
compensation [8] Hongren LI, Hydraulic control system, Beijing: National
defence industry press, 1981, 06, pp. 55-67.
6. Conclusion [9] Bao LIU, Modern control theory, Beijing: China machine
press, 2004, 07, pp. 272-292.
The force control circuit and the position control
circuit in one electro-hydraulic servo system are
compensated by unchanged structure theory
simultaneously, established system state-space model
on base of the compensated transfer function, this

77
 International Conference
on Computer Engineering and Technology

Session 2
 2009 International Conference on Computer Engineering and Technology
Features Selection using Fuzzy ESVDF for Data Dimensionality Reduction
Safaa Zaman and Fakhri Karray
Electrical and Computer Engineering Department
University of Waterloo
Waterloo, Canada
szaman@pami.uwaterloo.ca, karray@pami.uwaterloo.ca
Abstract—This paper introduces a novel algorithm for features
selection based on a Support Vector Decision Function (SVDF)
and Forward Selection (FS) approach with a fuzzy inferencing
model. In the new algorithm, Fuzzy Enhancing Support Vector
Decision Function (Fuzzy ESVDF), features are selected
stepwise, one at a time, by using SVDF to evaluate the weight
value of each specified candidate feature, then applying FS
with the fuzzy inferencing model to rank the feature according
to a set of rules based on a comparison of performance. Using a
fast and simple approach, the Fuzzy ESVDF algorithm
produces an efficient features set and, thus, provides an
effective solution to the dimensionality reduction problem in
general. We have examined the feasibility of our approach by
conducting several experiments using five different datasets.
The experimental results indicate that the proposed algorithm
can deliver a satisfactory performance in terms of classification
accuracy, False Positive Rate (FPR), training time, and testing
time.
Keywords-Features selection, features ranking, support
vector machines, support vector decision function, Sugeno fuzzy
inferencing model.
I. INTRODUCTION
Dimensionality reduction [11, 12] is an important topic in
machine learning. Elimination of useless (irrelevant and/or
redundant) features enhances the accuracy of the
classification while speeding up the computation. This
elimination also simplifies the classification by searching for
the subset of features that best classifies the training set, and
allows the extraction of perfectly interpretable rules, thus
improving the overall performance of the classifier and
overcoming many problems such as the risk of “overfitting.”
Moreover, it helps us to understand the data and reduces the
measurements and storage requirements.
Current dimensionality reduction methods can be
categorized into two classes: features extraction and features
selection. Features extraction [4, 5] involves the production
of a new set of features from the original features in the data
through the application some mapping. The dominant
features extraction techniques are Principle Component
Analysis (PCA), Linear Discriminant Analysis (LDA). On
the other hand, features selection [6, 7, 8, 9, 10] selects the
“best” subset of the original features. It reduces the number
of features and removes irrelevant, redundant, or noisy data.
In terms of features selection, several researchers have
proposed identifying important features through wrapper and
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.36
filter approaches. The wrapper method [13, 14, 15, 16]
exploits a machine learning algorithm to evaluate the
goodness of features or feature set. It provides better
performance in the selection of suitable features since it
employs a performance of learning algorithm as an
evaluation criterion. The most used wrapper methods are
Forward Selection (FS), Backward Elimination (BE), and
Genetic search. In contrast, the filter method does not use
any machine learning algorithm to filter out irrelevant and
redundant features; instead, it uses the underlying
characteristics of the training data to evaluate the relevance
of the features or feature set by some independent measures
such as distance measures, correlation measures, and
consistency measures [17, 18]. The most used techniques in
this area are the Chi-Square measure, information gain, and
odds ration. In general, wrapper approaches demand heavy
computational resources, but they can achieve better results
than filters due to the fact that they are tuned to the specific
interaction between an induction algorithm and its training
data. However, they tend to be much slower than feature
filters because they must repeatedly call induction algorithm
and must be re-run when a different induction algorithm is
used.
This paper introduces a new features selection and
ranking method. This method, Fuzzy Enhancing Support
Vector Decision Function (Fuzzy ESVDF), integrates the
Support Vector Decision Function (SVDF) and Forward
Selection (FS) approaches with the fuzzy inferencing model
to select the best features set as an application input. It is
simple and fast. Moreover, the Fuzzy ESVDF improves
overall system performance of many applications.
II. BACKGROUND
This section introduces a brief description of Support
Vector Machines (SVMs) and features ranking, FS approach
and fuzzy inferencing systems.
A. SVMs and Feature Ranking
SVM is a machine learning method that is based on
statistical learning theories [20]. The SVM classifier
computes a hyperplane that separates the training data in two
different sets corresponding to the desired classes. Since
there are often many hyperplanes that separates the two
classes, the SVM classifier picks the hyperplane that
maximizes the margin. Training SVM is a quadratic
81
optimization problem with bound constraints and linear
equality constraints.
n
1 n
max W (α ) = ∑ α i − ∑ αiα j yi y j K ( xi , x j ) (1)
i =1 2 i =1, J =1
n inferences or decisions using its reasoning mechanism, “the
Subject to C ≥ α i ≥ 0, ∑ α i yi = 0 (2)
i =1
For testing with a new data z :
s consists of three basic components:
f = w, φ ( z ) + b = ∑ α tj ytj K ( xtj , z ) + b (3)
j =1
s the history of solutions to date. Therefore, any new
w = ∑ α tj ytiφ ( xtj ) (4)
j =1
Where α is a lagrange multiplier, x is the input, y is the
output, b is a bias value, K is a kernel function, and C is
trade off parameter between error and margin.
The contribution weight of each feature can be calculated
using Support Vector Decision Function (SVDF). This
decision function is a simple weighted sum of the training
patterns plus a bias, b , as in (5). The value of F ( X )
depends on both the X i and Wi values, where the absolute
value of Wi measures the strength of the classification. If Wi
is a large positive/negative value, then the i th feature is one
of the major features for the positive/negative class;
otherwise, if it is close to zero, the i th feature does not
contribute significantly to the classification. Moreover, if
X i has a positive value, then it belongs to the positive class;
otherwise, it belongs to the negative class.
n is therefore required on the output in order to produce a crisp
F ( X ) = ∑ Wi X i + b (5)
i =1
B. Forward Selection (FS) Approach
The Forward Selection (FS) approach [19] selects the
most relevant variables (features/attributes) based on
stepwise addition of variables. In the FS approach, variables
are progressively incorporated into larger and larger subsets.
This approach yields nested subsets of variables, but it is not
guaranteed to find the optimal solution.
The FS approach starts with the initial variable being
selected for the features subset, and then the predictor
variables are added one by one to the features subset.
Through this process, if the added variable increases the
system evaluation criterion, it will stay in the features subset;
otherwise, it will be removed. The process continues until
either all variables are used up or some stopping criterion is
met. The FS approach does not necessarily find the best
combination of variables; however, it results in a
combination that comes close to the optimum solution. In
our case, the initial variable (feature) is the feature with
maximum weight value (the weight is calculated using (4)),
and the system evaluation criterion is the system
82
performance results (the classification accuracy and the
training time).
C. Fuzzy Inferencing Systems
A fuzzy inferencing model (also called knowledge based
system) is a system that is able to make perceptions and new
inference engine”, by interpreting the meaning of the new
coming information to the database within the capabilities of
the existing knowledge base [21]. An inference model
1. Database, which is a short term memory that contains
the current status of the problem, inference states, and
information that is generated from external sources
(sensors or human interfacing) is stored in the database.
2. Knowledge base, which contains knowledge and
expertise in the specific domain. Specifically, it contains
domain specific facts and heuristics that are useful for
solving problems.
3. Inference engine, which is the driver program of the
knowledge based system. Depending on the new coming
data to the database, the inference engine operates on the
knowledge in the knowledge base to solve problems and
arrives to conclusions. This reasoning mechanism also
involves a fuzzification process that transforms input
values from crisp numerical values into fuzzy sets, and a
defuzzification process that maps output control values
from fuzzy sets back to crisp numerical values.
There are various fuzzy inferencing models. The most
common ones are the Mamdani fuzzy model, and Sugeno
fuzzy model [21]. The output of the first model is a fuzzy set
that is obtained by aggregating the qualified fuzzy sets using
the min max compositional rule of inference. Defuzzification
control signal. However, the second model (Sugeno model)
generates crisp consequence for each rule and then the
overall crisp output is obtained using the weighted average
method. Sugeno model is used by the work of in this paper.
III. PROPOSED APPROACH (FUZZY ESVDF)
We propose a new features selection approach based on
SVDF and FS with a fuzzy inferencing model. This
approach, Fuzzy ESVDF, selects the features in a stepwise
way, one at a time, while evaluating the weight (rank) of
each specified candidate feature by using (4). Through this
process, if the added feature increases the system
performance, it will stay in the features subset; otherwise, it
will be removed. As shown in Fig. 1, the algorithm starts by
picking three features from the features set (S1) with the
highest weight values (S1 contains all the features with
weight values equal to or greater than one; the weight value
is calculated using (4)) and putting them in the features set
(S2) and then calculates the classification accuracy and
training time for features set (S2). The feature with the next
highest weight value from the features set (S1) is added to
the features set (S2) while calculating their performance
metrics. Through this process, two types of comparisons are
made: a local fuzzy comparison and a global fuzzy
comparison. The local fuzzy comparison compares the
performance of the features set (S2) with the performance of
the previous features set (S2); if the first value is less than
the second value, the added feature is ignored; otherwise, it
is kept in the features set (S2). In the global fuzzy
comparison, the classification accuracy of the features set
(S2) is compared with the global accuracy, which is equal to
the minimum of two values: the accuracy of all the features
and the accuracy of the features set (S1). If the classification
accuracy of the features set (S2) is equal to or greater than
the global accuracy value, the algorithm will stop and the
features set (S2) will be the selected features set.
The local fuzzy comparison is ranked according to a
fuzzy system that takes two inputs: the percentage of
increase or decrease in the training time as one input, and the
percentage of increase or decrease for the accuracy as the
second input. It compares the performance of the features set
(S2) with the performance of the previous features set (S2).
The first and the second input variables (percentage of
change in the training time and in the accuracy) are
represented by three fuzzy sets: “increase,” “same,” and,
[1] Calculate the Global Accuracy
Calculate the accuracy and training time of all (41) features
(Accuracy41, Train41),
Calculate the accuracy and training time of the features with
weight >= 1 (Accuracy, Train),
Pick the Global accuracy
If Accuracy41 >= Accuracy
Global = Accuracy
Else
Global = Accuracy41
End if
[2] Create the features set
Sort the features set(S1)in descending order depend on its
weight values,
Pick the first three features as an initial features set (S2),
Calculate the Accuracy and Training time of S2 (Accuracy1,
Train1)
If (Global equal or less than Accuracy1)
Exit;
Else
continue_loop= 1,
count_loop = 0;
Do while (continue_loop ==1) & (count_loop <= length(S1))
Add the next feature f(i) from S1 into S2,
Calculate the accuracy and training time of S2
(Accuracy2,Train2)
If (Accuracy2 less than Accuracy1) and (Train2 greater than
Train1)
Remove f(i) from S2,
count_loop = count_loop + 1;
Else
Accuracy1 = Accuracy2,
Train1= Train2,
count_loop = count_loop + 1;
If (Global equal or less than Accuracy1)
continue_loop = 0,
End if
End if
End while
End if
“decrease” with their corresponding membership functions
as shown in Fig. 2. “Increase” refers to the case where the
percentage of change (accuracy and time calculated by current selected
features – accuracy and time calculated by previous selected features) in the
training time and accuracy is slightly positive. This means
that the training time and accuracy slightly increase after a
feature is added. “Same” refers to the case where the training
time and accuracy remain almost the same. The system has
one output ranging from “0” to “1” where “0” represents a
non-important feature, and “1” represents an important
feature in the detection process.
The knowledge base is implemented by means of “if-
then” rules. Nine rules are needed to describe the system and
rank each feature as “important” or “non-important,”
according to the following rules:
1. If training time decreases and accuracy decreases,
then the feature is non-important
2. If training time decreases and accuracy has no
changes, then the feature is important
3. If training time decreases and accuracy increase, then
the feature is important
4. If training time has no changes and accuracy
decreases, then the feature is non-important
5. If training time has no changes and accuracy has no
changes, then the feature is important
6. If training time has no changes and accuracy
increase, then the feature is important
7. If training time increases and accuracy decrease, then
the feature is non-important
8. If training time increases and accuracy has no
changes, then the feature is non-important
9. If training time increases and accuracy increases,
then the feature is non-important
The global fuzzy comparison compares the classification
accuracy of the features set (S2) with the global accuracy.
The comparison is ranked according to a fuzzy system that
takes only one input variable (percentage of change in the
accuracy). This input variable is represented by three fuzzy
sets: “increase,” “same,” and “decrease” with their
corresponding membership functions, as shown in Fig. 3.
“Increase” refers to the case where the percentage of change
(selected features set accuracy – global accuracy) in the
accuracy is slightly positive. This means that the training
accuracy slightly increases after a feature is added. “Same”
refers to the case where there is no change in accuracy. The
system has one output ranging from “0” to “1” where “0”
represents a loop to continue, and “1” represents a loop to
stop.
The knowledge base is implemented with three “if-then”
rules. Only three rules are needed to describe the system and
decide whether or not to add features to the features set (S2):
1. If accuracy increases, then stop adding feature
2. If accuracy unchanges, then stop adding feature
3. If accuracy decreases, then continue adding feature

[3] The selected features set = S2

Figure 1. The Fuzzy ESVDF Algorithm.

83

Input 1: Time
Input 2: Accuracy
Output: Feature Rank
Figure 2. Sugeno Fuzzy Inferencing Model for the Local Comparison
IV. EXPERIMENTS AND RESULTS
For evaluating the performance of our proposed
approach, we choose five different datasets; four datasets are
taken from the UCI Irvine Machine Learning Repository
[22], and the fifth one is KDD Cup 1999 data [23]. In this
section, we initially describe the content of different datasets
and the experimental settings. Follow by some experimental
results and discussion.
A. Datasets Description
The Five real datasets are considered; four datasets are
taken from the UCI Irvine Machine Learning Repository
[22], and the fifth one is KDD Cup 1999 data [23]. The
objective is to select a subset for the features using the Fuzzy
ESVDF algorithm, and then to evaluate these selected
features using both Neural Networks (NNs) and Support
Vector Machines (SVMs).
The first dataset, the SPECT Heart Dataset, describes the
diagnosis of cardiac Single Proton Emission Computed
Tomography (SPECT) images [22]. It contains 267 samples,
of which 55 are normal (20.6 %) and 212 are abnormal (79.4
%). Each instance is characterized by 44 attributes.
The second dataset, Wisconsin Diagnostic Breast Cancer
(WDBC), describes characteristics of the cell nuclei present
in the image as either benign or malignant [22]. This dataset
contains 569 samples; of which 357 are benign samples
(62.74 %) and 212 are malignant samples (37.26 %). Each
instance is characterized by 30 real-value attributes.
84
In the third dataset, the Hill and Valley Dataset, each
record represents 100 points on a two-dimensional graph.
When plotted in order (from 1 through 100) on the Y co-
ordinate, the points will create either a Hill or a Valley [22].
This dataset contains 1212 samples; of which 612 are hill
samples (50.5 %) and 600 are Valley samples (49.5 %). Each
instance is characterized by 100 real-value attributes.
The fourth dataset, Wisconsin Breast Cancer (WBC),
describes characteristics of the cell nuclei present in the
image as either benign or malignant [22]. This dataset
contains 699 samples; of which 458 are benign samples
(65.52 %) and 241 are malignant samples (34.48 %). Each
instance is characterized by nine attributes.
The last dataset, KDD Cup 1999 data [23], contains
TCP/IP dump data for a network by simulating a typical U.S.
Air Force LAN to configure and evaluate intrusion detection
systems. This dataset contains 6000 samples; of which 3000
are normal samples (50 %) and 3000 are attack samples (50
%). Each instance is characterized by 41 attributes.
B. Experimental Settings
Our experiment is divided into two main steps. First, we
apply the proposed algorithm (Fuzzy ESVDF) to select the
appropriate features set for each dataset. Second, we validate
the results by using any classifier type.
In the first step, the proposed algorithms were applied ten
times with training and testing data. Each time, about 40% of
the samples were randomly selected as the testing dataset;
the remaining 60% were used as the training dataset.
NN and SVM classifiers are used to evaluate the proposed
algorithms in the second step. We carried out five validation
experiments for each different dataset (SPECT Heart
Dataset, WDBC dataset, Hill and Valley dataset, WBC
dataset, and IDS dataset). Each experiment was repeated ten
times with a random selection of the training and the testing
data with different ratios, which are ((training %) / (testing
%): 50/50, 40/60, 60/40, 30/70, and 70/30).
Input 1: Accuracy
Output: Action
Figure 3. Sugeno Fuzzy Inferencing Model for the Global Comparison
C. Experimental Results TABLE II. COMPARISON OF DIFFERENT DATASETS USING NNS

Fuzzy ESVDF was applied to the different datasets Dataset No. Attributes Acc. FPR
Training Testing
(SPECT Heart Dataset, WDBC dataset, Hill and Valley Time Time
dataset, WBC dataset, and IDS dataset) to select the best SPECT Selected Set (5) 74.14 % 0.082 33.940 0.006
features set for the application. To evaluate our approaches, Heart Complete Set (44) 66.77 % 0.189 325.534 0.030
we used NN and SVM classifiers to classify a record as Selected Set (4) 94.85 % 0.036 38.031 0.012
being either zero or one (binary classification). The results WDBC
Complete Set (30) 94.63 % 0.069 206.144 0.013
of the classifier SVMs for Fuzzy ESVDF for all datasets are
Hill and Selected Set (11) 77.93 % 0.085 105.638 0.019
presented in Table 1. The results of the classifier NNs for
Fuzzy ESVDF for all datasets are presented in Table 2. The Valley CompleteSet(100) 75.84 % 0.270 2488.83 0.047
different datasets are compared with respect to different Selected Set (3) 95.91 % 0.035 43.738 0.016
WBC
performance indicators: number of features, classification Complete Set (9) 95.05 % 0.059 72.475 0.015
accuracy, False Positive Rate (FPR is the frequency with Selected Set (7) 99.64 % 0.003 221.928 0.047
which the application reports malicious activity in error), IDS
Complete Set (41) 99.65 % 0.002 911.680 0.075
training time, and testing time. Table 3 compares execution
times for Fuzzy ESVDF approach for the five different
datasets. For the WDBC dataset, by using SVMs, the classification
accuracy, training time, and testing time based on the
D. Discussion selected features set are very near to the accuracy of the
As shown in Table 1 and Table 2, there is a dramatic system based on the complete features set. However, there is
reduction in the number of features for all datasets after the obvious improvement in FPR, which is reduced from
application of Fuzzy ESVDF. For the SPECT Heart Dataset, 0.128536 to 0.03932, as shown in Table 1. Table 2 shows
the number of features is reduced from 44 to 5 (cut nearly significant improvement in training time for the selected
88.6 %). For the WDBC dataset, the number of features is features. The proposed algorithm reduces the training time
reduced from 30 to 4 (cut nearly 86.7 %). For the Hill and from 206.144 sec to 38.031 sec. However, classification
Valley dataset, the number of features is reduced from 100 to accuracy, FPR, and testing time in both experiments (using
11 (cut nearly 89 %). For the WBC dataset, the number of the four features selected from the Fuzzy ESVDF algorithm,
features is reduced from 9 to 3 (cut nearly 66.67 %). Finally, and using the entire 30 features) are nearly the same.
for the IDS dataset, the number of feature is reduced from 41 For the Hill and Valley dataset, Table 1 shows a
to 7 (cut nearly 82.92 %). significant improvement in classification accuracy, FPR,
For the SPECT Heart Dataset, by using SVMs, the training time, and testing time for the selected 11 features
classification accuracy based on the selected features set is (attributes) from the complete 100 features set. On the other
76.73 %, which is better than the classification accuracy for hand, Table 2 shows that the classification accuracy for both
the complete features set, which is equal 69.43 %. Moreover, features sets (11 selected features and the entire 100 features)
FPR, training time, and testing time are also improved, as are nearly the same. However, there is obvious improvement
shown in Table 1. The NN classifier shows obvious in training time (it is reduced from 2488.825 sec to 105.6376
improvement in training and testing time. The five selected sec), testing time (it is reduced from 0.0186 sec to 0.0128
features from the Fuzzy ESVDF algorithm reduce the sec), and FPR (it is reduced from 0.270454 sec to 0.084815
training time from 325.53 sec to 33.94 sec and the testing sec).
time from 0.03 to 0.0064. The classification accuracy and For the WBC dataset, by using SVMs, the overall system
FPR are also improved, as shown in Table 2. performance is improved based on the selected features set,
as shown in Table 1. On the other hand, Table 2 shows no
significant improvement in classification accuracy, FPR, and
TABLE I. COMPARISON OF DIFFERENT DATASETS USING SVMS testing time between using the selected three features and all
Training Testing nine features. However, the training time is improved (it is
Dataset No. Attributes Acc. FPR reduced from 72.4746 sec to 43.7376) based on the selected
Time Time
SPECT Selected Set (5) 76.73% 0.063 0.33 0.01
features.
Finally, for the IDS dataset, for the SVM classifier,
Heart Complete Set (44) 69.43% 0.134 0.376 0.015
classification accuracy and FPR for the selected seven
Selected Set (4) 96.65% 0.039 0.452 0.00 features does not show significant differences to that of using
WDBC
Complete Set (30) 96.42% 0.129 0.476 0.012 all 41 features. However, there is an obvious improvement in
Hill and Selected Set (11) 69.13% 0.109 23.588 0.088 the training (it is reduced from 5.182 sec to 2.41 sec) and
Valley CompleteSet(100) 66.77% 0.591 42.45 0.152 testing time ( it is reduced from 0.17 sec to 0.054 sec), as
Selected Set (3) 96.75% 0.019 0.678 0.012
shown in Table 1. On the other hand, the NN classifier
WBC shows significant improvement in training and testing time.
Complete Set (9) 94.72% 0.049 0.85 0.016
The seven selected features from the Fuzzy ESVDF
Selected Set (7) 99.61% 0.002 2.41 0.054
IDS algorithm reduce the training time from 911.68 to 221.93 sec
Complete Set (41) 99.70% 0.006 5.182 0.17 and the testing time from 0.075 to 0.047. The classification
and FPR in both experiments are nearly the same.

85
 TABLE III. EXECUTION TIME COMPARISON FOR THE DIFFERENT
DATASETS
Dataset
No. of No. of Execution Time
Complete Set Selected Set (sec)
SPECT Heart 44 5 125.093
WDBC 30 4 87.215
Hill and Valley 100 11 3000.025
WBC 9 3 0.725
IDS 41 7 134.792 sec
Comparing the different datasets according to execution
time, Table 3 shows that the proposed approach, Fuzzy
ESVDF, becomes slow when the number of features
increases to 100 features. As is shown, when the number of
features is around 50 features, the algorithm is reasonably
fast and when this number doubles (to 100 features),
execution time increases greatly (more than doubling). Also,
in the case of WBC dataset, the number of features is nine
and when the number of features triples (the case of the
WDBC dataset), the execution time more than triples. On the
whole, this amount of time does not depend on the number
of features only; but, it depends on how fast SVMs are,
because the ranking approach depends on the system
performance (classification accuracy and training time) that
is calculated by SVMs. Moreover, the SVDF used in this
approach also depends on SVMs.
In summary, the experimental results demonstrate the
feasibility of the proposed approach. The proposed approach,
Fuzzy ESVDF, for features selection based on SVMs and FS
with the fuzzy inferencing model, gives the best performance
in terms of training and testing time while retaining high
classification accuracy regardless of the classifier used.
Consequently, the selected features subset is representative
and informative and, thus, can be used to replace the
complete features. In addition, this approach is considered to
be a selection features approach regardless of the type of
classifier used, making this approach a suitable features
selection method for many applications. Moreover, the
proposed algorithm is simple and does not require that many
parameters be initialized, facilitating the retention or
modification of the system design and allowing this model to
be used in a real time environment.
V. CONCLUSIONS AND FUTURE WORKS
SVDF is used to rank the input features by giving a
weight value for each of them; however, by using the
weights only, as proposed in previous works [14, 16], we are
unable to specify the appropriate features set for a detection
process because selecting features with the highest rank
values (weight) can not guarantee that combining these
features can create the best features set based on the
correlations among candidate features. Moreover, limiting
the number of selection features to a specific value (e.g., 6 or
11 as it is mentioned in the previous works) as an indicator
for highest rank value may affect system performance. Our
new approach overcomes these limitations by proposing a
Fuzzy Enhanced Support Vector Decision Function (Fuzzy
ESVDF). The Fuzzy ESVD approach improves the
86
classification process by optimizing the selected features
SVDF and studying the correlation between features through
the application of the Forward Selection (FS) approach. The
fuzzy inferencing model is used to accommodate the
learning approximation and small differences in decision
making steps in the FS approach.
The proposed approach, Fuzzy ESVDF, applied the
SVDF based on FS with the fuzzy inferening model to select
the most effective features set for an application. The
proposed approach (Fuzzy ESVDF) has many advantages
that make it attractive for many features selection
applications.
First, employing a reduced number of features by SVMs
may be more advantageous than other conventional features
selection methods [13, 15, 24, 25]. The advantage becomes
more conspicuous for many applications as our experiments
show. With SVMs, satisfactory performance could be
obtained much more easily than with other approaches.
Second, evaluating features weights through SVDF and
studying the correlation between these features through the
application of the FS algorithm enables the approach to
select efficiently the appropriate features set for the
classification process.
Third, ESVDF is considered to be a selection features
approach regardless of the type of classifier used, making
this approach a suitable features selection method for other
applications, as we showed through our experiments.
Finally, this approach is simple and efficient and it does
not require parameters initialization.
To evaluate the proposed approach, we used SVM and
NN classifiers with five different datasets (SPECT Heart
Dataset, WDBC dataset, Hill and Valley dataset, WBC
dataset, and IDS dataset). The experimental results
demonstrate that our approach can reduce training and
testing time with high classification accuracy. Thus, it
combines good effectiveness with high efficiency. It
produces an efficient features subset, so it provides an
effective solution to the dimensionality reduction problem in
general. On the other hand, the efficiency of Fuzzy ESVDF
depends on how SVMs are able to classify the dataset, which
may obstruct the modification and maintenance processes
and impede the use of this approach in some types of
applications. Moreover, it can not guarantee the optimal
solution in terms of minimizing the number of features to the
least number, but in all situations, it gives a satisfactory
features number with excellent performance results.
Future work will involve planning to investigate the
possibility and feasibility of implementing our approach in
real time applications. We are planning also to improve our
approach and decrease its execution time.
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87
 2009 International Conference on Computer Engineering and Technology

PDC: Propagation Delay Control Strategy for

Restricted Floating Sensor Networks

Liu Xiaodong
College of Information Science and Technology
Ocean University of China, Qingdao, China
liuxiaodong.ouc@gmail.com

Abstract—Restricted Floating Sensor (RFS) proposed by Yunhao III describes our experiment and discusses results we obtain
Liu is an important model in offshore data collection. There is a from experiment. Finally, the conclusion is given in Section
need for highly precise, real-time in many offshore monitoring. IV.
In this paper, we propose a Propagation Delay Control (PDC)
strategy for RFS networks. This strategy can modify the
transmission delay adaptively depending on users together with II. DESIGN OF PDC
the changeful condition of wireless links. The experiment shows In this section, we mainly describe the design of PDC in
our work is succeeding in changing propagation delay by orders. detail. In this paper we assume all nodes transmit data at a
fixed frequency, let p stands for the fixed transferring rate of
Keywords- Bit error rate; Propagation delay; Sensor network
nodes. Throughout this paper, we denote D as the length of
I. INTRODUCTION packet data item and denote H as the length of header.
There are significant interests in analyzing the siltation of The architecture of PDC is shown in Fig. 1 c). Firstly we
the estuary and harbor. For example, H. H. Harbor, the second give the design of the FW Module. Let vij denotes node of the
largest harbor for coal transportation (6.7 million tons per year) network, where i stand for the layer, and j for the sequence
in China, currently suffers from the increasingly severe
problem of silt deposition along its sea route (19 nautical miles number of the same layer. Let V = {vij } .
long). The sea route has always been threatened by the silt Let S (vi0 j0 ) = {vij | vij ∈ V , i = i0 + 1, j ∈ J i0 j0 } denote the
from the shallow sea area. Monitoring sea depth costs this
harbor more than 18 million US dollars per year. In [1], aggregation of all child nodes of vi0 j0 in the RFS network,
Yunhao Liu proposed a Restricted Floating Sensor (RFS)
model. By locating such sensors, the sea depth can be where J i0 j0 contains all the sequence number of these nodes.
estimated efficiently without the help of extra ranging devices. Because of the movement of nodes, S (vi0 j0 ) and J i0 j0 are
To adjust propagation delay on request is required by RFS variables. For each node according to BER , the definition of
networks. For example, the amount of siltaanytion in H. H. the node’s weight for transmission time (weight, for short) is
Harbor is mainly affected by tide and wind blow. The highly
1
variable nature of wind brings more intensive effects. Records given as follows: w(vij ) = (1)
show that strong winds with wind forces of 9 to 10 on the (1 − BER) ( H + D )
Beaufort scale hit H. H. Harbor from Oct. 10th to Oct. 13th in
3
2003. The storm surge brought 970,000 m of silt to the sea w(vij ) is the average number of
It is easy to know that
route, which suddenly decreased the water depth from 9.5m to packets required for successfully transferring D bits data.
5.7m and blocked most of the ships weighing more than 35 Then we give the definition of First Weight as follows:
thousand tons. In this situation, the siltation of the Harbor
changes in a moment. There need a shorter propagation delay Definition 1: to any node vi0 j0 in the RFS network, the
to ensure the safety of the channels. On the other hand, a longer
response interval is necessary to reserve energy in better First Weight (FW) of node vi0 j0 is specified by the maximum
weathers. of FW of it’s child nodes, that is:
In this paper we propose a Propagation Delay Control
strategy (PDC) to control the transmission time. This strategy fw(vi0 j0 ) = max{ fw(vi0 +1, j )} + w(vi0 j0 ) (2)
gets the Waiting Time (WT) by modifying the First Weight j∈J i0 j0
(FW) and the Second Weight (SW) dynamically depending on
orders from users. Our experiment site lies in the east coastal So it is straight forward the first weight of leaf nodes is:
area of Qingdao, as shown in Fig. 1 a). We organize this paper
as follows: in section II, we discuss the PDC strategy. Sections fw(vdepth, j ) = w(vdepth , j ) (3)

This project is funded by the Chinese National High Technology Research

and Development Plan (863 program) under Grant No. 2006AA09Z113, by
Key Project of Chinese National Programs for Fundamental Research and
Development (973 program) under Grant No. 2006CB303000.

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DOI 10.1109/ICCET.2009.232
 FW
BER model FW

T AT AT
ΔT model SW

d SW SW
model
PDC Strategy

a) Our experiment site b) Topology model for RFS networks c) The architecture of PDC strategy

Figure 1. PDC strategy for RFS networks

Then we design the SW Module. As discussed in the subtree, a message carrying the information of ΔT ( L j ) and
introduction, nodes in RFS network float within a restricted
area. The topology is variable because of the restricted sw(v1, j ) will be passed to nodes of subtree L j . To nodes on
motility. Thus an amended method is needed to deal with the different layers, the following expression is used to work their
variability.Firstly we give a definition which is important in own transmission time:
describing the variability:
Definition 2: to any node vi0 j 0 in the RFS network, let sw(vik ) fw(vik )
at (vik ) = × ΔT ( L j ) + (7)
d (vi0 +1, j , vi0 j0 ) denotes the shortest distance ever detected sw(v1 j ) p
between vi0 +1, j and vi0 j0 . Let d min (vi0 +1, j ) denotes the Theorem 1: the propagation delay of L j is less than T.
distance between vi0 +1, j and it’s nearest parent node. The
Proof: to the root node v1, j , the at (v1, j ) is
Second Weight (SW) of node vi0 j0 is specified by the
fw(v1, j )
summation of w(vi0 j0 ) and the weighted sw(vi0 j0 ) , that is: ΔT ( L j ) + . To any other node in the subtree
p
sw(vi0 j0 ) = tw(vi0 j0 ) + fw(vi0 j0 ) (4) sw(vik ) fw(vik )
L j , at (vik ) is × ΔT ( L j ) + . From
d min (vi0 +1, j ) sw(v1 j ) p
Where tw(vi0 j0 ) = max{ × sw(vi0 +1, j )} . d min (vi0 +1, j )
j∈J i0 j0 d (vi0 +1, j , vi0 j0 ) sw(vi0 j0 ) = max{ × sw(vi0 +1, j )} + max{ fw(vi0 +1, j )}
j∈J i0 j0 d (vi0 +1, j , vi0 j0 ) j∈J i0 j0

It is straight forward that sw(vdepth , j ) = 0 (5)

We have sw(vi ,k ) < sw(vi−1, j ) , where i >1
In the PDC strategy, each node should transmit their SW to
their parent node. And the sink node will finally receive sw(vik )
and i, j , k ∈ J i0 j0 . That <1 is . For
sw(v1, j ) . The transmission time of nodes will be worked out sw(v1 j )
according to the FW and SW.
fw(vi0 j0 ) = max{ fw(vi0 +1, j )} + w(vi0 j0 )
j∈J i0 j0
Finally we give the design of WT unit. Let L1 stand for the
subtree with the root of v1, j . With the limitation of propagation That is fw(vi ,k ) < fw(vi−1,k ) , Then we have

delay (denoted by T ), the sink work out the limit of at (v1, j ) > at (vik ) where i > 1

propagation delay of L j : ΔT ( L j ) = T −
sw(v1 j )
(6) As shown in (6), at (v1, j )  = T +
( fw(v 1, j ) − sw(v1 j ) )
p p
fw(v1, j ) From (4) we know sw(vi0 j0 ) = tw(vi0 j0 ) + fw(vi0 j0 ) ,
The adjust time of v1, j : at (v1, j ) = ΔT ( L j ) +
p d min (vi0 +1, j )
For tw(vi0 j0 ) = max{ × sw(vi0 +1, j )} ≥ 0
It is clearly thatΔT ( L j ) and wt (v1, j ) is affected by j∈J i0 j0 d (vi0 +1, j , vi0 j0 )
many factors, among which T is the most dominating.
We have
After getting the upper limit of propagation delay for each

89
 fw(v1, j ) − sw(v1 j ) = fw(v1, j ) − (tw(v1, j ) + fw(v1, j ) ) ≤ 0 60 70

( fw(v ) − sw(v1 j ) )
60
50

) =T + 1, j
≤T
50

Thus we have at (v1, j

40

Requested Response Time

Interval of Packets
40

p 30

30

From (7) we know that at (vik ) is also dominated by T .

20
20

10
10

From theorem 1 we know the adjust time of the root node is the
longest in it’s subtree. This ensures the propagation delay is
0 0
10 30 60 30 10 10 30 60 30 10
Interval of Packets Requested Response Time

within the limitation.

Figure 2. the Adjustability of Response Time on Request
From the introduction we know there are various conditions
leading to changes of the propagation delay. We will discuss
(7) is efficient in these different conditions. TABLE I. CONDITIONS OF WEATHERS

Factors
The propagation delay should be shorten when storm Conditions
wind scale wave height(m)
comes, that is t < T . one 1-2 0.5

From (6) we know ΔT ′( L j ) < ΔT ( L j ) . two 8-9 3

From (7) we know the transmission time of each node can
fw(vik )
be shortened by × (ΔT ( L j ) − ΔT ′( L j ))
fw(v1 j )
In this case, under the time-sensitive requirement, all nodes
just transfer data gathered within the limitation of delay in
stead of waiting for every child node. In the same way we can
change the propagation delay back to normal to decrease the
energy consumption when the storm is over.
From the discussion we know (7) is suitable for all these
conditions. This algorithm modifies transmission time of nodes
depend on the limitation of delay afforded by users. And from
(9) and (10) we know that because of differences over BER,
FW, degree and etc, the transmission time of each node will be
different. So different nodes won’t send data at the same time
and this will decrease the probability of collision and ease
network congestion. All of these will be helpful to dampen
congestion, reduce the number of collisions resulting in fewer
retransmissions and could converge the propagation delay to a
certain set point.
Except T, all information is stored in the memory in this
strategy. The sink only needs to send a message containing the
new limitation of propagation delay (denoted by t ) when there
is a need to change the propagation delay. Each node will work
out the new adjust time easily by the following algorithm 3:
III. EXPERIMENT
[2]
In our experiment, we take LEPS introduced by SUN
Limin et al. as the routing protocol. In this section, we evaluate
the performance of PDC strategy.
We are deploying the working system in Qingdao, China.
We use TelosB motes and Tiny OS as our development basis.
Current system consists of 23 sensor nodes deployed in the
field, reporting sensing data continuously to the base station.
The data centre of Ocean Sense has been launched and most of
our data can be seen on internet. With limited power resources,
it is vital for RFS nodes to minimize energy consumption
during radio communication to extend the lifetime of the
network. So we will evaluate the performance of PDC. Here
we compare PDC with other transmission time control
mechanisms and discuss their effect on efficient transmission
and energy saving. Because the condition of channel between
nodes is variable in different weathers, we compare the
performance PDC in terms of different wind scale and wave
height. These conditions are shown in table I.
Fig. 9 displays the adaptability to the change of propagation
delay. These three figures are in different weather conditions.
From Fig. 9 we get the adaptability in condition one is better
than in other two worse conditions. This is mainly because the
interval between nodes caused by bad weathers.
IV. CONCLUSION
In this paper we proposed a Propagation Delay Control
strategy for large scale RFS networks. The experiment shows
our work is succeeding in changing propagation delay by
orders. However the complete system is designed to scale to
hundreds of sensors covering the sea area off Qingdao. Thus
our future work will focus on designing an improved strategy
for large-scale RFS networks.
REFERENCES
[1] Zheng Yang, Mo Li, Yunhao Liu, Sea Depth Measurement with
Restricted Floating Sensors. The 28th IEEE Real-Time Systems
Symposium[C], Tucson,2007: 469-478.
[2] Tingxin Yan, Limin Sun, Principle and Performance Evaluation of
Routing Protocol in TinyOS[J]. Computer Engineering, 33:112-114.
[3] Raghunathan V.,Schurgers C.,Park S.,et. Energy Aware Wireless
Microsensor Networks[J]. IEEE Signal Processing
Magazine,2002,19(2):40-50.
[4] Spragins J D., Hammond J L., Pawlikowski K. elecommunications:
Protocols and Design [M]. Boston; Addison Wesley Publishing
Company,1991.
[5] He Tian, Blum Brian M, Stankovic John A.,Abdelzaher T. AIDA:
Adaptive Application Independent Aggregation in Sensor Networks
[J].ACM Trans on Embedded Computing System, 2003, 3(2): 426-457.

90
 2009 International Conference on Computer Engineering and Technology

Fast and high quality temporal transcoding architecture in the DCT domain
for adaptive video content delivery

Vinay Chander1 Aravind Reddy2 Shriprakash Gaurav3

vinay87@gmail.com aravind_k_iiitm@yahoo.co.in gaurav.agnos@gmail.com

Nishant Khanwalkar4 Manish Kakhani5 Shashikala Tapaswi6

nis.iiitm@gmail.com manishkakhani@gmail.com stapaswi@iiitm.ac.in
1, 2, 3, 4, 5, 6
Department of Information Technology,
Indian Institute of Information Technology and Management, Gwalior, MP, INDIA

Abstract variety of end devices that exist in today’s internet and

Video communication over the Internet is
becoming increasingly popular today. Transmission of
videos over networks has always been a problem due
to reasons like high bandwidth requirement, lossy
channels (especially wireless links, low bit rate two
way communication links etc), huge file sizes and the
large variety of end terminals with different
constraints. In this paper, we limit ourselves to
solving the problem of bandwidth and end terminal
frame rate requirements in the context of video
content distribution over the internet. We propose an
efficient temporal transcoding architecture in which
motion change is considered. For motion vector
composition, both bilinear interpolation vector
selection (BIVS) and forward dominant vector
selection (FDVS) are used. Our architecture operates
entirely in the DCT domain, unlike some
architectures, which operate partially or fully in the
pixel domain, thus avoiding computationally
expensive operations of inverse DCT and DCT. The
re-encoding errors are also minimized. The macro
block coding types are re-judged to reduce drift
errors. We have implemented the algorithms and
carried out experiments for Mpeg 1 video sequences
and the results are found to be promising.
Keywords: Video transcoding, DCT domain,
motion based frame skipping, motion vector
composition and prediction error re-calculation
1. Introduction
The eminence of digital video on the internet is
increasing by the day. Video content delivery has
many issues associated with it. This is due to the large
also because of the varying constraints of the channels
that make up networks [7]. In this paper, we attempt
to help with the problem of video content delivery to
heterogeneous end devices over channels of varying
capacities by proposing a fast and efficient temporal
transcoding architecture, which operates completely in
the compressed domain [3, 12]. An overview of the
other approaches for video content adaptation can be
found in [1, 2 and 6].
A video may be considered as a stream of frames
played at quick succession with very short time
intervals, which is viewed and interpreted by the end
user(s). This fact brings about an important feature
which is exploited to reduce the transmitted data size.
First of all, short time interval between consecutive
frames means that the contents should be very close to
each other, which can be made use of by dropping the
less important frames .Since the final user of the video
stream is a human, the biological features of human
visual system should be considered and the priority in
preserving data should be based on these features. In
temporal transcoding of videos, pre-encoded frames
may be dropped from incoming sequence to freely
adjust the video to meet network and client
requirements [4, 5 and 14].
Thus we propose an architecture which adapts the
incoming compressed video to the end terminal’s
frame rate constraint as well as the bandwidth of the
available channel by using a motion based frame
skipping algorithm [5, 11] to skip frames. This
method involves calculation of motion activity of the
frames .Motion activity gives a measure of the motion
in a frame and is defined as the sum of the motion
vector components in that frame. We use the modified
definition of motion activity [11], which is found to be
better. If the motion activity is larger than a given
threshold, the frame cannot be skipped, since it has

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DOI 10.1109/ICCET.2009.174
considerable motion, and so transcoding this frame
improves the quality and smoothness of the video
sequence delivered. The architecture is described in
greater detail in the next section.
Our architecture allows skipping of B frames as
well as P frames. When the P frames are dropped, the
motion vectors that refer to the dropped frames
become invalid and need to be recalculated with
respect to their new reference frames that are part of
the transcoded bit stream. Also, the quantized DCT
coefficients of residual signal of non-skipped frames
become invalid as they may refer to the reference
frame(s) which no longer exist(s) in the transcoded
bit-stream. So, it becomes necessary to re-compute the
new set of quantized DCT coefficients with respect to
the past reference frame(s) that will act as reference in
the transcoded bit-stream.
Thus, for performing the recalculations and
validating the transcoded frames, we present our
scheme to re-judge the macro block coding types to
minimize the drift errors [4]. We calculate the new
motion vectors using motion vector composition. We
implement a scheme, where in both bilinear
interpolation vector selection (BIVS) [8] and forward
dominant vector selection (FDVS) [9] are used. The
method which is appropriate for the macro block
under consideration is employed to make the process
computationally cheap yet, achieving good results
.The pixel-domain approach for re-calculating the
prediction residual involves high computation due to
the computationally expensive operations of inverse
DCT, DCT. Also, re-encoding errors are incurred in
performing DCT and re-quantization. We re-calculate
the new quantized DCT coefficients of the prediction
residuals by processing the quantized DCT coefficients
available from the incoming stream even in the case of
MC macro blocks. This is achieved using the block
translation method [13, 15]. This differs from [4], in
Figure 1. Transcoding Architecture
92
which inverse quantization is carried out, thus further
minimizing the re-encoding errors. Thus we compute
the quantized DCT coefficients for non-skipped
frames entirely in the DCT domain.
To summarize our work, in this paper we present a
fast frame skipping transcoding architecture that
operates entirely in the DCT domain, which
accelerates the transcoding process. The smoothness
and quality of the transcoded video stream is also
maintained by using an efficient motion based frame
skipping algorithm. The rest of the paper is organized
as follows. In Section 2, we explain our transcoding
architecture. Section 3 describes the macro block re-
encoding scheme, which covers three parts: (i) Re-
judging macro block types under different situations
(ii) Motion vector composition and (iii) Recalculation
of the residual signal. Section 4 briefly summarizes
our implementation and Section 5 presents the
experimental results obtained. Finally, the conclusion
and future scope is given in Section 6.
2. Transcoding architecture
The block diagram of our transcoding architecture
is as depicted in Figure: 1.
Firstly, the input bit stream is parsed by a variable
length decoder (VLD) which performs the extraction
of header information, frame coding modes, macro
block coding modes, motion vectors and the quantized
DCT information for each macro block. This is
followed by the calculation of motion activity for each
frame in the video. For this, we use the modified
definition of motion activity, which is briefly described
below [11].
……. (1)
(MA)m is the motion activity of a macro block, k is
a properly tuned constant and |xm| and |ym| are the
motion vector components. Since an intra macro block
is produced when there are many prediction errors
(namely, the macro block is largely different from the
reference area in the previous frame), they are
assigned the maximum motion activity value, equal to
the maximum size of the motion vectors, which
corresponds to the search range used by the motion
estimation procedure. In this way, we take into
account of intra macro blocks also in the motion
activity computation. The motion activity of a frame is
calculated by summing up the motion activities of all
the macro blocks of that frame, as given by equation
(2).
….. (2)
Once the motion activities are computed, the next
step involves application of a motion based, frame
skipping algorithm as described in [11].The algorithm
is presented below:
Algorithm for frame skipping
Motion-based Policy (frame f):
if(f = first frame) Thr(f)=0;
else Thr(f)= (Thr(f-1)+ MA (f-1))/2;
if MA(f)<= Thr(f) skip f
else transcode f
smoothness and quality of the output video are
maintained to a very good extent. The results are
presented and described in section 5. The final list of
the frames to be skipped includes both B frames as
well as P frames.
Since B frames are non-reference frames, they are
first dropped from the bit stream and no re-encoding is
required in this case. The dropping of P frames is
performed next. This order is followed to avoid
redundant re-encoding of frames. Dropping P frames
requires a re-encoding scheme because the motion
vectors and prediction residuals of frames that refer to
the dropped P frame become invalid. Our re-encoding
scheme is presented in the following section. The P
frame skipper (portion of figure: 1, which follows the
B frame skipper) includes a switch S, which updates
the DCT domain buffer for the quantized DCT
coefficients of the residuals, depending on the coding
mode of the macro blocks. When the switch opens, the
transcoder performs the motion compensation in the
DCT domain. After the new motion vectors and
prediction residuals are calculated, the output is fed to
a variable length coder (VLC), thus producing the
final bit stream.
3. Re-encoding architecture

Thus, after performing this step, an initial list of
frames that can be skipped are available. Then, we
select a value N (which we name as the quality factor),
which is done in such a way that the frame rate as well
as the bit rate is reduced sufficiently to meet the
requirements. This value of N gives the number of
frames to skip per GOP (Group Of Pictures). After
this, the frames contained in the list are sorted by their
motion activities. Then, using the value N, the final
list of frames that are to be skipped from the incoming
bit stream is obtained, that is the N frames with the
least motion activities amongst a GOP, are skipped.
By experiments, it is observed that till N=3, the
When P frames are dropped, the frames that are
part of the transcoded bit stream need re-encoding.
Consider a GOP (in the display order),
IBBPBBPBBPBBPBB. Suppose the 7th frame (P
frame) is dropped, the 2 B frames that precede it (5th
and 6th frames), the B frames (8th and 9th frames)
that appear after it and the subsequent P frame (10th
frame) need to be modified.
Macro block Type Conversions and re-encoding:
Case 1: Intra macro blocks need no changes, as
they do not contain references.

Figure 2. New prediction errors computation for non-MC macro blocks

Description: Rt is the frame being modified ; Rt-1 is the skipped frame and Rt-2 is the new temporal
reference of Rt. MBt is the macro block ,which is being modified ; MBt-1 is the skipped macro block and
MBt-2 is the new reference area of MBt. Q[DCT(et)] is the original error term, Q[DCT(est)] is the
s s
modified error term and similarly Ut and Vt are original vectors of MBt , U t and V t are their modified
values .

93
 Case 2: In a B frame that appears before the
dropped frame, if the macro block is (Forward)
predicted, then it requires no changes. But if the
macro block is (Backward) predicted or (Forward +
Backward) predicted, then it is converted into an Intra
macro block and is replaced by the referenced picture
area .It is found that the increase in bit rate due to this
is not significant, since the percentage of these macro
block types are quite low. If the referenced area does
not lie within the macro block boundary, then its
quantized DCT coefficients can be calculated from the
incoming quantized DCT coefficients of its
neighbouring macro blocks (available from the DCT
domain buffer).The Block Translation method, that is
available in literature [13, 15] achieves this. And in
case the neighbouring macro blocks are not Intra
coded, they can be found out by recursively tracing the
vectors. For this purpose, motion vector composition
(as covered in 3.2.1 and 3.2.2) may be required.
Case 3: In the frames that appear after the dropped
frame, if the macro block is (Backward) predicted,
then no change is needed. In case of Forward
predicted macro blocks, if the referenced area lies
within the macro block boundary, we have the
following sub-cases. (figure: 2):
3.1. Intra coded macro block: If the referenced
macro block (MBt-1) is Intra coded (which is not the
case in figure 2), then MBt is converted into an Intra
macro block.
3.2. Inter coded macro block: If the referenced
macro block (MBt-1) is Inter coded (Forward
predicted), then the vector of the referenced macro
block is simply added to the vector of MBt, which
gives the modified vector:
Ust = Ut + Ut-1 ......(3)
Vst = Vt + Vt-1
…..(4)
Where Ust and Vst are the modified components of
the motion vector of MBt. But if the referenced area
Figure 3. New prediction errors computation for non-MC macro blocks
94
does not lie within the macro block boundary (figure:
3), then we have the sub-cases 3.2.1 and 3.2.2, which
follow:
Sub-cases 3.2.1 and 3.2.2 cover the motion vector
composition scheme used. We implement a scheme,
where in both bilinear interpolation vector selection
(BIVS) and forward dominant vector selection
(FDVS) are used. The method which is appropriate for
the macro block under consideration is employed to
make the process computationally cheap yet, leading
to good results.
3.2.1. MB1t , MB2t-1, MB3t-1, MB4t-1 are all Inter
Coded: If all the involved macro blocks (MB1t, MB2t-
1, MB3t-1, MB4t-1) are Inter Coded , then we use the
Bilinear interpolation [8] to find the resultant vector.
Hence, the final modified vector can be found by
simple vector addition (equations 3, 4). The new
prediction error is calculated entirely in the DCT
domain [15] unlike some architectures that operate
partially in the pixel domain [10].The linear property
of DCT is used and since DCT (et) and DCT (et-1) are
divisible by quantizer step-size, the newly quantized
DCT coefficients of prediction error for MBt are given
by:
Q[DCT(est)] = Q[DCT(et)] + Q[DCT(et-1)] …..(5)
where,
Q[DCT(et-1)] = Q[DCT(MBt-1)] - Q[DCT(MBt-2)]
… (6)
Where Q[DCT(MBt-1)] and Q[DCT(MBt-2)] can be
found by the block translation method .
3.2.2. One of the four (MB1t , MB2t-1, MB3t-1,
MB4t-1) is an Intra macro block: If one of the four
(MB1t , MB2t-1, MB3t-1, MB4t-1) turns out to be an
Intra macro block , then we find out if there exists an
Inter macro block, among the neighboring ones and
whether the referenced area overlaps more than 3/4th
in area with it . If it exists, then FDVS [9] is used to
find the resultant vector. Hence, the final modified
vector can be found by simple vector addition
(equations 3, 4). Otherwise the macro block is re-
encoded as an Intra macro block. There by, a good
tradeoff between bit rate reduction and video quality is
achieved.
Case 4: In the frames that appear after the dropped
frame, if the macro block is (Forward + Backward)
predicted, then the Forward vector needs to be
modified and is done in the same way as discussed
above for case: 3. After the new forward vector is
found, the new quantized DCT coefficients can be
found out by the equation given below:
Q[DCT(est)]= Q[DCT(et)] + (Q[DCT(et-1)]/2) ...(7)
Which can be derived as follows:
We have the 3 equations as follows:
Q[DCT(est)] = (Q[DCT(MBt)] - Q[DCT(MBt-2)])/2
+(Q[DCT(MBt)] - Q[DCT(MBt+1)])/2 .....(8)
Q[DCT(et)] = (Q[DCT(MBt)]-Q[DCT(MBt-1)])/2
+(Q[DCT(MBt)]-Q[DCT(MBt+1)])/2 .....(9)
Q[DCT(et-1)] = Q[DCT(MBt-1)] - Q[DCT(MBt-2)]
.....(10)
Eq:8 – Eq:9 – (Eq:10)/2 gives us the required
equation .
4. Implementation
In this section, we give a brief overview of our
implementation. We have carried out our
implementation for Mpeg-1 videos. The Mpeg-1 video
stream is taken as the input along with the quality
factor (N), which is decided by the channel capacity
and the target frame rate. The input is processed to get
the details of video stream including the required
header information, the motion vectors and the
quantized DCT coefficients of each macro block. This
is done using mpeg_stat, which is a free tool available
on [16]. We have implemented our algorithms in C++,
which includes four header files namely,
declarations.h, frame_modifier.h, block_dct.h and
Huffman.h containing functions for motion activity
calculation, motion vector composition, re-calculation
of prediction errors and computation of the number of
bits in the modified frames (this helps in evaluating
the size of the output video stream).Thus the output
generated by our modules include:
95
1. The new values of bits per frame.
2. The quantized DCT coefficients of the
modified frames.
3. The new motion vectors.
4. The re-judged coding types of the macro blocks
We reconstructed the new frames by performing
inverse quantization and inverse DCT to compare
them with the frames from the original video
sequence. The results are presented in the next
section.
5. Experimental results
We conducted experiments over mpeg-1 videos to
evaluate the performance of our transcoding
architecture and the results for three sample mpeg-1
video sequences are presented. The three video
sequences, ACT60.mpg, ADVERTISMENT.mpg
and SCHONBEK.mpg are compressed mpeg-1 videos,
each with a resolution of 160×120 and having 15
frames per GOP with the following sequence:
IBBPBBPBBPBBPBB. The average PSNR (peak
signal to noise ratio) values for N=3, 4 frames per
GOP, are calculated and presented below. Table 1
gives the results for the video sequences (percentage
reduction in the stream size, average PSNR values,
and input and output frame rates for the N=3 and
N=4).
From the results obtained, we observe that the
average PSNR value decreases with increase in the
value of the quality factor (N), which is a direct
consequence of increase in the drift errors with the
increase in the value of N. The result obtained is found
to be promising. We also took observations on the
smoothness of the output video, by showing the input
and output video sequences to a group of 30 people.
From their feedback it is observed that the output
video is found to be very good till quality factor (N) =3
(for a GOP of size 15). We also present few images
below, to show the comparison between the visual
quality of the reconstructed reference frames (for N=3,
N=4) and the original frame taken from the input
video. Figure 4(a) is a frame from the original
sequence, ACT60.mpg and 4(b), 4(c) correspond to
the reconstructed frames for N = 3 and 4 respectively.
Similarly figures 5(a), 5(b) and 5(c) are images taken
from the original and reconstructed sequences of
ADVERTISMENT.mpg respectively.
 Table 1. Average PSNR, % reduction in stream size, change in frame rate for 3 sample
mpeg-1 video sequences
Input video Input Quality Factor (N =3) Quality Factor (N=4)
sequence frame % Averag Output % Averag Output
rate reduction in e PSNR Frame reduction e PSNR Frame
(fps) stream size (dB) rate in stream rate
(fps) size (fps)
ACT60.mpg 29.97 26.32 37.19 23.97 36.31 35.82 21.97

ADVERTISME 23.97 28.85 36.53 19.17 35.22 34.29 17.57

NT.mpg

SCHONBEK.mp 29.97 26.79 35.47 23.97 35.55 33.78 21.97

g

(a) Original frame (b) for N=3 (c) for N=4

Figure 4. Sample frame taken from ACT60.mpg

(a)Original frame (b) for N=3 (c) for N=4

Figure 5. Sample frame taken from ADVERTISMENT.mpg

6. Conclusion & future work
We have proposed a temporal transcoding
architecture, which helps in solving the problem of
video transmission over networks, which constitute
channels of low bandwidth and end terminals with
varying frame rate constraints. The architecture skips
frames (both reference and non reference frames) on
the basis of motion and is low in complexity because
the transcoding is carried out completely in the DCT
domain. Also, the re-encoding errors are low because
the quantized DCT coefficients are directly
manipulated upon. The Macro block coding re-
judgement scheme and the re-calculation of the
prediction residuals reduce the cumulative errors.
From the results we have obtained, this architecture
works well in terms of the output video quality for
both videos with high motion as well as low motion,
while having a low computational complexity.
One of the areas that still seem to be open is the
design of efficient frame skipping policies. The

96
literature consists of frame skipping policies which are
mainly defined by motion information in an
experimental way. It would be interesting to
investigate this problem in an analytical way by the
use of tools like dynamic programming and
randomization to design new frame skipping
strategies.
References
[1] Ahmad I., Xiaohui Wei, Yu Sun and Ya-Qin Zhang,
“Video transcoding: an overview of various techniques and
research issues,” IEEE Transaction on Multimedia,
Volume:7, Issue: 5, on page(s): 793- 804, Oct. 2005
[2] A. Vetro, C. Christopoulos, and H. Sun, ‘Video
Transcoding Architectures and Techniques: An Overview”,
IEEE Signal Processing Magazine, vol.20, no.2, pp.18-29,
Mar.2003.
[3] Chia-Wen Lin, Yuh-Reuy Lee, “Fast algorithms for
DCT-domain video transcoding,” International Conference
on Image Processing, 2001 Proceedings. 2001, Volume:1,
on page(s): 421-424, Publication Date: 2001.
[4] Chunrong Zhang, Shibao Zheng, Chi Yuan and Feng
Wang, “A novel low-complexity and high-performance
frame-skipping transcoder in DCT domain,” IEEE
Transactions on Consumer Electronics, Volume: 51, Issue:
4, on page(s): 1306- 1312, Nov. 2005.
[5] H. Shu and L. P. Chau, “Frame-skipping Transcoding
with Motion Change Consideration”, Proceedings of the
2004 International Symposium on Circuits and Systems,
vol.3, pp.773-776, May 2004.
[6] J. Xin, C.W. Lin, and M.T. Sun, “Digital Video
Transcoding”, Proceedings of IEEE, vol.93, no.1, pp.84-97,
Jan.2005.
[7] Jens Brandt, Ing. Lars Wolf, “Multidimensional
Transcoding for Adaptive Video Streaming,” Proceedings of
the 17th International workshop on Network and Operating
Systems Support for Digital Audio and Video
(NOSSDAV'07), june-2007.
97
[8] J. N. Hwang, T. D. Wu and C. W. Lin. “Dynamic Frame-
Skipping in Video Transcoding.”, IEEE Transactions on
Consumer Electronics, Vol. 44, pp. 88-98, February 1998.
[9] J. Youn, M. T. Sun and C. W. Lin, “Motion Vector
Refinement for High-Performance Transcoding”, IEEE
Transactions on Multimedia, Vol. 1 (1), pp. 30-40, March
1999.
[10] K.T. Fung, Y. L. Chan, and W. Siu, “New architecture
for dynamic frame skipping transcoder,” IEEE Transactions
on Image Processing, vol. 11, pp. 886–900, August 2002.
[11] M.A. Bonuccellit, F. Lonetti and F. Martelli,
“Temporal transcoding for mobile video communication,”
The Second Annual International Conference on Mobile and
Ubiquitous Systems: Networking and Services, 2005,
MobiQuitous 2005, pp 502- 506, July 2005.
[12] Susie Wee, Bo Shen, John Apostolopoulos,
“Compressed-Domain Video Processing” Hewlett-Packard
Laboratories Technical Report HPL-2002-282 to be
published in CRC Handbook on Video Databases, October –
2002.
[13] T. Shanableh and M. Ghanbari, “Hybrid DCT/pixel
domain architecture for heterogeneous video transcoding,”
Signal Processing: Image Communication, vol.18,
September 2003.
[14] V. Patil and R. Kumar, “An Arbitrary Frame-Skipping
Video Transcoder,” IEEE International Conference on
Multimedia and Expo., ICME 2005, pp 1456-1459, July
2005.
[15] V. Patil and R. Kumar, “A DCT domain frame-skipping
transcoder,” IEEE International Conference on Image
Processing, ICIP 2005, Volume: 1, on page(s): I-817-20,
11-14 Sept. 2005.
[16] mpeg_stat. A video analyzing tool for mpeg-1 videos
http://bmrc.berkeley.edu/ftp/pub/multimedia/mpeg/stat/
 2009 International Conference on Computer Engineering and Technology

Electricity Demand Forecasting Based on Feedforward Neural Network

Training by A Novel Hybrid Evolutionary Algorithm
Wenyu Zhang Yuanyuan Wang
College of Atmospheric Sciences, Lanzhou School of Mathematics & Statistics, Lanzhou
University, Lanzhou 730000, China University, Lanzhou, 730000, China
yuzhang@lzu.edu.cn liangjzh07@lzu.cn
Jianzhou Wang Jinzhao Liang
School of Mathematics & Statistics, Lanzhou School of Mathematics & Statistics, Lanzhou
University, Lanzhou, 730000, China University, Lanzhou, 730000, China
wjz@lzu.edu.cn Chejxh07@lzu.cn

Abstract networks, such as slow training rate, easy to trap into

Electricity demand forecasting is an important index
to make power development plan and dispatch the
loading of generating units in order to meet system
demand. In order to improve the accuracy of the
forecasting, we apply the feedforward neural network
for electricity demand forecasting. Inspired by the idea
of Artificial Fish Swarm Algorithm, in this paper we
proposed one hybrid evolutionary algorithm which
based on PSO and AFSA methods through crossing
over the PSO and AFSA algorithms to train the
feedforward neural network. This proposed method
has been applied in a real electricity load forecasting,
the results show that the proposed approach has a
better generalization performance and is also more
accurate and effective than the feedforward neural
network trained by particle swarm optimization.
1. Introduction
With the establishment of the electric power market,
a high forecast precision is much more important than
before. The inaccuracy or large error in the forecast not
only means that load matching is not optimized but
also influences the stability of the power system
running. The traditional models of the load forecasting
such as the time series model, regression analyses
model [1] are too simple to simulate the complex and
fast change of the power load [2].
In order to improve the accuracy of the forecasting,
in this paper, we apply the feedforward neural network
for electricity demand forecasting. Artificial neural
networks technology is an effective way to solve the
complex non-linear mapping problem. Feedforward
neural network is a kind of neural networks, which has
a better structure and been widely used [3]. But there
are still many drawbacks if we simply use neural
local minimum point, and bad ability on global search
etc [4].
In recent years, the swarm intelligent methods such
as Particle Swarm Optimization (PSO) algorithm,
artificial immune algorithm and Artificial Fish Swarm
Algorithm (AFSA) are applied in the function
optimization and parameters optimization. These
algorithms reflect different better properties with their
characteristics such as scalability, fault tolerance,
adaptation, speed, autonomy, and parallelism in the
different applications. Tao[5] applied the PSO
algorithm in Optimization, but the PSO algorithm has
the disadvantages such as: sensitive to initial values,
easily trapping into local optimum, premature
convergence, parameters selection problems, slow
convergence in the later stage of the evolution.
The AFSA has strong ability of avoiding local
extremum and achieving global extremum, its usage is
flexible and convergence speed is fast. This algorithm
has been applied in nonlinear function optimization,
parameter estimation, short-term load forecasting of
power system, etc. AFSA and PSO are much similar in
their inherent parallel characteristics, whereas
experiments show that they have their specific
advantages when solving different problems. What we
would like to do is to obtain both their excellent
features by synthesizing the two algorithms. This paper
proposes a novel approach based on the above
mentioned idea. The method called AFSA-PSO-
parallel-hybrid evolutionary (APPHE) algorithm. The
main idea of the hybrid algorithm is to divide the
particles into two, the first particles execute the AFSA
algorithm while the second particles execute the PSO
algorithm simultaneously, and then find the largest
fitness in the two systems. The whole system only has
a largest fitness and a best position, the best solution
should be transmitted back to PSO populations.
Afterward the two subsystems execute the PSO

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 98

DOI 10.1109/ICCET.2009.76
algorithm simultaneously, and then find the largest where q is the number of total training samples,
fitness in the two systems. Output the best solution and
yi − Ci is the error of the actual output and desired
k k
stop if the termination criterion is satisfied. And then
we apply the AFSA-PSO-parallel-hybrid evolutionary output of the ith output unit when the kth training
algorithm to train the feedforward neural network. The sample is used for training. We defined the fitness
results which are compared with feedforward neural function of the ith training sample as follows:
network trained by particle swarm optimization (PSO)
algorithm show much more satisfactory performance, fitness( xi ) = E ( X i ) (2)
converges quickly towards the optimal position,
convergent accuracy and can avoid overfitting in some When the APPHE algorithm is used in evolving
extent. weights of feedforward neural network, every particle
The remainder of this paper is organized as follows: represent a set of weights and bias.
Section 2 provides a brief description of Multi-layer
feed forward ANN, PSO, AFSA and APPHE
algorithm. Section 3 describes the research data and 2 .2 The PSO algorithm
experiments. Section 4 summarizes and analyzes
empirical results and discusses the conclusions and The PSO Algorithm was first proposed by Kenney
future research issues. and Eberhart in 1995 [9] and could be performed by
the following equations:
2. Methodology t +ϕ1⋅rand1⋅(pid −xid)+ϕ2 ⋅rand2 ⋅(pgd −xid)](3)
vid(t +1)=k⋅[vid()
2.1. Multi-layer feedforward neural network xid (t + 1) = xid (t ) + vid (t + 1) (4)
where i = 1, 2, … , m. d = 1, 2, … , D , rand1 and
An FNN, consists of an input layer, one or more
hidden layers and an output layer. Every node in each rand 2 are random numbers uniformly distributed
layer is connected to every node in the adjacent layer.
Supposed that the input layer has n nodes; the hidden within [0,1] . vid ∈ [−vmax , vmax ] and vmax is a
layer has H hidden nodes; output layer has O output designated value. Moreover, in order to guarantee the
nodes. In this paper, the hidden transfer function and convergence of the PSO algorithm, the constriction
the output transfer function are both sigmoid function. factor k is defined as follows
The computed output of the ith node in the output layer 2
is defined as follows [7]: k= (5)
H n
2 − ϕ − ϕ 2 − 4ϕ
yi = f (∑ (wij f (∑ v jk xk + θvj ) + θwi )), i = 1, , O (1) Where ϕ = ϕ1 + ϕ2 , ϕ > 4 . Usually, when the
j =1 k =1
constriction factor is used, ϕ is set to 4.1
where yi is the output of the ith node in the output ( ϕ1 = 2.05, ϕ2 = 2.05 ), and the constriction factor k
layer; xk is the input of the kth node in the input layer; is 0.729.
wij is the connective weight between nodes in the
2.3. Standard AFSA Algorithm
hidden and output layers; v jk represents the connective
weight between the nodes in the input and hidden Artificial Fish Swarm Algorithm was first proposed
in 2002 [6]. In this paper, we adopt [10] to describe the
layers; and θ wi (or θ vj ) are bias terms that represent
AFSA Algorithm.
the threshold of the transfer function f .
2.3.1 The structure of algorithm and definition
The learning error E can be calculated by the
following formulation [8]: Suppose that the searching space is D-dimensional
and N fish form the colony. The AF individual state
can be expressed with vector: X = ( x1 , x2 , … , xD ) ,
q O
Ek
E=∑ where Ek = ∑ ( yi − Ci ) ,
k k 2

k =1 q * O i =1 where xi (i = 1,…, D) is the variable to be searched

for the optimal value; AF food consistence at present
position can be represented by: FC = f ( X ) , and

99
FC is the objective function; the distance between the forward a step to the fellow centers. Mathematically
AF individuals can be expressed as: xck − xik
xinextk = xik + Random( Step ) (8)
d i , j = X i − X j ; Visual represents the vision Xc − Xi
distance; Step is the maximum step length and δ is Otherwise executes the behavior of searching food. If
crowd factor. n f = 0 , AF executes the behavior of searching food.
(3) Following behavior
2.3.2 The description of behavior
Let X i denote the AF states at present, and find
(1) Searching behavior
X min , in which FCmin is the minimum value of its
We randomly select a new state X j
in current state
fellows in the near fields (d i , j ≤ Visual ) , if
X i ' s visual field. If FCi > FC j in the minimum
problem, it goes forward a step in this direction; FCmin / n f < δ FCi , which means that the fellow
otherwise, select a state X j randomly again and X min has high food consistence and the surrounding is
judge whether it satisfies the forward condition. If it not very crowded, forward a step to the fellow X min .
can’t satisfy after try _ number times, it moves a
Mathematically
step randomly. The mathematical expression is as x m in k − x ik
follows: x inextk = x ik + Random ( Step ) (9)
X m in − X i
⎧ x jk − xik
⎪xinext = xik + Random(Step) FC j < FCi Otherwise executes the behavior of searching food. If
⎪ X j − Xi n f = 0 , AF executes the behavior of searching food.
⎨ (6)
⎪ (4)Bulletin
⎪⎩xinextk = xik + Random(Step) FCi ≤ FC j Bulletin is used to record the optimal AF's state and
the AF food consistence at present position. Update the
Where k = 1,2, ⋅⋅⋅, D ; x jk , xik and xinext represent the bulletin with the better state of the AF’s, the final value
k element of state vector X , X and AF’s state of the bulletin is the optimal value of the problem, the
j i
state of which is the optimal solution of the system.
vector at the next X inext respectively.
time
Random ( Step ) represents a random number in the 2.3.3 Select the behavior
range [0, Step ] , FC i , FC j are food consistence of According to the character of the problem, the AF
evaluates the environment at present, and then selects
state X i , X j . Meaning of symbols in following an appropriate behavior. For example, the simplest
formula is same with these. way is trial method. Evaluate the values derived by
(2) Swarming behavior swarming behavior and following behavior, and
An AF with the current state Xi seeks the implement the behavior whose result is the minimum.
The acquiescent behavior is searching food.
companion's number in its current neighbourhood
where satisfy d i , j < Visual ; and calculate their 2.4. AFSA-PSO-parallel-hybrid evolutionary
centre X c , FCc denotes the food
position algorithm
consistence of the center position and n f denotes the The performance of the novel algorithm is described
as follows:
number of its fellows in the near fields, if n f ≥ 1, X j 1. Divide the particles into two, and then initialize
explores the center position of its fellow. AFSA and PSO subsystems respectively.
Mathematically 2. Execute AFSA and PSO simultaneously.
nf 3. Find the best solution in the two systems, and then
xck = (∑ x jk ) / n f (7) transmit the best solution back to PSO populations.
j =1 4. The two subsystems execute PSO algorithm
simultaneously.
If FC c / n f < δ FC i , which means that the fellow 5. Memorize the best solution as the final solution and
center is high and surroundings is not very crowded, stop if the best individual in one of the two

100
 subsystems satisfies the termination criterion. If not,
go to step 2. The flow chart of the APPHE algorithm
is shown in figure 1.
Fig.1 Flow chart of APPHE algorithm
3. Application
3.1 Problem description
In this paper, we adopt [11] to analyse the problem.
Supposing that a time series {xk }, k = 1, 2, , N , is
given, using the delays method, we represent the data
in d -dimensional space by vectors
X k = [ xk , xk +1 , , xk + ( d −1) ],
where d is the embedding dimension. Time series
prediction is to deduce the future values of a time
series according to its past values. We can use the map
f to make the prediction. The prediction can be
described as
X k +T = f ( X k ) (10)
f :(xk , xk+1, , xk+(d−1) ) →(xk+T , xk+1+T , , xk+(d−1)+T ) (11)
xk + ( d −1) +T = g ( xk , xk +1 , , xk + ( d −1) )
g is an unknown function, and T is the prediction
step. Here T = 1 means one-step-ahead prediction,
and T > 1 means multi-step prediction. In this work,
we try applying feedforward neural network to
estimate the unknown function g .
3.2 Simulation
We collected 28-day hour load series from a state of
Australia, a 672-observation series as experiment data
analysis and forecasting will be done with it. In this
paper we use an FNN with the structure of 5–5–1 to
address the problem. Multi-layer feedforward ANN is
trained with 26-day hour load series and then forecast
the next two days.
Supposed that every weight in the network was
initially set in the range of [− 60 ,60 ] , and all the bias in
the network were set in the range of [−50,50 ] . The
maximum velocity assumed as 2 and the minimum
velocity as -2. The population size is 45. The maximal
iterative step is 500. In the AFSA algorithm parameters
setting, try _ number = 15 , δ = 1.3 , Visual = 1 . 4 ,
Step= 1.5 . And the embedding dimension d is 5.
Simulations are performed in matlab7.1 with a 2.8GHZ
Pentium PC.
Two error metrics are employed to evaluate the
prediction performance, one is the related index
defined as:
n
∑ [Y i − Y i ] 2
R 2
= 1− i =1 (13)
n
∑
i =1
[Y i − Y ] 2
Where Yi is the actual data, Yi is the forecasting
data, Y is the average of the time series
{Yi , i = 1, 2, , n} , n is equal to 24. The closer R 2
is to the value 1, the more satisfactory performance is.
The other is the relative error defined as:
n
| Yi − Yi |
∑i =1 Yi
.
n
The results are as follows. Figure 2 is the predictions
of the next two days’ load series. From figure 2, we
can know that the predictions based on APPHE-FNN
algorithm are closer to the actual data than PSO-FNN
algorithm’s. The process of training via the APPHE-
FNN algorithm and PSO-FNN algorithm are showed in
figure 3. In figure 3, it shows that the PSO-FNN may
rapidly stagnate, and the solution no longer improves
(12) anymore, while the APPHE-FNN can still search
solution progressively till the global optimum is found.
It’s found from table 1 that the related index and the
relative error of the APPHE-FNN algorithm for the
predicted 2 consecutive days are obviously much better
than PSO-FNN algorithm’s. Through the comparison
analysis, it can be seen that the APPHE-FNN
algorithm has a more accurate forecasting capacity and
considerably better convergence than the PSO-FNN.
101
 that the proposed method can be used to many other
Table 1 The prediction results about R 2 and relative error
for 2 consecutive days complex time series forecasting such as financial series
Algorithm 27th 28th and hydrological series forecasting.
2 0.92 0.86
APPHE-FNN R
Relative error 2.71% 3.82%
Acknowledgment
2 0.85 0.81 The research was supported by the NSF of Gansu
PSO-FNN R
Province in China under Grant (ZS031-A25-010-G).
Relative error 3.92% 5.64%

11000
References
actual data
PSO-FNN [1] Niu Dongxiao, Cao Shuhua, Zhao lei, et al. Power Load
10500
APPHE-FNN Forecasting Technology and Its Application. Beijing: China Electric
10000 Power Press, 2001.
Electricity Load Forecasting(MW)

[2] Dongxiao Niu, Jinchao Li, Jinying Li, et al. Daily Load
9500 Forecasting Using Support Vector Machine and Case-Based
9000
Reasoning, 2007 Second IEEE Conference on Industrial Electronics
and Applications, 2007, pp:1271-1274.
8500 [3] Changrong Wu, Mingquan Ye, The application of BP neural
networks in hospital multifactor time series forecast, Fujian
8000
Computer, No.1, pp.38-39, Jan 2005.
7500
[4] Shu-xia Yang, Ning Li, Power Demand Forecast Based on
Optimized Neural Networks by Improved Genetic Algorithm,
7000 Proceedings of the Fifth International Conference on Machine
Learning and Cybernetics, Dalian, 13-16 August 2006, pp:2877-
6500
0 5 10 15 20 25 30 35 40 45 50 2881.
hours [5] Tao Xiang, Xiaofeng Liao, Kwok-wo Wong, An improved
Fig.2 The prediction results based on APPHE-FNN algorithm particle swarm optimization algorithm combined with piecewise
200 and PSO-FNN algorithm respectively linear chaotic map, Applied Mathematics and Computation, Volume
APPHE-FNN 190, Issue 2, 15 July 2007, pp:1637-1645.
180 PSO-FNN
[6] Li Xiaolei, Shao Zhijiang, Qian Jinxin. An optimizing method
160 base on autonomous animates fish-swarm algorithm [J].Systems
Engineering Theory and Practice.2002, 11:32-38.
140
[7] Chern-Hwa Chen, Jong-Cheng Wu and Jow-Hua Chen,
120 Prediction of flutter derivatives by artificial neural networks, Journal
Fitness

100
of Wind Engineering and Industrial Aerodynamics, In Press,
Corrected Proof, Available online 7 April 2008.
80
[8] Jing-Ru Zhang, Jun Zhang, Tat-Ming Lok, Michael R. Lyu, A
60 hybrid particle swarm optimization–back- propagation algorithm for
feedforward neural network training, Applied Mathematics and
40
Computation, Volume 185, Issue 2, 15 February 2007, pp: 1026-
20 1037.
0
[9] J.Kennedy and R.Eberhart, “Particle swarm optimization”, in
0 50 100 150 200 250 300 350 400 450 500 Proceedings of the IEEE International Conference on Neural
Iteration
Networks (Perth, Australia), IEEE Service Center, Piscataway,
Fig.3 Fitness curves of FNN based on APPHE algorithm pp:1941-1948, 1995.
and PSO algorithm respectively [10] Yi Luo, Juntao Zhang and Xinxin Li, The Optimization of PID
Controller Parameters Based on Artificial Fish Swarm Algorithm,
Proceedings of the IEEE International Conference on Automation
4. Conclusions and Logistics August 18 - 21, 2007, Jinan, China, pp:1058-1062.
[11] Xiaodong Wang, Haoran Zhang, Changjiang Zhang , et al, Time
In this paper, feedforward neural network trained by series prediction using LS-SVM with particle swarm optimization,
Advances in Neural Networks, 2006, Vol.3972, pp: 747-752.
AFSA-PSO-parallel-hybrid evolutionary algorithm is
proposed for electricity demand forecasting. The
results show that the proposed algorithm has a better
ability to escape from the local optimum and a better
predicting ability than PSO-FNN algorithm. And the
high precision has a significant impact on the
economic operation of the electric utility since many
decisions based on these forecasts have significant
economic consequences.
It should be pointed out that, although the processes
are focused on electricity load forecasting, we believe

102
 2009 International Conference on Computer Engineering and Technology
Investigation on the Behaviour of New Type Airbag*
Hu Lin,Liu Ping
College of Mechanical Engineering
University of Shanghai for Science and Technology
Shanghai, China, 200093
Hulin888@sohu.com
Abstract -This research a new type airbag system (NAB) was
developed in this study which consists of two flat layers and a
middle layer with tube-type structure. The objective of the new
airbag system is to enhance the occupant safety in passenger car
collisions with reduction of injury risks for small and OOP
occupants. A prototype of the airbag was built up with volume 40
litters. The performance of the prototype airbag was investigated
using static deployment test and sled crash test with Hybrid III
dummy. A FE model of the airbag was developed and validated
using results from the airbag deployment test and sled crash test.
In order to determine the potential for occupant protection,
computer simulations of the OOP occupant with NAB and the
normal driver-side airbag (DAB) are carried out using HYBRID-
III FE model. Results indicated that the NAB airbag need less gas
than the DAB to get the same load and displacement curve, and
the leading velocity of NAB is lower than that of DAB. In the
OOP simulation, the acceleration of dummy head using NAB is
smaller than that using DAB.
Index Terms - Sandwiched airbag, OOP occupant protection,
sled crash test, finite element model
I. INTRODUCTION
Airbags have been proven to be very helpful in protecting
drivers and passengers in many cases of automotive crashes.
However, side-effects may be serious if the system is not
carefully designed, manufactured and used[1]. One common
side-effect is that it may harm occupants, especially for
children and small women, when it deploys improperly on
crash (Chritina et al., 1998; John et al., 1993; Alex et al.,
1995). Because to achieve occupant protection during a crash
using a fully-deployed airbag to dissipate the frontal crash
forces experienced by the driver over a larger body area and
gradually decelerate the occupant’s head and torso to prevent
contact with other interior surfaces, the airbag itself must
deploy rapidly in less than 50 milliseconds. Consequently, an
occupant positioned extremely close to the airbag module at
the time the airbag begins to inflate is exposed to highly
localized forces. Two phases of airbag deployment have been
associated with high, injury-causing localized forces: the
punch-out phase and the membrane-loading phase. According
to the findings of biomechanics research, field automotive
accident and laboratory test, chest and neck are the most
vulnerable objects subjecting injuries during airbag interacting
with OOP occupant. It has been found that chest injury often
associates with airbag “punch out” force and occurs at very
early stage of the airbag deployment, but neck injury is more
*
Supported by Shanghai Leading Academic Discipline Project,
Project N.J50503
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.60
Huang Jing
The State Key Laboratory of Advanced Design &
Manufacture for Vehicle Body
Hunan University
Changsha, China ,410082
luckycitrine@163.com
likely induced by the “membrane” loading at relative late
deployment time[2]. It is believed that the side-effect would
be minimized if the early development pattern of the airbag
can be controlled. Proper development of the airbag system
depends on several factors, including the structure of the
airbag, the inflation system, the ignition system and the
ignition time etc. Development of the traditional airbag would
require a powerful inflation system and quick response of the
ignition system, which put high requirements on the systems
and thus resulting high costs. To reduce possible side-effect of
the airbag system and to have suitable requirement on the
inflation system, a sandwiched tube-type airbag system (NAB)
[3]was developed (Zhong, 2005) and evaluated in this paper.
II. METHOD AND MATERIALS
A prototype of the NAB was developed, which consists of
an upper layer flat airbag, a lower layer flat airbag and a
middle layer of tube-type airbags. The upper layer flat airbag
is designed to touch the occupant on crash, and the middle
layer tube-type airbags are designed to support the upper layer
and the lower layer. The whole NAB’s volume is about 40 L.
In the event of a crash, the middle layer is first and mostly
inflated while the other two layers are inflated later and less
rapidly. Two groups of deployment tests using DAB and NAB
were performed to support and validate the computational
modeling efforts and compare their deployment properties;
then the sled test and virtual tests were used to estimate the
protection efficiency of the NAB; finally the parametric study
was carried out to find the more sensitive and critical design
variables to occupant injuries.
Experimental Set-up - An experimental inflation device
was designed as shown in Figure 1. The experimental devise
can be used to inflate the airbag and provide gas with
prescribed pressure and leakage.
The whole deployment system mainly consists of an air
compressor, a tank, fast-acting valves and sensors (shown in
figure 1). The air compressor 1 provides compressed air to the
tank 4. The outlet of fast-acting valve is connected to the inlet
of the new structure airbag. Pressure sensor 9 and 6 are
separately used to transfer tank pressure signal and airbag
inside pressure signal to computer. Safety valve 3 prevented
the system from pressure overloading. Solenoid valve 2
functions as a switch between air compressor and tank. A/D
board 8 does the conversion from pressure voltage signal to
numerical signal. The NAB system and traditional airbag
103
system are tested with the device in different loading
conditions. The features of the airbag deployments were
examined.

(a) (b)
Fig.2 (a) out of position 1 (b) out of position 2

III. NUMERICAL MODEL AND EXPERIMENTAL VALIDATION

Fig.1 Layout of the airbag test system: 1.Air Compressor 2.Solenoid Valve
3.Safety Valve 4.Air Tank 5.Fast Acting Valve 6,9. Pressure Sensor NAB model - The NAB system is modeled with 18408
7.Fixed Airbag 8.A/D Board.
Belyschko-Tsay membrane elements. Material properties of
Airbag deployment test - The behaviors of the NAB
Nylon 66 are assigned to the membrane elements. Contact
system are investigated using an experimental device as
interface is defined between the airbag and the dummy’s head,
shown in Figure 1. The NAB system and traditional airbag
neck,chest and hands. Self-contact interface is also defined
system are tested with the device in conditions of static and
within the airbag system.
dynamic deployment: (1) static deployment without impact,
Static deployment -The high speed films from static airbag
and (2) dynamic deployment with impact.
deployment are shown in Figure 3a. The dynamic stiffness of
Sled crash test - To examine the actual behavior of the
the airbag system is estimated through dropping composite
NAB system in protecting occupants, sled crash tests are
blocks onto the inflated airbag. Experimental results show that
carried out by using a 50% HYBRID-III dummy with standard
by adjusting the pressure alone equivalent dynamic stiffness
safety belt at impact speed of 35 km/h. as shown in Figure 4a.
properties may be obtained with different types of airbags. It
Two energy absorbing square metal beam of the size
means that the NAB system has the potential to provide
120×120×500 in mm with thickness of 1.2 mm. The airbag is
similar load-displacement feature as the traditional airbag
mounted at an angle of 60 degrees with horizontal plane. The
system would do although the NAB system requires much less
tested NAB system is inflated with gas storage on the sled,
gas. Figure 3b shown the simulation results which are similar
providing a prescribed pressure to the system. The
with that from deployment tests.
acceleration signal of the dummy’s head and the test sled is
measured and recorded.
Virtual testing - Simulations of the static and dynamic
deployment of the NAB system were carried out by using LS-
DYNA program to study the feature of inflation process. Then,
computer simulation was also conducted with the HYBRID-III t=0ms t=10ms t=20ms t=30ms t=45ms
dummy FE models. In response to side-effects of an airbag in (a)
low- and moderate-severity crashes, FMVSS 208 issued by
NHTSA in May 2000, proposed that static OOP tests should
be a mandatory requirement starting in 2003. These tests
include performance requirements to ensure that airbags
t=0ms t=10ms t=20ms t=30ms t=45ms
developed in the future do not pose an unreasonable risk of
(b)
serious injury to OOP occupants[4]. In the FMVSS-208
Fig.3 (a)The airbag during inflation test; (b) Simulation of the deployment
NPRM, for the 5th percentile female "low risk deployment", process of the NAB
the two test positions for OOP situations are as shown in Sled test - Figure 4a shows the 50th percentile HYBRID III
Figure 2. dummy response and contact with NAB at 50 ms in the sled
According to above requirement, the 5th percentile adult crash test. The measured acceleration peak value is 16 g at the
female dummy OOP virtual tests were used to do parametric dummy head center of gravity.
analysis. The driver position 1(~ISO 1) is to obtain maximum
neck loading (membrane) and the driver position 2 (~ISO 2) is
to obtain maximum chest loading (punch out) from the
deploying air bag. Finally, the virtual test results with DAB
and NAB were compared.

(a) test at 50 ms (b) simulation at 50 ms

104
Fig.4 (a) Sled crash test of the NAB with HIII dummy; (b) Simulation of the Pa) at time 0.035s. DAB reached its peak value (0.1445×MPa)
interaction of the NAB with the dummy
at time 0.037s. So , NAB can use less gas to reach the same
From virtual testing using FE model, the dummy response airbag inner pressure.
and contact with the airbag at 50 ms is shown in the Figure 4b,
which is comparable with the crash test result. The time
history plot of head accelerations measured from test is
presented and compared in Figure 5, and the magnitude and
duration of the head acceleration curves are reasonable good.
The validated NAB module model will be applied in the OOP
occupant simulation as discussed in the next paragraph.
200
Head acceleration (m/s2)

Simulation
160 Sled test

120 Fig.7 Volume history plot

80

40

0
0,000 0,020 0,040 0,060 0,080 0,100 0,120
Time (s)

Fig.5 The time history plots of dummy’s head acceleration from sled crash
test and simulation of the sled test
Comparison of deployment property of NAB and DAB -
To compare the NAB and normal DAB’s deployment property
Fig.8 Pressure history plot
and assess the relative potential of different airbag designs to
cause injury, two groups of static deployment simulations Ⅳ. AIRBAG-DESIGN PARAMETRIC STUDY
were conducted. The first group of simulation is to measure
the leading-edge speed between the normal DAB and NAB, Design variables and control levels - Some of the design
the normal DAB and NAB have the same volume (40 L), parameters that might affect the airbag module performance in
drivers' OOP conditions are: airbag structure, vent hole,
either their inflators’ parameters. Figure 6 shows the different
inflator tank pressure characteristics such as the slope and the
leading-edge velocity results from the static deployment
simulation. From the time history plot of the leading-edge peak, venting during inflation, cover break out force, fabric
velocity, the peak value of the normal DAB is 53.9 m/s at time material property e.g. density, modulus of elasticity, porous
property, etc[5]. The parametric study was carried out in two
0.0225 s, and the NAB is 27.6 m/s at time 0.0025 ms. Reed et
steps, In the first step, some initial design parametric about
al. recorded some ARS 3 (Abrasion Rating System) abrasions
to human skin at a contact speed of 234 km/h (65m/s), so the NAB’s structure have been carried out, mainly to judge the
possible abrasion injury caused by NAB is lower than ARS 3, influence of tether and vent hole to NAB. The collocation of
tether, the length of tether, the mass flow rate, the size and
and is lower than the possible abrasion injury caused by DAB
location of vent hole are selected as design variables[6]. The
also.
design variables and their control levels are showed in Table
1.
Table 1 Design Variables and control levels
Control level
Design Variable
1 2 3 4
A=Tether collocation Col 1 Col2 Col 3 Absence
B=Tether Length 100% 90% 80% 70%
C=Vent hole Area 1110mm2 80% 70% 110%
D=Vent hole
Pos1 Pos2 Pos3 Pos4
circumferential Position
E=Vent hole Radial
Fig.6 Leading-Edge Velocity comparison Pos1 Pos2
Position
In the second group of simulation, the normal DAB’s
F=Mass flow rate 100% 70%
volume is 60 L and the NAB’s volume is 40 L, their leakage
In the second step, based on the research results of the first
area was set to the same value (1152 mm2), also use the same
step of the parametric study, the influence of inflator
inflator’s parameters. Figure 7 shows volume history plot of
characteristics and fabric material properties are discussed,
two types of airbag. NAB reached its maximum volume value
and the design variables and their control levels are showed in
at time 0.036 s, normal DAB reached its maximum volume
Table 2.
value at time 0.039 s. Figure 8 shows the pressure history plot
Table 2 Design Variables and control levels
of two types of airbag. NAB reached its peak value (0.214M

105
 Control level Parametric virtual tests results - The virtual tests results
Design Variable
1 2 3 for parametric study are shown in Table 3.
G= inflator Pressure peak (kpa) 120 160 200 Table 3 virtual test results
H=inflator Pressure slope (ms) 0 4.5 6.5 Dummy Chest My
NO. HIC36 Fz(N) Pcom Nij
I=fabric material Density (kg.m-3) 541 721 901 pos G (Nm)
J=fabric material Porous property 0 50 100 1 8.5g 182.0 650 15.0 0.019228 0.264300
Object function definition - According to the findings of 2 2.7g 271.1 1200 -35.0 0.021403 0.883049
biomechanics research, field automotive accident and 3 4.3g 191.0 370 17.5 0.017891 0.208264
laboratory test, head, chest and neck are the most vulnerable 4 9.0g 243.8 1250 25.0 0.022394 0.483455
objects subjecting injuries during airbag interacting with OOP IP 5 36g 195.1 1500 31.0 0.057342 0.586598
occupant. In order to valuate NAB’s protection performance 6 9.0g 1131 3600 28.5 0.264523 1.111706
synthetically, the minimizing occupant injuries, US NCAP 7 3.0g 244.6 1600 28.0 0.020036 0.593016
Injury Index— PCOMB and neck injury criteria N ij are chosen as 8 11.5g 266.4 2250 25.0 0.024956 0.741187
9 13.0g 187.5 800 30.0 0.021954 0.399734
the object functions. PCOMB can presents both HIC and chest
1 7.8g 273.1 1500 75.0 0.023451 0.870469
acceleration as only one function. It is formulated as following 2 7.5g 279.6 950 68.0 0.023699 0.683555
[7]. 3 27.5g 352.8 1800 90.0 0.047185 1.044563
Object Function = MIN ( P ) COMB
(1) 4 8.2g 318.6 1180 95.0 0.026487 0.917027
Where, OOP1 5 18.0g 302.3 1400 75.0 0.031829 0.844696

PCOMB = PHEAD + PCHEST − ( PHEAD × PCHEST )

6 18.0g 1061 -2100 81.0 0.225341 1.063818
(2)
7 23.0g 297.1 1150 95.0 0.036827 0.909295
PH E A D and PCHEST are defined as following: 8 10.0g 482.7 -1900 96.0 0.042145 1.109046
9 10.5g 333.7 1750 90.0 0.028680 1.031676
1
PHEAD = (3) 1 7.0g 217.7 1050 -28.0 0.020170 0.729635
1 + EXP ( 5.02 − 0.00351 × HIC36 ) 2 8.0g 223.4 1120 -27.0 0.020887 0.731283
1 3 28.0g 263.4 1300 -29.0 0.042293 0.810461
PCHEST = (4)
1 + EXP ( 5.55 − 0.00693 × CHESTG )
4 7.8g 220.4 1220 -26.0 0.020650 0.740663
OOP2 5 10.0g 242.0 1450 -30.0 0.022805 0.865515
N ij can present both upper neck force and upper neck 6 11.5g 246.8 1300 -32.0 0.023882 0.859642
moment, it is formulated as following: 7 20.0g 216.8 1300 -22.0 0.029033 0.695707
Fz My 8 11.5g 246.7 1500 -31.0 0.023877 0.894795
N ij = + 9 12.0g 268.7 1400 -38.5 0.025381 0.991972
Fcritical M critical (5) Analysis of results - The Design Exploration Tools of
Simulation matrix - After formulation of design problem, iSIGHT is used to analyze the previously mentioned virtual
to provide enough representative test data, a design of test results, the ANOVA, pareto plots and main effects graphs
experiment (DOE) analysis was conducted to design the test are recorded. ANOVA is the statistical analysis of the
matrix; for the first step of parametric study, the orthogonal contribution of each factor or their interactions to the variance
array is defined as L16 ( 4 × 2 ) matrix. And for the second
4 3 2
of a response [8]. R in ANOVA means the coefficient of
step of parametric study, the orthogonal array is defined as L9 determination, in another word it is a measure of the accuracy
4
( 3 ) matrix. of the model fit. The pareto plots indicate the relative effect of
The traditional test work generally changes one design each design variable on occupant injury. The main effects
variable’s value at one time and other variables maintain graphs indicate the desirable direction for each design
invariable, so the number of experiments will be very big and variable.
can’t get the mutually affecting relations of these variables. In order to make NAB provide good protection for occupant
The method by using orthogonal array to arrange experiments under all conditions, the choice of design variables should be
permits many variables change simultaneously; it can reduce considered synthetically. Based on above three scenes
the experiment numbers greatly and can be possible to parametric analysis, we can draw a conclusion that ‘the
estimate the effect of these variables more precisely. pressure slope of inflator’ and ‘the porous property of fabric’
Simulations are accomplished in each case of table 3 and are sensitive to occupant neck injury, ‘the pressure peak of
table 4 using validated LS-DYNA simulation models, the 5th inflator’ is sensitive to occupant head and chest injuries. And
percentile female dummy seated in three postures: in-position, the predicted desirable direction for each design variable are
out-of-position 1 and out-of-position 2. According above as follows: the pressure peak of inflator should be chosen
orthogonal array experiment matrix, LD-DYNA simulations lower level, the pressure slope of inflator should be chosen
lower or medium level, the density and porous property of
are not required 3 × 34 times but only 27. fabric should be chosen medium or higher level.
Ⅴ. RESULTS AND ANALYSIS Effectiveness of NAB - Adjust the parameters of NAB to
the direction of design improvement according to above

106
parametric analysis, and a series of virtual tests were carried
out with 5th percentile female dummy and 50th percentile
male dummy sitting on position 1 and position 2, 60 L normal
DAB and 40 L NAB are used respectively, then total 8
simulations were carried out. The purpose is to verify NAB’s
protective efficiency to different stature drivers.
The normalized values with respect to percentages of
injuries caused by DAB are shown in figures 9. The recorded
injury criteria include: 15ms HIC, 3 ms clip, chest deflection,
neck tension, neck compression and Nij.
5th-OOP1 5th-OOP2
1
STAB injury/DAB injury
[3] Zhong, Z H and He, W. Sandwiched Tube-Type Airbag, Patent
CN200410046609.9, 2005.
[4] Raj Roychoudhury, Dana Sun, Mohamed Hamid , Craig Hanson. 5th
Percentile Driver Out of Position Computer Simulation, SAE paper, No
2000-01-1006.
[5] William Mu, Driver out-of-position injuries mitigation and advanced
restraint features development. ESV 17th Conference.
[6] J Huang, Z Zhong, L Hu, and D Liu. Modeling and Simulation of
Sandwiched Tube-Type Airbag and its Optimization using Design of
Experiments, Part D: J. Automobile Engineering, Proc. IMechE Vol. 221.
[7] Seybok Lee, Soongu Hong. A Shape and Tether Study of Mid-Mounted
Passenger Airbag Cushion Using Design of Experiments. 9th
International MADYMO User's Meeting, Como,Italy . 10-
50th-OOP1 50th-OOP2 11th,October,2002 , page1-5.
[8] Neter, J., Wasserman, W., Kutner, M., Applied Linear Statistical Models:

0.9 Regression, Analysis of Variance, and Experimental Designs. Irwin

0.8 Publishing. Boston, MA, 1990. R. P. Van Zuydam. A driver’s steering aid
0.7 for an agricultural implement, based on an electronic map and real-time
0.6 kinematics DGPS, Computer and Electronics in Agriculture, 24(2005),
0.5
0.4 p153-163.
0.3
0.2 Hu Lin was born in Hunan province P.R.China on September 4,
0.1
0 1978.Hu Lin got doctor degree of mechanical engineering from
HIC15 3ms-Clip Chest-D Neck-T Neck-C Nij Hunan University in 2008, and focuses on the research of
Automotive Safety and Electronics. He has published 25 articles.
Fig.9 NAB Injury comparison with DAB (Corresponding author to provide mobile phone: 086-
It is seen that all the injury ratios are less than 1, which 15821431148 ;E-mail: hulin888@sohu.com )
indicate that the NAB could decrease the risk of OOP
occupant’s injury and provide better protection.
Ⅵ. CONCLUSION
Results from computer simulation and sled crash tests show
that the NAB system has good potential to provide effective
protection to occupants as the traditional airbag system would
do. At the same time, the NAB system would require less gas
for the airbag to take the same amount of space, the inflation
system can be made smaller and less expensive. At least two
advantages may be obtained. Firstly, the system can be
exploded more rapidly to take as much space as possible
between the occupant and the automobile interiors. Secondly,
the upper layer which would impact the occupant becomes
softer, which is desirable to avoid unexpected injuries to the
occupants.
Numerical results also indicate that the NAB system would
give less harm to the occupant if the ignition takes place
properly. However, the actual performance of the NAB system
in commercial product has to be examined with an actual
inflation system activated by a matched ignition system.
Nevertheless, the authors believe that the concept and
prototype of the NAB system is worthwhile to be exploited for
improvement of the airbag for occupant protection. Further
investigation will be performed subsequently to develop a new
airbag product.
REFERENCES
[1] M. U. Khan, M. Moatamedi, A review of airbag test and analysis.
International Journal of Crashworthiness, Volume 13, Issue 1 February
2008 , pages 67 – 76.
[2] Jörg Hoffmann, Michael Freisinger, Mike Blundell, Manoj Mahangare,
Peter Ritmeijer, Investigation Into the Effectiveness of Advanced Driver
Airbag Modules Designed for OOP Injury Mitigation. In: Proceedings of
the 20th International Technical Conference on the Enhanced Safety of
Vehicle, paper NO. 07-0319-O

107
 2009 International Conference on Computer Engineering and Technology

Performance Evaluation of PNtMS: A Portable Network Traffic Monitoring

System on Embedded Linux Platform

Mostafijur Rahman, Zahereel Ishwar Abdul Khalib, R. B. Ahmad

School of Computer and Communication Engineering,
Universiti Malaysia Perlis,
P.O. Box 77, d/a Pejabat Pos Besar,
01007 Kangar, Perlis, Malaysia.
Email : mostafijur21@hotmail.com

Abstract— The principal role of embedded software is the
transformation of data and the interaction with the
physical world stimulus. The main concern in developing
embedded software for network application is the lack of
published best practice software architecture for
optimizing performance by means of reducing protocol
processing overhead, memory usage, and power
consumption. This paper presents the implementation ,
operation and performance evaluation of the portable
network traffic monitoring system (PNtMS) on an
embedded Linux platform. The system has been designed
to capture network packets information from the network
and performs some statistical analysis, which includes
providing data on the volume and types of traffic
transferred within a LAN, traffic generated per node,
number of traffic going through or comming from a
system or application which is causing bottleneck, and the
level of peak traffic. These data are then stored into the log
files, and the traffic information can be shown through
web browser or onboard LCD. Result shows that PNtMS
is performing at par with an existing network protocol
analyzer with minimal usage of RAM ( 578 KB), low end
processor (133MHz) and storage(less than 1GB).
Keywords- embedded linux; network monitoring; single
board computer.
I. INTRODUCTION
Rapid growth of hardware technologies brings large
variety of smaller hardware architectures and platform
orientation that has been leading a large demand of
embedded software. So, programmers are focusing
more and more on to developing software on embedded
system to make it portable and platform independent.
The principal role of embedded software is the
transformation of data and the interaction with the
physical world [1]. Embedded softwares are marked
with the stamps as: timeliness, concurrency, liveness,
reactivity, and heterogeneity [2]. Because of the
resource limitation in terms of processing power,
memory and power consumption, the PNtMS was
developed to make efficent usage of the limited
resources.
The Internet has been growing dramatically over the
past several years, in terms of the amount of traffic
usage as well as connectivity. With this rapid growth,
the Internet is used for increasingly diverse and
demanding purposes, which lead to frequent changes in
network status, yet raised performance is required.
Therefore, it is crucial to monitor the network, in order
to understand the network behavior and to react
appropriately. Thus it will help us to design and provide
more efficient network for the future network. The
principal work of network monitoring software includes
collecting data from the Internet or intranet and analysis
of those data. The common features of a network
monitoring software includes, providing data on the
volume and types of traffic transferred within a LAN,
traffic generated per node, number of traffic going
through or comming from a system or application which
is causing bottleneck, and the level of peak traffic [3].
According to a survey, commercial Embedded Linux
owns approximately 50 percent more share of the new
project market than either Microsoft or Wind River
Systems [4]. The Technology Systems (TS) [5] provides
Single Board Computer (SBC) with TSLinux operating
system(OS). It is built to develop applications for a very
small target that does not require a keyboard, video,
floppy disks, and hard drives. Linux is a multi-tasking,
multi-user, multi-processing operating system and
purposely made for the required application and target
hardware, and thus attempts to be the optimized form of
the kernel for a specified application. The primary goal
for this work is to see how TSLinux could cope with the
limitations inherent in a low end embedded platform in
producing reliable embeddded traffic monitoring
system.
The main problem of developing embedded
software is inadequate software architecture and get
better performance in order to reduce protocol
processing overhead, memory usage, and power
consumption. In this paper, the design, implementation
and performance evaluation of the PNtMS into

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 108

DOI 10.1109/ICCET.2009.37
embedded system will be discussed. The system has
been designed to capture network packets information
from a switch and perform some statistical analysis,
then store into log files. The traffic information can be
shown through web browser or onboard LCD.
The proposed work was to develop portable network
monitoring and protocol analysis software into an
embedded Linux board. One of the main
requirements for this work was to write the code as
generically as possible so that it could be ported to other
Linux based SBC’s, like TS-5000 and TS-7000 serieses
etc. The hardware platform setup and the device driver
are expected to serve as a development platform for
implementing a well designed PNtMS.
II. RELATED WORK
Nowadays researchers are focusing their research on
small devices. They overlook the size, weight, cost,
interchangeability and consistency of the hardware and
try to make it portable and reliable with better
performance. Numerous works have been done in the
the embedded system area.
Work by Li and Chiang [6] proposed the
implementation of a TCP/IP stack as a self-contained
component, which is independent of operating system
and hardware. For adapting TCP/IP stack as a self-
contained component for embedded systems, zero-copy
mechanism has been incorporated for reducing
protocol-processing overhead, memory usage and
power consumption. In this mechanism, data from a
Network card is directly received in the user buffer and
the data from the user buffer is directly sent to the
network card.
The navigation system for an autonomous
underwater vehicle using TS-7200 was developed by
Sonia thakur [7]. The objectives of the work were the
implementation of a driver for the external ADC and the
GPS receiver on a Linux SBC and to demonstrate the
use of such a setup in autonomous navigation system.
Ahmad Nasir Che Rosli [8] implemented the face
reader for biometric identification using SBC (TS-
5500). In his research the system was able to capture
face image, execute image preprocessing (such as color
space conversion, motion analysis, grayscale
modification and image scaling), and to perform
biometric identification based on iris detection. It is
used for security purposes.
R. B. Ahmad and W. M. A. Mamat [9] implemented
a web-based data acquisition system using 32bit SBC
(TS-5500). In their research the system was able to take
analog reading through sensor and convert it into digital
109
data; the data transmission and data access is done by
TCP or UDP connection and web page respectively.
This system is used to monitor river or beach, and large
scale area such as in agricultural and environmental
fields. It can be used to read analog sensor such as for
sewer or septic early warning system.
III. TS-5400 SBC
The Technology Systems (TS) provides different types
of Single Board Computer (SBC). Among the model
TS-5400 was choosen for this research, because of its
compatible architecture. The TS-5400 SBC runs on a
AMD Elan520 processor at 133 MHz (approximately
10 times faster than 386EX based products) with
dimension 4.1" X 5.4". The processor AMD Elan520
was designed with a memory management unit (MMU)
that supports Linux and many other operating systems.
It shows worst case interrupt latencies of under 7
microseconds [10]. The core support is 32-bit
instruction set. The other features of TS-5400 are it is
fanless with temperature range -20° to +70°C and
power requirement 5V DC @ 800 mA. It has 16MB of
high speed SDRAM, 2MB flash disk, dual 10/100 Mbps
ethernet interface-autosense, compact flash (CF) card
interface as IDE0, matrix keypad interface on DIO2,
alphanumeric LCD interface , 3 PC/AT standard serial
ports with 16 Byte FIFOs support. As a general purpose
controller, it provides a standard set of peripherals [5].
Here CF card in the socket appears as a hard drive to the
operating system.
TSLinux and DOS being an embedded distribution
OSs, are installed into (CF) and onboard chip
respectively. The PNtMS was developed into TS-Linux.
TSLinux is an open source project and compact
distribution based on GPL and GPL like licensed
applications and was developed from “Linux From
Scratch” and Mandrake. Although TSLinux is fairly
generic, it is custom made to be used on a Technologic
System's Single Board Computer, and is unsupported in
any other use [11]. The version of TSLinux is 3.07a.
For development, TSLinux providers provide
development tools on their web site. The features of
TSLinux includes, Glibc (V2.2.5), Kernel (V2.4.18 and
2.4.23), Apache web server with PHP, Telnet server and
client, FTP server and client, BASH, and other basic
utilities. The total footprint is less than 18 MB (requires
32 MB or larger CF card).
 IV. SOFTWARE DEVELOPMENT
A. Preliminary setup
In TS-5400, Keypad(4x4) and LCD panel(24x2) are
used as I/O device. In this system keypad driver module
needs to be mounted into the kernel module. Because of
this, a device driver file TSKeypad is created into the
/dev/SBC/ location. Here Tskeypad file is used for input
funtion. Again there are three ways to transfer file from
desktop PC to TS-5400 using FTP, ZMODEM, and
Secure copy. To activate FTP and secure copy ,
dropbear script is run as a deamon. To start program
automatically the network file is needed to be
configured as DHCP for eth0 NIC.
B. Development environment
It is usual practice for most embedded system
developments not to support onboard compilation. The
cross platform compilation is the mechanism used to
develop the firmware. The same goes with the target
platform for this research, whereby the board requires
cross compilation for its application development. The
TSLinux 3.07a operating system running onboard does
not integrates a C compiler in its set of supported
utilities. Apart from that, since the kernel version of the
target OS (kernel 2.4.23) does not match the kernel
version of the development platform ( kernel 2.6.20-16).
The compatibility of the developed application has to
resort to chroot during the compilation to ensure with
the C libraries.
Figure 1. Embedded Linux development host and target system
interconnection setup [12]
On the host machine, minicom (a text-based modem
control and terminal emulation program for Unix-like
operating system) is used to communicate with the
target embedded system through serial port. This
particular setup is common in embedded linux
110
development, but it is not the only possible setup that
should work. The communication between the
development host and the target hardware is shown in
Figure-1.
C. Operation of the system
Our focus is to reduce memory usage and CPU
processing, as well as power consumption. To adapt
PNtMS components for resource-limited embedded
system, the pre-allocated memory are used rather than
allocating them at run time if buffers are needed at
memory managemet. The functionality of the PNtMS
constitutes processing of TCP/IP network traffic with
respect to network protocol analysis and traffic
monitoring. Here the gathered packet headers were
analysed based on the header information at the TCP/IP
protocol suit [13]. Figure 2 shows the components
element of PNtMS.
Figure 2. Component involve in the PNtMS
The structural breakdown of the PNtMS can be
generically segregated into three parts which can be
mapped into different layers of the OS. The first part is
probe, that has to do with capturing all incoming
packets from the Network. This part operates at the
network layer and captures data physically through the
Network Interface Card ( NIC ). This functionality is
realized through the usage of libpcap library package
[14]. Basically, this part forces the NIC to run in
promiscous mode.
The probe itself doesn’t process packet header at all,
but transfers packets to storage without any packet
loss.[15] The next layer (kernel layer) contains the
functionality for data acquisition and places the data
into a special memory region (called kernel buffer) to
be read and used by a seperate user-application program
(third layer). The packet filter extracts each packet
header information and stores into data buffer for
further analysis. After a while analyzer analyzes all
captured data. In the analyzer part, the available hosts
are selected and updated their information. Then host
information are sorted according to their total capture
bytes. The maximum 31000 packets can be captured
and more than 200 hosts information can be stored in
every interval. All of these are performed because of
memory limitation. After analysis, all hosts information
is saved into a file for monitoring through a web
browser. Some statistical data are shown through 24x2
LCD panel. The System is controlled by 4x4 matrix
keypad. Figure 3 shows
Figure 3. The complete System Setup
Figure 4. Web based traffic statistics into TS-5400
the complete system setup of PNtMS with TS-5400. In
this system, keypad is used to set input data such as
bandwidth(Kbps) and time interval. If the original
bandwidth is greater than inputed one, alarm is
displayed into LCD, when the traffic is peak. Within the
111
inputed time interval, packets are captured then
analyzed and displayed. Keypad can control the
program such as start and stop , and control the system
such as restart and shutdown. Some system information
also can be shown through LCD panel; such as system
IP, memory and CPU usage, and browsable web address
that shows total traffic information. The browser was
developed using HTML and PHP code. Some special
option are set through the web browser; such as options
to shutdown and restart the system. Also we can see
ethernet status, process status, disk usage, memory
usage through the web browser. Figure 4 and 5 show
some statistical information of the available traffics and
hosts.
Figure 5. Web based individual host statistics into TS-5400
V. RESULT AND PERFORMANCE EVALUATION
This section presents the results and performance
evaluation of PNtMS after being integrated into
TSLinux OS and Wireshark (V0.99.4). Table 1 shows
the experimental platform that we used to evaluate the
software. We compared our embedded software with
Wireshark because it is renown. In this, the traffic
measurement and the analysis were carried out at
Embedded Kluster Lab, University Malaysia Perlis. The
experimnt assumes that the traffic is typical Internet
traffic, and the characteristics of the analysis results can
be applied to the whole Internet.
The PNtMS program is used to capture packets and
store into a buffer to analyze. Each packet and available
host size limits ares set at 94 and 66 bytes respectively.
We tried to use limited memory and display traffic
 information as possible as we can. On the other hand, in
Wireshark, each packet size limit is 65535 bytes, so it
can store more information about a packet. Eventhough
wireshark is high performance protocol analyzer with
more features we want to evaluate the performance of
PNtMS with it.
All packet headers of the packets transferred on the
network during a 1 h period on September 15, 16, 17
and 23, 2008 were captured from the same network.
The duration was rather short, but we did several
experiments at different times and the results showed
very similar traffic characteristics.

Table 1. Exprimental Platform

TS-5400 PC
Architechture AMD Elan520 at 133 MHz Intel(R) Pentium(R)4 CPU 2.66 GHz
processor, 16MB RAM (Avg. Processor, 256MB RAM, 80GB secondary
14.23MB already used), 1GB storage
secondary storage
Type Embedded SBC Desktop PC
Dimention 4.1" x 5.4" Larger than TS-5400
Weight and portability Less than 1 Kg. and portable More than 1 Kg. and not portable
OS TS-Linux 3.07a UBUNTU 7.04
Linux Kernel 2.4.23 2.6.20-16-generic
NIC 10/100 Ethernet Realtek RTL8139 Family PCI Fast Ethernet
I/O interface Keypad (4X4) and LCD(24X2) Keyboard and Monitor

Table2. Performance Comparison

TS-5400 PC
Software Name PNtMS Wireshark (0.99.4)
Compiled With gcc (2.95) , libpcap GTK+ 2.10.11, Glib 2.12.11, libpcap
0.9.5 in development 0.9.5, libz 1.2.3, libpcre 6.7, ADNS,
environment GnuTLS 1.4.4, Gcrypt 1.2.3, Port Audio
<=V18, gcc 4.1.2
Execution Type Sequential Sequential
Avg. CPU Usage (%) 0.725 3.150
Avg. RAM Usage (%) 4.175 14.675
Avg. Packets/Sec. 7.347 6.661
Avg Packet size 470.883 439.958
Avg. Data Rate 3.378 KBps 2.862 KBps

a) Network Layer Protocol b) Transport Layer Protocol c) Application Layer Protocol

Figure 6. Different layers protocols statistics.

Figure 6 clearly demonstrate that the PNtMS can capture takes place through TCP protocol. A small number of
three types of network traffics and protocols capture state ICMP(62 packets) packets are also captured and put into the
using the percentage. In the Figure 3b percentage of overall statistics. At the same time Wireshark captured
TCP(65%) traffic dominates as majority of communication network and transport layer protocol such as, IP(76.88%),

112
ICMP(0.26%), TCP(59.69%) and UDP(15.97%); services
such as, DNS(0.83%), Netbios(11.74%), SSH(17.90%) and
HTTP(25.96%). This is only the glimps of what PNtMS
presents while performing the protocol analysis. From the
analysis we can conclude that PNtMS is successfully
developed into limited usable memory ( 578 KB), processor
and storage.
VI. CONCLUSION
Network performance measurement is an important
aspect of network management. In this article, we compared
and analyzed PNtMS with desktop base high performance
network analyzer (Wireshark). Our work focuses on how to
obtain better performance by using limited resources on TS-
5400 and show more statistical analysis result and make
PNtMS user friendly. From the result we can conclude
about the evaluation and validation of the PNtMS through
the implementation and protocol analysis. The challenging
future work is to use database into CF(1GB) for monitoring
long term traffic history. The most challenging future work
is to extend PNtMS so that it can be used to monitor and
analyze switched network such as fast ethernet and gigabit
ethernet and make it to perform better.
Hardware Architecture and Software Development," in
Proceeding of the 2nd International Conference on
Informatics, Petaling Jaya, Selangor, Malaysia, 2007.
[10] M. Breinich, "Online book on real-time simulation of a
satellite link using a standard PC running Linux as operating
system,” Salzburg, Austria, 2002.
[11] Technologic System Inc., "Technologic System Web Page,"
2008.
[12] C. Hallinan Embedded Linux Primer: A Practical Real-
World Approach. New York: Prentice Hall PTR, 2007.
[13] D. C., Lynch, and M. T. Rose, Internet System Handbook.
Reading, MA, USA: Addison-Wesley, 1993.
[14] V. Jacobson, C. Leres, and S. McCanne, "Tcpdump," in
Manual Page, 1997.
[15] T. Kushida, "An empirical study of the characteristics of
Internet traffic," Computer Communications, vol. 22, pp.
1607-1618, 1999.

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[1] L. Xuejian, Y. Jing, and W. Minghui, "A heterogeneous

evolutional architecture for embedded software," in
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[2] E. A. Lee, "Embedded Software " in Advances in Computers,
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[3] J. W. K. Hong, S. S. Kwon, and J. Y. Kim, "WebTrafMon:
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Acquisition System Using 32 Bit Single Board Computer:

113
 2009 International Conference on Computer Engineering and Technology
PB-GPCT
A Platform-Based configuration tool
Yan huiqiang, Tan runhua, Shi kangyun, Lu fei
Institute of Design for Innovation, Hebei University of Technology, TianJin, China
E-mail: yanhuiqiang@ scse.hebut.edu.cn
Abstract
In this paper, the Platform-based generic product
configuration tool(PB-GPCT) is presented which is
developed by Institute of Design for Innovation, Hebei
University of Technology. The PB-GPCT is designed to
be used by sales engineers for the configuration of
complex products, so that customers can get
customized products in tolerable time. Comparing with
other configurators, platform is introduced into PB-
GPCT. Being a structure-based and domain
independent system, the PB-GPCT can be used in
different companies without doing any modification. In
order to achieve this goal, knowledge-based
configuration methods are employed. The
configuration model, Configuration knowledge and
configuration constraints are discussed in this paper.
1. Introduction
With the competitiveness become more and more
violent in industries, companies have to support Mass
Customization to keep low cost and meet individual
requirements at the same time. Product families design
is the most important method to implement Mass
Customization. Product families design usually is
based on platform. Meyer and Lehnerd define a
platform as a “set of common components, modules, or
parts from which a stream of derivative products can
be efficiently created and launched” [1]. Muffato
defines platform similarly as: “a relatively large set of
product components that are physically connected as a
stable sub-assembly and are common to different final
models” [2].So that the platform is the components and
the constraints among them from which different
products can be derived. There are several methods for
designing a platform. Simpson et al. give two types of
platform design methods: (1) top-down and (2)
bottom-up [3]. Another way of categorizing platform
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DOI 10.1109/ICCET.2009.20
design methods is to distinguish between module based
and scale based platform[4]. For brevity, the design
approach of platform is not discussed in this paper. The
relationship between product families and platform is
showed in Figure 1. In terms of customer’s
requirements, firstly, platform is determined; then
different products are derived by selecting different
components which obey the constraints among
platforms and components.
Platform 1
Product family
Component 1
Component 2 Product 1
Platform 2
User requirements
Component 3 Product 2
Component 4 Product 3
Platform 3
Component n
Platform m
Constraints
Product n
Figure 1. Relationship between platform and
product family
Mittal & Frayman defines configuration as a form
of design, which selects assembly of components from
a set of pre-defined components to meet customer’s
requirements[5]. Configuration is described as selecting
objects and their relations from a set of objects and a
set of relations according to customer’s requirements
[6]
. Both definitions emphasize product is composed of
components and relationship among components;
configuration is choosing components from a set of
pre-defined components to construct the products.
Corresponding to these definitions, a product
configuration is defined as a tool which supports the
product configuration process so that all the design and
configuration rules which are expressed in a product
configuration model are guaranteed to be satisfied [7].
Configuration model is the product model which is
scoped within the conceptualization of configuration
domain. The configuration model is defined as set of
pre-designed components, rules on how these can be
114
combined into valid product variants and rules on how
to achieve the desired functions for a customer [8]. But
from viewpoint of product family design, it is different
as stated above. To this end, platform-based
configuration is introduced into PB-GPCT.
This paper is organized into 4 sections. In section 1,
the concept of platform-based configuration is
introduced. In section 2, the technique relevant to
configuration is discussed. In section 3, the
maintenance and analysis of constraints is presented.
Lastly, the implementation of PB-GPCT and future
works is discussed.
2. Configuration model
2.1. Configuration model definition
PB-GPCT defines the configuration model (CM) as:
CM = ({P1,…Pm}, {C1,…Cn},{R1,…Rs}). Ci is the
element of the product model which is called
component type; Pi is platform which is composed of
component instances and component types; Ri is
constraints among platforms and component types.
Corresponding to this, configuration defines as:
Solution space= f(CMi,{R1,…Rn}). CMi is the
configuration model which the platform has been
designated in term of customer’s requirements.
Namely, configuration is the process to instantiate
configuration model.
2.2. Configuration model representation
UML (Unified Modeling Language) is the leading
industrial object-oriented modeling language for
software engineering. It is also widely accepted to
describe product model, so that PB-GPCT selects
UML to describe configuration model. OCL (Object
Constraint Language) is a formal language used to
describe expressions on UML models. CCL
(Configuration Constraint Language) is based on OCL
which is used to describe the constraints of the
configuration model. Part of configuration model of
configurable PC expressed by UML and CCL is
showed in Figure 2. The configuration model is
composed of components, relations among components
and rules represented by CCL.
Component type. There are two kinds of
component types. One is virtual component which does
not have a physical correspondence, e.g. HD-Unit and
CPU. The other is physical component which typically
has a bill-of-material associated with it, e.g. VideoCard
and MotherBoard.
Component instance. When all the properties of
component are given values, we call it Component
115
instance. Virtual component and physical components
all have component instance associated with it.
However, the properties of physical component are
given by product experts before configuration, the
properties of virtual component are assigned by sales
engineers at the configuration phase.
Figure 2. Part of configuration model of
configurable pc
Relations. The taxonomies of relations are Part-of
and Is-a. ComponentA part-of componentB means that
componentB is part of componentA, e.g. CPU,
VideoCard and HD-Unit are parts of MotherBoard.
ComponentB Is-a componentA means that
componentB is a kind of componentA, e.g. IDE-Unit
and SCSI-Unit are two kinds of HD-Unit.
Rules. Rules not only include the constraints among
component types, but also include the constraints
among component instances. The taxonomies of
constraints are unary, binary, global and incrementally
constraints [9].
Platform. Platform consists of common elements
from which a range of products can be derived. For a
configurable PC, the platform elements are
MotherBoard and CPU. Sales engineer first selects
platform (determine the MotherBoard and CPU)
according to customers requirements, then based on the
platform other components are chosen.
2.3. The taxonomies of Configuration
knowledge
Being knowledge extensive system, knowledge
representation is critical to configuration tool.
Configuration tasks are summarized to have the
following characteristics [10]:
(1) A set of components in the application domain.
(2) A set of relations between the domain
components.
(3) Control knowledge about the configuration
process.
 3. Configuration constraints
How to write and evaluate constraints is core of PB-
GPCT. Maintenance and valuation of constraints is
show by Figure 5.

Figure 3. Configuration knowledge

Corresponding to this viewpoint, the configuration

knowledge is categorized into component knowledge,
Figure 5. Maintenance and valuation of
constraint knowledge and control knowledge.
However, as stated above, PB-GPCT is different from constraints
this view. The Configuration knowledge of PB-GPCT
is showed in Figure 3. 3.1. Edit constraints
Component knowledge. It is the element of
product, which describes product architecture utilizing The Object Constraint Language (OCL) is an
UML. expression language based on first-order logic which
Constraint knowledge. It describes the rules enables the definition of constraints on object-oriented
among components types utilizing CCL. models. However, writing constraint expressions in
Component Instance knowledge. Component OCL is an error prone task. CCL is sub collection of
Instance is component type which is instantiated by OCL which can be written easily using the edit
product experts. In general, component instance is windows showed in Figure 6.
mass-produced component.
Platform knowledge. It is the sub collection of
component instances from which individual products
can be derived.
Control knowledge. It is used to control the
process of configuration. In terms of configuration
policy, different control knowledge will be required.
According to Figure 3, all the knowledge is
managed by product experts except for control
knowledge . The main tables which stores
Configuration knowledge are showed in Figure 4.

Figure 6. CCL constraint edit windows

The following are two constraint examples in

Figure 2.
Constraint1:MotherBoard.port = CPU.port
Constraint2:MotherBoard.port = HD-Unit.port
Comparing with OCL, the following grammars of
OCL[11] are not supported by CCL.
Let Expressions. The let expression allows one to
define a variable that can be used in the constraint.
Figure 4. Main tables storing configuration Tuples. A tuple consists of named parts, each of
knowledge base which can have a distinct type.

116
 Messages. It is used to specify the communication
between classes.
For brevity, the detail of CCL is not discussed in
this paper.
3.2. Constraint patterns
Writing constraint expressions is a time-consuming
task, though users can do it flexible using edit window
showed in Figure 7. To address these issues, PB-GPCT
constraint patterns provides a repository of constraint
patterns that encapsulate the expertise needed to easily
and quickly add constraints to a model. Constraint
patterns are predefined constraint expressions that can
be instantiated and subsequently configured for
developing constraints in a concise way. The
taxonomies of constraint patterns are Atomic
Constraint Patterns and Composite Constraint
Patterns[12]. The following patterns are supported now.
(1) AttributeValueRestriction pattern.
(2) UniqueAttributeValue pattern.
(3) Exists pattern.
(4) ForAll pattern.
(5) IfThenElse pattern.
3.3. Analysis of constraints
After finish inputting the constraints, PB-GPCT has
to ensure the form of the constraints is correct through
Lexical analysis and Syntactic analysis.
(1) Lexical analysis. It distinguishes between the
identifiers and the keywords of CCL. Let Expressions
is not supported by CCL, so that the identifiers are
component type names, component instance names and
the properties of the component types. The result of
this process is a sequence of the tokens which will be
the input of the Syntactic analysis.
(2) Syntactic analysis. The task of syntactic analysis
is to determine whether constraint expressions are
correct input by user. PB-GPCT employed the LL (1)
[13]
approach to do the analysis because Left recursive
and Backtracking do not exist in the grammars of CCL.
3.4. Valuation of constrains
At configuration phase, PB-GPCT has to evaluate
the configuration items, which selected by user, to
ensure to be consistent with constraints. In order to
realize this function, constraints are transformed into
first logic sentence [14]at the first step; then the first
logic sentence is transformed into specific language to
execute to evaluate the constrains.
117
4. Implementation
The configurator PB-GPCT employs the popular
language C#, which is more efficient than java; and
uses SQL server 2000 as the Database. The main GUI
is showed in Figure 7. In order to support extensibility
and upgradeability, Plug-in architectures are employed,
since plug-in architectures can quickly customize
applications by mixing and matching the plug-ins, or
write new plug-ins for missing functions. Constraints
analysis and constraints evaluation are main plug-in
component of the system.
Figure 7. The system of PB-GPCT
5. Conclusions and Future works
In this paper, the theory of PB-GPCT is presented.
It has been applied to TIANJIN NO. 2 MACHINE
TOOL CO., LTD. Using this software, the average
designing time has been declined from more than one
month to about one week.
However, there will be numerous constraints for
complicated products in configuration model.
Consequently, contradictory constraints may be
defined inadvertently. A future extension might
support constraint consistency check.
Another task for the future will be support 3-D
configurations which make WYSIWYG(What You See
Is What You Get) feasible in the system.
6. Acknowledgment
It is a project supported by Natural science in
Tianjin City (07JCZDJC08900) and National Natural
Science Foundation of China (50675059).
7. References
[1] Meyer, M. H. & Lehnerd, A. P. The power of product
platforms. The Free Press. 2000. New York, NY. ISBN 0-
648-82580-5.

[2] Muffato, M. Introducing a platform strategy in product
development. International . 60-61. pp. 145-153. 1999.
Journal of Production Economics.
[3] Simpson, T., Maier, J., & Mistree, F. Product platform
design: method and application. Research in Engineering
Design. vol. 13. No. 1. pp. 2-22. 2001.
[4] Simpson, T.W. Product platform design and
customization: Status and promise. AIESAM, Special issue:
Platform product development for mass customization. Vol
10. No. 1. Jan 2004.
[5] Mittal, S. & Frayman, F. (1989). Towards a generic
model of configuration tasks. In the 11th IJCAI, pages 1395-
1401, San Mateo, CA, Morgan Kaufman.
[6] A.GuenterandC.Kuehn. “Knowledge-Based Configurati
on-Survey and Future Directions”. In XPS-99 Proceedings,
Lecture Notes in Artificial Intelligence No.1570,Springer-
Verlag, Wuerzburg, 1999.
[7] Görel Hedin, Lennart Ohlsson, John McKenna.In
Proceedings of the 8th International Symposium on System
Configuration Management (SCM-8), Brussels,July20-21,
1998. LNCS-1439, pp 107-126, Springer-Verlag.
[8] K. OLDHAM, “Modelling Knowledge used in the Design
of Hosiery Machines”. Proceedings 33 International
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MATADOR Conference, D.R. Hayhurst (ed.) Springer-
Verlag, July 2000 93-98.
[9] Ander Altuna, Alvaro Cabrerizo. “Co-operative and
Distributed Configuration”. NOD 2004 September 27-30
2004 Erfurt, Germany.
[10] A.GuenterandC.Kuehn. “Knowledge-Based Configurati
on-Survey and Future Directions”. In XPS-99 Proceedings,
Lecture Notes in Artificial Intelligence No.1570,Springer-
Verlag, Wuerzburg, 1999.
[11] OMG. Object Constraint Language Specification 2006.
[12] M. Wahler, J. Koehler, and A. D. Brucker. Model-
Driven Constraint Engineering. Electronic Communications
of the EASST, 5, 2007.
[13] Chen ying.Principle of Compilation. 2006.Beijing.
[14] A. Felfernig, G. Friedrich, and D. Jannach. Generating
product configuration knowledg Internationa bases from
precise domain extended UML models. In Proceedings of the
Conference on Software Engineering and Knowledge
Engineering (SEKE’2000), pages284-293, Chicago,USA,
2000.
 2009 International Conference on Computer Engineering and Technology
A feasibility study on hyperblock-based aggressive speculative execution model
Ming Cong, Hong An, Yongqing Ren, Canming Zhao, Jun Zhang
Department of Computer Science and Technology Key Laboratory of Computer System and Architecture
University of Science and Technology of China
Hefei, 230027, China
mcong@mail.ustc.edu.cn, han@ustc.edu.cn, {renyq, zcm, junzh}@mail.ustc.edu.cn
Abstract—Speculation execution model which executes sequential
programs in parallel through speculation is an effective technique
for making better use of growing on-chip resources and exploiting
more instruction-level parallelism of applications. However,
accompanied high communication overheads and roll-back
penalties can not be neglected. This paper focuses on analyzing the
feasibility of aggressive speculation execution model and finding
an appropriate degree of “aggressiveness” under hyperblock-
based execution model. We analyze the characteristic of control
dependences and data dependences between adjacent hyperblocks,
and propose a quantitative analysis method to detect data
dependences on hyperblock-based execution model, and then
evaluate the feasibility of aggressive speculative execution model
on 8 applications from SPEC2K. Our experiments show most
applications can get high prediction accuracy on control-flow from
hyperblock-based prediction mechanisms, especially SPECFP.
Furthermore, we analyze factors which impact expected prediction
depth and find depth depends more on application than predictors.
Keywords-hyperblock; speculative execution; prediction;
control dependence; data dependence
I. INTRODUCTION
Modern CMOS technology brings the increasing number
of transistors on one chip, so how to effectively utilize the
growing resources and exploit more parallelism to accelerate
applications is an urgent problem for computer architects.
However, to expose potential of instruction-level parallelism
(ILP), control-flow and data-flow constraints inherent in a
program must be overcome. Speculative execution [1] which
executes programs aggressively has become a mainstream
technique to reduce the impacts of dependences in high
performance microprocessors.
The block-based execution model [4] has been proposed
to enlarge the instruction window, which may achieve high
ILP and high resource utilization. Recent works on
computer architectures, such as TRIPS and Multiscalar, use
block-atomic(tasks in Multiscalar [4]) execution, in which
each block is fetched, executed, and committed atomically,
behave like a conventional processor with sequential
semantics at the block level. A hyperblock [2][3] is a set of
predicated basic blocks combined by compiler in which
control flow only enters from the top, but may exit from one
or more locations. With hyperblock, larger instruction
window and more ILP can be achieved than basic block.
Similar to the multi-issue in superscalar, block-based
execution model utilizes additional resources to execute more
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.30
Chinese Academy of Sciences
Beijing, 100080, China
blocks simultaneously on the processor substrate, with all but
one executing speculatively. It’s obvious that the feasibility
of aggressive execution model depends largely on the
effectivity and prediction accuracy of speculation. However,
as speculation in high-ILP processors become more
aggressive, the number of mis-speculation increases with
growing numbers of inflight instructions/ blocks. Although
many mechanisms have been proposed for speculative
execution, efficiency is still limited because of mis-
speculation penalties, high communication overheads and etc.
This paper focuses on the analysis of hyperblock-based
aggressive execution model. We concentrate on finding a
tradeoff and maximum potential of aggressive execution that
can be exploited, and analyze dependent behaviors of
applications under this model. At the aspect of control-flow,
we evaluate performance of three branch predictors, and
estimate the expected prediction depth, and then analyze the
effecting factors of aggressive speculation with control-flow.
In the view of data-flow, we propose a quantitative analysis
of data dependence on hyperblock-based execution model,
and analyze the distributions of data dependences between
hyperblocks and their impacts on the depth of prediction.
We evaluate the feasibility of aggressive speculative
execution model on 8 applications from SPEC2K. Our
experiments show that most applications have good
predictability, and high prediction accuracy on control-flow
can be gained by using hyperblock-based branch prediction
mechanisms, especially SPECFP applications. Furthermore,
we analyze factors which impact expected prediction depth,
and find that it depends more on applications themselves
than predictors, and expected prediction depth differs
depending on the characteristics of each application.
The rest of this paper is organized as follows. Section II
describes the aggressive execution model in detail, and
analyzes it in the aspects of both control dependences and
data dependences. Section III experimentally evaluates and
analyzes the feasibility of aggressive execution, which
corresponds to the part in section II. Section IV introduces
related works on aggressive execution models. Finally in
section V we make our conclusions.
II. HYPERBLOCK-BASED AGGRESSIVE EXECUTION
MODEL
Current researches of ILP processors focus on exposing
more inherent parallelism in an application to obtain higher
performance. To effectively exploit ILP in programs, the
119
dependences between instructions need to be detected and
avoided to prevent pipeline stalls, among which control
dependence and data dependence are of the most importance.
A. Speculation execution on the control-flow
1) Differences between hyperblock branch predictor
and conventional predictor
Branch predictions are made based on branch history
information, so that the accuracy of predictors depends on
the frequency of historical representation of previous branch
target address. Branch predictions in hyperblock-based
model have high parallelism, but their mechanisms are much
different with those in conventional superscalar processors
from following aspects.
First, in superscalar model, each branch has only one exit
so that one bit of exit information is enough for representing
taken or not taken (T/NT); while in block-based model, each
block has several exit points, predicting the exit taken out of
multiple possible exits is a multi-way branching problem,
some bits of exit ID or target address of the branch should be
reserved as the exit information.
Secondly, dedicated adder in the fetch mechanism and
branch target buffer (BTB) of RISC architectures can be
used to compute PC relative target addresses before they are
computed by ALU(s) in execution stage; but in hyperblock-
based model, the variable block sizes and different target
addresses which exit may correspond to force us to predict
target addresses of all exit point.
Thirdly, A return address stack (RAS) used for return
address prediction makes it more accurate, and the type of
branch instruction can be easily got from a pre-decoder in
RISC architectures, but the type of exit points in blocks is
hard to get before block committing. So we need a
mechanism to predict the type of exit points.
2) Design space of hyperblock-based branch predictors
In this section we describe the design space of
hyperblock-based branch predictors in detail. Based on
conventional branch predictors and the characteristics of
hyperblocks, we consider a two-level predictor [5] which
predicts the first branch that will be taken in a hyperblock.
As shown in Figure 1, the first level predicts the exit point,
and then the second level produces the branch target address.
Figure 1. 2-level hyperblock-based branch predictors
a) Exit predictor: Corresponding to branch behaviors
of conventional branch predictors, the predictors for exit [7]
can be organized around following methods.
Global predictor: Global predictor indexes the table of
bimodal counters with the recent global history integrated
with branch instruction addresses to get branch behavior. In
hyperblock-based model, we replace the T/NT stored in
Pattern History Table (PHT) with exit numbers of each
block; branch behaviors in History Register Table (HRT) are
also replaced by recent exit history of blocks. In addition, for
120
the accuracy of prediction, we use two HRT, one for
updating the prediction information, the other for recovering.
Local predictor: Local branch predictor keeps two tables.
One is a local BHT indexed by the low-order bits of each
branch instruction’s address; it records T/NT history of N
most recent executions of each branch, which is different
from global predictor. The other is a PHT indexed by the
generated value from the branch history in BHT. Local
prediction is slower than global prediction because it requires
two subsequent table lookups for each prediction, but it may
be favorable for deep prediction of a certain local branch.
Tournament predictor: Since different branches may be
predicted better with either local or global techniques,
tournament predictor uses a choice predictor to dynamically
select the best method between two prediction for each
branch, it is nearly as accurate as local predictor, and almost
as fast as global predictor. The choice prediction is made
from the table of 2-bit prediction counters indexed by path
history; processor trains counters to prefer the correct
prediction whenever the local and global predictions differ.
b) Block target address prediction
Block branch target address determined by the exit
instruction is the starting block address of next task. After
the exit is predicted, we use both predicted exit and branch
address to determine the target. Conventional methods
usually use BTB and RAS, in which BTB handles prediction
of branch and call exit, RAS predicts address of return exits
with known types of each exit, this will certainly degrade
prediction accuracy and increase its complexity. So we only
use BTB for the prediction, which is easy and versatile.
Figure 2. Structure of the BTB
As shown in Figure 2, each item of BTB maintains target
addresses of several exit and hysteresis bits, indexed by the
block address and exit ID which was predicted by branch
predictor, to obtain the Block Target Address.
3) Evaluation of three branch predictors
In this section we evaluate prediction accuracy of exit
predictors with block target address predictor presented
previously on 8 benchmarks from SPEC2000. Both Global
predictor and Local predictor are configured to be 16384
96%
94%
92%
Prediction Accuracy
90%
88% Global
86% Local
84% Tournament
82%
80%
78%
76%
ammp art bzip equake gzip mcf vortex parser
Figure 3. prediction accuracy of three branch predictors
 Branch Predictor with Global-Predictor Branch Predictor with Local-Predictor Branch Predictor with Tournament-Predictor
100%
f

100% 100%

f
90% 90% 90% ammp
Prediction Accuracy

Prediction Accuracy

Prediction Accuracy
80% 80% 80% art
70% 70% 70% bzip
60% 60% 60% equake
50% 50% 50% gzip
40% 40% 40%
mcf
30% 30% 30%
20% 20% 20% vortex
10% 10% 10% parser
0% 0% 0%
1 2 3 4 5 6 7 1 2 3 4 5 6 7 1 2 3 4 5 6 7
Prediction Depth Prediction Depth Prediction Depth
(a) (b) (c)
Figure 4. Prediction accuracy of different branch predictor

entries, Tournament predictor contains 8192 entries for both

global and local predictor; The BTB has 1024 entries. III. EXPERIMENTAL EVALUATION
From Figure 3, it is clear that global predictor performs
better than local predictor on art, bzip, gzip and mcf in which A. Methodology
global histories is predominant; but equake and vortex on Our experiments are performed on the TRIPS toolchain
which local histories have more impact perform better with which supports hyperblock-based multi-level speculation.
the local predictor. The tournament predictor takes full TRIPS is a block-atomic execution model, the size of
advantage of global and local histories, achieves better instruction window can reach up to 1024 by speculation. It
performance than both of them. contains compiler (Scale [3]), functional simulator tsim_arch
and cycle-accurate simulator tsim_proc [6]. Tsim_proc can
B. Speculative execution on the Data-flow generate trace files containing all events of the simulating
Data dependence is an important factor that influences process. Although our experiments are based on TRIPS, our
effectivity of multi-level speculative execution. In this research is not limited to this architecture but aims at all
section, we conduct a quantitative analysis of data current hyperblock-based execution models. We use 8 whole
dependence on hyperblock-based execution model, and benchmarks written in C from SPEC2000 benchmark suite,
analyze its impact on the depth of prediction. To be able to including 3 float point benchmarks: art, ammp, equake, and
obtain benefits from data speculation, we must execute 5 integer benchmarks: gzip, mcf, parser, vortex, bzip2.
instruction streams correctly, so we make two assumptions
before the analysis: (1) A perfect branch predictor is B. The Speculative execution on control dependence
assumed. (2) No complexity of hardware implementation is between hyperblocks
taken into accounts. These assumptions would not affect the Large instruction window is built to issue more
analysis of the natural characteristic of programs. independent instructions per cycle, but the effective size of
The dependent relation between instructions in traditional instruction window is limited by the depth of control-flow
processors can be described by dependence-distance (the speculation, so we evaluate the feasibility of predicting more
number of instructions between data producers and aggressively and analyze the appropriate depth of prediction.
consumers); but it is no longer applicable for measuring First, we evaluate prediction accuracy with different
hyperblock-based model, because different impacts of prediction depths; then we depict the depth distribution and
instructions on separating producer and consumer, and predictability for all applications; finally, we analyze the
instructions in the same block can execute in parallel. expected prediction depth of different applications based on
instructions distribution, which can help us adopt appropriate
∞

Dependence depeth = ∑ Pi × i (1) prediction depth while studying aggressive execution model,
and we analyze impacts of expected prediction depth on
i =1
different predictors and configurations.
We define the term “Dependence depth” to measure the 1) Evaluate predictors with certain prediction depth
degree of dependences between blocks. There i denotes the We evaluate prediction accuracy of global and local
distance from current block; Pi denotes the proportion of predictors with certain prediction depths (range from 1~7),
instructions which have dependence with instructions in the as in figure 4, the prediction accuracy decreases while
previous ith block to the total number of instructions within prediction depth increases, but the depressive gradient slows
current block. Dependence depth describes the dependence down with deeper depths, and most applications still keep a
strength between blocks. The larger dependence depth means high prediction accuracy even the prediction depth is up to 7.
the weaker dependent strength, so that we can exploit more This demonstrates deeper prediction is feasible to most
parallelism from inter-blocks. And it is proportional to the applications. Global and local predictor both have strengths
potential depth of speculated execution, which means on different types of applications with small depths, but local
programs could benefit from speculating more blocks predictor performs better with the increase of depth, which
according to larger dependence depth. indicates that it is more powerful for deeper speculative
execution. Tournament predictor performs best as it can
adapt to pattern of applications with both previous predictors.

121
 100%

Instructions distribution in blocks

40
90%
35
Proportions of blocks

80%
70% 30
60% >10 25 rd_reg insts
50% 6~10
load insts
20
40% 2~5
15 normal insts
30% 0~1
20% 10
10% 5
0%
0
ammp art bzip equake gzip mcf vortex parser
mcf gzip vortex bzip2 parser art ammp equake
Figure 5. Proportion of blocks VS. prediction depth
Figure 8. Numbers of dependence instructions

2) Distribution of Prediction depth on blocks Figure 7 describes the expected prediction depth with
We introduce a quantitative method for denoting the tournament predictor which is configured with 16x the size
potential of deep prediction of distinct applications of old PHT. From the results, we can see configurations of
intuitively. We predict each block with unbounded depths predictors have less impact on the expected prediction depth.
until the prediction cannot continue; thus each application In most applications, low prediction accuracy is not made by
can be divided into several block sequences with various the conflict on PHT, but comes from the characteristic of
block numbers. According to a statistical analysis of applications, we should consider more improvements on the
different prediction depths among various sequences, we can predictors in order to better adapt to patterns of applications.
estimate the potential of deep prediction for applications.
Figure 5 illustrates the proportion of blocks that predicted C. The Speculative execution on Data dependence
with different prediction depths (0~1, 2~5, 6~10, 10~∞) on 1) Numbers of dependent instructions in applications
tournament predictor, among which the depth 0 indicates Data dependence between hyperblocks arises from the
blocks cannot be predicted. Vortex, art and ammp have high load or register-read instructions (dependent instructions).
proportions with deeper prediction depth; bzip and gzip are Figure 8 presents a statistic of the numbers of overall
just the opposite in which the former group can be predicted instructions, load instructions and register-read instructions
aggressively and the latter are less predictable. within hyperblocks. Although these values are closely related
3) Expected Prediction Depth to compiles, we can still find that the number of dependent
Prediction depth distribution can not reflect actual quality instructions is considerable in hyperblock, from approxima-
of prediction depth for applications, so we further introduce tely 18% in gzip to 52% in ammp. We would not achieve the
the Expected Prediction Depth, mean value of prediction anticipated speedup if we only increase speculative depths
depth distribution that reflects the magnitude of prediction without considering the impact of data dependence.
and even prediction accuracy. 2) Distribution of data dependences
Figure 6 shows the expected prediction depth evaluated Previous statistic of instructions number only reflects the
under the condition that size of local and global PHT is ubiquitous of data dependence under block-level execution
configured to 16348, we can see that the types of predictors model, but cannot completely substitute for the data
have big impact on the expected prediction depth. For each dependence behaviors. Results may be inequable while a
predictor, applications has distinctive results, we can see block has data dependences with blocks located in different
values of ammp, art, mcf and vortex exceed 10, and others place, so we further analyze the distribution of data
are around 5, which also shows that these applications can dependence (Figure 9). Most data dependence of a
accommodate well to such prediction model. hyperblock comes from its previous adjacent blocks, our
14 experiments show 20% or even 40% [parse] of data
Expected Prediction Depth

12
dependence locates in two adjacent blocks, and an average of
10
Global 40% or even 60% [parse] locates in six contiguous blocks.
8

6
Local These differences are totally attributed to the natural
Tournament
4
characteristics of applications.
2 The unbalanced distribution on data dependences is not
0 what we expect, as we cannot make subsequent blocks
ammp art bzip equake gzip mcf vortex parser
executed even increasing the speculation depth because of
Figure 6. Expected prediction depth the serious dependence between adjacent blocks.
100%
16
90%
Expected Prediction Depth

14 >64blocks
80%
Expected speculation depth

12 33~64blocks
70%
10 17~32blocks
Old 60% 11~16blocks
8
New 50% 9~10blocks
6 7~8blocks
40%
4 5~6blocks
30%
2 3~4blocks
20%
<=2blocks
0 10%
ammp art bzip equake gzip mcf vortex parser 0%
mcf gzip vortex bzip2 parser art ammp equak e

Figure 7. Expected prediction depth with different configured predictor Figure 9. Distribution of data dependence in applications

122
 45
40
V. CONCLUSIONS
Data Dependence Depth

35
Hyperblock-based execution model can tremendously
30
25 improve the size of instruction windows for high ILP and
20 good performances. Since enlarging a hyperblock would
15
10
become more difficult under the constraints of compiler
5 technology and inherent characteristics of applications, we
0 introduce a preliminary evaluation of aggressive execution
mcf gzip vortex bzip2 parser art ammp equake
model. Our experiments concentrate on the characteristics of
Figure 10. Data dependence depth in applications dependence distribution and prediction depth in views of
70
speculative execution both on control-flow and data-flow.
mcf Under this model, most applications have good predicta-
Expected speculation depth

60

50
gzip bility, high prediction accuracy and expected prediction
vortex
40 bzip2
depth with control-flow prediction mechanisms; and
30 parser expected prediction depth which reflects the degree of
20
art predictability mainly depends on applications themselves
10
ammp
equake
rather than predictors. Moreover, although prediction
0
accuracy on control-flow decreases while prediction depth
increases, it remains high within acceptable bounds. Finally,
.5
%

%
.5%

.5%

.5%

.5%

.5%
27
20

25

30

35

40

45

50
22

32

37

42

47
<=
<=

<=

<=

<=

<=

<=

<=

we introduce a quantitative method for denoting speculative

<=

<=

<=

<=

<=

Figure 11. Expected speculation depth VS. Different dependence depth execution potentials for distinct applications intuitively, and
analyze the distributions of data dependences between
3) Dependence Depth and Expected Prediction Depth hyperblocks and their impacts on depth of prediction. As we
Figure 10 shows dependence depth of applications, figure observes, many applications have high expected prediction
11 shows expected speculation depth under the constraints of depth, but whether they are suitable for aggressive executing
certain data dependence proportion. As in equake, the depends on data dependences between adjacent hyperblocks.
expected speculation depth is only up to 5 with less than
47.5% data dependence because data dependence within 6 ACKNOWLEDGEMENT
blocks is 48.696% (Fig. 9), bigger than 47.5%. Applications This research was supported financially by the National
of SPECINT have low speculation depths less than 10(4~8 Basic Research Program of China under contract
on average), in contrast, applications of SPECFP have better 2005CB321601, the Natural Science Foundation of China
speculation depth that up to 64 with 50% data dependence. grant 60633040, the National Hi-tech Research and
IV. RELATED WORK Development Program of China under contract
2006AA01A102-5-2, the China Ministry of Education &
Speculative executions on hyperblocks mainly focus on Intel Special Research Foundation for Information
two directions. One is resolving control-flow between Technology under contract MOE-INTEL-08-07.
hyperblocks such as Multiple Branch Predictors and Region
Predictors, the other is Predication which converts control REFERENCES
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Multiscalar processor, and subsequently refined by Jacobson um on Code Generation and Optimization, 2006, pp. 185-195.
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leave a code region (such as a Multiscalar task). The local, Bringmann, “Effective Compiler Support for Predicated Execution
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proposed from Jacobson et al. [5] T.Y. Yeh and Y. Patt. Two-level adaptive branch prediction. In
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123
 2009 International Conference on Computer Engineering and Technology
Parallel Method for Discovering Frequent Itemsets Using Weighted Tree Approach
Preetham Kumar
Department of Information and Communication
Technology
Manipal Institute of Technology,
Manipal, India
prethk@yahoo.com
Abstract— Every element of the transaction in a
transaction database may contain the components such as item
number, quantity, cost of the item bought and some other
relevant information of the customer. Most of the association
rules mining algorithms to discover frequent itemsets do not
consider the components such as quantity, cost etc. In a large
database it is possible that even if the itemset appears in a very
few transactions, it may be purchased in a large quantity.
Further, this may lead to very high profit. Therefore these
components are the most important information and without
which it may cause the lose of information. This motivated us
to propose a parallel algorithm to discover all frequent
itemsets based on the quantity of the item bought in a single
scan of the database. This method achieves its efficiency by
applying two new ideas. Firstly, transaction database is
converted into an abstraction called Weighted Tree that
prevents multiple scanning of the database during the mining
phase. This data structure is replicated among the parallel
nodes. Secondly, for each frequent item assigned to a parallel
node, an item tree is constructed and frequent itemsets are
mined from this tree based on weighted minimum support.
Keywords-attribute;cost;component;parallel;Weight
I. INTRODUCTION
The goal of knowledge discovery is to utilize those
existing data to find out new facts and to uncover new
relationships that were previously unknown, in an efficient
manner with minimum utilization of the space and time.
Mining association rules is an important branch of data
mining, which describes potential relations among data items
(attribute, variant) in databases, the well-known Apriori
algorithm [3] was proposed by R. Agrawal et al., in 1993.
Mining association rules can be stated as follows: Let I={i1,
i2, … im } be a set of items. Let D, the task-relevant data, be
a set of transactions, where each transaction T is a set of
items such that T ⊆ I. The quantities of items bought are not
considered. Each transaction is assigned an identifier, called
TID. Let A be a set of items, a transaction T is said to
contain A if and only if A ⊆ T. An association rule is an
implication of the form AÎB, where A ⊆ I, B ⊆ I, and
A∩B= . The rule AÆB holds in the transaction set D with
support s, where s is the percentage of transactions in D that
contain AUB (i.e., both A and B). This is taken to be the
probability, P(A∩B). The rule AÆB has confidence c in the
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.194
Ananthanarayana V S
Department of Information Technology
National Institute of Technology Karnataka
Surathkal, India.
anvs1967@gmail.com
transaction set D if c is the percentage of transactions in D
containing A that also contain B. This is taken to be the
conditional probability, P(B|A). That is,
Support(AÆB)=P(A∩B)=s, Confidence(AÆB)=P(B|A) =
Support(AÆB)/Support (A)=c.
Mining of association rules is to find all association rules
that have support and confidence greater than or equal to the
user-specified minimum support and minimum confidence
respectively [2]. This problem can be decomposed into the
following sub problems:
All itemsets that have support above the user specified
minimum support are discovered. These itemset are called
the frequent itemsets.
For each frequent itemset, all the rules that have user
defined minimum confidence are obtained.
The second sub problem, i.e., Discovering rules for all given
frequent itemsets and their supports, is relatively
straightforward as described in [1].
Discovering frequent itemsets, considered as one of the
most important tasks, has been the focus of many studies in
the last few years. Many solutions have been proposed using
sequential or parallel algorithms based on user defined
minimum support. However, the existing algorithms depend
heavily on massive computation that might cause high
dependency on the memory size or repeated I/O scans of the
datasets. The published parallel implementations of
association rule mining inherited most of these problems in
addition to the new costly communication cost most of them
need. Parallel association rule mining algorithms currently
proposed in the literature are not sufficient for large datasets,
and new solutions, that do not heavily depend on the
repeated I/O scan, less reliant on memory size, and do not
require a lot of communication costs between nodes, still
have to be found. In addition, the existing algorithms
discover all frequent itemsets based on user defined
minimum support without considering the components of the
transaction such as weight or quantity, cost and other
relevant information of the customer which lead to profit.
This motivated us to design a new parallel algorithm that
is based on the concept of Weighted Tree.
II. PROPOSED METHOD
This algorithm is divided into two phases. The first one
requires only one scan of the database and generates a
124
special disk-based data structure called Weighted Tree. In
the second phase, the Weighted Tree obtained is reduced to
contain only frequent items and then all the branches of the
tree are sorted according to the increasing order of the
frequency. This tree is called an ordered Weighted Tree. This
tree is replicated among nodes and mined in parallel.
The weighted minimum support is the minimum weight
an itemset has to satisfy to become frequent. If an itemset
satisfies user defined weighted support then we say that it is
weighted frequent itemset.
Consider a sample database given in Table 1, in which
every element of each transaction represents either quantity Figure 1. Branch of a Weighted Tree
or cost of the respective attribute or item.

Table 1. Sample Database

Figure 2 represents the Weighted Tree corresponding to
the Sample Database given in the Table 1.

It may so happen that an itemset appears in a very few

transactions in a large quantity or cost which leads to profit
will not qualify as frequent itemset based on user defined
minimum support. This results in a lose of information. In
the sample database given in the Table 1, if user defined
minimum support is 2 transactions then an item D is not
frequent and will not appear in the set of frequent itemsets
even though if it is bought in large quantity and leads to
Figure 2 .Weighted Tree for Table1
more profit than other frequent items.
Structure of Weighted tree
The idea of this approach is to associate each item with all
transactions in which it occurs.The Weighted tree has two The Parallel Algorithm
different nodes and its branch is shown in the Figure 1.
The first type of node labeled with attribute contains This algorithm involves 3 steps. They are
attribute name and two pointers, one pointing to the nodes A. Construction of Weighted Tree
containing transaction ids and weights and another is a child B. Removal of infrequent attributes of Weighted Tree
pointer pointing to the next attribute. This node represents C. Arrange the attribute lists of Weighted Tree in an
the head of that particular branch. increasing order of their weighted minimum support
The second type of node has 2 parts. First part is labeled TID D. Parallel mining of frequent itemset at different nodes for
represents a transaction number or id and second part of assigned items based on weight.
which is labeled weight, indicates quantity purchased in that
transaction or cost or other components. This node has only A. Algorithm for constructing Weighted Tree
one pointer pointing to the next object having this particular Input: The database D
attribute. Output: Weighted Tree
for each attribute weight w in a transaction t ∈ D do
begin
Create a node labeled with weight w and add this

125
 node to the respective attribute node.
end
B. Removal of infrequent attributes of Weighted Tree
Input : w_min_sup = weighted minimum support
Output: Reduced Weighted Tree.
for each attribute in a Weighted tree do
begin
if sum(weights of all nodes) < w_min_sup then
remove that branch from the tree
end
For example, In the above case if we consider min_sup=3
then only attributes A and C are frequent in the database. The
attributes B and D are found to be infrequent.
If we consider w_min_sup = 10 then the attributes A, C
and D will be frequent in the database. The reduced
Weighted Tree of Table 1, is shown in Figure 3.
Figure 3. Reduced Weighted Tree of Table 1
C. Arrange the attribute list in an increasing order of
their weighted minimum support
In Figure 4, we see that attribute D ahs weight 10, C has 12
and A has 19. The attribute lists are sorted in increasing
order of their weights and is shown in Figure 4 and is called
Ordered Weighted Tree.
The ordered Weighted Tree is replicated among all
parallel nodes. By doing so a full set of transaction database
126
containing only frequent items would be available to each
processor to generate all globally frequent patterns with
minimum communication cost at parallel node level. This
replication is executed from a designated master node.
Figure 4. Ordered Weighted Tree
D. Parallel mining of frequent itemset at different nodes
for assigned items based on weight.
This approach relies on distributing frequent items among
the parallel nodes. After ordering the frequent items by their
support, starting from the least frequent, each processor
successively receives one item. The process is repeated until
all items are distributed. In other words, if we have m
processors, and n trees need to be built, assuming m< n then
processor1 builds the tree for the least frequent item.
Processor 2 builds tree for the next least frequent item, and
so on up to processor m. After that processor 1 takes item
m+1 and so on until all n items are distributed.
Each parallel node is responsible for generating all
frequent patterns related to the frequent items associated to
that node. To do so each node reads sub-transactions for each
frequent items directly from the Ordered Weighted Tree,
then builds independent and relatively small trees for each
frequent item in the transaction database called item tree.
Each parallel node mines separately each one of these item
trees as soon as they are built. The trees are discarded as
soon as mined. Finally, all frequent patterns generated at
each node are gathered into master node to produce the full
set of frequent patterns.
The each node of the item tree consists of item number
and its count and two pointers called child and sibling
pointers. The child pointer points to the following item and
sibling pointer points to the sibling node.
In our example, if we need to mine the Ordered Weight
Tree in Figure 4, using 2 processors machine with weighted
minimum support is equal to 10, the starting node for each
parallel node would be D, the first frequent item. Processor 1
finds all frequent patterns related to items D and A.
Processor 2 would generate all frequent patterns related to
item C.
The first an item tree is built for item D. In this tree for D,
all frequent items which are more frequent than D and share
transactions with D participate in building the tree. The tree
starts with the root node containing the item D. For each sub-
transaction containing the item D with other items that are
more frequent than D, a branch is formed starting from the
root node.
If more than one frequent items share the same prefix,
they are merged into one branch and their count fields are
incremented. Figure 5 illustrates trees for frequent items D,
C and A.
Figure 5. Item Trees for items D, C, A
The tree for any frequent item say f contains only nodes
labeled with items that are more frequent or as frequent as f
and share at least one transaction with f. Based on this
definition, if A has weight greater than B then tree for B is
larger than the tree for A. In other words, the higher the
support of a frequent item, the smaller its tree is.
Input: A Ordered Weighted tree, w_min_sup
Output: set of all frequent itemsets at node i.
for every assigned item I, construct item tree for I
containing transactions associated with I at node i.
Traverse item tree for I
for each maximal path from a root to a leaf in the tree for
an item I
begin
T= {elements from I to leaf with weight}
if sum of weights of elements of [T] >= w_min_sup
MI= {elements from I to leaf with weight}
end
127
Apply downward closure property[1] to get all frequent
itemsets from MI which contains all maximal frequent
itemsets. If w_min_sup =10 then applying above algorithm,
we see that at node 1, Tree for D does not contain any other
node except itself. Therefore it is ignored. Similarly, tree
corresponding to A contains only one node and is also
ignored. Therefore at node 1 only frequent itmestes are one
itemsets, {D} and {A}.
Similarly, at node 2, item tree corresponding to C is built.
The maximal path corresponding to this tree is {A, C}. The
weight of {AC} = 18 is greater than user defined weighted
minimum support. Hence it is frequent. Also all its subsets
are frequent by using downward closure property.
Hence Fw = { {A}, {C}, {D},{ A, C}}.
III. THEORETICAL ANALYSIS
A. Construction of Weighted Tree
In a transaction database D, if there G transactions, then G
transactions have to be read by the algorithm to construct
Weighted Tree. Therefore, this tree can be constructed in
O (G) steps.
If the average length of the transaction t is equal to m
weights, then there will Gm nodes in the tree. In the worst
case, if there are M weights and G transactions in D, then
there will be GM nodes in a Weighted Tree.
B. Reduction of Weighted Tree
If there are m attributes, then reduction of Weighted Tree is
in O (m).
If there are k infrequent items then Reduced Weighted Tree
will contain (m-k) attribute lists with at most G (m-k) nodes.
C. Arrange the attribute list in an increasing order of
their weighted minimum support
If there are n attributes in the reduced Weighted tree then
arranging them in an increasing order of the weight is in
O (n2).
D. Parallel mining of frequent itemset at different nodes
for assigned items based on weight.
If there are n frequent items then there will be n item trees.
Each processor will construct one tree at a time. If there are
m processors and n trees, m<n then this step is in O( n/m).
Merits
In our research we have implemented this algorithm and
found that this algorithm is better than Apriori and FP-tree
algorithm.
(i)This method is space as well as time efficient than Apriori
since it involves candidate generation method and requires
multiple scan over the database. Whereas, our method
requires only one scan of the database.
(ii) The algorithm FP-Tree requires 2 scans to discover all
frequent itemsets and mining patterns involve construction of
the condition based tree, which involves header table. Our
method uses item tree to mine maximal frequent itemsets and
does not involve header table.
IV. CONCLUSION
The Parallel algorithm for discovering frequent itemsets
based on weight is a new method and is found to efficient
when compared to Apriori and FP-tree. As such, we are still
working on it with the aim of extending the application of
this algorithm to various kinds of databases.
V. REFERENCES
[1] Arun K Pujari “Data mining Techniques”: Universities Press(Indis)
Private Limited. Hyderbad, 20003, India.
[2] J. Han and M. Kamber, “Data MiningConcepts and Techniuqes”: San
Franscisco, CA:. 2004, Morgan Kaufmann Publishers.
128
[3] Han, J., Pei, J., Yin, Y. “Mining Frequent Patterns without Candidate
Generation”, Proc. of ACM-SIGMOD International Conference
Management of Data. Dallas, TX, 2000, 1-12.
[4] Han J, Pei Jian, Runying Mao(2004) ” Mining Frequent Patterns
without Candidate Generation: A Frequent Pattern Tree Approach” ,
Data Mining and Knowledge Discovery, Kluwer Academic Publishers,
Netherland” 53-87.
[5] Agrawal Charu and Yu Philip,(1998) “ Mining Large Itemsets for
association rules. Bulletin of the IEEE Computer Society Technical
committee on Data Engineering, 21, No.1.
[6] Ananthanarayana V. S, Subramanian, D.K., Narasimha Murthy
M.(2000). “Scalable, distributed and dynamic mining of association
rules” In Proceedings of HIPC'00. Springer Verlag Berlin,Heidelberg,
559-566.
[7] O.R.Zaiane, M. El-Hajj, and P. Lu. “Fast parallel association rule
mining without candidacy generation” In Proc. of the IEEE 2001
International Conference on Data Mining, San Jos, CA, USA,
December 2001
 2009 International Conference on Computer Engineering and Technology

Optimized Design and Implementation of Three-Phase PLL Based on FPGA

Yuan Huimei, Sun Hao,Song Yu

College of Information Engineering, Capital Normal University, Beijing, 100048, China
yuanhmxxxy@263.net, sh_1983@sina.com

Abstract has analyzed the influence of three-phase PLL

An optimized method to design and implement digital
three-phase phase-locked loop (PLL) based on FPGA is
presented in this paper. The PLL fits in electric power system
as well as other fields. At first, principle and basic structure
of the PLL including phase discriminator, loop filter and
voltage controlled oscillator (VCO) are introduced, then
these modules are designed in VHDL language with blocking
design method. In order to save logic resource of FPGA, an
optimized method called chip area sharing is adopted. At the
same time, a new method which combines CORDIC
algorithm with look-up table algorithm is put forward in this
paper to generate sine function. This new method can both
increase calculating speed and guarantee accuracy of
results. At last, the design is implemented on Altera’s FPGA
chip EP1C12Q240C8. It is verified that the design can save
logic resource of FPGA largely, and it is also proved that the
PLL can track the variation of frequency and lock the basic
phase well.
1. Introduction
Flexible electric power transmission system which is
widely used in electricity system requires accurate and
real-time voltage phase information of the system, so
phase tracking is one of the most important
components in the system[1]. The frequency and phase
of voltage in the electric network are indispensable.
Analog or digital phase-locked loop (PLL) is widely
used in general instruments to achieve zero detection
[2]. However, when the rectifier notch appears in the
signal, the method mentioned above cannot do
anything about it. Discrete fourier transform (DFT) is
the most commonly used method to detecting
frequency and phase, because this way could suppress
harmonic wave well[3]. But using asynchronous
sampling, which means the width of sampling window
is not equal to integer times of the signal, would cause
measurement error because of spectrum leakage [4]. [5]
detecting the error of phases when interference, such as
asymmetry, harmonic and migration etc., exists in
voltage signal. Three-phase PLL would obtain better
performance, if system can separate positive and
negative voltage and feedback positive voltage[6]. This
algorithm usually is implemented using DSP
technology by software because of its complexity. But,
in this way, it will take a lot of CPU time, and the
performance is limited. To implement three-phase PLL
using field programmable gate array (FPGA) by
hardware is a new design scheme. This method is a
pure hardware approach to parallel process signals and
able to achieve high performance.
The research of principle, system components and
implementation algorithm based on FPGA is
represented in this paper at first. Then, the components
of this three-phase PLL, such as phase discriminator,
loop filter and voltage controlled oscillator (VCO),
etc., are designed using VHDL. In order to save logic
resources of FPGA, this design is improved by using
chip area sharing improvement scheme. A new
algorithm which is combined with CORDIC algorithm
and look-up table is proposed to generate sine function
aiming to increase computing speed and improve the
accuracy of results. At last, this improvement design is
inspected on ALTERA’s FPGA Cyclone
EP1C12Q240C8. The result shows that this
improvement scheme is able to increase the computing
speed, reduce the usage of logic resource and track the
variation of frequency and lock the base phase well.
2. Principle of three-phase PLL
PLL is composed by three main components: phase
discriminator (PD), loop filter (LF) and voltage

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DOI 10.1109/ICCET.2009.74
controlled oscillator (VCO), which is shown in ud = uαuuα −uβuuβ
Fig.1.Three-phase equilibrium voltage can be =Us KL sin(ω0t +θ1 −ω0t −θ2 ) (5)
described as follows formula (1).
=Us KL sin(θ1 −θ2 )
Assuming PLL starts to lock, we can know, phase
error (θ1 − θ 2 ) is zero or infinitesimal, so the value of
KL cos θ sine function in (5) may be equal to the phase error
proximately. Then, formula (5) could be rewrite as
ud = U s K L (θ1 − θ 2 ) = K d (θ1 − θ 2 ) (5)
1 Where, K d = U s K L .
s
In order to guarantee both the performance of LP
ω0 and stability of the dynamic system[7], proportional-
integral (PI) filter is used in this paper. As the result,
KL sin θ the transfer function of LP would be described as
U f (s) K
= Kp + l （7）
Fig.1 Block diagram of three-phase PLL U d (d ) s
⎛ ⎞ Where, K p is proportional gain and Kl is integral
⎜ sin(θ ) ⎟ gain.
⎜ ⎟
2π From the Fig.1 we can see that the input of VCO
uabc = U s ⎜ sin(θ − ) ⎟ (1)
⎜ 3 ⎟ u f (t ) is the output of adaptive low-pass filter. In order
⎜ ⎟
⎜⎜ sin(θ + 2π ) ⎟⎟
to get a transfer function of the phase of PLL, the VCO
⎝ 3 ⎠ module should generate an output of phase signal θ 2 .
The output of PLL, θ (t ) can be stated as follows:
Where, uabc = [ua ub uc ]
T
θ(t) = ∫ [ω0 +Δω(λ)]dλ =ω0t + ∫ Kvuf (λ)dλ
t t
could be transformed as （8）
−∞ −∞
T
uαβ = ⎡⎣uα uβ ⎤⎦ in αβ 0 coordinate through transform Then, using θ 2 (t ) = θ (t ) − ω0 t to replace the formula
matrix A, where (8), we can obtain
θ 2 (t ) = ∫ K v u f (λ )d λ
t

⎛ 1 1 ⎞ （9）
−∞
1 − ⎟
2⎜ 2 2 In this way, the VCO is able to be simply expressed as
A= ⎜ ⎟ (2)
3⎜ 3 3⎟ an integrator with a gain K v , and the transfer function
⎜0 − ⎟ of this integrator is
⎝ 2 2 ⎠
θ 2 (s) K v
Then, according to vαβ 0 = Avabc , we can get = （10）
U f (s) s

⎧ua = U s sin(ω0 t + θ1 )
⎨ (3)
⎩ub = U s cos(ω0 t + θ1 )
Where, θ1 (t ) = ω1 (t )t + ϕ1 (t ) − ω0 t , and ω0 which is
assumed as a constant here is center frequency of
VCO.
PD’s feedback input uuα and uu β are the cosine
and sine function of θ with the gain of K L , which are
shown as formula (4):
⎧⎪uuα = K L cos θ
⎨ (4)
⎪⎩uu β = K L sin θ
The output of PD is expressed as follows:
3. FPGA design of three-phase PLL
3.1. FPGA design of PD
From Fig.1, we can know, in the calculation of
three-phase PLL, three-phase voltage needs to be
converted to two-phase voltage through the matrix
operations at first. Then, this two-phase voltage should
multiply with the closed-loop feedback and do addition
u
operator to get the d-axis component d we need.
However, if we process following the design described
as Fig.1, we need do multiplication 10 times, addition
10 times and trigonometric calculation 4 times totally.

130
It will take up a lot of logic resource in that way. After is locked by PLL. The design of this PLL system is
analyzing the design described above, we find out if shown in Fig.2.
we use d, q coordinate transformation directly, there is

T
⎡⎣ud uq ⎤⎦ = Tdq uabc 4. Optimization of design
（11）
⎡ 2 2 ⎤ As we described above, to complete all of the
⎢cosθ cos(θ − 3 π ) cos(θ + 3 π )⎥
Where, Tdq = ⎢ ⎥ . computation would use nine multipliers, but it is a
⎢ sin θ sin(θ − 2 π ) sin(θ + 2 π ) ⎥ waste of logic resource of FPGA. After analyzing
⎢⎣ 3 3 ⎥⎦ the
Then, we could obtain cos θ

1 3 Kl
ud = ua cos θ + ub [− cos θ + sin θ ] ua
2 2 uaubuc
(12) ub
⎡ 1 3 ⎤ uc θ1
+ uc ⎢ − cos θ − sin θ ⎥ 2π
⎣ 2 2 ⎦ cos(θ −
3
)
θ

We only need to implement the formula (12), and Kv

the program only needs to do multiplication 5 times, θ2 1
2π s
addition 5 times and trigonometric calculation twice to 3 cos(θ + )
2 3
complete the calculation. In this way, the workload and
complexity of computation are greatly reduced, and a sin θ
Kl
lot of logic resource of FPGA is saved at the same
time. Fig.2 Block diagram of FPGA design of
three-phase PLL
3.2. FPGA design of LF and VCO
system, we find that the process has its time sequence,
so to resolve this problem, we use time division
The result of input voltage signal θ1 and θ 2 of the
multiplexing (TDM) technology to reuse one multiplier
system in (6) implements the unit negative feedback of
iteratively. Every multiplication will have a fixed slice
the system’s output. And the closed-loop feedback of
to complete computation. In this way, the whole
the system is
system only need one hardware multiplier and logic
resource of the FPGA is able to be saved greatly.
Kd K p ( z) z (c − z )
Φ( z ) = = Kc (13) Another important part of this system is sine, cosine
1 + Kd K p (z) z − az + b
2
function generator which is usually implemented by
Kd Kv K p Kd Kv K p c + 2 using the method of look-up table. However, look-up
Where, Kc = , a= ,
table would take up a lot memory resource of FPGA.
1 + Kd Kv K p Kd Kv K p + 1
One of the solutions is CORDIC algorithm [8].
1 K
b= , c = 1 − l T and T is sampling CORDIC works by rotating the coordinate system
1 + Kd Kv K p Kp
through constant angles until the angle reduces to zero.
frequency. The angle offsets are selected such that the operations
Analyzing the formula (13) can shows when on X and Y are only shifts and adds. CORDIC would
T > 2 K p / Kl , this PLL system is unstable because the not take up any memory of FPGA, but it also have flaw
opened-loop zero point is out of the unit circle. which is the accuracy of CORDIC is limited by its
Nevertheless, K p , Kl and T decide the dynamic own. A new scheme which is combined with CORDIC
performance of the system. and look-up table is proposed in this paper in order to
At the end of the system, a sine and cosine function improve the CORDIC or look-up table individual.
generator is designed in order to supply the feedback Because of the symmetrical characteristic of sine and
for this system, according to phase information which cosine function, we only need to build up a table

131
including the information of a quarter of one period to period of fundamental wave, if the input has steady
minimize the table [9]; also it can improve the phase. The results show that the utilization of resource
accuracy of CORDIC. Table 1 lists the usage of logic on FPGA is greatly reduced after optimization. And
resource on ALTERA’s FPGA Cyclone the processing speed also meets the requirement of the
EP1C12Q240C8 in the way of look-up table, CORDIC system.
and combined way.
The usage of logic resource in this system is greatly
reduced by improvement scheme mentioned above.
Fig.3 shows the design of three-phase PLL after
optimization. In Fig.3, the number 1, 2, 3 …9 means
the slice taken up by calculation.

Table1 Usage of FPGA’s logic resource

Memory（
Method LEs M4K
ROM）（bit） Fig.4 Hardware simulation result of this system
Look-up table 242 180224 4
CORDIC 1335 0 0
Combined 675 9216 3
Table2 Usage of resources on EP1C12Q240C8 before
method and after optimization of three-phase PLL
Rate of
Resource Usage of resource
ua ub sin θ cos θ uc EP1C12Q240C8
on FPGA resource utilization/
%
LES (Look-up table) 12060 2546 21
1
cosθ Kv Kl Kl Memory/Bit （ Look-up 239616 13824 6
2π 2π s
cos(θ − ) ua cos θ + ub cos(θ − ) table）
3 3
2π LES（CORDIC） 12060 1210 10
+uc (θ + )
3
3 Memory/Bit （ 239616 0 0
2
θ1 − θ2
2π CORDIC）
cos(θ + )
3 LES （ Combined 12060 983 8
θ method）
Memory /Bit （ 239616 9216 4
Fig.3: Block diagram of FPGA design of Combined method）
three-phase PLL after optimization

5. Results of experiment
We use VHDL to implement this design and verify
the design on ALTERA’s FPGA Cyclone
EP1C12Q240C8. Here, we set K v = 128 , K p = 0.5 and
sampling frequency T = 12.8KHz . Table 2 lists the
usage of resource on Cyclone EP1C12Q240C8 before
and after optimization of this three-phase PLL. Fig.4
shows the result of hardware simulation.
In Fig.4, clk is the system clock, here we set it as
25KHz, u1, u2, u3 represents the three-phase voltage;
and the dataout is the output of this three-phase PLL.
In the picture, we can see that this three-phase PLL
could stably output phase information on the third
6. Conclusion
This paper has proposed an optimized method to
design and implement digital three-phase PLL based
on FPGA. Basic modules of the PLL including PD, LF
and VCO are designed in VHDL language as IP cores;
and an optimized method called chip area sharing is
adopted in order to save logic resource on the chip. A
new method which combines CORDIC algorithm with
look-up table algorithm has been put forward to
generate sine function, and results show this new
scheme is able to both increase calculating speed and
guarantee accuracy of results. It is also proved that the
PLL can track the variation of frequency and lock the
basic phase well.

132
Acknowledgment
The authors would like to thank Beijing Science &
Technology Government for financially supporting this
work by project of Beijing Technology New Star (No.
2006B58).
References
[1] Sang-Joon Lee, Jun-Koo Kang, Seung-Ki Sul.A New
Phase Detecting Method for Power Conversion Systems
Consid-ering Distorted Conditions In Power System,
Industry Applications Conference Conf. Record of the
IEEE’99[C]. 1999: 2167- 2172.
[2] G. C. Hsieh, J. C. Hung. Phase-Locked Loop techniques-
a survey, IEEE Transactions on Industry Electronics,1996,
6(43): 609-615.
[3] IEEE Working Group Report. Synchronized Sampling
and Pastor Measurements for Relaying and Control, IEEE
Transactions on Power Delivery, 1994, 19(1): 339-442.
133
[4] hadke A.G., Thorp J. S, Adamiak M.G. A New
Measurement Technique for Tracking Voltage Phasors,
Local System Frequency, and Rate of Change of Frequency,
IEEE Transactions on Industry Electronics,1996, 43(6): 609-
615.
[5] Kyo Chung.A Phase Tracking System for Three Phase
Utility Interface Inverters, IEEE Trans. on Power Electronic
2000. 2000: 431-438.
[6] J. Lee, J. K. Kang, S. K. Sul. A New Phase Detecting
Method for Power Conversion Systems Considering
Distorted Conditions in Power System, Thirty-Fourth IAS
Annual Meeting IEEE Conf. Record ′99.1999: 2167-2172.
[7] YANG Shi-Zhong, Li Xiao-Chun. Foundation of Phase
Lock Technology, Beijing: People Post-office Press. 1978, 8:
100-102.
[8] Uwe Meyer-Baese. Digital Signal Processing with Field
Programmable Gate Arrays, Second Edition, Springer Press
Ltd, 2004, (2) : 79-82.
[9] Wu Po, GUO Yu-Hua, WEN Liang-Yu. Improvement
and Implementation of One Phase Power Phase-Locked Loop
based on FPGA, Power Electronics, 2007, (4): 80-82.
 2009 International Conference on Computer Engineering and Technology
Research on the Data Storage and Access Model in Distributed Environment1
Wuling Ren
College of Computer and Information
Engineering, Zhejiang Gongshang
University, Hangzhou, Zhejiang, 310018,
China
rwl@zjgsu.edu.cn
Abstract
In order to improve the data access efficiency and
enhance the security in IT systems, a layered access
control model with classified distributed storage in
distributed environment is presented[1,2]. In the model,
information is stored in numbers of physical
distributed storage nodes. The storage architecture is
logically organized into four classes, which includes
active data storage, static data storage, backup data
storage and data warehouse. Data access efficiency
is increased because most applications will
concentrate on active data storage or static data
storage where the most recently and frequently used
data are stored. Data security access is controlled
through a 3-layered access control model based on
RBAC model[3,4]. The three layers include operation
layer, data logic security shield layer and application
layer. Operation layer and application layer are to
control the authorized access for system managers
and application end users respectively. The data
logic security shield layer is to shield the difference
among the database safety control organizations. It
is appreciated after the model has been applied to
Zhejiang Fuchunjiang Limited Company.
1. Introduction
It is particularly important while the information
security in distributed computing system. The
security system guarantees data can and only can be
accessed by the authorized users, whatever the
request comes from domestic or external. The
encrypted centralism authorization provides trustable
login, it keeps privacy and consistency of data in
system. Backup and restore quickly and reliably is a
mandatory requirement of digital system. Data is
well protected by good security design and
implementation. The security system is the critical
part of data centre in the distributed computing
environment.
1
Foundation item: Project supported by the National Science & Technology Pillar Program, China(No.2006BAF01A22), and the Science &
Technology Research Program of Zhejiang Province, China (No.2006C11239)
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.89
Pan Zhou
College of Computer and Information
Engineering, Zhejiang Gongshang
University, Hangzhou, Zhejiang, 310018,
China
zhoupan@pop.zjgsu.edu.cn
In this paper, we study the architecture of data
center according to the characteristic of distributed
computing environment. A security access control
model is presented based on the above study.
2. The architecture of data storage in
distributed computing environment
The design of data central of distributed
computing environment must consider the security of
information. The data central collects the massive
primary data and all kinds of regeneration
summarized information in each subsystem, therefore
the design of data central must realize security
control of each kind of database information, prevent
the invasion of illegal user, and maintain the integrity
and uniformity in the procedure of data gathering,
storage, transmission and processing in each
subsystem of distributed computing environment. [6]
According to information characteristics, we
discuss the data storage architecture structure of data
central system structure. The information may be
roughly divided into four kinds in the distributed
computing environment.
The first kind of information is the foundation data
in the distributed computing environment, including
system parameter, product information, the material
information, customer information, supplier
information, staff code, department code in each
subsystem and so on. These information is shared in
various subsystems, and has high frequency to use. In
the distributional data management environment, we
must adopt effective method to maintain the integrity
and uniformity of visiting to these data in various
subsystems.
The second kind of information is the model
meta-data in each subsystem[2], including t the model
of process control standards and its executing log.
This is the main information in distributed computing
executing environment. The information is so much
134
that we must control their depositing cycle. In the
regular circumstances, we only record and retain the
information which exceptionally occurs or not yet
completely processing in the system. That
information is the basis of dynamic adjusting the
system procedure, modeling the executing
parameters and the management control standard.
The third kind of information is the process
information. The primary data gathers from each
related subsystem. One of the tasks of distributed
computing system is to process this information
effectively.
The fourth kind of information passes through the
distributed computing environment processing,
compiles processing the history data material, which
we call the data warehouse.
The data central data storage system is composed
by five parts, as figure 1 shows. One of the parts is
the current data storage. It deposits the information
which is used presently and in high frequency in the
system, such as the system foundation information,
the primary data and so on. Another part is the
historical data storage. It deposits the information
which is used presently and in low frequency in the
system. These two parts constitute the active data
storage. Another part is the static data storage. It
deposits information which id used in the lowest
frequency in the system, such as some original
documents processing in several years ago. The last
part is the backup data storage. We must set up the
backup system for the above data storage in order to
prevent the system from accident collapses and lose
of data, and guarantee the security of information.
Figure 1. The architecture of data storage in
distributed computing environment
3. The design of database security
The data security including two aspects, one is the
data protection (or we call anti-damage nature),
another aspect is the data security. The former avoids
the unintentional destruction of information (for
135
example: the operation mistakes, such as incautious
deletion, the computer hardware breakdown, hard
disk expiration and so on). The latter refers to the
authority of data, namely who can use the
information and how to use it.
3.1 Information protection
In the distributed computing environment, the
information of active data storage and static data
storage is the protection object[7]. The information in
the reserve data storage is their backup, in order to
prevent the system from accident collapses, and
guarantee the security of information. The active data
storage is organized by the database form, and the
storage medium is the hard disk; Static data storage
and reserve data storage is organized by file form,
and the medium is the hard disk, the magnetic tape or
CD-ROM.
The method of data transfer includes various date
storages as followings:
Data transfer：Information transfers between the
active data storage and historical data storage. It
means information transferred among the database or
tables. We can use programs to realize the procedure.
Data export／load：Information transfers between
the static data storage and historical data storage. We
can use the database tool EXPORT/IMPORT which
the database management system provides to realize.
They will dump the data to text file. If the database
management system has not provided this kind of
tool, then it need program to realize this function
Data backup／recovery：
(1) Information transfers between the active data
storage and reserve data storage . It can use the hard
disk or the magnetic tape as the reserve medium to
backup the information by the copy/restore tool
which provided by the database management system
or the operation system. Thus the backup information
is stored in the hard disk as files.
(2) Information transfers between the static data
storage and reserve data storage. It can use the file
copy tool to realize the procedure.
3.2 Information Access Control
In the distributed computing environment system,
the complete safety control model is as figure 2
shows. Except administrator, other operators can
only be able to visit the distributed computing
environmental information through the application
subsystem.
In figure 2, system configure file, parameter file
refer to the operation system-level file which need
security control and should be controlled by the
application system.
File system safety and security control is to carry
out the security control of system configure file,
parameter file with the safety control mechanism of
operating system, with the encryption method which
defined by user.
The data central management system provides the
data central safety control level. Primary cognizance
data central information authorized operation has the
same effect as well as data central manager's security
control.
The main function of the data logic security shield
level is to shield the difference among the database
safety control organization, the file system safety
control and the security mechanism, so as to provide
the consistent support way for the application system
security control mechanism.
The security control layer is responsible for the
authority of the application system. It takes the role
of privilege grant. It decides functions the operator
can use, which data the operator can access or modify,
how long the privilege is available. Application
operating is granted by the security control layer. A
triple is used to describe the three objects: (operator,
operation, operated object). The security control
layer determines its granting by matching the certain
triple in triple database. Application system defines
other 2-truple and triples, such as (operator, manager
of operator), to extend flexibility, which delegates
privileges to other operator or group. Effective time
can be also added in the triples to solve the
temporary authorized problem.
Figure 2. The 3-layered access control model in
distributed computing environment
136
4 Summary
This paper discussed the architecture structure of
data centre in the distributed computing environment,
and presented a layered security access control model.
In classified distributed storage model, the data
distributed in each physical storage node is organized
into four components which includes active data
storage, static data storage, backup data storage and
data warehouse. The layered access control mode is
presented based on RBAC model. Security control is
distributed in three layers, namely management layer,
operation layer and application layer. The system has
been applied to Zhejiang Fuchunjiang limited
company. It was well evaluated.
5.References
[1] K. Hammer. Web Services and Enterprise Integration,
EAI Journal, 11, 2001.
[2] Bruce Lownsbery, Helen Newton.The Key to Enduring
Access: Multi-organizational Collaboration on the
Development of Metadata for Use in Archiving Nuclear
Weapons Data.
[3] David F. Ferraiolo, Role-Based Access Control, Artech
House, 2003.
[4] J. Kwon, C.-J. Moon, Visual modeling and formal
specification of RBAC constraints using semantic web
technology, Knowl. Based Syst.(2008),
doi:10.1016/j.knosys.2008.02.007 I.
[5] Soomi Yang, An Efficient Access Control Model for
Highly Distributed Computing Environment.
Distributed Computing - IWDC 2005, pub: Springer
Berlin/Heidelberg, doi:10.1007/11603771.
[6] Yu Hwanjo, Yang Jiong, et al., Classifying Large Data
Sets Using SVMs with Hierarchical Clusters,
SIGKDD’03, Washington, DC, USA. 2003, 8.
[7] Shigeki YAMADA,Access Control for Security and
Privacy in Ubiquitous Computing Environments.
IEICE Transactions on Communications 2005
E88-B(3):846-856; doi:10.1093/ietcom/e88-b.3.846.
[8] Ninghui Li, Mahesh V. Tripunitara, Security analysis in
role-based access control. ACM Transactions on
Information and System Security (TISSEC), v.4 n.9,
Nov 2006.
 2009 International Conference on Computer Engineering and Technology
An effective classification model for cancer diagnosis using micro array
Gene expression data
Dr.V. Saravanan,
Department of Computer Applications,
Karunya School of Science and Humanities,
Karunya University,Coimbatore,India
tvsaran@hotmail.com
Abstract-Data mining algorithms are commonly
used for cancer classification. Prediction models
were widely used to classify cancer cells in human
body. This paper focuses on finding small number
of genes that can best predict the type of cancer.
From the samples taken from several groups of
individuals with known classes, the group to which
a new individual belongs to is determined
accurately. The paper uses a classical statistical
technique for gene ranking and SVM classifier for
gene selection and classification. The methodology
was applied on two publicly available cancer
databases. SVM one-against- all and one-against-
one method were used with two different kernel
functions and their performances are compared
and promising results were achieved.
Keywords: Classification; SVM one-against- all;
SVM one-against-one; Gausssian; RBF.
I. INTRODUCTION
Micro array technology has made the modern
biological research by permitting the
simultaneous study of genes comprising a large
part of the genome [1]. In response to the rapid
development of DNA Micro array technology,
classification methods and gene selection
techniques are been computed for better use of
classification algorithm in microarray gene
expression data [2][3].
The goal of classification is to build a set of
models that are able to correctly predict the class
of different objects. The input to such models is
a set of objects (i.e., training data), the classes,
which these objects belong to (i.e., dependent
variables), and a set of variables describing
different characteristics of the objects (i.e.,
independent variables). Once such a predictive
model is built, it can be used to predict the class
of the objects for which class information is not
known. The key advantage of supervised
learning methods over unsupervised methods
like clustering is that by having an explicit
knowledge of the classes the different objects
belong to, these algorithms can perform an
effective feature selection if that leads to better
prediction accuracy.
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.38
R.Mallika,
Department of Computer Science
Sri Ramakrishna College of Arts and Science for
Women, Coimbatore, India
mallikapanneer@hotmail.com
In classification analyses of microarray data,
gene selection is one of the critical aspects
[4[5][6][7]. Efficient gene selection can
drastically ease computational burden of the
subsequent classification task, and can yield a
much smaller and more compact gene set
without the loss of classification accuracy [8][9].
In microarray classification analyses, the main
objective of gene selection is to search for the
genes, which keep the maximum amount of
information about the class and minimize the
classification error [10].
With the help of gene expression obtained from
microarray technology, heterogeneous cancers
can be classified into appropriate subtypes and
the challenge of effectiveness in cancer
prediction lies when there are high dimensional.
Supervised machine learning can be used for
cancer prediction, which uses a part of the
dataset as training set and the uses the trained
classifier to predict the samples in the rest of the
data set to evaluate the effectiveness of the
classifier. [11]. With DNA microarray data,
selecting a compact subset of discriminative
genes from thousands of genes is a critical step
for accurate classification. Selection of important
genes using statistical technique was carried out
in various papers such as Fisher Criterion,
Signal-to-Noise, traditional t-test, and Mann-
Whitney rank sum statistic [12], chi-squared test,
Euclidean distance [13] and the some of the
classification algorithms used were SVMs, k-nn
[14], Genetic algorithms (GA) [16] Naïve bayes
(NB)[15]. This paper used the two publicly
available cancer dataset Lymphoma and Liver.
In 2003, Tibshirani [17] successfully classified
the lymphoma data set with only 48 genes by
using a statistical method called nearest shrunken
centroids and used 43 genes for SRBCT data.
Lipo Wang [11] 2007 proposed an algorithm in
finding out minimum number of gene up to 3
genes with best classification accuracy using C-
SVM and FNN.
Tzu-Tsung [27], 2008 proposed a classification
method to classify the causality of a disease is of
two stages. Gene Selection mechanism with
137
individual or subset gene ranking as the first
stage and classification tool with or without
dimensionality reduction as the second stage.
This paper proposes an efficient methodology
using statistical model for individual gene
ranking and data mining models for finding
minimum number of gene rather than thousands
of genes, which can be used to give good
classification accuracy. Furthermore, the paper
compares the SVM one-against- all, SVM one-
against-one classifiers with kernel functions
gausssian and RBF (heavy tailed RBF) using two
databases for Lymphoma cancer and Liver
cancer. Their performances are compared with
their respective training time and accuracy.
II METHODOLOGY
A. Gene Ranking -ANOVA p-values
ANOVA is a technique, which is frequently
used in the analysis of microarray data, e.g. to
assess the significance of treatment effects, and
to select interesting genes based on P-values.
[18]. The ANOVA test is known to be robust and
assumes that all sample populations are normally
distributed with equal variance and all
observations are mutually independent.
The approach chosen in this paper is the one-way
ANOVA that performs an analysis on comparing
two or more groups (classes) for each gene and
returns a single p-value that is significant if one
or more groups are different from others. The
most significantly varying genes have the
smallest p-values.Of all the information
presented in the ANOVA table, if the p value for
the F- ratio is less than the critical value ( ),
then the effect is said to be significant. In this
paper the α value is set at .05, any value less
than this will result in significant effects, while
any value greater than this value will result in
non-significant effects. The very small p-value
indicates that differences between the column
means are highly significant.
The probability of the F-value arising from two
identical distributions gives us a measure of the
significance of the between-sample variation as
compared to the within-sample variation. Small
p-values indicate a low probability of the
between-sample variation being due to sampling
of the within-sample distribution, small p-values
indicate interesting genes. The paper uses the p-
values to rank the important genes with small
values and the sorted numbers of genes are used
for further processing
138
B. Gene Selection-SVM one-against-all and one-
against-one
SVMs are the most modern method applied to
classify gene expression data, which works by
separating space into two regions by a straight
line or hyper plane in higher dimensions. SVMs
were formulated for binary classification (2
classes) but cannot naturally extend to more than
two classes. SVMs are able to find the optimal
hyper plane that minimizes the boundaries
between patterns [19]. SVMs are power tools
used widely to classify gene expression data
[21][22]. How to effectively extend SVM for a
multi-class classification is still an ongoing
research issue [23]. This paper gives effective
methodology to classify a multi-class problem
.To extend SVM for multi-class classification,
SVMs were designed with SVM one-against-
one, SVM one-against-all. This paper efficiently
uses the varieties of SVM such as one-against-
all method (SVM-OVA) and SVM one-against-
one (SVM-OAO) method with heavy trailed
RBF and Gaussian kernel function.
The SVM-OAA constructs ‘n’ binary SVM
classifier with the ith class separating from all
other classes. Each binary SVM classifier creates
a decision boundary, which can separate the
group it represents from the remaining groups
For training data t = (X1, Y1), (X2, Y2)…(Xt,
Yt); Xn ∈ Rn and n=1…t, Yn =1.. K is the class
labels corresponding to Xn. The nth SVM solves
Min ½ (wi)Twi + C Σ tj=1 ξji
i i
w biξ
(wi)Tφ(xj) +bi ≥ 1 - ξji , if yj =i,
(wi)Tφ(xj) +bi ≤ -1+ ξji, if yj ≠ i,
and ξji ≥ 0.j =1…t, (1)
Where each data xi is mapped to the feature
space by function φ and c the penalty parameter.
The Radial basis function (RBF) is the most
popular choice of kernel functions used in
Support Vector Machines, which can be
represented by
K (Xi, Xj) ≡ e-γ | Xi – Xj |2 (2)
Where Xi is the support vector of the ith class and
Xj is the support vector for the new higher
dimensional space and γ is the tuning parameter.
RBF kernel can be able to give the same decision
as that of RBF network. [23].
The SVM-OAO method was first used by [28]. It
constructs n (n-1)/2 classifiers in which each one
is trained on the data for 2 classes. The
classification problem for the training data with
ith and jth class is shown as
 The average 5-fold accuracy for each run was
Min ½ (wij)Twij + C Σ tj=1 ξjij calculated and average error rate in training were
w b ξj
ij ij ij calculated. Table .1 shows the average Training
(wij)Tφ(xt) +bij ≥ 1 - ξjij , if yt =i, accuracy for training all possible gene pairs. In
(wij)Tφ(xt) +bij ≤ -1+ ξjij, if yt ≠ j, all the cases SVM one-against-all was superior
to all other methods were well proved.
ξjij ≥ 0. (3) The method used the gene subsets that achieved
100% CV accuracy for the training samples and
III RESULTS AND DISCUSSION retrained the classifier. Then the classifier was
used to predict the samples in the testing dataset.
The proposed methodology was applied to the
publicly available cancer database namely Liver Fig.1 plots the gene pairs (4,6) that shows a good
and Lymphoma cancer. This section reports the separation of 3 classes-DLBCL, FL, CLL with
experimental results of all the datasets exhibiting clear boundaries, which can be able to predict for
the SVM varieties such as one-against-all and a doctor for the 3 subtypes of lymphoma.
one-against-one with different kernel functions Furthermore Table.2 shows the best prediction
RBF and Gaussian. accuracy for the gene pair (4,6) was achieved
using SVM one-against-all method with RBF
A. Lymphoma dataset kernel function (OAA-RBF) from the top 20
genes. It should be noted that all the genes after
The methodology was applied to the lymphoma ranking were given numbers in ascending order.
dataset with 4026 genes and 62 samples with 3
class namely DLBCL, CLL and FL, the subtypes
pf Lymphoma cancer.
The dataset was with few missing data. The K-
Nearest neighbor algorithm as used by [18] with
k=3 was used and the missing data were filled.
Half of the samples were picked randomly for
training and all the samples for testing. For the
training dataset with 4026 X 31 dimensions
ANOVA p-value was calculated for each gene
and the top ranked genes were selected .All
possible combinations of the top n genes were Figure.1 gene expression level of gene pairs showing good
generated. For n number of top genes all possible separation of different classes for Lymphoma dataset
combinations are n (n+1)/ 2!Using these
combinations, the databases were trained using TABLE.2. COMPARATIVE RESULTS OF TESTING
ACCURACY OF ALL ALGORITHMS- LYMPHOMA
SVM one-against-all and one-against-one. DATA
The performance of the classifier was validated
using cross validation (CV) technique with 5- Top OAA- OAO-
folds. For 5 folds, the samples in the training gene OAA-RBF GAU OAO-RBF GAU
dataset was randomly divided into 5 equal parts,
10 96.77 91.94 95.16 91.94
classification was performed for 5 runs, using the
20 98.39 95.16 96.77 95.16
4 parts as training and the other as testing.
30 98.39 96.77 98.39 96.77
TABLE.1 AVERAGE 5 FOLD CROSS- 50 96.77 96.77 98.39 98.39
VALIDATIONACCURACY OF ALL ALGORITHMS- 100 96.77 98.39 93.55 98.39
LYMPHOMA DATA
Top OAA OAA OAO OAO In comparison with previous works, proposed
gene RBF GAU RBF GAU method proves well for the number genes needed
10 91.97% 91.54% 91.43% 91.04% to achieve best accuracy. Comparisons of results
20 91.32% 88.98% 91.02% 88.62% with previous work are shown in Table.3
30 92.04% 88.50% 91.95% 88.70%
50 91.99% 88.18% 91.90% 88.71%
100 90.33% 86.49% 90.32% 87.14%
Each time the classifier was trained, a different
test set was used, so that over 5 runs of the
classifier, all the samples were used as test set.

139
TABLE .3 RESULT COMPARISON- LYMPHOMA DATA

Method Accuracy No of genes

Proposed method 98.39% 2
Extreme Learning
machine (24) 97.33% 10
Bayes with 93% 30
Local SVM (26)
SVM-KNN (25) 96% 50

Figure.2 Plot showing best separation for liver dataset that

B.Liver Dataset
Liver dataset (http://genome-
www.stanford.edu/hcc) has 2 classes HCC and
non-tumor liver for 1648 genes for 156
observations in which 82 are from HCC and 74
from non-tumor livers. Unlike the Lymphoma
dataset, which were with the prediction of
subtypes in Lymphoma cancer, the Liver dataset
were from the cancerous and non-cancerous
tissue. The problem is to predict whether the
samples are from cancerous or non-cancerous
tissue. The methodology adopted for lymphoma
achieved the best test accuracy for Liver data
Fig.3 show the results for each SVM variety. The
figure shows the number of gene pairs that
achieved 100% CV accuracy using the top 100
genes for Liver and Lymphoma dataset. As
observed in the figure, SVM one-against-all with
RBF kernel function has higher or nearing 100%
training accuracy gene pairs for the liver and
lymphoma dataset.

top 100 genes

dataset was applied. Randomly half of the 400
observation was selected for training and all 300
samples for testing. 200 OAA-RBF
100
As with that of Lymphoma dataset, it was well 0
OAA-GAU
proven that SVM one-against-all with RBF OAO-RBF

Liver
Lymphoma
kernel can achieve better prediction accuracy OAO-GAU
compared to all other varieties for liver dataset as
well. This is depicted well in Table.4
.
TABLE.4 COMPARATIVE RESULTS OF TESTING Figure.3 Maximum number of gene pair that achieved 100%
ACCURACY OF ALL ALGORITHMS FOR LIVER DATA
CV Accuracy for Lymphoma and Liver dataset.

Top OAA- OAA- OAO- OAO- IV. CONCLUSION

gene RBF GAU RBF GAU
10 94.87 94.87 94.87 94.23 In this paper an efficient classification method
20 97.44 96.15 94.87 96.79 for cancer diagnosis problem using microarray
30 95.51 96.33 96.69 95.15 data was presented. The paper used a classical
50 94.87 94.87 94.87 94.23 statistical technique for gene ranking. From the
results of the two datasets, it was found that
100 92.31 92.95 96.79 91.67
SVM one-against-all classifier with RBF kernel
function (SVM OAA-RBF) achieves better
From the Plot in Fig.2 the gene pairs (13,23) for
classification results. On application of SVM
liver dataset, a doctor can able to diagnose that a
one-against-all and one-against-one to the
patient has HCC if and only if the expression
Lymphoma and Liver datasets it was found that
level is less than 0.75 and greater than –0.3
SVM one-against-all with RBF kernel function
otherwise the tissue is of a patient without tumor.
outperforms with the other. Though the
computational time for training was longer,
SVM one-against-all could achieve the best
prediction accuracy. The results were promising
when compared with previous works as well.
The future work shall extend with different
classifier and gene ranking methods

140
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 2009 International Conference on Computer Engineering and Technology

Study and Experiment of Blast Furnace Measurement and Control System Based on
Virtual Instrument

Li Shufen Liu Zhihua

Beijing Union University Beijing university of chemical and technology
Beijing, China Beijing, China
zdhtshufen@buu.com.cn Liuzhihua714@sina.com

Abstract—This article mainly introduces a blast furnace This system has the following functions:
measurement and control system, which based on the virtual system parameters setting function
instrument development platform LabView. The hardware Real-time measurement and display function
takes the ADAM4000 series data acquisition module of Real-time control function
Advantech as the core. The system can replace the massive Sound and light alarm function
display instruments, finish monitoring the furnace conditions Data storage, browsing and report printing functions.
in the production process, and realize to manage the historical
data. B. Hardware design
The measurement and control system is mainly
Keywords-LabView; Data Acquisition Module; Instrument; composed of sensors, measuring transmitter, acquisition
the Furnace Condition
card, computer, printer, the executive drives and actuator.
The application of sensors should consider two factors: the
I. INTRODUCTION site environment and the measurement request. The
Virtual Instrument is a new technology which is ADAM4000 series of data acquisition module developed by
developed in the 1990s. It is the integration of computer and Advantech can accept the standard signal from the sensor so
bus technology, microelectronic, measurement and control as to omit some of the transmitter. High-performance
technology. LabView is a virtual instrument development computer is used in the host computer. The actuator is the
platform for many fields, which is developed by the drive circuit used to drive the regulating valve. Diagram of
American NI Corporation. It has a simple graphical the basic system components is shown in figure 1.
programming environment and powerful hardware-driven
features, and more a friendly visual simulation interface. It furnace
has been widely used in various fields, such as: fault
diagnosis, detection, process control, equipment design and
so on[1]. sensor annunciator executor
This system takes the blast furnace system of the No.1
Iron making of Tang gang as the design model. As the site
signal is complex and multi-species, the amount of signal
acquisition is large. Even worse, the operating mode has
ADAM4018 ADAM4050 ADAM4021
manual control and PLC control, the data records have
computer records and manual records, of which the data
amount is also great. All these make it difficult to measure
and control the condition of blast furnace. The ADAM4000
series signal acquisition module is characterized as the ADAM4520
acquisition of many kinds of signals, band rate-setting, work
stability and RS-485 communication function. So it is very
suitable for the measurement objects.
host
II. SYSTEM FUNCTIONS AND HARDWARE DESIGN
Figure 1. Diagram of the basic system
A. System Functions
In this system, the serial COM2 was used and the band
In the production process of blast furnace, it is necessary
rate system was set as 9600b/s. Since hot-air and the furnace
to measure all the signals of each furnace point, such as the
bottom temperature is high, we can use the S-type
temperature, the pressure and other physical quantities, to
temperature sensor. Its temperature ranges from 500 to
ensure the quality and safety of the production. In addition, it
1750 and it can convert signal below +22mV. The
also needs to show, record and storage the measuring
remaining temperature sensors adopted K-type, the
information.

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 142

DOI 10.1109/ICCET.2009.242
temperature ranges from 0 to 1000 and it converts signals
below +50mV. Through the compensation lead, temperature
sensor directly connected to the analog input module 4018.
Output of the module was automatically selected,4021
analog output module set output range 0~5V to control
regulating valve, while 4050 digital input and output module
set the working methods as “40”.
III. SOFTWARE DESIGN
A. software function
1. display the parameters of furnace points: the workers
can know all points’ pressure, temperature, flow rate and the
furnace condition through the host computer screen.
2. set the system alarm limit : in order to ensure high-
quality products and safety production, we can set the quality
assurance and security assurance of the relevant parameters
,respectively.
3. flashing alarm and manual operation: the alarm will be
operated if any of the parameters of measurement points
exceed the set limit, and then notify the workers what the
reason is. Meanwhile, if necessary, we can switch to manual
operation to directly adjust output signal for emergency.
4. data management: the operators need to register in the
display interface before they began to work, and procedure
will record the furnace conditions every 10 seconds.
Everyday these data will form a daily document. In each of
the data records, it contains both the workers’ number and
their working time, and in the event of fault or when it
alarms the current data can be stored and printed. These
measures can make it convenient to query, analysis and
investigate the production data.
B. Level of Software Function
According to the practical production, the level of
software modules is based on the problems which may
happen on the production process and the requirement for
production management. So the software mainly has the
following functions:
1. signal acquisition and display, the site temperature,
pressure, flow, location and other signals can be shown.
2.the equipment control, after filtering and giving PID
operations to the acquisition signals, we can output the
control signals, then control the air valve opening, and then
to adjust the parameters such as temperature and pressure to
achieve the objective of adjusting furnace condition.
3.single-function, including setting parameters, human-
computer interaction, data management and other functions.
IV. MAJOR TECHNICAL PROBLEMS AND SOLUTIONS
It is a key process to operate and debug software for the
system. In the practical production, if directly applying the
system, there will inevitably cause great loss. So it must
have a quite accurate simulation system to complete the
debugging system before running in the field.
Before operating the system, we must finish the hardware
setting as the requirement and complete the hardware
143
detection because this will affect the stability of the system.
For example, if there is an error on the hardware module
address, the corresponding parameter will have an error too,
and this will cause serious consequence to the production.
So it is very important to correctly set up the hardware and
software[2].
I have also met some problems in the experimental
debugging process, the followings are some solutions:
1) 4250 operates normally and hardware connects
correctly, but the computer detects no module. The reasons
are:
a The module address is duplicated.
b Serial band sate is not consistent with the module
band rate.
2) Module operates normally, but the collection data
shows abnormal, even with special symbols. The reasons
are:
a Error on interception of character or operation of
the software program.
b The signal source for the data acquisition module
has hardware error.
c No power supply for the signal source.
3) ADAM drivers can find the modules, but LabView
procedures can’t detect any modules[3], the reasons are:
a The ADAM driver is not closed, it has been
occupying the serial.
b LabView procedure has errors.
4) System running very slow. the reasons are:
a Data acquisition delayed too long
b Interruption in data collection process
c The data acquisition module which needs to scan
responses disordered.
d Enter wrong direction.
e The impact of other procedures.
V. DATA PROCESSING
Through displaying each point’s acquisition data, we can
use the key parameters to complete the control ,and set the
representative parameters(such as: there are four signals T1,
T2, T3, T4 of the furnace top temperature, take T1 as
representative parameter ) as the alarm limits.
Software system will read the key parameters, compare
with the settings and execute PID calculations then output
the adjusting signals. Storage signals have also taken the
ways of representative parameters, which can save a lot of
space to the data files. And it can also display the scattered
data in a centralized way in the function of data analyze.
VI. CONCLUSION
Based on the virtual instrument development platform,
the measurement and control equipment can significantly
improve the testing efficiency, increase testing functions and
improve system performance and to improve the accuracy
and precision control. Virtual instrument has already got a
very universal application in developed countries, while in
china the traditional instruments are still in the separation
from computer. Therefore, it will be very conducive for the
development of china’s measurement and control technology
to use virtual instrument technology to transform traditional
equipment. And it will greatly improve the production
efficiency and reduce the production costs.
REFERENCES
.
144
[1] ZhaoYong, The status and development trend of virtual instrument
software platform technology. Foreign Electronic Measurement
Technology, 2002(1).
[2] XuYun, Virtual instrument data acquisition based on serial
communication. Instrument technology and sensor, 2002(1).
[3] YangLeping, LabView programming and application, Electronics
Industry Press..
 2009 International Conference on Computer Engineering and Technology
A New Optimization Scheme for Resource
Allocation in OFDMA based WiMAX Systems
Arijit Ukil, Jaydip Sen, Debasish Bera
Wireless Innovation Lab, Tata Consultancy Services
BIPL, Saltlake-91, Kolkata, India
arijit.ukil@tcs.com
Abstract— In this paper, an efficient scheme to optimize
resource allocation for dynamic OFDMA based WiMAX systems
is presented, which provides priority estimated resource
allocation with individual QoS provisioning. Our proposed
scheme dynamically assigns OFDMA resources to the users to
optimize the overall system performance in heterogeneous traffic
and diverse QoS based WiMAX systems. To achieve this goal, we
have introduced priority indexed long-term (PILTPF) resource
allocation algorithm, which dynamically allocates the OFDMA
resources to the users to meet their QoS requirement, which is
dependent on user’s derived priority profile. It also considers the
time diversity gain achieved in long-term computation of
Proportional Fair (PF) metric. PILTPF algorithm emphasizes on
individual user’s true priority in allocating system resources in
order to maintain as well as optimize the requirements of
different QoS classes, existing in the current and next generation
broadband wireless networks like WiMAX. Simulation results
show considerable improvement of performance due to priority
based resource allocation with long term PF calculation with
respect to traditional PF algorithm.
Keywords— WiMAX, OFDMA resource allocation, QoS, long-
term fairness, time-diversity gain, scheduling
I. INTRODUCTION
The next generation broadband wireless applications
require high data rate, low latency, minimum delay, real-time
applications; in short highly demanding Quality of Service
(QoS), which cannot be realistically provided unless the
limited system resources, bandwidth and transmitter power are
intelligently used and properly optimized. Optimization
approaches basically attempt to dynamically match the
requirements of data-link connections to the physical layer
resources available to maximize some system metric. The
optimization objective and algorithm developed so far for
dynamic resource allocation in OFDMA systems mostly
consider instantaneous gain in system performance, where the
system is either to attain maximum aggregate throughput or to
provide fairness among the users or to have a trade-off
between them. It was shown in [2] that in order to satisfy the
objective of maximizing the total capacity, each subcarrier
should be allocated to the user with the best gain on it, and the
power should be allocated using the water-filling algorithm
across the subcarriers. The principle of the resource allocation
taken in [3] is to allow the user with least proportional
capacity to use that particular subcarrier. Large gain in
throughput can be achieved through multiuser raw rate
maximization by exploiting multi user diversity (MUD) but
simultaneously fairness must be guaranteed. Proportional Fair
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.10
(PF) based optimization heuristically tries to balance the
fairness among the users in terms of outcome or throughput,
while implicitly maximizing the system throughput in a
greedy manner. PF optimization problem as instantaneous
maximization of the sum of logarithmic user data rate in
multi-carrier systems is considered in [4], which selects a
user-carrier mapping where the logarithmic sum of the user is
maximized.
It is very much computationally expensive to satisfy each
user’s instantaneous data rate requirement in an optimum way.
In order to be practically feasible, previous research work [5]
proposes low complexity suboptimal algorithms. But the
approach of resource allocation based on instantaneous
guarantee of QoS does not consider the temporal diversity of
mobile wireless channel. Considering the channel dynamics
and the fine granularity of OFDMA systems in both frequency
and temporal domain, it is quite obvious that mean QoS
guarantee in long term for resource allocation would provide
better performance, specifically for delay-tolerant traffic.
Another degree of freedom in QoS guarantee is the priority
of the user. A good resource allocator or scheduler generally
depends on the business model of the operator. QoS guarantee
is an important feature. Mostly, user satisfaction of a well-
served user increases only marginally by increasing the QoS
level even further. However, if the QoS level is declined
below some threshold, the satisfaction level drops
significantly. Premium users are the users with higher priority,
who enjoy the privilege to get the best and uninterrupted
service even if it has bad channel condition, affected the most
with degraded service quality. By assigning considerable
resources to premium users like UGS class in WiMAX, the
limited system resources can not be properly utilized, which is
not what the service provider would like. So an integrated
scheme to simultaneously satisfy the individual QoS (user
requirement) as well as provide system performance
maximization (service provider requirement) is very much
required. So, it is quite likely that QoS provision should play a
major role in deciding resource allocation and should be
coupled together.
From this perspective we propose an efficient resource
allocation scheme with integrated QoS provisioning by
evaluating and assigning priority index to each user and a
biased optimization based on the assigned priority index. The
proposed PILTPF algorithm attempts to allocate the OFDMA
subcarriers to the users to achieve its minimum mean data-rate
requirement within few frame duration with a degree of
145
biasness to the higher priority users in heterogeneous traffic Pkn is the transmit power for nth subcarrier when transmitted
condition. The objective of the PILTPF algorithm is to
simultaneously provide long term fairness, QoS and system P
to kth user, which is equal to . Let Rknt = f( hknt )be the
optimization in order to satisfy the requirement both user and N
the service provider. PILTPF algorithm is very much suitable instantaneous achievable rate for kth user when nth subcarrier is
and practical for wireless broadband systems like WiMAX,
allocated at tth time instant and Rkt is the achieved data rate
LTE,IMT-A for its simplicity and mean QoS guarantee
feature. Simulation results also show the performance of the kth user at allocation epoch t and Rkt is expressed as,
improvement by PILTPF algorithm over unbiased N
B
conventional Proportional Fair algorithm.
The paper is organized as follows. In section II system
Rkt =
N
∑ρn
knt f (hknt ) (1)
model and problem formulation are presented. In section III
Unbiased Instantaneous Proportional Fair Optimization  
 hknt × Pkn 
2
problem is discussed. PILTPF algorithm and the concept of
f( hknt ) = log 2 1 +
priority indexing of the users are presented in section IV.

 B 
Simulation results roughly based on Mobile WiMAX Scalable  NT × 
OFDMA-PHY and analysis of the results are presented in the  N 
next section. Section VI provides the summary, conclusion  
and future scope of work. N  hknt × Pkn 
2

Rkt = ∑ ρ kn x log 2 1 +  (2)

II. SYSTEM MODEL AND PROBLEM FORMULATION  B 
 
n =1
NT ×
OFDMA is the standard multiple access scheme for next  N 
generation wireless standards like WiMAX, LTE, IMT-A. N K
OFDMA is basically Multi-user-OFDM. OFDMA is
characterized by a fixed number of orthogonal subcarriers to ∑∑ ρ
n =1 k =1
knt =N (3)
be allocated to the available users [1]. OFDMA resource
allocation algorithms dynamically assign mutually disjoint where ρ knt is the subcarrier assignment matrix at allocation
subcarriers to the users by taking the advantage of MUD to epoch t, if the nth subcarrier is assigned to kth user at tth time
meet the some specified system performance objective with instant, then ρ knt equals to 1, otherwise 0. In practice the
the help of knowledge of the channel condition available from
achieved data rate is less than that of what equation (2) suggests
channel state information (CSI). Multiuser OFDMA system
as there exists a few dB SNR gap. SNR gap in simplified [5]
architecture with Subcarrier Allocation and Dynamic Priority
Index Estimator module is shown in Fig. 1. A single cell term can be approximated as ∆ gap = -ln(5BER)/1.6. Then Rkt
downlink wireless cellular network with one base station (BS) becomes:
serving total K users is considered, each of which requires
 
certain QoS guarantee. The interference from adjacent cells is N  hknt × Pkn 
2
treated as background noise. The total available bandwidth B
is partitioned into N equal narrowband OFDMA subcarriers;
Rkt = ∑ρ ktn x log 2 1 +
 B 
 x ∆ gap (4)
 
n =1
B NT ×
the the subcarriers are denoted as s1, s2 ….sN, where sn =  N 
N As shown in Fig. 1, OFDMA subcarrier allocation architecture
Hz and in order to overcome frequency selective fading si is consists of three components, namely, Subcarrier allocation
chosen to be sufficiently smaller than the coherence Module, Priority Index Estimator and Scheduler. Priority
bandwidth of the channel. Each OFDMA subcarrier n, Index Estimator calculates the unique priority of each of the
belonging to user k, is subject to flat fading, path loss and user based on maximum buffer size; minimum data rate
shadowing with channel gains H = { hkn }. In addition, the requirement, maximum tolerable delay and QoS class of the
user and assigns each user with a unique priority index. Let
signals suffer from AWGN noise, which has Gaussian
distributed with zero mean with psd N0 and total noise power the available QoS classes be Qm , where 1≤m≤M. For an
is assumed as N T . Subcarrier allocation epoch is assumed to example, in WiMAX there exists five (M=5) QoS classes;
be less than the channel coherence time. Perfect channel UGS, ertps, nrtps, rtps and BE. UGS and ertPS class users are
characteristic is assumed in the form of channel state given the highest priority and should be satisfied with their
information (CSI). The QoS of the kth user is described by the QoS requirement all the time. PI k denotes the priority index
minimum individual rate requirement = [γ1, γ2, ……….., γK] of kth user and PI k is function of the maximum buffer size,
in bits/sec. Total available Transmitter Power is P. PILTPF
algorithm is simplified by equally distributing the total Signal to Noise Ratio (SNR), maximum latency and QoS class
available transmitted power as performance can hardly be of the kth user:
deteriorated by equal power allocation to each subcarrier [6].

146
PI k = f (Qk , bs k , γ k , δ k ) (5) R
k* = arg max knt (8)
where bs k is the maximum buffer size and δ k maximum Rkt
k
tolerable delay.
Subcarrier Allocation Module (Fig. 1) with the help of 1 1
Rkt = (1 − ) Rkt −1 + Rkt −1 (9)
PILTPF algorithm allocates OFDMA subcarriers to the users. ∆τ ∆τ
The scheduler decides, in each frame, how to schedule the The constraint of the system in order to provide QoS is:
packets for transmission. Basically, subcarrier allocation
module does the frequency domain mapping of the user to the
Rkt ≥ γ k ∀k (10)
subcarriers, scheduler does the time domain mapping of the The optimization and sub-optimization schemes (6-10)
users to the time-slots. Priority Index Estimator considers discussed so far deals with maintaining QoS guarantee
system constraints and based on user’s QoS class maps the instantaneously and without any provision to provide
QoS metric of delay, buffer-size, minimum data rate privileged service to the higher priority users. Traditional PF
requirement to a priority value, unique to each users. can only support a loose delay-bound QoS requirement, which
is not suitable for real-time multimedia services where the
III. UNBIASED INSTANTANEOUS PROPORTIONAL FAIR delay bound QoS requirement is stringent, which demands
OPTIMIZATION modification of the traditional approach of PF optimization to
Proportional Fairness (PF) in OFDMA system maximizes handle diverse QoS based traffic. PF is also not the most
the sum of logarithmic mean user rates [5, 7]. PF scheduler optimized solution for delay-tolerant applications, like nrtps,
maintains fairness by the law of diminishing return. BE class traffic in IEEE 802.16. To take full advantage of the
time-diversity gain, inherent to the nrtps and BE class of
Subc traffic, a QoS class-biased proportional fair optimization has
User1 arrier
1
to be introduced, which is discussed in the next section.
T
S
r
c
… h
a IV. PRIORITY BASED PROPORTIONAL FAIR OPTIMIZATION AND
buffer1 n ALGORITHM
User… e
K d s
Subc
u arrier
m Unbiased instantaneous Proportional Fair Optimization as
l i
e
N t
described in (6-10) is based on instantaneous computation of
bufferK
r t Wirel proportional fair metric ( PF ) and the resultant optimization
QoS e ess
K, r does not consider the time diversity and priority of the users. PF
calculates achievable data rate of individual user instantaneously
Priori
Subcarri CS
(8-9). The mean data rate achieved ( Rkt ) is computed as
ty
Index er I moving average to imply a notion of low pass filter [7] for the
Estim Allocatio {h
kn} purpose of providing fairness.
QoS1 ator P n
I
Wireless channel is normally very much dynamic in nature.
k Long term optimization for system performance improvement
Fig. 1. Priority based Resource Allocation module yields time diversity gain. The priority of the user, which is
N K
basically the monotonically increasing assurance of QoS, should
PF t = max ∑∑ ln R kt (6)
be taken into account to provide the premium users’ QoS
guarantee. Traditional PF optimization (6-10) does not consider
n =1 k =1
user’s priority and treats every user equally and unbiased way.
Equation (5) can be more generalized when PF is defined
This kind of subcarrier allocation may deprive the higher priority
as follows [6]:
user with bad channel condition to maintain its QoS. Priority

 ∑ Rknt  index is already introduced in equation (5). The simplest way of
PF t = ∏ 1 + n∈N incorporating user priority is to give the chance to the highest
 (7)
priority user to get allocated by the best of the subcarriers. But
k∈K  (∆τ − 1) Rkt 
  that will be the strict priority based subcarrier allocation. The
advantages incurred by considering the parameters like
where PF t is the proportional fairness index at tth instant,
maximum buffer size, maximum latency should be taken in to
∆τ is average window size or the period between successive account, which relaxes the strict priority and reflects the true
priority.
allocations and Rkt is the average data rate achieved of the
PILTPF algorithm exploits the time diversity as well as
user k at the preceding allocation instant. The above equation considers the priority of the user by calculating priority index
(7) is the optimal PF subcarrier allocation, but can not be based PF metric in long term to meet the constraint of minimum
implemented due to its high computational complexity. The rate requirement. PILTPF has two parts and operates
suboptimal form [7] is where subcarrier n is allocated to k* sequentially. First part estimates the user’s priority based on (5)
user when and the next one is subcarrier allocation.

147
 Priority estimation evaluates the dynamic priority of the priority of the user is estimated by PI k , which converts the
user based on its current QoS class, available buffer–size,
traditional PF optimization as weighted PF optimization.
delay limit and minimum data rate requirement. PI is highest
for UGS class and for rest PI is estimated as below: Higher the priority of the user, higher the magnitude of PI k
Qk × γ k and more is the chance to get the subcarrier allocated even in
PI k = (11)
(bs kMAX ) (
− bs k t × δ MAX − δ t
) bad channel condition and high Rkt . Equations (11-15)
describes PILTPF optimization scheme. It can also be noted
where bskMAX , bsk t , δ MAX , δ t are the maximum buffer-size, from (13) that more time diversity gain would be achieved
used buffer-size, maximum delay-limit and elapsed delay for with the increasing value of δ kMAX , in fact.
kth user at current allocation instant (t). For, NDC traffic,
δ MAX − δ t = 1 . It can be clearly seen that: Lim P (ω k → γ k ) = 1
δ kMAX →∞
Above equation is the condition of absolute convergence to
PI k → ∞ , when bs kMAX − bsk t
or δ MAX − δ t → 0 the QoS requirement with probability one. This is due to the
Equation (11) is based on the notion that user’s priority will fact that in wireless mobile environment over long duration,
increase as the urgency for user’s allocation to resources time diversity gain becomes high and the mean channel
becomes higher. This (11) can also be termed as urgency condition ( Rkn ) follows similar distribution according to the
based priority assignment, which attempts to avoid QoS
violation to the maximum extent as well as minimize the Bernoulli’s Law of Large Numbers, which justifies the
outage probability. It can be noted in traditional resource intuitive interpretation that the expected value of a random
variable is basically long-term average when sampled
allocation, priority mostly depends on the value Qk only, repeatedly and can well be considered as independent and
which is termed as strict priority, where as, in priority index identically distributed (i.i.d) random variable with mean µ k .
estimation function (11), all the QoS related PHY and MAC
parameters are taken into account, which reflects the true Then theoretically, Lim TDgain = µ k
δ kMAX →∞
priority of the user and it is highly dynamic.
Subcarrier allocation part assigns the appropriate PILTPF optimization utilizes this performance gain for its
subcarrier to the user to optimize the system performance. It attempt to converge to γ k (14) by relaxing the QoS constraint
exploits the TD gain whenever possible. The optimization is of minimum data rate as statistical mean value. Maximum
modified PF. Subcarrier allocation module selects k* user to system optimization is obtained when all each user’s
assign n-th subcarrier based on:
Tk
allocation is complete only when t = δ kMAX , ∀k , which
PI k × Rknt exploits the time-diversity gain of each and every users.
k* = arg max (12) Based on the optimization scheme (11-15) the proposed
k Rkt t =1 PILTPF algorithm is described as below:
Tk T
where, Rkt = E ( Rkt ) t =1 (13) Step1: Set initial mean achievable data rate: E ( Rkt ) =ε
t =1
The constraint of optimization utilizes the TD gain. Instead of and t=0 , where ε is a small number.
instantaneous data rate guarantee, it assures long-term average Step 2: Find
PI k , ∀k as per (11)
data rate.
Tk Step2: Find the user k* as per (12) for all the subcarriers
E ( Rkt ) t =1 ≥γk (14) Step3: Calculate the data rate achieved as
N
Tk = ∂ kMAX
∑ρ
(15)
per Rkt = knt × Rknt for all the users at tth instant
PILTPF optimization as described does not depend on the n =1
moving average value of Rkt nor does the PF metric is Step4: Find the mean data rate achieved by the kth user at tth
instant.
calculated instantaneously. Instead it calculates both the PF T
metric and Rkt in a window of Tk and allocates the Step5: if E ( Rkt ) t =1 ≥ γ k , find next best k, else continue

subcarrier to the user in a way to maintain its minimum
average data rate requirement within Tk time duration (14).
If Tk = T f , where T f is the allocation instant; the problem is
purely proportional fair. More the value of Tk , the more
ergodic the optimization scheme becomes and with higher
granularity in T f , better optimization can be obtained. The
Step6: t= t+ T f and go to Step 2
V. SIMULATION RESULTS AND ANALYSIS
To investigate the performance of PILTPF algorithm,
simulation results under the system parameters and simulation
scenario given in Table 1 is presented in this section. The
system parameters are roughly based on Mobile WiMAX
Scalable OFDMA-PHY. Frequency Reuse factor of 1 is taken

148
so that all the subcarriers can be assigned to the users. VI. SUMMARY AND CONCLUSION
Subcarrier allocation instant is taken as equal to be equal to We have proposed an efficient but simple OFDMA
the frame duration (5ms), which is assumed to be less than resource allocation algorithm, which has shown the
coherence time of the channel. A random heterogeneous mix characteristics of better performance and QoS guarantee in
of UGS,ertps, ertps, nrtps, BE traffic with varying QoS metric long-term than the conventional suboptimal PF algorithm in
like delay, buffer-size, minimum data rate are taken for QoS diversified heterogeneous traffic condition. PILTPF is of
simulation purpose. Simulation results are shown in Fig. 2, 3 very practical importance and can well be implemented for
and 4. next generation broadband wireless systems like LTE,
Fig. 2 is showing the comparison of achievable data rate of WiMAX, IMT-A to improve the overall system
the users by PILTPF and PF algorithm. It also shows the performance.
minimum data rate requirement of individual user as QoS
profile. Fig. 2 clearly shows that PILTPF algorithm follows
QoS profile for all the users where as in PF algorithm some of
the users get deprived and achieved very less data rate than
their minimum requirement due to the inherent feature of its
instantaneous PF metric computation. If the deprived user, say
10th user (fig. 2) is a high priority customer, then the
allocation is very much unacceptable from service provider as
well as user perspective. In Fig. 3 a bar chart comparing the
performance of PILTPF and PF algorithm is depicted. From
Fig. 2 and 3, it can be interpreted that as PILTPF considers
both the priority of the user and the time diversity gain, so it
results in better performance both in terms of throughput and
QoS guarantee. The unevenness in assigning subcarrier by Fig. 3. Chart comparison between PILTPF and PF
conventional PF algorithm becomes more visible when more
number of users exists. In that highly complex scenario of
large number of users, it is depicted in Fig.4 that PILTPF at
least attempts to follow the QoS profile in order to preserve
the importance of priority of the users. However, in a less
number of users and good channel condition throughout the
entire cell, the difference between the PF and PILTPF
diminishes.
TABLE I
Simulation Parameters
Available Bandwidth 1.25 MHz
Number of users 20/30
Number of sub-carriers 72
BER 10-3
SNR_gap -ln(5BER)/1.6
Channel model Rayleigh Fig. 4. User Achievable throughput by PILTPF and PF algorithm when
Modulation 16QAM number of users = 30
Frequency reuse factor 1
Channel sampling frequency 1.5 MHz REFERENCES
Maximum Doppler 100Hz [1] Ahmad R, et al. “Multi-Carrier Digital Communications Theory and
Applications of OFDM”. 2nd ed. Springer.
[2] Rhee et al. “Increase in capacity of multiuser OFDM system using
dynamic subchannel allocation”. IEEE VTC., pp. 1085–89, May 2000.
[3] Z. Shen, et al, “Adaptive resource allocation in multiuser OFDM
systems with proportional rate constraints,” IEEE Trans. Wireless
Communication, vol. 4, no. 6, pp. 2726–2737, Nov. 2005.
[4] Hoon Kim and Youngnam Han, “A Proportional Fair Scheduling for
Multicarrier Transmission Systems”, IEEE Communication Letters, pp.
210-212, vol.9, no.3, March 2005.
[5] Abolfazl Falahati, Majid R. Ardestani. “An Improved Low-Complexity
Resource Allocation Algorithm for OFDMA Systems with
Proportional Data Rate Constraint”, ICACT, pp. 606-611, 2007.
[6] Wei Xu, et al."Efficient Adaptive Resource Allocation for Multiuser
OFDM Systems with Minimum Rate Constraints", IEEE ICC, pp.
5126-5131, 2007.
[7] Christian Wengerter, Jan Ohlhorst, Alexander Golitschek Edler von
Elbwart., “Fairness and Throughput Analysis for Generalized
Fig. 2. User Achievable throughput by PILTPPF and PF algorithm when Proportional Fair Frequency Scheduling in OFDMA.”, IEEE VTC,
number of users = 20 Vol.3,pp.1903-1907, 2005.

149
 2009 International Conference on Computer Engineering and Technology

An Integration of CoTraining and Affinity Propagation for PU Text

Classification

Na Luo1,2, Fuyu Yuan3 , Wanli Zuo1

1
College of Computer and Science and Technology,
JiLin University, Changchun, JiLin, 130012, China
2
Department of Computer,
Northeast Normal University, Changchun, JiLin,130117, China
3
State Key Laboratory of Electroanalytical Chemistry,
Changchun Institute of Applied Chemstry Chinese Academy of Sciences, 130022, China
Luon110@nenu.edu.cn, wanli@jlu.edu.cn, yuanfuyu@yahoo.com.cn

Abstract excluding the positive class (e.g. sample of non-

Under the framework of PU(Positive data and
Unlabeled data), this paper originally proposes a
three-setp algorithm. First, CoTraining is employed for
filtering out the likely positive data from the unlabeled
dataset U. Second, affinity propagation (AP) approach
attempts to pick out the strong positive from likely
positive set which is produced in first step. Those data
picked out can be supplied to positive dataset P.
Finally, a linear One-Class SVM will learn from both
the purified U as negative and the expanded P as
positive. Because of the algorithm's characteristic of
automatic expanding positive dataset, the proposed
algorithm especially performs well in situations where
given positive dataset P is insufficient. A
comprehensive experiment had proved that our
algorithm is preferable to the existing ones.
1. Introduction
Traditionally, a general binary text classifier is built
by employing some algorithm on the positive dataset
and negative dataset. This kind of algorithms are
termed supervised learning algorithm [1]. However,
Traditional supervised learning algorithms are thus not
directly applicable because they all require both
labeled positive and labeled negative documents to
build a classifier. For instance, in order to construct a
”homepage” classifier, one needs to collect a sample of
homepages (positive training examples) and a sample
of non-homepages (negative training examples).
Collecting negative training examples is especially
delicate and arduous because (1) negative training
examples must uniformly represent the universal set
homepage should represent the Internet uniformly
excluding the homepages), and (2) manually collected
negative training examples could be biased because of
human’s unintentional prejudice, which could be
detrimental to classification accuracy. For above
reason, PU classification has been an important
problem.
This paper has two main contributions. The first is
the proposal of employment CoTraining for purifying
unlabeled dataset. Unlike former PU algorithm [2,3]
exploiting reliable negative dataset, CoTraining iterates
to purify unlabeled dataset by filtering out some likely
positive examples. The behind rationality is that
purifying the unlabeled data by filtering out a small
fraction positive data is much easier than exploit a
small but reliable negative dataset. Moreover, the
likely positive data being filtered out can also be made
use of supplementing positive dataset. Our second
contribution is the employment of affinity propagation
(AP) on likely positive data for expanding positive
dataset. Finally, a linear SVM will train both on the
purified unlabeled dataset as negative examples and the
expanded positive dataset as positive examples. Thus,
PU problem is transformed to supervised learning
problem.
The remainder of the paper will be organized as
follow: Some related works is presented in section 2.
Details about the proposed algorithm can be found in
section 3. A number of comparative experiments have
been made in section 4. Section 5 will briefly make
some conclusions.
2. Related Works

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 150

DOI 10.1109/ICCET.2009.131
 Probably Approximately Correct (PAC) learning each other by filtering out likely positive examples
from positive and unlabeled examples is proposed. It from the unlabeled dataset of the counterpart. In
defines a PAC learning model from positive and algorithm 1 positive dataset is denoted by P, unlabeled
unlabeled statistical queries. Liu Bing employs a dataset is U, arbitrary document is d, classifier is S, and
weighted logistic regression to PU and originally the purified unlabeled dataset is denoted as pU, L
defines a measure on unlabeled dataset. PEBL is represent the likely positive dataset which is filtered
another two-step approach involving in PU [3]. Unlike out from U. Classify(S,d) signify the label that S
other methods, it attempts to explore the distinguishing assigned to d.
words between positive and unlabeled corpus, then Algorithm 1: CoTraining Algorithm
using these distinguishing words to obtain reliable Input: (P,U)
negative set (RN). Liu Bing etc. reviewed several two- Output: (L,pU)
step PU approaches and proposed his Biased-SVM [2]. 1: Randomly split U set into two set U1, U2, subject to
Combining CoTraining framework with PU is first U1∪U2=U, U1∩U2={}
proposed by Blum and Mitchell [4]. Another 2: Build SVM classifier S10 with P as positive and U1
comprehensive study on Co-Training is made by as negative
Nigam and Ghani [5]. 3: Build SVM classifier S20 with P as positive and U2
as negative
3. The Proposed Algorithm 4: L={}; Q={}; R={}; i=0;
5: while(true)
As mentioned in introduction section, this paper had 6: Q={Classify(S2i,d)==positive, d∈U1}
proposed both the adoption of CoTraining for 7: U1=U1-Q
purifying unlabeled dataset and affinity propagation 8: R={Classify(S1i,d)==positive, d∈U2}
(AP) for expanding positive dataset. Specifically, the 9: U2=U2-R
proposed algorithm can be divided into three steps: 10: if(Q={} and R={})
(1) Purify unlabeled dataset with CoTraining by 11: break
filtering out likely positive set. 12: end if
(2) Employ affinity propagation on likely positive 13: L=L∪Q∪R; i=i+1;
set for supplementing positive dataset. 14: Build SVM classifier S1i with P as positive and
(3) Perform a linear SVM on the purified unlabeled U1 as negative
dataset as negative and the expanded positive dataset 15: Build SVM classifier S2i with P as positive and
as positive. U2 as negative
16: end while
3.1. CoTraining Step 17: L={Classify(S1i,d)==positive and
Classify(S2i,d)==positive, d∈L}
In order to obtain a reliable negative dataset, former 18: pU=U1∪U2
algorithms used to first assigned negative "pseudo-
label" to all unlabeled data and then train on it together 3.2. Affinity Propagation Step
with positive dataset. The result classifier is used again
to assign "pseudo-label" to unlabeled data. Obviously, For the purpose of expanding positive dataset,
such classifier is trained and classifying on the same affinity propagation is adopted. So far as we known,
unlabeled dataset. Theoretically, this classifier is not there is no existing algorithm involving with
likely to assign a confident label. This embarrassment supplementing positive dataset.
can be indeed avoided by out CoTraining algorithm The advantages of affinity propagation clustering[6-9]
which use two individual classifier trained on two over other clustering methods lie in that it’s more
different datasets to purify U iteratively. stable for different initializations. In affinity
Remarkably, the way CoTraining differs from propagation clustering, two kinds of message are
traditional CoTraining [5] is that the two basic exchanged between data points, each of which takes
classifiers are based not on two different feature spaces into account a different kind of competition. Messages
but on two different training datasets of the same can be combined at any stage to decide which points
feature space. are exemplars and, for every other point, which
Specifically, CoTraining consists of two individual exemplar it belongs to. The “responsibility” r(i,k), sent
SVM learners (base classifier) which are built on the from data point i to candidate exemplar point k,
same positive dataset and the different unlabeled reflects the accumulated evidence for how well-suited
datasets. The two base classifiers "help" or "co-purify" point k is to serve as the exemplar for point i, taking

151
into account other potential exemplars for point i. The
“availability” a(i,k), set from candidate exemplar point
k to point i, reflects the accumulated evidence for how
appropriate it would be for point i to choose point k as
its exemplar, taking into account the support from
other points that point k should be an exemplar. To
begin with, the availabilities are initialized as a(i,k)=0,
and the responsibilities are initialized as r(i,k)=0, Then,
the responsibilities and availabilities are iteratively
computed as:
r (i, k ) ← s (i, k ) − max{a (i, k ' ) + s (i, k ' )}
'
k ≠k
a(i, k ) ← min{0, r (k , k ) + ∑ max{0, r (i , k )}}, i ≠ k
i ≠i &i ≠ k
' '
a (k , k ) ← ∑ max{0, r (i ' , k )}
i' ≠k
Where s(i,k) reflects the similarity between the data
points i and k. For all i ≠ k, s(i,k) can be set to be the
negative Euclidean distance, namely, s (i, k ) = − xi − xk 2 ;
while for all i=k, s(k,k) is a varying parameter, and the
initialized values of s(k,k) for all ks are set to be equal
to each other because all data points are equally
suitable as exemplars. The affinity propagation takes a
real number s(k,k) as its input. The number of
identified exemplars (number of clusters) is influenced
by the initialized value of s(k,k). As reported in the
literature, the shared value of s(k,k) is set as the median
of the input similarities (resulting in a moderate
number of clusters) or their minimum (resulting in a
small number of clusters). However, the true number
of clusters may be a widely changeful value, but not
exactly the moderate number or the small number. So
in our design, we set the initialized value of s(k,k)
varying from minijs(i,j) to their maximum maxijs(i,j),
namely:
s (k , k ) = min s (i, j ) + α ( max s (i, j ) − min s(i, j ))
i, j i, j
where α ∈ [0,1] .
To quantify the labels, we assign 1 to positive label
and -1 to negative, pU is the purified unlabeled dataset,
and L is likely positive dataset. In this way, AP pick
out the supplementary set (sP) from likely positive
dataset.
Algorithm 2: Affinity Propagation Algorithm
Input: (P,pU,k,L)
Output: (aP)
1: sP={};
2: for each d ∈ T
3: Rank documents in pU and P according to its
similarity to d;
4: s ( k , k ) = min s (i , j ) + α ( max s (i , j ) − min s (i , j ))
i, j i, j i, j
5: sum=0
6: for each nd ∈ s ( k , k )
7: if( nd ∈ P )
8: sum=sum+similarity(nd,d)*(1);
9: else if( nd ∈ pU )
10: sum=sum+similarity(nd,d)*(-1);
11: end if
12: end for
13: if(sum>0)
14: sP = sP ∪ {d };
15: end if
' 16:
17:
end for
aP = sP ∪ P;
After AP step, positive dataset is expanded with sP,
and this technique is especially suited for situations
where the given positive dataset is insufficient.
3.3. Semantic-based Feature Extraction in
SVM Algorithm
A word possibly has many meanings, WordNet
expresses a meaning with a synonym sets. If two
features in one category have a common synonym set,
we can say that these features represent this category
greatly [10]. In this paper, the semantic feature
extraction is used by WordNet through the above
examples. First, Documents meaning were determined
by overlapping semantics of Documents, Next,
documents meaning found out the features which
formed documents vector. The algorithm provides the
method applying semantic extraction to form positive
document vector for PU problems. Algorithm finds the
semantic for all the words in the positive set, and the
repeated semantics are the meaning of the multi-sense
words in the positive set, which represents the meaning
i, j
of the document. The words corresponding to these
semantics construct the important features represented
positive examples, so they can represent the document
meaning greatly. Forming the vector of positive
document by the words is to avoid losing the meaning
of document with reducing the dimension of document
vector, and ensure the later classifier. In the process of
semantic feature extraction, multi-sense words are
disambiguated and selected by correct semantic.
Currently, SVM is the most popular algorithm in
text classification. The principle of minimizing the
structure risk enables SVM to be suitable for high
dimension text data. However, One-Class SVM
algorithm uses only the positive set to train the
classifier. In this paper we had improved the original
algorithm to form the document vector of positive
examples by above method of semantic feature
152
extraction, and use the document vector to construct concerned, the goal of this step is to purify U by
the classifier. The algorithm of semantic-based feature filtering out the positive example. To measure the
extraction in One-Class SVM is as follows: effect of being purified, the popular F can be a good
choice. In our experiment, after CoTraining step, the
Algorithm 3: Semantic-based Feature Extraction in precision and recall of negative data in pU (purified
One-Class SVM Algorithm unlabeled dataset) are calculated and then combined as
1: Set all semantics appeared in LP follow: f=2*recall*precision/(recall+precision). For
hm_allSyn=NULL, Set semantics appeared the second step, it attempts to expand P with a
repeatedly in LP hm_crossSyn=NULL supplementary set (sP) that is extracted from LP (likely
2: For each document d in LP positive dataset). Obviously, the effect of the second
3: For each word w in d step totally depends on the precision of positive data in
4: For each semantic s of w this supplementary set (sP). Finally, for the third step
5: If not s∈hm_allSyn aiming to build an accurate classifier, F score in test
6: hm_allSyn+=s dataset is employed as measure of the final classifier’s
7: else if not s∈hmcrossSyn accuracy.
8: hm_crossSyn+=s
9: For each document d in LP 4.3 Experiments
10: For each word w in d
11: t=tfidf(w) Experiments are comparing the methods of
12: For each semantic s of w document frequency and semantic feature extraction
13: if(t≠0 and s∈hm_crossSyn) for one-class SVM algorithm, and the process is as
14: output w to the file of document vector of follows:
positive examples
15: sP=LP∪{d} Express data set as the file
16: break of sequence of woks

4. Empirical Evaluation Training set Test set

4.1. DataSets Training the set U Training the set P

We used two popular text collections in our Forming the tfidf vector
experiments. The first one is the Reuters-21578, which
One-Class SVM One-Class SVM classifier Test document
has 21578 documents collected from the Reuters classifier using using semantic feature
newswire. Of the 135 categories, only the most semantic feature extraction based on
represented by
extraction document frequency SVM vector
populous 10 are used. Each category is employed as
the positive class, and the rest as the negative class.
This gives us 10 datasets. The second collection is the
Usenet articles collected by Lang. we use each Fig 1: The system flow chart
newsgroup as the positive set and the rest of the 19 In the experiment, we take two experiments as γ is
groups as the negative set, which creates 20 datasets. 0.1 and 0.7. Table1 provides the average F score of all
For each dataset, 30% of the documents are randomly categories of classifier and shows that our method
selected as test documents. The rest(70%) are used to increases the F score by 11.15 percent as γ is 0.7, and
create training sets as follows: γ percent of the
by 6.45 percent as γ is 0.1. So we draw the conclusion
documents from the positive class is first selected as that out three-step method improves the performance
the documents from the positive set P. Te rest of the of One-Class SVM algorithm, especially when the
positive documents and negative documents are used positive examples are few.
as unlabeled set U. In the data preprocessing, after Table 1 Average F of One-Class SVM classifier
remove stopwords and stemmer, we use tf-idf express datasets γ Average F Previous Best F
document vector. 0.1 0.735 0.716
Reuters
0.7 0.885 0.780
4.2. Evaluation Measure 0.1 0.742 0.632
20Newsgroup
0.7 0.892 0.774
There are three measures evaluate the effects of
three steps respectively. So far as CoTraining step is

153
 At last, we had trained the final SVM classifier
from the purified unlabeled dataset (pU) as negative
and the expanded positive dataset as positive. In order
to test the final classifier, it is used to classify the data
in 10 test dataset and the F score corresponding to each
dataset are listed in Table2. The result proves that the
combination of three steps is superior to the former PU
algorithms. Moreover, even if the given positive
dataset is insufficient, the algorithm still performs well.

Table 2 F Score of 10 datasets after three steps

γ acq com crude earn grain interest money ship trade wheat Avg
0.7 0.9380 0.8023 0.8705 0.9829 0.9211 0.8073 0.8745 0.8353 0.8822 0.7514 0.8666
0.5 0.9544 0.8527 0.8935 0.9809 0.9021 0.8217 0.8683 0.8742 0.9433 0.8333 0.8924
0.3 0.9642 0.7863 0.9049 0.9779 0.9305 0.7896 0.8541 0.7853 0.8701 0.8000 0.8663

5. Conclusions [3] H.Yu, J. Han, K.C. Chang, “PEBL: Positive example

This paper combines CoTraining in PU
classification, and learns the problems of PU text
classification based on affinity propagation and
semantic feature extraction. Its goal is that use these
algorithms can improve the performance of classifier.
Affinity propagation can improve the performance of
PU classifier when the positive examples are few.
From the comparison with the method, One-Class
SVM increases the F score by 10.183 percent, which is
an important measure of classifier.
Acknowledgement
This work was supported by National Nature
Science Foundation of China under Grant
No.60803102, and Technological Development
Projects of JiLin Province under Grant No.20070533.
References
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classifiers using positive and unlabeled examples”, In: Proc.
of the 3rd IEEE int’l Conf on Data Mining, IEEE Computer
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New York, NY, 2006, pp.429-436.
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image retrieval results”, In: Proc. of the 14th Annual ACM
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Web Intelligence(WI2004), IEEE Computer Society, Beijing,
2004, pp.166-172

154
 International Conference
on Computer Engineering and Technology

Session 3
 2009 International Conference on Computer Engineering and Technology
Ergonomic Evaluation of Small-screen Leading Displays on the Visual
Performance of Chinese Users
Yu-Hung, Chien
Department of Product Design
Ming Chuan University
5 De Ming Rd., Gui Shan District, Taoyuan
County 333, Taiwan
roland@mail.mcu.edu.tw
Abstract
Leading display represents a mechanism for
exhibiting temporal information instead of spatial
information to overcome the limitations of small-screen
mobile devices. Previous studies examining this area
focused only on information presented by leading
displays, the use of such devices may be influenced by
the context of tasks and adaptability of users was
somehow disregarded. Actually, mobile interaction is
often a secondary task performed while doing
something else; the attention of leading-display users
cannot always be assumed to be only on reading
leading-display information. Therefore, this
investigation performed a dual-task experiment (a
search task for static information and a reading task
for leading display information) to examine the effects
of leading-display factors on the visual performance of
users during different stages of usage (whether current
usage is the first, second, third, fourth, or fifth day of
usage) for a small screen. The results showed that
leading display design factors did not distract
participants from static information search tasks but
did affect participant reading comprehension on
leading displays. Speed and presentation mode
significantly influenced the participants’ reading
comprehension. Consequently, the possible
applications of leading displays and the implications of
these findings on reading Chinese text are discussed.
1. Introduction
The future is certainly looking mobile and wireless.
Ubiquitous computing and personal networks are
believed to be the near-future paradigms for our
everyday computing needs, essentially giving access to
any information, anywhere at any time.
The rapid development of ubiquitous computing has
led to the increasing use of mobile devices to read text
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.21
Chien-Cheng, Yen
Graduate School of Design Management
Ming Chuan University
5 De Ming Rd., Gui Shan District, Taoyuan
County 333, Taiwan
ccyen0706@yahoo.com.tw
information. However, display space on mobile devices
is at a premium, with the amount of information that
can be displayed on the small screens at one time
extremely limited. One possibility in overcoming this
limitation is to use a leading display, where screen
space can be traded for time to present temporal
information in a dynamic manner.
Previous Chinese dynamic-display studies have
examined the effects of several leading-display factors
on the visual performance of users, including
presentation mode, speed, font size, text/background
color combination, jump length, and Chinese
typography [1-4, 6-8]. According to the results of the
studies [1-4, 6-8], speed and presentation mode were
the two most critical factors for visual performance.
In most leading-display studies, the reading
comprehension of participants in each set of reading
conditions has been measured only once. An
assessment of the adaptability of users to leading
displays, that is, the connection between visual
performance and previous experience in using leading
displays, requires further investigation. Moreover, the
majority of leading-display studies have been
conducted under idealized single-task conditions,
resulting in a lack of information on how changes in
the context of tasks affect the ability of users to
perform effectively. Actually, mobile interaction is
often a secondary task performed while doing
something else; the attention of leading-display users
cannot always be assumed to be only on reading
leading-display information. Frequently, dynamic
display is a means to present notifying information to
users while they are reading static information on
displays. Currently, in leading display used on
websites, for instance, the string of text moves from
right to left sequentially along a single line within a
small screen, which is widely used to show additional
notifying information. In case of presenting notifying
information, users would be performing other task with
157
attention focusing on the dynamic displays. For
instance, users can read dynamic information showed
on a single-line display of a facsimile machine, and the
tasks of searching and pressing keypads have to be
accompanied with simultaneously. Consequently, it is
important to address the disconnection between the
actual use of leading display and the effects of leading-
display factors on users’ visual performance in terms of
dual-task scenario.
Figure 1. Example of the experimental
interface used in this study
2. Method
We conducted a 3 × 2 × 5 repeated-measures dual-
task experiment to examine the effects of speed at
250, 350, and 450 characters per minute (cpm). Speed
settings were based on the studies of Chien & Chen
[2], Lin & Shieh [3],Shieh & Lin [4], Wang, et al. [6],
Wang & Chen [7], and Wang & Kan [8]; presentation
mode (character-by-character or word-by-word) was
based on studies by Chien & Chen [2], Lin & Shieh
[3], and Shieh & Lin [4]; and adaptability (day 1–5 of
use) determined the visual performance of Chinese
users in using leading displays to accomplish dual
tasks on small screens.
The dual tasks, which consisted of a static Chinese-
character search task and a leading-display information
reading task, were conducted on a Sony Ericsson P910i
Smartphone (Fig. 1), with all text material displayed on
a 208 × 320 resolution touch-screen. The smart phone
was placed on a 75 cm high table with a desk
synchronization stand. The smart phone was positioned
at an incline of approximately 105°. The distance from
the center of the screen to the desktop was 8 cm, while
the distance from the participants’ eyes to the center of
158
the screen was 40 cm. An environment involving both
leading and static displays was constructed. Text was
presented in black on a light gray background. The
leading display could be maximized to display ten
Chinese characters of 14-point Chinese typography on
a single-line display. Fig. 2 shows the interface design
in which the leading and static displays are presented
simultaneously on a smart phone.
Figure 2. Photo of the experimental interface
showed on an real-world P910i device
Twelve college students (native Chinese speakers)
from Taiwan were selected as participants. During the
experiment, participants searched for 10 of the same
characters among 100 Chinese characters on the smart
phone screen and simultaneously read a 30-character
leading-display passage. In each trial, participants were
required to finish the search and reading tasks in 30 s.
During this period, the leading display repeated
continuously. At the end of the time period, the number
of discovered target characters was recorded.
Participants were also asked to respond to two
multiple-answer, multiple-choice comprehension
questions based on the leading-display content. At each
session, participants performed two trials under each
condition and repeated the same experimental
procedure on days 1–5.
 Table 1. Scores of search and comprehension for each independent variable
Independent Static-search Leading-display
variable score comprehension
Mean SD Mean SD
Speed (cpm) 250 0.95 0.11 0.88 0.14
350 0.95 0.09 0.84 0.20
430 0.94 0.08 0.72 0.25
Presentation mode Character-by-character 0.95 0.09 0.76 0.22
Word-by-word 0.94 0.10 0.87 0.17
Adaptability Day 1 0.93 0.12 0.82 0.16
Day 2 0.93 0.09 0.83 0.18
Day 3 0.96 0.07 0.83 0.15
Day 4 0.95 0.07 0.80 0.27
Day 5 0.94 0.11 0.79 0.23

Table 2. ANOVA table for mean comprehension of the leading display

Source Type III Sum of Squares df Mean Square F P
Speed 1.732 2 0.866 14.919 0.000 a
Mode 1.084 1 1.084 21.595 0.001 a
Day 0.127 4 0.032 0.666 0.619
Speed × Mode 0.007 2 0.003 0.055 0.947
Speed × Day 0.581 8 0.073 1.634 0.127
Mode × Day 0.141 4 0.035 2.390 0.065
Speed × Mode × Day 0.302 8 0.038 1.253 0.279
a
Significantly different at =0.01 level.

3. Results Previous studies of visual performance for leading

Analysis of variance (ANOVA) was applied during
the statistical analysis of the experimental data and
Tukey’s HSD (honesty significant difference) test was
used for post hoc comparison. The level of significance
was set at α=0.05.
In this study, we found that no factors significantly
affected the static-information search task score, and
none of the interactions among factors was significant.
In contrast, presentation mode and speed significantly
affected reading comprehension (F1,11= 21.60, p< 0.01;
F2,22= 14.92, p< 0.01 ). Reading comprehension was
significantly higher for the word-by-word presentation
mode than the character-by-character format and was
also higher for the 250-cpm speed compared to 350 or
450 cpm. The adaptability factor did not significantly
affect reading comprehension, and none of the
interactions among factors was significant. Table 1
shows the means and standard deviations of static-
information search scores and leading-display reading
comprehension and Table 2 shows the ANOVA
analysis for the mean comprehension of the RSVP
display.
4. Discussions and Conclusions
displays focused only on assessing the adequacy of
leading display design [1, 2, 4]. However, a leading
display does not usually appear in isolation and is
designed to accompany static information. The effect
of leading display design on visual performance for
static information thus is also an important issue.
According to the analytical results presented here,
none of the leading display design factors distracted
participants from the static search task, but they did
influence comprehension. The impact of leading
display factors on the static search task was relatively
low. This finding suggests that users can receive more
information using a well designed leading display on
small-screen devices without sacrificing visual
performance efficiency on a static-information search
task.
For the leading display, users had the highest
reading comprehension using the word-by-word
presentation mode at a speed of about 250 cpm. This
finding is in accordance with previous Chinese leading-
display studies [1-4, 6, 7]. Chinese script belongs to the
logographic system in which characters are used as
writing units, and no salient boundary exists between
characters. The character-by-character presentation
mode on the leading display required that subjects
partially use cognitive resources to divide the
characters into the smallest meaningful units. However,

159
the word-by-word format allowed the addition of
spaces between words, making the boundaries between
the words more salient; this helped users to divide the
Chinese characters into Chinese words, thereby
lowering the cognitive load and promoting reading
comprehension of dynamic text [2-4]. In terms of user
adaptability to leading displays and speed, previous
studies on leading displays have shown no significant
improvement in reading comprehension with time of
use increments, and 250 cpm is far slower than the
average Chinese reading speed of 580 cpm [5].
Leading displays present a trade-off between space and
time, which is a novel presentation technique to most
users [3, 4]. Thus, users may need more practice to
familiarize themselves with this novel, temporal-
domain, and dynamic technique. Further studies are
needed to investigate how users acquaint themselves
with and accept this novel presentation technique by
increasing time of use to improve reading efficiency for
Chinese text dynamically displayed on a small screen.
The leading of dynamic information displays has
recently been identified as a means of presenting text
on electronic device screens. The results of this study
demonstrate user comprehension of a variety of leading
display designs. Suggestions made in this study may
assist interface designers in developing effective
leading displays that promote improved user
Comprehension for small-screen mobile devices.
[5] F. Sun, M. Morita, and L.W. Stark. Comparative patterns
of reading eye movement in Chinese and English. Perception
and Psychophysics, 37 (6): 502–506, 1985.
[6] A.H. Wang, J.J. Fang, and C.H. Chen. Effects of VDT
leading-display design on visual performance of users in
handling static and dynamic display information dual-tasks.
International Journal of Industrial Ergonomics, 32 (2): 93–
104, 2003.
[7] A.H. Wang and C.H. Chen. Effects of screen type,
Chinese typography, text/background color combination,
speed, and jump length for VDT leading display on users’
reading performance. International Journal of Industrial
Ergonomics, 31 (4): 249–261, 2003.
[8] A.H. Wang and Y.F. Kan. Effects of display type, speed,
and text/background colour-combination of dynamic display
on user’ comprehension for dual tasks in reading static and
dynamic display information. International Journal of
Advanced Manufacturing Technology, 23 (1–2), 133–138,
2004.

5. Acknowledgment
The authors gratefully acknowledge the financial
support from the National Science Council, Taiwan,
Republic of China under Project NSC 97-2221-E-130-
021-.

6. References
[1] C.H. Chen and Y. H. Chien. Effect of dynamic display
and speed of display movement on reading Chinese text
presented on a small screen, Perceptual and Motor Skills,
100: 865–873, 2005.

[2] Y.H. Chien and C.H. Chen. The use of dynamic display to
improve reading comprehension for the small screen of a
wrist watch. Lecture Notes in Computer Science, 4557: 814–
823, 2007.

[3] Y.C. Lin and K.K. Shieh. Reading a dynamic presentation

of Chinese text on a single-line display. Displays, 27 (4–5):
145–152, 2006.

[4] K.K. Shieh and Y.C. Lin. Dynamic Chinese text on a

single-line display: Effects of presentation mode. Perceptual
and Motor Skills, 100: 1021–1035, 2005.

160
 2009 International Conference on Computer Engineering and Technology

Curvature-based Feature Extraction Method for 3D Model Retrieval

Yujie Liu, Xiaolan Yao, Zongmin Li

School of Computer Science and Communication Engineering,
China University of Petroleum (east of China)
Dongying 257061, P.R.China
e-mail: liuyujie@hdpu.edu.cn; yaoxiaolan2002@163.com; lizm@hdpu.edu.cn

Abstract—Curvature on the surface of 3D mesh model is
an important discrete differential geometrical descriptor.
It can show the curving degree very well for those models
with curving pieces and the ones with extreme points or
extension components. In this paper, we use mean
curvature and corresponding coordinates of the vertexes
on the surface as the feature descriptor of model. The
descriptor we defined describes models accurately and
keeps invariant for transformation and rotation.
Additionally, we bring EMD (Earth Moving Distance)
method into the similarity measure frame. This
comparison way is very efficient and especially adapt to
these conditions: different numbers of the two features,
variant or indefinite dimension of the features. Thought
several retrieval experiments, the results implied that this
descriptor could be used to find the exact model from the
PSB model library. Particularly for models which are
made up of curving pieces and those models with extreme
points or parts, the result showed much better than other
feature descriptors.
Keywords—Mean Curvature; Feature descriptor;
EMD; 3D model retrieval
I. INTRODUCTION
Computer graphics technology and some 3D
scanning devices are all improved largely in the past
decades. More and more 3D models appeared.
Meanwhile much more 3D modeling software is
developed significantly and 3D models are not only in
large number but also in complex format of data type.
How to find the exact model to satisfy the practical
needs has become a heated topic. So, 3D model
retrieval has always been an important research issue
during the past decades. It has also been implemented
in many practical areas prevalently. How to describe a
model accurately is the key problem in the retrieval
subject. Generally, the feature we can extract from the
model is to describe the object accurately. Traditional
classification of these model features goes through two
main periods: from the model shape to the content of
3D models.
The shape of the model is fundamental and the
lowest level feature. So there are many methods
extracting features through the models’ surface shape
attribute. Distance or geodesic distance on the surface
[8, 9], area of pieces [13], volume [10], normal
direction [4, 7], and so on, which are all the shape
character. As we known, shape is easy to be received to
humans’ perception directly. Especially, in this paper,
we introduce the curvature on surface of the model to
show the model well, and use it in retrieval further.
In the paper, we present a method based the
curvature on the mesh model surfaces. It got well result
by doing a retrieval experiments on 314 models in PSB.
The paper left parts are organized as follow: Section 2
shows the previous works; Section 3 presents our
feature extraction method in detail; Similarity
measurement is described in Section4 particularly;
Section 5 is our results show and analysis during
experiment process; At last, there is our conclusion in
Section 6.
II. RELATED WORK
3D model retrieval methods are in the increasing
trend in the past years. Many methods are based on
shape of 3D model, which often use the surface
geometric features to describe models. In [8, 9] Osada
proposes a descriptor named shape distribution. He
adopts the shape function of model, which is related to
the geometric characters such as distance of the random
two points on surface, square root of the triangle area
made up from the random three points on surface, cube
root of volume of the random tetrahedron made up from
model and so on. Then he computes values of one of
the functions and gets a statistic to describe the model.
However, this means is not very accurate and maybe
suit to classify models rough. Afterwards, Mahmoudi et
al. make use of curvature function to get seven views of
the model to describe the model in [10]. Similarly they
also compute a statistic about the function values. There
are many other means based on this statistic way to
show characteristic of models, such as the normal
vector on the surface of models [11, 12], histogram
[13], cord-based method [14], and so on[15,16].
So in this paper, we concentrated the geometrical
and topological features on the surface, which show the
shape characteristics directly, to describe the model. So,
we try to introduce curvature into the 3D mesh models.

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DOI 10.1109/ICCET.2009.49
 We used the Meyer method to compute the
curvature of each vertex on the mesh surface, choose
the N biggest curvature vertexes, and combined them
with the standard coordinate position of each vertex to
show the overall model[7]. This descriptor is invariant
to transformation, rotation, projection and scaling under
canonical coordinate system. Especially, for those
models with smooth surface or distinct topological
features, our method shows the better results.
Because of our feature is not only related to the
curvature value, but also based on the coordinate vertex
dates. The conditional similarity comparison method is
not satisfied our needs. In the paper, we applied the
EMD method to measure the similarity between
different models. This distance belongs to the engineer
implementation concept; moreover, it can measure the
two models with different dimension features and
weights. WAN Li-Li used this method to compute
distance between the spatial distributions of different
components in every two models [11]. Features and
weights in this method can be defined up to you. The
speed of the computation is so fast and it is a well
metric tool.
III. CURVATURE FEATURE EXTRACTION
The common used geometrical characters such as
normal vector direction or some other surface angles
information often has simple computation and these
features is not considered as the feature of one object
directly. They are usually known as the assistant tool
combined with other features. In my method, we
compute the differential geometrical attribute—mean
curvature, apply it to describe the model directly.
A. Mean curvature computation
Based on the above principle of the finite element
method, we compute the Voronoi area of the 1-ring
neighborhood triangles on the model surface, and then
compute the mean curvature on each vertex [7].
Firstly, we must use the Voronoi area on surface of
the model. Figure 2 shows the definition of the
Voronoi briefly. For each point P on the triangle mesh
surface, connect the circumcenters of its 1-ring
neighborhood triangles, the area which be constructed
is the Voronoi area of the vertex P.
Secondly, the Voronoi areas depend on the types of
mesh triangles. For the non-obtuse triangle, we can
compute the area by the formulate bellow:
1
∑ (cot α ij + cot βij ) xi − x j
2
ΑVoronoi =
8 j∈N1 (i )
Α
And then we use the mixed area Mixed to compute
the mean curvature vectors by the following formulate:
1
K ( xi ) =
2ΑMixed
∑
j∈N1 ( i )
(cot αij + cotβij )( xi − x j ) (2)
Where N1 (i ) stands for the neighbor triangles of
vertex
xi , and x j is the neighbor of xi , α ij and βij are
xi x j
the opposite angles of the edge respectively,
these angles relationship is shown in figure1.
At last, according to formulate(3), which shows the
x
relationship between the curvature vector of vertex i
and its value, we can get the mean curvature of every
vertex on the triangle mesh surface easily.
K ( xi ) = 2κ H ( xi )n( xi ) (3)
Note: in all above formulates, the bold font means
vector.
B. Our descriptor
Our descriptor is defined by the following
formulate:
F ( M i ) = ( P, κ H ) , i = 1, ⋅⋅⋅n;
P = {( x j , y j , z j )}, j = 1, 2, ⋅⋅⋅, N (4)
κ H = {k j }, j = 1, 2, ⋅⋅⋅, N
Where, F is the descriptor of any model M i in our
library. It is made up of two sets: P and H , the
κ
former is the coordinates of the vertex on surface of the
model, and the latter is mean curvature. In our
experiment, after analysis of all models we choose
from PSB, we set N equals to 50 firstly. So, our
descriptor is the feature in 50 dimensional.
Figure1 shows the curvature result of one face
model with curving pieces, where the left picture
presents the change of the curvature on the model
surface, and the color from red to blue stands for the
curvature value from big to small changing; The right
one shows the vertexes of the whole model, and the
green ones are those feature point we have chosen
according to the curvature value.
(1
)
Figure1. The curvature of one face
162
 Table1 shows the time our descriptor computation P = {( p1 , w p1 ),( p2 , w p 2 ), ⋅⋅⋅,( pm , w pm )}
cost, we only list four models with different size in Let be
vertexes and faces, and record the time spent to
the first signature with m clusters, where
pi is
compute the mean curvature of all vertexes and the N
biggest ones respectively. It is not difficult to see that wpi
representative of the cluster and is the weight of
the largest model s1734 will only spent 2.3 seconds
approximately on computing the N biggest vertexes; Q = {(q1 , wq1 ), (q1 , wq1 ), ⋅⋅⋅, (qn , wqn )}
the cluster;
and for the small size model s0661, it only spent D = [ d ij ]
0.000858 seconds on computing the feature we want. is the second signature with n clusters. And ,
So, this descriptor is suitable to the library with large dij
number of models, because it has the lower time cost the ground distance matrix where is the ground
and feasible to describe different models. All the time q p
listed is for Intel Celeron M 430 1.73 ghz with 512 distance between cluster i and j . The goal of the
RAM. F = [ f ij ] fij
EMD is to find the flow with the flow
TABLE1I. TIMING OF OUR METHOD NEEDED
between
qi and p j that minimize the overall cost [12].

∑ ∑
Model Verte Faces Time(all Time(N m n
i =1 j =1
fi j di j
ID x vertexes) biggest) EMD ( P , Q ) = (5)
∑ ∑
m n
s0661 71 138 0.000796 0.000858 f
i =1 j =1 i j
s0838 1981 3907 0.022880 0.025905
B. Our Measurement
s0961 21013 40384 0.237080 0.279705
In our method, the signature is composed of the
s1734 31649
160940 1.883898 2.267323 mean curvature and the coordinate values for each
8
vertex on the model surface. We recognized the mean
curvature as the weight of the signature, and defined
IV. EMD SIMILARITY MEASURE
the Euclidian distance between the coordinates of
In this section, we introduce the EMD method to vertexes as the ground distance function. Run the
measure our descriptor and then used it into retrieval process of the EMD comparison, the dissimilarity
experiment. between the different models can be valued and
Traditional comparison method uses Euclidian computed fast.
distance, Minkowski distance，Hausdorff distance and In the experiment, we defined that the
so on. SIG_MAX_SIZE equals 100. Besides our descriptor is
These means are limited to the dimension of chosen the N biggest mean curvature, it needs to make
features and related to the numbers of the feature sure the N will be smaller than 100. If we need to deal
points in two models will be compared. So, for the with other large mesh models, we can change these
sake of avoiding this problem, we adopt the earth constraints and make the SIG_MAX_SIZE much
mover’s distance (EMD) means common used in larger.
engineering implementation in our method.
V. EXPERIMENT RESULT
A. Method Presentation
Our experiment is based on the triangle mesh
Earth mover’s distance can be dating back to the models in PSB benchmark database [13]. We choose
transport problem, and then be implemented in many 314 models from the 1814 models and it contains 47
areas. It can evaluate the dissimilarity between two categories and 37 non-empty classes.
multi-dimensional distribution in some features space After the curvature computation, we applied the
where distance measure between single features, which EMD method to compare the similarity and use the
we call the ground distance is given [12]. model in this database to finish the retrieval process.
Intuitively, given two distributions, one can be seen The evaluation tool we used is also in the PSB
as a mass of earth properly spread in space, the other as benchmark. The experiment results were compared to
a collection of holes in that same space. Then, the the Shape Distribution and EGI descriptors. It
EMD measures the least amount of work needed to fill improved that our method got the well result,
the holes with earth. Here, a unit of work corresponds especially for those models with curving pieces and
to transporting a unit of earth by a unit of ground those with extreme points or parts, the retrieval result
distance. shows much better.

163
A. Retrieval result
The figure 2 shows part of our results for five
models: head, feline, spider, bottle, and face. The first
model plot is the model we want to retrieval, and the
left three are the results showed in order of ascending
dissimilarity.
Figure2. Retrieval results
We don’t list all the retrieval result, because each
result all contains 314 models and they are in great
number. By the Precision-recall diagram we can
compute the precision and recall for each retrieval with
all the models listed in the retrieval results.
While we compute the P-R curves, we compare our
method with the Shape distribution and EGI (Extend
Gaussian image) respectively and get one plot showed
in Fig8 below.
B. The whole efficiency
Applied all the 314 models to do the retrieval
experiment, got the precision-recall plot of all the
models and compared it with the other two methods,
the result showed that our descriptor is much more
excellent than the other two.
Figure3. Average precision and recall of all classes
164
Figure4. Average precision and recall of all models
From the precision-recall plots of all models (fig. 3)
and all classes (fig. 4) in our experiment, it is notable
that the red line is on above of the green and blue ones.
We can improve the mean curvature descriptor has
higher retrieval quality than shape distribution and EGI
methods.
VI. CONCLUSION
In this paper, we propose a new descriptor for the
3D triangle mesh model. The descriptor combined the
vertexes’ mean curvature on the surface of the model
and the coordinate position of these vertexes. Using
EMD method to compare features we computed in our
experiments, we found that the descriptor showed well
results and especially for those models with curving
pieces and those with extreme points or extending
parts.
The biggest virtue of our method is to describe the
curing pieces and the extreme points or parts of the
model accurately. Moreover, through the retrieval
experiment, our method got well results. On the other
hand, the weakness of the descriptor is that it does not
adapt to the complex models or other models with
plain pieces.
In our future work, we can continue to express the
model with other differential geometrical operators
such as Gaussian curvature, normal curvature and
direction on the mesh surface. We can also combine it
with other features to get much better feature
expression and try to solve the inaccurate description
for other general models with plain pieces.
ACKNOWLEDGEMENTS
This work is supported by National Key Basic
Research Program of China (2004CB318006) and
National Natural Science Foundation of China
(60533090, 60573154).
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 2009 International Conference on Computer Engineering and Technology

A New Method for Vertical Handoff between WLANs and UMTS in

Boundary Conditions
Majid Fouladiana, Faramarz Hendessic, Alireza Shafieinejadc,
Morteza Rahimib, Mahdi M.Bayatc
a
Azad Islamic University, Saveh, Iran c , b Azad Islamic University, Qazvin, Iran
c
Dept. of Electrical and Computer Engineering Isfahan University of Technology, Iran
majidfoladi@yahoo.com, hendessi@cc.iut.ac.ir, a.shafieinejad@ec.iut.ac.ir
m18.rahimi@gmail.com mahdimbayat@gmail.com

Abstract estimated by the received signal strength of each cell.

We propose a new vertical handoff method in which
the number of signaling and registration processes is
lowered as a result of reduced utilization of home
agent and correspondent nodes while users are mobile.
Throughput performance of the integrated network is
improved by reducing handoff delays and packet
losses. Special conditions such as boundary condition
also considered. Performance and delay evaluations
which done by ns-2 are presented to demonstrate the
effectiveness of the proposed scheme.
1. Introduction
Nonhomogenous networks contain a number of
wired networks with different technologies. The need
for transparent services to end users makes vertical
handoffs between these networks unavoidable.
Generally, there are two categories of wireless
networks, those such as 3G and 2.5G which provide
low bandwidth services over a wide geographical area,
and those such as WLANs which provide high
bandwidth services over a small geographical area.
One of the most challenging problems in these
networks is QoS which must not experience significant
changes during the handoff process. Thus we need to
employ an efficient algorithm for handoff in terms of
initializing and completing the handoff operation [1,2].
Vertical handoff can be divided into upward and
downward cases [1]. In the downward case, the mobile
node (MN) enters a smaller cell from a larger one such
as 3G to WLAN, and in upward case, the MN enters a
larger cell form a smaller one (i.e. WLAN to 3G).
Downward handoff usually has low delay sensitivity
while in the upward case since the MN switches to a
cellular network before disconnecting from the WLAN
delay is not a serious problems. However, in upward
handoff the MN tries to maintain its connection with
the WLAN because of better QoS and lower cost.
In horizontal handoff, the only parameter for
selecting a network is channel quality which is
In the vertical case, in addition to channel quality,
other parameters such as service cost, bandwidth,
network delay, traffic, MN speed and movement
patterns, must be considered in selecting a suitable
network. After selecting the proper network, the MN
initiates the handoff and connects to the target network.
The handoff process mainly depends on the network
architecture and its hardware and software elements.
Different algorithms and protocols have been proposed
for this process. Management of handoff protocols in
the network layer is based on MIP1, which introduced
two new elements, Home Agent and Foreign Agent to
address the problems of mobility [3,4]. These agents
provide synchronization, updating and authorization of
user connections and the CoA of the foreign network.
1.1 Nonhomogenous integrated networks
ETSI has proposed two methods for connecting
WLANs and 3G networks, loose and tight, as depicted
in Fig. 1.
Fig. 1: Tight and loose coupling architectures.
In a loose connection, a separate path is provided for
3G traffic, thus the WLAN traffic is not transferred via
the 3G network and routing to the Internet is via the
WLAN gateway. In this case, 3G networks and
WLANs can employ different protocols for
authentication and accounting [5]. In a tight
1
-Mobile IP

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DOI 10.1109/ICCET.2009.166
connection, a WLAN is connected to the 3G network
in the same way as other radio access networks. It
becomes a part of the 3G core and all WLAN traffic is
transferred via 3G networks.
2. The Proposed method
In our approach the mobile node (MN) locations in
the UMTS network are estimated as the MN nears the
WLAN. At an appropriate time, a message is sent to
the MN to turn on its WLAN interface. In this scheme
the UMTS’s SGSNs are considered as FA and also the
WLAN’s APs in each SGSN are considered as a B
Node for that SGSN. While entering a WLAN from a
UMTS network, the MN gets a new IP address from
the WLAN.
After assigning the new address, all of the packets
from the CN with the old IP address, which is valid for
the UMTS network, must be tunneled to the MN by
means of this new IP address. On the other hand all
packets from the MN to the CN must be sent using the
old IP address, routed to the AP, then UMTS, and
finally forwarded to the MN. We first describe the
process of turning on the WLAN interface, and then
we explain the details of our protocol for upward and
downward ones.
We estimate the location of the MN using 3GPP
methods which can be categorized in four major
groups [6]:
Cell identifier
OTDOA2
UTDOA3
GPS4
Regardless of the method employed, while
connecting to the MN, the CN requests the MN
position from the SRNC via the Iu interface. The
SRNC collects the required information from the
LMU, evaluates the MN position and finally responds
to the CN [7]. Fig. 2 shows UMTS/WLAN vertical
handoff architecture
Fig. 2: UMTS/WLAN vertical handoff architecture.
2 old MN IP address (which is valid in the UMTS
-Observed Time Difference Of Arrival
3
-Uplink Time Difference Of Arrival
4
-Global Positioning System
167
In our scheme, we employ a database of user and
WLAN information, denoted the WUD to RNC unit.
The WLAN WUD contains the geographical position
and general specifications of the networks such as
bandwidth, delay and cost, while for users we store an
access and priority list of those WLANs a user can
connect to. As mentioned above, when a CN requests
the position of a MN from the SRNC, the SRNC
computes the approximate position of the MN and
compares it with that stored in the WUD. If the
difference is below the threshold (Rth), a message is
sent to the MN to turn on its interface. A comparison
can be done based on the inequality
( x − x 0 ) 2 + ( y − y 0 ) 2 < ( R − R th ) 2 (1 )
where ( x, y ) , ( x0 , y0 ) , R and Rth show the current
MN position, center of the WLAN, radius of the
WLAN, and the threshold, respectively.
In this method, upward and downward handoffs
differ because it is not possible to use 3GPP estimation
when the MN is in a WLAN network. In the next
sections, we explain the upward and downward
handoff techniques.
2.1 Downward handoff
We assume the MN is connected to the CN via Node
B and moves in the UMTS network. The SRNC
regularly evaluates the MN position and relays this
information to the CN. If the MN nears a WLAN
boundary, the SRNC signals the MN. The MN turns on
its WLAN interface and searches for advertisement
messages from nearby WLAN APs. Next, the MN
determines if there is an AP with better quality than the
UMTS network according to a weighted function [7] of
the AP parameters.
⎡ ⎤
f n = W n = ∑ ⎢ ∏ E sn, i ⎥ ∑ f s , j ( w s , j ) N (Q sn, j ) (2)
s ⎣ i ⎦ j
If the MN does not find a suitable AP with a cost
function better than current UTMS, it ignores the
handoff process and continues its current connection.
Otherwise, the handoff process shown in Fig. 3 is
initiated. The AP sends the registration request which
contains the new CoA address to the FA, and then the
FA registers the MN and responds to the AP. In
addition, the FA sends a binding update message to
Node B to cleanup the allocated resources. Afterwards,
the FA is informed of the new IP address of the MN
(which is valid in the AP domain).
The received packets from the CN which have the
domain), are tunneled to the new IP address by the FA.
Since the source address of the packets from the MN to
the CN is the old MN address, these packets must first 2.2 Upward handoff
be routed to the AP through the new IP address and
then forwarded to the CN. We consider the MN is connected to the CN in the
WLAN network. In this case, only one upward handoff
is required. While the MN is in the WLAN network
and connected to the CN via the AP, it must
continuously measure the strength of the received
signal and compare it with the threshold Sth. If the
measured strength is below Sth for at least a threshold
interval (Tth), the MN will send a handoff request to
Node B. If the system rejects the request, the MN must
find another network, otherwise it will be
disconnected. At the beginning of the process, Node B
forwards the request to the FA which registers the MN
and then informs Node B. Then all packets sent
between the MN and CN are transmitted through Node
B. Moreover, the FA sends a binding update message
to the AP to cleanup the allocated resources. This
process is illustrated in Fig.5, 6.

Fig. 3: The UMTS to WLAN handoff process.

Fig.5: The WLAN to UMTS handoff process

Fig. 4: Signal flow for downward handoff
M4 Sending target network information, M5: MN
signaling flow, M11-13: Data path for the neighboring
WLAN, M14-15: Data path from the MN to CN.

168

Fig. 6: Signaling flow in an upward handoff starting in the
WLAN, M1-4: Data path between the CN and MN before handoff.
2.3 Boundary conditions
In this section, we consider the case where the
WLAN is not covered by one Node B. For the situation
where the WLAN is covered by two Node Bs, we
identify two cases based on the number of FAs related
to these Node Bs, i.e., the Node Bs have a common FA
or different FAs. For these cases the downward
handoff is the same as in Section 2.1 and upward
handoff is the same as in Section 2.2, but when the
connection begins in a UMTS network, it is different
than in Section 2.2. This is discussed in detail below.
2.3.1 Two Node Bs under one FA
According to Fig.7 , the MN connects to the CN via
the first Node B and then with a downward handoff
enters the WLAN and accesses the CN through the AP.
Then the MN exits from the WLAN and enters the
second Node B domain. This process is the same as in
Section 2.2 in spite of the existence of two different
Node Bs. Moreover, the protocol is completely
transparent to the HA and CN which do not need to
know the new IP address of the MN. This part of the
protocol was simulated and the results show an
improvement in handoff delay.
169
Fig. 7: Special handoff when the source and destination FAs are
similar.
Fig.8: Signaling for upward handoff when the source and
destination FAs are similar,M1-5: Data path between the CN and
MN before handoff.
2.3.2 The case of two Node Bs under two
FAs
When a Node B is under coverage of two different
FA (fig. 9) before and after of handoff, while in
WLAN and the received signal strength is below SRth,
MN received advertisement message from B Node2
and then send handoff request to B Node2, this
message must be forwarded along a path B Node2-FA2-
FA1. This process needs a registeration in CN which
causes more delay than previous handoffs. Thus for
reducing the effects of packet loss, we offer propose
that FA1 routed the packets with MN destination via B
Node1 and in addition send a copy of them to FA2.
As the strength of received signal form AP in MN
become less than treshold, MN sends registeration
request to B Node2 which in its turn forwarded to FA2
and then HA repectively. Affer registeration in HA,
responses was returned to FA2 and then to B Node2.
For correcting MN address in packets from CN, HA
must send updating message to CN too. After this
message valid addresss of MN is the one which was
got from FA2.
For cleanup the resources allocated for MN in B
Node1, FA2 send a connection updating message to
FA1 which in its turn forwarded to B Node1. After
receiving updating message, FA1 must send all of
packets with MN destination to FA2 until CN registers
the new address of MN. This process is illustrated in
Fig. 10. To eliminate the problem of packet lost during
handoff delay, MN sends a message containing the ID
of the last received packet to FA2 via B Node2 which
make it possible for FA2 to forward the received packet
with MN destination to MN. Thus the packet lost
during handoff is minimized.
Fig. 10: Upward handoff signaling for different source and
destination FAs,M1-5: Data path between the CN and MN before
handoff.
Fig. 11: Downward handoff delay in tight coupling.
170
3. Performance Results
We use NS-2 for simulation [8]. IEEE 802.11b was
selected for the WLAN in addition to a UMTS
network. Handoff delay is computed as the difference
in time between receipt of the last packet in the
previous network and the first packet in the new
network.
Figs. 11 and 12 show the delay in downward
handoff. Fig.11 shows a tight architecure between the
AP and SGSN. An improvement was achieved relative
to the MIP protocol. Similar results are shown in
Fig.12 in which thw FA and AP do not have a wired
connection so packets between them are routed through
the Internet. In this case, the Internet delay was
selected randomly between 0 and 55 ms.
For upward handoff, both tight and loose connections
were considered, First the MN initiates connection
with the CN in the WLAN and then enters the UMTS
network. These results are shown in Figs. 13 and 14 for
tight and loose connections, repectively, Second, the
MN connects to the CN via the SGSN address and
enters the WLAN. Then it initiates handoff to UMTS
during the connection with the CN. It is not necessary
to register in this case and a reconnect request to SGSN
is sufficent. Figs.
The above results considered a WLAN that is
completly located within a UMTS network. The results
where the WLAN is located between two Node Bs
under one FA are shown in Figs. 15 and 16 for tight
and loose connections, respectively.
Fig. 12: Downward handoff delay in loose coupling.
 Fig. 13: Upward handoff delay in tight coupling
Fig. 15: Upward handoff delay in tight coupling in the case of two
Node Bs with one FA.
4. Conclusions
We proposed a new vertical handoff method which
does not require any modifications of the WLAN
structure. The only change is in UMTS where SGSN
must tunnel packets to the MN after receiving the
new MN address. While in the WLAN, the MN has
two addresses, the UMTS address which is used as
the source and destination of transmitted packets, and
the CoA address which belongs to the WLAN and is
used for routing packets to the CN via the AP while
in the WLAN. Special conditions such as boundary
condition also considered This method reduces the
interaction between the CN and HN. The processing
time and handoff delay were both reduced, in both
coupling tight ones which AP has wired connection
with SGSN was considerably reduced and cause a
higher network throughput.
5. References
[1] Zhang, Q., Guo, C., Guo, Z., and Zhu, W.,
“Efficient mobility management for vertical handoff
171
Fig 14: Upward handoff delay in loose coupling
Fig. 16: Upward handoff delay in loose coupling in the case of two Node
Bs with one FA.
between WWAN and WLAN, ,” IEEE Commun.
Mag., vol. 41, pp.102-108, 2003.
[2] Siddiqui,F.,Zeadally,S.,“Mobility management
across hybrid wireless networks: Trends and
challenges,” Computer Commun., vol. 29, no. 9, pp.
1363–1385, May 2006.
[3] Perkins, C. and R. Glenn, “IP Mobility Support
for IPv4”, RFC 2404, Aug. 2002
[4] El Malki, K., et al., Low Latency Handoffs in
Mobile IPv4,Internet Draft, Aug. 2005.
[5] Salkintzis, A.K., “Interworking technique and
architecture for WLAN/3G integration toward 4G
mobile data networks”, IEEE Wireless Commun., vol.
11, no.3, pp. 50-61, June 2004.
[6] 3GPP TS25.331, Radio Resource Control (RRC):
Protocol Specification, Version 6.6.0, June 2005.
[7] McNair, J., and Zhu, F., “Vertical handoff in
fourth-generation multinetwork environments”, IEEE
Wireless Commun., vol. 11, no. 3, pp. 8-15, June
2004.
[8] Network simulator. URL http://www.isi.edu/
nsnam /ns/.
 2009 International Conference on Computer Engineering and Technology
Research on Secure Key Techniques of Trustworthy Distributed System
Ming He
1.Institute Command Automation, PLA Aiqun Hu
Science and Technology University; Information Science and Engineering
2.Information Science and Engineering School,
Institute, Southeast University, Southeast University,
Nanjing,China Nanjing, China
e-mail: blue_horse@126.com e-mail: aqhu@seu.edu.cn
Abstract— To arrive at the goal of intensifying the
trustworthiness and controllability of distributed systems, the
core function of secure algorithms and chips should be fully
exerted. Through building the trustworthy model between
distributed system and user behaviors, constructing the
architecture of trustworthiness distributed systems,
intensifying the survivability of services, and strengthening the
manageability of distributed systems, the secure problem of
distributed systems is to be radically solved. By setting up the
trustworthy computing circumstance and supplying the
trustworthy validation and the active protection based on
identity and behavior for trustworthy distributed system, we
will reach the goal of defending the unaware viruses and
inbreak. This research insists that the security, controllability,
manageability, and survivability should be basic properties of
a trustworthy distributed system. The key ideas and techniques
involved in these properties are studied, and recent
developments and progresses are surveyed. At the same time,
the technical trends and challenges are briefly discussed.
Keywords-trustworthy distributed system; trustworthiness;
controllability; survivability
I. INTRODUCTION
At present, the techniques of distributed system are
excessive, miscellaneous, and the cost of implementation is
great. The influence on the performance of distributed
system is increasingly complex. The overstaffed abuse is
gradually revealed. For example:
1) distributed system is built on the insecure terminal
system
The most prominent secure problem of terminal system is
prone to suffer from the erosion by the worm virus and
Trojan horse. Because the bulk of terminal systems do not
adopt enough safety precautions, some important programs
and files will be destroyed. Moreover, other goal systems
will be attacked by the worm virus and Trojan horse.
Therefore, these will lead to the drop of performance of the
whole distributed system. Distributed system is confronted
with the serious crisis of trust[1].
2) distributed system is scare of the trusted safeguard
measures
Practices indicate that the worm virus and Trojan horse
could not be kept out with firewall, IDS, and antivirus
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 172
DOI 10.1109/ICCET.2009.100
Hangping Qiu
Institute Command Automation,
PLA Science and Technology
University,
Nanjing, China;
e-mail: qiuhp8887@vip.sina.com
software. On the one hand, these products make high
requirement for administrators. A few users are able to fall
short of this kind of demand. On the other hand, as far as the
architecture is concerned, these products are extra add-ons.
They ascribe the passive defensive forms and can not cope
with the secure menace which is increasingly variational.
The secure problems of distributed system need be solved
and provide more reliably and simply controllable means to
construct the trustworthy environment [2].
3) distributed system is devoid of controllability and
manageability, most remote-behavior is unpredictable
At the present time, capability of distributed systems is
insufficient in mangy conditions such as user behaviors,
run-states, controllability and manageability of system
resource. At the same time, those capabilities are absolutely
necessary not only for the security of distributed systems
but also for health and continuance of development.
Therefore, new ideas are needed to resolve the problems
such as security and function of distributed systems. To
arrive at the goal of intensifying the trustworthiness and
controllability of distributed systems, through building the
trustworthy model between the distributed system and user
behaviors, constructing the architecture of trustworthiness
distributed systems, intensifying the survivability of
services, and strengthening the manageability of distributed
systems, the core function of secure algorithms and chips
should be fully exerted. The secure problem of distributed
systems including protection, creditability and
manageability is to be radically solved. The security of
service of distributed systems is improved effectively and
development of Electronic Commerce and Electronic
Government is promoted healthily and fleetly.
In this paper, we present a new scheme for constructing a
trustworthy distributed system and make research on secure key
techniques. In section1, we analyse the situation of security in
distributed systems and points out the necessity to build
trustworthy distributed systems. In section 2ˈwe introduce
the notion of trustworthy distributed system. Through
building the trustworthy model between the distributed
system and user behaviors, constructing the architecture of
trustworthiness distributed systems, intensifying the
survivability of services, and strengthening the
manageability of distributed systems, the security problem of
distributed systems is radically resolved in section 3. The
significance of this paper is present in section 4.
II. OVERVIEW OF TRUSTWORTHY DISTRIBUTED SYSTEMS
At the present time, there are different cognitions to the
trustworthy distributed systems: based on dependable
authentication, based on conformity of exiting security
technologies, based on trustworthiness of distributed system,
based on trustworthiness of service supported by distributed
system. The divergence to opinion may result in illegibility
of definition on trustworthy distributed system and
aggrandize difficulty of estimation for maneuverability of
resolvent, as well as difference between requirement and
development is magnified.
In this paper, distributed system, user behavior and its
result is divinable and controllable in the trustworthy
distributed system. Status of behavior can be supervised and
unconventional action can be managed, while aftereffect of
action can be estimated. In another word, the trustworthiness
of distributed system has a set of attributes that security and
survivability must be ensured in user’s view and the
manageability of distributed system must be supported in
designer’s view. Around maintenance of trustworthiness and
manageability of behavior between components those three
attributes of trustworthy distributed system can be
amalgamated to arrive at the goal of trustworthiness of
distributed system, while the conceptions of security,
survivability, controllability and manageability are
decentralized and isolating in traditional sense.
Unlike conventional researches on security, survivability
and controllability, which are isolated and separated, the
three properties are closely related in trustworthy distributed
system, and are formed an organic whole around
trustworthiness maintenance and behavior control, which are
divided into three parties, input, process and output, as
depicted in Figure 1.
Figure 1. Trustworthiness maintenance and behavior control.
The role of security is to reduce vulnerabilities on the
chain of trustworthiness gathering, spreading and processing.
Survivability can be thought of as a special control
mechanism, i.e. resources management and schedule under
the circumstance of existing misbehaviors. Controllability
provides detailed mechanisms to monitor systems states and
control misbehaviors. Trustworthiness information is
collected by several methods, such as behaviors analysis,
states detection and correlation computing. Then,
trustworthiness information is stored on efficient format for
173
quickly querying and updating, and spread to corresponding
components for correlation computing.
To complete trustworthiness of distributed system, there
are four problems: 1) Trustworthiness of remote user:
trustworthiness of user is the precondition of security in
distributed system and needs identity authentication terminal
system so that termination and user can be controlled
separately by distributed system. 2) Trustworthiness of
remote platform: trustworthiness of remote platform contains
trustworthiness of identity and computing environment.
Distributed application is secure only when the platform is
credible, otherwise, the remote node may be counterfeit or
controlled by Trojan horse. 3) Trustworthiness of remote
task: when the distributed application id executed, it must be
partitioned to several independent physical modules which
need to verify identity and trustworthiness of behavior each
other. In that way, initiator of task can affirm that the task
has been done inerrably. 4) Trustworthiness of remote action:
comparing with trustworthiness of task, trustworthiness of
action requires to forbid some actions and to restrict activity
of remote user. Controllability of remote action refers to
distribution and controllability of authority. To achieve the
security strategy of whole system, authority of termination
must be restricted. The last two problems are more difficult.
III. KEY TECHNOLOGY OF SECURITY
Around algorithm, technology and production of the
distributed system, this research has four steps: (1) a
trustworthy model of distributed system and user behavior is
present based on exiting security technology through
analyzing requirement of trustworthy distributed system. (2)
a core chip of security is designed and completed. (3) an
archetype is built to intensify security of trustworthy
distributed system. (4) an estimation theory of trustiness is
present to verify the archetype. The first is the main step in
the research that the effect to security of distributed system
is studied according to the characteristic of user behavior.
The study on security chip is the foundation of substrate
hardware. All the study is based on the second step. The
perfect archetype is built through amalgamating pivotal
security technologies such as security arithmetic, security
protocol and intrusion prevention. The third step is the
difficulty in the whole study. The last step is the soul of all
the study in which the archetype is verified using
trustworthiness estimation to prefect mechanism of
trustworthiness of distributed system.
A. Trustworthy model
Trustworthy model is the pivotal process in
development of system. In the TCSEC of Us Department of
Defense formal description, verification and covert channel
analysis is needed form Standard B. How to build
trustworthy model which analyzes distributed system and
user behavior availably is the precondition to study the
trustworthy distributed system. From identity, behavior,
content and computing environment, the foundational
method which verifies system trustworthiness is studied
through amalgamating method of verifying trustworthiness
and traditional Take-Grand model, introducing trusted
subject, restricting acquirement and authorization used only
by trusted subjects and adding regulation to verify
trustworthiness in model, as a result trustworthy model of
distributed system and user behavior is built. There are two
primary advantages in that model. Firstly security
vulnerability of system can be analyzed by mathematics
model because the requirement of trustworthiness of system
is described abstractly and exactly without implementation
details; secondly security trustworthiness of system is
improved by the use of formal description and verification.
Figure 2. a kind of analysis model of trustworthy model.
The elements in analysis of trustworthiness are present in
fig.2. Estimation of trustworthiness of behavior contains
trustworthiness of behavior and identity, while the latter is
based on trustworthiness of content such as capability of
protection and service, recommendation of trust and record
of behavior.
The evaluation of user's behavior is comprehensive
evaluation for user's behavior, which is reflected by user's
past behaviors. In order to get evaluation result effectively,
we first subdivide user's behavior into behavior attributes,
such as security behavior attribute and performance behavior
attribute. Then we subdivided behavior attributes once again
into more small data unit, namely behavior evidences. We
use the layered, subdivided and quantitative idea to convert
the complicated and comprehensive evaluation of user's
behavior into the measurable and computable evaluation of
behavior evidences. Thus the evaluation of user's behavior in
the trustworthy distributed system can be solved effectively.
So this method is feasible in engineering.
B. Secure Kernel chip
It is the foundation to trustworthy distributed system on
hardware that SOC technology is adopted to design secure
kernel chip, also security algorithm and key storage are
completed inside chip. A security chip which can be applied
to different cryptographic algorithms is needed imminently
because of different applications with dissimilar
cryptographic algorithms. In this paper, a secure kernel chip
SOC which is constituted with kernel of RISC and multi-
processor is designed, also that chip measures up to the
standard of TPM. The chip can be applied to different
cryptographic algorithms while the cost and efficiency of
cryptographic algorithm are affected lightly.
C. Enhanced security architecture
Consulting exiting architectures of security operation
system this paper puts forward the architecture of
174
trustworthy termination system based on building
trustworthy model through secure kernel chip plays an
important role in the controllability of security system, then
enhanced security architecture is built in trustworthy
distributed system. That architecture, which realizes
trustworthy access controllability and supports flexible
security strategy, can ensure the security of mechanism.
(1) Enhanced security architecture based on P2P
There is not a central manager in the architecture of P2P
to manage nodes and users. Therefore, their trustworthiness
of identity must be authenticated by the trusted third part
(usually it is the center authentication, CA) and every node
can authenticate destination node through proof supported
by trusted third part (usually it is the signature for certificate
of platform and user identity by CA).
Furthermore, secure kernel chip (SKC) supports
authentication for trusted proxy in own node and verifies its
trustworthiness when the computer system starts. Every
node can verify trustworthiness of trusted proxy on the
platform of destination through EK after destination is
verified through authentication with signature of CA, since
authentication of platform contains its authenticated key.
(2) Enhanced security architecture based on C/S
In distributed system based on topology of P2P
authentication to destination is needed to verify particularly
in the light of trusted model due to the lack of uniform view
of authentication. The latter behavior is safe in the view of
the owner of authority but unsafe in the view of other node,
because other node may be controlled through the
authorization once it is accredited to destination node.
This problem is avoided easily in distributed system
based on C/S for database management of user and authority
is centralized and the view of authorization is consistent for
all nodes. If authorization is distributed only by server and
client must request authority from server, this problem is
solved easily.
In the architecture based on C/S, user’s information
including authorization and identity is managed by a trusted
centralized management platform exiting in the server,
while a trusted proxy is disposed on the client. The server
verifies identity of client through authorization with its
signature. Authorization of client encapsulated in SKC with
encryption is requested from server which supports the key
of decryption. All the authorization of client must be
distributed by server and client has no prerogative to
authorize any authorization to other client. Therefore, server
can know which authorization the client has at any moment
and grant or retract some authorization to client.
A simple architecture of trustworthy distributed system
which avoids disadvantage like traditional affixed security
mechanism is present in fig.3. To research architecture of
trustworthy distributed system, there is one point to be
realized: physical positions of nodes in distributed system
are dispersed, so that security service is consistent in the
view of system.

Figure 3. A architecture sketch map of trustworthy distributed system.
Compared with traditional security architecture, security
controllability is applied in enhanced architecture of
trustworthy distributed system in four aspects: 1) security of
password: using secure kernel chip of terminal system,
hardware-encrypted platform like TCP is completed.
Security algorism of key is ensured while encryption or
decryption of data and digital signature authentication are
completed, profiting from flexible ability of chip to support
security algorism, perfect machination of key management
and security defense. 2) security of identity authentication: it
is no possibility that counterfeit user can operate system
illegally because security level is heightened and
certification has three steps “USB Key + password + biology
keystroke characteristic” , also the computer is used
exclusively. Based on the certification technology PKI, the
identity of platform is safe and the risk of defense that
counterfeit system intrudes distributed system is reduced,
since the identity of platform is the unique sequence code of
secure kernel chip with 64-bit in enhanced security system. 3)
security of permission assignment: in enhanced security
system inner process scheduling and outer process access is
verified strictly under the instruction of trustworthy model so
that permission of system is insured to be safe. 4) security of
access control: by the trusted monitor in enhanced terminal
system, trustworthiness of any access is verified to insure the
trustworthiness of identity, behavior and computing
environment in the access from subject to object. 5) security
of system audit: all kinds of behavior or event logged in
enhanced security system support history of events and
supervise afterwards. Based on the enhanced terminal system,
taking advantage of C/S architecture and simplifying strategy
of authority controllability, the whole view of authority
controllability is built on the server. Monitoring of client
behavior is completed using trusted proxy of client. After
those measures, security management in the whole
distributed system is put into effect to every host and user.
D. Evaluation theory of trustworthiness
It is not possible to build a perfect trustworthy distributed
system, so the quantitatively trustworthiness estimation of
distributed system is valuable. Quantitative research on
trustworthiness could find the weakness and risk in the
distributed system and improve them. Quantifying of
trustworthiness is also at exploring stage.
Trustworthiness evaluation is performed based on some
preconfigured models, with logic instructions generated to
drive particular control actions, such as forwarding control,
attack prewarning, survivability control, and immunity [8].
175
Therefore, evaluation theory of trustworthiness which
contains estimation of security, survivability and
manageability is the precondition to monitor and revises
system. It is also insurances of performance of the whole
trusted distributed system. Though there is no absolute safe
system, the ultimate goal of estimation trustworthiness of
distributed system is not to remove weakness completely but
support a scheme to balance service and security for
administrator and a measure to defense attack actively such
as that through building mechanism to describe behavior of
attack, aggressive behavior is taken out from plentiful
normal behaviors and access controllability of terminal
system is completed. Emphasizing architecture and
quantitative analysis on survivability of distributed system
based on the transform between problem and space,
estimation of trustworthiness is transformed to a method
frame of classic problem.
IV. CONCLUSIONS
Trustworthiness is an important aspect to the study on
distributed system. In that paper, trusted computing
technology and trustworthiness management technology of
networks are integrated to resolve the conjuncture with
distributed system and enhance the ability to dispose states
dynamically so that enhanced trustworthy security system is
built. That system supports a basal strategy to intelligent
self-adaptive controllability of system security and service
quality. The study in that paper hasn’t published at present
at home and abroad. The prominent theoretical significance
and practice value is present as follows:
1) It is a helpful reference to design and complete the
security system and it is also a direction for study on theory
and method of security controllability in distributed system.
2) User’s controllability to distributed system is enhanced
to insure that system authentication is used exactly and
reduce the risk of system out of control bring by virus and
Trojan horse.
3) Security can be separated from management. Enhanced
security of distributed system is helpful for increasing
authentication of security laws and deterrence to hacker.
4) Based on trustworthiness of distributed system, the
trustworthiness of network and grid system is improved and
the development of remote cooperation and distributed
application is accelerated.
REFERENCES
[1] Cyber Trust [EB/OL]. http://www.nap.edu/catalog/6161.html,2006.
[2] LIN Chuang, REN Fengyuan. New network with trustworthiness and
controllability as well as expansibility[J]. Software Journal, 2004,
15(12): 1815-1821.
[3] LIN Chuang, PENG Xuehai. Research on network architecture with
trustworthiness and controllability[J]. Journal of Computer Science
and Technology, 2006, 21(5): 732-739.
[4] LIN Chuang, PENG Xuehai. Research on trustworthiness of
network[J]. Computer Journal, 2005, 28(5): 751-758.
[5] LIN Chuang, WANG Yang, LI Linquan. Random model method and
evaluation technology of network security[J]. Computer Journal,
2005, 28(12): 1943-1956.
[6] TIAN Junfeng, XIAO Bing, MA Xiaoxue, et.al. Trustworthy model [7] SHEN Changxiang, ZHANG Huanguo, FENG Dengguo, et.al,
and analysis in TDDSS[J]. Computer research and development. 2007, Summarazation of information security[J]. China Science:
44(4): 598-605. E(Information Science), 2007, 37(2): 129-150.
[8] Chuang Lin, Xue-hai Peng, Research on Trustworthy Networks,
China Journal of Computer, May, 2005, 28(5): 751-758,

176
 2009 International Conference on Computer Engineering and Technology
WebELS: A Multimedia E-Learning Platform for Non-broadband Users
Zheng HE, Jingxia YUE and Haruki UENO
National Institute of Informatics, Japan
hezheng@nii.ac.jp
Abstract
In this paper, a multimedia-support e-learning
platform called WebELS (Web-based E-Learning
System) for non-broadband users is introduced. The
system includes three main modules: editor, player
and manager. Both asynchronous and synchronous
e-learning are supported on this platform. Moreover,
some technologies are implemented to make the
multimedia contents possible for users without
wideband network at hand: the multimedia content
was converted into slide serials synchronized with
audio and cursor action, the multimedia content
download is streaming-like, and an offline player
was embedded into the multimedia content.
1. Introduction
The rapid development of computer and Internet
technologies has made e-learning become an important
education method. One of the most attractive features
of e-learning is its capability to integrate different
media such as text, picture, audio and video to create
multimedia content, which can provide flexible
information associated with instructional design and
authoring skills to motivate the learning interests and
willingness inside students. [1-3]
Generally, to support multimedia contents there are
three things that a network has to provide: bandwidth,
consistent quality of service and multipoint packet
delivery. Today's multimedia applications have a wide
range of bandwidth requirements: 64kbps for telephone
quality audio, 128kbps to 1Mbps for video conference,
1.54Mbps for MPEG video, etc. Therefore, for users in
non-broadband areas or without wideband network at
hand, the multimedia e-learning with acceptable quality
seems to be an impossible task. [4]
WebELS system presented in this paper provides a
multimedia e-learning platform for non-broadband
users. It includes three main modules: editor, player
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.108
and manager. Some technologies are implemented to
make multimedia contents possible for users without
wideband network: image-based slide synchronized
with audio and cursor movement, media streaming-like
downloading and offline player embedded in the
downloaded content. So that teachers can create or edit
their own multimedia content by transferring or
integrating raw materials, then achieve the contents on
WebELS server and manage course list or access
permission. Students can access those contents
anywhere, anytime without wideband network to study
as their own styles. [5]
2. System Overview
2.1. Features
WebELS (Web-based E-Learning System) is an
online e-learning platform aiming to not only assist
teachers to develop, deliver or archive multimedia
contents on the web, but also provide students various
materials and flexible tools to study in their own
learning styles. Some main features are list below:
x Browser-based: no download and pre-installation
are necessary.
x Always-on and Real-time: teachers can archive
their lectures or even carry out virtual classroom
over the Internet by WebELS.
x Flexible authoring: all necessary tools to create
contents online are provided.
x Broad file type support: text, audio, video, flash
contents with presentation pointer are supported.
x Unicode: multi-lingual is supported.
2.2. Functions
2.2.1. Editor. The WebELS editor aims to provide
editing toolset for users to develop their own
multimedia contents from raw materials in various
formats such as: .DOC, .PPT, .PDF, JPG, .GIF, etc.
Contents title and category should be assigned at first.
177
Then those materials are converted into image series side and perform data transformation with database
which are called as slides in this paper. Moreover, server. Database server is to provide data storage for
users can make attractive multimedia contents by achieved contents, supports contents editing, manage
inserting audio synchronized with cursor movements or user information, and verify course dependency. And
some video and flash clips into one slide. all user account information is maintained by SQL
database server.
2.2.2. Player. The WebELS player provides users a
friendly platform to learn multimedia contents online or 3. Multimedia Contents
offline. Usually contents can be presented as the
following delivery modes: pure slide, video clips, audio 3.1. Image-based content
and cursor synchronized slide. The player was designed
as a standard media player and users can easily control An e-learning course based on multimedia contents
their contents. Zoom-in function is available so that may be highly bandwidth intensive and students using
details on slide can be displayed clearly. non-broadband network may have difficulty in playing
and retrieving such contents. The WebELS content is
2.2.3. Manager. The WebELS management toolset presented by a sequence of image frames instead of
was developed to help users manage the achieved video stream. Not only the contents originally prepared
contents including content category, content permission, in PDF, PPT or DOC formats but also those pictures
and log and registration information. used to describe lecture scenes (such as GIF, JPG and
PNG) will be automatically converted into a kind of
2.3. Architecture image format by the WebELS editor before they are
uploaded to the WebELS server.
In WebELS system, implementation of editor,
player and manager is based on a typical
Browser/Server (B/S) structure. Under this structure,
all user interfaces are implemented on web pages. Main
tasks are executed by server side and only small work
is carried out on browser side, it will much lighten the
loads on client-side computers. Moreover, this kind of
system can be accessed and used without downloading
or installing other tools beforehand. Figure 1 describes
the architecture of WebELS system.

㪪㫋㫌㪻㪼㫅㫋㫊 㪙㫉㫆㫎㫊㪼㫉

㪫㪼㪸㪺㪿㪼㫉 㪙㫉㫆㫎㫊㪼㫉 㪘㪻㫄㫀㫅 㪙㫉㫆㫎㫊㪼㫉

㪠㫅㫋㪼㫉㫅㪼㫋

㪮㪼㪹㩷㪪㪼㫉㫍㪼㫉
㪘㫇㪸㪺㪿㪼㩷㪟㪫㪫㪧㩷㪪㪼㫉㫍㪼㫉

㪫㫆㫄㪺㪸㫋㩷㪪㪼㫉㫍㫃㪼㫋㩷㪚㫆㫅㫋㪸㫀㫅㪼㫉

㪡㪪㪧㩷㪧㪸㪾㪼㫊 㪟㪫㪤㪣㩷㪧㪸㪾㪼㫊

㪪㪼㫉㫍㫃㪼㫋㫊

㪛㪸㫋㪸㪹㪸㫊㪼㩷㪪㪼㫉㫍㪼㫉

㪤㫐㪪㪨㪣 㪜㪻㫀㫋㫆㫉 㪝㫉㫆㫅㫋 㪯㪤㪣

㪛㪸㫋㪸㪹㪸㫊㪼 㪘㫇㫇㫃㪼㫋 㪝㫀㫃㪼㫊 㪝㫀㫃㪼㫊
Figure 1. Architecture of WebELS system
Web server is responsible for users’ requirements. A
customized HTTP server handles web-based user
interface includes HTML and JSP pages. Java Servlets
response the interaction with the Java tools on client-
Figure 2. Comparison of picture quality between
image-based and video contents
Figure 2 compares picture quality between image-
based contents provided by WebELS system and video
contents played by Windows Media Player. Obviously,
the image-based picture with 800×600 pixels is clearer
and brighter than video picture with 512×342 pixels.

178
So if users have to limit the contents size due to through sound pick-up outfits were encoded into the
network, image-based contents would be a substitution. SPEEX format. Figure 4 illustrates how the SPEEX
Figure 3 is an example of image-based contents with audio files are implemented on WebELS system.
enough resolution to ensure complete details in the
picture. By freely zooming or dragging such content Select one slide to be edit
based on high-quality images, students can see those
fine details which are difficult to recognize in common
video contents. Use sound pick-up No
outfits?

Read audio data from files

Yes

Read audio data from

sound pick-up outfits Decode audio data to raw
format

Encode audio data to

SPEEX format

Save SPEEX audio data

into the selected slide file

Figure 4. Implementation of SPEEX audio files

3.3. A synchronized cursor

In WebELS system, a cursor file is also embedded

Figure 3. WebELS contents with zoom-in function
into each slide of the contents to simulate lectures’
action. Users can edit the slide and point a cursor onto
corresponding slide to tip the topic they are talking on,
or show the points they want to strengthen. Each slide
has its own timeline in order to: i) record the cursor
positions (x, y) chronologically; ii) store the
information (s, f) for the embedded audio stream file
simultaneously, in which s is the current state (play or
stop) and f is frame numbers.

3.2. SPEEX audio files Cursor Timeline Audio stream

(x1,y1) (x3,y3) (x6,y6) (xi,yi)

Audio is indispensable part in an e-learning course, t0 t1 t2 t3 t4 t5 t6 t7 tI tn
and vocal files with high quality assist students to (s0,f0) (s1,f1) (s7,f7)
(s4,f4) (s5,f5)
understand lectures more clearly and actively.
Normally, to improve the audio quality would result in f1=f4 f5=f7
large size and heavy load of the contents, which is
unpractical for non-broadband users. Therefore, an
audio format with high quality as well as low size is Figure 5. Synchronization of cursor positions with
necessary. SPEEX is an audio compression format audio stream
designed especially for speech. It is well-adapted to
Internet applications and provides useful features that As Figure 5, the cursor positions are synchronized
are not present in most other codec. SPEEX provides a with audio stream in each slide file by the time shaft.
free alternative to expensive proprietary speech codec For examples: i) when t=t0, (s, f)=(play, 0) and (x,
and lowers the barrier of entry for voice applications. y)=(0, 0), the audio stream is started, and there is no
In WebELS, the audio is embedded into the contents cursor action; ii) when t=t1, (s, f)=(stop, f1) and (x,
by WebELS editor. The audio datum either recorded y)=(x1, y1), the audio stream is paused and the cursor is
and saved previously or simultaneously transmitted pointed to the location (x1, y1) on this slide; iii) when

179
t=ti, (s, f)=(play, fi) and (x, y)=(xi, yi), the audio stream
has processed fi frames and the cursor is pointed to the
location (xi, yi) on this slide.
Also in Figure 5, a continuous audio stream can be
dispersed at different cursor positions according to the
frame numbers recorded by the timeline. Therefore the
duration of audio stream is normally shorter than cursor
action although both of the actions are simultaneously
recorded. Compared with real-time recording system,
the file size of the audio stream which was edited based
on timeline was decreased.
3.4. Comparison
Comparison was carried out between video-based
and WebELS content as Table 2. The original content
was a 42-minute video file with AVI format which was
real-time recorded in classroom. Then the video based
content was edited by WebELS editor and converted
into the image-based content with audio and cursor
files attached. From the table, the content size was
much decreased while the image resolution was
improved. Moreover, both of the two contents were
tried out by using 56K modem under the same
condition, and only the WebELS contents can play
smoothly.
Table 2. Comparison between video-based and
WebELS contents
Video-based WebELS
Size 154MB 3.9MB
Image resolution 512×342 800×600
Streaming quality Time lag and delay Smooth and stable
server named GNUMP3d for video-based content and
WebELS server on the same server machine. Then the
Bandwidth Controller (tool for bandwidth controlling)
was installed in the client-side computer to simulate
various network bandwidths. Figure 6 shows the time
delay which was necessary to open the two kinds of
contents under different network bandwidths. When the
bandwidth is less than 256Kbps, it was hardly to open
the video-based content; WebELS content could be
opened with some delay during playing. And if the
bandwidth is under 10Mbps, WebELS contents were
easier to open compared with video-based contents.
4. Streaming multimedia
4.1. General
Streaming multimedia system can not only ensure
that teachers create appealing multimedia contents but
also allow students to access multimedia contents
without wideband network. WebELS server supports
streaming-like download.
Basically, the WebELS contents consist of image-
based slide series and each slide includes the following
information: image, timeline, audio and cursor. See
Figure 7. The data of contents are stored into each slide
file as separate data packages instead of one complete
file. Therefore, a streaming-like download process is
available based on such data packages. And the audio
data as well as correspondent cursor data are divided
into some clips, and those clips are also considered as
buffering packages.
㪮㪼㪹㪜㪣㪪㩷㪚㫆㫅㫋㪼㫅㫋

80
Video Content
70 㪪㫃㫀㪻㪼㩷㪈 㪪㫃㫀㪻㪼㩷㪉 㪪㫃㫀㪻㪼㩷㫀 㪪㫃㫀㪻㪼㩷㪥
WebELS Content
60 㪪㫃㫀㪻㪼㩷㫀
Time delay (s)

50 㪠㫄㪸㪾㪼
40
㪫㫀㫄㪼㫃㫀㫅㪼
30 㪚㫌㫉㫊㫆㫉 㪚㫌㫉㫊㫆㫉 㪚㫌㫉㫊㫆㫉 㪚㫌㫉㫊㫆㫉
㪺㫃㫀㫇 㪺㫃㫀㫇 㪺㫃㫀㫇 㪺㫃㫀㫇
20

10
㪘㫌㪻㫀㫆 㪘㫌㪻㫀㫆 㪘㫌㪻㫀㫆 㪘㫌㪻㫀㫆
㪺㫃㫀㫇 㪺㫃㫀㫇 㪺㫃㫀㫇 㪺㫃㫀㫇
0
56K 256K 512K 1M 10M
Network bandwidth (bps)
㪚㫃㫀㫇㩷㪈 㪚㫃㫀㫇㩷㪉 㪚㫃㫀㫇㩷㫁 㪚㫃㫀㫇㩷㪤

Figure 6. Comparison of the time delay under

various network bandwidths Figure 7. Data structure of WebELS content

Furthermore, the two kinds of contents were played 4.2. Implementation

under network with bandwidths of 56Kbps, 256Kbps,
512Kbps, 1Mbps and10Mbps to test their sensitivity to Basically, after one data package was downloaded,
network condition. Firstly we installed the streaming the client side’s player would begin to display the slide

180
as one seamless stream. Simultaneously, download of
the data package for the next slide was continued.
Moreover, within the data package of one typical slide,
the size of audio data usually takes a higher percentage
than image, timeline and cursor data. So we also adopt
streaming technology to audio data included in one
slide. Namely, an audio clip cursor data should be
downloaded at first, then the slide is displayed and the
users can start to study. While the first audio clip is
playing, the next audio clip is being downloaded.
Suppose a WebELS content consists of N slides
(i=1, 2, …, N), and in each slide the cursor and audio
data are divided into M clips (j=1, 2, …, M). The
process of streaming download in WebELS system is
described in Figure 8.
Start
Set i=1
Download image and
timeline of slide i
Set j=1
Download cursor clip j
and audio clip j
Set j=j+1
Is j>M?
No
Yes
Set i=i+1
No
Is i>N?
Yes
End
Figure 8. Implementation of stream-like download
5. Offline Player
WebELS system also provides an offline player for
students without settled network connection, it has the
following features:
x The offline player was embedded into WebELS
contents so all contents are like portable textbook.
x The interface style is same as online viewer so that
the offline learning mode can ensure the maximum
feasibility for users without network.
x The contents bundled with the offline player were
encrypted in order to protect their copyright.
After users decide to download the e-learning
contents, the WebELS server will automatically create
a container which includes three main parts: encrypted
181
WebELS player, necessary libraries and WebELS
contents. Then such container was converted to a .JAR
package by the WebELS server. When the download
was finished, the .JAR package was unpacked and
saved on the client-side computer. The following figure
describes the implementation process of the WebELS
offline player.
㪝㫆㫉㪼㪾㫉㫆㫌㫅㪻㩷㫆㫇㪼㫉㪸㫋㫀㫆㫅 㪙㪸㪺㫂㪾㫉㫆㫌㫅㪻㩷㫆㫇㪼㫉㪸㫋㫀㫆㫅
㪚㫆㫇㫐㩷㫃㫀㪹㫉㪸㫉㫀㪼㫊
㪪㫋㪸㫉㫋㩷㪻㫆㫎㫅㫃㫆㪸㪻 㪚㫉㪼㪸㫋㪼㩷㪺㫆㫅㫋㪸㫀㫅㪼㫉 㪚㫆㫇㫐㩷㫇㫃㪸㫐㪼㫉 㪧㪸㪺㫂㩷㪺㫆㫅㫋㪸㫀㫅㪼㫉
㪜㫅㪺㫉㫐㫇㫋㩷㪺㫆㫅㫋㪼㫅㫋㫊 㪚㫆㫇㫐㩷㪺㫆㫅㫋㪼㫅㫋㫊
㪝㫀㫅㫀㫊㪿㩷㪻㫆㫎㫅㫃㫆㪸㪻
㪬㫅㫇㪸㪺㫂㩷㪺㫆㫅㫋㪸㫀㫅㪼㫉
㪙㪼㪾㫀㫅㩷㫋㫆㩷㫊㫋㫌㪻㫐
Figure 9. Implementation of the offline player
6. Conclusions
The paper introduces an e-learning system for non-
broadband users. In order to ensure the multimedia
learning materials accessible under the network with
limited bandwidth, a kind of image-based contents
combined with synchronized audio and cursor was
created by an editor module provided by web server.
Streaming-like download was implemented so that such
multimedia contents can be displayed smoothly even
on client-side non-broadband network. Moreover, an
offline player was embedded in e-learning content.
The WebELS system has been launched at Internet
URL http://webels.ex.nii.ac.jp/. Services have been
made available on the Internet and its functions are
continuously extended to cover user’s requirements.
7. References
[1] C.A. Carver Jr., R.A. Howard and W.D. Lane,
“Enhancing Student Learning Through Hypermedia
Courseware and Incorporation of Student Learning Styles”,
IEEE Transaction on Education, Vol.42, No.1, pp.33-38.
[2] P.M. Lee and W.G. Sullivan, “Developing and
Implementing Interactive Multimedia in Education”, IEEE
Transactions on Education, Vol.39, No.3, pp.430-435.
[3] R. Jain, “A revolution in education”, IEEE Multimedia,
Vol.4, No.1, pp.1-5.
[4] “Web Design Guidelines for Low Bandwidth”,
http://www.aptivate.org/webguidelines/Multimedia.html
[5] H. Ueno, “Web-based Distance Learning for Lifelong
Engineering Education - A Personal View and Issues”,
Journal of Information and Systems in Education, Vol.1,
No.1, pp.45-52.
 2009 International Conference on Computer Engineering and Technology
Implementation and Improvement Based On Shear-Warp Volume Rendering
Algorithm
Li Guo
College of Electronic Engineering，University
of Electronic Science and Technology of
China，Chengdu， 610054
dlbasin@163.com
Abstract
The shear-warp algorithm is one of the fastest
algorithms based on CPU for volume rendering so far,
but it achieves this rendering speed only by sacrificing the
resulting image quality. This paper presents a new
method that makes the resulting image quality much
improved, by mending the resampling values used during
the process of image compositing ， nearly not affecting
the efficiency of rendering. In this method, two adjacent
slices are defined as front slice and back slice
respectively, according to the direction of the viewing ray.
A cost function is defined for describing the two adjacent
slices’ effect to the slab between them. The values of the
resampling points in the front slice and back slice are
inputted to the cost function as its parameters, and the
resulting value can reflect the information of the slab
more effectively. Compositing this resulting value as
resampling value will improve the image quality
significantly, and overcome the drawback of image-
aliasing in the traditional shear-warp algorithm.
1. Introduction
The process of building screen image by object’s
geometric model is called rendering, which is the most
important part during the processes of the scientific
computing visualization. Rendering includes surface
rendering [1~3] and volume rendering. The surface
rendering algorithm can usually reach a fast speed;
unfortunately, it cannot reflect the whole primal data and
the specific characters.
Volume rendering is also called direct volume
rendering. It transforms the 3D data field to final 2D
image for displaying directly, not needing construct the
geometric describing of object’s surfaces. So, volume
rendering is considered as the most promising rendering
technology in the field of scientific computing
visualization. At the present time, volume rendering
mostly includes ray-casting [4] algorithm based on image
space, splatting [5] algorithm based on object space and
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.9
Xie Mei
College of Electronic Engineering，University
of Electronic Science and Technology of
China，Chengdu， 610054
xiemei@ee.uestc.edu.cn
shear-warp [6~7] algorithm based on both image and object
spaces. Owing to the need of resampling after changes of
the viewing direction, both the ray-casting and splatting
methods have the problems of large amount of calculation
and a long time for computing. Therefore, it is difficult to
achieve the requirements of interactive applications by
using these two methods. The shear-warp algorithm uses
a method that factorizes the projection transformation in
the 3D discrete data field to two steps: shear for 3D data
field and warp for intermediate image. As a result, the
process of resampling is changed from 3D space to 2D
plane, and the nearly interactive speed of rendering is
achieved based on graphics workstations by reason of the
significantly reduced amount of calculation. However,
because the resampling just occurs inside of the slices and
only make a single slice’ value to determine the value of
the slab between the two adjacent slices, image quality
usually cannot meet the requirements by reason of the
image-aliasing. Based on the problems of traditional
shear-warp algorithm, this paper presents a new method
by mending the resampling values for image compositing
to make the image quality much improved, with nearly
not affecting the rendering speed. Experiments have
proved this new method’s effects.
2. The shear-warp algorithm
The basic principle of the shear-warp algorithm is
shown in figure 1 [8].
(a)
182
 1). Transform the volume data to shear space by
translating each slice using Mshear.
2). Project the volume into a distorted 2D intermediate
image in shear space by compositing the resampled slices
together.
3). Transform the distorted intermediate image to
resulting image by warp operation using Mwarp.

3. The improvement
(b) 3.1. Image compositing
Figure 1: (a) Standard and (b) Factorized In this section, we firstly introduce the foundation
viewing transformation
theory of image compositing. Image compositing is to
Let Mview be the view matrix that transforms points from
form the value for each pixel in image along each viewing
object space to image space. The shear-warp algorithm is
ray; by transforming the resampled volume data to color
based on the idea of a factorization of the view matrix
value and opacity value during each slice and compositing
Mview into a shear component Mshear, a warp component
them together. The order of the compositing can be front-
Mwarp and a principle axis transformation matrix P
to-back or back-to-front. In this paper we use the front-to-
M view=M warp •M •P
shear back order as shown in figure 2.
with . As for a parallel
projection, since the shear is parallel to the slices, the cin cout
shear factor (Mshear) just contains a transformation that ci
Eye
simply translates each volume slice, in such a way that the
viewing direction becomes perpendicular to the slices. So αi
the shear matrix Mshear is described as follows: α in α out

⎛1 0 sx 0 ⎞ Figure 2: Front-to-back order compositing

⎜ ⎟
0 1 sy 0 ⎟ We specify colors and extinction coefficients for each
Mshear = ⎜
⎜0 0 1 0⎟
scalar value s of the volume data by transfer functions
⎜⎜ ⎟ c ( s ) and τ ( s ) . The color emitted from one point of the
⎝0 0 0 1 ⎟⎠
volume is determined by c ( s ) τ ( s ) ; thus, the intensity I
The sx and sy represent the translating factors in x-axis
along a viewing ray parameterized by x from 0 to D is
and y-axis respectively, decided by the viewing direction.
given by:
Let Mview3 be the upper-left 3x3 submatrix of Mview, as D x
described like: I = ∫ 0
τ ( s ( x ) ) c ( s ( x ) ) e x p ( − ∫ τ ( s ( x , ) ) d x , ) d x (1)
0

⎛ m11 m12 m13 ⎞

⎜ ⎟
Mview3 = ⎜ m21 m22 m23 ⎟ s(x)
⎜m m33 ⎟⎠
⎝ 31 m32
Then, the values of sx and sy are computed by: si=s(id)

m22m13 − m12m23
sx =
m11m22 − m21m12
id (i+1)d x
m11m23 − m21m13
sy = d
m11m22 − m21m12 Eye
x
The warp factor (Mwarp) that transforms the intermediate Figure 3: Sampling of s(x) along a viewing ray
image to resulting image can be decided by Mview, Mshear While the viewing ray is divided up to n equal segments
−1
M warp = M view • M shear • P −1 of length d=D/n and the s value are constant for each ray
and P with . Because the
segment as shown in figure 3 [8], the formula (1) can be
warp is after the projection, it becomes a 2D operation and
approximated by:
much faster to compute.
Briefly, the shear-warp algorithm consists of three
primary steps as follows:

183
 n −1 i −1
occurs just inside of each slice, and the scalar value of
I ≈ ∑ τ ( s (id )) c ( s (id )) d exp( − ∑ τ ( s ( jd )) d )
i=0 j =0
each ray segment is to be constant, as the precondition of
n −1 i −1 image compositing theory mentioned before, with the
≈ ∑ τ ( s (id ))c ( s (id )) d ∏ exp( −τ ( s ( jd )) d )
i=0 j=0
value of s=sf. That is to say the slab’s value is determined
only by the front slice with s=sf, having nothing to do
n −1 i −1

∑ c ∏ (1 − α
≈ i j )
with the back slice. Since the single slice can not represent
i=0 j=0 the slab effectively, the resulting image quality is usually
With the opacity α i of the i-th ray segment defined by poor for users’ requirements.
(i+1) d Considering the relatively poor image quality of the
α i = 1 − exp(− ∫ id
τ ( s ( x ))d x ) traditional shear-warp algorithm, literature [8] introduced a
≈ 1 − e x p ( −τ ( s (id )) d ) pre-integration method, and literature [9] used a resampling
≈ τ ( s (id )) d method of real-time interpolation in the middle section,
but they both increased the amount of calculation. In the
and the color ci defined by
(i+1)d
proposed algorithm, we use the two adjacent slices to
c i ≈ ∫ id
τ ( s ( x ))c ( s ( x ))d x describe the slab, as shown in Figure 5.
≈ τ ( s (id ))c ( s (id )) d viewing ray
≈ α ic ( s ( i d ) )
sb back slice
Then, the final formula of image compositing can be
described as follows:
s = f(sf，sb)
α o u t = α in + (1 − α in ) α i
sf front slice
c o u t = c in + (1 − α in ) c i
(2)
As demonstrated above, it is easy to know the fact that Figure 5: Improved resampling for composting
compositing image by formula (2) is based on the As the slices front and back both contribute to the slab
precondition of each ray segment’s scalar value being between them, we define a cost function f to describe the
constant. effect. Parameters are resampling values of the front and
back slices. The return value s represents the value of the
3.2. Improved method slab, that is, the value for compositing. The more cost
There are two kinds of resampling in shear-warp function close to the real situation, the better the effect of
algorithm. One is resampling in each slice during the rendering. Here we define the cost function as
process of image compositing in shear space; the other is s=f(sf,sb)=w*sf+(1-w)*sb ，where w is the scale factor.
resampling in the intermediate image while warping it to This function is relatively simple and could be a good
the resulting image. We just discuss the first situation in reflection of the slab’s information. When composting, the
this paper. slab’s value s, not sf, is used as the scalar value of the ray
segment. The proposed method has little impact on the
viewing ray
rendering speed, but the image quality is improved a lot
compared with the traditional method attested by our
sb back slice
experiments
s=sf slab
sf front slice 3.3. Results
The traditional and presented shear-warp algorithm are
simulated respectively in Matlab 7.0 on a workstation
Figure 4: The scalar value of a slab in the (Pentium(R) 4 2.40GHZ CPU, 256M memory) using the
traditional shear-warp algorithm same CT slices images coming from West China Center
We define two adjacent slices as a front slice and a of Medical Sciences, Sichuan University. Figure 6 shows
back slice, along the viewing ray direction. Between the the image quality comparison
two slices, a slab is formed as figure 4 illustrated. Scalar
values sf and sb are the values of the resampled points in
the front slice and back slice respectively. The length of
each ray segment is determined by the distance of two
primal adjacent slices and the direction of the viewing ray.
After correction, the length of the ray segment also (a) Traditional elevation
represents the length of the slab. In the traditional shear-
warp method, the resampling operation in shear space

184
 Acknowledgment
The authors would like to thank Zhen Zheng for
helping so much in the daily study, thank Wu Bingrong
for providing the medical images for simulation, and
thank Peng Gang for instructing writing papers in English.
(b) Proposed elevation
Finally, thank for all people in 903 staff room.

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IEEE Computer Graphics and Application, 8(3), 1988,
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[5] Westover L, “Footprint Evaluation for Volume
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[6] Lacroute P, Levoy M, “Fast Volume Rendering Using
a Shear-Warp Factorization of the Viewing
Transformation”, Computer Graphics Proceedings,
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[7] Lacroute P, “Fast Volume Rendering Using a Shear-
Warp Factorization of the Viewing Transformation”,
Technical Report: CSL-TR-95-678, 1995
[8] Schulze J P,Kraus M,Lang U, “Integrating Pre-
integration into the Shear-Warp Algorithm”,
Proceedings of the 2003 Eurographics/IEEE TVCG
Workshop on Volume Graphics Tokyo:ACM
Press,2003,pp.109-118
[9] Sweeney J, Mueller K, “Shear-Warp deluxe: the
Shear-Warp algorithm revisited”, Proceedings of the
Traditional Proposed Symposium on Data Visualization
(e) Partial magnified image Barcelona:Eurographics Association,2002,pp.95-104
Figure 6: Traditional and proposed algorithm’s
image quality comparison
The simulation results show that images produced by
the traditional method of rendering have some visible
stripes, but in the proposed algorithm, the stripes
disappeared, the image quality is improved.

4. Conclusions
By using two adjacent slices’ value to define the slab’s
value, which is more in line with the actual situation, the
proposed algorithm has improved a lot as to the traditional
algorithm. The principle and implementation results both
show that it is effective to obtain a higher image quality,
and almost does not affect the rendering speed

185
 2009 International Conference on Computer Engineering and Technology

Conferencing, Paging, Voice Mailing

via Asterisk EPBX
Ale Imran1, Mohammed A Qadeer2
1
Dept of Electronics Engg, Aligarh Muslim University, Aligarh, India
2
Dept of Computer Engg, Aligarh Muslim University, Aligarh, India
{aleimran, maqadeer}@zhcet.ac.in

Abstract—This paper is intended to present theoretical and such as paging, (which may be from one-to-one or many-
implementation details of various important features that to-one, depending on the usage requirements) Interactive
are generally associated with an Asterisk based Voice voice responses (IVR), Conferencing, Voicemail, Music
Exchange i.e. Conferencing, Paging and Voice mailing. on hold to name a few.
Besides this we will be concentrating on the web based
approach for the configuration of the hard phones that will
be used at the clients end. Our approach follows the client-
server model for all subsidiary procedures. The application
is developed in C language and is compatible with all the
versions of Linux, essentially providing PC to PC data and
voice communications.

Keywords-Asterisk, VoIP, PBX, Voicemail, Conferencing,

Paging, ST-302 hardphone.

I. INTRODUCTION
Over the last few years Voice over Internet Protocol
(VoIP) has became an important player in supporting
telephony. The contribution in achieving this goal clearly
goes to the various advantages that the technology offers.
Besides being almost free, it provides various other
features which one could say are almost patented with it..
To talk of a few would certainly include a reduction in the
bandwidth requirements and availability of a large number
of features like for example selective call forwarding and
rejection [4].
The development of Asterisk Based Voice Exchange
which works on VoIP takes into consideration the various
complexities associated with a conventional Private
Branch Exchange (PBX). A conventional circuit switching
based PBX is not only expensive but limited in terms of
functionality as well. Asterisk is a complete phone system
in software. It can replace large and expensive phone
systems powering thousands of extensions, or it can help
users save money on long international call usages.
Because being implemented in software, it’s extremely
versatile, easy to customize as and easy to extend.
Asterisk is different for several reasons, the most
important being its all software approach. Instead of
switching analog lines in hardware, it routes and
manipulates Voice over Internet Protocols (VoIP) Packets
in software [4]. The backbone of the system generally
becomes an IP enabled network, and phones can be
hooked into that. However it also supports old analog
phones using gateway devices.
Asterisk provides more than what one would expect
from a conventional PBX. We get a variety of features
Fig 1: Overview of Asterisk based system [4]
On top of that we can get interfaces to the operating
system and programming languages for the extreme in
power, optional web based administration interfaces,
configuration in SQL databases or flat files, detailed call
logging into a database and many more features. Hence to
summarize it up, we can say that with Asterisk you can:
• Provide basic service to Analog and Digital
phones.
• Develop a call routing logic in order to choose a
least expensive way to route a particular call.
• Route Incoming and Outgoing voice calls over
standard voice lines or the internet.
• Provide voicemail and teleconferencing services
• Develop complex or simple interactive menus
• Operate small or large queues for call centers and
announcing the estimated hold time to the callers
• Call other programs on the system .
II. COMPLETE ASTERISK SYSTEM
A. Set Up of the Work
Asterisk is an open source converged
telecommunications platform designed to allow different
type of telephony hardware, middleware and software to
interact with each other consistently. It provides multiple
layers, managing both TDM and packet voice at lower
layers while offering a highly flexible platform for PBX
and telephonic applications. Asterisk can bridge and
translate different type of VoIP protocols such as SIP,

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DOI 10.1109/ICCET.2009.209
MGCP, IAX and H.323[2]. In order to set up a Asterisk
based Private branch exchange (PBX), we require the
following 3 major components [3];
• Asterisk based PBX
• Phones at the clients end which may be soft or
hard depending upon the requirements.
• VoIP gateway service in order to enable a
particular user to call others who might be on
PSTN or on the same IP network.
We begin implementing our voice exchange by compiling
the Asterisk system. The following commands help us in
compiling the Asterisk.

cd/root/ale/asterisk
make
make install
make samples
make progdocs

Fig 3: Picture showing ST-302, the IP enabled hardphone.

B. Asterisk PBX configuration

Fig 2: Snapshot after the complete installation [4]
Once it has been successfully installed we can start
Asterisk on the server by running the following
commands:
Now we need to create one user in the iax.conf file.
This is because the phone is using IAX protocol for being
connected with the Asterisk server. So now we create a
new user say by the name of user1. This one is now going
to be used with the ST-302 IP phone.

/root/ale/asterisk-vvvc.

Now since here we are interested in having hard phones

at the client’s end ,we have the choice of selecting a
particular hard phone among the various available ones
and configuring it . ST-302 is an IP enabled hard phone
which uses an Inter Asterisk Exchange (IAX) protocol. It
has got two RJ-45 ports, one for connection with a PC
and another for connection with the existing IP enabled
network. We put the Ethernet cable from our network in
the port labeled RJ-45 and
use the another Ethernet cable to connect our computer
with the phone. Then we put this cable in the RJ-45 jack
labeled PC.
Fig 4: IAX.conf file
type=friend means that the user can make & receive
calls, host=dynamic means that the IP is dynamic through
a DHCP server, allow=all means that the line which this
user will be using, could support all audio codec’s,
context=test shows that the user is working with the
extensions in this context of the configuration file
extensions.conf.

187
 Now lets have a look on the extensions.conf file that
we ‘ll be using for setting up the various extensions.
Fig 5: Extensions.conf file
[test]
Exten=>100,1,answer()
Exten=>100,2,dial(IAX/user1)
Exten=>100,3,hangup()
Exten=>200,1,answer()
Exten=>200,2,dial(IAX/user2)
Exten=>200,3,hangup()
In the extensions.conf file, working in the [text] section we
have got 2 phone numbers 100 and 200, and for each of
them we have created 3 extensions . When somebody dials
a number 100,the call will be answered by the Answer
application.The next application which will be executed is
the Dial application.User1 is the user which we are going
to use for the ST-302 IP phone. The last extension is using
the Hangup application. Its purpose is to hang up the line
after the conversation is over. Infact its always a good
idea to use this application in the dial plans. So when
somebody dials the number 100, the ST-302 phone will
start ringing and call will be connected to this phone.
III. CONFIGURATION ON THE ST-302 IP PHONE
The configurations of the ST-302 IP phone could be
done with the help of Web Interface and for this we need
to know the IP address of the phone .
A. Making connections through the Web Interface
The web interface has two access modes ie ordinary
and super. The first does not give the access to the settings
considering the IAX ,however the super mode gives you
the access to all the settings ,including the one for the IAX
protocol. The one implemented for the configuration of
ST-302 is super access mode since it allows IAX settings.
Fig 6: Web Interface for the configuration of hard phone.
188
Here configuring the ST-302 with the help of web
interface allows to perform the following operations :
1) Audio Settings: gives a user free hand as far as
selection of various codec’s for the hardphone is
concerned. It may generally depend on the design aspect
and for our implementation we have set this to G.711.
Besides this, it allows for various other added features as
well like for example voice activity detection(vad),
automatic gain control(agc) ,automatic echo cancellation
(aec), jittre size and various others.
2) IAX Settings: The following parameters summarize
to give up the complete IAX settings:
a) Use service: Enabling this particular feature
allows calls to made through the gatekeeper, if however
this feature is disabled only IP to IP calls could be made.
b) Local port: this is the port on which phone will
negotiate registration information with the server.
Generally by default it’s 4569.
3) Network Setings: includes the features generally
associated with the networks. It includes local ip, subnet
mask, dns, iptype, ppp id and ppp pin.
4) Dial Plan Settings: In this file actions are
connected to extensions. Each extension belongs to a
context, either the default context or a specific context
that we have created like incoming IAX calls, long
distance outgoing PSTN calls, local calls etc [4].
IV. FEATURES OF ASTERISK
Asterisk based telephony solutions offer a rich and
flexible feature set. Asterisk offers both classical PBX
functionality and advanced features, and interoperates
with traditional standard based telephony systems and
Voice over IP systems. Asterisk offers the advanced
features that are often associated with large, high end
proprietary PBX’s. Here in this paper we will be
concentrating in details on the paging, voicemails and
conferencing applications of the Asterisk enabled Voice
exchange [3].
Fig 7: Management & Configuration Modules of Asterisk
A. Call Conferencing means that no mandatory downsampling is required.
The Call conference is an Asterisk solution based These two factors in fact provide a rather significant boost
PBX system. The call conference provides conference to the VoIP base conferencing.
room system for use by all users. The call conference
provided by Asterisk based PBX has the following B. Voicemail
features: Security passwords to control access to who can
This feature enables users to record messages for
call into a conference bridge, unlimited simultaneous
incoming calls that are not answered within a specified
conferences, with unlimited participants. Joining a
number of rings, or receive busy treatment, or are
particular conference is as simple as dialing the extension.
transferred directly to the voicemail. Asterisk comes with
Incoming or outgoing calls may be transferred to a
a voicemail storage with storage capacity of more than
conference, or it may be directly dialed [4].
thousand hours and it can be retrieved from any remote
phone or it could be attached with the email as .WAV file
While installing Asterisk on the server, we need to
or to the voice messaging system repository for retrieval
execute the following commands for enabling the
from a phone [4]. By accessing voice portal from any
conferencing feature via the server:
phone user can listen, save, delete or reply to a message
Cd/usr/src/asterisk
received from a one or more than one group members
Cd/app_conference
with introductory messages. Moreover users have the
Make clean
option of marking a particular message as urgent or
Make
confidential.
Make install
The first thing we need to do is to create the mail box for
Once the conferencing feature has been installed,
asterisk to use, with the help of the following utility
compilation of the conferencing application, we need to
Usr/src/asterisk/addmailbox.
make the following changes in the configuration files :
Moreover we also need to edit the configuration files of
[Conferences]
voicemail ie
Exten=>s,1,wait(1)
vi voicemail.conf
Exten=>s,2,digit timeout,5
Exten=>s,3,response timeout,8
Now we need to locate the IAX section where we have
Exten=>s,4,background
added the entries for the extensions, the lines should like
Exten=>s,5,waitexten(20)
this:
Exten=>s,6,goto(s,4)
Exten=>1, 1, dial (IAX/100,20, tr)
Exten=>XXXX,1,DBget(pass=conferences/${exten});
Exten=>2, 1 dial (IAX/200,20, tr)
Exten=>XXXX,2,gotoIf($[“xxx${pass}”=” xxxNONE”
Exten=1000,1,dial(IAX,100&IAX,200,20,tr)
Exten=>XXXX,3,read(secret,pls-enter-conf-password,10)
Exten =>1000,2,voicemail,u9999
Exten=>XXXX,30,conference(${EXTEN}/MTV)
Exten=>1000,102,voicemail,b9999
Exten=>XXXX,31,hangup
Exten=>XXXX,40,playback
Now what we have done over here is that when extension
Exten=>XXXX,41,hangup
1000 rings ,the first thing we do is dial phone number
1and phone number2 ie 100 and 200 make them ring for
[confhelper]
20 seconds. If the extension is not answered in 20
Exten=>in,1,answer()
seconds, the second entry will be executed which is a
Exten=>in,2,background()
voicemail. The mailbox is specified by u9999 at the end
Exten=>in,3,response timeout()
of the line. Moreover there are two more things noticeable
Exten=>in,4,hangup()
over here, the first is that the priority number has jumped
Exten=>out,1answer()
to 102.This is in fact quite a useful feature of Asterisk
Exten=>out,2,background
which signifies that when the call comes and if a person
Exten=>out,3,response timeout()
is on, the priority will be jumped to+101(n+101).This
Exten=>out,4,hangup()
gives a priority of 102.The second thing to notice is that
the mailbox number at the end of the line is preceded by
[context]
b9999,this indicates a busy message should be displayed
Exten=>999999,1,goto(conf-conferencename,join,1]
to the user and the user should be allowed to leave a
message.
The conferencing feature provided by the Asterisk offers
the advantage of doing conferencing without any zaptel
hardware as well doing native codec streaming, which

189
However until here, a person can leave a voice message,
but in order to access it, we need to add the following
lines:
Exten=>1001,1,ringing
Exten=>1001,2,wait(2)
Exten=>1001,3,voicemailMain,s9999
Now when an extension 1001 is dialed, the following
happens
• The phone gets a ringing tone
• There is a two second wait(the phone is still getting
the ringing tone)
• The call is answered and goes straight to the
voicemail menu for mailbox 9999.
C. Paging
This feature supports system wide paging and single
phone intercom or unlimited parking of calls
simultaneously. Call parking enables a user to hold a call
and to retrieve it from another station within the group.In
order to park a call, a user presses the flash hook and dials
the call park feature code.The call is parked and the caller
is held.[4]In order to retrieve the call,the user can go to
any phone in the group and has to dial the call retrieve
feature code,followed by the user’s extension id. Hence as
a result the call is retrieved and connected to the
retrieving user.
1) One-to-Many Paging:
[One_way_page_group]
Exten=>1,1,IAX Addheader(Call-Info:answerafter=0)
Exten=>1,2,Page($,{One_way_paging_list})
Exten=>1,3,Hangup.
The above written configuration will allow users to one
way page (broadcast) to all the extensions defined in the
variable”One_way_paging_list”, which infact could be
defined as
One _way_paging_list=>IAX/100 & IAX/200.
2) One-to-One Intercom:
We first need to define a macro and then use it in the one
to one intercom context.
[macro-pageext]
Exten=>s,1,chanisaval(${ARG1}/JS)
WHERE J:is for dump& Sis for ANY call
Exten=>s,2IAX Addheader( Call-Info:answerafter=0)
Exten=>s,3,Dial(${ARG1})
Exten=>s,4,No Op()
Exten=>s,5,hangup
Exten=>s,102,hangup
[INTERCOM GROUP]
Exten=>*5XX,1,macro(pageext,IAX,/${EXTEN:1})
Exten=>*5XX,2,hangup
190
The above written configurations will allow user
intercom with any extension by dialing *5XX.
3) One-to-Many Intercom:
[Two_way_intercom_GROUP]
Exten=>**2,1,IAXheader (Call-Info:answerafter=0 )
Exten=>**2,2,Page(${Two_way_intercom_list}/d)
Exten=>2,3,hangup
The above written configuration will allow user to do two
way intercom to all the extensions defined in the
variable”Two_way_intercom_list as follows
Two_way_intercom_list=>IAX/100 & IAX/200
V. SUMMARY
We expect that the design and implementation aspects
presented in this paper, will be a valuable developing
guide for similar kinds of applications. Future work could
proceed along the following lines:
• Using an Asterisk based server for connecting two
remote locations.
• Implementing IVR, Multiple Auto Attendants,
Music-on-hold and various other features of the
Asterisk based PBX.
This project has provided us with an invaluable
experience related to VoIP and collaborative efforts. In
fact, Asterisk has proven to be a viable PBX for future
research studies.
REFRENCES
[1] Taemoor Abbasi, Shekhar Prasad , Nabil Seddigh, Ioannis
Lambadaris “A comparative study of the SIP & IAX voice protocols” in
CCECE/CCGEI, Saskatoon, May 2005
[2] Md. Zaidul Alam, Saugata Bose, Md. Mhafuzur Rahman,
Mohammad Abdullah Al-Mumin”Small office PBX using Voice over
IP” in ICACT I2-14 FEB,2007
[3] Jim Van Meggelen, Leif Madsen , Jared Smith”Asterisk the Future
of Telephony” Second Edition, August 2007
[4] Mohammed A Qadeer & Ale Imran, “Asterisk Voice Exchange-An
alternative to conventional EPBX” in Proc. IEEE ICCEE 2008, Dec 20-
22, 2008
 2009 International Conference on Computer Engineering and Technology

A new Mind Evolutionary Algorithm based on Information Entropy

Yuxia Qiu Keming Xie

The college of Management Science and Engineering The college of Information Engineering
Sanxi University of Economic and Finance Taiyuan University of Technology
Taiyuan, P. R. China, 030006 Taiyuan, P. R. China, 030024
qyxljl@yahoo.com.cn kemingxie@tyut.edu.cn

Abstract—A new self-adaptive Mind Evolutionary Algorithm original winners for superior subpopulations and those with
based on information entropy is proposed to improve the lower score are the winners for temporary subpopulations.
algorithmic convergence especially in the late evolutionary Similartax operator completes the competition inside
time. As studied, the evolution of a population is a process subpopulation on subspace and realizes local optimization. In
with entropy reducing and the population entropy can be this process, individuals are produced by normal
used to reflecting the evolution state. Thus the self-adaptive distribution with some variance around each winner and the
strategy can be realized by building population entropy one with highest score is the new winner replacing the old
computing module to estimate the region including global one in following steps. Dissimilation operator completes the
optimal solution. In this way, the exploring of the algorithm competition between subpopulations on whole solution space
and realizes global optimization. In this process, competition
is more purposeful and sufficiently and the performance is
occurs between the winners of subpopulations. Some with
improved. The experimental results show the algorithm is lower score are washed out and replaced by new ones
valid and advanced. scattered at random in solution space to keep the global
searching ability of population. Therefore, prematurity is
Keywords- Evolutionary computing; Mind Evolutionary
avoided and the population evolves to the global solution.
Algorithm (MEA); information entropy
B. The Population Eentropy
I. INTRODUCTION When MEA is used to solve an optimal problem, the
evolution begins with a stochastic population and then the
Mind Evolutionary Algorithm (MEA) simulates the
‘similartax’ operator is used to local research and the
evolutionary process of human being’s mind. It extracts the ‘dissimilation’ operator is used to guarantee the global
virtues from GA and ES and overcomes their disadvantages. ubiquitous , that is ‘wide explore’ and ‘accurately scan’
‘Similartax’ and ‘dissimilation’ operators are presented and hand-in-hand research mode. As shown in Figure 1, the
monolayer population evolution is improved to multilayer evolution is a process that the research region becomes more
population evolution. Since memory and directional and more clear.
learning mechanism are introduced, search efficiency has
been increased greatly [1]. Now, MEA has been successfully
applied to solve some practical problems [2, 3].
As a new type of evolutionary algorithm (EA), MEA
succeeds basic property from traditional EA, good or not
good. For example, it usually takes algorithms a short time to
find a good solution but an opposite long time to find the
best solution. That means at the late evolutionary time the
convergence will deteriorate. Many scholars take this as their
research object and some delightful achievements [4]. We try
to solve the problem by introducing information entropy and
hope to improve other performance of MEA at the same time.
II. THE POPULATION ENTROPY OF MEA

A. Basic MEA
The whole solution space is divided into some subspaces
by MEA and ‘similartax’ and ‘dissimilation’ operators are
used to realize the evolutionary operation. Figure 1. Sketch Map for Movement of Populations
When “learning” starts, individuals are randomly
scattered in solution space and their score are calculated The optimal process is just similar as the
respectively. Those with higher score are kept as the communication process in information theory. The former

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DOI 10.1109/ICCET.2009.43
introduces negative entropy into system just like the latter
does. Based on the definition of information entropy, the
entropy reduction process is described as following: at the
beginning of evolution, the population is absolutely
stochastic and the least knowledge of optimal solution is
known. So, the information entropy is biggest. Then, with
evolutionary operators running, the population moves to the
optimal region and the individuals are more centralized and
the entropy reduce accordingly. In the late time of evolution,
the population unlimitedly approach optimal solution and
individuals are centralized at it, that means the entropy is
smallest. Take the analysis as base, MEA algorithm can be
improved by introducing entropy to adjust research region.
Suppose the solution region can be cut into M areas,
marked as z1 , z 2 , Λ , z M , stochastic experiment is done.
Drew out an individual x from a population and find out
which area it belongs to. If the probability that x comes
from area zi is pi , where i = 1,2,Λ , M , pi ≥ 0 ，
M Where C = 1 .
i = 1,2,Λ , M and ∑p i =1
i = 1 . Hence it is uncertain that a
individual falls into which arear.
Definition 1 The population entropy as a measurement
for the denseness of individuals in a population is defined as
the following formula.
M
H P = −∑ pi ln pi
i =1
C. The Population Entropy Estimation
Accordingly, the entropy of a population H P (t ) is
impossible to be computed truly before the optimal solution
is got but be estimated. Here a strategy is proposed to
accomplish the task.
Suppose f min and f max respectively describe the minimum
and maximum fitness value that MEA algorithm has
explored so far and λ = f max − f min .The number of super
subpopulations is NS and let M= NS. Then the solution
region is plotted into M areas,
[ f min + α i ⋅ λ , f min + α i+1 ⋅ λ ] . i = 1,2,Λ , M . Where
α1 = 0, α M +1 = 1 + ε , 0 ≤ α i ≤ 1 , i = 2,3,Λ , M , α i > α i−1 and
ε is a small positive number.
If M is the population size and mi is the number of
th
individuals from i area, the probability of an individual
th
comes from i area can be approximately calculated as
following.
m PEMEA is done to analyze how the algorithmic
pˆ i = i ， i = 1,2,Λ , r （2）
M
Hence, the estimate of population entropy can be
computed.
r i =1
Hˆ P = −∑ pˆ i ln pˆ i
i =1
The minimum and maximum value of Ĥ P are 0 and
ln M .
III. SELF-ADAPTIVE MEA BASED ON POPULATION
ENTROPY (PEMEA)
As analyzed above, if the population entropy is
calculated, it’s not difficult to estimate the distance between
the population and the optimal solution. In MEA, variance
σ is a key factor to control the population’s research zone.
So the self-adjustment of σ can realize self-adjustment the
algorithm.
Accordingly, a new research strategy can be proposed.
Suppose σ i is the similartax variance of i th generation,
formula (4) describes the parameter self-adjust strategy of
the new algorithm.
σ i2 = C ⋅ Hˆ P (i ) (4)
ln M
Based on above idea, the PEMEA algorithm steps are
described as following:
Step1 Set evolutionary parameters: population size,
subpopulation size and conditions for end;
Step2 Initialization: scatter individuals composing
(1) initial population in whole solution space;
Step3 Similartax operation: new individuals are
produced by normal distribution with variance around each
winner, material calculation follows formula (5). And the
individual with highest score is the new winner replacing
the old one in following steps;
x k = N ( x k −1 , σ k ) = x k −1 + σ k ⋅ r （5）
Where r is a random numeral, x k is the new (or son)
individual in k th generation, x k −1 is the old (or father)
individual in (k − 1) th generation and σ k is the similartax
variance. σ k is calculated according to formula (4).
Step4 Dissimilation operation: realize global
optimization, some with lower score are washed out and
replaced by new ones scattered at random in solution space;
Step5 Conditions for end: if the end conditions are
filled, turn to step6; else repeat step3 and step 4;
Step6 Output evolutionary result, algorithm ends.
IV. SIMULATION EXPERIMENT
Three classic numerical optimization test functions are
applied. And the contrast experiment between MEA and
performance has been improved.
• De Jong’s function 1
3
f ( xi ) = ∑ xi2 xi ∈ [−5.12, 5.12]
（3）
• Rosenbrock
f 2 ( x ) = 100( x12 − x 2 ) 2 + (1 − x1 ) 2 xi ∈ [ −2.048, 2.048]
192
 • Sharffer’s F6
[sin 2 ( x12 + x 22 )1 / 2 − 0.5]
f 3 ( x ) = 0.5 − xi ∈ [−100,100]
[1 + 0.001( x12 + x 22 )]2
The parameter value is set as shown in table 1 and the
algorithms are respectively run 100 times to optimize test
function 1 to 3. If the error between the evolutionary result
and optimal solution less then 0.0001, it is accepted that
algorithm convergent, otherwise not convergent. The
experimental result is recorded in table 2.
Run the algorithms to optimize function 4-5 with
different coefficient C1 to analyze the affect of the
coefficient to MEA. Thereby the advantage of the new
algorithm on parameter adjustment is test. The results are
showed by figure1-4.
From table2, it can be easily figured out that both
PEMEA and MEA has good convergence, but the speed of
the former is apparently faster than the latter with absolute
precision and arrive theory optimal solution indeed.
It can be find out from figure 2 and figure 5 that the
performance of MEA is precarious because affected by
parameter badly but PEMEA overcome the shortcoming. As
shown in figure 3, 4, 6 and 7, it is the off-line performance
but not the on-line performance is improved greatly because
PEMEA take one best individual as its representative and
ignore the average state.
In one word, the performance of MEA is improved
largely in PEMEA.
V. CONCLUSIONS
The idea of information entropy in information theory is
introduced into the MEA operator design and a new
algorithm is proposed as PEMEA. According to the analysis
on experimental result, it can be concluded that the research
strategy based on population entropy improved the
performance including robust, convergence speed and
precision of MEA greatly and PEMEA is an efficient and
stable optimization algorithm.
ACKNOWLEDGMENT
This work was supported by a grant from the Specialized
Research Fund for the Doctoral Program of Higher
Education, P.R.C (No. 2006112005）and Visiting scholar
foundation of Shanxi province, P.R.C（No.2004-18）.
REFERENCES
[1] Chenyi Sun, Yan Sun, Lijun Wei, “Mind-Evolutionary-based
Machine Learning: Framework and the Implementation of
Optimization,” In Proc. Of IEEE INES’98, pp. 355-359, Sept., 1998.
[2] Chuanlong Wang and Chengyi Shun, “Analysis on the Convergence
of MEBML Based on Mind Evolution,” Journal of Computer
Research and Development, pp. 838-842, 2000.
[3] Yuxia Qiu, Gang Xie and Keming Xie, “Multi-model Parallel Tuning
for Two Degree-of-Freedom PID Parameters,” Proc. of 6th
International Symposium on Test and Measurement, pp. 1162-1165,
June, 2005.
[4] Keming Xie ， Yiuxia Qiu. Convergence Analysis of Mind
Evolutionary Algorithm Based on Sequence Model. System
Engineering and Electronics. 2007, 29(2): 308-311

TABLE I. CONTROL PARAMETERS OF THE ALGORITHMS

Parameter MEA PEMEA
Population size N=50 N=50
Sub-population size M=10 M=10
Maximum generation number G=100 G=100
variance σ k = C1σ k −1 σ k = 2 C 2 ⋅ Hˆ k (i )

TABLE II. EXPERIMENTAL RESULT

De Jong's Sharffer’s
Test function Rosenbrock
function 1 function 6

algorithm MEA PEMEA MEA PEMEA MEA PEMEA

Convergence
100 100 100 100 100 100
ratio (%)
Average
convergence
16 6 39 13 20 11
generation
number
Figure 2. Optimization of Rosenbrock function with PEMEA and MEA.
Optimal theory
0 0 1
solution
Optimal
1.17e- 5.64e- 1.78e- 2.05e-
experimental 1 1
7 59 6 17
solution

193
Figure 3. On-line Performance of PEMEA and MEA with Rosenbrock Figure 6. On-line Performance of PEMEA and MEA with Sharffer’s F6
function

Figure 7. Off-line Performance of PEMEA and MEA with Sharffer’s F6

Figure 4. Off-line Performance of PEMEA and MEA with Rosenbrock
function

Figure 5. Optimization of Sharffer’s F6 with PEMEA and MEA

194
 2009 International Conference on Computer Engineering and Technology

An Encapsulation Structure and Description Specification for

Application Level Software Components

Jin Guojie, Yin Baolin

State Key Laboratory of Software Development Environment
Beihang University, School of Computer Science
100083 Beijing, China
{jinguojie, yin}@nlsde.buaa.edu.cn

Abstract—Traditional research of software components mainly
aims to components at the level of program and source code,
due to which the coupling degree between the components and
the runtime environments is hard to lower, thereby limiting the
reuse ability of the component. The application level
components containing comparatively integral granularity of
business function are more suitable for a larger scale of
application, while there is still lack of reference model and
description specification in this field. The inherent deficiency
of program level components is analyzed followed by the
observation of the unique requirements of application level
components, then a novel reference model is proposed. The
main advantage of the model is to propel the integration and
reuse of legacy software resources by means of transforming
currently existing applications into application level
components with the wrappers providing standard interfaces.
The generalization and applicability of the model are evaluated
by real cases.
Keywords-software component; application level component;
reference model; specification
I. INTRODUCTION
Component technology is of growing importance in
software engineering and is considered to be an critical
measure to enhance the reuse ability of software modules.
Software components can be classified into two main
categories according to the functional granularity: the
components at program code level and others at application
level[1]. Traditional software components research mainly
aims to the previous category, resulting in the formation of a
variety of software standards in business world, including
CORBA, EJB, COM/COM+, etc[2]. Among all of these
standards, the component granularity is limited to
fundamental software elements like code segments, functions,
and objects/classes, which are difficult to accommodate
relatively integral business feature. The long term application
practice indicated that the reuse ability of such program level
components is limited to a low level because of their inherent
deficiency as follows:
(1) The component granularity is too fine to undertake a
complete item of business function, like the orders
management task in a ERP system which is composed by a
large number of small code level components working
together and relying on each other. Therefore, for an unique
component, its reuse degree in a new business environment
is significantly low. This means that there is high coupling
degree among the code level components, which is a
situation violating the initial objective of component
engineering[3][7]. This is currently, as we observed, a
bottleneck with no effective solution.
(2) The program level components are developed mainly
by skillful programmers with a specified type of
programming language, which has a strong dependency with
the developing environment and operating platform. As
technology and tools evolving rapidly, there are few
components developed today may remain reusable after a
certain number of years [4].
The novel component model named Application Level
Component(ALC) is proposed in this paper. It is defined as a
software module at the granularity of a standalone executable
software application, and consisting an integral item of
business function [7][8]. Being compared with the program
level components, application level components can consist
relatively more complete function granularity, and the
dependency between the components and their runtime
environments can reasonably be lowered to a certain extent,
so the coupling degree among the components becomes
more loose which is more suitable for domain modelling.
Although application level components show advantages
in many aspects discussed above, there is still lack of
research attention towards them. In current literatures only a
few related concept keywords can be found, like “software
level components”, “components of complete applications”,
etc[5][7][8]. However, the component model and interface
standard have not been commonly recognized, thereby
causing the absence of a common basis for component
analyzing and evaluating. In this paper the specialty and
characteristic of application level components is first
explored, and then a reference model in terms of the
encapsulation structure and specification is presented. In the
end, we established several demonstrations for real practice
to manifest the conclusion that the model applies to a large
scope of domain engineering and can be made to be the basis
of application level component-based design, integration and
testing.
II. SURVEY ABOUT APPLICATION LEVEL COMPONENTS
As a specific subclass of software components,
application level components should meet the requirements
that all software components have in common. First of all, a
general reference model should be built to model the
components falling into this category, which involves a

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DOI 10.1109/ICCET.2009.8
structural model, an interface specification, and a description
mechanism to represent necessary information about the
component’s function and usage[1][4]. As the ALCs are tend
to be formated at higher granularity with more complete
domain logic, they have many unique characteristics which
should be deliberately took into account while designing the
reference model. To make the model more applicable and
acceptable, a survey of the ALC is done by comparing them
with traditional software components, and the unique
characteristics of ALCs are summarized as follows:
(1) The core issue of the ALC’s model is to provide the
ability of encapsulating and reusing current existing software
and systems. As large number of legacy software and
systems in business practice have to be abandoned only
because they fail to keep following the changeful external
requirements, the reuse of numerous functional modules
inside these systems is a key path to reduce duplicate coding
and investment waste.
(2) An interprocess mechanism for communicating
among ALCs is needed. As standalone executable programs,
the ALCs within a system should communicate with each
other in a way that supporting interprocess data transferring.
By contrast, traditional software components are primarily
defined as functions or objects, so only function call and
object request are needed for the communicating of program
level components.
(3) The invocation and assembling style of the ALC has a
close link with the business requirements which it undertakes.
While traditional components are defined and assembled in
terms of program codes, the ALCs are executable programs
whose life circle starts from the creation of the
corresponding process instance and ends when the process
halts. It is a essential difference between traditional
component and ALC, so there must be provided with
specific instructions to control the startup and termination of
ALC though its interface. Other necessary instructions
besides them are supplied in the same way.
Conclusions can be drawn from the comparison that there
are essential differences between traditional components and
ALCs according to many factors from the internal structure
to interface requirement.
III. REFERENCE MODEL
Considering the above survey about the ALC’s
characteristics and properties, an novel ALC oriented
reference model was developed and is now still being
improved. The design target of the model is to represent a set
of necessary description information about ALCs and to
facilitate the reuse of legacy software systems through the
integration and encapsulation of business elements such as
applications and forms with a uniform interface standard.
The structure of the reference model is shown in Fig. 1, and
the main components are as follows:
(1) the applications and forms to be encapsulated,
(2) a description file representing description information,
(3) an parser program administrating the execution,
(4) a data area maintaining runtime data environment,
(5) the component interfaces exposed to callers.
196
Form B
Form C
Application
Form A B
Application
C
Begin End
Application Application
A Wrapper
Data Area
Component
Parser
Component Description File
Description Data Controlment
Interface Interface Interface
Figure 1. Structural Model of Application Level Component
A. Applications and Forms
To encapsulate currently existing software and systems,
the property Applications is certainly to be a significant part
of the model. From the ALC point of view, applications can
be on any scale from single executable program files, like a
primitive “Hello world” printing routine, to complex
software suites, like Microsoft Office or an OA/ERP system.
Furthermore, a conclusion came out after an intensive study
that numerous documents and electronic forms in business
practice have been left up for years and years, which contain
business information of great use, e.g., Microsoft Excel
spreadsheets(.xls) and Word documents(.doc) are all of this
category. The term Forms is used to stand for all sorts of
electronic files containing information about visual layouts
and manual interactions for a given set of business content.
Forms are usually used as a companion of a dedicated
type of application and they work collaboratively to perform
business tasks. Therefore, they can be treated as joint entities
of business tools, that is, a basic run unit can be defined as an
application plus a corresponding form, where the form is
mainly responsible for visual representation of the business
data, and the application handles the parsing and loading of
the form as well as the interactions with the operators. For
example, an Excel spreadsheet containing records of staff
payment can be parsed and loaded by the main program of
Excel from Microsoft Office, so the records can be shown on
the screen display and then be viewed and modified through
the graphical interface. After manipulation, the user’s
operation result is saved in the spreadsheet and can then be
extracted out.
Considering more complicated business tasks involving
multiple applications and forms, several run units can be
encapsulated in one ALC and be routed according to a
specified property named Sequence. The run units are
working in an collaborative way to accomplish the business
functions. For a more flexible usage, advanced flow
structures like choices and loops can be used to indicate the
execution sequence of the run units.
 To facilitate the encapsulation of existing applications, a
common interface standard should be established for all the
applications to be encapsulated. Considering the simplest
case, the common tasks that the applications carried out in
each run unit can be described as: (1) fill in the form
template with initial business data (-write); (2) load and
display the working form for browsing and modification(-
exec); (3) as the user ends operating, extract data from the
processed form and output them with a standard style(-read).
Therefore, the calling interface of the applications can be
summarized as a collection of command params {-write, -
exec, -read}. Any program in accordance with this standard
can be directly integrated into a component, like application
A and B in Fig. 1. For the applications and legacy systems
incompatible with the interface standard, a type of wrapper
technology[8] can be used to provide an external layer called
“application wrapper”, which dominates the program
internally and providing external interface that meets
demands (shown as application C and its wrapper in Fig. 1).
In this way, a broad scale of legacy systems can be
transformed into standard ALCs without touching the inside,
and then can be reused in new business environments.
B. Component Description File
The description file is a repository for a series of
properties representing the necessary information about the
component. Beside the items of Applications, Forms and
Sequence discussed above, some other properties should also
be included. The properties in the description file can be
classified into to main categories: the descriptive properties
and the data-related properties, all shown in Table1.
The descriptive properties are designed to indicate some
generic description information as follows:
(1) ID: an unique identity code assigned to each
component to ensure the uniqueness in designing and
running environment. Here UUID protocol can be combined
to meet the demand.
(2) Name & Description: properties that simply provide
the textuary name and brief information about usage. Other
materials like versions and created time can also be put here.
TABLE I. CONTENT OF THE COMPONENT DESCRIPTION FILE
Category Property Name & Usage
Descriptive ID - a unique identity code
properties Name - textuary name
Description - brief information about usage
Type - whether the component is UI-driven
Applications - the executable programs to be
encapsulated
Forms - the documents/spreadsheets to be encapsulated
Sequence - the execution route for multiple run units
Forms Mapping - data coupling between the
component and the inside forms
Constraint Conditions for Form Data - the conditions
that should be satisfied after each time of form
manipulation
Constraint Conditions for Component Data - the
conditions that should be satisfied for the input and
output data of the component
Data-related Input & Output Data Fields - data fields that the
properties component receives and outputs
197
(3) Type: indicates whether the component has interactive
user interface. It gives a strong hint for determining the
appropriate running environment of the component, i.e., a
type of applications with visual forms and dialogs are usually
to be manipulated by staff, while others without any user
interface are more likely to execute automatically. For
example, components providing dialogs for business data
browsing and maintaining naturally fall into the first
category, and the components doing job for batch printing or
email sending usually refer to the latter.
(4) Forms Mapping: indicates the mapping relation
between the data fields exposed in the component interface
and the data within the inside forms. It is defined as a 3-
tuple{ component data name, form name, data name in the
form}, which describes the rules about how to insert the
component data into each forms and extract them vice versa.
(5) Constraint Conditions for Form Data & Component
Data: a set of logic expressions to keep validity during
runtime. They describe the constraint conditions to be judged
when each form is submitted or when the component ends up.
The expressions are composed by data fields that
representing the component’s current state, so the component
itself can learn the erroneous situation timely and do error
handling or emit exceptions to the caller.
The data-related properties are used to describe the
input/output data fields of the component. Each data field is
designed as a 2-tuple{Name, Type}, where Name is the data
field’s identification, and Type is the data type including
{string, int, float, datetime, recordset}. To enhance the
ALC’s power of data processing, “recordset” type is
designated to abstract a typical set of relational data records
in business practice.
C. Component Parser
To administrate the run units in an ALC, a specific
module is needed to take charge of the scheduling of the run
units according to the description file. Meanwhile, other
standard business-independent tasks like data I/O and
interface handling can all be undertook by this module. This
is the element called component parser. It works as the call
entry of the ALC and conducts the running procedure during
the component’s whole life circle as follows:
(1) as the component is launching, read in the input data
as its original working data.
(2) during runtime, schedule all run units according to the
logic expressions of property Sequence. In each step, the
form written in with current component data is delivered as a
command parameter to the application for manipulating.
When the application ends, the parser reads out data from the
form and saves them as intermediate working data which is
to be processed in the next run unit. If there is no satisfied
expression for next step, the parse output the current
component data and exits.
(3) handle runtime communication with the caller.
(4) to ensure validity, the parser evaluate the expressions
of Constraint Conditions for Form Data after each run unit step,
and Constraint Conditions for Component Data before exit.
(5) when the component ends up, export the working
results which can be manipulated by the caller.
 The above tasks of the component parser are all based on
the content of the description file which are absolutely
business-independent, so the parser itself is kept away from
any business logic and gains a common sense in all ALCs.
D. Data Area
Data area is a region allocated in memory. It stores all the
intermediate working data between multiple run units. Each
item in the region is a 2-tuple {DataName, DataValue},
where DataName is corresponding to a component data
field’s name and DataValue contains the current value.
Data area is created and maintained by the parser. As the
component launches, the parser allocates a piece of the
memory and fills it with all the original input data. During
runtime, the output of each run unit is took over by the parser
and then be updated into the data area. When the component
ends running, the values kept in the data area are definitely
the final work result.
E. Component Interfaces
The component interfaces are exclusively the only entry
exposed to the users for calling and controlling. In this way,
the internal details can all be hidden to the outside, thus
giving the component a “black-box” feature. The interfaces
of the ALC model are defined as a set of instructions
supporting the minimum collection of calling and integration
requirements. Classified by functional type, there are three
categories of instructions shown in Table 2.
(1) Descriptive instructions: to query the basic
description properties of the component, which are necessary
for use.
(2) Data-related instructions: to provide a uniform rule to
handle data I/O between the component and its caller. Here a
platform-independent XML file format is used to facilitate
data exchange. The data file is assigned as the parameters of
the instructions --load-data and --save-data, which simply
contains the data names and values in an intuitionistic format.
(3) Controlling Instructions: to provide methods for
controlling startup and termination of the component.
Furthermore, a particular feature is assigned to the model for
dumping the intermediate running state to a disk file which
can be loaded from to resume the previous state. This feature,
carried out by the component parser as a standard function,
can enable the operators to save multiple versions of work
effort at will, and revert to one in some situation, e.g., a data
crash after an unexpected system halt.
TABLE II. CONTENT OF ALC COMPONENT INTERFACE
Type Instruction Name & Parameters
Descriptive instructions --get-id
--get-name
--get-description
--get-type
--get-input-data / --get-output-data
Data-related instructions --load-data <data_filename>
--save-data <data_filename>
Controlling Instructions --exec <input_datafile> <output_datafile>
--terminate
--dump-image <image_filename>
--load-image <image_filename>
198
Figure 2. Structural Model of Application Level Component
Generally, the descriptive instructions are mainly used in
system assembling stage, i.e., for the system developer to
identify the usage of the component, while the data-related
and controlling instructions are mainly used in system
running stage to invoke the proper execution behavior of the
component.
IV. SPECIFICATION OF THE REFERENCE MODEL
An concrete description specification of the ALC model
is presented as the UML class diagram shown in Fig. 2. Each
of the structural elements and interfaces in the model is
abstracted as a class in the diagram. As a whole, an ALC is
abstracted as a core class ApplicationLevelComponent.
The class Sequence is used to describe the execution
steps of run units. It is formed as a collection of expressions
like <Source [condition] => Destination>, where: (1) Source
is the current run unit represented as a pair of
“Application:Form”. (2) Condition is a logic expression
indicating the routing path and flow condition between every
two run units. The condition is composed by the component
data fields which can be evaluated at run time. (3)
Destination is the run unit to be executed next when the
condition is satisfied.
Other description elements include the classes of
Interface describing the collection of interface instructions,
InputData and OutputData describing the input/output data
fields, FormsMapping describing the data coupling between
the forms and the component, and classes of
FormConstraintCondition and Input/Output DataConstraintCondition
describing various types of conditions in the model. All these
classes and their relations in the diagram form an explicit
description specification of the ALC reference model.
V. EVALUATION
The evaluation is done by the development of several
typical business systems fully assembled by ALC
components built with the method proposed here. As an
example, Table 3 shows the description file of the “loan
forms filling component” in the newly developed Housing
Fund Reviewing System(HFRS). The component contains
three forms to be filled in by the fund applicants, including a
loan contract form (Contract.doc), a credit rating report
(Credit.doc), and a house mortgage evaluating report
(Eval.doc). As before the development of HFRS, there had
been a long time for the staff to manually process the
application using emails and some desktop office suites like
Microsoft Word, so a collection of standard business forms
had been accumulated then. Since the forms are familiar to
the staff, they are desired to be reused in the new system of
HFRS. This is just the situation that the ALC model applies
to, and the forms are encapsulated and transformed into
ALCs using wrapper technology at low cost.
A wrapper for Microsoft Word is then be developed,
shown in Table 3 as “msword-wrapper.exe”, which invokes
and communicates with the Word application through the
opening interface supported by Microsoft Office via
COM/OLE[2]. Given the wrapper implemented in a short Figure 3. UI Loan Contract Form
period, all the components are developed simply by writing
distinct description files.
The HFRS, as a typical medium scale system popular in
business domain, is constructed by totally 24 ALCs, of
which 75% are UI-driven components constructed with the
wrapper for Microsoft word (shown in Fig. 3 and 4), and
others are components towards automatical business tasks
including document printing and email sending. As a statistic,
the development of HFRS using traditional method would
cost approximately a minimum workload of 1.5 man-month,
while with our method was done in 25 days (including the
development of several wrappers for about 10 days). It got a
efficiency improvement by 45%. Considering that the Figure 4. UI of House Mortgage Evaluating Report
wrappers are reusable resources commonly used in business
domain, future development can be further accelerated. The method has been employed in several typical
business applications like loans reviewing systems,
VI. CONCLUSION AND FUTURE WORK professional titles declaring and management systems, as
The reference model and description specification for well as revenue administration systems, where the model’s
application level components have provided the expected applicability and validity were preliminary evaluated. A
advantages in evaluation. A novel method is proposed to wider evaluation will be made involving a larger scale and
encapsulate and reuse current existing software and system scope of systems for continuously improving the model.
resources, which is a way to promote the reuse level of REFERENCES
software modules. The model is simple and flexible to use by
providing a standard framework clearly figuring out the [1] Short, K., “Component Based Development and Object Modelling”,
Texas Instruments Software, USA, 1997.
business specific properties to be focused on.
[2] SunSoft, “Component-Based Software with Beans and ActiveX”,
http://java.sun.com/javastation/whitepapers/javabeans/javabean_ch1.html
TABLE III. DESCRIPTION FILE OF LOAN FORMS FILLING COMPONENT
[3] Redolfi, G. Spagnoli, L. Hemesath, P. Bastos, R.M. Ribeiro, M.B.
ID: 5588ce2e-0808-11dc-9a1b-00a0c90a8199 Cristal, and M. Espindola, A., “A Reference Model for Reusable
Name: Loan Forms Filling Component Components Description”, Proc. of 38th Annual Hawaii International
Description: ALC for filling loan forms by applicants(V1.0, 5/11/2008) Conference on System Sciences, 2005
Type: Manual
InputData: [4] Geisterfer, C.J.M., and Ghosh, S., “Software component
OutputData: string Name, string Address, string Phone, string Sex, string specification : a study in perspective of component selection and
SN, string HouseAddress, string HouseSN, datetime AppDate, ... reuse”, International Conference on Commercial-off-the-Shelf
Applications: msword-wrapper.exe (COTS)-Based Software Systems, 2006
Forms: Contract.doc, Credit.doc, Eval.doc [5] Graubmann, P. and Roshchin, M., “Semantic Annotation of Software
FormsMapping: Name = [Contract.doc].AppliciantName, Address = Components”, 32nd EUROMICRO Conference on Software
[Contract.doc]. AppliciantAddress, …, Name = [Credit.doc]. Engineering and Advanced Applications, 2006
AppliciantName, Sex = [Credit.doc]. AppliciantSex, SN = [Credit.doc].
[6] Meling, R. Montgomery, E.J. Sudha Ponnusamy, P. Wong, and E.B.
SN, …, HouseSN = [Eval.doc]. HouseSN, …
Mehandjiska, D., “Storing and Retrieving Software Components: A
Sequence: Begin => msword-wrapper.exe:Contract.doc , msword-
Component Description Manager”, Software Engineering Conference,
wrapper.exe:Contract.doc => msword-wrapper.exe: Credit.doc, 2000
msword-wrapper.exe: Credit.doc => msword-wrapper.exe: Eval.doc, [7] Liu Y, Gao Y, Wang YH, Wu L, Wang LF, “A Component Based
msword-wrapper.exe:Eval.doc => End Geo-Workflow Framework: A Discussion on Methodological Issues”,
FormConstraintConditions: [Contract.doc].AppliciantName == [Credit.doc]. Journal of Software, 2005, 16(8)
AppliciantName == [Eval.doc]. AppliciantName, … [8] Yang L, “Combining Mechanism of a Domain Component-Based
InputDataConstraintConditions: Architecture Based on Separate Software Degree Components”,
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HouseSN != NULL && ……

199
 2009 International Conference on Computer Engineering and Technology
Fault Detection and Diagnosis of Continuous Process Based on Multiblock
Principal Component Analysis
Libo Bie, Xiangdong Wang
School of Information Science and Engineering,
ShenYang University of Technology, ShenYang, 110178, China
bielibo81758@yahoo.com.cn, xdwang@sut.edu.cn
Abstract
The fault detection and diagnosis of continuous
process is very important for the production safety and
product quality. Owing to its no need to know much
about the process mechanism and exact process
model ,the data driven method, typically the principal
component analysis (PCA) has attracted much
attention by chemical researchers for monitoring
process. PCA is powerful in fault detection ,however, it
has difficulties in diagnosing fault correctly in complex
process. In this paper, the multiblock principal
component analysis (MBPCA) is applied for fault
detection and diagnosis in continuous process, which
uses the integral PCA to detect fault and block
contribution and variables contribution to diagnose
fault. The simulations on the Tennessee Eastman
process show that the proposed method can not only
detect fault quickly ,but also find the fault location
exactly.
Keywords: fault detection; fault diagnosis;
principal component analysis; multiblock principal
component analysis; block contribution
1. Introduction
On-line monitoring of continuous process
performance and fault diagnosis are extremely
important for plant safety and good product quality. In
the last decade, many approaches have been applied to
this field, which could be classified into three
categories: methods based on principle model, methods
based on knowledge and methods based on data-driven.
For the model method, many factors, such as high
process nonlinearity, high dimensionality, and the
complexity of a process can make it very difficult to
develop; whereas for the knowledge method, the lack
of knowledge may confine its application. As an
alternative, the data driven method, which only needs
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DOI 10.1109/ICCET.2009.107
historical data of normal operation condition has
attracted much interest by chemical engineers.
One of the data-driven methods which obtains
widespread availability is principal component
analysis(PCA)[1]. Because of the favorable features of
PCA, their method can handle high dimensional and
correlated process variables. The main point of the
approach is to use PCA to compress normal process
data and extract information by projecting the data
onto a low dimensional score space. The confidence
limits of Hotelling T2 and SPE statistics in those
subspaces can be used to detect fault and the variable
contribution plot was introduced for fault diagnosis.
Simulation results show these methods are simple and
powerful[2].However, the fault diagnosis based on
variable contribution plot in PCA method may make
mistakes, especially when the process is complex and
the number of monitoring variables is large. To
enhance the PCA model’s explanation and fault
diagnosis ability ,researchers[3,4] propose the approach
of multiblock principal component analysis(MBPCA)
or hierarchical principal component analysis(HPCA).
The key idea they proposed decomposes the large-scale
system into function-independent block or
site-independent block. When a fault is detected, first
judge which block the fault locates in and then
determine the fault position in that block ,thus the fault
diagnosis ability has been improved.
In this paper, a novel continuous process fault
detection and diagnosis technique based on MBPCA
and block contribution[5] is proposed. The confidence
limit of SPE statistics in the integral PCA model is
used to detect fault and the block contribution and
variable contribution plots are applied to diagnose the
fault. The rest of the paper is organized as follows. In
section 2, the concept of PCA is introduced. In section
3, the approach of MBPCA, combined with block
contribution and variable contribution is developed to
monitor process and diagnose fault. The method is
demonstrated on the Tennessee Eastman process in
section 4.
2. Principal Component Analysis(PCA)
As a data-driven method, PCA has been used
successfully for detecting faults in handling highly
correlated variables. The essence of PCA is projecting
the process data onto the principal component space of
lower dimensionality to compress data and extract
information. Fault detection can be carried out by
confidence limit of SPE or T2 statistics and in simple
increase the difficulty of analysis and diagnosis. Many
approaches have been proposed for selecting the
number of the principal components to be retained in a
PCA model[7]. In this paper, the simple but widely used
cumulative percent variance(CPV) method is proposed ,
which judges that the cumulative percent variance
captured by the first k principal components is bigger
than the predetermined limit. As follow,
k m

process fault diagnosis can be realized by variable （ ∑λ i ∑λ i ） × 100% >65%--90% (6)

contribution plot. The data matrix X(n × m), which has i =1 i =1

been collected under normal operation and
standardized to zero mean and unit variance comprises
n samples and m process variables. PCA is intended to
construct the independent variables of less numbers to
reflect the information of the original data matrix as
more as possible, which are linear combination of the
original m variables. The information is measured by
variances and PCA actually on the eigenvalue
decomposition of covariance matrix C[6].
X T X
C= n − 1
Cpi=λi pi (2)
where C defines the covariance matrix of X , pi
represents the orthogonal and unit eigenvector of the
covariance matrix C; λi is the eigenvalue according to
reducing order. Thus PCA can transform the matrix X
into the following form:
m
Having established a PCA model based on normal
process data, future behavior can be referenced against
this “in-control” model. This comparison can be
carried out in the principal component space or the
residual space, corresponding the T2 statistic or the
SPE statistic.
For concision, the formula (5) can be rewritten as
the following form:
X=TkPkT+ TePeT (7)
where Tk（n × k） and Pk（m × k） are the principal
component score and loading matrices. Te（n × (m-k
(1)
）) and Pe（m × (m-k）) are the residual score and
loading matrices respectively.
Suppose a new process sample x is collected, the
squared prediction error(SPE), which indicates the
amount by which a sample deviates from the model, is
defined by
SPE=eeT=(x- x )(x - x )T=x (I- Pk PkT) xT (8)

∑t and
T
X＝t1p1 +t2p2 + …
T T
+tmpmT＝ i pi (3)
t=xPk (9)
i =1
where pi is the same as the above definition and named x =tPkT (10)
as principal component loading vector and ti is the e=x- x (11)
corresponding score vector. where t, x , e, I are the score , prediction value,
ti=Xpi (4) prediction error and unit matrix respectively.
PCA is aimed to explain the main variation of the As PePeT= I- Pk PkT, the formula(8) can be rewritten
data matrix X by choosing the proper anterior k as
principal component score vectors. Supposed k SPE=eeT= xPePeT xT (12)
principal components(k<m) are selected. Matrix X can According to the determinant standard, SPE can’t
be reconstructed as exceed the confidence limit Qlim under the normal
k condition, otherwise the abnormal state may appear in
∑t p
T
X=t1p1 +t2p2 + …
T T
+tkpkT+E＝ ＋E (5) the process. Assuming the normal distribution of x, the
i i
i =1 confidence limit Qlim for the SPE statistic can be
calculated using the following formulas.
where E（n × m）is residual, which is the difference 1

= θ 1 ⎡1 + ⎤ h0
2 θ 2 h0 2 θ 2 h0 ( h0 −1)
+
between original data and data reconstruction with the Ca
Q lim
anterior k principal components, usually describing the ⎢⎣ θ1 θ12 ⎥⎦
noise information. m
θ = ∑ λ i
The number of the principal components retained i j
j = k +1 (13)
in the PCA model is a critical issue in fault detect .If 2 θ 1θ 3 (i=1,2,3)
the number is large, the model will be h0 = 1 − 3θ 2 2
overparameterized and include noise. On the other
hand ,if the number is small, that will not effectively
capture the information of the original data set and may

201
where λi is the eigenvalue of the covariance matrix C,
Ca is the standard normal deviate corresponding to the
upper a percentile.
When SPE of the new sample exceeds the
confidence limit of the PCA model , the abnormal
situation may occur and can be judged, but can’t be
assured what place the fault appears in the process.
That is the task of the fault diagnosis and can be
realized by the variables contribution plots in the
simple process. The idea behind contribution plots is
simple. When monitoring a process, each variable will
have a unique impact on the SPE and T2. The larger the
contribution of an individual variable, the more likely
it is that the variable is the source of the excursion. The
contribution of the SPE of the jth process variable at the
ith observations is as:
Qij=eij=(xij- x ij)2 (14)
Although variables contribution plots are effective
to diagnose fault in simple process, it has difficulties in
complex process. When highly correlated variables are
monitored, significant contributions may appear across
several of them and the variables contribution plots
method can’t determine the exact location of the fault.
The next section is attempted to resolve this problem
by using the concept of block contribution and
multiblock principal component analysis(MBPCA).
3. Fault Detection and Diagnosis Based on
MBPCA and Block Contribution
Multiblock principal component analysis (MBPCA)
or the hierarchical principal component analysis
(HPCA) divides the large-scale process into several
blocks to enhance the model’s ability of explaining and
diagnosing. Westerhuis JA et al [8] showed the scores
in the MBPCA can be calculated directly from the
regular PCA. S. Qin, S. Valle et al [5] approved the
loadings can also be calculated from the regular PCA
algorithm and pointed further that the equivalence
between multiblock analysis algorithms and regular
PCA indicated that the scores and loadings in the
MBPCA can be obtained by simply grouping the
scores and loadings in the regular PCA. Qin, S. Valle
et al also proposed the concept of block contribution to
assist fault diagnosis. In this section, fault detection
and diagnosis will be accomplished by the MBPCA
and block contribution. The basic procedures can be
summarized as follows.
1 Collect process data under normal condition,
pretreat and build the regular PCA model of all
variables.
2 Block the scores and loadings acquired by the
procedure 1 according to process knowledge and
calculate the confidence limit of block SPE.
202
3 Collect a new sample, monitor the process
using the SPE in procedure 1 and a fault is detected if
SPE is out of limit continuously.
4 Once a fault has been detected, calculate the
block SPE. The block whose SPE is out of limit
severely can be considered as the source of fault or
mainly affected by the fault.
5 Plot the variable contribution of that block and
localize the root cause of the fault with process
knowledge.
3.1 Blocking
Suppose the regular PCA mentioned in section 2
has been established to get the principal component
loading matrix Pk（m × k） and residual loading matrix
Pe（m × (m-k）). As defined in [5], prior knowledge
can be used to divide the m variables into B blocks,
which the bth block contains mb variables. Loading
matrix Pk and Pe can also be parsed as follows.
Pk1 Pe1
Pk= : P e= : (15)
Pkb Peb
: :
PkB PeB
where Pkb (mb × k) and Peb (mb × (m-k)) are the
principal component loading and residual loading
matrix of the bth block respectively.
Given a new sample x, separate the data vector into
B blocks and the bth block is xb (1 × mb).
3.2 Block SPE statistics
Similar as equation (12), the block SPE of the bth
block is
SPEb=ebebT= xbPebPebT Peb PebT xbT (16)
Note that PebPebT is not the idempotent matrix.
Since SPEb is a quadratic form of a multivariate normal
vector x, it can approve that SPEb is distributed
approximately as gbχ2(hb), where
gbspe =tr{(PebTPebΛ)2}/ tr{PebTPebΛ} (17)
hbspe =(tr{PebTPebΛ})2/ tr{(PebTPebΛ)2} (18)
and Λ is the diagonal matrix composed of the residual
eigenvalues, namely Λ=diag{λk+1, … λm}. Therefore
the confidence limit for SPEb is δb=gbspeχa2(hbspe) for a
given confidence 1-a.
3.3 Variable contribution in a block
The jth variable in the bth block is defined as
SPEb,j=(xb,j- x b,j)2 (19)
where xb,j is the jth variable of the bth block and x b,j is
the prediction value of xb,j by the PCA model.
4. Case Study
The proposed approaches are applied to the
well-known benchmark process, Tennessee Eastman
process, which was developed by Downs and Vogel [9].
As shown in Fig 1, the process consists of five major
units: a reactor, a product condenser, a vapor-liquid
separator, a recycle compressor and a product stripper.
As the plant is open loop and unstable, many control
schemes have been proposed and in this paper we used
the control scheme in [10]. The process has 12
manipulated variables, 22 continuous process
measurements and 19 composition measurements..
Fig .1. Tennessee Eastman process
In this paper, 22 continuous variables are selected
17 Underflow 20 Compressor power
18 Temperature 22 CW temperature
19 Steam flow
Calculate the control limit of block SPE statistics under
the 99% confidence limit respectively.
Then the first abnormal case is considered and the
process is simulated for 20 hours. The fault 1 A/C
composition ratio in feed stream 4 with a step change
was introduced at the 5th hour and concluded at the
5.5th hour, namely the fault 1 is introduced between the
100th and the 110th sample. The SPE chart under fault 1
is shown in Fig 2. The figure shows SPE oversteps the
99% confidence limit at sample 105, detecting the fault
quickly. To compare the traditional PCA with the
proposed method for fault diagnosis, variables
contribution plot of SPE at sample 105 is present at Fig
3. It displays there are 3 variables contribute most for
the SPE, which are variable 11(separator temperature),
variable 20(compressor power), variable 21(reactor
cooling water inlet temperature). The difference of
contribution among them is not obvious and it is
difficult to determine which is mostly affected by the
fault, If we only consider the variable which has the
maximum contribution as the excursion of the fault, it
will be believed that the variable 21 is the source of
fault or mostly affected by the fault. Making such an
decision will lead to check around the compressor and
obviously make a mistake. In fact, the fault 1 is located
in the material section. So the fault diagnosis based
only on the variable contribution plot can’t determine
correctly the position of the fault, especially in the
complex process.
800

for monitoring and the sampling interval is set to 3 700

minutes. Collect data in the normal operation condition 600

500

during the 25 hours simulation time, which generates

PE

400
S

500 samples as the reference set. Ten principal 300

components are selected, which captures 69.87% of the 200

variation in the reference set. Control limit of SPE 100

99%confidence limit

statistics is 16.357 which is based on the 99% 0 50 100 150 200

Samples
250 300 350 400

confidence limit. The process is divided into five Fig .2 . SPE chart for fault 1
25
blocks, namely the material block, the reactor block,
the separator block, the stripper block and the 20
te

compressor block. These 22 variables are grouped into

nra

15
u
tribtio

five blocks according to their locations in the plant, as

Econ

10
SP

shown below. 5

一. Material 二 Reactor 三 Separator

0
0 5 10 15 20 25
1 A feed 6 Feed rate 11 Temperature Variables

2 D feed 7 Pressure 12 Level Fig.3. Variable contribution plot at

3 E feed 8 Level 13 Pressure sample 105 for fault 1
4 A and C feed 9 Temperature 14 Underflow To resolve the problem, we applied the proposed
21CW temperature method to this case. After a fault has been detected, we
四.Stripper 五.Compressor first draw the block contribution of SPE of each block
15 Level 5 Recycle flow and determine the most likely block containing fault by
16 Pressure 10 Purge rate comparing their ultralimit. Then we draw the variable

203
 Block 1 Block 2 Block 3
20 150 20

contribution plot of that block and combine the process 15

100
15

knowledge to diagnose fault. Fig 4 is the plot of block 10 10

50
SPE under fault 1. It shows that the fault 1 is serious 5 5

0 0 0
causing all the blocks having obvious out-of-limit and 0 200 400 0 200 400 0 200 400

Block 4 Block 5
block 1(material block) is most likely the source of the 20 20

fault as it has the maximum ultralimit. The variable 15 15

10 10
contribution plot of SPE at sample 105 in block 1 is 5 5

shown in Fig,5. It is apparent that the most 0

0 200 400
0
0 200 400

contribution ratio for SPE is the variable 1(A feed), so Fig.7. Block SPE chart for fault 2
the fault is produced through the variable 1 or affects it
140

120

most. Having knowing the process knowledge that the 100

te
nra
A/C ratio should be kept at a fixed value and the great

tio
80

cn
o u
trib
60

E
excursion of variable 1 just compensates the change of

P
S
40

A/C ratio in stream 4, the operator can easily confirms 20

the fault is produced through the abnormal variable 1 2 3

Variables in bloc k 2
4 5

4(A and C feed), thus determining the location of fault

exactly. Fig.8. Variable contribution of block 2 at
800
Block 1
150
Block 2
150
Block 3
sample 102 for fault 2
600
100 100
400

200
50 50
5. Conclusions
0 0 0
0 200 400 0 200 400 0 200 400

Block 4 Block 5
150
400 In this paper, we apply the MBPCA to detect and
100 300

200
diagnose fault. First build the PCA model of all
50

0
100

0
variables to detect fault and then determine which
0 200 400 0 200 400
block the fault locates in by the block contribution.
12
Fig.4. Block SPE chart for fault 1 Finally diagnose the fault according to the variable
10
contribution of that block combined with the process
knowledge. Simulation on the TE process shows the
te

8
ib n
tio
u ra

6
effectivity of this method.
P
SEo
cntr

0
1 2
Variables in bloc k 1
Fig.5. Variable contribution of block 1 at
sample 105 for fault 1
The other abnormal case is tested for the fault 2
reactor cooing water inlet with a step change. As same
as the case 1, the process is simulated for 20 hours,
introducing the fault at 5th hour and concluding at 5.5th
hour. The SPE for fault 2 in Fig 6 shows the
out-of-limit is at sample 102, detecting fault timely.
Fig 7 is the plot of block SPE and it displays fault 2 is
not serious as fault 1. The trend in block 2 appears
strongest and is most likely the site of fault. Fig 8
shows variable 9(reactor temperature) is affected most
by the fault. With process knowledge, the empirical
operator is inclined to suspect the reactor cooling water
inlet temperature has an abnormal change, thus finding
the cause of fault.
250
3
4
References
[1] Jackson J E.A., User’s Guide to Principal Component,
New York: Wiley-Inter-Science, 1991.
[2] Gang Chen and ThomasJ.McAvoy Predictive On-line
monitoring of continuous processes
[3] Wangen, L.E., and Kowalski, B.R. A multiblock partial
least squares algorithm for investigating complex
chemical systems[J]. J.Chemometr., 1988 3:3-20.
[4] Ra nnar, S., MacGregor,J.F.,and Wold,S.Adaptive batch
monitoring using hierarchical PCA[J], Chemometrics
Intell.Lab.Syst 1998,41:73-81
[5] S. Qin, S. Valle, and M. Piovoso, “On unifying multi-block
analysis with applications to decentralized process
monitoring,” J. Chemometrics, vol. 15, pp. 715–742,
2001.
[6] Jincheng fan et.al Data Analysis Science publication
[7] W. Li, H. Yue, S. Valle, and J. Qin, “Recursive PCA for
adaptive process monitoring,” J. Proc. Cont., vol. 10,
pp. 471–486, 2000.
[8] Westerhuis JA, Kourti T, MacGregor JF. Analysis of

multiblock and hierarchical PCA and PLS models.

200
J.Chemometrics 1998; 12: 301–321.
150
[9] Downs, J. J. and Vogel, E.F. A plant-wide industrial
P
SE

100
process control problem[J]. Computers and chemical
50
99% c onfidenc e limit
Engineering, 1993, 17(3):245255.
0
0 50 100 150 200 250 300 350 400
[10] Self-Optimizing control of a large-scale plant:the
Samples
Tennessee Eastman process Larsson, T., et al., Ind.
Fig.6. SPE chart for fault 2 Eng.Chem. Res., Vol. 40, pp. 488

204
 2009 International Conference on Computer Engineering and Technology
Strong Thread Migration in Hetereogeneous
Environment
Khandakar Entenam Unayes Ahmed, Md. Al-mamun Shohag, Tamim Shahriar, Md. Khalad Hasan, Md. Mashud
Rana
Dept. of Computer Science & Engineering
Shahjalal University of Science & Technology
E-mail: tanvir-cse@sust.edu, shohag_2002@yahoo.com, subeen@acm.org, khalad-cse@sust.edu, masudcse@yahoo.com
Abstract—This paper provides a complete framework for
thread migration using JPDA. In our framework neither we
lose portability nor do we insert any artificial code. The system
developed based on our framework needs no extra involvement
of programming to continue the whole migration system. It is a
powerful autonomous system for heterogeneous environment
maintaining its portability since Java Virtual Machine (JVM)
is not modified.
Keywords- distributed computing; strong thread migration;
sending agent; receiving agent; capture debugger; restoration
debugger.
I. INTRODUCTION
In Java Programming Language the mobility is
considered during design. Source code is compiled into a
transient state called bytecode (an instruction set that is very
similar to the one of a hardware processor). The Java Virtual
Machine then interprets the byte code into an executable
form. This makes the developers’ work easy where a user
can run and compile his code once and then can run in any
machine with JVM. This makes Java convenient for
implementing distributed applications. The Object
Serialization allows java to be transmitted or stored which
enables a process to be started at new hosts with an initial
state that is always starts from the beginning of the process.
Our concern is here where we can start the process from the
point at which the process is transmitted. That is we need to
transmit/store the state of the execution. So we have to
serialize/deserialize the thread. So our goal is to develop a
complete framework for thread migration. Thread migration
or thread serialization has many applications in the areas of
persistence and mobility, such as check pointing and
recovery for fault tolerance purpose, dynamic
reconfiguration of distributed systems, dynamic load
balancing and user’s traveling in mobile computing system.
In this project the JPDA (java platform debugging
architecture) is used to capture and restore the execution
state for dynamic thread migration. The focus in this project
is on the migration of the execution state instead of focusing
on transfer of objects since this facility is provided by the
JAVA’s transport protocol.
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.207
II. RELATED WORKS
M. Fuad et. al. [2] developed provides a mechanism of
strong thread migration. Their mechanism is based on a pre-
processor that inserts code into the program to be migrated.
It can suspend the thread only inside run method, not in any
other method. The second limitation is code growth rate. T.
Sekiguchi et. al. [3] utilizes the java exception handling
mechanism. It first identifies the methods that are to be
migrated. Then it adds a primitive migratory for those
methods. When migration takes place a NotifyMigration
exception is generated and is handled by try – catch block.
At that time local variables are stored in state object. When
any NotifyMigration exception is thrown it is caught in the
current method and propagated to the calling method. It uses
an artificial program counter. The code growth rate is high
due to the code insertion by Pre-processor. S. Bouchenak et.
al. [1] modifies the JVM. In their framework they used two
techniques named Type Inference and Dynamic de-
optimization. In Type Inference they create user defined
stack. When it serializes a thread it maintains all information
about local variables, operand stack, operand and pc. In
Dynamic de-optimization technique it uses the user defined
stack named type stack and type frame to restore the states.
Though the full concern was on reducing performance
overhead but they lose the portability. A. Dorman [8] also
uses JPDA for state capture and restoration. His complete
focus was on capturing and rebuilding the execution context
of running objects and not on particular means of
transportation.
III. JAVA PLATFORM DEBUGGING ARCHITECTURE(JPDA)
A debugger developer may hook into JPDA at any layer.
Since the JDI is the highest level and easiest to use
developers are encouraged to develop debugger using JDI.
Debuggers are able to ‘hook into’ programs running in the
JVM by requesting notification of events fired by the
framework, e.g. variable modification, exception, stepping,
method entry and method exit events. These events are
placed in an ‘EventQueue’, from which the debugger may
consume and further query the running program. While the
JPDA provides access to information not normally available
in standard Java applications, it also limits access in other
areas. In querying the JVM for running objects, the debugger
205
is not able to obtain direct object references, limiting the
ability to use an object in the same manner as a conventional
program. All references and values returned by the JVM are
mirrors of the actual values. For the most part this poses few
problems, as it is possible to both get and set local instance
variables. A notable exception is that of the current execution
point within a stack frame. There exists a location() method
for the retrieval of the execution point, but in an effort to
enforce security constraints within the Java environment
there is no accompanying setLocation().
IV. JVM CHARACTERISTICS
Before you begin to format your paper, first write and
save the content as a separate text file. Keep your text and
graphic files separate until after the text has been formatted
and styled. Do not use hard tabs, and limit use of hard returns
to only one return at the end of a paragraph. Do not add any
kind of pagination anywhere in the paper. Do not number
text heads-the template will do that for you.
Finally, complete content and organizational editing
before formatting. Please take note of the following items
when proofreading spelling and grammar:
JVMDI - Java VM Debug Interface
Virtual Machine
Back-End
JDWP - Java Debug Wire Protocol
Front-End
User Interface
JDI - Java Debug Interface
Figure 1: Java Platform Debugging
Architecture
The JVM data areas that are related to our project are
described below:
• A Java stack is associated with each thread in JVM. A
new frame is pushed onto the stack each time a java method
is invoked and popped from the stack when the method
returns. A frame includes a table with the local variables of
the associated method, and an operand stack that contains the
partial results (operands) of the method. A frame also
contains registers such as the program counter (pc), and top
of the stack.
206
• The heap of the JVM includes all the java object
created during the lifetime of JVM. The heap associated with
a thread consists of all the objects used by the thread (objects
accessible from the thread’s java stack).
• The method area of the JVM includes all the classes
that have been loaded by the JVM. The method area
associated with a thread contains the classes used by the
thread (classes where some methods are referenced by the
thread’s stack).
We access java stack, heap and method area using JPDA
as it provides convenient way to access them.
V. PROJECT GOALS
Using JPDA and byte code modification this project will
build up a complete framework for the dynamic thread
migration. The total project follows the following steps.
• The capturing of the execution state of a process using
JPDA.
• Display of all capturing states for research purpose.
• The byte code of the class is transferred using TCP/IP
protocol.
• Build up an agent that will receive the byte code in
receiver and then execute the byte code under the debugging
mode.
• Successfully suspend the process and modify the byte
code and restore all of the variables and resume the process.
VI. OVERALL DESIGN
In this project the design process concentrates on state
capture and restoration of running objects along with the
transportation of mobile objects. The capture and restoration
is done using JPDA without inserting any code or modifying
the JVM. The JPDA allows stopping the execution of any
running objects and accessing the exact location of execution
state and then store the location along with the local
variables. The total capture and storage are done by
developing a debugger using JDI named ‘CaptureDebugger’.
Then the mobile object is serialized along with the stored
value of local variables. Both execution and transmission are
monitored by an agent named ‘SendingAgent’. There will be
also a receiving agent that will receive the serialized data and
then run the mobile object in listening mode that will be
traced by another debugger in sender side named
‘RestorationDebugger’. RestorationDebugger stops the
mobile object in remote machine and then restore the state
and values of the local variables and starts the process. The
system consists of two agents named sendingagent and
receivingagent for sender and receiver respectively.
Sendingagent is responsible for executing the Algorithm: SendingAgent
‘CaptureDebugger’ and user can take decision of migration 1. Start
using sendingagent. The total migration process is 2. Check for any request for migration.
maintained by sendingagent. The receivingagent is always in 3. If there is a request to migrate
listening mode by running an accepthread which can receive then goto step 4.
the mobile object along with the variables. Then the Else goto step 2.
receivingagent will be responsible for running the received
4. Setup Connection with target
mobile object in suspend mode that will be restarted only
machine.
when the second debugger named ‘RestorationDebugger’
will be connected. RestorationDebugger will restore the 5. Send the class file of mobile object to
captured values and restart the process. Four major target machine.
components that will be greatly responsible for making the 6. Execute CapturDebugger.
migration success are: 7. Check wheter the mobile object
started in target machine.
• Sendingagent: Responsible for executing 8. If the mobile object is started
‘CaptureDebugger’ and migrating the mobile objects along then goto step 9.
with captured information(execution state, values of local Else goto step 7.
variables etc). It will also responsible for setting up 9. Execute RestorationDebugger.
connection with the receiver and launching the second 10. Check for further migration request.
debugger named ‘CaptureDebugger’ that will trace the 11. If more request(s) then goto step 4.
mobile objects in remote machine (receiveer). Else goto step 12.
12. Exit
• CaptureDebugger’: It will stop the running process and
will capture the execution state (pc), thread name, signatures
and local variables. It will dump all captured information in a
stack list named stackframelist.
B. Receiving Agent
• Receivingagent: It will be responsible for receiving the
mobile object along with captured information and executing Receivingagent will start its function by opening a socket
the mobile object in suspension mode. It will also notify the in a specific port number. Then it will start a thread named
sender about the start of execution of mobile object. AcceptThread. AcceptThread will first accept the server
socket and then get the class name, size and class file sent by
• RestorationDebugger: It will be started by the sendingagent of sender. Create a file named class name and
sendingagent. It is responsible for tracing the mobile object then execute the class using exec() method of untime class in
in remote machine. It will then restore the captured suspended debugging mode. Then it will send a notification
information and then restart the mobile object. about the start of the mobile object to the sendingagent of
sender. This thread will continue its execution until the
server socket becomes null.
A. Sending Agent
Sending agent will start its duty only when the migration
decision will take place. At first it will set up connection Algorithm: ReceivingAgent
with receiving agent using stream socket connection. Then 1. Start
using this socket it first read the class file of the mobile 2. open a server socket
object and then send the class file along with its name and 3. Start a thread named AcceptThread.
size. Then Sendingagent execute the debugger 4. check for the connection.
‘CaptureDebugger’ using the method exec() of Runtime 5. if connection is closed
class. It will always keep itself in communication with then goto step 6
receiver until the migration is complete. Now it will wait for else goto step 4.
the notification that will come from receiver whether the 6.Exit
mobile object has started or not in the receiver. If it gets the
notification then it will execute the second debugger named Figure 3: Algorithm for Receiving Agent
‘RestorationDebugger’

207
C. Capture Debugger
Algorithm: AcceptThread
CaptureDebugger connects with debuggee running in 1. Start
listening mode at specific port number. It first finds out the 2. Accept the server socket
connector type by matching the specification provided 3. Get the class name, size and class file
through argument with connection specification provided by 4. Create a new class file with the same
VM. Then it connects with debuggee using this connector. name and content of the class file
To capture information the debuggee must be passive instead received.
of active. That’s why the CaptureDebugger first 5. Run the class file in specific port
passivate/suspend the debugge. To capture information it address
extracts all threads that are in debuggee. Then it finds out 6. Send a notification to the
main thread, as our first concern is with single thread, which SendingAgent
will later see the light of working with multithread. It then 7. Check the server socket
dumps the stack from this thread. Stack contains all frames,
if sever socket is null
which are popped recursively. Then from the frames the
then goto step 8
program counter (pc), method name, method signature, local
variables and its corresponding values. All information is else goto step 2
stored using a user-defined stack named stackframelist. Since 8. Exit
CaptureDebugger acts as an individual process simply Figure 5: Algorithm for AcceptThread
executed by sendingagent, so a handshaking is required to
make process communication through which we can transfer
the captured information to the sendingagent. D. Restoration Debugger

RestorationDebugger connect with debuggee running in

Algorithm: CaptureDebugger suspension mode at specific port number. It first finds out the
1. Start connector type by matching the specification provided
2. Find specific connector through argument with connection specification provided by
3. Check whether the connection is open VM. Then it connects with debuggee using this connector.
4. If the connection is open After attaching with TargetVM it will first resume the
then goto step 13 TargetVM by default. In the mean time the EventHand
Else goto step 5 thread will be started by this debugger. Then EventHand will
5. Check whether the debuggee is trace all of the method call events and will search for a
present specific method call that is inserted just after variable
6. If the debuggee is present declaration in a method. If the specified method is found
then goto step 7 then it will suspend the TargetVM. Now it will search for the
Else goto step 13 main thread and will also dump the main thread’s stack. The
7. Attach debugger with debuggee frame will be dumped too. Then the frame will be
8. Suspend the debuggee distinguished according the method signature and method
9. Extract all threads in debuggee name. Then it will update the local variables using the
10. Find main thread StackFrameList which is obtained from capture debugger.
Then it will resume the TargetVM. Here it is not guaranteed
11. Extract all frames
that the process will be started from the exact desired
12. Capture all states
location but it is ensured that all of the local variables are
13. Exit updated successfully. And also it should be noted that the
Figure 4: Algorithm for CaptureDebugger debugger will resume the process after changing the PC.

208

Algorithm: RestorationDebugger
1. Start
2. Find specific connector
3. Check whether the connection is open
4. If the connection is open
then goto step 18
Else goto step 5
5. Check whether the debuggee is
present
6. If the debuggee is present
then goto step 7
Else goto step 18
7. Attach debugger with TargetVM
8. Resume TargetVM
9. Start eventhand thread
10. Search for the sentinel method
11. If the method is found
then goto step 12
Else goto step 10
12. Suspend TargetVM
13. Get StackFrameList from
SendingAgent
14. Find main thread from all threads
15. Dump frame from stack
16. Update frame by StackFrameList
17. Disconnect debugger
18. Exit.
Figure 6: Algorithm for Restoration
Debugger
209
VII. CONCLUSION
Our proposed framework provides a new and different
way for strong thread migration. Using JPDA we provide
system independent feature in our program (as JPDA is part
of standard Java). Though we might develop a faster
mechanism for strong thread migration modifying the JVM,
we didn’t do so as we want to use our system in
heterogeneous environment, hence not to sacrifice
portability.
VIII. REFERENCES
[1] Sara Bouchenak and Daniel Hagimont. “Zero Overhead Thread
Migration.” INRIA Technical Report No. 0261. May 2002.
[2] M. Fuad and M. Oudshoorn. “AdJava - Automatic Distribution of
Java Applications.” In Michael Oudshoorn, editor, Twenty-Fifth
Australian Computer Science Conference, volume 4, pages 65 – 75,
Melbourne, Australia, 2001. Australian Computer Society.
[3] T. Sekiguchi, H. Masuhara, and A. Yonezawa. “A Simple Extension
of Java Language for Controllable Transparent Migration and Its
Portable Implementation.” In Coordination Models and Languages,
pages 211–226. 1999.
[4] Sun Microsystems. “The JavaTM Virtual Machine Specification -
Second Edition, 1999”.
http://java.sun.com/docs/books/vmspec/2ndedition/html/VMSpecTO
C.doc.html.
[5] Sun Microsystems. “Java Platform Debugger Architecture Overview,
2000.” http://java.sun.com/j2se/1.3/docs/guide/jpda/jpda.html.
[6] E. R. Harold. Java I/O. O’Reilly, 1999.
[7] M. Dahm. “Byte Code Engineering Library,” 04/12/2002 2002.
http://bcel.sourceforge.net/.
[8] Andrew Dorman, “Execution Context Migration within a Standard
Java Virtual Machine Environment.”
 2009 International Conference on Computer Engineering and Technology
A DSP-based active power filter for three-phase power distribution
systems
Ping Wei, Zhixiong Zhan,Houquan Chen
Department of Information Engineering, Nanchang University,
wp620125@yahoo.cn zhixiong3090@163.com choq521@163.com
Abstract
This paper presents a new digital signal
processor (DSP)-based control method for shunt
active power filter (SAF) for three-phase power
distribution systems. Compared to conventional
analog-based methods, the DSP-based solution
provides a flexible and cheaper method to
control the SAF. The proposed technique
requires fewer current sensors than other
solutions which require both the source current
and the load current information. Furthermore,
the proposed technique provides both harmonic
elimination and power factor correction. Finally,
the simulation and experimental result also
shows that both controller techniques can
reduce harmonics in three-phase electric
systems drawn by nonlinear loads and reduce
hardware components.
Keywords: Harmonic distortion; Shunt Active
power filter; Digital signal processors
1. Introduction
There has been a continuous proliferation of
nonlinear type of loads due to the intensive use
of electronics control in all branches of industry
as well as by general consumers. Conventional
rectifiers are harmonic polluters of the power
distribution system. Non-linear loads, especially
power electronics loads, create phase
displacement and harmonic currents in the main
three-phase power distribution system both make
the power factor of the system worse.
In recent years, digital signal processors (DSP)
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 210
DOI 10.1109/ICCET.2009.140
are being used in a variety of applications that
require sophisticated control. DSP-based
controller for active power filters has been
proposed in some papers in which general
purpose and floating-point DSPs are used. A
DSP-based three-phase active power filtering
solution is proposed in this paper. The proposed
technique uses a fixed-point DSP to effectively
eliminate system harmonics and it also provides
reactive power compensation or power factor
correction.
This paper also presents the application of
DSP-based controllers for SAF for three-phase
distribution systems. The system considered in
this paper is shown in Fig. 1. Section 2 of this
paper provides the fundamentals of SAF and the
structure of the controller is discussed. In
Section 3, simulation results verifying the
concept are presented.
2. Active power filter system
The main objective of the SAF is to
compensate harmonics, reactive power, neutral
current and unbalancing of non-linear loads
locally such that a.c. mains supplies only unity
power-factor sinusoidal balanced three-phase
currents. The SAF draws the required currents
from the a.c. mains to feed harmonics, reactive
power, and neutral current for balancing of load
currents locally and causes balanced sinusoidal
unity power-factor supply currents under all
operating conditions.
Fig. 1 shows the basic SAF scheme including
a set of non-linear loads on a three-phase
distribution system. In this paper, the SAF
consists of three single phase inverters. The
switches of SAF must support unipolar voltage
and bipolar current, an IGBT with anti-parallel
diode is needed to implement each switch.
the capacitor, and at the same time act as the low
pass filter for the AC source current. Then the
SAF must be controlled to produce the
compensating currents if1, if2 and if3 following the
reference currents i*f1, i*f2 and i*f3 through the
control circuit.

Fig.1 an SAF and nonlinear loads considered in this paper

The current which must be supported by each Fig.2.The proposed shunt active power filter

switch is the maximum inductor current. The

maximum voltage which must be supported by 2.2 System modeling
controllable switches is the maximum dc bus
voltage. The load may be either single phase, The representation of a three-phase voltages
two phase or three phase and non-linear in and currents of the VSI in Fig. 2 are as follows:
nature. In this paper, we consider three single the voltages Vf1, Vf2 and Vf3 supplied by the
phase uncontrolled diode bridge rectifiers with inverter as a function of the capacitor voltage, Vc,
resistive–capacitive loading as non-linear and the state of the switches G1, G3 and G5 are:
unbalanced loads. This load draws a non- ⎡Vf 1⎤ Vc ⎡−2 1 1 ⎤ ⎡G1⎤
⎢Vf 2⎥ = ⎢ 1 −2 1 ⎥ ⎢G3⎥
(1)
sinusoidal current from the utility. ⎢ ⎥ 6 ⎢ ⎥⎢ ⎥
⎣⎢Vf 3⎦⎥ ⎢⎣ 1 1 −2⎦⎥ ⎣⎢G5⎦⎥

2.1 Description of proposed filter Where, G1, G3 and G5 represent three logic
variables of the three legs of the inverter. The
As shown in Fig.2, the SAF system consists inverter conduction state is represented by these
of a three phase voltage inverter with current logics. Then the SAF currents can be written as:
regulation, which is used to inject the dif 1
Lf1 = VS 1 − V f 1 (2)
compensating current into the power line. The dt
VSI contains a three-phase isolated gate bipolar dif 2
Lf 2 = VS 2 − V f 2 (3)
transistor (IGBT) with anti-paralleling diodes. dt
The VSI is connected in Parallel with the dif 3
Lf 3 = VS 3 − V f 3 (4)
three-phase supply through three inductors Lf1, dt
Lf2 and Lf3. The DC side of the VSI is connected Where, if1, if2 and if3 are SAF currents and Vs1,
to a DC capacitor, Cdc that carries the input Vs2 and Vs3 are the supply voltages. The voltage
ripple current of the inverter and the main in the DC capacitor can be calculated from the
reactive energy storage element. The DC SAF currents and switching function as follows:
capacitor provides a constant DC voltage and the
∫ [G i
1 + G 3if 2 + G 5if 3] (5)
VC = 1 f1
real power necessary to cover the losses of the C
system. The inductors Lf1, Lf2 and Lf3 perform The set point of the storing capacitor voltage
the voltage boost operation in combination with must be greater than the peak value of the line

211
neutral mains voltage in order to be able to shape
properly the mains currents.
2.3 Capacitor voltage with PI controller
The basic operation of this proposed control
method is shown in Fig. 3. The estimation of the
reference currents from the measured DC bus
voltage is the basic idea behind the PI controller
based operation of the SAF. The capacitor
voltage is compared with its reference value,
V*c, in order to maintain the energy stored in
the capacitor constant. The PI controller is
applied to regulate the error between the
capacitor voltage and its reference. The output of
PI controller is multiplied by the mains voltage
waveform Vs1, Vs2, Vs3 in order to obtain the
supply reference currents i*s1, i*s2, i*s3. Then the
supply reference currents are proportional to the
mains voltages. The three-phase compensating
reference current of SAF (i*f1; i*f2; i*f3) are
estimated using reference supply currents and
sensed load currents.
rectifier commutations. In this paper, we give
several simulation results with uncontrolled
rectifier at α=0°.
The simulation results in steady state
operation are presented. Fig.5 shows the
performance of the SAF system using PI
controller. Waveform of the source current
without SAF is shown in Fig. 5(a). Fig. 5(b)
shows the source current with SAF
superimposed to the supply voltage. Fig.5(c)
shows the compensation current. The capacitor
voltage superimposed to its reference is shown
in Fig.5 (d). It is noticed that the supply current
in phase with the supply voltage, and the
capacitor voltage follow its reference.
Fig.4. SAF connected in parallel with nonlinear loads.

Fig.3.The proposed control system

3. Simulation results

A number of simulation results with different

operating conditions were developed. In Fig. 4,
SAF is connected in parallel with nonlinear load.
Then, the three-phase controlled rectifier with
resistive load has to be compensated by the SAF.
In order to limit maximum slope of the rectifier Fig.5. Simulation results of proposed SAF using PI controller.

current, a smoothing inductor, Lr, has been (a) Source current without filter. (b) Source current and

inserted before the rectifier. Also, it is prevented source voltage with filter. (c) Compensating current. (d)

the inverter saturation even in correspondence of Capacitor voltage with its reference.

212
4. DSP implementation and Experi-
mental results
A laboratory prototype of the active filter has
been built to evaluate the performance of the
proposed active filter and its implementation in
the TMS320F240 DSP. Fig. 6 shows the block
diagram of the experimental setup. Control and
gating signals for the switches of the active filter
are generated on a TMS320F240 DSP. The DSP
is connected to a computer through a XDS510PP
emulator. The Texas Instruments (TI)
TMS320F240 processor is a fixed-point 16-bit
processor, operating at 20 million instructions
per second (MIPS), and has the capabilities of
using advanced PWM motor control techniques
for adjustable speed drives. The XDS510PP is a
portable scan path emulator capable of operating
with the standard test interface on TI DSPs and
Microcontrollers. This portable emulator works
of the computer parallel port.
Fig.6. Experimental setup of proposed DSP controlled
active filter.
Fig. 7 shows experimental waveforms for the
load condition of uncontrolled rectifier. Fig. 7(a)
shows the supply voltage and current without
ASF. Fig. 7(b) shows the supply phase voltage,
supply current and its reference. It is clear from
Fig. 7(b) that the supply current is almost sin
waveform and follows the supply voltage in its
waveform shape with almost a unity
displacement power factor. Fig. 7(c) shows the
compensating current. From these figures, it is
clear that the effectiveness of the proposed
controller for active power filter.
213
Fig.7. Experimental results of proposed SAF. (a) Source
current and voltage without filter. (b) Source current, its
reference and source voltage. (c) Compensation current.
5. Conclusions
The focus of this paper is to present a novel
DSP controlled active filter to cancel harmonics
generated in three-phase industrial and
commercial power systems. A three-phase PWM
controlled shunt active filter was designed to
inject harmonic currents with the same
magnitude and opposite phase to that of the
nonlinear load currents in order to cancel
harmonics at the point of common coupling
(PCC) with the utility. The control scheme using
three independent hysteresis current controllers
has been implemented. The operation and
modeling of the SAF have been described. A
laboratory prototype has been built to verify the
performance of the active filter. The feasibility
of the approach was proven through the
experimental results.
References
[1] H.Akagi, New trends in active filters for power
conditioning, IEEE Trans. IA 32 (6)
(November=December 1996) 1312–1322.
[2] Fujita H, Akagi H. The unified power quality
conditioner: the integration of series-and shunt-active
filters. IEEE T Power Electr 1998; 13(2):577–84.
[3] Texas Instruments-Configuring PWM Outputs of
TMS320F240 with Dead Band for different Power
Devices, Application Report SPRA289, 1997
[4] Texas Instruments-TMS320C24x DSP Controllers
Reference Set, Vol. 1 (CPU, System and Instruction
Set), March 1997, Revision A.
[5] Texas Instruments-Dead-Time Generation on the
TMS320C24x, Application Report SPRA371, 1997.
[6] Kamran F, Habetler T. Combined deadbeat control of a
series parallel converter combination used as a
universal power filter. IEEE T Power Electr 1998;
13(1):160–8.
[7] Dastfan A, Gosbell V, Platt D. Control of a new active
power filter using 3-d vector control. IEEE T Power
Electr 1999; 14(4):5–12.
[8] Chandra A, Singh B, Al-Haddad K. An improved
control algorithm of shunt active filter for voltage
regulation, harmonic elimination, power factor
correction, and balancing of nonlinear loads. IEEE T
Power Electr 2000; 15(6):495–503.
[9] Jeong S, Woo M. DSP-based active power filter with
predictive current control. IEEE T Ind Electron 1997;
44(3):329–36.
[10] Buso S, Malesoni L, Mattavelli P. Comparison of
current control techniques for active filter applications.
IEEE T Ind Electron 1998; 45(5):722–9.
[11] J. Matas, L. Garcia deVicuna, J. Miret, J.M. Guerrero,
M. Castilla, Feedback linearization of a single-phase
active power filter via sliding mode control, Power
Electron. IEEE Trans. 23 (1) (2008) 116–125.
[12] K. Nishida, M. Rukonuzzman, M. Nakaoka, Advanced
current control implementation with robust deadbeat
algorithm for shunt single-phase voltage-source type
active power filters, Electric Power Applic. IEE
Proceedings 151 (3) (8 May. 2004) 283–288.
[13] T. Toshihiko, H. Eiji, I. Norio, O. Yasushi, Y.
Masayoshi, A novel real-time detection method of
active and reactive currents for single-phase active
power filters, Power Electronics Specialists
Conference, 2007 (PESC 2007), IEEE, 17–21 June,
2007, pp. 2933–2938.
[14] D. Lindeke, S.A. Mussa, F.P. de Souza, I. Barbi, Single
phase active power filter controlled with a digital signal
processor – DSP, Power Electronics Specialists
Conference, 2004 (PESC 04), 2004 IEEE 35th Annual
vol. 2, m20–25 June 2004, pp.1024–1027
[15] Texas Instruments-TMS320C24x DSP Controllers
Evaluation Modul, Technical References, 1997.

214
 2009 International Conference on Computer Engineering and Technology
Access Control Scheme for Workflow
Gao Lijun Zhang Lu
computer school
Shenyang institute of Aeronautical Engineering
Shenyang, China
e-mail: gaolijun0610@163.com
Abstract—Access control is an important security measure in
workflow system. Based on the analysis of many access
control models, by emphasizing the weakness existing in the
present workflow system, a new access control model is
proposed with the introduction of task set, task flow, time
constraint and the formalized definition and relevant
algorithm. It overcomes the shortcomings of the traditional
model based on the role access control that has bad dynamic
adaptivity of authority assigning, no flow control and time
constraint. The model can therefore enhance the security
and practicability of the workflow system.
Keywords- workflow; tasks; task flow; time constraint
I. INTRODUCTION
Workflow is an efficient method of complex multitask
cooperative modeling with extensive application in the
enterprises’ informationization, and the most remarkable
characteristic is to divide big task into smaller tasks which
can be finished by many men cooperatively, therefore
effective task control model is needed to manage and
control the access of these cooperation staff. That’s to say,
it can not only forbid users executing unauthorized tasks
but also ensure authorized users to execute tasks smoothly.
If the system fails to provide sufficient security protection
for these cooperative staff, it is inevitable that some
personnel can execute illegal operation by the
convenience of work. The widespread mainstream model
is based on the RBAC˄role-based access control˅ model,
however the existing RBAC can’t express complicated
workflow access control constraint, especially the
shortage of control mechanism of workflow, so it fails to
control the interactive relations among tasks (e.g. order,
recurrent time constraint, etc.), and can’t satisfy all
control demands if the existing RBAC model is
introduced into workflow system straightly.
II. The shortage of RBAC model in the
workflow system
RBAC, put forward by Sandhu, etc. is the basic
framework of RBAC96[1] model, which has wide
application in the access control field, that main idea is
modify the traditional two-layer authorization
structure—user-permission into three-layer authorization
structure—user-role-permission, therefore through
relating user with permission to simplify the difficulty of
authorization management. RBAC model has its own
advantage, but has some weakness in the workflow area.
A. The dynamic property of assignment and
revocation of permission is bad, and the granularity
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.120
Xu Lei
partition of permission has bigger localization
RBAC is a kind of access control model tending to be
static, which can’t be related to the transaction process of
application tightly, so it’s difficult to authorize and revoke
the permission timely and rightly according to the need of
transaction process dynamically.
Furthermoreˈ permission is relevant to the role
directly in the traditional RBAC model, which determined
the granularity of permission can be specified up to the
role level, and the minimum permission constraint is the
the minimum permission constraint of role[2].
B. Shortage of task process control capability and the
periodic constraint capability
Take the case of a simplified firm equipment
purchasing workflow in this section, the faults of RBAC
in the workflow system are pointed out. The equipment
purchasing workflow consists of five tasks: filing an
application for purchasing(task1), making purchase
bill(task2), scrutinizing purchase bill(task3), confirming
purchase bill(task4), executing purchase(task5). The task
executing constraint of a real workflow instance is as
follows: (1) is project manager pr executes task1, financial
clerk cl executes task2, general manager role ma and
financial executive role su cooperate to execute task3, role
c1 executes task5;(2)the execution series of task is task1ˈ
task2ˈtask3ˈtask4ˈtask5,and if user A has executed task1 or
task2ˈthen A is forbidden executing task3 and task5; but if
A has executed task1ˈ then A must execute task4. task3
should activated by ma, then by su;(3) The time constraint
period is set from Jan, 2008 to Oct, 2008, and task2 can be
executed from 9 o’clock to 16 o’clock on the 15th every
moth, furthermore the execution time length of task2 is
regulated not over 30ms; the figure 1 shows the role layer
structure; in the fig. 2, user represents the user collection
of executing task, the number on the arc represents the
user number needed in this task.
.
ma
pr s
cl
Figure 1᷏ Role Hierarchy Structure
215

Figure 2ˊ Purchasing workflow
The task execution of workflow system is limited by
some conditions, including the task order constraint, task
cooperation constraint, task time constraint, which are all
hard to resolve through traditional RBAC, not to say that
the RBAC failed to recognize the task status which results
to unable to trace the accomplishment of task and
increases the difficulties of calculating the start time of
follow-up task.
III. Introducing the access control model of time
constraint T-R&TBAC based on the role and task
˄Time-Rose & Task Based Access Control˅
The above section analyzes the faults of role access
control model in the workflow systemˈbut the basic
reason is the three-level access control structure which
determined these congenital faults of RBAC. In the
workflow system, this fault is more obvious because the
concept of task is very distinct. In order to solve the
above-mentioned problem, this structure must be changed,
so it’s natural to introduce the concept of task set into
RBAC model clearly, and make the task execution limited
by time constraintˈso it’s easy to think that modify the
original structure into four-level one, and finally form a
access control model with time constraint
-----T-R&TBAC which based on the role and task, just
like the following fig. 3 illustrate.
 Definition 4 the time and status function
Figure 3᷏ Simplified T-R&TBAC Model
The model gives specific role to all users of the whole
workflow system, and assigns tasks to every role, also the
minimum access executing permission of every task. In
such a kind of access control structure, permission is not
directly related to the role, while having to be related
through task. No matter which user is given with any role
to register , it must have the permission during the
executing permission task instance period, so it’s natural
to realize the dynamic assignment and revotion of
permission, moreover specify the granulites of permission
up to the task level, and improve the dynamic adaptability
of workflow system dramatically.
T-R&TBAC model inherits the characteristics of
RBAC3[1], on which adding the workflow control
mechanism that make the current task can obtain the
prescriptive permission only in accordance with the
constraint conditions set by the model. The constraint
conditions include time constraint, task flow constraint,
216
task cooperative constraint, permission assignment
constraint, task conflict constraint, etc. The following
gives the formalization description of model and the
related algorithm realizing the above-mentioned
constraints.
Definition 1 T-R&TBAC model structure
U={u1, u2…um} is the set of all users; R={r1,r2…rn} is
the set of all roles[3,4]; T={t1,t2…to} is the set of all
divided tasks; P={p1,p2…pp} is the set of all access
permission authorities; S={s1,s2…sq}is the set of all
sessions.
UA Ž U u R ,a many-to-many mapping from user
setto role set, representing the user’s role assigned;
RT Ž R u T, a many-to-many mapping from role set to
task set, representing the role’s task assigned; TP Ž T u
P, a many-to-many mapping from task set to permission
set, representing the task’s permission assigned; SA Ž S
u U, a function mapping from session set to user set,
representing the affiliation between session and user;
RH Ž R u R define the partial order on R, which called
the role hierarchy.
Definition2 Defination of conflicting role and
conflicting task set
CR={cr1, cr2,…crh},the conflicting role set, cri  Rˈ
and make
u  U , cr  CR :| role _ set (u ) ˆ cr |d 1 ˈ
where the role_set is the function assigning role to user.
CT˙{ ct 1, ct 2,…cth},the confilicting task setˈ
cti  Tˈand make
r  R, ct  CT :| task _ set (r ) ˆ ct |d 1 ˈ
where task_set is the function assignning task to role.
Definition 3 Logic control structure of task
TSUCC˙{…tsucc iˈ…tsucc j }, TSUCC Ž T is the
front ordered set of present active task, which is the task
set must been finished before finishing this task.
TNEXT˙{ …tnextpˈ…tnextq }, TNEXT Ž T is the
back ordered task set of present active task, which is the
task set must been finished after finishing this task.
TM={ti |i  N}ˈTM is the set of all time point of
visual world, ti  TM represents one time point which
should not be accordance with real time, but can set a
function mapping from ti to real time, and i, j  N , 
tiˈtj  TMˈi<j œ ti<tjDŽ
TR={(tiˈtj)| tiˈtj  TMˈi<j}ˈis the time zone between
two time points.
Currenttime˖ ‡ o TM˖is a global atomic sub
function, which return current time.
StartTime˖T o TM˖ StartTime˄taski˅= tsˈts  TM
is the beginning time of task.
GetTaskTR˖T o TR˖GetTaskTR˄taski˅= trˈtr
 TR is the time zone of task permitted to activate.
TaskTimeLength˖T o N˖ TaskTimeLength (taski)=
nˈn  N is natural number(s˅,which can get the time
length of task permitted to be activated.
Definition 5 T-R&TBAC system status
 state={U,R,T,P,UA,RT,PT,SA,RH,CR,CT,Currenttim While ˄ʽ˄Currenttime/ ' T ˅˅ ˅
//rejudge the constraint conditions at ' T interval
e,StartTime,TSUCC,TNEXT,TaskStation}ˈ
{True ( finished 
False ( unfinished  {
TaskStation ˙ is the task
If˄ ʽConstraint_valid(state )
accomplishment status, this 16 variables set illustrates the
status of the task at that occasion.  
||GetTaskStation˄ task GetSuccTask
Definition 6 authorizing constraint judgement (taski))==false
Authorizing constraint judgement is a one variable ||ʽ(StartTime˄taski˅IN GetTaskTR (taski))
function, judging whether meeting the constraints of user,
role(including role cooperation), permission, inheritance, ||ʽ(CurrenttimeˉStartTime<TaskTimeLength
confliction, time, etc. (taski))˅
PerAccessFlag = False ˗
Constraint_valid:state o {True
False ˖
If˄ʽPerAccessFlag˅
True (VWDWXVLVDFFHSWHGE\V\VWHP
Constraint_valid(state)={False (VWDWXVLVQRWDFFHSWHGE\V\VWHP {DelTaskFrom (taski);
In order to discuss conveniently, we set a task DelTaskPermission˄taski˅;}
executing flag variable. Else
True ( WDVNLVSHUPLWWHGWRH[HFXWHG {BackWriteNextUser˄GetNextTask˄taski));
PerAccessFlag={False (WDVNLVQRWSHUPLWWHGWRH[HFXWHG Č
The resolution scheme and relevant algorithm of task flow
control and periodic constraint
Definition 7 global function definition
GetRoleTask: R o T:GetRoleTask˄r˅=T’ˈT’ Ž T get
the role’s task.
AddTaskTo:T o AT: AddTaskTo(taski)=AT, AT is
the active task set, AT Ž T,taskj  T add the task into
active task set.
DelTaskFrom: T o AT: DelTaskFrom (taski)= AT,
taskj  T delete the task from active task set.
GetSuccTask˖T o Tsucc˖GetSuccTask˄taski˅˙
Tsucc, Tsucc Ž T is front ordered set obtaining the current
active task.
GetNextTask˖T o Tnext˖GetNextTask˄taski˅˙
Tnext, Tnext Ž T is back ordered set obtaining the current
active task.
{True
GetTaskStation˖T o False ˖GetTaskStation (task )=
i status.
GetTaskPermission˄taski˅;}
}
If with the above judgments of  task, the
PerAccessFlag is still True, which show that the task is
feasible and obtain the permission, then access to relevant
resources; if PerAccessFlag is False, showing the
constraints are not satisfied, so the task will be deleted,
and the relevant resources access permission is also been
deprived.
č Conclusion
The work flow control mechanism is introduced into
the T-R&TBAC model based on the RBAC model, which
makes the workflow constrained by time, task flow, task
cooperation, permission assignment, task confliction, etc.
The extended model overcomes many inherent
shortcomings of traditional model, combining with
workflow system seamlessly, enhancing the security
greatly. To a real system, further study should been made
to solve the time efficiency and recovery of security
( WKHWDVNLVDFFRPSOLVKHG
{True
False ( WKHWDVNLVQRWDFFPSOLVKHG [1] Ravi S. Sandhu, Edward J. Coyne, Hal L. Feinstein and
, obtaining the
accomplishment of current task.
BackWriteNextUser˖Tnext o HEADNODE ˖
BackWriteNextUser˄taski˅=HeadNode˄which can be
empty˅,filling the user of the task in the back ordered
task assigned user area, indicating the back ordered task
of this task should be accomplished by the user.
GetTaskPermission: T o P:GetTaskPermission˄taski˅
=P’ˈP’ Ž Pˈtaskj  T obtain the limited permission of the
task.
DelTaskPermission˖T o P˖DelTaskPermission˄taski˅
˙DP, DP Ž P delete the permission of the task.
Algorithm description:
PerAccessFlag=True
AddTaskTo(taski)
GetTaskHeadStation(taski)
Charles E. Youman. Role-Based access control models[J]. IEEE
Computer, 1996,29(2)˖3847.
[2] Reinhardt A. Botha, Jan H. P. Eloff.Access Control in
Document-centric Workflow Systems An Agent-based
Approach[J].Computers and Security,2001,20(6):525-532
[3] Trent Jaeger. On the increasing importance of constraints.
In: Proceedings of the Fourth ACM Workshop on Role-Based
Access Control[C]. Fairfax, Virginia, USA: ACM, 1999: 33-42
[4] Ahn,G.-J.The RCL2000 language for specifying role-based
authorization constraints [Ph.D. Thesis]. Fairfax, VA: George
Mason University, 1999.
[5] Fang Chen, Ravi S. Sandhu. Constraints for role-based
access control. In: Proceedings of the ACM RBAC Workshop. MD:
ACM Press, 1996. 39-46.
[6] Gao lijun, Xu Lei. The reseach on the turning point choose
and the security recovery algorithm in TRBAC [J].Shanghai˖
Computer Engineering,2007,33(9):15

217
 2009 International Conference on Computer Engineering and Technology
A Mathematical Model of Interference between RFID and Bluetooth in
Fading Channel
Junjie Chen Jianqiu Zeng
Beijing University of Posts and Beijing University of Posts and
Telecommunications, Beijing, Telecommunications, Beijing,
100876, China. 100876, China.
Chenjunjie78@gmail.com
Abstract—In this paper, the performance of Bluetooth
under the interference of RFID is investigated. A
mathematical model is proposed to quantify the
performance degradation of Bluetooth. The proposed
model carefully takes PHY (i.e., the channel model, the
transmitted power, the distance and the modulation) and
MAC (i.e., the frequency hopping, the packet format and
the traffic load) into account. Packet Error Rate (PER),
the average delay and the throughput are selected as the
performance metrics and are analyzed in this paper. In
final, the theoretical analysis is justified by the
simulation.
Keywords-RFID; Bluetooth; Interference;BER; PER
I. INTRODUCTION
Bluetooth chips are embedded into diverse
products such as notebook computers, cellular phones,
digital cameras and headsets to facilitate the inter-
device data exchanging. Bluetooth uses frequency
hopping spread spectrum (FHSS) and hops on 79
channels in 2.4 GHz ISM band. In recent years, the
academia and industry paid much attention to RFID
technology and it will be deployed in the place (such as
the campus, the shopping mall and the hospital) where
many people use Bluetooth-enabled devices. Since the
transmitted power of Bluetooth is very limited (1 mW
in general), RFID devices emitting on a relatively large
power (up to 4000 mW EIRP) bring significant
interferences and interrupt the data exchange in
Bluetooth piconet. Therefore, the performance of
Bluetooth under the interference of RFID is
worthwhile to be evaluated to pave the way for
developing coexistence algorithms.
The interference related to Bluetooth or RFID is
extensively investigated in some previous works. In
[1], PER of 802.11b under the interference of RFID is
analyzed. The interference between 802.11b and
Bluetooth is widely researched and many anti-
interference methods are proposed in the literature such
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.193
Yuchen Zhou
Hangzhou branch of China
Telecom, Hangzhou, 310000,
China.
as [6]-[8]. The mutual interference between Bluetooth
and Zigbee is presented in [9]. The inter-piconet
interference within the Bluetooth network is illustrated
in [10]. Furthermore, IEEE specifically established
Working Group 802.15.2 to research the coexistence
problem of WLAN and WPAN operating on 2.4 GHz
band.
Although many works related to Bluetooth are
completed, to the author’s knowledge, the research
result on the interference between Bluetooth and RFID
not appears in the public literature. Therefore, in this
paper, a mathematical model is established to quantify
the performance of Bluetooth in the presence of
RFID’s interference. Packet Error Rate (PER), the
average delay and the throughput are selected as the
performance metrics. The theoretical and simulation
result of PER, the average delay and the throughput are
compared.
The rest parts of this article are organized as
follows. In section II, the interference in PHY is
analyzed and Bit Error Rate (BER) of Bluetooth under
the interference is obtained. Then, in section III, the
MAC sublayer is taken into account and the collision
time is figured out. Based on the BER and the collision
time, in section IV, the performance metrics (i.e., PER,
the average delay and the throughput) are formulated.
In final section, the simulation and theoretical results
are compared to justify the proposed mathematical
model.
II. PHY LAYER INTERFERENCE
Path loss is defined as the difference of the signal
strength in the unit of decibels between the transmitter
and the receiver. Given the required parameters such as
the distance and the frequency, the path loss can be
figured out easily by the path loss model. The
following 2.4 GHz indoor path loss model is used for
Bluetooth device [3].
218
 ⎧40.2+20log10 (d), 0.5 ≤ d ≤ 8 m (1) Q(a, b) = Q1 (a,b)
Lp, Bluetooth = ⎨ ∞ (9)
⎩58.5+33log10(d/8), d > 8 m = ∫ x ⋅ exp ⎡⎣ −( x 2 + a 2 ) 2 ⎤⎦ ⋅ I 0(ax)dx
b
where d in the unit of meter denotes the distance ∞ β
a
between the transmitter and the receiver of Bluetooth. = exp(−(a 2 + b 2 ) / 2) × ∑ ( ) I β (ab) (b > a > 0 )
Path loss follows the free-space propagation (path loss β =0 b
exponent is 2) up to 8 meters and then attenuates more Besides, in above equations, a and b is defined as:
rapidly (path loss exponent is 3.3). The model is not
Eb Eb
(1 − 1 − ρ ), b = (1 + 1 − ρ )
2 2
effective below 0.5 meter due to near-field effects. a=
Similarly, the path loss of RFID can be obtained by its 2No 2No
own path loss model. According to [5], the path loss (10)
model of 2.45 GHz RFID is given by: where Eb/No is the ratio of energy per bit to power
Lp, RFID = −147.6 + 20log(d)+ 20log(f) − 10log(GTGR) (2) spectral density (PSD) of the noise (or the
where d meters is the distance between the RFID interference). And the correlation coefficient ρ in
interferer and the Bluetooth receiver, and f in the unit above equations is given as:
of Hertz is the operating frequency. GT and GR are sin(2πβ )
ρ= (11)
antenna gains of the RFID interferer and Bluetooth 2πβ
receiver, respectively. The transmitted power of where β is the modulation index, and IEEE 802.15.1
Bluetooth is denoted by Pt in the unit of dBm and the standard [2] specifies a minimum modulation index of
received power of Bluetooth is denoted by Pr dBm. It 0.28 and a maximum modulation index of 0.35.
is known that the received power is the difference of From Eq. (6) to (11), BER as the function of Eb/No is
the transmitted power and the path loss: obtained, but BER as the function of SIR is required
Pr = Pt − Lp, Bluetooth (3) for the interference analysis. GFSK modulation of
When the distance between the RFID device and the Bluetooth has a bandwidth time (BT) of 0.5; therefore,
victim receiver is given, the path loss of RFID signal Eb/No in previous equations can be converted to SIR
can be calculated by Eq. (2). Suppose the transmitted as the following formula.
power of RFID is Pti dBm, therefore the interference Eb Eb × B Eb × (2 / Tb) 2Pr
power Pi in Bluetooth receiver is: = = = = 2SIR = 2γ (12)
N0 N0× B Pi Pi
Pi = Pti − Lp, RFID (4) Using SIR to replace Eb/No in Eq. (10), obtained:
The signal-to-interference ratio (SIR) is the ratio of the
a = γ (1 − 1 − ρ ), b = γ (1 + 1 − ρ )
2 2
received power and the interference power. Since the
noise power is very weak relative to the interference
power, the noise can be neglected in this case. Hence, (13)
the signal to interference and noise ratio (SINR), which So now, according to the parameter a and b is Eq. (13)
is used to calculate BER in general, approximates to and ρ in Eq. (11), solving the Bessel and Marcum Q
the SIR and is given by: function, BER of Bluetooth under the interference of
RFID can be figured out, and it is the function of SIR
γ = SINR ≈ SIR = Pr − Pi (5)
as follows.
Bluetooth modulator uses Gaussian Frequency Shift Pb = f (γ ) (14)
Keying (GFSK) and the envelop detection, therefore
the BER is: where SIR can be obtained easily by Eq. (1) to (5).
BER of Bluetooth as the function of SIR is illustrated
Pb = Q(a,b) − 0.5 ⋅ I0(ab) ⋅ exp( −(a 2 + b2 ) 2 ) (6) in Fig. 1, in which the solid and dotted line stand for
where Iβ(x) is β order modified Bessel function of the BER with the minimum and maximum modulation
first kind and Q(a, b) is Marcum Q function. Both are index, respectively.
specified in the following formulas. The Bessel 0
Bluetooth BER vs.SIR
10
function is: Index=0.28
∞
( x / 2) β + 2 k Index=0.35
I β ( x) = ∑ , x ≥0 (7) −1

k = 0 k !Γ( β + k + 1)
10

where the Gamma function is defined as follows.

BER

−2
∞ 10
Γ( p ) ∫0
t p −1e − t dt , p >0 (8)
The Marcum Q function is: 10
−3

−4
10
−2 0 2 4 6 8 10
SIR (dB)

219
 Figure 1. Bluetooth’s BER versus SIR
The BER obtained in Eq. (14) is the result in the
AWGN channel. However, in the real world,
considering the effect of the multipath, the received
signal has Rayleigh distributed amplitude in the fading
channel. The received power of the Rayleigh fading
follows the exponential distribution.
1
p ( x ) = e − x / Pr (15)
Pr
where Pr is the average received power of the signal,
i.e., the received power based on path loss and
shadowing alone, and the x is the instantaneous
received power. To simplify the analysis, the
interfering power of RFID is assumed to be constant;
therefore SIR is also exponentially distributed. As
illustrated by Eq. (14), the BER is the function of the
SIR. Since the SIR is randomly changing, the BER is
also a random variable. Although BER is drastically
changing, the average BER of Bluetooth can be
determined. According to the theory of probability, the
mean of BER can be formulated as follows.
+∞
Pb = ∫ Pb ( x / Pi ) p ( x)dx
−∞
(16)
− ( a +b ) x 2 2
1 +∞ 1 −
= ∫ (Q(a,b) − I0(ab) ⋅ e 2 )e Pr dx
Pr −∞ 2
III. MAC SUBLAYER INTERFERENCE
A. The Time Domain Collision
When Bluetooth piconet collocates with RFID
devices, the Bluetooth packet may overlap with the
RFID packets in time domain. As Fig. 2 illustrated, the
time domain is divided into 625-us-long slots by
Bluetooth system and each packet can occupies 1, 3 or
5 slots. To simplify the analysis, the one slot packet is
used. The slot time is denoted by TS, but the Bluetooth
packet only occupies TB of the slot time to transmit, in
general, the TB is 366 us long. Fig. 2 is the collision
model of RFID and Bluetooth packet in time domain.
In this model, the traffic load (denoted by λ) is taken
into account. For RFID, the idle time (denoted by Tidle
in Fig. 2) and the transmission time (denoted by TR in
Fig. 2) has the following relation.
1
Tidle = ( − 1)Tbusy, λ ∈ [0,1] (17)
λ
Figure 2. The time domain packet collision model
220
Since Bluetooth transmits packets in the slot, the
traffic load of Bluetooth is exactly the probability of
packet occurrence in one slot. That is, if the traffic load
of Bluetooth is λ , then the probability of packet
occurrence in one slot is exactly λ. Bluetooth and
RFID are asynchronous due to its different PHY and
MAC technologies. The packets of them have a time
offset which is a random variable and is denoted by X
in Fig. 2. It is a continuous variable and is uniformly
distributed between 0 and LR (the average inter-arrival
time of two consecutive RFID transmissions).
X ∼ U (0, LR ) (18)
In the downlink, RFID reader sends out a modulated
carrier to power up the tags as well as to carry the
command message. It is noted that the RFID reader
still sends out an unmodulated carrier to power up the
passive tag when the tag transmit the response packet
in the uplink. Hence, the RFID command packet in
forward link and the response packet in reverse link
have an equivalent interference to the Bluetooth
packet.
The collision time is defined as the time interval in
Bluetooth packet which is overlapped by the RFID
packet and is the time duration of the interference.
According to the standard, the 2.45 GHz RFID operate
at the data rate of up to 40 Kbps, and only the header
of each RFID packet will occupy the time period
greater than one time slot of Bluetooth. Hence, the
RFID packet is definitely longer than the Bluetooth
one. The collision time, TC, can be calculated as the
following equations.
[1] TB ≤TR & TB ≤Tidle
⎧TB , 0 ≤ X ≤ TR − TB
⎪TR − X ,
⎪ TR − TB < X < TR (19)
TC = ⎨
⎪0, TR ≤ X < LR − TB
⎪⎩TB + X − LR, LR − TB ≤ X ≤ LR
[2] TB ≤TR & TB >Tidle
⎧TB, 0 ≤ X ≤ TR − TB
⎪TR − X , TR − TB < X ≤ LR − T B
⎪ (20)
TC = ⎨
T B − Tidle, L R − TB < X ≤ TR
⎪
⎪⎩TB + X − LR, TR < X ≤ LR
B. The Frequency Domain Collision
Bluetooth system occupies from 2.4 to 2.4835
GHz, but in order to comply with out-of-band
regulations in each country, a guard band is used at the
lower and the upper band edge. The operating
frequency is:
f = 2402+ k MHz, k = 0,...,78 (21)
Each RF channel is 1 MHz wide and 79 channels are
available. Bluetooth system adopting FHSS can hop on
the 79 channels uniformly; therefore, the probability of
each channel being occupied is Pf =1/ 79.
RFID also adopts FHSS to avoid the inband
interference and can hop on 100 channels, each of
which has a bandwidth of 1 MHz. The operating
frequencies of the RFID are:
fC = (2931 + m) × fCH ,
fCH = 0.8192 MHz, m = 0,1,...,99
(22)
where fCH is the frequency spacing, and m is the
channel number ranged from 0 to 99. Since the
frequency spacing is less than the bandwidth, each
channel of RFID is partially overlapped with its
adjacent channels.
Figure 3. The frequency domain collision model
Since the channels of RFID and Bluetooth are
unaligned, the offset of two different channels is a
random variable that denoted by X in Fig. 3. To
simplify the analysis, we assume that the power of
Bluetooth and RFID is uniformly distributed in the
occupied channel. The shadowing area in the figure is
the area which the two channels are overlapped with.
And the symbol S denotes the ratio of the shadowing
area and the power of the whole signal. The S can be
easily obtained according to the offset X as the
following formula illustrated.
⎧0, X <0
⎪ X, ∈ (0,1]
⎪ X (23)
S=⎨
⎪ 2 − X, X ∈ (1,2]
⎪⎩0, X >2
The mean power of one signal is overlapped can be
figured out as follows.
1 2
S = ∫ XdX + ∫ (2 − X)dX = 1 (24)
0 1
Hence, the probability of frequency collision of
Bluetooth and RFID is the product of the mean
overlapping area and the probability of the Bluetooth
channel occupied. It can be calculated by:
PC = Pf × S = 1/79 (25)
IV. PERFORMANCE OF BLUETOOTH
UNDER THE INTERFERENCE
A. Packet Error Rate
The collision time TC is the time interval of one
Bluetooth packet overlapped with the interfering RFID
packet, which is analyzed in section III. However, if
the packet of RFID and Bluetooth are transmitted by
different frequencies, the inband interference is
negligible. Therefore, the actual collision time taken
221
into account both the time domain and the frequency
domain is the product of the collision time and the
probability of frequency collision. So the actual
collision time is:
T = TC × PC = TC /79 (26)
The actual collision time T is a random variable,
which is the function of the time offset (i.e., the
variable X). Since the time offset is uniformly
distributed on 0 to LR as Eq. (18) illustrated. By
deconditioning with the random variable, the average
collision time without the effect of the time offset is
given by the following integral.
1 LR
LR ∫ 0
T ′ = E[T ] = T ( X )dX (27)
The PER can be derived from the BER and the
average collision time. Suppose the BER with and
without the interference is denoted by Pb and Pb0,
respectively, and Tb is the bit duration of Bluetooth.
The packet error rate (PER) is:
Pp = PER = 1 − (1 − Pb)T ′ / Tb (1 − Pb0)(TB −T ′) / Tb (28)
B. The Average Delay and the Throughput
The master and the slave of Bluetooth use Time
Division Multiplexing (TDD) to realize the
bidirectional link. At the same time, if multiple slaves
are present in the piconet, Time Division Multiple
Access (TDMA) is adopted to enable the
communications between one-to-multiple nodes. Since
the Bluetooth is transmitting according the time slots,
if one data packet is detected to be error, this packet
will be retransmitted when next transmitting turn of the
node. The retransmission time should be the multiple
of the slot time.
τ = iTS, i = 1,2,.... (29)
For voice packet of the Bluetooth, Forward Error
Correct (FEC) is used to combat the bit error in the
packet. Even if the errors are uncorrected, the voice
packet cannot be retransmitted since the voice is delay-
sensitive and can tolerate some bit errors. The
retransmissions are required when the data packet is
detected to contain some errors in the receiver.
Assumed that the number of retransmission is
unlimited, therefore the retransmission is following the
Geometric distribution. The probability of one
successful packet transmission after (k-1)
retransmission is:
P (k ) = (1 − Pp) Pp k −1 (30)
Since the required times of packet transmission is
Geometric distributed, the average transmission times
for a successful packet transmission is:
∞
1
m = ∑ (1 − Pp ) Pp n −1 × n = , (0 < Pp < 1)
n =1 1 − Pp
(31)
Under the assumption that one master and only one
slave are present at the network, the master and the
slave use TDD to realize the duplex. The delayed slot VI. CONCLUSION
number i in Eq. (29) is the double of the average In this paper, a mathematical model is proposed to
transmission times m in above formula. quantify the performance degradation of Bluetooth
2 system under the interference of 2.45 GHz RFID
i = 2m = (32)
1 − Pp system. The PER, the average delay and the throughput
Hence, the average delay time of Bluetooth data packet of Bluetooth are selected as the performance metrics.
is: The factors that can impact the interference include
2TS PHY (the transmitted power, the distance, the
τ= (33) modulation scheme and the channel model) and MAC
1 − Pp (the frequency hopping pattern, the packet format) are
Another performance metric, the throughput, can be taken into account. The result of this paper can provide
calculated easily from the average delay as follows. In the criteria for coexisting between RFID and
below formula, the symbol Spayload is the payload length Bluetooth. The anti-interference algorithms can be
of each packet. designed based on the analysis of this paper.
λ ⋅ Spayload
ρ= (34)
τ VII. REFERENCES
[1] IEEE Std. 802.15.1, “Part 15.1: Wireless Medium Access
V. SIMULATION COMPARED WITH THE Control (MAC) and Physical Layer (PHY) Specifications for
THEORETICAL RESULT Wireless Personal Area Networks (WPANs),” 2005.
[2] IEEE Std. 802.15.2, “Part 15.2: Coexistence of Wireless
Comparison of theoretical result and simulation Personal Area Networks with Other Wireless Devices
1
Theoretical
Operating in Unlicensed Frequency Bands,” August 2003.
Simulation
[3] ISO/IEC 18000-4, “Radio frequency identification for item
0.8 management, Part 4: Parameters for air interface
communications at 2.45 GHz,” Sept. 2006.
PER of Bluetooth

0.6 [4] Klaus Finkenzeller, RFID Handbook, 2nd ed., John Wiley &
Sons, 2003.
[5] N. Glomie, R.E. Van Dyck, A. Soltanian, A. Tonnerre and O.
0.4
Rebala, “Interference Evaluation of Bluetooth and IEEE
802.11b Systems,” Wireless Networks, vol. 9, pp 201-210,
0.2 2003.
[6] Nada Golmie and Frederic Mouveaux, “Interference in the 2.4
0
GHz ISM Band: Impact on the Bluetooth Access Control
1 1.5 2 2.5 3 3.5 4 4.5 5 Performance,” in Proceedings of IEEE ICC’01, Helsinki,
Interference distance (meter) Finland, June 2001.
Figure 4. Simulation results are compared with the theoretical [7] N. Golmie, R.E. Van Dyck, and A. Soltanian, “Interference of
results. Bluetooth and IEEE 802.11: Simulation modeling and
The simulation is done and the result is illustrated performance evaluation,” in Proceedings of the Fourth ACM
International Workshop on Modeling, Analysis, and
in figure 4. In the simulation, the distance between the
Simulation of Wireless and Mobile Systems, MSWIM’01,
RFID interferer and the Bluetooth receiver is changed Rome, Italy, July 2001.
from 1 meter to 5 meters. The PER is inversely [8] Soo Young Shin and Hong Seong Park, “Packet Error Rate
proportional to the interference distance. The Analysis of Zigbee under WLAN and Bluetooth Interference,”
theoretical result of the mathematical model is also IEEE Transactions on Wireless Communications, vol. 6, pp.
obtained and is shown in Fig. 4 too. As Fig. 4 2825-2830, August 2007.
[9] Carlos de Morais Cordeiro and Dharma P. Agrawal,
illustrated, the simulation result is very close to the “Interference Modeling and Performance of Bluetooth MAC
theoretical result. Therefore, the proposed Protocol,” IEEE Transactions on Wireless Communications,
mathematical model is verified. vol. 2, pp. 1240-1246, November 2003.

222
 2009 International Conference on Computer Engineering and Technology
Optimization Strategy for SSVEP-Based BCI in Spelling Program Application
Indar Sugiarto
Department of Electrical Engineering
Petra Christian University
Surabaya, Indonesia
e-mail: indi@petra.ac.id
Abstract— This paper describes an optimization strategy for
steady state visual evoked potential (SSVEP)-based brain-
computer interface (BCI), especially in a spelling program
application. In this application, there are at least three
components for implementing a complete BCI application:
stimulator, signal processing, and application (spelling
program). Ideally, those three components should run on
different processing units in order to obtain optimum
performance. But integrating those three components in one
computer system also gives advantages: make it easier for the
subject to concentrate and simplifies the system configuration.
There are two main parts that need to be optimized: the
flickering animation and the spelling system. When optimizing
the flickering animation, we will focus on the display driver
technology and programming aspects. The optimization of
spelling system will be focused on the layout and
representation of the letter matrix. We tested our program on
several computers for the following parameters: frequency
range, frequency resolution, and frequency stability. It can be
concluded that using a computer monitor as the stimulator, the
maximum synthesizable stimulator frequency is always half of
its minimum refresh-rate, no matter which software
technology is applied (DirectX or OpenGL). The maximum
synthesizable frequency of up to 30 Hz with frequency
resolution 0.11Hz is achieved. We have tested our system on
106 subjects during CeBIT 2008 in Hannover, Germany. Mean
accuracy for the spelling system is 92.5%. Therefore, the
optimization strategy described here led to a stable and
reliable system that performed effectively across most subjects
without requiring extensive expert help or expensive hardware.
Brain-Computer Interface, EEG, SSVEP, spelling program
I. INTRODUCTION
Brain-computer interface (BCI) systems are designed to
enable humans to send commands to a computer or machine
without using the normal output pathways of the brain [1],
[2]. Signals from the brain, usually nonelectrical signals from
the surface of the scalp (EEG activity), are classified and
then translated into commands using certain digital signal
processing (DSP) algorithms. To enable a signal processing
system recognizing specific features of the brain signals in
the cue-based BCI system, e.g. steady state visual evoked
potential (SSVEP)-based BCI, a certain stimulator is
required. In an SSVEP-based BCI, frequencies of the evoked
potentials match the frequencies of stimuli, which are usually
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.189
Brendan Allison, Axel Gräser
Institute for Automation
University of Bremen
Bremen, Germany
e-mail: allison@iat.uni-bremen.de, ag@iat.uni-
bremen.de
elicited using flickering light [3]. In the spelling program
application, the DSP algorithm translates those frequency
responses into commands for controlling cursor movement
and character/letter selection. The problem arises when using
general purpose computers, e.g. laptops, for implementing a
complete BCI application including stimulator, signal
processing, and application (spelling program). Ideally, those
three components should run on different processing units in
order to obtain optimum performance. But integrating those
three components in one computer system also gives
advantages. At least, providing the stimulator, i.e. flickering
light, and the application (spelling program) in the same
display/screen makes it easier for the subject to concentrate
and it also simplify the system configuration. These
advantages are important, because end users need a BCI that
does not require elaborate hardware (such as customized
LED boards or separate computing systems) or expert help
(such as to find working SSVEP frequencies or adapt the
system to each user).
This paper will be presented in the following outline. At
the beginning, software architecture will be presented and
followed by its optimization strategies. Then, the software
analysis result will be presented and discussed.
II. METHOD
A. Software Architecture
There are three components required for running an
SSVEP-based BCI system in a spelling program application
as depicted below:
Figure 1. Three components required for running SSVEP-based BCI
system in a spelling program application: a stimulator, a signal processing
unit, and the spelling program. In addition, the spelling program also
provides mental feedback for the BCI user.
223
 Of course, any BCI also requires a fourth component, word/phrase. For this purpose, we create five
namely sensors to acquire the brain signal [1], but these are flickering boxes on the monitor screen with label:
not modified in this work and are not mentioned further here. UP, DOWN, LEFT, RIGHT, and SELECT.
In this work, we used spatial filter called Minimum Energy We can arrange the letters in many forms and orders. One
Combination described in [4] as the core for the signal simple way is the square matrix with ascending order as
processing unit, and in this paper we focus on optimization shown below:
for stimulator and the spelling program. To integrate those
three components in a single computer, we created two
programs working together and connected via TCP/IP
socket. The first program, which is called the Display
Program, is responsible for displaying flickering animation
as the stimulator and also displaying the letter matrix for the
spelling program as well as the visual feedback for the user.
The second program, which is called the Signal Processing
Program, is responsible for signal acquisition, feature
extraction and classification, and command generation.
Commands generated by the second program will be sent to
the first program through a network connection. The
following diagram shows this architecture. Both programs
are written in C++. Figure 3. One simple way to organize letters is in a square matrix. The
cursor position is indicated by red-shaded cell.

The above configuration can be optimized in the

Figure 2. Three components required for running SSVEP-based BCI
system in a spelling program application: a stimulator, a signal processing
unit, and the spelling program. In addition, the spelling program also
provides mental feedback for the BCI user.
B. Optimization Strategies
In the Display Program, there are two main parts that
need to be optimized: the flickering animation and the
spelling system. The spelling system works as follows:
• We provide collection of letters arranged in a
matrix and a moveable cursor within this matrix.
This cursor can be moved up, down, left, or right
according to the command interpreted by the Signal
Processing Program from user’s EEG signals.
• When the user wants to select a letter to form a
word, he/she must concentrate his/her gaze on the
corresponding stimulus. And if the Signal
Processing Program is able to detect this intention,
the letter at the cursor position will be selected and
displayed on the appropriate position. In this way,
the collection of selected letters will form a
following way.
• Since a word formation is achieved first by moving
the cursor around, it is better if the home position of
the cursor is located at the center of the matrix. In
addition, since we only utilize four possible
movement (up, down, left, and right), it is better if
the distance of a letter from the center is kept as
near as possible.
• The fact that many words composed mainly by
vowels, it is reasonable to put vowels’ letter such as
A, I, U, E, O around the base of the cursor position.
Detail analysis using letter probability of words
done by Thorsten Lüth [5] shows that letter E is the
most commonly used letter. Based on such
probability analysis, we can construct a letter matrix
in an irregular but efficient way.
The following figure shows the optimization result from
the aforementioned approaches. Note that this result can be
expanded with additional characters if required.
Figure 4. Characters are arranged in a rhombus layout in order to achieve
higher efficiency for cursor movement.
The next part that needs to be optimized is the stimulator
itself. From our previous work [6], it is revealed that

224
animations with plain texture will elicit better SSVEP
response than animations with checkerboard texture. That is
why we use black-and-white animation with plain texture for
the stimulator.
In order to achieve robust and high resolution flickering
frequencies, we developed our program with the following
approaches:
• Each of five flickering boxes has its own thread and
corresponding timer. Since we run the program in
Windows platform, we use high resolution timer
called Multimedia Timer which has resolution
down to 1ms. The timer intervals are calculated as
follows:
Interval = 1/(2*fLed) (1)
where fLed is the flickering frequency of the
animation
• All of those threads are synchronized in one process
and we set the process with high priority above all
other Windows processes. In addition, we add
CPU’s mask-affinity to this process in order to
utilize the first CPU’s core and give the second core
to the Signal Processing Program.
• Two graphics display technologies are utilized and
compared: DirectX 9.0 and OpenGL 2.0. Using
these technologies, our program performance is
improved significantly because those technologies
will maximize the utilization of the graphics card.
• When developing program with OpenGL approach,
we used QGraphics Framework from QT [7]. We
then optimized QGraphicsView class with the
following parameters:
o ScrollBarPolicy : ScrollBarAlwaysOff
o Interactive : false (no interaction with
other GUI events)
o CacheMode : CacheBackground
o OptimizationFlags : DontClipPainter and
DontAdjustForAntialiasing
Here is the screenshot of our SSVEP-based BCI for
spelling application.
Figure 5. Integrated SSVEP-based BCI for spelling application.
225
III. RESULT AND DISCUSSION
We tested our program on several computers for the
following parameters: frequency range, frequency resolution,
and frequency stability. We also tested our program using
subjects during CeBIT 2008 in Hannover and collected
useful information such as spelling speed, accuracy, ITR,
and many important neuro-psychological parameters.
The following graph shows the comparison between
DirectX and OpenGL approach.
Figure 6. Comparison between OpenGL and DirectX method for
displaying flickering animations on the computer screen.
It can be concluded that using a computer monitor as the
stimulator, the maximum synthesizable stimulator frequency
is always half of its minimum refresh-rate (for CRT monitor)
or response-time (for LCD-TFT monitor), no matter which
software technology is applied (DirectX or OpenGL). The
maximum synthesizable frequency of up to 30 Hz is
adequate since the optimum flickering frequency for low-
frequency SSVEP-based BCI is around 15 Hz [8], [9].
To measure frequency resolution, first we have to
measure timer accuracy during run time. In order to produce
flickering animation at 17Hz, according to equation (1), we
have to set the timer interval of 1/(2*17) = 29.4ms = 29ms
(integer value). Using Windows Multimedia Timer and
executing the program as a normal priority process, we
found that for timer interval of 29ms, the standard deviation
value is just 0.46ms. It means one may expect that the
flickering frequency produced by using this Multimedia
Timer may be shifted about 0.55Hz above or below the
expected frequency. This value is calculated in the following
way:
Frequency shift =
(2)
We then increased the process’ priority and it yielded
standard deviation interval of about 0.1ms, which means that
we got the increasing frequency resolution from 0.55Hz to
0.11Hz.
To measure flickering stability against variation of
animation size and number of animation objects, we
conducted experiments with various CPU rating and by
varying the animation size and number of animation objects
displayed simultaneously on the screen. The following graph
shows the result from this experiment.
Figure 7. Comparison of optimum animation (LED) size for three
computers in four different procedures: 1)Single non-textured object,
2)Single checkerboard-textured object, 3)Three non-textured objects, 4)Six
checkerboard-textured objects.
It can be seen that the highest performance is achieved by
HP Compaq nx9420 while the lowest performance belongs
to Acer Aspire 5510. According to Windows Experience
Index of Windows Vista, HP Compaq nx9420 has the value
of 5.1, Acer Extensa 5620 has the value of 4.6, and Acer
Aspire 5510 has the value of 3.6. It seems that the software
performance, as indicated by the size of animation object as
well as the number of animation object displayable
simultaneously on the screen without disturbance, is greatly
influenced by the computer performance.
We have tested our system on 106 subjects during CeBIT
2008 in Hannover. During the experiment, the subject was
asked to spell five words: ‘BCI’, ‘Chug’, ‘Siren’, ‘Brain
Computer Interface’, and one free spelling. To average
subjects needed two minutes to spell ‘BCI’ (122.06 sec.),
‘Chug’ (139.3 sec.) and ‘Siren’ (111.13 sec.) and eight
minutes to spell ‘Brain Computer Interface’ (475.63 sec.).
Mean accuracy for all five words was 92.72%; mean ITR
13,852 bits/minute.
IV. CONCLUSION
The optimization strategy for the Bremen SSVEP-Based
BCI in a spelling program application has been presented
226
and the resulting performance of the system has been
evaluated. The optimization strategy in this paper is focused
on the display performance of the stimulator and the speller
program. We summarize the optimization strategy as
follows: using advance graphics driver technology (DirectX
and OpenGL), optimizing the multi-thread feature of a dual-
core processor; using high resolution timer (Windows
Multimedia Timer); and selection of a computer with high
CPU rating. However, external factors such as light
reflection and interference will also affect the overall
performance of the spelling system using SSVEP-Based
BCI. Future work should address these two concerns, and try
to further improve SSVEP BCI performance while
minimizing the needs for expert help, external LEDs, or
expensive hardware.
REFERENCES
[1] J.R. Wolpaw, N. Birbaumer, D.J. McFarland, G. Pfurtscheller, and
T.M. Vaughan, "Brain-Computer Interface for Communication and
Control," Clinical Neurophysiology 113, 2002.
[2] Brendan Allison, Dennis McFarland, Gerwin Schalk, Shi Dong
Zheng, M.M. Jackson, and J.R. Wolpaw, "Towards an Independent
Brain-Computer Interface Using Steady State Visual Evoked
Potentials," Clinical Neurophysiology 119, 2008: 399-408.
[3] J. Ding, G. Sperling and R. Srinivasan, “Attentional Modulation of
SSVEP Power Depends on The Network Tagged by the Flicker
Frequency,“ Cerebral Cortext - Oxford University Press., July 2006,
Vol 16:1016-1029.
[4] Ola Friman, Ivan Volosyak, and Axel Gräser, "Multiple Channel
Detection of Steady-State Visual Evoked Potentials for Brain-
Computer Interfaces," IEEE Transactions on Biomedical Engineering,
2007, Vol. 54, No. 4, 742-751.
[5] Thorsten Lüth, “Tools zur Verbesserung der Leistung von Brain-
Computer Interfaces,” Diplomarbeit, Universität Bremen, 2006.
[6] B. Allison, I. Sugiarto, B. Graimann, A. Gräser, “Display
Optimization in SSVEP BCIs,” Computer-Human Interaction 2008,
April 2008, Florence, Italy.
[7] Trolltech, Inc., “Creating Cross-Platform Visualization UIs with Qt
and OpenGL,” Trolltech Whitepaper.
[8] Gary Garcia, “High frequency SSVEPs for BCI applications,”
Computer-Human Interaction 2008, April 2008, Florence, Italy.
[9] Y. Wang, R. Wang, X. Gao, B. Hong, and S. Gao, “A Practical VEP-
Based Brain-Computer Interface,” IEEE Transactions on Neural
Systems and Rehabilitation Engineering, June 2006, Vol. 14, No. 2.G.
 International Conference
on Computer Engineering and Technology

Session 4
 2009 International Conference on Computer Engineering and Technology
A Novel Method for the Web page Segmentation And Identification
Jing Wang1,Zhijing Liu1
1School of Computer Science and Technology, Xidian University, Xi’an 710071, China
E-MAIL: wangjing@mail.xidian.edu.cn, liuprofessor@163.com
Abstract—A method of page segmentation and recognition
based on generalized hidden Markov model is present in this
paper, according to the page content as well as the structural
configuration. It can effective carry on the division to the
homepage, thus distinguishes each homepage block part
with GHMM. The experimental result indicated that,
compares with the original page segmentation algorithm,
this algorithm operation efficiency enhanced by 14.3%, and
the effect of segmentation has significantly improved.
Keywords: Web Page Segmentation; A Generalized Hidden
Markov Model (GHMM); Web Page Identifacation
I. INTRODUCTION
Along with the rapid and continuous development of
the Internet technology, Web already became in this
world the biggest information origin. But how to extract
the needed information is a necessitous problem. Page
segmentation and recognition is an important part in the
web page processing, and we accurately deal with web
information only when page is segmented nicety.
Therefore, the algorithm of page segmentation and
recognition is introduced in this paper.
According to the homepage manifestation and function,
the web pages can be divided into three categories. One
kind is the subject homepage, and it is composed by the
massive non-link text and the few hyperlinks text,
expresses a central subject together; One is label page,
also may be called the leader page, which is mainly
composed of text hyperlinks. The labels of page can be
classified into navigation label (navigation bar) and theme
labels, whose role is to generalize theme of its linked web
pages. Also some kind of homepages are composed by the
few texts and pictures, is called the picture homepage. A
subject pages can be divided into different areas,
including general navigation area, the theme of the text
area, the theme labels area. The so-called page
segmentation is to divide a subject page into each region.
The most popular methods of web page segmentation
are web DOM tree marking[1], page location coordinates
of entities[2], and VIPS page segmentation algorithm[3].
These methods are mainly considering the layout and
structure of web page, but not take the features of web
content in different areas into account. Therefore, these
methods are too reliant on the structure of web page. So a
method based on generalized hidden Markov model is
presented in this paper, and it is considering the structure
and the content of web pages. There are many advantages
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.149
for web pages segmentation using HMM, such as accurate
modeling and high recognition rate. It has been a new
research direction in recent years. Because of its easy-
established, good adaptability and high accuracy
extraction, it has become the concern of the researchers.
II. THE ANALYSIS OF WEB PAGE
Considering the traditional HMMs limitation that the
approaches only consider the semantic term as observed
emission feature, we use multiple emission features (term,
layout, and formatting) instead of single emission feature
(term). Web information contains other emission features
such as format, layout, etc, which could be observed to
help improve state transition estimation for HMM. In this
way ,we can consider the Web Page Segmentation from
the content and structure of web page . After the Web
page segmentation, it would be easier to identify and
automatic extraction for machines
A. The structure of Web page
It is the characteristics of web page that the logical
content interdependence block can be organized together.
We may think the page is composed by the different
content block. The text demonstrated which in the
different block also can demonstrate because of its
content according to certain form. For example, if the
relevant text is the hyperlink indicated that it will be
different from the color of the text in general; headline
fonts will be displayed over the text great. There are other
layouts such as font, size, italics, background color, and
background images. Because there is more embellishment
for letter than text in web page, it is more suitable for web
page's features taking these factors into consideration.
B. The text attribute of Web page
For the text processing by traditional HMM, the output
of each observation is released by the term shift. But
regarding take homepage content block as condition
GHMM, the transition probability between the blocks is
decided by the released observation characteristic. The
text in each block is expressed with a vector form.
Because ,the frequency of each word present in different
content documents has certain rule that the higher
frequency is, the more importance it is. And different
character words can divide different content text. So we
extract character words constitutes character vectors with
VSM model. Vector Space Model (VSM)[4] is considered
229
one of most popular model for representing the feature of
text contents. In this model, t1 , t2 ,… , tn refer to a system of
n-dimensional coordinates. w1 ( d ), w2 ( d ),… , wn ( d ) are the
relevant values of coordinates. A document d is expressed
as a vector composed of a serial of cross words, as follow:
V (d ) = (t1 , w1 (d );…… ; ti , wi ( d );…… ; tn , wn ( d ); )
where ti denotes the i-th keyword and wi ( d ) is the weight
of ti in document d .
III. GHMM
Web page generally can be divided into navigation,
Theme text, User Interaction, Copyright, the Websites
Label regions and so on. Although there respective
position is may be different according to each website
design style, it does not affect the content of the division.
So we can establish a hidden Markov model of five states.
In this model, five states itself are not directly visible, but
the each observation vector belongs to a certain state. In
other words, all the observation vectors are divided into
these five states, which reflect the changes in HMM, thus
recording the individual difference between homepage
different regions. So the HMM is introduced. So we can
establish a Hidden Markov Models of five states for the
HMM,as the following figure1 shown.
Figure 1. The state transition of GHMM
In this model, five state itself is not directly visible, but
the each observation vector belong to a certain state.
Namely,all the observation vector is divided into five
a) N is the number of states in model. These N states
are S = {S1 , S 2 , , S N } , and suppose the state at time t
is qt , so we can know q t ∈ S .
b) M is the number of probability observable
symbols. V = {V1 , V2 , , VM } , and the observable
symbol at time t is ot , where ot ∈ V .
c) π : the initial state probability distribution, and
π = {π 1 , π 2 , , π N } , where π i = P (q1 = S i ), 1 ≤ i ≤ N .
d) A: the hidden state transition probability
distribution. A = ( aij ) N × N , where
aij = P (qt +1 = S j | qt = Si ),1 ≤ i, j ≤ N (1)
e) B: the emission probability distribution.
{
B = B ,B ,
1 2
,B
Z
} , B = (b (k ))
j s N ×M
.
b j (k s ) = P (ot , s = Vk | qt = S j ) 1 ≤ j ≤ N ,1 ≤ k ≤ M (2)
where N is the number of states in the model, Z is the
number of attribute sets of observation symbols, MS is the
number of distinct observation symbols of the attribute s,
1 • s • Z, and T is the number of symbols observed. A is N
× N matrix of the state transition. Transition
probability aij , the probability of being in state j at time
t+1 given that we were in state i at time t. Emission
probability b j ( k s ) , the probability of observing the
symbol v s k of the attribute s given that we are in state j.
The GHMM defined as = (A, B, ) find out the state
transition sequence I = i1 , i2 , , iT which maximizes P
(O, I | ).
P (O, I | λ ) = P (O | I , λ ) P( I | λ )
(3)
= π i1 bi1 (o1 )π i2 bi2 (o2 ) π iT −1 biT −1 (oT −1 )
In GHMM an observation symbol k is extended from
its mere term attribute to a set of attributes as
k1 , k2 , , k z . We consider the overall feature as a linear
combination of these attributes, that is
z

states, which reflects the changes in HMM. HMM do use b j (k ) = ∑ α s ∗ b j (ks ) , where α s is the weight factor for
s =1
the method to record the differences of web page in z
different area. the s-th attribute, ∑α s = 1 , 0 ≤ α s ≤ 1 ,1 ≤ s ≤ z .
A common goal of learning problems that use HMM is s =1
to figure out the state sequence that has the highest Equation (3) now becomes
probability of having produced an observation sequence, P(O, I | λ) =
which can be efficiently solved by Viterbi algorithm [5]. (4)
⎡ z
⎤ ⎡ z
⎡z ⎤ ⎤
For GHMM based learning problems, the goal is to find πi ⎢∑αs ∗bis (o1,s )⎥ ai i ⎢∑αs ∗bis (o2,s )⎥ ai
⎢∑ αs ∗ bi
s
(oT , s ) ⎥
T −1iT
⎣ s=1 ⎦ ⎣ s=1 ⎦ ⎣ s=1 ⎦
1 1 12 2 T
the most probable state sequence using the following
extended Viterbi algorithm. And the related definition as For GHMM based learning problems, the goal is to find
follow: the most probable state sequence using the following
Formally, a GHMM consists of the following five extended Viterbi algorithm. Here we denote an
parts [6]: observation symbol k with attributes k1 , k2 , and k3 ,
which correspond to format, layout attribute and the text

230
feature vector of VSM respectively. We consider the Area of Web page Method(1) Method(2) Method(3)
emission probability as a linear combination of these Navigation 69.4% 75.4% 82.4%
Theme Text Theme 81.9% 85.3% 89.3%
attributes, that is, Labels 61.8% 70.2% 81.7%
b j ( k ) = α1 ∗ b j ( k1 ) + α 2 ∗ b j (k2 ) + α 3 ∗ b j ( k3 ) (5) User Interaction 81.2 87.6% 92.5%
Copy Right Website 90.3 93.2% 96.6%
where α1 , α 2 , and α 3 are the weights for term attribute, Label 66.5 70.7% 76.3%
format attribute, and layout attribute respectively, and
α1 + α 2 + α 3 = 1 , 0 ≤ α i ≤ 1 . The results indicate that the accuracy of improved
method is improved by about 20 percent in navigation
IV. EXPERIMENTAL RESULT AND ANALYSIS area and Theme Label area. Because these two areas have
significant features such as the text of related topics link
Based on the above discussion, in order to verify this are displayed in the form of hyperlinks.
page segmentation and the identification method
presented in this paper, we randomly selected 200 online V. CONCLUSION
pages, of which the 150 are used as the training set, while
According to the characteristics of web pages, a new
the remaining 50 are used as the test set. page segmentation method based on improved GHMM is
Here, we take the recall and accuracy ratio as
presented in the paper. The results also show that the new
evaluation standards. When we compare segmentation
method on the page segmentation and recognition
results of the different class of web page, we use the F
accuracy is improved markedly. In the future, we will
measure which combines recall (R) and precision (P), and improve the effective learning of GHMM, so that the
the recall, precision and F measure are defined as:
model can be fully automated without lots of manual
numbers of correct Web segmentation
R= intervention.
numbers of all Web segmentation
numbers of correct Web segmentation ACKNOWLEDGEMENTS
P=
numbers of possible Web segmentation This research project was promoted by the National
2× R × P Science & Technology Pillar Program No.
F= 2007BAH08B02.
R+P
The experiment shows the compared result of the REFERENCES
accuracy of three extraction methods. (1) DOM tagging [1] Chakrabarti S. Integrating the document object model with
tree method; (2) VIPS page segmentation method; (3) the hyperlinks for enhanced topic distillation and information
method based on GHMM. Table 1 has given the extraction. Proc Tenth International World-Wide Web
comparison results of these three methods in average Conference, p 211–220, 2004.
[2] Salton G, Buckey C. Term-weighting approaches in automatic
accuracy. text retrieval. In: Readings in Information Retrieval. Morgan
Kaufmann; 2005. p 323–328.
TABLE 1.THE THERE METHODS IN AVERAGE PRECISION [3] Osawa Y, Benson NE, Yachida M. KeyGraph: Automatic
Method P R F indexing by co-occurrence graph based on building construction
Method(1) 73.2% 76.8% 74.4% metaphor. Trans IEICE 2003;J82-D-I:391–400.
Method(2) 78.8% 79.3% 79.1% [4] Weimin Xue, Hong Bao, Weitong Huang et al, “Web Page
Method(3) Classification Based on VSM”, Proceedings of the 6th World
85.2％ 82.6％ 83.8％
Congress on Intelligent, Control and Automation, pp.6111-6114,
June 21-23, 2006.
[5] Du Shi-ping. The Viterbi algorithm of mixture of HMM2 [J].
From table 1, we can see that the method based on Journal of Yunnan University, 2006, 28(2):98-102.
GHMM has the highest recognition accuracy. The novel [6] Jinlin Chen, Ping Zhong, Terry Cook. Detecting Web Content
method takes the layout and the content of pages into Function Using Generalized Hidden Markov Model. IEEE
account, so it has the higher accuracy rate than other Proceedings of the 5th International Conference on Machine
Learning and Applications. 2006.
methods. The results are shown in table 2.
TABLE2. THE PRECISION OF THERE MENTHOD IN
DIFFERENT AREAS OF WEB PAGE

231
 2009 International Conference on Computer Engineering and Technology
Disturbance Observer-based Variable Structure Control on the
Working Attitude Balance Mechanism of Underwater Robot
LI Ming (1) LIU Heping(2)
School of Electronic Information Engineering, Henan University of Science & Technology (1)
Department of Precision Machinery, School of Mechatronics Engineering and
AutomationShanghai University ()
lim@mail.haust.edu.cn
Abstract
When an underwater robot is to work under the
water, the manipulator has to reach out. This can
change the center of the gravity and cause the robot
pitching. It’s difficult to make a maneuver for a pitched
robot. An automechanism was developed to balance
the pose of the robot. A sliding mode variable structure
controller was designed to control the balance device,
in which a disturbance observer was used to observe
the disturbances. This reduced the switching-gain of
the control system greatly and weakened the chattering
more of course. A practical result was achieved for the
control of pitch change of the robot caused by the
manipulators and detecting pan/tilt and indicated the
practicability of the balance device including its
control approach. As the device is a single-axis system,
a double-axes system is needed for the balance of
pitching and rolling.
abstract is to be in fully-justified italicized text, at
the top of the left-hand column as it is here, below the
author information. Use the word “Abstract” as the
title, in 12-point Times, boldface type, centered
relative to the column, initially capitalized. The
abstract is to be in 10-point, single-spaced type, and
up to 150 words in length. Leave two blank lines after
the abstract, then begin the main text.
1. Introduction
Nowadays, the application of the underwater robot
becomes more and more popular. The working mode
ranges from cleaning, kelp reaping, drilling, cutting,
sampling, gripping in civil fields to mining and mine
sweeping, anti-terrorism and explosive-proof in
military fields. With the developing of our society and
the progressing of science, more missions would be
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.202
finished by underwater robots to substitute aquanauts
for human being. Intelligent underwater robot with
ability of perceiving, thinking and decision-making
and with a strong multi-function will be developed
rapidly.
Robot is a complicated dynamic system with
serious nonlinear property. For an underwater robot
with open-frame structure, a strong coupling will take
place among the movements in each degree of freedom
and a strong nonlinear movement exists always. It is
unpractical to set up an accurate dynamic model. A
number of tests are necessary to estimate the numerous
hydrodynamic coefficients. And the properties are
different between static water and flowing water. The
control of position and attitude of underwater robot is
an arduous task.
According to reference [1], the fixed coordinate
system EӶӯӮ and motion coordinate system Oxyz
are setup, and both origins are on the center of gravity
of ROV, as shown in Fig. 1. When not working, the
manipulator equipped by the robot furled usually
beneath the robot shown in Fig. 2. At this time, the
robot is on the balance position with tiny buoyancy.
The center of gravity coincides with the center of
buoyancy in x-y plane. A distance B0 named
metacentric height is designed in z axis to produce an
uprighting moment.
When the robot moving,
x G0 =x F0
yG0 =y F0
z F0  z G0 =B0
232
 cosine law basically, the viscous resistance and
E
additional mass make a influence on the movement. In
Ӷ addition, the operating of the pan/tilt with sonar, lights
and camera and the dragging of the neutrally buoyant
ӯ O tether can impose inestimably on the attitude of the
robot. So, it’s direct and rapid to control the attitude of
the robot, in which all of the disturbances are
Ӯ y considering.
x
z
Figure 1 Fixed and motion coordinate systems

Figure 3 Reaching out of the manipulator and

the operating of the detecting pan/tilt

Figure 2 Underwater robot with two

manipulators and detecting pan/tilt

The robot is driven by propellers and moves to

appointed position. When needed, the manipulators
reach out from the bottom of the robot to carry out
various works, as shown in Fig. 3. The barycenter of
the manipulator arms varies from w0(xm0,ym0) in
original state to w(xm,ym) in working state. This makes
the whole barycenter of the robot transfer from
G0(xG0,yG0,zG0) to G (xG,yG,zG).
The underwater robot test platform described in this
paper has two manipulators symmetrical to x axis in
structure. During the process of reaching out from the
bottom of the robot, the manipulator has only a change
of the barycenter in x axis. The move of the barycenter
of the robot certainly causes a pitching on the robot.
When adjusting the position of the robot for the
convenience of manipulator’s work, it is difficult to
operate a pitching robot accurately under the water. To
solve the problem, a balance mechanism was
developed (shown in Fig. 4) to balance and
compensate for the move of the gravity center of the
robot based on the pitching angle and angular velocity
measuring by the attitude transducer. Why don’t adjust
the center of gravity of the robot according to the
change of the barycenter of the manipulator arms?
Although the movement of the manipulator keeps to a
Figure 4 Balance mechanism of the attitude
2. Modelling
The mass of the balance body of the balance
mechanism of the attitude is m2, pulled by a step motor
along the guiding rails. When the barycenter of the
robot system is at the balance, the position of the
barycenter of the balance body is assumed as the origin.
The change of the position X of the balance body
causes a shift of the center of gravity of the robot and
the pitch angle T of the robot is changed accordingly.
Assumed that when pitching and corrected, the robot
turns only around the y axis and other movements
don’t exist. The moment of inertia of the robot is Jy.
The turning equation of the robot is:
J y
T=  MT  m 2 X+f1

233
 where M is the hydrodynamic coefficient when The sliding line is chosen as:
turning around y axis and f1 is the disturbances s=ce1  e2 (5)
including the change of the moment of inertia , the The analysis of stability is shown as follow:
viscous resistance and additional mass caused by the
move of the manipulators, balance object, detecting Let f =f  fˆ , e 2 =e 2  eˆ 2 , and
pan/tilt. The disturbance produced by the dragging of 1 2 1 2 1 2
the neutrally buoyant tether is also included. V1 s Δ V2 = f  e 2
Let a M / J y , u m 2 X / J y and f f1 / J y
2 2K1 2
Lyapunov function is selected as:
The equation can be re-written as:
 1 2 1 2 1 2
T=  aT  u+f (1) V=V1  V2 s  f  e 2 (6)
2 2K1 2
3. Disturbance Observer-based Variable From equation (5) and (2), It’s attainable that:
structure Control  1  e 2
s=ce ce 2  ae2  u  f
Putting (4) into it, then:
The desired value of pitching angle is Td 0 , then s ce2  ae 2  ]e1  K sgn(s)  fˆ  f
the error and its derivative are defined as:
e1 Td  T T
=(c  a)e  ]e  K sgn(s)  f
2 1

e2 e T , e e1  ,
T =(c  a)(s  ce1 )  ]e1  K sgn(s)  f
1 2
Putting into (1) yields to the error state equation: =(c  a)s  c(c  a)e1  ] e1  K sgn(s)  f
ª e1 ºª0 1 º ª e1 º ª 0 º ª0º =(c  a)s  e (c(a  c)  ] )  K sgn(s)  f
«e » 
«0 a » «e » « 1» u  «1 » f (2) 1
¬ 2¼ ¬ ¼¬ 2¼ ¬ ¼ ¬ ¼ 
  a)s  se1 (c(a  c)  ])  K s  fs
ss=(c 2

That is e 2 ae 2  u  f
d (c  a)s2  se1 (c(a  c)  ])  ( f  K) s
According to the theory of sliding mode variable max
structure [2], the movement of the state of system is Consequently:
forced along the sliding mode surface by the switching  d (c  a)s2  se (c(a  c) ])  (f
V  K) s (7)
1 1
of the control parameters. When the change of the max
parameters of control object and the external In addition
 = 1 f f  e e
disturbances are applied, the sliding mode is
unchangeable and robust. But the chattering of the V2 2 2
sliding mode control is inevitable when the control is K1
(8)
switching. To reduce the chattering, a disturbance 1   ˆ
observer is adopted to estimate the disturbances so that = f( f  f)  e 2 (e 2  eˆ 2 )
the chattering can be weakened [3] [4] [5] [6]. K1
From (3),,then
Let f̂ be the estimate of the disturbances while ê 2
ˆ
denote the estimate of e 2 and the gains k1 and k2 are f=k 1 (e 2  e 2 )
ˆ k1e 2 and
selected by the pole placement method. The eˆ 2 fˆ  aeˆ 2  u  k2 (e2  eˆ 2 ) fˆ  aeˆ 2  u  k2e 2
disturbance observer is designed as As the disturbances produced by the moves of
ª fˆ º ª0 0 º ª fˆ º ª 0 º ª k1 º manipulators and pan/tilt as well as the dragging of the
« » « » « »  « » u  « » > e2  eˆ 2 @ (3) neutrally buoyant tether are slow movements, it can be
«¬ê 2 »¼ ¬1 a ¼ ¬ê2 ¼ ¬1¼ ¬k 2 ¼ thought as that:
The sliding mode controller is defined as:
 . Putting into (8) together with (2), then:
f=0
u=]e1  K sgn(s)  fˆ (4)
­D se1 ! 0
Where ] = ®
¯E se1  0
and K represents the switching gain.

234
  =  1 f fˆ  e (  ae  u  f  fˆ  aeˆ  u  k e )
1

V2 2 2 2 2 2 0.5
K1
=  f e +e (aeˆ  ae  f  fˆ  k e )
0

2 2 2 2 2 2 -0.5

=  f e 2  ae 22  e 2f  k 2e 22 )

State response
-1

-1.5

 =  (a  k )e 2
V (9) -2
2 2 2
From equation (7) and (9), it can be found that: -2.5

while equation (10) -3

k 2 t a -3.5
0 1 2 3 4 5 6 7
atc t(s)

Figure 5 Response of the error state

­D t c(a  c) if se1 t 0
]= ® (10)
¯ E d c(a  c) if se1 d 0
250

K t f 200
max

k1 ! 0 150

control input
 d 0 had been be verified. This means
is satisfied, V
that the sliding mode control is stable. 100

It is the adoption of the disturbance observer that

the condition of K t f  fˆ could be satisfied so 50
max
easily that the switching gain K is depressed greatly
0
and weaken the chattering therefore. 0 1 2 3 4 5 6 7
t(s)

Figure 6 Control input

4. Simulation
1

The control Algorithm was simulated on

0
Matlab®/Simulink® [7][8]. The mass of the robot is
175kg. The center of gravity of the manipulator arms -1
that has a mass of 0.5kg varies from 0 to 900mm while
working, which produces a max pitching moment up to -2

0.45N.m. In order to balance the moment as well as the

e2

-3
other disturbances, the balance object is designed to
have a mass of 2.2kg and slide a distance from Ё50 to -4
Ѐ250mm, which will cause a balance moment from
Ё0.11N.m~Ѐ0.55N.m. Since it is the pitching of the -5

whole robot induced by the movements of the -6

-0.1 0 0.1 0.2 0.3 0.4 0.5 0.6
manipulators, detecting pan/tilt and the dragging of the e1
neutrally buoyant tether that has a slow velocity, the
condition of slow velocity mentioned-above for sliding Figure 7 e1 - e2 plane trajectory
mode controller is satisfied. The pitching of high
frequency induced by wave is not taken into account.
[9][10]
Assumed that the disturbance applied upon the
robot obey a law of sines.
f=5  0.3sin(0.5St)
The results of the simulations presented here are
promising shown in Fig.5 to Fig.8.

235
 7
0LFUR6WUDLQ'0*;'DWD)LOH9HUVLRQ'DWH
ILOHFUHDWHG7LPHILOHFUHDWHG
6 $QJOHV3LWFK


4

f and ef

3LWFKGHJUHH
3

















2 

1

0
WV
0 1 2 3 4 5 6 7
t(s)

Figure 8 Estimate of the disturbance

Figure 10 Pitching change of the robot controlled
5. Experiment by the balance mechanism

The operating experiment of the underwater robot 6. Conclusion

was made in a static water pool. The manipulator
turned out from the bottom of the robot at a angular Aimed at the pitching of the robot caused by the
velocity of 150̓/s. The detecting pan/tilt pitched at a move of the manipulator, detecting pan/tilt and tether,
angular velocity of 60̓/s(without rolling movement). a balance mechanism was developed and a sliding
While the balance mechanism didn’t work, the max mode controller was designed to control the device.
pitching angle of the robot would reach about 10.3 Meanwhile a disturbance observer was defined in the
degrees as shown in Fig.9. Nevertheless, the pitching controller to estimate the disturbances. The amplitude
angle would be compensated within 4.5 seconds when of the switching gain was reduced greatly and the
the balance mechanism was working as shown in chattering in the sliding mode control was weakened
Fig.10. The max pitching angle reached only 5.6 certainly. A good effect had been achieved upon the
degrees at 0.4 second. The control goal had been pitching of the robot induced by the manipulator and
implemented basically. detecting pan/tilt. This shows that the balance
mechanism as well as the control approach is
0LFUR6WUDLQ'0*;'DWD)LOH9HUVLRQ'DWHILOHFUHDWHG
7LPHILOHFUHDWHG$QJOHV3LWFK
applicable. As it is a single axis device, the pitching
can only be compensated around the y axis. For the
 robot with a barycenter disturbance along the y axis, a

two DOF balance mechanism is needed to compensate
the pitching round the y axis and the rolling round the

x axis.

3LWFKGHJUHH

 7. Acknowledgment



The research projects concerned in this paper were
financially supported by State Leading Academic























Discipline Fund and Shanghai Leading Academic
 Discipline Fund of Shanghai University (Project No.
WV
BB 67 and No. Y0102), National Natural Science
Foundation of China (Project No. 60605028) and The
National High-tech Research and Development
Figure 9 Pitching change caused by the moves of
Program (Project No. 2007AA04Z225).which are
the manipulator and pan/tilt without the work of
greatly appreciated by the authors.
balance control
8. References

236
[1] Li D.P., Motion and Modelling of Ship, Harbin
Engineering University Press, Harbin, 1999.
[2] Yao Q. H., Huang J. Q. and Wu H S, Variable Structure
Control System, Chongqing University Press, Chongqing,
1997.
[3] Chang Jeang-Lin, Wu Tsui-Chou Robust disturbance
attenuation with unknown input observer and sliding mode
controller Electrical Engineering, v 90, n 7, September, 2008,
pp.493-502.
[4] Kawamura, Atsuo; Itoh, Hiroshi; Sakamoto, Kiyoshi
Chattering reduction of disturbance observer based sliding
mode control IEEE Transactions on Industry Applications, v
30, n 2, Mar-Apr, 1994, pp.456-461.
[5] Oudghiri, Mohammed, Chadli, Mohammed, and El
Hajjaji, Ahmed, “Lateral vehicle velocity estimation using
fuzzy sliding mode observer”, 2007 Mediterranean
Conference on Control and Automation, MED, 2007
Mediterranean Conference on Control and Automation,
MED, 2007, pp. 4433910.
[6] Wang Ying, Xiong Zhenhua, and Ding Han, “Robust
controller based on friction compensation and disturbance
observer for a motion platform driven by a linear motor”,
Proceedings of the Institution of Mechanical Engineers. Part
I: Journal of Systems and Control Engineering, v 220, n 1,
2006, pp.33-39.
[7] Yu. S., Zhao Y.Z,. “Simulation study on a friction
compensation method for the inertial platform based on the
disturbance observer”, Proceedings of the Institution of
Mechanical Engineers, Part G: Journal of Aerospace
Engineering, v 222, n 3, 2008, pp. 341-346.
[8] Liu J K, MATLAB Simulation for Sliding Mode Control,
Tsinghua University Press, Beijing, 2005.
[9] Heping LIU, Zhenbang GONG, “Upper Bound Adaptive
Learning of Neural Network For The Sliding Mode Control
of Underwater Robot”, Proceedings of the International
Conference on Advanced Computer Theory and Engineering ,
2008, Phuket, pp. 446-451.
[10] Heping LIU, Zhenbang GONG, “The Anti-wave
Control of Small Open-frame Underwater Robot”,
Proceedings of the 3th International Conference on
Intelligent System & Knowledge Engineering, 2008, Xiamen,
pp. 653-658.

237
 2009 International Conference on Computer Engineering and Technology

ADAPTIVE OFDM Vs SINGLE CARRIER MODULATION WITH FREQUENCY

DOMAIN EQUALIZATION

Kaur Inderjeet* Thakur Kamal* Kulkarni.M# Gupta Daya$ Arora Prabhjyot*

Dept of CSE, Dept of ECE Dept of ECE Dept of CSE Dept of ECE
# $
*Institute of Technology & Management National Institute of Technology Delhi College of Engineering
# $
*Gurgaon,India Suratkal,India, Delhi, India
inderjeetk@gmail.com kamalthakur12@gmail.com mkuldce@gmail.com daya_gupta2005@yahoo.co.in aroraprabh@gmail.com,

Abstract— The aim of the present paper is to compare multi-
carrier and single carrier modulation schemes for wireless
communication systems. In both cases the fast Fourier
transform (FFT) and its inverse are utilized. In case of OFDM
(orthogonal frequency division multiplexing), the inverse FFT
transforms the complex amplitudes of the individual sub-
carriers at the transmitter into time domain. At the receiver
the inverse operation is carried out. In case of single carrier
modulation, the FFT and its inverse are used at the input and
output of the frequency domain equalizer in the receiver.
Different single carrier and multi-carrier transmission systems
are simulated with time-variant transfer functions measured
with a wideband channel sounder. In case of OFDM, the
individual sub-carriers are modulated with fixed and adaptive
signal alphabets. Furthermore, a frequency-independent as
well as the optimum power distribution are used. Single carrier
modulation uses a single carrier, instead of the hundreds or
thousands typically used in OFDM, so the peak-to-average
transmitted power ratio for single carrier modulated signals is
smaller. This in turn means that a SC system requires a
smaller linear range to support a given average power. This
enables the use of cheaper power amplifier as compared to
OFDM system.
Keywords- OFDM, BER, minimum mean square error,
QAM, LOS, IFFT.
I. INTRODUCTION
In this paper the wideband frequency-selective radio
channels is used for investigating the transmission of digital
signals. Frequency-selective fading caused by multipath time
delay spread degrades the performance of digital
communication channels by causing intersymbol
interference, thus results in an irreducible BER and imposes
an upper limit on the data symbol rate. The performance of
single carrier and multi-carrier modulation schemes will be
compared for a frequency-selective fading channel
considering un-coded modulation scheme.
Un-coded OFDM loses all frequency diversity inherent in
the channel: a dip in the channel erases the information data
on the subcarriers affected by the dip and this information
cannot be recovered from the other carriers. This mechanism
results in a poor Bit Error Rate (BER) performance. Adding
sufficiently strong coding spreads the information over
multiple subcarriers. This recovers frequency diversity and
improves the BER performance.
The performance of OFDM can be improved
significantly by using different modulation schemes for the
individual sub-carriers. The modulation schemes have to be
adapted to the prevailing channel transfer function. Each
modulation scheme provides a trade off between spectral
efficiency and the bit error rate. The spectral efficiency can
be maximized by choosing the highest modulation scheme
that will give an acceptable (BER). In a multipath radio
channel, frequency selective fading can result in large
variation in the received power of each carrier.
The paper is organized as follows: In section II the fixed
and adaptive OFDM transmitters are described. A
description of a single carrier system with frequency domain
equalization in section III is followed by simulation results in
section IV.
II. ADAPTIVE OFDM TRANSMISSION
The block diagram of the OFDM transmitter used is shown
in Fig. 1. Binary data is fed to a modulator which generates
complex symbols on its output. The modulator either uses a
fixed signal alphabet (QAM) or adapts the signal alphabets
of the individual OFDM sub-carriers. Both, signal alphabets
and power distribution can be optimized corresponding to
the channel transfer function. Propagation measurements of
radio channels with fixed antennas show that the transfer
function varies very slowly with time. Because of this
reason, it is assumed that the instantaneous channel transfer
function can be estimated at the receiver and can be
communicated back to the transmitter. The third block
transforms the symbols into time-domain using inverse fast
Fourier transform (IFFT) at the transmitter. The next block
inserts the guard interval. The output signal is transmitted
over the radio channel. At the receiver, the cyclic extension
is removed and the signal is transformed back into
frequency domain with an FFT. Prior to demodulation, the
signal is equalized in frequency domain with the inverse of
the transfer function of the radio channel corresponding to a
zero-forcing equalizer.
Two different adaptive modulator/demodulator pairs are
considered in this paper: In modulator A, the distribution of

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 238

DOI 10.1109/ICCET.2009.243
Fig. 1 Block diagram of a) an OFDM and b) a single carrier transmission system with frequency domain equalization
bits on the individual sub-carriers is adapted to the shape of
the transfer function of the radio channel. Modulator B
optimizes simultaneously both, the distribution of bits and
the distribution of signal power with respect to frequency.
The algorithms for the distribution of bits and power are
described in [7]. The adaptive modulators select from
different QAM modulation formats: no modulation, 2-PSK,
4-PSK, 8-QAM, 16-QAM, 32-QAM, 64-QAM, 128-QAM,
and 256-QAM. This means that 0, 1, 2, 3 ... 8 bit per sub-
carrier and FFT block can be transmitted. In order to get a
minimum overall error probability, the error probabilities
for all used sub-carriers should be approximately equal.
In case of modulator A, the distribution of bits is carried out
in an optimum way so that the overall error probability
becomes minimum. The algorithm for modulator A
maximizes the minimum (with respect to all sub-carriers)
SNR margin (difference between actual and desired SNR
for a given error probability).
Modulator B optimizes the power spectrum and distribution
of bits simultaneously. The result of modulator B is that the
same SNR margin is achieved for all sub-carriers. The
obtained SNR margin is the maximum possible so that the
error probability becomes minimum. Therefore, modulator
B calculates the optimum distribution of power and bits.
The results of the optimization processes of both modulator
A and modulator B are shown in Fig. 3. For comparison, the
upper diagram gives the absolute value of the transfer
function. For the specific example presented in Fig. 3, both
modulators yield the same distribution of bits. Furthermore,
the power distribution and SNR is shown for both
modulators.
III. SINGLE CARRIER TRANSMISSION WITH
FREQUENCY DOMAIN EQUALIZATION
The lower part of the block diagrams in fig. 1 shows the
considered single carrier transmission system. The figure
shows that the basis concepts for single carrier modulation
with frequency domain equalization and OFDM
transmission are almost similar. The main difference is that
the block “inverse FFT” is moved from the transmitter to
the receiver [l]. Therefore, single carrier modulation and
OFDM without adaptation exhibit the same complexity.
239
Since also in case of single carrier modulation the FFT
algorithm is used, a block wise signal transmission has to be
carried out. Like in an OFDM system, a periodic extension
(guard interval) is required in order to mitigate interblock
interference.
In contrast to adaptive OFDM, for single carrier modulation
a fixed symbol alphabet is used in order to realize a constant
bit rate transmission.
There is however, a basic difference between the single and
multi-carrier modulation schemes: In case of the single
carrier system, the decision is carried out in time domain,
whereas in case of the multi-carrier system the decision is
carried out in frequency domain. In case of the single carrier
system, an inverse FFT operation is located between
equalization and decision. This inverse FFT operation
spreads the noise contributions of all the indivi.dua1 sub-
carriers on1 all the samples in time domain. Since the noise
contributions of highly attenuated
Sub-carriers can be rather large; a zero-j-forcing equalizer
shows a poor noise performance. Because of this reason, a
minimum mean square error (MMSE) equalizer is used for
the single carrier system. The transfer function of the
equalizer H, (w,t) depends on the SNR of the respective sub-
carriers
S/Nlr(w, t) at the input of the receiver:
1 S / N | r ( w, t )
H e ( w, t ) = ⋅
H ( w, t ) S / N | r ( w, t ) + 1
H(w, t) denotes the time-variant transfer function of the
radio channel. For large SNRs the MMSE equalizer turns
into the zero-forcing equalizers which mu1tiplies with the
inverse transfer function.
The main advantage of single carrier modulation compared
with multi-carrier modulation is the fact that the energy of
individual symbols is distributed over the whole available
frequency range. Therefore, narrowband notches in the
transfer function have only a small impact on error
probability. Furthermore, the output signal of a single
carrier transmitter shows a small crest factor whereas an
OFDM signal exhibits a Gaussian distribution.
 IV. SIMULATION RESULTS The results show that an enormous improvement in
In the present paper following systems are compared using performance (12 to 14 dB) is obtained from OFDM with
measuring transfer function of the channels: adaptive modulation. Adaptive OFDM shows also a
i Single carrier modulation with minimum mean square significant gain compared with single carrier modulation.
error (MMSE) frequency domain equalizer But only a gain of less than 0.5 dB is achieved using an
ii OFDM with fixed modulation optimized power spectrum for OFDM instead of a
iii Sub-carriers and frequency-independent power frequency-independent. Because of this small difference, it
distribution, OFDM with optimized modulation is recommended to use a constant power spectrum in order
schemes and frequency-independent power distribution to save computational or signaling effort.
(modulator A)
iv OFDM with optimized modulation schemes and Length of FFT interval 256 samples
optimized power distribution (modulator B).
Length of guard interval 50 samples
For all transmission systems a complex base band
RF bandwidth 5MHz
simulation is carried out with ideal channel estimation and
synchronization. No over sampling was used since only Average data rate 16.8 Mbps
linear components (except the detectors) are assumed in the
transmission systems. The temporal location of the FFT Noise figure of the receiver 6 dB
interval with respect to the cyclic extension at the receiver Number of transmitted bits 2·105
(i.e. the time synchronization of the OFDM blocks) is
optimized so that the bit error ratio becomes minimum. For
both, single carrier and multi-carrier modulation, QAM Table 2: Simulation Parameters
schemes with different bandwidth efficiencies are used.
For the LOS measurements also, a significant gain (5 to 6
Measurement 1 2 3 4 dB) is obtained from adaptation, but the gain is smaller than
Distance of
100m 100m 250m 250m in the NLOS case. This results from a higher coherence
antennas bandwidth of the LOS radio channel transfer function.
Propagation Particularly in the NLOS case with single carrier
LOS LOS NLOS NLOS
conditions
Omni Omni
modulation, a high gain (7 to 9 dB) compared with fixed
Base station Sectional Sectional OFDM is obtained. In case of the LOS channels single
directional directional
antenna fixed fixed
fixed fixed carrier modulation yields only a signal gain of 1 to 2 dB.
Omni Omni Omni Omni
User terminal
directional directional directional directional Additional simulations show that the gain from adaptive
antenna
fixed mobile fixed mobile
modulation increases when higher-level modulation
Carrier
1.8 GHz schemes are used. Furthermore, adaptive OFDM is less
frequency
Average sensitive to interblock interference due to an insufficient
77.4 dB 66.1 dB 112 dB 105.5 dB
attenuation long guard interval than fixed OFDM and single carrier
Delay spread 0.41 µs 0.34 µs 1.43 µs 0.74 µs modulation [7]. This can be explained by the fact that in the
adaptive system, bad channels are not used or only used
Table 1: Parameters of radio channel propagation measurements with small signal alphabets so that a small amount of
interblock interference is not critical. But adaptive OFDM
Simulation results for four typical radio channels at a carrier exhibits also some disadvantages: The calculation of the
frequency of 1.8 GHz are presented. Table 1summarize the distribution of modulation schemes causes a high
parameters for all measurements. In case of the mobile computational effort. Additionally, the channel must not
scenarios (measurements 2 and 4), the user terminal antenna vary too fast because of the required channel estimation. A
was moved over a distance of 1 m with a low velocity. rapidly varying channel causes also a high amount of
Examples of the simulation results are presented in Fig. 2 signaling information with the effect that the data rate for
and 3. The figures show the bit error ratio as a function of the communication decreases. Furthermore, an OFDM
the average transmitted power. In all the examples shown, signal exhibits a Gaussian distribution with a very high crest
16- QAM (bandwidth efficiency: 4 bit/symbol) is used for factor. Therefore, linear power amplifiers with high power
single carrier modulation and fixed OFDM (systems 1 and consumption have to be used.
2). In case of adaptive modulation, the average bandwidth If channel coding is included in the transmission system
efficiency is the same as in case of fixed modulation. also, it has been shown in [1, 2] that OFDM with fixed
Therefore, only transmission systems with the same average modulation schemes shows approximately the same
data rate are compared. The main parameters of the performance as single carrier modulation with frequency
simulations are shown in Table 2. domain equalization.

240

Fig 2: Simulation results for a line-of-sight (LOS) radio channel with a)
a fixed (measurement 1) and b) a mobile (measurement 2) user terminal
antenna
The better performance of adaptive OFDM compared with
single carrier modulation results due to the capability of
adaptive OFDM to adapt the modulation schemes to sub-
channels with very different SNRs in an optimum way. In
order to improve the performance of single carrier
modulation, the latter can be combined with antenna
diversity using maximum ratio combining [8].
V. CONCLUSION
By using adaptive modulation schemes for the individual
sub-carriers in an OFDM transmission system, the required
signa1 power can be reduced dramatically compared with
fixed modulation.
Simulations show that for a bit error ratio of a gain of 5 to
14 dB can be achieved depending on the radio propagation
scenario. Also with single carrier modulation a significantly
better performance is obtained than with OFDM with fixed
modulation schemes. But adaptive OFDM outperforms
241
Fig 3: Simulation results for a non-line-of-sight (NLOS) radio channel with
a) a fixed (measurement 3) and b) a mobile (measurement 4) user
single carrier modulation by 3 to 5 dB. In addition to the
modulation schemes (bit distribution) also the power
distribution of adaptive OFDM can be optimized. But
simulations reveal that from the optimum power distribution
only a small gain of less than 0.5 dB is obtained. Therefore,
it is recommended to refrain from optimizing the power
distribution since either additional computation or additional
signaling for the synchronization is needed.
With adaptive OFDM and single carrier modulation, higher
gains - compared with conventional OFDM-are obtained for
NLOS channels than for LOS channels. Since NLOS radio
channels exhibit usually higher attenuation, this property is
of particular advantage. Furthermore, the simulation results
yield no significant differences between radio channels with
fixed and mobile user triennial antennas.
VI. REFERENCES
[1] H. SARI,G . KARAMA, ND I. JEANCLAUDEA:
An analysis of orthogonal frequency-division
multiplexing for mobile radio applications. In
Proceedings of the VTC '94 in Stockholm, pp. 1635-
1639 (2006).
[2] H. SARI,G . KARAM AND I. JEANCLAUDEF:
frequency-domain equalization of mobile radio and
terrestrial broadcast channels. In Proceedings of the
Globecom '94 in San Francisco, pp. 1-5 (2004).
[3] B. HIROSAKAI, . YOSHIDA0, . TANAKAS.,
HASEGAWAK, . INOUE, AND K. WATANABEA:
191.2 kbps voice band data modem based on
orthogonally multiplexed QAM techniques. In IEEE
International Conference on Communications, pp.
661-665 (1995).
[4] D. HUGHES-HARTOGS: Ensemble modem
structure for imperfect transmission media. U. S.
Patent 4,679,227 (1987).
[5] P. S. CHOWJ, M. CIOFFI AND J. A. C.
BINGHAMA: practical discrete multitone transceiver
loading algorithm for data transmission over
242
spectrally shaped channels. IEEE Trans. on
Communications 43 (1995), pp. 773-775.
[6] CZYLWIKC: Comparison of the channel
capacity of wideband radio channels with achievable
data rates using adaptive OFDM. In Proceedings of
the 5th European Conference on Fixed Radio
Systems and Networks ECRR '96, Bologna, pp. 238-
243 (1996).
[7] CZYLWIK: Adaptive (OFDM for wideband radio
channels. In Proceedings of the GLOBECOM '96,
London, pp. 713-718 (1996). G. KADEL: Diversity
and equalization in frequency domain - a robust and
flexible receiver technology for broadband mobile
communication systems. In Proceedings of the IEEE
Vehicular Technology Conference '97, Phoenix
(1997).
 2009 International Conference on Computer Engineering and Technology

A bivariate C 1 cubic spline space on Wang’s refinement

Huan-Wen Liu Wei-Ping Lu

Faculty of Mathematics & Computer Science Department of Computer Science
Guangxi University for Nationalities Guangxi Economic Management Cadre College
Nanning 530006, P.R. China Nanning 530006, P.R. China
mengtian29@163.com luweiping06@163.com

Abstract In this paper, we consider a kind of refined triangula-

In this paper, by using the technique of B-net method and
the minimal determining set, the dimension of the space of
bivariate C 1 cubic spline functions on a kind of refined tri-
angulation, called Wang’s refinement, is determined, and a
set of dual basis with local support is given.
tion ΔW which was first proposed by Wang [13], where the
dimension of the bivariate C 2 quintic spline space S 52 (ΔW )
was given. By using the technique of minimal determining
set, the dimension of the space S 31 (ΔW ) is determined and a
set of dual basis with local support is constructed.
2. Notation and Preliminaries

1. Introduction 2.1 Preliminaries

 
Let Δ be a regular triangulation of a simply connected 1 , v2 , v3 ∈ T , accord-
For any s ∈ S dr (Δ) and T (l) := v(l) (l) (l)

polygonal domain Ω in R2 , i.e., Δ is a set of closed triangles ing to the theory of Bernstein-Bézier polynomials by Farin
whose union coincides with Ω such that the intersection of [4], the restriction of s in T (l) can be expressed as
any two triangles in Δ is either empty, a common edge or   
v(l) ,v(l) ,v(l) d! i j k
a vertex. Let V, VI , VB , E, E I , E B and F denote the set of s(x, y)|T (l) = ci jk1 2 3 αβ γ , (2)
i! j!k!
vertices, interior vertices, boundary vertices, edges, interior i+ j+k=d
edges, boundary edges and triangles in Δ, respectively.
where (α, β, γ) is the barycentric coordinates of (x, y) with
Given 0 ≤ r < d, the space of bivariate splines over the
respect to the triangle T (l) , defined by
triangulation Δ is defined by
(x, y) = αv(l) (l) (l)
1 + βv2 + γv3 , α + β + γ = 1,
S dr (Δ) = {s ∈ C r (Ω) : s|T (l) ∈ Pd , l = 1, 2, . . . , |T |} , (1)
 
v(l) ,v(l) ,v(l)
where T is a triangle in Δ and Pd is the linear space of
(l) and ci jk1 2 3 are called the B-net coefficients of s(x, y) with
bivariate polynomials of total degree d. respect to
the triangle

T (l) . It is clear that each B-net coef-
As we know, for an arbitrary triangulation Δ, the dimen- v(l) ,v(l) (l)
2 ,v3
ficient ci jk1 is associated with a corresponding domain
sion of space S dr (Δ) is known only for those cases with point  
d ≥ 4r + 1 by Alfeld and Schumaker [1], d ≥ 3r + 2 by v(l) ,v(l) (l)
2 ,v3
Hong [6] and d = 4 and r = 1 by Alfeld et al. [2], respec- ξi jk1 1 + jv2 + kv3 )/d.
:= (iv(l) (l) (l)

tively. The dimension of space S dr (Δ) with low degree d  

verse smoothness r is quite difficult to determine and poorly
understood, since it depends not only upon the topological
variants but also the geometrical shape of the triangulation
as pointed out in several references, see Diener [3] and Mor-
gan and Scott [10]. This dependence of geometric structure
results in the fact that the dimension of S 31 (Δ) over an arbi-
trary triangulation becomes an open problem though several
results have been obtained for some special triangulations,
such as Farin [5], Lai [7] and Liu [9].
v(l) ,v(l) ,v(l)
For convenience, the set of all the domain points ξi jk1 2 3
will be denoted by Dd,Δ . For each ξ ∈ Dd, , let λξ be the
linear functional such that for any spline s ∈ S d0 (),
λξ s = the B-net coefficient cξ of s at point ξ.
If S is a linear subspace of S d0 (), then M ⊆ Dd, is said
to be a determining set for S provided that
s ∈ S and λξ s = 0 for all ξ ∈ M implies s ≡ 0. (3)

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DOI 10.1109/ICCET.2009.15
 (l)
v1(l)
v 1

w3(l) a
w3(l) c
d

w (l)
2
w2(l)

w1(l) w1(l)

v2(l) v3(l) v2(l) v3(l)

(l)
(l)
Figure 1: The refined triangulation T W of a single triangle. Figure 2: A minimal determining set for S 31 (T W ) marked
with , ,  and .

M is called a minimal determining set (MDS) for S if there (l)

is no other determining set for S with the cardinality being Following Wang [13], let T W shown in Figure 1 be a
(l)
smaller than M. It is easy to see that if M is a MDS for S , refining subdivision of T formed by the following steps:
then dim S = M, where M denotes the cardinality of M. Step 1. Take three interior points w(l) (l) (l)
1 , w2 , and w3 in
We now introduce the C 1 smoothness condition between each triangle T (l) respectively, where w(l) 1 (l) 4 (l)
j = 7 v j + 7 v j+1 +
two adjacent triangles, which is a special case of the general 2 (l)
and v(l) (l)
7 v j+2 j+3 := v j , j = 1, 2, 3; l = 1, 2, · · · , |T |.
C r smoothness condition given by Farin [4].
Lemma 2.1. Let T (1) = [v1 , v2 , v3 ], T (2) = [v1 , v2 , v4 ] be Step 2. Join v(l) (l) (l) (l)
j to w j and v j to w j+1 , respectively,
two adjacent triangles in R2 with a common edge [v1 , v2 ]. where w(l) (l)
j+3 := w j , j = 1, 2, 3; l = 1, 2, · · · , |T |.
Suppose p(x, y) agrees with p1 (x, y) ∈ P3 in T (1) , and with We therefore obtain the refined triangulation ΔW =
|T | (l)
p2 (x, y) ∈ P3 in T (2) , assume l=1 T W of the original triangulation Δ.
 3! i j k
p1 (x, y) = ci[vjk1 ,v2 ,v3 ] αβ γ, (4)
i! j!k! 1 1 1 3 A minimal determining set for S 31 (T W(l) )
i+ j+k=3
 3! i j k
p2 (x, y) = ci[vjk1 ,v2 ,v4 ] αβ γ, (5) In this section, we establish a theorem concerning MDS
i+ j+k=3
i! j!k! 2 2 2 (l)
for S 31 (T W ).
where (α1 , β1 , γ1 ) and (α2 , β2 , γ2 ) are the barycentric coor- Theorem 3.1. Let M be the set of domain points associ-
(l)
dinates of (x, y) with respect to the triangles T (1) and T (2) , ated with the following 16 B-net coefficients for S 31 (T W ):
  
v(l) ,w(l) ,w(l)
respectively. Then p(x, y) ∈ C 1 (T (1) T (2) ) if and only if 1) ci jkt t+1 t+2
, i + j + k = 3, i > 1, t = 1, 2, 3,
the following holds:  
v(l) (l) (l)
t ,wt+1 ,wt+2
2) c111 , t = 1, 2, 3,
 
ci[vj01 ,v2 ,v4 ] = c[v 1 ,v2 ,v3 ]
, i + j = 3, v(l) ,v(l) ,w(l)
i j0
3) c111t t+1 t+2 , t = 1, 2, 3,
 (l) (l) (l) 
[v1 ,v2 ,v3 ] 1 ,v2 ,v3 ] 1 ,v2 ,v3 ]
w ,w ,w
ci[vj11 ,v2 ,v4 ] = αci+1 j0 + βc[v
i j+10 + γc[v
i j1 , i + j = 2, 4) c1111 2 3 ,
(6) where v(l) (l) (l)
4 := v1 , w5 := w2 , w4
(l) (l)
:= w(l)
1 , these domain
where (α, β, γ) is the barycentric coordinates of v4 with re- points are marked with , ,  and  in Figure 2. Then M
spect to the triangle T (1) . (l)
is a MDS for S 31 (T W ).
(l)
Proof : Suppose that the B-net coefficients of S 31 (T W )
2.2 Wang’s refined triangulation listed in 1) - 4) are set to zero, then
i) By using the C 1 smoothness
  (l) conditions
 (Lemma 2.1)
Let vi (i = 1, 2, · · · , |V|) be vertices of Δ, e j ( j = along the edges v(l) (l) (l)
t , wt+1 , vt , wt+2 , t = 1, 2, 3, all the
1 , v2 , v3 (l =
1, 2, · · · , |E|) be edges of Δ, and T (l) = v(l) (l) (l)
B-net coefficients associated with the domain points in
1, 2, · · · , |T |) be triangles of Δ. D1 (v(l)
t ), t = 1, 2, 3, must be zero.

244
 ii) It is noted that Proof: We choose the following domain points to con-
struct a determining set P for S 31 (ΔW ):
w(l) (l) (l) (l)
3 = α1 v1 + β1 w2 + γ1 v3 a) Three B-net coefficients to determine all the B-net co-
efficients associated with the domain point in D1 (v) around
with (α1 , β1 , γ1 ) = (0,
 (l)2, −1).
1
 By using C smoothness con-
(l) each vertex v ∈ V in the original triangulation Δ. The re-
ditions across edge v1 , w2 , we have lated three domain points are marked with  in Figure 3.
    b) For each edge in the original triangulation Δ, choose a
v(l) ,w(l) ,w(l) v(l) ,w(l) ,v(l)
c1111 2 3 = 2a − c1111 2 3 . domain point associated with that edge. The related domain
  point is marked with  in Figure 3.
v(l) ,w(l) c) For each triangle T (l) in the original triangulation Δ,
(l)
t+1 ,wt+2
Thus a = 0. Similarly the B-net coefficients c120t ,
we choose four domain points, each of them is the interior
t = 2, 3, indicated by are annihilated.
domain point located in the triangle which does not include
iii) It is also noted that
any edge of Δ. These domain points are marked with  and
 in Figure 3.
2 = α2 v1 + β2 v2 + γ2 w3
w(l) (l) (l) (l)
We now set all B-net coefficients of s ∈ S 31 (ΔW ) asso-
with (α2 , β2 , γ2 ) = (−2, −1, 4). By using C1 smoothness ciated with all domain points in P to zero. Then it follows
conditions (Lemma 2.1) across edge v(l) (l)
1 , w3 , we have
from a) that all B-net coefficients in D1 (v) around every ver-
tex v ∈ V must be zero. And by using Theorem 3.1, all the
 (l) (l) (l)   (l) (l) (l)   
v ,w2 ,w3 v ,v2 ,w3 v(l) ,v(l) (l)
2 ,w3
remaining B-net coefficients in each triangle are zero. So s
1
c111 = −2c201
1
− c1111 + 4b. must vanish identically inside each subtriangle. This mean

v(l) (l) (l)
 that P is a determining set and then dim S 31 (ΔW ) ≤ |P|. It is
t ,wt+1 ,wt+2
Thus b = 0. Similarly the B-net coefficients c102 , easy to see that the total number of B-net coefficients in the
t = 2, 3, indicated by ♦ are annihilated. determining set P is 3|V| + |E| + 4|T |. This completes the
iv) Noticing that proof of this theorem.
Theorem 4.2.
v(l) (l) (l) (l)
1 = α3 w1 + β3 w2 + γ3 w3
dim S 31 (ΔW ) = 3|V| + |E| + 4|T |. (8)
with (β3 , α3 , γ3 ) = (−1,
 0, 2),  by using C smoothness con-
1

ditions across edge w(l) , w(l) Proof: Using the lower bound formula by Schumaker
2 3 , we have
[11], we have
   
v(l) ,w(l) ,w(l) w(l) ,w(l) ,w(l) 
|VI | 
c1111 2 3
= −c1111 2 3
+ 2c.  2

 
dim S 31 (ΔW ) ≥ 10+3|E I |−7|VI |+ ( j + 2 − jei )+ , (9)
w(l) ,w(l) ,w(l) i=1 j=1
Thus c = 0. Similarly the B-net coefficients c012t t+1 t+2 ,
t = 2, 3, indicated by  are zero. And then we have where |VI | denote the number of interior vertices in ΔW , |E I |
   (l) (l) (l)  the number of edges in ΔW and ei the number of distinct
v(l) ,w(l) (l)
2 ,w3 w ,w2 ,w3
c1201 = −c1201 + 2d. slopes assumed by these edges. Noticing that

By using Lemma 2.1 and item ii), we obtain d = 0. Simi- |E I | = 9|T | + |E I |, |VI | = |VI | + 3|T |,

w(l) (l) (l)
t ,wt+1 ,wt+2
larly the B-net coefficients c021 , t = 2, 3, indicated and
by  are zero.  (l) (l) (l)  

|VI | 
2 
w ,w ,w
v) It follows from above that items t = c300t t+1 t+2 , ( j + 2 − jei )+ = [(3 − ei )+ + (4 − 2ei )+ ] = 0,
1, 2, 3, are forced to zero by Lemma 2.1. i=1 j=1

Therefore s ≡ 0. Noticing that T W (l)

is a quasi-cross-cut we have
(l)
partition, according to [12], we know dim S 31 (T W ) = 16, the
set M is a MDS for S 31 (T W(l)
). The proof is completed. dim S 31 (ΔW ) ≥ 10 + 6|T | + 3|E I | − 7|VI |.
Using the Euler Theorem
4 Main Results
|T | = |E I |−|VI |+1, |E| = 2|E I |−3|VI |+3, |V| = |E I |−2|VI |+3,
Theorem 4.1. Let Δ be an arbitrary triangulation and ΔW we have dim S 31 (ΔW ) ≥ 3|V| + |E| + 4|T |. Which together
be the Wang’s refined triangulation of Δ, then with Theorem 4.1 yield that
dim S 31 (ΔW ) ≤ 3|V| + |E| + 4|T |. (7) dim S 31 (ΔW ) = 3|V| + |E| + 4|T |.

245
 nan Key Laboratory for Computation and Simulation in Sci-
ence and Engineering.

References

[1] Alfeld, P., Schumaker, L.L. (1987). “The dimension of

spline spaces of smoothness r for d ≥ 4r + 1”. Con-
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[2] Alfeld, P., Piper, B., Schumaker, L.L. (1987). “An ex-
plicit basis for C 1 quartic bivariate spline”. SIAM J.
Numer. Anal., Vol.24, pp.891-911.
[3] Diener, D. (1990). “Instability in the dimension of
spaces of bivariate piecewise polynomials of degree
2r and smoothness r”. SIAM J. Numer. Anal., Vol.27,
pp.543-551.

Figure 3: A MDS for S 31 (ΔW ) marked with , ,  and . [4] Farin, G. (1986). “Triangular Bernstein-Bézier
patches”. Comput. Aided Geom. Des., Vol.3, pp.83-
128.
The proof of the theorem is completed.
[5] Farin, G. (2006). “Dimensions of spline spaces over
It is clear that the set P constructed in Theorem 4.1 is a
  unconstricted triangulations”. J. Comput. Appl. Math.,
MDS for S 31 (ΔW ). Denote P = ξ1 , ξ2 , . . . , ξ|P| . For each
Vol.192, pp.320-327.
ξ j ∈ P, we define spline B j ∈ S 31 (ΔW ) to satisfy
[6] Hong, D. (1991). “Spaces of bivariate spline functions
λξi B j = δi j , for all ξi ∈ P. (10) over triangulations”. Approx. Theory Appl., Vol.7,
The set {Bi , i = 1, ..., |P|} obviously forms a dual basis of pp.56-75.
S 31 (ΔW ), which is commonly called the dual basis corre- [7] Lai, M.J. (1996). “Scattered data interpolation and
sponding to P. approximation by using bivariate C 1 piecewise cu-
We now analyze the support properties of the basis bic polynomials”. Comput. Aided Geom. Des., Vol.13,
function Bi . Suppose ci is the B-net coefficient of Bi which pp.81-88.
is set to 1. If ci is associated with a domain point lying in
D1 (v) around a vertex v ∈ Δ, then the support set of Bi is all [8] Lai, M.J., Schumaker, L.L. (2007). Spline Functions
triangles of Δ sharing

v. If ci is associated with a domain over Triangulations. Cambridge University Press.
w(l) ,w(l) ,w(l)
 (l) in (l)ξ111(l) 
1 2 3
point lying , then the support set of Bi is the [9] Liu, H.W. (1996). “The dimension of cubic spline
triangle v1 , v2 , v3 . If ci is associated with a domain point space over stratified triangulation”. J. Math. Res. Exp.,
 
v(l) ,w(l) ,w(l) Vol.16, pp.199-208.
lying in ξ111t t+1 t+2
(t = 1, 2, 3), then the support
 set(l)of (l)Bi
is the union of those subtriangles located in v(l) [10] Morgan, J., Scott, R. (1975). “The dimension of piece-
1 , v2 , v3
wise polynomial”, manuscript.
t , wt+1
sharing vertices v(l) , w(l)
(l)
 t+2 . If ci is associated with a
v(l) ,v(l) ,w(l) [11] Schumaker, L.L. (1979). “On the dimension of
domain point lying in ξ111t  t+1 t+2 (t = 1, 2, 3), we consider
space of piecewise polynamials in two variables”. In:
two situations. Firstly, if v(l) (l)
t , vt+1 is a boundary edge of
  Schempp, W., Zeller, K.(eds.): Multivariate Approxi-
Δ, then the support set of Bi is the triangle v(l) (l) (l)
t , vt+1 , wt+1 . mation Theory. Birkhäyser Verlag, pp.396-411.
 (l) (l) 
Secondly, if vt , vt+1 is an interior edge of Δ, then the
  [12] Wang, R.H. (1975). “The structural characterization
support set of Bi is the the union of v(l) (l) (l)
t , vt+1 , wt+1 and and interpolation for multivariate splines”. Acta Math.
 (l) (l) (l)   
vt , vt+1 , w̃t+1 , where v(l) (l) (l)
t , vt+1 , w̃t+1 is the subtriangle
Sinica, Vol.18, pp.91-106.
   (l) (l) (l) 
sharing a common edge v(l) t , vt+1 with vt , vt+1 , wt+1 .
(l)
[13] Wang, T.J. (1992). “A C 2 -quintic spline interpolation
scheme on triangulation”. Comput. Aided Geom. Des.,
Acknowledgements. The first author is supported by the Vol.9, pp.379-386.
Natural Science Foundation in Guangxi (0575029) and Hu-

246
 2009 International Conference on Computer Engineering and Technology

Fast Shape Matching Using a Hybrid Model

Gang Xu, Wenxian Yang

Department of Electrical and Electronic Engineering
North China Electric Power University
Beijing 102206, China
E-mail: xugang@ncepu.edu.cn, ywxzgs81@163.com

Abstract—A hybrid model is proposed to finish image shape the image quickly among the given test images and the
matching from coarse to fine, which is composed of three image has the greatest similarity to the template image. After
parts: rough matching, accurate matching, and optimum edge extracting, a fast strategy is given as the rough measure
matching search. According to the partial Hausdorff distance, of the similarity between the template and images and a
a fast strategy for rough matching and a new improved partial novel partial Hausdorff distance is proposed to compute the
Hausdorff distance for accurate matching are presented as the shape similarity accurately. Finally, a new genetic algorithm
measures of the degree of shape similarity between the based on fuzzy logic, which can adaptively regulate the
template and images. At the same time, a new genetic probabilities of crossover and mutation, achieves the
algorithm based on fuzzy logic, which can adaptively regulate
optimum shape matching with higher search speed and
the probabilities of crossover and mutation, is used to search
quality.
the optimum shape matching quickly. The experimental results
show that the model achieves the shape matching with higher II. THE HAUSDORFF DISTANCE
speed and precision compared with the traditional matching
algorithms and can be used in real-time image matching and On the basis of reviewing the conventional and existing
pattern recognition. improved Hausdorff distances, two new measures of shape
similarity between the template and images are proposed to
Keywords-shape matching; Hausdorff distance; genetic meet the real-time image matching.
algorithm; fuzzy control
A. Conventional HD and Partial HD
I. INTRODUCTION Given two finite point sets:
A = {a1 , a 2 , " , a p } and B = {b1 , b2 ,", bq } , the HD is defined
The combination of Hausdorff distances and genetic
algorithms can effectively detect rotating, and scaling of as follows:
objects in image shape matching.
The Hausdorff distance (HD) [1] measures the mismatch H ( A , B ) = m ax ( h ( A , B ), h ( B , A )) , (1)
of two sets and is more tolerant to perturbations in image
matching for it measures proximity rather than exact
superposition. However, the conventional Hausdorff , (2)
distances require high computational complexity and are not h ( A , B ) = m a x m in a − b
a∈ A b∈ B
suited to the practical applications. Some modified Hausdorff
distances have been proposed recently. Huttenlocher
proposed the partial Hausdorff distance [2], a directed , (3)
h ( B , A ) = m a x m in b − a
modified Hausdorff distance (MHD) was introduced by b∈ B a∈ A
Dubuisson [3], a robust Hausdorff distance was used in [4],
two different improved Hausdorff distances were introduced ⋅
is any norm on the points of A and B, we employ
in [5],[6]. the Euclidean distance in this paper. The distance between a
The genetic algorithm (GA) [7] is such a search and point and a set is defined as the minimum distance from the
optimization method, which has developed to stimulate the point to all points of the set. The directed
mechanism of natural evolution and is powerful in finding distance h( A, B) identifies the largest distance from the point
the global or near global optimal solution of optimization
a ∈ A to B. H ( A, B) is defined as the maximum of
problems. The genetic algorithm has found kinds of
applications successfully and has shown to be of great h( A, B) and h(B, A) , the computational complexity
promising. The Hausdorff distance and genetic algorithm is O ( p ⋅ q ) .
were applied to object detection in images [8], the The directed partial Hausdorff distance (PHD) is defined
applications of Hausdorff distances and genetic algorithms as follows:
were researched in image shape matching as in [6],[9].
Based on the above-mentioned articles, a hybrid model
hl ( B, A) = Lthb∈B h(b, A) , 1 ≤ l ≤ q , (4)
for fast shape matching is introduced in this paper to search

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DOI 10.1109/ICCET.2009.53
 Where Lthb∈B denotes the Lth ranked value of h(b, A) .
B. Two Novel Measures of Shape Similarity Based on the
Partial Hausdorff Distance
For fast and effective image matching, two novel
similarity measures are given on the basis of [5],[6].
1) A Novel Partial Hausdorff Distance for Accurate
Measurement: Whether the conventional or partial
Hausdorff distance, the distances between every point of the
template B and the matched image A are computed, this
computational cost limits Hausdorff distance’s practical
applications. In fact, for each point of B, in the matched
image only one point or several points in its neighborhood
have the smaller distance with the point, thus it is
unnecessary to compute each distance. For higher matching
speed and accuracy, a novel partial HD is proposed as
follows:
For the point b j (1 ≤ j ≤ q ) in B, the small
neighborhood M2 (m2×m2), which correspond with b j in A,
is searched to find the feature points and the directed
Hausdorff distance h(bj , A) from b j to A is the smallest
distance from b j to these points; if there are not points in the
neighborhood, h(bj , A) is a given maximum. Ranking h(bj , A)
dimension. It’s better to adopt the genetic algorithm to
search the optimum in this model. In view of the
shortcomings of simple genetic algorithm (SGA), such as
premature convergence and slow convergence, a new
genetic algorithm based on fuzzy logic is used which can
regulate the probabilities of crossover and mutation
adaptively.
III. THE GENETIC ALGORITHM WITH PROBABILITIES OF
CROSSOVER AND MUTATION SELF-ADAPTION BASED ON
FUZZY CONTROL
The convergence speed and solution quality are directly
influenced by the probabilities of crossover Pc and mutation
Pm. It’s known that the self-organizing genetic algorithm has
higher robustness, faster convergence and the global
optimization. There have been some Pc and Pm self-adaption
algorithms, of which the algorithms based on fuzzy logic
have better performances.
A new genetic algorithm is proposed in [10] and
adaptively regulates Pc and Pm by inquiring the table based
on fuzzy control, the differences of the population average
fitness and standard deviation between two adjacent
generations are inputs, Pc and Pm are outputs. Two self-
adaption normalized operators are given. This algorithm has
better performance and we use it to search the optimum.

from small to large, the lth value is h2 ( B, A) which is used IV. THE HYBRID MODEL FOR SHAPE MATCHING
for accurate similarity measure. Compared with [6], this The hybrid model composed of these algorithms can
distance has higher accuracy with the same speed. finish shape matching through two parts: the rough matching
2) A Fast Strategy for Rough Measurement: Initialize and the accurate matching. The process of shape matching is
the counter Mt0=0. For the point b j in B, the small shown as Fig.1; the flow diagrams of rough matching and
accurate matching are shown in Fig.2 and Fig.3.
neighborhood M1(m1×m1), which correspond with b j in A,
is searched to find the feature points. If there are points, Mt0
add one, if there are not points, Mt0 is unchanged.
Normalized value h1 ( B , A) = 1 − Mt 0 / q is called as the fast
similarity measure between A and B. Two images are
similar only when Mt 0 / q ≥ l1 × q , 0 ≤ l1 ≤ 1 is true.
3) The Optimum Model for Shape Matching: Quickly
retrieving the given test images by rough and accurate
matching, image rotation (rotation coefficient θ ) and scale
(scale coefficient m) are considered. Fig.1. The process of shape matching.
For A and B are the sets of the edge points of the
matched images and the template respectively. Equation
t = (θ , m ) is the parameter of template rotation and scale.
The optimum model is defined as follows:

min h (t ( B ), A ) , (5)
t

Where t ( B) = m ⋅ ⎡cos(θ ) − sin(θ ) ⎤ ⋅ B .

⎢ sin(θ ) cos(θ ) ⎥
⎣ ⎦
Therefore, the key is to find the optimum
parameter (θ , m ) to get the best matching. The genetic
algorithm is widely used for the optimization problems
including objective functions are discontinuous, high Fig.2. The flow diagram of rough matching.

248
 V. RESULTS AND ANALYSIS
The algorithms are implemented by Matlab7.0 in the
Platform of Intel (R) Core (TM)2 Duo CPU (2.00GHz). The
test images are showed in table 2.

TABLE II. TEST IMAGES

1 2 3

4 5 6
Fig.3. The flow diagram of accurate matching.

A. The Implementation Steps

1) Edge Extracting: Canny operator is employed.
2) Definition of Objective Functions: The functions for 7 8 9
rough and accurate matching are minimized and defined as
follows:

f1 (θ , m) = h1 (t (B), A) = 1− Mt0/ q f1 ∈ (0,1) , (6)

10 11 12
f 2 (θ , m) = h2 (t ( B ), A) , (7)

Where t ( B) = m ⋅ ⎡cos(θ ) − sin(θ )⎤ ⋅ B .

⎢ sin(θ ) cos(θ ) ⎥
⎣ ⎦
3) Parameters and Ranges: Population size N, the
largest iterative number NM, the initial value of Pc and Pm 13 14 15
etc.
4) Optimum Results Search: The genetic algorithm is
adopted.
B. Computational Efficiency Analysis
The computational times of addition and subtraction (+/-
), multiply and divide (×/÷), and comparison of the fast 16 17
strategy and Hausdorff distance in this paper are showed in
table 1 compared with the Hausdorff distance in [6],[9]. (The
worst situation is considered in the partial Hausdorff distance
and fast strategy. Generally M1 is a little bigger than M2,
M1 and M2 are much less than p).

TABLEⅠ. COMPUTATIONAL EFFICIENCY ANALYSIS

Algorithms +/- ×/÷ Comparison

HD reference [9] 6q×p 6q×p 2q×p The template is the same as NO.2 image; NO.1-10 are
reference [6] 3q 3q M2+q
PHD
this paper 3q×M2 3q×M2 M2×q+q
fishes with kinds of shapes (128×128); NO.11-15 are images
the fast strategy after rotating the template 15, 90, 167, 245, and 315 degree
q 1 M1×q counter-clock-wise respectively; NO.16 is the image after
(in this paper)
narrowing the size of the template one times and rotating it
From table1, the methods proposed in this paper lessen 90 degree; NO.17 is the image after doubling the size of the
the computational cost to a great degree. template and rotating it 45 degree.

249
 Setting parameters: N is 20, NM is 100, the coding length 0.25

of θ and m is 10 and 5 respectively with binary coding, and SGA

GA
the initial value of Pc and Pm is 0.6 and 0.05 respectively.

Standard deviation
The selection, crossover, and mutation operator are adopted
0.2
as in [11]. Each algorithm has 10 operations to overcome the
randomness in GA.
A. Comparison Between Several Hausdorff Distances
Take the matching between the template and NO.11 for 0.15

example. Adopting the same genetic algorithm (proposed in

this paper), the Hausdorff distance and fast strategy proposed
in this paper are compared with which as in [6],[9], shown in
table 3. Noting that the running time is relative because it is 0.1
0 10 20 30 40 50 60 70 80 90 100
influenced by kinds of external factors such as the computer Generation
running time. Fig.5. The changes of standard deviation.

TABLE III. COMPARISON BETWEEN SEVERAL HAUSDORFF From Fig.4 and Fig.5, the optimized function has lower
DISTANCES
average fitness and higher standard deviation in GA. The
Optimal results lower population average fitness means that the average
（ θ ，m）
Algorithms Running time distance between individuals of each generation and the
HD reference [9] 27min 39s (15.1320, 0.9839) template is smaller and the search quality is better in the
reference [6] 2min 58s (15.4839, 0.9839) process of searching. The higher standard deviation shows
PHD
this paper 3min 1s (14.7801, 0.9839) that the individuals are more dispersive, which are very
the fast strategy
2min 50s (14.3109, 1.0323) helpful to wider search.
(in this paper)
2.5

From table3, the HD given in this paper has higher

Crossover probability

matching accuracy as well as faster speed and the fast 2

strategy has lower computational complexity. The
advantages become more obvious when the images are
larger. 1.5

B. The Changes of Population Average Fitness, Standard SGA

Deviation, Crossover Probability Pc, and Mutation GA
1
Probability Pm
Take the rough matching between the template and
NO.11 for example, the changes of population average 0.5
fitness, standard deviation, Pc, and Pm of GA given in this 0 10 20 30 40 50 60 70 80 90 100

paper are shown in Fig.4, Fig.5, Fig.6, and Fig.7 respectively Generation
compared with SGA. Fig.6. The changes of crossover probability.

1 0.24

0.22
0.95 SGA
Mutation probability

0.2
GA
Average fitness

0.9 0.18

0.16
0.85
0.14
0.8
0.12
SGA
0.75 0.1 GA

0.08
0.7
0.06

0.65 0.04
0 10 20 30 40 50 60 70 80 90 100 0 10 20 30 40 50 60 70 80 90 100
Generation Generation
Fig.4. The changes of population average fitness. Fig.7. The changes of mutation probability.

250
 From Fig.6 and Fig.7, the changes of Pc and Pm in GA match with the template exactly is removed. The parameters
are frequent with the changes of average fitness and standard show that the matching has low error and high quality.
deviation, whereas Pc and Pm in SGA remain in the process
of searching,, which show that the GA given in this paper VI. CONCLUSIONS
has a strong self-adaptive ability. The hybrid model produced in this paper has features as
The parameters of SGA: ( θ , m) = (17.0088, 0.9839), shown below:
GA: ( θ , m) = (14.3109, 1.0323), the average optimum • Quickly and effectively finishing the shape matching
iterative times of SGA and GA are 74 and 51 respectively. from coarse-to-fine with the fast strategy and
improved HD.
C. Results and Analysis of Rough and Accurate Matching • The genetic algorithm with the probabilities of
The results of rough and accurate matching are shown in crossover and mutation self-adaption based on fuzzy
tables 4 and 5 (n denotes the optimal generation, √ denotes control is adopted to get higher speed, better quality,
the image is selected, Y denotes the image is exactly and stronger self-adaptive ability.
matched with the template). In conclusion, compared with other similar matching
methods, this model can be used for image recognition and
TABLE IV. RESULTS OF ROUGH MATCHING matching in practice. Its stability is not that great yet and will
be studied next.
NO. n
θ m
The fast
Selected
(degree) measures REFERENCES
1 87 45.5719 0.8145 0.0002 ×
2 71 2.5513 0.9597 0 √
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4 87 181.2317 0.5 0.0003 × Comparing images using the Hausdorff distance
5 98 349.0909 0.5 0.0002 × undertranslation, In: Proc. CVPR ’92., 1992 IEEE Comput.
6 92 217.478 0.8871 0.0002 × Soc.Conf. on 15–18 June, Comput. Vision Pattern Recogn.,
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13 83 168.9150 1.0565 0 √ algorithms using robust Hausdorff distance measures, IEEE
14 61 247.0381 1.0323 0.0001 √ Transactions on Image procession, December 1999, pp. 425-
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[6] Zang Tiefei, Shen Tingzhi, Chen Jianjun, and Gu Jianjun, The
TABLE V. RESULTS OF ACCURATE MATCHING Application of Improved Hausdorff Distance and Genetic
Algorithm in Image Matching, Journal of Beijing Institute of
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θ m HD Matched Technology, 2000, 20 (6), pp. 733-737.
(degree)
2 62 1.9795 0.9718 0 Y [7] Holland J.H., Adaption in Natural and Artificial Systems, The
3 95 203.6657 0.8992 0.0005 N University of Michigan Press, Ann Arbor, M I, 1975.
11 58 14.7801 0.9839 0 Y [8] Liu Jianzhuang, Xie Weixin, Gao Xinbo, Ma Kun, Hausdorff
12 67 90.7918 0.9839 0 Y Distance Based Object Matching with Genetic Algorithms,
13 66 167.5074 1.0202 0 Y Acta Electronica Sinica, 1996, 24 (4), pp. 1～6.
14 54 246.5103 1.0081 0 Y
15 79 315.6598 0.9839 0 Y [9] Zhang Wenjing, Xu Xiaoming, and Su Jianfeng, An Improved
16 49 91.1437 0.5 0 Y Algorithm for 2D Shape Matching Based on Hausdorff
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106-109.
[10] Li Qing, Zhang Wei, Yin Yixin, and Wang Zhiliang, A new
From table 4, NO.2, 3, 11, 12, 13, 14, 15, 16, and 17 are adaptive algorithm for regulating the probabilities of
selected after rough matching. The image 3 is selected crossover and mutation, Control and Decision, 2008, 23 (1),
which is similar to the template; other images are kinds of pp. 79-83.
affine transformations of the template and are also selected. [11] Yang Shuying, Image Pattern Recognition——VC++
The matching results fit with human vision. Implementation, Tsinghua University Press, Beijing Jiaotong
On the basis of rough matching, eight images of University Press, 2005.
transformations of the template are gotten by accurate
matching as shown in table 5. The image 3 which does not

251
 2009 International Conference on Computer Engineering and Technology
A Multi-objective Genetic Algorithm for optimization
of Cellular manufacturing system
H. kor, H. Iranmanesh, H. haleh, and S. M. Hatefi
Department of Industrial Engineering, College of Engineering
University of Tehran
Tehran, Iran
hkor@ut.ac.ir, hiranmanesh@ut.ac.ir, haleh24@hotmail.com, s_m_hatefi@yahoo.com
Abstract—Cellular manufacturing (CM) is an important
application of group technology (GT) in which families of parts
is produced in manufacturing cells or in a group of various
machines. In this paper, a genetic algorithm approach is
proposed for solving multi-objective cell formation problem.
The objectives are the minimization of both total moves
(intercell as well as intracell moves) and the cell load
Variation. In this paper, authors used a SPEA-II method as
well known and efficient standard evolutionary multi-objective
optimization technique. This hybrid method presents the large
set of non-dominance solutions for decision makers to making
best solution. The efficiency of multi-objective GA-SPEA II is
illustrated on a large-sized test problem taken from the
literature.
Keywords— Cellular Manufacturing System, Genetic
Algorithm, Multi-Objective, Pareto set, SPEA II.
I. INTRODUCTION
The Cellular manufacturing (CM) is the application of
group technology (GT) in manufacturing systems.
Recent surveys [1,2] indicate that the practical
implementation of a cellular manufacturing system involves
the optimization of many conflicting objectives. Given the
size of the cellular manufacturing research output, it is
surprising that relatively few solution methodologies
explicitly address the multi-objective version of the
problem. Even when multiple objectives are considered,
optimization is normally achieved through the aggregation
of the objectives into a single composite objective, as
indicated in the reviews by Mansouri et al. [3] and
Dimopoulos [4]. This technique does not provide the system
designer with a set of alternative trade-off solutions, which
is the main aim of a typical multi-objective optimization
process.
II. DEFINITIONS
A general multi-objective optimization problem can be
described as a vector function ݂ that maps a tuple of m
parameters (decision variables) to a tuple of n objectives.
݂ଵ Cell load variation:
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 252
DOI 10.1109/ICCET.2009.212
Formally [5]: Where x is called the decision vector, X is the
parameter space, y is the objective vector, and Y is the
objective space.
The set of solutions of a multi-objective optimization
problem consists of all decision vectors for which the
corresponding objective vectors cannot be improved in any
dimension without degradation in another. These vectors
are known as Pareto optimal. Mathematically, the concept
of Pareto optimality is as follows: Assume, without loss of
generality, a maximization problem and consider two
decision vectors ܽǡ ܾ ‫ܺ א‬Ǥ
. Then, a is said to dominate b (also written as a>b) if
‫ א ݅׊‬ሼͳǡ ʹǡ͵ǡ ǥ ǡ ݊ሽǣ݂௜ ሺܽሻ ൒ ݂௜ ሺܾሻ  ‫ר‬
‫ א ݆׌‬ሼͳǡʹǡ͵ǡ ǥ ǡ ݊ሽǣ݂௝ ሺܽሻ ൐ ݂௝ ሺܾሻ
III. MATHEMATICAL MODEL
According to the literature, the most fundamental
objectives for the CFP are Minimization of intercell flows
and cell load variation, or their combinations. We can
benefit from lower parts transfer cost due to the
minimization of intercell flows and higher within-cell
machine utilization due to the minimization of cell load
variation. Several measures have been used to evaluate the
objective function of the CFP, such as grouping efficiency
proposed by Chandrasekharan and Rajagopalan [6], group
efficacy by Kumar and Chandrasekharan [7] and bond
energy by Miltenburg and Zhang [8]. However, these
measures are only suitable for those CFPs whose machine–
part flow chart is a 0–1 matrix denoting the manufacturing
relationship between machines and parts, without
considering other important design factors such as
processing time, production requirements and available time
on machine in a given period, etc. In our research, we
propose to minimize intercell flows and cell load variation
in a consideration of the processing time and available time
on machine in a given period. The mathematical model is
given as [9]:
 ௠ ௖ ௣
As SPEA (Strength Pareto Evolutionary Algorithm) [5]
݂ଵ ൌ ෍ ෍ ෍ ‫ݔ‬௜௟ ሺ‫ݓ‬௜௝ െ ݉௟௝ ሻଶ ሺͳሻ forms the basis for SPEA2, we give a brief summary of the
௜ୀଵ ௟ୀଵ ௝ୀଵ algorithm here. For a more detailed description the
݂ଶ ǣ –‘–ƒŽ‘˜‡• interested reader is referred to Zitzler [5].
௣ ௞೔ ିଵ ௣ This approach was conceived as a way of integrating
݂ଶ ൌ ߠଵ ෍ ෍ ȁܿ௞ െ ܿ௞ାଵ ȁ ൅ ߠଶ ෍ ݉௜ ሺʹሻ different MOEAs (Multi Objective Evolutionary
௜ୀଵ ௞ୀଵ ௜ୀଵ
Algorithms). SPEA uses an external archive containing
nondominated solutions previously found (the so-called
Where: external nondominated set). At each generation,
ܿ௞ The cell number in which operation k is nondominated individuals are copied to the external
performed on part i taking into consideration nondominated set. For each individual in this external set, a
the sequence of operations. strength value is computed. This strength is similar to the
ܿ௞ାଵ The cell number in which operation (k+1) is ranking value of MOGA [12], since it is proportional to the
performed on part i taking into Consideration number of solutions to which a certain individual
the sequence of operations. dominates. In SPEA, the fitness of each member of the
݇௜ the total number of operations to be current population is computed according to the strengths of
performed on pan i to complete its all external nondominated solutions that dominate it. The
processing requirements fitness assignment process of SPEA considers both
c The number of cells. closeness to the true Pareto front and even distribution of
‫݌‬ The number of parts. solutions at the same time. Thus, instead of using niches
݉௜ The total number of intracell moves based on distance, Pareto dominance is used to ensure that
performed by part i in order to complete its the solutions are properly distributed along the Pareto front.
processing requirements, taking into Although this approach does not require a niche radius, its
consideration the sequence of operations. effectiveness relies on the size of the external nondominated
ߠଵ ǡ ߠଶ The fractions representing the weights set. In fact, since the external nondominated set participates
attributed to the intercell and intracell moves in the selection process of SPEA, if its size grows too large,
respectively (ߠଵ and ߠଶ are assumed to be 0.7 it might reduce the selection pressure, thus slowing down
and 0.3 respectively as suggested by the search. Because of this, the authors decided to adopt a
Dimopoulos [10] for comparison purposes). technique that prunes the contents of the external
m the total number of machines nondominated set so that its size remains below a certain
‫ݐ‬௜௝ the processing time (hour/piece) of part j on threshold. The approach adopted for this sake was a
machine i clustering technique called average linkage method [13].
ܶ௜ the available time on machine i in a given SPEA II is also a revised version of SPEA whose pseudo
period of time code is shown in Algorithm 2 [14]. SPEA II has three main
ܰ௝ the production requirement of part j in a differences with respect to its predecessor [14]: (1) it
given period of time incorporates a fine-grained fitness assignment strategy
ܹ [ܹ௜௝ ] is an mൈp machine-part incidence which takes into account for each individual the number of
matrix, where ܹ௜௝ is workload on machine i individuals that dominate it and the number of individuals
to which it dominates; (2) it uses a nearest neighbor density
induced by part j and is equal to ሺ‫ݐ‬௜௝ ൈ ܰ௝ ሻȀܶ௜
estimation technique which guides the search more
X [ܺ௜௟ ሿ is an m X c cell membership matrix,
efficiently, and (3) it has an enhanced archive truncation
where ܺ௜௟ ൌ ͳ if ݅ ௧௛ machine is in cell l and 0 method that guarantees the preservation of boundary
otherwise. solutions. The overall algorithm is as follows:
M ݉௟௝ is a c X p matrix of average cell load, Algorithm 1 (SPEA2 Main Loop)
where Input: N (population size)
σ௠ ௜ୀଵ ܺ௜௟ ൈ ܹ௜௝ ഥ (archive size)
݉௟௝ ൌ ܰ
σ௠௜ୀଵ ܺ௜௟ T (maximum number of
According to the above model, we solve the problem by generations)
GA-SPEA II method and present set of non-dominated Output: A (nondominated set)
solutions. The result would be present in section 5. Step 1: Initialization: Generate an initial
population P0 and create the empty
archive
IV. SPEA II TECHNIQUE തതത଴ = Set t = 0.
(external set) ܲ

253
Step 2: Fitness assignment: Calculate fitness
values of individuals in ܲ௧  and ܲഥ௧ .
Step 3: Environmental selection: Copy all
nondominated individuals in ܲ௧ and ܲഥ௧
to ܲത௧ାଵ If size of ܲത௧ାଵ exceeds N then
reduce ത୲ାଵ by means of the truncation
operator, otherwise if size of ത୲ାଵ is less
than N then fill ത୲ାଵ with dominated
individuals in ܲ௧ and ܲഥ௧ .
Step 4: Termination: If t ൒ T or another
stopping criterion is satisfied then set A
to the set of decision vectors represented
by the nondominated individuals in ഥ୲ାଵ .
Stop.
Step 5: Mating selection: Perform binary
tournament selection with replacement
on ത୲ାଵ in order to fill the mating pool.
Step 6: Variation: Apply recombination and
mutation operators to the mating pool
and set ܲത௧ାଵ to the resulting population.
Increment generation counter (t = t + 1)
and go to Step 2.
V. GENETIC ALGORITHM
The general outline of GA is summarized below [11]:
Algorithm 1: Genetic algorithm
Step 1 Generate random population with n
chromosomes by using symbolic
representation scheme (suitable size of
solutions for the problem).
Step2 Evaluate the fitness function of each
chromosome x in the population by
using the proposed objective functions.
Step3 Create a new population by iterating
loop highlighted in the following steps
until the new population is complete
(i) Select two parent chromosomes
from a population according to their
fitness from step2. Those chromosomes
with the better fitness will have chosen.
(ii) With a preset crossover probability,
crossover operation will perform on the
selected parents and to form new
offsprings (children). If no crossover
was performed, offsprings are the exact
copy of parents. Here, Multi-point
crossover is used while partially
matched crossover is employed for
Problem.
(iii) With a preset mutation probability,
mutation will perform on new offspring
254
at each gene. Chosen genes are swapped
to perform mutation process.
(iv) Place new offsprings in the new
population.
Step 4 Deliver the best solution in the current
population. If the end condition is
satisfied, stop.
Step 5 Go to step 2
VI. NUMERICAL EXAMPLE
In order to demonstrate efficiency of our model, authors
use a large scale of test problem that taken from literature
[9] .The authors purposely use this test problem in order to
assess the validity of their approach in a large-sized test
problem, since multi-objective cell-formation problems of
this type do not exist in the literature. In fact, authors in
different models and objective used this test problem. Gupta
et al. [9] In addition, the authors provided a corresponding
set of non-dominated solutions for comparison purposes. As
already indicated in table I, this set of non dominated
solution also, compared with Dimopoulos’s solution.
VII. RESULTS & DISCUSSION
Multi-objective GP-SPE II is an algorithm for
automatically producing sets of non-dominated solutions for
multi-objective CFP. The set of evolved solutions covers
the entire trade-off range of the objectives considered. This
is because it’s using of SPEA II methodology that solved
the multi objective problem and produce the set of non
dominated solution. Also, the set of non-dominated
solutions produced by multi-objective GP-SPEA II provides
the decision maker with a reasonably complete picture of
the potential design trade-offs, which is the aim of a natural
multi-objective optimization Process. The set of solutions
provided by multi-objective GP-SPEA II provides a good
starting point for other decision-maker activities that would
lead to an informed decision.
In the research of Dimopoulos, a Cell Formation Problem
test problem [9] solved by GP-SLCA algorithm. He gets the
set of non dominated solution that consist of 43 solutions.
On the other hand, in this paper, authors get 36 solutions.
By comparing these solutions, we perceive solution number
36 in this paper can dominate the solution numbers 23,…,
43 that get by Dimopoulos’algorithm. In the other solutions
that get by two algorithms, the solution is near to each
other.
VIII. CONCLUSION
This research aim to implementation of SPEA II algorithm
as a state of art method in multi objective problem. So, by
this study authors want to show the efficiency of this
algorithm in CMS environment. As it can be seen in
previous section, authors compare this algorithm with GP- algorithm to comparing and presenting the best algorithm
SLCA that presented by Dimopoulos [10]. Authors suggest for this type of problems.
to researcher to study about NSGA II and another efficient

TABLE I. OBJECTIVE FUNCTION VALUE FOR THE SET OF NON-DOMINATED SOLUTION AND COMPARISON WITH DIMOPOULOS [10] RESULT

DIMOPOULOS GP- DIMOPOULOS GP-

Our GA-SPEA II Our GA-SPEA II
Solution SLCA Solution SLCA
No. Cell–load Total part Cell–load No. Total part Cell–load Total part Cell–load
variation moves variation  moves variation moves variation

1 2.016 25.8 2.016 25.8 23 1.115 83.1 0.763 94.8

2 1.822 28.6 1.822 28.6 24 1.105 83.3 0.741 111.3

3 1.769 34.5 1.665 31.7 25 1.078 84.2 0.739 122.5

4 1.665 35.2 1.661 32.1 26 1.008 84.3 0.738 122.9

5 1.661 35.6 1.634 33.8 27 1.008 85.9 0.635 123.6

6 1.634 38.7 1.613 34.1 28 1.003 89.6 0.621 131.2

7 1.565 43.9 1.565 38 29 0.989 94 0.482 132.5

8 1.509 44.4 1.558 38.8 30 0.989 94 0.417 148.8

9 1.507 45.7 1.48 40.4 31 0.902 98.7 0.378 156.2

10 1.476 47.7 1.472 42 32 0.845 100.8 0.364 162

11 1.472 51.8 1.456 44.6 33 0.843 104.5 0.362 170.8

12 1.373 55.3 1.383 45.5 34 0.842 108.2 0.31 170.9

13 1.346 58.2 1.315 49.9 35 0.168 113.6 0.3 174.8

14 1.315 60.4 1.228 53.2 36 0 113.7 0.28 180

- -
15 1.313 65 1.157 63.6 37 0.224 183.2
- -
16 1.312 66.1 1.136 69.2 38 0.185 186.4
- -
17 1.303 67.6 1.009 70.2 39 0.14 200.7
- -
18 1.28 68 0.997 70.3 40 0.128 208.8
- -
19 1.173 69.4 0.982 79.5 41 0.093 217.1
- -
20 1.159 76.4 0.932 82.8 42 0.051 230.2
- -
21 1.158 76.8 0.892 85 43 0 238.7

22 1.149 76.9 0.885 89.9  

255
 TABLE II. MACHINE-CELL CONFIGURATION FOR THE SOLUTION IN TABLE I
Solution W1 W2 W3 W4 W5 W6 W7 W8 W9 W10 W11 W12 W13 W14 W15
1 C1 C1 C1 C1 C1 C1 C1 C1 C1 C1 C1 C1 C1 C1 C1
2 C2 C1 C2 C2 C2 C2 C2 C2 C2 C2 C2 C2 C2 C2 C2
3 C3 C3 C3 C3 C3 C3 C3 C3 C3 C3 C2 C3 C3 C1 C3
4 C1 C2 C3 C3 C3 C3 C3 C3 C3 C3 C3 C3 C3 C3 C3
5 C3 C1 C3 C3 C3 C3 C3 C3 C3 C3 C2 C3 C3 C3 C3
6 C3 C1 C3 C3 C3 C3 C3 C2 C3 C3 C3 C3 C3 C3 C3
7 C4 C3 C4 C4 C4 C4 C4 C4 C4 C4 C2 C4 C4 C1 C4
8 C1 C2 C3 C4 C4 C4 C4 C4 C4 C4 C4 C4 C4 C4 C4
9 C1 C2 C4 C4 C4 C4 C4 C4 C4 C4 C3 C4 C4 C4 C4
10 C4 C1 C4 C4 C4 C4 C4 C3 C4 C4 C2 C4 C4 C4 C4
11 C4 C1 C2 C4 C4 C4 C4 C3 C4 C4 C4 C4 C4 C4 C4
12 C2 C1 C3 C4 C5 C5 C5 C5 C5 C5 C5 C5 C5 C5 C5
13 C2 C1 C3 C5 C5 C4 C5 C5 C5 C5 C5 C5 C5 C5 C5
14 C1 C2 C3 C5 C5 C5 C5 C4 C5 C5 C5 C5 C5 C5 C5
15 C6 C1 C3 C6 C6 C6 C6 C5 C6 C6 C4 C6 C6 C6 C6
16 C5 C1 C5 C5 C5 C4 C5 C3 C5 C5 C2 C5 C5 C5 C5
17 C5 C1 C2 C5 C5 C3 C5 C4 C5 C5 C5 C5 C5 C5 C5
18 C2 C1 C3 C4 C5 C6 C6 C6 C6 C6 C6 C6 C6 C6 C6
19 C2 C1 C3 C4 C6 C6 C6 C5 C6 C6 C6 C6 C6 C6 C6
20 C1 C2 C3 C6 C6 C6 C6 C4 C6 C6 C5 C6 C6 C6 C6
21 C5 C1 C4 C7 C5 C5 C5 C3 C5 C5 C2 C5 C5 C5 C5
22 C1 C2 C3 C6 C6 C4 C6 C5 C6 C6 C6 C6 C6 C6 C6
23 C2 C1 C3 C4 C5 C6 C7 C7 C7 C7 C7 C7 C7 C7 C7
24 C2 C1 C3 C4 C5 C7 C7 C7 C7 C7 C6 C7 C7 C7 C7
25 C2 C1 C3 C4 C5 C7 C7 C6 C7 C7 C7 C7 C7 C7 C7
26 C1 C2 C3 C4 C7 C7 C7 C6 C7 C7 C5 C7 C7 C7 C7
27 C2 C1 C3 C4 C7 C6 C7 C5 C7 C7 C7 C7 C7 C7 C7
28 C1 C2 C3 C4 C7 C7 C7 C5 C7 C7 C7 C7 C6 C7 C7
29 C7 C1 C5 C3 C7 C6 C7 C4 C7 C7 C2 C7 C7 C7 C7
30 C7 C1 C5 C3 C7 C6 C7 C4 C7 C7 C2 C7 C7 C7 C7
31 C1 C2 C3 C4 C5 C8 C8 C7 C8 C8 C6 C8 C8 C8 C8
32 C1 C2 C3 C4 C8 C7 C8 C6 C8 C8 C5 C8 C8 C8 C8
33 C1 C2 C3 C4 C8 C8 C8 C6 C8 C8 C5 C8 C7 C8 C8
34 C1 C2 C3 C4 C8 C6 C8 C5 C8 C8 C8 C8 C7 C8 C8
35 C1 C2 C3 C4 C5 C7 C8 C7 C8 C8 C6 C8 C8 C8 C8
36 C2 C1 C3 C4 C5 C8 C9 C7 C9 C9 C6 C9 C9 C9 C9

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256
 2009 International Conference on Computer Engineering and Technology

A Formal Mapping between Program Slicing and Z Specifications

Fangjun Wu
School of Information Technology
Jiangxi University of Finance and Economics
NanChang, 330013, China
wufangjun@jxufe.edu.cn

Abstract transforming, model checking, and software security,

This paper represents a research effort towards
the formal mapping between program slicing and Z
specifications. General aspects of program slicing are
considered, i.e. definitions of program slicing, nodes
and edges of program dependence graphs, and
program slicing algorithms. With this approach, we
provide not only a precise semantic basis for program
slicing but also a sound mechanism for reasoning and
verification about program slicing.
1. Introduction
Program slicing, originally introduced by Weiser
M., is an effective technique for narrowing the focus of
attention to the relevant parts of a program during the
debugging process [1]. It is used to extract statements
and predicates from original program P, which might
influence the variables of V at a special program point
p, and make a new program (called by slice), where <p,
V> is called slicing criterion. Thus we can analyze the
original programs P by the slice. The research,
development and applications of program slicing
technique have been carried out for more than twenty
years. Many researchers have done much in program
slicing, obtained lots of theoretical results, and
developed lots of practical applications [2]. Till now,
researchers have proposed forward slicing & backward
slicing, static slicing & dynamic slicing,
intraprocedural slicing [3] & interprocedural slicing [4],
sequence program slicing & current program slicing,
object-oriented program slicing, aspect-oriented
progran slicing, conditioned program slicing, relevant
slicing, union slicing, hybrid slicing, amorphous slicing,
denotational slicing, chopping, dicing, specification
slicing [5,6] and so on. Nowadays, program slicing has
been widely used in program analyzing, understanding,
optimizing, debugging, testing, measuring, reusing,
maintenance, reverse and reengineering, etc.
Although program slicing has been widely studied
in literatures, very little work is involved in its
formalization [7]. Major difficulties in formalizing
program slicing lie in how to describe various
definitions, and how to deal with program slicing
algorithms. To help alleviate this situation, this paper
represents a research effort towards the semantic
formalization for program slicing technique using Z
schema calculus [8-10]. The reasons for choosing the Z
notation are as follows. The specification language Z
includes both a means of specifying data types, based
on sets, and a means of specifying constrains using
predicate logic. The Z notation is used recently not only
in academic domain, but also for industrial
development of high-integrity systems such as safety-
critical software. Z has been used for a number of
digital systems in a variety of ways to improve the
specifications of computer-based systems. Many
textbooks on Z are now available. The teaching of Z has
become of increasing interest. It is thus attractive to
consider it used in formalization of program slicing.
2. A formal mapping between program
slicing and Z specifications
The most related concepts and definitions of
program slicing are known from [1] and [3], the former
due to Weiser M., while the latter due to Ottenstein K.
J., here referred to as Weiser’s slicing and Ottenstein’s,
respectively.
Definition 1 Weiser’s slicing problem is stated as
follows: For a given program point point and a set of
variables V in program P, find a projection Q of P by
deleting zero or more statements that when P and Q
run on the same initial state, the sequences of states
that arise at point point in the two programs have
identical values for all variables in V, where Q is called
by slice, and <point, V> is called slicing criterion.

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DOI 10.1109/ICCET.2009.122
 Definition 2 Ottenstein’s slicing problem is stated
as follows: For a given point point and a set if variables
V in program P, a program slice is a subset of the
statements and control predicates of the program P that
directly or indirectly influence the variables of V at a
special program point point, where <point, V> is called
slicing criterion.
From Definition 1 and Definition 2, we know that
Weiser’s slice is executable, while Ottenstein’s slice
does not necessarily constitute an executable program.
Henceforth, unless explicitly stated, a program slice
implies Ottenstein’s slice.
To slice programs, an original program
OriginalProgram must exist. Generally, programs are
composed of statements and variables. Therefore, free
types Statement and Variable are introduced:
[Statement, Variable].
For Weiser’s slicing problem, if original program
OriginalProgram and slice SliceProgram have the
same initial state, then the sequences of states that arise
at point point in OriginalProgram and SliceProgram
have identical values for all variables in VariableSet.
Thus two free types states of program States and values
of variables Value are introduced: [Value, States].
Weiser’s slicing problem can be described using
schema WeiserProgramSlicing.
For the Ottenstein’s slicing problem, given slicing
criterion <point, VariableSet>, to assure that slices
contain all of statements that might affect values for all
variables in VariableSet, a Boolean type Affect::=Yes |
No is introduced to decide whether the variable is
affected or not, in which Yes means that the variable is
affected, No means not. The Ottenstein’s slicing
problem can be described using schema
OttensteinProgramSlicing.
258
Several formalisms have been used to represent
program dependences, such as: program dependence
graphs (PDG), extended program dependence graphs,
system dependence graphs (SDG), value dependence
graphs, architectural dependence graphs, UML class
diagram dependence graphs and specification
dependence graphs (SpDG). Among them, program
dependence graph is the basic, while others are
extended on it by adding some specifical
characteristics. In following section, we will outline the
features of program dependence graphs. Although
different definitions of program dependence
representations have been given, they share some
common features. Two typical definitions of program
dependence graphs are Ottenstein’s definition[3] and
Horwitz’s[4]. Henceforth, unless explicitly stated,
program dependence graphs imply Horwitz’s.
Definition 3 Ottenstein’s program dependence
graph is composed of control dependence graph (CDG),
data dependence graph (DDG) and control flow graph
(CFG).
Definition 4 Horwitz’s program dependence
graph is a modification of Ottenstein’s, and it is a
directed graph whose nodes are connected by several
kinds of edges, namely control dependence edges, def-
order dependence edges , loop independent edges and
loop carried edges.
Free types are introduced to describe nodes and
edges respectively: [Node, Name, Line, Arc].
A node has a shape. Node_Shapes will normally
be an enumerated type which holds the possible shapes
that nodes can have on a given graph type.
Node_Shapes::=circle | ellipse
The node will also have a type.
Node_Types::=entry | assignment |
controlpredicate | InitialState | FinialUse | others
Many of the characteristics of dependence edges
are similar to those of a node. Similarly, dependence
edge also has shape, and their possible shapes can be
described by the enumerated set Arc_Shapes.
Arc_Shapes::=solid | solid_with_a_hash_mark | dashed For the reason that program dependence graphs
| medium_boldface | heavy_boldface are the basic, we will take them as example to discuss
Similarly, dependence edges have type, too, and in the following part. All kinds of nodes and edges are
their possible types can be described by the depicted respectively.
enumerated set Arc_Types. Firstly, nodes entry are formalized by schema
Arc_Types::=loop_carried_flow_dependence|loop_ind EntryNode.
ependence_flow_dependence | def_order_dependence | Secondly, initial state and final use of variables are
control_dependence formalized by schemas InitialStateNode and
In addition, in order to specify whether an arc is FinalUseNode respectively.
directional or not, we can add arrowhead to it. There
are two possible types of end: plain and arrow.
Ends::=plain | arrow
Although there are a lot of different kinds of
nodes, they have some common features. We define
these common features as a generic node, represented
by state schema GenericNodes. Declaration part
contains name, shape, type of nodes.

Similarly, different kinds of edges also have some

common features. We define these common features as
a generic arc, represented by state schema GenericArcs.
Declaration part contains name, shape, type, direction
of nodes, the source and target of dependence edges,
whether an arrowhead is included or not, etc.

Similarly, nodes assignment and nodes

controlpredicate are formalized by schemas
AssignmentNode and ControlPredicateNode
respectively.

259

After formalized various kinds of nodes, we are
going to formalize control dependence edges, which
are formalized by schema ControlDependenceArc.
On the basis of formalization of nodes and edges,
we will formalize program dependence graphs by
schema PDG.
So far, two methods of computing program slices
have been developed. One is the Weiser’s method,
which iteratively solves dataflow equations derived
from inter-statement influence[1]. The other lays its
base on graph reachability algorithms[4]. Many
researchers are doing work along the graph reachability
260
algorithms line, on which our work in this paper is also
based. This operation can still be performed in time
linear in the size of the slice by performing a single
traversal of the PDG starting from the set of points,
which can be formalized by schema SlicingPDG.
3. Conclusions
This paper represents a research effort towards the
semantic formalization for program slicing technique
using Z schema calculus. All schemas in this paper are
checked using the ZTC type-checker package [11] and
Z User Studio[12]. This formalization could be helpful
in correct understanding of different types of slicing
and also the correct application of a desired slicing
regime in a rigorous way. Furthermore, there are a
number of existing tools on the market which do
manipulate Z specifications (for example, to perform
type-checking). Hence inference and verification of
program slicing become possible.
Acknowledgements
This research has been supported by the Natural
Science Foundation of Jiangxi (Project No.
2007GQS0495) and Science and Technology
Foundation of the Education Department of Jiangxi
(Project No. GJJ08353 and [2007] 434).
References
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Software Engineering, vol.16, no.5, pp. 498-509, 1984.
[2] Tip F, “A survey of program slicing techniques”,
Journal of Programming Languages, vol. 3, no. 3, pp.
121-189, 1995.
[3] Ferrante J., Ottenstein K. J., Warren J. D., “The program
dependence graph and its use in optimization”, ACM
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vol. 9, no. 3, pp. 319-349,1987.
[4] Horwitz S., Reps T., Binkley D., “Interprocedural
slicing using dependency graphs”, ACM Transactions
on Programming Languages and Systems, vol. 12, no. 1,
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[5] Wu Fangjun, Yi Tong, “Slicing Z specifications”, ACM
SIGPLAN Notices, vol. 39, no. 8, pp. 39-48,2004.
[6] Wu Fangjun, Research on Z formal specifications
slicing. Jiangxi nanchang: Jiangxi university of finance
& economics, 2006.
[7] Danicic S, Binkley D, Gyimothy T, Harman M,
KissAkos O L., “Formalizing executable dynamic and
forward slicing”, Proceedings of 4th Workshop on
Source Code Analysis and Manipulation (SCAM 2004),
pp.43-52, September 2004.
[8] Spivey J. M., The Z notation: a reference manual:
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[9] International Standards Organization, Z formal
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andards/
[10] Miao Huaikou, Li Gan, Zhu Guanming, Software
engineering languageZ. Shanghai: Shanghai
Science and Technology Information Publishing House,
1999.
[11] Jia X, ZTC: A Type Checker for Z Notation. User’s
Guide. Version 2.03, August 1998. Division of Software
Engineering, School of Computer Science,
Telecommunication, and Information Systems, DePaul
University, USA, 1998.
[12] Miao Huaikou, Liu Ling, Yu Chuanjiang, Ming Jijun,
Lili, “Z User Studio: an integrated support tool for Z
specifications”, APSEC 2001, pp. 437-444, December
2001.
 2009 International Conference on Computer Engineering and Technology
Modified Class-Incremental Generalized Discriminant Analysis
Yunhui He
Department of Communications Engineering
Nanjing University of Information Science and Technology
Nanjing, China, 210044
yunhuihe@163.com
Abstract—In this paper, we propose an efficient method for
resolving the optimal discriminant vectors of Generalized
Discriminant Analysis (GDA) and point out the drawback of
high computational complexity in the traditional class-
incremental GDA [W. Zheng, “Class-Incremental Generalized
Discriminant Analysis”, Neural Computation 18, 979–1006
(2006)]. Because there is no need to compute the mean of
classes and the mean of total samples in the proposed method
as needed in the traditional class-incremental GDA, the
computational complexity is reduced greatly. The theoretical
justifications of the proposed batch GDA and the class-
incremental GDA are presented in this paper.
Keywords- class-incremental generalized discriminant
analysis; orthogonalization; difference space; kernel method
I. INTRODUCTION
The classical linear discriminant analysis (LDA) [1] was
nonlinearly extended to the Generalized discriminant
analysis (GDA) [2] by mapping the samples from input
space to a high-dimensional feature space via the kernel trick
[3,4]. However, the LDA often encounters the small sample
size (SSS) problem [5] when the dimensionality of samples
is greater than the number of samples. The GDA methods
also suffers from this problem since the dimensionality of
samples in feature space is much greater than the number of
the samples. A numerous methods have been proposed to
deal with the SSS problem of LDA [6-8] for resolving the
optimal discriminant vectors in the range space of total
scatter matrix and the null space of within-class scatter
matrix. For solving GDA, the traditional approach is the use
of the singular value decomposition (SVD) [2]. However, the
SVD-based GDA algorithms suffer from the numerical
instability problem due to the numerical perturbation [9].
Recently, Zheng [10] proposed a numerically stable
algorithm for batch GDA and class-incremental GDA in the
case of the small sample size problem by applying only QR
decomposition in feature space. However, the algorithms in
[10] are not optimal in terms of computational complexity
because the mean of classes and the mean of samples must
be computed before the QR decomposition is applied. To
overcome this drawback, in this paper, we propose an
efficient method to resolving the optimal discriminant
vectors of the batch GDA and class-incremental GDA
respectively, by directly performing the Gram-Schmidt (GS)
orthogonalization procedure [11] in the difference space
(DS) [12] using the kernel trick. We call the proposed
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 262
DOI 10.1109/ICCET.2009.28
method GS-DS GDA as an abbreviation. Because there is no
need to compute the mean of classes and the mean of total
samples in GS-DS GDA as needed in the class-incremental
GDA, the computational complexity is thus reduced greatly.
In the next section, the Gram-Schmidt orthogonalization
in feature space is introduced in Section 2. The batch GS-DS
GDA algorithm and class-increment algorithms are proposed
and the theoretical justification is proofed in Section 3 and 4
respectively. The conclusions are drawn in Section 5.
II. GRAM-SCHMIDT ORTHOGONALIZATION IN FEATURE
SPACE
In kernel method, the samples are transformed into an
implicit higher dimensional feature space F though a
nonlinear mapping Φ(x) . In implementation, the implicit
features vector in F dose not need to be computed
explicitly, and it is just done by computing the inner product
of two vectors in F with a kernel
function k ( x, y ) = Φ ( x ) T Φ ( y ) [3,4]. Let the sample matrix
be X = [ x1 ,..., xn ] which becomes
X Φ = [Φ( x1 ),..., Φ( x n )] in feature space.
By using QR decomposition, the matrix X Φ can be
expressed as
X Φ = QR (1)
where Q is column orthogonal matrix and R is an upper
triangular matrix
Because the columns of matrix Q are orthonormal, we
have
K = X ΦT X Φ = R T Q T QR = R T R (2)
where K is a n × n kernel matrix which can be computed
using kernel function as ( K ) ij = k ( xi , x j )
K is symmetric positive semi-definite matrix. From
(2), the triangular matrix R can be obtained by performing
Cholesky decomposition of K [11].
If samples are linearly independent, then upper
triangular matrix R is positive definite and we obtain
X Φ R −1 = Q (3)
From above analysis, we can see that performing the
Gram-Schmidt orthogonalization in feature space is actually
equivalent to performing a Cholesky decomposition of the of covariance matrix S i of class i . This relation can be
kernel matrix K .
further expressed as
III. THE BATCH GS-DS GDA ALGORITHM R(SiΦ ) = span{Φ(b1i ),...,Φ(bnii −1 )} i = 1,...,C (10)
In the traditional class-incremental GDA algorithm, the Let all samples be
column vectors (Eq. (4.9) in [10]) used for solving GDA are
not optimal, because the mean of classes and the mean of {Φ(x11),...,Φ(x1n1 ),Φ(x12 ),...,Φ(xn22 ),...,Φ(x1C ),...,Φ(xnCC )} .
total samples must be computed, which thus leads to a great The n −1difference vectors are computed as
increase of computational complexity. To overcome this
Φ(d l1 ) = Φ( xl1+1 ) − Φ( x11 ) for class 1,
drawback, in this section, we propose the optimal vectors on
which the orthogonalization is performed for resolving the where l = 1,2,..., n1 − 1
optimal discriminant vectors of the batch GDA. Φ (d kii ) = Φ ( xki i ) − Φ ( x11 ) for class i = 2,..., C , where
Let the training set has C classes, where each class i has
ki = 1,2,..., ni
ni samples and the number of total samples is n = ∑i=1 ni .
C (11)
Likewise, these n −1 difference vectors span the
Let Φ ( xki ) be kth sample in ith class. μiΦ and μ Φ are the difference space which equals the space of all eigenvectors
mean of classes and the mean of total samples respectively. corresponding to the nonzero eigenvalues of covariance
All samples are independent. The between class scatter matrix S tΦ of total samples. From this relation we obtain
matrix S bΦ , within-class scatter matrix S wΦ and total scatter
R(StΦ ) = span{Φ(d11 ),...,Φ(dn11 −1 ), Φ(d12 ),
matrix S tΦ are defined as: (12)
C ...,Φ(dn22 ),...,Φ(d1C ),...,Φ(dnCC )}
S bΦ = ∑ ni ( μ iΦ − μ Φ )( μ iΦ − μ Φ ) T (4)
i =1 By comparing the definitions in (11) and (9), we obtain
C ni C
Φ(d l1 ) = Φ( xl1+1 ) − Φ( x11 ) = Φ(bl1 ) for class 1 where
SwΦ = ∑∑(Φ( xmi ) − μiΦ )(Φ( xmi ) − μiΦ )T = ∑ SiΦ (5)
i=1 m=1 i=1 l = 1,2,..., n1 − 1
C ni
StΦ = ∑∑(Φ(xmi ) − μΦ )(Φ(xmi ) − μΦ )T = SbΦ + SwΦ (6) Φ(dkii ) = Φ( xki i ) − Φ( x11) = (Φ( xki i ) − Φ(x1i )) +
i=1 m=1
Ni ni
(Φ(x1i ) − Φ(x11)) = Φ(bkii −1) + Φ(d1i )
where SiΦ = ∑∑(Φ( xmi ) − μiΦ )(Φ( xmi ) − μiΦ )T is the for class i = 2,..., C where ki = 2,..., ni (13)
m=1 m=1
From (8) and (10), we obtain
covariance matrix of class i .
C
The ranks of S bΦ , S wΦ and S tΦ are C −1, n− C and n −1 R(SwΦ ) = ∪ R(SiΦ ) = {Φ(b11),...,Φ(bn11−1),
i =1 (14)
respectively for independent samples.
Because S wΦ and S tΦ are positive semidefinite matrices, Φ(b ),...,Φ(b
2
1
2
n2 −1 ),...,Φ(b ),...,Φ(b
C
1
C
nC −1 )}
from Eq.(5), we have Now we rearrange all difference vectors
C Φ(d ) ( i =1,...,C, ki = 1,2,...,ni ) in sequence as follows
i
ki
N ( S wΦ ) = ∩ N ( S iΦ ) (7)
i =1 { Φ ( d 11 ),..., Φ ( d n11 −1 ), Φ ( d 22 ) − Φ ( d 12 ),...,
which leads to
C C Φ ( d n22 ) − Φ ( d 12 ),..., Φ ( d 2C ) − Φ ( d 1C ),..., (15)
Φ Φ ⊥ Φ ⊥ Φ
R(S ) = N (S ) = (∩ N (S )) = ∪ R(S ) (8)
w w
i =1
i
i =1
i
Φ ( d nCC ) − Φ ( d 1C ), Φ ( d 12 ),..., Φ ( d 1C )}
From the definition of difference space [12], the following From (13), it can be seen that the vectors in (15) equal in
ni − 1 difference vectors span the difference space of class sequence
i in feature space: {Φ(b11 ),...,Φ(bn11−1 ),Φ(b12 ),...,Φ(bn22 −1 ),...,Φ(b1C ),
Φ ( bkii ) = Φ ( xki i +1 ) − Φ ( x1i ) ( ki = 1,..., ni − 1) (9) (16)
...,Φ(bnCC −1 ),Φ(d12 ),...,Φ(d1C )}
From the relation in [12], these ni − 1 difference vectors From (14) and (16), it can be seen that the leftmost n − C
span the difference space of class i which equals the space Φ
vectors in (12) span the R( S w ) .
of all eigenvectors corresponding to the nonzero eigenvalues
Since from (12) all n −1 vectors in (15) span the
R( StΦ ) and leftmost n − C vectors span R( S wΦ ) , by

263
performing Gram-Schmidt orthogonalization procedure on
⎡0 0 0⎤
vectors in (15), the last C −1 orthonormal vectors lie in ⎢1 0 0 ⎥⎥ n1 th row
R ( S tΦ ) ∩ N ( S wΦ ) which are exactly the optimal ⎢
discriminative vectors searched by GDA/KGS method in ⎢ 1 ⎥ ( n1 + n 2 ) th row
BC +1 ( n −1)× ( C −1) =⎢ ⎥
[10]. ⎢ 0 ⎥ C −1
To summarize above derivation, the n −1 difference ⎢ 1 ⎥ ∑ n k th row
vectors in (12) can be represented as follows ⎢ ⎥ k =1
[ Φ ( d 11 ),..., Φ ( d n11 −1 ),..., Φ ( d 12 ),..., Φ ( d n22 ),..., ⎢⎣ 0 0 0 ⎥⎦
The other elements of the matrix B are zeros.
Φ ( d 1C ),..., Φ ( d nCC )]
(17) DΦ = [Φ(d11 ),...,Φ(d n11 −1 ), Φ(d 22 ) − Φ(d12 ),...,
= [ Φ ( x ),..., Φ ( x ), Φ ( x ),..., Φ ( x ),...,
1 1 2 2
1 n1 1 n2
Φ(d n22 ) − Φ(d12 ),...,Φ(d 2C ) − Φ(d1C ),...,
Φ ( x ),..., Φ ( x )] A
C C

Φ(d nCC ) − Φ(d1C ), Φ(d12 ),...,Φ(d1C )]

1 nC (19)
where
⎡− 1 − 1 − 1⎤ = [Φ( x11 ),...,Φ( x1n1 ), Φ( x12 ),...,Φ( xn22 ),...,
⎢1 0 0 ⎥⎥
⎢ Φ( x1C ),...,Φ( xnCC )]AB = X Φ AB
⎢ 0 1 ⎥
A n × ( n −1 ) = ⎢ ⎥ To perform the orthogonalization in feature space on the
⎢ 0 ⎥ column vectors of matrix DΦ in (19), we need to compute
⎢ 0 ⎥
⎢ ⎥ the kernel matrix
⎣⎢ 0 0 1 ⎦⎥
The vectors in (15) can be represented as follows
KD = DΦT DΦ = BT AT X ΦT X Φ AB = BT AT KAB. (20)
where K = X Φ X Φ
T
[Φ(d11 ),...,Φ(dn11−1 ),Φ(d22 ) − Φ(d12 ),...,
Then performing Cholesky decomposition of K D
Φ(dn22 ) − Φ(d12 ),...,Φ(d2C ) − Φ(d1C ),...,
K D = RDT RD (21)
Φ(dnCC ) − Φ(d1C ),Φ(d12 ),...,Φ(d1C )] (18) where R D is the triangular matrix.
=[Φ(d11 ),...,Φ(dn11−1 ),...,Φ(d12 ),..., The orthonormal vectors are
Q D = [q1Φ , q 2Φ ,..., q nΦ−C +1 ,..., q nΦ−1 ] = DΦ R D−1 (22)
Φ(d ),...,Φ(d ),...,Φ(d )]B
2
n2
C
1
C
nC For a given sample x , the nonlinear optimal
where discriminative feature is computed as follows
B( n−1)×( n−1) = [ B1 , B2 ,..., BC , BC +1 ] yΦ = QDT Φ( x) = ( RD−1 )T DΦT Φ( x) = ( RD−1 )T K x (23)
⎡1 0 0⎤ where the n-1 by 1 column vector K x = DΦT Φ(x) is
⎢0
⎢ 1 0 ⎥⎥ computed using kernel function,
B1 ( n −1)× ( n1 −1) =⎢ ⎥ K x = DΦT Φ( x) = BT AT X ΦT Φ ( x)
⎢ ⎥
⎢ 1 ⎥ ( n1 - 1)th row = B T AT [k ( x11 , x),..., k ( x1n1 , x), k ( x12 , x),..., (24)
⎢⎣ 0 0 0 0 ⎥⎦
m −1
k ( xn22 , x),..., k ( x1C , x),..., k ( xnCC , x)]T
⎡− 1 − 1 − 1⎤ ∑ n k th row Then the last C-1 elements of vector yΦ is the
⎢1 0 0 ⎥⎥ k =1
⎢ nonlinear optimal discriminative feature of sample x
⎢0 1 ⎥ y opt = [ y Φ n − C +1 ,..., y Φ n −1 ]T
Bm ( n −1)×( nC −1) =⎢ ⎥
0 1⎥ The batch GS-DS GDA is summarized as follows
⎢ (1) Compute the matrix A, B in (17), (18) and kernel
⎢ ⎥
⎢ ⎥ matrix K using kernel function in (20).
⎣⎢ 0 0 0 ⎦⎥ (2) Compute the kernel matrix KD in (20).
(3) Performing Cholesky decomposition of KD in (21) to
where m=2,3,…,C obtain the upper triangular matrix RD.
(4) Compute the nonlinear optimal discriminative

264
 feature using (23) and (24). C
It can be seen from above algorithm that there is no need ⎡− 1 − 1 − 1⎤ ∑ nk th row
to compute the mean of classes and the mean of total ⎢1 0 0 ⎥⎥ k =1
samples as needed in the GDA/KGS in [10]. The ⎢
computational cost of the GS-DS GDA is thus saved greatly.
~ ⎢0 1 ⎥
IV. CLASS-INCREMENTAL GS-DS GDA
BC +1( n−1)×( nC +1 −1) =⎢ ⎥
⎢ 0 ⎥
In this section, we propose the class-incremental GS-DS ⎢ 0⎥
GDA algorithm in cases where a new class is inserted into
⎢ ⎥
the data set. The class-incremental GDA proposed in [10] is ⎢⎣ 0 0 0 ⎥⎦
time-consuming because the mean of total mean and the
mean of new class must be computed when a new class is C +1
dynamically added into the data set. However, in the class- where n = ∑k =1
nk
incremental GS-DS GDA, this drawback is overcome.
C +1
Let xi be i th samples in the new class C+1. The ⎡0 0 0⎤
⎢1 0 ⎥⎥ n1 th row
∑
C +1 0
number of total samples becomes n = ni . ⎢
⎥ ( n1 + n 2 ) th row
i =1
⎢ 1
( n −1 )× C = ⎢
The Eq.(17) becomes BC +2 ⎥
[Φ (d11 ),..., Φ (d n11 −1 ),..., Φ(d12 ),..., Φ(d n22 ),..., ⎢ 0 ⎥ C
⎢ 1 ⎥ ∑ n k th row
Φ(d1C ),..., Φ(d nCC ), Φ (d1C +1 ),..., Φ (d nCC++11 )] ⎢ ⎥ k =1
(25) ⎢⎣ 0 0 0 ⎥⎦
= [Φ ( x ),..., Φ ( x ), Φ( x ),..., Φ( x ),...,
1
1
1
n1
2
1
2
n2 The other elements of the matrix B are zeros.
~ The difference vectors in Eq.(19) become
Φ( x1C ),..., Φ ( xnCC ), Φ( x1C +1 ),..., Φ( xnCC++11 )] A ~
DΦ = [Φ(d11 ),...,Φ(dn11−1 ),Φ(d22 ) − Φ(d12 ),...,
⎡− 1 − 1 − 1⎤
Φ(dn22 ) − Φ(d12 ),...,Φ(d2C ) − Φ(d1C ),...,
⎢1 0
⎢ 0 ⎥⎥
n = ∑i =1 ni Φ(dnCC ) − Φ(d1C ),Φ(d2C+1 ) − Φ(d1C+1 ),...,
C +1
where ~ ⎢0 1 ⎥
An×( n −1) =⎢ ⎥ (27)
⎢ 0 ⎥ Φ(dnCC++11 ) − Φ(d1C+1 )， Φ(d12 ),...,Φ(d1C )， Φ(d1C+1 )]
⎢ 0⎥
⎢ ⎥ = [Φ(x11 ),...,Φ(x1n1 ),Φ(x12 ),...,Φ(xn22 ),...,
⎢⎣ 0 0 1 ⎥⎦
~~ ~ ~~
The Eq.(18) becomes Φ(x1C ),...,Φ(xnCC ),Φ(x1C+1 ),...,Φ(xnCC+1 )]AB = X Φ AB
[Φ(d11),...,Φ(dn11−1 ),Φ(d22 ) − Φ(d12 ),..., We obtain the kernel matrix K D~
Φ(d ) − Φ(d ),...,Φ(d ) − Φ(d ),...,
2 2 C C ~ ~ ~ ~ ~ ~ ~ ~ ~ ~~ ~
n2 1 2 1 KD~ = DΦT DΦ = BT AT XΦT XΦ AB = BT AT KAB (28)
C+1 C+1
Φ(d ) − Φ(d ),Φ(d ) − Φ(d
C
nC
C
1 2 1 ),..., where
~ ~ ~
C+1 C+1 C+1
(26) K = X ΦT X Φ (29)
Φ(d nC+1 ) − Φ(d 1 ),Φ(d ),...,Φ(d ),Φ(d )]
2
1
C
1 1
Then performing Cholesky decomposition of K D~
=[Φ(d11 ),...,Φ(dn11−1 ),...,Φ(d12 ),...,Φ(dn22 ),...,
K ~
D
= R T
~
D
R ~
D
(30)
~
Φ(d1C ),...,Φ(dnCC ),Φ(d1C+1 ),...,Φ(dnCC++11 )]B where RD~ is the triangular matrix.
where The orthonormal optimal discriminant vectors of class-
~ ~
B( n −1)×( n −1) = [ B1 , B2 ,..., BC , BC +1, BC + 2 ] incremental GS-DS GDA are
~
where B1 , B2 ,..., BC are the same as in (18). QD~ =[q~1Φ , q~2Φ ,...,q~nΦ−C ,...,q~nΦ−1 ] = DΦ RD−~1 (31)
~ The optimal discriminative feature of GDA can be
The BC +1 is a ( n − 1) × ( n C +1 − 1) matrix and BC + 2 is a
obtained as in batch GS-DS GDA using the Q D~ .
( n − 1) × C matrix as follows
We can see that the kernel matrix K in (20) can be
~
reused in the kernel matrix K in (29), and the matrices

265
 ~ ~
A and B can be reused in A and B in the case of class-
increment. Thus, the computational cost for solving the
class-increment GDA is saved greatly.
The class-incremental GS-DS GDA is summarized as
follows
(1) Compute the matrices A~ , B~ in (25), (26)and kernel
~ 10, 1299–1319.
matrix K using kernel function in (29).
(2) Compute the kernel matrix K ~
D in (28).
(3) Performing Cholesky decomposition of K ~
D
in (30)
to obtain the upper triangular matrix R D~ .
Compute the nonlinear optimal discriminative feature
using the last C column vectors of QD~ .
V. CONCLUSIONS
We have proposed an efficient method to resolving the
optimal discriminant vectors of the batch and class-
incremental GDA for the small sample size problem, by
performing the orthogonalization procedure in feature space
only once on difference vectors. The theoretical justifications
are presented. Since there is no need to compute the class
means and total mean of sample as needed in the traditional
class-incremental GDA, the proposed algorithms are more
efficient. In the class-incremental GS-DS GDA, the most
computational results can be reused when a new class is
dynamically inserted into the data set. Thus, the
computational complexity is reduced greatly compared with
the traditional class-incremental GDA.
266
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New York: Wiley, 1973.
[2] Baudat, G., & Anouar, F.. Generalized discriminant analysis using a
kernel approach. Neural Computation, 2000, 12, 2385–2404.
[3] Sch¨olkopf, B., Smola, A., & M¨ uller, K. R.. Nonlinear component
analysis as a kernel eigenvalue problem. Neural Computation, 1998,
[4] S.T. John and C. Nello, Kernel Methods for Pattern Analysis,
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[5] P.N Belhumeur, J.P. Hespanha, and D.J. Kriegman, “Eigenfaces vs.
Fisherfaces: Recognition Using Class Specific Linear Projection,”
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662.
 2009 International Conference on Computer Engineering and Technology
Controlling Free Riders in Peer to Peer Networks by
Intelligent Mining
Ganesh Kumar. M Arun Ram. K
Computer Science and Computer Science and
Engineering Engineering
Sri Venkateswara College of Sri Venkateswara College of
Engineering, Sriperumbudur Engineering, Sriperumbudur
Chennai, India Chennai, India
mganeshmail@gmail.com arun@websrs.org
Abstract—In a typical Peer to Peer system, each node is owned
and operated by an independent entity, and the nodes
collectively form a self-organizing, self-maintaining network
with no central authority. The peer to peer Technology has
attracted a significant amount of interest because of its
capacity for resource sharing and content distribution. The
existence of high degree of free riding is a serious threat to P2P
network. This paper deals with the issue of identifying free
riders and other kind of peers who subvert the system.
Additionally, this paper also proposes a basic data warehouse
structure for a peer to peer system and analyzes peer behavior
using data mining functionalities. Results prove that the free
riders and malicious peers get automatically eliminated from
the system.
Key Words- Data mining; peer to peer network; Free Riders;
Malicious peers; Reputation
I. INTRODUCTION
The advent of peer-to-peer (P2P) file sharing systems
heralds a new era in the field of Internet technology. While
these systems alleviate the scalability problem that has
dogged the client-server model, they present new data
management problems. It is widely believed that the success
of P2P file sharing systems depends upon the quality of
service offered by such systems. Accordingly most of the
present research in P2P systems has been concentrated on
issues such as efficient data placement, fast file lookup, data
replication etc.
We argue that, in addition to the quality of service,
there is another key aspect that impacts the success and
continued sustenance of P2P systems. It is the quality of the
data present in the system. For a file sharing system, no
matter how excellent the lookup capabilities of a system are,
or what file download speeds it offers, if the system does
not have a large and growing number of interesting files, it
will eventually fail to attract or retain users. Unfortunately,
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 267
DOI 10.1109/ICCET.2009.109
Ananya A.R
Computer Science and
Engineering
Sri Venkateswara College of
Engineering, Sriperumbudur
Chennai, India
ar_ananya@yahoo.co.in
research on developing mechanisms to maintain or enhance
the quality of data is yet to receive much attention from the
P2P research community.
This problem is exemplified by the phenomenon of
free riding in many P2P file sharing systems. A recent study
on Gnutella file sharing system shows that as many as 70%
of its users don't share any files at all. This means that these
users use the system for free. This behavior of an individual
user who uses the system resources without contributing
anything to the system is the first form of the Free Riding
problem. Such users are referred to as free riders. The study
further indicates that not all file sharers share popular and
desirable files. It shows that as many as 63% of the peers,
who shared some files, never answered a single query. This
implies that these file sharers did not share any desirable
files. This is a second form of the Free Riding problem,
wherein users share some files that are not useful.
The free riding problem affects the system in two
significant ways. First, the number of files in the system
becomes limited or grows very slowly. The number of
popular files may become even smaller as the time goes by.
This adversely affects user's interest in the system and they
eventually pull out of the system. When users who share
popular files pull out of the system, the system becomes
poorer in terms of the amount of files shared. This is a
unproductive cycle and it may eventually lead to the
collapse of the system. Second, if only a few peers share
popular files, all the downloading requests are directed
towards those peers. This causes those peers to become hot
spots, overloading their machines and causing congestion on
their network. Peers frequently experiencing CPU overloads
or network congestion due to the P2P system may exit the
system if it affects their other routine activities.
In order to maintain the productivity and ensure the
healthiness of a P2P file sharing system, there is a need for
mechanisms that can help in securing cooperation from its
users by encouraging them to share popular files.
Surprisingly, none of the existing P2P file sharing systems,
to our knowledge, offer or incorporate mechanisms that
effectively encourage their users to share files of interest
with other users in the system.
In this paper, we explore a new approach that
integrates Data Mining with Peer to peer system which
attempts to discover and extract new knowledge from the
recorded data and information. This data is normally stored
in databases, and can be of different nature such as peer id
and reputation of the peer. The knowledge learned is
represented in forms of rules, such as classification rules,
prediction rules, association rules or clusters of rules. These
results can be often used for identifying the peer behavior.
II. RELATED WORK
[1] explains the details about various problems in
peer to peer networks and how they can be solved using
reputation concepts. [4] Mainly concentrates on the various
issues concerning free riding and gives us the formulas to
identify the free riders based on the popularity, size and
number of files shared by a peer. [2] and [3] give a broad
outlook of a distributed way of identifying and isolating the
free riders in the peer to peer system.
III. DATA WAREHOUSE FOR PEER TO PEER
SYSTEMS
Fig.1 Data cube for a Peer to peer network
A data warehouse is a repository of information
collected from multiple sources, stored under a unified
schema, and which usually resides at a single site. Data
warehouses are constructed via a process of data cleansing,
data transformation, data integration, data loading, and
268
periodic data refreshing. A data warehouse is usually
modeled by a multidimensional database structure, where
each dimension corresponds to an attribute or a set of
attributes in the schema, and each cell stores the value of
some aggregate measure, such as count or sales amount. The
actual physical structure of a data warehouse may be a
relational data store or a multidimensional data cube. It
provides a multidimensional view of data and allows the
pre-computation and fast accessing of summarized data.
Fig.1 gives a basic description of how a data cube for a peer
in a P2P network look like. The peers with whom the peer i
has interacted is given in X-axis, the reputation gained is
given in Y-axis, time is given in Z-axis.
A. Data warehousing Operations
1. Roll-up: The roll-up operation (drill-up) performs
aggregation on a data cube, either by climbing-up a concept
hierarchy for a dimension or by dimension reduction
2. drill-down: Drill-down is the reverse of roll-up. It
navigates from less detailed data to more detailed data.
Drill-down can be realized by either stepping-down a
concept hierarchy for a dimension or introducing additional
dimensions.
3. Slice and Dice: The slice operation performs a selection
on one or more dimension of the given cube, resulting in a
sub cube.
4. Pivot (rotate): Pivot is a visualization operation which
rotates the data axes in view in order to provide an
alternative presentation of the data.
B. Super Marts
The main highlight of Peer to Peer Computing is its
distributiveness. The properties of data marts and
distributiveness can be easily cross produced to evolve a
new concept of Super Marts. In Kazaa, all peers are
clustered into groups and a Super peer is assigned to every
group. The main role of the super peer is to monitor the
activities taking place in the particular group. We propose a
new concept of using data marts for these super peers so that
the activities of the group can be effectively managed. The
super marts are directly controlled by a central Data
Warehouse called Super Warehouse. The main task of Super
Warehouse is to periodically update the information about
the peer’s activities. The interaction between the super peer
(with respect to KAZAA network), Super Mart and Super
Warehouse enables the system to identify and eliminate the
peers exhibiting malicious behavior. [7]At the same time
good peers who effectively increase the life time of the
system are awarded with certain incentives.

Fig. 2. Super mart and Super warehouse
IV. DATA MINING IN P2P
The kinds of patterns that can be discovered
depend upon the data mining tasks employed. By and large,
there are two types of data mining tasks: descriptive data
mining tasks that describe the general properties of the
existing data, and predictive data mining tasks that attempt
to do predictions based on inference on available data. The
data mining functionalities and the variety of knowledge
they discover are briefly presented in the following list:
A. Characterization
Data characterization is a summarization of general
features of objects in a target class, and produces what is
called characteristic rules. The data relevant to a user-
specified class are normally retrieved by a database query
and run through a summarization module to extract the
essence of the data at different levels of abstractions. For
example, one may want to characterize the peers who play
with him or the strategies that are used by a peer regularly.
Free riders and malicious peer can be easily found using
characterization. With concept hierarchies on the attributes
describing the target class, the attribute-oriented induction
method can be used, for example, to carry out data
summarization. Note that with a data cube containing
summarization of data, simple OLAP operations fit the
purpose of data characterization.
B. Discrimination
Data discrimination produces what are called
discriminant rules and is basically the comparison of the
general features of objects between two classes referred to
as the target class and the contrasting class. For example,
one may want to compare the general characteristics of the
peers who have cheated more than 30 times with those
269
whose cheated lesser than 5 times. The techniques used for
data discrimination are very similar to the techniques used
for data characterization with the exception that data
discrimination results include comparative measures.
C. Association analysis
Association analysis is the discovery of what are
commonly called association rules. It studies the frequency
of items occurring together in transactional databases, and
based on a threshold called support, identifies the frequent
item sets. Another threshold, confidence, which is the
conditional probability than an item appears in a transaction
when another item appears, is used to pinpoint association
rules. For example, a peer is interested to know what move
the peer is going to take if he is going to download a file.
For this the peer need to analyze the opponent peers history.
With all these information the peer can easily find the next
move. The discovered association rules are of the form: P -
> Q [s,c], where P and Q are conjunctions of attribute value-
pairs, and s (for support) is the probability that P and Q
appear together in a transaction and c (for confidence) is the
conditional probability that Q appears in a transaction when
P is present.
D. Classification
Classification analysis is the organization of data in
given classes. Also known as supervised classification, the
classification uses given class labels to order the objects in
the data collection. Classification approaches normally use a
training set where all objects are already associated with
known class labels. The classification algorithm learns from
the training set and builds a model. The model is used to
classify new objects. For example, we might want to
classify our agents into 'Good' or 'Malicious' or ‘Free riding’
categories with regard to their reputation. The category or
'class' into which each peer is placed is the 'outcome' of
classification process.
Case based reasoning is an apt classifier for peer to peer
network, which uses the previous history for the process of
classification. This method is more efficient because of the
possibility of unsupervised classification. To solve a current
classification problem (to find a free rider), the problem is
matched against the cases in the case base, and similar cases
are retrieved. The retrieved cases are used to suggest a
solution which is reused and tested for success. If necessary,
the solution is then revised. Finally the current problem and
the final solution are retained as part of a new case.
E. Prediction
Prediction has attracted considerable attention
given the potential implications of successful forecasting in
a business context. There are two major types of
predictions: one can either try to predict some unavailable
data values or pending trends, or predict a class label for
some data. The latter is tied to classification. Once a
classification model is built based on a training set, the class
label of an object can be foreseen based on the attribute
values of the object and the attribute values of the classes.
Prediction is however more often referred to the forecast of
missing numerical values, or increase/ decrease trends in
time related data. The major idea is to use a large number
of past values to consider probable future values. For
example if the peer is interested in predicting the chance of
downloading without any interruption, he might simulate
the whole transaction exactly with the history of the
opponent. [5] Has effectively used the prediction technique
to identify a winning strategy using data mining. We can use
the same [6] Game theoretic technique to identify the
malicious peer using data mining and Nash equilibrium
concept.
F. Clustering
Similar to classification, clustering is the
organization of data in classes. However, unlike
classification, in clustering, class labels are unknown and it
is up to the clustering algorithm to discover acceptable
classes. Clustering is also called unsupervised classification,
because the classification is not dictated by given class
labels. There are many clustering approaches all based on
the principle of maximizing the similarity between objects
in a same class (intra-class similarity) and minimizing the
similarity between objects of different classes (inter-class
similarity). For example if system is interested in finding the
type of peer available into malicious and free rider, the
system can use the history of each peer to form a cluster.
Then any clustering algorithm can be used to cluster the
peer types fig .3 shows an example cluster for peer type.
G. Outlier analysis
Outliers are data elements that cannot be grouped in a given
class or cluster. Also known as exceptions or surprises, they
are often very important to identify. While outliers can be
considered noise and discarded in some applications, they
can reveal important knowledge in other domains, and thus
can be very significant and their analysis valuable. By
analyzing the outliers peer can easily identify the outliers
and isolate them from the system.
270
Fig.3 Peer Cluster
H. Evolution and deviation analysis
Evolution and deviation analysis pertain to the
study of time related data that changes in time. Evolution
analysis models evolutionary trends in data, which consent
to characterizing, comparing, classifying or clustering of
time related data. For example a peer can find the change in
the strategy of each peer with respect to time and how the
reputation gets changed. Deviation analysis, on the other
hand, considers differences between measured values and
expected values, and attempts to find the cause of the
deviations from the anticipated values.
It is common that users do not have a clear idea of the kind
of patterns they can discover or need to discover from the
data at hand. It is therefore important to have a versatile and
inclusive data mining system that allows the discovery of
different kinds of knowledge and at different levels of
abstraction. This also makes interactivity an important
attribute of a data mining system.
V. RESULT
Experiments on an existing peer to peer network
were analyzed. Java environment was used to develop the
peer to peer structure. Initially the experiment was done
with 20 nodes, and later on scaled up to 100 nodes.
Existence of behavior analyzer increased the interaction
with good peer, indirectly isolating malicious and free riders
in the peer to peer system. Fig 4 and 5 proves the
observation mentioned above. At the starting point each
peer holds an INR initial reputation value (we took INR =
30) then with respect to the peers interaction the reputation
value changes. The queries from a good peer are given
higher preference, which indirectly suppresses the action of
the free riders. And the query replies from a good peer will
be given more weight, which indirectly reduces the
malicious peer’s action. So our concept of integrating data
mining with peer to peer network greatly increases the
lifetime of the p2p system by directly and indirectly
eliminating free riders.
Fig.4. Interaction graph for good peer
Fig.5. Interaction graph for Free rider or malicious peer
CONCLUSION
While there are several ongoing research projects
on improving the quality of service in P2P file sharing
systems, there hasn't been much research to counter the
problem of free riding effectively, which is essentially a
data quality issue. To address the free riding problem in P2P
systems, we have introduced this concept to measure the
usefulness of every user to the system. We have proposed a
free riding control scheme based on the general data mining
functionalities. We expect that this paper to trigger further
research in this area of P2P systems.
271
ACKNOWLEDGEMENT
We would like to thank our department staffs for
teaching us the latest trends in Peer to peer networks and our
Head of Department for permitting us to experiment our
peer to peer software in college research cell.
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Strategize As Well As Predict Nodes’ Behavior In Peer-to-Peer Networks”.
Proceedings of the 11th International Conference on Parallel and
Distributed Systems (ICPADS'05).
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Networks”. Euro-Par 2003 Parallel Processing.
[8]. J. Han, M. Kamber, “Data Mining: Concepts and Techniques”,
Harcourt India / Morgan Kauffman, 2001.
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[12] http://www.wikipedia.org
 2009 International Conference on Computer Engineering and Technology

Servo System Modeling and DSP Code Autogeneration Technology for Open-CNC

Cao Shukun, Zhang Heng, Song Li, Ai Changsheng, Ze Xiangbo

School of Mechanical Engineering
University of Jinan
Shandong Province, China
E-mail: caoshukun@126.com

Abstract—With the progress in science and technology, the algorithm, often have to re-writing code based on specific
modernization manufacturing process have become hardware specification and the ability to achieve. As a result,
increasingly demanding in CNC. In this paper, it has chosen such a fracture of the development process has brought a lot
Permanent Magnet Synchronous Motor (PMSM) and Digital of questions: the specification is not explicit; it can’t carry on
Signal Processor (DSP) to design the servo control system for a test to the whole system; the too late detection of design
Open-CNC. Based on the analysis of the Open-CNC structure, flaw brings in high risk; the factors of artificial mistake are
movement principle, fundamental theories of vector control, it increasing and so on [1].
has built the PMSM vector control module in MATLAB, and A new and high-efficiency DSP developing flow will be
introduces the control principle in detail. In addition, a new
introduced in this paper.
and high-efficiency DSP developing flow is introduced. The
flow realizes the auto -generation of DSP code with the use of II. OVERALL DESIGN OF OPEN CNC SYSTEM
MATLAB, Simulink, Real-Time Workshop, DSP and TI’s
development environment-CCS. It realizes the integration
A. Project Design of Open CNC System Based on DSP
from design to implementing, and shortens the developing
period greatly. The PMSM control system designed using this X-Y table is a key component to realize X and Y
flow now has worked well on the TMS320F2812EVM. At last, coordinated movement, widely used in the machine
it gives the communications circuits between PC and DSP. manufacture and processing [2]. In our research, we plan to
achieve the X-Y table movement firstly. Its working
Keywords: Open-CNC; Permanent Magnet Synchronous principle is that the direction control of X, Y is adopted the
Motor(PMSM); CCS; Digita Signal Processor servo motor, through gear reducing the speed and screw
driving motion, to drive the X-Y table to do the motion. The
I. INTRODUCTION servo system function of the X-Y table is that transfer the
motion and the dynamic force from servo drive equipment to
From the view of cybernetics, the technical requirements the actuating mechanism, to achieve the tools cutting
of the CNC system can be summed up as the servo system campaign.
stability, static and dynamic features. Because the
qualifications of CNC machine, such as the speed and the
accuracy, to a large extent is decided by the servo system
performance. Now, servo systems are used more widely,
such as in industrial controllers and household appliances.
Performance, function and price of servo products are being
concerned more and more. Digital controlled AC servo
systems will be the trend, which uses Permanent Magnet
Synchronous Motors (PMSM) as power units and high
quality control strategy, based on DSP.
For the servo motor control, the most direct way is to set
up a mathematical model servo motor, and then create a
MATLAB simulation of the servo motor control model,
which is just for analysis. However, only the analysis is not
enough, we should achieve the control of the servo motor. At
present, the use of DSP digital control system is the
development trend of the control system design, but the
traditional DSP development process has great disadvantages.
As the design phase of the system, designers only consider
how to get the "perfect" algorithm model in line with the
performance, but in the realization phase, the engineers were
concerned how to market the product as soon as possible,
system design and realization lack of effective
communication. When programmers get the design Figure 1. The DSP-based Open CNC system diagram

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DOI 10.1109/ICCET.2009.200
 Permanent Magnet Synchronous Motor has lots of they can choose the generate code commend in MATLAB to
advantages, such as simple structure, small size, light weight, generate the code of the module. In the code generated
small loss, high efficiency, and so on. Compared with DC process, MATLAB will provide some coherent message,
motor, it does not have mechanical commutate and brush; such as warnings, mistakes, which can help to correct the
compared with the asynchronous motor, it does not need coherent module or the parameters. The utility system
reactive exciting current, and thus it has high power factor, development flow diagram using the CCSLink is shown in
the little loss of the stator current and stator resistance, and Fig. 2.
the rotor parameters being measured, the rotor having bigger CCSLink provides three ways to achieve the data
air gap, good control performance. So the servo motor we exchange between MATLAB and CCS, DSP. One way is
choose Permanent Magnet Synchronous Motor (PMSM) to “ the connected object to RTDX (real-time data exchange)”.
design. The servo control structure based on DSP is shown in “the connected object to RTDX” provides real-time
Fig. 1. communication channel between DSP and MATLAB. The
We design three-dimensional profile of the parts firstly, use of channel exchange data real-time with the DSP rather
and then extract the processing path. According to the than being to stop the procedure implemented on the DSP.
needed displacement, we calculate the pulse count that DSP So, in the exchange of the data between the DSP and host,
generates. DSP generating the pulse can be implemented by the author used the RTDX (Real-Time Data Exchange, real-
software. When the machine is working, the processing time data exchange) technology. RTDX in the
information can feed back real-time, which forms a closed- implementation of the application process can help the
loop. developers transmit and accept the data between the host and
the objectives DSP, and also observe real-time and stored
B. The Foundation of the PMSM Control Module data through an Application Programming Interface (OLE
1) The Introduction of Embedded Target for TI C2000 API). The design uses the TDS560 to support the high-speed
DSP data link RTDX ™, the speed is up to 2MB/s [3].
The Embedded Target for TI C2000 DSP is developed by
Mathworks and TI, and provides a method which can
integrate the MATLAB, Simulink and Texas Instruments
Express DSPTM tools, TI C2000 DSP to develop systems.
Making use of this tool, we can operate TI DSP memory or
register as MATLAB variables, that is to say, the whole
objective of DSP for MATLAB seems to be transparent. We
can send the data from CCS to MATLAB, and can also send
the data from MATLAB to CCS. CCSLink supporting any
target board can identified by CCS, including TI's DSK,
EVM board and target DSP (C2000, C5000, C6000) board
developed by users, and provides a powerful support for the
debugging and testing of TI DSP applications system design.
The toolbox can generate codes automatically online, and
speed up the development of prototype and embedded
system. According to the CCS document requirements of the
project, the toolbox can generate the C code accord with the
TI DSP requirements, and to verify and achieve real-time the
algorithm automatically on the DSP of TI C2000 series. The
MATLAB 7.0 support F2812eZdsp and F2407eZdsp, and
integrate the modules of PWM, ADC and CAN in the form
of modules, that have provided a great convenience for the
development and application of the users).
Developers will be able to complete the operation of CCS
in the MATLAB environment. Developers can use the
powerful visualization, data processing and analysis function Figure 2. The utility system development flow diagram make use of the
of MATLAB to analyze and process the data from the CCS C2000 embedded target module
and TI DSP, however, to complete these operations is very 2) The module foundation of the PMSM
difficult in the DSP integrated development environment
CCS. According to the vector control concept, it is to set up the
When developers make use of the CCSLink module to PMSM control module. The SVPWM vector control method
design the utility system, they should set up the Simulink is that used the actual magnetic flux being generated by the
module correctly. If the module has some mistakes, the inverter under the different switch module to approach the
connection between the MATLAB and the CCS will be not basic circle flux, which is generated by AC motor when it
built. After building the module, the developers should supply the three-phase sine wave voltage, and compared the
connect the DSP with PC, and give the DSP power. Then result to design the switch state of the inverter.

273
 The key of vector control is to control the magnitude of ### Build complete
the stator current vector and location of the space (frequency ### Downloading COFF file
and phase). used the method id = 0 to control, so the control ### Downloaded: servo motor. out
component of d-axis is 0, which will help to simplify the Then it starts the CCS automatically. CCS will compile
control algorithms, and let the electromagnetic torque and assemble the generated C code, connect and download
proportional to the stator current, at the same time will not the C code to 2812EVM board to run. The left side of the
cause the PMSM bad performance because of CCS interface is the debugging tools and the project
demagnetization, and easy to implement. Its main drawback management window, where display the projects built in the
is that with the increase of output torque, power factor rapid process of generating C code. The projects include the source
decline. As the control component of d-axis is 0, the stator files (source folder), library files (Library folder), connected
windings can be decoupled with the d-axis, so it can control files (F2812.cmd), and other documents generated. On the
the electromagnetic torque well, which is similar as the right side of the CCS are the disassembling files generated
control principle of the permanent magnet DC motor. the codes. The bottom of the CCS is the monitoring window.
PMSM control module consists of four parts: the Now, it display as starting the model, operational
detection module of the speed and the location, PI controller procedures), the lower left-hand corner shows the state of
of the speed ring and the electric current loop, coordinates F2812EVM CPU, at this time shows RUNNING. The
transform modular, and SVPWM module and the inverter PMSM control project in CCS is shown in Fig. 4.
module. The PMSM control module built in MATLAB is
shown in Fig. 3.
3) The PMSM control code generation from MATLAB to
CCS
To set the parameters of the real-time workshop. First, it
is to implement the target configuration. Set specific
parameters are as follows:
x System target file: ti_c2000_grt.tlc or
ti_c2000_ert.tlc;
x Template make file: ti_c2000_ert.tmf;
x Make command: make_rtw.
The parameters of real-time workshop are complex, due
to limited space, such as code optimization, or some other
CCS IDE tab settings will not introduce them. The following
are given some important settings in this design.
x The code generation target type of the TI C2000
target selection must choose C2000;
x The incorporate DSP/BIOS of the TI C2000 code
generation must be not chosen;
x There are some options, such as the byte type of the
TI C2000 complier and the frequency of the TI
C2000, which must be corresponding to the setting
of the 2812EVM development board.
After setting the parameters, knob down “Build &Run”,
then the MATLAB commend window will display the codes Figure 4. The PMSM control project
being generated, at the end, we can see as follows:
…… TMS320F2812EVM can run the auto-generation code
correctly, and achieve the control of PMSM, which verify
the correctness of code.
C. The Communication Circuit Design between DSP and
PC
TMS320F2812 not only integrates a synchronous serial
port on the chip, but also integrates serial asynchronous ports,
and this feature has simplified the PC serial communication
circuit design. Serial Communication Interface (SCI) is the
asynchronous communication interface using double line
correspondence, known as the UART port. To reduce the
Figure 3. The PMSM control module serial communication cost of the CPU, the ports of F2812
support the 16 serial receiving and transmitting FIFO. SCI
### Building Code Composer Studio (tm) project … receiver and transmitter send data by a dual-level buffer.

274
They have independent enable functions and the break bit,
can operate independently, and also operate in full-duplex
mode at the same time [4].
In order to ensure the integrity of the data, SCI module
checks the interruption, polarity, limits and frame error of
docking data received. By designing program to set the 16-
bit baud rate of the control register, different communication
rate of SCI is collocated.
In general, in order to ensure the universal application of
the system, it’s better that serial communication interface is
compatibility with the electrical standard bus (such as RS-
232C). Therefore, the design uses MAX3232 to design the
circuit between DSP and the PC machine. The
communication circuit design is shown in Fig. 5. Serial
RS232 asynchronous communication is used between PC
and the DSP. SCITXD is a sending data pin of SCI, and
SCIRXD is a receiving data pin of SCI [5]. The selection of
the design is MAX3232 whose voltage +3.3 V that is the
same to the output voltage of TMS320F2812, so the no-
electricity conversion simplifies the circuit design.
Figure 5. The communication circuit design between DSP and PC
III. CONCLUSIONS
This paper describes the vector control theory of
Permanent Magnet Synchronous Motor, and according to the
theory of vector control, we make use of DSP to realize the
three closed-loop control of Permanent Magnet Synchronous
Motor’s currents, the speed, and the location. Experiments
have proved that the application of MATLAB for DSP
system-level development ideas effectively to improve the
275
traditional design methods. MATLAB system-level design
environment will help early detection of an error in the
design and to deal with the ever-increasing challenge of the
complexity of the system, which facilitates the design of
complex application system based on DSP. Through the
process flow, the entire design process don’t need to input
the code, can get the highly optimized auto-generation code.
The PMSM control system designed using this flow now
has worked well on the TMS320F2812EVM.
However, CCSLink only for debugging of DSP program,
and if it's TI and Math Works jointly developed another tool
ETTIC6000 (2000) used in conjunction, can be directly the
Simulink model from to generate TIC6000 (2000) of
Executable code[6], that is integrated, unified environment
MATLAB to be completed by DSP development of the
whole process.
ACKNOWLEDGMENT
The study was supported by Key Subject Research
Foundation of Shandong Province under Grant #XTD0713.
REFERENCES
[1] Qi Xinggang, Zhao Gang, and Li Yuan, “Auto-generation of
DSP code on MATLAB/Simulink bench,” China
Measurement Technology, vol. 31, no. 1, pp. 87-88, January
2005.
[2] Zhang Haiwei, “Modeling and simulation of CNC machine
tool servo system for X-Y table,” Northwest A&F University,
November 2007.
[3] Texas Instruments Incorporated, “TI DSP CCS
guide,”Tsinghua University Press, 2005.
[4] Wan Shanming. “TMS320F281XDSP principle and
application,” Beijing University of Aeronautics and
Astronautics Press, 2007.
[5] TI company. “TMS320F2812 digital signal processors data
manual,” http://focus.ti.com.cn/cn/dsp/docs/, July 2007.
[6] Wang Yang, Guo Lili, and Fan Liping, “Research of DSP
developing method based on MATLAB,” Control
Engineering, vol.13, no. S0, pp. 123-126, May 2006.
 2009 International Conference on Computer Engineering and Technology
Extending Matching Model for Semantic Web Services
Alireza Zohali
Department of Computer Science
Sama Technical &Vocational Training School,
Khorasgan Branch, Isfahan,Iran
Alireza_Zohali2006@yahoo.com
Abstract— The Semantics-based Web Service Matching Model
is proposed in this paper to improve the performance of Web
Service discovery. Semantics-based Web Service Matching is a
two-phase method including semantic similarity matching and
qualitative filtering. Function-based semantic similarity
matching used of matching algorithm in order to finding the
most proper services. And also, used for selecting the best
service among results of qualitative filtering.
Keywords-component; Web Service; Function-based Semantic
similarity matching; matching algorithm; qualitative filtering;
Service Discovery
I. INTRODUCTION
Development of Service-Oriented Architecture (SOA) to
provide software systems and Web Services brilliant growth,
make use of a proper and practical procedure to discover and
finding the services, very essential. It's clear that finding out
the interested services, without using mechanize methods is
very difficult and time consumer. This issue is similar to
search in web pages without using browsers. Service
consumer can be a user, another service or a program. Thus
using of automated mechanisms to finding out services is
very important.
Services are the autonomous elements and often
platform-independent that are described, registered and
discovered and due to their integrity can provide a program.
The most sensible ability of service-oriented systems is the
possibility of software elements assembly in order to
building a grid of services with loose-coupling that
dynamically make processes and handy programs.
Basically, SOA is a designing style for sharing, reuse and
services interchange in a distributed environment. Web
serviced through use of integrity features and reusability, the
best solution for development of wide software scale is for
enterprises. But by fast growth of the number of accessible
services, the people will face to problems such as find out
proper service and above selection to make their programs
[1].
In respect to complexity of finding proper service in a
Query, using of different techniques and procedures have
been proposed in recent years. Among this, most basic
methods stressed on functional, semantic and qualitative
matching that remove the need of user by submitting
functions.
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.127
Kamran Zamanifar , Naser Nematbakhsh
Department of Computer Science
Isfahan University
Isfahan, Iran
{zamanifar ,nemat }@eng.ui.ac.ir
Semantic matching means to find out services With
regard to functions meaning and concepts and service
capabilities in used domain and context. Qualitative
matching means to finding services that client and server
can agree on offer procedure with respect to qualitative
scales.
Since Universal Description, Discovery and Integration
( Services specifications registry platform) that briefly calls it
"UDDI" , does not use of any semantic information in
services registry and their searching abilities emphasis only
on key words, can limit capability and efficiency of queries.
Therefore, semantic matching web service is proposed to
increase efficiency of query. Moreover, incorporation of
results similarity matching and qualitative filtering that used
in proposed model of this study can modify the accuracy of
proper web services [5]. The related work is first introduced
in Section 2. Basic method and idea will describe in section3.
Section4 represents the proposed model and Section5, shows
the future works and conclusions.
II. RELATED WORK
Web services provide a standard means of interoperating
between different software applications, running on a
variety of platforms and/or frameworks. W3C use the
definition as follows: A Web service is a software system
designed to support interoperable machine-to-machine
interaction over a network. It has an interface described in a
machine- process able format (specifically WSDL). Other
systems interact with the Web service in a manner
prescribed by its description using SOAP messages,
typically conveyed using HTTP with an XML serialization
in conjunction with other Web-related standards.
Recent years several papers have been represented for
solving the problem. Matchmaker proposes a solution based
on DAML-S, a DAML language for describing web
services. And the discovery query in DAML-S is compared
with the results that have features compatible with the
request in semantic [2].LARKS describe services as
functional parameter input, output. The matching engine of
the matchmaker agent contains five different filters: context,
profile, similarity, signature and constraint filter [3].
276
 III. MAIN IDEA AND METHODS -- Stability
In order to implement the efficient service matching, a -- Response Time
service matching model is includes four steps: -- Reliability
1) All services are defined according to the ontology of -- AcessedTimes
Web Service to generate semantic description, and when -- Grade
they are published, they are insured to have the same
description schema.
2) The semantic information of the Web Service
Description is extended according to the specification of
UDDI, and the extended semantic information is stored in
UDDI as tModel.
3) The set of Web Services is se3ected according to three
matching steps, which are as follows:
a) Input parameter matching
b) Output parameter matching
c) Well-defined matching by user
4) The best Web Service is selected according to
qualitative filtering method on the set obtained in step3 [4].
In the following will introduce the methods used in these
4 steps in details.
A. Semantic Descriptions and Publication of Web Services
In order to give a standard form for the providers and
requestors to describe and locate Web services, two types of
Ontology are used in the model.
SWSD (Semantic Web Service Description Ontology)
contains the semantic description of primary information, Figure1. Semantic Web Service Description Ontology SWSD
functional information, and other non-functional
information of Web services 3) Mapping from SWSD to UDDI
QoWSO(Quality of Web Service Description Ontology) Like the mapping from DAML-S Profile to UDDI,which
describes the qualitative description of Web Services. is used in the registration of WSDL in UDDI Registry, the
UDDI provides the Web Service specification for sharing mapping from SWSD to UDDI includes:
of information and applications. In order to semantically The mapping from Provider Information in SWSD to
development of UDDI, will store the semantic and Contacts of Business Entity in UDDI ,the mapping from
qualitative information of Web service into the UDDI .In Primary Information as service Name and text Description
the matching model, we try to improve the quality in order in SWSD to name and description in Business Service, the
to enhance the capability of UDDI. mapping from SWSD service semantic description to
1) Semantic Web Services Descriptive Ontology tModel ,the mapping from service quality description to
The four types of information described in SWSD for tModel. The detailed mapping is shown in Fig .2.
service discovery are [4]: B. Function-based semantic similarity matching of Web
-- Primary Information and Provider Information Service
-- Functional Description Function-based semantic similarity matching of Web
-- Quality Description Service includes the semantic similarity matching of input
-- Other Attributes Description and output, which is called IO matching for short. Assume
The structure of Semantic Web Service Description that the input set and output set of service request are RI and
Ontology SWSD is shown in Fig .1. RO respectively, and the input set and output set of
2) Quality of Web Service Description Ontology advertised service are AI and AO respectively, therefore
The model applies a Quality of Web Service Description if RO ⊆ AO which means that advertised service can
Ontology (QoWSO) to construct the consistent description provide all the outputs required by the service request, then
of Web service quality (historical statistical information and
output matching is successful; and if AI ⊆ RI which means
the up-to-minute information of Web Service quality).
QoWSO is described by the class QoWSProfile, which is that service request can provide all the inputs needed by the
advertised service, then input matching is successful.
described by five quality parameters [4]:

277

Figure2. Mapping from SWSD to UDDI [6]
1) Matching Algorithm
The main rationality behind our algorithm is that an
advertisement satisfies a request when the advertisement
provides all the functions (Inputs/Outputs) of the request's
service. Also, the user can specify weights for all
parameters, so the users can tell which parameters in their
requests are most important and must be provided, and
which parameters are not necessarily exactly matched.
The value of weights is limited as a decimal in [0, 1] i.e.
0 < ω <1. The default value for every weight is 1, which is
an important factor for matching.
2) Main Control Loop
Requests are matched against all the advertisements
stored in the registry in the main control loop of the
algorithm as shown in Fig.3.
When the score (match degree figured out) of a match
between the request and an advertisement is higher than ω
which is a user specified threshold in the request, we add the
advertisement and score to the result list. Because the
services may have multi input or multi output, we define
ScoreListo and ScoreListi, as two lists to store the match
degree of every output and input respectively.
Then we use the degrees acquired above and weights of
every parameter given in the request to compute the overall
score.
278
Figure3. Main control loop
3) I/O Matching
The I/0 matching is based on the degree assignment logic
as follow [2].
Exact:
If request R and advertisement A are equivalent
concepts, we label the match as Exact. Of course, it is the
best level since the request and advertisement are exactly
the same.
Sub-concept:
If request R is sub-concept of advertisement a
(inductive advertisement) which is the second matching
level.
Super-concept:
If request R is super-concept of advertisement A (the
advertisement can provide only some specific cases of what
the request desires) which is the third matching level.
Fail:
There are no relation between the advertisement A and
request R. This is the lowest level which represents an
unacceptable result.
Since the requests may have multi-output, we return a list
containing the degrees of every output. And the list is used
to compute the overall score of the matching with other
parameters and their weights later.
The matching between inputs follows the above
algorithm Fig.4 As outputs matching algorithm. The
difference is just in that inputs of request are matched
against the inputs of the advertisement.
4) Computing the Overall Score
After the I/O matching, we should let the user know an
overall score of the selected advertisement in the result list.
And we accomplish the work with the function Fig.3:
Compute Overall Score (request, ScoreListo, ScoreListi)

Figure4. Algorithm for outputs matching
We get the weight for every parameter from the request.
Then we define the variable used to compute the overall
score as follow
ScoreListo : list of matching degrees of outputs, which is
the return value of the outputs matching
ScoreListi : list of matching degrees of inputs, which is
the return value of the inputs matching.
SOi :(0 < i < |scoreListo|): match degrees of outputs in
scoreList0;
SIi :(0 < i < | scoreListi |): match degrees of inputs in
scoreListi;
WOi (0 < i < |scoreListo|): the weights of the outputs;
WIi (O < i < |scoreListi |): the weights of the inputs;
ScoreI : weighted average degree of inputs matching;
ScoreO : weighted average degree of outputs matching;
overallScore : the overall score of the match
Then we compute the match degree from (1), (2) and
compute matching overall Score From (3)
| Scorelist i |
∑W Ii * S Ii
Score I = i =1
| Scorelist i |
(1)
∑W
i =1
Ii
| Scorelisto |
5) Sorting Rules
Arrange the result in a reasonable order. If the scores in
advertisement are the same, we would sort as follow:
Over all Score> ScoreO > ScoreI
C. Semantic filtering on QoWS
Use Semantic filtering on QoWS to find the best
service in the case of existing a lot of functional similar
Web Services. The base of Semantic filtering on QoWS is to
compare the measured value of service quality parameters in
the QoWS between different services. The service quality
parameters include:
- Stability
- Response Time
- Reliability
- AcessedTimes
- Grade
The service quality parameters are classified as two kinds:
One has the higher measuring value such as Reliability,
AcessedTimes and Grade the other has the smaller measure
such as Response Time and Stability.
Finally can obtain the best service by compare the
measured value and quality parameters.
IV. ARCHITECTURE OF THE MODEL
The architecture of the service matching model is shown
in Fig.5. The four function modules are Web Service
Publication Agent, Web Service Search Agent, UDDI
Adapter, and QoWS Certificate Authority. The two libraries
are SWSD and QoWS Ontology Base, and UDDI Registry.
A. Web Service Publication Agent
The function of Web Service Publication Agent is to
process the interaction between Service Provider and UDDI
Adapter, and implement service publication. And the main
part of Web Service Publication Agent, which is named
SWSD Generator, is designed to generate the SWSD
description of Web Service according to Web Service
information and Specification of Web Service Description
SWSD.
B. Web Service Search Agent
The function of Web Service Search Agent is to process
the interaction between Service Requestor and UDDI
Adapter, and implement service query. It includes the
following three parts:
- SWSD Parser

∑W
- QoWSO Parser
Oi * S Oi - Web Service Matching Engine
Scoreo = i =1
| Scorelist o | (2) 1) SWSD Parser

∑W
Extracts the Web Service attributes from SWSD
Oi description file, generate the corresponding object instance,
i =1 and provide the necessary data to Function-based semantic
similarity matching filter.
Score i + Score o
OverallSco re = (3)
2

279
 2) QoWSO Parser
It extracts the QoWS information from QoWSO
description file, generate the corresponding object instance,
and provide the necessary data to QoWS Semantic filter.
3) Web Service Matching Engine
It matches the Service Requests and the advertised
services registered in UDDI Registry to obtain the list of
proper candidate services. It consists of Function-based
semantic similarity matching filter and Quality of Web
Service Semantic filter.
Figure5. The Architecture of Web Service Matching Model
Function-based semantic similarity matching filter: It
first gets the attributes of required service from SWSD
Parser, and then executes semantic similarity matching
between the attributes and SWSD function descriptions
which are extracted from UDDI Registry by UDDI Adapter,
and generates a list of functionally similar Web Services.
Quality of Web Service Semantic filter: It first uses
QoWS descriptions of Web Services in the list of
functionally similar Web Services, which are extracted from
UDDI Registry by UDDI Adapter, gets the attributes of
QoWS from QoWS Parser, then execute the QoWS
semantic filtering, and finally return the result to the
Requestor.
C. UDDI Adapter
The function of UDDI Adapter is to process the
interaction between other parts and UDDI Registry, which
includes:
280
1) Find out the business services referred by the tModels
in the list of function-based semantic similarity matching
and obtain the list of matched Web Services.
2) Create the business service information structure of
UDDI and store the information in UDDI Registry, by using
Web Service information provided by Web Service
Publication Agent and URL of QoWS description obtained
from QoWSCA, and applying the mapping method from
SWSD to UDDI. And transfer the Web Service ID to
QoWSCA.
3) Invoke the interface of UDDIfind_tModel() to search
all tModels belonging to swsdSpec in UDDI Registry for
function-based semantic similarity matching.
D. QoWSCA
QoWSCA provides Quality Certification to the Web
Services published in UDDI Registry.
V. CONCLUSION
How to give up the proper service description in UDDI
and how to match customer service request with service
description, become the major challenges in Web Service
discovery. Development of Web Service matching model
proposed in this paper uses semantic description and quality
description of Web Service based on ontology as the basis
of service matching, and its matching algorithm is based on
this logic that when a notice can satisfy a request that ever
functions (Input/Output) provides request service. Moreover
a user can specify some weights for all parameters and can
say that which parameter of his/her request is premier and
should provide and which one is unnecessary.
The service matching method composed of two phases.
In the first phase, the proper Web Services that satisfy the
functionality matching of the desired service is found, and in
the second phase, the best one is selected from Web Services
set obtained in the first phase using semantic filtering. Using
these descriptions and matching phases, the performance of
Web Service matching can be improved effectively.
REFERENCES
[1] R. A. Tsalgatidou, T. Pilioura,, "An Overview of Standards and
Related Technology in Web Services" [J].Distributed and Parallel
Databases, 2002(12),pp.135-162
[2] M. Paolucci, T. Kawamura, T.R. Payne, and K. Sycara,"Semantic
Matching of Web Services Capabilities",International Semantic Web
Services Conference, 2002.
[3] K. Sycara, S. Widoff, K. Klusch, and J. Lu, "LARKS:Dynamic
Matchmaking among Heterogeneous Software Agents in
Cyberspace," Autonomous Agents and Multi-Agent Systems, 2002,
pp. 173-203.
[4] J. Fan, B. Ren, L. Xiong, "An Approach to Web ServiceDiscovery
Based on the Semantics", Proceedings of 2005 International
Conference on Fuzzy Systems and Knowledge Discovery, 2005, pp.
1103 - 1106.
[5] UDDI Universal Description, Discovery and Integration (UDDI 2.0).
UDDI Technical White Paper,http://www.uddi.org/ specification.htm.
[6] F. Leymann and S. Pottinger ,"Rethinking the Coordination Models
of WS-Coordination and WS-CF", Proceeding of the Third European
Conference on Web Services(ECOWS'05), 2005
 2009 International Conference on Computer Engineering and Technology
Sound absorption measurement of acoustical material and structure using the echo-
pulse method
Sun liang Hou hong Dong liying Wan fangrong
College of Marine,NWPU
Xi an ,China
Email:slyxj1996@163.com
Abstract—A new approach is proposed for measuring the
sound absorption coefficient of material at normal incidence,
that bases on the echo-pulse method in a circular tube. Two
types of acoustical material and structure were employed:
foam and pipe exit. Two different waveforms called the
Butterworth and the damped sine pulse are successfully
generated and used as incident pulses in the experiment of
sound absorption measurement in a circular pipe. The
absorption coefficient of the acoustical material measured by
the echo-pulse method is consistent with that of B&K 4206
impedance tube measurement. Discrepancy between them is
totally less than 10% in broadband frequency range from 200
Hz to 5k Hz, mainly due to the different internal tube
diameters of these two tubes and the fitting of the sample, i.e.
the difference between the sample diameter and the internal
tube diameter. Since the incident sound pulses are all with the
features of regular waveform, short duration and broadband
frequency spectrum, sound absorption measurement results
conducted by the echo-pulse method are more reliable than
that of B&K 4206 impedance tube method especially in low-
frequency range. Further validation of this new method is also
given through the absorption coefficient measurement of pipe
exit. The echo-pulse method is relatively simple as only single
microphone is used for measurements. It is believing that this
method could also be used for in-situ measurement of sound
absorption coefficient in outer space.
Keywords- absorption coefficient; echo-pulse; acoustic
material; pipe exit
I. INTRODUCTION
Acoustic materials can be characterized by their acoustic
surface impedance, sound reflection coefficient or sound
absorption coefficient. This paper deals with the
measurement of sound absorption coefficient. According to
Garai[1],it can be measured by three basic methods: usually
reverberation room ,impedance tube, and reflection. The
reverberation room method presents the Sabine absorption
coefficient, which is an approximation of the absorption
coefficient for random incidence. The impedance tube
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 281
DOI 10.1109/ICCET.2009.104
method[2],based on the transfer function method established
by Chung and Blaser[3,4],yields the plane wave absorption
coefficient at normal incidence in the frequency domain. Its
errors and limitations have been analyzed by several
authors[5-6].
Many times there exists the requirement to measure the
sound absorption coefficient at oblique incidence or in
situ.Although there have been proposals to extend the
transfer function method in situ[7],most of the acousticians
use the reflection method.Even,there are two international
standards to measure in situ the sound absorption coefficient
of surfaces employing the reflection method[8,9].
Recently, Jing and Fung [10] developed an impulsive
sound source which can generate repeatable sharp pulses
whose durations were as short as 0.5ms.
From open literatures, there are few works about
designing and generating a good-waveform sound pulse in
sound absorption measurement by the echo-pulse method.
In this paper, our purpose is to examine an alternative to
the transfer function method, based on measuring the sound
pressures of incident and reflected wave at normal incidence.
II. EXPERIMENTAL SETUP
As shown in Fig.1, the experimental setup includes
computer, an NI DAQ6062E data acquisition board, a
BNC2029 adaptor, power amplifiers, a loudspeaker-pipe
system and a 1/4in MPA416 condenser microphone. The
loudspeaker is a 5in Hi-Vi mid-frequency woofer which is
positioned in a column steel enclosure whose length is 15
cm. The sound wave from the loudspeaker is guided through
a 20 cm conical reduction to a straight pipe 27mm in
diameter. The microphone, which is flush mounted into the
pipe wall and placed at 24cm from the open end of the tube,
is used to measure the sound pressure of incident and
reflected pulse. The sample is placed vertically at the end of
the straight pipe, backed by a rigid steel cover. The AI and
AO channels of the NI DAQ 6062E board are programmed
to be synchronized and have the sampling rate of 100k Hz.
 Sample
Power&amplifier Computer
Mic

DAQ Board
Power amplifier (NI DAQ6062E)

Loudspeaker

AI
Adaptor
(NI BNC2090)
AO

Figure 1. The sound source configuration and the test equipment

Second, the spectrum of the driving signal can be

III.
computed by the spectrum of the designed pulse H y (ω )
OUTLINE OF THE PRINCIPLES OF MEASUREMENT

A. generation of sound pulses and the resolved frequency response H (ω ) .

According to the requirement of measurement of sound
absorption coefficient, two different sound pulses referred to
as the Butterworth and the damped sine have been
generated[10-11].Both the principle and process of pulse
generation are only summarized here:
First, using the impulse response of a digital 5th-order
the Butterworth filter and about 1ms duration as the
excitation signal to drive the loudspeaker-pipe system with
open terminus, we record the response signal at microphone.
H e (ω ) and H r (ω ) denote the Fourier transforms of
excitation and response signals, respectively. Then, the
frequency response of the system is given by
H r (ω )
H (ω ) =
H e (ω )
H y (ω )
H x (ω ) = (2)
H (ω )
Applying the inverse Fourier transform to (2), the driving
signal of the loudspeaker-pipe system will be obtained.
The generated pulses of the Butterworth, the damped sine
and their frequency spectrums are shown in Fig. 2-1 and
Fig.2-2, respectively. It shows that the generated pulses have
the characters of regular waveform, short duration and
broadband frequency range. These pulses are suitable for
absorption coefficient measurement by the echo-pulse
method.
(1)

0.25
2.5

2 0.2
Sound pressure(Pa)

1.5
Amplitude(Pa)

0.15
1

0.5 0.1

0.05
-0.5

-1 0
0.3295 0.33 0.3305 0.331 0.3315 0 1000 2000 3000 4000 5000 6000 7000
Time(s) Frequency(Hz)

Figure2-1. The Butterworth pulse and its spectrum

282
 1.4

2.5
1.2
2

1.5 1
Sound pressure(Pa)

Amplitude(Pa)
1
0.8
0.5

0 0.6

-0.5
0.4
-1

0.2
-1.5

-2 0
0.3295 0.33 0.3305 0.331 0.3315 0 1000 2000 3000 4000 5000 6000 7000
Time(s) Frequency(Hz)

Figure2-2. The damped sine pulse and its spectrum

range from 100 Hz to 6k Hz. The difference between them

is less then 5%.
B. Absorption coefficient measurement
The sound-absorbing materials of sponge and foam are
1.2
examined in the loudspeaker-pipe system. The specimens are
made of the thickness of 16 mm and the diameter of 27 mm 1
(inner diameter of pipe),
The reflection coefficient of a test specimen is calculated Absorption coefficient 0.8

by the ratio of the reflected sound pressure to the incident

sound pressure. 0.6

p (ω ) −
W (ω ) =
0.4
(3)
p + (ω ) 0.2

Where p (ω ) and p (ω ) are the Fourier transforms

− +
0 Mean value of three times measuring by damped sine impulse
of the reflected and incident sound pressures, respectively. Mean values of three times measuring by butterworth impulse
1000 2000 3000 4000 5000 6000 7000
Considering no transmission wave, the absorption Frequency (Hz)
coefficient can be solved as
A(ω ) = 1 − W 2 (ω )
(4) Fig. 3 The absorption coefficients measured by the
The complete separation of the incident and reflection Butterworth and the damped sine pulses
pulses and high signal-to-noise ratio (SNR) are essential for
the echo-pulse method measurement. In this experiment, the It can be seen from Fig.4-1and Fig.5-1that the
enough distance between the microphone and the measured absorption coefficients measured by the echo-pulse method
specimen will ensure the incident and reflection pulses (in man-made pipe) and the transfer function method (in
separation. The time-domain averaging of pulse reflection B&K 4206 impedance tube) agree well, not only for the
measurement will increase the SNR. Butterworth pulse, but also for the damped sine pulse.
However, the curve by the Butterworth pulse is much closer
IV. EXPERIMENTAL RESULTS to that by the B&K 4206 impedance tube, which is probably
due to the much flatter frequency response from 400Hz to
A. foam material 4k Hz of the Butterworth pulse than the damped sine pulse.
Fig. 3gives the curves of the average absorption Anyway, the difference of the absorption coefficients
coefficients measured by the Butterworth pulse and the measured by this method and the transfer function method
damped sine pulse. It shows in Fig. 4 that the absorption are within 10%.
coefficients measured by two pulses agree well in frequency

283
 1
Echo-impulse method
0.9 0.35 Impedance tube method

0.8
0.3
Absorption coefficient

Absorption coefficient
0.7
0.25
0.6

0.2
0.5

0.4 0.15

0.3
0.1
0.2
0.05
0.1 Echo-impulse method
Impedance tube method 0
0
0 1000 2000 3000 4000 5000 6000 7000 50 100 150 200 250 300 350 400 450 500
Frequency (Hz) Frequency (Hz)

Fig.4-1 Comparison of absorption coefficients measured by the Butterworth pulse and B&K 4206
Fig. 4-2 The partially enlarged drawing of Fig. 4-1
0.8

0.7
1
0.6

0.8 Absorption coefficient 0.5

Absorption coefficient

0.4
0.6
0.3

0.4 0.2

0.1
0.2
0

0 -0.1
Echo-impulse method Echo-impulse method
impedance tube method -0.2 impedance tube method
-0.2
1000 2000 3000 4000 5000 6000 7000 100 150 200 250 300 350 400 450 500
Frequency (Hz) Frequency (Hz)

Fig.5-1 Comparison of absorption coefficients measured by the damped sine pulse and B&K 4206
Fig. 5-2 partially enlarged drawing of Fig. 5-1

B. Pipe exit
To further verify our new method, measurement of 2 J1 (ω ) 4 π /2

π∫
apparent absorption coefficient of the pipe exit is also Z (ω ) = 1 − + j sin(ω cos( x )) sin 2 xdx
ω 0
presented here. An open tube has the advantage that its
(5)
reflection coefficient can be predicted theoretically through
the theoretical calculation of radiation impedance of pipe Where ω is the angular frequency normalized by the
exit. The radiation impedance of a circular pipe exit with an sound speed and the pipe diameter. The reflection
infinite flange has been derived by Morse and Ingard[12] coefficient as well as the absorption coefficient of the pipe
and that of the unflanged one by Levine and Schwinger[13]. exit can be computed by:
According to Morse and Ingard[12] ,the acoustic Z (ω ) − 1
impedance of a pipe exit with an infinite flange is given as W (ω ) = (6)
Z (ω ) + 1
below:
2
α (ω ) = 1 − W (ω ) (7)
Here, the measurement results of the two impulses will
be presented directly.

284
 0.9
1
0.8
0.5
0.7
0
0.6
Absorption coefficient

Absorption coefficient
-0.5
0.5

-1
0.4

-1.5
0.3

0.2 -2

0.1 -2.5
echo-impulse method echo-impulse method
theoretical solution theoretical solution
0 -3
0 1000 2000 3000 4000 5000 6000 0 1000 2000 3000 4000 5000 6000
Frequency(Hz) Frequency(Hz)

(a) (b)
Figure6 absorption coefficient of pipe exit measured by impulse method (a)Butterworth (b)damped sine

From above two curves, one can see the measured
absorption coefficient agrees well with the theory of Morse
and Ingard[12],although slight fluctuations occur in two
measured curves, that are mainly caused by background
noise from outside of the tube. However, for the narrow
frequency band of damped sine impulse, its measurement
result is not better than that of Butterworth impulse.
V.
A novel method to measure the sound absorption
coefficient using the echo-pulse technique is proposed in
this paper. The main part of the measurement system is a
circular pipe with a sound pulse source at one end, the
specimen at the other end. The sound source emits the
designed pulse as incident wave which should have a
regular waveform and a broadband frequency response. A
microphone is flush mounted into the pipe wall. The
distance between the microphone and the measured
specimen should be large enough to separate the incident
pulse and reflected pulse. Compared with the result of
B&K4206 impedance tube measurement, the proposed
method has good repeatability and high precision, although
different internal tube diameters are used in our research.
Besides this ,the effectivity of this new method is also
verified through the measurement of the absorption
coefficient of pipe exit.
It is believing that this method could also be used for in-
situ measurement of sound absorption coefficient in space.
ACKNOWLEDGMENT
This work was financially supported by the National
Natural Science Foundation of China with Grant
No.10674111 and doctorate foundation of Northwestern
Polytechnical University with Grant No.Cx200802.
REFERENCES
[1]Garai M, “Measurement of sound-absorption coefficient in situ:the
reflection method using periodic pseudo-random sequences of maximum
length,”.Appl Acoust,vol. 39, pp. 119–139, 1993.
[2]ISO 10534-2, Acoustic-determination of sound absorption coefficient in
impedance tubes,Part2:transfer function method.Geneva, Switzerland:ISO
1996.
[3]Chung J Y,Blaser DA. “Transfer function method of measuring in duct
acoustic properties I.theory,”.J Acoust Soc Am, vol. 68, pp. 907–913, 1980.
[4]Chung J Y,Blaser DA. “Transfer function method of measuring in duct
acoustic properties II.experiment,”.J Acoust Soc Am, vol. 68, pp. 914–921,
CONCLUSIONS 1980.
[5]A.F.Seybert and B.Soenarko, “Error analysis of spectral estimates with
application to the measurement of acoustic parameters using random sound
fields in ducts,” J Acoust Soc Am, vol. 69, pp. 1190–1199, 1981.
[6]H.Boden and M.Abom, “influence of errors on the two-microphone
method for measuring acoustic properties in ducts,”. J Acoust Soc Am, vol.
79, pp. 541–549, 1986.
[7]Nocke C,Mellert V. “Brief review on in situ measurement techniques of
impedance or absorption,”. In Proceedings of the forum acusticum ,2002.
[8]CEN/TS 1793-5.Road traffic noise reduction devices-test method for
determing the acoustic performance-Part 5:intrinsic characteristics-in situ
values of sound reflection and airborne sound insulation,2003.
[9]ISO13472-1.Acoustic measurement of sound absorption properties of
road,Surfaces,in,situ-Part,I:extend surface method.
[10]X.Jing and K.Y.Fung, “Generation of desired sound impulses,”.
J.Sound and Vib. vol. 297, pp. 616–626, 2006.
[11]Xu shihua ,Houhong. “A method to develop a dignally-controlled
sound impulse generator,”.Journal of elementary electroacoustics, vol.31,
pp. 9–13, 2007.
[12] P.M.Morse and K.U.Ingard, Theoretical Acoustics(McGraw-Hill,New
York,1968).
[13]H.Levine and J.Schwinger, “On the radiation of sound from an
unflanged circular pipe,”. Phy.Rev , vol. 73, pp. 383–406, 1948.

285
 2009 International Conference on Computer Engineering and Technology

Parallel Design of Cross Search Algorithm in

Motion Estimation
ZHANG Fan
China University of Geosciences (Beijing), the School of Geophysics and Information
Technology, Beijing, 100083
68kilo@163.com

Abstract—In order to facilitate the H.264/AVC implementation in
application system with low bit rate; its coding algorithm should be
optimized. In this paper, through analysis of the motion vector
distribution in the video image and the serial search strategy, a type of
parallel program based on the cross search algorithm is presented,
designed and implemented. Finally this design program is played in
the multi-core computer equipment, and the experimental result
shows: on the premise of the coding speed of the cross search
algorithm after paralleling has been greatly improved without
affecting the quality of H.264 coding.
Keywords—cross search; parallel design; motion estimation;
H.264/AVC
I. INTRODUCTION
II. MOTION ESTIMATION TECHNOLOGY
A. Motion estimation fundamental
For motion image the most situation is that only very few
part of images in motion, the content difference between two
neighboring video frame in same identical scene is not obvious,
or that the content of the latter and the preceding have many
overlap parts, that is to say that the two frame have strong time
relativity. The basic idea of motion estimation (as shown in
figure 1) is to divide each frame of image sequence into many
non-overlapping macro blocks, and that all the block-pixel
displacement are the same, then for each block in current frame
find the most similar block namely matching block in given
scope to the front or after frame according to certain matching

H .264/AVC is a video code standard established by ITU-T

and the MPEG expert group JVT, this standard can obtain
very high coding efficiency, especially has distinct
criteria, and figure out motion displacement from the relative
position of the matching and current block and the motion
displacement is the motion vector of current block. Finally
enhancement in the low code rate aspect compared to MPEG-4, using the motion vector for motion compensation in reference
suits to the needs of low broad band and high quality network
frame and the difference will be lineal coded with the motion
video. But to improve coding efficiency H.264/AVC used
vector after DCT change, quantization and run coding then
many algorithms with high computed complexity which make
spread out with form of bit-stream.
the computation of coding and decoding very big, this becomes
a main bottleneck of H.264/AVC code standard application.
In a continuous video sequence, there is a lot of relevance current frame
between the adjacent frames, which is called frame
redundancy. Through reducing the frame redundancy the
efficiency of video encoding can be greatly improved, because
H.264/AVC adopts that various block partition modes use the
motion compensation with 1/4 pixel accuracy synchronously,
so comparing with the previous coding standard, its motion Y
Cr
estimation accuracy is greatly improved, but at the same time reference frame + difference
Cb
the calculation amount is also greatly increased. Through
analyzing the various algorithm modules of H.264/AVC
encoder it can be seen: the motion estimation module of
DCT+quantization+run
H.264/AVC encoder is the main time-consuming module, and coding…
in the single frame of reference, its waste time occupies 60% of
the total encoder; in the 5 frames of reference, its waste time motion vector
lineal coding
can occupy as high as 80%, therefore, how to reduce the MV

calculation amount of the module is one of the hot spots in the Fig.1 motion estimation basic idea
encoder algorithm.
Obviously, the smaller the difference, the more favorable it is
to be compressed. Therefore, the main objective of motion
estimation is to make the residual value between the forecast

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 286

DOI 10.1109/ICCET.2009.65
and current block as little as possible. The selection of the
search window size (the scope of the search) has great impact to
the search results: the greater the scope of the search, more
possibility to obtain smaller residual block, but it will bring
about more time spending. Usually the searching scopes can be
selected are: ± 7,±16,±32 and so on. Because the
block-matching needs pixel comparison between the before and
after frame, so if full search algorithm (that is compare with all
possible points in exhaustive reference frame region and then
find the overall merits) is used, the computation is very large.
B. Block matching function
The accuracy of motion estimation relies on the matching
criteria used in block-matching process and in which the most
important is the block distortion measure (BDM) between two
matching pixel blocks. Here assuming that the frame motion of
objects is limited in the search window (2 w × 2w) . Following
the definitions of these BDM and the corresponding matching
criteria are given: m and n are the scale to calculate BDM pixel
block (corresponding to the width and height), U ( k , l ) is the
object in before frame, U R (k + p, I + q) is the matching
block of current frame and ( p, q ) is the frame motion vector
of objects (displacement vector).
1) Cross-correlation function (CCF)
m n
∑∑U (k , l )U
k =1 l =1
R (k + p, l + q)
M 1 ( p, q ) = 1 1
⎛ m n 2 ⎞2 ⎛ m n ⎞2
⎜ ∑∑U (k , l ) ⎟ ⎜ ∑∑ U R (k + p, l + q ) ⎟
2
⎝ k =1 l =1 ⎠ ⎝ k =1 l =1 ⎠ motion vector distribution in the video sequence is uniform is
Matching criteria: the pixel block corresponding to the least
M 1 ( p, q) (− w ≤ p ≤ w,− w ≤ q ≤ w) in the region is
searched to be optimal matching block, d h = p and d v = q
separately denotes the horizontal component and vertical
component of the motion vector.
2) Mean square error (MSE)
2 concentrated distributed in the vicinity of the central region
1 m n
MSE ( p, q) = ∑∑ [U (k , l ) − U R (k + p, l + q)]
mn k =1 l =1
Matching criteria: the pixel block corresponding to the
maximal MSE ( p, q ) ( − w ≤ p ≤ w,− w ≤ q ≤ w) in the
region is searched to be optimal matching block, d h = p and
d v = q separately denotes the horizontal component and
vertical component of the motion vector.
3) Majority average error function (MAE)
1 m n
MAE ( p, q ) = ∑∑ | U (k , l ) − U R (k + p, l + q) |
mn k =1 l =1
Matching criteria: the pixel block corresponding to the least
MAE ( p, q ) (− w ≤ p ≤ w,− w ≤ q ≤ w) in the region is
searched to be optimal matching block, d h = p and d v = q
287
separately denotes the horizontal component and vertical
component of the motion vector. Here optimal matching is
corresponding to finding the maximal point of NCCF.
After the SAD (Sum of Absolute Differences) criteria
appeared, it rapidly replaced MAE and adopted by each kind of
motion estimation algorithm, only because its matching effect
is equivalent to MAE but get rid of the needless division and the
computation is largely reduced.
N −1 N −1
∑∑ | I
SAD(i, j) = 2 (m, n ) − I1 (i + m, j + n ) |
m = 0 n =0
This is because that for the blocks with same shapes size,
those N Z are same and it is not necessary to divide it; on the
other hand, in the calculation process when the block SAD is
found more than the current least it will be dropped out, thereby
the calculation will be significantly reduced. Therefore in this
paper SAD is used to be block-matching criteria.
C. Choice of motion estimation search algorithm
To design reasonable search templates and search strategies,
the research of the motion vector distribution in the video
sequence is very important. For the people, the knowledge of
the motion vector distribution in the video sequence and the
development track of the template search algorithms are
synchronous. In the early template search algorithms, such as
the template search algorithms (TTS), the motion vector
distribution in the video sequence was generally believed to be
uniformly, that is, any one within the frame reference search
scope could be the best match block of the current coding
block. However, with in-depth study, this knowledge that the
proved to be inaccurate. Through the statistic to the motion
vector distribution in a large number of video sequences, it is
found that the motion vector distribution of the video sequences
are basically in the center, and with the motion vectors location
farer the number of them is smaller. This is the video sequence
motion vector concentrated distribution rule generally known
currently. By analyzing, it is found that the motion vectors are
with a different ratio. Based on the research result, in this paper
the cross search algorithm is regarded to be an example and is
carried out a parallel design.
III. CROSS SEARCH ALGORITHM PARALLEL DESIGN
A. Search algorithm parallel design background
Through the research of the current search algorithm, it is
found that the main difference between the algorithms is the
aspect of selecting search point on the template, that is, the
shape and size of the search template are different. In the search
range with same size, in order to achieve faster search speed
and better search result with fewer search points, the different
template fast search algorithms have different starting point and
ideas when designing the search template. But they have one
common characteristic: all the search templates must go
through the descending, successive transformation process. In
other words, no matter whether or not it is necessary, they will
complete the entire search template change process NAME: Parallel-Cross-shaped
step-by-step. The deficiencies of these algorithms can be Begin
summarized as follows: the search template can not be selected Select search starting point;
optionally, and it is can not be processed flexibly based on the For each i par-do //the first search
motion type of the video image. No matter what motion type of Computer SAD (i);
the video image, it must be firstly searched by the large For (i=1; i<=4; i++)
template and then oriented by the small template; or firstly Select min ( SADL (i ) ); //find the point i responding
searched with larger search radius, and then gradually reduce
the smallest SAD in the large cross image
the search radius. This search strategy is undoubtedly a waste
For (i=1; i<=4; i++)
for the video image with small motion type, thus affecting the
speed of the search. Farther, when using large template to Select min ( SADS (i ) );//find the point i responding
search, the search direction can be misled because the larger the smallest SAD in the small cross image
search radius, thus affecting the accuracy of the search.
B. Cross search algorithm parallel design If SADL (i ) <=SAD (0) and SADS (i ) <=SAD (0)
When searching the large and small cross templates together then
participate in the search, then according to the different location //SAD (0) search the SAD value of the starting point
of the block with the smallest matching error SA the different The block of SAD (0) is the best current matching block
templates will be adopted to complete the subsequent searches Else
(under normal circumstances, the search distance of the large If SADL (i ) > SADS (i ) then
cross template is 2 times of the small). When the block with the
smallest SAD value in the search center, it will be considered FindLengthS = FindLengthS × 2
the best matching block and the end of the search; When the FindLengthL = FindLengthL × 2
block with the smallest SAD value in the large cross template,
the search distance of the large and the small templates SADL (i ) → SAD(0)
separately improved to 2 times of the original search distance, Else
then the form that the large and the small templates together FindLengthS
participate in the search will be adopted for the next step search, FindLengthS =
until the block with the smallest SAD value in the search
2
center; When the block with the smallest SAD value in the FindLengthL
FindLengthL =
small cross template, the search distance of the large and the 2
small templates separately reduced to 0.5 times of the original
SADS (i ) → SAD(0)
search distance, then the form that the large and the small
templates together participate in the search will be adopted for Do Parallel-Cross-shaped
the next step search, until the block with the smallest SAD //recursively execute the above parallel cross
value in the search center. The idea is shown in figure 2. algorithm
End IF
End IF
C. Time complexity analysis of cross search parallel
algorithm
In the multi-processor system environment, assuming that
the time complexity of common cross search algorithm
is O (N ) . Then for the parallel cross search algorithm, in the
best case, its algorithm time complexity is O (log 2
N ) ; and in
the worst case, its algorithm time complexity is O (N ) .
Therefore, in a multi-system environment, the time complexity
of the parallel cross search algorithm
O(log 2 N ) + O( N )
is O (S) parell = , and comparing to serial
2
cross search algorithm, the implementation efficiency of
parallel cross search algorithm has been obviously improved.

Fig. 2 cross search parallel algorithm IV. EXPERIMENT RESULT AND ANALYSIS
The cross search parallel algorithm is: The experimental conditions are: experimental flat roof is
JM9.3 of JVT; coding sequence is IBPBP format; the number

288
of frame reference is 5; coding speed are 30fps; the machine using the serial cross search algorithm to code and the other is
type is DELL XPS 370; translate and edit circumstance is using the parallel cross search algorithm presented in this
VC6.0; and the testing sequence is “Foreman”, “Stefan”, paper, and the complexity of testing sequence, image quality
“Akiyo” of the
QCIF (176 × 144 ) format.
and bit rate are separately compared with, which is shown in
table 1.
Two test groups are done in this experiment, one group is
TABLE1 SERIAL AND PARALLEL CROSS SEARCH ALGORITHM EXPERIMENT RESULT

(a) Foreman

Serial algorithm Parallel algorithm

Quantitative
PSNR bit rate time PSNR bit rate time
parameter
16 37.48 16022079 61.83 37.48 16022079 35.91
24 30.35 7027231 65.47 30.35 7027231 36.73
28 26.95 4288159 59.32 26.95 4288159 33.16
36 20.15 1263455 64.37 20.15 1263455 36.18
(b) Stefan
Serial algorithm Parallel algorithm
Quantitative
PSNR bit rate time PSNR bit rate time
parameter
16 37.89 7250180 57.65 37.89 7250180 31.82
24 31.33 2384252 53.63 31.33 2384252 30.81
28 28.51 1352196 53.89 28.51 1352196 29.94
36 22.95 480804 55.47 22.95 480804 29.73
(c) Akiyo
Serial algorithm Parallel algorithm
Quantitative
PSNR bit rate time PSNR bit rate time
parameter
16 39.75 1560993 35.12 39.75 1560993 20.56
24 33.76 466697 39.56 33.76 466697 21.78
28 30.98 252689 32.14 30.98 252689 17.07
36 25.23 88305 37.01 25.23 88305 20.50

Through analyzing the experimental data in table 1 it is easy
to be found: comparing with the previous serial cross search
algorithm, after paralleling the PSNR and the bit rate has not
change, but the coding time is reduced obviously,
consequently, the parallelizing of cross search algorithm is
validated and has notable effect to improve the coding speed of
H.264/AVC coder.
V. CONCLUSION
In this paper the motion estimation part which affects the
coding speed of H.264/AVC coder is optimized, that is, the
cross search algorithm is parallelized. Through the analysis of
the experimental data it can be seen that the performance of the
optimized motion estimation has obvious improvement under
the circumstance of enough hardware condition (exist several
operation units).
With the rapid development of computer network and
communication technology, the demand to provide high quality
network audio, video and other multimedia services is growing.
Because the unique applying environment of real-time video
transmission, traditional coding methods are no longer apply
with it. Through optimizing the coding speed of H.264/AVC
coder, it is able to meet market demand. Therefore improving
the coding speed of H.264/AVC coder will continually become
a hotspot of future academic research and its research results
also will have important impact on peoples’ living.
REFERENCES
[1] ITU Recommendation H.261. Video Codec for audio visual services at
px64k bits/5. 1990.
[2] ITU Recommendation H.263. Video Coding for low bit rate
Communication. 1995.
[3] ISO/IEC DIS 13818-2(MPEG-2). Generic Coding of moving pictures and
associated audio information. 1994.
[4] Draft ITU-T Recommendation H.264 ， Joint Video Team (JVT) of
ISO/IEC MPEG and ITU-TVCEq Document JVT C167，May 2002，
45∼49.
[5] ITU-T Recommendation H.261, Video Codec for Audiovisual Services at
px64 k bit/s ，Mar.1993，19∼21.
[6] Liqiang Han, Bo Yang, a method rapidly realize H.264 motion
estimation, Opto-electronics, 2006, 33(8), 94~97.
[7] Junhao Huang, Duyan Bi,and so on, H.264-based multi frame motion
estimation rapid algorithm, Computer engineering, 2006(32),238∼240.

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[8] Yonglin Ding, Haibin Shen, motion estimation algorithm comparison and
improvement in H.264/AVC, Digital TV & digital video,2005,17(2),
17∼18.
[9] Iain E.G.Richardson, new multimedia video coding technology of H.264
and MPRG-4 video compress, Changsha: National university of defense
technology publishing, 2004, 99~107.
[10] Lansun Shen, Li Zhuo, Video coding and low rate of transmission,
Beijing, Electronic Publishing Industry, 2001, 27~31.
[11] [Wensheng Wang, Ming Yang, Huijuan Cui, adaptive early exit
algorithm of H.264 motion estimation, Tsinghua University Journal
(Natural Science), 2004, 44(10), 28~31.

290
 2009 International Conference on Computer Engineering and Technology
Influences of DSS Environments and Models on Current Business Decision and
Knowledge Management
Md. Fazle Munim
MBA in MIS
Stamford University
Dhaka, Bangladesh
e-mail: sayket@gmail.com
Abstract-This paper mainly discussed on the influence of
Decision Support System – DSS´s environments and models on
current global business decision and knowledge management.
While describing the different stages of DSS development it
also clarifies the influencing environments as well as the
models of DSS. It is also identify the relation between Data,
Information and Knowledge.
Keywords-modeling; environment; support; relation;
knowledge; information
I. INTRODUCTION
"Information technology is no longer a business
resource, it is the business environment." and this is the
truth of modern business world. Increasing global
competition, are adding complexity and uncertainty of
several orders of magnitude to the modern business
environment and organizational style where the ongoing
advances in information technology (IT) have a great effect
to control over the uncertainty and challenges. Information
technology management (or IT management) is a
combination of two branches and they are information
technology and management. This aims at achieving the
goals and objectives of an organization through a proper
automation system. And in the automation system DSS or
Decision Support System have a vital role.
Decision Support Systems (DSS) are a specific class of
computerized information system that supports business and
organizational decision-making activities. A properly
designed DSS is an interactive software-based system
intended to help decision makers compile useful information
from raw data, documents, personal knowledge, and/or
business models to identify and solve problems and make
decisions.
Typical information that a decision support application
might gather and present would be:
• An inventory of all of your current information
assets (including legacy and relational data sources,
cubes, data warehouses, and data marts)
• Comparative sales figures between one week and the
next
• Projected revenue figures based on new product
sales assumptions
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.96
Fatima Binte Zia
BSc in Electrical & Electronics Engineering
Independent University
Dhaka, Bangladesh
e-mail: fatimabintezia@yahoo.com
• The consequences of different decision alternatives,
given past experience in a context that is described
II. EARLY STAGE OF DSS
Computerized decision support systems became practical
with the development of minicomputers, timeshare operating
systems and distributed computing. The history of the
implementation of such systems begins in the mid-1960s. In
a technology field as diverse as DSS, chronicling history is
neither neat nor linear. Different people perceive the field of
Decision Support Systems from various vantage points and
report different accounts of what happened and what was
important. As technology evolved new computerized
decision support applications were developed and studied.
Researchers used multiple frameworks to help build and
understand these systems. Today one can organize the
history of DSS into the five broad DSS categories, including:
Communications-driven, Data-driven, Document driven,
Knowledge-driven and Model-driven decision support
systems.
In the early days, DSS was filled with magic – we
thought that managers would be directly involved with
computers to evaluate scenarios and support their decision-
making! The dream lost steam and focus in about the mid
80’s. That was because “information” took over from
“decisions” in the use of IT. DSS was largely reduced to
spreadsheet reports and accessing and reporting historical
data. Executive Information Systems (EIS) that followed
DSS provided what became routine reporting systems for
historical data with their routine, non-imaginative, complex,
and even boring interfaces.
In 1980, Steven Alter published his MIT doctoral
dissertation results in an influential book. Alter's research
and papers (1975; 1977) expanded the framework for
thinking about business and management DSS. Also, his
case studies provided a firm descriptive foundation of
decision support system examples. A number of other MIT
dissertations completed in the late 1970s also dealt with
issues related to using models for decision support. Alter
concluded from his research (1980) that decision support
systems could be categorized in terms of the generic
operations that can be performed by such systems. These
generic operations extend along a single dimension, ranging
291
from extremely data-oriented to extremely model-oriented.
Alter conducted a field study of DSS that he categorized into
seven distinct types of DSS. His seven types include:
• File drawer systems that provide access to data
items.
• Data analysis systems that support the manipulation
of data by computerized tools tailored to a specific
task and setting or by more general tools and
operators.
• Analysis information systems that provide access
to a series of decision-oriented databases and small
models.
• Accounting and financial models that calculate the
consequences of possible actions.
• Representational models that estimate the
consequences of actions on the basis of simulation
models.
• Optimization models that provide guidelines for
action by generating an optimal solution consistent
with a series of constraints.
• Suggestion models that perform the logical
processing leading to a specific suggested decision
for a fairly structured or well-understood task.
III. MODELING DECISION MAKING
Decision support systems (DSS) disciplines deal with the
use of information technology to support human decision-
making processes. Michael Scott-Morton, who virtually
invented the discipline in the early 1970s, offered this
definition of DSS:
“Decision support systems couple the intellectual
resources of individuals with the capabilities of the
computer to improve the quality of decisions. It is a
computer-based support system for management
decision-makers who deal with semi-structured
problems.”
Like this his founding definition – like all definitions –
presents us with problems: who is the decision-maker? What
kinds of data serve as inputs to the decision-making process?
What does the decision-making process itself look like?
What kinds of risks and constraints are associated with the
decision-making process? How is the output of the decision-
making process a decision – evaluated, implemented and
tracked? Ultimately, no matter where we ground ourselves in
a discussion of decision support systems, we have to develop
and employ a model of decision-making: the set of activities
that DSS environments support. The key elements of this
model are fairly common, and include:
• A decision-maker: an individual or group charged
with making a particular decision
• A set of inputs to the decision-making process:
data, numerical or qualitative models for interpreting
that data, historical experience with similar data sets
or similar decision-making situations, and various
kinds of cultural and psychological norms and
constraints associated with decision-making
292
• The decision-making process itself: a set of steps,
more or less well-understood, for transforming the
inputs into outputs in the form of decisions
• A set of outputs from the decision-making
process, including the decisions themselves: a set
of criteria for evaluating decisions produced by the
process against the set of needs, problems or
objectives that occasioned the decision-making
activity in the first place.
IV. DECISION MAKING ENVIRONMENT:
In making important decisions, any information sources
that contain relevant important information are going to be
accessed and used, if possible. The decision maker uses the
whole network of information sources and variety of
available media. In most cases it is impossible to access or
produce all required information, so decisions are made
under circumstances of uncertainty and incomplete
information. Business decision support seems to have
common ground with other areas containing significant
analytical work: scientific research, military and political
intelligence, or criminal investigation – in all cases, there are:
• A problem situation which requires analysis in line
with general strategy and goals
• Assumptions
• Deficit of information (and time in many cases)
• Certain (usually big) amounts of diverse empiric
data which is chaotic in its nature and has to be
processed in some way for relevant facts and
findings
• Field knowledge is required to extract these facts and
findings
• The calculated facts and findings are carefully
evaluated against wider context - political, social,
ethical etc
• Growing IT support. Apart from needs for data and
information and their availability, knowledge
possessed or required by the decision making subject
is an important part of the decision environment
V. HOW DO DSS ENVIRONMENTS SUPPORT DECISION
MAKING:
DSS environments support the generic decision-making
model above in a number of ways:
• In decision preparation, DSS environments
provide data required as input to the decision making
process. This is interestingly enough; about all most
data mart and data warehousing environments do
today.
• In decision structuring, DSS environments provide
tools and models for arranging the inputs in ways
that make sense to frame the decision. These tools
and models are not pivot tables and other aspects of
data presentation found in query tools. They are
actual decision making tools, like fault tree analysis,
Bayesian logic and model-based decision-making
based on things like neural networks.
 • In context development, DSS environments again
provide tools, and provide the mechanisms for
capturing information about a decision’s
constituencies (who’s affected by this decision),
outcomes and their probabilities, and other elements
of the larger decision making context.
• In decision-making, DSS environments may
automate all or part of the decision-making process
and offer evaluations on the optimal decision. Expert
systems and artificial intelligence environments
purport to do this, but they work only in very limited
cases, because of some fundamental flaws in the
technology (namely, their inability to deal with non-
binary, or fuzzy, choices, like “it’s more likely that
we’ll lose market share than win it,” which is a rule
that no traditional AI-based system can code).
• In decision propagation, DSS environments take
the information gathered about constituencies and
dependencies and outcomes and drive elements of
the decision into those constituencies for action.
• In decision management, DSS environments
inspect outcomes days, weeks and months after
decisions to see if (a) the decision was implemented
/ propagated and (b) if the effects of the decision are
as expected.
VI. HUMAN SIDE OF IT MANAGEMENT DECISION
SUPPORT SYSTEM
In the field of business decision support, more and more
recent research has been concentrating on the human side of
the person-technology relation in decision making. It has
been shown in a variety of works that business decision
making environment is a unity of decision makers’
experience, beliefs and perceptions on one side, and decision
support tools and techniques – on the other side. The
information environment surrounding business activities and
decisions is getting increasingly complex due to growing
volumes of information of potential relevance to certain
business activities; increasing number of sources of such
information; and multiplying technologies for accessing and
handling data and information. The expected role of
information technologies (IT) is to filter and direct relevant
information flows and to provide reliable and flexible
support.
At the same time, every case of decision making for a
problem situation tests the existing support mechanisms and
provides valuable information for future situations, thus
creating new knowledge and experience for participants
involved, and in the case of right decision increasing
confidence in future actions. Alongside with technologies for
handling data and information, lately much attention has
been given to knowledge management (KM) models, their
relevance to decision activities and relations between data,
information and knowledge. In knowledge-intensive
activities, such as decision support, these relationships are
important in terms of efficient utilization of information
resources, and especially those supported and facilitated by
IT with its present capabilities, so it has been considered
293
worthwhile to take a look at the role of data, information and
knowledge in the context of managerial decision making,
and professional learning and experience.
Human side of IT management decision support, there
seems to be agreement that IT should act as:
• An enhancing instrument for decision search and
analysis as a high-level and knowledge-intensive
management activities
• A creativity stimulation and managerial learning
tool,
• An instrument for reduction of biased attitudes as
well as insurance from making fatal decisions,
• An instrument for maintaining, managing and
developing the explicit part of knowledge on
decision making – models, situations, scenarios, case
studies etc.
VII. UNDERSTANDING OF RELATIONS BETWEEN DATA,
INFORMATION AND KNOWLEDGE
Data →Information → Knowledge: Data is processed
into information, which is evaluated against existing
knowledge or stimulates creation of new knowledge in a
sense that missing links in the decision model are produced
and put in place (”the pieces finally fit”). There are relations
or sequence chains between data, information and
knowledge with the idea that better understanding of these
sequences might help producing better support for problem
situations. Here is some other relationship chain that data,
information and knowledge share:
Knowledge → Information → Data: this sequence
might be based on having the knowledge to look for
information and then turn it into data. For instance, in a
problem situation general and professional knowledge can
point to what information is needed to make the right
decision, what information is readily available, and what
information must be produced from some sources. This
information is then worked into decision data – prices to be
set, planned investment, resource distribution and
redistribution, budget structure and so on.
Data → Knowledge → Information: knowledge is
required to process data into information. Another possible
case: the content of data suggests ways (or produces new
knowledge) to extract information out of this data, e.g.,
group or query the data by some criteria which carry
business logic or other rationale.
Information → Knowledge → Data: knowledge is
required to get data from information, where data amounts to
final decision criteria: buy – don’t buy; accept proposal –
reject proposal; set the price etc. From the need for simple
outcome the situation can be worked backwards to track
what information would be needed to, for instance, estimate
the price, and what knowledge precedes the definition of this
information, its sources, completeness, Such decision
disassembly might help to explain better what exactly should
be supported and how to do it best.
Knowledge → Data → Information: probably this path
is possible only conditionally if we admit that having
knowledge we know where to look for data to produce REFERENCES
required information. [1] Angehrn A. (1993). Computers that Criticize You: Stimulus-Based
Information → Data → Knowledge: the ending phase Decision Support Systems. Interfaces, 23,
of a decision where decision information is processed and No.3, 3-16.
discussed into a decision which might be in a form of data –
a simple figure, a set of figures, text, choice, but it carries the [2] Borch O.J., Hartvigsen G. (1991). Knowledge-based systems for
strategic market planning in small firms.
load of preceding decision information, concepts and Decision Support Systems, 7, 145-157.
models, and its emergence leads to new knowledge added to
existing body. In all cases, the components of these [3] Alavi, M., & Joachimsthaler, E. A., “Revisiting DSS Implementation
sequences take some of the following forms: Research: A Meta-Analysis of the literature and suggestions for
Data: researchers,” MIS Quarterly, 16, 1, 1992, 95-116.
• Initial, “raw” unprocessed empiric data; [4] Berners-Lee, T., “The World Wide Web: Past, Present and Future,”
• Data available from previous experiences; August 1996, URL
• Data as a decision result. http://www.w3.org/People/Berners-Lee/1996/ppf.html, last accessed March
Information: 5, 2007.
• Readily available at the start of decision process; [5] Bush, V. "As We May Think", The Atlantic Monthly, 176, 1, July 1945,
• Obtained from processed data or other information; 101-108,
• Available outside the organization. http://www.theatlantic.com/unbound/flashbks/computer/bushf.htm
Knowledge:
[6] Chuang T.-T., Yadav S.B. (1998). The development of an adaptive
• Possessed at the start of the decision process; decision support system. Decision Support Systems, 24, No.2, 73-87.
• “know-how” of adding decision value: converting
raw data into information, or extracting final data [7]V.Lalith Kumar, Business Decision Making, Management and
from information; Information Technology, MBA Research, ICFAI Business School
Bangalore.
• Newly acquired in the decision making process.
• These relations are important in a sense that the most [8] P.G. Keen and M.S. Scott-Morton. Decision Support Systems: An
valuable asset in decision making situation is Organizational Perspective. Reading, Massachusetts: Addison - Wesley,
knowledge, and 1978.
• Any support for expanding the existing knowledge,
regardless of the sequence, is adding value and
quality to the
• Decision making process.
VIII. CONCLUSION
This paper will mainly help the DSS designer to
understand and develop a DSS system. In other words, the
decision maker has to develop sufficient knowledge on the
situation to make a confident enough decision, and that
eventually knowledge, not information or data alone, is
used in decision making. This knowledge must possess
completeness, as opposed to fragmented pieces gained in
separated situations which might be having very little in
common; completeness is also reflected in a decision process
where the deciding entity performs an iterative buildup of
decision support points towards a sufficient set to make a
decision. Regarding the simple support tools and techniques,
and decision makers’ ideas on the ideal decision
environment, a concept of “information control center” can
be developed for a decision making environment, where the
key information sources and most often used support tools
are always up and accessible just by few mouse clicks.
The possibilities of IT in facilitating problem solving
creativity are an important issue in itself; here the technology
has some proven points – idea generation, exchange and
testing mechanisms; growing sophistication of work styles;
support of teamwork and communication. This issue
deserves special attention and is going to be a part of further
research.

294
 2009 International Conference on Computer Engineering and Technology
A Method for Transforming Workflow Processes to CSS
XIAO Jing
Computer School of Wuhan University，
Wuhan Hubei 430079，China
xj21c111@yahoo.com.cn
CHEN Shu
Computer School of Wuhan University，
Wuhan Hubei 430079，China
Abstract In this paper, we proposed a method to
transform the workflow processes into CCS, and the
workflow process is described in activity network
charts. For the first, we used CCS to describe the six
basic logical relations of the workflow model, and then
based on the basic expressions, we illustrate how to
transform the whole workflow model to CCS process
expressions.We provide the exact formalization basis
for the workflow model, and express its operational
semantics in the form of synchronous tree, at last we
use the synchronous tree to verify the correctness of
the workflow processes.
Keywords: workflow process, CCS, activity network
charts, operational semantics, correctness
1.Introduction
With the advent of information age, more and
more enterprise would improve their competitiveness
rely on information technology, the workflow
technique plays a key role on the informatization of the
business system. At present, in this area, there are a lot
of related research and have achieved rich results in the
theory and practice. WFMC defined six basic elements
of workflow model that is sequence, AND-split, AND-
join, XOR-split, XOR-join and cyclic relation[1].Based
on the six elements, workflow model can be expressed
in several different ways. For example, Aalst expressed
the workflow model in the form of WF-net[2,3,4], it is
based on Petri net. Harel used statecharts and activity
diagram to describe the workflow process in [5].
Besides there are some other descriptive forms based
on ECA rules[6,7,8] or based on logic[9], and so on.
Among all of the descriptive forms of the workflow
model, the activity network charts[10] is the most
convenient and understandable for users. However, it
does not have exact semantics and formalization basis,
so it is difficult to analyze its properties and verify its
correctness.
CCS is a functional language proposed by
Milner[11,12], process is the basic element. It is
suitable for describing the concurrent system, and it
has a strict mathematical basis. In this paper, we used
This work was supported by a grant from the National High Technology Research and Development Program of China (863 Program) （No. 2007AA01Z185）
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.106
WU Guo-qing
Computer School of Wuhan University，
Wuhan Hubei 430079，China
the process expressions in CCS to describe the
workflow model which is expressed in activity
network charts, thus provide the exact formalization
basis for the workflow model, and then we describe its
operational semantics in the form of synchronous tree,
at last we verify the correctness for the workflow
processes.
2.Activity Network Charts
In this paper, we use activity network charts to
describe workflow process. The activity network charts
is directviewing, convenient, simple and practical, it is
sufficient to describe any business process. A process
model can be regarded as a oriented graph composed
of nodes and directed edges. Nodes denotes activities,
directed edges denotes the ordinal relations between
activities. Nodes can be classified into three categories
by its function: task node, control node and symbol
node. The task node represents all kinds of activities or
tasks which are elements in a business process. The
control node represents the logical relations among
task nodes. There is only two kinds of symbol nodes:
start node and terminal node, s denotes the start node
and it means the beginning of a workflow, t denotes the
terminal node and it means the finishing of a
workflow.
There are six basic logical relations among the
task nodes, other more complicated relations can be
composed of the six basic relations.
a) sequence relation: it indicates that an activity is
execute immediately after another activity.
b)AND-split: it indicates that generating more
than one parallel branches from one execution branch.
c)AND-join: it indicates that several parallel
branches gather as one brach to execute.
d)XOR-split: it indicates that choosing a brach to
execute according to the condition of several potential
branches.
e)XOR-join: it indicates that when several
parallel branches gather as a brach, it can be executed
as long as one of them satisfy the condition.
295
 f)cyclic relation: it indicates that a task is Activity Network Charts is semi formalized, it
executed for more than two times. does not have exact semantics, so it is difficult to prove
Graph notations of the six basic logical relations its correctness. In this paper, we use CCS to describe
are shown from Fig 1 to Fig 6: the activity network charts thus provide the theoretical
principle for verifying the workflow model in a
formalized method.
a1 a2 In the last section, we illustrated the six basic
logical relations, any complicated business process can
be composed of them, so how to use CCS to express
Fig 1 sequence the six basic logical relations seems a very crucial
problem.
a2
There is a key point for the translation, that is
a1 every task node in the activity network charts is
correspond to a process in the CCS. This is because
a3 they are clastic in terms of their granularity. A task
node can be refined as several sub activities, and a
process can contain several actions. Based on this key
Fig 2 AND-split
point, we proposed the method for the translation.
a1 Fig 1 shows the simplest relation, that is sequence
relation, it indicates that a2 is executed immediately
a3 after a1, if we regard a1 and a2 as processes, this
relation can be expressed as a=a1.a2.
a2 Fig 2 is AND-split relation, a2 and a3 are parallel
processes, this relation can be described in CCS as a =
Fig 3 AND-join
a1.(a2 | a3), it means that a2 and a3 is concurrent
immediately after a1 is executed.
a2 Fig 3 is AND-join relation, a1 and a2 are parallel
a1 processes, a3 can be executed only if a1 and a2 are
both finished. This relation can be described in CCS as
a3 a = (a1.Nil | a2.Nil).a3. Nil denotes null process, it
does nothing. The reason to add Nil after a1 and a2 is
that a3 should be executed only if a1 and a2 are both
Fig 4 XOR-split
finished. If either one is not finished, a3 can not be
executed. So adding Nil enables a1 and a2 to wait for
a1 each other.
a3 Fig 4 is XOR-split relation, a2 and a3 are
exclusive choice, that is after a1 is finished, either a2
a2 or a3 can be choosed to execute. So this relation can be
described in CCS as a = a1.(a2 + a3).
Fig 5 is XOR-join relation, that is a simple
Fig 5 XOR-join merge, when either a1 or a2 is finished, a3 can be
executed. So it can be described in CCS as a = (a1 +
a2 a2).a3.
Fig 6 is cyclic relation, it is more complicated, we
a1 a5 can expressed it with recursive process expressions. If
a´ is the loop activity, we can conclude that:
a3
a´ = (a2 + a3).(a4.a´+a5).
So the entire activity network charts can be
a4 described in CCS as a = a1. a´.
Table 1 summarized the six basic logical relations
with their CCS process expressions.
Fig 6 cyclic relation Based on the basic expressions, we can express
any complicated activity network charts with CCS, in
3.Use CCS to Describe Activity Network Charts the next section, we would illustrate the transform

296
method via an example about an office automation
system. s

Table 1 six basic logical relations

basic logical relations process expressions
sequence a = a1.a2
AND-split a = a1.(a2 | a3) a1 a2
AND-join a = (a1.Nil | a2.Nil).a3
XOR-split a = a1.(a2 + a3)
XOR-join a = (a1 + a2).a3
cyclic relation a´= (a2 + a3).(a4.a´+a5)
a = a1.a´

4.An Example About an Office Automation System a3

The example is an OA system, it is used for
managing the documents. For the first, a initial
document is drafted or entered, after it was checked, it
would be sent to the leader of the department and the
bureau, then both of the leaders would read it, the next
step is joint signature, at last the document would be a4 a5 a7
kept or sent back to recheck according to the result of
the last step. Table 2 shows all of the activities in the
flow:
Table 2 activities in the flow
activity ID activity name
a1 draft a document
a2 enter a document a6
a3 check
a4 Leader of the department read
a5 Leader of the bureau read
a6 joint signature
a7 recheck
a8 keep the document
Fig 7 is the work flow model of the system:
According to the method which we proposed a8
before, we can transform the whole workflow model to
CCS as below:
Since the start node and the terminal node are just
symbols that indicate the beginning and finishing of t
the workflow, they need not to be described in CCS.
There is a loop in the flow, so we should express Fig 7 the workflow model of the OA system
the cyclic activity for the first, if we define a´ as the
cyclic activity, then from the workflow model, we can In this section, we verify the correctness of the
conclude that: workflow model by means of checking the semantics
a´ = a3.(a4.Nil | a5.Nil).a6.(a7. a´ + a8) of the process expressions, these process expressions
If the whole flow is a, it can be described in CCS describe the workflow model in the form of CCS. Fig 8
as: shows the model to be verified. Obviously, it is not
a = (a1 + a2).a´ correctness in its structure. When we consider c2, a4
should be executed only if a2 and a3 are both finished,
5.Verification for Workflow Model
The workflow model does not have exact however, when we consider c1, a2 and a3 are exclusive
semantics, so it is difficult to excute formalized choice,so there would be a deadlock before a4.
verification. In this paper, we transform the workflow
model to CCS, as CCS has exact semantics, we can
verify the workflow model base on it.

297
 6.Conclusion and Future Work
a2 The workflow technique is a key role for the
a1 a4 informatization of the business systems. There are so
many ways to represent the workflow model. From the
c1 c2 view of the user, activity network charts is the most
a3
convenient and simplest method, and it is quite
intuitionistic and understandable. However it does not
Fig 8 the workflow model to be verified
have the exact formalization basis, so it is difficult to
When we transform the model to CCS, there are verify the correctness of its structure and function.
two result: To solve this problem we use CCS to describe the
α= a1.(a2 + a3).a4 six basic logical relations in the workflow model, and
β= a1.(a2.Nil | a3.Nil).a4 then propose a method to transform the whole
In the expressions above, αandβ both represent workflow model into the process expressions in CCS,
the workflow model. thus provide the exact formalization basis for the
Suppose that inβ, a2=μ.a2´,a3=ω.a3´, Fig 9 workflow model. To verify the correctness of the
and Fig 10 describe the operational semantics of αand model, we first transform it to process expressions,
β in the form of synchronous tree. In Fig 10, we elide then describe the operational semantics by means of
the null process Nil after a2 and a3. Becauseαandβ the synchronous tree, and at last check the conflicts
based on the semantics to judge the correctness.
describe the same workflow process, their operational
In this paper, we only illustrate the verification of
semantics should be consistent. Howere, the two
some basic properties of the workflow processes, the
synchronous tree denote different implementation of
method needs to be improved to verify the properties
the two process expressions, we can conclude that they
such as soundness and liveness. The issue about taking
have different operational semantics. So it can be
more complicated verification of the workflow model
proved that the workflow process in Fig 8 is not
needs further research. In the future we should
correctness consider about its structure, hence the work
concentrate on the verification for the properties and
flow process is unsoundness[15] ， soundness is
correctness based on CCS.
originally defined in[14].
REFERENCES
[1] Workflow Management Coalition, The Workflow Reference
a1 Model[R], Technical report WFMC TC1003, Jan. 19th, 1995,
http://www.wfmc.org.
[2] W.M.P, van der Aalst. Modelling and Analysing Workflow
a2 a3 using a Petri-net based approach[C]. Proc.2nd Workshop on
Computer-Supported Cooperative Work, 1994, pp:31-50.
[3] W.M.P, van der Aalst.Verification of Workflow Nets[J].
Application and Theory of Petri Nets. 1997, Lecture Notes in
Computer Science, Springer-Verlag,Berlin 1997, Vol.12, pp:
a4 a4 407-426.
[4] W.M.P, van der Aalst.Three good reasons for using a Petri-net-
based workflow management system[C]. In Information and
process integration in enterprises: Rethinking documents,
Fig 9 synchronous tree of α Kluwer Academic, Norwell,MA,1998,pp:161-182.
[5] D.Harel, H.Lachover, A.Naamad, A.Pnueli, M.Politi,
R.Sherman. STATEMATE: A Working Environment for the
Development of Complex Reactive Systems[J]. IEEE
a1 Transactions on Software Engineering, 1990, Vol.5, pp:176-
201.
μ τ [6] G.Kappel., P.Lang, S.Rausch-Schott, W.Retschitzegger.
ω Workflow Management based onObjects, Rules and Roles[J].
IEEE Data Engineering Bulletin, 1995, Vol.18, pp:11-18.
[7] G.Kappel., S.Rausch-Schott,W.Retschitzegger. Coordination in
Workflow Management Systems-A Rule-based Aproach[J].
a2´|a3 a2|a3´ a2´|a3´ Springer LNCS, 1998.
[8] G.Kappel., S.Rausch-Schott, W.Retschitzegger. From rules to
a4 a4 a4 rule patterns[J]. In caise96, Creta,Gr-eece, Springer-Verlag,
Berlin, pp:99-115.
[9] H.Davulcu, M.Kifer, C.R.Ramakrishnan, I.V.Ramakrishnan.
Logic Based Modeling and A nalysis of Workflows[C]. In
Fig 10 synchronous tree of β ACM Symposium on Principles of Database Systems; ACM
Press, 1998.

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[10] B.Scholz-Reiter, E.Stichel. Business Process Modeling[C].
Berlin: Springer-Verlag, 1996.
[11] R.Milner.A Calculus of Communicating Systems[J]. Springer-
Verlag, Lecture Notes in Computer Science 92, 1980.
[12] R. Milner. Communication and Concurrency[M]. Prentice-
Hall, 1989.
[13] Lu Ruqian.The Formal Semantics of Computer Language[M].
Bei Jing: Science Press, 1992.
[14] Van der Aalst, W.M.P. Verification of workflow nets. In
P.Azema and G.Balbo(eds), Pro.Application and Theory of
Petri Nets, Toulouse, France, June. Lecture Notes in Computer
Science, 1248, 407-426. Springer, Berlin, 1997.
[15] H.Verbeek, T.Baster, W.Aalst.Diagnosing workflow processes
using Woflan. Eindhoven University of Technology,
Eindhoven, Tech. Rep.:WP48, 2000.

299
 International Conference
on Computer Engineering and Technology

Session 5
 2009 International Conference on Computer Engineering and Technology
An Empirical approach of Delta Hedging in GARCH Model
Qian Chen
Statistics Department
Columbia University
New York City, U.S.A
qc2118@columbia.edu
Abstract—In this paper, we compare the costs and efficiency
arising from the delta-neutral dynamic hedging of options,
using two possible values for the delta of the option. The first
one is the traditional Black-Scholes delta, while the second one
is the GARCH option pricing delta, namely the delta of the
option in the generalized Black-Scholes model with a volatility
calibrated from GARCH model. Both our theoretical analysis
and empirical results of S&P 500 index options showed that
single-factor version of the GARCH delta showed better
performance than B-S delta (1973), especially when the
volatility are very high. We believe the improvement is due
largely to the ability of the GARCH model to describe the
correlation of volatility with spot returns. This allows the
GARCH model to capture strike-price biases in the
Black-Scholes model that give rise to the skew in implied
volatilities in the index options market.
Keywords:Delta hedging ; Garch; Time Series;Hedging Cost
I. BACKGROUND OF DELTA HEDGING IN TRADITIONAL
BLACK-SCHOLES
1.1 Black-Scholes Model
In option market, Black-Scholes gave out exact price for
the Vanilla options. It states that under perfect market
conditions and Geometric Brownian motion dynamics, the
only arbitrage-free time-t price of a strike-K expiry-T
call-option is
Call ( t ) = BS call
( S ( t ), T − t , K , r , σ )
where S(t) is the time-t price of a dividend-free stock, r is
the risk-free rate, σ is volatility (i.e. the standard deviation
of appropriately time-scaled returns), and the function is
given by B
− rτ
BS call
( S ,τ , K , r , σ ) = SN ( d1 ( S ,τ , K , r , σ )) − e
where N denotes the standard normal distribution
function and
ln(S / K ) + (r + σ 2 / 2)τ
d1 = , d 2 = d1 − σ τ
σ τ
1.2 Delta hedging in traditional Black-Scholes Model and
hedging Costs Analysis
Using BS model for hedging, however, may prove to be
very misleading. This is due to the misspecification of the
volatility smile which observed in capital market that the
option’s implied volatility depends upon the options’
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.226
Chengzhe Bai
Statistics Department
Columbia University
New York City, U.S.A
cb2549@columbia.edu
moneyness and time-to-maturity, which assumed constant in
the Black-Scholes model. That’s where local vol model
comes in to calibrate vol to the market.
A central quantity for hedging and risk-management is
the option’s sensitivity to changes in the stock-price; its
delta. The delta of an option is defined as the rate of change
of the option price with respect to the rate of change of the
underlying price. It is the number of units of the stock we
should hold for each option shorted to create a riskless
hedge. Mathematically, it is the partial derivative of the
option price with respect to the underlying price.
∂BS call
Δ=
∂S
An Attempting way to a show this is to ignore/forget that
S enters inside the N(…)-expressions which makes the
differentiation very easy: Δ=N(d1)
Delta can be used to hedge a holding of the derivative
with a position in the underlying security or vice-versa which
is named as Delta hedging. When implementing a delta
hedging strategy, an investor issuing an option buys a certain
amount of the underlying so that the overall portfolio,
compounded by the hedged and the hedging items, is
insensitive to changes in the underlying. The portfolio has
delta equals to zero by construction and it is said to be delta
neutral. As time passes, the portfolio tends to lose its delta
neutrality[1]. This in principle could be maintained revising
the portfolio continuously. However portfolio managers also
have an incentive not to hedge their risks, namely the cost of
hedging. Only if the cost of hedging for the option is lower
than the payoff from shorting the option, hedging increases
the profit[2].
1.3 The position to hedge
KN (d 2 ( S .τ , K , r , σ )) In this paper, the position we try to hedge is a short
position in a European style call option contract. A European
call option is the right to buy a stock at a given price: the
strike price of the option. This right can only be exercised on
its expiration date, this in contradiction to an American stock
option, where the right to buy can be exercised at any given
time before or on the expiration day. The short position
means that I ‘write’ the option contract and sell it to someone
else. This means we sell the right to buy. If the buyer decides
to exercise the option, I have to supply the share against the
strike price. In practice, the option is usually closed with a
cash-settlement. This means that I have to pay the difference
303
between stock price and the strike price if the stock price is
higher than the strike price.
A common way to hedge a short position in a European
call option contract is by keeping exactly D stocks. This
results in a portfolio –c + DS. For this portfolio,
∂∏ ∂c ∂S ∂c ∂c ∂S
=− +Δ =− + =0
∂S ∂S ∂S ∂S ∂S ∂S
Because the price of the portfolio is insensitive to change
of the stock price, as long as the amount of shares is adjusted
to D, the portfolio is riskless.
1.4 Profit and Loss analysis in traditional Black-Scholes
model
If it would be possible to adjust the amount of stock
continuously to the changes in the option delta, the hedge
would be perfect, i.e. the risk would be reduced to zero
without any costs. Because of the practical impossibility of
continuous hedging and the presence of transaction costs, the
hedge has to be adjusted with a certain time interval, e.g.
daily.
Adjusting the portfolio with a certain delay always
results in making hedging costs, with a bigger delay leading
to more costs. This is because if the price of the stock has
dropped, the more time goes by before one sells shares to
adjust the portfolio, the less money one receives. If the price
goes up, the opposite effect occurs and one will buy too
expensive. Therefore a proper estimation of volatility is a
crucial point to delta hedge.
During the traditional BS model, the profit and loss of a
portfolio of a long stock and short call option can be
represented as:
1 2 δS
S Γ (∑ τ − ( ) 2 ) …..(4)
2
P & LBS _ delta = −δC + ΔδS =
2 S
where C, Δ, Γ represent the options’ price, delta, and
gamma respectively in the B-S model.
Lemma: In BS model, whether the profit and loss is
negative or positive depends on whether the realized
volatility ( ∂S ) with respect to the implied volatility σ.
S τ
When we have short positions in calls and hedge it with long
shares of stock/Index, we actually are short the gamma, so
we will have net loss if the realized volatility is higher than
the implied vol in the BS model. Similarly, we long call and
short stocks/Index to have long gamma, we will have net
profit if the realized volatility is smaller than the implied vol.
With the true data of S&P 500 index, we can show the profit
and loss of delta hedging during different volatility sceneries
as follows:
TABLE 1. DELTA HEDGING P&L ANALYSIS UNDER TRADITIONAL B-S
MODEL(ELIDED)
We can see most of the time, we get negative P&L for
the portfolio of short call and long index when the realized
volatility is much higher than the implied vol we use in the
BS to calculate the hedge ratio delta. Alternatively, we can
see that the BS delta did not provide very good hedging
when the volatility is high.
304
Therefore, in this paper, we introduce a new hedging
strategy, GARCH delta, in which the estimation of
tomorrow’s volatility is needed when computing the amount
of shares Δ which is needed for the hedge. Compared with
the traditional calculation based on the assumption of
constant volatility in the Black-Scholes world while
delta-hedging a European call option, the new strategy
results in a smaller hedging error and therefore reduces the
hedging cost.
II. GARCH OPTION PRICING DELTA AND PROFIT &
LOSS ANALYSIS
2.1 Local Vol and P&L Analysis
Since we want use the same framework of local vol
model and local delta hedging for our GARCH delta
hedging. We want discuss the local vol model first. In local
vol model, Dupire [3] equation was widely used to calibrate
the local volatility to capture the volatility smile in the equity
market and overcome the limitation of BS delta hedging. In
Dupire equation, the local volatility is an unknown function
of time and stock. The calibration problems is the inverse
problem that consists inferring the function form the
market-quoted prices of liquid options, typically European
vanilla call and put options with various strikes and
maturities. Subsequently, the local volatility function thus
calibrate can be used to price exotic options or value
hedge-ratios consistently with the market.
Δ = Δlocal = ∂ S ∏ (t , S ;σ ) …………(1)
T ,K
∂C ∂C
+ rK
σ (K ,T ) = ∂T ∂K …………..(2)
1 2 ∂ 2C
K
2 ∂K 2
We can basically calibrate local vol in trinomial tree
specifically as follows, at time 0, we can get:
2Cal0 ( s j , t n )
σ (s j ,t n ) = …………. (3)
s 2j Fly0 ( s j , t n )
C0 (s j , t n+1 ) − C0 (s j , t n )
where: Calidar...spread...Cal0 (s j , t n ) =
Δt
C0 (s j − Δs, tn ) − 2C0 (s j,tn ) + C0 (s j + Δs, tn )
...spread...Fly0 (s j ,tn ) =
Butterfly
Δt
After we get the local vol value from equation (3), we
can input it with today’s stock price and time into equation
(1) to calculate the delta, which called local delta.
Similarly to BS P&L analysis, for the local volatility
model, the profit and loss is:
1 ∂S
P & LLocal_ delta = −δC + ΔlocalδS = S 2Γlocal(σ(t, S)2τ − ( )2 ) ..(5)
2 S
Compared with equation (5) and (6), we can get the
relationship of the two delta-hedging as:
P & LBS _ delta − P & Llocal _ delta = (ΔBS − ΔLocal )δS ……….(6)
In local vol models, it is well knows that the implied
volatility can be interpreted as an average of the local
volatilities on the most likely path s between (t,S) and (T,K).
Therefore, Coleman [19,p.9] proved that the following proxy
for the local delta:
ΔLocal = ΔBS + v BS ∂ K ∑ …………………………….(7)
We know in the market the volatility is negative skew, so
(7) implies Δ ≤ ΔBS
Local
Lemma: in a negative skewed vol model, whether the
profit & loss of BS is larger or smaller than the Local
according to whether the S∂ is positive or negative.
2.2 GARCH Delta Hedging and P&L Analysis
For GARCH delta, we can basically use the same method
of local vol. But we prefer to use the vol predicated from the
GARCH model instead of the local vol calibrated from the
market. We believe the GARCH predicated vol should be
closer than the local vol in our equation (5), since it is based
on the theory of regression to minimize the Residual of sum
of squared error of predicated vol and realized vol. Therefore
we expect that the P&L of GARCH deltas should have
smaller variability than the local vol or BS vol, or we can see
GARCH delta provide a better hedging strategy in terms of
smaller variance of P&L of our portfolio. Replace the vol in
equation of (5) and (6) we can get the following two
equations for our GARCH delta:
1 ∂S
P & LGARCH_ delta = −δC + ΔGARCHδS = S 2ΓGARCH(σ
2
τ − ( ) 2 ) …(5)
GARCH
2 S
P&L BS _ delta
−P&L GARCH _ delta
= (Δ BS
− Δ GARCH )δ S …...(6)
We can get the following table 2 to show the best delta
hedging strategy according to our analysis.
BS delta or GARCH delta:
1) In a Negatively skewed vol model, given a moderately
small rebalancing time interval, such as one day, the
GARCH delta provides a better hedge in a slow rally or a
fast sell-off, while the BS delta may provide a better hedge in
a fast rally or a slow sell-off.
2) In a Positively skewed vol model, given a moderately
small rebalancing time interval, such as one day, the BS delta
provides a better hedge in a slow rally or a fast sell-off, while
the GARCH delta may provide a better hedge in a slow rally
or a fast sell-off.
TABLE 2. OPTIMAL DELTA HEDGING STRATEGY IN NEGATIVE SKEWED
GARCH VOL MODEL
305
2.3 Market condition determination
We need three market conditions to determine which
delta hedging to use:
1. The daily Index price change: ;
2. The skewness of volatility model;
3. The volatility change speed or the GARCH predicated
volatility;
We can easily read the index price or stock price change
from the market, so it’s easy for us to determine the sign of
; for the skewness of local volatility model, we can
observe the market is highly negative skewed, which means
we can observe high frequent positive return and less
negative return, but some of the negative return could be
very huge, though with very low probability. Formally
testing the skewness could also be done as follows:
( x − u x )3
Skewness = E [ ]
σx3
When the value of skewness is larger than 3, we say the
distribution is positive skewed, alternatively, if it is less than
3, we say it is negatively skewed. For the volatility change
speed, it’s a little complicated, so basically we need to use
the GARCH model to make a judgment.
2.4 GARCH model and Volatility Change Speed
GARCH model [5] can be used to describe many
financial time series. It is a popular model for estimating
stochastic volatility and index return. It assumes that the
randomness of the variance process varies with the variance.
We adopt the computed GARCH volatilities into the
calculation of the option delta. We assume the GARCH
process to continue for the rest of the option’s life and find a
numerical way to approximate the option delta.
Under GARCH assumption we can get tomorrow’s
volatility from today’s volatility value.
Intuitively, under GARCH(1,1) model, the predicated
volatility is just the weight average of today’s predicated
volatility and today’s observed vol. To guarantee the stability
of the model, we should have the sum of the parameter alpha
and beta less than 1.
Our GARCH delta model based on one-step forecast of
the volatility. Suppose we have GARCH(1,1) model as
follows:
σ t2 = a0 + a1at2−1 + β1σ t2−1 + η ,0 ≤ a1, β1 ≤ 1, (a1 + β1 ) < 1
Assume that the forecast origin is h. For 1-step ahead
forecast, we have
σˆ h2+1 = a 0 + a1 a h2 + β 1σ h2
 2
Where a and σ are known at the time index h.
h h
2
Therefore, the 1-step ahead forecast is σ . The
h+1
strengths and weaknesses of GARCH models can easily
be seen by focusing on the simplest GARCH (1, 1)
2 2 2
model. First, a large a or σ gives rise to a large σ .
t−1 t−1 t
2
This means that a large a tends to be followed by
t−1
2
another large a , generating, again, the well-known
t
behavior of volatility clustering in financial time series.
Second, the model encounters the same weaknesses as
the ARCH model. For instance, it responds equally to
positive and negative shocks. In addition, recent
empirical studies of high frequency financial time series
indicate that the tail behavior of GARCH models remains
too short even with standardized Student-t innovations.
So we basically fit the realized volatility in our GARCH
model and predict the next day volatility and compared with
the BS vol. In this way, we can determine whether the
market change is speed or slow finally using the following
rules:
Fast : σˆ tGARCH
+1 ≥ σ tBS
Slow : σˆ tGARCH
+1 < σ tBS

III. DELTA HEDGING ANALYSIS IN REAL MARKET

The European call option we short sell has SP500 index
as its underlying. The reasons we chose stock index data in
stead of single stock data are:
Most similar research is done using stock index data and
therefore the result will be better comparable;
It is more realistic because most investors will hold a
diversified portfolio, similar to a major index;
The index pays no cash of stock dividends, has no stock 3.2 Model Selection
splits or secondary offerings.
The historical volatility (HV) is simply the standard GARCH(1,1),GARCH(1,2), GARCH(2,1) are commonly
deviation depending on selected sample window. It is used in volatility prediction in finance industry. We conduct
common market practice to use the HV estimator with a a set of adequate test to choose the most efficient model for
rather small data window, especially for the pricing of our volatility forecast.
options with short maturities. We find an estimation of HV Under the significance level 0.025, GARCH (1, 2) is
from a sample window for 20 trading days is convenient for rejected. Based on AIC and BIC data analysis, GARCH(2,1)
our trading strategy. The HV is computed from adjusted has the efficient variable selection, however the values of
closing prices of SP500. Furthermore we assume there is a AIC and BIC of GARCH(1,1) and GARCH(2,1) are not
constant risk-free interest rate of 4%. This is a realistic significantly different.
number and will not influence the results. Because we use TABLE 3. AKAIKE AND BAYESIAN INFORMATION CRITERIA
the same updating frequency (daily) for each of the models,
transaction costs will be roughly the same and are not taken GARCH(1,1) GARCH(1,2) GARCH(2,1)
into account. Then the delta is calculated daily using AIC -9.62E+03 -9.62E+03 -9.62E+03
Black-Sholes Equation.
BIC -9.60E+03 -9.60E+03 -9.59E+03
3.1 Eagle’s Arch Test As a result, although from the likelihood ration test with
In order to apply GARCH Model into our prediction, we alpha=0.025, however when alpha =0.05 ,GARCH(1,1) is
firstly conduct Eagle’s Arch Test to test if the time series of supported and The T Statistic column is the parameter value
return of S&P 500 are correlated. divided by the standard error, and is normally distributed for
large samples. The T-statistic measures the number of

306
standard deviations the parameter estimate is away from
zero, and, as a general rule, a T-statistic greater than 2 in
magnitude corresponds to approximately a 97.5 percent
confidence interval. The T-statistics in the table above imply
that the ARCH(2) parameter adds little if any explanatory
power to the model. On the other hand, from the comparison
of AIC and BIC values, since the two models are virtually
identical, thus, we accept the simpler restricted model, which
is essentially just the default GARCH(1,1) model.
Forecast:
3. 3 Best Delta Hedging Strategy under GARCH delta
hedging
According to our optimal hedging strategy in table 2, we
basically simplify our hedging strategy by combining the
stock change and vol change together. From the GARCH
model, we can get the confidence interval for the Index with
upper limit and lower limit. In our analysis, we consider the
situation of In-the-money call option with strike price
K=1550 and all the Index price is lower than K. If we use
such K value as guidance to determine our delta hedging
strategy, we can get the following basic rules:
If the predicated upper limit of the Index is higher than
K, we use the Delta+ to hedge; otherwise we just use the
Delta- to hedge. We can also sum.
TABLE 4. GARCH DELTA HEDGING STRATEGY
Upper Limit of Index Hedging Strategy
>K Delta +
307
<=K Delta -
Where: K=1550
With the new Delta Hedging Strategy, we can get the
following results:
TABLE 5. HEDGE COST COMPARISON BETWEEN TRADITIONAL MODEL &
GARCH FORECAST MODEL
Refer to the attachment for the detail calculation
IV. CONCLUSION
Black-Scholes model is the standardized model for
pricing the European options. However, using the BS for
hedging may prove to be very misleading due to the model
misspecification of the volatility, which is contradict to the
volatility smile[4] in equity market. So even with delta
neutral hedging strategy under the B-S model, we still will
have profit or loss due to the difference of realized volatility
and volatility we use in B-S model. Instead of using the local
volatility with calibration to option price, we use GARCH
delta with predicated volatility, which we believe should
have minimal error with volatility in B-S model. Therefore,
in theory we can get a better hedging performance than B-S
delta.
Empirical analysis on S&P 500 index options shows the
single-factor version of the GARCH model to be a
substantial improvement over the Black-Scholes (1973)
model. The GARCH model continues to substantially
outperform the Black-Scholes model even when the
Black-Scholes model is updated every period and uses
implied volatilities from option prices, while the parameters
of the GARCH model are held constant and volatility is
filtered from the history of asset prices.
REFERENCES
[1] Hua Zhang, An Empirical Investigation of GARCH Option Pricing
Model, Hedging Performance.
[2] Stéphane Crépey, Delta – hedging Vega Risk?, Université d’Evry, Jan
2004.
[3] B. Dupire, Pricing with a Smile, Risk, pp. 18-20, Jul. 1994.
[4] Emanuel Derman, Modeling the Volatility Smile, Columbia
University, 2006.
[5] Ruey S. Tsay, A course in Time Series Analysis, Second Edition,
University of Chicago, 2005.
 2009 International Conference on Computer Engineering and Technology
Multi-objective Parameter Optimization Technology for Business Process
Based on Genetic Algorithm
Bo Wang, Li Zhang, Yawei Tian
School of Computer Science and Engineering BUAA
Beijing,China
email_wangbo@yahoo.com.cn
Abstract- The idea of simulation analysis and multi-
objective evaluation was applied in parameter
optimization for business process. By analyzing the
parameters in business, four mainly optimal
parameters and their corresponding optimization
problems were determined. Some indexes for
synthetically evaluating the performance of process
were given, which could provide choice criteria for
optimization. Aiming at the characteristics of multi-
parameter and multi-objective of business process,
a multi-layer iterate optimization method based on
Genetic Algorithm was provided, which layered
disposal solution space according to the type of
optimal parameters and could not only enhance the
flexibility of selecting parameters but also avoid the
analysis for invalid parameter combinations. The
simulation results proved that this method could
supply balance and reasonable parameter
configuration scheme for enterprise decision-
maker.
Keywords- business process; multi-objective
optimization; parameter optimization; Genetic
Algorithm
I. Introduction
Enterprise business process optimization goal is to
adapt to the needs of the market competition,
significantly improve and enhance the competitiveness
of enterprises elements. Optimization content is
divided into structural optimization and parametric
optimization.[1] The current study of parameter
optimization focuses on resource scheduling and
quantity[1-5], but due to the lack of analysis and
confirmation of the optimal parameters, integrated
optimization is rarely seen which is aimed at the
implementation of activities, the import rate of source
products as well as the combinations of scheduling
strategy etc.
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 308
DOI 10.1109/ICCET.2009.22
On the other hand, the optimization of business
processes need to refer to the running time, cost,
resource utilization, queue length and so on a number
of performance indexes, the comprehensive evaluation
results of these related indexes is not a single solution,
but a group of balanced solutions which is the so-
called Pareto solution. Currently in the field of
business model parameters optimization, most of the
optimization methods[6-8] follow to a fixed technology
roadmap, that is, using countermeasures balance
principle to weigh on the relative importance of the
sub-goal and then composing to a single goal,
therefore, the final solution is over-dependent on user
preferences, not propitious to the synthetically
evaluation for optimization.
To deal with the problems above, we analyze the
optimal parameters based on both static and dynamic
configuration parameters and quantify four types of
process performance evaluation index calculated
according to simulation result. Finally, a multi-
objective parameter optimization method based on
genetic algorithm is promoted.
II. Selection of optimal parameters
The parameter optimization process of business
process is to simulate and analyze the model instances
with different configuration parameters, and select the
better after evaluate their performance indexes. So
determining the optimal parameters is a prerequisite to
the optimization. According to their effect, parameters
in business process model can be divided into the
following two categories:
1) Static configuration parameters, which can
directly describe some kind attributes of process
elements such as activities, products, resources etc, e.g.
running time of activities ,allocation of resources, as
well as the number of available resources for the
process and so on;
2) Dynamic configuration parameters, which are
provided for the simulation and can be used to reflect
the dynamic characteristics of business process
instances, such as the limit of simulation running time
and the followed resource strategy during simulation
and so on.
Among these, only the parameters that can be set up
by modeler in the modeling stage, directly controlled
by analyst in the phase of simulation the process and
have a decisive impact on performance can be selected
as decision-making variables, namely optimal
parameters.
A. Static configuration parameters
1) Import rate of source products.
Products in business process are divided into
source products, intermediates products and the final
products. Intermediate and final products are exported
by activities, the impact to the results of the simulation
process is determined by the implementation of related
activities, which means their properties belonging state
variables. On the other hand, source products as enter
to the process, are divided into single-source products
and multi-source products, [9] and the import rate of the
latter is subordinated to the probability distribution set
by modeler. During the simulation, distribution
parameters would affect the implementation of follow-
up activities and the control of the waiting queue for
the product. So import rate of multi-source products
belong to optimal parameters, and in the optimization
process the different probability distributions can
constitute different process instances.
2) Quantity of resources
The main attributes of resources include the
quantity, efficiency and cost etc [9]. The latter two are
objective reflection of resource state and belong to the
state variables. The quantity of resources is usually set
as a range and different quantity can be configured to
generate different process instances. Therefore, the
quantity of resources belong to optimal parameters,
and optimizing it can make a balance between waiting
and idle resources, and make the utilization rate of
resources and the cost of the process satisfy users. That
is, it can solve the rationality problem of distributing
resources for the enterprise.
3) Allocation of resources required for activities
implementation
During modeling, allocation of resources and
corresponding configuration of duration for activities
are set according to experience or work plan.
Allocation of resources not only determines the
duration of activities and the running time of process,
but it will also affect the use of related resources.
Therefore, allocation of resources required for
activities implementation belongs to optimal
parameters. Optimizing the allocation leads to
309
resources effective use under the condition of limited
resources.
In addition, in order to quantitatively descript the
non-linear relationship between duration of activities
and the allocation of resources, which can facilitate the
scheme selecting in the latter part of analysis and
optimization, we defined the Activities Implementation
Matrix. Its concrete form is as follow:
⎛ RC RC21
T
RCn1 ⎞ , n is the number of
AES = ⎜⎜ 11 ⎟⎟
⎝ DT12 DT22 DTn 2 ⎠
resource allocation scheme for the activities
implementation; the resource allocation vector
RCi1=(rc1 … rcm)is the allocation scheme of i type
resource; m is the largest number of types among
different resources involved in all schemes, rcj is the
number of resources required for the j kind
implementation of the resources allocation; the
duration vector DTi2=(dt1 dt2 dt3) is the activity
duration under the ith resources allocation; dt1 is the
probability distribution type of duration, dt2, dt3 are
corresponding parameters.
B. Dynamic configuration parameters
Business process scheduling strategy generally
adopted the combination of a variety of scheduling
rules (such as Highest Priority First Service, First
Come First Service). The various rules of the
combination need to have a certain order of priorities.
Using different combinations will make the simulation
results be various and impact the parallelism of
activities as well as the utilization of resource.
Therefore, selecting the appropriate combination of
rules with reasonable order to make the results of
simulation analysis optimal is just the target of strategy
scheduling optimization.
III. Optimization evaluation indexes
The implementation of the business process is a
dynamic process, therefore evaluation is mainly
focused on the dynamic indexes during the running of
business process, time, cost, resource utilization as
well as products waiting queue are the top four aspects.
So, we promote the following optimization indexes:
1) Utility of performance time PTU
1 k ci
PTU = ∑ ∑ ADurationij
k i =1 j =1
2) Utility of performance cost PCU
m RNum j
PCU = ∑ ∑ ( RCost
i =1 j =1
ij × RUsedTime ij )
3) Utility of resource utilization RUU
1 n m
RUU =
m*n
∑ ∑ RUNum
i =1 j =1
( t i , j ) / RNum j
 4) Utility of product queue PHU
1 n m
PHU = ∑∑ PQueueLen( j, ti )
m * n i =1 j =1
ADurationij is the running time of each activity; k
is the number of final products, ci is the number of
activities on the path; m is the number of resource
types, RNumi is the number of the i th type resource;
RCostij is the cost of resource; RUusedTimeij is the
actual used time of the resource; n indicates that the
entire project cycle or running time of the process is
divided into n equal parts; RUNum(ti,j) indicates the
quantity of resources of the j th type and its actual
amount being used at the time of ti; PQueueLen(j,ti) is
the number of the j th type non-final products at the
time of tj.
Some of these indexes are relevant, e.g. the cost
will be higher as the time becomes longer if the
quantity of resources is certain. But the time is often
saved at the cost of increasing the number of resources
associated, and the costs may not reduce.

IV. MLIO-BGA
Drawn from the above analysis, business process
parameters optimization has the characteristics of
multi-parameter and multi-objective. As the modern
enterprise is a complex social system, the number of
optional programs of parameters configure (the size of
the issue space) is huge in the process of optimization.
With some relevance among the evaluation indexes,
some optimal solutions maybe missed if simply
convert multi-objects into a single one under measures
balance principle. Therefore we promote a MLIO-BGA
(Multi-Layer Iterate Optimization Based on Genetic
Algorithm) method to solve it. Details are as follows:
A. Multi-layer iterate algorithm flow
MLIO-BGA uses bottom-up strategy, that is, 1) Through the dynamic creation of population to
according to the type of optimal parameters divide the make sure the efficiency of the individuals of the
main part of optimization into multilayer genetic population
iteration, and each layer optimize one type of 2) Increase flexible approach of the method
parameters using genetic algorithms. The optimal implementation: increasing or decreasing
configuration of all the parameters comes after the corresponding levels of iteration can adapt to the
highest level iterative end. Figure 1 is the optimization changes in the number of type parameters
process of Stream-like process [1], the process without B. Comparison between process instances based on
the layer of multi-source products import rate Weighted & Superior Relationship
optimization is for Project-type process [1].
The core idea of MLIO-BGA is layered disposal Suppose that the weight vector set for each
k
performance indexes is ω = (ω1 , ω 2 , ω k ) ,
according to the type of optimal parameters. The aim is
to determine the solution space of the lower level
∑ω
i =1
i =1 ,

inherit iteration dynamically according to the scheme
of parameters configuration expressed by each
individual in higher level inherit iteration. The main
advantages of MLIO-BGA are as follows:
and real number ε meets the condition 0 <ε ≤ 1. After
simulation analysis, we got a k-dimensional real vector
Sim(X) made up by k items of performance indexes,
that is Sin(X)=(x1, … xk). Then, instance X is(ω, ε)
superior than instance Y(recorded as X εω Y ) iff:

310
 k method for business process based on genetic
∑ ω ( I ≥ ( x , y )) ≥ ε
i =1
i i i algorithm. Practice has proved that it can provide
x≥ y
Where ( I ≥ ( x, y )) = ⎧
1 user. In the future, we will learn from other evaluation
⎨
⎩0 x< y
C. Calculate fitness
Suppose that the solution space is divided into k
layers, contemporary population is Pop which is also at
the m th layer, the outside-advantage set is R, the size
of Pop is N, the size of R is M, and Process (a) means
the instance created according to the combination of
individual a which is in the lowest layer population
and its upper individuals. The method for the
assignment is:
1) If m = k, then Pop is the lowest population (Step 7):
i) Con=Pop∪R,n = 0;
ii) Get individual xi in Con in turn, and compare it
with each individual yj (1 ≤ i,j ≤ N + M) in Con
based on the (ω, ε) relationship. When
Pr ocess( xi ) εω Pr ocess( y j ) , set n = n +1;
iii) fitness( xi ) = n ( N + M + 1) (1)
2) If m ≠ k (step 10):
i) Con=Pop∪R, if x∈Con, Bottom (x) is the set
made up by individuals in x’s solution set, and
Num (x) is the number of elements in the Bottom
(x);
ii) Get individual xi in Con in turn:
a) n = 0;
b) Get individual p in Bottom (xi) in turn, and
compare it with each element q in
Bottom(x1 ) ∪ Bottom(x2 ) ∪ Bottom(xM + N )
based on the (ω, ε) relationship. When
Pr ocess( p) εω Pr ocess(q ) , set n = n +1;
c) fitness ( xi ) = n ( Num + 1) (2)
N +M
Where Num = ∑ Num( x
j =1
j )
As n in formulae (1) and (2) increasing in degrees
along with the correlative advantage or disadvantage
relationship, it is certain that the better individual is
given a greater fitness value, so as to enhance its
probability to be chosen later.
V. Conclusion
This paper analyzes four types of optimal parameters
in business process those are input rate of source products
quantity of resources, allocation of resources required for
activities implementation and scheduling strategy. as
well as. It also gives a number of evaluation indexes
for process performance. With this understanding, we
promote a multi-objective parameter optimization
311
balanced and reasonable parameters configuration for
methods to further improve the evaluation system and
take full advantage of collateral calculation of genetic
algorithm to further enhance the efficiency of the
optimization.
ACKNOWLEDGMENT
This paper is supported by National High
Technology Research and Development Program of
China (863 Program, Grant No. 2006AA04Z165), PR
China.
REFERENCES
[1] Tan Wenan. A study and development for Dynamic
Optimizing Enterprise Process Technique and its Supporting
Environment. Thesis of Doctor Degree, BUAA, 2001. （in
Chinese）
[2] Tiwari, A. Vergidis, K. Majeed, B. . Evolutionary Multi-
objective Optimization of Business Processes. IEEE
Congress on Evolutionary Computation. 2006
[3] Zheng Yan, Lan Boxiong, Wang Wei. Aggregate
modeling for enterprise resource optimization. Computer
Integrated Manufacturing Systems, 2006,12(10):1561-1569.
（in Chinese）
[4] Redell O.,Martin T. Exact best-case response time
analysis of fixed priority scheduled tasks. In Proc. of the
IEEE 14th ECRTS. 2002. 165-172.
[5] Jiang Sijie, Xu Xiaofei, Li Quanlong. An Optimal
Algorithm for A Class of Multi-resource Scheduling
Problems. High Technology Letters, 2003, 13(1):76-80. （in
Chinese）
[6] Tan Wenan, Zhou BoshengResearch, Li Shuming, Wang
Qiang. Research on a Simulation-Based Auto-Optimized
Technique in Enterprise Process Model. Journal of Software,
2002,13(4):706-712. （in Chinese）
[7] Kacem I., Hammadi S.,Borne P. Approach by localization
and multiobjective evolutionary optimization for flexible job-
shop scheduling problems. IEEE Transaction on Systems,
Man, and Cybernetics, Part C,2002, 32:1-13.
[8] Chen Cheng, GU Yuqing. A Goal-Based Approach for
Process Instance Evolution. Journal of Software,2002,
13(8):1345-1351.
[9] Zhou Bosheng, Zhang Sheying. Visual Process Modeling
Language. Journal of Software, 1997, 8(z1):535-545. （ in
Chinese）
 2009 International Conference on Computer Engineering and Technology
Analysis and Design of an Access Control Model Based on Credibility*
Chang Chaowen1,2, Wang Yuqiao2 , Liu Chen2
1. School of Electronic and Information Engineering, Xi’an Jiaotong University, Xi'an, China
2. Institute of Electronic Technology, Information Engineering University,Zhengzhou,China
ccw@xdja.com
Abstract
An access control model based on credibility is
proposed. The Community Authorization Service (CAS)
is an authentication technology which allows service
providers to delegate some policy authority to the
virtual organization. While in open distributed systems,
provider join or leave the domain frequently, so
access-control policy in the server changes quickly. To
reduce the server load, threshold value of credibility
and a weight factor are applied in the model, and use
Analytic Hierarchy Process (AHP) to analyze the
weight of every rule. Providers adjust the threshold
value and weight factor instead of changing policies in
the server. First the server authenticates the requestor
and gives him credibility, then synthesizes the
credibility and trustworthiness. If the synthesis of
credibility achieves the threshold value, the server
gives him a certificate. The authentication policies and
trustworthiness are both used in the model, thus
flexibility and robustness are the advantages of this
model.
1. Introduction
Access control is a technology on the basis of
identification [1]. It can authorize qualified users to
access resources. According to the structure among the
users, the server and resources, there are three models:
“push” model, “pull” model, “agent” model [2]. As
shown in Figure 1:
*
This work was supported by National High Technology Research and Development Program of China (863 Program, No.2007AA01Z479)
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.87
Traditional access control mechanisms are
centralized and operate under a closed world
assumption in which all of the parties are known. In
open distributed systems, the user entities have a lot of
autonomy and arbitrariness. In many cases, the service
provider and the service requestor are strangers and do
not have a preestablished trust relationship. The
service provider must be able to authenticate the
unfamiliar users and then determine whether the
requestors have enough privileges to access the service.
Thus the access control model in open distributed
system must solve these problems: 1) Authenticate a
stranger who wants to access the resources, 2) Because
the users in open system join or leave the domain more
often, the strategy must be flexible.
In this paper we present an access control model
based on credibility. On the basis of CAS model, there
are two aspects of the improvement: 1) Use the
concept of trust threshold value. The service providers
register their threshold value of trust on the servers. If
a requestor's trustworthiness achieves this value in the
server, he can access to the service provider. 2)
Combine trustworthiness and authentication together.
Most access control models do not really combine the
two factors together. In our model, the server has a
database which contains the authenticate rules. When
the server authenticated the requestor, it gives a value
(credibility) to the requestor. The value is depended on
the rules it passed. Then a module will synthesize the
value and requestor’s trustworthiness as a final
credibility.
This paper is structured as follows: in Section 2, the
background and the related work is introduced. In
Section 3, an access control mode based on credibility
is described and the simulation model that improves
the control is presented. In Section 4, gives a method
to calculate the weight of the authenticate rules. Finally,
summarizes the work about this article and discuss
future work.
312
2. Related Work
Kerberos [3] is an authentication system that uses a
central server and a set of secret key protocols to
authenticate requestors and give both requestor and
application server a secret key for use in protecting
further communications. Firstly, the requestor
authentication is to the Key Distribution Center (KDC),
which gives it a Ticket Granting Ticket; this step
occurs infrequently. For each service the requestor
needs to contact, it must then contact the Ticket
Granting Service (TGS) which responds with a Ticket
(TKT) that is service-specific. The requestor then
contacts the service, providing the TKT. Often, the
KDC and the TGS are collocated. However, it is
possible to replicate the KDC and the TGS in order to
handle large numbers of users.
The approach taken by Firmato is the use of a
network grouping language which is customized for
each managed firewall at that firewall [4]. The language
used is independent of the firewall and routers used,
but it is limited to packet filtering. Policy updates are
equivalent to policy initializations in that they require a
reloading of all the rules on the affected enforcement
points. Finally, the entire relevant policy rule-set must
be available at an enforcement point; this may cause
scalability problems with respect to the number of
users and policy entries.
Role-based Access Control which is called RBAC [5]
has become a classical model. It reduces the
complexity and cost of security administration in large
scale network. Each user is assigned one or more roles,
and each role is assigned one or more privileges that
are permitted to users in the role. Although RBAC
conceptually allows for many different models and
implementations for management and policy
specification, none of the RBAC-based systems we are
aware of use flexible credentials, allow credential
composition from different administrative front-ends,
or decentralized management [6].
Community Authorization Service (CAS) is a
security mechanism based on the GSI Group
authorization service which is designed by Globus
project team [7]. It assumes that all members trust the
CAS server in the virtual organization, and the CAS
server manages the strategies of access control. CAS
server has a server access control strategy, resource list,
access control strategy, organization members list, and
so on. CAS is a “push” model. In an open system,
because the members join or leave frequently, resulting
in the CAS server access control strategy has been
changed quickly, and this costs the server a lot of time.
313
3. An Access Control Model Based on
Credibility
The definition: There are a lot of rules in the
authentication mechanisms, due to the different
degrees of importance; we need a mechanism to
measure it. In this paper, we use authentication
trustworthiness to describe the differences [8].
Definitions1. Trustworthiness: The quantitative
denotation of trust level relied on the integrity, ability,
or character of a person or thing. Using T to symbolize
this, T ∈[0,1]. T 0 symbolizes the initial value of the
users. If T = 0, it shows that the user is completely
discredited, if T = 1 that stands for a fully credible
user.
Definitions2. Authentication Certainty Factor A(u).
Figure the credibility about someone who has passed
the authentication. Using A(u) to symbolize this,
A(u)∈[0,1] .
Definitions3. Credibility Threshold Value λ , Stand
for a value that an authenticated requestor’s A(u) must
be over, λ∈[0,1].
Definitions4. Weight factor of authentication α .
When the user passes the authentication; he will
acquire the authentication Certainty Factor A(u), then
the server will multiply A(u) by α, the value of α is
decided by the service provider.
In the CAS model, policy and trust certificates are
kept in a centralized repository. If the service provider
join or leave the domain, the policy must be changed
either [9] [10]. The cost is usually very high in an open
distributed system. The improved model can reduce
the cost by using trust threshold value and weight
factor which can easily influence the policy. First, the
resource providers register λ & α in the server. If
anyone wants to gain access to the resources, he must
be authenticated by the server. Second, in the
improved model, both the A(u) (Authentication
Certainty Factor) and T (the user's trustworthiness) will
be considered as the reference. The terminal
trustworthiness of the requestor can be calculated by
this formula: T = αA(u) + (1 - α) T 0. This combination
of trust and certification mechanism can enhance the
security and reduce risks. The improved model
consists of three parts: the request for service (users),
CAS Server, the provider of resources (other users).
CAS Server includes rules of the certification system,
integrated modules of trust, list of resource providers
and trust threshold value λ.
The structure of the model is shown in Figure 2:
 Software OS: Window XP with SP2；
The experiment simulates downloading files. The
purpose is to compare the influence of the threshold
value. In this experiment, suppose there is one service
provider meeting the request. A total of 50 users send
their requests. 50 users were divided into five groups,
each contains 10 users, and the trustworthiness was
0.90, 0.80, 0.70, 0.60, and 0.50. λ = 0.55.
Results are shown in Figure3. X-axis symbolizes
the number of the requestors. Y-axis symbolizes the
success ratio of download services. Red curve stands
for the model using threshold value, while blue is not
adopting the strategy.

Following presents the step-by-step processing of a

request by this architecture:
Step1 Service provider registered services in the
CAS server and submitted their authentication weight
factor, then initialization trust threshold value λ.
Step2 When a user wants to request service; he can
query in the server and find the provider. If he chooses
a provider, he must put his request and certificate to
the server.
Step3 Consistency verification will verify the
signature first, and then it will check the legality and
validity of the certificate. If it is legality and validity, From Figure3 we can see that, when the number of
consistency verification sends certificate to the rules of users increases, both success ratios decreased
the certification system. gradually; if we use the strategy of threshold value, the
Step4 The certification system calculates the red curve declines more slowly, while the blue curve
trustworthiness of the requestor by T = αA(u)+(1-α) T 0. which is not adopting threshold value declines quickly.
If T≥λ ,the server gives a permission to the requestor. The result shows that when we use trust threshold
Otherwise, refuse to authorize the request. value, the success rate of the downloading will be
Step5 Requestor generates a new certificate and improved.
embeds the permission in the expansion of the
certificate, and then submits it to the service provider. 4. Calculating the Weight Ratio of the
The provider confirms the certificate issued by CAS
Rules
server, and determines the privileges according to his
local security strategies.
The authenticate rules is stored in the database of
Analysis and simulate model.
the server. Some rules are more important than others.
Because the authentication system contains a lot of
How to distinguish their differences? We use Analytic
method, such as the identification of ID card, password
Hierarchy Process (AHP) to analyze the weight of
authentication, fingerprint recognition, and so on, we
every rule [11]. The AHP is about breaking a problem
should describe the difference by weight ratio.
down and then aggregating the solutions of all the sub
From the formula T = αA(u)+(1-α) T 0 , it is known
problems into a conclusion. It facilitates decision
that when α = 0, T = T 0, certification of the user by making by organizing perceptions, feelings, judgments,
CAS server becomes useless. Thus let α > 0 more and memories into a framework that exhibits the forces
often; when α = 1, T = A(u), it means that the service that influence a decision. Suppose that C1, , C2 ,…, C9,
provider will not consider of the user.
stand for the importance of rule E1, rule E2, …, rule E9.
Environment of the experiment
The following table shows their symbolization:
Hardware PC: Tsinghua Tongfang 3500 D; CPU:
Intel Pentium 4 processor speeds 2.66 GHz; memory
512 M;

314

degree αij meaning
1 Ci is as important as C j
3 Ci is more important than C j (a little)
5 Ci is more important than C j (a lot)
7 Ci is much more important than C j (a little)
the number of users, the trust threshold value. It is
difficult to simulate the actual work fully. Moreover,
when the CAS server updates the rules, the weight of
the rules is also changed. The process is dynamic,
maybe there are some unpredictable mistake. How to
set the threshold value is also a problem for the
providers. We hope that future access-control systems
will solve these problems.

9 Ci is much more important than C j (a lot) 6. References

[1] Adam Hess and Kent E. Seamons.An Access Control
2,4,6,8 The importance is between two levels above Model for Dynamic Client-Side Content. ACMAT’03, June
2-3, 2003.
1,1/2,…,1/9 Change “important” to “weak” [2] Rich Baker, Bob Cowles, etc, Conceptual Grid
Authorization Framework and Classification, 2003
[3] S. P. Miller, B. C. Neuman, J. I. Schiller,and J. H. Saltzer,
The weight ratio is calculated by constructing
Section E.2.1: Kerberos Authentication and Authorization
comparison matrix. Suppose there are 4 rules, every System,M.I.T. Project Athena, Cambridge,
element in the matrix stands for αij : Massachusetts(December 21, 1987).
⎡ 1 3 2 5⎤ [4] S. P. Miller, B. C. Neuman, J. I. Schiller,and J. H. Saltzer,
⎢1 / 3 1 1 / 2 2⎥⎥
Section E.2.1: Kerberos Authentication and Authorization
U =⎢
System,M.I.T. Project Athena, Cambridge, Massachusetts
⎢1 / 2 2 1 4⎥ (December 21, 1987).
⎢ ⎥ [5] D.F.Ferraiolo, D. R.Kuhn, and R.Chandramouli. Role-
⎣1 / 5 1 / 2 1 / 4 1⎦ Based Access Control Artech House, ISBN I-58053-370-
1,2003.
Construction of the comparison matrix as follows: [6] ANGELOS D. KEROMYTIS and JONATHAN
M.SMITH. Requirements for Scalable Access Control and
U C1 C2 C3 C4 W Security Management Architectures. ACM Transactions on
Internet Technology, Vol. 7, No. 2, Article 8, Publication,
C1 1 3 2 5 0.476 May 2007.
[7] Markus Lorch,Bob Cowles,Rich Baker,et al.Conceptual
C2 1/3 1 1/2 2 0.155 Grid Authorization Framework and Classification.Global
Grid Forum.2004:21-22.
C3 1/2 2 1 4 0.155
[8] Wang Lunwei,Liao Xiangke,and Wang Huaimin.
C4 1/5 1/2 1/4 1 0.081 Research on Authenticati0n Trustworthiness Theory. Journal
of Computer Research and Development,42(3):501 ～
506,2005. Vol.50,No.S1,181～ 184.Oct, 2004(in Chinese).
[9] E.Bertino and P.A.Bonatti.TRBAC: A Temporal Role-
λ = 4.022 ， puts λ and the rank n = 4 to the Based Access Control Model.ACM Transaction on
λ − n Information and System Security,2001 Vol.4(3):191-223.
formula
CI =
n − 1 ， calculating CI = 0.0073 . [10] Rich Baker, Bob Cowles, etc, Conceptual Grid
CI Authorization Framework and Classification, 2003.
CR = = 0 . 008
When n = 4 , the RI = 0.89 . < 0.1.
RI [11] W.K.Edwards. Policies and Roles in Collaborative
R R2 R 3 R 4 Applications. In proceeding of ACM Conference on
So the weight ratio of 1 , , , is 0.476,
0.155, 0.155, 0.081. Computer-Supported Cooperative Work. Cambridge,MA.
1996: 11-20.

5. Conclusions
Because the simulation model based on the
assumption that contains more pre-conditions, such as

315
 2009 International Conference on Computer Engineering and Technology
Modeling of Rainfall Prediction over Myanmar Using Polynomial Regression
Wint Thida Zaw, Thinn Thu Naing
University of Computer Studies
Yangon, Myanmar
wintthida@gmail.com, ucsy21@most.gov.mm
Abstract—Myanmar is an agricultural country and its
economy is largely based upon crop productivity. The
occurrence of extreme precipitation variability may lead to
significantly reduce crop yields and extensive crop losses.
Thus, rainfall prediction becomes an important issue in
Myanmar. Regression has since long been a major data
analytic tool in many scientific such as behavioral sciences,
social sciences, biological sciences, medical sciences,
psychometrics and econometrics for predicting. Multi
variables polynomial regression (MPR) is one of the statistical
regression method used to describe complex nonlinear input
output relationships. In this paper, MPR is applied to
implement the precipitation forecast model over Myanmar.
Myanmar receives its annual rainfall during the summer
monsoon season which starts in June and end in September.
The model output result is station wide monthly and annual
rainfall amount during summer monsoon season. The
proposed model results are compared with the result produced
by multiple linear regression model (MLR). From the
experimental results, it is observed that using MPR method
achieves closer agreement between actual and estimated
rainfall than using MLR
Keywords- statistical modeling, polynomial regression,
rainfall froecasting
I. INTRODUCTION
Weather forecasting has emerged recently where the
specific climate depending on geographical location is to be
predicted. An accurate monthly and annual forecast can help
farmers to increase agricultural productivity in good rainfall
years and by giving information to adopt drought resistant
crop varieties and appropriate crop, soil and water
management practices in anticipated drought years. Thus
rainfall forecasting becomes an important factor in the
domain of agriculture.
Regression technique is the most commonly used
statistical technique for prediction of rainfall. MPR provides
an effective way to describe complex nonlinear input output
relationship. We implement rainfall prediction model with
the use of empirical statistical technique, MPR.
We use 37 years (1970-2006) datasets of the global
climate data such as sea surface temperature (SST), India
ocean dipole (IOD), and southern oscillation index (SOI),
oceanic Nino index (ONI) and premonsoon month rainfall
amount over 17 Myanmar weather stations as local weather
data. There are 15 predictors in total.
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.157
Using these past regional and global weather data,
prediction model is built with the use of second-order MPR.
We implement monthly and annual rainfall forecast model
using second-order MPR based on 15 predictors. And we
find all possible subsets of 15 candidate predictors and then
choose the best combination of predictors offering the lowest
mean absolute error (MAE).
It is possible to forecast monthly and annual summer
monsoon rainfall amount (in mm) with acceptable accuracy
with one month ahead. The resulted rainfall amounts are
intended to help farmers in making decision concerning with
their crop.
II. RELATED WORK
Accurate and timely weather forecasting is a major
challenge for the scientific community. Rainfall prediction
modeling involves a combination of computer models,
observation and knowledge of trends and patterns. Using
these methods, reasonably accurate forecasts can be made
up. Several recent research studies have developed rainfall
prediction using different weather and climate forecasting
methods.
Regression is a statistical empirical technique and is
widely used in business, the social and behavioral sciences,
the biological sciences, climate prediction, and many other
areas.
N. Sen [1] has presented long-range summer monsoon
rainfall forecast model based on power regression technique
with the use of EiNino, Eurasian snow cover, north west
Europe temperature, Europe pressure gradient, 50 hPa Wind
pattern, Arabian sea SST, east Asia pressure and south
Indian ocean temperature in previous year. The experimental
results showed that the model error was 4%.
S. Nkrintra, et al. [2] described the development of a
statistical forecasting method for summer monsoon rainfall
(SMR) over Thailand using multiple linear regression and
local polynomial-based nonparametric approaches. SST, sea
level pressure (SLP), wind speed, EiNino southern
oscillation index (ENSO) and IOD were chosen as
predictors. The experiments indicated that the correlation
between observed and forecast rainfall was 0.6.
T. Sohn, et al. [3] has developed a prediction model for
the occurrence of heavy rain in South Korea using multiple
linear and logistics regression, decision tree and artificial
neural network. They used 45 synoptic factors generated by
the numerical model as potential predictors.
316
 M. T. Mebrhatu et al. [4] modeled for prediction where
categories of rainfall (below, above, normal) in the highlands §1 2
x11 x21 ... xn1 x11 x11x21 ... x11xn1 x212
x21x31 x21x41 ... x21xn1 ... xn21 ·
of Eritrea. The most influential predictor of rainfall amount ¨ ¸
¨1 x12 x22 ... xn2 x12 x12x22 ... x12xn2 x22 x22x32 x22x42 ... x22xn2 ... xn22 ¸
2 2

was the southern Indian ocean SST. Experimental results X ¨ ¸

showed that the hit rate for the model was 70%. ¨... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ¸
¨1 x1t x2t ... xnt x1t x1t x2t ... x1t xnt x2t x2t x3t x2t x4t ... x2t xnt ... xnt ¸¹
2 2 2
H. Hasani et al. [5] proposed human height prediction ©
model based on multiple polynomial regression that was (3)
used successfully to forecast the growth potentials of height YT  y1 y 2 ... yt 
(4)
with precision and was helpful in children growth study. ET E0 E1 ...
Vaccari et al. [6] modeled plant motion time-series and (5)
nutrient recovery data for advanced life support using multi The least square estimates of the regression parameters
variable polynomial regression. are calculated using equation (6).
The following sections present an approach to the Eˆ (X T
X ) 1 X T
Y (6)
development of rainfall forecasting system. Firstly, we ŷ
discuss polynomial regression in section 3. Next, we The least square prediction is calculated as follows.
describe how to forecast rainfall with the use of MPR in Yˆ XÊ (7)
section 4. We report some experimental results in section 5. n (8)
Finally, we conclude the paper in section 6. MAE 1
n ¦i 1
y i  yˆ i

III. POLYNOMIAL REGRESSION

Regression is a statistical empirical technique that utilizes
the relation between two or more quantitative variables on
observational database so that an outcome variable can be
predicted from the others.
Polynomial regression produces a polynomial describing
the relationship between any set of inputs and corresponding
output. Polynomial regression model which contains more
than two predictor variables is called MPR.
The polynomial model is called as second-order
polynomial model with n predictor variables if a model
which contains n predictors expressing with the first and
second powers of the predictors.
The polynomial regression can become numerically
unstable, especially if the degree is high or the function
domain is not centered at zero. If the polynomial degree is
too low, it will not be able to represent the complexity of the
function being learned [7].
Thus, we find the statistical relationship between rainfall
amount and other climatic data with the use of second-order
MPR equation which contains added terms and nonlinear
cross product interaction of n predictors expressing with the
first and second power of the predictors. This is in the form
of equation (1).
yt = 0 + 1x1t +...+ nxnt + n+1x1t2 + n+2 x1tx2t + …+ 2nx1txnt +
2n+1x2t2 + 2n+2x2tx3t +…+ 3n-1 x2txnt +…..+ xnt2+ e (1)
where yt represents the predictand, representing SMR at time
t. x1t, x2t… xnt are predictors representing premonsoon month
monthly mean temperature, monthly rainfall, IOD, SOI, SST
at time t. 1t, 2t, … are least square regression parameters
estimated from the observed data. e denotes the error that is
difference between predicted and actual precipitation
amount.
The second-order MPR model used in this study is in the
matrix form of
Y = X + e (2)
IV. RAINFALL FORECASTING USING MPR
The general processes of forecasting rainfall amount are
1. Collecting data
2. Building models using all possible combinations of
predictors
3. Choosing the best model
4. Validity check
The overview of rainfall forecasting system is
illustrated in Figure 1.
Data are collected first and then data are separated into
training data and test data.
The next step is the building second-order MPR models
with the use of training data. In this paper, we consider all
possible combinations of the predictors. If the number of
predictors chosen is n, there will be (2n-1) distinct
regressions models. We find the least squares estimates of Ê
from equation (6) over training data. We calculate the least
squares prediction ŷ from equation (7). This procedure is
repeated until all possible combinations of the predictors
have been examined.
In selecting the best model, MAE of each model is
calculated by using the equation (8). The predictive skill of
the models is ranked according to MAE. The models are
ranked in the ascending order of MAE values. The model
with the lowest values of MAE is ranked first and the model
with the highest value is ranked last. The first rank prediction
model is chosen as the best regression model which contains
small group of predictors that can give an optimal prediction.
Finally, the model built over training period is tested with
test data to verify how much accuracy the model can give.

317
 Now we will report experiments and give graphs for monthly
Build Choose rainfall amount during rainy season which starts from June and ends
Trainin Best Model
Data Model in September. The following experiment uses data over 1990-2005
as training data. Table 2 describes the predictors used in prediction
and the error over training period.
Validity
Check TABLE II. SELECTED MODEL SKILL FOR PATHEIN STATION OVER
TRAINING PERIOD
Test
Station Month MAE Predictors used
Figure 1. Overview of the forecasting system. 0.02
Pathein June P3, P5, P8, P12
July 4.48 P5, P6, P7, P11
V. EXPERIMENTAL RESULTS August 1.61 P5, P6, P8, P10
In this section, the prediction results will be reported. September 1.3 P2, P7, P9, P15
Due to the lack of space, now we will give graphs for
Pathein rain gauge station located in lower Myanmar where Figure 2 shows the comparison between actual and forecast of
rain fall prediction is needed more for agriculture planning June rainfall amount over training period 1990-2005 of Pathein rain
and management although we take and analysis the regional gauge station.
data from 17 rain gauge station over Myanmar.
For monthly precipitation forecast, we use regional
rainfall amount taken from this rain gauge station and large
scale data such as East India SST, SOI, ONI taken from
[8],[9],[10].The following table shows the details of the
predictors used for monthly rainfall prediction.
TABLE I. DETAIL OF PREDICTORS

Predictor Symbol Period Location

ONI P1 MAM 5oN-5oS 120o 170oW

ONI P2 AMJ 5oN-5oS 120o-170oW

IOD P3 April
90°E-110°E
10°S-0°N
IOD P4 May

East India
P5 April
SST 50°E 70°E 10°S-
10°N
East India
P6 May
SST
Figure 2. Comparisons between predicted and actual June rainfall amount
Previous for Pathein Station over taining period
SOI P7
two months 17.5°S 149.6°W,
12.4°S 130.9°E
SOI P8
Previous The model evaluation is performed using test data of
month 2006. The trained model is tested with the use of 2006 test
Previous data and the test result is compared to the actual rainfall
SST P9 15°-20° N 85°-90° E
month amount. Figure 3 illustrates 2006 monthly rainfall prediction
Previous amount compared with actual rainfall amount in June, July,
SST P10 10°-15° N 85°-90° E August and September at Pathein rain gauge station which is
month
the area of rice grown most. Figure 4 shows the test result in
Previous
SST P11
two months
5°-10° N 85°-90° E graphical form.
Previous
SST P12 5°-10° N 90°-95° E
month
Previous
SST P13 5°-10° N 95°-100° E
month
Previous
Temperature P14 Predicted Station
month
Previous
Rainfall P15 Predicted Station
month

318
 Figure 6 illustrate the comparisons of actual and forecast
annual SMR over training period.

4000

3500

3000

SMR Amount (in mm)

2500

2000

1500

1000

Figure 3. Comparison between actual and prediction Pathein monthly 500

rainfall amount with test data of 2006
0

90

91

92

93

94

95

96

97

98

99

00

01

02

03

04

05
19

19

19

19

19

19

19

19

19

19

20

20

20

20

20

20
Actual
Year
Model Forecast

1400 Figure 5. Comparisons between predicted SMR using second-order MPR

and Actual SMR for Bago station over training period
R a in fa ll Am o u n t (in m m )

1200
1000
800
600 4000
400 3500

S M R A m ou n t (in m m )
200 3000
0 2500
June July August September 2000
2006 Summer Monsoon Month 1500
Predicted Actual 1000
500
Figure 4. Comparisons between actual and predicted Pathein monthly 0
rainfall amount with test data of 2006

05
90
91
92
93
94
95
96
97
98
99
00
01
02
03
04
20
20
20
19
19
19
19
19
19
19
19
19
19
20
20
20
Actual
Year
For Pathein station, the prediction accuracy we got is Model Forecast
over 99% in training period and 93% with test data of 2006.
Myanmar receives its annual rainfall during summer
monsoon season which starts from June and ends in Figure 6. Comparisons between forecast SMR using MLR and actual
September. The annual SMR variability impacts on crop SMR for Bago station over training period
productivity. Thus we implement annual SMR prediction
model with the use of four predictors. The details of these
predictors are described in Table 3. The model evaluation is performed using test data of
2006. Table 4 describes the test results got from two
methods.
TABLE III. DETAIL OF PREDICTORS USED IN ANNUAL SMR FORECAST

Spatial Correlation TABLE IV. BAGO STATION TESTING RESULT FOR 2006 SMR
Symbol Predictors Period
Domain with SMR
Monthly Estimated
96°30'E Estimated
P1 Rainfall May -0.28582 Bago Station
17°20'N Actual Bago Bago
amount
2006 SMR
Monthly Station 2006 Station
96°30'E using
P2 Mean May
17°20'N
-0.21075 SMR 2006 SMR
Temperature second-
using MLR
50°E -70°E order MPR
10°S-10°N,
P3 Mean IOD JFM -0.2245 2806 mm 2870 mm
90°E-110°E
10°S-0°N 2795 mm (accuracy of (accuracy of
17.5°S, 99.6%) 97%)
149.6°W,
P4 SOI May -0.48543
12.4°S,
130.9°E With MPR, the estimated 2006 SMR at Bago Station is
2806 mm against actual of 2795 mm. The forecast for 2006
Now we will report and give graphs for the prediction of Bago Station SMR by MLR is 2870 mm.
annual SMR prediction over Bago station. Figure 5 and

319
 From these experimental results, it is observed that using
second-order MPR method achieves closer agreement
between actual and estimated rainfall than using MLR in the
prediction of annual SMR forecast over Bago rain gauge
station.
VI. CONCLUSION
Weather affects the economy, health and agriculture, not
only in Asia, but across the whole planet. Rainfall is a key
climatic determinant of agriculture production in Myanmar
which can affect the economy. We have developed rainfall
prediction model based on 15 predictors using second-order
MPR. As a result of several experiments, the predicted
rainfall amount is close to the actual values. We also
developed annual SMR prediction model with four
predictors only. The model results are intended to use in the
areas such as crop planting and yield, water management and
reservoir control. The main aim of the development of the
forecast model is to help in farm management and water
management.
In this implementation of MPR model, all possible
subsets of predictors have been examined. But we use only
test data of 2006. Thus, we can’t grantee that the best model
with the test case of 2006 which can offer the smallest
prediction error with the test data of the year except 2006.
Therefore, to get the robust model which can offer the
smallest prediction error for any year, test data of more than
one year can be used.
320
REFERENCES
[1] N. Sen, ” New forecast models for Indian south-west
Monsoon season Rainfall”, in Current Science, vol. 84, No.
10, pp.1290-1291, May 2003.
[2] S. Nkrintra, et al., “Seasonal Forecasting of Thailand Summer
Monsoon Rainfall”, in International Journal of Climatology,
Vol. 25, Issue 5, American Meteorological Society, pp. 649-
664, 2005.
[3] T. Sohn, J. H. Lee, S. H. Lee, C. S. Ryu, “Statistical
Prediction of Heavy Rain in South Korea”, in Advances in
Atmospheric Sciences, Vol. 22, No. 5, pp.703-710, 2005.
[4] M. T. Mebrhatu, M. Tsubo, S. Walker, “A Statistical Model
for Seasonal Rainfall Forecasting over the Highlands of
Eritrea”, in International Crop Science Organization.
[5] H. Hasani, M. Zokaei, A. Amidi,“A New Approach to
Polynomial Regression And Its Application to Physical
Growth of Human Height”.
[6] D.A. Vaccari, J. Levri, “Multivariable Empirical Modeling of
ALS Systems Using Polynomials,” in Life Support and
Biosphere Science,vol. 6 pp. 265-271,1999.
[7] D.Stronger, P.Stone, “Polynomial Regression with Automated
Degree: A Function Approximator for Autonomous Agents”,
Internaional Journal on Artificial Intelligence Tools, pp. 474-
480, November 2006.
[8] http://www.cru.uea.ac.uk/.
[9] http://www.cpc.noaa.gov/.
[10] http://www.jamstec.go.jp/
 2009 International Conference on Computer Engineering and Technology

New Similarity Measure for Restricted Floating Sensor Networks

Feng Yuan, Liu Xiaodong, Ding Xiangqian

College of Information Science and Technology
Ocean University of China
Qingdao, China
liuxiaodong.ouc@gmail.com

Abstract—Similarity-based search has been a hot research

topic for a long history, which is widely used in many
applications. The large scale Restricted Floating Sensor (RFS)
network is an important mod-el in offshore data collection [1].
This paper constructs a new fuzzy similarity measures based on
distance. This work provides a theoretical guidance for
constructing a fuzzy query processing strategy for our RFS
networks. From [2] we know our work over distance and
similarity is a reasonable generalization and extension of other
Fuzzy Sets.

Keywords-restricted floating sensor network; fuzzy set;

similarity
Figure 1. RFS sensor

I. INTRODUCTION
There are significant interests in analyzing the siltation of
the estuary and harbor. For example, H. H. Harbor, the second
largest harbor for coal transportation (6.7 million tons per
year) in China, currently suffers from the increasingly severe
pr-oblem of silt deposition along its sea route (19 nautical
miles long). The sea route has always been threatened by the
silt from the shallow sea area. Monitoring sea depth costs this
harbor more than 18 million US dollars per year [1]. In 2007,
Yunhao Liu proposed a Restricted Floating Sensor (RFS)
model [1]. By locating such sensors, the sea depth can be
estimated efficiently without the help of extra ranging devices.
Similarity-based technologies are important in RFS Figure 2. Our experiment site
networks: firstly, RFSs are deployed on the sea collecting data
of silt. The event detection can be considered as the pattern
number. Let V = {vij } . The topology is shown in Fig. 3. Let
search over time series data, which highly demands fast
retrieval to keep the safety of sea route. Thus the retrieval of S (vi0 j0 ) = {vij | vij ∈ V , i = i0 + 1, j ∈ J i0 j0 } denote the agg-
similar time series is important to RFSs [1]; secondly, RFS
suffers long propagation delay and high-energy consumption. regation of all child nodes of vi0 j0 in the RFS network, whe-
There need a scheme can effectively exploit the sharing re J i0 j0 contains all the sequence number of these nodes.
among queries to greatly reduce the communication cost. Thus
similarity-aware query techniques are important to RFSs [3]. Because of the movement of RFSs, the relation betw-
Our experiment site for the RFS network lies in the east een S (vi0 j0 ) and vi0 j ( j ∈ J i0 j0 ) is variable. For example
S (v12 ) can equal to {v22 , v23 , v24 } or {v23 , v24 } and
coastal area of Qingdao. RFS sensors and experiment site are
shown in Fig. 1 and Fig. 2. Distinguished from traditional stat-
ionary or mobile sensor networks, nodes in the RFS network v31 belongs to S (v21 ) or S (v22 ) . As we know fuzzy theory
will generally not be stationary at their original places, but
is efficient in treating imprecision. So in this paper we
float in a restricted area on the sea [1]. This quality poses
construct two general similarity measures based on distance
some unique challenges. For example, let vij denotes RFS of on VIFSs. Our work is a reasonable extension of those on
the network, where i stand for the layer, and j for the sequence Fuzzy Set (FS), Intuitionistic Fuzzy Set (IFS) and Interval-Va-

This project is funded by the Chinese National High Technology Research

and Development Plan (863 program) under Grant No. 2006AA09Z113, by
Key Project of Chinese National Programs for Fundamental Research and
Development (973 program) under Grant No. 2006CB303000.

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 321

DOI 10.1109/ICCET.2009.177
 monitoring [8], moving object tracking [9], and financial data
analysis [10]. Many measures of similarity between fuzzy sets
have been proposed and researched. Hyung et al. presented
two similarity measures between fuzzy sets and between
elements [11]. Then Chen also made a comparison of
similarity measures of fuzzy values [12]. Li et al. proposed
similarity measures of IFS and applied similarity measures to
pattern recognition [13]. Mitchell discussed that the Dengfeng
and Chuntian operator may give counter-intuitive results and
proposed a modification of the Dengfeng-Chuntian operator
[14]. Liang et al. reviewed some existing measures of
similarity and proved these measures are not effective in some
cases. Then they proposed some new similarity measures and
discussed their relationship [15]. Hung et al. presented another
new method for similarity measures between IFSs. They
present a method to calculate the distance between IFSs on the
basis of the Hausdorff distance. Then they generate a new
Figure 3. Topology model for RFS networks
similarity measure to calculate the degree of the similarity
lued Intuitionistic Fuzzy Set (IVIFS). Furthermore it is gui- between IFSs by this distance [16].
dable on constructing similarity-based technologies for RFS In this paper, we propose a new measure of distance on
networks. In this paper, we propose a new measure of distance VIFS and discuss their corresponding similarities based on
on VIFS and discuss their corresponding similarities based on distance. This is important to future works on distance theory.
distance, that is s = 1 − d . We organize this paper as follows: III. NEW SIMILARITY MEASURE BASED ON DISTANCE
in section 2, we introduce some related work on FS and
similarity. It worth noting that from [2] we know all existed A. Preliminaries
fuzzy sets, such as: FS, IFS, IVIFS and IVFS, are subset of In the following, we firstly recall some basic notions and
VIFS in the view of isomorphic imbedding mapping. This definitions. Throughout this paper, let X is a nonempty
ensures that our work about distance and similarity is a
definite set, | X |= n .
reasonable extension of FS, IFS and IVIFS. Furthermore they
[2]
are useful and guidable on constructing new measures of Definition 3.1.1. An interval number over [0,1] is
distance and similarity on all these existed fuzzy sets. In − +
section 3, at first, we review some basic notions and give some defined as an object of the form: a = [a , a ] with the
− +
definitions which are quite useful in the following work. Then property: 0 ≤ a ≤ a ≤ 1 . Let L denote all interval
we mainly construct some measures of similarity based on
numbers over [0,1] . We define the definition of relation (it is
distance. From examples we make sure some existing
similarity measures between IFSs are involved in our new specified by “ ≤ ”): a ≤ b ⇔ a − ≤ b − and a + ≤ b + . We
measures. This is important to future works on distance can easily prove that “ ≤ ” is a partially ordered relation on
theory.
L . So < L, " ≤" > is a complete lattice where
II. RELATED WORKS a ∨ b = [a − ∨ b − , a + ∨ b + ]
Since L.A.Zadeh introduced FS in 1965 [4], a lot of new
theories treating imprecision and uncertainty have been a ∧ b = [a − ∧ b − , a + ∧ b + ]
− + − +
introduced. Some of them are extensions of fuzzy set theory. ∨ a = [ ∨ ai , ∨ a i ] ∧ a = [ ∧ a i , ∧ a i ]
K.T.Atanassov extended this theory, proposed the definition i∈I − i i∈I i∈I , i∈I − i i∈I i∈I .
of IFS [5] and IVIFS [6], which have been found to be highly Let 1 = [1,1] denotes the maximum of < L, " ≤" > and
useful to deal with vagueness out of several higher orders
fuzzy sets. Then in 2005, we proposed VIFS and proved sets 0 = [0,0] denotes the minimum of < L, " ≤" > .
such as: FS, IFS, and IVIFS, are all subsets of VIFS in the
Then we introduce the definition of interval-valued
view of isomorphic imbedding mapping. This ensures that any
work on VIFSs is a reasonable extension of FS, IFS and intuitionistic fuzzy sets on L .
[2]
IVIFS. Definition 3.1.2. A general interval valued intuitionistic
Measures of similarity between fuzzy sets, as an important fuzzy sets A on X is an object of the form:
content in the fuzzy theory, are widely used and cited in A = { x, μ A ( x),ν A ( x), x ∈ X }
literatures. Similarity-based search remains as a hot research
topic even today due to its wide usage in many new applicatio- Where μA : X → L and ν A :X →L
ns, including network traffic analysis [7], sensor network
0 ≤ μ A ( x) ≤ ν A ( x) ≤ 1 , ∀x ∈ X
C
With the property:

322
For briefly, we denote + + +
d 2i ( A, B) = ν A ( xi ) − ν B ( xi )
μ A ( x) = [ μ − A ( x), μ + A ( x)] ∈ L ,
− + − +
ν A ( x) = [ν − A ( x),ν + A ( x)] ∈ L Then we have d1i , d1i , d 2i , d 2i ,
In paper [2], we defined the relation “ ≤ ”: (i = 1,2,..., n) are weak distance functions.
−
A ≤ B ⇔ μ A (x) ≤ μ B (x) and ν A ( x) ≥ ν B ( x) . Prove: 1. Firstly we will prove d1i ( A, B) is weak
[2] distance function
Definition 3.1.3. A mapping
S : VIFSs ( X ) × VIFSs ( X ) → [0,1] . For
− − −
d1i ( A, B) = μ A ( xi ) − μ B ( xi ) ， then we have
S ( A, B ) is said to be the degree of similarity between − −
d1i ( A, B) = d1i ( B, A)
A∈ VIFSs ( X ) and B ∈ VIFSs ( X ) , if S ( A, B ) satisfies
− −
the properties: If A = B ， it follows that μ A ( xi ) = μ B ( xi ) ， then
(1) If A = B , S ( A, B ) = 1 ; −
we have d1i ( A, B) = 0
(2) S ( A, B ) = S ( B, A) ;
If A ≤ B ≤ C ，it follows that
(3) If A ⊆ B ⊆ C , A, B, C ∈ VIFSs ( X ) , then
μ A − ( xi ) ≤ μ B − ( xi ) ≤ μ C − ( x i )
S ( A, B) ≥ S ( A, C ) , S (C , B ) ≥ S ( A, C ) .
Then we have
Then we will give the definition of the weak distance
which is quite important to measures of similarity. μ A − ( xi ) − μ B − ( xi ) ≤ μ A − ( xi ) − μ C − ( xi )
Definition 3.1.4. A mapping − −
That is d1i ( A, B) ≤ d1i ( A, C )
d w : VIFSs( X ) × VIFSs( X ) → [0,1] . d w ( A, B) is said
+ − +
to be the weak distance A ∈ VIFSs ( X )
between 2. in the same way, we can prove that d1i , d 2i , d 2i
are weak distance functions.
and B ∈ VIFSs ( X ) , if the mapping d w ( A, B ) satisfies the
It worth noting that while n > 1 ，these 4n weak distance
properties: functions are not distance functions. This is because we can
(1) d w ( A, B ) = d w ( B, A) ;
only get μ A − ( xi ) = μ B − ( x i ) from
(2) If A = B , d w ( A, B) = 0 ;
μ A − ( xi ) − μ B − ( xi ) = 0 other than A = B .
(3) If A ⊆ B ⊆ C , A, B, C ∈ VIFSs ( X ) , then
Theorem 3.2.2. let
d w ( A, B) ≤ d w ( A, C ) , d w (C , B) ≤ d w ( A, C ) .
1. f ( x1 , x 2 ,..., x s ) : [0,1] s → [0,1] with properties:
It is easy to see this weak distance function is a
generalization of traditional distance functions. From 1) let x1 = x 2 = ... = x s = 0 ，then
definition 3.1.3 and 3.1.4 it is clear that
f ( x1 , x 2 ,..., x s ) = 0 .
S ( A, B) = 1 − d w ( A, B) .
2) to any xi (i = 1,2,..., s ) , f is a decreasing function.
Thus measures of similarity are equivalent to measures of
distance. Then in the next section, with distance and weak 2. to any d i ( A, B ) (i = 1,2,..., s ) is a weak distance
distance functions we propose some measures of similarity function.
and this will be useful in sensor networks.
Then to any ( A, B ) ∈ VIFS ( X ) ,
B. New similarity measure based on distance
d ( A, B) = f (d1 ( A, B), d 2 ( A, B),..., d s ( A, B))
In this section, we will propose a measure of similarity
between VIFS. Firstly we propose a distance function on is a weak distance function.
VIFS. Prove：
− + − + From property 2 we get:
d1i ，-- d1i ， d 2i ， d 2i are
Proposition 3.2.1. Let
d i ( A, B) = d i ( B, A) ， i = 1,2,..., s
VIFS ( X ) × VIFS ( X ) → [0,1] ， (i = 1,2,..., n)
Then we have d ( A, B ) = d ( B, A)
Where to any ( A, B ) ∈ VIFS ( X ) , there are
if A = B ，then d i ( A, B ) = 0 ， i = 1,2,..., s
− − −
d1i ( A, B) = μ A ( xi ) − μ B ( xi ) From property 1.1) we have
+ + + d ( A, B ) = f (0,0,...,0) = 0
d1i ( A, B) = μ A ( xi ) − μ B ( xi )
if A ≤ B ≤ C ，then
− − −
d 2i ( A, B) = ν A ( xi ) − ν B ( xi ) d i ( A, B) ≤ d i ( A, C ) ， i = 1,2,..., s

323
 then we have 3) let A ≤ B ≤ C , ( A, B ) ∈ VIFS ( X )
f (d1 ( A, B), d 2 ( A, B),..., d s ( A, B ))
ν A −ν C − ≥ ν A − −ν B − ≥ 0 ≥ μ A − − μ B − ≥ μ A − − μ C −
−

≤ f (d1 ( A, C ), d 2 ( A, C ),..., d s ( A, C ))
ν A + −ν C + ≥ ν A + −ν B + ≥ 0 ≥ μ A + − μ B + ≥ μ A + − μ C +
that is d ( A, B ) ≤ d ( A, C )
Hence
It worth noting that d ( A, B ) usually isn’t a distance
function. For example, let
ν A − −ν C − ≥ ν A − −ν B −
s = 1 , f ( x) = x , d1 ( A, B) = μ A ( x1 ) − μ B ( x1 ) , μ A− − μC − ≥ μ A− − μ B −
then we have d ( A, B) = μ A ( x) − μ B ( x) is only a weak
ν A + −ν C + ≥ ν A + −ν B +
distance function.
= W (θ1 ,θ 2 ,...,θ n ) , where
Definition 3.2.3. Let d ( A, B ) μ A + − μC + ≥ μ A + − μ B +
1. To any ( A, B ) ∈ VIFS ( X ) we have
φi − , φi + , ϕ i − , ϕ i +
{ ( ) ( )
From definition 3.6 we know all satisfy
− − − − − − −
θ i = E i [φ i μ A − μ B , ϕ i ν A − ν B ], their properties 2), hence
+
E i [φ i
+
(μ A
+
− μB
+
), ϕ (ν
i
+
A
+
−ν B ]
+
)} φi + ν A + − ν C + ≥ φi + ν A + −ν B +

W : [0,1] n → [0,1] with properties:

2. ϕi + μ A + − μ B + ≥ ϕi + μ A + − μC +
1) x1 = x 2 = ... = x n = 0 ⇔ W ( x1 , x 2 ,..., x n ) = 0 φi − ν A − − ν C − ≥ φi − ν A − − ν B −
2) To any xi (i = 1,2,..., s ) , W is a decreasing function.
ϕi − μ A − − μ B − ≥ ϕi − μ A − − μC −
3. θ i : [0,1] × [0,1] → [0,1] (i = 1,2,..., n) with
− +
properties: And in definition 3.6, Ei , Ei all satisfy their properties
1) θ i ( x, y ) =0⇔ x= y=0 2), hence
− − − − − − −
2) θ i is a increasing function to any variable. E i [φ i ( μ A − μ B ), φ i (ν A − ν B )]
− − − − − − −
4.
− +
E i , E i : [0,1] × [0,1] → [0,1] (i = 1,2,..., n) with ≤ E i [ϕ i ( μ A − μ C ), ϕ i (ν A − ν C )]
+ + + + + + +
properties: E i [φ i ( μ A − μ B ), φ i (ν A − ν B )]
− +
1) E i ( x, y ) = 0 ⇔ x = y = 0 ⇔ E i ( x, y ) = 0 + + + + + + +
≤ Ei [ϕi ( μ A − μC ), ϕi (ν A −ν C )]
− +
2) Ei , Ei is a increasing function to any variable. From properties 2) of W we have
5. φ i , φ i , ϕ i − , ϕ i + : [0,1] → [0,1] (i = 1,2,..., n)
− +
d ( A, B ) ≤ d ( A, C )
with properties: Hence, d ( A, B ) is a distance function.
1) φ i − ( x) = 0 ⇔ φ i + ( x ) = 0 ⇔ x = 0 E.g. 3.2.5. let
1 n
−
⇔ ϕi ( x) = 0 ⇔ ϕi ( x) = 0
+
1. W = ∑ xi
n i =1
2) φi − , φi + , ϕ i − , ϕ i + are increasing functions. x+ y
2. θ i ( x, y ) =
Theorem 3.2.4. d ( A, B ) in definition 3.2.3 is a distance 2
function. + +
3. E i ( x, y ) = max( x, y ) = E i ( x, y )
Prove: 1) let ( A, B) ∈ VIFS ( X ) ，if A = B ，then
μ A − − μ B − = 0,ν A − − ν B − = 0, 4. φ i − ( x ) = φi + ( x ) = ϕ i − ( x) = ϕ i + ( x ) = x
1 n
μ A + − μ B + = 0,ν A + − ν B + = 0. Then d ( A, B) = ∑ [max(H ) + max(K )]
2n i =1
− + − +
From properties 2) of W , θ i , Ei , Ei , φi , φi , Where
ϕ i − , ϕ i − we know A = B ⇔ d ( A, B) = 0 − −
H = μ A ( xi ) − μ B ( xi ) , ν A ( xi ) − ν B ( xi )
− −

2) from definition 3.6, it is easy to prove that + + + +

d ( A, B ) = d ( B, A) . K = μ A ( xi ) − μ B ( x i ) , ν A ( x i ) − ν B ( xi )

324
 Let
− +
μ A ( xi ) = μ A ( xi ) = μ A ( x i )
ν A ( xi ) = ν A + ( xi ) = ν A ( xi ) i ∈ {1,2,..., n}
−
Then we have
d ( A, B ) =
1 n
∑ [max( μ A ( xi ) − μ B ( xi ) , ν A ( xi ) − ν B ( xi ) )]
n i =1
The similarity measure based on this distance is:
s ( A, B ) =
1 n
1 − ∑ [max( μ A ( x i ) − μ B ( x i ) , ν A ( x i ) −ν B ( x i ) )]
n i =1
This is the similarity proposed by Hung et al..
IV. CONCLUSION
In this paper, we propose a new measure of similarity on
VIFSs. This measure is useful on all existed fuzzy sets. This
paper gives a example to show that this corresponding
similarity measure based on our distance can involve many
existing similarity measures. And these measures are
applicable on all existed fuzzy sets. Based on this we will
focus on constructing a fuzzy query processing strategy for
large scale RFS networks, which is important in our
monitoring.
There are still some similarity measures, such as:
Dengfeng-Chuntian measures, etc. can not be involved in this
general similarity measure. Thus there still need a
improvement of our measure.
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 2009 International Conference on Computer Engineering and Technology
3D Mesh Skeleton Extraction Based on Feature Points
Faming Gong
College of Computer and Communication Engineering
China University of Petroleum (East China)
Dongying Shandong, China
e-mail:gfming@hdpu.edu.cn
Abstract—A novel efficient skeleton extraction algorithm is
proposed, which is based on feature points extraction and Reeb
graph theories. Because of the topological facility of feature
points, a model can be divided into several branches according
to them. One feature point can present one branch. So we just
extract the other point of the branch – that could be skeleton
point, connect these points with their corresponding feature
points, then we can get all branch skeletons. Finally,
connecting branch skeletons through connecting skeleton
points according the topological relationship of each skeleton
point preserving, then 3D model’s skeleton can be extracted.
Without pre-processing stages and without input
parameters, this algorithm can automatically extract the
skeleton of 3D models. Theoretical analyses and experimental
results show that our method has a lower computing
complexity, and meets the requirement of extracting nice-
looking and affine-invariant skeletons efficiently.
Keywords-skeleton; skeleton point; skeleton extraction
I. INTRODUCTION
As a widely used representation of 3D shapes, polygonal
mesh is fundamental for computer graphics. However, with
the development of computer graphics, polygonal mesh
could not meet the need for many applications. As a higher
level shape description, topological skeletons are proposed
and then used in diverse fields. Many approaches have been
developed for extracting skeletons of 3D models. In
comparison to traditional skeleton extraction [1,2],
approaches based on Reeb graph theories, present the
advantage to preserve the topological properties of the shape
[3].
Skeleton extraction, based on Reeb graph theory, is
necessary to construct a Reeb graph by defining a continuous
scalar function appropriately. Lazarus and Verroust [4]
defined the function by the geodesic distance from a source
vertex to any other vertex in the mesh. Due to the lack of
stability, Hilaga etc. [5] proposed to calculate the integral of
geodesic distance all over the mesh. However, when more
than one connected components are contained in the model,
the function will not work and lead to much more errors.
Huang Kunwu [6] improved the function and solved the
above problem. Unfortunately, these functions will generate
an important amount of critical points. In [7] Julien Tierny
etc. developed a new method to generate Reeb graphs of
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.71
Cui Kang
College of Computer and Communication Engineering
China University of Petroleum (East China)
Dongying Shandong, China
e-mail:laura855@163.com
given models. In the method, they defined the mapping
function based on geodesic distance and feature points to
guarantee invariant to transformation and decomposition and
avoid generating an unmanageable set of critical points. Due
to its complexity with O(n2 ) steps, we propose a new method
to extract skeleton.
This paper is structured as follows. Firstly, we introduce
the algorithm for extracting feature points. Secondly, we
define our mapping function. Thirdly, we present our
algorithm for skeleton points extraction. Finally, we
comment on experimental results and end the paper with
concluding remarks.
II. FEATURE POINT EXTRACTION
Feature points are mesh vertices located on extremities of
prominent components. As they reveal the most significant
features of the shape, they are used in the mapping function
computation as origins for geodesic distance evaluation. To
extract them, we use a quite straightforward algorithm,
which is proposed in [7], based on topological tools. In this
paper, we will improve the algorithm aim at particular
stages.
To extract feature points, we first extract the mesh most
distant vertices as origins. Here we propose a new
algorithm—the minimum distance algorithm, described in
the following section.
A. The minimum distance algorithm
Let n be the number of vertices of the given connected
triangulated surface T. Let v s1 and vs 2 be the most geodesic
distant vertices of T. Let V be the vertices set of T. Now we
describe the algorithm that we used to compute v s1 and vs 2
as follows:
Step 1: Get the center point (marked v c ) of the mesh：
for all v i∈{1,..., n} ∈ V , mark the coordinate of v i as
(x i , y i , z i ) ,then compute vc as follow:
( x 1 , y 1 , z 1 ) + ( x 2 , y 2 , z 2 ) + ... + ( x n , y n , z n ) .
vc =
n
Step 2: If v c ∈ V then Step 3
else Step 4
326
 (a) Set of Vs (b)E1 (c)E2 (d) Feature points set F

Figure 1. Feature points set extraction overview

end if
Step 3: Compute the distance from v c to vi∈{1,..., n} ∈V
'
by distance formula, get the nearest vertex (marked as v c )
Step 4: Get the longest geodesic distance vertex (marked
v temp ) of v 'c :
for all v i∈{1,..., n } ∈ V do Moore-Dijkstra's algorithm to
model, and the center of the interface (the face between two
sub-parts) must be the skeleton point.
In this paper, skeleton points can be computed for the
whole mesh using a step by step process, described in
algorithm 2. Let T be the vertexes set of the mesh. We divide
T into two heaps, respectively the set of visited vertices T v
and the set of candidate vertices for visiting T c . Let F be the

δ (vc' , vi ) (geodesic distance between v 'c and vi )

set of feature points and V n be the set of neighbors of
compute
vertex v . The algorithm of skeleton extraction proceeds in
Iterate the above steps until there are two approximating the follow three steps.
results of δ (v s 1 , v s 2 ) . Step 1: Get vertexes of interfaces.
Tv = ∅
B. Extracting feature points set Tc = T
In the above section, we have extracted the most geodesic For T c ≠ ∅ do (1) at each step:
distance vertices v s1 and v s 2 (Fig. 1 (a)). In this section, we
(1) ∀ v ∈ T c , get v ’s corresponding feature point:
will use the two vertices as origins to extract feature points
set. The strategy proceeds in the following three stages. For F ≠ ∅ do compute f m ( v )
Firstly, for each vertex in the mesh T, we compute two
Remove v from Tc
geodesic mapping function (mark fg1 and fg2 ), whose origins
Add v to v
T
are the mesh most distant vertices. Secondly, both sets of
f f Initiate C v ← {} as the set of vertices that on the same
local extrema (minima and maxima) of g1 and g 2 (Fig. 1
(b) and Fig. 1 (c)) are computed, respectively marked E1 and interface with vertex v
E2. Then the set of feature points is both included in E1∪ While v
T ≠ ∅
do
E2. Finally, we get the set of feature points F of T (Fig. 1 (d)) for V n ≠ ∅ do
by intersect E1 and E2.
if f m ( v )! = f m ( v ' ) ( ∃ v '∈ V n ) Then (2)
III. MAPPING FUNCTION End if
To compute visually meaningful topological skeletons, it (2): Add v to Cv
is necessary to define a continuous scalar function, which for V n ≠ ∅ do
reveals the most significant parts of the mesh by generating
critical points as less as impossible. Considering these issues, If f m ( v ) == f m ( v ' ) ( ∀ v ' ∈ (V n ∩ T c ) ) then
we define our mapping function as the function which has Remove v' from Tc
been proposed in [7]. The function is defined as follow:
f m ( v ) = 1 − Min δ (v, v f ) Add v' to Tv
Vf i ∈ F i
End if
F: the feature points set, v f i : one feature point, End while
For C v ≠ ∅ do save Cv as the subset of Vs (the set of
δ (v, v f ) : the geodesic distance from v to v fi . In the rest
i

of the paper, we denote the function fm and compute it in
each vertex of T to get the geodesic distance to the closest
feature point.
IV. SKELETON EXTRACTION
As feature points highlight the most significant features
of the shape, they can discriminate visually meaningful sub-
parts of a given connected component. From our experience,
each feature point represents one different sub-part of the
subset of vertices which have the same result of f m () )
Step 2: Combine interfaces with small enough distance
For all interfaces at each step: FindBoundary()
∀v ∈ int erface1
If v has neighbors which also be the vertices of the other
interfaces then combine these interfaces
End if
Compute the centers of all interfaces.
Step 3: Dealing with circuits of skeleton

327
 For all v ∈ M s ( M s is marked as the set of skeleton
points) at each step: deleteF()
vi ∈Ms
v j ∈M s
(d) Skeleton of model
If vi and v j connects the same feature point
Figure 3. Skeleton of cetacean
v p then
Fig. 2 shows the example graph of interfaces obtained
If vi is the center of one interface then with this strategy, described in Step 1. The number of the
feature points will be the number of the interfaces. In fig. 3,
Connect vi and v p example of skeleton extraction overview is shown. We get
Else if v j is the center of one interface then the center of interfaces (Fig. 3 (a)) by computing the average
of the coordinates of all vertices in each interface. By dealing
Connect v j
and v p with all center points, we can delete the unnecessary points
and get the real skeleton points (Fig. 3 (b)). However, we
Else if both vi and v j are not center points then cannot avoid generating skeleton with circuits (Fig. 3 (c))
If the number of feature points connect until dealing with Step 3. Finally, we get the satisfied
skeleton (Fig. 3 (d)).
with vi > the number of feature points connect with v j then
V. EXPERIMENTS AND RESULTS
Connect vi and v p
In this section, we present and comment on experimental
Else results obtained with our method. Presented models are
Connect v j and v p connected triangulated surface extracted from 3D free model
database.
End if
Given an input connected triangulated surface T, let n be
End if
the number of vertices in T. Feature points extraction
End if
executes in O(n × log(n)) steps. The mapping function is
performed in O(|F|×n×log(n)) steps, with |F| be the number
of extracted feature points. Extracting skeleton points
executes in O(n×（|F|+u+v）) steps, with u be the largest
number of neighbors of the vertex in mesh and v be the
largest number of vertices of each interface. Dealing with
Figure 2. Examples of Interfaces circuits of skeleton is realized in O(|F|×|S|) steps, with |S| be
the number of the skeleton points. Consequently, we can
state that the overall complexity of our method is O(n ×
log(n)) steps.
Our experiment is executed in VC++ under Windows XP
on a PC with P4-CPU and 512M of RAM. Table 1 shows the
computation times corresponding to the models presented in
(a) The center of interfaces
Fig.4.
TABLE I. COMPUTATION TIMES.

模型 顶点 特征点 计算时间

(b) Skeleton points

Hand 1665 5 0.984s
Ring 287 4 0.021s

Candlestick 2100 2 1.72s

Cat 872 15 0.591s

(c) Skeleton with circuits cetacean 2563 4 3.387s

328

Figure 4. Topological skeletons of standard models
VI. CONCLUSION
In this paper, we presented an automatically algorithm
without pre-processing stages and without input parameters
for mesh skeleton extraction. This algorithm is based on
geodesic distance evaluations and feature points extraction.
So that it is stable and invariant to model rotation, translation
329
and geometric transformations. In our experiment, we
improve the feature point extraction algorithm and propose a
new method for computing the maximum geodesic distance
vertexes of the mesh efficiently, and we also develop a novel
efficient method for extracting skeleton points. Theoretical
analyses and experimental results show that the algorithm
reduces the computing complexity, improves the efficiency
and meets the requirement of topological skeleton extraction.
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 2009 International Conference on Computer Engineering and Technology
Pairings Based Designated Verifier Signature Scheme for Three-Party
Communication Environment
Han-Yu Lin
Department of Computer Science
National Chiao Tung University
Hsinchu, Taiwan
e-mail: hanyu.cs94g@nctu.edu.tw
Abstract—A designated verifier signature (DVS) scheme
has the property that only the designated verifier
specified by the signer can check the validity of the
signature instead of anyone else. Meanwhile, the
designated verifier can not use this proof to convince any
third party that a signature is generated by the claimed
signer. Consider the application of three-party
communication environment. One party may has to
generate a signature such that only the other two is
capable of verifying it solely. In this paper, we propose a
novel DVS scheme for three-party communication
environment from pairings. The proposed scheme has the
following advantages: (i) Each of the two designated
verifiers can independently check the validity of the
signature without cooperating with the other; (ii) The
proposed scheme is efficient in terms of the executed
pairings, since it only needs one pairing computation to
generate a DVS for two designated verifiers; (iii) The
proposed scheme satisfies the security requirements of
unforgeability, non-transferability and source hiding.
Keywords-designated verifier; digital signature scheme;
bilinear pairings; public key system
I. INTRODUCTION
Since Diffie and Hellman [2] proposed the first
public key system based on the discrete logarithm
problem in 1976, public key systems [4, 10, 13] have
been extensively studied. A digital signature scheme [3,
10] is one fundamental cryptographic technique which
primarily aims for providing authenticity [14] and
non-repudiation [8]. Traditionally, a digital signature
can be verified by anyone else with the signer’s public
key. Yet, in some special applications such as the
electronic voting [9, 12] and the electronic auction [6,
17], the signatures are not suitable for the public
verification. To fulfill such requirement, Jakobsson et
al. [5] introduced the concept of designated verifier
proofs and in a sense proposed a designated verifier
signature (DVS) scheme in 1996. Although the resulted
signature is publicly verifiable in their scheme, only the
designated verifier can be convinced that a signature is
generated by the claimed signer. On the other hand, the
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.34
Tzong-Sun Wu
Department of Computer Science and Engineering
National Taiwan Ocean University
Keelung, Country
e-mail: ilan543@gmail.com
designated verifier can also create a valid signature
which is computationally indistinguishable from the
one issued by the original signer for any third party.
Consequently, the designated verifier can not use this
proof to convince any third party that a signature is
created by the claimed signer. In 2003, however, Wang
[16] pointed out that their scheme is insure for that a
malicious signer can easily cheat the designated
verifier.
In 2004, Saeednia et al. [11] proposed a strong
designated verifier signature scheme in which the
signature verification requires the assistance of the
designated verifier’s private key to compete. Therefore,
only the designated verifier has the ability to verify the
signature. Generally speaking, a DVS scheme should
satisfy the following security requirements:
1) Unforgeability:
Given the identifier of some designated verifier,
say, U1, it is computationally infeasible for any
malicious adversary to forge a valid DVS with respect
to U1 without knowing the signer’s private key.
2) Non-transferability:
In a DVS scheme, only the designated verifier can
be convinced that a signature is created by the claimed
signer. The designated verifier can not convince any
third party that a signature is issued by the claimed
signer, since he can also forge a valid signature which
is computationally indistinguishable from the one
generated by the original signer.
3) Source Hiding:
It is also referred to as the signer’s anonymity.
That is, given a message and its corresponding DVS, it
is computationally infeasible to determine the identifier
of real signer from the original signer or the designated
verifier.
Consider the application of three-party
communication environment. One party may has to
generate a signature such that only each of the other
two is able to independently verify the signature.
330
Obviously, traditional DVS schemes of single-verifier a b c abc
Pr[A(g, g , g , g ) = e(g, g) ; (a, b, c) ← Z q* ,
setting are not well-suited here for that the total
computational complexities double. In this paper, we 1
a b c
(g, g , g , g ) ← G1] ≤
.
propose a bilinear pairings based novel DVS scheme Q(k )
for three-party communication environment. The The probability is taken over the uniformly and
proposed scheme not only fulfills the above security
independently g∈G1 and (a, b, c)∈ Z q* and over the
requirements, but also allows each of the two
designated verifiers to solely verify the signature. As to random choices of A.
the required pairing computation, the signer and each
B. Our Scheme
designated verifier only have to perform once for
generating and verifying the signature, respectively. The proposed DVS scheme can be divided into
The rest of this paper is organized as follows. We two phases: the signature generation and the signature
demonstrate the proposed DVS scheme in Section 2. verification phases. Initially, the system determines the
Some security analyses and the performance evaluation following public information:
will be discussed in Section 3. Finally, a conclusion is p, q: two large primes such that q | (p − 1);
given in Section 4.
G1 , G2 : two groups of the same order q;
II. THE PROPOSED DVS SCHEME g: a generator of order q over G1;
In this section, we first briefly review some e: a bilinear pairing, e: G1 → G2;
n
related definitions and then introduce the proposed h1 : a one-way hash function, h1: G2 → {0, 1} ;
DVS scheme. h2: a one-way hash function,
n n
A. Related Definitions h2: Z *p × Z *p × {0, 1} → {0, 1} ;
Let G1 and G2 denote two groups of the same
Each user Ui chooses his private key xi∈ Z q* and
prime order q. We say that e: G1 × G1 → G2 is a
computes the corresponding public key as
bilinear map if it satisfies the following properties:
yi = g xi mod p. Details of each phase are described
1) Bilinearity:
below:
e(P1 + P2, Q) = e(P1, Q)e(P2, Q);
e(P, Q1 + Q2) = e(P, Q1)e(P, Q2); The signature generation phase: Let Us be the signer
a b ab and {Ua, Ub} the two designated verifiers. For signing
e(P , Q ) = e(P, Q) for P, Q∈G1 and a, b∈ Z q* .
the message M, Us randomly chooses two integers k,
2) Non-degeneracy: r0 ∈ Z q* and computes
If g is a generator of G1, then e(g, g) is a generator
σ = e( y a , yb ) x s , (1)
of G2.
C1 = ( y a yb ) M g r0 mod p, (2)
3) Computability: k
C2 = g mod p, (3)
Given P, Q∈G1, the value of e(P, Q) can be C3 = h1(σ), (4)
efficiently computed by a polynomial-time algorithm.
r1 = h2(C1, C2, C3) mod q, (5)
Definition 1 (Bilinear Diffie-Hellman (BDH) r2 = k − xsr1 mod q. (6)
problem) Here, (r0, r1, r2) is the signature for M. Us then sends
Given a BDH instance (g, A, B, C)∈G1 where A =
a b c
(M, r0, r1, r2) to designated verifiers Ua and Ub.
g , B = g and C = g for some (a, b, c)∈ Z q* , compute
abc
The signature verification phase: Upon receiving it,
e(g, g) ∈G2. Ua (or Ub) first computes
Definition 2 (Bilinear Diffie-Hellman assumption) K = g r2 y s r1 mod p , (7)
For every probabilistic polynomial-time algorithm xa xb
A, every positive polynomial Q(⋅) and all sufficiently
σ = e( y s , yb ) ( = e( y s , y a ) ), (8)
large k, the algorithm A can solve the BDH problem and (C1, C3) as Eqs. (2) and (4). Then, Ua (or Ub) can
1 independently verify the received signature (r0, r1, r2)
with an advantage at most , i.e.,
Q(k ) by checking
r1 = h2(C1, K, C3) mod q. (9)

331
If it holds, the signature (r0, r1, r2) for the message M is
valid.
III. SECURITY ANALYSES AND COMPARISONS
In this section, we first analyze the security of our
proposed scheme and then evaluate its performance.
A. Security Analyses
We discuss some security considerations with
respect to the proposed scheme from the perspectives
of correctness, unforgeability, non-transferability and
source hiding.
1) Correctness
A DVS scheme for three-party communication
environment is correct if one party can generate a valid
signature such that only the other two can check the
validity of the signature when all involved parties
follow the steps of the scheme. We prove the
correctness of our proposed scheme as Theorem 1.
Theorem 1. With the signature (r0, r1, r2) for the
message M, each of the two designated verifiers Ua and
Ub can independently check its validity with Eq. (9).
Proof: From the right-hand side of Eq. (9), we have
h2(C1, K, C3)
= h2(C1, g r2 y s r1 mod p , C3) (by Eq. (7))
= h2(C1, g r2 + x s r1 mod p , C3)
k of Eq. (9).
= h2(C1, g mod p, C3) (by Eq. (6))
= h2(C1, C2, C3) (by Eq. (3))
= r1 (mod q) (by Eq. (5))
which leads to the left-hand side of Eq. (9).
Q.E.D.
2) Unforgeability
To forge a valid DVS with respect to the
designated verifiers Ua and Ub, an attacker may first
randomly choose a message M' and (k', r0', r2') ∈ Z q*
and then attempt to derive r1' from Eq. (7). However,
he has to solve the discrete logarithm problem (DLP)
[2, 7] which is computationally intractable. Even if he
could derive r1', the forged signature (r0', r1', r2') for
M' would not pass the test of Eq. (9) without knowing
the correct hash value of shared session key σ between
the signer and the designated verifiers. On the other
hand, any malicious designated verifier having the
knowledge of the shared session key can not forge a
valid DVS under the protection of DLP and one-way
hash function (OWH) [1, 14], either.
3) Non-transferability
A DVS scheme for three-party communication
332
environment achieves the security requirement of
non-transferability if both the designated verifiers can
create a valid signature which is computationally
indistinguishable from the one created by the signer. In
other words, the designated verifiers can not convince
any third party that a signature is issued by the claimed
signer. In our proposed DVS scheme, each of the two
designated verifiers Ua and Ub can first compute
xa(M − M') and xb(M − M') where M' is an arbitrarily
chosen message, respectively. Then, Ua and Ub
cooperatively create a valid signature (r0', r1, r2) where
r0' = r0 + ∑
xi ( M − M ′). It is easy to show that the
i = a,b
parameter ( ya yb ) M ′ g r0' mod p is equal to
( ya yb ) M
g r0
mod p as follows:
( ya yb ) M ′ g r0'
∑ xi ( M − M ′)
M′ r0 i = a ,b
= ( y a yb ) g g
∑ xi M ′ ∑ x i ( M − M ′)
= g i =a ,b g r0 g i =a ,b
∑ xi M
= g r0 g i =a ,b
= ( y a yb ) M g r0 (mod p).
Therefore, the forged DVS (r0', r1, r2) will pass the test
4) Source hiding
In the proposed scheme, any third party having the
knowledge of the shared session key σ can check the
validity of the signature. Nevertheless, a forged DVS
generated by the two designated verifiers has the
identical distribution as the one issued by the original
signer. Consequently, given a message M and its
corresponding DVS, it is computationally infeasible for
any third party to identify the actual signer from the
original signer or the designated verifiers, even if he
knows the shared session key.
B. Performance Evaluation
In this subsection, we compare the proposed DVS
scheme with Susilo et al.’s one [15] in terms of
executed pairings which are considered to be the most
time-consuming operation in pairings-based systems. It
is believed that reducing the number of such
computation helps the practical implementation. To
obtain a fair result, we assume that only one designated
verifier is involved in each scheme. The detailed
comparison is shown as Table 1. From this Table, it
can be seen that our proposed scheme has a better
performance than Susilo et al.’s one by one pairing
 computation as a whole.
TABLE 1. PERFORMANCE COMPARISON (IN NUMBER OF REQUIRED
PAIRINGS)
Phase Signature Signature
Total
Scheme generation verification
The proposed scheme 1 1
Susilo et al.’s scheme 1 2
IV. CONCLUSIONS
In this paper, we have proposed a novel DVS
scheme for three-party communication environment
from pairings. The proposed scheme only allows two
designated verifiers to independently check the validity
of the signer’s signature. The proposed scheme is
shown to have a better performance than Susilo et al.’s
one and also fulfills the security requirement of
unforgeability, non-transferability and source hiding.
Even if the two designated verifiers reveal their secrets,
they can not convince any third party that a signature is
created by the claimed signer, because they can
conspire to forge a valid signature which is
computationally indistinguishable from the one
generated by the original signer.
ACKNOWLEDGEMENT
This research was supported in part by the
National Science Council of Republic of China under
the contract number NSC 97-2221-E-019-019.
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efficient strong designated verifier signature scheme,”
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333
 2009 International Conference on Computer Engineering and Technology
A Novel Shared Path Protection Scheme for
Reliability Guaranteed Connection
①
Jijun Zhao, ①Weiwei Bian, ①②
College of Information & Electronic Engineering
① ②
Hebei University of Engineering, Handan Polytechnic College
Handan, China
Email: zhaojijun@china.com, bianweiwei2008@163.com, liukaironghua@163.com, wangsujian2006@163.com
Abstract—In this paper, we study the shared path protection
(SPP) problem in wavelength division multiplexing (WDM)
optical networks, and propose a novel SPP algorithm with
differentiated reliability (DiR) called SPP-DiR. Differing from
previous work, SPP-DiR algorithm effectively adapts to the
diversity demands of customer requests and has higher
resource utilization than SPP algorithm. Therefore, SPP-DiR
algorithm outperforms the previous work. Simulation results
have shown to be promising.
Keywords-WDM optical networks; shared path protection;
differentiated reliability; resource utilization
I. INTRODUCTION
As the advent of Wavelength Division Multiplexing
(WDM), hundreds wavelength channels carrying data at
rates in the order of terabits per second can be multiplexed
into a single fiber. In this circumstance, many new services
utilizing vast bandwidth of optical layer is gradually
growing, such as OVPN, BOD, SLA, et al. These new
services provide multiple levels of service performance in
the same network. At the same time, in the
connection-oriented networks such as WDM optical
networks, dynamic provisioning of reliability guaranteed
connections with protection against single network link
failures is an important issue and has received much study
in recent years.
Consistently with these developments, the concept of
optical circuits with differentiated reliability (DiR) was
introduced by the researchers in [1] and [2] and applied to
provide multiple reliability degrees in WDM rings using
dedicated path protection (DPP) switching. DiR concept
guarantees each connection has a minimum reliability
degree that meets the customer requirement, so it reduces
the entire network cost [3]. While in the aspect of resource
utilization (RU), shared path protection (SPP) has better
performance than DPP, so this paper extends the DiR
concept which is applied to DPP in [1] and [2] to SPP.
Many literatures refer to SPP [4-7], but they consider SPP
from the aspect of shared risk link group (SRLG) or in the
background of availability or quality of service (QoS).
Some literatures refer to SPP with DiR[3][8], which is studied
from the point of time efficiency or in the aspect of segment
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 334
DOI 10.1109/ICCET.2009.133
Lirong Wang, ①
Sujian Wang
protection, but RU comparison between SPP and SPP with
DiR is not considered. In our study, a novel SPP algorithm
with DiR, which is called SPP-DiR algorithm, is proposed
and its RU is studied.
The rest of the paper is organized as follows. Section 2
gives the SPP-DiR network model. Section 3 presents a
novel SPP-DiR algorithm. Section 4 gives the simulation
results. Section 5 concludes the paper.
II. THE SHARED PATH PROTECTION ALGORITHM WITH
DIR (SPP-DIR) NETWORK MODEL
The basic idea of SPP is that when two primary paths
are link disjoint and their corresponding backup paths have
overlapped links, then these overlapped links can be shared.
Although SPP saves backup resources, it does not adapt to
the diversity demands of customer requests, so for the
purpose of considering different customer requests, SPP is
studied from the point of DiR in this paper.
Fig. 1 shows an example to illustrate SPP-DiR network
model. We assume that there are two primary paths p1 and
p2 and their corresponding backup paths b1 and b2. These
two primary paths correspond to two different customer
requests. There are three circumstances: first is that when
the reliability of customer request is smaller than the
reliability of primary path, backup path is not needed;
second is that when the reliability of customer request is
bigger than the reliability of primary path and is smaller
than joint reliability of primary and backup path,
overlapped backup link C-D can be shared; third is that
when the reliability of customer request is bigger than the
reliability of primary path and is bigger than joint reliability
of primary and backup path, customer request is rejected.
We can conclude from above mentioned three
circumstances that the first circumstance can save backup
resources because backup path need not to be established;
the second circumstance can share backup link because
joint reliability of primary path and backup path is bigger
than the reliability of customer request, and primary paths
are link disjoint, thus backup resources are saved; the third
circumstance rejects the customer request, thus backup
resources are also saved, but blocking probability are
increased correspondingly (RU is considered in this paper,
while blocking probability is not included).
Figure 1. Sample network
III. PROPOSED NOVEL SPP-DIR ALGORITHM
A. Problem Definition
The network topology is G (N, L, W) for a given WDM
network, where N is the set of nodes, L is the set of fiber
links, and W is the set of available wavelengths per fiber
link. Numbers |N|, |L| and |W| denote the node number, the
link number, and the wavelength number respectively. The
following are some important notations: j, fiber link in G;
costj, dynamic cost for link j, determined by the current
network state; crn, connection request n; pn, primary path
for crn; bn: backup path for pn; rj, reliability of link j; r(q),
reliability of path q, and r(q)=Π∀j∈qrj; r(pn,bn), joint
reliability of primary path pn and backup path bn, and
r(pn,bn)=r(pn)+(1-r(pn))r(bn); ru, reliability required by
customers.
We assume each link of network has a random value of
reliability and Fig. 1 is taken as an example for the purpose
of illustrating above mentioned notations. When connection
requests of customer arrive dynamically, if r(p1)>ru and
r(p2)>ru, b1 and b2 need not to be established; if
r(p1)<ru<r(p1,b1) and r(p2)<ru<r(p2,b2), the overlapped
link C-D between b1 and b2 can be shared; if r(p1)<ru and
r(p2)<ru, and r(p1,b1)<ru and r(p2,b2)<ru, customer
requests are rejected.
B. Link-Cost for Computing Path
After illustrating all the circumstances of selecting path,
how to select a path is considered. In SPP algorithm,
sharing backup resources are only considered, so Dijkstra
algorithm is applied to select primary path and backup path.
While in SPP-DiR algorithm, three different circumstances
are considered because of different reliability of customer
request, so new link cost is designed to meet DiR demand.
New link cost is shown in (1). Among (1), rj denotes the
reliability of link j. Logarithm is used to make link cost be
additive and -1 is multiplied to make link cost be positive (rj
is between o and 1), so the standard shortest path algorithm
(Dijkstra algorithm) can be applied to compute the minimal
cost path. It is obvious that the path with minimal cost will
335
be the path with maximal reliability. We assume connection
request crn arrives at a given time. Adjust link cost
according to (1) before primary path is computed (if link j
has not spare wavelength, then link cost is a maximum),
then Dijkstra algorithm is applied to compute primary path.
Adjust the link cost according to (1) before backup path is
computed (if link j is in the primary path or link j has not
spare wavelength, then link cost is a maximum), then
Dijkstra algorithm is applied to compute backup path.
cos t ∀j∈L = − log r j (1)
C. Procedure of SPP-DiR Algorithm
1) Compute primary path pn
a) Adjust link cost according to (1);
b) Compute primary path pn using Dijkstra
algorithm;
c) If pn is found and r(pn)>ru, establish a connection,
update the network state, and return pn;
d) If pn is found and r(pn)<ru, go to step (2);
e) If pn is not found, block the connection, update the
network, and return NULL.
2) Compute backup path bn
a) Adjust link cost according to (1);
b) Compute primary path pn using Dijkstra
algorithm;
c) If bn is found and r(pn,bn)>ru, establish a
connection, update the network state, and return
(pn,bn) (backup resources sharing is carried out
in this step);
d) If bn is found and r(pn,bn)<ru, or bn is not found,
block the connection, update the network, and
return NULL.
IV. SIMULATION RESULTS AND ANALYSIS
In our simulation, two typical physical topologies are
adopted: the 14-node NSFNET network with 21
bi-directional links (as shown in Fig. 2) and the 16-node
Mesh-torus network with 32 bi-directional links (as shown
in Fig. 3), and each link of these two topologies has 4
wavelengths. Assume the network nodes have full
wavelength conversion capacity.
In these two topologies, the reliability of each link is
randomly distributed between 0.96 and 1, and the reliability
required by customers is randomly distributed between 0.95
and 0.99[9]. Connection requests arrive according to an
independent Poisson process with arrival rate β, and the
connection holding time is negative, exponentially
distributed 1/μ, namely, the network load isβ/μErlang.
Fig. 4 shows occupancy numbers of backup resource in
SPP algorithm and SPP-DiR algorithm. We can find out that
backup resource numbers increase gradually and backup
resource numbers in SPP algorithm are bigger than backup
resource numbers in SPP-DiR algorithm as the network
load increases. The reason is that SPP-DiR algorithm not
only considers the backup resources sharing as same as the
SPP algorithm, but also has possibility which backup path
need not to be established. Although the selected paths in
SPP-DiR algorithm may be longer than selected paths in
SPP algorithm (that is, the backup resource occupancy in
SPP-DiR is larger than the backup resource occupancy in
SPP), as a result of the possibility which backup path need
not to be established in SPP-DiR algorithm, so the backup
resource numbers in SPP-DiR algorithm is smaller than
backup resource numbers in SPP algorithm, that is, RU in
SPP-DiR algorithm is higher than RU in SPP algorithm. Fig.
5 shows the higher RU than Fig. 4 because each node
degree in Fig. 3 is greater than each node degree in Fig. 2. It
is possible that connection request can select a higher
reliability primary path in Fig. 3 than in Fig. 2, so the
probability which backup path need not to be established in Figure 4. Backup resources vs. network load for NSFNET
Fig. 3 is greater than the probability in Fig. 2.

Figure 2. 16-node, 21-link NSFNET

Figure 5. Backup resources vs. network load for Mesh-torus

V. CONCLUSIONS
In this paper, we studied shared path protection with
differentiated reliability in WDM optical networks, and
proposed a novel dynamic shared path protection algorithm
called SPP-DiR. Simulation results show that backup
resource occupancy in SPP-DiR algorithm is smaller than
backup resource occupancy in SPP algorithm, that is,
SPP-DiR algorithm has higher RU than SPP algorithm.
ACKNOWLEDGEMENT
This research is supported by Science Research Project
Figure 3. 16-node, 32-link Mesh-torus Program of Hebei Education Department (No. 2008110).
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 2009 International Conference on Computer Engineering and Technology

Generalized Program Slicing Applied to Z Specifications

Fangjun Wu
School of Information Technology
Jiangxi University of Finance and Economics
NanChang, 330013, China
wufangjun@jxufe.edu.cn

Abstract
Program slicing, a well-known technique,
originally introduced by Weiser M., has already been
widely used in numerous software engineering areas.
With the development of research, the concept of
program slicing is extended, and its application
domains are broadened. This paper proposes a method
for slicing Z formal specifications based on relations
selection, relations projection and relations connection.
This method does not take any time to construct
dependence graphs, which decreases chance of making
a mistake. Finally, it is applied to automate theorem
proof, one of main artificial intelligence activities.
1. Introduction
With the development of research, the concept of
program slicing, an important technique to analyze and
[1]
understand programs , is extended, and
application domains are broadened. Specification
slicing is a generalization of program slicing,
[2]
introduced by Oda T. and Araki K in 1993 year. It
allows designers to analyze dependencies and perform
[6-8]
& Yi T. , and Ingo Bruckner & Heike Wehrheim
[9,10]
.
Although specifications slicing have arose some
attentions and resolve specifications slicing to some
degree, some important features of Z language are
omitted in these literatures, such as Oda T. & Araki K.
omitted control dependences, and ignores calling
context of called schemas; Leminen J. limits form of
predicates and ignores declaration part of schemas, and
so on.
Current existing specification slicing methods still
fail to solve specification slicing. As more and more
specifications become formal, development of
techniques and tools to support analysis of formal
specifications is urgently required.
2. Generalized program slicing applied to
Z specifications
its In the following part, we will slice Z formal
specifications by virtue of relational algebraic calculus
[11]
. For the purpose of discussion and self-contained,
we will introduce some notations [12,13], such as relation
selection (denoted by V), relation projection (denoted

–
testing and debugging activities at the specification
by ), relation joint (denoted by f).
level.
There are many researchers on formal For a schema s in Z formal specification fs, three
[2]
specifications slicing, such as Oda T. & Araki K. , kinds of relations are defined.
[3,4] [5]
Chang J. & Richardson D. J. , Leminen J. , Wu F. (1) Binary relation MOD(s): contains predicate p and

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DOI 10.1109/ICCET.2009.123
 variables that possibly directly defined in p;
step 4 For all variables {vi1 , vi 2 ,..., vik } defined at p,
(2) Ternary relation USE(s): contains predicate p and
variables that possibly directly defined and used
in p; – 1, 3
((V 1 ' p 'š
{vi 1 ,vi 2 ,...,vik }Ž 2
MOD( s ) )f USE ( s ))
1 1š
are
2‹ 2
(3) Binary relation CONTROL(s): contains predicate
p and predicates that p possibly directly control computed to find possible use of variables when
depends.
variables {vi1 , vi 2 ,..., vik } are defined. The first and
For the given slicing criterion C=<p, V>, the Z

formal specification slice fsC can be computed by third columns of new relation are input as new slicing

algorithm 1, where p is a predicate and V is a set of criterion NewC. Subsequently, the Z formal
variables in the Z formal specification, s is a schema in
specification slice fs NewC with respect to NewC has
the Z formal specification.

Algorithm 1 A novel approach to slicing Z formal to compute. Then fsC fsC ‰ fs NewC .
specification fs with respect to slicing criterion C=<p,
V> step 5 For all variables {vi1 , vi 2 ,..., vik } defined at p,

step 1 fsC =‡. look up in MOD() to find whether they are defined at

step 2 If all variables of V are before-state variables, other sites or not. If they are defined at other sites, then

then the slice result fsC is empty; otherwise go to add those predicates to the slice result fsC . Those

step 3. predicates and variables {vi1 , vi 2 ,..., vik } form new

step 3 If some variables {vi1 , vi 2 ,..., vik } are defined slicing criterion NewCp. At the same time the Z formal

at p, others {v j1 , v j 2 ,..., v jl } are not defined at p, specification slice fs NewCp with respect to new

where {vi1 , vi 2 ,..., vik } ‰ {v j1 , v j 2 ,..., v jl } V , slicing criterion NewCp is computed. Then

fsC fsC ‰ fs NewCp .

{ v i 1 , v i 2 ,..., v ik } and {v j1 , v j 2 ,..., v jl }

cannot be both empty. That is to step 6 For all variables {v j1 , v j 2 ,..., v jl } not defined

V1 MOD( s ) z I
say ' p 'š
{vi 1 ,vi 2 ,...,vik }Ž 2
and at p, – (V1 v ji 2 MOD( s )) is computed to find

V1 ' p 'š MOD( s ) I . Then possible predicates that define them, where 1 d i d l .
{v j 1 ,v j 2 ,...,v jl }Ž 2

Those predicates and variables {v j1 , v j 2 ,..., v jl } form

fsC fsC ‰ { p} ‰ –2 (V 1 ' p 'CONTROL( s ))
new slicing criterion NewCnp. At the same time the Z

339
 specifications with respect to slicing criteria <pi,
formal specification slice fs NewCnp with respect to
checked_out> and <pj, stock>, where i and j means any
new slicing criterion NewCnp is computed. Then predicate that contains variables checked_out and
stock.
fsC fsC ‰ fs NewCnp .

step 7 If operation schemas are sliced, then system

invariant has to be considered. If some variables of V

also exist in system invariant, then system invariant

should be contained in the slice result fsC , otherwise

not.

step 8 If state schemas are sliced, then system

invariant has already been considered. Therefore,

Proof:
system invariant need not be dealt with separately. Firstly, slicing schema Lib with respect to slicing
criteria <pi, checked_out> and <pj, stock>, where i and
j means any predicate that contains variables
3. Theorem proof application
checked_out and stock.

One of the obvious advantages of formal methods
[13,14]
is automated theorem proof . How to decrease
search space is still a problem needed to solve. This
paper will try a novel method to theorem proof on the
According to algorithm 1, relations MOD (Lib),
CONTROL (Lib) and USE (Lib) are computed and
shown as Table 1, Table 2 and Table 3 respectively.

basis of formal specifications slicing. The process of Using Algorithm 1, we get the slice results as {p2, p3,
finding the preconditions that affect conclusions is the
p4, p5, p6} and {p3, p7} respectively.
process of slicing specifications.
[13]
For example, in schema Lib of library system ,
we consider whether variable available can be Table 1 Relation MOD (Lib)
represented by stock and checked_out or not, which Predicates Variables that possibly directly defined
can be described by Theorem 1. p1 staff, borrowers
p2 checked_out
Theorem 1 Libavailable=stock\(dom p3 checked_out, available
checked_out). p4 checked_out, available
p5 checked_out
Theorem 1 means that the numbers of available
p6 checked_out
borrow books is equal to the difference of those of all
p7 book_info
books and those of current borrowed books. The
process of proving theorem 1 is the process of slicing

340
 = (available ˆ (dom checked_out))

Table 2 Relation CONTROL (Lib) ‰(available\(dom checked_out)) (5)

Predicates Predicates that possibly directly be = available (6)
control depended
=The left side of Theorem 1
p1 ‡
p2 ‡ For equation (1), (dom
p3 ‡ checked_out)‰available=stock, replace stock with
p4 ‡
(dom checked_out)‰available equation getting (2).
p5 ‡
p6 ‡ According to distribution law
p7 ‡
(U‰W)\V=(U\W)‰(W\V), equation (2) is transformed

Table 3 Relation USE (Lib) into equation (3).

Predicates Variables that Variables that For equation (4), (dom
possibly directly possibly directly used
checked_out)ˆavailable=I, replace I with (dom
defined
p1 staff borrowers checked_out)ˆavailable getting equation (5).
p2 checked_out staff, borrowers
According to definition of intersection and
p3 checked_out available, stock
p3 available checked_out, stock difference operations (UˆV)‰(U\V)=U, equation (5)

p4 checked_out available is transformed into equation (6).

p4 available checked_out
Thus Theorem 1 is settled.
p5 checked_out p, maxbooks
p6 checked_out Last_checked_out
p7 Book_info stock 4. Conclusions

This study proposes a method for slicing Z formal

Secondly, specifications based on relation calculus, and applies it
to automated theorem proof. Automated theorem proof
the right side of Theorem 1
is one of main artificial intelligence activities. How to
=stock\(dom checked_out) (1) apply specifications slicing to automated theorem

=((dom checked_out)‰available) proving is still in the initial step, and needs deeply
study, moreover it has practical meaning.
\(dom checked_out) (2)

=((dom checked_out)\(dom checked_out)) Acknowledgements

‰(available\(dom checked_out)) (3)
This research has been supported by the Natural
=‡‰(available)\(dom checked_out)) (4) Science Foundation of Jiangxi (Project No.
2007GQS0495) and Science and Technology

341
Foundation of the Education Department of Jiangxi [10] Bruckner I., Wehrheim H. Slicing object-Z
(Project No. GJJ08353 and [2007] 434). specifications for verification. LNCS, Vol.3455, 2005,
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http://www.ics.uci.edu/~djr/abstracts/slicing-ta.html
[5] Leminen J. A. Slicing and slice based measures for the
assessment of functional cohesion of Z operation
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[9] Bruckner I. Slicing CSP-OZ specifications. In:
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Proceedings of the 16 Nordic Workshop on
Programming Theory. Oct 2004, pp.71-73.

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 2009 International Conference on Computer Engineering and Technology

PC Based Weight Scale System with Load Cell for Product Inspection

Anton Satria Prabuwono, Habibullah Akbar Wendi Usino

Faculty of Information and Comm. Technology Faculty of Information Technology
Locked Bag 1200, Ayer Keroh, Melaka, Malaysia Budi Luhur University, Jakarta, Indonesia
e-mail: {antonsatria, habibullah_ra}@utem.edu.my e-mail: wendiu@bl.ac.id

Abstract—Personal computer (PC) based weight scale system is manufacturing inspection. For the initial experiment, we
a digital measurement system for checking the weight of apply static weighing process in our laboratory.
product in manufacturing process. The system will connect
directly to the measurement device and will display the result
automatically. The load cell has been applied as a II. WEIGHT SCALE SYSTEM FOR INSPECTION
measurement sensor. The interface allows users to interact In the area of mass production, products are weighed
with the system. A software user interface along with the signal using industrial weighing systems, which are machines that
conditioning and data acquisition hardware has been weigh a product dynamically. Fig. 1 describes product flow
fabricated. The system has been successfully tested to obtain
and the weighing process in the manufacturing environment.
various load measurements. A few suggestions have also been
drawn up for the improvement of the system. As the system is
easily customizable, its seamless integration is always possible
in a highly automated industry. Thus, there is a great potential
for this system to be used in numerous industries, where load
measurement form is a part of the manufacturing process.

Keywords-load cell; weight scale; inspection

I. INTRODUCTION
Measuring load is an important and essential part of
many industrial and commercial operations. It is crucial to
have accurate measurements of load, as small errors,
Figure 1. Product flow and weighing process.
occurring repeatedly, can lead to substantial loss of revenue.
There is a constant need for knowing the exact weight of The weight of the product is estimated while the product
many items, e.g. food, ingredients for production, has been carried over a weigh table by a transport system.
pharmacology, chemistry, technology, etc. The type and the Normally, the transport system is a conveyor belt type. The
number of products that require weight control are weigh table is mounted on a load cell, which is
increasing. Consequently, the legal requirements of uncontrollable weighing device capable of weighing an
government bodies internationally are trying to maintain the article. A signal processing module acquires the electrical
same constant pace. In production, this means high accuracy signal from weighing device and estimates a value of weight
and efficiency of weighing are also constantly high on the for the passing product as its output [5].
agenda. Continuation of this trend brings benefits for both
the customer and the producer. The manufacturing III. SYSTEM DEVELOPMENT
efficiency is increased and hence profitability whilst PC based weight scale system is proposed to ease the
package quality and quantity are assured to the customer’s product inspection in controlling the weight of product
satisfaction [1]. directly to the computer after the measurement. Fig. 2
Having in mind industrial weighting applications where provides the theoretical approach of weight measurement
6,000 divisions are needed [2] for the equipment (external system [6].
divisions) a conversion resolution of at least 60,000
divisions (10 internal divisions for each external) with 50 or
more readings per second, at least for static weighting
applications [3]. For dynamic weighting a faster reading rate
is required but with lower resolution [4]. In this paper, we
intend to develop PC based weight scale system that
provides weight information of product in the

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DOI 10.1109/ICCET.2009.182
 TABLE I. LOAD CELL SPECIFICATION
Rated 5 Kg Sensitivity 1.9452 mV/V
capacity
Combined <0.03% R.O. Zero balance 0.013 mV/V
error
Zero tempco 0.03% Rated output 0.03%
R.O./10°C tempco R.O./10°C

Input 403.57 Ω Output 350.02 Ω

impedance impedance
Insulation 2000 MΩ @ Maximum 15V
Figure 2. Theoretical approach of measurement. impedance 50 VDC excit. voltage
Temperature -20°C ~ 60°C Safe overload 150% R.L.
The main purpose of this system is to generate the range
Creep <0.03% R.O. Non <0.03% R.O.
weight information that is placed on the load cell. The (30min) repeatability
output is a differential voltage, which is dependent on the
transfer function of the cell itself. According to this transfer 2) Instrumetation amplifier: An instrumentation
function, the load cell provides a certain voltage for a amplifier is a type of differential amplifier that has been
certain load level. The output of the load cell is then outfitted with input buffers, which eliminate the need for
amplified using a differential instrumentation amplifier. The input impedance matching and thus make the amplifier
instrumentation amplifier output signal is fed to an analog to particularly suitable for use in measurement and test
digital converter (ADC) that will provide digital output. equipment. Additional characteristics include very low DC
Furthermore, the data will be programmed in offset, low drift, low noise, very high open-loop gain, very
microcontroller. Finally, this digital information is sent to a high common-mode rejection ratio, and very high input
PC directly via the standard serial port [7]. In this research, impedances. Instrumentation amplifiers are used where
the system is divided into hardware and software great accuracy and stability of the circuit both short and
frameworks. long terms are required.
A. Hardware Framework Although the instrumentation amplifier is usually shown
schematically identical to a standard operational amplifier
The hardware consists of load cell sensor,
(op-amp), the electronic instrumentation amplifier is almost
instrumentation amplifier, ADC, microcontroller, serial port
always internally composed of three op-amps. These are
communication and PC.
arranged so that there is one op-amp to buffer each input (+,
1) Load cell: A load cell is a transducer which converts −), and one to produce the desired output with adequate
force into a measurable electrical output. The most common impedance matching for the function.
weight scale implementation uses a bridge type load cell The most commonly used instrumentation amplifier
sensor, with voltage output directly proportional to the circuit is shown in the Fig. 4. The gain of the circuit is:
weight placed on it [8]. A typical load cell bridge is
illustrated in Fig. 3. Vout ⎛ 2 R1 ⎞⎟ R3
= ⎜1 +
V2 − V1 ⎜⎝ Rgain ⎟⎠ R2 (1)

The ideal common-mode gain of an instrumentation

Figure 3. Basic circuit of load cell.
The detailed specification of this load cell can be found in
the Table 1.
amplifier is zero. In the circuit shown, common-mode gain
is caused by mismatches in the values of the equally-
numbered resistors and by the non-zero common mode
gains of the two input op-amps. Obtaining very closely
matched resistors is a significant difficulty in fabricating
these circuits, as optimizing the common mode performance
of the input op-amps.
Instrumentation amplifiers can be built with individual
op-amps and precision resistors, but are also available in
integrated circuit (IC) form from several manufacturers. An
IC instrumentation amplifier typically contains closely
matched laser-trimmed resistors, and therefore offers
excellent common-mode rejection. This project used OP
07C as instrumentation amplifier.

344

Figure 4. Typical instrumentation amplifier.
3) Analog to digital converter (ADC): ADC is a device
that converts continuously varying analog signals from
instruments and sensors that monitor conditions into binary
code for the computer. The 8-bit ADC 0804 IC considered
able meet the requirement from network that will be made.
An ADC takes analog voltage as an input and provides a
digital output. This digital information is in binary format
consisting of 0s and 1s only. The ADC finds a fractional
binary number that gives the closest approximation to the
fraction formed by the input voltage and reference. The
following is the conversion equation [9]:
V x = Vref ( a1 2 −1 + a 2 2 −2 + a3 2 −3 + ..... + a n 2 − n )
Where, Vref is the ADC reference voltage, Vx is the input
analog voltage and n is the number of bits.
It is noteworthy that there is an inherent uncertainly in
the input voltage producing a given ADC output, and this
uncertainty is given by:
ΔV = Vref 2 − n (3) graphical user interface (GUI) which is developed using
Therefore, it is very important to take this into account
when using an ADC in industrial instrumentation and
control systems, and in defense sector applications, where
100% accuracy is required. Fig. 5 shows simple schematic
diagram of ADC 0804.
Figure 5. Schematic diagram of ADC 0804.
4) Microcontroller: The AT89S51 provides the
following standard features: 4K bytes of flash, 128 bytes of
RAM, 32 I/O lines, two 16-bit timer/counters, five-vector
two-level interrupt architecture, a full duplex serial port, on
chip oscillator and clock circuitry. In addition, the AT89S51
is designed with static logic for operation down to zero
frequency and supports two software selectable power
saving modes. The idle mode stops the CPU while allowing
the RAM, timer/counters, serial port and interrupt system to
continue functioning. The power down mode saves the
RAM contents but freezes the oscillator disabling all other
chip functions until the next hardware reset.
5) Serial port:Recommended standard-232 (RS-232) is
a standard interface approved by the Electronic Industries
Association (EIA) for connecting serial devices [10]. In
other words, RS-232 is a long established standard that
describes the physical interface and protocol for relatively
low-speed serial data communication between computers
and related devices. An industry trade group, the EIA,
defined it originally for teletypewriter devices. The serial
ports on most computers use a subset of the RS-232C
standard. Fig. 6 shows RS-232 for the 9-pin D-type
connectors.
(2)
Figure 6. The 9-pin D-type of RS-232 connector.
B. Software Framework
The software frameworks consist of database and
SQL server 2005 and visual C++ 2005. The software system
uses COM port to receive digitized signal from
microcontroller and convert the hexadecimal data into
decimal format. Fig. 7 shows the user interface of PC based
weight scale system.
Figure 7. User interface of the system.
345
 IV. RESULT REFERENCES
The performance of PC based weight scale system has [1] Halimic, M., Balachandran, W. and Enab, W., “Fuzzy Logic
been verified with static process environment. In this Estimator for Dynamic Weighing System”, Proc. The 5th
environment, there will be test on hardware and application IEEE Int. Conf. on Fuzzy Systems, Vol. 3, 1996, pp. 2123-
design to run together before introducing them into the real 2129.
[2] A 20-bit (1ppm) Linear Slope Integrating A/D Converter,
application. Fig. 8 presents the linearity of the load cell Application Note 260, National Semiconductor.
sensor that has been applied in the system. The graphic [3] A High Performance Industrial Weighting System,
shows that the system is very linear and suitable to be used Application Note 295, National Semiconductor.
to measure weight of a product. The system is able to [4] Higino, J. and Couto, C., “Digital Filtering in Smart Load
measure the weight up to 5 kilogram. Cells”, Proc. the 21st IEEE Int. Conf. on Industrial
Electronics (IECON), 1995, pp. 990-994.
[5] Halimic, M. and Balachandran, W., “Kalman Filter for
Dynamic Weighing System”, Proc. the IEEE Int. Symposium
on Industrial Electronics (ISIE), Vol. 2, 1995, pp. 786-791.
[6] Panthala, S., Islam, N. and Habib, S. A., “Automated
Industrial Load Measurement System”, AU Journal of
Technology, Vol. 10, No. 1, 2006, pp. 23-28.
[7] Prabuwono, A. S. and Akbar, H., “Automated Weight
Measurement System Based on Serial Communication”, Proc.
the 3rd Int. Conf. on Mechatronics (ICOM), 2008.
[8] Johnson, C. D., Process Control Instrumentation Technology,
7th Ed., Prentice-Hall, Englewood Cliff, NJ, USA, 2003.
[9] Boylestad, R. L. and Nashelsky, L., Electronic Devices and
Circuit Theory, 7th Ed., Prentice-Hall, Englewood Cliff, NJ,
Figure 8. Linearity of the load cell sensor. USA, 1999.
[10] http://cnx.org/content/m12293/latest/
A complete design of PC based weight scale system has
been proposed, describing both hardware and software.
However, it is possible to enhance the system further. The
following suggestions are made for the improvement of the
system: first, the range of measurement can be expanded to
cover the weight of product in real manufacturing. Second,
analog to digital converters with higher resolution (e.g., 16-
bit devices) can be applied to increase the accuracy of the
measurement system. Third, experiment can be done with
the dynamic environment of weighing process that simulates
the manufacturing environment as closely as possible.
V. CONCLUSION
As the conclusion, this paper has contributed to the
development of a PC based weight scale system for the
purpose of product inspection. A software user interface, the
signal conditioning and data acquisition hardware are
successfully developed where the system can obtain various
load measurements. A few suggestions have also been
drawn up for the improvement of the system.
The system is seamless integration and always possible
in a highly automated industry. Thus, there is a great
potential for this system to be used in numerous industries,
where load measurement form is a part of the manufacturing
process.
ACKNOWLEDGMENT
The authors would like to thank Technical University of
Malaysia Melaka (UTeM) for financial support and Afzan
Omar for her contribution in this research.

346
 2009 International Conference on Computer Engineering and Technology
Short-Term Electricity Price Forecast Based on Improved Fractal Theory
CUI Herui YANG Li
Department of Economics and Management, North China Electric Power Universiy, Baoding,
071003, China
cuiherui1967@126.com
Abstract
This paper discusses the basic principle of the
fractal theory and the basic thinking of how to use it to
forecast the electricity price, then builds electricity
price forecasting model based on improved fractal
model and solve the model on this basis. Finally, this
fractal model is used to forecast the short term
electricity price.
1. Introduction
With the marketing trend of the global electricity,
electric power industry is growing gradually from the
perfect monopoly to competition , also as the core
factor of electricity market, the electricity price has
changed correspondingly, it is determined by the
gaming of participants of electricity market, no longer
by government's administrative order. Cause
participator's benefit directly affected by electricity
price, electricity price forecast has important
significance to every participator, as for government
and the regulation organs, it is used to help to make
macro economic policy; for investor, it is very
important to decide investment; for generation
companies who gain profit through electricity price
bidding, it is also the life-and-death. Form every
meaning; electricity price forecast is one of the most
problems of electricity market which is urgent to be
resolved.
At present, the researches of electricity price
forecast at home and abroad can be divided into two
kinds: mathematical model and market simulation.
Though market model contains the structural details of
the whole market and the forming mechanism of power
price of this market, this method's lots of market data
need become the limitation of it's application in power
system. Mathematical model mainly has several kinds
as following: (1) forecast based on time series analysis,
such as autoregressive model, autoregressive moving
average model, threshold autoregressive model. (2)
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.73
forecast based on factor analysis, such as linear
regression method, artificial neural network method,
model matrix and comprehensive evaluation method.
(3) forecast based on econometrics, such as Holt and
Winters’s equation, Markov theory of economic
prediction. (4) forecast based on Combination. [1][2]
This paper tries to use a new method named fractal
theory to forecast the electricity price.
Taking non-regular and nonlinear thing as research
object, studying on the regularity, hierarchy and scale
invariance hidden in complex natural and social
phenomenon, crossing the natural science, social
science and thinking science, fractal theory is a new
method to study on the complex object. Based on the
self-similarity between part and whole, fractal theory
establishes the global property from the part, along the
direction from microscopic to macroscopic. This
theory takes the relationship between regional and
global as research object, emphasizes the global's
dependence to part, and searches for the bridge through
it can deduce and transient from part to whole. As the
whole is made up by the part, we can understand the
whole through part. One of the fractal theory
philosophical foundations is that the unit and element
of the system contains the property and information of
the whole system. Self-similarity and scale invariance
are the base of fractal theory. [3][4][5]
2. Modeling of electricity price forecast
The change of commodity prices, including stock,
has fractal feature. [6] [7] The distribution about
commodity prices per unit time obey the symmetry and
stable distributions with special dimension, so the
fractional theory can be used for price analysis and
prediction. [8][9][10] And the same to the electricity
price. So analyzing the regularity of the electricity
price using fractal method, the research can start from
the electricity price itself, other than the electricity
price's influencing factors.
347
2.1. Analysis of the fractal property of N—related object's number, such as price, index,
electricity price based on R/S etc;
C—undetermined constants;
Before the construction of the fractal model to D—fractal dimension.
predict price, the first must to do is to judge whether Usually, take the numbering of time as “r”,
the data has the fractal property and can use the fractal r1=1,…and rn=n, and take electricity price value as
model. This paper uses the R/S analysis, and it’s “N”, N1 is equal to electricity price value when r1=1,
theoretical calculations are as follows. …and Nn is equal to electricity price value when rn=n.
Take {Xi}(i =1,2,…,n)as a time series, the average At present, in the general application of fractal
_ method, D is constant, and this type of fractal method
of it is X ,and the standard deviation is S. is called constant dimension fractal, manifested as a
_
1 n
X= ∑Xi
n i =1
(1) line in double logarithm coordinate system. Any two
data points (Ni,ri)and(Nj,rj) on the straight line
2
determine the fractal parameters, namely fractal
1 n ⎛ _
⎞
S= ∑
n i =1 ⎝
⎜ Xi − X ⎟
⎠
(2) dimension D and constant C of this line.
⎡N ⎤
ln ⎢ i ⎥
(8)
X(t,n) is cumulative deviation,
D = ⎣⎢ N j ⎦⎥ = ln N i − ln N j
t
⎛ _
⎞ ⎡r ⎤ ln r j − ln ri
X (t , n ) = ∑ ⎜ X i − X ⎟ (3) ln ⎢ i ⎥
i =1 ⎝ ⎠ ⎣⎢ r j ⎦⎥
Definite the difference between the maximum and
C = Nir D (9)
the minimum obtained by the formula as R,
As negative number can not take logarithm , there is
R = max X (t , n) − min X (t , n) (4) a need to add a constant to all the N value to eliminate
1≤ t ≤ n 1≤ t ≤ n
And the rescaled range analysis is the negative number , in another word, there is a need

(R / S)n = (c × n)H
(5)
Calculate the logarithm of the above equality,
Log(R / S )n = Log(a ) + H × Log(n) (6)
The slope between Log(R/S) and Log(n) obtained
through the linear regression is Hurst index. The
introduction of Hurst index is to determine the random
distribution property and the fractal property of the
data sequence.
When H=0.5, time series show the property of
random and independent, and that is a sequence of
random walk. When 0 ≤ H <0.5, the time series is anti-
persistence sequence, also known as "mean eversion".
When 0.5<H ≤1, the time series show the trend of an
enhance persistent sequence, and have the property of
fractal.
2.2. Model construction
2.2.1. Related formula of fractal distribution. Power
exponent distribution can be used to describe and
express fractal distribution.
C distribution, in fact, in the practical application, what
N= (7)
rD
In the formula:
r—Characteristic Linearity, such as time, length, etc;
to make a translation for all the N value firstly, when
some of Ni are negative numbers. Sometimes, such
translation is made firstly for a better analysis and
predication.
This paper uses variable dimension fractal model to
get over the weakness of the above constant dimension
fractal method, that it is unable to deal with the non-
linear functional relationship in double logarithm
coordinate system.
Now it will show that any functional relationship
N=f(R), between N and R, can be transformed into
C
fractal distribution N = , make:
rD
C
f ( R) = (10)
rD
So,
r = [C f ( R )]
1D
(11)
That is, through the transformation above, R gets the
form of fractal distribution.
2.2.2. Transformation. Although any functional
relationship can be transformed into the form of fractal
have known are just some data points, not the
functional relationship, so can not get the concrete
form.At this time, it can make a series of

348
transformations called cumulative sum, and choose a value have good linear relation, so as to the D and C
transformation, through which the data can be in fairly value. In this paper, the D and C value is calculated by
good agreement with fractal distribution model, and curve fitting analysis according to the
can be treated with fractal distribution. Transformation TABLECURVER firstly, other then take the fractal
method as following: parameters as fixed.
First, drawing the initial data point,(Ni,ri) (i The process of model construction as follows:
=1,2,…,n), on the double logarithm coordinates, in The first and the second step is the same as the
general case, these data points cannot be in fairly good originally model.
agreement with fractal distribution model, so, Third, calculate the D and C value through curve
arranging these data into a basic series. fitting analysis according to the Tablecurver;
{Ni} = {N1,N2,N3…} (i =1,2,…,n) (12) Finally, calculate the S value and forecast the
Based on the above basic series, other series can be electricity price correspondingly.
constructed. Such as to construct the first order
cumulative sum series S1, just need to make S11= N1, 3. Empirical Analysis on electricity price
S12= N1+N2, S13= N1+N2+N3,and so on. The Second forecast
Order, third order can be made as the same, that is:
{S1i}= {N1, N1+N2, N1+N2+N3…}(i =1,2,…,n) (13)
3.1. Judgment of the fractal property of
{S2i} = {S11,S11+S12,S11+ S12+ S13…}(i =1,2,…,n) (14)
Second, establish the fractal model of the cumulative electricity price
sum series. Taking the first order cumulative sum
series S1 as example, draw the data (S1i,ri)(i The following is the test sequence based on HURST
=1,2,…,n)on the double logarithm coordinates to get a index constructed by the price value from January
discrete fractal distribution model. According to the n 1th,2007 to October 31th, 2007 including the price of
data points, what obtained is a segment constant 300 trading days, and divided into 20 groups labeled as
dimension fractal model made up of n-1 line, and Ia(a=1,2,3…20) with the length 15, that is n=15.
calculating the fractal parameters of every line Element in every subinterval labeled as Nk,a ,in
according to formula (8) and (9). which a is subinterval marker from 1 to 20, and k is
Third, choose the transformation with best result, element marker from 1 to 15 in every subinterval.
15
and determine the fractal parameters correspondingly. The average of Ia is : ea = 1 ∑ N k ,a ,and the
Drawing the data points of each order cumulative sum 15 k =1
series on the double logarithm coordinates and cumulative deviations of Ia
comparing with a fractal distribution model, then can
∑ (N − ea ) ,and the range of Ia is
k
choose a transformation with best result, and determine is X k ,a = i ,a
the fractal parameters correspondingly. i =1
Finally, determine the fractal distribution model, Ra=Max(Xk,a)-Min(Xk,a),in which k is from 1 to 15.
forecast using extrapolation calculation. The standard deviation of Ia is S I ,
a
2.2.3.Improvement.This paper make a improvement
× ∑ (N k ,a − ea ) ,thus, can get the value of the
1 15
S Ia =
2
to the above mentioned model, which take a pair of
fractal parameters determined by the known data as the 15 i =1
fractal parameters of all the unknown future price rescaled range RIa/SIa, and the average of it labeled as
value to realize price forecast. (R/S)n.
The unreasonable place is that as the fractal With the increase in the value of N, repeat the above
parameters of the known data are not always able to process until N = 300.
represent the unknown future price value’ accurately, The slope obtained according to
the prediction results have bigger deviation. From the log(R/S)n=H×log(n)+log(a) and least square method is
table 1, it can get the same conclusion. D value Hurst index. After derivation, H=0.69.
presents monotonically increasing tendency, while C H>0.5 implies that today's events have a true impact
value presents monotone decreasing tendency with the on tomorrow, the data shows the fractal characteristics.
increase of r. As D and C value have obvious changes,
the above mentioned model is wrong for it take the 3.2. Prediction
fractal parameters as fixed. Through the research of
sets of different data, it can find that the lnr and D

349
 Step 1. Collect original data. Select the Day Ahead
Hourly LMP Values from the PJM of the America as
the original data to construct fractal model. The
original data and the fractal parameters of this data can
be seen in table 1.
Step 2.Calculate the D and C value through curve
fitting analysis according to the TABLECURCER and
SCILAB.
Table 2 The fractal parameters predictive value
R lnr D C
16 2.772588722 1.8269746 32.33224
17 2.833213344 1.835935 31.518581
18 2.890371758 1.8442091 30.79839
19 2.944438979 1.8518937 30.115215
Step 3.According to the S 2 = N i r and D, C,
D
calculate S2 and N value correspondingly.
Make a comparison between predictive value and
the real value. Only one day, that is 6.29, the predictive
value has big error as 16.7%, other days’ predictive
values’ error is very small. Therefore, the fractal model
is relative ideal in short term electricity price forecast,
and will be of much help for price forecasting in future.
4. Conclusions
In this paper, the fractal model is used to forecast the
short time electricity price.
First, using the R/S analysis method, this paper
analyzes the fractal characteristic of electricity price to
determine whether the price data has the fractal feature
or not. After derivation, H=0.69, implies that the time
series show the trend of enhance persistent sequence,
and have the property of fractal. So the fractal model is
can be used to do prediction.
Second, this paper establishes a fractal model based
on the cumulative sum and makes an improvement,
using the Day Ahead Hourly LMP Values as the
original data to construct forecasting model. The
comparison between the predictive value and the real
value shows that the application of the fractal theory
can improve the precision of prediction, and the fractal
model is suitable. This forecasting model possesses
350
broad application in electricity price forecasting in
power market.
References
[1] Yang Bo,Zhao Zunlian,Chen Yunping,Han Qiye,
Forecasting Methods of Day-ahead Trading Marginal Price,
High Voltage Engineering, Vol.33,No.7，2007(In Chinese)
[2]Lin Zhiling,Gao Liqun,Zhang Dapeng,Zhang Qiang,
System Marginal Price Forecast Based on the Dynamic
Neural Network, Journal of Northeastern University (Natural
Science)Vol.27,No.1，2006(In Chinese)
[3]Yin Yuliang,Li Peizheng,Kang Yuyun,Gu Zonglei, The
Development and Research Status of Fractal Theory ,Science
Information,No.15,2007(In Chinese)
[4]Li Peng, The Development and Application of Fractal
Theory, Collected Works Science and Education,
No.1,2008(In Chinese)
[5]Liu Ying ， Hu Min, She Guiying, Li Xiaobing, Liu
Xiaolin, Fractal Theory and Its Application, Jiangxi Science,
Vol.2415, No.24,2006(In Chinese)
[6]Liu Yan,Wang Li,Ma Guangwen ， On-grid Price Risk
Identification to the Generation Company，Water Power，
Vol.33.No.1，2007(In Chinese)
[7]Liu Haibo,Yi Dongyun, The Research of Stock Price
Forecast Basing on Wavelet Analysis and Fractal Theory,
Statistics and Advisory, No.5,2006(In Chinese)
[8]Lu Fangyuan, Fractal Theory and Its Application in
Economic Management, Journal of Henan College of
Financial Management Cadres,No.3,2004(In Chinese)
[9]Liu Xiaocheng,Wang Yanming, Oil Price Forecast Based
on Fractal Theory, Journal of Xi'an Shiyou University,
Vol.15, No.4,2006(In Chinese)
[10]Zhang Ying,Lu Ying, Yao Yan, The Price Forecast of
Fashion Commodity Based on Fractal Theory, Journal of
Shengyang University of Technology(Social Science
Edition), Vol.1,No.2, 2008(In Chinese)
[11]http://www.pjm.com
 Table 1.The fractal parameters
priceN
time ri S2 D C
（$/MWhr）
6.11 1 40.53 241
6.12 2 38.71 420.71 -0.803792964 241
6.13 3 40.57 640.99 -1.038486636 204.8176
6.14 4 38.99 900.26 -1.180712275 175.1896
6.15 5 35.34 1194.87 -1.268730643 155.0658
6.16 6 36.51 1525.99 -1.341618573 137.9017
6.17 7 35.61 1892.72 -1.397149649 124.8412
6.18 8 33.38 2292.83 -1.436156029 115.7161
6.19 9 31.75 2724.69 -1.465132876 108.9494
6.20 10 30.40 3186.95 -1.487366358 103.755
6.21 11 33.52 3682.73 -1.517046539 96.9011
6.22 12 34.06 4212.57 -1.54483922 90.65372
6.23 13 27.12 4769.53 -1.551357499 89.1972
6.24 14 30.03 5356.52 -1.566184306 85.86872
6.25 15 31.28 5974.79 -1.5832714 82.08258

Table 3. 6.26～6.39 Day ahead hourly LMP values predictive value

time 6.26 6.27 6.28 6.29
S2 5123.0951 5716.1397 6360.8183 7029.1321
predictive value（$/MWhr） 35.03 36.57 35.78 28.88
real value（$/MWhr） 35.42 38.59 36.97 34.7
Error（%） 1.1 5.23 3.22 16.7

351
 2009 International Conference on Computer Engineering and Technology
BBS Sentiment Classification Based on Word Polarity
Shen Jie, Fan Xin, Shen Wen, Ding Quan-Xun
Information Engineering College of Yangzhou University,
Yangzhou, JiangSu. China
fanxin2632@163.com, shenjie@yzu.edu.cn, shenye1982@163.com
Abstract
Sentiment classification is an applied technology
with great significance. It can help people find right
reviews in a more efficient way. In this paper, we
present a novel efficient method for BBS sentiment
classification. Through extracting sentiment-bearing
words from WordNet using the maximum entropy, a
ranking criterion based on a function of the probability
of having Polarity or not is introduced. The words with
polarity are selected as features, which are processed
with SVM classifier at the following step. The
experimenta1 results show that our method achieves
high performance.
1. Introduction
BBS forum has become an important place for
people to retrieve information and deliver of
comments. The reviews are mixed in terms of positive
and negative opinion polarity. If you want to read some
negative reviews on some products just to pinpoint
possible drawbacks and don’t want to spend time
reading positive reviews, you need to classify them by
sentiment. Sentiment classification is an applied
technology with great signification. Its main idea is to
represent the overall opinion polarity of a review as
positive or negative that the author would assign to.
Sentiment classification is not the same as topic
classification. Topic classification is based on the topic
keywords. The researches in this domain have been
studied extensively in the past and a number of
techniques have been proposed, such as excellent
supervised methods, feature extraction method and
feature selection mechanism[1][2][3]. Sentiment
classification is mainly based on the sentiment-bearing
words and the traditional methods of topic
classification are unsatisfactory in it. Some effective
solutions were proposed. Bo Pang[4] applied Naïve
Bayes, Maximum Entropy and Support Vector
Machine classification techniques to the identification
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 352
DOI 10.1109/ICCET.2009.13
of the polarity of movie reviews. Their best results
came using Unigrams, calculated by SVM at 82.9%
accuracy. Wei-Hao Lin[5] proposed one kind of
statistical model, distinguishing the viewpoint by
analyzing the words’ usage. Xu Jun et al. [6] used
simple Bayes and the maximum entropy method to
classify the reviews on news with relatively high
performance. Tang Hui-feng et al.[7] had considered
different methods of feature extraction, feature
selection and classifier selection. But these researches
only considered about lexical category. Because
Chinese language usage is very complicated, these
methods given by the researchers introduced many
features unrelated with author’s sentiment, and ignored
some low frequency part-of-speech with sentiment-
bearing tendency, such as interjection, adverbial.
Sentiment classification is mainly based on the
sentiment-bearing words. If we use the methods of
topic feature selection for sentiment classification, a lot
of noise features will be introduced, which affects the
classification. The maximum entropy method is widely
applied in the natural language processing domain, in
part-of-speech tagging[9], Sense Disambiguation [10]
and so on. In this paper, the semantic tendency
identification was studied. We identified whether the
words had semantic tendency by using maximum
entropy model. The words with polarity were selected
as features and our Support Vector Machine classifier
was built.
2. Identifying sentiment-bearing words
based on maximum entropy model
Maximum entropy is a general technique for
estimating probability distributions from data. The
over-riding principle in maximum entropy is that when
nothing is known, the distribution should be as uniform
as possible, that is, have maximal entropy. In this paper,
we will regard the sentiment-bearing words’
identification as a classification issue. Maximum
entropy model is used to computer (p1) the probability
that the word is a sentiment-bearing word and (p2) the
probability that the word is not a sentiment-bearing ⎧ pλ ( f / | word ) = 1/ Z λ ( word ) exp(∑ λi fi ( word , f / ))
word. If p1 is bigger than p2, the word is a sentiment- ⎪ i
⎨ (1)
bearing word. Maximum entropy model is mainly based Z
⎪ λ ( word ) = ∑/ ∑i i i
exp( λ f ( word , f /
))
on feature selection and parameter estimation. The ⎩ f

feature selection mainly seeks for the constraints,

selects the features which are important to feature Z λ ( word ) is a normalizing constant determined
∑
template of maximum entropy. Then the parameters’
value of each feature is estimated and the model is built. by the requirement that f/
pλ ( f / | word ) = 1 for
According to the BBS document’s characteristics, we
choose the appropriate feature set and label each word’s
all word . For each feature f i we introduce a
semantic tendency value. The words with polarity are parameter λi which denotes the weight. We compute
selected as features. Here the knowledge of which word
has polarity is what we should obtain by maximum the value of λi where λi achieves its maximum by
entropy. Through analyzing BBS documents, we GIS(Generalized Iterative Scaling) algorithm. Thus we
mainly consider of several kinds of factors as folling: can obtain probability distribution of each sentiment-
bearing word. The algorithm is as following:
{1. Part_of_speech
Input: document
2 .Position (where the word appears)
Output: all the sentiment-bearing words
3. Word’s length begin
4. Frequent (how many times the word appears) the document split into words using the
participle system
5. Style (whether the word is in special symbol for each word
such as {} begin
and “”}
{Suppose
f / ( word ) = 1
Firstly we hand-labele sentiment-bearing words in get the feature vector based on feature template
the training set. Each document is represented by a set λ
put the value of i according to the training
of sentiment-bearing word set θ ( d i ) . We also produce
data into maximum entropy formula(1)
the sentiment-bearing word set ϑ ( d i ) for each computer (p1) the probability that the word is a
document by the automatic generating program. We use sentiment-bearing word;}
{ Part_of_speech, Pos, Len, Fre, Style} as feature end
template. Take the word as an event and build feature begin
vector. With this constraint to features, we calculate the
{Suppose
f / ( word ) = 0
words’ most superior probability distribution
get the feature vector based on feature template
p( f / ( word j ) | word j ) by using the maximum λ
put the value of i according to the training
/
entropy model. f ( word j ) denotes the result of data into maximum entropy formula(1)
computer (p2) the probability that the word is
word j labeled. It has two kind of values. not a sentiment-bearing word; }
end
⎧1, word labeled sentiment-bearing p1 > p 2
f / ( word ) = ⎨ if
⎩0, word not labeled sentiment-bearing the word is a sentiment-bearing word
Each feature of the feature template is computed else
by f ( word， f ( word )) . The value of this function
/
the word is not a sentiment-bearing word
is 0 or 1. If the word’s feature is the same as sentiment- endfor
end
bearing words and f ( word ) = 1 , the value of the
/
Figure 1.The algorithm of identifying
function is 1. If the word’s feature is not the same as
sentiment-bearing words and f ( word ) = 0 , the
/ Sentiment-bearing words
value of the function is also 1. In other situations the
value of function is 0. 3. Svm classification based on sentiment-
According to the maximum entropy theory, we
bearing word identifying
build the model.

353
 Classifying the BBS documents based on the
sentiment-bearing words can be seen as a classification
problem involving two classes. Given a set of
documents D = {d1 , d 2 , ⋅⋅⋅, d n } . We pre-define
classes, C = {c1 , c2 } . c1 denotes the positive review
andc2 denotes the negative review. We implement
SVM classifier to label document c1 or c2 . Support
vector machines (SVM)[11] is a state-of-art supervised
kernel method for machine learning. It is successfully
to text classification. The basic idea behind kernel
methods is to embed the data into a suitable feature
space F via mapping function: X → F , and then use
a linear algorithm for discovering nonlinear patterns. In
sentiment classification, our goal is to seek an
optimum value with the restrains.
Before carrying on the classification, a document
should be represented by a set of word count features.
We use the vector space model (VSM).
V (d ) = (t1 , w1 ; t2 , w2 ; ⋅⋅⋅; tn , wn )
ti (i ∈1, 2, ⋅⋅⋅, n) . t denotes the feature. w denotes
i i
the weight of ti in d . The words which have the
sentiment-bearing tendency are selected as the features
by using the maximum entropy model.
4. Experiment
4.1. Experiment datasets
Our experiment used 2476 reviews from Yangzhou
University Venus forum in recent years. These
reviews were consisted of 1543 positive reviews and
933 negative reviews, not including neutral reviews.
For our experiment purposes, we removed the html
tags and transformed them into the unified document
format and manually labeled, 50% for training and
50% for testing, from which positive and negative are
also listed. The training documents and the testing
documents were not overlapped. We used Binary[8]
for feature weight and Chinese Academy of Science’s
ICTCLAS system for participle and part-of-speech
tagging. It is different from topic classification. Some
negative terms, exclamation terms and so on are
removed as stop words in the topic classification. But
these words often have the semantic tendency and are
not removed in the sentiment classification. We did
three experiments, respectively using the traditional
topic feature selection, lexical-category feature
selection and the method we proposed..
4.2. Experiment results
For the topic feature selection, all the features are
selected to represent documents, and the SVM classifier
is constructed to carry on the classifying. This method
is shown to achieve good results on the task of topic
classification. For the lexical-category feature selection,
we consider the probabilities which part-of-speech is
the sentiment-bearing word. We use each kind of part-
of-speech for the feature selection respectively.
Through analyzing of the BBS documents, we
discovered that the sentiment is mainly illustrated by
the noun, verb, adjective and adverb words. The
experiments were implemented respectively selecting
one from the noun, verb, adjective and adverb words
and all of them. The method we proposed is that the
maximum entropy model should be firstly instantiated
by training samples before constructing our classifier.
The parameter value is obtained by the maximum
entropy model’s feature template. When a new
document comes, it is considered as an ordered list of
words. The sentiment-bearing tendency words are
,
selected for features. The figure 2 is the maximum
entropy model used to identify sentiment-bearing
words. We can see from the figure the identifying of the
sentiment-bearing words is consist of training process
and testing process.
Figure 2.The maximum entropy model used to
identify sentiment-bearing words
We extracted 6740 words from the documents for
testing, including 2659 positive examples and 4081
negative examples. The positive example denoted that
this word has the semantic tendency; negative example
denoted that this word does not have the semantic
tendency or it is neutral. With five cross-validations on
the reviews dataset, we obtain the accuracy by using
maximum entropy model to identify sentiment-bearing
words. As table 1.
Table1.The accuracy of identifying
sentiment-bearing words by using maximum
entropy model
354

From the BBS documents, we discovered that many
words did not have the sentiment-bearing tendency. But
when they were co-occurrences with some others, they
would have the sentiment-bearing tendency. And the
training data was not all-sided. Therefore, error
happened while estimating feature parameter, which
affected accuracy of identifying sentiment-bearing
words. The sentiment classification result is in table 2.
From table 2, we can see that the accuracy of
sentiment classification by using topic feature selection
was low. Sentiment classification is mainly based on
the words with polarity. The topic feature selection that
did not separate the sentiment-bearing words from the
ordinary words introduced massive irrelevant feature
items which affected the sentiment classification. As
following, the example about a piece of review to the
movie “An American Crime” is given:
Example: It has been a long time that I had not
watched a movie which can touch me. Fortunately, I
saw the new movie “An American Crime” of this year
in that raining night which I would never forget.
Although each role in the movie was very common,
which can be found everywhere in normal life, I was
deeply moved by it.
Table 2. Results for three methods
The example above is an obvious positive attitude,
but only a few words represent sentiment tendency,
such as touch, fortunately and move. When we used
topic feature selection, a lot of non-correlated words
were introduced which affected sentiment
classification Sentiment classification is based on the
sentiment-bearing words. So we would improve the
classifier if we select words with polarity as features.
Only one kind of part-of-speech selected as
features can reduce the noise features. From table 2,
the accuracy was improved by using noun, verb,
adjective and adverb respectively. And the results of
355
using noun and verb were better than using adjective
and adverb. But linking them, classified precision is
better than any of single part-of-speech. When we
selected single part-of-speech words as features, the
features was not all-sided and the viewpoint the
reviews expressed was not distinct. So we should
combine more kinds of part-of-speech. After
combining noun, verb, adjective and adverb, accuracy
was raised. But because of BBS document’s non-
conventional written, some other part-of-speech words
which are seldom used also express the sentiment,
such as interjection and condition. So we need to
select more part-of-speech words which are likely to
be sentiment-bearing words. When we select more,
more unrelated words will be introduced. So it is
contradictory.
Although the accuracy of identifying sentiment-
bearing words by using maximum entropy was not
very high, the accuracy of classifying reviews was
better. By using the maximum entropy model, we
removed the unrelated words. At the same time, we
took out the shortage that part-of-speech is not all-
sided. All the words which were identified to be the
sentiment-bearing words could be used as features
whatever its part-of-speech. Our method can
effectively help people locate the required reviews in
the Internet.
5. Conclusion
In this paper, a novel efficient method for BBS
sentiment classification is presented. By using
maximum entropy model to extract words with polarity,
it is optimized that the features represented of
documents. We find that selecting the words with
polarity as features can improve the performance of
BBS sentiment classification. But because the training
data was not all-sided, which made error in constructing
our feature template and computing the parameter value
of features. And the results of classification were
influenced by the error.
Some shortages need to be improved. For example,
it is only considered whether a word had the semantic
tendency. We have not considered that some words’
sentiment-bearing tendency maybe change with some
special words around or in different linguistic
environment. In future, we would like to improve our
training data and our maximum entropy model. The
polarity difference of the words in variable
environment needs to be considered and more features
need to be extracted according to the maximum
entropy.
 Acknowledgement
Shen Jie, Fan Xin , Shen Wen and Ding Quan-Xun
acknowledge the support by National Natural Science
Foundation of China 60673060.
6. References
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CA,1999, 42-49.
[3] Lewis D.. Naive bayes at forty:The independence
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[4] Bo Pang, Lillian Lee, and shivakumar Vaithyanathan.
Thumbs up? Sentiment classification using machine
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[5] Wei-Hao Lin, Theresa Wilson, Janyce wiebe and
Aleaxander Hauptmann. Which side are you on?
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[6] Jun Xu,, Yu-Xin Ding, Xiao-Long Wang. Sentiment
Classification for Chinese News Using Machine
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[7] Hui-Fen Tang, Song-Bo Tang ,Xue-Qi Cheng. Research
on Sentiment CIassification of Chinese Reviews Based
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[8] Lan M, S Y Sung et al.A Comparative study on Term
Weighting Schemes for Document Categorization[A].
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[9] Hong Lin, Chun-Fa Yuan, Shu-Jun Guo. A Chinese Part
of Speech Tagging Method Based on Maximum Entropy
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24(1): 14-16
[10] Xiao-Rong Chen, Jin Qin. Maximum Entropy-Based
Chinese Word Sense Disambiguation. Computer
Science [J], 2005, 32(5)
[11] JOACHIMS, T. 1999. Making large-scale SVM
learning practical. In B. Schlkopf, C. Burges, and
A. Smola(eds.), Advances in Kernel Methods –
Support Vector Learning, pp. 169–184. MIT
Press.
 2009 International Conference on Computer Engineering and Technology
Applying eMM in a 3D Approach to e-Learning Quality Improvement
Kattiya Tawsopar
Information Technology, Graduate School
Sripatum University
Bangkok, Thailand
e-mail: kattiya@gmail.com
Abstract—The E-Learning Maturity Model (eMM) is a
framework for quality improvement, by which institutions can
assess and compare their capability to sustainably develop,
deploy and support e-learning. This paper presents a
three-dimensional (3D) approach to e-Learning quality
improvement. In the approach the eMM is applied in
“Diagnosis” phase as an assessment tool for e-Learning process
improvement in institutional context where the key elements
necessary for improvement in e-learning activities are
identified. The “Development” phase of the 3D approach
concentrates on putting together improvement or change
packages to target areas of deficiency. In strategic point of
views, the packages are translated into implementation plans
in a short term, a mid term, and a long term. In “Delivery”
phase of the approach, the main focus is the human resource
and marketing efforts for implementing the change packages
in operational point of views. The 3D approach described can
be beneficial in guiding individual institution's understanding
of their e-learning capability and providing educational
institutions with a roadmap for e-learning process
improvement as well as providing a framework for strategic
and operational planning and investment.
Keywords-component; E-learning, Process Improvement,
e-Learning Maturity Model, eMM, e-Learning Life Cycle.
I. INTRODUCTION
Today e-learning is an important role for learning
Management in Thailand. E-learning can be usefull in
helping students keep the conceptual knowledge more
durable than classroom learning. Besides, learning
management is also in the same standard. In particular,
checking and supporting the students' knowledge can be
done automatically and they can learn everywhere at all
times [1]. Some researches in Thailand we found that the
main problem in the development of e-learning in Thailand
[2] is instructors are not interested to designing teaching
materials, including a lack of time to do this.
As a measure to elevate quality e-Learning to a universal
standard in order to achieve a mutual working objective,
universities in Thailand should lay more special emphasis on
the significance of development in terms of Capability
Maturity. In connection to this, the Capability Maturity
Model Integration (CMMI) is to be adapted and applied
direct onto improvement’s process, hence it is considered as
the most effective means in upgrading working system
within the organization.
978-0-7695-3521-0/09 $25.00 © 2009 Crown Copyright
DOI 10.1109/ICCET.2009.221
Kittima Mekhabunchakij
Information Technology, Graduate School
Sripatum University
Bangkok, Thailand
e-mail: kittima.me@spu.ac.th
This paper presents a three-dimensional (3D) approach to
e-Learning quality improvement. In the approach the eMM is
applied in “Diagnosis” phase as an assessment tool for
e-Learning process improvement in institutional context
where the key elements necessary for improvement in
e-learning activities are identified. The “Development” phase
of the 3D approach concentrates on putting together
improvement or change packages to target areas of
deficiency. In strategic point of views, the packages are
translated into implementation plans in a short term, a mid
term, and a long term. In “Delivery” phase of the approach,
the main focus is the human resource and marketing efforts
for implementing the change packages in operational point
of views.
This paper is organized as following, Section 1
Introduction. Section 2 explains the Maturity Thinking and
e-learning Maturity Models. In Section 3 discusses the
e-Learning Maturity Model (eMM) model , Section 4
presents A 3D Approach in Evaluation e-learning Maturity
Model Section 5 discusses Using eMM in e-learning
Development 3D View and Section 6 Conclusion and an
acknowledgement.
II. MATURITY THINKING AND E-LEARNING MATURITY
MODELS
Capability maturity models typically identify five or six
‘maturity’ – or ‘capability’ – levels, from low to high. The
number of organisational ‘foci of assessment’ – the
viewpoints through which the organizations are examined –
for “Maturity Models” varies depending on the model.
Probably the best known derivative maturity model is the
CMM (Capability Maturity Model) for software engineering.
Also, other software engineering maturity models have been
developed, including ‘Standard Process Improvement and
Capability dEtermination’ (SPICE), the basis of ISO 15504.
The recent development of the CMMI model (CMMI 2000,
SEI 2002) – covers not only software development, but
product development. This is an indication of the continued
importance of maturity models in product development.
In the area of e-Learning development and
implementation, and management, there are numerous
studies with specific aspects with some sort of e-Learning
measures for improving the evaluation of the capability and
maturity in e-Learning.
Marchall and Mitchell [3] they apply the CMM and
SPICE in the area of e-Learning in order to explore whether
357
similar insights could be generated for institutions engaged -Development,
in online delivery of teaching. They suggest that this model -Support,
offers a means for institutions to identify systemic weakness -Evaluation,and
in their e-Learning development, delivery and management - Organization.
that potentially can inform future resourcing and strategic
priorities.
Petch, Calverley, Dexter and Cappelli [4] they were
designed to evaluate the operational viability of a method
based on the e-Leaning Maturity Model developed at the
University of Wellington, New Zealand, which in turn was
derived from Carnegie Mellon’s widely accepted Capability
Maturity Model. A successful benchmarking effort should be
able to inform an institution’s planning and resourcing
processes and the outcomes.
In specific area of education Lutteroth, Luxton-Reilly,
Dobbie and Hamer [5] has proposed a maturity model for
computing education which is inspired by the SEI’s CMM.
CMM can be used to rate educational organizations
according to their capability to deliver high quality education Maturity gauged in color-coded scale
on a five level scale.
The work of Moazzam Baig, Sidra Basharat, and Manzil- Figure 1. Process Dimensions and in E-Learning Maturity Model (eMM)
e-Maqsood focuses on the development of a maturity
framework for higher educational sector that would enable The eMM divides the capability of institutions to sustain
education providers to improve quality of the existing and deliver e-learning into five major categories or process
educational processes and also aid the cost-effective areas in Figure 1 that indicate clusters of strongly related
development of value-added and practical processed that processes.
have been overlooked in the past [6]. For this purpose they
have selected CMM as our base model and People Capability IV. A 3D APPROACH IN EVALUATING E-LEARNING
Maturity Model (P-CMM) and CMMI as helping models for MATURITY MODEL
quality improvement in higher education sector.
Donovan [9], When viewed in its totality, your e-learning
Errol Thompson suggests the international literature in
priorities are essential elements representing the E-Learning
improving student learning in a subject and how to assess the
Tactical Process Model. This paper presents a three-
effectiveness of these learning strategies [7]. Thompson
dimensional (3D) approach to e-learning quality
proposed a model based on the principles of the CMM in the
design perspective in order to encourage the learners to improvement by applying eMM in “Diagnosis,”
reflect on their learning and to evaluate the effectiveness of “Development,” and “Delivery” phases.
their learning . Based on the operational principles in Figure 2,
e-Learning Development Lifestyle is naturally composed of
III. THE E-LEARNING MATURITY MODEL Strategy, Requirements Analysis, Design and
Implementation.
The e-Learning Maturity Model (eMM) model was
developed in New Zealand based on two complementary
models, the Capability Maturity Model (CMM) from the
Software Engineering Institute (SEI 2002) and SPICE
(Software Process Improvement and Capability
Determination).
Capability is perhaps the most important concept in the
eMM. The eMM can be used by organizations to “assess and
compare their capability to develop, deploy and support
e-learning”, according to [6], [8]. As to Figure 1, Process
Dimensions and e-Learning Maturity Model (eMM) pop up
the following illustrative descriptions.
The authors identify five dimensions of capability:
-Delivery,
-Planning, Figure 2. e-Learning Development Lifecycle
-Definition,
-Management, and In the Diagnosis phase the eMM is applied as an
- Optimisation. assessment tool. In the context of e-learning there are five
Five e-Learning KPAs were also proposed: process areas for the institution to consider in process
-Learning, improvement: Learning, Development, Co-ordination,

358
Evaluation, and Organization. This phase uses eMM as
a ‘lens.’ to identify any potential gaps which exist in the
current set of eMM practices in each process area.
The key elements are necessary for improvement in
e-learning activities are identified and prioritized in the
Diagnosis phase.
The Development phase of the 3D approach concentrates
on putting together improvement or change packages to
target areas of deficiency. In strategic point of view, the goal
of this phase is to describe the evolution of the institution as
a whole by continually planning the change packages. The
planning is emphasized in progression on the institution’s
Process Dimensions in eMM: Delivery, Planning, Definition,
Management and Optimization. For implementation of
change packages, we show an approach to define the
implementation based on an e-learning Tactical Process
Model (eTPM). The eTPM outlines the priorities in moving
e-learning from strategy to delivery and continuous
improvement. It is the infrastructure that contains and
supports the e-learning programs and courses that the
institution develops and delivers. In a holistic view, the
institution’s e-learning priorities are essential elements
representing the eTPM in three views:
-Instructional architecture,
-Technical architecture, and
- Organizational environment.
critical priorities and processes can be focused to assure
implementation aligning development and delivery to the
e-learning strategy and operations in institutional context.
V. USING EMM IN ELEARNING DEVELOPMENT 3D VIEW
This topic involves the adaptation of eMM for
developing e-Learning, which signifies in the form of 3D
view that comprises Diagnosis, Development and Delivery,
as related earlier under the 4th topic - 1) Diagnosis : After
obtaining results derived from analyzing and summarizing
definitions in 5 process areas [3] , Table 1 show an example
view of maturity indexed for each phase of e-learning
development, representing the 2D view is then awarded.
In relation to this, the vertical bar graph in Figure 4 is
consequently built to bring up more clear-cut pictures. The
higher climb-up thus witnesses the more efficiency of
Process and vice versa. For example, Record 1, Column 1,
bar graph of Learning Process in terms of Delivery rockets
up outstandingly, revealing how efficient it keeps going on.
On the contrary, it is obvious that Management and
Optimisation drop so heavily down to an astonishing extent
that such unfortunate signs remind the need of prompt
improvement.
TABLE I. EXAMPLE VIEW OF MATURITY INDEXES FOR EACH
PHASE OF E-LEARNING DEVELOPMENT

Management

Optimisation
Definition
Planning
Delivery
Learning 3.10 3.00 1.50 1.00 1.00
Development 3.00 2.14 1.57 1.14 1.14
Support 2.83 2.00 1.33 1.50 1.00
Evaluation 1.67 2.67 0.67 1.00 1.00
Organisation 3.00 2.22 1.89 1.11 1.00

Figure 3. A 3-D View of e-Learning Maturity Model

In Delivery phase, the institution’s eTPM is then

translated into Delivery Plans in a short term, a mid term,
and a long term. Each Delivery Plan defines priorities and
processes for implementing a specific change package;
evaluating the delivery process itself; and evaluating the
effects of the change. In this phase, there are three
interrelated priorities in considerations: program-level
development, course-level development of learning products,
and delivery at the program and course levels.
The 3D approach described in this paper can be
beneficial in guiding an individual institution's understanding
of their e-learning capability in eMM perspective, and
Figure 4. In seach of deficiencies based on the maturity levels of the
providing the institution with a framework for quality “Learning”
improvement of e-learning development and delivery
following the institution’s eTPM. Using this approach the

359
 TABLE II. SAMPLE DATA FOR FIGURE 5. VI. CONCLUSIONS

Management

1.03 Optimisation
The e-Learning Maturity Model is applied to Thai

Definition
Planning
Delivery
university’s development system for evaluation and
comparison purposes of e-Learning. In other word, this
so-called eMM is somewhat capability in bringing university
up to international recognition standard. It is three-
2.72

2.41

1.39

1.15
dimensional representation, i.e. Diagnosis, Development and
Delivery, that originates quality improvement of university’s
Learning 3.10 3.00 1.50 1.00 1.00 e-Learning. By this, we trust the qualified universities will
7.1% 6.9% 3.4% 2.3% 2.3% have to pass through the good developing system as Watts
Humphrey [10].
Development 3.00 2.14 1.57 1.14 1.14
At this initial stage of evaluation and comparison, none
6.9% 4.9% 3.6% 2.6% 2.6% can tell exactly which level each university in Thailand is in,
Support 2.83 2.00 1.33 1.50 1.00 since data collection is still getting on and incomplete due to
limited assignment time.
6.5% 4.6% 3.1% 3.4% 2.3% Decidedly, we choose eMM to adapt the process of
Evaluation 1.67 2.67 0.67 1.00 1.00 development for Thailand’s universities in order to learn how
3.8% 6.1% 1.5% 2.3% 2.3% advanced they presently are prior to pushing them to higher
level. Lastly, in the analyst’s humble opinion, this model
Organisation 3.00 2.22 1.89 1.11 1.00 will be bettered for universal acceptance and, conclusively,
6.9% 5.1% 4.3% 2.6% 2.3% will benefit all Thai universities in their future developments.
TOTAL 13.60 12.03 6.96 5.75 5.14 43.49 ACKNOWLEDGMENT
Avarage 2.72 2.41 1.39 1.15 1.03
Our sincere thankfulness to my adviser and instructors
for their support in conducting this research and my deepest
appreciation to Graduate School, Sripatum University.

REFERENCES
[1] K., Tawsopa , M. Kittima, “An Evaluationof Open Source E-Learning
Systems incorporated with OSMM,” Proceedings of the 6th
International Conference on e-Business 2007 (INCEB2007).
[2] S. Supanee, R. Arnut, “The Strategic Move to Higher Education
Reform,” APRU DLI2006, http://www.cmsthailand.com/
[3] S. Marshall, G. Mitchell, “Applying SPICE to e-learning: an e-
learning maturity model?,” In Proceedings of the 6th conference on
Australasian computing education, Vol. 30,pp. 185-191. Dunedin,
New Zealand: Australian Computer Society, Inc.
[4] J. Petch, G. Calverley, H. Dexter, & T. Cappelli, “Piloting a Process
Maturity Model as an e-Learning Benchmarking Method,” In
Figure 5. Colored Maturity Diagnose based on eMM Proceedings of the 5th European Conference on e-Learning (ECEL
2006). Academic Conferences Limited.
Table 2, Sample data for Figure 5. gives a better vision [5] C. Lutteroth, A. Luxton-Reilly, G. Dobbie, & J. Hamer, “A maturity
by means of circular diagrams in various colours, which are model for computing education,” In Proceedings of the ninth
Australasian conference on Computing education, Vol. 66,pp. 107-
created from percentage calculation. 114. Ballarat, Victoria, Australia: Australian Computer Society, Inc.
[6] B. Moazzam, B. Sidra, M. Manzil, “A Maturity Model for Quality
2) Development : Upon completing the 1) Diagnosis, now Improvement in Higher Education (ICMHK2006).
we come to focus on the Deficiency of Process that is visible [7] E. Thompson, & E. Thompson, “Using a subject area model as a
from the graph in Figure 5 (Coloured Maturity Diagnosis learning improvement model,” In Proceedings of the 8th Austalian
based on eMM). It enables us to realize which terms should conference on Computing education, Vol. 52, pp. 197-203. Australian
be bettered in order to achieve 3) Delivery : As seen from Computer Society, Inc. Darlinghurst, Australia, Australia.Albeanu
the graph, the group of smaller circles are situated mostly on [8] A. Grigore, “E-Learning metrics. UNESCO Chair in Information
Technologies at University of Oradea,” The 2nd International
the below-left. It reflects the inefficient state of Process, no Conference on Virtual Learning. ICVL 2007, University Street No.1,
doubt. So, it is vital for the prioritization of Process. That 410087, ROMANIA.
is to say the development should be performed in the sequent [9] L. Donovan, “Implementing E-Learning. In George M. Piskurich
manner by starting from the smallest circle first. Once it (Ed.),” The AMA Handbook Of E-learning: Effective Design,
becomes larger and can be moved adjacently to the upper- Implementation, And Technology Solutions, pp. 317-323. New York
right of graph, it means efficiency of that Process now stays : AMACOM/American Management Association.
at the satisfactory level of high stability. [10] H. Watts, “An IT Metrics and Productivity Journal Special Edition,”
Copyright©2006 Computer Aid, Inc.

360
 2009 International Conference on Computer Engineering and Technology
Research on Automobile Driving State Real-time Monitoring System Based on
ARM
Hongjiang He
Department of Information &Electronic Engineering
Hebei University of Engineering
Handan, Hebei province, China
E-mail: zhangyaminhb@sina.com
Abstract—Research on Automobile Driving State Real-time
Monitoring System is introduced in this paper, which takes
S3C2410（ARM920T）microprocessor as the core and takes
Linux as the operating system. This system adds the fault
diagnosis expert system to the traditional automotive driving
state monitoring system, which is only storing and displaying
the driving state parameters on the base of the real-time
monitoring for the automobile driving state. Employing the
control method of the communication between the embedded
system and CAN field bus; the proposed system can efficiently
collect the status data related to the security performance of
the automobile and integrate them. Fault diagnosis expert
system is implemented for real time monitoring and
diagnosing. It can also give some reasonable suggestions to the
driver by comparing and analyzing the collected data, which
makes this system more humane. The information can be
displayed and the audio alert can be sounded through a
human-machine interface. This system has a good prospect in
the field of automotive electronics.
Keywords-ARM;Linux opeating system;real-time
monitoring; fault diagnosis expert system;Controller Area
Network(CAN)
I. INTRODUCTION
The sharp development of modern transport brings in a
great facility to people as well as increases the radio of traffic
accidents. Vehicle traveling data record (which is also called
automobile block box) can accurately record the state
parameters of the automotive traveling process so it has been
widely used in many countries and areas. Statistics shows
that using the vehicle traveling data recorder can succeed in
reducing the radio of traffic accident by 37 to 52 percent.
The using of vehicle traveling data record has been certified
to receive good results. It manifests in two aspects. On one
hand it effectively restraint violations such as fatigue driving
and speeding; on the other hand it can improve the drivers’
sense of responsibilities. At the same time the data recorded
by the vehicle traveling data record can offer a real, effective
and scientific legal basis for the analysis of traffic accidents.
The national standards of vehicle traveling data record in our
country was officially launched in September 1, 2003, which
presented that the vehicle traveling data record has entered a
practical stage in our country[1].
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.162
Yamin Zhang
Department of Information &Electronic Engineering
Hebei University of Engineering
Handan, Hebei province, China
E-mail: zhangyaminhb@sina.com
However, the traditional vehicle driving data record can
only display and store the driving state parameters, it can’t
analyze the parameters. This system adds the fault diagnosis
expert system on the base of the traditional vehicle traveling
data record. By comparing the automobile driving
parameters with the normal parameters which stored in the
automotive normal driving state parameters library, the fault
diagnosis expert system can give the driver some rational
recommendations, which makes this system more humane.
II. HARDWARE STRUCTURE
This system takes the popular 32-bit embedded system
nowadays and takes the microprocessor produced by
Samsung Company. The microprocessor mainly includes an
S3C2410 chip based on ARM920T core and the
corresponding external interface circuit. The system mainly
includes modules such as data acquisition, data storage, fault
diagnosis, CAN communication, data output, human-
machine interface and audio alert and so on. The workflow
of the system is as follows: firstly the data acquisition
module achieves the collection of the data and then deals
with it briefly; secondly the processed data will be sent to the
main control module, where the data will be analyzed and
diagnosed; finally the fault diagnosis expert system analyzes
the data and then gives some correspondent sound alarms
and rational suggestions. The communication between the
data acquisition module and the main control module is
achieved by the CAN bus. The overall system hardware
structure is shown in Fig. 1.
A. Data Acquisition Module
The data acquisition module mainly consists of input
channel, I/O ports, counter and A/D converter. As an
intelligent CAN node, the function of the data acquisition
module is to collect the state signal which has a great
influence to the vehicle safety travel. And then the signal
will be converted and processed. At last the processed signal
will be transferred to the main control processing unit for the
further processing by the CAN field bus. All the signals
collected are based on existing on-board sensors. The
structure of one of the signal collection units is shown in Fig.
2.
361
 K9F1208 the SRAM; for another thing it also has the ability to store
the data after the power down, which is the characters of
both of the Flash and the EEPROM. Further more some
K9F1208 multifunctional FRAM has the power capability. So the
FRAM is very suitable for this system. It will not only
FRAM simplify the system’s circuit and lower the cost, but also can
enhance the reliability of the system.
Peripheral The character of the driving state data is large volume
and slow refresh. This system refresh every 20 seconds, so
the system should record the average real-time data in the
mentioned above. It should contain the record volume 360
Keyboard USB and RS232 hours at least, and record the date as the style of year, month
CPU interface
and day, and record the time as the style of hour, minute and
S3C2410
second. This system uses 2 pieces of K9F1208 flash memory
LCD produced by Samsung as storage device. One is used for the
XF-4240
storage of driving state; the other is used for program
storage.
CAN interface
C. CAN Communication Module
Firstly CAN was designed for automobile inspection and
control system by BOACH automotive company, Germany
[2]. The structure of CAN Bus Interface is shown in Fig. 3.
The system bus controller uses SJA1000 produced by
Intelligent Intelligent Intelligent Philips semiconductor. It is an independent CAN controller,
unit of signal unit of signal … unit of signal which is mainly used for zone network control of moving
collection collection collection
target and general industrial environment. 82C250, the CAN
bus driver, is the interface between CAN controller and
physical bus. It was designed for automotive high speed
Speed Oil … Brake application firstly. It can implement the function of
Pressure
differential sending and receiving.
Figure 1. Hardware structure
D. Data Output Module
The signals collected can be classified three kinds: In order to implement the function of detection and
analog quantity, switching value, instantaneous quantity. The discrimination after the traffic accident, the system should be
collection of the analog quantity, such as water temperature, able to output the data stored in memory into the peripherals,
oil temperature, fuel capacity, oil pressure and engine speed such as: the police’s hand-hold PDA, PC or printer.
and so on, is very difficult. The switching value mainly According to the national standard, this system provides both
includes brake, headlights, left-turn light, right-turn light, standard USB and RS232CD interface, so it can implement
backup light, high beam, low beam, ignition, front door, the final output.
back door, brake pressure and hand braking. All the
switching values are isolated by the optical coupling E. Human Machine Interface Module
isolation. The instantaneous quantity includes speed and The system set an LCD to display the driving parameters,
mileage. such as instantaneous speed, high speed in 1minute, high
speed in 10 minute, time, date and total driving mileage. The
B. Data Storage Module real-time collected data is sent to memory at the same time is
According to the standards of national institute, the data
stored in the vehicle traveling data recorder should contain Sensor Data
two parts: one is the automotive real-time data, which refers storage
to the data within 20 seconds before and after the traffic
accident, the other is the automotive historical data, which
contains the automobile and driver’s traveling state in 360 Signal CAN
hours. The automotive real-time data storage has a strict conditioning interface
requirement to the memory. Regarding to the automotive
high real-time, the memory should satisfy not only the high
reliability and speed, but also can store the data without of Digital signal
AD
losing after the power down and can be rewrite for many processor
conversion
times. Because of this, this system chooses the FRAM for
the data storage module. There are some reasons for this. For
one thing the FRAM has the high speed and rewrite times of Figure 2. Signal collection unit

362

S3C2410
SJA1000 82C250
Figure 3. CAN Bus Interface
showed on LCD screen, which is refreshed every
0.5Seconds. In order to view the different contents and input
the driver’s correspondent information, the system set some
key-press. They are menu, up key, down key, right key, left
key, enter key, cancel key.
F. Sound Alarm Module
The system uses XF-4240 speech synthesis module
developed by kedaxunfei to realize sound alarm. The module
has the function of text-to-speech. The Encoding supports
GB2312 、 GBK 、 BIG5 and Unicode. Communication
supports UART, SPI and I2C [3]. The system uses GB2312
encoding and receives the processor’s sending data by I2C so
it can implement the sound alarm suggestion for the speed
and fatigue driving.
III. SOFTWARE STRUCTURE
The software design chooses Linux as operating system
because Linux has such advantages as open source code,
cutting conveniently and convenient transplantation. The
development tool chooses VC++6.0.
The function of main modules is as follows:
• Module of CAN communication: finishing the basic
communication protocol and data sending.
• Module of real-time monitoring: comparing the
current automotive driving parameters with the
standard parameters of normal working state.
According to the content of the discrimination of the
data it can give correspondent monitoring and
alarming. It is the preliminary diagnosis of the
running state, which has a strong real-time fault
handling.
• Module fault diagnosis: on the basis of real-time
monitoring module, according to the expert system
method, this module can make a further diagnosis
for the driving state. And then it can make an early
warning for the possible fault or the diagnosis for the
current fault so that it can give some reasonable
suggestions.
• Database of monitoring data: storing the driving data
sent by CAN communication module. The data is
collected by the intelligent signal collecting unit and
is used for the real-time monitoring and fault
diagnosis module.
• Module of sound alarm: making an early warning for
the driver by sound alarm, LCD and image. It can
help the driver deal with the problem in time.
• Module of human-machine interface: processing
user’s key-press and demonstrating the driving state
and fault.
363
A. Fault Diagnosis Module
The vehicle fault diagnosis expert system is referred to an
intelligent method which applies the artificial intelligence
technology to simulate the humane thinking process of
solving the problem so as to diagnosis the automotive fault
[4]. The traditional vehicle fault diagnosis expert system has
some kinds of knowledge representation as follows:
1) Production: Production which is also called rule-
based representation is the most commonly used method of
knowledge representation in artificial intelligence. It is the
procedural knowledge-centric, and is usually used to
express the cause-and-effect relationship of the knowledge.
It has a natural, clear character.
2) Semantic network: Semantic network is a network
used concept and semantic relation to express knowledge.
Semantic network can concatenate all kind of things
together organically. It can clearly describe the semantic
relation among the things. At the same time it put emphasis
on the expression of semantic relation between the
knowledge. It conforms to the process of human associative
thinking and has the characters of information sharing and
natural expression. However, this knowledge representation
method can’t express the deep-seated judgment knowledge
very well.
3) Frame-based: Frame-based is a data structure which
represents the qualitative state. It can represent the structure
relationship between the knowledge. But the frame-based
isn’t suitable for the representation of the procedural
knowledge.
The basic structure of the expert system consists of six
parts, such as human-machine interface, knowledge base,
inference engine, interpreter, global database and knowledge
acquisition. The structure is shown in Fig. 4.
The expert system used in this system compare the
collected data during the automobile driving with the
Knowledge
engineer User
Human-computer
interface
Professional Information of
knowledge why, how
Knowledge Interference Interpreter
acquisition engine
Knowledgeb Global
ase database
Figure 4. Structure of expert system
automotive normal driving parameters stored in the libraries.
By the comprehensive comparing and analyzing the data
mentioned above, the system can give the driver appropriate
predictive failure and diagnostic output. Therefore the
system adopts the rule-based reasoning approach which has a
directly presentation, unified style, easily understood and
convenient interpreted. The system structure is shown in Fig.
5.
As the fault diagnosis of the automotive driving is a
fuzzy knowledge and doesn’t have a strict definition or an
accurate mathematical analysis, so in the system design we
adopt a production rule to express the expert knowledge [5].
The general expression of the rule is as follows: IF P THEN
Q, P (also called antecedent) presents the precise of the
production rule, it gives a precondition whether the
production rule will be used or not; Q (also called
consequent) is a group of conclusions or operations, it gives
the correspondent conclusion or action which should be
executed when P is true.
Comparing the current automotive driving state
parameters with the data stored in the normal driving state
parameters library, the fault diagnosis expert system can give
predictive failure and fault diagnosis output. It will help the
driver aware whether all the automotive parameters are
normal or not. It plays a good guide for the vehicle
maintenance and safe driving.
The system has two modes: recording mode and
communication mode. The recording mode is when the
vehicle is running the system periodically collects all the
states and the driving information of the current driver, such
as speed, driving time, time of fatigue driving, record of
speeding, stopping time and so on. It also provides sound
alarm for the fatigue driving and speeding and presents the
date, time and speed. And then it stores the collected data
and parameters of date and time to the data memory. The
communication mode is when the vehicle is stopping the
system exchanges the data through the communication
interface and PC. The data can be uploaded or downloaded.
When using USB for communication, the system provides
Oil Water Fuel
… Brake
pressure temperature consumption
Data fusion program of multi-sensor
Data incidence matrix
Automotive
Fault normal
diagnosis driving
Inference engine
expert parameters
library
Fault early Interpreter
warning Diagnostic
output
Figure 5. Structure of predictive failure and diagnositic output
364
the keyboard to choose the correspondent operations. When
using RS232 for communication, the control of data
transmission is completed by PC. Some of the strong real-
time works such as time calibration are completed by PC.
The work flow of the system is as follows:
• Initialization and self-checking after the power up.
• Authentication for the driver or administrator.
• LCD display interface.
• Determine whether the vehicle is driving or not
according to the pulse of speed sensor and then run
into the correspondent mode.
• Collecting, analyzing, storing and displaying all the
signals when the vehicle is driving, locking the keys
and shielding all of the communication interfaces at
the same time; if the vehicle is stopping the system
unlock the keys and open the communication
interfaces.
• Determining whether the driver is speeding or
fatigue driving according to the driving speed and
time and give correspondent suggestions of sound
alarm.
• The expert system analyzes and compares all the
automotive driving parameters with the normal
parameters and then gives some reasonable
suggestions.
ACKNOWLEDGMENT
The system is an embedded system which takes S3C2410
microprocessor as the core and takes Linux as the system.
The system realizes LCD display for the parameters of
vehicle driving state by the multi-sensor monitoring for the
vehicle driving state. Further more it adds predictive failure
and diagnosis system to the traditional system. It can give
some reasonable suggestions to the driver by comparing the
parameters to the normal parameters. So the system is more
humane. The system has a small volume, complete function,
high reliability, high performance-price ratio character and is
very suitable for on-board. It also can improve the reliability
of the vehicle drives safety. Therefore it has a good
application prospect.
REFERENCES
[1] Shaoyong Ruan, Jiadong Zhang and Kai Huo, “The design of Vehicle
Traveling Data Record Based on ARM”, Journal, Machinery &
Electron, No 4, 2006, pp. 50.
[2] Yuntao Rao, Jijun Zou, and Yongyun Zheng, Theory and Technology
Application of CAN field bus, Buaapress, Beijing, June.2003.
[3] Kedaxunfei, XF-S4240 voice synthesis module data manual,
Kedaxunfei Information Technology Co., Ltd., Anhui china, 2006.
[4] Lili Zhang, Jiangwei Chu, Tiangang Qiang, and Daming Han, “Key
technologies study and development for automobile fault diagnosis
expert system”, Journal, Application Research of Computers, Vol 25,
No.6, June.2008, pp. 1634
[5] Shuzi yang, Knowledge based diagnosis reasoning, Tsinghua
University Press, Beijing, December.1993.
 2009 International Conference on Computer Engineering and Technology
Information Security Risk Assessment & Pointed Reporting: Scalable Approach
D.S. Bhilare Dr. A.K. Ramani
School of computer science School of computer science
Devi Ahilya University Devi Ahilya University
Indore, India Indore, India
e-mail: bhilare@hotmail.com e-mail: ramaniak@yahoo.com
Abstract-Network Managers of Higher Educational
Institutes, are well aware of general Information Security
issues, related to Campus Networks. There are well
developed security metrics, giving exhaustive list of security
controls, required to mitigate different risks. Accordingly,
various security measures and technologies are being
deployed. However, at present, not enough attention is being
paid on measuring the effectiveness of these controls and
overall state of security in the institution. In this study,
attempt is made to build a metric based assessment and
reporting plan, specific to the needs of an academic
environment. Proposed assessment metric facilitates
iterative implementation, by prioritizing each metric.
Secondly, to reduce response time, a novel approach of
pointed reporting is suggested, where responsibilities are
distributed across the institution, based on relevant roles. In
this approach, security exceptions are reported directly to
the predefined roles, responsible for that particular security
control. This pointed reporting, delivers message to the right
person in minimum time, resulting in improved response
time. The proposed assessment metric and pointed
reporting structure, will improve overall security
governance. As security measures and practices can be
assessed systematically and remedial actions can be taken in
less time, which is so crucial for effective security
governance.
Keywords- information security; security assessment;
pointed reporting; distributed defense; iterative
implementation.
I. INTRODUCTION
Today’s Campus Networks are complex grouping of
technology (including hardware, software, and firmware),
processes, students, faculty and staff, all working together
to provide institutions with the capability to process,
store, and transmit information on a timely basis to
support various academic and administrative functions.
It is well known that you cannot mange some thing
which can not be measured. Therefore, in order to
improve the security levels it is necessary that we
understand the strength and weakness of the practices
being followed. Though, Information Security is an
emerging area but there are enough solutions and products
available which are being deployed at various levels.
There are Information Security practices and policies in
place for quite some now. But measuring of effectiveness
of these products and practices is one of the major
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 365
DOI 10.1109/ICCET.2009.218
Dr. Sanjay Tanwani
School of computer science
Devi Ahilya University
Indore, India
sanjay_tanwani@hotmail.com
challenges in Information Security Management. Many
institutions, invest in security technologies, policy
documents, staff training, but often find no correlation
between increased spending on such initiatives and a
better overall security record [1].
There are number of incidences which shows the
potential for manipulating and exploiting technologies
commonly utilized by universities and colleges today [2].
In this paper, information security assessment plan is
proposed, keeping in view the expectations of academic
institutions & relevant regulatory requirements. Basic
objective of this plan is not only to provide a checklist of
security metric but to provide an inbuilt evaluation &
role-based response system. Proposed metric, addresses
specific requirements of three levels of institutions,
namely small, medium and large. This approach facilitates
Iterative implementation, and serves as a starting point for
small institutions, for protecting their valuable
information assets. Another important issue addressed in
this paper is exception reporting. Exceptions found during
assessment & continuous network monitoring, are
reported directly to the linked role as specified in the
proposed metric, by an e-mail or SMS alerts. Each metric
is associated with a role and corresponding
responsibilities. This reporting system should reduce
response time required for taking remedial action.
The rest of this paper is organized as follows. In
section II, related works are shown. Section III.A,
identifies various roles necessary to implement proposed
assessment plan and their key responsibilities. In section
III.B, proposed assessment plan and reporting system is
described. Section III.C, describes assessment, reporting
process and possible applications of outcome of
assessment. Finally in section IV, conclusion along with
future work is presented.
II. RELATED WORK
Saydjari [4] has given pros and cons of considering
risk as a base metric. One good property of risk as a
security metric is that it directly addresses possible threats
and damages. One shortcoming is that the metric doesn’t
explain how to rectify threats.
Policy-Based metrics look at quantities like number
of unauthorized login attempts, files accesses, and so on.
These metrics may end up measuring the inadequacy of
user training more than it measures actual system security.
Incident-Based metrics look at the actual successful
attacks that occur, the frequency and the real damages.
None of the approaches mentioned above provide
inbuilt role-based assessment mechanism and exception
handling, neither considers issues pertaining to small
academic institutions having limited resources. Most of
the approaches provide a generalized list of metric
without defining associate roles and responsibilities. NIST
publications[6] provide a broad categorization of roles.
Secondly, there has been little consideration for academic
environment, as generally focus is on industry.
III. INFORMATION SECURITY RISK ASSESSMENT
AND POINTED REPORTING
In order to propose a robust and flexible assessment
metric, it is essential that we understand necessary
measures required in general to protect institutional
information assets. These measures are needed to
accomplish institutional objectives, protect information
assets, fulfill legal responsibilities, and protect interest of
various stake holders. Indian IT Act 2000, amended on
October 16, 2006 describes legal obligations of the
institutions [9].
Proposed metric facilitates iterative implementation
and pointed reporting. Proposed plan can be implemented
on iterative basis, as security culture and awareness
matures in the institution. This will assist small and
medium sized institutions particularly, in assessing their
existing security plans and assuring an acceptable level of
security to begin with. This will also improve exception
handling, as messages are delivered immediately and
directly to the associated role. This effective
communication process, where information is sent to right
person in time, will reduce time taken in planning and
implementation of remedial action. This will improve
overall security management, as assessment outcomes are
acted upon quickly.
Another contribution is creation of new roles which
were non existent in the traditional IT setup earlier and
association of existing and new roles with each metric.
These new roles are necessary to manage this complex
and developing discipline. Key responsibilities and
accountabilities of these new roles are also defined and
data base of their e-mail addresses and mobile numbers is
maintained. As every metric is associated with a unique
role, there is no conflict among roles and no time is
wasted in taking actions. These provisions will help in
assuring a more secure environment with effective
implementation and monitoring.
A. Identification of Roles and their Job description
In order to implement an efficient and effective
information security plan, a suitable organization structure
is essential. For a normal routine management, a
366
centralized structure is more suitable, but for effective
exception handling and quick reaction, traditional
hierarchical system does not serve the purpose. Therefore,
a role-based direct reporting system is proposed, where
exceptions needing immediate attention are conveyed to
the right person in minimum possible time. Time taken to
respond a particular event is very critical, particularly in
Information Security Management. Secondly, as
institutions are answerable and responsible for
compliance with existing laws, it is crucial that
responsibilities and accountabilities are clearly defined.
Therefore, a formal organizational structure, having clear
identification of relevant roles, and their respective
responsibilities & accountabilities is suggested. In view of
the complexity and requirements of this new discipline,
there is a need for new roles in addition to the existing
ones. Accordingly, new roles are suggested and their key
responsibilities, specific to the needs of Indian
universities are described as under.
I. Vice-Chancellor/Executive-Council
Executive-Council comprises prominent persons
from society & academics, in addition to governor
nominees.
Key Responsibilities: a)Responsible for the overall
information security of the University; b)Manages
strategic, operational and financial risks; c)Promotes and
supports information security initiatives as part of the risk
management; d)Establishes that Risk reporting, controls
and review functions are in place; e)Ensures the
University systems comply with applicable law,
regulations and ethics; Approves necessary budgets
II. Information Security Task Force
This body comprises University senior
Academic, Administrative and IT representatives, who
will co-ordinate the management and implementation of
information security measures.
Key Responsibilities: a)Supports the Director IT services
& Chief Information Security Officer in ISA initiatives;
b)Approves methodologies and processes for information
security; c)Co-ordinates the implementation of specific
information security measures for new systems or
services.
III. Deans & Head of Departments
Key Responsibilities: a) Monitor and report to the VC
on compliance with mandatory information security
policies within their faculty/department; b)Take
appropriate disciplinary actions relating to users who
breach IT security policies; c)Make proposals for
necessary funding to meet information security
commitments; d)Develop and implement additional
security policies specific to their faculty/department, in
coordination with the Chief Information Security
Officer(CISO); e)Make business continuity plan in
coordination with Director IT and CISO.
IV. Registrar (Head of Administration, Finance,
Development, Establishment etc.)
Key Responsibilities: a)Accountable to the VC regarding
information security risk management; b)Ensures
information security risks are managed to an acceptable
level; c)Responsible for legal aspects and acts as an
interface with external world.
V. Legal Advisor
Key Responsibilities:a)Advises VC/Registrar about
applicable laws, regulations; b)Ensures that all third party
contract documents include appropriate provisions with
respect to information security; c)Understands various
international cyber laws and its implications.
VI. Application Owners
The application Owner is the University Employee
responsible for the particular application. For example,
the Deputy Registrar (Exam.) is owner of the result
processing application.
Key Responsibilities: a)Accountable for protecting the
information assets within the systems they own;
b)Develop access policies for systems they own; c)Ensure
new applications comply with security policies; d)Notify
all system security issues to the Chief Information
Security Officer.
VII. Director Information Technology Services
The director reports to the Vice-Chancellor and is
responsible for the provision of enterprise information
services to the University, including; the management of
the University’s networks and related IT Services.
Key Responsibilities: a)Ensures information security is
addressed as part of all IT projects; b)Maintains an up-to-
date record of major information security risks;
c)Develops Information Security Policies, Guidelines,
Processes and Standards; d)Ensures infrastructure,
systems and applications implemented & maintained;
e)Coordinate with ISTF & CISO.
VII. Chief Information Security Officer
Key Responsibilities: a)Collaborates and liaises with all
information security stakeholders; b)Formally assesses
information security related risk & develops mitigation
plan; c)Develops information security policies;
d)Coordinates security awareness initiatives.
VIII. Information Security Officer
A technical expert assists the CISO and other users in
technical matters.
Key Responsibilities: a)Oversees monitoring to detect
breaches of security related policies; b)Manages the
response to any security incidents; c)Maintains
professional relationships with international security
bodies; d)Develops or customizes in house security
solutions; Monitors online resources and provides
appropriate e)security consultancy;
IX. Network Administrators and IT Staff
Key Responsibilities: a)Prepare procedures that
implement the IS security policies in their local
environment; b)Maintain administrative accounts;
c)Applying all relevant security patches; d)Develop
367
procedures, guidelines and standards; e.g. hardened server
configurations;
X. External Consulting Agencies
Key Responsibilities: a)Ensure proper information
security management; b)Ensure that all tools used or
deployed are certified or follow mutually agreed
standards; c)Take responsibility of proper conduct of their
employees;
XI. User (Faculty, Staff and Student)
Comply with University security policies as published on
the University web site.
In addition to above, new roles may be created,
depending upon changes in technical or managerial skill
requirements. This distribution of responsibilities has dual
advantage: as institutions are answerable and responsible
for any violations of prevailing laws, structure proposed
above will pinpoint non performing roles. Second
advantage will be swift communication of messages to the
right person in less time so that overall reaction time is
reduced. Thus, non ambiguous roles and responsibilities
will help in effective implementation of the information
security plans. After identifying and describing required
roles & responsibilities, now assessment metric necessary
to measure effectiveness of security controls and
practices, along with associated roles and level is
proposed in the following section.
B. Proposed Assessment Metric
While proposing the metric, efforts are made to
ensure that the metric: a) Enables consistent, comparable,
and repeatable assessments of security controls; b)
Facilitates cost-effective assessments of effectiveness of
security controls; c)Promotes a better understanding of the
risks to organizational operations, organizational assets,
individuals, and other organizations; d)Generates
comprehensive and reliable information to support
security assurance decisions.
Proposed metric covers various issues pertinent to
University Environment, identified on the basis of policy
documents of various universities [10,11,13].
Implementation of the full set of metric described
below may not be practical for even large institutions.
Therefore, an incremental approach is proposed, where
institutions may begin with a base set of metric which is
subset of full metric. Over the period, as institutions
mature and get more resources, full set of metric may be
implemented.
Based on the guidelines published by various
standards agencies NIST ( 800-53)[7], ISO 17799, Policy
documents of various universities[10,11], Indian IT Act
2000 [9], UGC/AICTE guidelines and the requirements of
academic environment as discussed above, the following
metric is proposed in Table I. There are three columns in
the table namely role, indicator and control. Control
column, describes security measures to be assessed. Role
column, describes roles responsible for a metric. Each ISTF: Information Security Task Force
metric is associated with a unique role, so that there are REG: Registrar, LA: Legal Advisor
no ambiguities, and plans are implemented smoothly. In DN: Deans & Head of Departments
order to assist start up institutions or institutions in early DSW: Dean Student Welfare
phase of Information Security implementation, level of AO: Application Owners
metric is shown in the indicator column. Base line metrics DIT: Director Information Technology Services
which should be implemented in the first phase are CISO: Chief Information Security Officer
indicated by “S”. Medium sized institutions may use ISO: Information Security Officer
additional metrics indicated by “M”. NA: Network Administrators and IT Staff
Indicator Column:
D. Coding structure used in the metric S: Indicates base metric, Starting Point
Role Column: M: Applicable to Medium sized University/College
VC: Vice-Chancellor/Executive-Council L: Applicable to Large Universities

Table I: Role Based Information Security Metric

Sno Role Indi- Control
cator
1. VC M Number of institutional functions, Number of functions for which protection is planned
2. VC L Estimated financial loss from security incidents
3. VC M Percentage of service down time due to security incident
4. VC S Number of key information assets, Number of assets for which protection is planned
5. VC S Number of external compliance/legal requirements, How many of them have been addressed?
6. VC Number of departments
S
Number of departments having business continuity plan
7. CISO Percentage of users whose access privileges have been reviewed during this reporting period
M
(Application users, Application owners, Retired/Terminated/Suspended employees)
8. CISO L Number of known security risks that are related to third party relationship
9. CISO S Preparation of management report with target values for chosen metric
10. CISO S Percentage of systems and applications that perform password policy verification
11. CISO S Percentage of systems where vendor-supplied accounts and passwords have been changed
12. CISO S Percentage of computer where configuration changes are done as per policy
13. CISO S Percentage of system where event and activity logs are: a)maintained b)monitored
14. CISO S Percentage of system for which log size and retention period have been specified
15. CISO S Percentage of system with a)firewall b)IDS/IPS c)malicious code protection
16. DIT L Percentage of remote users who access network using secure communication methods
17. DIT M Percentage of new users, undergone basic security training before using network
18. DIT M Percentage of users who completed periodic refresher training as required by policy
19. DIT M Mean time from vendor patch availability to patch installation
20. DIT L Percentage of software changes that were reviewed for security impacts
21. DIT M Percentage of backup media stored offsite in secure storage
22. DIT S Percentage of systems for which approved configuration setting have been implemented as per policy
23. DIT S Percentage of systems that are being monitored for configuration policy compliance
24. DIT S Percentage of computers whose configuration is compared with a trusted baseline
25. DIT S Percentage of systems with critical assets where restoration from a stored backup has been tested
26. DIT S Percentage of used backup media sanitized prior to reuse or disposal
27. DN S No. of department wise security breaches by the students & no. of cases where action has been taken
28. DN S Percentage of equipment, which are protected from power failures
29. DSW L Percentage of foreign students for whom background check is carried out
30. DSW Number of incidents where students transmitted obscene material to colleagues
S
Number of incidents reported to proctorial board
31. DSW M Number of social engineering incidences resulted in financial loss to students
32. ISO S Percentage of systems with account blocking parameters are set as per policy
33. ISO S Percentage of systems with automatic timeout is set as per policy
34. ISO S Percentage of systems where permission to install non-standard software is limited

368
 35. ISTF L Percentage of performance reviews that include IS related issues
36. ISTF L Percentage of critical information assets stored in encrypted form
37. ISTF M Percentage of Security roles for which responsibilities, and authority are assigned
38. ISTF L Total number of meetings where IS was on the agenda
39. ISTF M Percentage of staff assigned responsibilities from IS policies and controls
40. ISTF M Percentage of IS policy compliances reviews with no violations
41. ISTF M Percentage of critical assets & functions for which cost of compromise has been quantified
42. ISTF M Percentage of third-party agreements that have been reviewed for IS requirement compliance
43. ISTF M Percentage of systems with critical assets that use stronger authentication than user-id and password
44. ISTF S Percentage of systems & applications where user privileges are role-based
45. ISTF M Percentage of mobile devices that are examined before granting network access & use encryption
46. ISTF M Percentage of passwords and PINS that are encrypted in accordance with policy
47. ISTF M Number of hacking attempts from university domain reported
48. ISTF S Periodic comparative review of various critical IS metric
49. ISTF S Percentage of systems where configuration do not deviate from approved standards
50. ISTF S Percentage of systems with critical information assets have been backed up
51. ISTF S Percentage of security incidents that were managed in per established policies & procedures
52. ISTF M Percentage of vulnerability assessment findings that have been addressed since last reporting period
53. REG L Number of total incidents, Number of incidents that did not cause damage beyond limit
54. REG L Percentage of third party agreements where Information Security is part of the agreement
55. REG L Percentage of users who have undergone background checks
56. REG M Number of required internal/external audits, Number of required internal/external audits completed
57. REG M Number of audit findings, Number of audit finding resolved
58. REG M Number of employees handling confidential information & signed non-disclosure agreement
59. REG M Percentage of department heads who have ensured compliance with IS policy and controls
60. REG M Number of identified risks, Number of risks having mitigation plan
61. REG M Percentage of information assets that have been classified as per policy
62. REG S Percentage of departments with business continuity plan
63. REG M Percentage of continuity plans that have been reviewed, tested and updated
64. REG S Percentage of critical assets that have been reviewed for physical risks & mitigation is planned
65. REG S Percentage of critical assets, reviewed for environmental risks such as fire, flood, earthquake etc
66. REG S Percentage of sections, where information processing services are protected physically.
67. REG M Percentage of host servers that are protected from becoming relay hosts

The metric proposed above can be implemented in a
phased manner, iteratively. Initial round of assessment will
give an idea of present state of security, in the institution
and areas where more attention is required. Accordingly,
risk mitigation strategies can be planned and
implemented. This cycle may be repeated, till full metric
implementation is achieved. This process would also lead
to enhancement in the proposed metric.
C. Assessment and reporting
Figure 1, gives an overview of the assessment and
reporting procedure. Assessment of present security
measures is carried out using proposed assessment metric
and various other inputs described as under:
• Applicable Laws, Information Security Policy and
Mission statement are considered, while adapting
proposed metric by any institution,
• Third party product reviews, which are available
publicly for the products being used,
• Threshold values for the assessment metric, which
are arrived at, on the basis of level of security desired
by the institution,
• Out come of the online network monitoring &
analysis,
• Role database with e-mail addresses and mobile
numbers, required for sending exception alerts.
Whenever, threshold values are violated for a particular
metric, or an online network monitoring software detects,
some suspicious activity, an exception condition occurs.
This exception triggers a search in the data base for
getting the associated role & contact information for that
particular exception. After getting required information
message is sent by an SMS or e-mail. Based on the
information and situation analysis, remedial action is
planned and implemented.

369
 Figure 1: Information Security Risk Assessment and Pointed Reporting

Info. Security Role Database Mission Applicable

Policy with e-mail and Statement Laws
Third party mobile no.
Product
evaluation

Threshold
Values for Online Network
Metrics Monitoring
I

Metric Based Implementation

Assessment

Analysis
Remedial plan

Exceptio Role-Based
n? SMS/e-mail

Routine Reports

The outcome of above assessments can be used to: a)

Identify potential problems or shortcomings of present
measures; b) Prioritize risk management plans; c) 1.
Confirm that problems identified earlier are addressed; d)
Justify budgetary provisions. 2.
IV. DISCUSSION & CONCLUSION
3.
Proposed metric based assessment and reporting plan
has been designed as per the specific needs of an
academic environment. Additional roles are created and 4.
their key responsibilities & accountabilities are defined,
which is necessary to manage this complex and evolving 5.
discipline. Each metric is associated with a predefined
role. As each metric is prioritized, an iterative assessment 6.
can be planned. Secondly, exception handling is
7.
distributed across the institution, and approach of pointed
reporting is adopted. Security exceptions are reported 8.
directly, without wasting any time to the predefined roles
responsible for that particular security control. This 9.
pointed reporting helps in reducing response time, as right
person is involved and more time is available for planning 10.
and implementation. The proposed assessment metric and
pointed reporting structure, will improve overall security 11.
governance. Reduction in response time is very crucial for
12.
effective security governance.
Future work: Design of an automated process to
assess vulnerability score using open data bases. Another 13.
issue is design of a model projecting an optimal
investment plan, keeping in view the financial value of
potential tangible or intangible gains or losses.
REFERENCES
Berinato, S., “The state of information security 2003”,
CIO Magazine 17, 2 (Oct. 15, 2003).
M. Peter Adler, “Uniform approach to information
security compliance”, EDUCAUSE Review, pp46–61,
vol. 41, no. 5 (September/October 2006):
Thomas Bellocci, CERIAS Tech Report 2001-34
Information Assurance in Networked Enterprises, No.
01-05 Purdue University West Lafayette, IN 47907
O. Sami Saydjari, QoP’06, October 30, 2006,
Alexandria, USA. ACM 1-59593-553-3/06/0010.
Swanson, M., Bartol, N., “Security Metrics Guide for
IT Systems”, NIST Sp. Publication 800-55, Jul., 2003.
National Institute of Standards and Technology,
http://csrc.nist.gov/publications
Ron Ross, NIST SP-800-53A, Guide for assessing the
security controls, December 2007
Swanson, M. Security Self-Assessment Guide for IT
Systems. NIST Spl Publication 800-26, Nov., 2001.
Information Technology ACT 2000,
www.eprocurement.gov.in/news/act2000mod.pdf
Washington university policy document,
http://www.wustl.edu/policies, October 2006.
Auckland University Newzeland, Security policy and
organization http://www.auckland.ac.nz, Jan 2008
Mohammad Saad Saleh, Saad haj bakry, “Using ISO
17799: 2005 IS Management, Stope View with six
sigma approach”, Int. Journal of network mgt, 2006.
University of Houston, Information Security Manual,
www.uh.edu/infotech, 2008

370
 2009 International Conference on Computer Engineering and Technology
An Extended Algorithm to Enhance the Performance
of the Gridbus Broker with Data Restoring Technique
Abu Awal Md. Shoeb, Altaf Hussain, Md. Abu Naser Bikas, and Md. Khalad Hasan
Department of Computer Science & Engineering
Shahjalal University of Science & Technology
Sylhet, Bangladesh
{johnshoeb, altaf_sust_82, bikasbd}@yaho.com, khalad-cse@sust.edu
Abstract—There are various types of Grids have been developed
to support different types of Applications. The Gridbus broker
that mainly focused on Data Grid mediates access to distributed
resources by discovering suitable data and computational
resources, job monitoring, accessing local or remote data sources
during job execution, collecting and presenting results. In this
paper, we present an enhanced version of Grid Service Broker
scheduling on Global Data Grids with job restoration point. In
the present version of scheduling algorithm, if any job associated
with an application becomes failed during its runtime then the
scheduling algorithm just marked the job as failed and reset. But
it does not keep any track of the percentage of the work has
already been done by the job. In Contrast, our proposed
enhanced algorithm utilizes a restore point to store the
proportion of task completed by an executor. In case of failure, it
starts the job from that restored point rather than its initial
point. From the experimental result, we have found that our
proposed algorithm increases the performance of the present
scheduling algorithm.
Keywords- Gridbus Broker, Data Grid, Data Restoration
Technique, Data Restoration.
I. INTRODUCTION
Grid computing is very useful in the large scientific
projects. A grate number of researcher organization and
communities using grid computing to carry out their sheared
resource out come. However the using of the power of Grid
Computing remains to be a challenging problem for the users
due to the complexity involved in the creation of the
application and deployment on distributed computational
resources. The Gridbus broker [1] is an advanced service-
oriented meta-scheduler for data Grids. It supports different
Grid middleware and services. It takes care of many functions
that Grid applications require such as resource selection, job
scheduling and management, data access to any distributed
Grid resources and handling faults that may occur during
execution, collecting result and presenting them. In Gridbus
broker the task commands are encapsulated within an Agent
which is dispatched to and executed on the remote machine.
On completion of execution, the Agent returns any results to
the broker and provides debugging information. The
Monitoring component keeps track of job status – whether the
jobs are queued, executing, finished successfully or failed.
Here Gridbus broker does not keep track of the calculated
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 371
DOI 10.1109/ICCET.2009.233
result of a failed job. In this paper we have present an
enhanced version of Grid Service Broker scheduling on
Global Data Grids with the failed job restoration point.
This paper is organized as follows: Section II discusses the
Related Works and Section III shortly discusses Gridbus
Broker Architecture and the problem scope of Gridbus Broker
Architecture and Scheduling. Section IV elaborates our
proposed algorithm to counteract the problems of Gridbus
Broker Scheduling. Section V illustrates the Experimental
Result and Performance Analysis of algorithm with some
examples. We conclude in section VI.
II. RELATED WORK
As the scheduling of independent jobs on distributed
heterogeneous resources [2] is a well-studied problem, so here
we discussing only a few major work which deal with Grid. In
the context of Data Grid, the Storage Resource Broker (SRB)
[3] from San Diego Supercomputing Centre (SDSC) provides
middleware for storing datasets over a network and accessing
them. Although it does not deal with applications executing
directly.
The Gridbus broker extends the Nimrod-G [4] computational
Grid resource broker model to distributed data-oriented grids.
Nimrod-G specializes in parameter-sweep computation.
However, the scheduling approach within Nimrod-G aims at
optimizing user-supplied parameters such as deadline and
budget [5] for computational jobs only. Like Nimrod-G, the
AppLeS PST [6] supports deployment of parameter-sweep
applications on computational grids, but its adaptive
scheduling algorithm emphasizes on data-reuse.
Harold, Shazia, Liao, Krishna, Buyya developed a Grid
Thread Programming Environment (GTPE) [7] for the
Gridbus Broker, a software grid resource broker that provide
transparent access to grid services. However, it does not
provide any way to save the output of thread at any point of it
runtime.
Venugopal, Buyya, Winton [1] focuses on a resource
scheduling strategy within a Data Grid but concentrated on
adaptive scheduling algorithms and brokering for
heterogeneous resources that are shared by multiple user jobs.
Similar to [1], our work focuses on a resource scheduling
strategy within a data Grid but we concentrate on runtime
thread scheduling algorithms and brokering for heterogeneous
resources that are shared by multiple user jobs. However our
work focuses on failed job and reusing the execution time by
creating restoration point. Our modified ‘File based Grid
Thread Implementation’ provides a way of saving the output
of a thread at any point of its life time.
III. GRIDBUS BROKER OVERVIEW
The Gridbus broker inputs are considered as task and the
parameter values associated with the task. The parameters can
be static or dynamic. Among the other component the Grid
Index Information service (GIIS), gathers information from
remote information services i.e. availability and the properties
of compute resources. The Data Service Catalogue organizes
the data files as Logical File Names (LFNs). Network weather
service (NWS) obtains the information about the current
available bandwidth between the data sources and the compute
resources. Global Access to Secondary Storage (GASS)
obtains the inputs or parameter file form the secondary storage
if required. Logical and Physical File Names (LFNs), each
LFN maps to one or many Physical File Names (PFNs)
somewhere on the Grid and the Job Monitor keeps track of job
status, whether the jobs are queued, executing, finished
successfully or failed. A task is further considered as sequence
of commands that focuses user requirements. The task along
with its associated parameters decomposed in to jobs. A job is
unit of work that is send to a Grid node. The set of jobs along
with the set of service nodes are an input to the scheduler. For
jobs requiring remote data, it interacts with a NWS to obtain
the information about current available bandwidth between the
data sources and the compute resources. It then uses this
information to schedule jobs by optimizing on the amount of
data transfer involved.
In the current implementation, the jobs are dispatched to the
remote node through the Actuator component. The Actuator
submits the job to the remote node using the functionality
provided by the middleware running on it. The Actuator has
been designed to operate with different Grid middleware
frameworks and toolkits such as Globus 2.4 [9] that primarily
runs on Unix-class machines and Alchemi [10], which is a
.NET based grid computing platform for Microsoft Windows-
enabled computers. The Gridbus broker scheduling algorithm
schedules a job depending compute resources and data host
associated with a job that is to be deployed to an executor.
Firstly it gathers the information about the compute resources
and data hosts using GIIS and Data Service Catalogue and
determines the job completion rate for every compute resources
for its history. For a job to be deployed, identifies the data
hosts associated with that job and for every pair of data and
compute resources, it finds the pair with earliest job completion
rate and if such a pair exist the scheduler assigns the job to that
pair. After the job has successfully finished at the remote node
(i.e. executor), its returns any results to the Broker. If any job
became failed, it is marked as failed and reset for rescheduling.
A. Problem Scope of Gridbus Broker Architecture And
Scheduling
When a job is executed in the remote node (i.e. executor),
the Agent returns any results to the broker. If any jobs are
372
failed during execution, the Agent does not return the output
file (i.e. remote data) to the broker, it returns only the status of
the job whether the jobs are finished successfully or failed. If a
job becomes failed the job is marked as failed and reset for
rescheduling without considering the job has been done so far.
We all know that, in a network or Internet based application
there is a high possibility that a GRID node can be interrupted
for the inconsistent nature of the network. In the existing
implementation there is no mechanism to save the percentage
of job result to be recovered when the job is failed. As the
GRID job can be a huge one so the lost of the job can cause a
critical time factor. For example, scheduler submits a job to a
Grid node that is 30 minutes long. If that job is lost or failed
after 27 minutes of successful execution then the scheduler will
reassign the job without using the results or data of the last 27
minutes. So the job will take roughly 57 (27+30) minutes
rather than 30 minutes.
IV. PROPOSED SOLUTION
Our proposed solution, we have kept the track of running job.
Suppose, the remote node is failed after 90% of the job is
completed, then the Agent will return the output file (i.e.
remote data) with the job status to the broker. Since the job
status is failed, the job is reassigned to another remote node
with the last generated output file (i.e. last remote data) to
restart the job from the failed point (i.e. from the 90%, not
from the beginning).
In our application (File Based GRID Thread Implementation),
we created a random access file associated with each thread
and writing the results to that file and thus creating restoration
point. The file only contains the last two computed output of a
thread. The file size depends on the type of application but in
our application we managed it to be quite small. This
generated file is stored in the Manager Node. Manager can
reassigns that thread to any executor node with the same
thread ID from the thread pool. A priority for each thread can
be explicitly specified when it is created or submitted. Threads
are scheduled on a Priority and First Come First Served
(FCFS) basis [8], in that order. When the Manager distributes
the thread to any Executor node, the node first check if there is
any file is available with its own thread id. If any file is
available for that thread, the node then recovers the resultant
value of that thread from the file created and starts execution
from the recovered value. If no file is found for that thread
then the thread will start execution from the beginning and
also create a file in the Manager Node using Thread ID (i.e.
filename) as Thread ID remain same throughout the lifetime of
an application.
In the global grid environment, these generated restoration file
can be stored in the remote data host assigned with that for
that Grid node or can be stored in some different host
depending on the bandwidth and cost of data transfer etc. For
the jobs requiring remote data, the scheduler interacts with the
Figure 1: Modified Adaptive scheduling algorithm for Data Grid
 Initialization
1. Identify characteristics, configuration, capability,
and suitability of compute resources using the Grid
information services (GIS).
2. From the task definition, obtain the data query
parameters (if present), such as the logical file
name
a. Resolve the data query parameter to obtain the list
of Logical Data Files (LDFs) from the Data Catalog.
b. For each LDF, get the data sources or Data Hosts
that store that file by querying the Data Catalog.
Scheduling Loop
Repeat while there exists unprocessed jobs. [This step
is triggered for each scheduling event. The event
period is a function of job processing time,
rescheduling overhead, resource share variation, etc.]:
3. For each compute resource, predict and establish
the job consumption rate or the available resource
share through the measure and extrapolation strategy
taking into account the time taken to process previous
jobs. Use this estimate along with its current
commitment to determine expected job completion
time.
4. Repeat if there exists any previously failed job
a. Select the restoration file host (data host) that
contain LDF associated with that job.
b. Repeat from 6(b) to 6(d).
c. If there exits such a resource pair, then assign the
job to compute resource and remove it from the
Failed-Jobs-List.
5. If any of the compute resource has jobs that are yet
to be dispatched for execution and there is variation in
resource availability in the Grid, then move such jobs
to the unassigned-Jobs-List.
6. Repeat until all unassigned jobs are scheduled or all
compute resources have reached their maximum job
limit.
a. Select the next job from the Unassigned-Jobs-List.
b. Identify all Data Hosts that contain the LDF
associated with the job.
c. Create a Data-ComputeResource-List for the
selected job:
• For each data host, identify a compute resource that
can complete the job earliest given its current
commitment, job completion rate, and data transfer
time using current available bandwidth estimates.
d. Select a data host and compute resource pair with
the earliest job completion time from the Data-
ComputeResource-List.
e. If there exist such a resource pair, then assign the
job to the compute resource and remove it from the
Unassigned-Jobs-List.
7. End of scheduling loop.
Network Weather Service (NWS) [11] to get the current
bandwidth condition.
We can create a new instance for the restoration file in the
Logical File Names (LFNs) which maps to Physical File
Name of the restoration file stored in the appropriate location
373
or data host. This instance can be updated and can use for the
selection of the restoration file host. The Gridbus broker can
resolve the restoration files by querying the Data Service
Catalogue.
In the case of job failure, the LFNs instance can be updated
that map the Physical File Names of the restoration files. When
the scheduler reschedules that failed job for execution, the
restoration file can be assigned with that job. This technique
enables the new executor can retrieve the saved file from the
data host before execution and stat execution from that
successful executed point. Thus this data restoration technique
saves the execution time for a failed job, as we did not need to
start executing the job from the very beginning.
Depending on this approach, we modified the existing
scheduling algorithm of the Grid bus broker. The modified
Gridbus broker scheduling algorithm is stated in Figure: 1. We
added a new module (Module: 4) in the existing algorithm for
rescheduling the failed job. This algorithm gives us a way of
rescheduling the failed job so that we can utilize the saved
output the thread executed before being failed.
V. EXPERIMENTAL RESULTS AND PERFORMANCE
ANALYSIS
Here, we have considered thread as job and implemented a
file based GRID thread implementation using Alchemi Toolkit
so that we can restore the lost or failed thread result. The
details description of the file based Grid thread
implementation is beyond the scope of this paper. Here we
have found that our modified approach gives better
performance than the existing approach in almost every case.
The performance of modified approach depends on how long a
thread is executed before it is being failed. The much longer a
thread is executed before it is being failed the much better the
performance is obtained by our modified file based
implementation.
To evaluate the performance analysis we have developed one
Application program in two different approaches. One of them
is existing approach (where failed thread starts its execution
from its initial state) and another one is our file based
approach. In second approach, we can save the output of any
thread. This technique enabled us to create the restoration
point of any thread. By utilizing these restoration points, we
can start the failed job form the last saved point.
Here we have shown different performance comparison table
where our modified approach gives better performance than
the existing approach in every case. The performance of
modified approach depends on how long a thread is executed
before it is being failed. The much longer a thread executed
before it is being failed the much better performance is
obtained by our modified file based implementation.
In both program, same input is used to compare between
them, both give the same result as well. Both application give
the total number of factors of a given input value (e.g. 10^10).
For comparing between the two implementations we have
taken different datasets in two different scenarios.
A. Scenario-1 (Considering No Thread Failure)
1) Here, different number of thread has been executed
using both algorithm where all the threads successfully
executed in the Executor node without any interruption and
returned the result to the Manager node.
Different number of threads has been interrupted and datasets
are taken using both algorithm.
TABLE III. TIME SAVED BY THE MODIFIED ALGORITHM WITH
COMPARED TO THE EXISTING ONE WHEN MULTIPLE THREADS ARE
INTERRUPTED

TABLE I. OVERHEAD OF OUR MODIFIED ALGORITHM WITH Previous Algorithm Modified Algorithm
COMPARED TO THE EXISTING ONE WHEN NO THREAD IS INTERRUPTED Test Thr Is Time Per Total Time Per Total
Cas ead Inte Thread Time Thread Time
Test # of Time Taken (min) Overhead e No rrup (sec) (min) (sec) (min)
Cases Thread (sec) ted?
Previous Modified
Algorithm Algorithm 1 1 Yes 36.843750 26.562500
2 No 17.718750 01:13.28 17.890625 01:03.2
1 3 0:55.37500 0:56.328125 0.953125 3 No 18.703125 125 18.765625 50000
2 5 1:35.81250 1:39.140625 3.328125 2 1 Yes 37.906250 27.328125
3 8 2:41.71875 2:43.906250 2.1875 2 No 17.781250 01:35.09 18.796875 01:14.1
4 15 4:25.62500 4:31.437500 5.8125 3 Yes 39.406250 375 28.046875 71875
3 1 Yes 41.937500 26.343750
2 Yes 40.515625 02:04.96 25.843750 01:18.0
Table 1 shows a slight overhead of our modified algorithm. 3 Yes 42.500000 875 25.843750 31250
This overhead indicates the time access file. From the above
comparison, we clearly see that the exiting procedure is better
when the number of thread is small but when the number of
thread increases the modified approach finish the thread almost 140

Total Time Taken(sec)

at the same time. File read and writes overhead is negligible 120
when number of thread increases. 100
80
B. Scenario-2 (Threads Are Terrupted) 60
Two approaches have also been compared where one or 40
more threads have been failed during execution time and 20
rescheduled by the manager. 0
For all the test cases we use: 0 1 2 3 4
Input Value = 1000000000
No of Failed Thre ad
Average Thread Execution Time = 19 sec (measured without
Existing Implementation Modified Implementation
interruption)
Time between the interrupt and restart a thread = 5 sec
Condition 1 Figure 2: Comparison between the two implementation in case of thread failure
No of Thread = 1
The Thread is interrupted at different time counting from the From the Table 3 and Figure 2, it is clearer that, modified
start of the execution. algorithm gives better result when one or more thread fails
during execution and performance is enhanced when the
degree of failure of thread increases in GRID environment.
TABLE II. TIME SAVED BY THE MODIFIED ALGORITHM WITH
COMPARED TO THE EXISTING ONE WHEN ONLY ONE THREAD IS As we can save the result of a failed thread and have shown
INTERRUPTED AT DIFFERENT TIME that this file based implementation gives better performance
and if we consider this approach in job we can say that our
Test Interrupted Time Taken (sec) Saved
Cases Time Time proposed modified algorithm will obviously give better
Previous Modified
(nth Sec) Algorithm Algorithm (sec) performance than the existing Gridbus broker scheduling
1 5 30.56250 23.125000 7.437500 algorithm.
2 10 32.65625 23.015625 9.640625
3 15 36.65625 27.453125 9.203125 VI. CONCLUSION
In this paper, a modified algorithm has been presented for
the scheduling of the Gridbus broker. We implemented an
From the above comparison, it is clear that in every case extended GRID thread application program where we showed
our modified approach gives better performance (i.e. time that using file based modified algorithm gives better time
improvement) than the exiting approach. It is also clear that improvement with compare to the existing thread
modified implementation performs well when a thread fails implementation. As Grid thread is the last unit of the job that is
after maximum execution of its lifetime. executed in the executor node and if the file based approach is
Condition 2 used for failed job then our modified Gridbus broker algorithm
No of Thread = 3 gives better performance than the existing Gridbus broker

374
scheduling algorithm. Thus, it has the potential to push the Systems,Volume 15, Numbers 5-6, pp. 757-768, Elsevier Science,
Grid computing one step further. Amsterdam, Netherlands.

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[1] Venugopal, S., Buyya, R., Winton, L.: A Grid Service Broker for
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Conference (Middleware 2004), Toronto, Canada (2004)

[2] M. Maheswaran, S. Ali, H. J. Siegel, D. Hensgen and R. F.Freund,

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[3] C. Baru, R. Moore, A. Rajasekar, and M. Wan, The SDSC Storage

Resource Broker, in Proceedings of the 1998 conference of the
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(CASCON’98), Toronto,Canada, October 1998, ACM Press, New
York, USA.

[4] D. Abramson, J. Giddy, and L. Kotler, High Performance

Parametric Modeling withNimrod/G: Killer Application for the
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Cancun, Mexico, IEEE Computer Society Press, Los Alamitos,CA,
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[5] R. Buyya, D. Abramson, and J. Giddy, an Economy Driven

Resource Management Architecture for Global Computational
Power Grids, Proceedings of the 2000 International Conference on
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(PDPTA 2000), June 26-29, 2000, Las Vegas, USA, CSREA
Press,Las Vegas, USA, 2000.

[6] H. Casanova, G. Obertelli, F. Berman, and R. Wolski, The AppLeS

Parameter Sweep Template: User-Level Middleware for the Grid,
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Computer Society Press, Los Alamitos, CA, USA.

[7] Harold Soh, Shazia Haque, Weili Liao, Krishna Nadiminti and
Rajkumar Buyya, “GTPE: A Thread Programming Environment
for the Grid”, Proceedings of the 13th International Conference on
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Allied Publishers, Mumbai, India, ISBN: 81-8424-109-7),
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[8] Srikumar Venugopal, Rajkumar Buyya and Lyle Winton, A Grid

Service Broker for Scheduling e-Science Applications on Global
Data Grids, Concurrency and Computation: Practice and
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[9] Ian Foster and Carl Kesselman, “Globus: A Metacomputing

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[10] A. Luther, R. Buyya, R. Ranjan, and S. Venugopal, Alchemi: A

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375
 International Conference
on Computer Engineering and Technology

Session 6
 2009 International Conference on Computer Engineering and Technology
Prediction of Ship Pitching Based on Support
Vector Machines
SUN Li-hong
Collegeof Foundation Science Harbin University of
commerce
E-mail address: honghong03050305@163.com
Abstract—ship pitching influences mostly ship motion, it’s
important to study ship pitching modeling and prediction in
order to improve ship’s seaworthiness. Based on the random
character of ship movement, this paper put forward a method
for prediction of ship pitching movement with SVM. Based on
the phase-space reconstruction theory, the method, the
characteristic, and the selecting of the key parameters in the
modeling is discussed. Using support vector machine model to
predict ship pitching series, the RBF kernel function is
introduced, which simplified the course of solving non-linear
problems, It is shown by the study case that the model
proposed in the paper has better generalization and prediction
accuracy.
Keywords- phase space reconstruction; support vector
machine; ship pitching; prediction; RBF kernel function
1 Introduction
ship navigates in the wind wave, as result of wind, wave
and flow, it swings back and forth inevitably. Badly sway
can make employee to go down work efficiency on the ship,
passengers feed the fishes, goods damage, naval battle
effectiveness falls. Marine sway has a six degrees of freedom
in common: ship’s rolling, pitching, heaving motion and so
on. Thereinto rolling and pitching influence most for ship
motion . marine rolling capability is an important index for
ship enduring wave[1,2]. Existing many reducing sway
equipment aim at decreasing rolling, now an effective
method for pitching isn’t existence. Generally people make
shipping to evade dangerous period and reduce accident
applying prediction technique. Now many countries attach
importance to the modeling prediction of ship movement and
develop many research. Home and overseas researchful
method may sum up two big genus, frequency domain
analytical method and time domain analytical method.
Among frequency domain analytical method convolutional
method was often used, in time domain analytical method
kalman filtering and time series analysis[3] method. It was
researched to Use time series analysis to predict extremely
short-time motion attitude. It needn’t know any priori-
information for ocean wave and the state equation of
shipping motion, it’s only to know historical data of ship
movement and seek rules to forecast. Anciently the method
of predicting the ship were time series, Periodogram,
artificial neural network, grey system theory[4-9] and so on.
It’s important to study the theory of ship pitching, which can
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.24
SHEN Ji-hong
Science College Harbin Engineering University
E-mail address: shenjihong@hrbeu.edu.cn
help us know the disciplinarian of pitching and master and
use it to service for the ship navigate.
SVM becomes new study focus after neural network in
the machine learning field. The support vector machine
(SVM) was initially developed by Vapnik and his group to
solve classification problems and has been successfully
applied to a number of real world problems. Some
applications of SVMs for classification are isolated
handwritten digit recognition speech recognition, and face
detection in images. Recently, its applications have been
extended to the domain of regression problems. In the case
of regression, SVMs have been applied to benchmark time
series prediction tests, the Boston housing problem and
financial forecasting. In most of these cases SVM
generalization performance either matches or is significantly
better than competing methods.
SVM is based on the structural risk minimization
(SRM) principle[10-12], which has been shown to be superior
to traditional empirical risk minimization (ERM) principle.
SRM minimizes an upper bound on the expected risk while
ERM minimizes the error on the training data. highly
generalized SRM results in the regularized ERM with the "-
insensitive loss function. In this paper, we apply support
vector machine to ship pitching movement prediction, by
extending support vector machine regression (SVMR) for the
estimation of linear and nonlinear regressions, first
reconstruction the pitching phase space, and then the SVM
forecast model of reconstruction time series is introduced to
ship pitching series, finally predicting the future motion. At
the same time SVM compared to improved GM(1,1) model
,the precision of SVM is higher than improved GM(1,1)
model, it provides a new approach for pitching motion time
series forecasting.
2 time series phase reconstruction
For single variable time series, Packard put forward a
method of reconstruction phase, its main idea is to change
one dimension time series into many dimensions time series
through delaying time and embedding dimensions, so
reconstructing dynamics system. Supposing pitching
movement time series xt , t = 1, 2, , n , if we can select
embedding dimensions m and delaying time τ ,
reconstructing the phase is
yi = {xi , xi +τ , xi + ( m −1)τ } (1)
379
 Among (1), i = 1, 2, , n − ( m − 1)τ 。 So every yi forms SVM regression performs linear regression in the
a phase point of m dimension phase ； the above all higher dimensional feature space using ε -insensitive loss
m dimension phase constitutes a phase, while the line and, at the same time, tries to reduce model complexity by
T
between phase points depicts the track of ship pitching minimizing w w : This can be described by introducing
motion in phase space. Working out correlation dimension ς
(non-negative) slack variables i , i to measure the
ς*
of reconstructing phase space can estimate if original time
series exist fractal character. The key of phase space deviation of training samples outside ε insensitive zone.
reconstruction is to how to ascertain delay-time τ and Thus, SVM regression is formulated as minimization of the
following functional:
embedding dimension m . Usually people adopted auto- m
1
correlation function algorithm to calculate τ , using G-P min* wT w + C ∑ (ς i + ς i* ) (4)
algorithm by Grassburger and Procacia putting forward G-P w, b ,ς i ς i 2
i =1
algorithm to calculate m . ⎧ yi − [ wT Φ( xi ) + b] ≤ ε + ς i
⎪⎪
3 Regression algorithm of SVM Subject to ⎨[ wT Φ( xi ) + b] − yi ≤ ε + ς i*
3.1 regression estimate algorithm based on SVM ⎪
⎪⎩ς i , ς i ≥ 0
*

SVM initially is used to solve mode identify, using its
classification algorithm to realize better generalization
function. With the introduction of ε -insensitive loss
function [10]，SVM already was extended to solve the non-
linear regression estimate. In regression, the goal is to
estimate an unknown continuous-valued function based on a
finite number set of noisy samples. Assumed that the
sample points {( xi , yi )} (i = 1, 2, m), m is sample numbers,
xi is input vector, yi is the target value. Considering sample
to take on non-linear relation under most condition, through
the flowing method estimating function f : Given a set of
centered input vectors, the idea is to first map the original
input vectors into a higher dimensional feature space using a
non-linear mapping and then to calculate the linear
regression, so reaching the effect of non-linear regression in
original space. In SVM regression, the input xt , t = 1, 2, , n
is first mapped onto an m dimensional feature space using
some fixed nonlinear mapping Φ , and then a linear model is
constructed in this feature space. The linear model
f ( x, w) is given by
f ( x, w) = wT Φ ( x) + b = ( w, Φ( x)) + b （2 ）
where C is a positive constant (regularization
parameter) Parameter C determines the trade off between
the model complexity (flatness) and the degree to which
deviations larger than 1 are tolerated in optimization
formulation (4).For example, if C is too large (infinity), then
the objective is to minimize the empirical risk only, without
regard to model complexity part in the optimization
formulation (4).
3.2 Regression estimating function expression
By adding Lagrangian multipliers α i and α i* , the
Lagrange function is described as follows
1 m m
L = wT w + C∑ ς i + ς i* − ∑ α i (ε + ς i − yi + wT Φ ( xi ) + b)
2 i =1 i =1
m m
−∑ α i* (ε + ς i* + yi − wT Φ ( xi ) − b) − ∑ (ηiς i − ηi*ς i* )
i =1 i =1
Where α i , α i* ,ηi ,ηi* Lagrangian multipliers, they are
nonnegative. According to the KKT condition[13] theorem in
optimization theory, the solution of the optimization is
given by the following condition
⎧ ∂L m

where Φ ( x) represents a high-dimensional feature ⎪ ∂b = 0 ⇒ ∑ (α i − α i* )

⎪ i =1
space, which is nonlinearly mapped from the input ⎪
⎨0 ≤ α i , α i ≤ c, i = 1, 2, m （5）
*
space xt ,and b is the “bias” term. w is weights vector, ⎪
⎪ ∂L = 0 ⇒ W = (α − α * )Φ( x )
m
( w, Φ( x)) is inner-product. Often the data are assumed to be
⎪⎩ ∂W ∑ i i i
zero mean, so the bias term in expression (2) is dropped. i =1
The quality of estimation is measured by the loss function This optimization formulation can be transformed into
L( y − f ( x), x) . SVM regression uses a new type of loss the dual problem, and its solution is given by
function called ε-insensitive loss function proposed by 1 m m
max* = − ∑∑ (α i − α i* )(α j − α *j ) K ( xi , x j )
Vapnik. α i ,α i 2 i =1 j
L( y − f ( x), x) = y − f ( x) ε = m m
−ε ∑ (α i + α i* ) + ∑ yi (α i − α i* )
⎧⎪0 y − f ( x, w) − ε i =1 i =1
⎨ (3)
⎪⎩ y − f ( x, w) − ε ⎧m
⎪∑ (α i − α i ) = 0
其它 *

The empirical risk is: Subject to ⎨ i =1 (6)

1 n ⎪α , α * ∈ [0, C]
Remp (ω ) = ∑ L( yi , f ( xi , ω )) ⎩ i i
n i =1

380
 Where K ( xi , x j ) = Φ ( xi )T Φ ( x j ) is kernel function, it
depicts the inner-product of higher feature space, Usuall it
may select satisfying Mercer condition. Also, SVR can
estimate a non-linear function. by employing a kernel
function, The regression function estimated by SVR can be
written as the following kernel expansion:
m
f ( x) = ∑ (α i − α i* ) K ( x, xi ) + b
i =1
Selecting a particular kernel type and kernel function
parameters is usually based on application-domain
knowledge and may reflect distribution of input values of
the training data. familiar kernels function were polynomial
kernel, radial basis function kernel and so on. So f in (7) is
decided by α i , α i* . According to the character of SVR, The
sample points that appear with non-zero coefficients in (7)
are called support vectors (SVs). (7) depicts a convex
programming, which every solution of which is global
optimization solution. SVR avoids poor fitting and over
fitting of data through (6), so SVM is a universal and
flexible tool to solve regression problem.
This paper the realization of SVM adopts the structure
like Fig1, where α i − α i* is network weigh, x1 , x2 , xm is
input vecter, y is output value. Its structure is similar to
ANN, its numbers of hidden nodes are the numbers of
SVM.
y Step 3 calculating α i , α i* to use LIBSVM algorithm.
α1 − α1* α 2 − α 2* α j − α *j
……….
x1 x2 … ….
Fig1 the structure of SVM
4 Applipplication of SVM for ship pitching
movement
4.1 reconstruction of pitching data
This paper used data imitating is pitching angle time
series for some ship, it is real measured by ship model pool
experiment. Supposed sampling interval of real ship data is
0.5s, choosing 60 data (30s every group), these gradual
pitching angle connects a long series, series is
{ x(t ), t = 1, 2, n} , n = 60 . Time lag is decided by
autocorrelation function, let the time τ which
autocorrelation function first vias zero be delay time; let
embedding dimension m = 2,3, , for every m , according
to G-P algorithm, drawing a figure about correlation
dimension; from the curve of embedding dimension m , we
can know when correlation dimension gets saturation, it is
the best embedding dimension of phase space. This paper
faces pitching angle series, τ = 1 , m = 12 。
4.2 training algorithm and constitution of kernels
Realization of classical SVM algorithm can summarize
to solve a nonlinear programming with linear constraints,
the algorithm is complexity, convergence rate is slow. Later
many scholars did a series of improvements, but
(7) applicability isn’t enough good. This paper uses LIBSVM
algorithm provided by Chang Chichun and Lin Chihjen in
2001 year, the method used for reference the merit of other
method, affording an ascertain method for a working
dataset, so optimization problem becomes a typical
quadratic performance with analytic solutions. Comparing
to of ordinary optimization problem, it uses much less EMS
memory and computing rate is quick, the precision of result
is high. Selection of kernels affects mostly pitching
movement prediction, this paper test indicates RBF kernels
can achieve higher precision in ship pitching movement
prediction. Besides, parameter C also affects capability of
SVM.
43 SVR steps for ship pitching prediction
The regression based SVM which is to predict the Ship
pitching is described in the following:
Step 1 Choosing samples and numerical normalization
processing for the pitching angle series.
Step 2 establishing training and test samples and
reconstructing phase space for them.
Step 4 combining Eq.(7) and using test samples to predict
the future pitching angle.
5 Experimental results
60 Simulated datasets are from ship pitching angle in
30s, according to τ = 1 , m = 12 , reconstructing pitching
angle series, the former 42 databases are used to training,
the latter 18 datum are used test data. Training samples are
xm the following format.
⎡ x1 x1+τ x1+ ( m −1)τ ⎤ ⎡ x2 + ( m −1)τ ⎤
⎢ ⎥ ⎢ ⎥
⎢ x2 x2 +τ x2 + ( m -1)τ ⎥ ⎢ x3+ ( m −1)τ ⎥
X =⎢ ⎥ Y =⎢ ⎥
⎢ ⎥ ⎢ ⎥
⎢ ⎥ ⎢ ⎥
x x
⎣ n -1− ( m −1)τ n −1− ( m − 2)τ xn −1 ⎦ ⎣ n ⎦ x
Based on above learning sample learning SVM
regression function. The selection of parameters for SVM
model usually based on fact, under the control of fitting and
forecasting precision, getting model parameter C , ε , σ by
optimal combination, where ε controls the error fitting,
consequently controlling the number of SVM and the ability
of generalization, if ε is bigger, the number of SVM is few,
precision isn’t high relatively. In this paper ε = 0.01 ，
C , σ control complexity of model, thereby, a ‘good’ value
for C can be chosen equal to the range of output (response)
values of training. However, such a selection of C is quite
sensitive to possible outliers, so we propose instead the
following prescription for regularization parameter:
381
 C =10000 σ = 0.5 .in order to compare to SVM model, in
this paper selecting grey GM(1,1) model, grey GM(1,1)
model is to establish model facing to data approximately
obeying exponential law, its merit is a few datasets needed
and reduces the calculation quantity. This presents empirical
comparisons for SVM Regression and GM(1,1) model,
simultaneity using MATLAB language programme to
simulate and learn in computer, forecasting curve is in Fig2
and Fig3. result indicates: the forecasting time interval of
GM(1,1) model is 7-8 second, while SVM can forecast
much longer time, We evaluated the root mean relative error
is e1 = 0.5094 e2 = 0.1795 for every method, ∵ e1 > e2 ,so
SVM method is better than GM(1,1) model forecasting
method.
paper established forecasting model of ship pitching angle
series using phase space reconstruction and SVR theory.
Phase space reconstruction mined information of single
variable sequences, the establishment of SVM forecasting
model vias learning history samples, so SVM is self-
learning process for history samples, when training samples
are finite, SVR model took on good generalization
capability and overcome disadvantage of local optimization,
so the forecasting method can improving the forecasting
precision. Imitate results show that SVM selecting RBF
kernels is applied to short-time forecasting for ship pitching
motion, it got higher calculating precision than GM(1,1)
model. At the same time the SVM model is simple and
convenient in modeling and training rate is quick.

GM(1,1) model and original

0.5 REFERENCES
original data [1] TAO Rao-sen.Sea-keeping of Ships[M].shanghai:shanghai jiao tong
GM(1,1)forecasting university press,1985.
0
[2] ZHAO Xi-ren. Application of random process[M].harbin: Harbin
Engineering University Press,2003.
-0.5 [3] George E.P.Box. Time series analysis: forecasting and controlling
pitch angle

[M].Gu Lan interpret.Beijing:China Stat Press,1997.

[4] ZHAO Xiren, PENG Xiuyan.Study Status Quo of Extremely Short-
-1
Time Modeling and Predicting of Ship Motion[J]. ship engineering,
2002,(3):4-8.
-1.5 [5] WANG Huihua,LIU Wenhua,WANG Hang. Short-term Prediction of
Ship Motion Based on Neural Networks[J].computer
simulation,2006,23(5):18-20.
-2 [6] SHEN Jihong , SHANG Shouting , ZHAO Xiren. Functional
0 2 4 6 8 10 12
time t/s transformation GM( 1 ,1) model constructed for ship pitching [J].
Figure 2 GM(1,1)model and original data journal of harbin institute of technology,2001,33(3):291-294.
[7] LI Hui,WU Guofu,QI Quanyue. Multivariate Autoregressive Model
SVM forecasting ang original in Time Series for Movement State of Naval Vesse [J].journal of
0.5 system simulation, 2003,15(5):617-620.
SVM forecasting [8] CAI Feng, WANG Lin, SHI Aiguo. Extreme short term prediction of
original ship motions based on phase-space reconstruction [J].journal of
0
hydrodynamics,2005,20(6):780-784.
[9] MA Jie,LIU Xiaohe and so on. Multi-level recursive prediction
-0.5 method for pitching movement of large-sized naval vessels [J].ship
pitch angle

engineering,2006,28(1):5-8.
[10] LI Yinghong,WEI Xunkai, LIU Jianxun.engineering application for
-1 SVM [M].Beijing: weapon industry Press,2004.
[11] SUN Deshan, WU Jinpei, XIAO Jianhua. The Application of SVR to
-1.5
Prediction of Chaotic Time Series[J]. journal of system
simulation,2004,16(3):519-520.
[12] DIAO Xiang, LI Qi. On-line Training Algorithm and Its Application
-2 Based on Weighted SVR [J].journal of system
0 2 4 6 8 10 12
time t/s
simulation,2007,19(17):3970-3973.
Figure 3 SVM model and original data [13] Vapnik V·The Nature of Statistical Learning Theory[M]·NewYork:
Springer Verlag, 1999.

6 Conclusion
It is in testing and exploring moment for SVR model
application in ship pitching movement prediction. This

382
 2009 International Conference on Computer Engineering and Technology
The Methods of Improving the Manufacturing
Resource Planning(MRPⅡ) in ERP
WenChao Jiang
School of Information Management & Engineering,
Shanghai University of finance and economics
Shanghai China
Email：wenchaojiang@gmail.com
Abstract-This paper focus on the methods to improve the
traditional MRPⅡprocess according to the requirement of the
time base competition. By finding out the defections in the
process of traditional MRPⅡ, such as using the fixed lead-time,
using the fixed secure stock, and lacking of time management,
then we building 5 models and integrating these models into the
process of MRPⅡ to make all the parameters which correlate
with the environment of the enterprise become dynamic
parameters, and make the MRPⅡ obtain the ability of time
1
management.
Key words: improve, MRPⅡ, method, model, TBC
Ⅰ INTRODUCTION
Since 1990s, the market has changed greatly. As the
competition become severer, the life cycle of products become
shorter, and the demand of customers become more multiples,
the competitive advantage of manufacturing have to be shifted
form price, quality, and service to the adaptability of
manufacture. The one who have grasped time in the market
would predominate in the competition. All of these mean that
we are now in the “Time Base Competitive age”. In such a
competitive environment, how to reduce the time of
manufacturing, and put the products which loved by
customers into the market has became a very important
advantage in competition.
The MRPⅡ(Manufacturing Resource Planning) which born
in America at the end of 1970s is a very good method which
has been used in enterprise resource and manufacturing
The topic of this paper from the China National Social Science Fund project,
the approval number of this project is: 07AJY024, and this project is the
"211" key project engineering (211-6-44) of China Shanghai University of
Finance and Economics,
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.18
Jingti Han
School of Information Management & Engineering,
Shanghai University of finance and economics
Shanghai China
Email：hanjt@mail.shufe.edu.cn
management. Now, the MRPⅡ is the core of ERP(Enterprise
Resource Planning) System which has been widely used in
manufacturing industry. Nevertheless, there are some
defections in the method of MRP Ⅱ used to form
manufacturing plans in the Time Base Competitive
environment, such as the setting of fixed lead-time, the setting
of fixed secure stock, lacking of guidance in capability
resource planning, and ignores the time management in the
manufacturing management.
This paper focus on improving the process which MRPⅡ
used to form manufacturing resources plans, and resolving the
problems exist in the MRPⅡ at the TBC age. The final
purpose of this paper is to improve the MRPⅡ by integrating
5 models into the process of forming plans, and make the
thoughts of TBC combined with the theory of MRPⅡ.
Ⅱ THE DEFECTS OF TRADITIONAL MRPⅡ IN TBC
As the competitive environment of market become more
and more uncertain, the competition of enterprises shifted to
time base competition, the time become the most powerful
resource which make the enterprises to gain advantage in
competition. In the Time Base Competitive environment, how
to produce the goods wanted by customers in the shortest time
with the lowest cost and how to guarantee sufficient supply in
the lowest level of stocks become the most important points in
manufacturing management. So the defections of traditional
MRPⅡ which born at the end of 1970s become more and
more obvious, the defects are listed underneath:
1) The traditional MRP Ⅱ make the system
administrator having the tendency to set a longer
lead-time which used to form manufacturing plans.
The lead-time is a very important parameter in the MRPⅡ
383
system. Usually, the administrators like to set a longer
lead-time, because using the longer lead-time make the MRP
Ⅱ system form plans whatever how big or small the demand
is. The longer lead-time doesn’t reflect the real capability of
manufacturing, and the longer lead-time increase the difficulty
of market forecast that would increase the risk of
manufacturing.
The using of longer lead-time to form plans would also
increase the unfinished products’ stocks and secure stocks,
these stocks occupy too much circulating capital what would
decelerate the running speed of logistic system. Even worse,
the longer lead-time covers up the resource restriction existing
in the process of manufacturing, which go against the wish of
the enterprise to build an adaptability manufacturing system.
2) The traditional MRPⅡ use the fixed lead-time as
parameters, that do not satisfy the requirement of building
a adaptability manufacturing system.
The lead-time was set by the administrators when initialized
the MRPⅡ system, and the lead-time in the MRPⅡ system
would not be changed when the initialization had finished. So
the traditional MRPⅡ ignores the situation that the using of
elastic workers and the advance of manufacturing capability
would reduce the time of manufacture when using the fixed
lead-time to form manufacturing plans.
3) The precondition of forming the manufacturing plans
is the assumption of infinite manufacturing capability.
The traditional MRPⅡ system ignores the restriction of
manufacturing capability when forming the plans. So the CRP
(Capability resource planning) is a capable adjustment after
event, this adjustment is a complex and difficult task, and it
lack of guidance. The adjustment of capability without
guidance would make the administrators harder to decide how
to distribute the finite resource, and it often makes the
administrators adjust the capability without considering time
reduction.
4) The traditional MRPⅡ lack of restriction on secure
stocks.
The secure stocks are the stocks set in order to prevent the
halt of manufacturing system which cause by the fluctuation
of demands, and the secure stocks correlate with the fluctuant
degree of demands. The quantity of secure stocks is another
parameter set when the MRPⅡ system is initialized. Like the
lead-time, the quantity of secure stocks is also a fixed
parameter in traditional MRP Ⅱ system. It ignores the
384
situation that the demand is fluctuant. It should set bigger
secure stocks when the fluctuant degree of demand was big,
and set smaller secure stocks when the fluctuant degree of
demand was small.
Ⅲ IMPROVE ASSUMPTION
In order to make the improvement more strictly, there are
some assumptions of the enterprise environment that should
be set first:
1) The running environment of MRPⅡ is MTO (Make
to Order).
2) For the finished goods the MRPⅡ use the direct
quantity; For the semi-finished goods or purchase
material the MRPⅡ use the fixed quantity, and the
fixed quantity is the most economical quantity.
3) For the finished goods and semi-finished goods the
companies do not set the secure stocks.
4) For the purchase material the companies set the secure
stocks to prevent the halt of manufacturing system.
5) The demands of goods satisfy the normal distribution
in the planning period, and the demands are all
independent.
Ⅳ IMPROVE METHOD AND THE MODELS
In order to satisfy the requirement of time base competition
and improve the traditional MRPⅡ system, there are five
models should be integrated into the traditional MRP Ⅱ
system. These five models are:
A. The Model of Resource Balance in Main Production
Scheduling
1） The description of the model
The demands of customers are changeable in the time base
competitive environment, if the company keeps too many
finished products, it would suffer great losses when the
demands changed greatly. So the model of resource balance
should be integrate into the MRPⅡ system when it forming
the MPS(Main Production Scheduling) plan. The resource
balance model is a model that balances the finite resource
while fully considering reducing the inventory of finished
products and lowering the production costs. The purpose of
this model is to improve the MPS, and make the MPS satisfy
the requirement of TBC.
2） The description of the symbol in the model
Dik - The demand of the product i in the period k; Pijk - The
quantity of product i that produce in the period j in order to

satisfy the demands in the period k;
product that produce in the period of j;
normal working hours for producing the i product in the
period of j; Ot ij -The demand of overtime working hours for
producing the i product in the period of j;
restriction of normal working hours in the period of
j; Ot j -The restriction of overtime working hours in the period
of j; Wp -The unit cost of normal working hours; Op -The
traditional MRPⅡ system, that is one of its defects. So we
Pij -The quantity of i
should integrate the secure stock forecasting model in to the
traditional MRPⅡ system to make the secure stock become a
Wtij -The demand of
dynamic parameter. The model is a statistical model that uses
the fluctuation of demand to forecast the secure stocks.
2) The description of the symbol in the model
Pj -The average total demands of j purchase material
Wt j -The
within a certain period; Rij -The average demand of i product
which correlate with the j purchase material within a certain
period; σ ij -The standard deviation of demand of i product
which correlate with the j purchase material within a certain

unit cost of overtime working hours; S i -The unit inventory period； Ω j -The standard deviation of total demands of j

cost of the i product; Ci -The manufacturing cost of i product; purchase material within a certain period； nij -The ratio of

demand between the i product and the j-related material;

ni -The working hours for producing per unit i product；
S j -The number collection of the product which relate to the j
n -The variety of products; T -The number of the periods。
3） The model of resource balance purchase material; CSL -The probability that there is no
n T T n T T n T n T
stagnation in the production within the production cycle；
minZ =∑∑∑CiPijk +∑∑∑(k − j)SiPijk +∑∑Wp×Wtij +∑∑Op×Otij
i=1 j=1 k=j i=1 j=1 k=j i=1 j=1 i=1 j=1 LT j -The time interval of procuring the j material;
k

∑P j =1
ijk =Dik i = 1,2,......, n; k = 1,2,......T LTs j -The shortest time interval of procuring the j material；
T

∑n P
k= j
i ijk ≤Wtij +Otij i = 1,2,......, n; j = 1,2,......, T LTm j -The average time interval of procuring the j material；
n

∑Wt
i =1
ij ≤ Wt j j = 1,2,......, T ; LTl j -The longest time interval of procuring the j material;
n

∑ Ot
i =1
ij ≤ Ot j j = 1,2,......, T ; ss j -the secure stock of j material；B-The collection of all

procuring material
Pijk , Wtij , Otij ≥ 0, Pijk is integer
3) The model of secure stock forecast
This model is base the assumption that the distributions of
The Pijk , Wtij and Otij is the decision variable in the model.
the demands are the normal distribution and all these
T
distribution are independent.
The Pij( P =
ij ∑ Pijk ) is the decision variable that we want get
k= j
ss j = LT × Pj + LT × Ω j × F −1 (CSL ) j∈B；
finally, it tell us how many i products should be produced in
the j period. Pj = ∑ nij Rij ；
B. The Model of Secure Stock Forecast i∈S j

1) The description of the model Ωj = ∑n σ

i∈S j
ij
2
ij
；
As we know the quantity of secure stock is fixed in the

385
 LTs j + 4 LTm j + LTl j ；
LT j =
6 LTp j = ∑ (Qti + ⎢ j
C. The Model of Producing Lead-time Calculates
1) The description of the model
The producing lead-time is a fixed parameter in the
traditional MRPⅡ system. The model of producing lead-time
calculates is the model use to calculate the producing time
according to the capability of manufacturing system. So if we
integrate the model into the MRPⅡ system, it would make
the producing lead-time become a dynamic parameter which
changes when the environment of manufacturing changes.
2) The description of the symbol
i -The No. of working center; P -The collection of all
products and purchase material; S j -The No. collection of the
working center that the j material involved; Pti - The
preparing hours for the i working center prepare to process per
product or material ； Mti - The working hours for the i
working center to process per product or material; Qti -The
waiting time before the i working center process the material；
Tt i -The time for the i working center to transport the material
or product to the next working center； nk -The production
volume of the k product; mkj -The ratio of requirement
between the k product and the j material； R j -The collection
of the product which need the j material ；
Wi -The rated
working hours of the i working center per day； LTp j -The
production lead-time of j product or material, the unit of this is
the day； PLT j -The production lead-time of j product or
material, the unit of this is the week； WT -The per week
working hours in the working calendar.
3) The model of producing lead-time calculates
386
⎡ ∑ nk × mkj × ( Pti + Mti ) ⎤
⎢ k∈R ⎥ j∈P；
⎥ + Tti )
Wi
i∈S j
⎢ ⎥
⎣ ⎦
The symbol-[ ] means give the smallest interger greater than
or equal to the result of formula
⎡ LTp j ⎤
PLT j = ⎢ ⎥ j∈P
⎣ WT ⎦
The symbol-[ ] means give the smallest interger greater than
or equal to the result of formula
D. The Model of Purchase Lead-time Forecast
1) The description of the model
The purchase lead-time is another fixed parameter in the
traditional MRPⅡ system, but the purchase lead-time would
be change when the condition of purchase is changed. The
model of purchase lead-time forecast is the model that using
the BP neural network to forecast the purchase lead-time
according to the real procuring condition. The first step of
building this model is to build a BP net work, then using the
training set to train the network. When the neural network had
been trained, we can use the network to forecast the lead-time
of purchase.
2) The step to build the model
The Fist step: Using the method such as fishbone figure or
other analytical tools to find out the factors that affect the
purchase in the real procuring environment. Then quantify
these factors, and let them to reflect the real purchase
condition.
The Second step: According to the effect factors to construct
frame of the training set of different purchase material. Then
build the training sets according to the history of purchase.
These training sets are used to train the BP neural networks.
The Third step: Build the structure of the BP neural
network. According to the demand of the forecast, choosing
the structure that contain tree hidden layer, multi-input and
single output (see the Figure 1)
Figure 1 the structure of the BP neural networks
The Forth step: Determining the transmission function in
the hidden layer of the BP neural networks.
 In the first and second hidden layer, we use the tansig We have the training set already now, then we can use this
function as transmission function. train set to train our neural network. We use the Matlab to
build this model.
1 − e− n
a(n) = tan sig (n) =
1 + e− n
In the third hidden layer, we use the purlin function as
transmission function.

a (n) = purelin(n) = n

The Firth step: Using the train set to train the neural
network what we have built. After training, we can use the Figure 2 the curve of training performance

network to forecast the purchase lead-time, and all we have to Now we have the neural network which had been trained,
do is just input the indexes which represent the purchase then we can use the model to forecast the purchase lead-time
condition. of the material.
3) ٛ Example e.g we put the index X1=1.0;X2=0.05;X3=0.5;X4=4
Table1. The train set of the model X5=1;X6=0.5 in to the model, we will get the result that the
No. X1 X2 X3 X4 X5 X6 Lead-time purchase lead-time of this material in this purchase
1 100 0.05 50 air 1 1500 5 condition is 3.3 days.
2 200 0.10 150 air 1 1250 5 E. The Model of Resource Balance in Material Requirement
3 100 0.03 30 train 2 1750 8 Plans
4 200 0.05 80 train 1 1250 6 1) The description of the model
5 100 0.10 100 air 1 700 2 The model of resource balance in the MRP is a model that
6 100 0.05 50 train 1 1500 5 integrates the technology of PERT and CPM into the MRP.
7 200 0.04 75 air 1 1500 4 The model considers the manufacturing activity as a project,
8 300 0.05 100 ship 2 1500 11 then using the BOM list and the lead-time to form a net of the
9 100 0.06 50 Ship-truck 1 1500 9 manufacturing plan. By using the network we can find out the
10 100 0.05 50 train 1 1500 5 critical path, and then we can balance the resource of
11 120 0.08 75 Ship-truck 2 450 6 manufacturing. In addition, we can implement the time
12 240 0.1 120 train 4 450 5 management in manufacturing system.
13 150 0.07 90 air 2 500 4 2) The operation step of the model
14 120 0.10 120 truck 1 500 3

15 360 0.10 150 truck 2 500 6

Table 1 is a part of the train set for the model of one

material’s purchase leading-time forecast. All of the data we in
the table we gain in a company. And according to the
investigation, there are six factors would affect the purchase
leading-time of the company for one of is purchase material.
X1: the purchase quantity of the material; X2: The condition
of the work force about the company which produce the Figure 3 the operation step of the Resource Balance model

material; X3: the efficiency of the work force about the The First step: Considering the activity of
company which produce the material; X4: The transportation manufacturing as a project and all the results of produce and
method. X5: the weather condition. X6: distance. And the purchase material as nodes. Then form the networking of
table 1 is just a part of the training set. manufacturing plan by using the results of MRP and the
results of lead-time forecast. The line between two nodes

387
represents the producing activity or the procuring activity.
The Second step: Using the result of producing
lead-time calculating and the result procuring lead-time
forecasting as the time need of activities ( t (i, j ) ) in the
networks.
The Third step: Set t E (1) = 0 , then using the formula
to calculation:
The earliest time to receive the material or product j
t E ( j ) = max{t E (i ) + t (i, j )}
( i , j )∈I
The latest time to receive the material or product i
t L (i ) = min {t L ( j ) − t (i, j )}
( i , j )∈I
The earliest start time of the activity (i, j)
t ES (i, j ) = t E (i )
The earliest finish time of activity (i, j)
t EF (i, j ) = t E (i ) + t (i, j )
The latest start time of the activity (i, j)
t LS (i, j ) = t L ( j ) − t (i, j )
The latest finish time of activity of (i, j)
t LF (i, j ) = t L ( j )
The Forth step: Finding out the critical activity
S (i, j ) =0, S (i, j ) = t LS (i, j ) − t ES (i, j )
Then calculate the time need of entire project, and
calculate the relaxation time of the non-critical activities:
r (i, j ) = t E ( j ) − t E (i ) − t (i, j )
The Fifth step: Calculating the starting time of the entire
project according to the time of the critical activity. Then
consider balancing the manufacturing resource without
prolong the time of the critical path at first. If the resource
can not be balanced in the time of the critical path, then
considering prolong the time of the critical path. Then
amend the networks and the MRP plans.
The sixth step: Distributing the networks and the MRP
plans which had been amended after the resource balance to
the department of purchase and the department of
388
manufacture.
Ⅴ INTEGRATE THE FIVE MODELS INTO THE MRPⅡ
Figure 4 The operational model of the improved MRPⅡ
Now, we considering how to integrate these five models
into the MPRⅡ, figure4 show the operational model of the
improved MRPⅡ.
We integrate the model of resource balance in main
production scheduling into the process of MPS. When the new
MRPⅡ forming the MPS, it consider the resource restriction
meanwhile. So the process of forming the MPS is the process
to balance the resource in the planning period and calculate
the most economical quantity of the finished products which
produce on each week or month. Here, the economical
quantity is the quantities which meet the demand in every
week or month with balancing the resource restriction in
every week or month, and it is the quantities which are set
by fully considering reducing the inventory of finished
products and the production costs.
Base on the MPS which have been formed, the MRPⅡ
begin to form the MRP(material resource planning). We then
integrate the Model of Secure Stock Forecast, the Model of
Producing Lead-time Calculates, and the Model of Purchase
Lead-time Forecast into the process of MRPⅡto form the
MRP.
The secure stock forecast model fetches the demand
forecast information from the ERP, and calculates the average
purchase leading-time by using the history purchase data, after
having all the information the model need, it would work out
the most appropriate secure stock of every purchase material.
All of these secure stock quantities would be used in forming
the MRP. By integrate this model into the MRPⅡ make the
secure stock which is a fixed parameter in traditional MRPⅡ
become a dynamic parameter that correlate with fluctuation of
demand.
The Model of Producing Lead-time Calculates fetches the
information of the work center and the BOM from the file
stored in the ERP system, and receives the producing
requirement of one particular product in a period from the
MPS. Then, the model can work out the producing lead-time
of every producing activity. After we integrate this model into
the MRP Ⅱ , while the productivity of the manufacturing
system changing or the produce quantities changing, the
producing leading-time will change. The leading-time become
a dynamic parameter in the new MRP Ⅱ , this make the
leading-time more accurate to reflect the time requirement for
producing. This make the producing orders which formed by
MRP have more value of guidance.
The purchase lead-time would be change when the
condition of purchase is changed. So we integrate the Model
of Purchase Lead-time Forecast into MRP Ⅱ , and before
forming the purchase material order, the system first ask for
the information of the purchase condition. The model fetches
some part information from ERP, and call for information
which can’t be obtained in the ERP system form the purchase
department, such as asking the employee of the purchase
department the question like “which vehicle to choose to
transport the purchased material?”, “what the weather would
like during the purchase period?” After having enough
information this model need, it can forecast how many days
this purchase order would take. So after integrating this model
into the process of MRP, the purchase lead-time of one
particular material become a dynamic parameter which
changes while the condition of material purchase changes.
The last model we want to integrate into the MRPⅡ
process is the Model of Resource Balance in Material
Requirement Plans. We combine this model with the CRP
(capability resource planning). Combine this model with the
CRP, we now can find out the time requirement of every
production activity, and find out when the production activity
begin or finish, then we can form the PERT networks which
contain the time information of each production activity and
information about the relation among production activities
which are contained in the process of producing one particular
product. By using the PERT networks, the system can easily
389
find out the critical activities and the shortest time needed to
produce every product. Then, by using the technology of
resource balance of PRET and CPM the system can balance
the resource distribution just adjusting the relaxation time of
the non-critical activities. These balance processes wouldn’t
increase the shortest time required to product the product.
After resource balance, the system can delivery the PERT
producing activities networks of every product every
workshop. By doing this, the manager of each workshop can
know exactly when the activities that belong to their
workshop beginning, even when these activities should be
finished. To form the producing activities networks and
delivery those to the workshop make all workers in the
workshop understand how important they are, because they
can see the fact that the efficiency of their workshop
determines the on time finishing of the whole manufacture
project. And this reflects the concept of management by
objectives.
Ⅵ CONCULUSION
The purpose of this paper is to provide methods to improve
the traditional MRPⅡ system according to the demand of the
time base competition. The manufacturing lead-time, the
purchase lead-time and the secure stock become dynamic
parameter in the MRPⅡ by integrating the 5 models into the
traditional MRPⅡ, and by integrating the Model of Resource
Balance in Material Requirement Plans into the MRPⅡ make
the ERP obtain the ability of time management in
manufacturing management.
REFERENCE
[1] M.N.Cawdery: “The lead time and inventory control problem” London
Operational Research Quarterly Vol. 27 No.4, ii.
[2] Sunil Chopra, Peter Meindl: “Supply chain management-strategy,
planning, and operation”. Pearson Education Press.
[3] David R. Anderson, Dennis J. Sweeney, Thomas A. Williams: “An
Introduction to Management Science (Quantitative approaches to decision
Making) 11th Edition”. Thomson Press.
[4] SimonHaykin: “Neural Networks (Second edition). ” Pearson Press.
[5] Kong Chen，Yang Ge: MRPⅡ/ERP principles and application（second
edition）. Ting hua university press, 2006.
[6] WanXia Mei，ShiHua Ma：The problem of lead-time reduce in the (Q,r)
stock. Chinese Management Science. 2007
 2009 International Conference on Computer Engineering and Technology
A New Model for Classifying Inputs and Outputs and Evaluating the DMUs
Efficiency in DEA Based on Cobb-Douglas Production Function

S. M. Hatefi, F. Jolai, H. Kor, and H. Iranmanesh

Department of Industrial Engineering, College of Engineering
University of Tehran
Tehran, Iran
s_m_hatefi@yahoo.com, fjolai@ut.ac.ir, hkor@ut.ac.ir, hiranmanesh@ut.ac.ir

Abstract—In conventional data envelopment analysis it is
assumed that the input versus output status of each chosen
performance measures is known. In some conditions finding a
statue of some variables from the point view of input or output
is very difficult; these variables treat as both an input and
output and are called flexible measures. This paper proposes a
new model based on Cobb- Douglas production function for
classifying inputs and outputs and evaluating the performance
of Decision making units by considering flexible measures. The
model is capable to provide marginal output elasticities and
marginal rates of output transformation. The presented model
is evaluated comparing with the recent model in the literature.
The result shows that the measure efficiencies of our model are
statistically closer to true efficiencies and have higher rank
correlation with true efficiencies. We illustrate the application
of proposed model with the real data found in literature. Also
results obtained from simulated and real data show that there
are high correlation between our model and the recent model.
Keywords-DEA; variable status; flexible measures;
performance evaluation
I. INTRODUCTION
Charnes et al. [1] were the first to describe the DEA
model (the CCR model), and used a mathematical
programming model to identify the efficiency frontier based
on the concept of the Pareto optimum when using multiple
measures. Following Charnes et al. , a number of different
DEA models have now appeared in the literature [2]. Recent
years have seen a great variety of applications of DEA.
Applications of DEA in hospitals, banks, maintenance crews
can be seen in Cooper studies [3].
There is a literature –albeit a relatively small one- which
is directed to quantitative estimate of RTS and scale
elastisities in DEA models. Banker et al. [4] developed a
variable returns-to-scale (RTS) variation; the multiplicative
model was developed by Charnes et al. [5] in which the data
are transformed using a logarithmic structure. Some authors
survey qualitative characterization in the form of identifying
whether RTS are increasing, decreasing or constant.
Examples are the treatment of scale elastisities in Førsund
[6] and Banker and Thrall [7]. However, Førsund [6]
showed that there are difficulties in using the standard DEA
models, to obtain scale elasticity estimates. But in
multiplicative DEA models, Banker et al. [8] show how
exact (numerically valued) elasticities of scale can be
obtained from multiple input-output variables. Ruggiero [9]
developed an alternative methodology based on CCA and
Cobb-Douglas production function to measure technical
efficiency. He provided marginal output elasticity and
marginal rates of output transformation by CCA approach.
Bardhan et al. [10] was jointed data envelopment analysis
and statistical regressions for Cobb-Douglas production
function estimation and efficiency evaluation in simulation
study.
In the conventional application of DEA, it is assumed
that the status of performance measures from point view of
input or output is specified. But in some condition there are
variables or measures that treat as both an input and output.
For example consider performance evaluation of bank
branch operations, such as discussed in Cook et al. [11,12].
In this study output variables are the standard counter
transactions such as deposits and withdrawals, and input
variables are resources such as various staff types. In
evaluating performance of each bank branch to attract
investments, a measure such as the number of high value
customers treats as both an input and output. From one view
this factor is considered as proxy for future investment,
hence can be classified as an output. Also it can be input
that aids the branch in generating its existing investment
portfolio. Such discussion can be made, as well, for
measures such as deposits. Examples of flexible measure in
evaluating performance of university, hospital and highway
maintenance crew can be seen in [13].
This paper presents a new model for classifying flexible
measures and evaluating DMUs based on Cobb-Douglas
production function. In this model flexible measure is
classified to input or output as efficiency of each DMU is
optimized. The model is capable to provide marginal output
elasticities and marginal rates of output transformation
while in standard DEA models there are problems to
obtaining marginal elasticity and marginal rate of
transformation.
The rest of the paper is organized as follows. In section
II the DEA model establishes based on Cobb-Douglas
production function. The model provides marginal output

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 390

DOI 10.1109/ICCET.2009.231
elasticities and marginal rates of output transformation using
model. In section III, three scenarios design based on . .
simulated data and variable status classifies by our model
and Cook and Zhu model. In section IV verification of
presented model surveys by statistical methods in 2 0
comparison with Cook and Zhu model. Section V illustrates
application of the model in the real practical data. 1,2, … ,
Conclusions are discussed in Section VI. 0,1 , , , 1 4

II. PROPOSED MODEL Note that since either 0 or 1 , then measure

We have proposed a Cobb-Douglas production function. will end up either as an output or an input. Thus, model (4)
In this production function, s output variables is produced for DMU0 permits that DMU to choose which is best for
by using of m inputs variables and relationship between each measure l, whether to designate it as an output or input.
inputs and outputs variables is as bellow: Model (4) is clearly nonlinear. But it can be linear by way of
the change of variables for each , imposing the
1 constraints:

That 1,2, … , is input variables and 0

1,2, … , is output variables and is an efficiency 1 5
index 0 1 . Where 1 if and only if production Where M is a large positive number. These linear integer
is technically efficient. Now we assume there are L factors restrictions capture the nonlinear expression ,
that these factors don’t specify the statues of variables from without actually having to directly specify it in the
the point of view input or output. These factors are called
optimization model. Note that if 1 then ,
flexible measures [14]. We extend Cobb-Douglas
production function considering flexible measures as and if 0 , then 0. By this way of the changing
bellow: variables model (4) converts to a mixed integer linear
program such as bellow:

2
In this formula 1,2, … , is the weight of L
factors. For any of L factors we define 0,1 as binary
variables. If 1 the factor of l is considered as an output 2
variable otherwise is considered as an input variable. By
taking of logarithm from two sides of Eq. (2) we have:
. .
2
2 0
3 1,2, … ,
0
1
In Eq. (3) the value of is negative. While λ treat
0,1 , , , 1 6
between (0, 1], Ln(λ) treat in (-∞,0].
Using Eq. (3) as objective function, a nonlinear
programming model is proposed for evaluating the
The value of DMUs efficiency is calculated as bellow:
performance of every DMUj (j=1, 2, …, n) as bellow:

2
2 7

391
 And in this equation the amount of , , , are the
TABLE I. DATA GENERATIONA
coefficients of the objective function for optimizing the
efficiency of DMU0. Variable Type Distribution
Following Banker et al. [8] and Ruggiero [9] if we place X1 Input Uniform on [40,60]
X2 Input Uniform on [45,75]
, , in model (6), we can calculate the marginal X3 Input Uniform on [50,75]
output elasticities and marginal rate of output transformation Y1 Output Normal: 25, 6
for evaluating DMU0 from Eq. (7) as follow: Y2 Output Normal: 20, 9
Marginal output elasticity of input i in the production of Y3 Output NA
output j for DMU0 is as follow: U Efficiency Normal: 0, 0.15
a
All random numbers were generated using SPLUS. The efficiency component
g was generated as the value of 1-|U|, with the restriction g=1 for |U| > 0.4 . As a
, 8 result, 31 observations (31%) were technically efficient. NA represents not
applicable.

If in DMU0 is zero the marginal elasticity of input l in

the production of output r will calculate as follow:
, 9
And the marginal rate of output transformation for
outputs r and h in DMU0 is:
, 10
If in DMU0 is 1, the marginal rate of output
transformation for outputs r and l is:
, 11
One of the advantage of proposed model related to Cook
and Zhu model is that marginal elasticity of input (i) in the
production of output (r), marginal elasticity of factor(l) in
production of output (r), marginal rate of output
transformation for outputs of r and h, and the marginal rate
of output transformation for outputs of r and l can be
calculated.
III.SIMULATED DATA
The simulated data is generated with S-PLUS software.
For this simulation, 31 DMUs (%31) were generated to be
technically efficient with g=1, and 69 of DMUs (%69) are
inefficient with g<1. Assumed distributions and description
of the variables are reported in table I. In this table have
assumed that the outputs of , are produced by
inputs of , by using of the following function:
. .
12
For considering the verification of proposed model from
the point of classifying the status of variable and evaluating
the performance of DMUs three scenarios are considered. In
the first scenario which is an input variable is considered
as a flexible measure. In the second scenario which is an
output variable, is considered a flexible measure. And in
third scenario x and y are considered as a flexible
measures. These three scenarios were run by proposed
model and Cook and Zhu model .
Obtained results show in the first scenario was
considered as a flexible measure. The results of proposed
model shows that this variable is considered as an input in
%89 of DMUs, but in Cook and Zhu model it is a input in
%92 of DMUs. In the second scenario was considered as
a flexible measure and by using proposed model it is
considered as an output in the %96 of DMUs, but in Cook
and Zhu model it is considered as an output in the %86 of
DMUs.
In the third scenario by using proposed model is
considered as an input variable in %91 of DMUs and y is
considered as an output variable in %92 of DMUs. But in
the Cook and Zhu model is considered as input in %89 of
DMUs and y in %78 of DMUs is considered as an output.
These results show that proposed model is classified
better flexible measure than Cook and Zhu model in the
second and third scenarios.
IV. MODEL VERIFICATION
In this section we use two methods for comparing the
efficiency of these two models (Proposed model and Cook
and Zhu model). First we compare the rank correlation of
measure efficiency, which was calculated by two models,
with true efficiency. Then we use the Wilcoxon signed
ranks test and Wilcoxon rank sum test for surveying the
measure efficiency from true efficiency. For more
information about these statistical methods see [14].
A. Rank correlation
Here, using Spearman and Kendal Tau correlations, we
calculate the correlation between measure efficiency and
true efficiency. The coefficient correlation between
calculated efficiency and true efficiencies in proposed
model and Cook and Zhu model are showed in table II. The
results show that in all of the scenarios measure efficiency
of proposed model have higher correlations with true
efficiencies rather than Cook and Zhu model, also there are
high correlation between measure efficiencies in proposed
and Cook and Zhu model. For example in the first scenario,
Spearman and Kendal Tau correlation coefficients between

392
 TABLE II. SPEARMAN'S RHO AND KENDALL'S TAU CORRELATION

Scenario no. Spearman's rho Kendall's tau

MEa model 6 ME C-Z ME model 6 ME C-Z
True Efficiency 0.638 0.348 0.551 0.266
1
ME model 6 0.780 0.632
True Efficiency 0.676 0.345 0.564 0.264
2
ME model 6 0.723 0.570
True Efficiency 0.587 0.328 0.498 0.252
3
ME model 6 0.715 0.566
a
ME presents measure efficiency
C-Z present Cook and Zhu model
All correlation are statistically significant at 0.001

Beasley’s analysis, this latter variable was treated as both an

input and output. The optimal d and the optimal value of
TABLE III. WILCOXON TESTS
measure efficiencies calculated in the proposed model and
Scen Test type Test statistic Decisiona P-value Cook and Zuh model.
ario Results showed that flexible variable in 20 cases of
1 signed-rank Z = -6.9352 Reject H0 0 DMUs is considered as an output variable and in 30 cases of
2 signed-rank Z = -6.9021 Reject H0 0
3 signed-rank Z = -6.9352 Reject H0 0 DMUs considered input variable in Cook and Zhu model.
1 rank sum Z = -2.2551 Reject H0 0.0121 These values for proposed model are 10 and 40 respectively,
2 rank sum Z = -2.2778 Reject H0 0.0114 so this variable is considered as an input in many DMUs.
3 rank sum Z = -2.2057 Reject H0 0.0137 The spearman and Kendal Tue correlation coefficients
a
H0 : Absolute difference between measure and true efficiency in proposed
model ≥Absolute difference between measure and true efficiency in Cook and between measure efficiencies in proposed model and Cook
Zhu model and Zhu model are 0.758 and 0.635 respectively that shows
H1 : Absolute difference between measure and true efficiency in proposed
model ≤ Absolute difference between measure and true efficiency in Cook and
there is a high correlation between them. This correlation is
Zhu model signified statistically at α =0.0001.
VI. CONCLUSION
measure efficiency and true efficiency for proposed model In conventional data envelopment analysis we assume
are 0.638 and 0.551 respectively. But these values for Cook that measure status from point view of input or output is
and Zhu model are 0.348 and 0.226 respectively. known. But in real word there are some performance
measures that treat as both an input and output and
B. Wilcoxon signed ranks and Wilcoxon rank sum test specifying of their status is difficult. In this paper we
Now, we calculate absolute difference between measure established new model based on Cobb-Douglas production
efficiency and true efficiency for every DMUj of the function for classifying variable status from point view of
proposed model, D , and the Cook and Zhu model D j input or output and evaluating the performance of decision
1,2, … ,100 . For comparing error of two models we making units. This model is capable to provide marginal
consider H : µD µD and H : µD µD . H shows that output elasticities and marginal rates of output
error of proposed model is greater than Cook and Zhu transformation while in standard DEA models there are
model and H shows that error of proposed model is lower problems to obtaining marginal elasticity and marginal rate
than Cook and Zhu model. We used Wilcoxon signed ranks of transformation. Verification of our model evaluated with
test and Wilcoxon rank sum test for evaluation these Simulated data in comparison with Cook and Zhu model.
hypothesis. Founding results are showed in table III. Results obtained from simulated data show that: 1) Measure
These results reject H0. In the other words the value of efficiencies calculated from proposed model are statistically
measure efficiency in proposed model is closer to true and significantly closer to true efficiencies. 2) Measure
efficiencies statistically. efficiencies obtained from our model have higher
correlation (Spearman and Kendal Tau rank Correlation)
V. APPLICATION OF MODEL IN REAL PRACTICAL DATA with true efficiencies than Cook and Zhu model. 3)
In this section we exerted proposed model on two series Proposed model is classified better flexible measure than
of real data and have compared the results with Cook and Cook and Zhu model in second and third scenarios and in
Zhu model. First series of data related to Beasley 1990 that first scenario results are almost similar. Also there are high
used in [13]. In this data there are two inputs, General rank correlation between efficiencies obtained from
Expenditure and Equipment Expenditure and three outputs, proposed model and Cook and Zhu model .
consisting of three types of students. The ‘‘flexible Application of presented model illustrated in two real
measure’’ here is the Research Income. We note that in practical data. Also results obtained from real data showed

393
 that there are high correlation between presented model
and Cook and Zhu model.
REFERENCES
[1] Charnes, A., Cooper, W.W. and Rhodes, E., 1978,
Measuring the efficiency of decision-making units,
European Journal of Operational Research,2(6), 429–
444.
[2] Cooper, W.W., Seiford, L.M., and Tone, K., 2000, Data
Envelopment Analysis: A Comprehensive Text with
Models, Applications, References and DEA-Solver
Software, Kluwer Academic Publishers, Boston.
[3] Cooper, W.W., Seiford, L.M. and Zhu, J., 2004.
Handbook of Data Envelopment Analysis. Kluwer
Academic Publishers., Norwell, MA
[4] Banker, R.D., Charnes, A., Cooper, W.W., 1984. Some
models for estimating technical and scale in
efficiencies in data envelopment analysis. Management
Science, 30 (9), 1078–1092.
[5] Charnes, A., Cooper, W.W., Seiford, L. and Stutz, J.,
1982. A multiplicative model for efficiency analysis.,
Socio-Economic Planning Sciences, 6, 223–224.
[6] Førsund, F.R., 1996. On the calculation of scale
elasticities in DEA models. Journal of Productivity
Analysis, 7, 283–302
[7] Banker, R.D. and Thrall, R.M., 1992, Estimation of
returns to scale using Data Envelopment Analysis.
European Journal of Operational Research, 62, 74–84.
[8] Banker, R.D., Cooper, W.W., Seiford, L. M., Thrall, R.
M. and Zhu , J., 2004, Returns to scale in DEA
Models., European Journal of Operational Research,
154, 345–362
[9] Ruggiero, J. , 1998, A new approach for technical
efficiency estimation in multiple output production,
European Journal of Operational Research, 111, 369–
380
[10] Bardhan, I. R, Cooper, W. W. and Kumbhakar, S. c.,
1998, A Simulation Study of Joint Uses of Data
Envelopment Analysis and Statistical Regressions for
Production Function Estimation and Efficiency
Evaluation, Journal of Productivity Analysis, 9, 249–
278
[11] Cook, W.D., Hababou, M., Tuenter, H., 2000.
Multicomponent efficiency measurement and shared
inputs in data envelopment analysis: An application to
sales and service performance in bank branches.
Journal of Productivity Analysis 14(3), 209–224.
[12] Cook, W.D., Hababou, M., 2001. Sales performance
measurement in bank branches. Omega 29, 299–307.
[13] Cook, W. D., Zhu, J., 2007, Classifying inputs and
outputs in data envelopment analysis, European
Journal of Operational Research, 180, 692–699
[14] Sheskin, D. , 2000, Handbook of parametric and
nonparametric statistical procedures, Florida, CRC
Press.

394
 2009 International Conference on Computer Engineering and Technology
The Analysis and Improvement of the Price Forecast Model Based on Fractal
Theory
CUI Herui YANG Li
Department of Economics and Management, North China Electric Power Universiy, Baoding,
071003, China
cuiherui1967@126.com
Abstract
This paper discusses the basic principle of the
fractal theory and the basic thinking of how to use it to
forecast the electricity price, then build electricity
price forecasting model and make a analysis on the
prediction error to put forward improvement. Before
the application of fractal mode in the price forecast
field, there is a need to do R/S analysis of the data
series to derive its index Hurst; And in forecast
process, use variable dimension fractal parameters
calculated by curve fitting technique instead of
constant dimension fractal parameters.
1. Introduction
Fractal geometry is already become a new important
discipline after developing for more than 20 years
since Mandelbrot brings forward fundamental theory
of fractal geometry in 1970's, which has been widely
used in a lot of fields such as physics, chemistry,
biology, material science, computer science,
astronomy, meteorological phenomena, economy,
management science and art.
Taking non-regular and nonlinear thing as research
object, studying on the regularity, hierarchy and scale
invariance hidden in complex natural and social
phenomenon, crossing the natural science, social
science and thinking science, fractal theory is a new
method to study on the complex object. Based on the
self-similarity between part and whole, fractal theory
establishes the global property from the part, along the
direction from microscopic to macroscopic. This
theory takes the relationship between regional and
global as research object, emphasizes the global's
dependence to part, and searches for the bridge through
it can deduce and transient from part to whole. As the
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.75
whole is made up by the part, people can understand
the whole through part. One of the fractal theory
philosophical foundations is that the unit and element
of the system contains the property and information of
the whole system. [1][2][3]
Now many economists utilize the fractal theory to
explain the complicated and diversified economic
phenomena, for example, the fractal dimension
significance of the economical elastic, the fractal
dimension calculation of the change of price, the
fractals and fractal dimensions of national income, the
negative power distribution of capital and the property,
the R/S analysis of the economic system’s change
tendency, the fractal dimension measurement of the
economic chaos, all these new achievements have
infused new vigor for the economic. In domestic, many
scholars used the fractal model to forecast the price of
stock, electric power, grain and other fashionable
commodity. [4][5][6][7] However the established
models are not all suitable for the price forecasting, in
this article we discuss one commonly used fractal
model, point out its flaw and the possible
improvement.
2. Analysis on the prediction error of
fractal model
2.1. Description of the commonly used fractal
model
The price forecasting models based on fractal
theory,
include the models used fractal theory only and the
models used the combination of fractal theory and
other methods such as wavelet analysis, vector
machine and so on. These models have a common
basement as follow.
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2.1.1. Related formula of fractal distribution. Power
exponent distribution can be used to describe and
express fractal distribution.
C ⑴
N= D
r
In the formula:
r—Characteristic Linearity, such as time, length, etc;
N—related object's number, such as price, index,
etc;
C—undetermined constants;
D—fractal dimension.
Generally, the numbering of time is taken as “r”,
r1=1,and rn=n, and electricity price value as “N”, N1 is
equal to electricity price value when r1=1, …and Nn is
equal to electricity price value when rn=n.
At present, in the general application of fractal
method, D is constant, and so this type of fractal
method is called constant dimension fractal,
manifested as a line in double logarithm coordinate
system. Any two data points (Ni,ri)and(Nj,rj) on the
straight line determine the fractal parameters, namely
fractal dimension D and constant C of this line.
⎡N ⎤
ln ⎢ i ⎥
⎢ N j ⎥ ln N i − ln N j ⑵
D= ⎣ ⎦ =
⎡r ⎤ ln rj − ln ri
ln ⎢ i ⎥
⎣⎢ rj ⎦⎥
C = NirD ⑶
As negative number can not take logarithm , there is
a need to add a constant to all the N value to eliminate
the negative number, in another word, it must make a
translation for all the N value firstly, when some of Ni
are negative numbers. Sometimes, such translation is
made firstly for a better analysis and predication.
This paper uses variable dimension fractal model to
get over the weakness of the above constant dimension
fractal method, that it is unable to deal with the non-
linear functional relationship in double logarithm
coordinate system.
Now it will show that any functional relationship
N=f(R), between N and R, can be transformed into
fractal distribution N = C , make:
rD
C ⑷
f ( R) =
rD
So,
r = [C f ( R)]
1D
⑸ relationship, and the fractal parameters of every line
That is, through the transformation above, R gets the
form of fractal distribution.
2.1.2. Transformation Although any functional
relationship can be transformed into the form of fractal
distribution, in fact, in the practical application, what
have known are just some data points, not the
396
functional relationship, so can not get the concrete
form. At this time, it must to make a series of
transformations called cumulative sum, and choose a
transformation, through which the data can be in fairly
good agreement with fractal distribution model, so can
be treated with fractal distribution. Transformation
method as following:
First, drawing the initial data point,(Ni,ri) (i
=1,2,…,n), on the double logarithm coordinates, in
general case, these data points cannot be in fairly good
agreement with fractal distribution model, so ,
arranging these data into a basic series.
{Ni} = {N1,N2,N3…} (i =1,2,…,n) ⑹
Based on the above basic series, other series can be
constructed. Such as to construct the first order
cumulative sum series S1, just need to make S11=
N1,S12= N1+ N2,S13= N1+N2+N3,and so on. The Second
Order, third order can be made as the same, that is:
{S1i}= {N1, N1+N2, N1+N2+N3…} (i =1,2,…,n) ⑺
{S2i} = {S11,S11+S12,S11+ S12+ S13…}(i =1,2,…,n) ⑻
Second, establish the fractal model of the cumulative
sum series. Taking the first order cumulative sum
series S1 as example, draw the data (S1i,ri)(i
=1,2,…,n)on the double logarithm coordinates to get a
discrete fractal distribution model. According to the n
data points, what obtained is a segment constant
dimension fractal model made up of n-1 line, and
calculating the fractal parameters of every line
according to formula and .
Third, choose the transformation with best result,
and determine the fractal parameters correspondingly.
Drawing the data points of each order cumulative sum
series on the double logarithm coordinates and
comparing with a fractal distribution model, then can
choose a transformation with best result, and determine
the fractal parameters correspondingly.
Finally, determine the fractal distribution model,
forecast using extrapolation calculation.
2.2. Analysis on the prediction error
In rectangular coordinate system, if take lnS series’
logarithm values as longitudinal axis, and take lnr
series’ logarithm values as lateral axis, in essence, the
above model’ parameters D, is the slope between lnS
and lnr. There is a implied assumption of the model
that lnS and lnr shows a nearly perfect linear
segment have no difference. However, this hypothesis
is in fact untenable. Although lnS and lnr shows linear
relationship grossly, but still can not meet the stringent
assumptions. The difference between the fractal
parameters(as can be seen from the table 1,D and C
series) is small, but there is no any pair of
parameters(D and C) can effective on behalf of the enhance persistent sequence, and have the property of
whole’s fractal parameters, and this is the root causes fractal.
of the forecast model’s error.
In addition, there are also problems of its own 3.2. Prediction of the fractal parameters
algorithm, and the most critical one is that, according
to the chosen D and C, using N = C to forecast, D According to the analysis of the causes of
rD deviations, correspondingly, improvement of the
and C’ deviation will be enlarged in several times, model from the two aspects can be made: one, to raise
duing to the existence of power and the multiplication. the accuracy of fractal parameters and the other is to
The above two reasons make the existence of large change the algorithm of N = C . This paper will use
deviations of the forecast result. In Table 1, it can rD
clearly see. the first method to make improvement of the above
mentioned model.
3. Improvement No matter D and C are calculated by what
method(calculated by least square method or of just a
certain line segment’s), when using the above process
3.1. R/S analysis of projections, it always takes D and C unchanged, In
fact, it can find that D value presents monotonically
Before the construction of the fractal model to
increasing tendency, while C value presents monotone
predict price, the first must to do is to judge whether
decreasing tendency with the increase of r, as showed
the data has the fractal property and can use the fractal
in table 1.Therefore, the thinking is, first, establish the
model. This paper uses the R/S analysis, and it’s
functional relationship between r values and the fractal
theoretical calculations are as follows.
parameters to predict the D and C values of each phase,
Take {Xi}(i =1,2,…,n)as a time series, the average
_
so as to make it more approximate to real values and
of it is X ,and the standard deviation is S. improve its accuracy, and accord to N = C to forecast
_
1 n rD
X = ∑Xi
n i =1
(9)
prices, rather than the directly use of fixed fractal
2 parameters.
1 n⎛ _
⎞
S= ∑ ⎜ Xi − X ⎟
n i =1 ⎝ ⎠
(10)
3.3. Empirical analysis
X(t,n) is cumulative deviation,
Take one commodity as an example.
t
⎛ _
⎞
X (t , n ) = ∑ ⎜ X i − X ⎟ (11) First, choose the price of the commodity in 100
i =1 ⎝ ⎠ trading day to construct the test sequence. Through the
Definite the difference between the maximum and R/S analysis get the Hurst index which is 0.63.As 0.5<
the minimum obtained by the formula as R, Hurst index≤1，it implies that today's events have a
R = max X (t , n) − min X (t , n) (12) true impact on tomorrow, the data shows the fractal
1≤ t ≤ n 1≤ t ≤ n characteristics and can use fractal model.
And the rescaled range analysis is Then, select the second-order cumulative series to
(R/ S)n = (c × n)H (13)
construct the forecasting model. In the table 1, fractal
parameters are calculated by the formula and .
Calculate the logarithm of the above equality, Form the table 1, what can found is:
Log(R / S )n = Log(a ) + H × Log(n) (14) Every line segment’s fractal parameters including D
The slope between Log(R/S) and Log(n) obtained and C, has remarkable difference;
through the linear regression is Hurst index. The D value presents monotonically increasing tendency,
introduction of Hurst index is to determine the random while C value presents monotone decreasing tendency
distribution property and the fractal property of the with the increase of r.
data sequence. This paper calculates the D and C value through
When H=0.5, time series show the property of curve fitting analysis according to the
random and independent, and that is a sequence of TABLECURCER and SCILAB.
random walk. When 0 ≤ H <0.5, the time series is anti- Curve equation of r and D:
persistence sequence, also known as "mean eversion". y=(a+cx0.5+ex+gx1.5)/(1+bx0.5+dx+fx1.5+hx2) ⒂
When 0.5<H ≤1, the time series show the trend of an

397
 5. Conclusions
Table 2. Parameters of r and D curve equation
In the area of price forecast, fractal theory will be
a b c
broad prospects, but as the analysis in this paper, the
-0.03614436 -2.69000731 0.210340011 commonly used original prediction model’s precision
d e f is very poor, and in such circumstances, this paper put
forward and experienced an effective method to
2.732972396 -0.26390564 -1.25104128
improve the general model.
g h Firstly, using the R/S analysis method, this paper
0.095690633 0.219910658 analyzes the fractal characteristic of electricity price to
Curve equation of r and C： determine whether the price data has the fractal feature
or not. Only when 0.5<Hurst index≤1, it is able to use
y=(a+clnx+e(lnx)2+g(lnx)3+i(lnx)4)/(1+blnx+d(lnx) fractal model.
2+f(lnx)3+h(lnx)4) ⒃ Then, different from the previous model, this article
does not use the constant dimension fractal model, but
instead with the variable dimension fractal model. This
Table 3. Parameters of r and C curve equation paper tries to calculate the D and C value through
a b c curve fitting analysis according to the
TABLECURCER and SCILAB to improve forecast
128.7510436 -8.03692232 -884.548896
accuracy. Facts have proved that, compared to the
d e f original model, the improved forecasting model clearly
28.23779273 3906.908953 -5.20079553 has a better forecast accuracy.
g h i
-5596.01211 -48.6888513 2694.019303
References
j [1]Yin Yuliang, Li Peizheng,Kang Yuyun,Gu Zonglei, The
45.37515406 Development and Research Status of Fractal Theory ,Science
By substitution of r=11~15 into the above two curve Information,No.15,2007(In Chinese)
equations ⒂ and ⒃ , can obtain the D and C value, [2]Li Peng, The Development and Application of Fractal
and S and N value calculated by equation (1) Theory, Collected Works Science and Education, No.1,
correspondingly. 2008(In Chinese)

4. Comparisons [3] Liu Ying ， Hu Min, She Guiying, Li Xiaobing, Liu

Using the improved fractal model and the general
fractal model to do predict commodity prices as r = 11
~ 15.
4.1. The general fractal model
D ＝ － 1.53643797 ， C ＝ 40.12727517 ， and
according to the formula ①, S ＝ C/rD ＝
40.12727517×r1.53643797, and the prediction result as
table 4.
4.2. The improved fractal model
Xiaolin, Fractal Theory and Its Application, Jiangxi Science,
Vol.2415, No.24,2006(In Chinese)
[4]Lu Fangyuan, Fractal Theory and Its Application in
Economic Management, Journal of Henan College of
Financial Management Cadres, No. 3, 2004(In Chinese)
[5]Liu Xiaocheng,Wang Yanming, Oil Price Forecast Based
on Fractal Theory, Journal of Xi'an Shiyou University,
Vol.15, No.4,2006(In Chinese)
[6]Zhang Ying,Lu Ying, Yao Yan, The Price Forecast of
Fashion Commodity Based on Fractal Theory, Journal of
Shengyang University of Technology(Social Science
Edition), Vol.1,No.2, 2008(In Chinese)

The predicted results show in table 5.
Through the comparison between prediction value
and real value of the two models, it can find that the
improved fractal forecasting methods is clearly better
than the original prediction model.
[7]Liu Haibo,Yi Dongyun, The Research of Stock Price
Forecast Basing on Wavelet Analysis and Fractal Theory,
Statistics and Advisory, No.5,2006(In Chinese)
[8]Mao Fengmei, The Theoretical Calculations of Fractal
Dimension in the Financial System Fluctuations Based on

398
Fractal theory,Mathematics in Practice and Theory, Vol.36, Foreign Economic Relations & Trade, No.7,2005(In
No.12,2006(In Chinese) Chinese)

[9]Huang Zhao, Initial Analysis on Shanghai Stock Market
Based on Fractal theory, Accounting Research, No.12.2007
(In Chinese)
[10]Chen zhao,Wang Guocheng, The Research and
Application of Chaos Theory and Fractal Dimension, HLJ
[11]Meng Li,Wang Yue,Zhao Jing, The Foreign Exchange
Market Forecast Based on Fractal Theory,Journal of
Shengyang University of Technology(Social Science
Edition), Vol.1,No.2, 2008(In Chinese)

Table 1. The fractal parameters

Every line segment’ s fractal fractal parameters with least square
r N s2 parameters method
D C D C
1 21 191 / /
2 22 234 -0.292935892 191
3 20 297 -0.587993933 155.6725525
4 23 383 -0.883971838 112.4591241
5 25 494 -1.140523786 78.80159956
6 20 625 -1.29011696 61.94038379
7 21 777 -1.412181256 49.77249451
8 22 951 -1.513304911 40.88179959 -1.53643797 40.12727517
9 24 1149 -1.605776369 33.73028082
10 23 1370 -1.669683751 29.31141716

Table 4. Prediction result using the general fractal model

r 11 12 13 14 15
S2 1604.4385 1834.2136 2074.5381 2325.0287 2585.3427
Prediction N 13.4385 -4.6634 10.5494 10.1661 9.8234
Real N 24 22 25 21 22

Table 5. Prediction result using the improved fractal model

r 11 12 13 14 15
D 1.7243768 1.7650208 1.7952154 1.8169338 1.8318153
C 25.819216 23.451076 21.782076 20.596113 19.745076
S2 1613.2305 1883.3894 2177.6041 2490.1415 2817.3498
Prediction N 22.2305 23.3894 28.6041 24.1415 17.3498
Real N 24 22 25 21 22

399
 2009 International Conference on Computer Engineering and Technology
A Flash-Based Mobile Learning System
for Learning English as a Second Language
Firouz B. Anaraki
College of Internet Distance Education, Assumption University
Bangkok, Thailand
firouz@cide.au.edu
Abstract— This paper explains the development of a Flash-Based
mobile learning system for learning English as a second language
(FML4ESL). Adobe Flash CS3 was used to develop this system
which works on smart mobile phones and PDAs that support
Adobe Flash Lite. Twelve English lessons were developed for this
system. University students at Assumption University tried out
this system using their mobile devices for a period of 4 weeks. A
pretest, posttest, and surveys were used to evaluate the
effectiveness and usability of this system.
Keywords— Mobile English Learning
I. INTRODUCTION
Learning and the lifelong pursue of knowledge has become
one of the most essential activities in the current knowledge-
based economy, which is characterized by information age,
globalization, knowledge acquisition and transfer, and the
information and communication technology revolution.
Globalization requires new methods of knowledge acquisition
and to convey new skills and tools [1].
The traditional learning context is experiencing radical
changes and challenges. There are often occasions when
people want to learn exactly what they are interested in or
what they need to learn, without time and location boundaries.
Learners are demanding for better and improved access and
convenience at the lower costs and with the direct application
of contents to their working environment and needs.
Mobile technology is introducing new approaches to
learning due to the features that are defined by a number of
characteristics. They support learning activities that need just-
in-time (JiT) and true learning on demand.
II. LITERATURE REVIEW
One of the first projects where mobile phones were used in
language learning was developed by the Stanford Learning
Lab in 2000 [2]. They basically developed Spanish language
learning using both voice that could be played on mobile
phones and emails to support the study material. They
provided voice for vocabulary practice and pronunciation,
word and phrase translation, and quizzes to their students.
Their results indicated that voice vocabulary lessons and
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DOI 10.1109/ICCET.2009.183
quizzes had great potential if provided in small chunks
suitable for the small screen sizes of mobile devices.
Another project at a Japanese university utilized SMS to
deliver English vocabulary and their meanings to their
students [3]. They sent short lessons on separate and discrete
chunks to their students’ mobile devices 3 times a day. Each
lesson introduced a couple of new words daily and the new
words were recycled in subsequent lessons. Students were
tested biweekly and compared to groups that received
identical lessons by web and on paper. The results indicated
that the SMS students learned over twice the number of
vocabulary words as the web students, and that SMS students
improved their scores by almost twice as much as students
who had received their lessons on paper.
One more project in Japan utilized mobile phones
that played short audio clips in English to help with
pronunciation for English as a Second Language (ESL)
students [4]. The results indicated that that the learning
content should be in short learning units or sessions. For
instance, a unit on language vocabulary would best fit the
capabilities of mobile devices for learning in a brief period,
such as around 5 minutes. Another conclusion that was
considered very useful was to provide customization of
learning material for individual or group needs and learning
experience.
III. FLASH-BASED MOBILE LEARNING SYSTEM
The purpose of this research was to design and develop a
flash-based mobile learning system for learning English as a
second language using mobile devices capable of playing
Adobe Flash Lite. Most researchers in language learning have
mainly focused on a single medium, i.e. SMS (text) or MP3
(voice), to deliver the content to learners. This researcher
wanted to develop a multimedia mode of delivery of the
content to learners. This researcher with the assistance from
couple of professional ESL lecturers developed 12 mobile
English lessons to be used by a sample population of students
owning a Flash-based mobile phone or a PDA at Assumption
University.
400
 The objectives of the Flash–based English lessons for
mobile devices were as follows:
• Flash programming allows full control of the mobile
device and the lessons by the user.
• Students can listen to native speakers reading a passage
or a lesson.
• While listening, students can see and read the text on
the screen of their mobile device.
• Students can also learn spelling, grammar, and rules for
conversation.
• There are exercises in each unit (lesson) asking
students to practice what they have learned.

IV. DEVELOPMENT OF FML4ESL Fig. 1 Optimization of text under Adobe Flash

Very similar to the Learning Management System (LMS)
for the Internet based eLearning systems, the Mobile Learning
Systems (FML4ESL) needs to contain 2 main components,
namely the content (learning material) and the interface with
the learner. Currently this system has no management features
to be called Mobile Learning Management System. This
prototype system, however, could be considered as an
embryonic model for a fuller version with management of
user activities. In this section the two components of
FML4ESL are presented and discussed.
The content material should be optimized for the
small screens of mobile devices, usually between 2 to 3.5
inches. The contents should be grouped into small chunks of
data. Each chunk then could be displayed on the screen of the
mobile device. The content could be multimedia material; for
this research the researcher is making use of text and pictures
to display the contents as learning material and the voice of
native speaker for listening purpose. As the developed Flash-
Based Mobile Learning System is intended to be used for
learning English as a second language for the trial purpose, it
would be useful to show the pictures of English nouns or
actions for better understanding of the meaning of English
words by showing their pictures.
The sound associated with the text is in MP3 format with a
bit rate of 48 kilo bits per second (kbps) and the best quality
available under Adobe Flash. The higher the bit rate the larger
the file size is; the quality does not affect the file size but it
affects the loading time of the content into the mobile device.
The small delay (1 to 2 seconds) is acceptable for these
lessons as it provides a couple of seconds to the learner to read
and meditate upon the text.
The various combinations of bit rates and quality were
tested among 11 participants to select the best combination of
sound quality. The only limitation in this selection was the file
size, the researcher wanted to keep the file size for a lesson to
be around 1 megabyte (MB) in order that the older mobile
phones could play them the same as newer ones. Figure 2
shows the selection of sound quality for every slide.

The resolution of most modern mobile devices is

around 240x320 pixels. This researcher optimized the texts of
all the English lessons for the resolution of 240x320 pixels.
Mobile devices with lower resolutions display the text with
less clarity or sharpness, though still legible. The contrast and
brightness could be controlled by the user through mobile
device features. Figure 1displays how the content is optimized
for a screen resolution of 240x320 pixels on Flash CS3
development platform.

Fig. 2 Optimization of sound under Adobe Flash

The Flash-Based Mobile Learning System has a component

that controls the functions of a mobile device and the

401
interfaces with the learner. To provide the best functionality
and user interface to learners, the researcher consulted with
ESL lecturer who provided the contents and also a couple of
technology experts to find out the essential functions that
needed to be provided. Below are some recommendations
from experts.

The learner should be able to select a lesson, be able to play

a lesson slide by slide, and at the end of each slide, the learner
should have a choice to repeat the same slide or be able to go
forward to the next slide. At the end of a lesson (the last slide),
the learner can decide to repeat the same lesson from the
beginning all over again or simply quit the lesson and return
back to the normal mode of operation of the mobile device.
Based on the above recommendations, this researcher
decided to divide a slide into 4 distinct layers as shown in
figure 3.
Fig.5 Action script for the control layer for the last slide of a lesson
The prototype system was developed using Adobe Flash
CS3 package under Windows Vista operating System. The
developed Flash Mobile Learning System for Learning
English as a Second Language (FML4ESL) was in SWF file
format which can be played on any computer, including PCs,
Macintosh, Linux systems or any Flash enabled mobile
devices. By limiting each lesson to 8-10 slides, the SWF are
around 1 MB which could be played on most mobile devices.
Figure 6 displays a slide of a lesson on a mobile device.

Fig.3 The 4 layers of a slide in a unit (lesson)

As shown in figure 3, each slide of a lesson comprises 4

layers. The first layer, Layer p, holds the text and the picture
of the content on any slide. The second layer, the s layer,
contains the MP3 audio sound of the lesson which starts about
2 seconds after the text is displayed on the screen. The third
layer, the layer e, is an action script which is executed at the
end of each slide. This action script basically stops the slide
from going further. The last layer, the layer c, is the control
layer that controls the functionality of the knob on mobile
device. The action script for this control layer is shown on
figure 4. Fig.6 A slide of a lesson on a mobile phone

V. TRYING OUT THE FML4ESL

In order to conduct this research the author designed an
LMS website based on open source Moodle on one of the
servers at the university. Information about this research and
mobile English learning was provided on this LMS. Students
could apply to join this research online. Pretest, pre-survey,
posttest, post-survey were placed on this server.

Fig.4 Action script for the control layer for controlling the knob
This action script controls the left and the right movement
of the knob on mobile devices. The left click causes the same
slide to be replayed and the right click causes the next slide to
be continued. All slides follow the same logic except the last
slide in the lesson. The action script for the last slide is shown
in figure 5.
To attract students to participate in this research, a banner
was placed on the home page of the university website with a
link to the Moodle site, and posters were placed at various
schools at the university inviting students to join and to
participate in this research. The requirement for participating
in this research was to accomplish all the components of the
research which included:
• Answering to a pre-survey

402
 • Taking an online quiz (pretest) developed by the
content expert which had 20 questions and a time limit
of 20 minutes. Each question scored 1 point
• Downloading 3 lessons per week as provided and
listening to these lessons as often as possible during the
week for a total period of 4 weeks.
• Taking the same online quiz (posttest).
• Answering to a different survey (post-survey)

To encourage students to join and complete all the

requirements of this research, 10 gift vouchers for a values of
approximately US$ 20 each were provided and later were
given to 10 of the participants at random. A total of 182 filled
the online application to join the research and most of them
answered the pre-survey and/or took the pretest. This
researcher was in touch with the participants through the
Moodle forum and provided necessary information and
guidelines to them. Some students came to the office of the
researcher to solve their technical problems, i.e. downloading
or synchronizing problems of transferring the Flash lessons
(swf) to their mobile devices. Some students needed help to
install the Flash Lite to their mobile devices. Most questions
were answered through the forum of the Moodle.
Fig.7 Ranking of factors for mobile English learning
As shown in figure 7, learning at any place, lecturer’s
pronunciation, and learning at any time are the factors that are
most important for mobile English learning. Ease of use and
learning at any pace are also two other factors.
Another question in the pre-survey asked the participants to
rank (1 to 10) their English proficiency skills. Figure 8 shows
the responses of the 76 participants who also finished the
posttest and post-survey.

At the end of the 4 weeks, participants were requested to

take the posttest and answer to the post-survey. Finally 76 of
participants completed all the components of the research.
This sample population represented the population of
university students. They were from various schools in the
university from freshmen to seniors with various degrees of
English proficiency. Table 1 shows the descriptive data about
the participants.
Fig.8 Ranking of English proficiency skills before trying out
TABLE I
DESCRIPTIVE DATA ABOUT PARTICIPANTS After the 4-week of trying out of the lessons on their mobile
devices, participants were asked the same question in the post-
N Min. Max. Mean Std. Dev. survey. Figure 9 shows the response of the same 76 who
Pretest 76 4.00 17.00 10.74 3.18 finished all the components of the research.
Pretime 76 08:00 19:45 12:31 02:24
Posttest 76 10.00 20.00 16.50 2.98
Posttime 76 06:17 17:01 11:01 02:03

The one-way repeated-measure Analysis of Variances

(ANOVA) was used to test the data collected and to evaluate
the effectiveness of the Flash-based mobile English learning.
The standard univariate ANOVA indicates a significant time
effect, F(2,75) = 311.44, p < .01. This indicates a significant
improvement among all participants in this research.

Besides pretest and postest, there were a pre-survey and a

post-survey to evaluate the students’ attitude towards mobile
English learning. One of the question in the pre-survey asked
participants to rank (1 to 10) the importance of the 8 factors in
learning English using mobile devices. Figure 7 displays the
rankings.
Fig.9 Ranking of English proficiency skills after trying out
When figure 9 is compared with figure 8, participants felt
improvements in all 6 levels of the English proficiency skills.

403
 In the post-survey they were asked to rank (1-10) the impact
of Flash-based mobile learning on their English proficiency
skills. Figure 10 displays their responses.
Fig.10 The impact of mobile English learning on participants
As shown in figure 10, participants felt the most impact of
mobile English learning has been on their pronunciation
followed by listening skills and conversation. There have been
impacts in all levels of their English proficiency of
participants.
One important feature of the Flash-based mobile learning
system is its multimedia capability, i.e. users can read the text
on their mobile devices screens while listening to the same
text spoken by a native speaker. This researcher wanted to
explore if this feature was useful to participants. One of the
questions in the post-survey asked participants if this feature
of being able to read the text while listening was helpful to
them. Figure 11 displays their replies.
Fig.11 Multimedia feature helpful?
In the post-survey, there were 2 open ended questions to
seek the opinions of participants as they wished to express it.
One question asked participants to express their “likes” and
another question about their “dislikes” of using the Flash-
based mobile learning to learn English. Their likes included
“convenience”, “effective”, “just-in-time learning”, “easy to
404
learn”, “technologically advanced”, and “good for listening
and pronunciation improvements”. Their dislikes were “not
having enough lessons”, “the Flash-based mobile learning was
not available for all models of mobile devices”, and “not
being able to record their voice on the mobile device to
answer the questions.”
VI. CONCLUSIONS
The major findings of these research as follows:
1) Majority of students at Assumption University have a
positive attitude towards mobile learning and are
enthusiastic to learn English using their mobile
devices.
2) A Flash-Based mobile learning system is best
optimized with an audio bit rate of 48 Kbps and the
quality set to ‘best’.
3) The same Flash-Based mobile learning system could
be also utilized for learning other languages.
4) Smart mobile phones and PDAs could have a
positive effect in learning English as a second
language at all levels of English learning.
5) Multimedia features of mobile English learning are
most effective in English pronunciation, listening and
reading comprehension.
6) Convenience and just-in-time learning features of
mLearning make it ideal for learning foreign
languages.
7) The multimedia features of mLearning make
language learning more effective.
8) Students are willing to pay for mobile lessons for
their mobile lessons.
9) Mobile English learning is at its best in an hybrid
mode when supplemented with traditional English
learning.
REFERENCES
[1] Adam, N., Awerbuch, B., Slonim, J., Wegner, P., and Yesha, Y. (1997).
“Globalizing business, education, culture through the Internet”.
Communications of the ACM, 40(2), 115-121.
[2] Brown, E. (Ed.) (2001, January 8). “Mobile learning explorations at the
Stanford Learning Lab”. Speaking of Computers, 55. Stanford, CA:
Board of Trustees of the Leland Stanford Junior University.
http://sll.stanford.edu/projects/tomprof/newtomprof/postings/289.html
[3] Thornton, P., & Houser, C. (2005). Using mobile phones in English
Education in Japan. Journal of Computer Assisted Learning, 21, 217-
228.
[4] McNicol, T. (2004). ‘‘Language e-learning on the move”.
http://ojr.org/japan/wireless/1080854640.php [An online article,
Retrieved on December 7, 2007]
 2009 International Conference on Computer Engineering and Technology

Recognition of Trade Barrier Based on General RBF Neural Network

Yu Zhao Miaomiao Yang Chunjie Qi

College of Economics & Management,
Huazhong Agricultural University,
HZAU
Wuhan, Hubei, 430070, P.R. China,
E-mail:math052360@gmail.com.
College of Economics & Management,
Huazhong Agricultural University,
HZAU
Wuhan, Hubei, 430070, P.R. China,
E-mail:ymm19850815@163.com.
College of Economics & Management,
Huazhong Agricultural University,
HZAU
Wuhan, Hubei, 430070, P.R. China,
E-mail: gemz10186@126.com.

Abstract—Trade barriers correct externalities, but distort
trade as well, which brings negative effect on international
distribution of resources. Study on how to recognize there are
trade barriers or would-be threat of trade barriers in
international trade is of great necessity. Based on previous
studies, taking horticultural products as an example, this
paper designs a general RBF neural network model to detect
whether there are trade barriers among the main destinations
of Chinese horticultural products. The study shows that
general RBF neural network can identify trade barriers
correctly with an error probability about 10%, so it is a helpful
tool for identification of trade barriers.
Keywords-pattern recognition; trade barrier; RBF neural
network; Chinese horticultural products
I. INTRODUCTION
Trade barriers correct externalities, but distort trade as
well, which brings negative effect on international
distribution of resources. After China’s accession into WTO,
trade friction happens repeatedly. Patten recognition can help
to identify whether there is any threat from trade barriers so
as to take countermeasure. However, present artificial
intelligence technology is seldom applied into researches on
international trade friction. In the field of pattern recognition,
the increasing complexity of patterns requires better pattern
recognition method. And neural network shows great
advantage in parallelizability and fault-tolerance in this field.
The application of neural network in pattern recognition is to
match the pattern with eigenvector and construct network
model to classify samples into correct sample spaces.
Neural network has two functions: feature extraction and
statistical classification [1]. Gao Daqi and Yang Genxing
(2001) raised the idea that, by learning, the center and width
of Gaussian kernel function should automatically determine
and design IRBF consisting RBF network and LBF network
[2]. Han Min and Cui Pisuo (2006) put forward a dynamic
RBF training algorithm applied to pattern recognition, which
used regional shining-upon error function and the novelty of
resource allocation network (RAN) to dynamically adjust its
hidden nodes in order to ensure more efficient pattern
recognition [3]. Dang Changqing (2007) proposed a RBF
neural network to test traffic accident based on the improved
Boosting algorithm [4]. Zhao Jiewen applied RBF neural
network with genetic algorithm to classify the odor of
apples, which showed accuracy over 83% [5]. Ji Feng used
RBF network to evaluate personal credit, and measured the
amount of neurons in hidden layer and the center of the
radial basis function by introducing maximum input
sensitivity in training, and designed Sen-RBF to recognize
the two types of patterns [6]. Liang Wuke combined rough
set theory and RBF neural network to design a Rough
set-RBF neural network for examining the breakdown in
hydro-turbine generating unit [7]. He Tao and Zhou
Zheng’ou, by using higher order statistics and RBF neural
network, raised a new method to recognize the modulation
type of signals based on statistics pattern with higher
efficiency and accuracy, which is specific to digital
modulation of communication signals [8]. Zhao Liaoying
estimated generic conditional probability density with RBF
network, and classified data with Bayes classification. The
results showed that the method has a fast classification speed
with a similar recognition rate with Support Vector
Machines [9]. This paper combines those references and the
practical problems and designs the general RBF network
model.
II. MODEL AND ITS IMPROVEMENT
Radial Basis Function Network is a forward neural
network with powerful function in vector classification and
calculation. RBF network is composed of input layer, hidden
layer and output layer. The signal transmits from the input
layer to radial basis layer and brings out partial effect under
the effect of kernel function, and then form linear out in
output level.
The general RBF network uses radial basis function as
the basis of hidden layer. Hidden layer changes input vectors
and transmits low-dimensional pattern into high-dimensional
pattern so that the linear inseparable problem in former
pattern become separable in later pattern. General RBF
network is different with regularized RBF network. First, the
amount of radial basis function M is unequal to the amount
of samples P, M<<P. Second, the center of radial basis
function is not restricted to the data point, but subjected to
training algorithm. Third, diffusion constants of radial basis
are not unified which is also subjected to training algorithm.
Four, linear output function consists hedging points which
can make up the difference between mean value and target
value on the sample set of basis function.
The design of General RBF network composes of
structure design and vector design. Structure design mainly
relays on question on hand to determine the hidden notes.

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DOI 10.1109/ICCET.2009.25
Vector design should take data center of basis functions, In which cj, δj and wj denote the data center, developing
development constant and the weight of output nodes into constant and weight of output nodes of the basis function No.
consideration. j respectively, as in (7)-(9),
wj P
Δc j = η ∑ ei G ( X i − c j )( X i − c j ) , (7)
2
2
Φ(X, C1)
δ j i =1
W1,1
x1 W1,1
3
wj P 2
2
W1,2 W 2 Δδ j = η ∑ ei G ( X i − c j ) X i − c j , （8）
3
2
p ,1
3 δ j i =1
... ...W1,n W2,1
Φ(X, C2)
... Σ P
Δw j = η ∑ ei G ( X i − c j ) . (9)
xp 2 i =1
Wp,2
... 3
The above objective function is the sum of all errors
Wn,1 caused in training sample. The output vector modifier
formula is a kind of batch processing modification. That’s to
2 Φ(X, Cn ) say the output will be adjusted in every round. Define the
Wp,n
objective function in an instantaneous form, so the error
caused by inputting sample:
Figure 1. The structure of RBF neural network. E = 0.5ei2 . (10)
Kernel function in hidden layer should be radial Change the minimized vector modified formula in the
symmetry, so this paper uses Gaussian function as kernel objective model into
function in hidden layer, whose expression is wj
T Δc j = η e G ( X − c )( X − c ) , (11)
( X − Ci ) ( X − Ci ) δ j2 i j j

Φ i ( X , Ci ) = exp[ ]. (1)
2
2σ wj 2

Here Φi denotes the output of the hidden node NO. i; X

Δ δ j = η 3 ei G ( X − c j ) X − c j , (12)
δj
denotes output sample; Ci denotes central value of Gaussian
function. This paper uses supervised learning algorithms to Δw j = η ei G ( X − c j ) . (13)
train data centers, developing constant and weight of output
nodes in basis functions. III. CASE STUDY
Define the objective function as:
p 2 According to trade barrier issues to Chinese horticultural
E = 0.5 ∑ ei . (2) products in recent years, sample countries can be divided
i =1 into two groups. The first group consists Austria, Brazil,
Here p is the number of training samples; ei is the error Canada, Egypt, Germany, Israel, Japan, Korea, Kuwait,
signal aroused while inputting sample No. i; di is the distance Mexico, Netherlands, Philippines, Qatar, Russian, Saudi
between two data centers; Cj is the mean value of samples in Arabia, Spain, Turkey, United Arab Emirates, United
clustering domain. Kingdom, United States of America; the second group
M consists Argentina, Australia, Belgium, Chile, Denmark,
ei = di − F ( X i ) = di − ∑ w j G ( X i − C j ) . (3) Finland, France, Greece, India, Indonesia, Iran, Italy,
j −1 Malaysia, New Zealand, Norway, Pakistan, Singapore, South
To minimize the objective function, the correct value of Africa, Sweden, Switzerland, Thailand. In recent years,
vectors should be proportional to negative gradient. That’s to countries in the first group impose non-tariff barriers to
say, China, while the second group has some unserious trade
∂E conflicts with China [10].
Δc j = −η , (4) This paper assumes network output variable as Y. Y = 1
∂c j means that there is threat from trade barriers, while Y = 0
means there is not threat or the threat is small.
∂E
Δδ j = −η , (5) Pure economics and pure politics can not explain
∂δ j ubiquitous contradiction between politic and economy,
country and market, and society and person in market
∂E economy society. To consider those politic factors is
Δw j = −η . (6) necessary in studying economic issue, while to solve politic
∂w j
issue need economic methods. New political economy comes
into being in a context of international economy integration
and liberalization. As international trade is closed related

406
with political situation in all countries, explanatory variables
in existing literatures can not explain trade flow completely.
Model in this paper not only uses economic variables,
location variables and political variables, but also completes
political variable and cultural variable with public choice
theory and international politic economy theory. The
probability that horticultural products exports encounter with
non-tariff barriers mainly lies on agricultural characteristics
of import countries, economic characteristics and the
recognition rate is quite low. Table 3 lists data of 2007 as
testing samples and the simulation results, and the spread is
0.01. China does not have horticultural trade conflicts with
Singapore in 2007, so the real pattern should be 0, but the
simulation result is 0.98. All other simulation values are
right. Therefore, the erroneous recognition rate is only 10%.
TABLE II. SOME TRAINING SAMPLES AND THEIR GENERAL RBF
OUTPUT AND INPUT VALUE

preference of policy maker [11]. Model variables in the Country Input Output
paper are listed in table I. Argentina 0 0 1 1 1 0 0 0.09 0.18 0 0
Australia 1 0 0 0 1 1 1 0.05 0.08 0.02 0
TABLE I. OUTPUT VARIABLES AND THEIR MEANING
Austria 0 1 0 1 1 1 0 0.01 0.09 0 1
Variables Meaning Belgium 0 0 0 1 1 1 0 0.02 0.03 0.01 0
APEC APEC member countries or not Brazil 0 0 1 1 1 0 0 0.33 0.3 0.02 1
Parliamentary Parliamentary republicanism or not Canada 1 0 0 0 1 1 1 0.03 0.04 0.04 1
Presidential Presidential republicanism or not Chile 1 0 1 1 1 0 1 0.06 0.29 0 0
Multipartism Multipartism or not Denmark 0 0 0 1 1 1 0 0 0.07 0 0
Culture Western religious culture or not Egypt 0 0 1 1 1 0 0 0.18 0.68 0.01 1
Income High-income country or not Finland 0 1 0 1 1 1 0 0 0.11 0 0
Position Geographical position. France 0 0 1 1 1 1 0 0.16 0.06 0.03 0
Yield Yield rate of horticultural products of a country
Germany 0 1 0 1 1 1 0 0.07 0.04 0.06 1
Population Agricultural population rate of a country
Greece 0 1 0 1 1 1 0 0.07 0.24 0 0
Export Export rate of horticultural products of a country
India 0 1 0 1 0 0 1 1 1 0.04 0
The above indices cover economy, politic, cultural, Indonesia 1 0 1 1 0 0 1 0.13 0.82 0.11 0
location and policy variables. All output variables, APEC, Iran 0 0 1 1 0 0 1 0.2 0.49 0 0
Parliamentary, Presidential, Multipartism, Culture, Income Israel 0 1 0 1 1 1 1 0.02 0.05 0 1
and Geo-position, are two valued variables. APEC=1 means Italy 0 1 0 1 1 1 0 0.27 0.09 0.06 0
the country belongs to APEC, while APEC=1 means the Japan 1 0 0 1 0 1 1 0.14 0.07 1 1
other side. Parliamentary=1 means the country carries Korea 1 0 1 1 0 1 1 0.12 0.15 0.14 1
Kuwait 0 0 0 0 0 1 1 0 0.02 0 1
parliamentary republicanism, while 0 means not.
Malaysia 1 0 0 1 0 0 1 0.01 0.32 0.09 0
Presidential=1 means the country carries presidential
Mexico 1 0 1 1 1 0 1 0.2 0.43 0 1
republicanism, while 0 means not. Multipartism=1 means
Netherlands 0 0 0 1 1 1 0 0.03 0.06 0.07 1
there are no less than 3 parties in the country, while 0 means
New Zealand 1 0 0 0 1 1 1 0.02 0.16 0 0
less than 3. Culture=1 means the mainstream culture in the Norway 0 0 0 1 1 1 0 0 0.09 0 0
country is related with western religious cultures which refer Pakistan 0 1 0 1 0 0 1 0.08 0.95 0.01 0
to Catholic, Christian, Orthodox and Judaism, while 0 means Philippines 1 0 1 1 0 0 1 0.13 0.73 0.05 1
unrelated. Income=1 means the country belongs to Qatar 0 0 0 0 0 1 1 0 0.02 0 1
high-income country which is decided according to the data Russian 1 0 1 1 1 0 1 0.14 0.19 0.01 1
from World Bank, while 0 means not. Geo-position=1 means Saudi Arabia 0 0 0 0 0 1 1 0.02 0.17 0.01 1
the country is China’s neighbor, while 0 means not. Note: Data about level of national income are from World Bank Data Base; the yield rate of
As Briton carries constitutional monarchy, most horticultural products, agricultural population rate of a country, export rate of horticultural product
are calculated according to the data from FAO data base; political and custom data are from research
scholars think commonwealth countries such as Australia, website of Chinese international strategies and http://www.fmprc.gov.cn; Geo-position are got from
research website of Chinese international strategies and Chapter V: Transportation in International
New Zealand, Canada, as constitutional monarchy Trade of “International Trade Geography (Edition IV)” edited by Zhu Xianru; the data also refer to
APEC s secretariat official website: http://www.apecsec.org.sg.
country. Some scholars are opposite to the idea, but this
paper accepts it. Law in some countries in West Asia, 1.0
such as Saudi Arabia, Kuwait, Qatar and United Arab
0.8
Emirates prohibits parties to take part in politics, so the
paper takes these countries as non-partisan countries. All 0.6
western religions are with the same resource, so they
0.4
share some similarities. Western culture, custom and
values are generated from these religious cultures. This 0.2
paper uses data of 2004 and 2007, but there are only
0.0
output variables and input variables of 2004 listed in
table II. This paper uses Matlab 7.0 to design and realize -0.2
general RBF network. Network training and testing shows 1 2 3 4 5 6 7 8 9 10

that when vector spread ∈[0.02, 0.008], the erroneous Figure 2. The simulation error curve.

407
 TABLE III. TESTING SAMPLES AND THE SIMULATION RESULTS Network for Classifying Apples’ Scent,” Food Science,
Country Input mode Simulink
vol.25, No.2, 2004, pp.39-42.
Singapore 1 1 0 1 0 1 1 0 0.23 0.04 0 0.98 [6] Ji Feng, Li Yong, Song Jiashan, Fang Zhaoben, “Credit
South Africa 0 0 1 1 1 0 0 0.06 0 0.01 0 0 scoring model based on SenV-RBF,” Journal of University of
Spain 0 0 0 1 1 1 0 0.22 0.11 0.11 1 0.98 Science and Technology of China, vol.37, No.7, 2007,
Sweden 0 0 0 1 1 1 0 0 0.06 0.02 0 0 pp.767-772.
Switzerland 0 1 0 1 1 1 0 0.01 0.11 0 0 0.22
Thailand 1 0 0 1 0 0 1 0.08 0.86 0.05 1 0.98 [7] Liang Wuke, Zhao Daoli, Ma Wei, etc, “Fault diagnosis of
Turkey 0 1 0 1 0 0 1 0.27 0.53 0.01 1 0.98 hydroelectric units based on rough set and RBF network,”
UAE 0 0 0 0 0 1 1 0.01 0.07 0.01 1 0.98 Chinese Journal of Scientific Instrument, vol.28, No.10, 2007,
UK 0 0 0 0 1 1 0 0.02 0.03 0.13 1 0.98 pp.1806-1809.
USA 1 0 1 0 1 1 1 0.51 0.03 0.15 1 0.98 [8] He Tao, Zhou Zheng’ou, “Modulation recognition using
Note: The date sources are same with table 2. Improved RBF neural networks,” Signal Processing, vol.22,
No.4, 2006, pp.515-518.
IV. CONCLUSION [9] Zhao Liaoying, Li Xiaorun, Zhao Guangzhou, “Data
This paper applies artificial intelligence technology into classification based on density function estimated by radial
trade barrier recognition. Under the framework of political basis function,” Journal of Zhengjiang University
economic system, the quota system of the trade partners’ (Engineering Science), vol.41, No.7, 2007, pp.1088-1092.
effects on Chinese horticultural products is identified. [10] Wang Ping, Study on Chinese strategy of technical barriers to
Recognizing which country would impose trade barrier to agricultural products trade. Beijing: China Agriculture Press,
Chinese horticultural products by general RBF neural 2004.
network can help Chinese enterprises to take preventative [11] Zhao Yu, Zhang Yu, Qi Chunjie, “Analyze on factors
measures or countermeasures pointedly. The erroneous influencing the horticulture products export: based on new
recognition of RBF neural network is 10%. The political economics,” Ecological Economy, 2008, pp.92-95
classification capacity of RBF neural network is closely
related with the number of training samples, so the
performance of network can be improved by adding the
number of samples. In addition, pattern recognition is based
on the extracted characteristics of recognizing objects, so the
extracted characteristics should be representative and typical.
Therefore, fuzzy principles can be introduced to extract
objects’ characteristics in trade barrier recognition so as to
reduce the erroneous rate of recognition.
ACKNOWLEDGEMENT
It is a project supported by The Trade Barrier of Chinese
Horticultural Production Export and the Research of Its
Gambling Strategy (NCET050671), which was assisted by
the new centurial talent plan of Chinese Education
Department.

REFERENCES
[1] Lang K J, Witbrock M J, “Learning to tell two spirals apart,”
Proceeding of Connectionist Models Summer School, vol.1,
No.1, 1989, pp. 52-58.
[2] Gao Daqi, Yang Genxing, “Basic Principles of Pattern
Classification Methods Based on Improved RBF Neural
Networks,” Journal of East China University of Science and
Technology, vol.27, No.6, 2001, pp. 677-683.
[3] Han Min, Cui Pisuo, “A dynamic RBF neural network
algorithm used in pattern recognition,” Journal of Dalian
University of Technology, vol.46, No.5, 2006, pp. 746-751.
[4] Dang Changqing, Zhang Jinghui, Shen Zhiyuan, “Traffic
incidents detection based on Boosting method and RBF
neural network,” Computer Applications, vol. 27, No.12,
2007, pp. 3105-3107.
[5] Zhao Jiewen, Liu Muhua, Pan Yinfei, Zou Xiaobo, “The
Application of Genetic Algorithm to Optimizing Neural

408
 2009 International Conference on Computer Engineering and Technology

An Object-oriented Product Data Management

Fan Wang, Li Zhou
School of Mechanical Engineering
Shenyang Ligong University
Shenyang,Liaoning,110168, China
wangfan48382002@yahoo.com.cn, lizhou@imr.ac.cn

Abstract considered. The first one is the product structure

Product Data Management (PDM) system enables the
management of the entire product lifecycle and related
information about its design, its manufacturing and its in-
service. The interest of using the object-oriented approach
for modeling, specifying and implementing PDM systems
is detailed. Then an implementation method is proposed in
order to clarify the instantiation of UML diagrams into
database entities. Finally the system is implemented on
J2EE platform. The used UML diagrams for the modeling
and integration of product, process, and resource data is
detailed in order to argue the interest of an object-
oriented approach in such kind of PDM implementation.
1. Introduction
“Time to market” is becoming one of the most
important business and strategic key elements for product
development and manufacturing. Customers also want a
more customized product. In aeronautics, companies have
to reduce lead times and to develop a better-configured
product fulfilling customers requirements including an
higher level of quality and safety [1]. These topics
represent major challenges in turboprop manufacturing as
it is a niche market; in 2001, the sector total sales were
equivalent to the sales of General Motors alone. Section 2
presents the key features of Product Data Management
(PDM) systems and the basic needs for their
implementation. Section 3 proposes the overview of a
Unified Modeling Language (UML)-based method for
PDM implementation. Section 4 describes the used
approach for product structure modeling. Section 5 details
the modeling of design process and the specification of
the corresponding workflow. Finally, Section 6concludes
this research work and summaries some future
developments on UML-based method for PDM
implementation.
2. PDM features
PDM aims at managing and storing the product data
and also information related its entire lifecycle
(manufacturing, assembly, maintenance, etc.) [2,3]. In this
paper, two main functionalities of such a system will be
management and the second one is the workflow engine
[4,5]. Product structure manager organizes and stores the
whole product data including the parts list and the various
product configurations. It controls the data versioning and
the link between the parts of the bill of material and
associated documents (CAD or text files). It also manages
of data lifecycle status: available, frozen or obsolete.
According to the product structure definition, workflow
engine distributes the right data to the key user at the right
time. The workflow functionality allows the
implementation of ISO oriented processes in product
development in order to be sure that key tasks (such as
decision making, product review, etc.) are effectively
carried out by the right team member. Many others
functionalities are also currently available in PDM system
but they will not be discussed in this paper.
Functionalities such as product configuration
management, digital mock-up visualization, and
collaborative tasks management can be mentioned.
2.1. Modeling requirements of PDM
The added value of UML [6,7] for modeling and
specifying the product structure [8,9] and the workflow
[10,11] is considered in the paper. Its applications [12] in
Enterprise Modeling and Business Process Reengineering
[13] have led to the choice of UML for several reasons. It
efficiently supports the modeling and specification of the
product breakdown structure (parts) and related product
data (documents). It ensures a relevant modeling of
generic workflow with activities and transition criteria.
Then their implementation in workflow engine is fairly
simplified. Last, the overall class diagram of the VPM-
Chains system for specifying all kind of data to be
managed is efficiently supported by UML.
2.2. Brief survey of PDM implementation
methods
Often, PDM implementation methods are based on
information system modeling approaches [14,15,16].
Considering this subject, various methods have been
proposed for capturing the requirements [17,18],
modeling and structuring the data [10,19], specifying the

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DOI 10.1109/ICCET.2009.42
implementation [20,21]. Several of the above mentioned
research works are based on object-oriented methods. The
MOKA framework can also be mentioned as a good
example of UML-based approach [21]. Then the choice of
UML as methodological kernel for the PDM
implementation was evident [22] and a proposition of
UML-based method has been developed using the
experience of design chain specification and
implementation.
3. Object-oriented method for PDM
The implementation process of new chain includes
five major phases: 1)Clarification of the design chain: In
this phase, the subject and scope of the design chain are
defined. The requirements list of design chain is specified
including the use of UML use-case diagram; 2) Capture
and analysis of process and data: The capture of design
and production engineering processes and the modeling of
input and output data of each task are carried out. The
issue of this phase is deliverable summarizing the required
data in order to implement the chain with UML use-case,
class and activity diagrams; 3) PDM implementation: This
phase consists in the specification of tree structure
including product data and documents on the one hand
and in the definition of workflow including all the tasks
and data processing on the other hand. This workflow
tends to be created as a generic model with the build time;
4) CAx integration: The aim of this phase is to link the
numerous CAD and CAE systems used in design chain in
order to ensure a complete traceability and to improve the
quality of the design. Therefore, design and simulation
tasks must be clarified. Each task is identified and named
according to generic ID. CAx integration consists then in
specifying the name and type of task where the software
should be used including the data list required to carry out
the tasks. After a set of case studies and final validation,
all softwares to be used are clarified including the data to
be issued.
Indeed, both PDM implementation and CAx
integration phases need the complete achievement of the
capture and analysis of process and data phase. These two
major phases can be parallelized in order to reduce the
global implementation time.
4. Product structure modeling
All the information belonging to the product lifecycle
phases (design, manufacturing, assembly, etc.) are
managed and stored within the product structure. The
definition of this structure is based on metadata for the
data classification and configurable links for the
specialization of relationship between the metadata [23].
The product structure is considered as a hierarchical parts
list including: product, sub-assemblies, and basic
410
components. Regarding their types (CAx or MS-Office
files), documents are attached to these parts via metadata
links. In PDM, the product structure allows the
management of bills of material and product
configurations [24].
4.1. Specification of product structure
Two kinds of breakdown viewpoints have been
defined: 1) product structure based on design viewpoint; 2)
product structure based on assembly viewpoint. These two
viewpoints of the breakdown only partially provided a
clear understanding of the full product structure. This
modeling solution avoids duplicating data: the shared data
are integrated into the common part of the product
structure. The instantiated breakdown does not even have
to be browsed when searching specific document related
to a specific part used in a given activity. The query will
use a given part linking to the document and the activity.
The main goal is to allow the workflow engine to
automatically retrieve the right data without ambiguity.
The class diagram of the product structure (Fig. 1)
provides not only static information but also helps to
understand the methods and relationships existing
between metadata. The class diagram also specifies how
the product definition is clarified in the system and how
the user may handle it. The product structure will be
composed of Part class, its sub-classes and Document
class. The performer will be able to use these classes
during his activity. The relevant data will be characterized
with Feature class and its sub-class.
Product Type
0..1
1..1
Product Component Product Part
has
belongs to
contains 0..*
1..1
1..*
0..*
Product Series
Product Parameter
Product Node
1..1 contains 0..*
1..* assemble constraints
belongs to
1..*
1..1 1..*
1..* 0..1
Product
on
1..* assembed after
has
1..1
Related Document
Fig.1 Class Diagram of Product Structure
4.2. Product data and documents lifecycle
In product data management a key attribute is the
information status. It represents the current issue of a state
change of a data or a document during its lifecycle. It
provides a very important maturity criterion for data
transfer and persistency. Then user may have access or
not to the data and documents depending on their status.
For a document, four possible statuses exits (Fig. 2): 1)
“In progress” means that the data is currently being
modified by an activity (or that nothing has been done
yet); 2) “Shared” means that the data is mature enough to
be used by the process as input for other activity; 3)
“Released” means that this data is frozen and will not be
further modified in this configuration; 4) “Obsolete”
means that the data cannot be used as input for an activity.
A comment usually explains why data is not usable. The
owner of the data is in charge of it and promotes or
demotes the maturity level in the system.
Obsolete
[demote]
[promote] [promote]
In Progress shared Released
[demote]
Fig. 2 State diagram of a product document.
5. Design process modeling
The PDM system includes workflow functionality
[28]. The main goal is to build (model) and run the same
generic process with controlled data for each new
turboprop development. The workflow ensures
airworthiness authority and customers of ISO oriented
quality and information traceability. It also allows a
project team member to find every relevant input data of a
given activity.
5.1. Capture of activities and processed data
In order to clarify the workflow, interviews with
experts are carried out, and the process split up into basic
activities [25,26] according two criteria: an activity is
under the responsibility of one person and lasts only a few
days. Then all the business processes have to be redefined
with all the relevant data (input or output data of each
activity)[19]. A difficult point is to identify which
activities should be considered as “push” and which ones
as “pull”. At the end of “push” activities the following one
is triggered. Mainly due to concurrent engineering [29,27],
the need to pull data in order to launch activity is required:
411
activities may start as soon as the required input data are
available rather than waiting the end of an activity for
triggering the following one. This “pull” approach is
currently what the users really needed, but nowadays only
the “push” one is included in VPM-Chains system.
5.2. Specification of generic process
The build-time VPM-Chains functionality is used for
modeling “templates” describing generic design processes.
Then UML-based processes should be as generic as
possible in order to ensure a huge level of reuse [11]. It
includes generic sub-processes as specified in the class
diagram of the design process (Fig. 3). The process and
the product structure diagrams contain the generic names
of the relevant data (in the product structure, data is also
identified by the name of the father’s part). The link
between a product and the considered process enables
query on the needed data.
Software part
0..*
implements
0..*
Performer changed by Activity contains Process
0..* 0..1 0..*
0..1
Action
Fig. 3 Class diagram of design process
6 The implement of PDM
The PDM is developed on the J2EE platform. The
system can be divided into three sub-systems: product
data management, workflow management and design
process management.
 Fig.4 Representative Interfaces of QMIS
The product data management sub-system manages all
the product structure data and their related document. The
all the states of product document are managed here.
Workflow sub-system manages all the workflow models,
and all their instances are initiated, monitored in it. The
design workplace is an integrated client of both product
data management sub-system and workflow sub-system.
All kinds of actors anticipated a product development
project can use the design workplace sub-system to
complete their design work.
Three-tier software architecture, including user
interfaces, business logics and databases, is applied in this
system. The three-tier design enables more efficient
system development by modularizing tiers and separating
the application from the database, since this makes it
easier to modify one tier without affecting the other tiers.
The databases, which are the tier 1, are developed by
using MS SQL Server 2000. All files used in PDM are
managed through FTP service implemented by using
Microsoft IIS Service. The tier 2 is business logics, in
which the EJB technologies are employed by using
412
Weblogic 8.1. The session beans in the EJB container are
realized based on the defined use cases(Due to the limited
space, all the use cases are omitted in this paper), while
the entity beans are realized according to the class
diagram defined in previous sections. The tier 3 is user-
interfaces that are designed by using java swing. The
representative interfaces related to the quality
management of project process are shown in fig.4.
7. Conclusion
Object-oriented method provides a secured
management for CAD or CAE data linked with product
structure and workflow. The business process analysis has
highlighted the need for more flexible tools concerning
the build time (workflow modeling) and run time
(workflow engine) modules. UML-based method for
PDM implementation has been proposed and validated
within the business case study of design process. The
method efficiently guides project team and helps them in
carrying out complex design processes. Various
abstraction levels with the same kind of diagrams have
been described: the needs of business case study, the
global architecture, and the links between objects with
class diagrams. The user interactions with the system have
been clarified with use case diagrams and generic
processes with activity diagrams. Based on the experience
presented in the paper, UML can be considered as a
relevant and efficient notation enabling the modeling,
specification, and implementation of PDM system
especially concerning the product structure and workflow.
This modeling notation limits misunderstanding problems
for the various dimensions of the implementation phase.
Preliminarily to this step, enterprise modeling
methodologies such as ARIS, CIMOSA, GRAI or PERA
should be carried out for the global understanding of the
company organization and management, and for the re-
engineering its key business process. Considering the
modeling, specification and implementation steps of
dedicated system such as PDM, ERP, or SCM, UML can
be considered as an unquestionable supplementary
modeling language.
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 2009 International Conference on Computer Engineering and Technology
Study of 802.11 Network Performance and Wireless Multicasting
Biju Issac
School of Computing and Design
Swinburne University of Technology (Sarawak Campus)
Kuching, Malaysia
e-mail: bissac@swinburne.edu.my
Abstract—IEEE 802.11 proposed a number of standards and
each one had a specific goal. They can be configured in
infrastructure mode or ad hoc mode. In the study of
infrastructure mode, the focus was on 802.11b/g co-existence
along with other single mode operations to analyze the
throughput and traffic drop through simulation. In the study
of ad hoc mode, the focus was to analyze the behavior of ah hoc
networks with demand driven ad hoc routing protocols like –
Ad-Hoc On-demand Distance Vector Routing, Dynamic Source
Routing and Temporally-Ordered Routing Algorithm in terms
of throughput. Finally an investigation is done to study
wireless multicasting and its behavior.
Keywords-wireless network performance, wireless
mulitcasting, infrastructure mode, ad hoc mode.
I. INTRODUCTION
The 802.11 wireless networks can be broadly classified
into two modes, namely – infrastructure mode and ad hoc
mode. Infrastructure mode has wireless nodes
communicating with a central access point (AP), which is in
turn is linked to a wired LAN, to form an Infrastructure
Basic Service Set (IBSS). Here the AP acts as a relay
between wireless stations or between wired and wireless
stations. A combination of many BSS with a backbone
distribution system (normally Ethernet) forms an Extended
Service Set (ESS). In contrast, Ad hoc mode has wireless
stations or nodes communicating with one another without
an access point to form an independent BSS. Nodes of these
network act as routers that discover and maintain routes to
other nodes [1]. Infrastructure and ad-hoc mode of
operations are used in different circumstances.
The rest of this paper is organized as follows. Section 2
is on Infrastructure network and its performance, section 3
is on Ad-hoc network and its performance, section 4 is on
wireless multicasting and section 5 is the conclusion.
II. INFRASTRUCTURE NETWOK AND ITS PERFORMANCE
In a wireless infrastructure setup, there are two basic
components – access points and wireless stations. An access
point or base station functions as a bridge by connecting to a
wired LAN (Local Area Network) through Ethernet cables.
It receives data, buffers and transmits data between the
wireless and the wired network infrastructure.
A building may require several access points to provide
complete coverage and allow users to roam seamlessly
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 414
DOI 10.1109/ICCET.2009.161
between access points. A wireless network adapter connects
users via an access point to the rest of the LAN. A wireless
station can be using a PC card in a laptop, an ISA or PCI
adapter in a desktop computer, or can be fully integrated
within a handheld device. The different extensions to the
802.11 standard use the radio frequency band differently.
Some of the popular IEEE 802.11 extensions are as follows:
802.11b – specifies the use of DSSS at 1, 2, 5.5 and 11
Mbps. 802.11a – specifies the use of a frequency
multiplexing scheme called as Orthogonal Frequency
Division Multiplexing (OFDM) and it uses a physical layer
standard that operate at data rates up to 54 Mbps. 802.11g –
specifies a high speed extension to 802.11b that operates in
2.4 GHz frequency band using OFDM to obtain data rates
up to 54 Mbps and as well as backward compatible with
802.11b devices. This is now a popular standard. 802.11i –
recognizes the limitations of WEP and enhances wireless
security. It defines two new encryption methods as well as
an authentication method. The two encryption methods
designed to replace WEP include Temporal Key Integrity
Protocol (TKIP) and Advanced Encryption Standard (AES)
[1].
Related works on performance analysis can be found in
[2]-[3]. The simulation is performed through OPNET
software [4] for the analysis of different 802.11 standards.
We considered 802.11a, 802.11b, 802.11g and 802.11b with
g networks. Comparisons are done in terms of throughput
and other factors when they work in single and mixed mode.
The trajectory timing of the mobile node movement is
generally set between 0 and 49 seconds as the mobile nodes
move from one end to the other and back.
A. Mixed Mode (802.11 b and g) vs. Single Mode (802.11 a, b, g)
In this case, different scenarios of single mode (802.11
a, 802.11 b and 802.11 g) and mixed mode (802.11 b and
802.11 g) were created. After the simulation run, the data
traffic dropped and throughput were measured. We created
an 802.11b with 802.11g WLAN that consisted of two Basic
Service Set (BSS) networks – BSSLEFT and BSSRIGHT. The
data rate of 802.11g mobile node is set to 54 Mbps and that
of 802.11b node is set to 2 Mbps. The nodes communicate
to a Target Station T that is connected to an Ethernet LAN
where the access points of BSSLEFT and BSSRIGHT are
connected. In the initial setup, BSSLEFT consist of five
802.11g stations that transmit data (each over 2Mbps) to T
and an 802.11g AP (AP1). The total load of BSSLEFT thus
becomes around 13 to 14 Mbps. BSSRIGHT consist of five
Figure 1 The topology used in the scenario where 802.11g nodes with AP
(on right) co-exist with 802.11b nodes with AP (on left)

802.11b stations that do not transmit data and an 802.11b

AP (AP2). The wireless network topology used is as shown
in figure 1.
The five 802.11b stations moves toward BSSLEFT and
gets associated with AP1 and this happens between 5 and 21
seconds as shown in figure 2. The 802.11g station in
BSSLEFT now have to use ‘CTS-to-self’ protection
mechanism before transmitting their data frames at 54 Mbps
Figure 2. The throughput on the target station T using different
because the newly arrived 802.11b stations cannot decode configurations like 802.11b, 802.11g, 802.11 b with g and 802.11a.
802.11g data. This and association of 802.11b stations to
BSSLEFT causes the throughput rate to drop from 14 Mbps to
10 Mbps during 10 to 21 seconds. 802.11b supports, compared to the other configurations. In
Later the five 802.11b stations traverses back to the first half of simulation, the mobile nodes that are
BSSRIGHT and re-associates with AP2 during 21 to 30 moving from right to left AP and back to original position
seconds. Now the throughput rises to 14 Mbps as BSSLEFT are not transmitting anything, but only receiving.
becomes an all-802.11g WLAN. The five 802.11g stations In the second half of the simulation, the transmitting
on left hand side do a similar but reverse exercise. They mobile nodes makes move from left to right AP and then
move toward BSSRIGHT and associate with AP2 during 30 to back to original position. In 802.11b and 802.11g (mixed
33 seconds. Now the throughput drops down as they are mode), the transmitting 802.11g mobile nodes will only
disassociated with AP1 and are scanning for AP2. From 33 work at the throughput of 802.11b access points with ‘CTS-
to 44 seconds, all the five 802.11g stations attach to AP2 in to-self’ protection mechanism enabled, in the second half of
BSSRIGHT and as AP2 is a 802.11b AP, the throughput drops simulation.
down to less than 6 Mbps. From 44 to 49 seconds, the five The performance of the mixed mode operation is not that
802.11g stations traverses back to BSSLEFT. This causes a encouraging compared to the single mode operation. Even
short dip in the throughput as they have to disassociate with for single mode operation, where all devices are of the same
AP2 and reassociate with AP1. Thus the 802.11g stations standard, the packets dropped are high (and hence more
move to and fro between 30 and 49 seconds. delay) when the mobile nodes move from one access point
Figure 2 shows the above processes of 802.11b with to another and thus it would hamper the smooth handoff and
802.11g co-existence in detail and shows the ups and downs seamless connectivity.
in the throughput graph as measured on Target station T.
The throughput is comparable for all 802.11g and all
802.11a networks that support 54 Mbps. For all 802.11b
networks, the throughput is lower. In the other simulation
models, the 802.11 standard is changed on the devices to
accommodate comparison of standards. The network
structure and node movements still remain the same. Thus
wireless networks with all 802.11a devices, with all 802.11b
devices and 802.11g devices were created and simulations
were performed on them for throughput and performance.
The graph for data traffic dropped is shown in figures 3,
which is more of an inverse of figure 2.
The drop is quite consistent and uniform for all 802.11b
Figure 3. The data traffic dropped in WLAN configurations like 802.11b,
node networks. For other configurations the drops are 802.11g, 802.11 b with g and 802.11a.
comparable. This is because of the low data rate that

415
 III. AD-HOC NETWORK PROTOCOL AND ITS
PERFORMANCE
Ad-Hoc On-Demand Distance Vector Routing (AODV) is
a reactive protocol in ad hoc network, and the finding of
route is based on a route discovery cycle that includes a
network search that is broadcasted and a unicast reply
containing the discovered paths. When a node wants to
establish a communication link, it does a path-discovery
process to locate the other node. If a path to destination
exists in the source node’s routing table, it will use that Figure 4. The mobile ad hoc network topology used for simulation.
existing route to send data packets. Otherwise to start the
route discovery, the source node broadcasts a route request,
RREQ packet with its and destination’s IP addresses, study the performance and to analyze its behavior, under
Broadcast ID (or RREQ ID) and the sequence numbers of different number of nodes etc using existing models [6]-[8].
source and destination nodes. The RREQ ID is a per-node A review of current routing protocols for ad-hoc mobile
counter that is incremented every time the node initiates a wireless network can be found in [9].
new RREQ. This makes the RREQ ID together with the
A. Performance Comparison
source IP address quite unique and can be used to identify a
As shown in figure 4, a wireless server is configured
particular RREQ. When a node receives this RREQ, it
initially with DSR and 40 mobile nodes are configured as
creates a reverse path to the node and the hop count in the
work stations (that can generate application traffic) running
RREQ is incremented by one. Once the receiving node’s
DSR. They are placed as 20 nodes on one side and
sequence number is greater than or equal to the destination
remaining 20 nodes on the other side. Later the same
station’s sequence number as indicated in the RREQ, it
exercise is done with 80 nodes, by placing 40 nodes each on
generates a route reply, RREP (reply message) indicating
either side of the server. The simulation is done for 30
the route. The node then unicasts the message to its next hop
minutes (1800 sec).
towards source node. Thus the receiving nodes set a
The wireless nodes can communicate to each other
backward pointer to the source and generate a RREP unicast
within a cloud or to the server. They can also communicate
packet. The reverse route created as RREQ is forwarded and
with the wireless nodes in the other cloud. Figure 5 shows
is used to channel the RREP back to the source. AODV
the generated throughput of 40 and 80 node DSR networks
favors the least congested route and uses hello messages to
that run some applications like Telnet, Email or FTP
maintain node connectivity.
sessions. The mobile nodes are configured to eliminate all
Dynamic Source Routing (DSR) also uses source based
receivers that are over 1500 meters away.
routing rather than table based and it is source initiated
The graph shows an expected result that the throughput of
rather than hop-by-hop. A node wishing to communicate
80 node DSR network is greater than the 40 node DSR
issues a Route Request to all its neighbors and each
network, reaching a peak of around 24,000 bits (during 1200
neighbor in turn rebroadcasts this Request adding its own
sec) compared to 12,000 bits (during 500 sec) for 40 nodes.
address in the header of the packet. When a Request is
Secondly, a wireless server is configured with TORA and
received by the destination node or by an intermediate node
40 mobile nodes are configured as work stations running
in the path to destination, a Route Reply is sent back to the
TORA by placing 20 nodes on one side and remaining 20
sender along with addresses accumulated in the Request
nodes on the other side. Later the same exercise is done with
header. Thus the entire route is maintained in the packet
80 nodes, by placing 40 nodes each on either side of the
header. DSR is not that scalable to large networks.
Temporally-Ordered Routing Algorithm (TORA) is a
distributed and adaptive protocol that can operate in a
dynamic network [5]. For a particular destination, it uses the
‘height’ parameter to determine the direction of a link
between any two nodes. Thus multiple routes are often
present for a given destination. For a node to start
communication, it broadcasts a Query to all its neighbors,
which is rebroadcast through the network until it reaches the
destination. This node would reply with an Update that
includes its height with respect to the destination, which is
sent back to the sender. Each node that receives the Update
sets its own height to one greater than that of the neighbor
that sent it, which results in a series of directed links from
sender to destination in the order of height values.
We did OPNET network simulation of ad hoc network Figure 5. The throughput of ad-hoc network when using DSR with 40
demand driven protocols like DSR, TORA and AODV to nodes and 80 nodes.

416

Figure 6. The throughput of ad-hoc network when using TORA with 40
nodes and 80 nodes.
server. The simulation is done for 30 minutes (1800 sec).
Figure 6 shows the generated throughput of 40 and 80
node TORA networks that run some applications like
Telnet, Email or FTP sessions. The nodes are configured to
eliminate all receivers that are over 1500 meters away. The
throughput graph for 40 nodes and 80 nodes TORA
networks looks comparable.
Thirdly, a wireless server is configured with AODV and
40 mobile nodes are configured as work stations running
AODV by placing 20 nodes on one side and remaining 20
nodes on the other side. This is done in the default mode.
Later the same exercise is done with 40 nodes, with reduced
or light traffic. The simulation is done for 30 minutes (1800
sec). It is configured to eliminate all receivers that are over
1500 meters away. Parameters set for AODV (to reduce
routing traffic) are as follows: Route Discovery Parameter:
Gratuitous Reply – Enabled, Active Route Timeout: 30 sec
and WLAN data rate: 1 Mbps.
Figure 7 also shows the throughput comparison of all 3
protocols –DSR, TORA and AODV. AODV with gratuitous
reply enabled (light traffic/LT option) and DSR shows
comparable throughput. Default AODV shows a greater
throughput and TORA is in between.
A comparison summary on the three simulated protocols
can be given as follows. DSR is meant for wireless ad hoc
networks where the mobile nodes move with moderate
Figure 7. The Throughput for ad-hoc network when using DSR, TORA and
AODV (default and low traffic (LT)) with 40 nodes.
417
speed in comparison to packet transmission latency [7]. As
DSR packets need to contain full routing information, the
memory overhead is more compared to AODV. But it does
not make use of periodic router advertisement which saves
some bandwidth and power consumption. A direct result of
this is that router advertisement overhead would be nil and
topology changes are eliminated. As DSR allows multiple
routes to the destination, in the event of a link failure, the
other valid routes can be checked and this can prevent route
reconstruction, hastening the route recovery. If multiple
routes are not available, route discovery should be done to
find a route. As DSR assumes small network diameter, it is
not scalable to large networks. The requirement to place the
route in both route replies and data packets increases the
control overhead much more than in AODV.
AODV overhead is much lesser in comparison to DSR as
the route replies need to carry only the destination IP
address and sequence number. This also reduces the
memory overhead in comparison to DSR. AODV also uses
a route discovery mechanism that is similar to DSR. A big
advantage of AODV is its support for multicast
communication. But on the negative side it needs symmetric
links between nodes and cannot make use of asymmetric
links. TORA is a link-reversal algorithm that is suitable for
densely populated network with large number of nodes. It
creates directed acyclic graphs (DAG) to help route creation
and supports multiple routes for single source-destination
pair. Multiple routes reduce route reconstruction overheads.
TORA needs synchronized clocks for its operation and if the
external time source fails, the protocol would fail. Even
route rebuilding can incur lengthy delays because of
synchronization related oscillations [8]-[9].
IV. WIRELESS MULTICASTING
To understand the effects of multicasting, we configured
wireless networks that support unicasting and multicasting
using simulation. PIM-SM (Protocol Independent Multicast
– Sparse Mode) is the multicast routing protocol model
used.
Protocol-Independent Multicast (PIM) is a family of
multicast routing protocols that can provide one-to-many
and many-to-many data distribution over networks,
especially the Internet. PIM uses routing information
supplied by other traditional routing protocols such as
Border Gateway Protocol (BGP) and does not include its
own topology discovery techniques – hence it’s protocol
independent. PIM Sparse Mode (PIM-SM) explicitly builds
unidirectional shared trees rooted at a Rendezvous Point
(RP) per group, and optionally creates shortest-path trees
per source. PIM-SM generally scales fairly well for wide-
area usage. PIM relies on an underlying topology-gathering
protocol to populate a routing table called the Multicast
Routing Information Base (MRIB) with routes. The routes
in this table may be taken directly from the unicast routing
table, or they may be different and provided by a separate
routing protocol such as Multiprotocol Border gateway
protocol. (MBGP). Regardless of how it is created, the
primary role of the MRIB in the PIM protocol is to provide

Figure 8. The point-to-point link utilization and throughput for wireless
unicast and multicast network
the next-hop router along a multicast-capable path to each
destination subnet. The MRIB is used to determine the
next-hop neighbor to which any PIM Join/Prune message
is sent. Data flows along the reverse path of the Join
messages. Like all multicast routing protocols that
implement the service model from RFC 1112, PIM-SM
must be able to route data packets from sources to receivers
without either the sources or receivers knowing a priori of
the existence of the others [10].
IP multicasting [11] can be enabled or disabled for each
IP interface during simulation. In the WLAN scenario that
was simulated, the sender uses unicast and multicast to
setup a video conference with the three receivers that are
connected over a point-to-point link. In unicast separate
sessions are setup for each receiver. Therefore, the sender
sends three copies of each video packet, one for each
receiver as shown in figure 8, resulting in greater traffic.
In multicast only a single session is setup for all receivers.
Therefore, the sender sends only one copy of each video
packet, resulting in lesser traffic. So the multicasting
approach is more efficient as it reduces line congestion.
Figure 9 and 10 shows the wireless throughput in sender and
receive stations.
Figure 9. The wireless throughput in each of receiver stations
418
Figure 10. The wireless throughput in the sender station
V. CONCLUSION
This paper presents an analysis of different 802.11
network standards – single and mixed mode. It was
observed that the mixed mode (802.11b/g) networks can
bring down the network throughput drastically. The
performance analysis of the ad hoc network and its three
demand driven ad-hoc routing protocols, namely – AODV,
DSR and TORA were also done. The different protocol’s
performance varies depending on different factors like lesser
or higher number of nodes, light or heavy traffic etc. Lastly
wireless multicasting efficiency was also looked into.
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of ACM, 45(12), 2002, pp.31-37.
 2009 International Conference on Computer Engineering and Technology
A Novel Approach for Face Recognition Based on Supervised Locality
Preserving Projection and Maximum Margin Criterion
Jun Kong∗, Shuyan Wang, Jianzhong Wang, Lintian Ma, Baowei Fu, Yinghua Lu
Computer School, Northeast Normal University, Changchun, Jilin Province, China
{Kongjun,wangsy273,wangjz019,malt442,fubw561,luyh}@nenu.edu.cn
Abstract
In this paper, we propose a novel approach for face
recognition, that combine Supervised Locality
Preserving Projection (SLPP) with Maximum Margin
Criterion (MMC) for preserving the within-class
neighborhood structure of facial manifold and
meanwhile finding an optimal feature space for
classification. We also give an effective solution to the
eigenvalue problem. Our method can avoid the
preprocessing stage of resizing the original image
resolution and Principle Component Analysis (PCA)
projection, so there is no information lost. Experiment
results demonstrate the effectiveness of the proposed
approach on the ORL face database .
1. Introduction
Face recognition is a focus in the pattern
recognition and computer vision research area. In the
past few decades, numerous algorithms have been
proposed for face recognition. Among these methods,
Principle Component Analysis (PCA) [1] and Linear
Discriminant Analysis (LDA) [2] are the most popular
ones. As an unsupervised method, PCA finds a
subspace where the samples have the minimum
reconstruction error, while LDA employs class label
and searches for a subspace where the sample from the
same class are as compact as possible and meanwhile
the sample from the different classes are as far as
possible. However, a number of researches [3], [4], [5]
have shown that facial images lie on a nonlinear
manifold. Though the PCA and LDA are simple to
implement and efficiently computable, they fail to
detect the intrinsic structure of the image space.
Recently, some nonlinear techniques have been
proposed to discover the nonlinear structure of the
manifold, such as Isomap [6], LLE [7] and Laplacian
Eigenmap [8]. Isomap preserves global properties
∗
Corresponding author.Tel.:+86 431 84639647; fax:+86 431 84639647
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 419
DOI 10.1109/ICCET.2009.55
while LLE and Laplacian Eigenmap both preserve
local properties [9]. However, these manifold learning
approaches are developed for data embedding and
manifold visualization [10], [11], they are only for
train samples, and have no straightforward extension
for out of sample examples, since are not suitable for
classification.
Some LDA based methods [12], [13], [14], [15] are
suitable for practical application, such as pattern
classification. Given a new high-dimension data, it will
find the corresponding low-dimension point via the
optimal transformation matrix which is learned by
these methods. Unfortunately, these LDA based
methods are all linear methods, without consideration
of the nonlinear structure of the image space.
In order to overcome the above shortcomings, we
propose a effective method for face recognition which
combine the manifold learning with discriminant
analysis based approach. Locality Preserving
Projection (LPP) [16] is a linear approximation of the
nonlinear Laplacian Eigenmap, it can optimally
preserves the neighborhood structure of the face
manifold, and shares many of the properties of
nonlinear techniques such as Laplacian Eigenmap,
LLE. What’s more, an extension of LPP that is
Supervised Locality Preserving Projection (SLPP) [17]
takes class label into consideration which is more
efficient for classification. Maximum Margin Criterion
(MMC) [12], [13] is a good discriminant analysis
based method, it aims at maximize the average margin
between classes, that can not only obtains an optimal
low-dimension feature space for classification but also
solve the so-called small sample size (SSS) [18], [19]
problem.
We utilize the idea of combine SLPP with MMC.
First, within-class scatter matrix and between-class
scatter matrix are learned by SLPP and maximizing the
distance of interclass margin samples , respectively;
then MMC is used to search for a feature space that
makes intraclass compact and interclass separate;
Finally, we give an effective solution to the eigenvalue
problem. Our method avoids the preprocessing stage of
resizing the original image resolution and PCA
projection, thus, there is no information lost.
Experiment results show the effectiveness of our
proposed algorithm. The rest of paper is organized as
follows: In section 2, LPP and MMC are reviewed.
Our proposed algorithm is described in section 3.
Experiment and comparison results are presented in
section 4. Conclusions are summarized in section 5.
2. Review of Locality Preserving Projection
and Maximum Margin Criterion
Given a set of
N sample images
X ={xi ,i =1,2,", N | xi ∈R }, Each xi belongs to one of C
m
classes. Y = {yi , i = 1,2,", N | yi ∈ Rd } (d < m) are
corresponding projected vectors. A ∈ R m×d is the
transformation matrix, that is Y = A T X .
2.1. Locality Preserving Projection and its
extension
LPP is a recently proposed method for
dimensionality reduction and image recognition. It can
preserve the local structure of the image space by
considering the manifold structure. Let W be an
affinity matrix defined on samples. The objective of
LPP is to solve the following problem:
N
∑ (1)
2
arg min
A
yi − y j W ij
i, j
= arg min AT XLX T A
A
subject to A XDX A = I
T T
where L = D −W is the Laplacian matrix and
D ii = ∑ W ij , measures the local density around xi , I
j different classes. Through this two scatter matrix,
is an identity matrix. A is determined by the minimum
eigenvalue solution to the generality eigenvalue
problem.
An extension of LPP, SLPP has been developed to
enhance the performance of LPP in pattern recognition
by considering class label information into original
LPP. Different from LPP, SLPP computes W with the
constraint that we find one point’s k nearest neighbors
with class label the same as that point’s.
2.2. Maximum Margin Criterion
MMC is a discriminant analysis based method, aims
at maximizing the average margin between classes, it
can solve the SSS problem and is reported to be very
effective for face recognition. The interclass distance
(or margin) can be denoted as :
d (Ci , C j ) = d (mi , mj ) − (S(Ci ) + S(C j )) (2)
where mi and mj are class mean of ith class and jth
class respectively, d(mi, mj) is defined as the Euclidean
distance between mi and mj, S(Ci) and S(Cj) are defined
as the traces of the scatter matrices of the ith and jth
class, respectively. After some deduction, the average
margin between classes under projection A is:
(
J ( A) = tr AT ( S B − SW ) A ) (3)
The transformation matrix A that maximize (3) is
the learned by solving the following eigenvalue problem:
( S B − SW ) A = λA (4)
Both SB and SW are matrices of size m × m. However,
m is often very large in face recognition. e.g. given
face images of size 92 × 112, m equals 10304, it will be
very expensive to compute A. In [13] the author resized
the original image resolution of 92 × 112 to 23 × 28 for
computation efficiency in the experiment of face
recognition. However, it must lose some important
information in the process of resizing which has a bad
influence on the recognition result.
3. The proposed algorithm
In this section, our proposed algorithm which
combine SLPP with MMC is introduced. We learn
within-class scatter matrix based on SLPP which can
preserve with-in neighborhood geometry structure,
while between-class scatter matrix is computed by
maximize the distance of margin point belong to
MMC looks for an optimal feature space for
classification. Finally, a simple and efficient solution
to the eigenvalue problem is given.
3.1. Within-class scatter matrix
In order to make the projected vectors of the sample
of same class compact with each other, based on the
method of SLPP, we minimize the following cost
function:
1 N
∑
2
min yi − y j W ij (5)
2 i, j
1 N
∑
2
= min A T x i − A T x j W ij
2 i, j
420
 = min tr (AT X ( D − W ) X T A) Let L1 = D − W and L 2 = D '−W ' , ui and vi be the

{ exp( − xi − x j
2
t) xi ∈k1 ( j ) or x j ∈k1 ( i )
ith eigenvalue and eigenvector of L2 − L1 , respectively.
Wij = U = diag (u1 , u 2 , " , u N ) , V = [ v 1 , v 2 , " , v N ] , we
0 otherwise
can obtain L2 − L1 = ZZ T where Z = VU 1 2
VT.
S B − S W = X ( L2 − L1 ) X
T
where t is a constant parameter, k1(i) denotes the set of So (8)
k1 nearest neighbors of xi in the same class. Let
= FF T

SW = X ( D − W ) X T , and we call it the within-class where F = XZ , and FF T is of size m × m.

scatter matrix. Since the size of F T F is N × N, according to the
knowledge of linear algebra, let λ i and l i be the ith
3.2. Between-class scatter matrix non-zero eigenvalue and the corresponding eigenvector
of F T F , ai = (λi ) −1 2 Fli is the eigenvector
With the purpose of making the projected vectors of corresponding to the ith non-zero eigenvalue of FF T ,
the sample of different class far from each other, we let rank ( F ) = r , L = [l1 ,", lr ] , Λ = diag ( λ1 ," , λr ) ,
maximize the distance of margin points belong to
different classes, the cost function is as follows: A = [a1 ,", ar ] , we get:
1 N
∑
2
max y i − y j W ij ' (6)
2 i, j A = FLΛ−1 2 (9)
1 N T
∑
2
= max A xi − AT x j Wij ' Finally, we can obtain the transformation matrix
2 i, j from eigenvectors corresponding to the d largest non-
= max tr ( AT X ( D'−W ' ) X T A) zero eigenvalues of FF T in A.

{
1 xi ∈k2 ( j ) or x j ∈k2 (i )
Wij ' =
0 otherwise
where k2(i) denotes the set of k2 nearest neighbors of
xi in the different class. Let SB = X ( D'−W ' ) X T , and we
call it the between-class scatter matrix.
3.3. Projection
According to MMC, we will maximize the
following criterion:
J ( Aopt) = maxtr(AT (SB − SW ) A)
The transformation matrix A is obtained based on
(4).
3.4. An effective solve to the eigenvalue
problem
Since the dimension of facial image is often very
high, computing the eigenvector of SB -SW is time-
consuming. We give an efficient computation to the
transformation matrix A, it avoids the process of
resizing the original image which gives cause for the
information loss in the MMC.
For a new test sample xi, we can project it to the
subspace via transformation matrix A as yi = AT xi ,
then classify it by nearest neighbor criterion.
Compared with some other methods, our proposed
algorithm has the following advantages:
• It merges respective advantage of two methods,
not only preserves within-class neighborhood
geometry structure but also has excellent
discrimination ability, which is effective for
face recognition.
• Based on the method of MMC, it can solve the
SSS problem in face recognition and avoid
projecting the sample images into PCA
subspace in the preprocessing step, so without
(7) losing any important discriminative
information.
• We give an efficient solution to the eigenvalue
problem, makes it relate to the number of
training samples rather than the size of the
image, since our method avoids resizing the
original image of high resolution, there is no
information lost.
4. Experiment
To evaluate the proposed algorithm, we compare it
with several other methods, Fisherface [2], marginal
fisher analysis (MFA) [20], geometry preserving
projection (GPP) [21] (single model), MMC [12], [13],
SLPP [17], using the well-known ORL face database.

421
4.1. ORL face database

The ORL database contains 400 facial images, they

are 10 different grey images for each of 40 individuals. Figure 1. Examples of ORL face database
For some subjects, the images were taken at different
times, the facial expressions( smiling or not smiling,
open or closed eyes) and facial details( glasses or no
glasses) also vary. Each image has the resolution of
92 × 112 pixels with 256 grey levels per pixel. Twenty
sample images of four individuals in the ORL database
are displayed in Figure 1.
To see the influence of the number of train sample
to the recognition rate, we chose randomly n (n = 3, 4,
5) different images from each person for training and
the rest 10-n for testing, respectively. For each n, we
repeat this operation 15 times, the final recognition rate
is the average result over 15 random training sets.

4.2. Experiment setting

(a)
For all the methods except our proposed method,
we first resize the face images from 92 × 112 to 30 × 37
for computation efficiency in the experiment, then for
methods: Fisherface, MFA, SLPP, GPP (single model),
we preprocess them using PCA step, and N － C
principle components are retained.
We experiential set the parameters in the
experiment. We chose the value k1 ( 2 ≤ k 1 ≤ 4 )
according to the number of train sample of each
2
person, and k2=40. In the heat kernel exp(− xi − xj t) ,
we let constant parameter t=2. The nearest neighbor
classifier is used for final classification.

(b)
4.3. Experiment result

Figure 2 shows the plots of recognition rate vs.

reduced dimensions for the aforementioned methods.
The experiment result above demonstrates that our
proposed algorithm achieves the best recognition
performance.
We give some analysis on the above result:
• MFA, GPP (single model) and our proposed
algorithm are similar, they all preserve local
manifold structure and meanwhile search for
optimal feature space for effective classification.
However, our proposed algorithm outperforms
GPP (single model) a little, mainly because our
method avoids resizing the image and the process
(c)
of PCA projection, without losing any
information. Figure 2. Performance comparisons on the
ORL database (a) 3 train. (b) 4 train. (c) 5 train

422
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423
 2009 International Conference on Computer Engineering and Technology
Association Rules Mining Based On Simulated Annealing Immune Programming
Algorithm
Yongqiang Zhang
Hebei University of Engineering
Handan,China
yqzhang@hebeu.edu.cn
Abstract—Association rules mining is an important problem of
data mining, in this paper we propose a association rules
mining algorithm based on the simulated annealing immune
programming algorithm which combines each character of the
simulated annealing algorithm and immune programming
algorithm. Through the theoretical analysis and the
experiment we can find that this algorithm has better
robustness and the global and local ability of search, thus it
can quickly, effectively excavate more association rules which
meet the conditions.
Keywords-Data mining; Association rules; Simulated
annealing algorithm; Immune programming algorithm
I. INTRODUCTION
Association Rules Mining is an important area of
research of data mining, it is used to look for the interesting
relationship or association between the data items of the
given data. Association Rules Mining is firstly raised by
Agrawal, Imielinske and Swami. Well-known Apriori
algorithm was proposed by Agrawal and Srikant[1]. On the
many expansion of Association Rules Mining, Agrawal and
Srikant proposed a sequential pattern mining again[2],
Koperski and Han also studied the spatial association rule
mining[3] and so on. However, these traditional methods of
association rule mining existed various problems of slow
convergence and the low extraction rate of the right
associated rule. Moreover, the association rules mining need
to find mostly frequent patterns which meet the
requirements of support, it will face a great deal of space to
search. Therefore, this will require a high-speed, efficient
method for association rules mining.
Immune programming[4] is an effective algorithm of
global optimization, which is proposed based on the
evolutionary programming. In this algorithm the immune
operation was joined into the evolutionary programming, to
a certain extent, it can inhibit degradation phenomenon in
the completely random optimization process of evolutionary
programming and the possible fluctuation phenomenon in
the latter time of evolution. The core idea is that the
operation is completed through two steps which are
vaccination and immune operation based on reasonably
extraction of immune vaccine. The former is to increase the
individual's fitness, the latter is designed to prevent the
degradation of the group. Therefore, the algorithm has two
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.156
Shuyang Bu
College of Information and Electrical Engineering
Hebei University of Engineering
Handan,China
bushuyang840617@163.com
significant characteristics. Firstly, it's the characteristic
succession of evolutionary programming "random evolution,
survival of the fittest", which has a better global
optimization;Secondly, it introduces the feature of
"immunization" on the composition model of unfavorable
gene, which use awareness of the human brain on the
problem to avoid the search of complete random algorithm in
the region of no solution, thereby reduce the amount of
calculation and improve the speed of solving questions.
Because the simulated annealing algorithm has strong
ablity of local optimization and can make the search to
avoid falling into the local optimal solution. Therefore, the
algorithm of this paper combines the simulated annealing
algorithm based on the immune programming algorithm, to
achieve the global and local organic combination. While
this algorithm retains the feature of stronger ablity of local
optimization of simulated annealing algorithm, it also inhers
the feature of global optimization and the better
characteristics of ability of computing of immune
programming, and thus it can better adapt association rules
mining of data mining. Therefore, this algorithm provides a
novel solution of the problem for association rules mining
because of its powerful searching ability of the global and
local combination.
II. THE RELATED CONCEPTS OF ASSOCIATION RULES
Let I = {i1, i2 ,..., im} be the collection of item sets (called
item sets),and the task-related collection of data D is a group
of the transaction sets(called transaction database), in which
each transaction T is also a item set, that is, T ⊆ I. Let A be a
collection of item sets, if and only if T ⊆ I, called transaction
T include A. Association Rule is a kind of rule that contains
the following form: A⇒ B ,where
A ⊂ I,B ⊂ I,A ∩ B = φ .
Generally, we use two parameters to describe the
attribute of associate rules:
(1) Support: Let D have the transaction of the s% to
support the item sets of both A and B, s% is called the
support of association rules A ⇒ B .
Called as support ( A ⇒ B ) = P ( A ∪ B ) .
(2)Confidence: Let the transaction of D which support
the item sets of A have the transaction of the s% as well as
424
to support the item sets of B, c% is called the confidence of
association rules.
Called as confidence ( A ⇒ B ) = P ( B | A) .
In order to excavate a significant association rules, we
need to set two thresholds: minimum support and minimum
confidence. If one both meet the threshold of the two rules,
it is called the rules of strong. If item sets meet the
minimum support, that is, the emerging frequency of item
sets is greater than or equal to the product of the minimum
support and the total number of transaction of D, often
called as frequent itemset.
Association rules mining mainly includes two steps as
follows:
Step one: Find all the frequent item sets, the frequency
of these item sets should be at least equal to (pre-set) the
minimum frequency of support according to the definition.
Step two: It can generate strong association rules accord
to the collection of frequent item sets. According to the
definition these rules must meet the minimum threshold of
confidence.
III. SIMULATED ANNEALING IMMUNE PROGRAMMING
ALGORITHM
proportion α(0 ≤ α ≤ 1). The vaccine is extracted from the
prior knowledge of the problem, and it contains
informations and its accuracy which play an important role
on the efficiency of the implementation of this algorithm.
3) Immune selection. That is: we test individual which
was vaccinated. If the offspring fitness is smaller than the
parent generation, which show that resulted in serious
degradation in the genetic process, then this individual will
be replaced by the corresponding individual of the father. If
the offspring fitness is better than the parent generation, it
will enter into the next generation instead of the parent.
Immune selection plays a positive role on the convergence
of the algorithm, it can eliminate the negative influence and
has strong robustness. Immune selection has the effective on
inhibiting the poor adaptive modern and enhancing its role
in the strong, so it is the fundamental guarantee of
convergence.
B. The flow chat of the simulated annealing immune
programming algorithm
Initialization

A. The description and analysis of immune operator

Fitness calculation
This paper introduces a new operator based on
enlightenment of the life sciences in the standard
evolutionary programming algorithm, that is, immune Stop running and
operator. Immune operator is made up of three steps, they Y output the optimal
Conditional
value
are extraction of the vaccine, vaccination and immune judgement
selection. Vaccination is to improve the individual fitness,
and immune selection is to prevent degradation of the N
group. Specific description is as follows: Extraction of the
1) Extraction of the vaccine. This algorithm takes the vaccine
adaptive method to extract the vaccine. In the evolution
process of the groups, we extract effective information from
The operation of
the best individual genes of every generation in order to simulated annealing
produce vaccines. The procedure is: let the individual be x
= x1, x2, ..., xL, through the analysis of the best individuals
which are reserved from the former K-1 generations, we The operation of
gauss mutation
take the common characteristics and effective informations
of this best individual's gene x1, x2 as vaccines H.
2) Vaccination. Set up individual b ∈ Rn, b = (b1, ..., Vaccination
bn). Vaccinated to them means to amend some genes of the
b according to the prior knowledge, so that individuals
could have the higher adaptation degree with the greater Immune selection
probability. These prior knowledges can be probably value
range of the optimal solution's some components, they may
also be a certain restrictive relation between some The updating of
components. Moreover, the operation should meet the popolation
following conditions: if b is already the best individual,b
transfer to b with probability 1; if b is the worst individual, Figure 1 The flow chat of the simulated annealing
that is, every gene of b with is different from the best immune programming algorithm
individual, b transfer to b with probability 0. Let the Groups
B = (b1, ..., bN), Vaccinated to B is a sort of operation that
random select Nα = αN individuals according to a certain

425
 IV. THE ASSOCIATION RULES MINING
ALGORITHM BASED ON THE ALGORITHM OF THIS
PAPER
1) Initialization code: At present, antigen and antibody
in the general immune algorithm, that is: the coding
methods of solution of the problems mainly have three
kinds which are binary code, real-coded and character
encoding, there are also a few gray code. It is mainly
transaction database for the discovery of association rules,
in which the application aimed are some data. Therefore,
this paper adopts the character encoding. Each record in
data mining which will be dealt with is regard as each
antigen, candidate pattern which will be generated in the
searching process is regard as antibody. Assume that there
are transaction data as follows (the commodities were
bought by customers )
Table1 The coding of commodity
Commodity 1 Commodity 2 Commodity 3 Commodity 4
a b c d
a
a
d
i
e
j
h
d
For the table 1 the antigen coding is a, antibody
a ⇒ d can be defined as a structure, coding is (a,d).
N individuals are randomly generated in data set,thus
constitue the initial parent groups A0.
2) Fitness function: Because the support Supp(x) and
the confidence Conf(x) are the two important parameters in
association rules, we take the fitness function as follows:
F(x) = W × Supp(x)/Supp + the step of the optimization of algorithm in the different
s min
(1)
W × Conf(x)/Conf
c min
Where Ws + Wc = 1,Ws ≥ 0,Wc ≥ 0, Suppmin is the
threshold of support, Confmin is the threshold of confidence.
We can see that the greater the support and confidence, the
larger the value of fitness function.
3) Conditional judgement: If the current group has
included the optimal solution (that is, the antibodies which
have been got meet the minimum support and minimum
confidence at the same time) or have already met some
termination condition (such as setting a maximum iteration
numbers), the algorithm would stop running and output the
result; Otherwise continue.
4) Extraction of the vaccine: We extract the vaccines H
based on the best individual.
5) The operation of simulated annealing: We take fitness
as the energy in the simulated annealing algorithm. When
the value of fitness increases, we accept the current solution
to be the next current solution, or accept this solution by a
certain probability p = exp (-fitness / T), that is :
426
⎧ 1, f i + 1≥ f i
p = ⎨ (2)
⎩ exp[( f i + 1 − f i / T )], f i + 1 < f i
Where fi +1 is the fitness value of the offspring
individual, fi is the fitness value of the father individual,
temperature T acts as the control parameters which diminish
in the process of the algorithm.
6) The operation of gauss mutation: The operation of
gauss mutation is conducted for the current group of father
Ak, thus generates the new group Bk. Let Ak=(a1,a2,...,aN),
ai Ω ⊂ Rn, a i = (a i ,a i ,...,a i ) ,（i =1,2,…,N）
1 2 n
The operation of gauss mutation which is carried out for
individual ai generates the new individual bi as follows:
b ij = a ij + ξ ji , j =1,...,n j = 1 ,..., n (3)
Where ξ ij obeys the gaussian distribution N(0, δ ij ),
2
variance is calculated according to the formula (4)
δ ij = (Tk M j + m j ) /(Tk + f (a i )) (4)
The gauss mutation carried out for the group Ak is
conducted for some individuals of the groups according to a
certain probability of mutation. Mj and mj respectively
control the degree of mutation of the individual during the
early and late of evolution process. Or rather, they control
evolution stages. Therefore, the choose of these two
parameters have a great relationship with the shape and size
of Ω, and mj≠0.
7) Immunization: We carry out the operation of
immunization for the group Bk, thus have got the group Ck.
8) Immune selection: We carry out the operation of
immune selection for the Ck, thus have got the new father
generation Ak+1, then transfer to (2).
V. THE RESULT OF EXPERIMENT
In this paper, according to the record data of transaction
of customers in e-shopping we conduct the corresponding
mining of association rules. We can discover the habits of
customers who used to buy commodities by using the
algorithm of simulated annealing immune programming,
thus help businesses develop targeted marketing and
advertising plans and compile the proper list of commodities.
For example: the analysis result of market shopping will
help businessmen plan and design how to rationally put the
commodities inside market.
For some basic informations of customers and the name
of commodities(that is, the fields of database ), such as the
customer's age, occupation, income and the name of the
electronic goods and so on, we express them by means of experiment show: this algorithm has the feasibility and high
the character encoding. Some values of the attributes effectiveness.
encoded in the following table 2.
ACKNOWLEDGMENT
Table 2 The coding of attributes values
This paper is supported by National Natural Scieence
Name of attribute Wide of attribute Values of coding
Foundation of Hebei Province No: F2006000648 and No:
<20 a
F2008000752.
Age 20-39 b
40-59 c REFERENCES
>59 d [1] Agrawal R, Srikant R. Fast Algorithm for Mining Association Rules.
20K-29K a In Proceeding 1994 International conference Very Large Data
30K-39K b Base(VLDB 94). Santiago, Chile. Sept, 1994:487-499.
Year of income 40K-49K c [2] Agrawal R, Srikant R. Mining Sequential Patterns. In Proceedings
50K-59K d International Conference Data Engineering (ICDE,95). Taibei.Taiwan.
… … 1995:3-14.
Student a [3] Koperski K, Han J.Discovery of Spatial Association Rules in
Worker b Geographic Information Databases. In Procceeding 4th International
Occupation Manager c Sympothsis Large Spatial Databases (SSD,95). Portland,
Teacher d Me.1995:47-66.
… … [4] Agrawal R, Srikant R.Fast Algorithm for Mining Association Rules.
Computer a In Proceeding 1994 International conference Very Large Data
DV b Base(VLDB 94). Santiago, Chile. Sept, 1994:487-499.
Category of MP3 c [5] Meilian Liang, Jiahui Liang, Guo Chen.Association rule mining
commodity Mobile hard disk d algorithm based on arrificial immune system[J]. Computer
Anti-Virus Software e Applications.Aug.2004.
… … [6] Yang jie, Genhao Deng. Mining Association Rules Based on
Artificial Immune Algorithm. Computing Technology and
In the experiment the parameters of algorithm are: the Automation. Mar. 2005.
initial group is 50, the number of iteration are 100 times, the [7] Parthasarathy S, Zaki M J, Ogihara M. Parallel data mining for
probability of the gauss mutation is 0.6; The parameter Tk in association rules on shared-memory systems[J]. Knowledge and
the annealing selection meets the formula: Tk= ln[T0/k+ Information Systems, 2001; 3(1): 1-29.
1](where T0=100) , the threshold of confidence is 75%, the
threshold of support is 1%.
The following are some association rules excavated
which meet the minimum support and confidence:
Computer ⇒ Anti-Virus Software
[support=2%,confidence=90%]
Age （ X ， <20) ⇒ category of commodity （ MP3)
[support=4%，confidence=75%]
Age（X，20-39） ∩ Income of year（30K-39K）
⇒ category of commodity（computer）
[support=6%，confidence=80%]
...
From the result of the experiment we can see that this
algorithm could effectively find most frequent sets and
generate the association rules. These rules meet the actual
conditions, thus show that this algorithm is effective and
feasible for use in the association rules mining.
VI. CONCLUSION
This paper proposes a association rules mining
algorithm based on the simulated annealing immune
programming algorithm which combines the immune
programming algorithm and simulated annealing algorithm.
This algorithm has stronger robustness and the effectively
global and local ability of search. The result of the

427
 2009 International Conference on Computer Engineering and Technology

Processing Power Estimation of Simple Wireless Sensor Network Nodes by Power

macro-modeling

M. Rafiee, M. B. Ghaznavi-Ghoushchi, B. Seyfe

Electrical engineering dept.
Shahed university
Tehran, IRAN
{rafiee, Ghaznavi, seyfe}@shahed.ac.ir

Abstract— The operation of wireless sensor networks is
fundamentally constrained by available energy sources. In this
paper, we propose a numerical power estimation technique for
register transfer level. This technique allows estimating the
power dissipation of components in WSN1 to their statistical
knowledge of primary inputs/outputs. During power
estimation procedure, the sequence of an input stream is
generated using input metrics and numerical macro-model
function is used to construct a set of functions that maps the
input metrics of a module to its output metrics. Then,
simulation is performed and power dissipation is predicted by
a macro-model function. The results are effective and highly
correlated, with average error of 3.5%.
Keywords-component; Wireless Sensor Networks, Processing
Power, Macro-modeling
I. INTRODUCTION
Networked sensor systems are seen by observers as an
important technology that will experience major deployment
in next few years for a plethora of applications. Typical
applications include, but are not limited to, data collection,
monitoring, surveillance and medical telemetry [1-2].
Wireless Sensor Network consists of large number of sensor
nodes. These tiny nodes consist of sensing, data processing,
and communicating components. Sensor Network (SN) aims
to provide an efficient and effective connection between the
physical and computational worlds. Sensor-nets are enabling
previously impossible applications but since they are often
battery-powered -typically by a pair of AA alkaline batteries
that can supply 3 volts at 2000mAhr – sensor-net operations
are fundamentally constrained by energy availability.
The four main ways in which nodes consume energy are
sensing, communication, computation, and storage [3-6].
Energy constraints, coupled with the need for longer life-
times, have led to a variety of techniques for modeling [7]
and managing energy consumption.
Power estimation at RT level is crucial in achieving a
short design cycle. Macro-modeling is the major technique
for power estimation at RT-level.
The standard hierarchical simulation approach to RT-
level power estimation consists of three steps: 1) functionally
simulate the RT-level description and collect the input
1
Wireless sensor network
sequences for each circuit block. 2) Simulate each block at
gate or circuit-level using the collected input sequences. 3)
Add the power consumption for all blocks to produce the
power consumption of the whole circuit. The disadvantage of
this approach is that it requires the interaction between RT-
level simulators and low-level simulators; power evaluation
is actually done at gate-level.
Other technique for power estimation at RTL-level is
macro-modeling. In this technique, low-level simulations of
modules under their respective input sequence are replaced
by power macro-model equation evaluation.
Macro-modeling techniques use capacitance models for
circuit modules and activity profiles for data or control
signals [8-10].
In this paper, we focus on the problem of power
estimation at register transfer level (RTL) for processor
components of node in sensor network.
Various power estimation techniques can be divided into
two categories: probabilistic and statistical.
Probabilistic techniques [11-13] use the probabilities of
the input stream and their propagation into the circuit to
estimate the internal switching activities. These techniques
are very efficient, but they cannot accurately capture factors
like glitch generation and propagation. In statistical
techniques [14-16] the circuit is simulated under randomly
generated input patterns and the power dissipation is
monitored using a power estimator. Therefore, the power
values are used to estimate the power dissipation for every
input stream.
The used technique generates random input vectors and
its power dissipation is estimated using power estimator. The
sequence vectors were provided to estimate power
dissipation of a given circuit with certain statistical
constraints such as confidence levels and error. The average
power dissipation was estimated by simulating the circuit
using a large number of samples in these vectors.
The input/output (I/O) metrics of our macro-model are
the average input signal probability Pin, average input
transition density Din, input spatial correlation Sin, average
output signal probability Pout, average output transition
density Dout, and output spatial correlation Sout.
The rest of this paper is organized as follows. In Section
II we give the background for the input parameters of our
macro-model. In Section III, we describe structure of WSN.
In Section IV, we discuss about our macro-model

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 428

DOI 10.1109/ICCET.2009.91
construction. This macro-model is evaluated in Section V. nodes that do stimulation of the sensors, 2) sensor nodes that
The results depict in Section VI. Section VII summarizes our monitor events and 3) user nodes that query the sensors and
work. are the final destination of the target reports.
Node performs a simple task: take an analog sample, and
II. POWER MACROMODELING transmit the data over the wireless link to one or more
Our macro-model consists of a nonlinear function and receivers. Node module functionally consists of three
estimates the average power dissipation Pavg using equation concurrent modules, which are analogous to the four main
(1). components of the node, namely, Analog-To-Digital (ADC)
Pavg = f ( Pin , Din , Sin ) (1) Converter, Microcontroller, Encryption and Transceiver.
The ADC converts the analog signal into digital values.
The macro-model function f (.) is obtained by a given Transceiver Sender receives data from sensor module, or
module macro-modeling which maps space of input signal receives data from other node module or transmits the data to
properties to the power dissipation of a circuit. other node module through the wireless channel.
Given a circuit block with n inputs and a binary input Microcontroller controls the whole process by setting an
stream I={(i11,i12,…,i1n)},…,(il1,il2,…,iln)} of length l, these interface between the two other sections.
metrics are defined as follows: All the nodes perform the same function but at different
n l
frequency, data resolutions and data domains.
ikj ADC converts the analog value into 10-bit digitals.
Module C takes the data from the ADC, adds the address of
j =1 k =1
Pin = (2) the receiver node and provides the data payload for the
n×l Transceiver module. Transceiver can send data to next node
or sink by using IEEE 802.15.4 standard [17] (CSMA/CA
n l −1
policy).
Sink consists of two components, Receivers, and CPU.
ikj ⊕ ik +1 j Receiver receives data from concurrent node module and
j =1 k =1
Din = (3) sends to CPU Module.
n × (l − 1) Fig. 1 shows the overall structure of sample WSN
system. Fig. 2 depicts the structure of Node module in WSN.
n n l
Port
ink ⊕ inj Node
S tim u lu s

M onito r
1 FIFO
j =1 k =1 n =1
Sin = (4) Sink
l × n × (n − 1) Port FIFO
Node
The output metrics are the average output signal 2
probability Pout, average output transition density Dout and FIFO
output spatial correlation Sout. The macro-model function f in ECO System
equation (1) can be used to construct to a set of functions fA ,
fB and fC that maps the input metrics of a macro-block to its Figure 1. Overall eco system
output metrics Pout, Dout, Sout are derived in equations (5), (6),
and (7):
16 16
Pout = f A ( Pin , Din , Sin ) (5)
ADC

ADC

Dout = f A ( Pin , Din , Sin ) (6)

Transceiver

Sout = f A ( Pin , Din , Sin ) (7)

10 10

The sequence of an input stream is generated for a µc

desired input metrics, then the output stream sequence is
extracted by the output waveforms simulation technique, and 30 64
the power dissipation is predicted by macro-model function.
III. STRUCTURE OF WSN Encryption

A sensor network is modeled as a set of heterogeneous

entities. Sensor nodes deployed over the area of interest.
They are triggered by a certain set of stimuli that are
transmitted to a remote base-station. Three main types of
Figure 2. Node module
sensor nodes need to be created and supported: 1) target

429
 IV. MACROMODEL CONSTRUCTION TABLE I. ACCURACY OF POWER ESTIMATION

Several methods [18-20] have been proposed to construct

power macro-modeling. Circuits Average error %
In used static power estimation procedure, the input ADC-1 7.2
metrics is computed for an input stream. The output stream
ADC-2 4
sequence is derived from the output waveforms of each
module. The average power dissipation Pavg is predicted ADC-3 8
using equation (1). ADC-4 3
We define a power macro-model as a linear function for
ADC-5 9.6
the estimation of power dissipation.
ADC-6 5.1
Pavg = β 0 + β 1 Pin + β 2 Din + β 3 Sin (8) DES-1 0.59
DES-2 0.33
Where the coefficientsβ i are the unknown and are to be DES-3 0.17
determined during the characterization using regression DES-4 0.44
analysis as discussed in [21]. To improve accuracy, another DES-5 0.29
option is the function considering I/O statistics Pin, Din, Sin,
Pout, Dout, and Sout as independent variables in (9):
Power consumption used model of microcontroller is less
than 0.0001 mW hence effect of this power in total
Pavg = β 0 + β 1Pin + β 2 Din + β 3 Sin + β 4 Pout processing power is inconsiderable.
+ β 5 Dout + β 6 Sout (9) Fig. 3 illustrates power comparison between macro-
model and reference simulated power in ADC module.
Fig. 4 depicts power comparison between macro-model
The key difference between the (8), (9) models is the
and reference simulated power in DES module.
number of fitting coefficients that impact the achievable
accuracy and the characterization effort.
V. MODEL ACCURACY ANALYSIS
Power macro-model is built at RTL level. The accuracy
of the proposed model is evaluated on processor components
of wireless sensor network. For those modules, we generated
random values of Pin, Din, and Sin. The function f in
equation (1) is used to construct to a set of functions fA, fB
and fC that maps the input metrics of a module to its output
metrics Pout, Dout, Sout.
The power values predicted by macro-model are
compared with those from simulations, and average errors
are computed.
Experimental results show that our randomly generated
sequences are with accurate statistics and highly
convergence. We generated 40 sequences with 10, 16, 64 Figure 3. Power comparison between macro-model and reference
bits wide. The sequence length is 200 vectors for the simulated power in ADC module
modules
VI. RESULTS
It is obvious from Table I that the function is accurate for
estimating the average power for ADC, Microcontroller, and
DES modules. In Table I the first column represents the
name of modules. Column two give the average error for the
estimates obtained with our macro-model. Reference values
for the module’s power dissipation are obtained using time
delays from the Synopsys PowerCompiler. In our
experiments, the average error is 3.5%.

430

Figure 4. Power comparison between macro-model and reference
simulated power in DES module
VII. CONCLUSIONS
Sensor-net lifetime is severely constrained by the
available energy source, Hence in this paper we present a
new numerical macro-modeling technique for high level
power estimation. This procedure uses input/output (I/O)
statistical parameters at the architectural level. We compare
our macro-modeling results with Synopsys Power Compiler
power estimation tool and find accuracy of this technique is
good. Our model shows average error of 3.5%.
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 2009 International Conference on Computer Engineering and Technology

A Fault-tolerant Strategy for Multicasting in MPLS Networks

Huang Weili, Guo Hongyan

College of Information and Electrical Engineering
Hebei University of Engineering
Handan, P.R. China
E-mail: starshine_0819@163.com, guohongyan_0819@163.com

Abstract—Multicast is an effective way for increasing the
efficiency of network resources utilization. But when designed,
MPLS did not consider the support of multicast. So multicast
in MPLS has many problems. To solve this, we introduce and
propose a new multicast strategy in this paper that combines
ERM and AMFM. Compared with the previous multicast
algorithms, it simplifies LSP setup and has less forwarding
states. And it is fault-tolerant. So if implemented in MPLS
networks, it can get a better performance.
Keywords-MPLS; multicast;fault-tolerant; FSMM
I. INTRODUCTION
Increasing the efficiency of Internet resources utilization
is becoming more and more important. Several evolving
applications like WWW, video/audio on-demand services,
and teleconferencing consume a lager amount of network
bandwidth. Multicasting is a useful and effective operation
for supporting such applications. Using the multicast
services, data packets can be sent from one source to several
destinations by sharing the link bandwidth. Only one copy of
the data packet is needed while a number of copies of data
packets are necessary in a unicasting network environment.
So, multicast uses the bandwidth resource more effectively
and efficiently.
Besides, Multi-protocol label switching (MPLS) [1] [2]
has emerged as an elegant solution to meet the bandwidth
management and service requirements for next generation
Internet Protocol (IP) based backbone networks. We think
that Multicast and MPLS are two complementary
technologies, and by merging these two technologies
together, we can enhance the network performance and
present an efficient solution for multicast scalability and
control overhead problems; Multicast attempts to reduce
network bandwidth usage, while MPLS attempts to provide
users with needed bandwidth in an appropriate switched-like
manner. However, at the beginning, MPLS is designed
without any consideration for supporting multicast. For
supporting multicast in MPLS networks, problems arise
while mapping layer 3 multicast trees to layer 2 LSPs (Label
Switched Path). Thus a number of issues need to be
addressed, such as flood and prune, source/shared trees,
uni/bin-directional trees and encapsulated multicast. To solve
the difficulties generated by implementing multicast in
MPLS networks, reference [3] proposed a new framework
for supporting multicast in MPLS networks. And
accordingly, many algorithms have been proposed to support
multicast in MPLS networks.
Reference [4] proposed an algorithm called MMT to
solve the problems of scalability of multicast in MPLS
networks. By using this algorithm, forwarding states can
only be stored on branching node routers, which largely
reduces the forwarding states of the network. And reference
[5] gives out an algorithm called ERM which only selects
edge routers as the branching nodes, and in doing so, many
advantages are gained. To further reduce multicast state,
reference [6] proposed an algorithm named Aggregated
Multicast, which allows multiple multicast groups to share a
single delivery tree in a MPLS domain. For the consideration
of the robustness of multicast in MPLS, reference [7]
proposed an algorithm called AMFM which facilitates fault-
tolerance in a very elegant way. Though so many algorithms
have been proposed, they only aim at solving one kind of
problems without any consideration for other kinds of
problems. It may successfully solve problems it studies, but
other kinds of problems still exist. And through thoroughly
study, we can find that different multicast algorithms may
solve different problems and have different performance, but
the technologies they adopted can be very similar. So it is
impossible to combine different kinds of algorithms to solve
several different kinds of problems together and
simultaneously. For this purpose, a new strategy called fault-
tolerant strategy for MPLS multicasting (FSMM) is proposed
in this paper.
The remainder of this paper is organized as follows. In
this section, we present a brief introduction to Multicast and
MPLS, and state the reason and motive for our proposed
strategy. Section 2 describes in detail our proposed strategy.
And Section 3 is analysis and evaluations of our proposed
strategy. Section 4 is a summary followed by a list of
references.
II. A FAULT-TOLERANT STRATEGY FOR MPLS
MULTICAST
Through firmly comparison and thoroughly study, we
find that the algorithms ERM and AMFM are perfect
solutions of the problems which they study. What is more,
the technologies they adopted are found to be similar. So it is
very possible to combine them to get a better solution for
multicasting in MPLS networks. Based on this concept and
the algorithms of ERM and AMFM, we proposed a new
strategy for MPLS multicast (FSMM). Our new proposed

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DOI 10.1109/ICCET.2009.138
strategy has both the characteristics and advantages of ERM
and AMFM, and furthermore, it can further reduce the
forwarding states and get a better performance of the
networks. The Fig. 1 shows an example of our proposed
FSMM, and in the following of the section, we will describe
our proposed strategy in detail based on Fig. 1.
A. Construction and Destruction of the Multicast Tree
In this paper, we mainly focus on the intra-domain
routing algorithm because inter-domain routing protocols
allow each autonomous domain to have its own multicast
implementation. And we assume that in MPLS domain,
Multicast group members are directly attached to LERs
(Label Edge Routers). Then our new proposed strategy tries
to construct a tree whose branching nodes are only located at
LERs. And in our new proposed scheme we adopt source-
based tree; that is means we construct a multicast tree per
source. Source-based tree has an advantage over core-based
tree in address allocation, since each source can freely pick
any address and create a unique (S, G) state. Besides, in our
new proposed strategy, we use centralized-control, which
means we select a dedicate node called ‘Multicast
Information Manager’ (MIM). The role of our MIM is very
different from the ‘Core’ or ‘RP’ in CBT and PIM. It
functions more like an information and managing server, and
is responsible for group membership management in an
MPLS domain. It keeps a record of current active on-tree
edge routers and returns a list of candidates to a new
receiver. The merits of centralized-control include easy
implementation and simplified routing algorithms. In our
strategy, we avoid using the flood-and-prune approach for
members to join and leave the multicast tree. Instead we use
explicit join and explicit leave. For the density of a multicast
group is likely to be sparse compared to the size of network.
Explicit join and leave would be more efficient in such
environment. Besides, flood and prune are traffic driven, not
control driven. When a multicast flow arrives at the edge of a
network, it needs to set up a tree first, only after which the
Ts
MIM
R1
T
R4
R7
(g0,g1,g2) S
Multicast Information Manager
Label Edge Router
Figure 1. Illustration of FSMM
flow assignment algorithm can be executed to map the flow
onto an LSP. This will increase the latency for the delivery
of the very first packet. So we use explicit join and leave.
Based on the information stated above, it is enough now
to construct a multicast tree. However, considering the
robustness of our strategy, it is still not perfect. So for each
multicast source, we construct two multicast trees at one
time. One is for present use, and the other is a backup tree. If
one link of the present tree breaks or fails for some reasons,
we can transfer the multicast flow to the backup tree and
destroy the failed tree. By doing so, our strategy is fault-
tolerant. And its robustness is largely developed.
Take Fig. 1 as an example, and as stated above, the
construction procedure of multicast tree can be described as
follows.
(1) Select a node as the MIM of the network, which can use
the same procedure adopted by the algorithms MMT
and PIM.
(2) The Source informs the MIM that he is the first active
branching node of the multicast tree.
(3) If a new LER, say E1, wants to join the tree, it sends a
QUERY message to MIM.
(4) MIM returns an ANSWER message with a list of
candidates, in this example, the candidates only include
S.
(5) Based on the own routing table, E1 picks the best
candidate, say S, as the join point.
(6) E1 sends a JOIN message to S, and S updates its routing
table by creating an outgoing entry for multicast state (S,
G).
(7) If successful, S informs MIM that E1 is now an active
on-tree edge router through an ADD message.
(8) MIM inserts E1 in the active member list.
(9) If there are still LERs who want to join, then repeat the
procedure between step 3 and step 8, until the whole
multicast tree is constructed
E2 (g0,g1,g2)
R2
R6
R5
R8
R3
E1 (g0,g1)
Label Switching Router
Backup Tree Ts Multicast Tree T
433
(10) As the multicast data being transmitted by the
established multicast tree, we construct a backup tree
using the same procedure as stated above. This time, if
a new member wants to join the tree, it must select a
different LSP from that of the previous tree to get to the
join point.
As depicted in Fig. 1, and by the procedure stated above,
we construct two trees for the multicast source S. They are
tree T includes LSP (S-R4-R5-R6-E2) and LSP (S-R7-R8-
E1), and tree Ts, Ts includes LSP (S-R1-R2-E2) and LSP
(S-R3-E1). An edge node will leave a multicast tree when
two conditions are met. First, it detects that there is no
active member directly attached to it by Internet Group
Management Protocol (IGMP) report. Second, it does not
have any downstream peer. The leaving node will send a
SUBTRACT message to the MIM to update the member
list, and a PRUNE message to its upstream peer. When
there is no multicast packets transmitted through the tree.
Then the MIM destroys the tree and the backup tree.
B. Group Aggregation and Fault-tolerance
In our proposed strategy, in order to further reduce
forwarding states, we will use group aggregation. That
means when a new multicast data flow arrives at an ingress
LER, we will not immediately establish a multicast tree for
it. We first check if there is already a existing tree that can
cover all the members of the new multicast session. If there
is one, then we will transmit it through the tree. And if there
is none, we will establish a new tree for it. In this way,
several multicast sessions can share one multicast tree. For
example in Fig. 1, we assume there are three multicast
sessions from source S--g0, g1, and g2. g0, g1 include E1
and E2, g2 includes E2. Then these three sessions can share
the same multicast tree T. What is more, we can see that we
limit the branching nodes at edge routers of the domain; in
this way, there is no need for the LSRs to store any
multicast states. As for the LSRs, they do not care whether
a packet is a multicast packet or a unicast packet. It only
needs to forward it according to its incoming label. So, the
forwarding states in the domain are largely reduced.
For fault-tolerance, we construct two trees. One for
present use and the other backup tree for future use if the
present tree fails. If a failure occurs in the present tree, then
the MIM informs the source S that, from now on, every
multicast FEC (Forwarding Equivalence Class) that goes
through the failed tree T will be moved to the backup tree Ts.
Meanwhile, MIM will establish a new multicast tree as the
backup tree. Then, MIM will inform all the branching nodes
of the failed tree to delete all the forwarding states they
stored and destroy that tree.
III. ANALYSIS AND EVALUATION OF THE FAULT-
TOLERANT STRATEGY FOR MPLS MULTICAST
Through the procedure we adopted to construct multicast
tree, we limited the branching nodes of the tree only at edge
routers. By doing this, many advantages can be gained. The
advantages of our proposed strategy are itemized as follows:
434
A. Simplifies LSP Setup
The multicast tree structure requires establishing point-
to-multipoint LSPs or even multipoint-to-multipoint LSPs. In
current MPLS architecture, only point-to-point LSP has been
addressed. MPLS does not exclude other type of LSPs, but
no mechanism has been standardized for this purpose. The
design of efficient multicast-enabled LSPs layout is still an
intriguing issue for researchers. In our proposed strategy,
since the diverging nodes of the tree are only located at edge
LSRs, there is no need to create and maintain point-to-
multipoint or multipoint-to-multipoint LSPs. Instead, a tree
can be decomposed and mapped to multiple point-to-point
LSPs. So in the core network of a MPLS domain, the LSRs
do not need to care whether a packet is a multicast packet or
a unicast packet. That means it is no need to establish point-
multipoint or multipoint-multipoint LSPs. So the setup of
LSPs is much simplified.
B. Reduces the Forwarding States of the Network
In our proposed strategy, each branch of a multicast flow
can be aggregated with other unicast flows which share the
same ingress and egress LERs. Thus the scalability of MPLS
traffic engineering will not be compromised. And by group
aggregation, several multicast sessions can share a single
multicast tree, so the number of multicast trees of the
network is much reduced. So in a word, the total forwarding
states is much reduced.
C. Relaxes the requirements of the core LSRs
One of the reasons that IP multicast is not widely
implemented is because of the fact that many core routers in
the backbone are not multicast ready. As the core routers are
usually carrying out critical missions, they are unlikely to be
upgraded off-line in the near future. So our approach is
designed in such a way that it poses little or no multicasting
restrictions on core routers. In our proposed strategy, core
LSRs are not involved in the whole procedure, for them, it is
the same as in a unicast MPLS network.
D. Better robustness
In our proposed strategy, when a failure occurs, the tree
manager invokes the failure recovery module, which first
detects which aggregated trees are affected. The recovery
module retrieves the backup trees of the failed trees, and
switches the related multicast groups to the backup trees.
Note that this tree-switching can be done very quickly and
efficiently since we only need to change the group-tree
matching entries at the edge of the network. In this way, our
proposed strategy is fault-tolerant. So the robustness of the
network is greatly improved.
From the analysis above, we can see that our proposed
strategy can simplify the setup of LSPs and makes multicast
flows aggregatable with unicast flows, which together with
group aggregation much reduced the forwarding states of the
network. And our strategy is fault-tolerant, so it has a better
robustness. In a word, through our proposed scheme, the
MPLS multicasting can get mangy advantages and gain a
better performance.
 IV. CONCLUSIONS
In this paper, we proposed an overview of a prominent
scheme for fault-tolerant multicasting in MPLS networks.
We identify the weakness of the previous schemes and
proposed a novel architecture, FSMM, for fault-tolerant
multicast provisioning. The idea is based on ERM and
AMFM. By our proposed strategy, LSP setup is simplified,
and the multicast flows can be aggregated with the unicast
flows in core networks. Together with group aggregation, the
forwarding states are much reduced. Besides, it relaxes the
work of the core LSRs, and enables them to do more critical
jobs. So the performance of the network is improved. What
is more, it is fault-tolerant, so we can get a better robustness
of the network. In a word, our proposed strategy can better
support multicast in MPLS networks.
For the future, we want to integrate QoS and MPLS-TE
(traffic engineering) considerations in our scheme to provide
a prominent multicast architecture. We believe that such an
435
architecture will be necessary to meet the requirements of
future networks for high performance and reliability.
REFERENCE
[1] E. Rosen, A. Viswanathan, and R. Callon, “Multi-protocol label
switching architecture”, IETF RFC3031, January 2001
[2] Rosen E, Viswanathan A, Callon R, “Multiprotocol label switching
architecture”, IETF RFC3031, 2001.
[3] Ooms D, Sales B, Livens W, et al. “Overview of IP multicast in a
multi-protocol label switching environment”, IETF RFC3353, 2002.
[4] Boudani A, Cousin B, “An effective solution for multicast scalability:
the MPLS multicast tree”, IETF Internet Draft, 2003.
[5] Yang B J, Mohapatra P, “Edge router multicasting with MPLS traffic
engineering”, 10th IEEE International Conference on Networks,
Mumbai India, 2002.
[6] Fei A, Cui J, Gerla M, et al. “Aggregated multicast: an approach to
reduce multicast state”, Global Telecommunications Conference, San
Antonio US., 2001.
[7] Cui J H, Faloutsos M, Gerla M, “An architecture for scalable,
efficient, and fast fault-tolerant multicast provisioning”, Network
IEEE, 2004
 2009 International Conference on Computer Engineering and Technology

A Novel Content-based Information Hiding Scheme

Jun Kong1, 2, Hongru Jia1, 2 , Xiaolu Li1, 2, Zhi Qi 1

1
Computer School, Northeast Normal University, Changchun, Jilin Province, China
2
Key Laboratory for Applied Statistics of MOE, China
E-mail: , kongjun@nenu.edu.cn, lixl605@nenu.edu.cn

Abstract data is decided with the number of wavelet

This paper presents a novel content-based hidden
transmission method for secret data to improve the
security and secrecy. In the proposed method, the
secret data is encrypted by chaotic map before
embedding. Then the cover image is segmented by
watershed algorithm and fuzzy c-means clustering.
At last we extract the feature of each region and
embed the secret data into the cover image
according to the result of feature extraction. Our
method can overcome the disadvantage of block-
based steganographic techniques. Experimental
results show that the security and performance of the
proposed scheme are high.
1. Introduction
With the rapid growth in communication
technology, more and more digitalized data are
distributed and transmitted over the Internet. For
these exposed data, security becomes one of the main
issues for the public channel. Many methods have
been proposed [1–2] to provide a way of data hiding
called steganography that embeds the secret data into
a cover carrier confidentially so that only the
intended receiver can detect and reveal the secret
message. In steganography, the quality of the stego-
image must be sufficiently high so as to make it
visually indistinguishable from the cover image, so
that an unintended observer will not be aware of the
very existence of the secret data, thereby maintaining
the secrecy of the embedded information.
Many block-based steganographic techniques
have been proposed in the literature [3-5]. Wang et al.
[3] proposed an image hiding method based on best-
block matching and k-means clustering, the method
splits the entire secret image into multiple non-
overlapping blocks. For each block, they search for
the best similar block from a series of numbered
candidate blocks that are generated from the cover
image. Kang et al. [5] divide an image into blocks of
prescribed size, and each block is decomposed into
one-level wavelet. The capacity of the hidden secret
coefficients of larger magnitude.
By these block-based methods, the cover image is
divided into a number of fixed non-overlapping
blocks. Then the feature of each block is extracted
and the secret data is embedded into the cover
images according to the result of feature extraction.
However, the main drawback of this technique is that
it doesn’t consider the content of the cover image.
There may exist different features in one block. For
example, one block can include some smooth areas
and some high texture areas, but when we calculate
the features for this block, the block is treated as a
whole to deal with. So the block-based segmentation
method is inaccurate. If the blocks are too small, it
can’t reflect the feature of an area.
To overcome this drawback, a novel content-
based hidden transmission scheme is present. First,
the secret data is encrypted by chaotic map and the
cover image is segmented into different regions.
After that, each region includes different content.
Then, the feature of each region is extracted by
calculating the entropy. At last, the secret data is
embedded into the cover image according to the
result of feature extraction.
The rest of this paper is organized as follows. In
Section 2, we shall briefly introduce the. The
proposed image hiding method is described in
Section 3, and experimental results are presented in
Section 4. Finally, conclusions are drawn in Section
5.
2. Relative technique
The basic ideas of the techniques used in this
paper are described briefly and the related work of
each topic is introduced.
2.1. Chaotic map
In mathematics and physics, chaos theory deals
with the behavior of certain nonlinear dynamic
systems that under certain conditions exhibit a
phenomenon known as chaos, which is characterized

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DOI 10.1109/ICCET.2009.167
by sensitivity to initial conditions. These special
characteristics make chaotic maps excellent
candidates for data hiding and encryption to increase
the security [6]. Chaotic signals, which are complex
and impossible to predict over long time, can be
generated by simple dynamic systems such as
logistic map and chebyshev map. In our scheme,
chaotic map is used for encrypting the watermark. It
provides the following features: (1) resistant to the
finite world length effect of the chaotic sequence, (2)
very unpredictable, and (3) resistant to repeated
group attack.
2.2. Watershed algorithm
Watershed algorithm is a widely used approach
for image segmentation [7], which is based on
immersion simulation: the topographic surface is
immersed from its lowest altitude until water reaches
all pixels. The output of watershed algorithm is
segmentation of cover image into a set of non-
overlapping regions. The length of the gradients is
interpreted as elevation information. During the
successive flooding of the grey value relief,
watersheds with adjacent catchment basins are
constructed. This flooding process is performed on
the gradient image, i.e. the basins should emerge
along the edges. Normally this will lead to an over-
segmentation of the image.
2.3. Fuzzy c-means clustering
The fuzzy c-means clustering (FCM) algorithm
assigns pixels to each category by using fuzzy
memberships. Let X = {xi , i = 1, 2,…, N | xi ∈ R d } denote
an image with N pixels to be partitioned into
c classes (clusters), where xi represents feature data.
The algorithm is an iterative optimization that
minimizes the objective function defined as follows:
c N
J m = ∑∑ ukim xi − vk
2
(1)
k =1 i =1
where uki represents the membership of pixel xi in
the k th cluster, vk is the k th class center, ⋅ is a
norm metric, m > 1 is a weighting exponent on each
fuzzy membership. The parameter m controls the
fuzziness of the resulting partition. The cost function
is minimized when pixels close to the centroid of
their clusters are assigned high membership values,
and low membership values are assigned to pixels
with data far from the centroid. The membership
function represents the probability that a pixel
belongs to a specific cluster. In the FCM algorithm,
the probability is dependent solely on the distance
between the pixel and each individual cluster center
437
in the feature domain. The membership functions
and cluster centers are updated by Eq. (2).
∑
N m
1 u x
uki = and vk = i =1 ki i
(2)
xi − vk 2 ( m −1) ∑
N m
∑ l =1 ( x − v ) u
c
i =1 ki
i l
In implementation, matrix V is randomly
initialized, and then U and V are updated through an
iterative process using Eq. (2).
3. The proposed method
3.1. Secret data encryption
In our scheme, the secret data is grayscale image
which size is 128 × 128. Before encrypting, secret
data is changed into a binary stream I (n) ∈ {0,1} . For
the encryption, as described in section 2.1, we use
logistic map to generate 1-D sequence of real
numbers that is used as a sequence key. The
following equation shows the logistic chaotic map:
yn +1 = g ( yn ) = μ yn (1 − yn ) (3)
where n = 1, 2,3… is the map iteration index and μ is
the system parameter, for 3.57 < μ ≤ 4.0 , the
sequence is nonperiodic, non-convergent, and very
sensitive to the initial value. The secret keys μ and
y0 are generated randomly as the parameter value
and initial condition of the logistic map, respectively.
The dynamic secret key generates a sequence of
different pseudorandom numbers at each transaction
session, so our chaotic encryption module can resist
the statistical characteristics of plaintext and
ciphertext attacks as mentioned in [8].
Because the stream generated by logistic map is a
sequence of real numbers, the output of the logistic
map in Eq. (3) is quantized into binary stream by the
following threshold:
⎧1, yn ≥ 0.5
f ( n) = ⎨ (4)
⎩0, yn < 0.5
So the normalized sequence created by logistic
map is f (n) ∈ {0,1} and the secret data is changed
into binary stream I (n) ∈ {0,1} . XOR operation is
used to encrypt the secret data, and the encrypted
data is obtained by
w(n) = f (n) ⊕ I (n) (5)
Figure.1 shows the result images after encrypting
the secret data.

(a) (b)
Figure 1. (a) The secret data, (b) The encrypted
secret data.
3.2. Embedding process
In this section the standard 256 × 256 color images
are used as cover images. Before embedding, we
convert RGB image to grayscale and segment the
cover image into different regions using watershed
algorithm. The output of watershed algorithm is
segmentation of cover image into a set of non-
overlapping regions. Figure. 2(b) demonstrates the
watershed result.
(a) (b)
Figure 2. (a) The cover image, (b) Partition result
using watershed algorithm
The watershed transformation constitutes one of
the most powerful segmentation tools provided by
mathematical morphology. But there exist a
disadvantage in the watershed algorithm. The result
of classical watershed algorithm on images is often
over-segmentation, as shown in Figure. 2(b). The
next step, we solve the over-segmentation problem
by merging the over-segmentation regions. The
Fuzzy C-Means (FCM) cluster has been applied
widely to image segmentation [9], and regarded as
one of the most promising methods. In our work, the
output of watershed algorithm is the segmentation of
cover image I G into a set of non-overlapping regions
denoted by Ri , i = 1, 2,… , n, where n is the number of
regions. To implement the merging of similar regions,
we use the region-based Fuzzy C-Means (RFCM)
clustering. The mean value, denoted by
mi , i = 1, 2,…, n of each region Ri , is needed. The
RFCM clustering algorithm in this paper is
formulated as:
c
J RFCM (U , v ) = ∑∑ uikm mi − vk
2
i∈Ω k =1
438
c
and ∑ uik = 1, Ω = {1, 2, … , n} . (6)
k =1
where the matrix U = {uik } is a fuzzy c-partition of I G ,
and uik gives the membership of region Ri in the
k th cluster ck , c is the total number of clusters and
set to 4 in our study. v = {v1, v2 , v3 , v4 ,} is the set of fuzzy
cluster centroids, m ∈ (1, ∞) is the fuzzy index. The
result image after using RFCM is shown in Figure.
3(a).
(a) (b)
Figure 3. (a) The result image after using RFCM, (b)
The result image without watershed lines.
After clustering, each of the region is classified
into a certain cluster, but the regions of watershed
lines. So we need to remove the watershed lines by
re-segmenting these regions. In this paper we exploit
the classical k nearest neighbor (KNN) classifier to
partition the regions needed re-segmentation. For
each data point to be classified in re-segmented
regions, KNN compute this point’s closest k training
samples in feature space. Then the data is classified
with the label most represented among these
k nearest neighbors. We choose k = 7 in the
experiments presented in this paper according to the
suggestion which Enas given in [10]. Figure. 3(b) is
result image without watershed lines.
We calculate the entropy of each region and
embed the secret data into cover image according to
the entropy values. Entropy is a statistical measure of
randomness that can be used to characterize the
texture of the input regions. Low entropy regions,
such as those containing a lot of black sky, have very
little contrast and large runs of pixels with the same
or similar pixel values. On the other hand, high
entropy regions such as an image of heavily cratered
areas on the moon have a great deal of contrast from
one pixel to the next. Entropy is defined as:
L −1
Entropy: e = − ∑ p( z ) log ( z )
i =0
i 2 i (7)
where zi is a random variable indicating intensity,
p(z ) is the histogram of the intensity levels in a
region, L is the number of possible intensity levels.
The color cover image is decomposed into three
sub-images indicated as R, G and B channels. The
embedding strength is based on the characteristic of values are higher than 46.8 dB, which is indicative of
Human Visual System which gives the following the high quality of the stego-images. Figure. 4 shows
rule: Disturbs are much less visible in highly the resulting images by content-based steganographic
textured regions than in uniform areas. So we embed scheme.
more information in highly textured regions than in
uniform regions. A threshold T is used to partition Table 1. The PSNR values of stego-images.
the two categories. If the entropy value of a region is Cover images PSNR
lower than T, the binary streams of the secret data is Lena 47.11
then inserted into the 2 least significant bit (LSB) F16 49.73
planes of the three channels of the region. If the Sailboat 47.82
entropy value of a region is higher than T, we embed
Pepper 46.87
4 bits into the 4-LSB planes of the three channels.
For each region, a flag is set to indicate the texture
feature. In highly textured regions, the flag is set to 1;
otherwise, it is set to 0. The locations and flags of
each region are as the secret key of the scheme.

3.3. Extraction process

At the receiver, the extraction process is the (a) (e)

reverse of the embedding process. For a stego-image,
the same secret key used during embedding is now
used to extract the secret data. For each region, if the
flag is 1, we extract 2-LSB planes of the three
channels; if the flag is 0, we extract 1-LSB plane of
the three channels. The original secret data is
obtained by combining the binary streams to an
(b) (f)
image.
We generate a chaotic sequence by the same key,
which was used in section 3.1. At last the XOR
operation is applied to decrypt the extracted secret
data, so the original secret data is obtained.

4. Experimental results
(c) (g)
In this section, we show some experimental
results to demonstrate the visual quality of our
proposed hiding method. The peak signal to noise
ratio (PSNR) is used to estimate the visual quality of
the stego-images, which is defined as follows:

M × N × max( I 2 (i, j )) (d) (h)

PSNR = 10 × log10 (
∑ i −1 ∑ j −1[ I (i, j ) − I ' (i, j )]2
M N
where I (i, j ) and I ' (i, j ) are the grey-scale values of
the cover image and the stego-image respectively.
Generally spearing, a larger PSNR value means a
smaller difference between the steto-image and the
original cover image. However, the difference is
unnoticeable by the human eye when the PSNR
value is higher than 30dB.
In our experiments, 4 cover images, “Lena”,
“F16”, “Sailboat”, “Pepper”, 256×256 each, are used
to check out the performance of the content-based
steganographic scheme. The PSNR values of various
stego-images are listed in Table 1. All of the PSNR
) (8)
Figure 4. Cover images (a)-(d) and stego-images (e)-
(h).
In experiment we attempt to hide more
information into the cover image to achieve overall
payload of each images. Table 2 shows that the
PSNR values of the cover images and the stego-
images are also high. We conducted experiments
using image “F16” and provide comparisons with the
block-based algorithm. For block-based algorithm,
the image is divided into a series of blocks with size
8 × 8 and the secret image is embedded into the cover
image in the same way with our scheme. Table 2
summarizes the performances between the two
methods. The fact is that the block-based method can

439
generate the stego-image with higher PSNR than our
method, but our scheme highly improves the
embedding capacity.
The threshold T plays a dominating role in
deciding the payload and the stego-image quality for
each image. Furthermore we test all images by using
various threshold values T where T=10,20,30,…,100
percent of the largest entropy value, respectively.
Figure 5 shows the payload capacity and PSNR
obtained by using different threshold T. In order to
maintain better stego-image quality and higher
hiding capacity, the threshold value T is set to be
seventy-five percent of the largest entropy value. In
deed, no difference can be perceptually told between
the original cover images and the stego-images.
Figure 5. The payload capacity obtained with
different threshold T.
Table 2. The comparison between the two methods
block-based method
Stego-
images Capacity(bits) Capacity(bits/pixel) PSNR(dB)
Lena 586369 2.98 31.54
F16 515755 2.62 35.62
Sailboat 608754 3.09 32.51
Pepper 597889 3.04 31.78
[4] Chih-Chiang Lee, Hsien-Chu Wu, “Adaptive
lossless steganographic scheme with centralized
difference expansion”, Pattern Recognition 41 (2008)
2097 – 2106.
[5] Kang Zhiwei, Liu Jing and He Yigang,
“Steganography based on wavelet transform and
modulus function”, Journal of Systems Engineering
and Electronics, Vol. 18, No. 3, 2007, pp.628–632.
[6] Zhang J, Lei T, Tai HM. “A new watermarking
method based on chaotic maps”. In: Proc IEEE
ICME’04, Taiwan, 2004.
[7] Luc Vincent, Pierre Soille, Watersheds in Digital
Spaces: An Efficient Algorithm Based on Immersion
Simulations, IEEE Transaction on Pattern Analysis
And Machine Intelligence, vol 13, No 6, (1991).
440
The payload capacity of the images “Lena” and
“pepper” are higher than “F16”and” Sailboat”, it
indicates that the payload of each image is different
according to the complexity of the cover image. The
higher the complexity of cover image, the higher the
payload.
5. Conclusions
This paper presents a novel content-based hidden
transmission method of secret data to improve the
security and secrecy. Our method can overcome the
disadvantage of block-based steganographic
techniques. Experimental results show that the
security and performance of the proposed scheme are
high.
10. References
[1] C.C. Chang, J.Y. Hsiao, C.S. Chan, “Finding
optimal LSB substitution in image hiding by
dynamic programming strategy”, Pattern
Recognition 36 (7) (2003) 1583–1595.
[2] Y.C. Tseng, Y.Y. Chen, H.K. Pan, “A secure data
hiding scheme for binary images”, IEEE Trans.
Commun. 50 (2002) 1227–1231.
[3] Ran-Zan Wang, Yao-De Tsai, “An image-hiding
method with high hiding capacity based on best-
block matching and k-means clustering”, Pattern
Recognition 40 (2007) 398 – 409.
Our method
Capacity(bits) Capacity(bits/pixel) PSNR(dB)
644538 3.28 30.83
561025 2.85 35.95
615937 3.12 31.79
670261 3.41 30.14
[8] Alvarez G, Montoya F, Romera M, Pastor G.
“Cryptanalysis of a chaotic encryption system”. Phys
Lett A 2000; 276:191–6.
[9] Bezdek J., Hall L., Clarke L, “Review of MR
image segmentation techniques using pattern
recognition”. Med. Phys. 20 (4), (1993)1033–1048.
[10] Enas, G., Choi, S., 1986. “Choice of the
smoothing parameter and efficiency of k-nearest
neighbour classification”, Computers and
Mathematics with Applications 12A (2), 235-244.
 2009 International Conference on Computer Engineering and Technology
Ambi Graph: Modeling Ambient Intelligent System
K.Chandrasekaran,I.R. Ramya, Syama .R
National Institute of Technology Karnataka (NITK), Surathkal.
kch@nitk.ac.in, ir.ramya@gmail.com, syamar@gmail.com
Abstract - In computing, ambient intelligence (AmI)[2,3]
refers to electronic environments that are sensitive and
responsive to the presence of people. In an ambient
intelligent world, devices work in concert to support
people in carrying out their everyday life activities,
tasks and rituals in easy, natural way using information
and intelligence that is hidden in the network
connecting these devices. The ambient intelligence
paradigm builds upon ubiquitous computing [4] and
human-centric computer interaction design. In this
paper, we introduce a notation, called Ambi Graphs, for
specifying the work flow in any ambient intelligence
system. An ambi graph is elegant when it is designed to
adapt to the wide range of users that work on it.
Keywords-Ambient intelligence, sensor, surrounding,
target, sensing, interpretation, action, element diagram,
computational diagram, ambi graph.
I. INTRODUCTION
Having a well-defined and expressive notation is
important to the process of any kind of development.
Standard notations make it possible for describing a
computing scenario or developing device architecture
and then unambiguously communicate these
decisions to others. A notation is a vehicle for
capturing the reasoning about the behavior and
architecture of a system. It is wholly a very well
defined approach which does not involve the use of
state diagrams or object diagrams. Characteristics of
ambient intelligence environments differ from
traditional ones due to various factors, like merging
with the surroundings, acting in response to a set of
sensed objects, and so on. A generalized approach to
represent every ambient intelligence system has to be
self-explanatory to every user. We find that the
notations reported in literature, such as State
diagrams and object diagrams are not sufficient to
describe an ambient intelligence system to its fullest.
A State diagram [1] is used to show the state space of
a system, the events that cause a transition from one
state to another and the actions that result from a state
change. Here we have extended the State diagrams
formally as Computational diagrams, that have the
same features of State diagrams and some more
features to model the characteristics to give a
generalized approach. Object diagrams [1] model the
dynamic behavior of an object in its lifetime. Here
we have used a notation called Element diagram to
specify real-time systems. The notation uses the
object diagram’s representations but with a much
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DOI 10.1109/ICCET.2009.203
sophisticated approach. In this paper, we introduce a
notation, called Ambi Graphs, for specifying the
work flow in any ambient intelligence system.
Though elements can be thought of as building
blocks to model ambi graph, element diagram
formalism does not provide mechanisms to model the
typical features of ambi graph. Some important
ambient intelligence device features are specifying
element location, properties, types of sensors used,
adaptability, etc. Currently, element diagrams do not
have clean mechanisms to specify the above features.
In this work, we extend the specification of an
element with a describer called tasks. Now each
element can independently describe its traits and
tasks completely. Using ambi graph, users can design
ambient intelligent devices by understanding how
exactly they want the device to function.
What is a notation and why it is used
A standard notation, that unambiguously expresses
different aspects of a system, is important to the
process of software development. The expressiveness
of a standard notation helps analysts and developers
to describe a computing scenario or to formulate
software architecture and then to communicate these
decisions unambiguously to other team members. An
expressive notation eliminates much of tedium of
checking the consistencies and correctness of the
architectural decisions of the developer by using
automated tools.
What is ambient intelligence?
Ambient Intelligence is a vision of a world where we
are surrounded by a huge amount of intelligent and
small devices, which are seamlessly embedded in the
user's environment. It is a vision on the future of
consumer electronics, telecommunications and
computing that was originally developed in the late
1990s for the time frame 2010–2020. A key to the
process of realizing an ambient intelligent system is
defined using the following stages:
• Sensing
• Interpretation
• Action
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II. AMBI GRAPH
The following three models are used to define an
ambient intelligence system. Element graph- one that
describes all the elements present in the model;
Computational graph- describes various
computational techniques used and the sequence in
which the system proceeds with various inputs; Ambi
graph- diagram that shows the actual flow of data
element graph and the computational graph to
describe the system as a whole.
Element Graph
In an ambient intelligence system (device), there has
to be a classification of different elements which
constitute not only the device but also the
surroundings. Here, surroundings play a vital role
because most of the existing ways of representation
has description of only how the device works. By
describing surroundings, we ensure that the changes
that are sensed are clearly monitored. In the element
graph we describe the various elements which
constitute any ambient intelligence device. The major
elements are as follows:
• Environment
• Sensor
• Target
To name each element, we use a semi-circular block.
This is placed over two other rectangular blocks. One
of which consists of the traits of the element and the
other, the tasks. Traits and tasks are used to describe
the two features of the elements- its characteristics
and its mode of operation related to it.
We connect the elements with arrows to show the
work flow. This is a trivial work-flow model. Here
we enhance on the traits and tasks rather than the
links from/to various elements. Since this approach is
dynamic, we express the traits and tasks within each
element as shown in Fig.1. Modification of data is
possible at any stage of development. It has to be
reflected in all the dependent models. The following
diagram is a generalized view of all the elements
mentioned above.
Figure1. Element Graph
442
Let us consider the very famous home intelligence
systems. When somebody enters the hallway, the
lights automatically get turned on. When they sit on
the couch, the coffee maker starts making coffee, etc.
Here our surrounding is the room. The possible traits
of this room could be the presence or not of anybody
around. Based on this, the task is the turning on or off
of the lights. If we consider the count of people as the
trait, the task could be how many people to prepare
coffee for.
Similarly, for sensor and target also we can have
traits and tasks. These are based on the computation
involved with respect to the entity that is sensed.
These shall be discussed later so as to provide a clear
idea because it follows the computations involved in
each step of sensing.
Computational Graph
Any intelligent system is known so for it's decision
making ability. We have a particular form of
representing our computations with details such as
what events and corresponding actions are
performed, the current state of the device, the
conditions which act like a split-road. So we
represent all these details in a very sophisticated form
in Fig.2.
Figure 2:Computational diagram
Each state is represented in the form of block arrows.
The upper half describes the event and action. The
lower half specifies the state name. Both, event and
action are separated by a ‘/’. Event is the occurrence
of a particular process. Action is the process of doing
something that can lead us to the next state.
Sometimes an ambiguity occurs when we have to
decide which the next state has to be based on a
specific condition. In such cases we use utility
functions that maps each possible state of a system or
object to a real scalar value on a common scale.
Utility function is specified in the circle. The value
returned by the utility function is written in a box at
the beginning of the next state. The start state and
final state are marked with an asterisk (*) and a hash
(#) respectively.
We use a policy based approach that separates the
policy from the implementation of a system and
therefore permit the policy modified without
changing the systems underlying implementation. In
the model being described we will have two policies
a low level policy and an efficiency level policy. A
low level policy basically describes the
computational model at a very abstract level. At the
efficiency level, the computational model described
is very specific. This approach helps maintain the
hierarchy. As ambient intelligence is a technique
used widely in the armed forces there should be
check on what is available to whom. This kind of a
policy based approach ensures that critical
information does not fall in the wrong hands.
Getting back to the home intelligence system, if
somebody sitting on the couch is the event and
making coffee is the action. If we had to choice
between making coffee and juice, we need a
condition. This condition could be sensing the
temperature. In on a clod day, the coffee maker can
serve us and on a hot day, nice cold juice could be
heaven!
Ambi Graph
The ambi graph is the core of any ambient intelligent
system. It shows the actual step by step process that
takes place. By making use of the element chart as
well as the computational chart the ambi graph gives
a very precise model of ambient intelligence. It is
very important to have such a model as the elements
and the computations alone cannot lead to the
understanding of a system. The ambi graph acts as a
container which uses both these models to give a
complete description. The model may roughly be
divided into two parts:
• Elements
• Computations
443
Figure 3. Ambi Graph
In the above diagram environment, sensor and target
are the element graphs specified by rectangular
boxes. The element diagram is specified inside the
rectangle. The computational model is specified
inside the rhombus. The arrows show the flow of data
between the element graphs and the computational
graph. Data received from elements or as a result of
certain computation is written on the arrows. This
allows the programmer to have a clear picture of the
flow of data along with all the processes and
transitions taking place in the system. The
programmer gets a better view of the whole scenario
and the things he has to take care of at any given
point of execution of the system.
The ambi graph has two forms a private form and a
public form. In the private form all the details are
described and the total workflow is represented.
Whereas in the public form only the outer picture or
abstract picture is represented.
In the case of our intelligent home, the turning on or
off of lights depends on certain data such as presence
or absence of people in the room. Similarly the
amount of coffee that has to be made depends on the
count of people. This data is represented on the arrow
headed lines in the above mentioned model. This data
is processed by the computational model which again
leads to a new piece of data leading to another
element.
III. CASE STUDY
WORKING OF A TREADMILL
As the famous saying by Albert Einstein goes,
“Example isn't another way to teach, it is the ONLY
way to teach." Treadmill is a device used to exercise.
It consists of a continuously moving belt. The
treadmill is programmable regarding simulated
course, distance, initial speed, and user weight, and
records approximate caloric burnt. Sensors
strategically positioned below the belt upper surface
regulate belt speed to keep a user toward the center of
the treadmill.
The treadmill works as follows. The upper surface of
the revolving belt upon which a user performs is
equipped, on the underside, with three sensors. The
sensor nearest the front of the belt is an acceleration
sensor. The acceleration sensor causes the speed of
the rotating belt to increase. Conversely, the
deceleration sensor slows the belt speed. The target
speed sensor maintains belt speed. Through these
mechanisms, the present invention automatically
adjusts to the momentary speed of the person,
through any course. No impact on the sensors stops
the treadmill.
Element diagram
adjusts to the momentary speed of the runner or
walker, through any course. No impact on the sensors
stops the treadmill.
Drive Motor
The driver/controller is itself controlled through
programmable circuitry having a display and

In a treadmill, we have identified the following keyboard with user input options. The controller
elements with respect to our modeling concepts. receives signals from a speed sensing device attached
They are as follows: to the AC induction motor to maintain the rotational
speed of the AC motor within pre-selected limits. The
• Acceleration sensor AC induction motor is attached to one or more
• Deceleration sensor flywheels and directly engages, through a drive
• Target speed sensor roller, a walking belt. The matched combination of an
• User AC motor driver/controller with appropriately sized
• Belt flywheels allows utilization of a variable speed AC
• Drive motor
induction motor for the direct drive of the treadmill
belt.
Acceleration sensor
It is placed below the belt. The Belt
acceleration sensor causes the
The treadmill has an endless belt
speed of the rotating belt to
entrained around a drive roller
increase when the belt is
and an idler roller. The belt has a
compressed against it by a user's
first or upper reach and a second
foot. An increase in speed of the belt thereby carries a
or lower reach extending
user back toward the midpoint of the belt upper
between the rollers. The belt has
surface, the most desirable position for a user to be
an outer surface and an inner surface. The endless
in.
belt encircles a support deck so that a user positioned
Deceleration Sensor on the outer surface of the upper reach is supported
The deceleration sensor is by the top side of the deck. The inner surface of the
disposed below the belt proximal upper reach is in a sliding relationship with the deck.
to the rear of the drive assembly
User
of the treadmill. When the belt is
Each user has different
caused to be compressed against
properties. For example, the
the rear sensor by a user's foot
weight of each user differs and so
strike, the belt communicates that information to the
does the walking speed. Certain
CPU which in turn slows the belt speed, providing
treadmills give an option to have
for the user to return to the midpoint of the rotating
different profiles for each user.
belt.
The amount of calories burnt is also kept a record of.
Target Speed Sensor Depending on the user’s position on the belt, the
The target speed sensor is speed of rotation of the belt varies.
disposed midpoint below the belt
Computational Graph
upper surface. The target speed
sensor maintains belt speed, as the In the computational graph of the treadmill we
user is ideally positioned when in describe the various computations and decision
the center. Through these makings taking place in the treadmill.
mechanisms, the present invention automatically

444
First let us see the low level policy based
computational graph which gives an overview of the
working without going into the intricate details. In
Fig.4 we describe where each sensor comes into
picture without giving an actual description of the
internal working and the use of motors.
Figure 4. Low level computational graph
In Fig.5, the efficiency diagrams, we look at each
sensor separately and show the actual computations
and interactions with the driver motor and the belt.
The acceleration sensor increases the speed of the
driver motor if the pace at which the user is moving
is greater than the pace of the belt. This is done so as
to make the user move towards the centre of the belt
where his target speed is calculated. Similarly the
deceleration motor slows down the driver motor if
the user is walking too slowly.
Figure 3. Efficiency level computational graph
Ambi Graph
Treadmills are used by different category of people.
Depending on the class and understanding required
by them, we propose two types of ambi charts. For
the high level a private one and for a comparatively
lower level a public one. Below is the complete
description of the two types.
445
Private Ambi Graph
We make use of a private ambi chart in order to make
every process visible to the user. This solely depends
on the kind of user. This level is preferred for the
advanced level users.
Figure 3. Private ambi graph
Public Ambi Graph
The public level is used for the reference of a normal
user. At this level only the basic knowledge of how a
treadmill works is required.
Figure 3. Public ambi graph
CONCLUSION
We have tried to give a notation for ambient
intelligence systems above. Such a system is very
useful in the analysis and planning stages. It provides
a very clear understanding of the system describing
all the functionalities and properties of it. It is also a
form of documentation for the system. Using such a
notation makes it easier for the programmer to design
the system.
REFERENCES
[1] Satyajit Acharya Hrushikesha Mohanty and R.K.
Shyamasundar “Mobicharts: A Notation to Specify Mobile
Computing Applications”
[2] Orit Zuckerman “Interactive Portraiture: Designing Intimate
Interactive Experiences”
[3] Paolo Remagnino Gian Luca Foresti “Ambient Intelligence: A
New Multidisciplinary Paradigm”
[4] Anand Raghunathan, Srivaths Ravi, Sunil Hattangady, and
Jean-Jacques Quisquater “Securing Mobile Appliances: New
Challenges for the System Designer
 International Conference
on Computer Engineering and Technology

Session 7
 2009 International Conference on Computer Engineering and Technology
Research on Grid-based Short-term Traffic Flow Forecast Technology
Wang Xinying Juan Zhicai
College of Computer Institute of Transportation
Science and Studies, Antai College of
Technology Economics & Management
Jilin University Shanghai Jiao Tong
Changchun, China University
xinying@jlu.edu.cn Shanghai, China
zcjuan@sjtu.edu.cn
Abstract
Short-term traffic flow forecast is an important
topic in the research field of intelligent transportation
systems. The article analyses the preliminary results in
the short-term traffic flow forecast, takes full
advantage of the characteristics of grid technology,
and builds a model based on chaos theory and neural
network. It uses grid resource management
mechanisms and migration strategies to solve issues of
effectiveness, and proposes the algorithm of minimum
price for Self-Governing Domain on the aspect of
resource management to improve the efficiency and
performance.
Short-term traffic flow forecast; grid; MPSGD
1. Introduction
With the continuous development of urban
transportation, short-term forecasting for urban traffic
flow becomes one of the most important issues of
Intelligent Transportation System (ITS). It is the
premise of and key to that of the realization of traffic
guidance and control [1]. Short-term traffic flow
forecast means real-time forecasts for the next time
t +Кt (Кt less than 15min), and even some time after
it, based on the information collection at the time point
t. Decision-making, according to this forecast, can
guide and control the traffic flow migration, coordinate
and balance the traffic flow, which plays an important
role in alleviating traffic jams. With regard to real-time
performance requirements, the typical cycle of traffic
control and guidance is usually to be 5min, so it is
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.64
Liu Xin Mei Fang
Network Center College of Computer
Jilin University Science and
Changchun, China Technology
lx@jlu.edu.cn Jilin University
Changchun, China
meifang@jlu.edu.cn
critical that accurate forecasts of traffic flow are made
within 5 minutes.
In this paper, traditional forecast model and
nonlinear dynamics theory with characteristics of the
short-term traffic flow are analyzed. Then a new
method is raised, that is the accuracy of the forecast is
improved based on the combination of Chaos theory
and neural networks is used to. At the same time, the
effectiveness of algorithm is solved by using of grid
resource management mechanism and the task of
migration strategy.
2. Short-term traffic flow forecast model
There are many short-term traffic flow forecast
models. According to principles, they can be divided
into four categories [2], the first category is based on
the analysis of mathematical forecast models,
including the History Average Model, Linear
Regressive Mode, Time Serial Mode, Kalman Filtering
Model, Maximum Likelihood Formulation Model, and
so on. In this kind of models, it is usually used
analytical mathematical model to describe the state
variable traffic trends, which is built on mathematical
statistics, and based on the assumption that the future
traffic data and historical traffic data has the same
characteristics. These items are simple models with
poor accuracy, especially not suitable for the section
with larger changes in the traffic flow. The second
category is knowledge-based intelligent model,
including non-parametric regressive forecast model
and neural network models, knowledge-based
intelligent model points to the view of the non-linear
and uncertainty characteristics in the transportation
systems, makes fits with the real traffic flow data, then
forecast traffic flow at the next time. Such methods of
449
forecasting are better, but with a large amount of
calculation. The third category is based on the
simulation of traffic, including model based on traffic
simulation and model based on dynamic traffic
distribution. These models realized the traffic
simulation, which is an important tool for the traffic
analysis, but with non-real-time characteristics.
Generally speaking they are used for the evaluation of
the existing models. The fourth one is the combination,
which combines these forecast models, enhances
strong points and avoids weaknesses, in order to
achieve better forecast results. This category earns the
catholic concern.
Transport system is essentially a giant complex
system, which is full of nonlinear interactions. And
traffic flow has the natures, such as high degree of
complexity, uncertainty and randomness. Therefore,
the traditional single forecast technology can no longer
meet the rising accuracy. As a matter of fact, the traffic
flow time series has the characteristics not only cycle
and trend, but also chaos. Chaos refers to a seemingly
random movement, random acts taking up in certain
nonlinear systems without any additional random
factors (intrinsic randomness). Forecast algorithm
based on Chaos theory requires large data sample [3],
resulted in the eager aid of computer in modeling. Now,
people try to set up integrated models based on chaos
theory and neural network, which makes space
reconstruction of the time series by using chaos theory,
then forecast with neural network. It has been proved
that [4], higher forecast accuracy can be achieved by
the models based on Chaos theory and neural network.
3. Enhance real-time performance by using
grid
It has advantages in principle, and is feasible in
practice, that integrated model based on chaos theory
and neural network is used in short-term traffic flow
forecast. However, it is real-time that the thorny
problem need to be solved. The algorithms of chaos
theory are so complex, that it will cause delay when
using it to analyze the traffic flow time series. It has
made effective that by improving the forecast
algorithm [5], building a new structure of the neural
network model [6], the introduction of genetic
algorithm [7] or other methods to improve the
effectiveness of the algorithm. But none can improve
the forecast efficiency fundamentally.
According to the model's basic theory, both the
forecast method based on neural network and based on
chaos theory are needed to deal with a lot of real-time
data in a limited time, that results in a higher demand
450
for computer processing power and calculation
strength. No single device can afford it. And it will
lead to high input costs, if a large number of hardware
upgrades. So this article brings about the method of
using grid technology to make full use of idle
computing resources in grid when computing, and to
solve effectiveness problem of the algorithm by using
grid resources management mechanisms and task
migration strategies.
Grid is an integrated computing and resource
environment, which integrates the recourses in
computer network into a huge supercomputer to realize
of the full range of access and sharing. To achieve high
performance computing and sharing of heterogeneous
grid resources, there must be a unified and effective
resource management mechanism for the network to
find and use the idle resources, share the computing
tasks of short-term traffic flow forecast. By conclusion
the resource management mechanisms [8], it takes
advantages of P2P and MDS, proposes the resource
management mechanism based on the minimum price
of self-governing domain (MPSGD), as shown in
Figure 1.
GGN1 SGD3
Node31 Node32
Grid System
GGN2 SGD1 GGN3 SGD2
Node11 Node12 Node21 Node22
Figure1. Structure of Minimum price SGD
Grid system is divided into some logically separate
autonomous regions, and each region has a dedicated
node, GGNi, which is in charge of grid resources
information services (GRIS) and grid directory
information service (GIIS). Each node in the
autonomous region i should obey active report strategy,
report its own state to the other ones periodically,
discover the neighbor’s corresponding information,
organize it in special data structure, and post it to the
service node GGNi. The GGNi in every domain should
not only store the information of SGDi, but also
change own information with other GGNi node
periodically, so that the whole system can provide
available resources information sheet. When the local
node need to migrate a task, it may only send a request
to the local services node GGNi directly. This method
has two strong points. One is that the discovery
mechanism is better than MDS, and the other is that
there is no duplicate data storage, reducing system
overhead.
In the division of regional autonomy, using a
minimum price law, nodes which have lower migration
cost will gathered in one domain. When it is too large
for a node to finish a computing task itself, it will ask
the GGNi for help. If there is a certain node in SGDi
which is lighter on load ,after query, migration will be
on. If there is no, the one with lighter load in other
SGD will be the successor. This method can ensure
that the task successfully finished with low cost..
4. Conclusion
It compares several models in the field of traffic
flow forecast. According to request of the real-time
and high accuracy, it draws a conclusion that the
model based on chaos theory and neural networks is
better for traffic guidance and control with grid. The
high efficiency is benefited from the minimum price of
Self-Governing Domain algorithm. Further research in
the theory is based on simulation system, to create a
higher accuracy in the real-time forecast system.
5. Acknowledgment
I would like to express my sincere gratitude to those
who have spent time for examining and revising my
paper. I also thank to Prof. Wang Demin and Prof. Jia
451
Hongfei for all the help and support and advices.
Thanks to my lab-mates for their comments and
suggestions. At last my thanks would go to my beloved
family for their loving considerations and great
confidence.
6. References
[1] Hu J M, et al. An applicable short-term traffic flow
forecasting method based on chaotic theory. Proc. of IEEE
6th International Conference on Intelligent Transportation
Systems (Vol.1)[C]. 2003:608-613.
[2] Wang Jin,Shi Qi-xin. The summery of the short-term
traffic flow predicts model. China Public Security,
2005(1):92-98.
[3] J.M. Lipton, K.P.Dabke. Reconstructing the State Space
Of Continuous Time Chaotic Systems Using Power Spectra.
Physics Letters A, 1996(201):290-300.
[4] Zhang Yumei,Qu Shiru,Wen Kaige. A Short-term Traffic
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 2009 International Conference on Computer Engineering and Technology
A Nios II Based English Speech Training System For Hearing-impaired
Children
Ningfeng Huang
School of Electronic Science
and Engineering
Southeast University
Nanjing, China wuhaining821@gmail.com
ningfengh@gmail.com
Abstract
We propose a novel feature system of language
training for hearing-impaired children’s speech
rehabilitation based on DE1 development board,
which contains an on-chip system called uc/os. The
system is imbedded with Nios II processors, which
could make our system stable, portable and cheap.
Considering the limited storage and calculation ability
of imbed system, at the same time, and we recognize
child’s voice by Single phoneme, we use LPCC as
Eigenvector to judge the similarity between speaker’s
voice and example’s.
1. Introduction
Hearing-impaired children have difficulty in
hearing, for which people may consider that they
cannot speak a word and will become so-called “deaf
and dumb” people. However, most of them actually
meet the requirements for learning to speak. Firstly,
most of the hearing-impaired children have healthy
speech organs. What’s more, using hearing aids to
amplify and modulate sounds, 95% hearing-impaired
children can compensate for their poor sense of sound,
as a result of which they can understand phonic
language and learn to speak. Thirdly, sense of vision,
feeling and motion contribute to learning phonic
language[1].
This work was supported by the Altera University Program.
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.159
Haining Wu Yinchen Song
School of Automation School of Biological Science
Southeast University and Medical Engineering
Nanjing, China Southeast University
Nanjing, China
songyinchen@gmail.com
To help hearing-impaired children learn to speak,
many researchers have developed various speech
training aids. Most of them are based on computer or
MCU+DSP[2-5]. This paper describes a Nios II based
English speech training system, which is designed for
the hearing-impaired children. The system is
developed on the Altera DE1 development board. DE1
board provides a lot of features to enable various
multimedia project developments for users. Cyclone II
FPGAs are perfectly suited as an embedded processor
or microcontroller when combined with Altera's 32-bit
Nios II embedded processor intellectual property (IP)
cores. Users can add many other functions to the
FPGA with additional IP cores available from Altera
Corporation and Altera's partners. Compared with the
computer or MCU+DSP based systems, system based
on Nios II is much more compact, stable and cheaper.
Considering the hearing-impaired children have
difficulty pronouncing and producing intelligible
speech, we design this system with several speech
training modules according to the feature of speech
training system for hearing-impaired children and the
connection between the acoustic character and
physiological character of the speech, such as
breathing training, vowel training and consonant
training.
This paper is organized as follows. In Section 2, a
brief introduction of the construction is presented. In
Section3, we describe the main algorithm we use to
recognize speech. In Section 4, we discuss about how
we realize it using Altera DE1 Multimedia
452
Development Board and the experimental results are
given in Section 5. Section 6 is conclusion of the
whole paper.
2. Construction of A System
The system is designed for hearing-impaired
children to rehabilitate the speech ability. Figure 1
shows the structure of the system. The system is
developed on the Altera DE1 board. Using VGA
monitor and mouse as interfaces between users and
system, the system provides users some introductions
and samples of pronouncing syllables by audio and
acquires the speech signal from the users by
microphone. The speech signal will be processed by
advanced speech signal processing and recognition
technologies. The system will make a comparison
between users’ speech and the sample speech and
eventually give users score by quantitating the
matching rate.
Figure1. Construction of the system
This training system consists of breathing training,
vowel training and consonant training[6].
Figure2. Structure of the training system
The breathing training is aimed to estimate the
length of the period that user pronounces a syllable at
his highest volume and help modulate his breathing.
The progress bar is used to display how much the time
the user have been pronouncing. We estimate that
children commonly could persist for 10 seconds.
In the vowel training and consonant training parts,
user could hear the sample speech and observe how the
453
speech organs perform during the complete process of
pronouncing a certain syllable with written instructions
both in Chinese and English step by step. The system
can response immediately after the user record his
speech. Then user could get the score from the
progress bar and replay his speech and the sample
speech, making a comparison on his own directly.
3. Speech Recognition
In a short period of time, voice can be considered to
be stable. Considering the low-frequency CPU of
embedded systems, limited storage resource
characteristics, we need to reduce the amount of
calculation to ensure real-time, we have adopted a
time-domain short-time voice processing as follows [7-
8]:
Figure3. Speech recognition process
Data samples at 8 kHz, the length of a frame is 240 (30
ms), frame shift 80 (10 ms). There are 10 groups of
girls’ voice samplings and 10 groups of boys’ voice
samplings of children between 8~9 years old.
Preprocessing, including sub-frame, pre-emphasis,
and endpoint detection. Sub-frame is to multiple the
original voice signal with a mobile window, between
the two overlapping, we selected rectangular windows
function as a window.
Rectangular windows formula:
w(n) = 1.0 ≤ n ≤ N − 1 (1)
(N is length of a frame)
The purpose of pre-emphasis is to increase the high-
frequency part of voice, which is the weaker part of
signal, and filter the baseline drift caused by DC
changes, filtering 50hz frequency interference. We use
FIR filter to realize it.
FIR formula:
data(n) = S (n) − 0.93* S (n − 1) (2)
The system detects endpoint by measuring the
sample’s short-term average amplitude and short-term
zero-crossing rate,
Short-term average amplitude formula:
N −1
M n = ∑ | S w ( n) | (3)
n=0
 Short-term zero-crossing rate formula:

Zn =
1
2
{∑ sgn[Sw (n)] − sgn[ Sw (n − 1)] } (4)

We use Autocorrelation to estimate the signal cycle

length, the formula is: (N is the length of a frame)
N −1−l
Rw (l ) = ∑S
n =0
w ( n) S w ( n + l ) l ∈ [1, N ] (5)

Then we use Durbin algorithm to calculate linear
prediction-based cepstral coefficient (LPCC) by the
recurrence formulas showed in reference [9]
We match the example voice’s LPCC vector with
speaker’s voice vector, each vector contains 16 LPCCs.
ai is example’s LPCC, and bi is speaker’s. After having
rate value, we qualify error to rate, in order to show
percentage of speaker’s similarity, as following
formulas shows:
Figure5. Each five vectors figure of [iə]
4. Hardware Implementation
This implement use FPGA to create system, based
on SOPC builder. Taking the NIOS II processor as a
core, we link many interfaces and user logic to the
Avalon bus. Because we bring many existing IP cores
to the design, many of which offered by Altera
corporation to support the DE1 board, development

∑
16 process is largely accelerated[10].
(ai − bi ) 2
error = i =1
(6) In this system, we use Mathwork’s simulink
16 software and Altera corporation’s DSP builder
Rate = 100 − 100 × log(error ) (7) software to design the voice short-time signal
processing user logic which realized by hardware. This
Besides, if error > 10, rate=0%, if error<1,
block links with NIOS II system through Avalon MM
rate=100%.
bus. In NIOS II system, we can obtain voice real-time
Considering the specific condition of Diphthong,
processing by using C language and this design has
like “oi”, “ei”, we analyze the LPCC graph simulated
been proved to be highly effective. It can reduce the
in MATLAB, noticing that most of them mainly
burden of the processing core and operation system to
contain two strands, each stands for corresponding
make the system more stable.
monophthong’s LPCC graph, So we take first five
The major functions of this model is to obtain the
vectors of each monophthong’s LPCC and calculate
real-time calculation of voice signal’s short time
error and qualify rate as above, call them Rate1, Rate2,
average magnitude Mn and short-time zero-crossing
then it turns out that Rate= (Rate1+Rate2)/2.
rate Zn which are defined in section 4. These two
Take [iə] for example; figure 4 shows the all vectors
parameters of voice signal are very important in
figure of [iə], figure 5 shows each monophthong’s five
judging the end-point of voice and further analysis of
vectors figure of [iə].
the voice.
3
This model is composed by Avalon MM write bus,
2 pre-emphasize filter, short time average magnitude
calculate block, short-time zero-crossing rate calculate
1 block, signal combination blocks and Avalon MM read
bus.
0

-1

-2

-3

-4
0 2 4 6 8 10 12 14 16

Figure4. All LPCC vectors figure of [iə]

454

Figure6. The entire design in dsp builder
Figure7. FIR filter symbol
Figure8. Sub-frame accumulation
After simulation, we can see the block work
satisfactorily in detecting the endpoint of the
pronunciation. From top to button, the fist one is wave
of two words, the second one is the signal filtered by
FIR filter, we can see DC part significantly reduced
high-frequency gain significant. The third one and the
forth one are the wave of the average amplitude and
zero-cross rate, observing the transition part between
two words, the amplitude is almost zero, the rate
remain high.
Figure9. Simulation result
455
5. Experiment
After realizing speech processing in Nios, we use
the wavread function, contained in Matlab, to convert
the sample signal to txt document, so c program can
read and calculate LPCC vector, and then show them
in Matlab.
In this figure, blue lines are speakers’ vector curve
of “i:” pronunciation, the red one is the average of
them, standing for the standard curve. The difference
of them is small and reasonable.
Figure10. Simulation result
6. Conclusion
This paper proposed a novel feature system of
English speech training for hearing-impaired
children’s speech rehabilitation. The system
developed on Altera DE1 development board, which is
embedded with Nios II processors, is much more stable,
portable and cheaper by comparison.
The training system consists of 3 sections as
follows: breathing training, vowel training and
consonant training. It aims to help modulate users’
breathing while talking and let them produce
intelligible speech.
This system adopts LPCC algorithm to determine
the accuracy of pronunciation, maintained by
programming c in imbedded Nios II processors. The
pre-processing part is realized by DSP builder, which
works as a block linked to Avalon MM bus.
The authors invited students from Nanchang Road
Elementary School (Nanjing) to record their voices
and made them into speech samples, given that users
of this system are hearing-impaired children. And the
authors also tested for a lot of times in order to make
sure of the system’s stability. Although the system is
currently designed for hearing-impaired children’s
learning English speech, it is expected that the
developed system could be a useful tool for hearing-
impaired people in speech training no matter how old
they are and what language they are going to speak.
7. Acknowledgment
We wish to gratefully acknowledge the advice,
comments and assistance we received from Professor
Tang Yongming of the School of Electronic Science
and Engineering of Southeast University. And we
deeply appreciate Ms. Liu Ruixue of Nanchang Road
Elementary School (Nanjing) for providing children’s
speech samples.
8. References
[1] Alsaka, Y.A.; Doll, S.; Davis, S., “Portable speech
recognition for the speech and hearing impaired,”
Southeastcon '97. 'Engineering new New Century'.,
Proc. IEEE, 1997, pp. 151-153
[2] M.L. Hsiao, P.T. Li, P.Y. Lin, S.T. Tang, T.C. Lee, S.T.
Young, “A computer based software for hearing
impaired children’s speech training and learning
between teacher and parents in Taiwan,” in 2001 Proc.
of the 23rd Ann. EMBS Int. Conf., Istanbul, pp. 1457-
1459.
[3] Weerasinghe, D.; Dias, D., "DSP-based techniques for
speech training of hearing impaired children,”
Communications, Computers and signal Processing,
456
2001. PACRIM. 2001 IEEE Pacific Rim Conference on,
vol. 1, pp. 51-54
[4] Y.Y. Shi, J. Liu and R.S. Liu, “Single-chip speech
recognition system based on 8051 microcontroller
core,” IEEE Transactions on Consumer Electronics,
Vol. 47, No. 1, 2001, pp. 149-153
[5] S. Phadke, R. Limaye, S. Verma and K. Subramanian,
“On design and implementation of an embedded
automatic speech recognition”, in Proc. of the 17th Int.
Conf. on VLSI Design(VLSID’04)
[6] I. Kirschning and M.T. Toledo, “Vowel & diphthong
tutors for language therapy,” in Proc. of the 6th
Mexican Int. Conf. on Computer Science(ENC’05), pp.
26 – 30
[7] Y. Yazama, Y. Mitsukura, M.Fukumi and N. Akamatsu,
“Vowel recognition method by using features included
in amplitude for mobile device,” in Proc. of the 2004
IEEE Int. Workshop on Robot and Human Interactive
Communication, Kurashiki, pp. 613-618
[8] W.M. Campbell, K.T. Assaleh and C.C. Brown, “Low-
complexity small-vocabulary speech recognition for
portable devices,” in 5th Int. Symposium on Signal
Processing and its Applications, ISSPA ’99, Brisbane,
pp. 619-622
[9] G.D. Wu and Z.W. Zhu, “Chip design of LPC-cepstrum
for speech recognition,” in 6th IEEE/ACIS Int. Conf. on
Computer and Information Science (ICIS 2007)
[10] DE1 Development and Education Board User Manual,
Version 1.1, Altera Co., San Jose, CA , 2006, pp. 1-45
 2009 International Conference on Computer Engineering and Technology

A New DEA Model for Classification Intermediate Measures and

Evaluating Supply Chain and its members

S. M. Hatefi, F. Jolai, H. Iranmanesh, and H. Kor

Department of Industrial Engineering, College of Engineering,
University of Tehran
Tehran, Iran
s_m_hatefi@yahoo.com , fjolai@ut.ac.ir, hiranmanesh@ut.ac.ir, hkor@ut.ac.ir

Abstract—Appropriate selection of intermediate measure
status in supply chain is an important requirement for the
effective management of supply chain. The data envelopment
analysis (DEA) is suitable tool for evaluating supply chain
performance. In the conventional DEA it is deemed that the
variable status of each performance measures from the point
view of input or output is known. In supply chain intermediate
measures treat as both inputs of buyer and outputs of seller.
Conventional data envelopment analysis cannot be employed
directly to measure the performance of supply chain and its
members, because of the existence of the intermediate
measures that connects the supply chain members. In this
paper we develop a DEA model to classify the status of
intermediate measures and evaluate the performance of supply
chain members. The proposed model for each DMU0 permits
that supply chains (DMUs) choose which is best for each
intermediate measure, whether to designate it as an output of
seller or input of buyer. Then we illustrate the application of
model in a numerical example.
Keywords-Buer-seller supply chain; intermediate measures;
flexible measures; performance evaluation
I. INTRODUCTION
Effective management of an organization’s supply
chains has proven to be a very effective mechanism for
providing prompt and reliable delivery of high-quality
products and services at the least cost. To achieve this,
performance evaluation of the entire supply chain is
extremely important. This means utilizing the combined
resources of the supply chain members in the most efficient
way possible to provide competitive and cost-effective
products and services. However, lack of appropriate
performance measurement systems has been a major
obstacle to effective management of supply chains [1].
There are studies on applying the standard DEA model, to
evaluate the performance of supply chain members. For
example, Weber and Desai [2] employ DEA to construct an
index of relative supplier performance. Models have been
established to estimate the exact performance of supply
chain members based upon single performance measures
[3]. Other performance evaluations have been focused on
studying the impact of management practices on supply
chain performance [4].
Within the context of DEA, there are a number of
methods that have the potential to be used in supply chain
efficiency evaluation. Seiford and Zhu [5] and Chen and
Zhu [6] provide two approaches in modeling efficiency as a
two-stage process. Along a related line, Färe and Grosskopf
[7] develop the network DEA approach to model general
multi-stage processes with intermediate inputs and outputs.
Zhu [8], on the other hand, presents a DEA-based supply
chain model to both define and measure the efficiency of a
supply chain and that of its members, and yield a set of
optimal values of the (intermediate) performance measures
that establish an efficient supply chain. The different kinds
of DEA models for evaluation of supply chain can be found
in [9, 10].
In the conventional application of DEA, it is assumed
that the status of performance measures from point view of
input or output is specified. But in some conditions there are
variables or measures that treat as both an input and output.
Consider the buyer-seller supply chain as presented in Liang
et al. [9]. In supply chain there are some measures between
seller and buyer that are called intermediate measures. In
this study, the seller has three inputs, labor, operating cost,
shipping cost and three outputs, number of three products
shipped. The buyer has another input (labor) in addition to
number of shipped products and two outputs, sales and
profit. Intermediate measures in supply chain are number of
shipped products. In performance evaluation of supply chain
members at the same time these measures treat as both
inputs (inputs of buyer) and outputs (outs of seller).
Examples of flexible measures in evaluating performance of
university, hospital and highway maintenance crew can be
seen in [11]. Recently, Cook and Zhu proposed the new
method for classifying inputs and outputs based on standard
DEA model (CCR model) [11].

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 457

DOI 10.1109/ICCET.2009.217
 However, such an efficiency approach cannot be applied ߤ௥ ǡ ߴ௜ ǡ ߱௟ ǡ ߨ௞ ൒ Ͳሺʹሻ
directly to the problem of evaluating the efficiency of
supply chains, because some measures linked to supply
chain members cannot be simply classified as “outputs” or  zl
“inputs” of the chain. In fact, with respect to those xi yr
Seller Buyer
measures, conflicts between supply chain members are
likely present. For example, the supplier’s revenue is an
output for the seller, and it is in the seller’s interest to ‫ݔ‬ƴ ୩
maximize it; at the same time it is also an input to the buyer
who wishes to minimize it. Simply minimizing the total Figure 1. Buyer-Seller supply chain
supply chain cost or maximizing the total supply chain
revenue (profit) does not properly model and resolve the
inherent conflicts. In this paper we develop a DEA model to Model (2) only considers the intermediate measures as
classify the status of intermediate measures from point view outputs of seller and ignores the effect of intermediate
of input or output and evaluate the performance of supply measures on buyer. Model (2) can’t present true
chain members. performance of supply chain. Zhu [8] also showed that if
The rest of the paper is organized as follows. Section II intermediate measures are treated as both inputs and outputs
introduces the DEA models for performance evaluation of in CCR model, all supply chain members become efficient.
supply chain. In section III, we establish the proposed model Overall efficiency of supply chain is affected performance
and show that the model is capable to classify intermediate of seller and buyer and the performance improvement of one
measures in supply chain. In section IV, we illustrate the supply chain member affects the efficiency status of the
application of proposed model with the numerical example other, because of existence of intermediate measures.
found in literature. Conclusions are discussed in Section V. Consequently, we may establish following DEA model
based on the average efficiency of the buyer and seller:
II. DEA MODELS FOR SUPPLY CHAIN EVALUATION
Consider Buyer-Seller supply chain as showed in Figure ͳ σ௅௟ୀଵ ߱௟ ‫ݖ‬௟଴ σ௦௥ୀଵ ߤ௥ ‫ݕ‬௥଴
‫ ݔܽܯ‬ቈ ௠ ൅ ௅ ቉
1., Where xi (i =1,2,…,m) is the input of the seller, and zl (l ʹ σ௜ୀଵ ߴ௜ ‫ݔ‬௜଴ σ௟ୀଵ ߱ƴ ௟ ‫ݖ‬௟଴ ൅ σ௄ ௞ୀଵ ߨ௞ ‫ݔ‬ƴ ௞଴
= 1,2,…,L) is the seller output. Also zl and š୩ƴ (k = 1, 2,…, σ௅௟ୀଵ ߱௟ ‫ݖ‬௟௝
K) are the inputs of buyer and yr (r = 1,2,…,s) is the output ܵǤ ‫ݐ‬Ǥ ௠ ൑ Ͳ݆ ൌ ͳǡʹǡ ǥ ǡ ݊
σ௜ୀଵ ߴ௜ ‫ݔ‬௜௝
of buyer.
σ௦௥ୀଵ ߤ௥ ‫ݕ‬௥௝
Suppose there are n supply chains or observations and we  ௅ ൑ Ͳ݆ ൌ ͳǡʹǡ ǥ ǡ ݊
interest to evaluate supply chain performance. The CCR σ௟ୀଵ ߱ƴ ௟ ‫ݖ‬௟௝ ൅ σ௄ ௞ୀଵ ߨ௞ ‫ݔ‬ƴ ௞௝
DEA efficiency of the supply chain is measured as : ߤ௥ ǡ ߴ௜ ǡ ߱௟ ǡ ߱ƴ ௟ ǡ ߨ௞ ൒ Ͳሺ͵ሻ

σ௦௥ୀଵ ߤ௥ ‫ݕ‬௥଴ Model (3) considers correctly intermediate measures in

‫ݔܽܯ‬ ௠
σ௜ୀଵ ߴ௜ ‫ݔ‬௜଴ ൅ σ௄௞ୀଵ ߨ௞ ‫ݔ‬ƴ ௞଴
performance evaluation of supply chain members, but it
σ௦௥ୀଵ ߤ௥ ‫ݕ‬௥௝ does not reflect the relationship between the buyer and the
‫ݏ‬Ǥ ‫ݐ‬Ǥ ௠ ൑ Ͳ݆ ൌ ͳǡʹǡ ǥ ǡ ݊ seller, because of the weights of intermediate measures as
σ௜ୀଵ ߴ௜ ‫ݔ‬௜௝ ൅ σ௄௞ୀଵ ߨ௞ ‫ݔ‬ƴ ௞௝
outputs of the seller ሺɘ୪ ሻ doesn’t equal to the weights of
ߤ௥ ǡ ߴ௜ ǡ ߨ௞ ൒ Ͳሺͳሻ
intermediate measures as inputs of the buyer ሺɘƴ୪ ሻ .
Following Liang et al. [9] in performance evaluation of
Model (1) only considers the inputs and outputs of the
supply chain model (3) considers the buyer and seller as two
supply chain system and ignores intermediate measures in
independent units and can’t characterize ideal efficiency of
performance evaluation of supply chain. Zhu [8] showed
supply chain. Liang et al. [9] set the weights of intermediate
that model (1) fails to correctly specify performance of
measure as the outputs of seller equal to the weights of
supply chain. One is to treat the intermediate measures as
intermediate measure as the inputs of buyer in model (3).
outputs in the following CCR model when DMU0 is under
With respect to this condition model (3) becomes nonlinear
evaluation:
programming problem. He showed that this new model can
be treated as a parametric linear program and obtained a
σ௦௥ୀଵ ߤ௥ ‫ݕ‬௥଴ ൅ σ௅௟ୀଵ ߱௟ ‫ݖ‬௟଴
‫ݔܽܯ‬ heuristic search for solving it.
σ௠ ௄
௜ୀଵ ߴ௜ ‫ݔ‬௜଴ ൅ σ௞ୀଵ ߨ௞ ‫ݔ‬ƴ ௞଴ In next section we establish new two stage DEA model
σ௥ୀଵ ߤ௥ ‫ݕ‬௥௝ ൅ σ௅௟ୀଵ ߱௟ ‫ݖ‬௟௝
௦
for classification intermediate measures and evaluating
‫ݏ‬Ǥ ‫ݐ‬Ǥ ௠ ൑ Ͳ݆ ൌ ͳǡʹǡ ǥ ǡ ݊
σ௜ୀଵ ߴ௜ ‫ݔ‬௜௝ ൅ σ௄
௞ୀଵ ߨ௞ ‫ݔ‬ƴ ௞௝ supply chain members.

458
 III. NEW TWO STAGE MODEL intermediate measures ሺɘ୪ ሻ as outputs of seller don’t equal
Consider two following DEA models. Model (4) is to the weights of intermediate measures ሺɘƴ୪ ሻ as inputs of
called output-oriented multiplier DEA model where the goal buyer in model (6). As mentioned model (6) considers buyer
is to minimize the inputs while keeping the outputs at their and seller as two independent units and can’t describe true
current levels. The optimal value to model (4) isɅ‫ כ‬൒ ͳ. performance of supply chain. Because of intermediate
When DMU0 is under evaluation, if Ʌ‫ כ‬ൌ ͳ then DMU0 is measures are the relationship between the seller and buyer
efficient. If Ʌ‫ כ‬൐ ͳ, then DMU0 is inefficient. we set ɘ୪ ൌ ɘƴ୪ ሺŽ ൌ ͳǡ ǥ ǡ ሻ .
Model (5) is called input-oriented multiplier DEA model Theorem 1. With respect this condition ሺɘ୪ ൌ ɘƴ୪ ሻ ,
where the goal is to maximize the outputs while keeping the model (7) ignores the effect of another inputs ሺšƴ ୩ ሻ of buyer
inputs at their current levels. The optimal value to model (5) in performance evaluation of DMU0.
isԄ‫ כ‬൑ ͳ. When DMU0 is under evaluation, if Ԅ‫ כ‬ൌ ͳ then Proof: Applying this condition to model (7) yields
DMU0 is efficient. If Ԅ‫ כ‬൏ ͳ, then DMU0 is inefficient. σ୐୪ୀଵ ɘ୪ œ୪଴ ൌ ͳ (seller) and σ୐୪ୀଵ ɘ୪ œ୪଴ ൅ σ୏୩ୀଵ Ɏ୩ šƴ ୩଴ ൌ
௠ ͳ(buyer). We conclude that σ୏୩ୀଵ Ɏ୩ šƴ ୩଴ ൌ Ͳ
‫כ‬
ߠ ൌ ‫ ݊݅ܯ‬෍ ߴ௜ ‫ݔ‬௜଴ ሺͶሻ Therefore intermediate measures can’t treat as both
௜ୀଵ outputs of seller and inputs of buyer in model (6). For
௦ ௠
solving this problem we define †୪ ‫ א‬ሼͲǡͳሽ as binary
෍ ߤ௥ ‫ݕ‬௥௝ െ ෍ ߴ௜ ‫ݔ‬௜௝ ൑ Ͳ݆ ൌ ͳǡ ǥ ǡ ݊ variables, where †୪ ൌ ͳ designates that zl is considered as
௥ୀଵ
௦
௜ୀଵ an output of seller and †୪ ൌ Ͳ designates it as an input of
buyer. Let ɘ୪ be the weight for each intermediate measure l.
෍ ߤ௥ ‫ݕ‬௥଴ ൌ ͳǡ ߤ௥ ǡ ߴ௜ ൒ Ͳሺߝሻ We establish the following mathematical programming
௥ୀଵ model:
௦
௠ ௦
‫כ‬
߶ ൌ ‫ ݔܽܯ‬෍ ߤ௥ ‫ݕ‬௥଴ ሺͷሻ ‫ ݊݅ܯ‬෍ ߴ௜ ‫ݔ‬௜଴ െ ෍ ߤ௥ ‫ݕ‬௥଴ ሺ͹ሻ
௜ୀଵ
௦ ௠ ௜ୀଵ ௜ୀଵ
ሺ‫ݎ݈݈݁݁ݏ‬ሻ
෍ ߤ௥ ‫ݕ‬௥௝ െ ෍ ߴ௜ ‫ݔ‬௜௝ ൑ Ͳ݆ ൌ ͳǡ ǥ ǡ ݊ ௅ ௠
௥ୀଵ ௜ୀଵ
௠ ‫ݏ‬Ǥ ‫ݐ‬Ǥ ෍ ݀௟ ߱௟ ‫ݖ‬௟௝ െ ෍ ߴ௜ ‫ݔ‬௜௝ ൑ Ͳ݆ ൌ ͳǡ ǥ ǡ ݊
෍ ߴ௜ ‫ݔ‬௜଴ ൌ ͳǡߤ௥ ǡ ߴ௜ ൒ Ͳሺߝሻ ௅
௟ୀଵ ௜ୀଵ

௜ୀଵ
෍ ݀௟ ߱௟ ‫ݖ‬௟଴ ൌ ͳǡ ߱௟ ǡ ߴ௜ ൒ Ͳሺߝሻ
Both models (4) and (5) identify the same efficient ௟ୀଵ

frontier, because Ʌ‫ כ‬ൌ ͳ if and only if Ԅ‫ כ‬ൌ ͳ. We establish ሺܾ‫ݎ݁ݕݑ‬ሻ

௦ ௅ ௄
the following cooperative linear programming problem.
෍ ߤ௥ ‫ݕ‬௥௝ െ ෍ሺͳ െ ݀௟ ሻ߱௟ ‫ݖ‬௟௝ െ ෍ ߨ௞ ‫ݔ‬ƴ ௞௝ ൑ Ͳ
௠ ௦ ௥ୀଵ ௟ୀଵ ௞ୀଵ

‫ ݊݅ܯ‬෍ ߴ௜ ‫ݔ‬௜଴ െ ෍ ߤ௥ ‫ݕ‬௥଴ ሺ͸ሻ ݆ ൌ ͳǡ ǥ ǡ ݊

௅ ௄
௜ୀଵ ௜ୀଵ
ሺ‫ݎ݈݈݁݁ݏ‬ሻ ෍ሺͳ െ ݀௟ ሻ߱௟ ‫ݖ‬௟଴ ൅ ෍ ߨ௞ ‫ݔ‬ƴ ௞଴ ൌ ͳ ǡ ߤ௥ ǡ ߱௟ ǡ ߨ௞ ൒ Ͳሺߝሻ
௅ ௠ ௟ୀଵ ௞ୀଵ

‫ݏ‬Ǥ ‫ݐ‬Ǥ ෍ ߱௟ ‫ݖ‬௟௝ െ ෍ ߴ௜ ‫ݔ‬௜௝ ൑ Ͳ݆ ൌ ͳǡ ǥ ǡ ݊

Thus, model (7) for performance evaluation DMU0
௟ୀଵ ௜ୀଵ
௅ permits that DMU to choose which is best for each
෍ ߱௟ ‫ݖ‬௟଴ ൌ ͳǡ ߱௟ ǡ ߴ௜ ൒ Ͳሺߝሻ intermediate measure , whether to designate it as an output
of seller or input of buyer. Model (7) is clearly nonlinear.
௟ୀଵ
ሺܾ‫ݎ݁ݕݑ‬ሻ But it can be linear by way of the change of variables
௦ ௅ ௄ Ɂ୪ ൌ †୪ ɘ୪ for each ݈, imposing the constraints:
෍ ߤ௥ ‫ݕ‬௥௝ െ ෍ ߱ƴ ௟ ‫ݖ‬௟௝ െ ෍ ߨ௞ ‫ݔ‬ƴ ௞௝ ൑ Ͳ݆ ൌ ͳǡ ǥ ǡ ݊
௥ୀଵ ௟ୀଵ ௞ୀଵ
Ͳ ൑ ߜ௟ ൑ ‫݀ܯ‬௟
௅ ௄ ߜ௟ ൑ ߱௟ ൑ ߜ௟ ൅ ‫ܯ‬ሺͳ െ ݀௟ ሻሺͺሻ
෍ ߱ƴ ௟ ‫ݖ‬௟଴ ൅ ෍ ߨ௞ ‫ݔ‬ƴ ௞଴ ൌ ͳ ǡߤ௥ ǡ ߱ƴ ௟ ǡ ߨ௞ ൒ Ͳሺߝሻ
௟ୀଵ ௞ୀଵ Where M is a large positive number. These linear integer
restrictions capture the nonlinear expression †୪ ɘ୪ ൌ Ɂ୪ ,
Following Chen and Zhu [6] , model (6) correctly without actually having to directly specify it in the
defines the efficiency of a Buyer-Seller supply chain and optimization model. Note that if †୪ ൌ ͳ then ɘ୪ ൌ Ɂ୪ , and
further can consider the effect of intermediate measures in if †୪ ൌ Ͳ , then Ɂ୪ ൌ0. By this way of the changing variables
performance evaluation of supply chain. But the weights of

459
 ߜ௟ ൑ ߱௟ ൑ ߜ௟ ൅ ‫ܯ‬ሺͳ െ ݀௟ ሻǡ ݀௟ ߳ሼͲǡͳሽǡ ‫݈׊‬
TABLE I. RESULTS FROM MODEL (9) FOR THREE INTERMEDIATE
MEASURES
Also this model is capable to provide efficiency score
ଵ
Supply chain no. d1 d2 d3 objective function
for seller and buyer. At the optima, ߠ஺‫ כ‬ൌ σ೘ ణ‫כ‬௫ and
1 1 0 0 0.0000 ೔ ೔ ೔బ
Ʌ‫כ‬୆ ൌ σୱ୰ୀଵ ρ୰‫› כ‬୰଴ present the efficiency scores for the seller
2 0 0 1 0.2231
and buyer respectively when DMU0 is under evaluation.
3 0 1 0 0.2500
Also overall supply chain efficiency of DMU0 can be
4 0 0 1 0.3721 ଵ ଵ
calculated by ൤ ൅ σ௦௥ୀଵ ߤ௥‫ݕ כ‬௥଴ ൨.
5 1 1 0 1.0435 ଶ σ೘ ‫כ‬
೔ ణ೔ ௫೔బ
6 1 0 1 0.0000
7 0 1 0 0.1667
IV. NUMERICAL EXAMPLE
8 1 1 0 0.2980
9 0 0 1 1.0000 In this section, we apply our model to the numerical
10 1 0 0 0.4419
example used in Liang et al. [9]. In this data the seller has
three inputs, x1 (labor), x2 (operating cost) and x3 (shipping
cost) and three outputs, z1 (number of product A shipped), z2
(number of product B shipped) and z3 (number of product C
model (7) converts to a mixed integer linear program shipped). The buyer has another input šƴ ଵ (labor) in addition
such as bellow: to œଵ ǡ œଶ ǡ œଷ and two outputs: y1 (sales) and y2 (profit).
௠ ௦ Table I. reports the results from model (9), where the
second, third and forth columns show the optimal d for
‫ ݊݅ܯ‬෍ ߴ௜ ‫ݔ‬௜଴ െ ෍ ߤ௥ ‫ݕ‬௥଴ ሺͻሻ
classifying the status of intermediate measures ሺœଵ ǡ œଶ ǡ œଷ ሻ
௜ୀଵ ௜ୀଵ
ሺ‫ݎ݈݈݁݁ݏ‬ሻ respectively and the fifth column, the optimal value to
௅ ௠ model (9). In this case, when supply chain 1 is under
‫ݏ‬Ǥ ‫ݐ‬Ǥ ෍ ߜ௟ ‫ݖ‬௟௝ െ ෍ ߴ௜ ‫ݔ‬௜௝ ൑ Ͳ݆ ൌ ͳǡ ǥ ǡ ݊ evaluation, z1 (number of product A shipped) treats as an
output of the seller and z2 (number of product B shipped)
௟ୀଵ ௜ୀଵ
௅ and z3 (number of product C shipped) treat as the inputs of
෍ ߜ௟ ‫ݖ‬௟଴ ൌ ͳǡ ߱௟ ǡ ߴ௜ ൒ Ͳሺߝሻǡ ‫݈׊‬ǡ ݅ the buyer. Supply chain 5 treat z1 and z2 as two outputs of
௟ୀଵ
the seller and z3 as an input of the buyer.
ሺܾ‫ݎ݁ݕݑ‬ሻ Table II. reports the efficiency scores for the seller ሺߠ஺‫ כ‬ሻ
௦ ௅ ௅
௄ , buyer ሺߠ஻‫ כ‬ሻ and the supply chain efficiency in model (9)
ߜ௟ ‫ݖ‬௟௝ െ ෍ ߨ௞ ‫ݔ‬ƴ ௞௝ ൑ Ͳ and Liang et al. cooperative model [9] .As mentioned, Liang
෍ ߤ௥ ‫ݕ‬௥௝ െ ෍ ߱௟ ‫ݖ‬௟௝  ൅ ෍ et al. obtained heuristic search for solving the nonlinear
௞ୀଵ
௥ୀଵ ௟ୀଵ ௟ୀଵ
݆ ൌ ͳǡ ǥ ǡ ݊ cooperative supply chain model. Note that model (9) is the
௅ ௅ ௄ mixed integer linear program. Results show that efficiency
෍ ߱௟ ‫ݖ‬௟଴ െ ෍ ߜ௟ ‫ݖ‬௟଴ ൅ ෍ ߨ௞ ‫ݔ‬ƴ ௞଴ ൌ ͳ scores for the seller and buyer and supply chain efficiency
௟ୀଵ ௟ୀଵ ௞ୀଵ of two models is extremely closer to each other.
ߤ௥ ǡ ߱௟ ǡ ߨ௞ ൒ Ͳሺߝሻǡ ‫ݎ׊‬ǡ ݈ǡ ݇
Ͳ ൑ ߜ௟ ൑ ‫݀ܯ‬௟
TABLE II. EFFICIENCY SCORES

model (9) Liang et al. model

Supply
chain no. Ʌ‫כ‬୅ Ʌ‫כ‬୆ Supply chain Ʌ‫כ‬୅ ߠ஻‫כ‬ Supply chain
1 1 1 1 1 0.894 0.947
2 0.972 0.805 0.889 0.924 0.801 0.862
3 0.8 1 0.9 0.691 1 0.846
4 1 0.628 0.814 1 0.628 0.814
5 0.676 0.436 0.556 0.676 0.573 0.625
6 1 1 1 1 1 1
7 1 0.833 0.917 1 0.819 0.909
8 0.770 1 0.885 0.747 1 0.873
9 0.5 1 0.75 0.5 1 0.75
10 1 0.558 0.779 1 0.596 0.798

460
 V. CONCLUSION
The current study develops new DEA-based model for
evaluating supply chain and its members. Recent DEA
models cannot be applied directly to the problem of
evaluating the efficiency of supply chains, because some
measures linked to supply chain members cannot be simply
classified as “outputs” or “inputs” of the chain. The
proposed model is capable to classify the status of
intermediate measures from the point view of input or
output. The presented model for performance evaluation
DMU0 permits that DMU to choose which is best for each
intermediate measure, whether to designate it as an output
of seller or input of buyer. Also, in this paper we provide the
score efficiencies for the seller, buyer and the supply chain
efficiencies. Our model is an important tool for the
managers in monitoring and planning their supply chain
operations.

REFERENCES
[1] Lee, H.L. and C. Billington. (1993). “Material management in
decentralized supply chains.” Operations Research, 41, 835–
847
[2] Weber, C.A. and A. Desai. (1996). “Determinants of Paths to
Vendor Market Efficiency Using Parallel Coor-dinates
Representation: A Negotiation Tool for Buyers.” European
Journal of Operational Research 90, 142–155
[3] Cheung, K.L. andW.H. Hansman. (2000). “An exact
performance evaluation for the supplier in a two-echelon
inventory system.” Operations Research, 48, 646–653
[4] Forker, L.B., D. Mendez, and J. Hershauer. (1997). “Total
quality management in the supply chain: What is its impact on
performance?.” International Journal of Production Research,
35, 1681–1702
[5] Seiford, L.M. and J. Zhu. (1999). “Profitability and
marketability of the top 55 US commercial
banks.”Management Science, 45(9), 1270–1288
[6] Chen, Y. and J. Zhu. (2004). “Measuring information
technology’s indirect impact on firm performance.”
Information Technology and Management Journal, 5(1–2), 9–
22
[7] Färe, R. and S. Grosskopf. (2000). “Network DEA.” Socio-
Economic Planning Sciences, 34, 35–49
[8] Zhu, J. (2003). Quantitative Models for Performance
Evaluation and Benchmarking: Data Envelopment Analysis
with Spreadsheets. Boston: Kluwer Academic Publishers
[9] Liang, L. , Yang, F., Cook, W. D., Zhu, J., 2006, DEA
models for supply chain efficiency evaluation, Ann Oper Res
, 145, 35–49
[10] Chen, Y., Liang, L., Yang, F. , 2006, A DEA game model
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data envelopment analysis, European Journal of Operational
Research, 180, 692–699

461
 2009 International Conference on Computer Engineering and Technology
A Novel Binary Code Based Projector-Camera System Registration Method
Jiang Duan1,2, Jack Tumblin2
1
Southwestern University of Finance and Economics, 2Northwestern University
Duanj_t@swufe.edu.cn, jet@cs.northwestern.edu
Abstract
Precisely relating pixels in the projector image to
pixels in camera image plays a significant role in
many projector-camera systems. In this paper, we
present a novel method of finding the correspondence
in a simple and efficient manner. Our registration
approach uses binary coding scheme, only need to
capture a few pictures and does not need to give any
assumption in the projector-camera system. Our
experimental results demonstrate the advantage of
using our registration methods.
1. Introduction
Finding the correspondence between pixels in the
projector image P and camera image I play a
significant role in many projector and camera systems,
like [1, 2, 3]. Consider a point (X, Y) in the projector
image, which is projected to a position located on the
projection screen. This process involves a perspective
transform whose parameters depend on the some
unknown factors such as the position and orientation
of the projector relative to the screen etc. The point on
the screen is then captured by a camera and appears at
a pixel location (x, y). This process involves another
perspective transform whose parameters depend again
on some unknown factors like the position and
orientation of the camera relative to the screen etc. The
goal of finding the correspondence is to relate the
position in projector image (X, Y) with the position in
camera image (x, y) without knowing the unknown
factors, i.e. for a position in the camera image at (x, y),
we would like to find out the point (X, Y) in the
projector image that corresponds to this position.
A category of finding the correspondence is the
homography based approaches like [1]. In this kind of
methods, 4 corresponding pairs of points in projector
image and camera image are usually required in order
to obtain projector-camera homography. The success
of this category of methods highly depends on how
accurate these points are located [4], as just 1 or 2
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.51
pixel error can make big difference in recovering the
projector-camera homography. In addition, the
assumption of using homography based approaches is
that the projector-camera system undergoes a
perspective transform. However, in practice, lens
curvature of both projector and camera as well as other
unknown factors can make the assumption fail, which
makes the projector-camera homography fail.
Other methods like [2, 3] use a lookup table to
store the correspondence between projector image and
camera image, i.e., each point in one domain is used as
an index to obtain the corresponding point in the other.
To determine this 2D mapping, a sampling of
corresponding points are obtained by projecting
uniformly spaced square patches on the scene and the
corresponding images are acquired. Then mapping
between the two coordinate frames are modeled using
piecewise second-order polynomials. This approach
can produce very accurate results as it can
accommodate for geometric distortions that may be
caused by the optics of the projector and the camera as
well as parallax due to the shape of the scene [3].
Similar with the work from [2, 3], in our work,
we use a novel binary code based method to recover
the correspondence between projector image and
camera image. However, we propose a simpler and
much more efficient method to achieve this aim.
2. Our binary code based registration
method
In order to relate the projector pixel position to
camera pixel position, an easy way is to use projector
to project certain pattern image onto the projection
screen and then use camera to capture the projected
pattern. Next, the positions of certain features in both
projector image and camera image are computed as the
sample points and are associated with each other. If the
correspondence of these samples between projector
image and camera image can be recovered accurately
enough, the correspondence of other position can be
recovered by interpolation. Thus, the key in our
projector-camera registration method is to precisely
462
recover the correspondence of these samples between
projector image and camera image.
In order to obtain the sample points, we can
divide the projector image space into uniformly spaced
rectangular patches. Each time, only one patch is lit
and a picture is captured. Then the centers of the
patches in both projector image and camera image are
computed as a pair of one of the correspondent sample
points. Using this way, to obtain N sample points, N
pictures are required, which makes this process slow
and inefficient. In our work, we use binary strip
patterns instead of the patch patterns described above
and propose an efficient coding scheme to make this
process efficient and achieve the same aim.
The binary strip patterns we used in our work are
shown as below in Figure 1. Patterns are sent to the
projector in the sequence from left to right and top to
bottom and corresponding pictures are taken. Thus,
each pixel either in the projector image or in the
camera image would have N chances of being either
turned on or turned off in the ideal situation, as long as
N pictures are taken. Based on this fact, our coding
scheme is described as following.
1. The pixels in the upper left picture represent
the most significant bit in the coding scheme and the
pixels in the right bottom picture represent the least
significant bit in the coding scheme.
2. If the pixel is turned on, the bit is stored as 1,
otherwise, the bit is stored as 0.
...
Figure 1. The strip patterns used in our experiment,
which are sent to projector in the sequence from left to
right and top to bottom
Let’s look at a simple example of using only 16
sample points in our project-camera registration to
illustrate how the coding scheme works. As described
above, in order to locate the position of 16 sample
points, we need to found out pixels belonging to 16
individual rectangular patches, based on which the
centers can be computed as the sample points. In this
case, only 4 pictures are required to determine the 16
individual rectangular patches. According to the
coding scheme proposed above, the 16 patches can be
easily determined based on pixels’ codes as shown
below in Figure 2 (Left). With this coding scheme, it is
463
easy to work out how many pictures are needed using
the following equation:
Pn log 2 Sn (1)
Where Sn is the sample points used in registration
and Pn is the picture number needs taking to obtain
these sample points. With this equation, it is easy to
work out that, 16 sample points needs 4 pictures, 64
sample points needs 8 pictures and so on. This is much
efficient with the naïve patch-by-patch-lit process. In
our work, we use 64 sample points to carry on the
projector and camera registration. The projector-
camera system we have used is a CANON LV-7325U
and the camera we used is the CANON Powershot G6
VPL-CS5.
Figure 2. Left: Coding scheme for 16 patches
computed using 4 pictures. Right: A selected camera
picture we used in our experiment
Using above method, the center of each patch that
is used as sample point in the projector images can be
easily determined, as the projector images are
produced using computer program and each pixel in
the images can be explicitly controlled to have two
possible values, which are either turned on or turned
off. However, this is not the same story in the camera
pictures. The point spread function and noises of the
sensor, the ambient light interaction as well as the non-
uniformity of the lighting coming from the projector
all contribute to make the pixels’ values varies across a
wide range and pixels cannot be explicitly turn on or
turn off as projector images (Figure 2 (Right)).Thus, it
is difficult to find a single threshold value that can
perfectly differentiate whether pixels in the captured
camera images are lit or not following the same style
as projector images, i.e., if pixels are lit in the projector
image, the corresponding pixels in the camera picture
should be determined as lit pixels using the threshold
value and vice versa. The difficulty would add
uncertainty in properly determining the centers of the
patches in the camera image.
According to the above analysis, we carried out
an experiment to determine pixels belonging to
different patches and pixels showing uncertainty that
are hard to determine which patches they should
actually belong to. We set the pixel values that are
above 90 to 1 and the pixel values that are below 70 to
0, and set the pixel values that are between 70 and 90
as uncertainty pixels. Following this scheme, in the
Figure 3 (Left), we show an example in which the
white pixels show these pixels whose codes value is
111010, the black pixels shows these pixels that
belong to other patches and the red pixels show pixels
with uncertainty. As seen from the Figure, pixels that
lie toward the edge of the patches of the patches show
more uncertainty in comparison with the pixel that are
located towards the center of the patches. The pixels
with uncertainty presents challenge in our projector
and camera registration method, as they would make
the calculated centers of the patches in the camera
images shift from the theoretical center.
Figure 3. Left: A recovered patch in for camera
pictures computed based on the scheme describe above.
In this image, the white pixels show these pixels whose
codes value is 111010; the black pixels shows these
pixels that belong to other patches; the red pixels show
these pixels that show uncertainty and that are hard to
determine which patches they should actually belong
to. Right: An image displaying the Ai(x, y) for a pair of
complimentary images (See Equation (2)).
In order to solve the problem, we propose to use
refined patterns shown in Figure 4. In the new pattern
scheme, there are two main changes. First, we add a
complementary pattern for each of the pattern shown
before, i.e., in Figure 4, the second row is
complementary element of the first row. Secondly, we
make grids in the pattern using gray pixels so that each
of the recovered patches is actually added with a
boarder. With the complementary pattern and the
added boarder, we can more certainly recover the
pixels belonging to a patch and use these pixels to find
the patch center as the simple point used in our
registration method. In Figure 5, we show some of the
camera pictures taken for the new patterns.
Figure 4. A few pattern selected form New pattern
scheme
464
Figure 5. Some of the pictures taken for the new
patterns
With the new pattern scheme, the centers of the
patches of the camera images can be determined using
the following procedures
1. Obtain the absolute value for each pair of
complimentary images i. i just represent the index of
the pairs and i ę Sn.
Ai(x, y) | Ii(x, y) - Iic(x, y) | (2)
Where Ii(x,y) represents the first image, the Iic(x,y)
represents its complementary image in the pair and Sn
represents the total pairs of the complimentary images .
Figure 3 (Right) shows an image which displaying the
Ai(x, y) for a pair of images. From this Figure, it can be
seen that the patches show high intensity, while the
boarder of each patch shows low intensity. This is
exactly the result from our pattern design. The
differences of complementary pixels produce high
value for Ai(x, y), while pixels around the patch board
gives low value for Ai(x, y) as they hold the same value
in both projector complimentary images and thus
produce very close value in the camera complimentary
images. As seen from the Figure, boarder pixels and
patch pixels are easier to be separated from each other.
This is important, as we pointed out before that
boarder pixels show high uncertainty. Using this way,
we can easily exclude these uncertain pixel and utilize
more certain pixels in finding the centers of the
patches.
2. As described in step 1, we can set a threshold
(we use 80 in our experiment) to easily separate
boarder pixels and patch pixels and assign 1 or 0 to
patch pixels according to following procedure:
If Ai(x, y) >T
If I1(x,y) – I2(x,y) > 0
I1(x,y) = 1
Else
I1(x,y) = 0
Else
Mark this pixel as boarder pixel
3. The coding scheme described before is used
to recover the pixels belonging to each patch based on
the detected certain pixels.
After recover the pixels belonging to each patch,
these pixels’ position can be averaged to find the
centers for each of the patch. However, in order to
compensate the drawback of using only a single
threshold in step 2 and improve the accuracy, we
propose to use the weighted patch pixel position to
achieve this aim as following:
4. Find the maximum value for Ai(x, y) as Aimax
5. Decide a certainty factor for each patch pixel as:
CF ( x , y ) ¦
CF i ( x , y ) / S n (3)
CFi ( x, y ) Ai(x, y) / Aimax
Where Sn is the number of pairs used.
6. The center of the patches can be computed as:
¦ CF ( x , y ) ˜ x
C kx For (x, y) ę k
¦ CF ( x , y )
¦ CF ( x, y ) ˜ y
C ky For (x, y) ę k
¦ CF ( x, y )
Where Ckx and Cky represent x and y coordinate of
the center for patch k.
After finding the correspondence of the centers
for both projector and camera images that would be
used as the sample points, we can interpolate to find
the correspondence for other projector and camera
pixels. In our work, the OpenGL texture mapping is
used for this purpose. In Figure 6, we show the result
of our projector and camera registration system. The
left top image is the projector image, The right image
is a camera image and the left bottom image is the
result after applying our geometric mapping. The
boarder of the image is empty because we do not
sample points in these regions. A future work for us is
to find out pixels correspondence in these regions.
Except these regions, projector pixels position and
camera pixel position are accurately related with each
other.
. Lasers and Electro-Optics Society, volume 1, pages
Figure 6. Registration results. The left top image is the
projector image; The right image is the camera image;
The left bottom right image is the result after applying
our registration method to the right image.
3. Conclusion
In our work, we propose a simple projector and
camera registration method, which makes no
assumption comparing with homography based method
(4) and can accommodate for geometric distortions that
may be caused by the optics of the projector and the
camera. It only needs to take a few images and simple
computation can achieve the accurate projector and
camera registration.
(5)
References
1. R. Sukthankar, R. Stockton, and M. Mullin.
Smarter Presentations: Exploiting Homography in
Camera-Projector Systems," Proceedings of the
(6) International Conference on Computer Vision, 2001.
2. S. Nayar, H. Peri, M. Grossberg, and P.
Belhumeur. A projection system with radiometric
compensation for screen imperfections. In IEEE
International Workshop on Projector-Camera Systems,
Oct. 2003.
3. M. Grossberg, H. Peri, S. Nayar, and Peter N.
Belhumeur. Making One Object Look Like Another:
Controlling Appearance using a Projector-Camera
System. Proc. Conf. on Computer Vision and Pattern
Recognition (CVPR), Washington, DC, IEEE
Computer Society, I, 452—459, 2004
4. S. M. Smith and J. M. Brady (May 1997).
SUSAN - a new approach to low level image
processing. International Journal of Computer Vision
23: 45-78.
5. M. Ashdown, M. Flagg, R. Sukthankar, J. M.
Rehg: A Flexible Projector-Camera System for Multi-
Planar Displays. CVPR (2) 2004: 165-172, 2004
6. R. Raskar, M.S. Brown, R. Yang, W. Chen, H.
Towles, B. Seales, and H. Fuchs. Multi-projector
displays using camera-based registration. In Proc.
IEEE Visualization, pages 161-168, Oct. 1999..
7. A. Majumder, Z. He, H. Towles, and G. Welch.
Achiev-ing color uniformity across multi-projector
displays. In T. Ertl, B. Hamann, and A. Varshney,
editors, Proc. IEEE Visualization, pages 117-124,
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8. K. Li and Y. Chen. Optical blending for multi-
projector display wall system. In IEEE Proc. 12th
281-282, 1999.
465
 2009 International Conference on Computer Engineering and Technology
Non-temporal Multiple Silhouettes in Hidden Markov Model for View Independent
Posture Recognition
Yunli Lee
Department of Media
Soongsil University
Seoul, South Korea
yunli@ssu.ac.kr
Abstract—This paper introduces a non-temporal multiple
silhouettes in Hidden Markov Model (HMM) for offering view
independent human posture recognition. The multiple
silhouettes are used to reduce the ambiguity problem of
posture recognition. A simple feature extraction of the 2D
shape contour based histogram is used for image encoding and
K-Means algorithm is applied for clustering and code-wording
of eight simple postures from multiple views. Therefore, 3D
volume reconstruction is not required, in return helps to
reduce the complexity of modeling and computational power of
feature extraction. HMM is trained to obtain view independent
recognition model using multiple silhouettes. A combination of
non-temporal multiple silhouettes, code-wording and HMM
methods in this proposed approach make it possible to
recognize human posture in view independent. The
experimental results demonstrate the effectiveness of non-
temporal multiple silhouettes in HMM for recognizing posture.
Non-temporal multiple silhouettes; Hidden Markov Model;
view independent; posture recognition
I. INTRODUCTION
Human gesture or action leads to the most active research
in computer vision community by offering several important
applications in our daily life, such as surveillance system,
human-computer interfaces, games, virtual reality and others.
These applications enable us to use hand or body gesture to
convey information to the system or to control the device
naturally without additional modal.
However, human body is not a rigid object and we could
present various kinds of gesture that composed by a
sequence of postures from each individual. Therefore,
posture is a basic element of human gesture or action. The
goal of our research is to recognize human posture in view
independent without restricting the actor or observer
position. Simple vision processing and feature extraction
supply an intuitive 2D shape contour based histogram, and
K-Means algorithm clusters the set of training posture
silhouette images. Then, the code-wording of eight simple
postures are trained by using the Hidden Markov Model
(HMM) which is supporting view independent posture
recognition.
In order to offer view independent posture recognition,
several approaches have been proposed. Feiyue et al. [1]
used stereo cameras of image pairs to propose a viewpoint
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.113
Keechul Jung
Department of Media
Soongsil University
Seoul, South Korea
kcjung@ssu.ac.kr
insensitive representation for action recognition. Each action
is represented in envelop shape, where this representation is
viewpoint insensitive under assumption of affine camera
projection model. Cen Rao [2] did a lot of related works on
view invariant analysis for human activity in his Ph.D. study.
He used trajectory of hand centroid to describe human action
that are performed by hand. Both Feng Niu and Mohamed
Abdel-Mottaleb [3], and Mohiuddin Ahmad and Seong-
Whan Lee [4], used multiple view image sequences of
combined features in Hidden Markov Model (HMM) for
human action recognition. They built HMM model for each
action in each viewing direction to characterize the variation
from the change of viewing direction. However, this
approach requires more HMM models of each action at each
viewpoint for view independent characteristic. Liu Ren et al.
[5] used extracted silhouettes from three video streams to
match the body configuration of the user against a database
of dance motion.
Different from previous works, we use synthetic models
in virtual environment that relies on silhouettes extracted
from seven virtual cameras to generate posture information
in various view. These clean non-temporal multiple
silhouettes are extracted automatically and clustered using
K-Means algorithm for code-wording purpose. The order of
cameras and code-wording are used to train the HMM for
posture recognition model where it can handles view
independent issue.
Section 2 describes the overview of our proposed HMM-
based view independent posture recognition system using
multiple silhouettes. We briefly explain the feature
representation and extraction of silhouette images in Section
3. A simple K-Means classification used to create the code
book is presented in Section 4. In Section 5, we introduce the
concept of HMM and how it is adapted to offer view
independent posture recognition. The experimental results of
posture recognition are discussed in Section 6. Some
thoughts about future work and conclusion are remarked in
Section 7.
II. PROPOSED SYSTEM ARCHITECTURE
The system architecture of this proposed system consists
of four parts. It contains a virtual studio of 3D model posture
capturing module which automatically generates a sequence
of non-temporal multiple silhouette images from seven
virtual cameras; a 2D shape representation and extraction
466
module that extracts the contour points and computes the A. Multiple Views Posture Databases
gradient point accumulation based on histogram bins Our multiple views posture database comprises of eight
template from silhouette image; a K-Means classification postures as follows
module that creates code-wording for non-temporal multiple 1. T Shape,
silhouette images of defined postures for training the HMM 2. Stand,
models; and a posture recognition module which uses Baum
3. Both Hands Front,
Welch and forward algorithm [6-7] for training and
4. Both Hands Up,
evaluating the input sequence of non-temporal multiple
silhouettes. The overview of the proposed system is 5. One Hand Up,
illustrated in Figure 1. 6. T Shape Foot Up,
In the virtual studio of 3D model posture capturing 7. T Shape Bend,
module, seven virtual cameras are used to capture the 3D 8. Crawl.
model posture. Seven units of camera are placed around the Each posture has been acquired from seven viewing
ceiling view. This module automatically generates seven position with 51 degree apart. Each posture was executed by
clean silhouettes from each camera view. Each posture is seven synthetic 3D models for six times. The sequence of
represented with seven non-temporal views in clockwise camera order is important to represent the posture
order where the cameras’ position is about 51 degree apart. information in view independent. However, we do not fix the
The contour points are extracted from the silhouette image starting position of the camera sequence. As long as the
and the gradient of each point on the contour to the center cameras sequence order is concerned, the starting position
point is computed and accumulated into 12 bins template. A does not affect the learning performance. The database
symbol which corresponds to a code-word in the code book comprises 2352 images. As a whole, eight HMMs are
created by K-Means is assigned to each silhouette image. A created to model eight postures that support view
sequence of code-word is formed by seven camera’s view independent using minimum number of cameras for training.
and used to train the HMM models. Without restricting the
III. FEATURE EXTRACTION
start position of the camera view and actor, six times
permutation of the sequence training dataset is applied. We assume clean silhouettes are extracted from the
Baum Welch algorithm [6-7] is used to obtain the HMM system. A simple posture representation of silhouette image
parameters from the training dataset. is used to extract the feature vectors. In this work, we extract
the posture contour from each silhouette, while the center
point of the posture contour is computed as a reference
center point of 12 bins template. The gradient between each
contour point and center is computed, and each bin from the
defined template accumulates the number of contour points
that overlaid on the corresponding bin. The feature named as
gradient point accumulation is extracted from contour points
of silhouette image. Let, Sv = {p1, p2, …, pn} be the contour
of silhouette image of a posture for each camera, where v is a
camera index, pi= (pxi, pyi) is a contour point, and n is the
total number of contour points, which is extracted using
boundary following algorithm. The center point cx,y, of the
contour is obtained by the following equation from Yiğithan
Dedeoğlu et al. [10]:

⎛ ∑n pxi ∑n pyi ⎞
4 3
5 2

1
cx, y = ⎜ i =1 , i =1 ⎟ (1)
⎜ ⎟
6

12
n n
⎝ ⎠
7

8
11
9 10

Fe a t u re s Re pre s e n ta ti o n f o r T S h a pe Po s tu re o f C 3 Vi e w
A 12 bins template is defined with 30 degree apart from
0.25

each bin. Then, the gradient G, between the center point cx,y
nts

0.2
do
Pi

0.15
m z
ie

of contour with respect to each contour point on the contour

a
l

0.1
o
Nr

0.05
0
1 2 3 4 5 6
Bin
7 8 9 10 11 12
is computed using below equation:

G ( p[i ], c x , y ) = arctan ( pyi −c y

pxi −c x ), ∀i ∈ [1, " , n] (2)

Human varies in shape and size. As such, we propose a

Figure 1. Overview of HMM-based view independent human posture non-temporal multiple silhouettes to represent a posture
recognition system. because each silhouette may have a different contour shape.

467
Therefore, normalization is required to standardize the
extracted features. The normalized accumulation of gradient
point for 12 bins template, B = {b1, b2, …, b12} is shown in
Figure 2. B is bin with 12 dimensions feature vectors that are
used to classify the contour shape of each silhouette image.
Fe atu re s Re pre se n tati on for T S h ape Postu re of C 3 Vi e w
0.25
Normalized Points
0.2
0.15
0.1
0.05
0
1 2 3 4 5 6
Bin
Figure 2. Bins distribution template and 2D shape representation graph of
T Shape.
IV. MAPPING FEATURE TO SYMBOL
We propose seven multiple silhouettes to represent a
posture, where each silhouette consists of 12 dimension
feature vectors. We map each silhouette image into code-
wording representation that forms a sequence of symbols for
a posture in seven viewpoints using K-Means clustering.
This result in a sequence of symbols represented as a posture
and used to train the HMM model subsequently. Hence, a
simple K-Means algorithm is applied to cluster the training
feature vectors posture into K clusters, which equals to the
size of code book. Consequently, the size of K clusters
equivalent to the number of HMM output symbols, M.
A. K-Means and Code-wording
Simple K-Means algorithm [8] was introduced for
clustering, where K is the number of cluster that cluster the
data using means information from given dataset. For the
silhouettes clustering, we initialized the means value of K
corresponds to the defined postures. Since K is equivalent to
M, these K symbols are used to generate code book for
assigning a sequence of code-wording symbols for defining
the input postures dataset to train the HMM models. Each
silhouette is transformed into the symbol which is assigned
to the cluster which is nearest to the means vector of the
feature space.
For posture recognition, we select non-temporal multiple
silhouettes of feature vectors for representative camera views
from each posture as code-word in the code book. The
extracted feature vectors would be mapped to a symbol,
which code-word of the most similar (minimal error) feature
vectors in the code book. Figure 3 shows the mapping result
for the defined postures from the clustering result of K-
Means, where K is defined as 56 by assuming the training
dataset of eight postures has seven different views for each
posture. The clustering result of simple K-Means is
reasonable even though some postures are mistaken.
7 8 9 10 11 12
Figure 3. The defined postures and symbols of K-Means clustering.
V. HIDDEN MARKOV MODEL
Hidden Markov Models (HMM) is a probabilistic state
transit model in which the system being modeled is assumed
to be a Markov process where the states sequence could not
be observed directly. There are widespread approaches to
probabilistic sequence modeling [6-7]. HMM is well known
as transition system which is able to attempt image
understanding through model of patterns. It can deal with
transitions in time, and also transition through another
pattern. Thus, by taking this advantage of transition through
another pattern, we learn a sequence of non-temporal
multiple silhouettes from different views of multiple cameras
for a defined posture that offers view independent human
posture recognition system.
The basic parameters of HMM with N states with M
observation symbols can be specified by λ = ( A, B, π )
where λ represents the HMM model. The detailed
explanation of HMM model parameters are as below:
z N is the number of states in the model where the states
are hidden
z M is the number of observation symbols correspond to
the physical output of the system being modeled
z A is a state transition probability matrix
z B is an observation probability for each state
468
z π is an initial state distribution
We propose only eight HMMs for recognizing the eight
defined posture where each model can offer view
independent by using a sequence of non-temporal silhouette
images from multiple cameras. The set of parameters of
HMM λ j , where j is number of HMMs of the proposed
recognition system, is learned by using Baum-Welch
algorithm [6-7] from the training dataset. To recognize a
given posture O, we evaluate P(O | λ j ) with forward
algorithm [6-7] for each j models and choose the maximum
probability of model c to recognize the posture:
Another testing dataset has been created using two new
3D synthetic models. The postures were captured two times
in same camera views and six times in new camera views.
Figure 4 shows two sets of test data for T Shape posture
which are generated from the new models in different
camera views and classified to obtain input observation
symbols. Table 2 shows the recognition results of testing
dataset. The mean accuracy of this proposed recognition
model is 89.8%.

c = arg max j P(O | λ j ) (3)

A. View Independent HMM

View independent posture recognition enables user to
perform posture command naturally without restricting the Figure 4. Two sets of new synthetic models with different camera views
observation position. We have chosen seven cameras to be of T Shape posture.
placed around the working space for capturing the posture.
Non-temporal sequence of images is ordered in clockwise- TABLE II. CONFUSION MATRIX FOR POSTURE TESTING MODELS
order and six times permutation of view sequences has been
applied to represent a posture where the user can perform the Posture 1 2 3 4 5 6 7 8
posture freely without any restriction. 1 14 0 0 0 0 2 0 0
These sequences are then used for training in HMM. The 2 0 15 1 0 0 0 0 0
sequence of cameras is important for HMM to understand 3 2 1 13 0 0 0 0 0
the posture in various views. HMM is able to learn the 4 2 0 0 14 0 0 0 0
relation between each view image of each observed camera 5 0 0 0 0 16 0 0 0
in order to support the view independent condition. 6 4 0 0 0 0 12 0 0
7 0 0 0 0 0 1 15 0
VI. EXPERIMENTAL RESULTS 8 0 0 0 0 0 0 0 16
The proposed framework has been tested using synthetic
dataset. The system has been trained using training data from VII. CONCLUSION AND FUTURE WORK
seven 3D models with different size and gender. The models
have been obtained from the 3D model database [9] and This work proposes an efficient 3D human model posture
Poser software application. Eight postures as defined in recognition using non-temporal multiple silhouettes without
Section 2.1 have been created using biped bone of 3D max any 3D volume reconstruction process. The contour points of
studio application for 3D models. silhouettes are extracted to represent the human posture.
We have used synthetic data to evaluate the accuracy of Each posture is composed from seven multiple contours of
proposed system. The synthetic postures dataset are divided silhouette views which are represented in gradient point
into two sets, one for training dataset, and another one is the accumulation feature. The gradient point accumulation is
testing dataset. Table 1 shows the confusion matrix obtained allocated into 12 bins template. A set of training defined
with five states of HMMs that learned from seven synthetic postures are clustered into K symbols that corresponds to a
3D models. The mean accuracy of the proposed recognition code-word of code book created by K-Means classification.
model is 97.7%. Seven sequences symbol of code-wording expressing a
defined posture. The recognition rate achieves average
97.7% of training models and 89.8% of new test models. The
TABLE I. CONFUSION MATRIX FOR POSTURES TRAINING MODELS HMM-based non-temporal multiple silhouettes show a
potential result for view independent human posture
Posture 1 2 3 4 5 6 7 8
recognition.
1 49 0 0 0 0 0 0 0 For future work, we plan to apply the non-temporal
2 0 47 0 2 0 0 0 0 multiple silhouettes of contour point features in HMM for
3 0 0 49 0 0 0 0 0
view independent posture recognition in real environment.
4 0 0 0 42 7 0 0 0
5 0 0 0 0 49 0 0 0
More postures definition and test data are required for
6 0 0 0 0 0 49 0 0 evaluating the proposed features and HMM-based multiple
7 0 0 0 0 0 0 49 0 silhouettes on the robustness of view independent human
8 0 0 0 0 0 0 0 49 posture recognition system.

469
 ACKNOWLEDGMENT
This work was supported by the ‘Seoul R and BD
Program (10581cooperateOrg93112)’.
REFERENCES
[1] Feiyue Huang, Huijun Di, Guangyou Xu, “Viewpoint Insensitive
Posture Recognition for Action Recognition,” Proceedings of
Articulated Motion and Deformable Objects (AMDO 2006), LNCS,
Vol. 4069, pp. 143-152, 2006.
[2] Cen Rao, “Invariance in Human Action Analysis,” Ph.D.
Dissertation of University of Central Florida, 2003.
[3] Feng Niu, Mohamed Abdel-Mottaleb, “View-Invariant Human
Activity Recognition Based on Shape and Motion Features,” IEEE
Sixth International Symposium on Multimedia Software Engineering,
pp. 546-556, 2004.
[4] Mohuiddin Ahmad, Seong-Whan Lee, “HMM-based Human Action
Recognition Using Multiview Image Sequences,” 18th International
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Conference on Pattern Recognition (ICPR 2006), Vol. 1, pp. 263-266,
2006
[5] Liu Ren, Gregory Shakhnarovich, Jessica K. Hodgins, Hanspeter
Pfister, and Paul Viola, “Learning Silhouette Features for Control of
Human Motion,” ACM Transactions Graphics, Vol. 24, No. 4, pp.
1303-1331, October 2005.
[6] Lawrence R. Rabiner, “A tutorial on Hidden Markov Models and
Selected Applications in Speech Recognition,” Proceedings of the
IEEE, vol. 77, no. 2, pp. 257-286, 1989.
[7] Lawrence R. Rabiner, B.H. Juang, “An Introduction to Hidden
Markov Models,” IEEE ASSP Magazine, vol. 3, pp. 4-16, 1986.
[8] Vitorino Ramos, Fernando Muge, “Map Segmentation by Colour
Cube Genetic K-Mean Clustering,” Proceedings of the ECDL, 2000.
[9] INRIA Gamma team research database, http://www-
c.inria.fr/gamma/gamma.php
[10] Yiğithan Dedeoğlu, B. Uğur Töreyin, Uğur Güdükbay, A. Enis Çetin,
“Silhouette-Based Method for Object Classification and Human
Action Recognition in Video,” 9th European Conference on
Computer Vision (ECCV 2006), LNCS, vol. 3979, pp. 64-77, 2006.
 2009 International Conference on Computer Engineering and Technology

Classification of Quaternary [21s + 1, 3] Optimal Self-orthogonal Codes

Xuejun Zhao(1),(2) Ruihu Li(2) Yingjie Lei(1)

(1) Department of Computer Science,College of Missile,Air Force Engineering University Sanyuan,Shaanxi 713800,PRC
(2) Department of Mathematics,College of Science, Air Force Engineering University Xi’an,Shaanxi 710051,PRC
e-mail:zlycczlycc@163.com; liruihu2008@yahoo.com.cn; lei@ieee.org

Abstract orthogonal codes over F4 with s ≥ 1 and t ∈ {0, 16, 19}

The classification of quaternary [21s + t, 3, d] codes with
d ≥ 16s and without zero coordinates is reduced to the
classification of quaternary [21c(3, s, t) + t, k, d] code for
s ≥ 1 and 0 ≤ t ≤ 20, where c(3, s, t) ≤ min{s, 3t} is
a function of 3, s, and t. Quaternary optimal Hermitian
self-orthogonal codes are characterized by systems of
linear equations. Based on these two results, the complete
classification of [21s + 1, 3] optimal self-orthogonal codes
for s ≥ 1 is obtained, and the generator matrices and
weight polynomials of these 3-dimensional optimal self-
orthogonal codes are also given.
meeting the Griesmer bound are unique under equivalence.
In this paper, we give complete classification of [21s+1, 3]
optimal self-orthogonal codes over F4 with s ≥ 1.
Let F4 = {0, 1, ω, $} be the Galois field with four
elements such that $ = 1 + ω = ω 2 , ω 3 = 1, and the
conjugation is defined by Γ(x) = x̄ = x2 . Let F4n be
the n−dimensional row vector space over the quaternary
field F4 . A k−dimensional subspace C of F4n is called a
quaternary linear [n, k] code.
The Hermitian inner product of u, v ∈F4n is defined as
n
X n
X
(u, v)h = uv T = uj vj = uj vj2 .
1 1

Keywords: quaternary linear code; self-orthogonal code; The Hermitian dual code C of C is defined as C ⊥ = {u ∈
⊥

optimal code; Griesmer bound.
1. Introduction
Since the pioneer work of MacWillams et.al. in [1] in
1978, people paid much attention on quaternary Hermitian
self-dual codes—a subclass of self-orthogonal codes ( SO
codes, for short) over the quaternary field F4 , and a vast
number of paper have been devoted to the study of quater-
nary self-dual codes, see the excellent survey of [2] and [3]
for an overview of these results and the references therein.
Recently, people begin to study optimal [n, k] self-
orthogonal codes of rate less than 12 and use these optimal
[n, k] SO codes to study quantum error-correcting codes and
self-dual codes, see [4-5]. In [6], Bouykliev et al. studied the
classification of binary optimal SO codes of length ≤ 40 and
dimension less than 10, and gave complete classification
of 3-dimensional optimal SO codes. In [7] Bouykliev et.
al. studied the classification of optimal SO codes of length
n ≤ 29 and dimension less than 7 over F3 and F4 , and
give complete classification of [n, 3] optimal self-orthogonal
codes over F4 for 6 ≤ n ≤ 29. Their results of optimal SO
codes over F4 show that some optimal SO codes meeting
the Griesmer bound are unique under equivalence.
In [8], we studied the classification of binary [15s + t, 4]
optimal SO codes for s ≥ 1 and t ∈ {1, 2, 6, 7, 8, 9, 13, 14}.
In [9], Ruihu Li proved that [21s + t, 3] optimal self-
F4n | (u, v)h = uv T = 0 for all v ∈ C}. A code C is self-
orthogonal if C ⊆ C ⊥ , and self-dual if C = C ⊥ . A quaternary
code is SO if and only if it is even,i.e. the weights of all
codewords in C are even, see [4] and [10]. An [n, k] SO
code C is called optimal if it has the highest weight among
all [n, k] SO codes.
Two quaternary codes C and C 0 are equivalent if one can
be obtained from the other by permuting the coordinates,
multiply some coordinates by non zero elements, or conju-
gate all the coordinates. If two matrices G and G0 generate
equivalent codes, we denote them as G ∼ = G0 .
We use 1n =(1, 1, · · · , 1)1×n and 0n =(0, 0, ..., 0)1×n to
denote the all-ones vector and the zero vector of length n,
respectively. And use iG = (G, G, · · · , G) to denote the
juxtaposition of i copies of G for given matrix G.
2. Quaternary Self-orthogonal Codes and Sys-
tems of Linear Equations
In order to present our main results, we need some
preparations.
A row (or column) vector is monic if its first non
zero coordinate is 1. Let α1 =(0, 1, 0)T , α2 =(1, 0, 0)T ,
α3 =(1, 1, 0)T , α4 =(1, ω, 0)T , α5 =(1, $, 0)T , α6 =(0, 0, 1)T ,
α7 =(0, 1, 1)T , α8 =(1, 0, 1)T , α9 =(1, 1, 1)T , α10 =(1, ω, 1)T ,
α11 =(1, $, 1)T , α12 =(0, 1, ω)T , α13 =(1, 0, ω)T ,
T T
α14 =(1, 1, ω) , α15 =(1, ω, ω) , α16 =(1, $, ω)T ,

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 471

DOI 10.1109/ICCET.2009.93
α17 =(0, 1, $)T , α18 =(1, 0, $)T , α19 =(1, 1, $)T , of G and C the code corresponding to LG . Let Y T = A3 LTG ,
T T
α20 =(1, ω, $) , and α21 =(1, $, $) . Then all the where Y = (y1 , y2 , ..., y21 ). Then Y = (y1 , y2 , ..., y21 ) is a
monic vectors of dimension 3 are these 21 vectors. projective weight vector of C. Since GL(3, F4 ) acts double
Definition 2.1 For a linear code C= [n, k, d] with genera- transitively on {α1 , · · · , α21 }, hence, in the following, we
tor matrix G, a matrix whose rows are formed by all monic can assume l1 ≥ l2 ≥ li , i = 3, 4, · · · , 21, l6 ≥ rj , j =
vectors of C is called a projective matrix of C and denoted 7, · · · , 21 as in [6], and without further explain.
as P (C) or P (G). Let P (G) = (β1T , β2T , · · · , βN T T
k
) and Let C be an [21s + t, 3, 16s + dt ] SO code with generator
W (C) = (w(β1 ), w(β1 ), · · · , w(βNk )) = (x1 , x2 , ..., xNk ), matrix G and define vector L = (l1 , l2 , ..., l21 ), where s ≥ 1,
W (C) is called a projective weight vector of C, and 1 ≤ t ≤ 20. Using the Griesmer bound, one can deduce
x1 , x2 , · · · , xNk are called projective weights of C. an upper bound of dt , hence the projective weight vector
If P = (aij ) is a matrix over F4 , its projection (P )p = of C can be assumed as Y = (y1 , y2 , ..., y21 ), where yi =
(bij ) is a binary matrix, where bij = 0 if aij = 0 and bij = 1 16s + dt + 2λi and λi ≥ 0 for 1 ≤ i ≤ 21. Thus we have
otherwise. the following equations
Fix G3 = (α1 , α2 , ..., α21 ), then G3 is a generator matrix 
l1
 
y1

of the Simplex code C3 = [21, 3, 16].  l 2   y2 
Let P2 and A2 = (P2 )p as Ak  .  =  . 
   
(1)
 ..   .. 
 
0 1 1 1 1 l21 y21
 1 0 1 ω $ 
  Then, for given Y , an [21s + t, k, 16s + dt ] SO code with
P2 =  1 1 0 $ ω ,

 1 ω $ 0 1  projective weight vector Y exist if and only if the above
linear equations (1) has non-negative integer solution.
1 $ ω 1 0
Let L0G = LG − s121 . Then, from Y T = A3 LT , one can
deduce that 21dt + 2(λ1 + λ2 + · · · + λ21 ) = 43−1 (l10 + l20 +
 
0 1 1 1 1
0
 1 0 1 1 1  · · · + l21 ), thus 2(λ1 + λ2 + · · · + λ21 ) = 43−1 t − 21dt . Let
Λ = (λ1 , λ2 , · · · , λ21 ), dt = 2at , δt = λ1 + λ2 + · · · + λ21 .
 
A2 =  1 1 0 1 1 .

 1 1 1 0 1  Then an [21s + t, 3, s43−1 + dt ] SO code with projective
1 1 1 1 0 weight vector Y exist if and only if the following system
of linear equations have integer solution L0G such that each
And, let J5 be the 5 × 5 all one matrix, denote P2 = P2,0 , s + li0 is non-negative.
P2,1 = P2 +J5 , P2,2 = P2 +ωJ5 and P2,3 = P2 +$J5 , their
 A3 (L0G )T = 2(at 1T
 T
projections are denoted as A2 = (P2 )p and A2,i = (P2,i )p , 21 + Λ ),
1 ≤ i ≤ 3, respectively. 2δt = 43−1 t − 21dt , (2)
λi ≥ 0, i = 1, 2, · · · , 21.

From the generator matrix G3 of C3 , we have P3 = P (C3 )
and A3 = (P3 )p as follow: Our classification of [21s+t, 3] optimal SO codes is based
  on the following lemma given in [9], its proof is similar to
P2 0 P2 P2 P2 the proof of Theorem 1.1 given by [8].
 0 1 15 ω15 $15 
 T
 Lemma 2.1 Suppose , s ≥ 1, 0 ≤ t ≤ 20 and n =
P3 =  P2 1 5 P 2,1 P2,2 P2,3  , 21s + t. Then, every [n, 3, d] quaternary code C with d ≥
 P2 ω15 T P2,2 P2,3 P2,1 
16s and without zero coordinates is equivalent to a code
P2 $15 T P2,3 P2,1 P2,2 with generator matrix G = ((s − c(3, s, t))G3 , H), where

A2 0 A2 A2 A2
 c(3, s, t) ≤ min{s, 3t} is a function of 3, s and t, and H
 0 1 15 15 15  has 21c(3, s, t) + t columns.
 T
 According to this lemma, the classification of [n, 3] opti-
A3 =  A2 15
 A2,1 A2,2 A2,3  .
 A2 15 T A2,2 A2,3 A2,1  mal codes is changed into determining c(3, s, t), and can be
reduced to the classification of [m + t, 3] optimal SO codes
A2 15 T A2,3 A2,1 A2,2
for m ≤ 21c(3, s, t).
Now, we can use systems of liner equations to characterize For given s ≥ 1 and 1 ≤ t ≤ 20. Denote the set of all
SO codes of given minimum distance as in [8]. the generator matrix G (determined above) of [21s + t, 3]
From the definition of equivalence of quaternary linear optimal SO codes as G[21s + t, 3], and let D[21s + t, 3]
codes, we can suppose an [n, 3] code C with generator matrix = {LG is the define vectors of G, G ∈ G[21s + t, 3]}. Then
G, where the columns of G are all monic. If the columns c(3, s, t) = −min1≤i≤21 {li − s | LG ∈ D[21s + t, 3]}.
of G have li copies of αi for 1 ≤ i ≤ 21, we denote G as Thus, this explains how to determine the generator matrices
G = (l1 α1 , l2 α2 , . . . , l21 α21 ) for short and call such G in of all [n, 3] optimal SO codes and how one can determine
order form, and call LG = (l1 , l2 , · · · , l21 ) the define vector c(3, s, t).

472
3. Results of Classification
In this section, we will use the results of the above two
sections to study the classification of optimal [n, 3] SO codes
for n = 21s + 1, s ≥ 1. Let M (n, 3) be the number of
non equivalent optimal [n, 3] SO codes, and M0 (n, 3) and
M1 (n, 3) be the number of non equivalent optimal [n, 3] SO
codes with zero coordinates and without zero coordinates,
respectively.
From [9], we know that there is only one [21s, 3, 16s],
hence, for n = 21s + 1, M0 (n, 3) = 1 and M (n, 3) =
M1 (n, 3) + 1. To save space, we only explain our classifi-
cation process for several [22, 3] optimal SO codes without
zero coordinates, other cases can be deduced similarly.
Let G and G0 be two 3×n matrices with define vectors L
and L0 , respectively. From the definition of equivalence of
quaternary code, we know that G and G0 generate equivalent
codes if and only if gG0 Dσ= G or = Γ(G), where g ∈
GL(3, F4 ), D is an n × n invertible diagonal matrix, σ is
a permutation that change gG0 D in order form and Γ is the
conjugate map.
A [21s + 1, 3] optimal SO code has minimum distance
16s + 2. Using MATLAB[11] program, one can easily check
that there are 102 solutions satisfying equation (1) if s = 1,
there are 381 solutions if s = 2, and there are 528 solutions
if s = 3. Thus, we give our classification results in three
cases.
Case 1.1 For s = 1, there are 102 solutions satisfying
equation (1), denote these 102 solutions as L1,i , 1 ≤
i ≤ 102. These 102 L1,i can be divided into four groups
according to the weight polynomials of their corresponding
codes. The first group are L1,i , 1 ≤ i ≤ 3; the second
group are L1,i , 4 ≤ i ≤ 12; the third group are L1,i ,
13 ≤ i ≤ 30; the fourth group are L1,i , 31 ≤ i ≤ 102. Two
codes with define vector in different groups have different
weight polynomials.
The three L1,i , 1 ≤ i ≤ 3, in the first group are as
follow:
L1,1 =(2,2,2,0,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1),
L1,2 =(2,2,0,2,0,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1),
L1,3 =(2,2,0,0,2,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1,1),
Let the code with define vector L1,i be C i and its generator
matrix be Hi , 1 ≤ i ≤ 3. Then H1 =(2α1 ,2α2 ,2α3 ,α6 ,...,α21 ),
H2 =(2α1 ,2α2 ,2α4 ,α6 ,...,α21 ). It is not difficulty to check that
   
100 100
 0ω0  H1 Dσ = H2 , g =  0ω0 
001 001
where D = diag{$, $, 1, 1, 1, 1, 1, $, 1, 1, 1, 1, $, 1, 1,
1, 1, $, 1, 1, 1, 1}, σ is a permutation that change gH1 D in
order form. Hence C 2 and C 1 are equivalent, they all have
weight polynomial 1 + 39y 16 + 24y 18 . Similarly, one can
check that C i and C 1 are equivalent for 1 ≤ i ≤ 3.
473
The nine L1,i , 4 ≤ i ≤ 12, in the second group are as
follow:
L1,4 =(2,2,1,0,1,2,1,1,1,1,1,0,1,1,1,1,1,0,1,2,1),
L1,5 =(2,2,1,1,0,2,1,1,1,1,1,1,0,1,1,2,0,1,1,1,1),
L1,6 =(2,2,1,1,0,2,1,0,1,1,2,0,1,1,1,1,1,1,1,1,1),
L1,7 =(2,2,0,1,1,2,1,1,1,1,1,1,1,1,1,1,0,0,2,1,1),
L1,8 =(2,2,0,1,1,2,1,1,1,1,1,0,0,2,1,1,1,1,1,1,1),
L1,9 =(2,2,1,0,1,2,1,0,1,2,1,1,1,1,1,1,0,1,1,1,1),
L1,10 =(2,2,1,1,0,2,0,1,1,1,1,1,1,1,1,1,1,0,1,1,2),
L1,11 =(2,2,1,0,1,2,0,1,1,1,1,1,0,1,2,1,1,1,1,1,1),
L1,12 =(2,2,0,1,1,2,0,0,2,1,1,1,1,1,1,1,1,1,1,1,1),
Let the code with define vector L1,i be C i and its
generator matrix be Hi , 4 ≤ i ≤ 12. Then
H4 =(2α1 ,2α2 ,α3 ,α5 ,2α6 ,α7 ,...,α11 ,α13 ,...,α17 ,α19 ,2α20 ,α21 ),
H5 =(2α1 ,2α2 ,α3 ,α4 ,2α6 ,α7 ,...,α12 ,α14 ,α15 ,2α16 ,α18 ,α19 ,α20 ,α21 ),
It is not difficulty to check that
   
100 100
 001  H5 D0 σ 0 = H4 , g 0 =  001 
010 010
where D0 = diag{1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, $, 1, ..., 1},
σ 0 is a permutation that change g 0 H5 D0 in order form.
Hence C 4 and C 5 are equivalent, they all have weight
polynomial 1 + 42y 16 + 18y 18 + 3y 20 . Similarly, one can
check that C i and C 4 are equivalent for 4 ≤ i ≤ 12.
The third group and the forth group can be discussed as
the above two groups similarly.
Let H1,1 = H1 , H1,2 = H4 , and let
H1,3 =(2α1 ,2α2 ,α3 ,α4 ,2α6 ,α7 ,2α9 ,α10 ,α11 ,α13 ,α14 ,2α16 ,α17 ,
α18 ,2α20 ,α21 ),
H1,4 =(2α1 ,2α2 ,2α3 ,2α6 ,2α7 ,2α10 ,2α11 ,α13 ,...,α16 ,α18 ,...,α21 ),
It is not difficulty to check that the codes with define
vector in the i−th group are equivalent to the code with
generator matrix H1,i , 1 ≤ i ≤ 4.
Case 1.2 For s = 2, there are 381 solutions satisfying
equation (1), denote these 381 solutions as L2,i = 2121 +
L02,i , 1 ≤ i ≤ 381. These 381 L2,i can be divided into
6 groups according to the weight polynomials of their
corresponding codes or the characteristics of L02,i . Two codes
with define vector in different groups have different weight
polynomials or their L02,i have different characteristics.
The first 102 solutions satisfying L2,i = 121 +L1,i , 1 ≤
i ≤ 102. Hence, these 102 solutions can be divided into four
groups according to Case 1.1. The other 279 solutions can be
divided into 2 groups, named as the fifth group and the sixth
group. These 2 groups are as follow: L2,i , 103 ≤ i ≤ 156;
L2,i , 157 ≤ i ≤ 381;
Let H2,i = (G3 , H1,i ), 1 ≤ i ≤ 4, and let
H2,5 =(3α1 ,3α2 ,2α3 ,2α4 ,α5 ,3α6 ,3α7 ,2α8 ,α9 ,2α10 ,3α11 ,α13 ,3α14 ,
2α15 ,...,2α21 ),
H2,6 =(3α1 ,3α2 ,3α3 ,2α4 ,3α6 ,3α7 ,α8 ,α9 ,3α10 ,3α11 ,2α13 ,3α14 ,α15 ,
2α16 ,2α17 ,2α18 ,α19 ,2α20 ,3α21 ),
It is not difficulty to check that the codes with define vectors
in the i−th group are equivalent to the code with generator
matrix H2,i , 1 ≤ i ≤ 6. Two codes in different groups are
inequivalent.
Case 1.3 For s = 3, there are 528 solutions satisfying (1),
denote these 528 solutions as L3,i = 3121 + L03,i , 1 ≤ i ≤
528. These 528 L3,i can be divided into 11 groups according
to the weight polynomials of their corresponding codes or
the characteristics of L03,i . Two codes with define vector in
different groups have different weight polynomials or their
L03,i have different characteristics.
The first 381 solutions satisfying L3,i = 121 +L2,i , 1 ≤
i ≤ 381. Hence, these 381 solutions can be divided into
six groups according to Case 1.2. The other 147 solutions
can be divided into 5 groups, named as the seventh group
to the eleventh group. These 5 groups are as follow: L3,i ,
382 ≤ i ≤ 384; L3,i , 385 ≤ i ≤ 411; L3,i , 412 ≤ i ≤ 483;
L3,i , 484 ≤ i ≤ 519; L3,i , 520 ≤ i ≤ 528.
Let H3,i = (G3 , H2,i ), 1 ≤ i ≤ 6, and let
H3,7 =(4α1 ,4α2 ,4α3 ,4α4 ,3α6 ,...,3α21 ),
H3,8 =(4α1 ,4α2 ,3α3 ,3α4 ,2α5 ,4α6 ,4α7 ,2α8 ,3α9 ,3α10 ,4α11 ,3α13 ,
4α14 ,2α15 ,3α16 ,4α17 ,3α18 ,2α19 ,4α20 ,3α21 ),
H3,9 =(4α1 ,4α2 ,4α3 ,3α4 ,α5 ,4α6 ,4α7 ,α8 ,3α9 ,4α10 ,4α11 ,4α13 ,4α14 ,
α15 ,3α16 ,4α17 ,3α18 ,1α19 ,4α20 ,4α21 ),
H3,10 =(4α1 ,4α2 ,4α3 ,2α4 ,2α5 ,4α6 ,4α7 ,2α8 ,2α9 ,4α10 ,4α11 ,4α13 ,
4α14 ,2α15 ,2α16 ,4α17 ,2α18 ,2α19 ,4α20 ,4α21 ),
H3,11 =4(α1 ,...,α4 ,α6 ,α7 ,α9 ,...,α11 ,α13 ,α14 ,α16 ,...,α18 ,α20 ,α21 ),
It is not difficulty to check that the codes with define
vectors in the i−th group are equivalent to the code with
generator matrix H3,i , 1 ≤ i ≤ 11. Two codes in different
groups are inequivalent.
From the above discussion, one can have
Theorem 3.1 If n = 21s + 1, s ≥ 1, then the following
holds:
(1) If s = 1, then M1 (22, 3) = 4. The four non equivalent
[22, 3, 16] optimal SO codes without zero coordinates have
generate matrices H1,i , 1 ≤ i ≤ 4, and have weight polyno-
mials W22,i (y) as 1+39y 16 +24y 18 , 1+42y 16 +18y 18 +3y 20 ,
1 + 45y 16 + 15y 18 + 3y 22 , 1 + 51y 16 + 12y 20 , respectively.
(2) If s = 2, then M1 (43, 3) = 6. The 6 non equivalent
[43, 3, 32] optimal SO codes without zero coordinates have
generate matrices H2,i , 1 ≤ i ≤ 6, and have weight polyno-
mials W43,i (y) as follow: W43,i (y) = 1+y 16 (W22,i (y)−1),
1 ≤ i ≤ 4, and 1 + 51y 32 + 12y 36 , 1 + 54y 32 + 6y 36 + 3y 40 ,
respectively.
(3) If s ≥ 3, then M1 (n, 3) = 11. The 11 non equiva-
lent [n, 3, 16s] optimal SO codes without zero coordinates
have generate matrices Hs,i , 1 ≤ i ≤ 11, and have
weight polynomials Wn,i (y) as follow: Wn,i (y) = 1 +
y 16(s−2) (W43,i (y) − 1), 1 ≤ i ≤ 6, 1 + 51y 16s + 12y 16s+4 ,
1 + 54y 16s + 6y 16s+4 + 3y 16s+8 , 1 + 57y 16s + 3y 16s+4 +
3y 16s+12 , 1 + 57y 16s + 6y 16s+8 , 1 + 60y 16s + 3y 16s+16 ,
respectively.
474
4. Conclusion
We have given the complete classification of quaternary
[21s+1,3] optimal SO codes for s ≥ 1, our results of [22, 3]
optimal SO codes are concordant with the results of [7]. Our
classification method given in Section 2 differ from that of
[6] and can be generalized to [n, k] optimal SO codes for
k ≥ 3.
5. Acknowledgement
This work is supported by Nation Natural Science
Foundation of China under Grant No.60573040, and Nation
Natural Science Foundation of Shannxi Province.
References
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W.C.Huffman(Eds.), Handbook of Coding Theory, Elsevier,
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[3] W.C.Huffman, On the classification and enumeration of self-
dual codes, Finite Fields Appl. vol.11, 451-490, 2005.
[4] A. R. Calderbank, E. M. Rains, P. W. Shor, and N. J. A. Sloane,
Quantum error-correction via codes over GF(4), IEEE Trans.
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[5] S.Bouyuklieva, Some Optimal Self-Orthogonal Codes and
Self-dual Codes, Discrete Math., vol.287:1-10, 2004.
[6] I.Bouyukliev, S.Bouyuklieva, T.A.Gulliver and P.Ostergard,
Classification of Optimal Binary Self-Orthogonal Codes. [On-
line]. Available: Http://users.tkk.fi/ pat/patric-pub-html.
[7] I. Bouyukliev, P.Ostergard, Classification of self-orthogonal
codes over F3 and F4 , SIAM J. Discrete Math., 19(2):363-
370, 2005.
[8] Ruihu Li, Zongben Xu, Xuejun Zhao, On The Classification
of Binary Optimal Self-orthogonal Codes. IEEE Trans. Inf.
Theory, pp:3778-3782, 2008.
[9] Ruihu Li, Research on Quantum Code and Self-orthogonal
Codes. Post-doctor Report. Xi’an Jiaotong University, May 29,
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[10] W.C.Huffman and V. Pless, Fundmentals of Error-Correcting
Codes, Cambridge University Press, Cambridge, UK, 2003.
[11] MathWorks. MATLAB R2006a. 2006.
 2009 International Conference on Computer Engineering and Technology
Performance Analysis of Large Receive Offload in a Xen Virtualized System
Hitoshi Oi and Fumio Nakajima
The University of Aizu, Aizu Wakamatsu, JAPAN
{oi,f.nkjm}@oslab.biz
Abstract
System-level virtualization provides us various advan-
tages including independent and isolated computing en-
vironments on which multiple operating systems can be
executed, and improved resource utilization. These benefits
come with performance overheads and network operation
is one of most typical cases. In this paper, we first present
the experimental results of network performance with LRO
under varying message sizes and maximum transmission
unit (MTU). We analyze the effectiveness of large receive
offload (LRO) in a consolidated execution environment
based on Xen. When SPECjbb, a CPU-intensive workload,
and network receiver are executed on separated domains,
LRO in physical and virtual NICs improve the throughput
up to 8 and 14%, respectively.
Keywords Virtualization, Network, Performance Analysis
I. Introduction
System-level virtualization technologies provide mul-
tiple independent computing environments on a single
physical platform. Each computing environment (virtual
machine, or domain in Xen) can run a (possibly) dif-
ferent operating system (guest OS) and resource isola-
tion between virtual machines (VMs) are guaranteed [1].
The advantages of virtualization come with performance
overheads. With hardware supports for CPUs, switching
between privileged and unprivileged execution modes is
relatively fast [2], [3]. However, accesses to I/O devices
are still costly operations and networking is one of most
typical cases. In Xen, paravirtualization (PV) model is used
for virtualizing I/O devices. With the PV device model,
multiplexing of a single I/O device among guest domains
is possible. In addition, the interface of the I/O device
for each guest domain can be abstracted. However, the
performance degradation of network interface in the PV
model is significant and various optimization attempts have
been made [4], [5].
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 475
DOI 10.1109/ICCET.2009.112
Large receive offload (LRO) is a technique to reduce
the overhead of handling received message packet [6],
[7]. Previously, we ported the LRO into a Xen virtualized
system and reported the results of preliminary performance
measurements [8]. In this paper, we will analyze the
performance of LRO in a Xen virtualized system in more
detail and investigate the interference with the workload
running on another domain on the same platform.
This paper is organized as follows. In the next section,
we explain the technical background of this work. In
Section III, experimental results and their analysis are pre-
sented. We show how network throughput is changed under
varying application message size, maximum transmission
unit and LRO scheme. We also show the effect of CPU
utilization by another guest domain. Previous work related
to this paper is presented in Section IV and we conclude
the paper in Section V.
II. Background of This Work
In this section, we present the background of this
work. First, we provide a brief description of the Xen
internal network architecture. Next, Large Receive Offload
(LRO), an implementation technique to reduce the network
operation overhead, is presented. The objectives of this
paper derived from our previous work are explained in
Section II-C.
A. Xen Internal Network Architecture
Figure 1 shows the architecture of the Xen internal
network. Each guest domain has zero or more virtualized
network interface called netfront. In the privileged domain
handling physical network interface (driver domain), there
is a counterpart for each netfront (netback). Netfront
and corresponding netback exchange packets by sharing
page frames pointed to by the descriptor ring. For event
notification, virtual interrupt mechanism (event channel)
through Xen hypervisor is used. This paravirtualized net-
work architecture of Xen has advantages, such as device
 Config. TP CLK INST LRO
isolation or transparency to the guest domain, but it incurs (Mb/s)
high network performance overhead. In [5], it is reported Linux 557.0 6.27 6.30 N/A
that the per packet CPU overhead (in clock cycles) in Xen Xen 505.4 9.71 7.16 N/A
pLRO 508.4 8.01 5.36 1.99
network is more than four times larger than that of native vLRO 615.3 9.87 7.49 4.92
Linux.
TABLE I. Effectiveness of LRO [8]. For the
Driver Domain Guest Domain abbreviations in Config. column, see Table III.
Bridge Application TP stands for throughput in Mbps. CLK and
Descriptor INST stand for the numbers of clock cycles
Ring and instructions in 1010 for a 10GB message
Physical Backend Frontend transfer. LRO is the number of packets com-
Interface Interface Interface bined by LRO.

Event Channel
NIC Xen VMM III. Experiments and Results
Fig. 1. Xen Internal Network Architecture In this section, we present the results and analysis of our
experiments. First, the hardware and software environment
used for the experiments are described. Next, we present
B. Large Receive Offload the measurement results of throughput for varying MTU
and LRO scheme. Finally, the performance interference
Large receive offload (LRO) was initially implemented between the packet processing overhead and the workload
by hardware on the Neterion 10GbE Ethernet card [6]. Jan- in other domain is investigated.
Bernd Themann implemented it in software as a patch for
the Linux kernel [7] and it has been adopted to the kernel A. Experimental Environment
tree from version 2.6.24. LRO combines the received TPC
packets and passes them as a single larger packet to the
upper layer in the network. By reducing the number of Component Description
packet processing operations, the CPU overhead is lowered CPU Xeon 3GHz
Memory 2GB
and the performance improvement is expected. NIC 1Gbps
Operating System Linux 2.6.18
C. Objectives of This Paper VMM Xen 3.1.1
Network Measurement Netperf 2.4.4
Java workload SPECjbb2001 v1.04
In our previous work, we ported LRO into the physical Guest Domains
and virtual NICs of a Xen virtualized system and measured vCPU 1/domain
its effectiveness [8]. As shown in Table I, LRO in the Memory 512MB/domain
physical NIC reduced the numbers of clock cycles and
TABLE II. Benchmarking Environments
instructions for the same amount of message transfer while
LRO in the virtual NIC increased the throughput.
Following these results, our objectives in this paper Table II shows the details of the machine used as
are as follows: (1) some NICs support larger maximum the message receiver. The hardware platform used has a
transmission units (MTUs) than standard 1500B. Larger Xeon processor, 2GB of memory and a Gigabiet NIC.
MTUs can reduce CPU utilization and lead to more This machine is connected to another machine used as
efficient message transfer. We show how a larger MTU a message sender by a cross-cable. The base operating
affects the effectiveness of LRO in Xen virtualized system. system is Linux kernel 2.6.18 and Xen 3.1.1 is used for
(2) In the previous results shown in Table I, the LRO in virtualization. This receiver machine is the same as the
the physical NIC reduced the CPU overhead of message one used in our previous work [8]. The machine used as
receive operation. We show that how this reduction of CPU the message sender is also a Linux 2.6.18 based machine,
overhead affects the performance of the workload in other but the main difference is the NICs. Previously, we used
domain sharing the hardware platform. (3) Similarly, we on-board NICs for both the sender and the receiver. In
investigate how the network performance is altered when this paper, we use NICs that support larger MTUs which
the workload in another domain is varied and whether LRO are connected through the PCI-Express bus. Also, we
can reduce the performance interference or not. previously used a network switch but in this paper a

476
cross-cable connects two machines. For measurements of
message throughput, we use the TCP STREAM TEST of
netperf [9]. To control the CPU utilization of the receiver
machine, we use SPECjbb2005 [10].
Abbrv. Configuration
Linux Native (Non-virtualized) Linux
Xen Xen-virtualized (no LRO)
pLRO LRO in Physical NIC
vLRO LRO in Virtual NIC
TABLE III. Configurations and Their Abbrevi-
ations
B. Standalone Performance
Standard maximum transmission unit (MTU) on the
Ethernet is 1500B. Some NICs support larger MTUs which
improve the transmission efficiency. However, the larger
the MTU, the fewer the packets to be sent which lead to the
fewer chances of LRO. Also, the application level message
sizes (e. g. the size of the message chunk with which
send() is invoked) affect the CPU overhead for packet
processing. In this section, we show the throughput and the
LRO rate (the number of receive packets divided by the
number of packets transferred to the upper network layer)
for the varying MTU, message size and LRO schemes
(Table III). In the virtualized configurations (Xen, pLRO
and vLRO), only the guest domain for the message receiver
is activated as the guest domain.
the throughput is almost constant for all configurations.
Throughput for all configurations are higher than those in
Table I. The primary factor of these improved throughput
should be the cross-cable connecting sender and receiver
machines (avoid the network switch used in the previ-
ous work). The amount of improvement varies from 3%
(vLRO) to 29% (Linux). The vLRO is least affected by
(possibly) the use of the cross-cable. This is understandable
because the LRO operations take place at the virtual NIC
which is distant from the physical NIC.
We also note other differences between the results in
this paper and those in the previous one. Previously, the
improvement of throughput by pLRO was less than 1%
but this time it is up to 8%. We may attribute this result
to the LRO rate of pLRO in Figure 5. Previously the
LRO rate in pLRO was 1.99 but now it is up to 2.70.
Again, these higher LRO rates should be due to the cross-
cable which transmits the packets at a higher rate than the
network switch. The difference between pLRO and vLRO
is shrunk compared to the previous one. vLRO achieved
22% and 21% higher throughputs than Xen and pLRO,
respectively, in the environment for the previous paper.
This time, throughput of vLRO is only 12% and 5% better
than Xen and pLRO, respectively.
850
800
750
Throughput (Mbps)

700
650
750
600 Linux
700 Xen
550 pLRO
650 vLRO
Throughput (Mbps)

500
600 450
550 400
64 128 256 512 1K 2K 4K 8K 16K 32K
500 Linux Message Size (Byte)
Xen
450 pLRO
vLRO Fig. 3. Throughput (MTU = 4000B)
400

350 The throughput for MTU = 4000B are shown in Fig-

64 128 256 512 1K 2K 4K 8K 16K 32K ure 3. We see great improvements of up to 15% in through-
Message Size (Byte) put over the MTU = 1500B cases in the native Linux.
Fig. 2. Throughput (MTU = 1500B) The larger MTU of 4000B also improves the network
performance of Xen-based configurations: the throughputs
of Xen, pLRO and vLRO are increased by up to 27%,
Figure 2 shows the throughput of four configurations 15% and 15%, respectively1 . Unlike MTU = 1500B cases,
for M T U = 1500. In the case of M T U = 1500B, LRO does not seem to improve the throughput of Xen-
throughput is quite low for the message sizes of 64 and
128B. However, for the message sizes of 256B or larger, 1 The cases for 128B or smaller messages sizes are excluded

477
based systems for MTU = 4000B. vLRO and Xen perform
almost the same and pLRO is actually worse than Xen
for the message sizes of 8KB or larger. The small benefit
of LRO should be attributed by the small LRO rates as
seen in Fig. 5: for the pLRO and vLRO cases, their LRO
rates are 1.2 or smaller and 1.5 or smaller, respectively. As
mentioned above, another thing we should note is that the
throughput of pLRO becomes lower than Xen and vLRO
for the message sizes of 8KB or larger. We do not have
a clear answer to this question and further investigation
is required. However, the Xen internal network passes
network packets by exchanging page frames whose size
is 4KB and the above phenomena may be related to this
page frame size.
850
800
750
Throughput (Mbps)
700
650
600
550
500
450
400
64 128 256 512 1K 2K 4K
Message Size (Byte)
Fig. 4. Throughput (MTU = 4500B)
The throughput for MTU = 4500B are shown in Fig-
ure 4. This is the largest MTU we could set in our system.
Similar to the MTU = 4000B cases, the native Linux takes
advantage of MTU = 4500B and 15% of improvement in
throughput is achieved. Unfortunately, Xen-based systems
we used cannot take advantage of this yet larger MTU. In
all three cases (Xen, pLRO and vLRO), the throughput are
even lower than MTU = 1500B cases. Please note that, for
the results shown in Figs. 3 and 4, MTUs of all Xen in-
ternal network components (bridge, netback and netfront)
are set to 4000B and 4500B, respectively. We consider this
poor network performance for MTU = 4500B as follows.
As briefly mentioned above, the backend transfers the
packet by exchanges page frames with the frontend of the
target domain. Since MTU is slightly larger than the size of
page frame, we have to use two page frames and only small
fraction of the second page frame is filled with payload.
These results suggest that even when the physical NIC
supports the jumbo frame feature, we have to set actual
MTU by taking the page frame size into consideration.
4.5
4
pLRO (MTU = 1500)
vLRO (MTU = 1500)
3.5 pLRO (MTU = 4000)
vLRO (MTU = 4000)
LRO Rate
3 pLRO (MTU = 4500)
vLRO (MTU = 4500)
2.5
2
1.5
1
64 128 256 512 1K 2K 4K 8K 16K 32K
Message Size (Byte)
Fig. 5. LRO Rates
Linux
Xen
pLRO LRO rates for pLRO and vLRO with MTUs of 1500B
vLRO
to 4500B are shown in Figure 5. As mentioned above,
for MTU = 1500B cases, pLRO rate is higher and vLRO
rate is lower than those reported in our previous work,
respectively. In all MTUs, vLRO achieves higher LRO
rates than pLRO. MTU = 4500B cases achieve higher
LRO rates than MTU = 4000B cases. Especially, vLRO
in MTU = 4500B achieves LRO rates of around 2.5. One
possible explanation is as follows. Again, due to the page
8K 16K 32K frame exchanging mechanism of the Xen internal network,
one packet of 4500B is divided into two part and then
merged at the virtual NIC. The LRO rates of vLRO for
MTU = 4000B (1.5 or smaller) support this idea because
a packet of 4000B can fit in a single page frame. Again,
further instrumentation and analysis are needed to validate
this idea. In all other large MTUs, LRO rates are quite
small and coincide with the little effectiveness of LRO in
these configurations.
C. Performance Interference
In the previous subsection, only the guest domain for
the message receiver was activated. In this subsection,
we also activate another guest domain for running the
SPECjbb benchmark to see the performance interference
between two guest domains. SPECjbb is modeled after
TPC-C benchmark [11] which is an OLTP workload, but
the main difference is that all the database tables are
stored in main memory (rather than HDDs in TPC-C).
To control the CPU usage of the SPECjbb domain, we
inserted short sleep periods in the transaction clients, which
is the same methodology used in benchmark suites for
consolidated systems [12], [13]. For the measurements in
this subsection, we use the message size of 1KB and
478
MTU = 1500B. We set the scaling factor of SPECjbb
W = 4, which is scaled down from the configuration of
the SPECjbb in the VMmark (W = 8 for vCPU = 2) [12].
700
Linux
650 Xen
pLRO
600 vLRO
Throughput (Mbps)
550
500
450
400
350
300
250
0 1000 2000 3000 4000 5000 6000 7000 8000
SPECjbb Score
Fig. 6. Consolidated Performance
The relationship between the SPECjbb score and net-
work throughput in four configurations are shown in
Figure 6. The right-most point for each line represents
the case where SPECjbb is executed without CPU usage
control. Due to the discrete even nature of SPECjbb and
unpredictability in system softwares (both Xen and Linux),
there are several “noises” on each graph. However, we
can find the followings from Figure 6: First, as expected,
native Linux achieves higher throughput when SPECjbb
consumes low CPU time, and vice verse. However, we
can also say that in the multiprogramming environment of
the native Linux, one CPU-intensive workload can use too
much CPU time which may result in performance degra-
dation of other workloads (in this case the lower network
throughput). In Xen-based cases, resource isolation works
as expected and such extreme case seen in the native Linux
case is avoided.
Second, the throughput of all cases degrease linearly
against the increasing SPECjbb scores. However, the rates
of throughput decreases change at around SPECjbb score
3000 for the Xen-based cases and 6000 for the native
Linux. These SPECjbb scores correspond to about 30%
(Xen-based cases) and 60% (Linux) of the physical CPU
time utilization. In the cases of Xen-based configurations,
two guest domains are running, and it looks to imply that
the network throughput decreases linearly until another
domain uses around 60% of its CPU time share. Further
investigation is necessary to confirm this hypothesis. Fi-
nally, the throughput improvement rate the pLRO ranges
from 4% to 8% while that by vLRO rangres from 7% to
14%.
479
IV. Related Work
Meron et. al. evaluated three optimization techniques for
the network performance in Xen [4]: the virtual network
interface, the data path between guest and driver domains,
and the virtual memory. They redefined the virtual network
interface so that it could utilize the hardwares support
from modern NICs such as TCP segmentation offloading
(TSO). In [1], the design of Xen is described in detail.
In this paper, they compared the network performance of
Xen with native Linux, VMware and User-mode Linux.
In addition to the standard MTU of 1500B, they used a
smaller MTU of 500B to present the network connection
in dial-up PPP client which is no longer used these days.
Santos et. al. analyzied the Xen receiving network path
by breaking down the CPU time into functional groups,
such as network device driver, grant copy, etc. [5]. They
applied various optimization techniques to these categories
and reduced the overhead. Some of the techniques they
used (such as reduction of packet copy operation) should
be applicable to our cases and we plan to do so. When mul-
tiple domains on a single platform are configured to form a
multi-tier system, inter-domain communication traffic can
be large and it could be the performance bottle neck of the
system. Jian Wang et. al. proposed XenLoop which is a
inter-domain communication channel [15]. It bypasses the
regular virtual network interfaces in Figure 1, yet provides
transparency in the sense that the application programs
can still use standard TCP/IP APIs. Vconsolidate is Intel’s
proprietary benchmark suite for evaluating consolidated
server. Apparao, et. al., evaluated how performance metrics
of each workload is changed under consolidation with
varying cache sizes and CPU affinity [13].
V. Conclusion
In this paper, we presented and analyzed the network
receiving performance of a Xen virtualized system with
LRO. When only the guest domain for the network receiver
was activated, LRO in the physical and virtual NICs
improved the throughput by 5 and 12%, respectively.
When another guest domain was activated and SPECjbb, a
CPU-intensive workload, is executed, the network through-
put was dropped linearly against the CPU utilization of
SPECjbb. However, throughout the ranges of the workload
mix used in this paper, LRO in physical and virtual NICs
improved the throughput by 4% to 8% and 7% to 14%,
respectively. Open questions and future work include the
following. First, compared to the standard MTU of 1500B,
we saw that MTU of 4000B improved the throughput of
both native Linux and Xen-based systems by up to 27%.
In the current system, we have to use the same MTU for
all the network components within Xen. When we use a
standard MTU (1500B) for the physical NIC, we apply
LRO there and use a larger MTU of the page frame size
(4KB). In this configuration, we may reduce the packet
handling overhead at the physical NIC and maximize the
throughput of the Xen internal network. Second, we saw
the changes in the relationship between the throughput
and SPECjbb score around CPU utilization of 60% and
reasons of this change should be identified. Lastly, various
optimization techniques mentioned in Section IV should
also be incorporated into our experimental system.
References
[1] Paul Barham et. al., “Xen and the Art of Virtualization,” in Pro-
ceedings of the ACM Symposium on Operating Systems Principles
(SOSP’03), pp164–177, October 2003.
[2] “AMD Virtualization,”
http://www.amd.com/virtualization .
[3] “Intel Virtualization Technology in Computing,”
http://www.intel.com/technology/
virtualization/ .
[4] Aravind Menon, Alan Cox and Willy Zwaenepoel, “Optimizing
Network Virtualization in Xen ,” in Proceedings of the 2006
USENIX Annual Technical Conference, pp15–28, May–June 2006.
[5] Jose Renato Santos, et. al., “Bridging the Gap between Software
and Hardware Techniques for I/O Virtualization,” in Proceedings
of the Usenix ’08, pp29–42, June 2008.
480
[6] Leonid Grossman, “Large Receive Offload implementation in Ne-
terion 10GbE Ethernet driver,” in Proceedings of the Ottawa Linux
Symposium, pp195–200, July 2005.
[7] Jan-Bernd Themann, “[RFC 0/1] lro: Generic Large Receive
Offload for TCP traffic,” Linux Kernel Mailing List archive,
http://lkml.org/lkml/2007/7/20/250.
[8] Takayuki Hatori and Hitoshi Oi, “Implementation and Analysis of
Large Receive Offload in a Virtualized System,” in Proceedings
of the Virtualization Performance: Analysis, Characterization, and
Tools (VPACT’08), April 2008.
[9] “Netperf: a network performance benchmark,” Hewlett-Packard
Company, Feb. 15, 1995.
[10] SPECjbb2005,
http://www.spec.org/jbb2005/ .
[11] TPC-C, http://www.tpc.org/tpcc/ .
[12] VMmark, http://www.vmware.com/products/
vmmark/
[13] Padma Apparao, et. al., “Characterization & analysis of a server
consolidation benchmark,” in Proceedings of the fourth ACM
SIGPLAN/SIGOPS international conference on Virtual execution
environments, pp21–30, March 2008.
[14] Aravind Menon, et. al., “Diagnosing performance overheads in
the Xen virtual machine environment,” Proceedings of the 1st
ACM/USENIX international conference on Virtual execution envi-
ronments (VEE05), pp13–23, June 2005.
[15] Jian Wang, Kwame-Lante Wright and Kartik Gopalan, “XenLoop:
A Transparent High Performance Inter-VM Network Loopback,”
in Proceedings of the 17th International Symposium on High
Performance Distributed Computing (HPDC’08), pp109–118, June
2008.
 2009 International Conference on Computer Engineering and Technology
An Improved Genetic Algorithm Based on Fixed Point Theory for Function
Optimization
Jingjun Zhang
Scientific Research Office
Hebei University of Engineering
Handan, Hebei, P.R.China
santt88@163.com
Abstract—This paper introduces triangulation theory into
genetic algorithm and with which, the optimization problem
will be translated into a fixed point problem. An improved
genetic algorithm is proposed by virtue of the concept of
relative coordinates genetic coding, designs corresponding
crossover and mutation operator. Through genetic algorithms
to overcome the triangulation of the shortcomings of human
grade, it can start from any point to find the most advantages.
Gradually fine mesh will be introduced the idea of genetic
algorithms so that the search area gradually decreased,
improving the efficiency of search. Finally, examples
demonstrate the effectiveness of this method. (Abstract)
Keywords- genetic algorithm; triangulation; relative
coordinates; gradually fine mesh; Fixed Point (key words)
I. INTRODUCTION
Genetic algorithm [1] (GA) is a computational model of
simulation of natural biological evolution. GA is based on
the criteria of survival of the fittest and carries on genetic
operations repeatedly to the population containing the
possible solution to generate new evolving population, and
simultaneously it searches the best individual of population
with the overall parallel searching methods to achieve the
optimal solution to meet the requirements. Comparing to
other search methods, GA has some merits, such as simple
and robust. And it can also search the optimal solution or
quasi-optimal solution with the overall parallel methods by
adjusting the direction of the search space continuously.
However, the algorithm also has some shortcomings, such as
slow convergence rate, prematurely and so many iterations.
To overcome these shortcomings, some improvement on
genetic algorithm has been made constantly, for example the
adjustment of population size[2], the optimization of control
parameters[3], an outstanding individual protection strategy
[4] and so on.
Genetic algorithm can converge to global optimal
solution theoretically, but in the actual design, there is no
mature principle and methods of designing the convergence
criteria. Instead, the designing of convergence criteria
basically relies on the experience and understanding of the
extent to the question, and the most common method is
taking the max generation as the convergence criteria or the
iteration will be terminated if average fitness of the continual
m- generation has not been improved obviously. Therefore,
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.116
Yuzhen Dong Ruizhen Gao Yanmin Shang
College of mathematics& physics
Hebei University of Engineering
Handan, Hebei, P.R.China
ddongyuzhen@163.com
it has vital significance to design objective criterion of
termination judgment.
This paper proposed an improved genetic algorithm
based on the fixed point. First of all, carrying on the
triangulation to the solution space and calculating the integer
label of the vertex, and then designing the encoding scheme,
crossover operator and mutation operator with the
conception of the relative coordinate. This algorithm
overcomes the shortcoming of man-made label of the
triangulation for fixed point and converges when all the
individuals are completely labeled simplexes.
II. ABOUT TRIANGULATION
So-called triangulation [5] is that making sure h > 0 and
with three groups of straight lines, such as {x1 = mh} ，
{x2 = mh} and {x2 = x1 + mh} (among them m is an
2
integer), the 2 Dimensional Euclidean space R will be
divided into many triangles and obtains the triangulation
2
of R .
2
Carrying on triangulation to the Euclidean Space R , the
g
goal is to find such a triangle under the self-mapping . The
first coordinate component of one vertex of the triangle
dropped, another vertex's second coordinate component
decreased, and the third vertex's two coordinate components
remained unabated. As a result of continuity of g , if this
triangle's diameter is small enough, the change of three
vertices will not differ too far. In this case, each vertex is an
approximate fixed point.
Supposing that x is a vertex of a triangle, its integer label
is determined according to l (x ) .
⎧ 0 g1 ( x) − x1 ≥ 0, g 2 ( x) − x2 ≥ 0
⎪
l ( x) = ⎨ 1 g1 ( x) − x1 < 0, g 2 ( x) − x2 ≥ 0
⎪ 2 g 2 ( x) − x2 < 0
⎩
The triangle with three different kinds of integer labels is
called a completely labeled simplex. Usually, we use the
man-made labels not relying on self-mapping g and find the
completely labeled simplexes though substitution of vertex
with the same label (as shown in Figure 1).
481
 obtained, they are
B ( a + ( n + 1) h, c + kh) ,C ( a + ( n + 1) h, c + ( k + 1)h) a
nd D ( a + nh, c + ( k + 1) h) . Therefore, we can calculate
the coordinates of other vertices of the square only if the
relative coordinate of A is given. In view of this fact, we take
the relative coordinate of an intersection point which is
decided by two groups of straight lines as genetic encoding.

Figure 1. The triangulation

4
In this paper, the thought of triangulation is introduced to
genetic algorithm and as a result, the improved genetic 3
algorithm avoids the shortcoming of man-made initial point.
Though evolution, narrowing the searching ranges gradually
to enhance the convergence rate and reduce the iteration 2
times
III. THE IMPROVED GENETIC ALGORITHM 1
We frequently meet the questions of solving the optimum
value of max f ( x ) ( a ≤ x ≤ b) without constraint 0 1 2 3 4
condition, and supposing that f (x ) has continuous Figure 2. .relative coordinates

derivative in the given interval. In this case, the question of Crossover operator: Using the overall arithmetic
solving the optimum value of f (x ) can be transferred to
crossover strategy, supposing that p1 = ( x1 , x2 ) and
(1) (1)

solve the zero point of f (x) and then it can be transferred to

p2 = ( x1( 2) , x2( 2) ) are two parent vectors with satisfaction
solve the fixed point of f (x) according
of x1 ≠ x1 and x2 ≠ x2 . s1 = ( z1 , z2 ) ，
(1) ( 2) (1) ( 2)
to g ( x ) = x + f ′( x ) . Obviously in the numerical
calculation, it is difficult to find a vector x satisfying the s2 = ( w1 , w2 ) are the two descendants obtained though
formula of g ( x ) = x .Under the circumstance, the crossover operator.
conception of ε fixed point [5] has been introduced. z1 = [α x1(1) + (1 − α ) x2(1) ⎤⎦
，
that ε > 0 , we say x ∈ R is a ε
n
z 2 = [ β x1( 2 ) + (1 − β ) x 2( 2 ) ⎤⎦
Supposing -fixed point
(approximate fixed point) of the self- ,
mapping g : R n → R n , if x − g (x) < ε . Here w1 = [ (1 − α ) x + α x ⎤⎦
(1)
1
(1)
2
，
x − g (x) expresses the vector mold in Euclidean space.
w2 = [ (1 − β ) x (2)
1 + β x ⎤⎦
(2)
2
Relative coordinates: Supposing that the definition .
domain of f ( x1, x 2) is that a ≤ x1 ≤ b, c ≤ x 2 ≤ d and In the formulas above ， α and β are random real
dividing the domain into many squares with two groups of number with satisfaction of 0 < α < 1 and 0 < β < 1 , [ ]
straight lines of {x1 = mh} and {x2 = mh} ( m is an is the integer operator.
integer).For a random point of intersection named A, we Mutation operator: In order to manifest the influence of
(1) (1) the integer label in genetic manipulation, the mutation
suppose that it’s coordinate is ( x1 , x2 ) , then there are
operation is as follows:
surely existing n, k satisfying the formula Taking the relative coordinate of A as the genetic encode,
then we can determine a square with A as a left bottom
of x
(1)
1 = a + nh , x = c + kh , and then (n, k ) is called
( 2)
1 vertex. Each vertex can search two aspects information; one
the relative coordinate of A. aspect expresses the parallel direction of the square and
Genetic encoding: selecting four adjacent points of a another describes the length of stride. Take vertex A as the
square randomly, according to the subdivision above, if the example, it has two directions, downward and towards left
relative coordinate of the lower left quarter vertex A of the (as shown in Figure 3), the length of stride for in this
square is given (as shown in Figure 2), recording that ( n, k ) , direction is the grid number with different label of A though
then the other three vertices` relative coordinates record the parallel migration. If the vertex happens to be in
B (n + 1, k ) , C (n + 1, k + 1) and D (n, k + 1) , boundary, then the length of stride to boundary outside is ∞ ,
or if searching in the boundary along a direction not to find a
simultaneously the coordinates of B, C and D can be
vertex with different integer label, we also suppose length of

482
stride along this direction is ∞ . If each direction length of
stride is ∞ , approaches to the origin. Chose the minimum
length of stride from 8 directions of four vertexes and obtain Start
a square though parallel moving according to this direction.
The coordinate of the left bottom vertex of the new square,
recording (n′, k ′) , is taking as the chromosome encoding According to precision request, design
after mutation of vertex ( n, k ) . First of all, this mutation h = {h1 , h2 ,", hn } ；
need to search the in formations of the vertices, then it
carries on specific operation according to the results of the
mutation. We call this mutation as neurons mutation that is a
guiding mutation. Carries on square cutting in half region Di , with two

groups of lines { x1 = mhi } ， {x 2 = mhi }

A D

Coding, initialization, p(0) , t<-0

B C
Calculates each individual to correspond the

Square four apexes the integer markings

Figure 3. Relative coordinates and vertex information

For example, as shown in Figure3, giving the square

corresponding with the mutation individual according to the
counter clockwise, the search information is given according
to the below order: first the horizontal direction, then vertical Whether has the entire
direction, under the circumstance, obtaining 8 integers,
supposing that are: 12, 23, 8, 15, 7, 12, 5, 6, choosing the sign square?
minimum one that should be 5. According to above
hypothesis order, 5 correspond with left direction. Suppose N
A’s encoding is ( n, k ) , the mutation encoding Y

becomes (n − 5, k ) . Carrying on search according to the

N
direction of coordinate connection though crossover Whether meets the
operator, this is carried on along the vertical or horizontal precision Y
direction between the selected individual.
The individual which obtains through crossover
Y
operation or mutation operation must be examined that
whether the four vertices integer labels of the square includes End
0, 1, and 2. If including the three labels, we call the square
the completely integer label square, the small region
determined by the square is the extreme point scope. Selection
In order to make sure the operation effectively, the
algorithm uses the thought of gradually thin cuts in half.
Design a series of h = {h1 , h2 ,", hn } , Crossover

satisfies hi % hi +1 = 0 according to the result precise degree.

In which % is the complement of a number operator, that is Mutation
the vertex which obtained in preceding cutting in half is still
in the latter thinner cutting in half. Firstly in the given scope,
determine a smaller scope which includes the optimal Obtains the new generation of
solution, then takes this scope as the new domain of individual p (t + 1)
definition and searches again. Reduce the search scope
gradually until the algorithm meets the requirements
precision.
Improvement genetic algorithm flow chart following
shown in Figure 4: Figure 4. The improved GA flow chart

483
 IV. EXAMPLE CONFIRMATIONS
The test function:
⎧
⎪ min f ( x1 , x2 ) = ( x1 + x2 )
2 2
⎨ ACKNOWLEDGMENT
⎪
⎩ s.t− 2 ≤ x i ≤ 2. (i =1,2)
The function achieves the minimum in (0,0) .
In this example, h = {1,0.1,0.001,0.0001} crossover
probability pc = 0.1 , mutation probability pm = 1 .The
completely label square obtains though the iteration, the
search scope are: (0 ≤ xi ≤ 1) ， (0 ≤ xi ≤ 0.1) ，
(0 ≤ xi ≤ 0.01) and (0 ≤ xi ≤ 0.001) in turn. From this,
the search scope reduces unceasingly with unceasing thinner
cutting in half. From this example the search scope are
contracting to (0,0) gradually, using the limit thought, the
optimum point is (0,0) .
V. CONCLUSION
This paper proposed an improved genetic algorithm
based on the fixed point. Carrying on the triangulation to the
solution space and calculating the integer label of the vertex
484
and then designing genetic operator, Population is used to
find a full-scale triangle as a more objective criteria for
convergence. Simulation gains the good result as expected.
The authors gratefully acknowledge the supports of
National Natural Sciences Foundation of China under the
project number 60875037 and Natural Sciences Foundation
of Hebei Province under No.F2007000710 and
No.E2008000731.
REFERENCES
[1] Holland, J. Adoptation in Natural and Artificial Systems [M].
University of Michigan Press, 1992,44~50
[2] Arabs J, Michalewicz Z,MulaWka J. GAVaPS--a Genetic
Algorithms
st with Varying Population Size.In:Proceedings of the
1 IEEE International Conference on Evolutionary
Computation(ICEC’94), 1994, 73-78
[3] Eiben A. E., Hinterding R., Michalewicz Z. Parameter Control in
Evolutionary Algorithms. IEEE Transaction on Evolutionary
Computation, 1999, 3(2): 124-141
[4] Farzad G,Gecil A. Application of random restart in genetic
algorithms[J].Information Sciences.1996,85(2):81~102
[5] Wang Zeke, pure fixed point algorithm foundation [M]. National
University of Defense Technology publishing house, 1993
 2009 International Conference on Computer Engineering and Technology

Example-Based Regularization Deployed to Face Hallucination

Hong Zhao1, 2 Yao Lu1 Zhengang Zhai1, 3 Gang Yang2
1
Beijing Laboratory of Intelligent Information Technology, School of Computer Science, Beijing
Institute of Technology, Beijing 100081 PRC, China
2
College of Mathematics and Computer Science, Hebei University, Baoding 071002, China
3
Chinese People’s Liberation Army Air Defence Forces Command College, Zhengzhou, China
E-MAIL: valen_zhao@bit.edu.cn, vis_yl@bit.edu.cn, zz_gang@bit.edu.cn, yanggang@hbu.cn

Abstract probability density function of the image (PDF) and

Regularization plays a vital role in ill-posed
problems. A properly chosen regularization can direct
the solution toward a better quality outcome. An
emerging powerful regularization is one that leans on
image examples. In this paper, we propose a novel
scheme for face hallucination. We target specially the
quality of highly zoomed outputs. Our work bases on
the pyramid framework and assigns several high-
quality candidate patches for each location in the
degraded image. We definite a global MAP penalty
function to reject all the problematic examples, and
then reconstruct the desired image using the patch
which is left after pruning. Experimental results
demonstrate that our approach can get better
resolution.
1. Introduction
Super-resolution is the process of combining
multiple low-resolution images to form a higher
resolution one. Numerous super-resolution algorithms
have been proposed in the past decades [1], [2], [3], [4],
[5], [6], [7]. Generally speaking, there are three
categories of approaches for super-resolution:
interpolation-based methods, reconstruction-based
methods and learning-based methods.
Face hallucination is an application of the super-
resolution techniques to human facial images whose
procedure is generally based on a single measured
image. Thus, the priori knowledge of the face in the
image is needed in order to compensate for the lack of
data. This leads to the concept of regularization [8], [9].
Regularization plays a vital role in inverse
problems, especially in ill-posed ones such as super-
resolution. In recent years, an emerging powerful
regularization that is drawing research attention is one
that leans on examples. Rather than guessing the
forcing a simple expression to describe the image, the
image examples can guide us in the construction of the
priors. The ways to exploit examples can be roughly
divided into two categories [10]: use them to learn the
parameters of previously defined regularization
expressions or use them directly for the reconstruction
procedure.
In this paper, we propose a new scheme for face
hallucination. We target specially the quality of highly
zoomed outputs, because the quality of the results in
Fig. 1 is not satisfying. Our algorithm bases on the
pyramid framework presented in Baker [1] and starts
by assigning several high-quality candidate patches
designed for each location in the degraded image.
Motivated by Elad [10], we do not use these candidate
examples to compute the reconstructed image directly.
By contraries, we definite a global MAP (maximum a
posteriori probability) penalty function for rejecting the
irrelevant examples. Only when one candidate is left,
does the reconstruction begin.
This paper is organized as follows: in section 2 we
describe the relevant theory and core algorithm; in
section 3 we present the experiment results; in section
4 we conclude this paper.
(a) (b) (c) (d)
Figure 1. Two results of Baker’s method.
(a)(c)Eightfold hallucinated (b)(d) Original
2. Theory and Algorithm

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DOI 10.1109/ICCET.2009.52
2.1 The Maximum a Posteriori Probability
Estimator
We model the formation of the measurements l
from the ideal image h by
l = Dh + η . (1)
This is the problem we target in this paper, where
D is the degradation operator and it may include blur,
decimation, geometrical warp and more. The symbol
η stands for an additive noise. For simplicity we will
assume that the noise is Gaussian, white and zero-
mean.
The method of maximum a posteriori probability
(MAP) practices the Bayesian approach and estimates
denote its feature:
FP i ( l ) = ( L i ( l ), H i ( l ), V i ( l ), H i2 ( l ), V i 2 ( l )) (6)
where i is from 0 to N. Meanwhile, the Parent Structure
vector can be written as:
Par i ( l ) = (FP i ( l ), FP i + 1 ( l ), " , FP N ( l ) ) . (7)
For each pixel ( x , y ) in the input image li
independently, we compare its Parent Structure vector
with all training Parent Structure vectors at the same
level i. By setting up a threshold T, we can select
several closest matching pixels in different training
images at the same level. In other words, all pixels that
satisfied
2
Par i ( l ( x , y )) − Par i ( h k ( x , y )) ≤ T (8)
2

h as follow:
are considered as the candidates of pixel l i ( x , y ) .
*
h MAP = arg max P (l | h ) P (h )
h
Consequently, we find several patches which are in the
= arg min ( − ln P ( l | h ) − ln P ( h )) . resolution we desired. These patches form the nearest-
h

(2) neighbor candidates set which we denote as Ω ( x , y ) .

Considering the model in Equation (1), based on
the Gaussianity of the noise and the fact that x is 2.3 A Global MAP Penalty
assumed to be known, thus, the likelihood function can
be written as follow: Once all the candidates have been found in every
1 ⎧ 1 2⎫ location, there are two classical way to reconstruct the
P (l | h ) = ⋅ exp ⎨ − ⋅ l − Dh 2 ⎬ (3)
M
( 2π ) σ2 M ⎩ 2 σ 2
⎭ image including MMSE estimating and MAP
where M is dimension of vector h. estimating. These pixel-based reconstructions are easy
Losing no generality, a convenient to describe to implement. However, their simplicity comes with a
the PDF of h is the Gibbs distribution, which price — the examples found contain many outliers
represents p(h) in an exponential form: with irrelevant content, and those may divert the
P ( h ) = Const ⋅ exp {− α A ( h ) } . (4) desired result. The work in Freeman [3], [4] considered
a pruning of the found examples by modeling the inter-
The constant in front of the exponential is a relations between adjacent/overlapping patches by a
normalization factor, guaranteeing that the integral random Markov network, and using of the Bayesian-
over all h is 1. The term A(h) is a non-negative energy Belief-Propagation (BBP) for choosing the proper
function, supposed to be low for highly probable examples in the recovery.
signals and high otherwise. In order to address the problem of outliers, we use
Using Equation (3) and Equation (4), we obtain: a global MAP penalty function to choose the
∧
h MAP = arg max P (l | h ) P ( h ) problematic examples and prune them out step by step.
h (5) The MAP penalty function is written as follow:

h
{ 2
= arg min l − Dh 2 + 2σ α ⋅ A( h) .
2
} ∧ ∧
ε (h ) = D h − l + ∑
2

∑
∧
wk( x , y ) R( x , y ) (h) − hk( x , y )
2

.
2 ( x , y ) k∈Ω ( x , y ) 2

2.2 Finding the Nearest-Neighbor Candidates (9)

Our method resembles the Baker’s and constructs
the training set which contains a lot of feature
pyramids. For each face image, we firstly create the
Gaussian pyramid; secondly, we get several pyramids,
including the horizontal first and second derivatives of
the Gaussian pyramid, the vertical first and second
derivatives of the Gaussian pyramid and the Laplacian
pyramid. Finally, we form a pyramid of features. For a
th
low resolution face image l at i level, we use FP (l) to
i
In this function, the first term stands for the log-
likelihood, with the assumption that the noise is white
and Gaussian. The second is the regularization term,
and it is defined by all examples found in one
location ( x , y ) . The weights wk( x , y ) are binary: ‘0’ for
a pruned example and ‘1’ for an active one. The
∧
operator R(x ,y) extracts a patch from the image h . The
inner summation is done over all found nearest–
neighbors, their indices taken from the set Ω ( x , y ) .
The outer summation runs through all pixels in the

486
training image. Expression (9) suggests that the
reconstructed image should agree with every found
example and in every location.
2.4 Pruning Problematic Examples
In the training stage we found several candidate
patches for each pixel location. However, some
irrelevant examples have been selected and put into the
candidate set. In order to improve the quality of the
super resolution, we must decide which one is the most
problematic example and throw it out from the
candidate set.
We start with the assignment wk( x , y ) = 1 for all
( x , y ) and k. Based on the following steps, we prune
one patch one time:
Step 1. For the current choice of weights and
current location ( x , y ) , the MAP value of ε (h) is
( x, y )
computed with the omission of patch hk
( x, y )
Step 2. Per each patch hk with wk( x , y ) = 1, we
compute the optimal output image minimizing the
modified MAP function
∧ ∧
ε (h) − ∑ ∑
( x , y ) k ∈Ω ( x , y )
w k( x , y ) R ( x , y ) ( h ) − hk( x , y )
Step 3. Among all examined patches, we prune the
one that gives the largest difference between the
reference penalty value by assigning wk( x , y ) = 0. The
patch discarded is considered to be the least compatible
with the remaining patches.
3. Experiment and Results
Our experiments are conducted with FERET [11]
database consisting of 596 images of 278 individuals
(92 women and 186 men). As a necessary preamble
step, we perform normalization manually. By doing so,
every image is cropped to 96×128 as the high-
resolution image. We compute low-resolution input
image by smoothing and down-sampling the
corresponding high-resolution image. We use a “leave-
one-out” methodology to test our algorithm.
Our target is to improve the quality of the highly
zoomed outputs. So, we only consider the case of
enhancing 12×16 pixel images eightfold to give
96×128 pixel images. The size of the searching
window is chosen to be 3 and the value of T is
assigned 35. We assume that the same part of the face
appears in roughly the same part of the image to save
the searching time. The initial values of wk( x , y ) for all
( x , y ) are 1. As might be expected, our algorithm
performs well, especially in the part of eyes and
mouth.
In Figure 2 we give the comparison of two choice
of the threshold T. Figure 3 presents the experimental
results of the fourfold zooming-up. In Figure 4 we give
the experimental results of the eightfold zooming-up.
In order to evaluate the hallucination effect of two
methods objectively, the paper used Peak Signal Noise
Ratio (PSNR) to measure their efficiency. For the
image of M×N, its PSNR defined as:
⎡ ⎤
⎢ 255 2 ⎥ (11)
PSNR = 10 log ⎢ M −1 n −1
⎥
⎢ 1 2 ⎥
⎢⎣ M × N ∑ ∑ mn ( X − X )
⎥⎦
mn
m = 0 N =0
where X mn , X mn is on behalf of the original image
∧ and the image to be assessed. Table 1 and Table 2
show the comparison of the PSNR of two methods and
demonstrate that our results are superior to those of
.
Baker’s method.
4. Conclusion
2 In order to reconstruct high-quality face images, we
. (10) must try to reject the irrelevant candidate patches. In
2 this paper, we use examples to obtain an effective
regularization for the problem of face hallucination.
The global MAP penalty function can exploit the
spatial coherence in some degree, considering the
unknown image as a whole. Thus, it can be used to
decide which one is the most unlikely candidate patch.
The effectiveness of our approach is demonstrated
by extensive experimental results. The results show
that the output images are certainly in the higher
resolution than that of Baker’s, especially in the part of
eyes and mouth.
Acknowledgements
This paper is supported by the Beijing laboratory of
intelligent information technology. The authors would
like to thank Ziye Yan and Yaozu An for the insightful
discussions.
487
 (a) (b) (c) (d) (a) (b) (c) (d)

Figure 2. Comparison between two different Figure 3. Comparison between our method
values of the threshold T. (a) Input 12×16 (b) and Baker’s method. (a)) Input 24×32(b) Our
T=80 (c) T=35 (d) Original 96×128 method (c) Baker’s method (d) Original
96×128.

(a) (b) (c) (d) (e) (f) (g) (h)

Figure 4. Comparison between our method and Baker’s method. (a)(e) Input 12×16 (b)(f) Our method
(c)( g) Baker’s method (d)(h) Original 96×128.

488
 Table 1. Comparison of PSNR for Fourfold Experimental Results in Figure 3.

Our method Baker’s method

Row 1 25.1726 24.5178
Row 2 25.4913 24.7310
Row 3 24.6323 24.5289
Table 2. Comparison of PSNR for Eightfold Experimental Results in Figure 4.

Our method Baker’s method

Left row 1 21.4741 21.2273
Left row 2 21.2619 20.6807
Left row 3 22.4739 22.0687
Left row 4 20.7809 20.6454
Right row 1 23.2435 22.6496
Right row 2 21.9418 21.6736
Right row 3 21.2839 20.8392
Right row 4 21.3963 21.2901

References algorithms”, Proceeding of the 1997 Conference on

Computer Vision and Pattern Recognition, 1997,
pp.137-143.
[1] S. Baker, T. Kanade, “Limits on Super-Resolution and
How to Break them”, IEEE Trans. on PAMI, Vol. 24,
No. 9, 2002, pp. 1167-1183.
[2] S. Baker, and T. Kanade, “Hallucinating Faces,”
Proceeding of Inter. Conf. on Automatic Face and
Gesture Recognition, 2000, pp.83-88.
[3] W. T. Freeman, and E. C. Pasztor, “Learning Low-
Level Vision”, Proceeding of ICCV, Vol. 2, 1999, pp.
1182-1189.
[4] W. T. Freeman, T. RJones, and E. C. Pasztor,
“Example-based Super-resolution”, IEEE Computer
Graphics and Applications, Vol. 22, No. 2, 2002, pp.
56 -65.
[5] C. Liu, H. Shum, and C. Zhang, “A Two-Step
Approach to Hallucinating Faces: Global Parametric
Model and Local Nonparametric Model”, Proceeding
of CVPR, Vol. 1, 2001, pp. 192-198.
[6] M. A. Turk , and A. P. Pentland, “Face recognition
using eigenfaces”, Proceeding of IEEE Conf.
Computer Vision and Pattern Recognition, 1991, pp.
586-591.
[7] X. Wang, and X. Tang, “Face hallucination and
recognition”, Proceeding of 4th Int. Conf. Audio- and
Video-Based Personal Authentication, IAPR,
University of Surrey, Guildford, U.K., Jun. 2003, pp.
486- 494.
[8] A. K. Jain, Fundamentals of Digital Image Processing.
Prentice-Hall, Englewood Cliffs, NJ. ,1989.
[9] R. L. Lagendijk, and J. Biemond, Iterative
Identification. 1991.
[10] D. Datsenko, and M. Elad, “Example-based single
document image super-resolution: a global MAP
approach with outlier rejection”, International Journal
of Multidimensional Systems and Signal Processing,
Vol. 18, No. 2, 2007, pp. 103–121.
[11] P. J. Philips, H. Moon, P. Rauss, and S. A. Rizvi., “The
FERET evaluation methodology for face-recognition

489
 2009 International Conference on Computer Engineering and Technology

An Ensemble Approach for Semantic Assessment

of Summary Writings
Yulan He∗ , Siu Cheung Hui† and Tho Thanh Quan‡
∗ Schoolof Engineering, Computing and Mathematics
University of Exeter
North Park Road, Exeter EX4 4QF, UK
y.l.he.01@cantab.net
† School of Computer Engineering

Nanyang Technological University

Nanyang Avenue, Singapore 639798
asschui@ntu.edu.sg
‡ Faculty of Computer Science and Engineering

Hochiminh City University of Technology

Hochiminh City, Vietnam
qttho@cse.hcmut.edu.vn

Abstract—Computer-assisted assessment of summary writings
is a challenging area which has recently attracted much interest
from the research community. This is mainly due to the advances
in other areas such as information extraction and natural
language processing which have made automatic summary as-
sessment possible. Different techniques such as Latent Semantic
Analysis, n-gram co-occurrence and BLEU have been proposed
for automatic evaluation of summaries. However, these techniques
are unable to achieve good performance. In this paper, we
propose an ensemble approach, that integrates two of the most
effective summary evaluation techniques, LSA and n-gram co-
occurrence, for improving the accuracy of automatic summary
assessment. The performance of the proposed ensemble approach
has shown that it is able to achieve high accuracy and improve
the performance quite substantially compared with other existing
techniques.
I. I NTRODUCTION
[6], n-gram co-occurrence [7] have been proposed. However,
most of these techniques are unable to achieve satisfactory
performance for assessing summary writings. In this paper,
we propose an ensemble approach, that integrates two of the
most effective summary evaluation techniques, LSA and n-
gram co-occurrence, for improving the accuracy of automatic
summary assessment.
Summary writings are usually assessed based on two cri-
teria, content and style. In this paper, the proposed ensemble
technique focuses mainly on content assessment. The rest of
the paper is organized as follows. Section II reviews some
of the techniques currently employed in summary evaluation.
The proposed approach is presented in Section III. Perfor-
mance analysis is discussed in Section IV. Finally, Section V
concludes the paper.

Summary writing is an important part of many English
examinations. The assessment of summary writings is nor-
mally conducted based on content overlapping and linguistic
qualities by comparing a student’s candidate summary with
a reference summary. However, manual assessment is always
a tedious and labor-intensive process. Furthermore, there is a
problem of inconsistency in human assessment that different
markers might give different grades for the same candidate
summary. To help alleviate such problems, there has been a
growing interest recently in automatic summary assessment or
computer-assisted assessment of summaries.
Computer-assisted assessment is a long-standing problem
that has attracted much interest from the research community
since the sixties and has not been fully resolved yet [1]. With
the recent success of e-learning and the advances in other
areas such as Information Extraction (IE) and Natural Lan-
guage Processing (NLP), automatic assessment of summary
writings has become possible. Some of the techniques such
as Latent Semantic Analysis (LSA) [2], [3], [4], [5], BLEU
II. S UMMARY A SSESSMENT T ECHNIQUES
This section reviews some of the most popular summary
evaluation techniques including those based on Latent Se-
mantic Analysis (LSA) [2], [3], [4], [5] and those based on
machine translation evaluation methods [6], [8], [7].
A. LSA Based Techniques
Landauer et al. [3] first developed Latent Semantic Analysis
(LSA) in the late ’80s with the purpose of indexing documents
and information retrieval. Automated assessment of natural
text was an interesting problem since that time. Landauer
modified LSA to assess natural text. LSA functions by using a
matrix to capture words and frequency of the words appearing
in a context. The matrix is then transformed using Singular
Value Decomposition (SVD). Cosine correlation is used to
determine the similarity. Based on the result of Landauer’s
experiment, LSA is capable of producing results that are
approximately as well as experts’ assigned scores as the scores
correlate with each other. However, LSA does not make use

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DOI 10.1109/ICCET.2009.90
of word order as Landauer claims that word order is not the
most important factor in collecting the sense of a passage [2].
A commercial summary evaluation system, Laburpen Ebalu-
aketa Automatikoa (LEA) [4], also makes use of LSA to derive
summarization scores. LEA is designed to address two types of
users, teachers and students. LEA allows teachers to manage
summarization exercises and inspect students’ answers, and al-
low students to create their own summaries. There is a support
tool that is available to help students write their summaries.
LEA evaluates summaries based on the combination of partial
scores in cohesion, coherence, adequacy, use of language and
comprehension.
Franzke et al. [5] at the University of Colorado at Boulder
developed Summary StreetTM , an automated tool to evaluate
the content of students’ summaries. Summary Street grades
students writing by comparing it with the actual text, eval-
uating it based on content knowledge, writing mechanics,
redundancy and relevancy. Based on the grading given by
Summary Street, feedback is given to help the student know
where his/her mistake is. The core of Summary Street is
the Knowledge Analysis TechnologiesTM (KAT) engine. The
KAT engine uses a modified version of Latent Semantic
Analysis (LSA).
B. Machine Translation Based Techniques
Perez et al. [6] modified the BLEU algorithm, which was
originally developed for ranking machine translation systems,
into one that is capable of marking students’ essay. The
modified BLEU algorithm is capable of assessing a student’s
essay for relevant information by matching it with the model
essay stored in the system. BLEU’s Brevity Penalty factor was
modified to increase the performance of the system, the results
of the modification showed that it was able to outperform the
original BLEU algorithm in terms of correlation. Based on
their evaluation, they had concluded that the modified BLEU
algorithm is capable of achieving reasonable correlation with
the human markers and it is more than sufficient to replace
keyword matching techniques in the assessment of students’
essays.
Lin et al. [8] conducted a study on using the two ma-
chine translation evaluation techniques, BLEU and NIST’s n-
gram co-occurrence scoring procedures, on the evaluation of
summaries. The main idea of the comparison is to measure
the closeness of the candidate to the reference summary by
using the weighted average of variable length n-gram matches
from that of BLEU. Based on the result of their experiments,
they had found out that unigram co-occurrence statistics is a
good automatic scoring metric as it is capable of constantly
achieving high correlation with human assessments.
Lin [7] also developed an automatic summary evalua-
tion program called Recall-Oriented Understudy for Gisting
Evaluation (ROUGE). The current version of ROUGE con-
sists of five different automatic evaluation methods, namely
ROUGE-N, ROUGE-L, ROUGE-W, ROUGE-S and ROUGE-
SU. ROUGE-N uses n-gram co-occurrences between the can-
didate and reference summaries, which is similar to the BLEU
491
algorithm in machine translation. N -gram with length greater
than one can be used to estimate the fluency of summaries.
ROUGE-L consists of matching two sequences by matching
their subsequence. The longer the matching subsequence, the
more similar the two sequences are. ROUGE-W is simi-
lar to ROUGE-L in which they both deal with matching
subsequences but in ROUGE-W weights are used. ROUGE-
S uses skip-bigram to estimate the similarity between two
summaries. Since ROUGE-L and ROUGE-W can only match
subsequence, ROUGE-S compensates this by being able to
match pairs of word in their sentence order with arbitrary gaps
in between them. ROUGE-SU is similar to ROUGE-S with
the addition of unigram based co-occurrence statistics. The
evaluation of ROUGE had shown that it correlates surprising
well with human evaluations.
III. P ROPOSED A PPROACH
In semantic assessment of summary writings, student so-
lutions are graded based on the number of content points
answered. Apart from those commercial techniques such as
LEA and Summary Street, there are mainly four summary
assessment techniques, namely LSA, BLEU, n-gram co-
occurrence and ROUGE. After evaluating these techniques,
we found that the overall performance of ROUGE is quite
poor compared with the other three techniques. We then built
the ensemble approach using LSA, n-gram co-occurrence and
BLEU. However, we found that BLEU produced low scores in
its performance when ensembled with other techniques. This
might due to the brevity penalty over penalization. As such,
the resultant ensemble approach only comprises LSA and n-
gram co-occurrence. Furthermore, as the LSA and n-gram co-
occurrence techniques have roughly the same performance,
both techniques will have very similar weights if the weighted
approach is used. Since the weights are similar, the use of the
unweighted approach will simplify the amount of processing
required by the ensemble approach.
Figure 1 shows the proposed ensemble approach which con-
sists of two major modules: pre-processing and ensembling.
Student Standard
Solution Solution
Pre-processing
Word Stop Word Synonym
Stemming Removal Replacement
Ensemble
N-gram Co-
LSA
occurrence
Ensemble
Score
Fig. 1. The proposed ensemble approach.
A. Pre-processing
Both the student’s candidate solution and the standard
(reference or model) solution will first go through the pre-
processing module. To avoid the problem that the student’s
candidate solution uses different words from the reference
summary, the pre-processing module aims to create a common
basis for comparison by converting all words used by the
candidate and reference summaries to a common one. There-
fore, the pre-processing module provides text pre-processing
functions such as converting synonyms into a common word,
eliminating grammatical differences and removing stop words.
For the first two functions, WordNet [9] was used. As for the
removal of stop words, a list obtained from the University of
Glasgow1 was used.
B. Ensembling
The ensemble approach comprises the modified LSA algo-
rithm and n-gram co-occurrence which are discussed in this
subsection.
1) Modified Latent Semantic Analysis (LSA): When apply-
ing Latent Semantic Analysis (LSA) to summary assessment,
first, a reference summary and a student’s candidate summary
is split into a set of sentences. For any summary, suppose
there are m distinct terms in n sentences. The summary
can be represented as a term-sentence (m × n) matrix X,
whose component xij is the weighted frequencies for how
often a term ti occurs in a sentence dj . The original matrix
X is then broken into the product of three new matrices
X = U ΣV T where U and V are the matrices of the left
and right singular vectors for terms and sentences respectively.
Σ comprises a diagonal of scaling factors. Some number k of
the scaling factors is retained and the matrices are recombined
using only the retained factors. Thus, the original matrix X is
approximated with a rank-k matrix Xk = Uk ΣVkT by setting
the smallest r − k singular values to zero (r is the rank of X).
The result is a compressed form of the original matrix in
which frequency values are approximated (raised or lowered)
depending on the number of factors used. After generating the
compressed matrix for a reference summary, a vector for each
sentence can be constructed by taking values in the matrix for
each term found in that sentence. A vector for each sentence
in the candidate summary can also be computed in a similar
way. The cosine distance between the reference vector and the
candidate vector can then be calculated as an indication of their
semantic similarity. A candidate sentence can be considered
as matched with a reference sentence if their cosine distance
is within an empirically determined threshold. The final score
is computed as the total number of matched sentences out of
the total number of sentences in the reference summary.
2) n-gram Co-occurrence: An n-gram is a subsequence of
n items from a given sequence. In our application here, n-
gram refers to a subsequence of n words in a sentence. An
n-gram of size 1 is a “unigram”; size 2 is a “bigram”; size 3 is
a “trigram”; and size 4 or more is simply called an “n-gram”.
1 http://ir.dcs.gla.ac.uk/resources.html
492
N -gram co-occurrence measures how well a candidate
summary overlaps with a reference summary using a weighted
average of variable length n-gram matches. First, the n-gram
match ratio is calculated as follows:
 
Sr ∈S n-gram∈Sr Countmatch (n-gram)
Cn =   (1)
Sr ∈S n-gram∈Sr Count(n-gram)
where S = {S1 , S2 , ..., SR } comprises all the sentences in
a reference summary. Countmatch (n-gram) is the maximum
number of n-grams co-occurring in a candidate summary and
a reference summary and Count(n-gram) is the number of n-
grams in the reference summary.
The n-gram co-occurrence
j statistics is defined as
n-gram(i, j) = exp ( n=i wn log Cn ) where j ≥ i, i
and j range from 1 to 4, and wn = 1/(j − i + 1). n-
gram(1, 4) is a weighted variable length n-gram match
score similar to the IBM BLEU score [10]; when i = j,
n-gram(i, i) is simply the average i-gram coverage score Ci .
3) Ensemble Approach: In the ensemble approach, the
scores of the individual techniques of LSA and n-gram co-
occurrence are used for the unweighted voting by taking an
unweighted average, i.e.,
LSA Score + n-gram co-occurrence Score
Ensemble Score =
2
A threshold value will be assigned to determine if the averaged
score is considered as a positive or negative solution.
IV. P ERFORMANCE A NALYSIS
In this section, we present the performance of the proposed
ensemble approach in comparison with other assessment tech-
niques. As LEA and Summary StreetTM are commercial and
patented techniques, we were unable to obtain their programs
for testing. However, the other techniques such as LSA,
BLEU, ROUGE and n-gram co-occurrence are compared. The
following six different types of tests are used to compare the
performance. The objectives of these tests are given below:
• Exact test - It is used to judge if the technique is capable
of providing a high score for totally related candidate
summary and reference solution.
• Opposite test - It is used to judge if the technique
is capable of providing extremely low score when the
candidate summary and reference solution are totally
unrelated.
• Content test - It is used to determine whether the tech-
nique is capable of producing a score that is proportional
to the number of content points present in the candidate
summary.
• Synonym test - It is used to determine if the technique
is able to evaluate the candidate summary based on their
content and not be influenced by the different synonyms
used in the summaries.
• Grammar test - It is used to determine if the technique
is able to evaluate the candidate summary based on their
content and not be affected by the different grammar used
in the summaries.
 • Student test - It aims to determine if the technique is LSA N-gram BLEU ROUGE Ensemble

capable of producing score that is closely related to the 1.2

one that is given by a human expert. The candidate sum- 1

maries used in this test are written by current students, 0.8

as opposed to those that are generated artificially used

Accuracy
0.6
for the above tests. Therefore, this allows us to test if
0.4
the technique is capable of accurately assessing real-life
summaries. 0.2

The six tests are used to evaluate the performance of the 0

Exact Opposite Content Synonym Grammar Student
ensemble approach in comparison with other base techniques. Techniques

All reference summary solutions used in the tests are obtained

from Cambridge O-Level English Language Examination [11],
[12]. The performance evaluation was conducted on 50 test Fig. 3. Performance comparison of the existing techniques and the proposed
ensemble approach.
samples (or student summaries) with 1 being the most accurate
and 0 being the most inaccurate for all the tests. All the test
samples were collected from a class of students taking the
Mid-Year Examination 2007 of Hillgrove Secondary School as it had the highest chances of producing them while having
in Singapore. These candidate summaries had been graded by the lowest probability for producing false negatives. On the
their O-Level English teacher. whole, the ensemble approach proves to be superior to the
Figure 2 shows the accuracy of the ensemble approach base techniques.
for each of the six tests versus the different settings of the
TABLE I
threshold value that defines the matching criteria. It can be C ROSS COMPARISON BETWEEN THE ENSEMBLE APPROACH AND OTHER
observed that the optimal threshold value is 0.7 as the accuracy TECHNIQUES IN FP AND FN.
starts to decline beyond this value. Method False Positive False Negative
LSA 0.094 0.228
Exact Opposite Content Synonym Grammar Student
N-gram 0.033 0.093
BLEU 0.026 0.194
1.2 ROUGE 0.003 0.476
Ensemble 0.124 0.046
1

0.8
Accuracy

0.6

0.4
0.2
0 integrates two of the most effective assessment techniques
0.3 0.4 0.5 0.6 0.7
Threshold
Fig. 2. Performance of the proposed ensemble approach vs threshold values.
Figure 3 shows a comparison of the ensemble system and
the base techniques on LSA, n-gram co-occurrence, BLEU
and ROUGE using their best performance parameters and
thresholds. It can be observed that the ensemble system is able
to outperform all the base techniques in all the tests except for
the content test. For the other tests, the ensemble approach is
capable of outperforming the other techniques by at least 0.003
and at most 0.774 in terms of accuracy. Based on the results of
the tests, the proposed approach is capable of producing equal
or higher accuracy compared to the existing techniques in all
tests except for one. Even though the proposed approach did
not perform as well in the content tests, its overall accuracy of
96% is still much higher than that of the existing techniques.
When comparing the chances of producing false positives
with the existing base techniques as shown in Table I, the
ensemble approach is slightly worse than the other techniques
V. C ONCLUSIONS
In this paper, we propose an ensemble approach which
0.8 0.9
of LSA and n-gram co-occurrence into an efficient technique
for automatic summary assessment. Performance comparison
between the proposed ensemble approach with other existing
techniques has also been conducted. The proposed approach
has achieved an overall accuracy of 96% as compared to the
best existing technique, BLEU, which has an overall accuracy
of 87%. For future work, as the techniques used and proposed
in this paper are mainly based on latent semantic analysis
or machine translation based evaluation techniques, we will
investigate the effectiveness of using machine learning or
statistical approaches for the assessment of summary writings.
In addition, as our current approach only focuses on semantic
assessment of contents, we also intend to develop a complete
summary assessment system by incorporating an English lan-
guage assessor and style checker.
R EFERENCES
[1] D. Perez, A. Gliozzo, C. Strapparava, E. Alfonseca, P. Rodriguez,
and B. Magnini, “Automatic assessment of students’ free-text answers
underpinned by the combination of a bleu-inspired algorithm and latent
semantic analysis,” in Proceedings of the 18th International FLAIRS
Conference, Clearwater Beach, Florida, May 2005.

493
 [2] T. Landauer, D. Laham, B. Rehder, and M. E. Schreiner, “How well can
passage meaning be derived without using word order? a comparison of
latent semantic analysis and humans,” in Proceedings of the 19th Annual
Conference of the Cognitive Science Society, 1997.
[3] T. Landauer, P. Foltz, and D. Laham, “Introduction to latent semantic
analysis,” Discourse Processes, vol. 25, pp. 259–284, 1998.
[4] I. Zipitria, J. Elorriaga, A. Arruate, and A. de IIarraza, “From human
to automatic summary evaluation,” in 7th International Conference on
Intelligent Tutoring System, 2004.
[5] M. Franzke and L. Streeter, “Building student summarization, writing
and reading comprehension skills with guided practice and automated
feedback,” Highlights From Research at the University of Colorado, A
white paper from Pearson Knowledge Technologies, 2006.
[6] D. Pérez, E. Alfonseca, and P. Rodrı́guez, “Upper bounds of the
BLEU algorithm applied to assessing student essays,” in Proceedings of
the 30th International Association for Educational Assessment (IAEA)
Conference, 2004.
[7] C.-Y. Lin, “ROUGE: a package for automatic evaluation of summaries,”
in Proceedings of the Workshop on Text Summarization Branches Out
(WAS 2004), Barcelona, Spain, July 2004.
[8] C.-Y. Lin and E. Hovy, “Automatic evaluation of summaries using n-
gram co-occurrence statistics,” in Proceedings of the 2003 Conference
of the North American Chapter of the Association for Computational
Linguistics on Human Language Technology, 2003, pp. 71–78.
[9] G. A. Miller, C. Fellbaum, R. Tengi, P. Wakefield, R. Poddar, H. Lan-
gone, and B. Haskell, WordNet: a lexical database for the English
language, Princeton University Cognitive Science Laboratory, 2006.
[10] K. Papineni, S. Roukos, T. Ward, and W. jing Zhu, “BLEU: A method
for automatic evaluation of machine translation,” in Proceedings of the
40th Annual Meeting of the Association for Computational Linguistics,
2002, pp. 311–318.
[11] K. Rajamanikum, English language (Yearly) Worked Solutions. Redspot
Publishing Singapore, 2000.
[12] J. Lee, O-Level English. Singapore Asian Publications (S) Pte Ltd,
2005.

494
 2009 International Conference on Computer Engineering and Technology

A fast reassembly methodology for polygon fragment

Gang XU, Yi XIAN

Department of Electrical and Electronic Engineering
North China Electric Power University
Beijing 102206, China
pxugang@163.com, xianyi0743@hotmail.com

Abstract—A new general methodology is introduced for

computer-aided reassembly of the polygon image fragments. II. REASSEMBLY METHODOLOGY BASED ON CORNER
Since polygon image fragment has a characteristic that it has DETECTION AND GRAY CORRELATION
obvious corners, this methodology takes advantage of such
characteristic, and detects the corners by Hough transform A. Fragment image pretreatment
and fuzzy reasoning. Meanwhile, it computes the length of each After we obtain an image with some fragments, as shown
side, compares the side length, finds out the sides with equal
Figure 1, each fragment is separated from the background
length in different fragments, and then computes the gray
using image segmentation algorithm. Subsequently, each
correlation of equal sides to determine the most optimal
matching location for reassembly. Experimental results show
fragment contour is obtained using following quite classical
that the reassembly methodology for polygon fragment based method.
on corner alignment and gray correlation is effective and fast.

Keywords—Fragment reassembly, corner detection, Hough

transform, fuzzy reasoning, gray correlation.

I. INTRODUCTION
The problem of image fragment reassembly is a novel
and typical application in the field of computer vision,
pattern recognition and image analysis. In the computer
vision domain, it can be approximately viewed as the jigsaw
puzzle problem which has been researched for some years,
[1], [2]. At present, the hard, tedious and repetitive works of Figure 1. Original fragments image.
reassembling image fragments are still completed artificially
in the fields of reassembling information fragments, judicial The color depth of each fragment decreases from 256 to
authentication and cultural relic fragments repair. So how to black and white under a threshold. Therefore, the whole
use computer to finish reassembly is one of the important fragment is white and its background black. Then we can
problems for many archaeologists and criminologists. execute contour trace algorithm to generate the fragment
The hot research topic aims at irregular curve edge contour in the form necessary for subsequent analysis. At the
fragment whose contour is unique. Therefore, we can same time, the gray values on the contours have been
reassemble the fragments using their contours, [3], [4], [5], recorded.
[6]. However, another type of fragment, irregular polygon B. Corner detection based on Hough transform and fuzzy
fragment, has seldom been researched. The most usual reasoning
example is paper fragment from paper shredder, whose sides
This paper aims at irregular polygon fragment, so the
are straight lines. Because the sides on polygon contour are
corner is defined as the intersection point between two
the same object in shape for computer, if we use the
straight lines, and a typical corner is shown in Figure 2a.
aforesaid contour matching methodology to deal with such
Because the corner is an important feature point on contour,
kind of fragment, there will be so much redundant and wrong
there are numbers of detection methods in the past. [7], [8],
matching-pair that the reassembly work will be failed. In
[9], [10], [11].
reaction to the irregular polygon fragment, we must take
The corner detection method based on Hough transform,
advantages of shape and gray information in fragment image,
which uses the characteristic that contour corner has two
so this paper proposes a polygon fragment reassembly
peaks in the Hough transform mapping space, can find out
methodology based on corner alignment and gray
the corner accurately. According to the principle of Hough
correlation, which proved to be correct by abundant
transform, the equation of the straight line which goes
instances.
through the origin can be represented as

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DOI 10.1109/ICCET.2009.144
 x sin θ + y cos θ = 0 . (1) arm i j 2 = max {θ j } . (3)
j∈[1,180 ]∩ j ≠ j 1

The formula has only one parameter θ , and

y Obviously, if pi is a corner, the values of arm i j 1 and
θ = arctg ( − ) . It illuminates that if one straight line
x arm i j 2 will be big or large. Hence, arm i j 1 and arm i j 2 are
which goes through the origin, all feature points on line will
good measures as evidences for pi to be a corner. To deal
have the same value θ , and a peak will be in mapping space.
If the origin is a corner, there will be two peaks in the with these evidences for existence of corner at semantic level,
mapping space, as shown in Figure 2b. we introduce fuzzy sets to represent them. The fuzzy
membership values for arm i j 1 and arm i j 2 are determined as
shown graphically in Figure 4a and Figure 4b.
Definition 2. Every point pi on contour has an angle with
front point and behind point, and the angle value is

δ i = ∠p i + L _ Hough p i p i − L _ Hough , (4)

then,

2
Figure 2. (a) A typical corner; (b) corner’s Hough transform mapping
space; there are two peaks in the mapping space.
δ i* = ∑ω
k = −2
k δ i+ k , (5)

where {ω k } are smooth weights. The fuzzy membership

value for δ i* is determined as shown graphically in Figure
4c.

Figure 3. A part of practical contour with noise, point P is a corner.

The corner is a partial feature on contour, so we can

execute Hough transform in a sliding window. In practical
application, the actual contour is always influenced by
contour extraction error and various kinds of noise, as shown
in Figure 3, so that the traditional corner detection method
must enlarge the sliding window to withstand noise. This
problem will make the amount of computation increase. In
this situation, this paper proposes a new corner detection
method based on Hough transform and fuzzy reasoning to
reduce the sliding window and withstand the noise.
In the method, we can choose a small sliding window
whose length is L _ Hough for Hough transform and obtain
the angle values of current point p i in mapping space, [11].
Definition 1. Every point on contour has 180 θj (here, the
quantification precision of Hough transform is 1 ), then the
maximal peak of the current computing point pi in mapping
space is
arm i j1 = max {θ j } ,
j∈[1,180 ]
and the second maximal peak is
Figure 4. Determining fuzzy membership values; (a)quality
arm i
j1
arm i j 2
fuzzy membership; (b)quality fuzzy membership;
δ*
(c)quality i fuzzy membership.
The conclusion whether current point p i is a corner has
two conclusion subsets {YES , NO } , and then the membership
value of p i with respect to class YES is calculated as
μ YES ( pi ) = μ big ( pi ) ∧ μ l arg e ( pi ) ∨ μ bend ( pi ) . (6)
From the formula, we can know whether the point p i is
a corner. Using fuzzy reasoning can strengthen anti-noise
robust of corner detection method based on Hough transform
and shorten the sliding window to reduce amount of
computation.
(2) C. Fragment contour rotation
Fragment contour rotation is a very important step for
fragment reassembly. After contour corner detection, we can
obtain the coordinates of corner, and we can also compute

496
the length and angle of every side. Subsequently, find out the
equal length sides, and calculate the difference between
equal length sides’ angle, as shown in Figure 5. The error
Δp of angle depends on quantification precision of Hough
transform, and rotation can be executed only γ ,
γ = α − β ± 2Δp . So we can obtain the optimal matching
angle between two fragments by 5 rotation computations. Of
course, rotation around the origin by an angle γ moves
point ( x, y ) via
⎡ Y ⎤ ⎡cos(γ ) − sin(γ ) ⎤ ⎡ y ⎤ .
⎢ X ⎥ = ⎢ sin(γ ) cos(γ ) ⎥ ⎢ x ⎥
⎣ ⎦ ⎣ ⎦⎣ ⎦
Figure 5. Rotation between two fragments.
Direct application of this formula probably reduces the
number of pixels on contour, because the rotation
computation possibly generates a small quantity of
coincidence points. In order to solve the problem, we can use
averaging method to determine the gray value at the
coincidence point, and then re-execute the contour trace
algorithm to rotated contour.
D. Gray correlation matching
In order to find out accurate matching location or
distinguish the true matching sides apart, we must compute
the gray correlation among equal length sides. The
measurement of gray correlation can be represented by
distance between two contour gray vectors. Suppose that the
equal length side has L pixels, the gray value of every pixel
is xi , and then the gray vectors of two sides are represented
as
X 0 = {x1 , x2 , x3 ,..., x L } ;
X l = {x1' +l , x2' +l ,..., xL' +l } .
Here, X 0 is the gray vector of the fixed fragment contour,
l can be set at will, and it is set as {− 2,−1,0,1,2} in this paper.
The distance between two gray vectors is
Dl ( X 0 , X l ) = X 0 − X l = ( X 0 − X l )( X 0 − X l )T .
Calculate and find out the minimal distance Dmin ( X 0 , X l )
to determine the accurate matching location. The
corresponding l of Dmin ( X 0 , X l ) is the optimal matching
location between two sides.
(7)
Figure 6. Gray correlation between matching sides; (a) the sides L1 and
L2; (b) the computing result of gray correlation between two sides.
There are two true matching sides, L1 and L2, as shown
in Figure 6a. From optimal matching location setting forward
38 pixels as the starting offset, shift backwards one pixel
every computation. And there are 50 computing values under
different offsets. As shown in Figure 6b, the optimal
matching location is determined at offset 38 whose
corresponding Dmin is 313.7.
E. Methodology represent and analysis
A polygon fragment reassembly system has been
developed to implement the aforementioned algorithm, and
the Figure 7 shows the flow chart of the system. After the
corner detection of all fragments, we can obtain some ( ≥ 2 )
length matching sides. If the number of matching sides
exceeds 2, the true matching sides will be easily
distinguished through gray correlation of matching sides.
The concrete matching process in system is as follow:
1) If the number of length matching sides is 2, fix one
fragment and move the other fragment to align the one
corner at the fixed matching side. Subsequently, rotate the
second fragment contour around the aligned corner and
compute the optimal matching angle, and then compute the
optimal matching location by gray correlation. Lastly, rotate
the whole second fragment to complete the reassembly
process.
2) If the number of length matching sides is n , n>2 , we
must firstly determine the true matching sides using gray
correlation of contour, and then execute the step (1).
(8) 3) Detect the corner of new fragment contour from step
(1), and set new fragment as the fixed fragment. Then find
out the matching sides to execute step (1) or (2), until all
(9) fragments have been reassembled.
The maximal amount of computation of reassembly
methodology attributes to Hough transform, because the
transform is executed at every point on contour. Suppose that
the fragment number is M , the pixel number in each
fragment is NPi , i ∈(1,2,...,M) , and length of each fragment
contour is NC i .Then Hough transform needs
(10) M
∑ NC × L _ Hough multiplication and addition. The other big
i =1
i
amount of computation attributes to fragment rotation, and it
497
 M −1
needs
∑ NP i
multiplication and addition. According to the
i =1
analysis, the amount of computation lies on the number and
size of fragments and L _ Hough . In this methodology, we
can use a smaller value of L _ Hough , so the amount of
computation can be reduced obviously. For a certain size
image, the ultimate value of amount of computation is
certain, as shown in Figure 8, and that is calculating all
pixels in image. But in practice, it is impossible. Therefore,
the polygon fragment reassembly methodology based on
corner alignment and gray correlation is effective and fast.
Figure 7. The flow chart of the actual reassembly system.
Figure 8. Relationship among computing cost, number of fragment and
length of sliding window.
498
III. EXPERIMENTAL RESULTS
In order to verify the possibility and effect of the
fragment reassembly methodology, a test was done with a
scan image from a part of newspaper, as shown in Fig. 1.
The computer’s CPU frequency is 2.8GHz, EMS memory is
768Mb, computing software is Matlab 7.0, and image size is
520×540 pixels.
The parameter values used in reassembly process was set
as follow:
a) two different values of L _ Hough : First 60, then
30;
b) M = 7.
We firstly set Hough transform sliding window,
L _ Hough = 60 . The corner detection method in
reassembly process was used with method based on Hough
transform and fuzzy reasoning and method based on Hough
transform, [9]. Both of the reassembly processes with
different corner detection methods were completed, and the
reassembly results were right. Both of the runtime we used
were less than 3 seconds.
Then, we set Hough transform sliding window,
L _ Hough = 30 .As above, we used the reassembly
methodology with different corner detection methods. This
time, only one corner detection method based on Hough
transform and fuzzy reasoning had completed the reassembly
process, and runtime we used was less than 2 seconds. And
the other method failed, because it couldn’t find the correct
corner. From these experiments, we know that the corner
detection method based on Hough transform and fuzzy
reasoning has the better anti-noise performance and faster
computing speed. They also prove that the polygon fragment
reassembly methodology based on corner alignment and gray
correlation is effective and fast. The correct reassembly
result is shown in Figure 9. We still have completed some
other experiments such as shown in Figure 10, and the
reassembly runtimes are shown in Table I.
(a) (b)
Figure 9. Fragment reassembly result; (a) the contour reassembly result;
(b) the complete reassembly result.
During the experiments, we find that the scan precision
of paper fragments is also an important reason for successful
reassembly and amount of computation. If the scan precision
is too low, incorrect matching will increase and even lead to
reassembly failure. Oppositely, if scan precision is too high,
the amount of computation will increase a lot. So we need to
choose an appropriate scan precision in actual experiment.
 TABLE I. RUNTIME IN THE DIFFERENT EXPERIMENTS [4] da Gama Leitao H.C., Stolfi J, “A Multiscale Method for the
Reassembly of Two-Dimensional Fragmented Objects”, IEEE
Type of Size M L _ Hough Runtime(s Transactions on Pattern Analysis and Machine Intelligence, Sep 2002,
image ) 24(9), pp. 1239-1251.
newspaper 500×700 9 30 <2
newspaper 500×1000 12 25 <3 [5] RAO Rui-ling, JIN Xue-feng, LU Huai-wei, “A Method of Contour
newspaper 1500×1000 18 30 <5 Matching Based on Chain Code”, Computing Technology and
Automation, Jun 2007, 26(2), pp. 34-37.
picture 1944×2592 28 21 <6
picture 3072×2048 40 25 <11
[6] ZHOU Feng, HUANG Xiao-ming, “Algorithm of 2D Contours
Matching Based on Angle-sequence”, Science Technology and
Engineering, Aug 2007, 7(15), pp. 3757-3760.
IV. CONCLUSION
In this paper, a new fragment reassembly methodology [7] E.R.Davies, MA, DPhil, etc, “Application of the generalized Hough
based on corner alignment and gray correlation is proposed transform to corner detection”, IEE Proceedings, Jan 1988, 135(1), pp.
for polygon image fragment. Using the obvious feature 49-54.
corners on polygon contour and gray correlation among
fragments, we can complete fragment reassembly process [8] Pritchard A.J., Sangwine S.J., Horne R.E.N, “Corner and Curve
Detection along a Boundary Using Line Segment Triangles”, IEE
correctly. This methodology proves to be fast and highly Colloquium on Hough Transforms, 7 May 1993, P2/1-P2/4.
effective both in theory and practical experiments.
REFERENCES [9] Liyuan Li, Weinan Chen, “Corner detection and interpretation on
planar curves using fuzzy reasoning”, IEEE Transactions on Pattern
[1] Freeman H , Garder L, “Apictorial jigsaw puzzles: the computer Analysis and Machine Intelligence, Nov 1999, 21(11), pp. 1204-
solution of a problem in pattern recognition”, IEEE Trans. Elec. 1210.
Comp, 1964, 13, pp. 118-127.
[10] Khattak N.S., Stockman G, “Shape and Structure for Image Matching
[2] Burdea B.G., Wolfson H.J., “Solving Jigsaw Puzzles by a Robot”, and Retrieval”, Proceedings of the IEEE International Conference on
IEEE Transactions on Robotics and Automation, Dec 1989, 5(6), pp. Machine Vision, Dec 2007, pp.79-84.
752-764.
[11] Chen Hongbo, Xu Xiaorong, “Corner Detection Based On Improved
[3] Papaodysseus C, Panagopoulos T, Exarhos M, etc, “Contour-Shape Hough Transform”, Computer Applications and Software, Feb 2008,
Based Reconstruction of Fragmented, 1600 BC Wall Paintings”, 25(2), pp. 57-60/76.
IEEE Transactions on Signal Processing, June 2002, 50(6), pp. 1277-
1288.

(a) (b) (c)

Figure 10. An example of picture fragment reassembly; (a) the initial fragment image; (b) the contour reassembly result; (c) the complete reassembly result.

499
 2009 International Conference on Computer Engineering and Technology
A Data Mining Approach to Modeling the Behaviors of Telecom Clients
LIU Xiaodong
Department of Computer Software Engineer
Shenzhen Institute of Information Technology
Shenzhen, P.R.China
liuxd@sziit.com.cn
Abstract：We begin this paper by describing our rationale and
overall design of an exceptional client model based on data mining
algorithm. Then, we continue by summarizing the simulation
details and describing the type of results obtained from
implementing the proposed system, which consists of three
heterogeneous data mining algorithms. The idea behind the model
is that three heterogeneous data mining algorithms are combined
in the phase of data processing and one algorithm’s outcome is
another algorithm’s input. By using the method of neural network,
the exceptional client attribute weight is calculated based on the
original dada. The characteristics of exceptional client are
identified accordingly by using the method of decision tree based
on the attribute weight. Then, the distinguishing model is
generated adaptively on the basis of clustering. The combination
of the three algorithms helps distinguish exceptional client
effectively. It can not only alarm the existing system but also
analyze and specify the data of exceptional client and consequently
supports the distinguishing system.
Key Words: Data Mining；Neural network；Decision Tree
；Clustering；Real Distinguishing
I. INTRODUCTION
Exceptional Client Distinguish(ECD）is used to detect
exceptional clients from common clients, which is very
popular in the field of financial and telecom. Generally,
client distinguish includes credit card exceptional client
distinguish and telecom exceptional client distinguish[1].
This paper is concerned with telecom client distinguish
technique. Traditional telecom exceptional client distinguish
technique depends on manual history data analysis by
telecom experts. The idea of this method is to find character
of exceptional client, and then apply the rule in the client
relation management system. This kind of work pattern has
two drawbacks. Firstly, the manual history data analysis by
telecom experts uses more person and money. Secondly, it
often misses the best chance to find exceptional client after a
long time for searching exceptional client.
A novel data mining based exceptional client distinguish
method (DMECD) is proposed in this paper, which
combines neural network algorithm, decision tree algorithm,
and clustering algorithm. This method can find exceptional
clients automatically, real time, and effectively, which gives
not only alarm to the system, but also character rules to the
client serve department. The following is the advantage of
DMECD comparing with traditional method:
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DOI 10.1109/ICCET.2009.164
（1）Real time detection. It can find exceptional clients
in a few days or even more short time, which is helpful for
CRM to adjust configure quickly.
（2）Automatically detection. Most of the exceptional
client distinguish job can be automatically done without
manual work, which means short time and great efficiency.
（3）Adaptation to new exceptional client detection.
DMECD can find interesting data attributes based on
decision tree although exceptional client’s character is
protean. This self-adapted attribute reduction method based
on decision tree algorithm can find not only all exceptional
clients and its variation, but also new exceptional client type.
The paper is organized as follows. In the next section the
principle and process that relates to our algorithm DMECD
(a Data Mining method based on Exceptional Client
Distinguish) is presented, followed by the experimental setup
used in this study and analyze the results in Section 3. We
conclude with a summary of our contributions and raise
several issues for future work in Section 4.
II. DMECD
A. Principles of DMECD
In this section, we present an algorithm DMECD (a Data
Mining method based on Exceptional Client Distinguish) for
distinguishing exceptional clients. Firstly, the proposed
method belongs to the family of exceptional distinguish
methods, which is one of the most important distinguish
methods[2]. The principle of exceptional distinguish is to
classify new data by the normal data model which is derived
from history data.
Secondly, self-adapted distinguish model method is used
in DMECD. The principle of this method is described by the
following figure 1. Data is collected from related systems
and cleaned in the part of Data Preparation. Then in the part
of Attribute Reduction, attributes are reduced by weights,
which are computed by specific algorithm. In the next part,
Characters of exceptional client behavior are generated by
specific algorithm. The result of this part can be used in the
fourth part and the fifth part in the sense of those characters
is high abstracts from raw data. In part 4, Self-adapted
distinguish model is generated by specific self-study
algorithm from history data and characters. In part 5,
DMECD can distinguish exceptional clients automatically
depends on the self-adapted distinguish model and characters
of behavior.
 1. Data 2. Attribute
Preparation Reduction
3. Characters of
Behavior
5. To distinguish
exceptional clients
4. Self-adapted
distinguish model
Fig.1 Self-adapted distinguish model generation
Self-adapted model methods based on exceptional
distinguish need specific algorithms. Clustering algorithm in
the field of data mining is one of the choosing methods[3].
The idea of clustering algorithm is that similar data belong to
one cluster. The distance between same type data is closer
than the distance between different type data. Clustering
algorithm can generate normal client cluster from history
data by adjusting algorithm parameters, while the distance of
exceptional client will be faraway from the center of normal
cluster.
Clustering algorithm parameters include attributes, such
as average fee, social circle. The selection of these
parameters is very important to the performance of clustering
algorithm. In this work, attributes reduction is implemented
with the neural network learning algorithm, which is a more
direct implementation of the theory. Some evidence exists
that DMECD works better in practice. The process of
DMECD for the study is discussed in detail in the next
section.
B. The process of DMECD
The process of DMECD includes model generation and
exceptional clients distinguish. Three algorithms include
neural network, decision tree, and clustering in the field of
data mining are used to generate models in the DMECD.
This paper focuses on a particular form of data
(attribute-value vectors), so a brief overview of relevant
concepts and terminology seems appropriate. We assume
that data is described using a fixed collection of features or
attributes. A single case is then a vector of values, one for
each attribute, which can also be thought of as a point in the
instance space in which there is one dimension for every
attribute.
Neural network, decision tree, and clustering are
constructed from a training set, which consists of objects.
Each object is completely described by a set of attributes and
a class label. The concept underlying a data set is the true
mapping between the attributes and the class.
The following will give more details about DMECD. We
assume a given set of N instances, D = ( xn , yn ) with n =
1,… , N, where xn is a multidimensional input vector, and
501
yn is the output variable. Let { Ai 1≤ i ≤ MaxAtt} be
the set of attributes of a domain, and for each Ai ,
{ Vij 1 ≤ j ≤ MaxAttVali} be its value set, where MaxAtt is
the number of attributes, and MaxAttVali is the number of
different values of Ai . Since the focus of this paper is on
classification problems, yn takes values from a set of
predefined values, which is yn ∈ {Cl1 , …, Clc } , where c is the
number of classes, xn takes values from Vij . A classifier C is
a function that is applied to the training set D to construct a
model C(D). The generated model is a mapping from the
input space X to the discrete output variable Y. When used
as a predictor C( xi , D), it assigns a y value to the
example xi .
To characterize correlation of yn and Ai , we first
propose to use weights of attributes. Classifiers
C1 ( D), C 2 ( D), …, C T ( D ) change weights of attributes and
weights of instances when they misclassify one instance.
Then rules will be constructed from the attributes. One of the
challenges in constructing rules is to choose criteria for
partitioning the instances at each node. Some examples of
the different node splitting criteria include entropy or its
variants, the chi-square statistic, the G statistic, and the GINI
index of diversity. Despite these efforts, there appears to be
no single node splitting that performs the best in all cases;
nonetheless there is little doubt that random splitting
performs the worst.
Our node splitting criteria is a variant of entropy. Prior to
introducing it we defined some notation used throughout this
section. Of N instances, N Cl k belong to class Clk .
∑ k NCl k = N . The decision rule at the node splits these
instances into V partions, or V child nodes, each of which
has N (v ) instances. In a particular partition, the number of
examples of class Clk is denoted by N Cl (v )
. ∑ k N Cl
(v)
= N (v) .
k k
Information gain is based on the calculation of entropy.
More specifically the information gain resulting from
splitting the instances based on attribute Ai can be written
as,
c N Cl k N Cl k v N (v) c
G( Ai )= [ ∑ − ( ) log( )] - [ ∑ − ( ) ∑−
k =1 N N v =1 N k =1
(v) (v )
N Cl N Cl
k k
( ) log( )] (1)
N (v) N (v)
The first term in Eq. (1) is the entropy at the parent node
and the second term is the weighted entropy of the child
nodes. wA j represents weights of attribute A j . We define
new information gain as:
 c N Cl k N Cl k wA j For i:= 1 to d do
wG( Ai )= [ ∑ −( ) log( )] - ( )
k =1 N N 1 − wA j {
(v ) (v) For k:= 1 to n do
v
N (v) c N N
[∑ − ( ) ∑ −( Cl k
(v)
) log( Cl k
(v)
)] (2)
v =1 N k =1 N N { Choose the attribute A j which has the smallest
weights.
The attribute which will be chosen is the one that results
in the largest attribute weight information gain. So the bigger Induce classifier Ĉ t , using training data D
wA j is, the more important it is. Refer to Table 1 for details
of the DMECD boosting algorithm. Calculate ε t of Ĉ t on the training data.
Table 1. DMECD Induce classifier Ĉ t , using training data D without A j
Given:
Calculate error rate εˆ t of Ĉ t on the training data.
A data set D contains n training instances xi with
Compare ε t and εˆ t
labels yi , d attributes A j , A Maximum trials, T. Base
learner, C. If ε t > εˆ t Then, Delete A j from D, j=j-1;
/*Initialization*/
Else Return Tree Ĉ t based on attribute A j
For j:= 1 to d do wA0j = 1 / d /* initial weights of all
attributes */ }

For t:= 1 to T do }
/* Self-adapted distinguish model */
{Induce classifier C t , using weighted training data and
C. Function Cluster( Ĉ t )
Calculate attribute error rate εAt of C t on the training {If stop Criteria( Ĉ t ) = True
data.
Generate a subcluster: Cluster j = Rule( D j )
If εAt > 1/2, discard C t and stop.
}
If εAt = 0, then C t gets infinite weight.
/*Re-weight attributes*/
The process of model generation is described by the
if C t ( xi ) ≠ yi then, For j := 1 to d do figure 2.

{if wAtj is concerned with C t ( xi ) = yi Neural net Decision tree

Data 1 Attribute 1 Character 1
Data 2
then wAtj +1 = wAtj /(2εAt ) ……
Attribute 2 Character 2
…… ……
Data n
else, wAtj +1 = wAtj /(2(1 − εAt )) . Attribute m Character m

Renormalizing so that ∑ j wAtj+1 = 1 Self-adapted distinguish model

Clustering
} Cluster 1
/*Construct the order of attribute Aj / Cluster 2
……
The order of attribute A j is obtained by comparing the Cluster m
t
attribute weights wA j . Fig.2 The process of model generation

/* Decision tree */

502
 Attribute reduction, character generation, and exceptional III. EXPERIMENTAL RESULTS
model are three main steps of DMECD. The following will In this section we present experiments where the
give more details about these steps. DMECD is used to one city telecom database[4]. First, the
（1）Attribute reduction experimental setting is described, and then, results of the
experiments are presented. The experiments are conducted
There are 141 exceptional clients attributes, which can be on one city telecom database, which contains 263654 records
divided into three parts that are client data, call behavior, and about telecom consumer.
consume data. Client data contains attributes such as
SUBSCRBID, AREAID, ORDERDATE, REMOVEDATE, We present accuracy values and check time for DMECD
and PRCPLNID(telecom product type). In the part of call in the testing dataset in table 2. D represents DMECD, while
behavior, there are MAXUSERNUM(the max calls of one T represents traditional system.
client in one month), AVGTIME(the average call time of TABLE 2. PERFORMANCE OF DMECD IN DISTINGUISHING EXCEPTIONAL
one client in one month), and MAXCIRCLENUM(the CLIENTS
number of different phone number that one client calls in one exceptional Check time Accuracy
month). ID clients （min） （％）
（％） D T D T
Consume data contains attributes, such as
1． 0.3 1 5 0 0
P_ACC1(client’s account of last month), P_ACC2(client’s
2． 0.6 2 20 67 46
account of this month), and WMAXCOST(client’s max
account in a week). 3． 1.2 8 40 72 53
4． 1.8 12 60 78 62
If 141 exceptional client attributes are all added into the
5． 2.4 16 80 82 65
model, the result of the model will be very complex and
sensitive. Such model will have a high probability to make a 6． 3.0 23 120 86 68
wrong prediction for the new data. So, neural network 7． 3.7 27 160 91 72
algorithm is used to calculate attributes weight. Only those
attributes which weight is bigger than the average can be From Table 1 one can see that DMECD works better
selected to the model. than traditional system. The accuracy of traditional system in
the training dataset is also lower than that of DMECD in the
The neural network algorithm adopted in this study was a
training dataset.
three-layer, feed-forward network with a back-propagation
algorithm. The input layer consisted of 141 neurons that DMECD and traditional system cannot find exceptional
corresponded with the 3 part features of each client; the clients in the training dataset when the proportion of
hidden layer consisted of 7 neurons decided experientially; exceptional clients is lower than 0.3％. But DMECD can
the output layer consisted of one neuron, whose output find exceptional clients in a real dataset, because the
values ranged from 0 to 1, indicating the likelihood of proportion of exceptional clients is higher than 3％in a real
exception. The network was trained by the environment.
Levenberg-Marquardt rule, and the training was complete
when the sum of square of error was less than 0.001. IV. CONCLUSIONS
The output of training sample is the weight order of 141 Our study highlighted some new aspects in detecting
attribute. The more big one attribute is, the more important exceptional clients from common clients, which is very
one attribute to the conclusion about one client is normal or popular in the field of financial and telecom. While we made
not. The output of neural network algorithm cannot give a several new observations that make three data mining
rule about the result. So we need decision tree algorithm to methods useful in the field of detecting exceptional, there are
describe the result. Because more attributes are added to still some issues that require investigation in future research.
decision tree algorithm can make a worse prediction for new
1. We find exceptional clients in the training dataset and
data, we use the neural network algorithm to reduce the
the experiments show this to be a successful approach. But
attributes. The detail will be described in the follow section.
we don’t compute the impact of noise data, which is exist in
（2）Client character generation the real environment. Are there methods to make our
algorithms more robust when the dataset is very noisy?
The output of neural network algorithm is the weight
order of 141 attribute. But how will we reduce the attributes? 2. How can our methods in the field of detecting
The key is that we need decision tree algorithm to describe exceptional clients be applied to other fields? How can our
the result. Because more attributes are added to decision tree three data mining algorithms be replaced by other
algorithm can make a worse prediction for new data, we use algorithms, such as k-nearest-neighbors?
the neural network algorithm to reduce the attributes. The Data mining classification algorithms have been
detail will be described in the follow section. demonstrated spectacular success for learning tasks, and we
expect to see them used significantly more than they are
today.

503
 REFERENCES [3] Shehroz S.Khan, Amir Ahmad. Cluster center initialization algorithm
for K-means clustering[J], Pattern Recognition
[1] Rui Xu, Donald Wunsch II. Survey of Clustering Algorithm[J]. IEEE Letters,2004(25):1293-1302.
Transactions on Neural Networks, 2005,16(3):645-678.
[4] Newman D. UCI Repository of machine learning databases[EB/OL].
[2] Donato Malerba. Mining Model Trees from Spatial Data[A]. Jaime G. http://www.ics.uci.edu/~mlearn/MLRepository.html, 2007.
Carbonel. Knowledge Discovery in Databases: PKDD 2005 [C]. Berlin
Heidelberg, Germany: Springer-Verlag Press, 2005. 169-180.

504
 2009 International Conference on Computer Engineering and Technology

Simulating Fuzzy Manufacturing System: Case Study

A. Azadeh, S. M. Hatefi, H. Kor

Department of Industrial Engineering, College of Engineering,
University of Tehran
Tehran, Iran
aazadeh@ut.ac.ir, s_m_hatefi, hkor@ut.ac.ir

Abstract—This paper presents an approach by considering
fuzzy parameters in simulation model to obtain fuzzy
throughputs in manufacturing systems. The basic subject of
simulation is the probabilistic approach to describing real
world uncertainty. However, in many cases we do not have
information that would be precise enough to build the
corresponding probabilistic model or there are some human
factors that prevent doing so. In such conditions the statistical
and mathematical tools of fuzzy set theory may be successfully
used. This paper applies t-test for validating the results
obtained from conventional simulation and fuzzy simulation in
comparison with the actual data of manufacturing system.
Then, we use fuzzy simulation to improve the performance of
manufacturing system by considering production constraints,
system limitations and desired targets. The design of fuzzy
simulation is discussed for an actual and large multi product
assembly shop.
Keywords- Intelligent Simulation; Performance
Improvement; Fuzzy Parameters; Fuzzy Simulation; Multi
Product Assembly Shop
I. INTRODUCTION
Computer simulations are exquisite tools for modeling
and analyzing the true performance of the evolving
production systems. An intelligent computer simulation
environment would allow the designers of manufacturing
systems to predict and provide the means to control the
relevant disturbances to an acceptable degree of
completeness. It would automatically enable us to foresee
the behavior of such systems in normal and increased
production situations [1-2].Furthermore, it should
intelligently guide us to a smoother and more efficient
performance for production systems.
An integrated computer simulation environment for
optimization of production systems is proposed in [3].
Reference [4] proposes an intelligent simulation
environment based on the idea of machine learning and
improvement mechanism. They describe the potential of
learning in simulation as means for extracting rules to be
used in expert systems. Reference [5] proposes a goal-
oriented simulation environment for optimization of
manufacturing systems. In a goal-oriented environment,
once the system is described and the goals specified, the
simulation system drives itself to goal achievement.
Several studies show the applications of intelligent
simulation in the area of production systems. Intelligent
simulation environments are also proposed for information
systems, just-in-time and aerospace manufacturing systems
in [6-9].
In simulation all natural uncertainties are usually
considered in stochastic sense. Fuzzy set theory provides
new tools to represent uncertainty. The fuzzy approach has
been used to assist the design of manufacturing systems.
Application to plant layout has been proposed in [10-11].
Fuzzy methods may also be used to choose the most
efficient maintenance approach .These methods were used
in [12-13]. Various applications have been issued from
fuzzy concept. In project management, where activity
duration times are assumed to be random fuzzy variables,
project scheduling is considered with mixed uncertainty of
randomness and fuzziness. One application of this approach
was published in [14].
A Fuzzy Inference System is proposed as an alternative
approach to handle effectively the impreciseness and
uncertainty that are normally found in supplier selection
processes. Applications of fuzzy set theory in supply chain
management are demonstrated in [15]. Fuzzy production
planning problems with vagueness parameters were
considered in [16-17].
Therefore it is the objective of this study to describe
fuzzy simulation because of uncertainties and human factors
that exist in manufacturing systems. Using fuzzy approach
is one of the best ways that reflects uncertainties in systems.
The simple and natural way to do this is to replace in our
model the probability densities by the similar fuzzy
intervals. The fuzzy simulation approach is verified and
validated t-test. Then, fuzzy simulation is used to optimize
the system performance considering production constraints,
system limitations and desired targets. The prescribed
framework is discussed for a large multi product assembly
shop with conventional and fuzzy simulation.

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 505

DOI 10.1109/ICCET.2009.216
 II. METHODOLOGY
We will be using fuzzy numbers for parameters in
probability density functions (probability mass functions,
the discrete case) and in this section we show how we obtain
these fuzzy numbers from confidence intervals [22]. Let X
be a random variable with probability density function (or
probability mass function) f ( x , T ) for single parameter T .
It is easy to generalize our method to the case where T is a
vector of parameters. Assume that T is unknown and it
must be estimated from a random sample X 1 ,..., X n .Let
Y u (X 1 ,..., X n ) be a statistic used to estimate T .Given
the values of these random variables X i x i ,1 d i d n , we
obtain a point estimate Tˆ y u (x 1 ,..., x n ) for T . We
would never expect this point estimate to exactly equal T so
we often also compute a (1  E )100% confidence interval
for T . We are using E here since D , usually employed for
confidence interval, is reserved for D  cuts of fuzzy
numbers. In this confidence interval one usually sets E
equal to 0.10, 0.05 or 0.01.
We propose to find the (1  E )100% confidence
interval for all 0.01 d E d 1 . Starting at 0.01 is arbitrary and
you could begin at 0.001 or 0.005 etc. Denote this
confidence interval as ¬ªT1 ( E ),T 2 ( E ) ¼º .
For 0.01 d E d 1 . Add to this confidence interval
ªTˆ,Tˆ º for 0% confidence interval for T . Then we have
¬ ¼
(1  E )100% confidence interval for 0.01 d E d 1 . Now
place these confidence intervals, on the top of the other, to
produce a triangular shaped fuzzy number T whose D  cuts
are the confidence intervals. We
have: T (D ) ª¬T1 (D ), T 2 (D ) º¼ For all 0.01 d D d 1 .In this
paper we obtain crisp number from triangular shaped fuzzy
T (D )  T 2 (D )
number by 1 for each D  cuts .
2 therefore sent to other shops outside the system and are
The fuzzy simulation is the process that produces
fuzzy outputs Y form fuzzy inputs X using a step-by-step
analysis. This process is shown in Figure 1. In fuzzy
simulation, at first we prepare (1  D )100% confidence
interval for all parameters in probability density functions.
Then a fuzzy simulation experiment is started for each
random number of the confidence interval (with
triangular random numbers in visual slam) and the results
{Y} are collected. The output results obtained for the input
value of the D -cut are placed together and the gaps are
filled to form an interval. That intervals are possible D -cut
for the output and so there are used to reconstruct the fuzzy
output.
506
Historical data
Next D Next D
Let 0 d D d 1
Prepare confidence interval
(1  D )100% for all parameters
in probability density functions
Fuzzy inputs
(Parameters)
Crisp outputs
Crisp inputs
Generate random
numbers from
confidence intervals
Discrete Event
simulation
Set of crisp outputs
Fuzzy outputs
Fuzzy D  cuts
Outputs
Figure 1. Modeling and fuzzy simulation
III. ACTUAL PROTOTYPE
To show how an intelligent environment could be
designed and developed, a large manufacturer of electric
motors was considered. The system being studied is
composed of three units: 1) coil operation shop, 2) core
insulation shop and 3) final assembly shop. The semi-
finished products from coil operation and core insulation
shops are required to go under special treatments and are
finally routed back to core insulation and final assembly
shops, respectively (Figure 2.). There are 21 manual, semi-
manual and fully automated workstations. The target was
defined as to determine the strategies required meeting the
increased demand by 30 percent. Historical data, general
engineering knowledge, specifications and layouts were
collected systematically. In addition, an identification form
was developed for each workstation. It contains valuable
information such as process times, buffer size, number of
operators and machines. It also reveals machine speed rates
and description of operations of a workstation. Performance
measures were defined as daily production rates in each the
three shops. Process times were collected by a time study
methodology and analyzed through a series of robust
statistical techniques. Other information such as machine's
 Figure 2. A general overview of the assembly shop
speed rates, buffer sizes, etc. was provided by historical
data and general engineering knowledge of supervisors.
Due to stochastic nature of the processes, the
distribution functions of several activities were evaluated.
Goodness of Fit techniques was applied to evaluate the best
distribution functions for all activities and processes of the
assembly shop. An integrated database was designed to
contain all of the above information. The conventional
simulation model was developed by AWESIM simulation
language . It contains detailed information about the
processes and operations of the assembly shop. To translate
the traditional simulation model into the prescribed
approach, the interacting systems namely: Insulating
Varnish and Machine shops should be modeled into the
simulation environment. To create a flexible and parametric
environment, a series of Visual SLAM variables and
network functions must be modeled into the simulation
program. Furthermore, all variables and constants must be
changed to global variables and mathematical expressions
and constraints should model system’s limitations such as
production capacity, machine tolerances, etc. The simulation
must be capable of generating various alternatives (by
changing resources and tolerances) until the optimum
solution is reported. The database should be capable of
interacting with the simulation environment through a series
of READ and WRITE statements. By designing a sequence
of rule based conditions, the model searches for the
optimum solution in relation to the defined objective. The
learned strategies are stored in the integrated database for
the future use and comparative analysis. This process is
repeated until the strategies required to increase the
throughput by 30 percent are reported. The conventional and
fuzzy simulation model must be verified and validated prior
to translation into intelligent environment. Therefore, it was
verified and validated against the actual assembly shop.
Daily production rate was chosen as the validation
parameter.
First we obtained fuzzy numbers for parameters in all
probability density functions. We run simulation with lower
507
and upper bounds of fuzzy parameters
(for D  cuts 0.01,.05, 0.1, 0.3, 0.5, 0.7, 0.9 ). Maximum
and minimum production rate of system are occurred at
D  cuts 0.01 . The assembly shop was simulated with
conventional and fuzzy approach for one working day and
repeated 100 times. System is idle in 30 percents of days
and it has production in 70 percents of days. In fuzzy
simulation we obtain 70 fuzzy throughputs with
D  cuts 0.01 and then defuzzyfy them. The outputs of 70
ordinary simulations and fuzzy simulations runs were
compared with the production rates of 70 working days.we
calculate absolute of differences between outputs of
ordinary simulation and actual production rates and call
them d1i (i=1,2,…,70) hence calculate absolute of
differences between outputs of fuzzy simulation and actual
production rates and call them d2i (i=1,2,…,70).
Hence null hypothesis H 0 : Pd d Pd is tested against
1 2
alternative hypothesis H 1 : Pd t Pd .Table I. show the
1 2
summary statistics for t-test. The results revealed that at
D 0.01 the null hypothesis is rejected. Also, from the
independent t-test, we conclude that behavior of fuzzy
simulation outputs have significantly greater similarity to
actual system than ordinary simulation outputs and fuzzy
approach improve accuracy of simulation in this case study.
Note that the equality of variances of H 0 : V d21 V d22
by F-test was tested prior to the t-test. Furthermore, the null
hypothesis was rejected at D 0.01 . Table II. shows
summary statistics for production rates for assembly shop,
conventional simulation and fuzzy simulation. Table III.
shows the classification of various alternatives in unit three
of the
assembly shop. The number in the boxes below workstation
number indicates the number of additional operators to a
particular workstation. For example, if one operator is added
to workstation number eleven, 12.96 percent increase in
production throughput would be expected. The alternatives
shown in Table III. are chosen from a set of more than 25
scenarios. In fuzzy simulation approach, each scenario is
separately designed, modeled, run and evaluated by the user.

Test
H 0 : Pd d Pd vs H 1 : Pd t Pd
1 2 1 2
2 2 2 2
H 0 : Vd Vd vs H1 : Vd z Vd
1 2
TABLE II. RESULTS OF SIMULATION AND ACTUAL PRODUCTION RATE
No. of observations
Actual production rate 70 days
Conventional simulation 70 days
Fuzzy simulation 70 days
This approach is a very time-consuming effort and may lead
to human errors. In addition, due to complexity of
manufacturing systems, a near optimum solution may be
introduced. However, by utilization of the intelligent fuzzy
simulation framework (discussed in this study) the user
directly obtains the best solution (alternative number 6)
without confusion in a shorter time. Furthermore, system
limitation is defined as allowance of addition of up to two
operators to workstation number 11 and one operator to
workstations number 12. Alternative number 6 may be
achieved through a series of learned rules and strategies and
flexible and integrated structures of the fuzzy simulation
model of the system being studied. Moreover, the intelligent
fuzzy simulation model considers system’s limitation and
by a rule-base learning mechanism locates the optimum
rather than near optimum solution. The reader should note
that the percent improvement in throughput remains almost
the same after one operator is added to workstations number
11 and 12. However in simulation approach alternative
number 7 or 8 may be selected if further analysis is not
performed. It means alternatives with the same outcome but
higher costs are proposed.
IV. SIGNIFICANCE
This study presents an integrated framework for using
fuzzy approach in simulation of actual system. It has been
shown that fuzzy simulation approach is more applicable in
uncertain conditions than conventional simulation and
behavior of fuzzy simulation outputs is statistically more
similar to actual manufacturing system than ordinary
simulation outputs. This result was retrieved by t-test and F-
test. Consequently, using fuzzy simulation, considering
system’s limitations and all endogenous and exogenous
factors such as management and economics factors, various
strategies were retrieved to improve the performance of
system. The distinct feature of this study is accomplished
by integration of computer simulation, obtaining fuzzy
numbers from confidence intervals, fuzzy simulation and t-
test. In addition, fuzzy simulation may be deployed in the
TABLE I. SUMMARY STATISTIC RESULTS
Statistics Df P-value
t-statistic = 2.4849 115.63 0.0072
F-statistic = 2.5704 (df1,df2)=(69,69) 0.0001
1 2
Mean Std. Deviation 95% Confidence Interval
72.81 31.99 [65.18, 80.44]
64.04 30.78 [56.70, 71.38]
74.69 14.18 [71.31, 78.07]
actual system because it reflects uncertainties better than
conventional simulation.
V. CONCLUSION
This paper represented an approach by considering fuzzy
parameters (fuzzy inputs) in simulation model to obtain
fuzzy throughputs in manufacturing systems. We used fuzzy
numbers from confidence intervals to obtain fuzzy
parameters in probability density function. Finding fuzzy
outputs is an iterative process that is guided in modeling and
fuzzy simulation. To illustrate the prescribed framework, the
actual system of fuzzy modeling manufacturing are
considered in comparison with the conventional simulation
results. T-test is verified the validity of conventional and
fuzzy simulation in comparison with the actual data of
manufacturing system. Conclusions showed that behavior of
fuzzy simulation outputs is statistically more similar to
actual manufacturing system than ordinary simulation
outputs.
Various alternatives and strategies retrieved to
improve the performance of system considering
management and economics factors [8] .To sum up, we
would like to point out the following:
x Fuzzy Simulation is a powerful tool to help
production managers in planning, controlling and
improving the performance of manufacturing
systems.
x In practice, it is often impossible or unnecessary to
use the probability distributions for mathematical
formalization of uncertainties available. Therefore, it
seems natural to use the fuzzy interval numbers
instead of precise probability distributions in the
cases of the real-world system modeling.
x In real-world fuzzy simulation reflects uncertainties
better than conventional simulation.
508
 TABLE III. SELECTED ALTERNATIVE BASED ON FUZZY SIMULATION RESULTS

Alternative Workstation Number Fuzzy production rate Percent improvement

Number 11 12 13 15 D cuts 0.01 throughput
1 - - - - [54,59.99,64] -
2 1 - - - [60,67.77,72] 12.96
3 1 1 - - [64,71.89,78] 19.83
4 1 1 1 - [63,72.1,78] 20.18
5 1 1 1 1 [64,72.12,78] 20.22
6 2 1 - - [71,82.74,89] 37.92
7 2 1 1 - [71,82.89,88] 38.17
8 2 1 1 1 [74,82.9,88] 38.19
9 1 2 1 1 [66,74.38,80] 23.98
10 1 1 2 1 [65,72.25,77] 20.43
11 1 1 1 2 [64,72.14,78] 20.25

Computer-Integrated Manufacturing, vol. 23, no.1, pp. 126-

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methods for plant layout design problem,” Robotics and

509
 2009 International Conference on Computer Engineering and Technology
Research of INS Simulation Technique based on
UnderWater Vehicle Motion Model
Cheng Jian-hua, Li Yu-shen, Shi Jun-yu
School of Automation
Harbin Engineering University
Harbin, Heilongjiang, China
ins_cheng@163.com
Abstract— Considering the dynamic error of Platform inertial
navigation system （ INS ） caused by vehicle mobile
movement, output information of underwater vehicle and
input information of INS are deeply analyzed. This paper gives
the transform relationship among geographical reference
frame, fixed and reference frame which are used to describe
movement of underwater vehicle. Then, logical join between
vehicle movement output information and INS’s input
calculation information are established so that INS can get the
navigation system close to actual underwater vehicle motion,
thus vehicle movement was reflected actual. Computer
simulation shows that the join between vehicle and INS is
correct. This is a good method for the movement adaptability
examination of INS.
Keywords- INS; underwater vehicle motion; space motion
equation; coordinate transformation
I. INTRODUCTION
INS has become the core system of marine navigation
system because of its autonomy and ability that can provide
high-precision position, velocity and attitude information for
a long time. It can be used not only in ensuring the safety of
underwater vehicle, surface ships but also in providing the
information base for shipboard(potential) weapon system
launching[1].
Comparing with the aerospace navigation system, marine
INS need provide high-precision information for several
days or even longer time. Therefore, the designers not only
need develop higher-precision inertial sensors, but also
reduce localization error by system technology. For example,
Schuler, Foucault and Earth oscillation of INS can be
damped by using horizontal and azimuth damp network,
comprehensive calibration can increase the ability of
navigation system greatly because it can realize gyro drift
correction, azimuth correction and position amendment[2].
Although systematic measures can improve the
autonomous navigation, it destroyed the Schuler adjusting
condition. In this case, vehicle movement, especially mobile
movement can lead to dynamic error of INS during state
switch and comprehensive calibration. Therefore, the
adaptability study of carrier movement on INS is also an
important technology research content of marine inertial
navigation technology in addition to improving the inertial
sensor accuracy and system technology.
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DOI 10.1109/ICCET.2009.129
INS characteristic testing with the real vehicle has a lot of
disadvantages such as high risk, high cost, long time, and
easily limited by environment and equipment [2]. Thus,
computer simulation became the most convenient in testing
INS by assumption of various motion state of underwater
vehicle. However, the traditional simulation method can not
reflect and trace the actual vehicle movement because of the
complexity of real underwater vehicle [3], we need establish
the analog simulation environment of INS based on the
motion state of actual vehicle. This is very important for
improving the movement adaptability of the carrier and
solving some overshoot such as switching state,
comprehensive calibration and other practical issues of INS.
II. DESCRIPTION OF UNDERWATER VEHICLE
MOTION MODEL
The underwater vehicle depends upon tail elevator, head
elevator and rudder to keep and change the depth and
direction. We usually regard the underwater vehicle as rigid
body in establishing the vehicle model. By using momentum
theorem of rigid body, the Newton law of motion and
hydromechanics principle, Taylor Series and differential
equation theory, the equation of 6-DOF motion of
underwater vehicle can be inferred.
The integral vehicle motion is very complex. Considering
the obtainment difficulty of the whole hydrodynamic
coefficient and the difficulty of the equation calculation, we
can simply the general space manipulation equation of
underwater vehicle reasonable according to the symmetry
and motion characteristic of the vehicle. If the static and
buoyancy has already been balanced, the hull is approximate
longitudinal and bilateral symmetrically well, the coupling
between the rudders is not considered, the second-order and
hydrodynamic coefficients of coupling terms are neglected
and so on, the space manipulation equation of underwater
vehicle accordance with the simplified conditions can be
obtained as[4][5]:
A(u v w p q r)T = ( f X fY f Z f K f M f N )T (1)
Where, A is the coefficient matrix ( A = ( a ij ) 6×6 )； u , v ,
w is longitudinal velocity, bilateral velocity, vertical
velocity of the underwater vehicle relative to the earth; p ,
q , r is transverse inclination angular velocity, pitch angular
velocity, course rotation angular velocity of the underwater
vehicle.
 According to the coefficient matrix, the solution of the
equation (1) is:
⎧ fX
⎪u = a
⎪ 11
⎪v = (a66a44 − a64a46 ) fY + (a26a64 − a24a66 ) f K + (a46 a24 − a44 a26 ) f N
⎪ a22 (a66 a44 − a64 a46 ) + a24 (a62 a46 − a66 a42 )
⎪ a55 f Z − a35 f M
⎪w =
⎪ a33a55 − a35a53
⎨ (a62a46 − a66a42 ) fY + (a22a66 − a26a62 ) f K + (a42a26 − a22 a46 ) f N
⎪ p =
⎪ a22 (a66 a44 − a64 a46 ) + a24 (a62a46 − a66a42 )
⎪ a33 f M − a53 f Z
⎪q =
⎪ a33a55 − a35a53
⎪  (a64a42 − a62a44 ) fY + (a24a62 − a22a64 ) f K + (a44 a22 − a42 a24 ) f N
⎪r = a22 (a66 a44 − a64a46 ) + a24 (a62a46 − a66a42 )
⎩
(2)
The 1-order ordinary differential equation of the motion
model can be get by combining auxiliary equation with
relationship of the motion[5]. Then, we can obtain the space
motion state parameter of underwater vehicle by using
numerical integration.
Besides obtaining the output motion information
according to the space motion equation and auxiliary
equation of underwater vehicle, another two parts of work
should be complete if we want to establish the connections
between the underwater vehicle motion and the inertial
navigation system input.
z The velocity information and the angular velocity
information of underwater vehicle should be transformed
from ship coordinate system to geographic coordinate system.
The attitude information of underwater vehicle should be
transformed form relative to fixed coordinate system to
relative to geographic coordinate system.
z The output information of underwater vehicle should be
transformed into the INS input information, that is to say, the
output information of the underwater vehicle should be
transformed into the input information of relative to the
inertial space.
III. COORDINATE SYSTEM TRANSFORMATION
OF UNDERWATER VEHICLE MOTION
E − ξηζ and
Fig.1 Fixed coordinate system
moving coordinate system G − xyz
The space motion description of underwater vehicle
widely adopted the system which is recommended by the
international Towing Tank Conference (ITTC) and
bulletined by Society of Naval Architects and Marine
Engineers. (SHAME), including fixed coordinate system
511
E − ξηζ and the moving coordinate system G − xyz (G is
the center of underwater vehicle, Gx points to the bow, Gy
points to the starboard and Gz points to the hull bottom. A
right-hand coordinate system is composed by them.)
The platform inertial navigation system usually adopts
geographic coordinate system (its movement with the
movement of carrier and the earth rotation is the horizontal
and azimuth reference) as the navigation system. Therefore,
we need transform the motion parameters of underwater
vehicle into that in geographic coordinate system by
coordinate transformation. Then, it can provide the actual
input calculation information for the inertial navigation
system.
2.1 The transformations between moving coordinate
system and fixed coordinate system
If the transformation matrix between moving coordinate
system and fixed coordinate system can be described by S ,
we can get[6]:
(ξ η ζ ) T = S ( x G y G z G ) T (3)
Because of any two rectangular coordinate system can
transform each other by three times Euler angular spinning,
then we can obtain S the coordinate transform matrix:
⎡cosψ cos θ cosψ sin θ sin γ − sinψ cos γ cosψ sin θ cos γ − sinψ sin γ ⎤
S = ⎢⎢ sinψ cos θ sinψ sin θ sin γ + cosψ cos γ sinψ sin θ cos γ − cosψ sin γ cos⎥⎥
⎢⎣ − sin θ cosθ sin γ cosθ cos γ ⎥⎦
(4)
Where ψ is the heading, θ is the pitch angle and γ is
the heeling angle.
2.2 The transformations between fixed coordinate
system and geographic coordinate system
According to the definition of each axis of fixed
coordinates system E − ξηζ and geographic coordinate
system G − xyz , the coordinate transformation matrix under
the condition of origin coincidence based on the following
rotation method can be get by:
Its corresponding state transition matrix is:
⎡ξ ⎤ ⎡ x ⎤ ⎡0 1 0 ⎤ ⎡ x ⎤
⎢η ⎥ = C E ⎢ y ⎥ = ⎢1 0 0 ⎥ ⎢ y ⎥ (5)
⎢ ⎥ o ⎢ ⎥ ⎢ ⎥⎢ ⎥
⎢⎣ζ ⎥⎦ ⎢⎣ z ⎥⎦ ⎢⎣0 0 − 1⎥⎦ ⎢⎣ z ⎥⎦
The definition of fixed coordinates system E − ξηζ
regards the earth as a horizontal plane. It doesn’t move with
the position changing of carrier, but the geographic
coordinate system will change by the earth rotation. It also
moves with the movement of carrier. When the position of
carrier changed, the equation can’t reflect the relationship
between the local geographic coordinate system and the
kinetic coordinate system of definition. Therefore we should
establish the changes of geographic coordinate system with
the movement of the carrier.
 IV. THE TRANSFORM BETWEEN THE MOVEMENT
z' INFORMATION AND THE INS INPUT INFORMATION
y'
The movement calculation information input of platform
y x' inertial navigation system is the acceleration which is along
o' z the geographic coordinate system and relative to inertial
space. The coordinate transformation only transmitted the
o x
movement information into the geographic coordinate
δϕ system. We still need join the underwater vehicle movement
oe output information in geographic coordinate system with the
δλ INS’s input calculation information. The motion model of
carrier plays a good role of bridge which connects motion
model of underwater vehicle with embedding INS[7].
Fig.2 the changes of geographic coordinate system with the movement of
the carrier Ax = Av sin H + Vd ⋅ cos H ⋅ Ah ⎫
⎬ (10)
As Fig.2 shown, because of the different origin of the Ay = Av cos H − Vd ⋅ sin H ⋅ Ah ⎭
geography coordinate system show different trends. The
vector angle between them is δθ . We can obtain each axis Where, Av is linear acceleration of underwater vehicle;
component of the vector angle δθ from Figure 2. The Ah is angular velocity of underwater vehicle; H is course
component is: angle of underwater vehicle; Vd is velocity of underwater
δθ x = −δϕ ⎫ vehicle.
⎪
δθ y = δλ cos ϕ ⎬ (6) According to the conversion relationship among moving
δθ z = δλ sin ϕ ⎪⎭
coordinate system, fixed coordinate system and geographic
coordinate system, we can transform the motion state
Similar to the definition of attitude angle, we can obtain information into that in the geographic coordinate system.
the rotation relationship of the geography coordinate system The transformation information of the rotational movement
with the movement of the carrier. and angle movement can be described as:
⎡ x⎤ ⎡ x' ⎤ ⎡ 1 δλ sin ϕ − δλ cos ϕ ⎤ ⎡ x'⎤ ⎡ Avx ⎤ ⎡ u ⎤
⎢ y ⎥ = C o ⎢ y '⎥ = ⎢− δλ sin ϕ
⎢ ⎥ o' ⎢ ⎥ ⎢ 1 − δϕ ⎥⎥ ⎢⎢ y '⎥⎥ (7) ⎢A ⎥ ⎢ v ⎥
⎢ vy ⎥ ⎢ ⎥
⎢⎣ z ⎥⎦ ⎢⎣ z ' ⎥⎦ ⎢⎣ δλ cos ϕ δϕ 1 ⎥⎦ ⎢⎣ z ' ⎥⎦ ⎢ Avz ⎥ ⎡G3T×3 0 ⎤ ⎢ w ⎥ (11)
According to formula (5) and formula (7), we can obtain ⎢ ⎥=⎢ T ⎥⎢ ⎥
the rotation transformation (denoted by C) considering the ⎢ Ahx ⎥ ⎣ 0 G3×3 ⎦ ⎢ p ⎥
movement of carrier in fixed coordinate system and ⎢ Ahy ⎥ ⎢q⎥
geography coordinate system. ⎢ ⎥ ⎢ ⎥
⎣⎢ Ahz ⎦⎥ ⎣⎢ r ⎦⎥
⎡ − δλ sin ϕ 1 − δϕ ⎤ Where Avx , Avy , Avz is the axial linear acceleration in
C = Co ⋅ Co ' = ⎢
E o ⎢ 1 δλ sin ϕ − δλ cos ϕ ⎥⎥ the geographic coordinate system of underwater vehicle;
⎢⎣− δλ cosϕ − δϕ − 1 ⎥⎦ Ahx 、 Ahy 、 Ahz is axial angular velocity in the geographic
2.3 The transformation between moving coordinate coordinate system of underwater vehicle.
The transformation of attitude information can be
system geographic coordinate system
described as:
According to the former relationship we have derived ⎡ϕ ' ⎤ ⎡ϕ ⎤
before between moving coordinate system and fixed ⎢θ ' ⎥ = C T ⎢ θ ⎥
coordinate system, moving coordinate system and (12)
⎢ ⎥ ⎢ ⎥
geographic coordinate system. Thus the rotation ⎢⎣ψ '⎥⎦ ⎢⎣ψ ⎥⎦
transformation (denoted by G) of the carrier between moving
coordinate system G − xyz and geographic coordinate We can see that all the inertial navigation system (INS)
input information can be obtained by formula (11) and
system o − xyz can be expressed by:
formula (12) by establishing the connection between motion
⎡ xG ⎤ ⎡ξ ⎤ ⎡ x⎤ information of underwater vehicle and inertial navigation
⎢ y ⎥ = S −1 ⎢η ⎥ = S −1C ⎢ y ⎥ (8) system input information. Dynamic simulation of INS can be
⎢ G⎥ ⎢ ⎥ ⎢ ⎥ realized on the condition of the establishment of relationship
⎣⎢ zG ⎦⎥ ⎣⎢ζ ⎦⎥ ⎣⎢ z ⎦⎥ between vehicle movement information and INS input. Then
According to the orthogonality of rectangular coordinates we can get the navigation input information of underwater
transformation, we can obtain: vehicle such as position, velocity and so on, considering
control rubber angle information, influenced by wind, water
G = ST ⋅C (9) and flow. Then we need study on the movement adaptability
of carrier for INS.

512
 V. COMPUTER SIMULATION
By using the computer simulation, we can not only study
on the movement adaptability of carrier for INS, but also
infer the connection between motion information of
underwater vehicle and the model of INS and verify the
correctness of the simulation program. Validated by
simulation we can set reservation input to formula (10) as the
reference information of the inertial navigation system
output.
Assign the inertial navigation system (INS) simulation
condition:
Initial Position ϕ 0 =30° ， λ0 =120° ； Platform initial
error angle α 0 = β 0 =0.3'， γ 0 =1'；it doesn’t consider the
gyro drift and accelerometer bias.
5.1 Simulation of suspension static state of underwater
vehicle
Assigning u =0.0kn, INS works on the level damp state.
The result of it is shown by Fig.3 and Fig.4.
Fig.3 East to velocity simulation Fig.4 North to velocity simulation
curves of suspended state curves of suspended state
(Curve 1 is the pure INS calculation result, Curve 2 is the
INS calculation result based on the vehicle motion)
Form the result of velocity simulation on the suspended
state, the result of INS’s normal calculation information and
calculation simulation result after the underwater vehicle
embedded overlap completely. It doesn’t produce error.
5.2 Simulation of uniform motion state of underwater
vehicle
Assigning u =5.8kn, INS works on the internal level
damp state. The result of it is shown by Fig.5 and Fig.6
Fig.5 East velocity simulation Fig.6 North velocity simulation
curves of uniform motion state curves of uniform motion state
Computer Simulation result shows that the results
between normal calculation and calculation after the
513
underwater vehicle embedded have some deviations, but
they reach stable soon. The main cause is because of slight
coupling between vertical plane and horizontal plane when
the underwater vehicle moves, but they regulate equilibrium
soon[5].
5.3 Simulation of uniformly accelerative motion state of
underwater vehicle
Assigning u =5.8kn, by increasing propeller thrust
constantly the underwater vehicle will be uniformly
accelerative motion. To ensure its accelerative is
a =0.005 m / s 2 , and the underwater vehicle will become
uniform motion state. INS works on the internal level damp
state. The result of it is shown by Fig.7 and Fig.8.
Fig.7 East velocity simulation curves Fig.8 North velocity simulation
of uniformly accelerative state curves of uniformly accelerative state
As we seen from the simulation curves, the result of
INS’s normal calculation curves and calculation simulation
curves after the underwater vehicle embedded is basically in
coincidence, but also both of them produce significantly
Schuler and Foucault periodical oscillation. This is because
that the underwater vehicle works on the condition of
uniformly accelerative motion state and the inertial
navigation system (INS) works on the no-damp state.
VI. CONCLUSION
According to the movement adaptability problem we face
in process of design and application of platform inertial
navigation system, we research simulation technology of
inertial navigation system (INS) based on the 6-dof space
motion of underwater vehicle. By correct algorithm such as
coordinate transformation and motion model embedded of
the carrier, it embedded into the platform inertial navigation
system successful. We can control the underwater vehicle
movement by giving the initial condition such as control
information, thrust information and so on. It can produce the
motion information close to the actual vehicle and provide
corresponding input information for INS. This provide not
only effective examination method for checking the
movement adaptability of INS used by underwater vehicle,
but also the powerful reference for the examination of the
other marine navigation system’s performance. Therefore, it
has strong theoretical significance and practical value.
VII. ACKNOWLEDGE
The financial support from National Natural Science
Foundation of China-NSFC under the grant NO. 60674087,
No. 60474046 and postdoctoral fund of Heilongjiang under
the grant NO.LBH-Z06108 are all gratefully acknowledged.
VIII. REFERENCES
[1] Xiong Zhengnan, Cai Kaishi, Wu Fengde. Huang Deming, Cheng
Lu, Inertial navigation system, National Defense Industrial Press,
1982.
[2] Liu Weiren, Zhuang Liangjie. Adaptive hybrid intelligent control of
an INS level damp network[J]. Journal of Harbin Institute of
Technology. 2005（37）:1586-1588.
[3] Shan Danping. The Design and Realization of Platform Inertial
Navigation System Simulator Based on COM. Master degree
dissertation of Harbin Engineering University. 2007.P33-P45.
[4] Yi Guangqu Theory on Inertial Navigation. Aeronautics Industry
Press, 1987.
[5] Pan-Mook Lee, Bong-Huan Jun. Simulation of an Inertial Acoustic
Navigation System With Range Aiding for an Autonomous Underwater
Vehicle[J]. IEEE JOURNAL OF OCEANIC ENGINEERING. 2007,32（2）
:327-340.
[6] Qi Xiaoqiang. Motion Model and Control Methods of UnderWater
Vehicle Hovering System[D]. Master degree dissertation of Harbin
Engineering University.2008, P15-29.
[7] Wang Wenwu. The design of Submarine Voyage Training Simulator
Based on Virtual Reality[J]. Journal of System Simulation. .2001（
13）:599-602.
[8] Goshen-Meskin D, Bar-Ithack I Y, Unified Approach to Inertial
Navigation System Error Modeling. Journal of Guidance , Control
and Dynamics, 1992, 15(3).
[9] Yuan Xin, Yu Jixiang, Chen Zhe. Navigation Systems. Aeronautics
Industry Press, 1993.

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 2009 International Conference on Computer Engineering and Technology

Modeling and simulation of Wireless Sensor Network (WSN) with SpecC and
SystemC

M. Rafiee, M. B. Ghaznavi-Ghoushchi, S. Kheirh, B. Seyfe

Electrical engineering dept.
Shahed university
Tehran, IRAN
{rafiee, Ghaznavi, kheiri, seyfe}@shahed.ac.ir

Abstract— As the complexity of systems is rapidly growing, the
designers are moving towards use System Level Design
Languages (SLDL) such as SystemC and SpecC. In this paper
we use SystemC and SpecC languages for modeling and
simulation of Wireless Sensor Networks (WSN). First we start
on simple ECO model and add modular sections as increases
as the complexity. In the next step, we add data encryption
using DES algorithm for security reasons. The future steps
include modeling the idel/active/work phases for nodes and we
add Attenuation-Transmission-power checking model for
estimation power dissipation among nodes. We also include
IEEE 802.15.4 protocol module and buffer insertion module.
We have simulated the proposed framework with simulation of
3, 7 and 11 nodes in 3000 ms. The fidelity of simulations with
SpecC and SytemC meet with 9.8% error and RMSD 4%. Our
preliminary results from deployment of the functional-proven
SystemC models into SystemC synthesizers, urges that it may
be directly used in early stages of WSN nodes synthesis.
Keywords-component; Wireless Sensor Network, System
modeling, SystemC and SpecC languasges
I. INTRODUCTION
Networked sensor systems are seen by observers as an
important technology that will experience major deployment
in next few years for a plethora of applications. Typical
applications include, but are not limited to, data collection,
monitoring, surveillance and medical telemetry [1]–[2].
Wireless Sensor Network consists of large number of sensor
nodes. These tiny nodes consist of sensing, data processing,
and communicating components. Sensor Network (SN) aims
to provide an efficient and effective connection between the
physical and computational worlds. In addition to
algorithms, hardware and software architecture will
significantly impact the effectiveness of SN. Furthermore,
SN design methodologies will have an important impact on
the cost and performance of SN. SN modeling and
simulation is too important. In SN, the environment should
be monitored entirely to obtain acceptable results. Different
approaches have used for simulation and modeling of SN
and WSN [4]. Traditional approaches consist of various
simulation tools based on different languages such as C,
C++, Java and partly matlab [4]. In the traditional techniques
the main focus is based on analyzing the performance of
wired and wireless sensor networks with analytical methods,
computer simulation, and physical measurement. However
many constraints imposed on sensor networks, such as
energy limitation, decentralized collaboration, and fault
tolerance necessitate the use of complex algorithms for
sensor networks. It appears that simulation is currently the
primary feasible approach to the quantitative analysis of
sensor networks [4]. On the other hand there is a gap
between simulation and synthesis in the traditional
approaches. Therefore the tasks done in the system
specification, performance evaluation are not used directly
during the synthesis. A New approach using SpecC is
reported in [5]. Some new tries started to use SystemC-AMS
[15]. The two main reasons for focus on SystemC are: One is
the existence of very suitable facilities for simulation of
concurrent and parallel phenomena's-where is strongly
observed in massive wireless sensor networks-in SystemC,
and the second is the near-to-hardware nature in SystemC
specifications which could lead to rapid prototyping and less
time to market in designs.
In this paper we explain available WSN simulation
techniques in section I. Traditional Approach on WSN
Simulation System modeling is explained in section II and
system is explained in section III. Modeling of ECO system
with SystemC and SpecC languages are covered in sections
IV and V. Modeling wireless sensor network and result are
described in sections VI, VII and VIII. Finally we conclude
the paper in section IX.
II. TRADITIONAL APPROACH ON WSN SIMULATION
There are different kinds of simulation tools specialized
for WSN. GloMoSim [14] is a scalable simulation
environment for wireless and wired network systems. It is
being designed using the parallel discrete-event simulation
capability provided by Parsec 1 [14]. Parsec is a C-based
simulation language, developed by the Parallel Computing
Laboratory at UCLA, for sequential and parallel execution of
discrete-event simulation models. It can also be used as a
parallel programming language. For developing protocols in
GloMoSim, one should have some familiarity with Parsec,
1
Parallel Simulation Environment for Complex systems

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DOI 10.1109/ICCET.2009.175
but not as an expert. Most protocol developers write pure C
code with some Parsec functions for time management.
OMNeT++ [16] is a discrete event simulation
environment. Its primary application area is the simulation of
communication networks, but because of its generic and
flexible architecture, is successfully used in other areas like
the simulation of complex IT systems, queuing networks or
hardware architectures as well. OMNeT++ provides
component architecture for models. Components (modules)
are programmed in C++, and then assembled into larger
components and models using a high-level language such
NED. Reusability of models comes for free. OMNeT++ has
extensive GUI support, and due to its modular architecture,
the simulation kernel (and models) can be embedded easily
into target applications.
III. SYSTEM MODELING LANGUAGES
System modeling in a large extent is a matter of handling
abstract and possibly incomplete information and trying to
evaluate different solutions based on the system model [5].
Ease modeling system depends on the semantics and syntax
of the used language. At present, there is no complete
language available for entire system modeling but there are
some System-Level Design Language’s (SLDL) being used
extensively for following system-level design methodology .
Two major candidates for system modeling are SpecC
and SystemC languages. SystemC is now based on IEEE-
STD 1666-2005 [9]. SpecC and SystemC share many
common features, such as dynamic sensitivity mechanism
for dynamic scheduling of execution sequence. SpecC is
better suited for the architecture exploration as compared to
SystemC in terms of profiling and determination of
execution sequence. The architecture refinement step
involving allocation, partitioning and mapping is easier in
SystemC compared to SpecC. SpecC and SystemC have
similar capabilities in terms of support for transaction
exploration, except for the determination of channel traffic
which is much feasible with SpecC on account of its
profiling capability. Both the SpecC and SystemC are
equally capable for transaction refinement and the
refinement process. SpecC and SystemC have similar
capabilities in terms of support for communication
exploration and equally capable for communication
refinement [6].
The SpecC language was specifically developed for the
specification and design of digital embedded systems,
including hardware and software portions. Built on the top of
the ANSI-C programming language, the SpecC language
supports concepts essential for embedded systems design,
including behavioral and structural hierarchy, concurrency,
communication, synchronization, state transitions, exception
handling, and timing. Since SpecC is a true superset of
ANSI-C, so every C program is also a SpecC program [7].
SystemC is a C++ based modeling platform supporting
design abstractions at the register-transfer, behavioral, and
system levels. The SystemC classes add the necessary
constructs to C++ for modeling systems and hardware at
various levels of abstraction-from the abstract untimed
models to cycle-accurate RTL models. The power of
516
SystemC is that it can be used as a common language by
system designers, software engineers, and hardware
designers. The language is an attempt at standardization of a
C/C++ design methodology, and is supported by the Open
SystemC Initiative (OSCI) [8]. IEEE-STD-1666-2005 is
SystemC Language Reference Manual Standard. This
standard provides a precise and complete definition of the
SystemC class library [9].
Beside other features, SystemC models a system with
logic threads running in parallel. However, its simulator does
not take advantage of the parallelism. It also recently is used
during direct model synthesis from system specification [18-
20]. The above mentioned reasons and features are our
motivation in choosing SystemC and SpecC for modeling
and simulation. Our future experiments show that the
synthesis is also very uniform using this approach.
IV. WSN MODELING
A sensor network is modeled as a set of heterogeneous
entities. Sensor nodes deployed over the area of interest.
They are triggered by a certain set of stimuli that eventually
result in a sensor report that are transmitted to a remote base-
station. Three main types of sensor nodes need to be created
and supported: 1) target nodes that do stimulation of the
sensors, 2) sensor nodes that monitor events and 3) user
nodes that query the sensors and are the final destination of
the target reports. We primarily used the approach proposed
in [5] then we entirely start from scratch and write down a
modular framework for simulation capable to include various
protocols, topology, security and more features.
A. Node
Node performs a simple task: take an analog sample, and
transmit the data over the wireless link to one or more
receivers. Node module functionally consists of three
concurrent modules, which are analogous to the three main
components of the node, namely, Analog-To-Digital (ADC)
Converter, Microcontroller, and Sender.
The ADC converts the analog signal into digital values.
Sender transmits the data through the wireless channel only
after forming a packet as shown in Fig. 1. Microcontroller
controls the whole process by setting an interface between
the two other sections.
In the transmitted packet, preamble marks the beginning
of the package, ADDR is the address of the receiving node,
PAYLOAD is the data and CRC is the cyclic redundancy
check. ADDR is added by the microcontroller and preamble
& CRC are added by the transceiver. All the nodes perform
the same function but at different frequency, data resolutions
and data domains. ADC converts the analog value into 10-bit
digitals. Module C takes the data from the ADC, adds the
address of the receiver node and provides the data payload
for the sender module. Sender is analogous to the
transceiver; it forms the packet by adding CRC and preamble
to the payload and transmits the packet to the Sink.
ADDR PAYLOAD CRC
Pre-amble
Figure 1. Package Format
 Port
Node
Node Transceiver

Stimulus

Monitor
1 FIFO
Sink
FIFO Port FIFO
Node
2
FIFO
ADC C
Port ECO System

Figure 4. Overall ECO System

Figure 2. Node module
B. Sink
Sink consists of two components, Receivers, and CPU.
Receiver receives data from concurrent node module and
sends to CPU Module. CPU first waits for the initial startup
commands from the monitor and then waits receiving data
from Receiver Module. It checks address of the receiver, if
address is right then deletes address from data and starts
sending 20-bit digital value to monitor.
C. Monitor and Stimulus
Monitor sends initial startup commands to the sink. Then,
the monitor starts waiting for the incoming data from the
station and after receiving the data packets it plots the motion
data of node. Stimulus basically provides the test bench with
the sample input data and performs the function of the
sensor.
V. OVERALL SYSTEM
The basic model we used for as primary simulation is
based on ECO [5]. This model is depicted in Fig. 4. The test
bench consists of ECO System, Stimulus and Monitor. The
ECO System consists of two similar Nodes and one Sink
module executing concurrently.
VI. COMPARISON ECO SYSTEM MODELING WITH
SYSTEMC AND SPECC
Network is simulated with SystemC and SpecC
languages. Simulation time for system model is 3000 ms.
Only difference between two codes is that because SystemC
language is using wait command in the Sender and Receiver
modules because SystemC is without handshake channel
command.
Fig. 5, shows comparision input and output received in
the Monitor modules for the input sine wave from two
networks.
The above results show that ECO system is successfully
modeled in SpecC and SystemC languages and the
simulation results were consistent. Any time varying input
signal can be easily recovered and monitored in the Monitor.
The maximum absolute value of differences in SpecC and
SystemC is about 9.8% and its RMSD2 is about 4%.
VII. MODELING WIRELESS SENSOR NETWORK MODELING
WITH SYSTEMC AND SPECC
Sensor node has two tasks in wireless sensor network: 1)
receiving data from sensor module, 2) receiving data from
other sensor node. Therefore for change ECO system to
wireless sensor network, we add Receiver process to Sender
module and produce transceiver module. In this method
received data from sensor transmit via other node(s) to sink.
Fig. 6, shows structure node and sink in wireless sensor
network.
To increase security in wireless sensor network, we add
DES [13] to our design. The algorithm is designed to
encipher and decipher blocks of data. Inputs of this block are
64 bits data and 64 bits key. Deciphering must be
accomplished by using the same key as for enciphering, but
with the schedule of addressing the key bits altered so that
the deciphering process is the reverse of the enciphering
process.
Therefore, output data from Microcontroller enters to
DES Encryption module, before sending to transceiver
module, and finally transmitted to the Sink module. In Sink
module after receiving data by receiver module, data
decoded by decryption and processed by CPU. Fig. 7, shows
structure of node and sink in wireless sensor network

Receiver
FIFO
CPU
FIFO
Receiver
Sink
Figure 5. Comparision input and Output in SpecC and SystemC language

Figure 3. Sink module

2
Root Mean Square Deviation

517
 TABLE I. DELAY (MS) DUE MODELING SENSOR NETWORK WITH
DIFFERENT TOPOLOGIES WITH SYSTEMC LANGUAGE

Node

Receiv er
Language SystemC

FIFO
Node_No
ADC

3 7 11
Topology

CPU
Transciev er
Primary wireless

FIFO
200 _ _
sensor network

Receiv er
Port

FIFO

Wireless sensor

FIFO
network with DES 400 700 1000
µc security
Sink
Wireless sensor
network with DES
Figure 6. Structure of node and sink in WSN security & 500 800 1100
Attenuation-
Before sending data, the signal level in all those nodes is Transmission-power
calculated according to the path loss formula 1/da. The Wireless sensor
attenuation exponent (a) is used in the path loss formula 1/da network with DES
security &
where d is the distance between transmitting and receiving Attenuation-
700 1200 1600
nodes. If the signal level in each node is larger than the Transmission-power
receiver sensitivity, then it can be detected. & IEEE 802.15.4
Transceiver can send data to next node or sink by using
IEEE 802.15.4 standard [12] (CSMA/CA policy) surrogate
use of Preamble and CRC. VIII. RESULT
IEEE 802.15.4 is a standard used for low rate personal These codes are simulated with SystemC and SpecC
area networks (PAN). It offers device level connectivity in languages. Simulation time for system model is 3000 ms. the
applications with limited power and relaxed throughput entire network is simulated with three, seven and eleven
requirements. Devices with IEEE 802.15.4 technology can nodes and one sink. The simulation result of the target
be used in many potential applications, such as home systems in various topologies are presented in this section.
networking, industry/environments monitoring, healthcare The primary wireless sensor network is simulated with
equipments, etc, due to its extremely low power features. three nodes.
The IEEE 802.15.4 offers device-level wireless connectivity Table I shows the Delay due four different topologies
at low cost. The low cost, here means lower manufacturing that using different number of nodes in SystemC language.
cost, lower installation cost and lower maintenance cost. The Table II shows the Delay due four different topologies
MAC protocol of 802.15.4 supports both contention-based that using different number of nodes in SpecC language.
medium access (Carrier Sense Multiple Access Collision
Avoidance (CSMA-CA)) and scheduled-based medium TABLE II. DELAY (MS) DUE MODELING SENSOR NETWORK WITH
access (Time Division Multiple Access (TDMA)) DIFFERENT TOPOLOGIES WITH SPECC LANGUAGE

simultaneously [12]. Language SpecC

In CSMA/CA algorithm, ack is sent to destination node Node_No
and if support ack by destination node, source node starts to 3 7 11
transmit data to destination node. Adding this algorithm to Topology
Primary wireless
wireless sensor network increases delay and lost path. sensor network
200 _ _
Wireless sensor
network with DES 400 600 800
Node security
Decryption
Transciev er

Receiv er

Wireless sensor
network with DES
ADC

security &
Port

500 700 900

Attenuation-
FIFO

CPU

Transmission-
Port

Port
power
Port

Wireless sensor
Port
Decryption
Encryption
Port

Receiv er

network with DES

µc security &
Attenuation- 600 900 1200
Sink Transmission-
power & IEEE
802.15.4
Figure 7. Structure node and sink in WSN

518
 IX. CONCLUSIONS
In this paper we have presented a framework for
modeling and simulation of Wireless Sensor Networks with
SystemC and SpecC language. SystemC and SpecC can be
used for system modeling but they include limitation for
system modeling. Simulation of the system-level scenario
shows that the total elapsed time strongly depends on the size
of the WSN .In the proposed approach the following features
are implemented and tested successfully: Simple ECO, Data
encryption with DES algorithm, Attenuation-Transmission-
power in sending modes of nodes, IEEE 802.15.4 protocol.
There are very wide range of other ideas may be stacked on
the top of this framework including, node unique
Identification, path registration, broadcast (multicast). In
total, accurate modeling of SpecC is better than SystemC.
REFERENCES
[1] I.F. Akyildiz, W. Su, Y. Sankarasubramaniam, and E. Cayirci,
"Wireless sensor networks: a survey",Computer Networks, 38(4), pp.
393–422, March 2002.
[2] D. Estrin, L. Girod, G. Pottie, and M. Srivastava, "Incrementing the
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[3] J. Feng, F. Koushanfar, M. Potkonjak. “Sensor Network
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[5] G. Sachdeva, R.Doemer, and P. Chou, "System Modeling: A Case
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05-12, University of California, June 15, 2005.
[6] L. Cai, S. Verma and D. D. Gajski, "Comparison of SpecC and
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[7] D. D. Gajski, J. Zhu, R. Dömer, A. Gerstlauer, and S. Zhao, “SpecC:
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[8] P. R. Panda, “SystemC a modeling platform supporting multiple
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[9] "IEEE STD 1666 - 2005 IEEE Standard SystemC Language
Reference Manual", IEEE, pp. 1- 423, 2006.
[10] M. S. Obaidat, D. B. Green, "On Modeling Networks of Wireless
Microsensors", Kluwer Academic Publishers, pp. 115 - 153, 2003.
[11] H. Karl, A. Willig, “Protocols and Architectures for Wireless Sensor
Networks”, John Wiley & Sons Ltd, 2005.
[12] L. Cheng, “IEEE 802.15.4 MAC Protocol Study and Improvement”,
A Dissertation Submitted in Partial Fulfillment of the Requirements
for the Degree of Doctor of Philosophy, Georgia State University,
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[13] R. G. Kammer, “DATA ENCRYPTION STANDARD (DES)”,
Federal Information Processing Standards Publication, 1999 October
25.
[14] L. Bajaj, M. Takai, R. Ahuja, K. Tang, R. Bagrodia, M. Gerla,
“GloMoSim: A Scalable Network Simulation Environment”,
University Technical Report 990027, UCLA Computer Science
Department, Los Angels, USA, 1999.
[15] M. Vasilevski, F. Pecheux, H. Aboushady, L. de Lamarre, “Modeling
heterogeneous systems using SystemC-AMS case study: A Wireless
Sensor Network Node”, IEEE International Behavioral Modeling and
Simulation Conference, BMAS'07, San Jose, CA, USA, pp. 11-16
,20-21 Sept. 2007.
[16] S. M. Bruschi, R. H. C. Santana, M. J. Santana, T. S. Aiza, “An
automatic distributed simulation environment”, Proceedings of the
36th conference on Winter simulation, Volume 1, pp. 378 – 385, 5-8
Dec. 2004.
[17] IEEE, IEEE 802.15.4 Standard Specification for Low-Rate Wireless
Personal Area Networks (LR-WPANs), May 2003.
[18] “Agility Compiler manual”, For Agility 1.2, Available:
http://www.celoxica.com.
[19] “SystemCrafter SC User Manual”, Version 3.0.0, Available:
http://www.SystemCrafter.com.
[20] “Getting Started with CoCentric SystemC Studio”, Version 2002.05-
SP1, synopsys, U.S.A, September 2002.
 International Conference
on Computer Engineering and Technology

Session 8
 2009 International Conference on Computer Engineering and Technology

Sub-micron Parameter Scaling for Analog Design

Using Neural Networks
A.A. Bagheri-Soulla M.B. Ghaznavi-Ghoushchi
School of Engineering, Shahed University School of Engineering, Shahed University
Tehran, IRAN Tehran, IRAN
ali.soulla@gmail.com ghaznavi AT shahed.ac.ir

Abstract—In present work we aimed to develop a NN in submicron region). What will happen for gain in
based approach to translate the design parameter from a scaling, while lambda, mobility and currents come with
submicron technology to a long channel one. The complicated relations?
proposed approach is able to find the superseded design
parameters in 1.2ȝm technology using the input
information, which are design parameters of Gain, Phase 2C ox W 2W 6
AV = μn μ p [7] (1)
( λn + λp )
Margin, Unity Gain Bandwidth and Power in TCMS 2
0.18ȝm. The training data are obtained by various L 2 L 6 I 1I 7
simulations in the HSPICE design environment with
TSMC 0.18ȝm process nominal parameters. The neural There is some approach to scale down an analog
network structure is developed and trained in the C++
circuit with circuit optimization techniques but these
based program. To observe the utility of proposed neural
network model it is tested through at least 50 samples.
techniques need to write circuit’s equations and they
Experimental results show validity of our approach in need optimization techniques that are complicated by
less than 0.5db error for gain, less than 1 degree phase them self [8]. There has been a large research effort in
error for phase margin, 0.1db for unity gain bandwidth the field of analog synthesis tools [9-11]. These tools
and 0.05db for power. aid to design circuits interactively by using special
libraries that include circuit topology, design rules,
analog design; sub-micron design; neural networks; macro-models, etc. they want to use a topology in some
parameter scaling. different technology with their library data.
So we want to relate design parameter in an analog
I. INTRODUCTION design like op-amp in two different technologies when
We have scaling theory on how we can scale down we scale down or scale up a design.
a MOS transistor’s parameters. Submicron transistor’s For this purpose we select a two stage op-amp
model has more parameter and should be more illustrated in Figure 1.
accurate. Some of the transistor’s parameter is variable
in higher level model and they affect the design
parameter like gain, phase margin, unity gain
bandwidth and power. So how we can relate design
parameter for example in an Op-amp in two different
technologies when we scale down or scale up a design.
Sub-micron transistor’s model has more parameter
and is more accurate. For example we can achieve an
acceptable result from level 1 simulation model for a
transistor whit 2ȝm channel length but level 1
simulation result is not acceptable at all for a 0.18ȝm
transistor. While level 49 or higher levels (like level 54)
are most suitable for 0.18ȝm and smaller feature sizes.
Level 1 model has about 8 parameters for a simulation,
LAMBDA and some of important parameters are
constant but in higher level e.g. Level 3 we have about
20 parameters and we have some variable parameters Figure 1. Figure 1: Two stage op-amp [7]
like LAMBDA. Here some problems arise for example
in equation (1) the gain of a two stage op-amp written it
in basic parameter as possible. If we accept gain
equation is true (it has not enough accuracy to be used

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DOI 10.1109/ICCET.2009.139
 II. SCALING
There are three types of scaling:
• Full Scaling (Constant Electrical Field): Ideal
model dimensions and voltage scale together by the
same factor S [1].
• Fixed Voltage Scaling: Most common model until
recently only dimensions scale and voltages remain
constant [1].
• General Scaling: voltages and dimensions scale
with different factors [1].
Because of lack in other scaling technique, we used
general scaling in our work. Full scaling can not be
used because voltage can not be scale as same as
technology and fixed voltage is not acceptable because
supply voltage and threshold voltage of each
technology is defined. Table 1 contains scaling relations
[1].
TABLE I. SCALING EQUATIONS [1]
III. THE MULTI LAYER PERCEPTRON NEURAL
NETWORKS
Multilayer Perceptron (MLP) is the most common
neural network model, consisting of successive linear
transformations followed by processing with non-linear
activation functions. MLP represents a generalization of
the single layer perceptron, which is only capable to
construct linear decision boundaries and simple logic
functions. However, by cascading perceptrons in layers
complex decision boundaries and arbitrary Boolean
expressions can be implemented. MLP is also capable
to implement nonlinear transformations for function
approximations. [2-3].
The network consists of a set of sensory units
(source nodes) that constitute the input layer, one or
more hidden layers of computation nodes, and an
output layer. Each layer computes the activation
function of a weighted sum of the layer's inputs. The
input signal propagates through the network in a
forward direction, on a layer-by layer basis. The
learning algorithm for multilayer perceptrons can be
524
expressed using generalized Delta Rule and gradient
descent since they have non-linear activation functions
[4]. In the general form of an MLP network, the xi
inputs are fed into the first layer of xh,l hidden units. The
input units are simply 'fan-out' units: no processing
takes place in these units. The activation of a hidden
unit (neuron j) is a function fj of the weighted inputs
plus a bias, as given in equation (2).
x pj = f j (¦w ji x pi + θ j ) = f j ( y pj ) (2)
i
Where wji is the weight of input i to neuron j, xpj is
input i, that is, output i from the previous layer, for
input pattern p and șj is the threshold value. The output
of the hidden units is distributed over the next layer of
xh,2 hidden units until the last layer of hidden units, of
which the outputs are fed into a layer of xo output units
[5].
IV. DEVELOPMENT OF NEURAL NETWORK MODEL
To find the most appropriate model for different
design parameter values, op-amp has to be simulated in
a large region of transistors channel width and other
variable parameters. The variable parameters for
simulations are W 1 ,...,W 8 ,C C , I bias . In our
simulation we assumed RC = 1/ gm 6 [7]. Adding the
output value Gain, Phase Margin, Unity Gain
Bandwidth and Power of the simulations. A program in
C++ generates net-lists in HSPICE format with random
value for variable parameters and then calls HSPICE
and simulates the circuit with HSPICE. Then the output
value gain, phase margin, unity gain and power from
HSPICE output are extracted. In the next step, program
scales up the same net-list and do simulation for new
net-list. The variable parameter and output value are
captured and saved in this step too.
Random values are generated around a core of a
feasible design with a deviation. For example we have
feasible design in this design parameters
are W 1 =W 2 = 1μ m ,...,W 6 = 12.3μ m .We took
these value as core parameters and with respect to their
value we generate random value, random generation
range for W 6 is from 2ȝm to about 25ȝm but this
range for W 1 is limited to 0.4ȝm to 2.5ȝm.The test data
were obtained randomly in the same range and they
were different from the training data. Over 1000 data
points were obtained from HSPICE simulations. The
simulation of the neural network was performed in the
C++ based program. 50 randomly made test data were
applied for testing. The NN applied structure is shown
in figure 2.
 Figure 2. NN structure used for training process

Generated samples format are as below: gain is in

db, phase margin is in degree, unity gain and power are
in 10log(x) format. Figure 3 shows error of network for
50 of test data. The result shows our approach can
estimate an acceptable result for scaled circuit design
parameters. In table 2 we illustrate a complete net-list
parameter values in 0.18ȝm and in table 3 we shows
design parameter in 0.18ȝm that HSPICE calculate Figure 3. error of NN for test data
them and scaled circuit design parameter that HSPICE
calculate and our model output value. V. CONCLUSION
The test results prove that the proposed neural
network model can decide scaled circuit parameter
TABLE II. NET-LIST VARIABLE VALUES.
accurately. In the applications of analog circuit design
Parameter Value we need a translator that can find the appropriate design
W1=W2 1.44um parameter in long channel from sub-micron one if we
W3=W4 3.26um want to use low level model in our work. These
W5 0.58um parameters must be determined by the designer with his
experience and knowledge. That is a very important
W6 6.34um
step forward for complex analog design process.
W7 6.20um Adding new input parameters to the neural network
W8 0.97um structure and obtaining more training data, the model
L1=…=L8 0.18um can produce more accurate results in a wider range,
Cc 0.015pF which can make the model an important tool for
Rc 39560.66 designers during the analog integrated circuit design
Ibias 0.601uA process.
VI. REFERENCES
TABLE III. DESIGN PARAMETER VALUE [1] Tsividis, Y., Opration and Modeling of The MOS
Transistor, McGraw-Hill, USA, 1999, pp. 290-301.
0.18ȝm 1.2ȝm 1.2ȝm
HSPICE HSPICE NN output [2] Hush, D., R., Horne, B., “Progress in Supervised Neural
Gain 1.89E+01 2.69E+01 2.793E+01 Networks”, IEEE Signal Processing Magazine, January
PM 5.12E+01 8.16E+01 8.162E+01 (1993), 8-39.
Unity GB 6.90E+01 6.19E+01 6.088E+01
Power -5.53E+01 -5.16E+01 -5.07E+01 [3] Geva, S., Sitte, J. A Constructive Method for Multivariate
Function Approximation by Multilayer Perceptrons, IEEE

525
Transactions on Neural Networks, Vol. 3 (4) (1992), 623-
624.
[4] Haykin, S., Neural Networks - A Comprehensive
Foundation, Prentice-Hall, USA, 1999, pp. 205-233.
[5] Avci, M., Babac, M. Y., Yildirim, T., Neural Network
Based MOS Transistor Geometry Decision for TSMC
0.18μ Process Technology, ICCS 2006, Part IV, LNCS 3994, pp.
615 – 622, 2006.
[6] TSMC 0.18ȝmҏҏtechnology parameters. From mosis, url:
http://www.mosis.org (Date:2003), (View:2008)
[7] M. Hershenson, S. Boyd, T. H. Lee, GPCAD: A tool for
CMOS op-amp synthesis, IEEE/ACM International
Conference on Computer Aided Design, San Jose, CA,
November 1998.
526
[8] FUNABA S., KITAGAWA A.,TSUKADA T,
YOKOMIZO G, A, Fast and Accurate Method of
Redesigning Analog Subcircuits for Technology Scaling,
Analog Integrated Circuits and Signal Processing, 25, 299±307, 2000
[9] R. Harjani, R. A. Rutenbar, and L. R. Carley, ``OASYS:
A framework for analog circuit synthesis.'' IEEE Trans.
Comput. Aided Des. Integrated Circuit. Syst. 8(12), pp.
1247-1266, December 1989.
[10] H. Y. Koh, C. S. Sequin, and P. R. Gray, OPASYN: A
compiler for CMOS operational amplifiers, IEEE Trans.
Comput.-Aided Des, Integrated Circuits. Syst. 9(2), pp. 113-
125, February 1990.
[11] G. Gielen, K. Swings, and W. Sansen, An intelligent
design system for analogue integrated circuits, Proc
European Design Automation Conf., pp. 169-173, 1990.
 2009 International Conference on Computer Engineering and Technology
An Improved Genetic Algorithm Based on Fixed Point Theory for Function
Optimization
Jingjun Zhang
Scientific Research Office
Hebei University of Engineering
Handan, Hebei, P.R.China
santt88@163.com
Abstract—This paper introduces triangulation theory into
genetic algorithm and with which, the optimization problem
will be translated into a fixed point problem. An improved
genetic algorithm is proposed by virtue of the concept of
relative coordinates genetic coding, designs corresponding
crossover and mutation operator. Through genetic algorithms
to overcome the triangulation of the shortcomings of human
grade, it can start from any point to find the most advantages.
Gradually fine mesh will be introduced the idea of genetic
algorithms so that the search area gradually decreased,
improving the efficiency of search. Finally, examples
demonstrate the effectiveness of this method. (Abstract)
Keywords- genetic algorithm; triangulation; relative
coordinates; gradually fine mesh; Fixed Point (key words)
I. INTRODUCTION
Genetic algorithm [1] (GA) is a computational model of
simulation of natural biological evolution. GA is based on
the criteria of survival of the fittest and carries on genetic
operations repeatedly to the population containing the
possible solution to generate new evolving population, and
simultaneously it searches the best individual of population
with the overall parallel searching methods to achieve the
optimal solution to meet the requirements. Comparing to
other search methods, GA has some merits, such as simple
and robust. And it can also search the optimal solution or
quasi-optimal solution with the overall parallel methods by
adjusting the direction of the search space continuously.
However, the algorithm also has some shortcomings, such as
slow convergence rate, prematurely and so many iterations.
To overcome these shortcomings, some improvement on
genetic algorithm has been made constantly, for example the
adjustment of population size[2], the optimization of control
parameters[3], an outstanding individual protection strategy
[4] and so on.
Genetic algorithm can converge to global optimal
solution theoretically, but in the actual design, there is no
mature principle and methods of designing the convergence
criteria. Instead, the designing of convergence criteria
basically relies on the experience and understanding of the
extent to the question, and the most common method is
taking the max generation as the convergence criteria or the
iteration will be terminated if average fitness of the continual
m- generation has not been improved obviously. Therefore,
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.249
Yuzhen Dong Ruizhen Gao Yanmin Shang
College of mathematics& physics
Hebei University of Engineering
Handan, Hebei, P.R.China
ddongyuzhen@163.com
it has vital significance to design objective criterion of
termination judgment.
This paper proposed an improved genetic algorithm
based on the fixed point. First of all, carrying on the
triangulation to the solution space and calculating the integer
label of the vertex, and then designing the encoding scheme,
crossover operator and mutation operator with the
conception of the relative coordinate. This algorithm
overcomes the shortcoming of man-made label of the
triangulation for fixed point and converges when all the
individuals are completely labeled simplexes.
II. ABOUT TRIANGULATION
So-called triangulation [5] is that making sure h > 0 and
with three groups of straight lines, such as {x1 = mh} ，
{x2 = mh} and {x2 = x1 + mh} (among them m is an
2
integer), the 2 Dimensional Euclidean space R will be
divided into many triangles and obtains the triangulation
2
of R .
2
Carrying on triangulation to the Euclidean Space R , the
g
goal is to find such a triangle under the self-mapping . The
first coordinate component of one vertex of the triangle
dropped, another vertex's second coordinate component
decreased, and the third vertex's two coordinate components
remained unabated. As a result of continuity of g , if this
triangle's diameter is small enough, the change of three
vertices will not differ too far. In this case, each vertex is an
approximate fixed point.
Supposing that x is a vertex of a triangle, its integer label
is determined according to l (x ) .
⎧ 0 g1 ( x) − x1 ≥ 0, g 2 ( x) − x2 ≥ 0
⎪
l ( x) = ⎨ 1 g1 ( x) − x1 < 0, g 2 ( x) − x2 ≥ 0
⎪ 2 g 2 ( x) − x2 < 0
⎩
The triangle with three different kinds of integer labels is
called a completely labeled simplex. Usually, we use the
man-made labels not relying on self-mapping g and find the
completely labeled simplexes though substitution of vertex
with the same label (as shown in Figure 1).
527
 obtained, they are
B ( a + ( n + 1) h, c + kh) ,C ( a + ( n + 1) h, c + ( k + 1)h) a
nd D ( a + nh, c + ( k + 1) h) . Therefore, we can calculate
the coordinates of other vertices of the square only if the
relative coordinate of A is given. In view of this fact, we take
the relative coordinate of an intersection point which is
decided by two groups of straight lines as genetic encoding.

Figure 1. The triangulation

4
In this paper, the thought of triangulation is introduced to
genetic algorithm and as a result, the improved genetic 3
algorithm avoids the shortcoming of man-made initial point.
Though evolution, narrowing the searching ranges gradually
to enhance the convergence rate and reduce the iteration 2
times
III. THE IMPROVED GENETIC ALGORITHM 1
We frequently meet the questions of solving the optimum
value of max f ( x ) ( a ≤ x ≤ b) without constraint 0 1 2 3 4
condition, and supposing that f (x ) has continuous Figure 2. .relative coordinates

derivative in the given interval. In this case, the question of Crossover operator: Using the overall arithmetic
solving the optimum value of f (x ) can be transferred to
crossover strategy, supposing that p1 = ( x1 , x2 ) and
(1) (1)

solve the zero point of f (x) and then it can be transferred to

p2 = ( x1( 2) , x2( 2) ) are two parent vectors with satisfaction
solve the fixed point of f (x) according
of x1 ≠ x1 and x2 ≠ x2 . s1 = ( z1 , z2 ) ，
(1) ( 2) (1) ( 2)
to g ( x ) = x + f ′( x ) . Obviously in the numerical
calculation, it is difficult to find a vector x satisfying the s2 = ( w1 , w2 ) are the two descendants obtained though
formula of g ( x ) = x .Under the circumstance, the crossover operator.
conception of ε fixed point [5] has been introduced. z1 = [α x1(1) + (1 − α ) x2(1) ⎤⎦
，
that ε > 0 , we say x ∈ R is a ε
n
z 2 = [ β x1( 2 ) + (1 − β ) x 2( 2 ) ⎤⎦
Supposing -fixed point
(approximate fixed point) of the self- ,
mapping g : R n → R n , if x − g (x) < ε . Here w1 = [ (1 − α ) x + α x ⎤⎦
(1)
1
(1)
2
，
x − g (x) expresses the vector mold in Euclidean space.
w2 = [ (1 − β ) x (2)
1 + β x ⎤⎦
(2)
2
Relative coordinates: Supposing that the definition .
domain of f ( x1, x 2) is that a ≤ x1 ≤ b, c ≤ x 2 ≤ d and In the formulas above ， α and β are random real
dividing the domain into many squares with two groups of number with satisfaction of 0 < α < 1 and 0 < β < 1 , [ ]
straight lines of {x1 = mh} and {x2 = mh} ( m is an is the integer operator.
integer).For a random point of intersection named A, we Mutation operator: In order to manifest the influence of
(1) (1) the integer label in genetic manipulation, the mutation
suppose that it’s coordinate is ( x1 , x2 ) , then there are
operation is as follows:
surely existing n, k satisfying the formula Taking the relative coordinate of A as the genetic encode,
then we can determine a square with A as a left bottom
of x
(1)
1 = a + nh , x = c + kh , and then (n, k ) is called
( 2)
1 vertex. Each vertex can search two aspects information; one
the relative coordinate of A. aspect expresses the parallel direction of the square and
Genetic encoding: selecting four adjacent points of a another describes the length of stride. Take vertex A as the
square randomly, according to the subdivision above, if the example, it has two directions, downward and towards left
relative coordinate of the lower left quarter vertex A of the (as shown in Figure 3), the length of stride for in this
square is given (as shown in Figure 2), recording that ( n, k ) , direction is the grid number with different label of A though
then the other three vertices` relative coordinates record the parallel migration. If the vertex happens to be in
B (n + 1, k ) , C (n + 1, k + 1) and D (n, k + 1) , boundary, then the length of stride to boundary outside is ∞ ,
or if searching in the boundary along a direction not to find a
simultaneously the coordinates of B, C and D can be
vertex with different integer label, we also suppose length of

528
stride along this direction is ∞ . If each direction length of
stride is ∞ , approaches to the origin. Chose the minimum
length of stride from 8 directions of four vertexes and obtain Start
a square though parallel moving according to this direction.
The coordinate of the left bottom vertex of the new square,
recording (n′, k ′) , is taking as the chromosome encoding According to precision request, design
after mutation of vertex ( n, k ) . First of all, this mutation h = {h1 , h2 ,", hn } ；
need to search the in formations of the vertices, then it
carries on specific operation according to the results of the
mutation. We call this mutation as neurons mutation that is a
guiding mutation. Carries on square cutting in half region Di , with two

groups of lines { x1 = mhi } ， {x 2 = mhi }

A D

Coding, initialization, p(0) , t<-0

B C
Calculates each individual to correspond the

Square four apexes the integer markings

Figure 3. Relative coordinates and vertex information

For example, as shown in Figure3, giving the square

corresponding with the mutation individual according to the
counter clockwise, the search information is given according
to the below order: first the horizontal direction, then vertical Whether has the entire
direction, under the circumstance, obtaining 8 integers,
supposing that are: 12, 23, 8, 15, 7, 12, 5, 6, choosing the sign square?
minimum one that should be 5. According to above
hypothesis order, 5 correspond with left direction. Suppose N
A’s encoding is ( n, k ) , the mutation encoding Y

becomes (n − 5, k ) . Carrying on search according to the

N
direction of coordinate connection though crossover Whether meets the
operator, this is carried on along the vertical or horizontal precision Y
direction between the selected individual.
The individual which obtains through crossover
Y
operation or mutation operation must be examined that
whether the four vertices integer labels of the square includes End
0, 1, and 2. If including the three labels, we call the square
the completely integer label square, the small region
determined by the square is the extreme point scope. Selection
In order to make sure the operation effectively, the
algorithm uses the thought of gradually thin cuts in half.
Design a series of h = {h1 , h2 ,", hn } , Crossover

satisfies hi % hi +1 = 0 according to the result precise degree.

In which % is the complement of a number operator, that is Mutation
the vertex which obtained in preceding cutting in half is still
in the latter thinner cutting in half. Firstly in the given scope,
determine a smaller scope which includes the optimal Obtains the new generation of
solution, then takes this scope as the new domain of individual p (t + 1)
definition and searches again. Reduce the search scope
gradually until the algorithm meets the requirements
precision.
Improvement genetic algorithm flow chart following
shown in Figure 4: Figure 4. The improved GA flow chart

529
 IV. EXAMPLE CONFIRMATIONS
The test function:
⎧
⎪ min f ( x1 , x2 ) = ( x1 + x2 )
2 2
⎨ ACKNOWLEDGMENT
⎪
⎩ s.t− 2 ≤ x i ≤ 2. (i =1,2)
The function achieves the minimum in (0,0) .
In this example, h = {1,0.1,0.001,0.0001} crossover
probability pc = 0.1 , mutation probability pm = 1 .The
completely label square obtains though the iteration, the
search scope are: (0 ≤ xi ≤ 1) ， (0 ≤ xi ≤ 0.1) ，
(0 ≤ xi ≤ 0.01) and (0 ≤ xi ≤ 0.001) in turn. From this,
the search scope reduces unceasingly with unceasing thinner
cutting in half. From this example the search scope are
contracting to (0,0) gradually, using the limit thought, the
optimum point is (0,0) .
V. CONCLUSION
This paper proposed an improved genetic algorithm
based on the fixed point. Carrying on the triangulation to the
solution space and calculating the integer label of the vertex
530
and then designing genetic operator, Population is used to
find a full-scale triangle as a more objective criteria for
convergence. Simulation gains the good result as expected.
The authors gratefully acknowledge the supports of
National Natural Sciences Foundation of China under the
project number 60875037 and Natural Sciences Foundation
of Hebei Province under No.F2007000710 and
No.E2008000731.
REFERENCES
[1] Holland, J. Adoptation in Natural and Artificial Systems [M].
University of Michigan Press, 1992,44~50
[2] Arabs J, Michalewicz Z,MulaWka J. GAVaPS--a Genetic
Algorithms
st with Varying Population Size.In:Proceedings of the
1 IEEE International Conference on Evolutionary
Computation(ICEC’94), 1994, 73-78
[3] Eiben A. E., Hinterding R., Michalewicz Z. Parameter Control in
Evolutionary Algorithms. IEEE Transaction on Evolutionary
Computation, 1999, 3(2): 124-141
[4] Farzad G,Gecil A. Application of random restart in genetic
algorithms[J].Information Sciences.1996,85(2):81~102
[5] Wang Zeke, pure fixed point algorithm foundation [M]. National
University of Defense Technology publishing house, 1993
 2009 International Conference on Computer Engineering and Technology
P2DHMM:A Novel Web Object Information Extraction Model
Jing Wang1,Zhijing Liu1
1School of Computer Science and Technology, Xidian University, Xi’an 710071, China
E-MAIL: wangjing@mail.xidian.edu.cn , liuprofessor@163.com
Abstract—Due to the difference between web page and plain
text document, the concept of web object is introduced in
this paper. Besides, the supposed state transition and the
emission symbol conditions are improved based on Pseudo
two dimension hidden Markov model (P2D-HMM), and a
novel web objects information extraction method is
proposed. Finally, through an example, it shows that the
proposed method has a very high precision for web objects
information extraction.
Keywords ： Web Object; Pseudo two-dimension Hidden
Markov Model (P2D-HMM); Information Extraction (IE)
I. INTRODUCTION
Along with the rapid and continuous development of
the Internet technology, web pages can provide a huge
amount of information. But how to extract the needed
information from the web is always a necessitous
problem. To resolve the problem, the web information
extraction (IE) is proposed. Now there are three common
models for IE: the model based on pattern dictionary, the
rule-based model and the hidden Markov model (HMM).
At present, the classic information extraction from
plain text document usually makes use of nature language
processing (NLP), and the extraction rules mainly are
based on the rational syntax among words [1]. Although
the web page may contain structured and plain text, it’s
always semi-structured text. And the information on page
is also dynamic and appears in different style, so the
classic information extraction for processing plain texts
cannot be directly applied to the web page. Of course, we
can transform each web page into a plain text document
by removing HTML tags and other irrelevant codes. But
treating web page as plain text is unwise obviously,
because a lot of information such as page structure and
layout will be lost [2].
In this paper, a new information extraction method
based on web objects is proposed and it can effectively
extract web objects from multiple heterogeneous web data
sources. For example, information about “author” objects
can be found in personal homepages, paper header and
even online databases. For the web objects features,
traditional HMM is extended to Pseudo two-
dimensional HMM model (P2D-HMM) for the web
objects information extraction.
The paper is organized as follows. Section 2 briefly
introduces the concepts of web objects, object block and
object elements and so on. Section 3 provides a detailed
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.147
presentation of P2D-HMM, and improves P2D-HMM for
web objects extraction and Section 4 is the experimental
results and analysis. Finally, Section 5 concludes this
paper.
II. WEB OBJECTS
Due to the differences between web page and plain text
document, we define the concept of web objects as the
principle data units about which web information is to be
collected, indexed and ranked [3]. Usually, web objects
are noun concept such as authors, papers, conferences or
journals. Of course, different types of objects represent
the different concepts information, and we suppose the
same style of objects have a common rational
pattern R ( a1 , a2 ,… , am ) , where the attribute
set A = {a1 , a2 ,… , am } is used to describe the objects.
For the web page creators are always displaying
semantically related information together, the information
about an object on web page is usually grouped together
too, and we call the collection of information which
relates to a single object is object block. For an object
block on a web page, we can further segment it into
atomic extraction entities called object elements. So the
object block Ei can be described as a sequence of
elements Ei =< ei1ei 2 eiT > , where eij only belongs to
a single attribute of the object, whereas an attribute can
contain several elements.
Figure 1. Web objects information extraction
As shown in the right of Figure 1, every section can be
consider as an object block, where “Bill Gates” is a web
object with the attributes name, title, organization. After
extracting the objects, the data shown in the left of Figure
1 can be obtained.
Given an object block Ei =< ei1ei 2 eiT > and its
relevant object pattern R(a1 , a2 ,… , am ) , we should
531
assign an attribute name from the attribute
set A = {a1 , a2 ,… , am } to each object element eij , so the
corresponding label sequence Li =< li1li 2 liT > of
block Ei can be gotten. After segmenting a given object
block into an object element set, the label operating can
be consider as a sequence data classification problem.
One of the most popular and effective methods is HMM
for the task [4]. For the web objects have several
attributes, this paper extended traditional HMM to P2D-
HMM, and makes it more appropriate for web objects
extraction. In the following section 3, we will have a
detailed discussion about P2D-HMM.
III. P2D-HMM
In general, Web page can be divided into Navigation,
Theme Text, User Interaction, Copyright, the Websites
Label areas and so on. Although the position of web page
may be different in the term of various web design styles,
but the content of the division is not affected. So we can
establish a Hidden Markov Models of five states for the
HMM.
In this model, five states themselves are not directly
visible, but the each observation vector belongs to a
certain state. Namely, all the observation vectors are
divided into five states, which reflect the changes in
HMM[5]. HMM do use the method to record the
differences of web page in different area
Although a one-dimensional HMM can be used in
information extraction, but a Web page is two-
dimensional, so two-dimensional HMM model should be
better reflected the differences of web page in different
area. In this way, we can enhance the accuracy and
efficiency of the model. So the two-dimensional hidden
Markov model is introduced in this paper.
Observing that visual layout structure of a Web page
reveals the logical relationship among different blocks,
this state transition sequence in which page elements
following a logical relationship, is rather stable and
appropriate .so we propose a VIPs method for the Web
page Segmentation and the result as the Figure2 showing.
(a) (b)
532
(c)
Figure 2. The states of P2D-HMM
As above Figure showing, traditional one-dimensional
HMM is extended to a two-dimensional hidden Markov
model (P2D-HMM), which contains a "super-state" and
each super-state has a HMM known as the "Sub-state".
The structure of Two-dimensional hidden Markov model
is actually a group of HMM will be embedded in another
state of HMM
P2D-HMM is no real sense of the "two-dimensional
hidden Markov model," because it is impossible for a
super-state in one state to transfer to another super-state,
the vertical transfer of the state is limited to super-states.
Figure 2 is typical two-dimensional Hidden Markov
Model for web information extraction. Obviously,Website
Label and Copyright area have less characteristics
relatively, and the other area has more ,especilly the
Theme text area.
Although more of the sub-state, the higher the
accuracy of modeling, but the complexity of computation
will also be increased. Practice has proved that the
number of sub-state is 2,3,5,2,2 in six super-state
respectively, under the precondition of computational
complexity being controled, we can build the model for
the webpage more accurately.
Formally, a P2D-HMM[6] consists of the following
five parts:
qid is the i th sub-state on d th layer, where
d ∈ {0, D} . And it represents super-state when d=0
and sub-state when d=1.
a) N o is the number of super-states in model. These N
states are q 0 = {qi0 } , 1 ≤ i ≤ N o
b) Π q0 = {π 0 ( qi0 +1 )} : the super-state initial
probability distribution, where π 0 ( qi0+1 ) represent the
super-state qi0 probability in time 0 .
c) A: the hidden state transition probability
distribution. Aq0 = {aq0 q0 } , where aq0 q0 is stand for the
i j i j
probability from the super-state q transferred to q 0j
0
i
 d)
i
{ ( )} is the distribution matrix of
Bq0 = bq0 Ot0,t1
i
P (O | Q , λk1 ) = bq(1 ) (o1, s )bq(1q)1 (o2, s )
k
1

T
k
1 2
bq(1 ) q1 (oT , s )
k
T −1 T
(2)
Observation probability, where Ot0 ,t1 is observation = bq1 (o1, s )∏ bq1
(k ) (k )
1 (ot , s )
1 t −1qt
t =2
vector concentrated in the row t1 and column t 0 .
Formula (1) multiplied by formula (2) is:
Embedded in the super-state of the hidden Markov
model includes the following elements: P(O, Q | λk1 ) = π q(1k )bq(1k ) (o1, s )aq( k1q)1 bq(1kq)1 (o2, s )
1 1 1 2 1 2
(k ) (3)
1) The number of sub-states in k th super-state is N 1 T
i∏ (aq( 1 ) q1 q1 bq(t−1) qt (ot , s ))
k k

and the set of sub-states is q1 = {qi1} . t =3

t − 2 t −1 t

For given the model λk , find out the state transition

1
2) The initial probability distribution of sub-states:
Π (k )
q1
= {π (k )
q1i
}; π (k )
q1i
is the probability of sub-state q i
1
sequence Q which maximizes P (O, Q | λk ) is our
1

in super-state k at time 0. objective. In GHMM an observation symbol k is

3)The probability matrix of states transfer: extended from its mere term attribute to a set of attributes
Aq(1k ) = {aq(1kq)1 } ; aq(1kq)1 is the probability of k super- as n1 , n2 , , nZ . We assume the attributes are
i j i j

1 independent from each other and consider a linear

state’s transfer from state qi to state q j1 . Z
combination of these attributes bij (n) = ∑ α ib (n ) .
s =1
s
s
ij s

4) B
(k )
= {bq1 (Ot0,t1 )} is the distribution matrix of
(k )
i Where α s is the weight factor for the ns attribute,
Observation probability, where Ot0 ,t1 is observation Z

vector concentrated in the row t1 and column t 0 .

and ∑α s
= 1, 0 ≤ α s ≤ 1 . So equation (3) now becomes:
s =1

Λ ( k ) = {Π (qk1 ) , Aq(1k ) , B ( k ) } representing the k-th k ⎡ ⎤

Z
If
P(O, Q | λk1 ) = π q(1 )bq(1 ) (o1, s )aq( 1q)1 ⎢ ∑ α s ibqs1q1 (o2, s ) ⎥
k k

super-state, P2D-HMM that can be defined

1 1 1 2
⎣ s =1 1 2
⎦ (4)
as λ = (Π q0 , Aq0 , Λ ) , where Λ = {Λ(1) , Λ( 2) , , Λ( N0 ) } ⎡ ⎤
T Z
i∏ (aq( 1 ) q1 q1 ⎢ ∑ α s ibqs1 q1 (ot , s ) ⎥)
k

The difference between HMM is reflected by the t =3

t −2 t −1 t
⎣ s =1 t −1 t
⎦
corresponding relation between their states and parted In (4), if we set α1 = 1 ， α s = 0(2 ≤ s ≤ Z ) , that is, we
observation vectors. Hence, it is the first thing to get a choose only one observation symbol set, then we can
group of observation vectors for the establishment of reduce (4) to (3). The improvement based on P2DHMM
HMM, which means that Web page segmentation is can be extended to Forward-Backward algorithm, Viterbi
needed. Here we adopt VIPs algorithm. algorithm and Baum-Welch algorithm, so as to resolve
Given sub-model λk in k-th super-state, get the state
1
three basic problems that must be solved for the model to
be useful in real-world applications. These problems are
sequence probability Q = q11q12 qT1 . evaluation, decoding and learning problems.
P(Q | λk1 ) = π q(1k ) aq( 1kq)1 aq(1kq)1 q1 aq( 1k ) q1 1
IV. WEB OBJECT INFORMATION EXTRACTION
1 1 2 1 2 3 T −2 T −1qT

T (1) To extract Web text information, first we need to

= π q(1k ) aq(1kq)1 ∏ aq(1k ) q1 1 process Web page according to their formats, and then
1 1 2 t −2 t −1qt
t =3 information is extracted from semantic understanding
(k ) level of Web text.
Where π is the probability of sub-state qi1 of k-th
q1i Web page information extraction based on P2D-HMM
super-state at the time i. aq(1kq)1 is the probability of k consists of the following two phases:
i j (1) Learn model parameters using statistical method
super-state’s transfer from state qi to state
1
q j1 . Given from training data
Step1:The content block of the Web pages
model λk , in the case of state sequence Q the probability
1
identification;
we propose multiple features (term, layout, and
of observation sequence O is:
formatting) as the emission of P2D-HMM , and separate
the web content block by Vips as observation sequences
of P2D-HMM.

533
 Step2: Web page pre-processing: Segment page into numbers of correct Web object identification
R=
several single object blocks. For every block Ei , segment numbers of all Web object identification
it into object element set Ei =< ei1ei 2 eiT > , and P=
numbers of correct Web object identification
numbers of possible Web object identification
labeling every eij ; the segmentation of the super-state
2× R × P
HMM embedded in the input sequence is adjusted by the F=
R+P
one-dimensional Viterbi algorithm to make
To verify the proposed method of information
P(O | Q, λ( k ) ) output the maximum value. extraction based on web objects, we randomly selected
Step3:The output of P (O | Q, λ ) 200 computer scientists’ homepages from the Internet,
(k )
as the
and choose the “author” of homepage as the object with
observation probability of λ . After completing all the
(0)
seven attributes such as name, affiliation, address, email,
observed sequence calculation, we can implement the phone, degree and other. We randomly selected 150 pages
one-dimensional Viterbi to output maximum value of as training set and the other 50 pages as testing set.
P(O | Q, λ( 0 ) ) .therefore we can identify hidden state In this experiment, three different approaches are
applied for the testing set: (1) the approach based on
behind super-state in P2D-HMM, and the content block conditional random field (CRF); (2) the approach based
E = E1 E2 EN on maximum entropy Markov model (MEMM); (3) the
Step4:Apply ML algorithm to learn P2D-HMM P2D-HMM approach based on web object. The average
parameters. In this step, the extended Baum-Welch extraction precision of each approach is shown in Table 1.
algorithm is used to calculate the initial probability,
transition probability and emission probability; it means TABLE I. THE COMPARED RESULT
to solve the learning problem of P2D-HMM. Approach P R F
(2) web objects information extraction based on the CRF 73.2% 76.7% 74.4%
learned P2D-HMM model MEMM 78.8% 76.5% 78.1%
P2D-HMM 85.2％ 83.2％ 82.6％
Step1:Pre-process the input of web page, and convert it
into a sequence of object blocks. The emission probability Table1 shows that the method proposed in this paper
of one block is the sum of emission probabilities of each has highest precision, because the precision based on
object element. Supposed block object block is higher than that based on single word, and
the labeling process is based on object elements instead of
sequence E = E1 E2 EN , where EK can be divided single word too. The detailed extraction precisions of the
into T-th Web object and Ek =< ek 1ek 2 ekT > . So the testing set are shown in table2.
probability of state j emission k-th block is: TABLE II. TABLE 2 THE DETAILED EXTRACTION PRECISIONS OF
T
bij ( Ek ) = ∑ bij ( Ekt )
THREE APPROACHES

CRF MEMM P2D-HMM

t =1
Name 69.4% 75.4% 82.4%
Step2:Baum-Welch algorithm is uesd for each well
Affiliation 81.9% 85.3% 89.3%
training model HMM to slove λ* = arg max P(O | λk ) , Address 61.8% 70.2% 81.7%
which can be identified the classes of the content blocks Email 81.2% 87.6% 92.5%
Phone 90.3% 93.2% 96.6%
in web page . 66.5% 70.7% 76.3%
Degree
Step3: Input the block sequence E = E1 E2 EN and
Web object Ek =< ek 1ek 2 ekT > ,the Viterbi In the table 2 ,the extraction precisions of Email and
addresses are increased sharply, that is because these two
algorithm is applied to find the state sequence attributes have the notable feature format. Therefore, we
λ * = arg max P(O | λk1 ) of maximal probability, which training less sample for such attributes, it can get good
matches the observation block sequence, and output the identification.
label sequence. VI. CONCLUSIONS
V. EXPERIMENTS AND ANALYSIS In this paper, a concept of web object is introduced for
Here, we take the recall and accuracy ratio as web-specific features, and a novel approach for web page
evaluation standards. When we compare segmentation information extraction is proposed based on P2D-HMM.
results of the different class of web page, we use the F The approach firstly segments page into some object
measure which combines recall (R) and precision (P), and blocks, and then uses P2D-HMM for web objects
the recall, precision and F measure are defined as: information extraction. The experimental results show

534
that the novel approach can significantly improve the web Knowledge and Data Engineering, VOL.18, NO.3. MARCH
2006:334-349.
object extraction accuracy.
[3] Z.Nie, Y.Zhang, J.-R.Wen et al. Object-level ranking: Bringing
order to web objects. In Proceedings of WWW Conference, 2005.
ACKNOWLEDGEMENTS [4] J.Zhu, Z.Nie, J.-R.Wen et al. 2d conditional random fields for
This research project was promoted by the National web information extraction. In Proceedings of ICML Conference,
2005.
Science & Technology Pillar Program No. [5] Jinlin Chen, Ping Zhong, Terry Cook. Detecting Web Content
2007BAH08B02. Function Using Generalized Hidden Markov Model. IEEE
Proceedings of the 5th International Conference on Machine
REFERENCES Learning and Applications. 2006.
[1] Lance A.Ramshaw, Ralph M.Weischedel. Information extraction. [6] Du Shi-ping. The Baum-Welch algorithm of mixture of HMM2
IEEE ICASSP 2005, 2005: 969-972. [J]. Journal of Yunnan University, 2006, 28(2):98-102.
[2] Shiren Ye, Tat-Seng Chua. Learning Object Models from
Semistructured Web Documents. IEEE Transactions on

535
 2009 International Conference on Computer Engineering and Technology
An Efficient Multi-Patterns Parameterized String Matching Algorithm with Super
Alphabet
Rajesh Prasad
Department of Computer Science & Engineering
LDC Institute of Technical Studies
Allahabad, India-212502
rajesh_ucer@yahoo.com
Abstract— In the parameterized string matching, a given
pattern P is said to match with a sub-string t of the text T, if
there exist a bijection from the symbols of P to the symbols of t.
This problem has an important application in software
maintenance where it is required to find equivalency between
two sections of codes. Two sections of codes are said to be
equivalent if one can be transformed into the other by
renaming identifiers and variables only. In this paper, we
extend single pattern exact shift-or string matching algorithm
to find all parameterized occurrences of multiple patterns P0,
P1, P2 …Pr-1, (r≥1), each of equal size m , in the text T. The set
of r multiple patterns is being handled by using the concept of
classes of characters. The new algorithm is named as multi-
pattern parameterized shift-or (MPSO) string matching
algorithm. We further extend MPSO by using the concept of
super alphabets. Implementation results show that by using a
super alphabet of size s, the algorithm (MPSO) is speeded-up
by a factor of s, where s is the size of the super alphabet (i.e. s
is the number of characters processed simultaneously). By
using multi-pattern parameterized string matching, the search
time is lesser than individual pattern searching in the text. We
also show the performance of super alphabet MPSO with
respect to duplicity present in the code. However these
algorithms are applicable only when pattern length (m) is less
than or equal to word length (w) of computer used (i.e. m≤w).
Keywords- Algorithm, finite automata,shift-or, bit-parallelism,
prev-encoding and parameterized matching.
I. INTRODUCTION
In the exact single pattern string matching problem, all
the occurrences of a given pattern P[0…m-1] in the given
text T[0…n-1] are to be reported. Many algorithms for
solving this type of problem exists [1, 2, 3]. In the multi-
pattern [4] string matching problem, all the occurrences of
multiple patterns P0, P1, P2 …Pr-1, (r≥1) in the given text T
are to be reported. In the present discussion, we assume that
all patterns have equal size m. In any bit-parallel algorithm,
multiple patterns can be handled in 3-ways. In the first way,
concept of classes of characters is used. In the second way,
we concatenate the first characters of each patterns, then
second characters of each patterns and so on to form a single
pattern. In the third way, we concatenate the patters as such.
In the present discussion, we consider the classes of
characters for handling the multiple patterns. In the
parameterized string matching [5], we have two disjoint
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.199
Suneeta Agarwal
Department of Computer Science & Engineering
Motilal Nehru National Institute of Technology
Allahabad, India-211004
suneeta@mnnit.ac.in
alphabets, Σ: for fixed symbols, and Π: for parameter
symbols. The symbols of P and T are taken from Σ∪Π. In
this type of matching, while looking for occurrences of P in
T, the symbols of Σ must match exactly whereas the symbol
of Π can be renamed. A given pattern P is said to match with
a sub-string t of the text T, if ∃ a bijection from the symbols
of P to the symbols of t. This problem has important
applications in software maintenance and plagiarism
detection. In software maintenance, it is required to find
equivalency between two sections of codes. Two sections of
codes are said to be equivalent if one can be transformed into
the other via one-to-one correspondence. A number of
algorithms have already been developed for exact string
matching problem, but only a few exist for parameterized
matching. In [6], parameterized on-line matching algorithm
for a single pattern was developed. This algorithm runs in
O(n log min (m, |Π|)) worst-case time. Another algorithm
was given in [7] that achieve the same time bound both in
average and worst case. In [8], shift-or string-matching
algorithm for parameterized string matching problem was
developed.
In this paper, we extend single pattern exact shift-or
string matching algorithm to find all parameterized
occurrences of multiple patterns P0, P1, P3 …Pr-1, (r≥1), each
of equal size m , in the text T. The set of r multiple patterns
is being handled by using the concept of classes of
characters. The new algorithm is named as multi-pattern
parameterized shift-or (MPSO) string matching algorithm.
We further extend MPSO by using the concept of super
alphabets. Implementation results show that by using a super
alphabet of size s, the algorithm (MPSO) is speeded-up by a
factor of s, where s is the size of the super alphabet (i.e. s is
the number of characters processed simultaneously). By
using multi-pattern parameterized string matching, the search
time is lesser than individual pattern searching in the text.
We also show the performance of super alphabet MPSO with
respect to duplicity present in the code. However these
algorithms are applicable only when pattern length (m) is
less than or equal to word length (w) of computer used (i.e.
m≤w). To the best of our knowledge, these generalizations of
shift-or string matching algorithm have not been studied.
II. PARAMETERIZED STRING MATCHING PROBLEM
We assume that pattern is P[0…m-1] and text is
T[0…n-1]. All the symbols of P and T are taken from Σ∪Π,
536
where Σ is fixed symbol alphabet of size σ and Π is
parameter symbol alphabet of size π. The pattern P matches
the text substring T[j…j+m-1] if and only if ∀ i ∈
{0,1,2…m-1}, fj (P[i] = T[j+i]), where fj(.) is a bijective
mapping on Σ∪Π. It must be identity on Σ but need not be
identity on Π. For example, let Σ = {A, B} and Π = {X, Y,
Z, W} and P = XYABX. P matches the text substring
ZWABZ with bijective mapping X → Z and Y → W. This
mapping can be simplified by prev encoding [5]. For any
string S, prev(S) maps its each parameter symbols to a non-
negative integer p, where p is the number of symbols since
the last occurrences of s in S. The first occurrence of any
parameter symbol in prev encoding is encoded as 0 and if
s∈Σ it is mapped to itself (i.e. to s). For example, prev (P) =
00AB4, prev(ZWABZ) = 00AB4. With this scheme of prev
encoding, the problem of parameterized string matching can
be transformed to traditional string matching problem, where
prev(P) is matched against prev(T [j…j+m-1]) for all j = 0, 1,
2...n-m. The prev (P) and prev(T [j…j+m-1]) can be
recursively updated as j increases with the help of lemma
given in [5].
III. SINGLE PATTERN SHIFT-OR STRING MATCHING
PROBLEM
Before presenting standard shift-or string matching
algorithm [1], we define the following terms:
- bm-1bm-2…b1b0 denotes bits of computer word of
length m.
- Exponentiation is used to denote bit repetition (e.g.
041=00001)
- C-like syntax is used for operations on the bits of
computer words: “|” is for bit-wise or, “&” is for
bit-wise and, “~” complements of all the bits. The
shift left operation, “<<r”, moves all bits to the left
by r and enters ‘r’ zeros in the right.
In the shift-or algorithm, an automaton is constructed as
follows: The automaton has states 0, 1, 2…m. The state 0 is
the initial state, state m is the final state and ∀ i = 0…m-1,
there is a transition from state i to state i+1 for character P[i].
In addition, there is a transition for every c ∈ Σ from and to
the initial state, which makes the automaton non-
deterministic. For example, for the pattern P = ba, the shift-
or automaton is shown in fig. 1.
a, b
b a
0 1
2
Figure 1. A shift-or automaton for P = ba.
The preprocessing algorithm builds a table B having one
bit mask entry for each c ∈ Σ. For 0 ≤ i ≤ m-1, the mask B[c]
has ith bit set to 0 iff P[i] = c otherwise it is 1. If the ith bit in
537
B[c] is zero, then in automaton, there is a transition from the
state i to i+1 with character c.
For searching, algorithm needs a bit mask D so that ith bit
of this mask is set to 1, if and only if state i in NFA is active.
For each text symbol c the state vector D is updated by D ←
(D << 1) | B[c], where D << 1, makes the state active and
then looks for transition on that state. If after the ith step
(after processing the ith symbol of the text), the (m-1)th bit of
D is zero, then there is an occurrence of P with shift i-m. If m
≤ w, then the running time of the algorithm is O(n). In
general, running time of the algorithm is O(n⎡m/w⎤).
IV. MULTIPLE PATTERNS HANDLING METHODS
In this section we explain the methods of handling
multiple patterns. There are three ways of handling multiple
patterns in any bit-parallel string matching algorithm. In the
present discussion, we assume that all patters are of equal
size (m).
In the first way, the basic idea is that each text character
is allowed to match to the character of position j in any of
the patterns. For example, if we have patterns “Hello” and
“world” then we form a single super pattern “{H, w}, {e,
o}, {l, r}, {l}, {o, d}”. This matches many text strings, for
example, “wello”, “Horlo” and “Helld”. This is a filter and
potential matches require verification. The algorithm does
not change at all, but only preprocessing changes. After we
have computed Bi table for each character of each patterns
Pi, we take the union of 1(set) bit to give resulting B. Rest of
the algorithm remains the same. This approach is applicable
for large number (r) of patterns.
In the second way, to search a set of patterns P0, P1…Pr-1
of length m in parallel, we concatenate the first characters of
each patterns, then second characters of each patterns and so
on to form a single pattern. The shift in D is not in one bit
but in r bits. i. e. D = D << r and instead of looking for the
highest bit dm-1 of the computer word, we consider all the r
bits corresponding to the highest positions, i.e. we replace
the old 10m-1 mask by 1r0r×(m-1) .This will automatically
search for r words of length m and keep all the bits needed
for each word. This approach is applicable when number of
pattern (r) is not very large and total length of pattern (after
catenation) is ≤w/2. In this approach, we do not get any
false matches.
In the third way, we just concatenate the patterns to form
a single pattern. In this case, the shift is of only one bit and
the bit mask is (10m-1)r. In this way D is replaced by D = (D
<< 1) & (1m-10)r. This way is applicable when number of
pattern(r) is not very large and total length of pattern (after
concatenation) is ≤w/2. In this way, we do not get any false
matches.
V. MULTI PATTERNS PARAMETERIZED SHIFT-OR STRING
MATCHING ALGORITHM
In this section, we discuss multi-patterns parameterized
shift-or string matching algorithm. This algorithm is
generalization of the algorithm explained in section 3. We 2 do P′[k] ← Prev-encode(P[k], m)
can generalize it in the following way: 3 for i ← 0 to π-1
(i) The set of patterns P0, P1, P2…Pr-1, (r ≥ 1) are encoded by 4 do pos[П[i]] ← -∞
prev-encoding and stored as prev(Pi), 0 ≤ i ≤ r-1. To 5 for k ← 0 to r-1
compute prev(Pi), an array prev[c] is formed, which for each 6 do for i ← 0 to σ+m-1
symbol c ∈ Σ, stores the position of its last occurrence in Pi. 7 do B[k][A[i]] ← ~0
For example, let pattern P = XAYBX on fixed alphabet Σ = 8 for k ← 0 to r-1
{A, B} and parameter alphabet Π = {X, Y}. Here prev(P) = 9 do for i ← 0 to m-1
0A0B4. 10 doB[k][P′[k][i]]←B[k][P′[k][i]]&~(1 << i)
(ii) For all j = 0, 1, 2...n-m, prev(T[j…j+m-1]) can be 11 for k ← 0 to r-1
efficiently prev-encoded by lemma 1. 12 do for i ← 1 to m-1
(iii) A table B is built such that all the parameterized pattern 13 do B[k][A[σ+i]]←B[k][A[σ+i]] &
prefixes can be searched in parallel.
(B[A[σ]] | (~0<<i))
To simplify indexing into B array, it is assumed that Σ =
14 for i ← 0 to σ+m-1
{0, 1…σ-1}, and prev encoded parameter offsets are
15 do B'[A[i]] ← ~0
mapped into the range {σ…σ+m-1}. For this purpose, an
16 for k ← 0 to r-1
array A[0…σ+m-1] is formed, in which the positions 0…σ-
17 do for i ← 0 to σ+m-1
1 are occupied by element of Σ and the rest positions are
18 do B'[A[i]] ← B'[A[i]] & B[k][A[i]]
occupied by prev encoded offsets. For example, if we take
19 D ← ~0
pattern P = XAYBX on Σ = {A, B} and Π = {X, Y} with σ 20 mm ← 1 << (m-1)
= 2, m = 5, then prev(P) = 0A0B4. Array A looks like as: 21 for i ← 0 to n-1
22 do c ← T[i]
A B 0 1 2 3 4
23 if c ∈ Π
0 1 2 3 4 5 6 24 then c ← i-pos[T[i]]
25 if c > m-1
Let P′ is prev(P) and T′ is prev(T). Searching for P′ in T′ 26 then c ← 0
can’t be done directly as explained below. Let P = XAXX 27 pos[T[i]] ← i
and T = ZZAZZAZZ then P′ = 0A21 and T′ = 01A21A21. 28 D ← (D << 1) | B'[c]
Clearly, P has two overlapping parameterized occurrences 29 if (D & mm) ≠ mm
in T (one with shift = 1 and other with shift = 4), but P′ does 30 then Verify match at position i-m
not have any occurrences in T′ at all. The problem occurs
because of when searching for all the length m prefixes of Following is the algorithm to evaluate prev-encoding of
the text in parallel, then some non-zero encoded offset p in pattern P while prev-encoding of the text is embedded in the
T′ should be interpreted as zero in some case. For example, search code.
when searching for P′ in T′[1…4] = 1A21, 1(from left) Algorithm 2 (Algorithm to Calculate prev-encoding)
should be zero. The solution to this problem is that, the Prev-encode (P, m)
lemma 1 should be applied in parallel to all m-length sub 1 for i ← 0 to π-1
strings of T′. This is achieved as follows: 2 do pos[Π[i]] ← -∞
The bit vector B[A[σ+i]] is the match vector for A[i], for 0 ≤ 3 for i ← 0 to m-1
i ≤ σ+m-1. If the jth bit of this vector is 0, it means that P′[j] 4 do c ← P[i]
= A[i]. If any of the ith least significant bit of B[A[σ]] is 0, 5 if c ∈ Π
corresponding bit of B[A[σ+i]] is also set to 0. This is 6 then c ← i-pos[P[i]]
achieved as follows: 7 if c > m-1
B[A[σ+i]] ← B[A[σ+i]] & (B[A[σ]] | (~0 << i)) 8 then c ← 0
which signifies that, for σ ≤ i ≤ σ+m-1, A[i] is treated as 9 pos[P[i]] ← i
A[i] for prefixes whose length is greater than A[i] and as 10 prev[P[i]] ← c
zero for shorter prefixes thus satisfies lemma 1. 11 return (prev(P))
We apply way I for handling the multiple patterns. In
this approach, we build a table B for each pattern and take VI. MULTI PATTERNS PARAMETERIZED SUPER ALPHABET
the union of zero bits to form a single table B. The SHIFT-OR ALGORITHM
algorithm 1 explains the concept. We first explain the concept of super alphabet [10] then
Algorithm 1 (Multi-Pattern Parameterized Shift-or) we explain the multi-pattern parameterized shift-or
MPSO (T[0…n-1], P[0…r-1][0…m-1]) algorithm using it. In the super alphabet approach,
1 for k ← 0 to r -1 processing of ′s′ number of symbols of the text T at a time

538
is done (i.e. we use a super alphabet of size σs, packing s 25 if c ∈ Π
symbols of T to form a single super symbol, where σ is the 26 then c ← j-pos[T[j]]
size of fixed alphabet Σ). There are two techniques to form 27 if c > m-1
super alphabets: First technique uses the concept of non- 28 then c ← 0
overlapping characters and second technique uses the 29 pos[T[j]] ← j
concept of overlapping characters. 30 C ← C | ((B′[c] & 1m ) << s-p)
In the present paper, we have taken the first technique of
forming super alphabets. In these two techniques the 31 p ← p+1
original automaton is not modified, but super alphabet is 32 prev[j] ← c
used to simulate finite automaton faster. Algorithm 1 given 33 D ← (D << s) | C
in section 4 is speeded-up by a factor of ′s′ by using the 34 for d ← m-1 to m+s-2
concept of super alphabets. In this algorithm, P is encoded 35 do mm ← 1 << d
in the same way as in section 4 but ′s′ symbols of the text T 36 if (D & mm) ≠ mm
are encoded at a time and its encoding is also embedded into 37 then Verify match at l×s-d-1
the search code. In the parameterized super alphabet shift-or 38 i←i+s
string matching algorithm, prev(T) is transformed into T′,
where for any s, T′ contains s consecutive symbols of VII. EXPERIMENTAL RESULTS
prev(T) and the vector D is modified as: We have implemented the algorithms explained in
D ← (D << s) | B′[T′[i]] sections 5 (MPSO) and sections 6 (MPPSO) in C and
Let C = {c1, c2, c3…cs} is a super alphabet consisting of s compiled with Borland C++ version 3.1 compilers. We
symbol of prev(T), then B′[C] = ((B[c1] & 1m) << s-1) | performed the experiments on the Pentium R D 2.80 GHz
(B[c2] & 1m) << s-2) |…| (B[cs] & 1m). If after the lth step, machine with 512 MB RAM, running Microsoft Windows
any of the m-1 to m+s-2 bits (from right to left) is 0, then XP. A text file of size 264244 characters long is taken as
pattern may occur at l×s-d-1, (where dth bit from right is a 0) input. The symbols of text and patterns are taken from
and hence need to be verified. For verification purpose, we Σ∪Π, which is fixed and equal to the set {a, b, c, d, e, f, g,
have taken the help of lemma 1 given in [5]. Following is h, i, j, k, l}. The algorithms can also be tested for different
the pseudo code of the algorithm. sets of symbols of different sizes. We experimented with
Algorithm 3 (Multi-Pattern Parameterized Super Alphabet parameterized super alphabet shift-or algorithm for s = 1(the
original algorithm), s = 3, s =5, s=7 and s = 8, while keeping
Shift-or)
r = 1. We also experimented with parameterized super
MPPSO (T[0…n-1], P[0…r-1][0…m-1], s) alphabet shift-or algorithm by increasing the size of the
1 for k ← 0 to r -1 parameter symbol alphabet (∏) and therefore decreasing the
2 do P′[k] ← Prev-encode(P[k], m) size of the fixed symbol alphabet (Σ) (because Σ∪Π is
3 for i ← 0 to π-1 fixed), while keeping the text size, pattern length (m), and
4 do pos[П[i]] ← -∞ super alphabet size (s) fixed. We also compare the
5 for k ← 0 to r-1 performance of multi-pattern parameterized shift-or
6 do for i ← 0 to σ+m-1 algorithm with single pattern parameterized shift-or
7 do B[k][A[i]] ← ~0 algorithm. For this comparison, we run multi-pattern
parameterized single times, and single pattern parameterized
8 for k ← 0 to r-1
r times. These algorithms only count the number of matches.
9 do for i ← 0 to m-1
Table 1 gives the timings for the multi-pattern
10 do B[k][P′[k][i]]←B[k][P′[k][i]]&~(1<< i) parameterized super alphabet shift-or (MPPSO) algorithm.
11 for k ← 0 to r-1 Fig. 2 gives the effect on processing time of algorithm
12 do for i ← 1 to m-1 MPPSO on increasing the size of the alphabet ∏ (i.e.
13 do B[k][A[σ+i]]←B[k][A[σ+i]] & increasing the duplicity in the code). Fig.3 compares the
(B[A[σ]] | (~0<<i)) performance of the multi-pattern parameterized shift-or
14 for i ← 0 to σ+m-1 algorithm (MPSO) and single pattern parameterized shift-or
15 do B'[A[i]] ← ~0 (PSO) algorithm, while keeping the pattern length (m) and
16 for k ← 0 to r-1 Σ∪∏ fixed.
17 do for i ← 0 to σ+m-1 TABLE I. PARAMETERIZED SUPER ALPHABET SHIFT-OR EXECUTION
18 do B'[A[i]] ← B'[A[i]] & B[k][A[i]] TIME(S) IN SECOND FOR VARIOUS SUPER ALPHABET (FOR FIXED PATTERN
LENGTH M = 5, NO. OF PATTERN = 1)
19 D ← ~0
20 i ← 0, l ← 0 Super s=1 s=3 s=5 s=7 s=8
alphabet
21 while i < n
Time 12.2091 10.1758 8.02087 7.012452 6.45301
22 do l ← l+1, p ← 1, C ← 0 (second)
23 for j ← i to i+s-1
24 do c ← T[j]

539
 [3] A.V.Aho and M.J. Corasick, “ Efficient String Matching: An aid to
bibliographic search”, Communication of ACM 18(6), pp. 333-340,
18 1975
16 [4] Leena Salmela, Jorma Tarhio and Jari Kytojoki, “ Multi-pattern
14 String Matching with q-grams: To appear in ACM Journal of
Experimental Algorithm”
12
Time (s) [5] B.S. Baker, “Parameterized pattern matching by Boyer-Moore type
10 algorithms”, In proceedings of the 6th ACM-SIAM Annual
8 Symposium on Discrete Algorithms, San Francisco, CA, pp. 541-550,
1995.
6
[6] B.S. Baker, “Parameterized duplication in string: algorithm and
4
MPPSO (s=2) application in software maintenance”, SIAM J. Computing, 26(5),
2 pp. 1343-1362, 1997.
0 [7] A. Amir, M. Farach and S. Muthukrishnan, “Alphabet dependence in
6, 6 5, 7 4, 8 3, 9 2, 10 1,11 parameterized matching”, Information Processing Letter 49(3), pp.
111-115, 1994.
Sigma, Pi [8] Kimmo Fredriksson and Maxim Mozgovoy, “Efficient parametrized
string matching”, In Information Processing Letters (IPL), 100(3),
pp. 91-96, 2006.
Figure 2. Effect on Processing Time of MPPSO on increasing parameter [9] G. Navarro and Mathieu Raffinot, “Fast and Flexible String Matching
symbol alphabet ∏. by Combining Bit-parallelism and Suffix automata”, ACM Journal of
70 Experimental Algorithms , 5(4), 2000.
[10] Kimmo Fredriksson, “ string matching with super alphabets”, In
60 Information Processing Letters (IPL), 87(4), pp. 201-204, 2003.
50
Time
40
30
PSO
20
MPSO
10
0
1 2 3 4 5
No. of Patterns (r)

Figure 3. Comparison of PSO and MPSO Algorithm (m = 5 and Σ∪∏ are

fixed).

VIII. CONCLUSIONS
We have shown that using a simple trick of moving to a
super alphabet decreases the search time of the algorithms.
The parameterized super alphabet shift-or (s = 3, s = 5, s =
7, and s = 8) algorithm is faster than parameterized shift-or
(s = 1).The experimental results show that when we increase
duplicity present in the code (i.e. increasing the size of ∏,
and hence decreasing the size of Σ), the searching time
increases. From fig. 3, we see that MPSO algorithm
performs better than single pattern algorithm. However
these algorithm works only when pattern length (m) less
than or equal to word length of computer used (i.e. m≤w).
Further extension of super alphabet parameterized shift-or
algorithm can be made for larger pattern size (i.e. for m >
w) and for regular expression searching.
REFERENCES
[1] R.A. Baeza-Yates and G.H. Gonnet, “A new approach to text
searching”, Communication of ACM, 35(10), pp. 74-82, 1992.
[2] R.S. Boyer and J. S. Moore, “A fast string-searching algorithm”,
Communication of ACM, 20(10), pp. 762-772, 1977.

540
 2009 International Conference on Computer Engineering and Technology
Research on Modeling Method of Virtual Enterprise in Uncertain Environments
Zhang Jihai
School of Management and Economics
Beijing Institute of Technology
Beijing, China
E-mail: zhangjihai@139.com
Abstract—Virtual Enterprise is a kind of temporary
organization which is made up of many enterprises. The
internal and external environments of virtual enterprises are
changing incessantly. Nowadays, it is short of the normalized
methods of describing this kind of organizations in uncertain
environments. Role theory is used to design role view which
associates with task view, workflow view, organization view,
agent view and interaction view. Finally, gains the modeling
method of virtual enterprise based on extended agent UML.
Keywords- Virtual enterprise; Agent UML; Role Theory
I. INTRODUCTION
World-wide economics makes the market become
international and global. Virtual Enterprise provides an
available approach for the enterprises to respond quickly the
requirements of global market (Roger Nagal and Rick Dove,
1991; Jiang Xingsong,1996). Based on the information
infrastructure, the enterprises can rapidly integrate the
resources, technology, men, organizations and cooperation to
develop and to produce the available products so as to win
the market chance and competitive (Dove,1995) .It makes
the manufacturing integration from in one enterprise into
multi-enterprises. On the other hand, it is difficult to design a
suitable virtual enterprise because of the uncertain
environments.
The uncertain environments of virtual enterprises mainly
involve two main aspects. The first one is the external
environment. For example, a virtual enterprise usually is
established for a kind of products. When the demand of the
product is changing, the virtual enterprise needs adjust its
organizational structure, workflow and members. The other
aspect is about the internal environment. Virtual enterprises
usually are made up of many enterprises. The members can
dynamically join and withdraw the virtual enterprises. In fact
the internal environments of virtual enterprises also
dynamically changes. The changes of internal environments
can also lead to change in organizational structure, workflow
and members. For these reasons, the adaptive ability of
virtual enterprises need be raised.
The adaptive capacity is reflected in the entire life cycle
of virtual enterprises. When virtual enterprise is established,
it needs dynamically adjust target products and decompose
productive tasks. In the operational phase, members can
dynamically join and withdraw the virtual enterprises, and
therefore the workflow and organizational structure of virtual
enterprises need be adjusted. It also needs research the
modeling method of virtual enterprises in uncertain internal
and external environments. This paper provides a new
modeling method based on role theory and Agent UML
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 541
DOI 10.1109/ICCET.2009.219
which improves the adaptive ability of the method. Agent
UML is a kind of agent modeling language which is based
on UML. [1][2][3]There are a number of frequently used
modeling methods, such as AUML and MESSAGE （
Methodology for Agent Oriented Software Engineering）
.This paper amends MESSAGE using role theory, and
provides modeling method of virtual enterprises based on
role theory. In this method task view, organization view,
workflow view, agent view are bound together by role view.
The method has a strong ability to adapt to the uncertain
environments through adding role view and amending task
view and workflow view.
II. TASK VIEW
Task view describes the relation between goals, tasks,
context, and the process of decomposing the main task. After
the leader of the alliance puts forward a new task, it needs
decompose the new task according to the method of
decomposition plan. Its main content is that complex
problems are decomposed to sub-problems. Finally, the
complex problems are decomposed to relative unattached
sub-problems, and all sub-problems constitute the complex
problem. The decomposed process of main tasks is shown on
Fig.1.
Figure 1.Task view
III. WORKFLOW VIEW
After drawing task view the complex problems are
decomposed to sub-tasks. Every sub-task usually represents
an activity. When drawing workflow view, every activity can
be assigned to one role. If the decomposed granularity is
very small, the roles related to the complex problem are very
numerous. The virtual enterprise will involve a lot of
members and be very complex, so it will be difficult to deal
with the so many members. When modeling a virtual
enterprise a method about virtual activities can be used. It
can assemble a virtual activity with basic activities according
to the assembling method, finally gets a workflow view. A
typical modeling process about virtual enterprise usually
includes four steps. Firstly, according to the characteristics of
the target product decomposes the task. Secondly, assembles
basic activities to the virtual activities according assembling
rules. Thirdly, assemble virtual activities to workflow model.
Fourth, defines the member roles according to the virtual
activities. A role usually is bound to one virtual activity.
Fifth, binds the candidates with the member roles. The
binding process is very complex which involves multi-stage
negotiation. After carried out the five steps, a workflow view
about the virtual enterprise can be drawn.
IV. ORGANIZATION VIEW
Organization view is used to describe the structure,
Figure 2.Organization view
competence, functions of virtual enterprises. It opens up
organizational structure of virtual enterprises. This paper V. AGENT VIEW
provides a method which builds organization view according
to role theory. Agent views describe the models of members. They also
The relation of members is described in organization reflect the relation between agent members, the targets,
view. The members of virtual organizations are looked as tasks, roles and resources. Fig.3 shows a typical agent view.
roles in this method. The organization based on role theory is G1
called role organization. The organization views of virtual A2

enterprises are made up of roles which are basic units. The

organizations of virtual enterprises show as the following <<wish>> R1

formula.
PS::={TASK,ROLE,AGENT,ROLERLN,
TASKROLELN,ROLEAGENTLN}
ROLE::={Role1,Role2,…,Rolen}
ROLE indicates the set of roles. ROERLN indicates the
relation of roles. Virtual enterprises usually contain three
kinds of roles. The first one is the leader of the alliance. The
second one is the basic roles. The last one is the control
roles. The organization of virtual enterprises is the set of the
three kinds of roles.
The main function of organization view is to describe the
rights and duties of objects of other views, such as workflow
view, agent view. The characteristics of role view make it
important in the modeling process of virtual enterprises. [5]
(1) Describes interactive mode of members. Every
member need take up its rights and duties which
are defined in roles.
(2) It describes the interactive activities of members.
(3) It’s a kind template that defines the binding
manners between agents and roles. The template
also supports that agents dynamically change its
roles.
(4) Role organization is a kind of organizational
plan which can be instantiated, generalized and
aggregated.
When modeling virtual enterprises, the organization view
needs describe organizational structure, basic units, agents
and the relation between agents and roles.
<<play>>
S1 A1
R2
<<play>>
<<provision>>
accesses
manages
Resource Resource
S2
1 2
Figure 3.Agent view
Agent view focuses on the descriptions of resources. The
resources really include the ability and potential of
members. Agent view covers all the inherent members of
virtual enterprises. Every agent usually includes the
following resources properties: capacity, potential and
effective time. The capacity is on behalf of the resources
ability of the agent which can be used to carry out the
productive task currently. Potential indicates the resources
ability of agent which can be used to carry out the productive
task in future. Potential can transform into actual capacity.
The attribute of every agent should involve names, ability,
potential, roles and tasks.
VI. INTERACTION VIEW
Interaction view (ID) can be described as a two-
tuples.ID:=(SD,WD). SD is on behalf of the communications
of roles in the alliance which can be described with Agent
UML. ID is made up of workflow views and is the set of the
interaction between roles. Interaction view mainly focus on
the control process of interaction. Fig.4 describes the
interactive process between two members.

542
 Initiator
<<participation>>
Collaborator
When building role view, another important concept is
<<pa rticipation>>
member
Travelling
Request 1
member role pattern which is an analytical and design pattern. This
pattern can be reused and combined. Every role pattern
Travel
describes a kind of repeatedly occurred collaboration and
1 Requirement 1 solution in a particular scene. Role pattern is the set of roles,
the relation of roles, and protocols. It is similar with plans
Travel
1..*
Arrangement
1..*
and mainly has four functions. Firstly, it defines the applied
range of role pattern. Secondly, it describes roles and the
relation of roles. Thirdly, the basic situation of agents can be
Figure 4.Interaction view found in role patterns. Finally, it describes the binding
Interactive protocol and the specification of messages relation between roles, tasks and agents.
can be described with interaction view of Agent UML. The A role pattern usually is suitable for special context whose
communications of roles are the basis of other collaboration applied range is very limited, and therefore it need assemble
which is based upon communication mechanism. When many simple role patterns into one complex role pattern.
involving the communications between roles it needs Role contact is imported to assemble simple role patterns. A
consider the specification of messages. typical role contact can be described as following form.
Role Pattern Contract {
VII. ROLE VIEW Name: the name of role contact
Role view is linked to the five other views. The Partner: the list of role patterns
information of other views can intensively display in the role Partner Relation: the relation of role patterns
view. In fact, role view describes the mechanism of virtual Coordination: the rules between role patterns}
enterprises from different aspects. It is an efficient technique to solve complex problems that
can be used to work out a role pattern. The combination of
TABLE I. THE TEMPLATE OF ROLES role patterns is a complex process which contains
Role ID Role Name
organizational integration, demand integration and resource
Role Role Description integration.
Information Role Inheritances
Role Data VIII. THE MODELING PROCESS OF VIRTUAL ENTERPRISE IN
Role Methods UNCERTAIN ENVIRONMENTS
Context Collaboration Collaboration Partner Role
Environment Entry Type
The target for modeling virtual enterprises is to describe
Role Binding Goal Task Goal Name Task Content the process of production and build the information system
Binding Name Reference for virtual enterprises. The modeling method of virtual
Agent Agent Name Binding Content enterprises can be described as following steps.
Binding Kind Reference (1) The productive tasks are decomposed according to
their characteristics, and a task view is drawn.
Role binding reflects the relation between roles, tasks and (2) Get virtual activities according to rules and algorithms
agents. When building a virtual enterprise the binding of combination. In detail, it needs connect virtual
process of roles and tasks is defined by the leader of the
activities with symbolic links. On the other hand, it
alliance according to demands and plans. The binding
needs defines the relation between roles and virtual
relation is very complex between roles and agents. When
assigning a role to an agent, the leader of the alliance needs activities. Every virtual activity usually binds a role.
negotiate with the agent. Role view can be described with (3) Defines roles according to the workflow view. In
Agent UML. Fig.5 displays a typical role view. details every member should be assigned suitable
rights and duties. Finally, gets organization view of the
virtual enterprise.
(4) Defines the model of agents which participate in
productive activities. In details gets the relation
between agents, roles and resources.
The above description indicates that the different views of
a virtual enterprise are related and the role view is the core
of the modeling method.
IX. CONCLUSION
Virtual enterprises are temporary organizations and
established for special products. Nowadays, it lacks the
standardized modeling method of virtual enterprises in
Figure 5.Role view uncertain environments. The method about extended Agent
UML and role theory can describe the virtual enterprises in

543
all directions. The method involves multi-dimensional
views, and role view is the core of the method. Task views [1]
describe the decomposing process of productive tasks.
Organization views describe the structure of virtual
enterprises and the relation between roles and agents.
[2]
Workflow views describe the productive process of target
product and the binding relation between virtual activities
and roles. Agent views describe the structure of members [3]
and the resource of virtual enterprises. Interaction views
describe the communication and interactive methods of
roles. The role views links target products, tasks, activities [4]
and resource. The method has strong adaptability in
uncertain environments. [5]
544
REFERENCES
B. Bauer, J.P. Mueller,and J. Odell,“An Extension of UML by
protocols for multi-agent interaction,”International
Conference on Multi-Agent Systems(C).ICMAS '00, Boston,
Massachusetts. 2002,pp.207-214.
JeanLuc Koning, MarcPhilippe Huget, Jun Wei,and Xu
Wang, “Extended Modeling Languages for Interaction
Protocol Design,” AOSE2001. Montréal, Canada. 2001.
Odell James J, Parunak H Van Dyke, and Bauer Bernhard.
“Representing Agent Interaction Protocols in UML,”
http://www.auml.org/, 2001.
Zhang Jihai, “Modeling and Simulation of National Economy
Mobilization System,”Beijing: Beijing Institute of
Technology Press,2008,pp.147-173.
Qi yan, Li Junshan, Xin Junmao,and Zhi Changqi. “RoMAS:
Role-based Model Method for Multi-Agent System,”National
University of Defense Technology,2002.
 2009 International Conference on Computer Engineering and Technology

Design of Intrusion Detection System Based on a New Pattern Matching Algorithm

ZHANG Hu
College of Information Technology
Anhui University Finance & Economics
Bengbu, China
e-mail: zhanghu2000@163.com

Abstract—Intrusion detection technology can help the system intrusion is found, the intrusion response module will
to deal with network attacks, extend the security management respond timely, give an alarm, cut off the connection and
ability of the system manager and increase the integrality of record the attack log; otherwise, it will make a detailed
information security foundation structure. Pattern matching record of protocol operation log.
algorithm is the core algorithm of intrusion detection system A detailed description of protocol processing module,
based on feature matching as well as an algorithm which is attack feature pattern matching module, log record module
universally used in current intrusion detection equipment. A and intrusion response module is given below.
design scheme of intrusion detection system based on pattern
matching algorithm is proposed in this paper. Meanwhile,
aiming at several key modules of intrusion detection system, a
detailed analysis of data acquisition module, protocol
processing module, feature matching module, log record
module and intrusion response module is also given in this
paper.

Keywords-intrusion detection system; pattern matching

algorithm; data acquisition module

I. INTRODUCTION
Due to limitations of computer network and vulnerability
of information system, hardware resources, communication
resources, software and information resources on computer
network system are damaged, altered and leaked or their
functions are invalid for foreseeable, unpredictable or baleful
reasons, which make the information system become
abnormal and even lead to system collapse and paralysis,
thereby causing enormous economic losses. A great deal of
practice has proved that as a newly emerging security
technology, it is not enough for intrusion detection to ensure Figure 1. Intrusion detection system based on pattern matching algorithm
the security of network system only depending on traditional
passive protection and a complete security strategy should
include real-time detection and fast response. A design of III. DETAILED DESIGN SPECIFICATION OF SEVERAL
intrusion detection system is proposed in this paper. MODULES

II. GENERAL DESIGN OF SYSTEM A. Data acquisition module

Shown as Fig 1, data acquisition module is responsible
for capturing various types of hardware frames from network
flow and handing these hardware frames to data pretreatment
module and then the data pretreatment module strips off
hardware frame heads and checks the integrity of messages,
after that, according to the application protocols messages
belong to, these hardware frames are sent to response
protocol analyzing and processing modules respectively.
Various protocol processing modules make corresponding
treatments according to concrete protocols, for example,
TELNET protocol has a process of packet. After that, feature
pattern matching starts to judge if the intrusion occurs. If the
Data acquisition module can collect data packets in three
ways, namely adopting shared concentrator, switch port
mirror image and monitoring interface box. If data
acquisition module wants to monitor the data which flow
through the network card but don't belong to its own host, it
must round the processing mechanism when the system
works normally and access to the network bottom layer
directly. First, set the working mode of network card as
promiscuous mode to make it receive data packets on target
MAC address not its own MAC address and then access to
the data link layer directly to capture relevant data, the data
should be filtered by application program not upper layers,

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DOI 10.1109/ICCET.2009.244
such as IP layer and TCP layer, etc, in this way, all the data
which flow through the network card can be monitored.
Data acquisition module is composed of three parts,
namely a data packet audiomonitor driver (it acquires
network data packets and transmits the data packets to the
application program which runs on the user layer without
modification), a low class dynamic link library (it runs on the
user layer and it can isolate application program from data
packet audiomonitor driver to make application program run
on different systems without modification) and a high class
static link library (it can be compiled together with
application program, use the services provided by the low
class dynamic link library and provide perfect monitoring
interfaces for application program).
B. Protocol processing module
This module includes all sub-modules aiming at concrete
protocol processing: HTTP protocol processing module, FTP
protocol processing module, TELNET protocol processing
module, POP3 protocol processing module, SMTP protocol
processing module, IMAP protocol processing module,
TFTP protocol processing module and PRC protocol
processing module.
(1) Some protocol packages. TELNET protocol, SMTP
protocol, IMAP4 protocol and POP3 protocol, etc, are
character-oriented protocols, and therefore, a single protocol
message transmitted from data acquisition module and data
preprocessing module needs to be cached by a protocol
processing module and a carriage return character is used to
represent the end of this process according to the command
end mode stipulated in the protocol, organize several
continuous protocol packages into a complete command
which can be detected by attack feature matching pattern, in
the process of package, the corresponding treatment of some
special keyed characters in concrete client protocol packages
can be made, including command character, space character
and backspace character, etc in the protocol itself.
(2) Preliminary judgment of intrusion behaviors. Some
simple intrusion behaviors can be judged in the protocol
processing module. Take FTP protocol processing module as
an example, after receiving FTP protocol messages, first
carry out the heat preservation integrity judgment and then
FTP computing operations and parameter lengths can be
defined by the user, which can be regarded as FTP long-
buffering overflow attack. An alarm signal is generated
directly to be sent to the attack log record module and
intrusion response module, it’s unnecessary to continue to
hand the alarm signal to the attack feature pattern matching
module, which not only enhances the real-time performance
of intrusion detection system but also reduces the resource
burdens of using attack feature matching pattern to detect
intrusion behaviors.
C. Pattern matching module
The concrete working process of pattern matching
module is essentially the matching process of rules and data
packages. After the dynamic organization and analysis of
rules and data package processing by package capture and
analysis module and preprocessor are finished, the detection
546
engine is called to carry out the real-time matching of data
packages and rules.
In the matching process, you can use the pattern
matching algorithm to match from low appearance
probability to high appearance probability of letters in the
pattern string in natural English, when a mismatch occurs,
use Badchar ( ) function which is similar to BM algorithm to
realize the forward jump of host string pointer and carry out
priority matching for low probability characters according to
probability, in this way, you can find as many mismatches as
you can and reduce as many comparison times as possible so
that you can use Badchar ( ) function as early as possible to
realize the forward jump of host string pointer.
Detection function Detect () will use corresponding rule
chain lists to check current packages according to different
protocols. First, it checks if the contents of rule heads RTN
(such as, source/destination address, protocol type and port
number, etc) match, if they don’t match, transfer to the next
node of the rule head chain list to continue to check, if none
of rule heads don’t match, which shows that defined
intrusions don’t occur, the function returns to 0. If RTN
match successfully, enter into rule option OTN chain list of
RTN to match, each option keyword corresponds to a
concrete processing function plug-in. If a match is found, it
shows that an intrusion occurs, and then call the
corresponding option keyword processing function, the
function returns to 1; conversely, if no match is found behind
traversal list OTN, it shows that no defined intrusions occur,
the function returns to 0.
When a feature matches rules with Content keywords
with payloads of data packages, it must call a pattern
matching plug-in, this plug-in will use efficient pattern
matching algorithms to match payloads of data packages to
find potential intrusions.
Finally, an output processing plug-in is called according
to detection results to log data packages according to the
mode set in advance and call concrete processing functions
for intrusion behaviors or give an alarm signal.
D. Log record module
The log record of intrusion detection system based on
attack feature pattern matching can be divided into two
types, namely attack log record and protocol operation log
record. Attack log record provides essential information for
the network manager after intrusion events occur. Protocol
operation log record provides a set of comprehensive,
operating system-independent and searchable log record,
which can be used to audit and trace intrusion behaviors
afterwards.
Figure 2. Call of log record module

Shown as Fig 2, when protocol processing module deals
with corresponding protocol messages respectively, if an
intrusion behavior is found, it will directly calls the attack
log record module to make an attack log record; if no
intrusion behavior is found, it calls the protocol operation log
record module to make a normal protocol operation log
record. Similarly, when attack feature pattern matching
module is matched to a certain attack feature, it will also call
the attack log record module.
1) Attack log record.
Attack log record module is used to record the necessary
information of intrusion behaviors detected by intrusion
detection system and its processing flow chart is shown as
Fig 3.
Figure 3. Flow chart of attack log record module
A complete attack log record should include the
information below:
Occurrence time of intrusion behavior, source address of
intruder, source port used by intruder, destination address of
intrusion, objective of intrusion and name of port attack
feature pattern.
2) Protocol operation log record.
Protocol operation log record module is a set of
comprehensive and open service protocol-oriented operation
547
log record independent of the log record of protected host,
which provides detailed network or host access information
for the system manager and can be used to trace the source of
intruder or the intrusion detection system based on audit, it is
independent of the log record of server itself, and therefore
even though a practiced hacker dodges the detection of
intrusion detection system, intrudes into the network or host
and clears away the logs in the intruded server, the network
manager can still find his whereabouts from the operation
log record of intrusion detection system.
Because of different operation contents of various
protocols, various protocols have different contents of
operation log, generally including the information below:
Protocol service name, occurrence time of operation,
source address of accessor, source port used by accessor,
destination address of access and destination port of access
the rest are concrete operation contents of each application
protocol
For example, the operation content of RPC protocol is as
follows: Remote program number, remote program version,
remote procedure number, mapping program number and
mapping program version.
E. Intrusion response module
Intrusion response module gives corresponding responses
for verified intrusion behaviors, including static measures,
for example, record attack data, store captured data, send
emails and messages to the manager, it can also take some
active dynamic preventive measures, cut off the intruded
connection and modify the access control of router, for
instance. Generally, the corresponding module of system
mainly include three very important functions, namely Alert
(), logtoTable (UNIT SourIP ， UNIT DestIP. CTime
TimeStamp) and TraceInstrusion (UNIT SourIP, UNIT
SourPort， UNIT DestIP, UNIT Destport). The function of
Alert () is to give real-time warning of intrusion behaviors,
the function of logtoTable (UNIT SourIP， UNIT DestIP.
CTime TimeStamp) is to set up a log of high danger class
intrusion behaviors in database and the function of
TraceInstrusion (UNIT SourIP, UNIT SourPort, UNIT
DestIP), UNIT Destport is to trace and take evidence of
intrusion behaviors if it proves necessary.
IV. CONCLUSIONS
Because many attack features are in accordance with the
idiomatic usage of natural English and the appearance
probability of English letters varies very much, according to
the above characteristics, a new pattern matching algorithm
is proposed in this paper, first sequence letters in the pattern
string from low appearance probability to high appearance
probability in natural English, and then match one by one
according to the algorithm, in this way, you can find as many
mismatches as you can, thereby reducing the comparison
times. Finally, on the basis of this algorithm, an intrusion
detection system model based on pattern matching algorithm
is put forward. In addition, a detailed analysis of data
acquisition module, protocol processing module, feature
pattern matching module, log record module and intrusion
response module of intrusion detection system is given in [4] Robert Graham, Protocol Analysis and Command Parsing vs. Pattern
this paper. Matching in Intrusion Detection System, http://www.networkice.com
[5] Neil Desai, Increasing Performance in High Speed NIDS,
REFERENCES http://www.linuxsecurity.com/resource-file/intrusion_
detection/increasing_Performance_in_high_ Speed_ NIDS.pdf
[1] Ulf lindvist, Phillip Brentano and Doug Mansur, “IDS Motivation,
[6] Moore, Strother J. The Boyer_Moore Fast String Searching
architecture, and An Early Prototype”, Computer Security
Algiruthm.Texas university. http://www.uteax.edu/moore/best-
Laboratory, US Davis: 160-171
ideas/string-Searching/index. html
[2] R. Bayer, Symmetric binary B-trees: Data structure and maintenance
[7] Jai Sundar Balasubramaniyan, Jose Omar Garcia-Fernandez, David
algorithms, Acta Information, 1972:34-56
Isacoff, Eugene Spafford, Diego Zamboni, “An architecture for
[3] Breslauer D, Efficient String Algorithmic [Ph.D. Thesis], Computer intrusion detection using autonomous agents”, CERIAS Technical
Science Department. Columbia University, NY: 231-255 Report 98/05, June 11. 1998:34-53

548
 2009 International Conference on Computer Engineering and Technology
To Construct Implicit Link Structure by Using
Frequent Sequence Miner (FS-Miner)
May Thu Aung
University of Computer Studies, Yangon
Myanmar
maythuaung.ucsy@gmail.com
Abstract—Generally, link analysis is based re-ranking web-page
retrieval in current web search engine. The primary reason lies
in the difference of link structures and data scale between global
web and website. In this paper, we propose a framework to
construct implicit link structure by using Frequent Sequence
Miner (FS-Miner) in the User Web Log. And then, the Page
Rank algorithm is applied to re-ranked Web Pages. The popular
web paths are used to extract in user’s navigation web log and
predict current user’s next step. Our system can solve the sparsity
of Web links and the diversity problem of user’s access behavior.
Moreover, our approach discovers users’ relevant documents to
improve the performance of search results.
Keywords- Web Usage Mining, Frequent Patterns, Sequence
Mining, Incremental Mining, Web Logs, Link Analysis, Log
Mining
I. INTRODUCTION
Data mining has gain a lot of attention nowadays. The
utilization of data mining techniques as a powerful data
extraction method brings up the popularity of data mining.
With the increasing popularity of the Web, it is no surprise
that Web mining has attracted many attentions. Web mining
is the use of data mining techniques to automatically
discover and extract information from Web documents. The
three categories of Web mining tasks are Web content
mining, Web usage mining and Web structure mining.
Structural Mining emphasizes on knowledge discovery
for the structure of the Web system, including the mining of
the user preferences on web browsing, the usage of the
different URLs in a particular websites, external structure
mining and internal structure mining. Structure mining is a
discovery patterns in the browsing and navigation to
understand users’ behaviors on the web. Within this
category, we focus on link mining and review methods
applied in Web structure mining such as HITs and Page
Rank. Several Link Analysis algorithms, based on the
popular Page Rank algorithm, play key roles in the context
of web search engines.
Global search engine such as Google and AltaVista has
been a great help for users to find desired information on the
ever growing Web. Google’s Page Rank is a widely applied
algorithm, which can be described as the stationary
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.145
Khin Nwe Ni Tun
University of Computer Studies, Yangon
Myanmar
knntun@gmail.com
probability distribution of a certain random walk on the web
link graph- the graph whose nodes are the individual Web
pages and directed edges are the hyperlink between pages.
The low performance of existing technologies lies in
several aspects: to find relevant information using content-
based similarity such as database approach need more time
and the searching keyword is restricted based on database
format. Link analysis technologies could not be directly
applied because the system will use the information
collected in web server to recommend the website based on
user interest. It is very time consuming and therefore a tool
(intelligent agent) is required to narrow down searching.
In general, the process of discovering useful the frequent
sequence is utilized by log mining methods such as
association rule mining, sequential pattern mining,
clustering and etc. A normal association rule mining
algorithm may fail to discover significant rules due to the
diversity. Instead of this association rule, they based on the
FP-growth algorithm to mine for generalized association
rules efficiently. These association rules are subsequently
pruned and applied to page re-ranking.
In this paper, a method is proposed to generate implicit
link structure based on frequent sequence pattern mining
from Web Logs. We describe an overview of frequent
pattern mining for extracting different types of patterns as
implicit link in a Web log. We can obtain page’s access
frequency and the traversal patterns of information finding
by mining the user logs. To compute the importance of
Web page, the link analysis algorithms use implicit links for
improving the performance of search result.
The rest of this paper is organized as follows. In Section
2, we introduce related works. In Section 3, we present
about frequent sequence patterns mining algorithm. In
Section 4, we describe mining technology to construct
implicit links structure and the re-ranking mechanisms in
site search. Finally, conclusions and future works are
discussed in Section 5.
II. RELATED WORK
Earlier works on association rule based log mining
focused on several applications. Nanpoulos et al. [1]
proposed a method for discovering access patterns from web
549
logs based on a new type of association patterns. They use a
candidate generation algorithm that requires multiple scans
of the database. Srikant and Agrawal [6] presented an
algorithm for finding generalized sequential patterns that
allows user-specified window-size and user-defined
taxonomy over items in the database. This algorithm also
required multiple scans of the database to generate
candidates.
Zheng Chen et al. [12] proposed an approach to
constructing implicit links by mining users’ access patterns,
and then apply a modified Page Rank algorithm to re-rank
web pages for small web search.
Traditionally, most of them use “full text search”
technology which a large amount of documents containing
the same keywords provided by the user and rank them by
keyword-similarity. Berkeley’s cha-cha search engine
organizes the search results into some categories to reflect
the underlying intranet structure [2].
Kleinberg’s HITs algorithm [5] (is also called modified-
HITs). This method does not separate the user logs into
sessions based on their tasks, thus many noise data will be
introduced into the link matrix.
Mobasher et al. [7] proposed a framework for web
mining using various web mining task and implemented a
prototype namely WEBMINER by applying the framework
as proposed. The problem has already pointed out from the
system is to perform cluster analysis on association rules
and sequential pattern discovery.
Mobasher et al. [8] proposed an effective technique for
capturing user profiles based on association rules discovery
and usage based clustering combining with current status of
an on-going activity to perform real time personalization
while Toolan and Kushmerick proposed techniques based
on web usage mining to deliver Personalized Site Maps that
are specialized to the interest of each individual visitor.
Renata Ivancsy [11] proposed PD-Tree algorithm for
determining whether a tree is contained by another tree. By
using a push down automaton, this can be done frequent
itemset discovery and the association rule mining with
different minimum support and confidence threshold values.
In mining phase, the frequent pattern discovery executed
iteratively on the whole dataset.
Some recommender systems [10] have been developed
to recommend Web pages. They used the Apriori algorithm
to mine for association rules over user’s navigation
histories. Their works focus on the discovery of frequent
item sets in transactional databases but avoid costly repeated
database scans and candidate generation.
constructed a new implicit link graph instead of the original
explicit link graph in Web. Thus we use a Frequent
Sequence Pattern Mining (FS-Miner) method to extract
implicit E ′ from Web logs by analyzing the observed users’
browsing behaviors.
Figure 1. A Framework for constructing Implicit Link
Generally, a web can be modeled as a directed graph G=
(V, E) represent pages in the Web. We construct directed
implicit graph:
G ′ = (V , E ′)
where V is the set of vertices represent the web page and
E ′ denote implicit links between pages. Each implicit
link I i , j ∈ E ′ assume that there exists implicit link
between pagei and page j . The entire Web log consists of
a number of browsing session and navigational links is
generated by the user. These implicit links are then re-
ranked by Page Rank algorithm.
A. Data Preprocessing
The starting and critical point for successful log mining
is data pre-processing. The pre-processing of Web logs is
usually complex and time demanding. It comprises four
different tasks: (i) the data cleaning (ii) the identification
and the reconstructing of users’ sessions (iii) the retrieving
of information about page content and structure, and (iv) the
data formatting. Each entry in raw web server log contains
time, IP address, and automatic created unique session
identifier, visited page, referrer and other data. Many entries
are considered as uninteresting items for mining and then
are removed. An example of web server logs can be
illustrated in Table 1.

III. GENERALIZED IMPLICIT LINK

CONSTRUCTION
We developed a general log mining model to discover
frequent sequence pattern for site search. This model is
illustrated in Figure 1. This model uses log file as the input
and output a set of frequent sequence pattern. We

550
 6. LogReader.OpenLogFile(LogFilePath,
TABLE I. SAMPLE WEB SERVER LOGS LogFileName, ModeRead
StrTypeOfLogFormat)
7. LogReader.StrReadLog
8. While NOT LogReader.EndOfStrReadLog
9. Retrieve IP Address in Log Attributes
10. Retrieve WebLink in Log Attributes
11. Add IP Address and WebLink to HT
12. logReader.StrReadLog
13. Loop
The remaining entries must be grouped by the visitors 14.
that performed them. We group consecutive entries to a user 15. LogReader.CloseLogFile
session or “transaction”. We support two criteria: 16. End Sub
1) A new session starts when the duration of the
Figure 2. Algorithm for Creating Header Table
whole group of traversals exceeds a time threshold.
2) The elapsed time between two consecutive In Figure 3, to generate frequent pattern, server logs must be
traversals exceeds a threshold. filtered. Many entries are considered uninteresting for
After pre-processing, we can get a set of n pages, P= {p1, p2, mining are removed. Once scan the header table (HT), each
…, pn}, and a set of m transactions T={t1, t2, …,tm}, where entry in frequent table has the name of the link and the
each ti ∈ T is a subset of P. count of that link.

B. Mining Step of FS-Tree 1. Read record in HT

In this section, let a sequence of pages
Seq= { p1 p2 .... pn } be a collection of items and a link li
from pi to pi +1 denote li = pi → pi +1 . We currently
assume that consecutive accesses from the same host during
a certain time interval come from the same user. The log file
has to be converted into two fields: SID that stores the
session id and InSeq sequences of page references accessed
by the user in a certain order. We allow duplicate items can
be either backward traversal or the same item can appear
more than once in the same sequence.
We consider any link that has the total number of links
in the web log file is greater or equal than the minimum
support count. If the number of link between pages is
greater than or equal to minimum support count, we called
as frequent link. If the number of link between pages is less
than minimum support count, we called as non-frequent
link. In order to choose the most suitable support for mining
user access pattern, when the minimum support is too small,
the number of implicit links is large. When the minimum
support is large, the impact of the Page Rank on the search
result is very weak. In Figure 2, a header table (HT) stores
several attributes about frequent and potentially frequent
links in the database. Each entry in the header table HT has
three fields: the name of the link, the count of that link and
linked list head pointer that points to the first edge in tree.
The algorithm for creating header table is illustrated below:
1. Declare Variables
2.
3. Sub HeaderTable (HT)
4.
5. Set LogReader= CreateObject
(“Log File Name”)
2. For each record in HT
3. Read fields (URL link)
4. RecordCount= RecordCount+1
5. Next record
6. If RecordCount>=min-sup
7. Save records
8. Else
9. Remove records
Figure 3. Algorithm for frequent link and non-frequent link table
C. FS-Tree Construction
The FS-Tree [3] is a compressed form for representing
sequences scanned from the input web log file. It is
compressed with three main steps. These steps are
performed for only frequent links (denoted as implicit
links).
Extracting Derived Paths: We extract its derived paths from
Header Table (HT) by following the linked list head pointer.
For each path in frequent sequence tree, we extract its path
from root of the tree until end of its path. We can call
derived paths of each link.
Conditional Sequence Base: By using the set of derived
paths of each link extracted in previous step, we construct
the conditional sequence base for each link by calculating
the frequency count of each link in the path.
Constructing Conditional FS-Tree: By using the conditional
sequence base, we create a tree and insert each of the paths
from the conditional base of each link into it in a backward
manner. We create this tree with necessary nodes, edge and
edge counts. This tree is called the conditional FS-tree for
that link.
Extracting Frequent Sequence Link Path: Given a
conditional frequent sequence tree of that link, we perform a
depth first traversal for that tree and return only sequences

551
link satisfying minimum support value. This link is called
recommendation link for user based on previous navigation
web logs.
Now, we described in detail the mining steps that
the final user log contains all implicit links. We separated
implicit link from explicit links by setting minimum support
count. This method generates implicit link pair ( I i , j ∈ E ′) ,
paths of the pattern should occur frequently in the log with
different in-between pages. Thus we can extract implicit
links by analyzing the observed explicit paths in the log file.
After frequent sequence patterns are generated, we
constructed the implicit link graph G ′ = (V , E ′) . We
normalized implicit links by representing frequent link
count. The resulting graph is used in subsequent link
analysis algorithm.
IV. RE-RANKING IMPLICIT LINKS
Link analysis algorithm play key role in web search
system. Link analysis technology has been widely used to
analyze the page’s importance, such as HITs [4] and Page
Rank [6, 2]. Page Rank is an algorithm of Google which
measures the importance of web pages. Page Rank
algorithm models the users’ browsing model, which
assumes that a user either follow a link from the current
page or jump to a random page in the graph. The Page Rank
measure effectively distinguishes high-quality web pages
from low-quality web pages. The Page Rank of pagei can
be defined as:
ε (P+ + P− )
R (i ) =
n
+ (1 − ε ) ∑ R ( j ) / outdegree ( j )
( j ,i )∈G
(1)
where
• ε is a constant usually set between 0.1 and 0.2 ,
• n is the number of nodes in G and
• outdegree(j) is the number of the edges leaving
page j .
Link analysis algorithm similar to Page Rank, we apply
it to re-rank the web pages after obtaining the implicit link
structure. To describe the graph, we construct an adjacency
matrix ‘A’ that used to compute the rank score of each page.
Generally, the rank score PRi of pagei is evaluated by a
function on the rank scores of all the pages that point
to pagei :
number of links on them, expect for a finite set of path
lengths that may be missing.)
V. EXPERIMENTAL RESULT
Traditional search engines are widely used by people to
find information on the web. Nevertheless, people often go
some specific web sites directly. The goal of our work is to
improve the search performance by analyzing the user
access log in the web site. At the pre-processing step, data
cleaning is performed by filtering out the access entries for
embedded objects such as images and scripts. After that, we
identify consecutive accesses from the same IP address
during a certain time interval is from a same user.
We use an efficient strategy for discovering frequent
patterns in sequence databases that does not require multi
scans of the database. Without the need for scanning and
reloading the entire database, this system scales up linearly
with the size of the input database. Our approach provides
significant performance advantages over full re-computation
even for relatively large size.
In web, hyperlinks could be divided into navigational
links and recommendation links. The latter is useful for link
analysis to enhance search. Some evidences should be given
to prove that the implicit links satisfy the recommendation
assumption. The prediction precision directly reflects the
correctness of the page recommendation. The precision
measures the degree to which the algorithm produces and
accurate result. In order to evaluate the precision measures
of search effectively:
P+
Prediction precision =
+ −
Where P and P are the numbers of correct and
incorrect predictions, respectively. Our experiment result
used from log file from the web proxy server, which
contains different web sites accessed by different users. The
result is shown in Figure 4.

PR i = ∑ PR . A[ j , i ]
j:I i , j ∈ E
j
(2)

In our system, we will not consider the web graph

represented by matrix ‘A’ that is not irreducible (i.e, there is
a directed path from every node to every other node) and
aperiodic (i.e, for all i,j, there are the paths with all possible Figure 4. Prediction of Precision by Pages

552
 We predicted the higher the minimum support threshold
of implicit link, the percentage of prediction precision
linearly increases. It indicates that our implicit link is
accurate and reflects users’ behavior and interest hyperlinks
to indicate our implicit links are accurate. We found that the
performance of implicit link based Page Rank is not worse
than explicit link-based Page Rank. In order to evaluate our
algorithm effectively, given a query Q, the precision
measures the set of the relevant pages to the query and the
number of pages.
VI. CONCLUSION AND FUTURE WORK
This paper navigates web paths in web log for improving
the performance of site search of users’ relevantly. The
basic idea can be interpreted jump to a page that is relevant
to the current page. Our method use frequent sequence
patterns to construct implicit link structure. And then, our
system applied Page Rank Algorithm to re-rank web pages
based on previous pages in the same user session. The
experiment results show that our system can effectively
improve the search performance. How to discover users’
relevant documents to improve the performance search
results, are main links of our future works. Since Page Rank
is an algorithm which is query independent, in the future we
plan to show that our algorithm improve a part of popular
queries’ precision.
Knowledge and Data Engineering Exchage Workshop
(KDEX’99) (Nov.).
[9] Page L., Brin S., Motwani R. and Winograd T. The
PageRank citation raning: bringing order to the Web.
Technical report, Stanford University Database Group,
1998.
[10] Page L., Brin S., Motwani R. and Winograd T. The
PageRank citation raning: bringing order to the Web.
Technical report, Stanford University Database Group,
1998.
[11] R.srikant and R.Agrawal, “Mining Generalized
Association Rules”, in Proceeding of the 21st Int’l
Conference on very Large Databases, Zurich,
Switzerland, Spetermber 1995.
[12] Yang Q., Zhang H.H. and Li T. Mining web logs for
prediction models in WWW caching and prefetching. In
Proc. of KDD’01, 473-478, August 2001.

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[2] Brin S. and Page L. The anatomy of a large-scale
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[3] C.Ruiz, E.Rundensteiner, and Maged EI.sayed, ‘FS-
Miner: Efficient and Incremental Mining of Frequent
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13-14,2004.
[4] Kleinberg J.M. Authoritative sources in a hyperlinked
environment. Journal of the ACM, 46(5); 10-25,1963.
[5] Levene M. and Wheeldon R. A Web site naviagation
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[7] Mobasher, B., Jain, N., Han, E.S., and Srivastava, J.
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[8] Mobasher, B., Cooley, R., and Srivastava, J. (1999).
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553
 2009 International Conference on Computer Engineering and Technology

Recognition of Eye States in Real Time Video

Lei Yunqi, Yuan Meiling, Song Xiaobing, Liu Xiuxia, Ouyang Jiangfan
Department of Computer Science, Xiamen University,
Xiamen, P. R. China
e-mail: yqlei@xmu.edu.cn

Abstract- A composite algorithm is proposed to identify the

opened and shut states of eye in real-time video. Sequentially,
each frame of the video is processed first to find and separate
the eye regions; then, the accumulation array of circular
Hough transform is applied for detecting circles to recognize
the opened state of the eye, and the upper eyelid curve is cal-
culated to identify its bending direction to recognize the shut
state; after that, if the state cannot be determined yet, judging
by the standard deviation of distance between eyelids, the eye
state of blinking is determined finally. An accuracy of 94% for
eye state detection is obtained in our experiments by using the
proposed algorithm. As an important part, the algorithm is
going to be utilized in a fatigue warning system for drivers.
Keywords- Real time video processing; pattern recognition;
eye state recognition; drowsiness detection; fatigue warning
system
I. INTRODUCTION
Eye state detection plays an important role in the fields
like human-machine interface and drowsiness detection of
drivers, etc. To detect the drowsiness of a driver, many
efforts have been made in the past decades [1,2,3]. The
methods already proposed can be summarized as to take the
following three measurements: (1) to monitor the movement
states of the driver (such as the activities of his/her eyelids,
the movement styles of head, the grip strength on steering
wheel, etc.); (2) to analyze the physiological signals (ECG,
EMG, etc.) of the driver; (3) to compute the running
parameters of the vehicle (speed, acceleration, driving trace,
etc.). Generally, the methods based on the later two are
complex and expensive since a lot of sensors are required.
Our study here is about the first measurement mentioned
above and is focused on the algorithm to implement a video
monitoring system to identify the opened and shut states of
eye in real-time video so as to determine the frequency of
blinking, which is to be used as an important reference of
getting drowsy of a driver. PERCLOS (the percentage of
time that eyes are closed in a given period) is the most-used
index of measuring drowsiness. High frequency of blinking
is strongly related to the getting drowsy of a man while high
PERCLOS is strongly related to the drowsy state of the man
[4]. The algorithm we proposed can be applied to determine
the blinking frequency and PERCLOS thus can be used in
fatigue warning system for drivers.
Fig.1 shows a general process for eye state identification
from real time video. The video frames are captured
sequentially, then for each frame, the face area is detected
Figure 1. Eye state identification
and extracted from the obtained frame, after that the eye
regions are separated and cut from the face area. Followed
by some pre-processing on eye regions, the exact eyes
without eyebrow or glasses borders are obtained. And then, a
proper algorithm is applied to identify the eye states, opened
or shut. The frequency of blinking and PERCLOS then can
be determined and fed into the fatigue warning system.
As stated above, to detect drowsiness of a driver by
identifying his/her eye states, we need real-time video to
compute the frequency of blinking and PERCLOS, thus, a
video camera must be installed properly on a car. In our
experimental scheme, a camera is installed behind the
windscreen and above the dashboard. In order not to disturb
the attention of driving and not to block the driver's sight, the
camera is set on driver's right front side (or on left front side)
other than directly point to the driver, as shows in Fig. 2.
There is a problem that should be taken into
consideration when capturing the video in a car. Generally,
the illumination level changes extremely wide from day time
to night time, the face area is very bright if the sunlight is
directly shined on the driver while it is almost totally dark at
night. To adapt to this difficult situation, the video camera
adopted should be of auto-iris-diaphragm and dual
functioned of both visible light and infrared light. Besides,
Infrared Light-Emitting Diodes (IR LEDs) should be used
for the infrared light supplement for the camera during the
nights or on cloudy dark days.
The rest of this paper is organized as follows. Section 2
presents the method of how to get the face area. Section 3
describes the way to extract images of eye region from the
face area. In section 4, a composite algorithm to identify eye
states is provided. Experimental results are shown and
analyzed in Section 5, and finally in Section 6, conclusions
and remarks are stated.
Figure 2. Our video capturing system

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 554

DOI 10.1109/ICCET.2009.105
 II. DETECTION OF FACE AREA
In the process for eye state identification (Fig. 1), after a
video frame is captured, detection of face area is conducted.
Till now, lots of approaches have been proposed in the field
of face and eye detection. Yang et al. [5] presented an
overview of face detection methods developed in recent
years. A scheme of boosted cascade algorithm using Haar-
like features as identification patterns was introduced by
Papageorgiou et al. [6]. Lienhart and Maydt [7] enhanced the
algorithm, Viola and Jones [8] used the algorithm in face
detection system. Harr-like features encode differences in
averaged intensities between different regions in an image
and they may contain rich information about an object, such
as an eye region is darker than that of cheek and the bridge of
nose is brighter than its sides. An illustration of Harr-like
features is shown in Fig. 3.
In this kind of boosted cascade algorithm, different Harr-
like features are selected and processed in different cascade
weak classifiers, a learning algorithm chooses and combines
those weak classifiers to generate a strong classifier. Fig. 4
shows a decision tree which displays the boosted cascade of
the classifiers. Each classifier at different stage is trained to
detect objects with interested patterns while rejecting a
certain fraction of non-object patterns. This cascade scheme
is called AdaBoost. When the Harr-like features are used,
then the scheme can be named as Harr-like+AdaBoost.
To create a ready classifier for face detection a large
amount of images with and without faces under different
conditions should be prepared to train the classifier. The
classifier in this paper we used is taken from the Open
Source Computer Vision Library (OpenCV) [9].
For our case, a driver's seating posture generally meets
the following conditions: (1) his/her head position is
relatively fixed and moves only in a small scope; (2) in most
of time the driver is looking forward to drive. Therefore, a
natural assumption for our process of face detection is that
when capturing the real-time video there is only a small
change of viewing angle and the face approximately is in a
front view to the camera, thus the faces appeared in frame
images are almost complete as a whole. As shown in Fig. 5,
for the images containing front view faces, the face areas can
be correctly obtained even for the girl with glasses.
In our processing, if no face area is detected in a frame,
then the frame is simply discarded. Although there are some
Figure 3. Harr-like features on a face
Stage 1 Y Stage 2 Y … Stage N Y Faces
N N N
Non-faces
Figure 4. A boosted cascade
555
Figure 5. Detection of face
small amounts of frames which containing faces are
discarded wrongly, the accuracy of 99% has been achieved
in our experiments for face area detection, this is quite
enough for our purpose.
III. TRACKING AND EXTRACTING OF EYE REGION
For two reasons, the eye on the left or on the right, only
the one which is closer to the camera is necessary to be
considered in our system. (1) Generally, the left and the right
eyes are always open or close simultaneously. Thus,
although occasionally there are cases that two eyes are not at
the same state, asynchronous or asymmetric eyelid
movements of the two eyes can be ignored with only
affecting identification results slightly. (2) When the driver
rotate his/her head in some angle, the eye more distant from
the camera may not appear as a whole or even disappear in
the video. If the two eyes on both sides are to be considered
at the same time, one more camera is necessary and this is
not economic.
To extract the eye region, a two-stage procedure is done
in our scheme. First, the coarse eye region is detected and
tracked in the scope of the face area, which has been cut out
by the cascaded classifier as shown in section 2. Then, the
exact eye region without eyebrow or glasses’ border is
extracted for further processing.
A. Tracking of eye region
To track an eye region correctly, determination of the
position of the eye region in a frame as the first one which
contains the eye is critical. Several methods were
implemented in our first stage of study to find the region, but
due to some negative factors, such as that the illumination
varies extremely widely, the eyes may be with glasses,
closed eyelids obstructed eyeballs thus hidden the important
clues to identify the region, etc., the results we obtained were
unsatisfactory. The methods we tried included the method
based on intensity projection, the method based on skin color
model and the method based on assumption that the eye
regions are the darkest on the face. The mentioned negative
factors form obstacles for those methods to produce desired
results.
We turn to the method of Harr-like+AdaBoost which is
already applied when detecting a face area in the former
section, but different in the training stage that the eye region
images instead of the face area images are used to train the
classifiers. This method is affirmed to be an effective way to
obtain eye regions in our experiments.
We notice in our study that sometimes several but not
unique candidate regions of eye are obtained by the Harr-
like+AdaBoost algorithm, especially, even a mouth region
which shows similar shape tends to be mistaken as an eye
region. To deal with this situation, we simply confirm an eye
region by checking whether it is in the upper part of face, if
not, just discard the region as non-eye one.
The algorithm is performed for each sequent frame until
the first eye region is found where an eye is detected at the
first time. For subsequent frames, the found eye region is
traced until there is no eye detected and a new procedure of
detection is followed from the beginning. The cycled work is
done repeatedly.
To track the eye region in subsequent frames, the region
obtained in the previous frame is enlarged to a bigger
rectangle area in the current frame, then the Harr-
like+AdaBoost algorithm is performed in this rectangle,
which covers the eye region acquired in the previous frame
due to continuous and small-scope movement style
assumption for drivers. Notice that, now the Harr-
like+AdaBoost algorithm is only performed in a smaller
scope other than on the whole detected face area. In this way,
it is much more efficient to detect the exact eye region and
this is the reason why we work with the tracking of the eye
region, it is really beneficial to reduce the computing time
and can assure the system to meet the requirements for
processing the real-time videos.
Only when the driver moves too quickly or turns head
with big angle thus the image of the eye disappeared or is not
identifiable, then the face detection and eye detection
procedures are to be performed again from the beginning for
the newly coming frames.
Some results of eye detection are presented in Fig. 6,
where the eye regions are visualized by a black rectangle.
The average detection accuracy of the eye regions achieved
for the normal cases is about 98%, but the detection accuracy
achieved for closed eyes and those eyes with glasses is about
95%, lower than for the normal cases.
B. Extracting of eye region
After the eye region is detected and traced, the exact eye
is going to be extracted before its states are identified.
Getting exact eye region means finding the eye by detecting
the edges of upper and lower eyelids. Many kinds of edge
detection methods exist. The Canny, Sobel, Prewitt and
Roberts edge detectors were performed in our study to search
the edges of eyelids. However, our experimental results show
that although the mentioned edge detectors perform well on
eyes under normal illumination, yet the edge information is
unstable for those eyes in the cases when the lights shine
strongly from side directions. Sometimes the edges even
cannot be searched out.
Figure 6. Detection of eyes
556
Figure 7. Obtaining of the exact eye regions
To fulfill the task of edge detection of eyes, a simple and
effective way is adopted here in our scheme. Initially, a
histogram equalization is carried out to increase contrast of
the eye region (Fig. 7b), the edges of eyelids and iris are
obviously darker than the non-eye regions after equalization.
Then, the image of the eye region is converted into binary
one by a dynamic threshold, which depends on the average
intensity of original eye region. The picture now contains not
only the eyelid surrounded region purely, but eyebrows,
glasses borders, or hairs etc., are also appeared in the picture
(Fig. 7c). Generally, those black segments appeared in the
binary image are not all connected, the moment and the
eccentricity of each segment in the image therefore are
computed to isolate the eyelids and the iris region. The
segment with moment at central is the possible exact eye
region while the other segments are possibly eyebrows or
glasses borders, thus are separated and erased (Fig. 7d).
It has been shown in our experiments that the approach
stated above can avoid the interference of glasses effectively
provided that the border of glasses is not connected with the
eyelids. Two examples are shown in the first two columns in
Fig. 7. Fig. 7 also shows that the exact eye regions can be
obtained correctly even under the various conditions of
widely changing illuminations.
IV. IDENTIFICATION OF EYE STATES
Regarding the identification of eye states, a variety of
approaches have been studied in recent years [10,11,12,13].
These methods can generally be divided into two categories:
the methods based on the characteristics of the eyes and the
methods based on pattern recognition. The former ones
utilize the features of eye corners, eyelids and iris, etc. to
recognize eye states. The intensity template, Fisher method,
intensity projection, searching of the upper eyelid, detecting
of iris through Hough Transform(HT) etc. are among the first
category. The methods of the second category generally
convert the problem of identifying eye states into a question
of classifying the opened or closed states, that is, consider
and recognize the patterns of eye states. The algorithms of
neural networks, the support vector machines, and the hidden
Markov model, etc., are among those methods. There are
difficulties to apply the second kind of methods in our real-
time system due to the complexity of training and the diverse
varieties of the images.
For our fatigue warning system, as long as the exact eye
region is obtained, it is more straightforward to use the fore
mentioned first category methods to identify the eye states,
opened or shut. Then, the blinking frequency and the time of
eye closing can be determined as the references of
drowsiness. We focus on identification of the two distinct
states of opened and shut while ignore the other ambiguous
eye states, although our approach stated in subsection IV.C
can be used to determine roughly the blinking extent of eyes.
Identification of the ambiguous eye states was also depicted
in [14] by Heishman and Duric.
Usually, to identify the states of eye, only a single
criterion is not enough. The environments are different when
capturing the video: the extremely widely changing
illumination, varied distance between the face and the video
camera, etc, thus the images we got maybe quite different.
Furthermore, people may have very different eye shapes,
some eyes are relatively round when opened while others are
relatively slender and irises do not expose much even when
the eyes are opened at the largest extent. We propose a
composite algorithm in the following to identify the states of
eye and the algorithm is verified in our experiments that it is
feasible to meet our requirements. The proposed algorithm is
composed of three procedures as follows.
a) The accumulation array of Circular Hough Transform
(CHT) is computed to find out whether a peak in the array
exists to detect iris, hence recognizes the opened states;
b) The direction of upper eyelid curve is calculated to
find out whether the eyelid is below a line to recognize the
shut states;
c) If the states cannot be determined yet, the standard
deviation of distance between the upper and lower eyelids is
computed to find out whether the deviation is bigger than a
threshold to judge the state as it opened or shut.
For each sequent frame that contains a face, the above
procedure is carried out to identify the eye states in the
frame. Notice that, once the eye state, opened or shut, is
determined, a new frame is always captured for continuous
processing of the real-time video, while if no eye or even no
face is detected in the frame, the process turns to capture a
new frame and the process start from the beginning.
A. Using the accumulation array resulted from CHT to
detect iris
An iris exists as a quite perfect round area in an eye.
Generally an iris exposes more than half when the eye
opened normally, that is, a dark region of more than half
round area would appear in an eye image. The Hough
transform then can be used to detect the iris. But, the result
of detecting circles by HT depends on the choice of some
parameters, which are unfortunately difficult to determine.
Xu and Oja [15] proposed a curve detection method of
Randomized Hough Transform (RHT) which is faster than
the traditional Hough transform. In our experiments, there
are many of closed eyes that are detected by HT have false
Figure 8. The resulted accumulation array from CHT
557
circles so they are mistaken to be opened eyes. Fig. 8 shows
some examples.
However, we have found that for the opened eyes, the
corresponding results of the accumulation array obtained
from CHT have obvious peaks while that kind of peaks do
not exit for the closed eyes, this is because that the iris circles
have more points on the edge than other fictitious circles.
Based on this observation, we check if there is a peak in the
accumulation array to know about the existence of iris, hence
can make decision for the state of an opened eye.
B. By the bending direction of upper eyelid to determine
the shut state
The upper eyelid bends downward when the eye shut
under plain face expressions (Fig. 9a). Therefore, we can
identify the closed state of an eye by computing the bending
curve of upper eyelid. If the curve representing the upper
eyelid is below the line from the inner eye corner to the outer
corner, the shut state of eye then can be identified.
However, when people smile or show some other
exaggerated expressions, the upper eyelids probably will be
upward arcs or almost straight lines (Fig. 9b). To deal with
those cases, we propose a standard deviation method and it
will be discussed in the next subsection.
After the exact eye region is obtained, the eye corners
can be easily located by scanning the region from left to
right. We discover that the middle part of eyelid could be
more precise to present the state of eye especially when the
parts that are near to corners tend to be unsteady. Taking this
factor into account, we let the both eye corners move M
points toward the central point, then we connect the moved
eye corners to form a reference line.
The upper eyelid cannot be expected as a perfect smooth
curve, there are noises on the images and the faces of people
are quite different. Hence, the upper eyelid is not required to
be absolutely above the reference line that connects eye
corners. Instead, statistical calculation needs to be
considered. Let:
where aσ is the total number of points above the line,
while bσ is the total number of points below the line. k is the
difference between them, and α represents a threshold value
which is chosen as 2.0 in our experiments. If k is smaller
than the threshold α , the state of eye is considered to be
closed; otherwise, further judgment is necessary as we will
describe in the next subsection.
Figure 9. a: shut eyes under natural expressions b: shut eyes under other
expressions
C. Standard deviation of distance between upper and
lower eyelids
For the states that cannot be decided yet by the methods
presented in the former two subsections, a final processing is
necessary to provide answers. A method of computing the
absolute distance between upper and lower eyelids is used in
many papers. The distance is compared to a threshold for
making judgment, but there is still no good measure worked
out about the choice of the threshold, while it is critical.
Based on the fact that the distance between upper and
lower eyelids is almost the same for a closed eye while it is
much more different for an opened eye, therefore the
standard deviation of distance between upper and lower
eyelids is calculated to decide the states of eye. Define
Where yiU , yiL means y-coordinate of upper eyelid and
lower eyelid in column i respectively, while d i is the distance
between them and d is the mean. s gives the standard
deviation of the distances.
Simply, the eye is regarded as opened if the standard
deviation s is bigger than a threshold, which is chosen as
1.0 in our experiments. Otherwise, the eye is considered as
closed. In our studies, the suitable threshold values tend to be
around a fixed one for different eyes.
In Fig. 10, the numbers below the eye pictures are the
standard deviation values of the corresponding eyes. It can
be seen that the numbers are smaller for closed eyes than for
opened eyes, thus the deviations can be utilized as the
measurement to discriminate the opened eyes from the
closed ones efficiently, as indicated in Fig.10.
If the eye regions are already correctly obtained, by the
composite algorithms stated in this section to detect eye
states, we reach an accuracy of about 97% in our
experiments. Compared with the results obtained by other
methods, such as by intensity template, Fisher method,
intensity projection, the averaged detection accuracies are
92%, 93%, 87%, respectively. Apparently, the resulted
accuracy of the method we propose is higher.
558
Figure 10. Standard deviation values
V. EXPERIMENTAL RESULTS
The fatigue warning system is to be used in car for driver,
so the experiments have been carried out in moving cars. The
videos are captured by using an ordinary webcam, with its
highest resolution (dpi): 800 x 600, maximum frame
frequency (FPS) 30. For not to disturb driving, the camera is
set not directly pointing to the driver (Fig. 2). The size of
frames captured is 320 x 240 pixels.
Table 1 provides some results of identification of eye
states in our system, representing the achievements obtained
by the whole process from face detection at the beginning to
states recognition at the end for sequent frames. In Table 1,
except the video of driver E, which is captured in laboratory,
other four videos were obtained in moving cars. 18 drivers
already took part in our experiments, the average eye state
detection accuracy is about 94%, precise enough for our
fatigue warning purpose.
Besides, our program is suitable for real time videos, for
20 frames can be processed in one second.
VI. CONCLUSIONS AND REMARKS
A composite algorithm is proposed to identify the
opened and shut states of eye in real-time video. The
algorithm comprises three main stages: face area detection,
eye region detection and eye state identification. Sequentially,
each frame of the video is processed in the first stage to find
the face area, then the eye region is separated; A new
composite method is proposed to identify the eye states, the
accumulation array of circular Hough transform is applied
for detecting circles to recognize the opened states of eye,
and the upper eyelid curve is calculated to identify its
bending direction to recognize the shut states; after that , if
the state cannot be determined yet, judging by the standard
deviation of distance between eyelids, the eye state of
blinking is determined finally. We have realized a satisfied
detection performance under real conditions in a car, an
accuracy of about 94% for eye state detection is obtained in
our experiments by using the proposed algorithm.
As an important part, the algorithm is going to be utilized
in a fatigue warning system for drivers.
ACKNOWLEDGMENT
 TABLE I. RESULTS OF IDENTIFYING THE EYE STATES [6] C.P. Papageorgiou, M. Oren, and T. Poggio, “A general framework
for object detection”, A general framework for object detection,
Opened eyes (frames) Shut eyes (frames) Accuracy 1998, pp. 55-562.
Counted by Detected Counted by Detected [7] R. Lienhart and J. Maydt, “An extended set of haar-like features for
observation correctly observation correctly
rapid object detection”, Image Processing.2002. Proceedings 2002
A 678 636 41 36 93.4% International Conference, vol 1, 2002, pp. I-900-I-903.
B 1285 1223 320 312 95.6%
C 866 803 118 106 92.3% [8] P. Viola and M. Jones, “Rapid object detection using a boosted
D 659 607 83 74 91.8% cascade of simple features”, Proceedings of the 2001 IEEE Computer
E 304 299 50 46 97.4% Society Conference, vol 1, 2001, pp. I-511-I-518 .
[9] Intel, “Open source computer vision library;
http://sourceforge.net/projects/opencvlibrary/”, 2001.
This work is supported by National High Tech Research
[10] J. Deng and F. Lai, “Region –based template deformation and
and Development (863) Programme of China rnasking for eye-feature extraction and description”, Pattern
(NO.2006AA01Z129). Recognition, vol 30, 1997, pp. 403-419.
[11] G. Chow and X. Li, “Towards a system for automatic facial feature
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drowsiness while driving by measuring eye movement - a pilot study”, actions for facial expression analysis.”, Proceedings of the IEEE
Proc. 5th Int. Conference on Intelligent Transportation Systems, Conference on Computer Vision and Pattern Recognition(CVPR’00),
2002 , pp. 156-161. vol 1, 2000, pp. 294-301.
[2] T. Hamada , T. Ito , K. Adachi , T. Nakano and S. Yamamoto, [13] Y. L. Tian, T. Kanade and J. F. Cohn , “Eye-state action unit
“Detecting method for drivers' drowsiness applicable to individual detection by Gabor wavelets”, The 3rd International Conference on
features”, Proc. Intelligent Transportation Systems 2003, vol.2, pp. Multi-modal Interfaces(ICMI’00), 2000, pp. 143-150.
1405-1410. [14] Ric Heishman and Zoran Duric, “Using image flow to detect eye
[3] Q. Ji and X. Yang, “Real-time eye, gaze, and face pose tracking for blinks in color videos”, Applications of Computer Vision, 2007, pp.
monitoring driver vigilance”, Real-Time Imaging, vol 8 , 2002 , pp. 52-52.
357-377.
[15] L. Xu and E. Oja, “A new curve detection method: randomized
[4] D. Dinges and R. Grace, “{PERCLOS}: A valid physiological Hough Transform(RHT”, Pattern Recognition Letters,1990,pp. 331-
measure of alertness as assessed by psychomotor vigilance”, Tech 338.
Brief FHWA-MCRT-98-006, 1998.
[5] M.-H. Yang , D. J. Kriegman, and N. Ahuja, “Detecting faces in
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Transactions on, vol 24(1), 2002, pp. 34-58.

559
 2009 International Conference on Computer Engineering and Technology
Performance Analysis of Postprocessing Algorithm and Implementation on
ARM7TDMI
Manoj Gupta, B.K.Kaushik
Department of Electronics & Electrical Engineering,
G. B. Pant Engineering College,
Pauri-Garhwal (Uttarakhand), INDIA
manojgupta35@yahoo.co.in, brajesh_k_k@yahoo.com
Abstract—Due to rapid advancement in multimedia
communication a growing need emerges for the
implementation of audio algorithm in real time. Filters are
often used in Digital Signal Processing applications. The filters
fast processing is usually done on DSP chips. The calculation of
the DSP operations like multiplication and accumulation
(MAC) and convolution should be fast enough so that the
audio can be heard in real time. In this paper we take up a post
processing algorithm for audio that is used to control the Bass
and Treble. For this purpose first order IIR filter is employed.
This filter is used due to low computational burden that it
places on DSP resources. The method that is presented here is
based on transforming a prototype analog filter to the desired
digital filter using the technique of Bilinear Transformation for
mapping the poles and zeros in Digital domain. The calculated
coefficients are used for implementation of audio algorithm for
ARM7TDMI. Implemented code is optimized for the real time
operations. The analysis is done in MATLAB for showing the
effect of Treble and Bass for different frequencies. The filter
mapped in digital domain is stable.
Keywords- Digital Filter, IIR filter, Bilinear Transform,
ARM Processor, Post processing, PCM
I. INTRODUCTION
Post processing algorithms are often employed to control
low (Bass) and high (Treble) frequency audio components
[1-7]. Bass and treble can be controlled by using the either
analog or digital filters. Analog filters are designed using op-
amps, resistors and capacitors. Bass and Treble can be
controlled by varying the values of resistor and capacitors
according to the circuit design. The disadvantage of analog
filter is bulkiness of circuit and precision limitations in
controlling the performance. However in digital filters these
disadvantages can be easily overcome [5]. Thus, digital filter
are often used for the implementation of post processing
algorithms [1-4].
We use a first order IIR filter for filtering process. To
compute filter coefficients bilinear transform method is used.
Bilinear transform converts analog domain filter into the
digital domain filter. S-domain poles and zeros are mapped
into the Z-domain poles and zeros. We use an input
interleaved PCM file, which is later de-interleaved using two
buffers for left and right channel data. This de-interleaved
data is passed separately through the IIR first order filter.
After IIR filtering the data is again interleaved by inserting it
into one data buffer. Finally, PCM interleaved data is sent to
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.179
Laxmi Chand
Division of Electronics & Communication Engineering,
Netaji Subash Institute of Technology,
New Delhi, INDIA
laxmisharma34@yahoo.co.in
output. MATLAB code is implemented for bass and treble
control.
Post processing algorithm is implemented in C-language
for ARM7TDMI and optimized for the real time. For this
project work the ARM simulation environment is used. The
simulator for ARM is available as µvision3 KEIL. The
environment of windows is used throughout the project.
The bass and treble controls are set to vary the level of
audio in a band of frequencies centered around 100Hz and
10,000Hz respectively. There are different kinds of curves,
with a wide range of frequencies where the boost is constant
until the 'slope' of the curve is reached. The values of the
gain are varied from -12dB to +12dB.
The audio should be played in real time, if there is any
time lag between playing the audio, no one can understand
the original audio. To play the audio in real time we need to
optimize the algorithm. Two types of optimization are used
in audio. One is for the memory or size reduction and second
is for increase the speed of execution. There is a trade off
between the speed and the size of memory. The algorithm is
optimized for the real time.
II. POSTPROCESSING AUDIO ALGORITHM
A. Bass & Treble Control
The tone control allows enhancing or reducing the bass
and treble independently. When enhancement is required, it
is possible to apply on the whole signal a negative gain (tone
attenuation) in order to avoid eventual saturations (the signal
could already be at his maximum dynamic that we can’t
know). For high and low frequencies a gain and a cut
frequency are configurable.
Different kinds of 'curves' are obtained for treble and
bass. The slope of the curve for treble increases as frequency
is increased for positive gains values. There is a wide range
of frequencies where the boost is constant until the 'slope' of
the curve is reached. Most head units claim to vary the level
by plus or minus 10dB. However, the range may be as high
as +12dB or -12 dB.
B. Calculation of the Filter Coefficients
The transfer function of first order IIR filter used is
(a0 + a1* Z−1 )
H(Z) = (1)
(1 + b1* Z−1 )
The coefficients a0, a1, b1 of the digital filter in Z-
domain is calculated by using Bilinear Transformation
method.
560
C. Bilinear Transform Dividing by (1 + G * X ) we obtain,
The bilinear transform converts a transfer function in the [(G + GX) /(1 + GX) + (GX − G) /(1 + GX)Z−1 ]
H ( Z) =
Laplace transform domain to one in the Z-transform domain, [1 + (GX − 1) /(1 + GX)Z−1 ]
by substituting: On comparing with (1), we obtain the cœfficients as
S = c*
( Z − 1) i.e. S=
2 (Z-1)
(2) (G + G * X) (G * X − G) (G * X − 1)
* a0 = ; a1 = and b1 = (8)
( Z + 1) T (Z + 1) (1 + X*G) (1 + G * X) (1 + X *G)
We get the above value by simple Jacobins
approximation. Here is a simple derivation. 2) Low pass filter
exp(ST/2) The transfer function of the IIR filter in S-domain is
Z = exp(ST) = (3)
exp(-ST/2) [G '* (S + ωc )] 2fs(1 − Z−1 )
H(s) = G * at S = where G ' = 1 G (9)
Now expand numerator and denominator to a 1 order st [(S + G '* ωc )] (1 + Z−1 )
approximation-equally factorizing Using Bilinear transform and inserting S from (5) in (9)
(1 + T / 2*S + ......) (1 − Z−1 )
Z= where T = 1 fs (4) [G '* 2fs * + G ' ωc ]
(1 − T / 2*S + .......) (1 + Z−1 ) [G '− G ' Z−1 ) + G ' X(1 + Z−1 ]
H(z) = G * = G*

So we can write Z =
(1 + T / 2 *S)
and S = 2fs *
(1 + Z−1 ) (5)
[2fs *
(1 − Z −1 )
(1 + Z−1 )
+ G '* ωc ]
[(1 − Z−1 ) + G ' X(1 + Z−1 ]

(1 − T / 2*S) (1 − Z−1 ) Separating the constant terms and coefficients of Z−1

The mapping thus generated has some good properties which [(G '+ G ' X) + (G ' X − G ')Z−1 ]
H ( Z) ) = G *
are: (1) It is one-to-one and order-preserving, and it maps the [(1 + G ' X) + (G ' X − 1)Z−1 ]
Dividing numerator and denominator by (1 + G ' * X )
left-hand S-plane in the Laplace domain to the interior of the
unit circle in the Z-domain. (2) The mapping is not linear. By
setting Z =exp(jΩ) and S=jω in the bilinear transform [(G '+ G 'X) /(1 + G ' X) + (G ' X − G ') /(1 + G ' X)Z−1 ]
H ( Z) = G *
equation, the relation between analog and digital frequencies [1 + (G ' X − 1) /(1 + G 'X)Z−1 ]

can be determined: ωana log = * tan ( Ωdigital 2 ) . There are

2 On comparing with (1) we obtain cœfficients as
T (G' + G '* X) (G '* X − G ') (G '* X − 1)
a0 = G * ; a1 = G * ; b1 = (10)
two methods of compensating for this frequency distortion. (1 + G'*X) (1 + G '* X) (1 + G '* X)
First, an existing analog filter is converted; the constant c can The first order IIR filter used is given by
be selected to make one the filters match at one particular (a0 + a1* Z−1 )
H(Z) = where H(Z) = y(Z) x(Z)
frequency (for instance, the filter's cutoff frequency). The (1 + b1* Z−1 )
filter's frequency specifications can be made in the digital
(a0 + a1* Z −1 )
domain and converted back to the analog domain (with an y(Z) x(Z) = ;
(1 + b1* Z−1 )
arbitrary choice of c). An analog filter is then designed to
meet these specifications; when converted back to a digital Thus, y(Z) = a1* x(Z) * Z−1 − b1* y(Z) * Z−1 + a0* x(Z) .
filter, the critical frequencies will be in the right places. On converting to sampled time domain, output y(n) is:
y(n) = a1* x(n − 1) − b1* y(n − 1) + a0 * x(n) . Here x(n-1) and
D. First Order IIR Filter Coefficients Calculation
y(n-1) are previous input and output values respectively.
1) High pass filter
The transfer function of the IIR filter in S-domain is
(S + ωc ) 2fs(1 − Z−1 )
E. Treble Filter
H(s) = G * at S = (6)
(S + G * ωc ) (1 + Z−1 )
where ωc is the cut of frequency. Using Bilinear transform
and inserting S from (5) in (6) we obtain
(1 − Z−1 )
[2fs * + ωc ]
(1 + Z−1 )
H(z) = G *
(1 − Z−1 )
[2fs * + G * ωc ]
(1 + Z −1 )
[(1 − Z−1 ) + (ωc / 2fs)(1 + Z−1 ]
= G*
[(1 − Z−1 ) + G(ωc / 2fs)(1 + Z−1 ]
[(1 − Z−1 ) + X(1 + Z−1 ]
= G* where X = pi * fc / fs (7)
[(1 − Z−1 ) + GX(1 + Z −1 ]
Separating the constant terms and coefficients of Z−1 Figure 1. High pass filter for Treble
−1
[(1 + X) + (X − 1)Z ]
= G* The gain curve in dB with variation in frequency for the
[(1 + GX) + (GX − 1)Z −1 ] high pass filter or the treble filter is shown in Figure 1. The

561
curve is shown for the positive and negative values of gain.
Initially the positive (GdB>0) and negative (GdB<0) gains
increases with increase in frequency, but after a certain limit TABLE I. VALUES GENERATED FOR THE fc =100HZ
both gains are constant. A positive and negative gain
corresponds to treble augmentation and reduction S.No. GdB Gain a0 a1 b1
respectively. Such type of curves can be obtained for 1 -12 0.251189 0.980977 -0.968214 -0.949191
2 -11 0.281838 0.983695 -0.970897 -0.954591
different cut of frequency values.
3 -10 0.316228 0.986129 -0.9733 -0.959429
The coefficients values for the high pass filter are given 4 -9 0.354813 0.98831 -0.975452 -0.963761
through equation (8). Here, G is given as G = 10GdB / 20 . 5 -8 0.398107 0.990261 -0.977378 -0.967638
6 -7 0.446684 0.992006 -0.9791 -0.971107
F. Bass Filter 7 -6 0.501187 0.993567 -0.980641 -0.974209
8 -5 0.562341 0.994963 -0.982018 -0.976981
9 -4 0.630957 0.99621 -0.983249 -0.979459
10 -3 0.707946 0.997324 -0.984348 -0.981672
11 -2 0.794328 0.998318 -0.98533 -0.983649
12 -1 0.891251 0.999207 -0.986207 -0.985414
13 0 1 1 -0.98699 -0.98699
14 1 1.122018 1.000708 -0.987689 -0.988397
15 2 1.258925 1.00134 -0.988312 -0.989652
16 3 1.412538 1.001903 -0.988869 -0.990772
17 4 1.584893 1.002406 -0.989365 -0.991771
18 5 1.778279 1.002855 -0.989808 -0.992663
19 6 1.995262 1.003255 -0.990203 -0.993458
20 7 2.238721 1.003612 -0.990555 -0.994168
21 8 2.511886 1.003931 -0.99087 -0.9948
Figure 2. Low pass filter for Bass 22 9 2.818383 1.004215 -0.99115 -0.995364
23 10 3.162278 1.004468 -0.9914 -0.995867
The gain (dB) curve with variation in frequency for the 24 11 3.548134 1.004694 -0.991622 -0.996316
low pass or bass filter is shown in Figure 2. The curve is 25 12 3.981072 1.004895 -0.991821 -0.996716
shown for the positive and negative values of gain. The
positive (GdB>0) and negative (GdB<0) gains increases with
decrease in frequency, but after the cut off frequency both
gains does not vary with further decrease in frequency. A TABLE II. VALUES OF COEFFICIENTS GENERATED FOR fc =1000HZ
positive and negative gain corresponds to bass augmentation
and reduction respectively. The coefficients values for the SNo. GdB Gain a0 a1 b1
low pass filter are given by equation (10). The cut of 1 -12 0.25119 0.458808 -0.095723 0.445469
2 -11 0.28184 0.497945 -0.103888 0.398167
frequency of bass filter is set at 100Hz.
3 -10 0.31623 0.538916 -0.112436 0.348649
4 -9 0.35481 0.581563 -0.121333 0 .297104
III. RESULTS
5 -8 0.39811 0.625693 -0.13054 0.243767
IIR first order filter has the less complexity to design and 6 -7 0.44668 0.671077 -0.140009 0.188914
are not as powerful as the higher order filters but from the 7 -6 0.50119 0.717458 -0.149686 0.132856
designing point of view first order filters are preferred for 8 -5 0.56234 0.764554 -0.159511 0.075935
treble and bass control. We have a PCM file input for the 9 -4 0.63096 0.812062 -0.169423 0.018515
10 -3 0.70795 0.859672 -0.179356 -0.03903
post process algorithm to control audio Treble and Bass. We 11 -2 0.79433 0.907069 -0.189245 -0.09631
implemented the filter, gave the input PCM file (PCM stereo 12 -1 0.89125 0.953943 -0.199024 -0.15297
48000Hz) and calculated filter coefficient (a0,a1,b1) for 13 0 1 1 -0.208633 -0.20863
different values of the gain in dB as shown in the Table-I, II 14 1 1.12202 1.044965 -0.218014 -0.26298
and III. The filter coefficients are given for different gain 15 2 1.25893 1.088591 -0.227116 -0.31571
values from -12 dB to +12 dB. 16 3 1.41254 1.13066 -0.235893 -0.36655
The coefficient values for cut of frequencies 100Hz, 17 4 1.58489 1.170993 -0.244308 -0.4153
1000Hz and 10000Hz are calculated. IIR filtering is 18 5 1.77828 1.209445 -0.25233 -0.46178
19 6 1.99526 1.245907 -0.259937 -0.50585
processed after data is de-interleaved. PCM input file has the 20 7 2.23872 1.280308 -0.267115 -0.54742
data for both left and right channels (interleaved data). IIR 21 8 2.51189 1.31261 -0.273854 -0.58646
filter implementation uses previous values as context and 22 9 2.81838 1.342804 -0.280153 -0.62296
saves those vales for the further calculations. 23 10 3.16228 1.37091 -0.286017 -0.65693
24 11 3.54813 1.396969 -0.291454 -0.68842
25 12 3.98107 1.421044 -0.296477 -0.71752

562
 TABLE III. VALUES GENERATED FOR fc =10000HZ B. Treble Curve
S.No. GdB Gain a0 a1 b1 The lower corner frequency is 1000Hz and the curves are
1 -12 0.251189 0.458808 -0.095723 0.445469 plotted for the gain ranges from +12dB to -12dB. When the
2 -11 0.281838 0.497945 -0.103888 0.398167 gain is positive or negative the curve is called boost or cut
3 -10 0.316228 0.538916 -0.112436 0.348649 curve respectively. The curve for +12dB to -12dB boost and
4 -9 0.354813 0.581563 -0.121333 0.297104
cut are shown in Figure 4.
5 -8 0.398107 0.625693 -0.13054 0.243767
6 -7 0.446684 0.671077 -0.140009 0.188914
The treble filter utilizes both feed forward and feed back
7 -6 0.501187 0.717458 -0.149686 0.132856 loop in the flow chart. Output rises from lower to higher
8 -5 0.562341 0.764554 -0.159511 0.075935 level with increasing frequency, and response is suitable for
9 -4 0.630957 0.812062 -0.169423 0.018515 bass cut or treble boost. Shelf level difference is dB(f1/f2).
10 -3 0.707946 0.859672 -0.179356 -0.039028 The computation should be mechanized if more than a few
11 -2 0.794328 0.907069 -0.189245 -0.096313 curves are required.
12 -1 0.891251 0.953943 -0.199024 -0.152967 trebbc.m
13 0 1 1 -0.208633 -0.208633
14 1 1.122018 1.044965 -0.218014 -0.262979 10
15 2 1.258925 1.088591 -0.227116 -0.315706
16 3 1.412538 1.13066 -0.235893 -0.366553
17 4 1.584893 1.170993 -0.244308 -0.415301
18 5
5 1.778279 1.209445 -0.25233 -0.461775
19 6 1.995262 1.245907 -0.259937 -0.505845

amplitude dB
20 7 2.238721 1.280308 -0.267115 -0.547423
21 8 2.511886 1.31261 -0.273854 -0.586464 0
22 9 2.818383 1.34804 -0.280153 -0.622957
23 10 3.162278 1.37091 -0.286017 -0.656927
24 11 3.548134 1.396969 -0.291454 -0.688423 -5
25 12 3.981072 1.421044 -0.296477 -0.717521

-10
A. Bass Curve
The curves are plotted for the gain range from +12dB to - 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2
12dB. When the gain is positive the curve is called boost frequency Hz 4
x 10
curve, whereas for negative gains it is called cut curve. The
curve for +12dB to -12dB boost and cut are shown in Figure Figure 4. Treble plot
3.
The Bass filter utilizes both feed forward and feed back C. High Pass Filter Plot
loop of the flow chart. Output falls from higher to lower
level with increasing frequency, and response is suitable for The high pass filter utilizes the feed forward portion of
bass boost or treble cut. the filter to realize a zero at DC and the feedback portion to
BASS FILTER.m
create a pole that pulls the rising response down to a flat
response. The response rises from zero at DC to a shelf at f1.
10 This characteristic is useful for DC offset removal filters. In
the curve below, f1 was set to 10, 20, 50, 100, 200, 500, 1K,
2K, 5K, and 10 KHz.
5 high filter.m
0
amplitude dB

-2
0

-4
amplitude dB

-5
-6

-10
-8

1 2 3 4 5
10 10 10 10 10
frequency Hz -10

Figure 3. Bass Curve -12

1 2 3 4
10 10 10 10
frequency Hz

Figure 5. High pass Filter plot

563
D. Low Pass Filter Plot low filter.m
0
The low pass filter utilizes only the feedback portion of
the filter to realize a pole that sets f1, the corner frequency,
-2
appropriately. In the curves below, f1 was set to 10, 20, 50,
100, 200, 500, 1K, 2K, 5K, and 10KHz. By enabling a zero
at -0.12, this adds extra attenuation needed to more closely -4

match the continuous time filter at 10KHz and above.

amplitude dB
low filter.m -6
0

-8
-2

-10
-4
amplitude dB

-12
1 2 3 4
10 10 10 10
-6
frequency Hz

-8 Figure 8. Zero plot

-10 IV. CONCLUSION

The algorithm for bass and treble was implemented
-12 successfully on ARM. The coefficients for the IIR filter are
1 2 3 4
10 10 10 10
frequency Hz calculated by using the bilinear transform method. Bilinear
transform method is used because, there is no difference in
Figure 6. Low pass plot pole and zeros mapping of the S-domain filter to Z-domain
filter. For different gain values and cut of frequencies, the
curves were plotted for bass and treble control. Optimization
E. Pole and Zero Plot
is done for the real time application. The following results
The pole and zero plots are shown in Figure 7 and 8 are obtained after simulation Code size =7842 KB; RO- data
respectively. =958 KB; RW data=48 KB; ZI data =1242 KB. The results
low filter.m obtained after optimization is Code size =4228 KB; RO- data
0
=168 KB; RW data=28KB; ZI data =642 KB.

-2
REFERENCES
[1] R.H.Yang, S.B.Chiah and W.Y.Chan, “Design and Implementation
-4
of a Digital Audio Tone Control Unit Using an Efficient FIR Filter
Structure,” Proc. IEEE TENCON Digital Signal Processing
amplitude dB

Applications, IEEE Press, Nov., 1996, pp. 273-277 vol. 1, doi:

-6 10.1109/TENCON.1996.608815
[2] R.Yang, B.Liu and Y.C.Lim, "A new structure of sharp transition FIR
-8
filters using frequency-response masking," IEEE Trans. on Circuits
and Systems, vol. 35, pp. 955-966, Aug. 1988.
[3] Y.Lian and Y.C.Lim, "Linear-phase digital audio tone control using
-10 multiplication-free FIR filter,” Journal of Audio Eng. Soc., vol. 41,
no. 10, pp. 791-794, Oct. 1993.
[4] L.Yong, “Frequency-response masking approach for the synthesis of
-12
1 2 3 4
sharp linear phase digital filters," IEEE Trans. on Circuits and
10 10 10 10 Systems, vol. 33, pp.357-364, Apr. 1986.
frequency Hz
[5] Y.C. Lim and L. Yong, “Frequency-response masking approach for
Figure 7. Pole plot digital filter design: complexity reduction via masking filter
factorization” IEEE Trans. Circuits and Systems II: Analog and
Digital Signal Processing, vol. 41, pp. 518 – 525, Aug. 1994.
[6] Y.C. Lim, "Linear-phase digital audio tone control,” Journal of Audio
Eng. Soc., vol. 35, pp. 38-40, jan/feb. 1987.
[7] P. Soardo, “A Versatile Tone Control Circuit and Preamplifier,” IEEE
Trans. on Audio, vol. 11, pp. 195-201, Nov. 1963.

564
 2009 International Conference on Computer Engineering and Technology

NURBS Interpolation Method with Feedrate Correction in 3-axis CNC System

Chen Liangji Li Huiying

School of Mechatronics Engineering Resources & Economic Trade Department
Zhengzhou Institute of Aeronautical Industry Zhengzhou Institute of Aeronautical Industry
Management Management
Zhengzhou, China Zhengzhou, China
chenjiaxiclj@163.com Lihuiying.lhy@163.com

Abstract—A real-time Non-Uniform Rational B-Spline

(NURBS) curve directly interpolation method is presented for C(u)=x(u)i+y(u)j+z(u)k 0≤u≤1, (1)
computer numerical control (CNC) machines to achieve the
goal of high-speed and highly accurate machining. To the where u is an arbitrary parameter and i, j, k is the unit vector
feedrate fluctuation of the interpolation method, the feedrate of axis orientation of x, y, z respectively.
correction method is also proposed. To implement the
A p-degree NURBS curve with parameter u can be
proposed methods, Taylor expansion and numerical
defined as follows
calculation methods are used. The given example shows that
the feasibility of the methods applied to achieve high-speed and
high-effect machining. n

Keywords-CNC; NURBS; interpolation; feedrate T

∑N
i =0
i, p (u )Wi Pi
C (u ) = [ x(u ) y (u ) z (u )] = n
, (2)
I. INTRODUCTION ∑N
i =0
i, p (u )Wi
At present, if one performs a cutting task for molds with
a CNC machine tool, firstly, the workpieces have to be where Pi is the ith 3-D control point; Wi is the corresponding
designed using CAD, and then cutter contact (CC) path is weight factor of Pi; (n+1) is the number of control points;
converted to the cutter location (CL) path by CAM. In order Ni,p(u), B-spline basis function with degree of p, can be
to guide the CNC machine to perform cutting along the CL calculated by the following formula
path, the CL paths usually are divided into a set of line or
circular segments(also called NC codes). These segments
will approximate the original CL path to a desired accuracy ⎧ ⎧1 (ui ≤ u < ui +1 )
if sufficient numbers of segments are used. To obtain more ⎪ N i ,0 = ⎨
accurate approximation and to reduce the contour error, the ⎪ ⎩0 ( otherwise )
size of the NC code should be increased, but this approach ⎪⎪ u − ui ui + p +1 − u , (3)
could lead to several drawbacks[1~6]. ⎨ N i , p (u ) = N i , p −1 (u ) + N i +1, p −1 (u )
Based on the above observation and analysis, it is evident ⎪ ui+ p − ui ui + p +1 − ui +1

that accurate high-speed manufacturing is hard to achieve if ⎪0

the conventional line and circular interpolators are used in
CNC systems. Therefore, developing a new type of
interpolator for CNC machine tools is of importance.
Since NURBS uses fewer data to represent the designed
curves and if both the CAD/CAM and CNC systems adopt
NURBS as the geometric representation, a considerable
amount of time spent on transferring data between
CAD/CAM and CNC machine can be saved. Because of the
advantages of NURBS, a real-time NURBS interpolator is
proposed for machining curves (or surfaces). Consequently,
feedrate can be increased so that the machining time will be
reduced.
II. NURBS REPRESENTATION
A general form to describe a parametric curve in the 3-D
space can be expressed as
⎪ = 0 (prescribed)
⎪⎩ 0
where [ui ,..., ui + p +1 ] is the knot vector. More information
about NURBS can be found in [7].
III. REAL-TIME NURBS INTERPOLATION
The key to approximate a parametric curve in real time
(namely command generation) is that the segmentation
should be based on equal increments of sampling period Ts,
rather than equal increments of the parameter u.
Accordingly, a method is needed to determine successive
values of parameter u(at each sampling period) such that the
ratio between △ s(increment of curve length during each
sampling period)and T is equal to a desired profile V(t) in a
feedrate control problem. The procedures for determining
successive values of u are summarized as follows.

978-0-7695-3521-0/09 $25.00 © 2009 IEEE 565

DOI 10.1109/ICCET.2009.11
Considering the feedrate V(t) along the curve, C(u) in Eq.(2), n n
V(t) can be expressed as
d 2C (u ) ∑ Ni , p (u )Wi ∑ Ni'', p (u )Wi Pi
= i =1 i =0
−
dC (u ) ⎛ dC (u ) ⎞⎛ du ⎞ du 2 ⎛ n
⎞
2

⎜ ∑ N i , p (u )Wi ⎟
V (t ) = =⎜ ⎟⎜ ⎟ , (4)
dt ⎝ du ⎠⎝ dt ⎠ ⎝ i=0 ⎠
n n
hence, ∑N ''
i, p (u )Wi ∑ N i , p (u )Wi Pi
i =0 i =0
2
−
du V (t ) ⎛ n
⎞
=
dt dC (u )
, (5)
⎜ ∑ N i , p (u )Wi ⎟
⎝ i=0 ⎠
du n n n
2∑ N i', p (u )Wi ∑ N i , p (u )Wi ∑ N i', p (u )Wi Pi
Eq.(5) can be processed as follows using Taylor expansion i=0 i =0 i =0
−
formula 3
⎛ n
⎞
⎜ ∑ N i , p (u )Wi ⎟
V (tk ) ⋅ Ts ⎝ i=0 ⎠
uk +1 = uk + , (6) n n n
2∑ N i', p (u )Wi ∑ N i , p (u )Wi Pi ∑ N i', p (u )Wi
dC (u )
du u =uk i=0 i =0 i=0
, (9)
3
⎛ n
⎞
where uk and uk+1 denote the value of u at t=tk=kTs and ⎜ ∑ N i , p (u )Wi ⎟
t=tk+1=(k+1)Ts, respectively. And also, the second-order ⎝ i=0 ⎠
interpolation algorithm is
where the general algorithm for mth order derivative of Ni,p(u)
is
V (tk ) ⋅ Ts
uk +1 = uk + −
dC (u ) ⎡ N m −1 (u ) N im+1,−1p −1 (u ) ⎤
du u =uk N i(,mp ) (u ) = p ⎢ i , p −1 − ⎥ (10)
⎢⎣ ui + p −1 − ui ui + p − ui +1 ⎥⎦
⎛ dC (u ) d 2C (u ) ⎞
V (tk ) ⋅ Ts ⎜
2
⋅2

⎝ du du 2 ⎟⎠ u = uk
and
4
. (7)
dC (u ) dC (u )
2⋅ = ( x' )2 + ( y ' )2 + ( z ' )2 . (11)
du u = uk du
The first and second derivative of C(u) with u are where
obtained as
x = x(u ) , y = y (u ) , z = z (u ) ,
n
dx(u ) ' dy (u ) ' dz (u )
dC (u ) ∑N '
i, p (u )Wi Pi x' =
du
,y =
du
,z =
du
. (12)
= i =0
n
−
∑N
du
i, p (u )Wi
i =0
IV. FEEDRATE CORRECTION METHOD
n n Eq.(7) can be written as follows according to Eq.(5)
∑ Ni', p (u )Wi ∑ Ni , p (u)Wi Pi
i =0 i =0
2
(8) uk +1 = uk + Tsu ′(tk ) + Ts 2u ′′(tk ) / 2 . (13)
⎛ n
⎞
⎜ ∑ N i , p (u )Wi ⎟
⎝ i=0 ⎠ The following numerical method is used to solve Eq.(13).
The back ward difference approximation method of [8]
and can be employed here to compute u ′(tk ) and u ′′(tk ) ,
which is

566
 u′(tk ) = (uk − uk −1 ) / Ts (14) y
P2 P1
and

u′′(tk ) = (uk − 2uk −1 + uk − 2 ) / Ts2 . (15)

0
Substituting Eqs.(14) and (15) into Eq.(13) will yield an R1
initial value of uk+1

uk +1 =2.5uk-2uk-1+0.5uk-2 .
However, the feedrate Vk resulting from uk +1 can be
calculated as follows
Vk = C (uk +1 ) − C (uk ) / Ts .
In fact, Vk may not be accurate. Let
feedrate, so the error of feedrate is | Vk − Vk | .
*
Assumed that
uk +1 = auk +1 +(1-a)uk ,
(16) x
0 P0
Figure 1. Example NURBS curve with knot vector[0,0,0,1,1,1].
The sampling period Ts of CNC system is 1ms, and the
*
desired feedrate Vk is equal to 600mm/min which is given
by the NC code. Without the correction of federate, the
maximum deviation error can reach to 10mm/min. The
situation can be immediately avoided if the proposed method
Vk* be the desired were added to NURBS interpolator. Assumed that
ε=6mm/min, 3 mm/min and 0.6mm/min, the feedrate of each
sampling period can be calculated as shown in Figs.2, 3 and
4 ,respectively. From the three figures we can see that the
fluctuation of feedrate is greatly decreased with the proposed
(17) method.
605
Feedrate(mm/min)

600
where
595

590
a= Vk
*
/ [bVk + (1 − b)Vk* ] b ∈ (0,1] . 585
0.0 0.2 0.4 0.6 0.8 1.0
Parameter u

Figure 2. Feedrate when ε is equal to 6mm/min.

If feedrate error | Vk − Vk | is enough small, such that the
*

error is smaller than the given tolerance ε, then the iterative 604

calculation described in Eq.(17) will be stopped and let

Feedrate(mm/min)

602

600

uk +1 = uk +1 .
598
(18) 596

594
0.0 0.2 0.4 0.6 0.8 1.0
Parameter u

V. EXAMPLE Figure 3. Feedrate when ε is equal to 3mm/min.

Fig.1 shows the curve for machining—a quarter of circle 601
with radius of 10 mm represented in third-order NURBS
Feedrate(mm/min)

600.5
form, and its control points are P0(10,0,0), P1(10,10,0) and 600
P2(0,10,0) with weights 1, 0.707 and 1 respectively. The 599.5

NC Code is described as follows: 599

G06.2 P3 K0 X10 Y0 Z0 R1 F600 598.5

0.0 0.2 0.4 0.6 0.8 1.0
Parameter u
K0 X10 Y10 Z0 R0.707
K0 X0 Y10 Z0 R1 Figure 4. Feedrate when ε is equal to 0.6mm/min.
K1
K1 VI. CONCLUSIONS
K1
The real-time NURBS interpolator in CNC system is
presented in this paper. To the feedrate fluctuation of the
interpolation method, the feedrate correction method is
proposed at the same time. Compared with the conventional
method, the advantages of the proposed approaches include
the following:

567
 • Using NURBS in CNC system, the machine tool can
control the cutters cutting along a curve or a surface
not line or circular arc segments approximation.
• NURBS uses fewer parameters (control points, knots
vector and weights) to represent the designed curves
(or surfaces), so that CNC memory can be saved.
Less data transmission between CAD/CAM and
CNC machine results in faster manufacturing speed
such that the goal of high-speed machining can be
achieved.
• With the feedrate correction method, the influence
on machining result from the fluctuation of feedrate
can be reduced. Similarly, the purpose of high-speed
and high-effect machining can be reached.
ACKNOWLEDGMENT
This research is wholly supported by the Chinese
Aeronautical Science Foundation (No.2008ZE55003)
through Zhengzhou Institute of Aeronautical Industry
Management. The author thanks for their support for this
research.
568
REFERENCES
[1] G. Qiyi, R. Zhang, and B.Greenway, “Development and implemen-
tation of a NURBS curve motion interpolator,” Robotics and
Computer-Integrated Manufacturing, vol.14, 1998, pp. 27-36.
[2] M. Cheng, M. Tsai, and J. Kuo, “Real-time NURBS command
generators for CNC servo controllers,”International Journal of
Machine Tools & Manufacture, vol. 42, 2002, pp. 801-813.
[3] M. Tikhon, “NURBS interpolator for constant material removal rate
in open NC machine tools,” International Journal of Machine Tools &
Manufacture, vol.44, 2004, pp. 237-245.
[4] M. Tsai, C. Cheng, and M. Cheng,“A real-time NURBS surface
interpolator for precision three-axis CNC machining,” International
Journal of Machine Tools & Manufacture, vol.43, 2003, pp. 1217-
1227.
[5] B.Bahr, X. Xiao, and K. Krishnan, “A real-time scheme of cubic
parametric curve interpolations for CNC systems,” Computers in
Industry, vol. 45, 2001, pp. 309-317.
[6] X. Zhiming, C. Jincheng, and F. Zhengjin, “Performance evaluation
of a real-time interpolation algorithm for NURBS curves,”
International Journal of Advanced Manufacturing Technology,
vol.20, 2002, pp. 270-276.
[7] L. Piegl, “On NURBS: a survey,” IEEE Computer Graphics &
Application, vol.11,no.1, 1991, pp. 55-71.
[8] S. Nakamura, “Applied numerical methods in C,” Prentice-Hall
International, Inc, USA, 1993.
 2009 International Conference on Computer Engineering and Technology
Implementation Technique of Unrestricted LL Action Grammar*
ZHANG Jing , JIN Ying
College of Computer Science and Technique, Jilin University, China, 130012
Zhangjing99@jlu.edu.cn, jinying@jlu.edu.cn
Abstract
Action Grammar is a grammar with action
symbols appearing in the right part of production rules,
which can be used to implement semantic analyzer
automatically. An action symbol represents the name
of a semantic routine. Implementation of action
grammar relies on specific parsing methods, such as
LL(K), recursive-descent or LR, and so on. However,
these parsing methods have their own restrictions on
grammars, therefore, in order to accomplish these
parsing methods certain equivalent transformations
have been done to make the grammars meet the
restrictions. This paper has proposed an
implementation technique of LL action grammar,
which requires no transformation to the original
grammar. Our approach includes following steps:
parsing the input string with back-tracking and
producing the parsing paths for each non-terminal
symbol, then constructing LL parsing table with the
parsing paths obtained, finally conducting semantic
analysis with LL parsing table. The main contribution
of this paper is that non-backtracking semantic
analyzer has been achieved without making any
transformations to the grammar.
1. Introduction
Software system can be divided into goal-oriented
and program-oriented. Program-oriented software
systems include interpreter, compiler, program
transformer, program optimizer, program analyst, and
so on, all complicate actions of these systems acted on
programs which were string data described by context-
free grammar or Backus-Nour Form notation[1].
Thereby, we must first transform the program into
some internal data structure, such as program tree. In
order to implement the transformation, the program
must do syntax analysis. However, the exsiting
frequently-used parsing methods all have
corresponding restrictions, and can only be used after
transfoming the grammar to satisfy their preconditions.
* This paper is supported by National Natural Science Foundation
for the Youth of China (No.606030313).
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.102
Unfortunately, there are some grammars for which no
amount of alternation will yield a right grammar, and
some transformations often followed by awful side-
effects: losing of readability and reliability[2]. For
example, the well-known and widely used Yacc, a
generator of LALR parser, have a restriction that there
should be no shift/reduce conflicts when generating the
LALR state machine. How to reslove the conflictions
is such a difficult work that there is no universal rules
and general algorithms and the phenomena that the old
one disappeared, and a new one springs up occur
frequently in resolving conflicts process[3,4]. A simple
method to resolve the restrictions problem of grammar
is using back-tracking technique, but it is more
difficult for semantics analysis[5].
During our implementation of ATLAS language[6],
hundreds of conflicts were found due to the great
amount of productions for the language, the conflicts
have never been reoslved by hand. A general efficient
tool was in need, which motivated this paper. The key
idea of our method is getting the parsing paths for each
non-terminal symbol after a backtracking parsing
firstly, then achieving semantics analysis using the
parsing paths rather than backtracking. A great amount
of transformations(may be endless) have been avoided
and any grammar can be processed by our method.
This paper is structured as follows: Action grammar
was introduced in section 2. Section 3 describes what
is a parsing path and how to use it to parse. A parsing
algorithm with backtracking, data structures needed
and a simple example were proposed in section 4.
Conclusions are made and some key issues are
discussed in Section 5.
2. Action grammar
Action grammar is a grammar with action symbols
inserted into the right part of its productions, each
action symbol represents the name of a semantics
routine. The goal of action grammar is to implement
semantic analyzer automatically. User can put semantic
routines into a parser so that semantic analysis can be
accomplished along with parsing.
569
 Since implementation of action grammars relies on L→ BB L→ < I NI T >B B
parsing, action grammar could be implemented by B→ a B B → a < AD Da > B
LL(K), recursive-descent, LR, LALR, and so on. The B→ b B B → b < ADDb >B
well-known Yacc is an implementation system of B→ c B → c < AD Dc >
action grammar based on LALR. Since each parsing
method has its own restrictions, each method is (a) (b)
applicable to specific group of grammars. In order to Fig.1.Grammar GL and its action grammar
use certain method, some transformation have to be Where INIT, ADDa, ADDb, ADDc all are semantics
made to the grammar if it is possible. Unfortunatelly, routines' name, which identifies the following code
not all transformations are effective and some separately:
transformations are rather inefficient. In this paper an
implementation of LL action grammar using two-pass •I NI T :{ ia: =0 ; ib : =0 ;i c : = 0 }
analysis has been given. • ADD a : { i a:= ia +1 }
A LL predictive parser has an input buffer, a stack, a • ADDb :{ ib : = ib +1 }
parsing table. The input buffer contains the string to be • ADD c : { i c:= ic +1 }
parsed, followed by $, a symbol used as a right end 3. Parsing path
marker to indicate the end of the input string. The stack
contains a sequence of grammar symbols with $ on the Suppose A is a non-terminal, and we have grammar
bottom, indicating the bottom of the stack. The parsing GA:
table is a two-dimensional array LL[A,a], where A is a [1] A → a
non-terminal, and a is a terminal. Each parsing action [2] A → a A a
was determined by A, the symbol on top of the stack, for a given sentence ''aaaaa'', the process of top-
and a, the current input symbol, LL[A, a], the table down parsing is as follows:
element related to A and a.
(A, aaaaa) →[1] (a, aaaaa) →#(, aaaa) →error
z If A = a = $, the parser pops A off the stack and
advances the input pointer to the next input As it was shown, production 1 of A was failed, the
symbol. parser should backtrack to the beginning, and try
z If A = a ≠ $, the parser halts and announces another production 2 of A, the continue process was:
successful completion of parsing. (A,aaaaa) →[2](aAa, aaaaa) →#(Aa, aaaa)
z If A is a non-terminal, the program consults entry
LL[A,a] of the parsing table LL. This entry will →[2](aAaa, aaaa)→#(Aaa, aaa) →[1](aaa, aaa)
be either an A-production of the grammar or an →#(aa, aa) →#(a, a)→# (,) →#
error entry. If, for example, LL[A,a] = p, p is the
number of production A → XYZ, the parser where [n] denotes production number used, #
replaces A on top of the stack by ZYX(with X on represents match operation of terminals. We call
top). If LL[A,a]= error, the parser calls an error <2,2,1> as parsing path for A.
recovery routine. The above grammar was not a LL(1) grammar,
because two productions of A started by the same
As to the action grammar, action symbols are added terminal a, so no LL parsing table could be constructed.
to the stack besides the normal syntax symbols. But with backtracking parsing and collecting of paths
Definition of LL parsing table doesn't change, but information, a new LL parsing table could be
some changes should be made to the meaning of constructed. For the grammar GA:
LL[A,a]. That is, except the above three possibilities, LL-ParseTab[ A, a] = < 2,2,1 >
the forth situation should be handled: So the semantics analysis could proceed determinately
z If A is an action symbol, the program executes without backtracking.
the routine represented by A, then A is popped Unrestricted LL action grammar was defined as
out after the routine had been finished. follows:
< aα, aμ , τ >→# < α , μ, τ >
For example, for the grammar shown in Fig.1.(a), a < Aα , aμ, τ >→[i]<riα, aμ, τ' >
program is needed to check whether any given string is i = first(τ(A)), ٛ τ'= τ[tail(τ(A)/A]
accepted by this grammar and count the numbers a, b,
< @α, μ, τ >→@ <α, μ, τ>
c occurred in the string separately. The LL action
where τ represents the parsing path list of nonterminals,
grammar is shown in Fig.1(b).
ri is the right part of the production i.

570
4. Parsing algorithm
Top-down parsing can be viewed as an attempt to
find a leftmost derivation for an input string.
Equivalently, it can be viewed as an attempt to
construct a parse tree for the input starting from the
root and creating the nodes of the parse tree in preorder.
Suppose there is a production A→X1 X2... Xn,
then the corresponding syntax tree is as follows:
A sentence.
x1 x2 ...... xn
A is called the father node of Xi, Xi is called the ith
son of A. If the name of the syntax symbol on one
node is a terminal, node is called terminal node, and if
the name of the syntax symbol on one node is a non-
terminal, node is called non-terminal node.
If there is no ambiguity of the language defined by
a grammar, then any legal sentence of this language
has a corresponding syntax tree. So the focus of our
method is how to build a syntax tree for a given
sentence.
4.1 Data structures
In order to implement LL action grammar, we have
defined following data structures.
(1)Syntax tree nodes
Node : NodeNum × Symbol × Product × ProductIndex
× FathNode × BroNode × SonNode
where Symbol ∈Terminal ∪ NonTerminal
NodeNum, Product, ProductIndex, FathNode,
BroNode, SonNode ∈ Integer,which represents the
current node number c, the current production p, the ith
symbol of production p, the input pointer, the father
node, the late brother node, the late son node separately.
So, a node is noted as (c, X, p, i, j, F, B, S).
In order to facilitate backtracking, node sequence
generated during the parsing were saved as stack.
(2)Productions
Production: NonTerminal × SymbolList
SymbolList : (Symbol, /, $)*
where "/", "$" are endmarkers of one production and
of all productions for each nonterminal respectively.
For example, for a grammar GA:
A → Ba | aA
B → b | bB
G was saved as :
A B a / a A / $
B b / b B / $ z c is a succ-backtracking node:
Specially, we use | p | to denote the length of the right
string of production p; | X | denotes the last production
of X. It is obviously that : if i > | p |, then production p
is successful(maybe temparily); if p > | X |, then all the
571
productions of X have been tried, node X is failed and
should be backtracked to its brother or father node.
We use pi to denote the ith symbol of the right
part of production p; First(X) to denote the first
production of non-terminal X.
(3)Input
Input is a character array to represent any given
sentence of the grammar, Input is the name of the array,
Input[i] represents the ith sysmbol of the sentence. If
Input[i] = "$"，the parser has reached the end of the
(4)Parsing paths
Parsing paths for each non-terminal can be gained
by searching the node stack generated during the
parsing. For example, if the node stack is shown as:
（1, X1, p1, i1, F1, B1, S1） bottom
（2, X2, p2, i2, F2, B2, S2）
（n Xn, pn, in, Fn, Bn, Sn）
top
For any non-terminal Y, Y's parsing path Yτ can be
determined by the following simple searching:
from bottom to the top,
repeat if Xk = Y then Yτ = Yτ •[pk].
where the intial Yτ = ε，"•" represents concatenation
operation of list structure.
4.2 Algorithm
The parse tree node that is being handled during the
parse tree's construction is called as the current node.
The parse tree nodes can be devided into four classes
according to different situations: new-terminal nodes,
new-nonterminal nodes, success-backtracking nodes
and fail-backtracking nodes. Suppose we use c to
represents the current node, then the key part of
constructing parsing tree backtrackingly is as follows:
z c is a new-terminal node:
if Xc = Input[j] then j = j+1; c = F ;
c.Flag = succ-backtracking node;
else delete c from the nodestack;
F.S = B; c = F;
c.Flag = fail-backtracking node;
z c is a new-nonterminal node:
oldX = Xc; c = c+1; Xc = First (oldX) ;
if Xc ∈ terminal then c.Flag = new-terminal node;
else c.Flag = new-nonterminal node;
add c to the nodestack;
if (i > | p |) & (p > | X |) then
if c is root then successexit ;
else c = F; c.Flag = succ-backtracking node
else if (i > | p |) then
 c = c + 1; Xc = pi +1; there is another alternative for A that we have not tried
if Xc ∈ terminal then but that might produce a match.
c.Flag = new-terminal node; In going back to A, we must reset the input pointer
else c.Flag = new-nonterminal node; to position 2, the position it had when we first came to
add c to the nodestack; A. We now try the second alternative for A to obtain
the tree of Fig.2.(f). The leaf a matches the second
z c is a fail-backtracking node:
symbol of ω and the leaf d matches the third symbol.
if (i > | p |) & (p > | X |) then
Since we have produced a parse tree for ω, we halt and
if c is root then failexit ;
announce successful completion of parsing.
else delete c from the nodestack;
So the parsing path of S is [1] and A is [3]. Then
F.S = B; c = F;
we can use the gotten parsing path to analysis
c.Flag =fail-backtracking node
semantics further whose process was omitted.
else if (i > | p |) then
if S ≠ 0 then c = S;
if Xc ∈ terminal then j = j-1; c = F; 5. Summary
c.Flag = fail-backtracking node;
else i = i - 1; Simplicity and facility are the goals for the
c.Flag = fail-backtracking node; application software. Even for system software such as
compiler, simplification of some phases is welcomed
4.3 A simple example during its working process. Our experience in
constructing the compiler for ATLAS language had
Consider the grammar Gs:
taught us that even for the most professional lingual
S → c A d [1]
specialist, transformation of grammar for an big
A → ab [2]| a [3] enough language is almost an impossible work since
and the input string ω = cad. To construct a parse tree there was no universal rules for such transformations.
for this string top-down, we initially create a tree However, any grammar could be used to LL parsing
consisting of a single node labeled S. An input pointer without any transformations using the method
points to c, the first symbol of ω. We then use the first proposed in this paper.
production for S to expand the tree and obtain the tree How to deal with ε-productions was not mentioned
of Fig.2.(b). in this paper just to be on the simple side. In fact, ε
S symbol can be considered as the wildcard character,
S S that is ε matches with any terminal. The only
c difference between them is that the input pointer
S c c
a doesn't move and no parse tree node generated when ε
symbol was met with.
(a) (b) (c) (d)
References
S S S
[1] A.V. Aho, R.Sethi and J.D. Ullman, Complilers:
c c c d Principles, Techniques, and Tools, Addison Wesley, New
Jersey, 1986.
a b a a [2] Jay, Earley, "An Efficient Context-Free Parsing
(f) Algorithm", Communications of the ACM, Vol. 13(2), Feb.
(e) (g)
1970. pp. 94-102.
Fig.2. Steps in building parsing path [3] Standard Test Language for All Systems-
Common/Abbreviated Test Language for All
The leaf, labeled c, matches the first symbol of ω, Systems(C/ATLAS). "IEEE Std 716-1995". 1995.
so we now advance the input pointer to a, the second [4] LIU Lei and GUO De-Gui, "The Implementation
Technology of Test Language ATLAS",Chinese Journal of
symbol of ω, and generate the next leaf, labeled A. We Scientific Instrument, Vol.25(S2), 2004.pp.16-19.
can then expand A using the first alternative for A to [5] D.E.Knuth, "Semantics of context-free languages",
obtain the tree of Fig.2.(d). We now have a match for Theory of Computing Systems, Vol.2(2), 1968 .pp.127-145.
the second input symbol so we advance the input [6] LI Ai-Ping, WANG Jia-Li and LIANG Hong-Yu, "A
pointer to d, the third input symbol, and compare d Study on YACC Confilct Resolving during ATLAS Syntax
againt the next leaf, labeled b. Since b does not match Analysis", Journal of Electrionic Measurement and
d, we report failure and go back to A to see whether Instrument, Vol.20(4), 2006. pp.97-101.

572
 2009 International Conference on Computer Engineering and Technology
Improving BER using RD Code for Spectral Amplitude Coding Optical CDMA etwork
Hilal Adnan Fadhil1, S. A. Aljunid2, R.Badlishah Ahmad3.
1,2,3
School of Computer and Communication, University Malaysia Perlis, 02000 Kuala Perlis,
Perlis, MALAYSIA
*
Corresponding author: eng_hilal_adnan@yahoo.com
Abstract- A new code for spectral-amplitude coding optical code-
division multiple-access system is proposed called Random
diagonal (RD) code. This code is constructed using code segment
and data segment. One of the important properties of this code is
that the cross correlation at data segment is always zero, which
means that Phase Intensity Induced "oise (PII") is reduced. For
the performance analysis, the effects of phase-induced intensity
noise, shot noise, and thermal noise are considered
simultaneously. Bit-error rate (BER) performance is compared
with Hadamard and Modified Frequency Hopping (MFH) codes.
It is shown that the system using this new code matrices not only
suppress PII", but also allows larger number of active users
compare with other codes. Simulation results shown that using
point to point transmission with three encoded channels, RD code
has better BER performance than other codes, also its found that
at 0 dbm PII" noise are 10-10 and 10-11 for RD and MFH
respectively.
Keywords- OCDMA, PII, MFH, BER.
I. INTRODUCTION
Fiber optic code division multiple access (CDMA) techniques
provide flexible solution for asynchronous high-capacity
communication in the field of digital local area networks. The
early incoherent optical CDMA (OCDMA) systems used
pseudo-orthogonal sequences to encode signals in the time
domain, but the codes were long and multiple access
interference (MAI) limited the number of simultaneous users.
Thus, spectral-amplitude-coding (SAC) of OCDMA systems
was proposed because of its ability to eliminate the influence
of MAI and to preserve the quasi-orthogonality between
network users [1, 2]. MAI exists in the OCDMA systems when
multiple users accessing the same medium using the same time
and frequencies for transmitting concurrent data streams. In
SAC-OCDMA systems, each user is assigned with a sequence
code that serves as its address. An OCDMA user modulates its
code (address) with each data bit and asynchronously initiates
transmission. Thus, this modifies its spectrum appearance, in a
way recognizable only by the intended receiver. Otherwise
only noise like bursts is observed [3]. Many code have been
used for OCDMA such as Modified Frequency Hopping
(MFH), prime codes [2], [4] and Hadamard codes [2], [5], etc.
However, since their in-phase cross-correlations were
relatively high, as compared with code weight, the phase-
induced intensity noise (PIIN) degraded system performance
severely. PIIN originated from the intensity noise caused by
the phase noise of the incoherent light fields that were mixed
and incident upon a photodiode [5]. The new proposed code is
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.57
used to improve the signal-to PIIN noise ratio, this study
introduce a code with zero cross correlation at data segment,
the systems that adopted this code had improved BER
performance and the encoding/decoding devices could be
implemented via fiber-Bragg grating (FBG) easily. However,
as the lengths of code words used in the frequency domain
became larger, the physical length of the FBG array became
impractical, and introduced unequal insertion loss for different
wavelength chips. In terms of code length a good set of code is
to obtain the maximum number of codes with maximum
weight and minimum length with the best possible
autocorrelation and cross-correlation properties. A code with a
larger size should give better BER performance. Hence,
unipolar {0,1} codes which have low out-of-phase
autocorrelation and cross-correlation values are used. Short
code length limit the addressing flexibility of the codes, while
long code length are considered disadvantage in
implementation, since either very wide –bandwidth source or
very narrow filter bandwidth are required, RD codes exists for
practical code length that are neither too short nor too long. .
This paper is organized as follows: in Section II we will discus
how the code is developed theoretically and also its properties;
simulation results for different OCDMA codes are given in
section III, and the conclusion is presented in section VI.
II. RANDOM DIAGONAL CODE PROPERTIES
Many OCDMA strategies have been proposed [4,5], where
one of the major concerns of designing RD code sequences is
MAI because the performance of such system is usually
interference limited. However, due to the characteristics of
optical signals, in the direct detection of OCDMA system the
signature sequence consists of unipolar (0,1) sequences. We
denote a code by N, W, λ where N is the code length, W is the
code weight, and λ is in-phase cross correlation. Let us define

λ = ∑ x i y i as the inphase cross correlation of two different
i =1
sequences X=(x1,x2,….x) and Y=(y1,y2,….y). When λ =1, it is
considered that the code possesses ideal inphase cross
correlation. The design of this new code can be preformed by
dividing the code sequence into two groups, which are code
segment and data segment.
Step1, data segment: let the elements in this group contain only
one “1” to keep cross correlation zero at data level ( λ = 0 ).
This property is represented by the matrix (K × K) where K
will represent number of users. These matrices have binary
573
coefficient and a basic Zero cross code (weight=1) is defined
as [Y1]. For example, for three users (K=3), Y1 can be 0 1 1 0 1
1 1 0 1 0 
expressed as   
1 0 1 and 0 1
0 0 1     
[Y1 ] = 0 1 0  0 1 1  1 0
[B](j) = [M ]( j) =
1 1 0 0 1
1 0 0    
: : :  : : 
: : :  : : 
where [Y1] –consists of (K × K) identity matrices.    
Notice, for the above expression the cross correlation between aj1 aj2 aj3  a j1 a j 2 
any two rows is always zero.

Step2, code segment: the representation of this matrix can be where j represents the value for K-th user (j=1,2…K), and the
expressed as follows for W=4: value of aj is either zero or one. The weights for code part for
0 1 1 1 0 both matrix [M], [B] are equal to W-1, so the total combination
of code is represented as (K × ) where K=3, =8, as given by
[Y2 ] = 1 1 0 0 1
1 0 1 1 0 [Z1], [Z1]=[Y1 Y2] ‫׀‬
where [Y2] consists of two parts -weight matrix part [W],and  0 0 1 0 1 1 1 0
basics matrix part [B]. Basic part [B] can be expressed as [ Z 1 ] = 0 1 0 1 1 0 0 1
1 0 0 1 0 1 1 0
0 1 1 
[B ] = 1 1 0
From the above basic matrix Z, determine the number of users
1 0 1
(K) and the code length (), as given by (K × ) matrix. Notice
1 0 that the code weight of each row is equal to 4, and the relation
0 1 
 between N and K for this case (W=4) can be expressed as
and weight part called [M] matrix =  1 0  which is

responsible for increasing number of weights. Let i= (W-3) and =K+5 (2)
1 0 
Mi =  0 1  where i represents number of Mi matrix on As a result we can find that for W=5,6, and 7 code, word
 
 1 0  length  can be expressed as K+7, K+9 and K+11
respectively. As a result the general equation describing
matrix [M], given by
number of users K, code length  and code weight W is given
as
[M]= M 1 M 2 M 3 .......M i (1)
 = K+ 2W -3 (3)
For example, if W=5, from Eq.(1) i=2, so that [M]= M 1 M 2
Finally, using the properties of RD code, the SNR for RD code
is derived mathematically as follows:
1 0 1 0 2
[M ] = 0 1 0 1
.  2ℜPsrW 
 
1 0 1 0 SR =   
2eBWPsr ℜ Bℜ 2 PsrW K
+ (K − 1 + W ) + 4 K BTn B
Notice that to increase the number of users simultaneously  2  2 ∆V RL
with the increase of code word length we can just repeat each (4)
row on both Matrixes [M] and [B]. For K-th user matrix [M]
and [B] can be expressed as where Psr is the effective power of a broadband source at the
receiver. The bit-error rate (BER) can be calculated using
Gaussian approximation [8]

1 SR (5)
BER = erfc
2 8

574
Table 1 shows the code lengths required by the different codes
to support a number of users, K, of 30, RD code offers better
performance than other codes in term of code length for same
number of users . Short code length limit the addressing
flexibility of the codes, while long code length are considered
disadvantage in implementation, since either very wide –
bandwidth source or very narrow filter bandwidth are required,
RD codes exists for practical code length that are neither too
short nor too long.

Table 1 Comparison of RD, MFH, OOC, Hadamard and Prime

Codes for the same number of users, K=30.

Code No. of user Weight Code

K W length N
(a)
OCC 30 4 364
Prime code 30 31 961
Hadamard 30 16 32
MFH 30 7 42
RD code 30 3 33

III. OCDMA SYSTEM DESCRIPTION

Figure (1) shows the setup of the proof-of- principle

simulation for the proposed scheme. The performances of RD,
MFH, and Hadamard codes are simulated by using the
simulation software OptiSystem Version 6.0. A simple
schematic block diagram consists of two users, as illustrated in
(b)
Fig. 1. Each chip has a spectral width of 0.8 nm. The tests were
carried out at a rate of 10 Gb/s for 20-km distance with the
ITU-T G.652 standard single-mode optical fiber (SMF). All the Fig. 1 Simulation setup of the proposed scheme a) Transmitter b) Receiver
attenuation α (i.e., 0.25 dB/km), dispersion (i.e., 18 ps/nm
km), and nonlinear effects were activated and specified
according to the typical industry values to simulate the real
environment as close as possible. The performances of the
system were characterized by referring to the bit-error rate
(BER). As shown in Fig.1 after transmission, we used a Fibre
Bragg grating (FBG) spectral phase decoder operates to decode
the code at data level. The decoded signal was decoded by a (a)
photo-detector (PD) followed by a 0.75 GHz low-pass-filter
(LPF) and error detector respectively. The transmitted power
used was 0 dBm out of the broadband source. The noise
generated at the receivers was set to be random and totally
uncorrelated. The dark current value was 5 nA, and the thermal
noise coefficient was 1.8 × 10-23 W/Hz for each of the
photodetectors. The eye pattern diagrams for RD, Hadamard
and MFH codes are shown in Fig.2 respectively. The eyes
diagram shown in Fig.2 clearly depict that the RD code system
gives better performance, having a larger eye opening. The
corresponding simulated BER for RD, Hadamard, and MFH
codes systems are shown in Fig.2. The vertical distance
between the top of the eye opening and maximum signal level (a)
gives the degree of distortion. The more the eye closes, the
more difficult it is to distinguish between 1s and 0s in the
signal. The height of the eye opening at the specified sampling
time shows the noise margin or immunity to noise.

575
 (b)
(c)
( c)
Fig. 2 Eye diagram of (a) one of the RD channels, (b) one of the Hadamard
channels, and (c) one of the MFH channels, at 10 G bit/s.
Using Eq.4, Fig.3 depicts the relation between the number of
users and the BER, for RD, MFH, and Hadamard codes, where
they have been plotted for different values of K (number of
users). This figure clearly shows that RD code results in a
much better performance, i.e (smaller BER) than MFH code
and Hadamrad code schemes. This is evident from the fact that
RD code has a zero cross-correlation while Hadamard code has
increasing value of cross-correlation as the number of users
increases. Note also that the calculated BER for RD was
achieved for W=7 while for MFH and Hadamard codes, the
calculated BER was for W=14, and W=64 respectively.
576
Fig. 3: BER versus number of users for RD, MFH and Hadamard code using
Psr =0 dBm, B= 80MHz at the operating wavelength of 1550nm
In OCDMA systems, phase induced intensity noise (PIIN) is
related to multiple access interference (MAI) due to the
overlapping of spectra from different users. Here, we will
analyze the relations between PIIN noise and received power.
We have already shown previously that MAI can be almost
reduced because of zero cross-correlation at data segment.
Thus the effect of MAI is not elaborated further. Figure 4
shows the relations between PIIN noise and received power
(Psr). The values of B, W, and ∆V are fixed (ie. B = 311
MHz, W = 5 and ∆ V = 3.75 THz) but Psr varied from -30
dBm to 20 dBm. When the received power increases the PIIN
noise for MFH and RD codes increases linearly. The PIIN
noise of RD codes family is less compared to that of MFH
code. As shown in Fig.4 the PIIN noise can be effectively
suppressed by using RD code family. This is because of the
superior code properties of RD code family, such as no cross-
correlation at data level. Also it has been shown that, when the
numbers of users are increased, the PIIN noise increased as
well.
 [5] F. R. K. Chung et al., “Optical orthogonal codes: design, analysis, and
applications,” IEEE Trans. Inform. Technol., vol. 35, pp. 595–604, May 1989

[6] Goursaud C., Julien-Vergonjanne A., Aupetit-Berthelemot C., Cances J.P.

and Dumas J.M., "Study of Parallel Interference Cancellation Structures for
Incoherent DSOCDMA Systems”, Asia-Pacific Conference on
Communication, Oct 2005

[7] S. Zahedi and J. A. Salehi, “Analytical comparison of various fiber optic

CDMA receiver structures, “J.Lightw. Technol., Vol 18, no. 5, pp.1718-1727,
Dec.2000

[8]- J. W. Goodman, Statistical Optics. New York: Wiley 1995

.4 PIIN versus Pr for versus OCDMA codes

IV. CONCLUSIONS

In this paper, we proposed a new code can effectively suppress

the PIIN, with a fixed in-phase cross correlation value of 1 at
code level and zero cross correlation at data level is presented.
The properties of this code are described and discussed with
the related equations. Based on the equations, the results of
system performance are presented. To conclude, the
advantages of the code can be summarized as follows: (1)
shorter code length; (2) no cross-correlation in data level which
minimized λ and reduced PIIN (Phase Induced Intensity
Noise); (3) data level can be replaced with any type of codes;
(4) more overlapping chips will result in more crosstalk; and
(5) flexibility in choosing , K parameters over other codes
like MFH code. The performance of the system with the
proposed RD code is analyzed by taking into account the effect
of the intensity noise, shot, and thermal noise sources. It is
found that RD code is the best in term of BER compared to
MFH code and Hadamard code. RD code can be an excellent
candidate for use in next –generation optical access networks.

REFERENCES

[1] Kavehrad, M., and D. Zaccarin. 1995. Optical Code-Division-Multiplexed

System Based on Spectral Encoding of Noncoherent Sources. Journal
Lightwave Technology. 13: 534-545.

[2] Yang, C. C., J. F. Huang, and S. P. Tseng. 2004. Optical CDMA Network
Codecs Structured with M-Sequence Codes over Waveguide-Grating
Routers. IEEE Photonics Technology Letters. 16 (2): 641-643.

[3] Z. Wei and H. Ghafouri-Shiraz, “Proposal of a novel code for spectral

amplitude-coding optical CDMA systems,” IEEE Photon. Technol.Lett., vol.
14, pp. 414–416, Mar. 2002.

[4] Z. Wei et al., “Modified quadratic congruence codes for fiber Bragg-
grating-based spectral-amplitude-coding optical CDMA systems,” J.
Lightwave Technol., vol. 19, pp. 1274–1281, Sept. 2001

577
 2009 International Conference on Computer Engineering and Technology
USS-TDMA:Self-stabilizing TDMA Algorithm for Underwater
Wireless Sensor Network
Guo Zhongwen Li Zhengbao
Department of Computer Science,
College of Information Science and Engineering Ocean University of China
Qingdao,China
guozhw@ouc.edu.cn lizhengbao@ouc.edu.cn
Abstract—Underwater wireless Sensor network is often
constrained by limited power, limited communication range
and high bit error rate (BER). We focus on designing a
TDMA service for a cube-based underwater wireless sensor
network. Such network is readily found in many
applications in the area of monitoring, hazard detection, and
so on. In this paper, we propose an underwater self-
stabilizing TDMA Algorithm for assigning time slots to
sensors. We present the TDMA algorithm customized for
different communication patterns, namely, broadcast,
convergecast and local gossip, which occur commonly in
sensor networks. The algorithm is self-stabilizing and energy
efficiency. Meanwhile, with the localization service, the
algorithm can be used in mobile sensor networks.
Keywords- Underwater Wireless Sensor Network, TDMA,
Broadcast, Convergecast, Local Gossip
I. INTRODUCTION
Underwater Acoustic Wireless Sensor Networks
(UAWSN) can be employed by a vast range of
applications, e.g., environment monitoring, data
gathering. So UAWSN has become a hotspot in wireless
sensor networks researching resent years. However, many
challenging technical issues are existent [2], such as high
latency and low bandwidth, because of using sound wave
for communication. Moreover, changeful environment
and ubiquitous noise in underwater environment bring
high bit error rate (BER) and exquisite latency changing.
All of these challenges made traditional protocols in
wireless ad-hoc networks unavailable in underwater
wireless sensor network.
Designing of medium access control (MAC) protocols
is a distinct challenge for UAWSN. The MAC protocols
for underwater acoustic networks are mainly proposed
from Aloha, CSMA, MACA and CDMA [3, 4, 5, 6, 7, 8].
But they have some shortcomings. Such as, RTS-CTS
may improve end to end latency greatly; the efficiency of
Aloha is not very high and for TDMA, synchronization of
all nodes is an intractable matter.
In this paper we present a new TDMA algorithm for
underwater wireless sensor network based on [1, 9].The
algorithm is a collision-freedom one and fairly utilizes the
bandwidth of acoustic channel among different sensors.
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 578
DOI 10.1109/ICCET.2009.54
Hong Feng
hongfeng@ouc.edu.cn
The base station manages the distribution of time slot and
keeps synchronization by sending diffusion message.
The rest of the paper is organized as follows. In section
Ⅱ, we discuss the network model and some assumptions.
In section Ⅲ, we give the details of our algorithm, present
the new TDMA algorithm for three communication
patterns. In section Ⅳ, we introduce the simulations of
the algorithm and the simulation results. We show that
our protocol is available in underwater environment and
performs better than other protocols, especially in high
traffic load case. Finally, we make conclusion and future
work in section Ⅴ.
II. MODEL AND ASSUMPTIONS
In this section, we propose the network models and
some assumptions in our algorithm. We assume that
network is plotted into some cubes, the sensors are
arranged in the vertexes of cubes and each sensor knows
its location. The base station is located in the left-top
position (at location (0,0,0) ). The structure of the network
is shown in Fig. 1. Further we assume each sensor has a
communication range and interference range. We
consider the communication distance is 1 and the
interface range is k ( k ≥ 1 away). Also, we assume that
the sensors are aware of their communication range and
interference range. The length of cube side equals to the
communication distance.
Figure 1. The structure of the network model
Now, based on these assumptions, given two sensors, i
and j, if i is in the interference range of j but i is not in the
communication range of j then i receives messages sent
by j with a low probability. However, if i receives another
message from a sensor n while j is sending a message, it
is possible that a collision between these two messages
can prevent i from receiving either of those messages.
Based on the definition of the interference range, it
follows that it is at least equal to the communication
range. As define in [9] the concept of collision domain.
The collision domain of sensor j includes the sensors that
are in the communication range of j and the sensors that
interfere with the sensors in the communication range of j.
Hence, if two sensors i and j are allotted the same time
slot then j should not be present in the collision domain of
i and i should not be present in the collision domain of j.
In this paper, we consider three commonly occurring
communication patterns: broadcast, convergecast, and
local gossip. In broadcast, a message is sent to all the
sensors in the network. Based on the internal classification
technique mentioned earlier, we consider the
communication pattern, local gossip, where a sensor sends
a message to its neighboring sensors within some distance.
And, based on the external classification technique
mentioned earlier, we consider the communication pattern,
convergecast, where a group of sensors send a message to
a particular sensor such as the base station.
III. THE ALGORITHM FOR TDMA SERVICE
We consider three common communication patterns and
give three different assignments of time slots. In section A,
we present the TDMA algorithm for broadcast. Then, in
section B, we present the TDMA algorithm for
convergecast and we present the algorithm for local gossip
in section C.
A. TDMA Service for Broadcast
Consider a cube grid network where a sensor can
communicate with sensors which are 1 away from it and
interfere with sensors which are k away from it, k ≥ 1 .(e.g.
Fig. 2, where k = 2 ) To present the TDMA algorithm, we
first present an algorithm for initial slot assignment. Then,
we present an algorithm for subsequent slot assignments.
1) Initial slot assignment: Let us assume the base
station sends a diffusing message at slot 0, the
sensors (1, 0 , 0 ) , ( 0,1,0) and (0,0,1) receive the message at
the same time. However, they can’t all forward the
diffusing message at slot 1 because the message would
collide in sensor (0,0,0) . So the sensor (1,0,0) is allowed to
forward the diffusing message at slot 1, the sensor (2,0,0)
is allowed to send it at slot 2. We find that if
sensors (2,0,0) and (0,1,0) forward messages at the same
time, the message would collide in the sensors (1,0,0) and
(1,1,0) (e.g., in Fig.2 while k = 2 ).Hence the sensor (0,1,0)
is allowed to forward the diffusing message at slot ( k + 1) .
Similarly the sensor (0,0,1) can send the diffusing message
at slot (k 2 + k + 1) . Generally to say, if a sensor receives a
diffusing message from its west neighbor, it is allowed to
forward the message at next time slot. While it receives the
message from its north neighbor, it will forward the
579
message after (k + 1) slots. If a sensor receives a diffusing
message from its top neighbor, it forwards the message
after (k 2 + k + 1) slots. In short, if the base station initially
send a diffusing message at time slot 0, the sensor ( x, y, z )
would forward the diffusing message at time slot
ti = x +(k +1) y +(k2 + k +1)z .
Figure 2. The TDMA assignment for broadcast where
communication range=1 and interference range=2. The numbers
associated with each sensor denote the slots at which it can send a
message.
2) TDMA Slots Once the initial time slots were
assigned to the sensors, we should calculate the next time
slot which the sensors can use. Let i and j to be the two
sensors in the network, i is in the collide-domain of j. we
proposed a algorithm that if i forwards the message at
initial slot t i , it can also send message at slot ti + c ∗ MCG,
the MCG captures information from the maximum collide-
domain. Because i and j shouldn’t collide in any slots,
which means ti + ci ∗ MCG≠ t j + c j * MCG . So | ti −t j |
couldn’t be a multiple of MCG. In this network model, the
max | ti −tj | is (k+1)(k2 +k+1) .Such a situation occurs if i is
( k + 1) away from j. Hence, the interval between slots
assigned to a sensor is (k +1)(k2 +k +1)+1.The algorithm for
assigning TDMA slots is shown in TableⅠ. You can see
the time slot assignment in Fig.2.
TABLE I. THE TDMA ALGORITHM (3.1) FOR BROADCAST
const P = (k + 1 )( k 2 + k + 1 ) + 1 ;
b
when sensor i receives a diffusing message from sensor j
if (j is west neighbor at distance 1)
Transmit after 1 slot.
else if (j is north neighbor at distance 1)
Transmit after (k + 1 ) slots.
else if (j is top neighbor at distance 1)
Transmit after (k 2 + k + 1 ) slots.
else
Ignore.
If sensor i transmit a diffusion message at time slot ti , i can transmit at
time slots, ∀ c : c ≥ 0 : t i + c ∗ Pb .
Theorem1 The initial slot assignment for broadcast in
algorithm3.1 is collision-free.
Proof We can assume that the base station (0,0,0) starts to
transmit at slot 0. Let’s prove the algorithm will get
collision-free in any two sensors. Considering two sensors
(x1, y1, z1) and (x2, y2, z2) , t1 = x1 +(k +1)y1 +(k2 +k +1)z1 and
t2 = x2 +(k +1)y2 +(k2 +k +1)z2 are the transmitting time slots.
Collision is possible only if the following conditions hold:
t1 = t 2 , ( x1 − x2 ) + (k + 1)( y1 − y2 ) + (k 2 + k + 1)(z1 − z2 ) = 0 (1)
The Manhattan distance between (x1, y1, z1) and (x2, y2, z2) is:
| x1 − x3 | + | y1 − y3 | + | z1 − z 2 |≤ k + 1
( x1 , y1 , z1 ) and (x2, y2, z2) are distinct, so the Manhattan
distance is at least 1:
| x1 − x3 | + | y1 − y3 | + | z1 − z 2 |≥ 1 (3)
We now show that the three conditions will not hold at
the same time. Firstly we consider | z1 − z2 |= 0 . In this
situation, the conditions will change to
(x1 − x2 ) + (k +1)(y1 − y2 ) = 0 (4)
| x1 − x3 | + | y1 − y3 |≤ k +1 (5)
| x1 − x3 | + | y1 − y3 |≥ 1
In [9], the author had proved this situation. Secondly,
we will prove that when | z1 − z2 |≥ 1 the threes conditions
still can’t hold at the same time. While | z1 −z2 |≥1, then
( x1 − x 2 ) + ( k + 1)( y 1 − y 2 ) = − ( k 2 + k + 1)( z 1 − z 2 ) (7)
| x 1 − x 3 | + | y 1 − y 3 |≤ k
We can see that
| ( x1 − x 2 ) + (k +1)( y1 − y 2 ) |<=| x1 − x2 | +(k +1) | y1 − y 2 |≤ k 2 + k
| − (k 2
+ k + 1 )( z 1 − z 2 ) | ≥ k 2
+ k +1
There occurs a contradiction in (7) 、 (8) 、 (9) and (10).
Thus, collision cannot occur in this algorithm.
Theorem2 The algorithm3.1 satisfies the problem
specification of TDMA.
Proof Consider two distinct sensors j and k such that j is
in the collision domain of k. t j + c1 * pb and t k + c2 * pb are
the time slots assigned to j and k respectively, c1, c2 ∈N and
Pb = (k +1)(k 2 + k + 1) +1 . From theorem1.1, we know that
t j ≠ tk . If t j + c1 * pb = tk + c2 * pb , then collision will occur only
if | tj −tk | is a multiple of pb . However, j is in the collision
domain of k, | t j − tk |≤ (k +1)(k 2 + k +1) , so | t j − t k |< pb .Hence, if
j and k transmit at the same time, then they don’t present in
the collision-domain of each other. This is a contradiction.
Thus, collision cannot occur in this algorithm.
B. TDMA Service For Convergecast
If we use the time slot assigned by the algorithm 3.1 to
transmit a message in the communication pattern
convergecast, we can find a problem that the delay will be
too long. e.g., in Fig.2 when the sensor (1,1,0) send a
message to the base station at slot 4, the message will be
forward at slot 23.At slot 45, the message will arrive in the
base station. We can see that the delay is 41 time slots. So
we should change the algorithm to reduce the delay for
convergecast. Thus, the TDMA algorithm customized for
convergecast is seen in Table2.
TABLE II. THE TDMA ALGORITHM (3.2) FOR CONVERGECAST
const P = ( k + 1 )( k 2 + k + 1 ) + 1 ;
b
when sensor i receives a diffusion message from sensor j
if (j is west neighbor at distance
Transmit after ( P − 1 )th slot.
(2) b
else if (j is north neighbor at distance 1)
Transmit after P － (k + 1 )th slot.
b
else if (j is top neighbor at distance 1)
Transmit after P － (k 2 + k + 1 )th slot.
b
else
Ignore.
If sensor i transmit a diffusion message at time slot t i , i can transmit at
time slots, ∀ c : c ≥ 0 : t i + c ∗ Pb .
C. TDMA Service for Local Gossip
(6) For local gossip, sensors need communicate with
neighbors, in order to reduce the delay, we increase the
interval period (P) to 2 [( k + 1 )( k 2 + k + 1 ) + 1 ] .In each
P, a sensor has two slots to send message, one is even, and
another is odd. Assume the slots assigned to the initiator at
0 and ( P − 1) .The details are in Table3.
(8)
TABLE III. THE TDMA ALGORITHM (3.3) FOR LOCAL GOSSIP
(9)
const P = 2 [ (k + 1 )( k 2 + k + 1 ) + 1] ;
(10) when sensor i receives a diffusing message from sensor j
if (j is west neighbor at distance 1
Transmit after 2 slots.
else if (j is north neighbor at distance 1)
Transmit after 2 (k + 1 )th slots.
else if (j is top neighbor at distance 1)
Transmit after 2 (k 2 + k + 1 )th slots.
else
+ Ignore.
if sensor i transmit a diffusion message at time slot t i , i can transmit at
time slots, ∀ c : c ≥ 0 : t i + c ∗ P ， ( ( P − 1) − t ) mod P + c * P
i
D. USS-TDMA Properties
1) Stabilization and Reliability: We now add
stabilization to the algorithm. We consider that the sensors
start at arbitrary clock values. We ensure that they recover
to states from where collision-free communication is
achieved. We modify that algorithm in section Ⅲ(A) to
obtain underwater self-stabilizing TDMA (USS-TDMA).
In USS-TDMA ,the base station periodically send a
diffusing message in its TDMA slots(calculated in the
algorithm 3.1).If a sensor receives the diffusing message, it
recalculates the time slot based on the algorithm 3.1, then
forward the message in its time slot. If a sensor doesn’t
receive a diffusing message for certain consecutive
number of time slots, the sensor shut down. It will wake up
580
when receives a diffusing message from its neighbor who
is closer to the base station. From then on, the sensors can
use the TDMA algorithm in section 3.1 to transmit
messages again. [9, 10]
2) Energy Efficiency: Energy efficiency can be
achieved in USS-TDMA as follows. A sensor remains in
active mode only in its allotted time slots and in the
allotted time slots of the sensors within its communication
range. In the remaining slots, the sensor can save energy
by turning off its radio and remaining in idle mode.
Suppose the communication range of a sensor is 1, then a
sensor will have at most 6 neighbors. Let P be the period
between successive time slots allotted to a sensor. A
sensor will have to be in active mode in its allotted time
slot and in the allotted time slot of its 6 neighbors, during
every period.
3) Supporting mobile sensor nodes: we know that
sensors in marine move following the water. USS-TDMA
can be used to deal with this situation while localization
service is available. In presence of mobile sensor nodes,
the allotted time slot used by a mobile sensor keeps on
changing due to its motion. Hence, a sensor should know
its current location and calculate its allotted time slot by
the position. In other word, if provided the localization
service, our algorithm can be applied to mobile sensor
network. [11, 12]
IV.
In this section, we compare USS-TDMA with the case
where CSMA is used and with the case where no MAC
layer is used in three communication patterns. The
underlying topology is a cube grid where the base station
Fig.3 (b) identifies the number of collisions that occur
in different algorithms. As expected, USS-TDMA is
collision free for all network sizes. By contrast, in CSMA,
about 12% of messages suffer from collisions.
Fig.3(c) identifies the maximum delay incurred in
receiving broadcast messages. Since all sensors may not
receive the broadcast message when CSMA is used, we
consider the delay when a certain percentage, 80-100%,
of sensors receives the broadcast message. As we can see,
the delay in USS-TDMA is only slightly higher.
B. Simulation for Convergecast
For convergecast, a set of sensors send a message to the
base station at the same time. In our simulations, we keep
the network size fixed at 10x10. We choose a subgrid of
varying size; sensors in this subgrid transmit the data to
the base station. We assume the subgrid sends the data to
the base station is in the opposite corner from the base
station. For these simulations, we consider the percentage
of sensors whose messages are received by the base
station, percentage of collisions and the maximum delay
incurred in receiving convergecast messages.
We can see in Fig. 4(a), the USS-TDMA performs
best in the three protocols; all the messages can reach the
base station. Compared with the CSMA, the number of
messages that reach the base station is less than 50%. And,
without the MAC layer, no data reaches the base station.
In Fig. 4 (b), although USS-TDMA is collision free,
there are a significant number of collisions with CSMA. In
SIMULATION RESULTS Fig. 4 (a) (c), the delay incurred by USS-TDMA is
reasonable and that the base station receives all the
messages sent by the sensors. The delay incurred for 75%
of the messages to reach the base station is infinity when
the field size is greater than 4.
16 USS-TDMA
14
USS-TDMA
80
USS-TDMA 30%

is in one vertex of a cube grid. The length of the cube, CSMA 12 CSMA 70 USS-TDMA 70%
CSMA 30%
14 No Mac Layer
CSMA 70%
Messages received by base station

10 60

Average Delay(in clock units)

12
Percentage of collisions

8 50

communication range and interference range is 1. 10

8
6

4
40

30
6

4 2 20

A. Simulation for Broadcast 2

0
0 2 4 6 8 10 12 14 16
0

-2
0 5 10 15 20 25
10

0
0 2 4 6 8 10 12 14 16 18 20
Size of the field sending data to base station Size of the field sending data to base station Size of the field sending data to base station

We conduct the broadcast simulations for different (a) (b) (c)

network sizes. In these simulations, we consider the Figure 4. Results for convergecast with communication range=1,
following metrics: number of sensors that receive the interference range=1
broadcast message, percentage of collisions and the
maximum delay incurred in receiving broadcast messages.
In Fig.3, we present our simulation results for C. Simulation for Local Gossip
broadcast. Fig.3 (a) shows the number of sensors that With local gossip, in a subgrid some sensors transmit
receive the broadcast message. We find that with the data to the sensors in another subgrid and the
CSMA/USS-TDMA, all sensors receive the message. neighbors of the sensors in the subgrid.In our experiments,
However, without the MAC layer, the number of sensors we keep the network size fixed at 10x10. And, we choose
that receive the message is less than 50%. different sizes of subgrid, sensors in this subgrid transmit
250
USS-TDMA
CSMA
No Mac Layer
14

12
USS-TDMA
CSMA
80

70
USS-TDMA 80%
USS-TDMA 100%
CSMA 80%
the gossip messages. For these simulations, we compute
CSMA 100%

the throughput in different traffic load and the percentage

200
10 60
Sensors receiv e m es s age

Percentage of collisions

Delay(in clock units)

8 50
150

100
6

4
40

30
of collisions.
2 20
50

0 10

0 -2 0
0 50 100 150 200 250 300 0 50 100 150 200 250 300 0 50 100 150 200 250
Size of the netwrok Size of the netwrok Size of the netwrok

(a) (b) (c)

Figure 3. Results for broadcast with communication range=1,
interference range=1

581
 0.7
Throughput Performance
USS-TDMA
CSMA
0.6
0.5
Throughput
0.4
0.3
0.2
0.1
0
0 0.2 0.4 0.6 0.8
offered load
1 1.2
(a)
Figure 5. Results for local gossip with communication range=1,
interference range=1
Fig. 5 (a) shows the efficiency of USS-TDMA and
CSMA in different traffic loads. When traffic load is low,
CSMA performs a little better than USS-TDMA but when
traffic load enhances, throughput of CSMA drops quickly
while throughput of USS-TDMA increases and keep
invariableness at last. Note that in Fig. 5(b) the percentage
of collisions with the CSMA decrease as the size of the
group performing local gossip increases. As discussed in
the case of convergecast, this is due to the fact that with
increased communication, many messages create
collisions in the initial part of the gossip communication
and then they are lost.
V.
In this paper, we presented a TDMA service for
underwater wireless sensor networks. As shown in Section
3, the service ensures that communication among sensors
is collision free. By contrast, previous CSMA based
techniques suffer significant collisions. We considered
three communication patterns: broadcast, convergecast and
local gossip, which occur frequently in underwater
wireless sensor networks. We showed how the TDMA
service could be customized for each of the
communication patterns. As discussed in Section 3, we
recommend that if the application requirements are
unknown, then the TDMA service for the local gossip be
used as the solution for local gossip provides substantial
benefit to broadcast and convergecast. Our solution
assumes that the underlying topology is a cube grid and
that each sensor is aware of its location in the grid. The
USS-TDMA algorithm proposed in this paper use a simple
routine such as diffusing computation to (re) validate the
slots assigned to a sensor. Further, our algorithms are
stabilizing [10] and reliability. Hence, as long as the
initiator of the diffusing computation does not fail, the
algorithm can recover from some states. The states include
sensors are improperly initialized, slots assigned to sensors
are corrupted and sensors have arbitrary clock drifts. This
is achieved by requiring that a sensor freeze (i.e., stop
transmitting messages) temporarily if it does not receive
the required revalidation message. When the revalidation
message is received, it can then continue to work
normally. With the localization service, out algorithm can
be applied to mobile sensor network [11, 12]. In the future
we will consider how the USS-TDMA scheme works
under general nodes distribution (not regular) and under
more than one source of packets in the same vertex.
40
USS-TDMA
ACKNOWLEDGMENT
35 CSMA
30
The experiment is supported by the Chinese National
Percentage of collisions
25
High Technology Research and Development Plan (863
20
15 program) under Grant No. 2006AA09Z113 and National
10
5
Science Foundation of China under Grant No. 60703082.
0
1.4 1.6 0 5 10 15 20
Size of the field sending data to base station
25
REFERENCES
(b) [1] Sandeep S. Kulkarni and Mahesh (Umamaheswaran) Arumugam.
“TDMA Service for Sensor Networks”. ICDCS’04 Workshop.
[2] L. Badia, M. Mastrogiovanni, C. Petrioli, S.Stefanakos, and M.
Zorzi. “An Optimization Framework for Joint Sensor
Deployment, Link Scheduling and Routing in Underwater Sensor
Networks”. SIGNOBILE 2007, 11(4):44-56, Oct 2007.
[3] Xiaoxing Guo, Michael R. and Frater, Michael J. Ryan. “A
Propagation-delay-tolerant Collision Avoidance Protocol for
Underwater Acoustic Sensor Networks. In proc”. MTS/IEEE
OCEANS’06, 2006.
[4] M. Molins, and M. Stojanovic. “Slotted FAMA: a MAC protocol
for underwater acoustic networks”. In Proc. MTS/IEEE
OCEANS’06, 2006.
[5] Borja Peleato and Milica Stojanovic. “A MAC Protocol for Ad-
Hoc Underwater Acoustic Sensor Networks. Proceedings of the
1st ACM international workshop on Underwater networks “
.WUWNet’06, pages 113-115, 2006.
[6] V. Rodoplu and M. K. Park. “An energy-efficient MAC protocol
for underwater wireless acoustic networks”. In Proc. MTS/IEEE
OCEANS’05. 2:1198-1203, 2005.
CONCLUSION [7] Chirdchoo, N. Soh, and K.C. Chua. “Aloha-Based MAC
Protocols with Collision Avoidance for Underwater Acoustic
Networks”. Infocom 2007, pages 2271-2275, may 2007.
[8] Dario Pompili, Tommaso Melodia, and Ian F. Akyildiz. “A
Distributed CDMA Medium Access Control for Underwater
Acoustic Sensor Networks”. In Submitted for Publication, Aug
2006.
[9] S. S. Kulkarni and M(U). “Arumugam SS-TDMA: A self-
stabilizing MAC for sensor networks”. Sensor Network
Operations, IEEE Press, 2006.
[10] E. W. Dijkstra. “Self-stabilizing systems in spite of distributed
control”. Communications of the ACM, 17(11), 1974.
[11] G. Agha, W. Kim, Y. Kwon, K. Mechitov, and S. Sundresh.
“Evaluation of localization services (preliminary report),” 2003.
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[12] T. He, C. Huang, B. Blum, J. Stankovic, and T. Abdelzaher.
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582
 2009 International Conference on Computer Engineering and Technology
Mathematical Document Retrieval for Problem Solving
Sidath Harshanath Samarasinghe
School of Computer Engineering
Nanyang Technological University
Singapore 639798, Singapore
sida0001@ntu.edu.sg
Abstract—Solving mathematical problems is both challenging and
difficult for many students. This paper proposes a document
retrieval approach to help solve mathematical problems. The
proposed approach is based on Kohonen’s Self-Organizing Maps
for data clustering of similar mathematical documents from a
mathematical document database. Based on a user query problem,
similar mathematical documents with their associated solutions
are retrieved in order to provide hints or solutions on solving the
user problem. In this paper, we will discuss the proposed
mathematical document retrieval approach. The performance of
the proposed approach will also be presented in comparison with
other clustering techniques.
Keywords-Knowledge Data Engineering, Data Mining,
Artificial Intelligence
I. INTRODUCTION
Solving mathematical problems is a challenging task
faced by many students studying in schools and universities.
When a student finds it difficult to solve a particular
mathematical problem with his existing knowledge, he will
usually refer to the textbooks and study the related examples,
or consult his peers for the solution. However, such approach
is quite inefficient and sometimes difficult. To help students
solve mathematical problems, a number of computer algebra
systems (CAS) such as Maple [1] and Mathematica [2] are
available which can provide step-by-step solutions to certain
mathematical problems. However, these systems are unable
to provide solutions to most mathematical problems which
comprise the problem description, mathematical expressions
and diagrams. Furthermore, general search engines such as
Google [3] and Yahoo [4] are also unable to provide hints or
help for mathematical problem solving.
In this paper, we propose a document retrieval approach
to help solve mathematical problems. The proposed
approach is based on Kohonen’s Self-Organizing Map
(SOM) [5] for data clustering of similar mathematical
documents from a mathematical document database, which
could be created manually or downloaded from the Internet.
Based on a user query problem, similar mathematical
documents with their associated solutions are retrieved in
order to provide hints or solutions on solving the user’s
problem.
The rest of this paper is organized as follows. Section 2
reviews the related work on mathematical retrieval systems.
Section 3 discusses the proposed approach. Section 4
978-0-7695-3521-0/09 $25.00 © 2009 IEEE
DOI 10.1109/ICCET.2009.69
Siu Cheung Hui
School of Computer Engineering
Nanyang Technological University
Singapore 639798, Singapore
asschui@ntu.edu.sg
presents the experimental results. Finally, the conclusion is
given in Section 5.
II. MATHEMATICAL RETRIEVAL SYSTEMS
Mathematical documents contain special features [6] that
make them not so effective to be retrieved by existing search
engines such as Google and Yahoo. To tackle this problem,
various mathematical document retrieval systems such as
Math Search Project [7] and Zentralblatt Math [8] are
currently under development. However, these systems are
not developed with the objective for mathematical problem
solving.
In this research, we investigate clustering techniques
such as Agglomerative Hierarchical Clustering (AHC) [9]
and partitioning algorithms (e.g. K-means [10]) for
mathematical document retrieval. In particular, the
Kohonen’s Self-Organizing Map is developed for our
retrieval engine. SOM is a neural network based approach
that can be used to discover groups of data in a data set by
extracting the hidden topological structure in the data. SOM
has the ability to organize its nodes into clusters based on the
similarity between them. The SOM is specifically capable in
handling high dimensional data as well as data with noise,
which are characteristic features generally encountered in
document data. Thus, SOM is very suitable for document
clustering. In addition, quite different from most other
clustering techniques, SOM is able to order input data both
locally as well as globally [11], which leads to form better
data clusters.
III. PROPOSED APPROACH
Fig. 1 gives the overall process for our proposed
document retrieval approach. In the proposed approach, a
mathematical document database is first constructed. Each
mathematical document contains the question part and
solution part as shown in Fig. 2. Each question or solution
part of the document may consist of any of the following
three types of data, namely, plain text, mathematical
expressions and diagrams. For example, in the question part,
plain text is used for problem description. For mathematical
expressions (or formulae), they are represented and stored in
the database using Mathematical Markup Language
(MathML) [12]. This aims for effective semantic extraction
of mathematical expressions. For some mathematical
problems, they also contain diagrams (e.g. on geometrical
figures) in the question part that illustrate the problems.
583
 Plain expression. There are two different markups in MathML:
Text
Text content markup and presentation markup. We have used
Preprocessing
presentation markup for mathematical expression
Feature Preprocessing Transformation
Selection representation as it coveys many semantics about the
Mathematical mathematical expressions in addition to the visual
Math. Expression
Model
appearance of them.
Expressions Preprocessing The proposed document retrieval approach consists of the
Generation
following steps: Feature Selection, Preprocessing,
Transformation, Model Generation and Retrieval. Note that
Mathematical Knowledge
Document in the proposed approach, we only train the question part of
Database the mathematical documents for knowledge generation by
SOM. The generated knowledge model will then be used for
Query
Input Query query processing for finding similar mathematical
Knowledge
documents.
Retrieval
Base
User Retrieval Retrieval A. Feature Selection
Client Result
In this step, we identify the useful features from the
Figure 1. Proposed approach. mathematical documents. These features should be able to be
used to distinguish different mathematical documents. Two
Question types of document contents, namely, plain text description
and mathematical expressions will be processed for the
extraction of representative features. Question diagrams are
usually stored as images which make it a complicated task
for feature extraction. However, diagrams are generally
associated with text description for the corresponding
explanation. As such, feature extraction from both plain text
description and mathematical expressions would be
sufficient for our document retrieval purpose.
The figure above shows the region bounded by the curve B. Preprocessing
π Preprocessing aims to remove noisy, erroneous, and
y = tan x , the x-axis and the line x = . Use the identity
4 incomplete data, and prepare the representative features that
sec 2 x = 1 + tan 2 x to find, correct to 2 decimal places, the are suitable for knowledge generation. There are two steps in
volume obtained when the region is rotated through 2π radians preprocessing: Text Preprocessing and Mathematical
about the x-axis. Expression Preprocessing.
1) Text Preprocessing
Solution This step preprocesses the descriptive text using text
The area of each circular area generated is
2 2 preprocessing techniques. It consists of the following steps:
πr = π (tan x) .
(1) Noise Removal removes noisy data in the descriptive
To find the volume obtained, we need to use definite integrals in
π text including numberings and capital letter labels. (2)
the interval: 0 to . Therefore, the volume generated is: Tokenization tokenizes the text string resulted from noise
4
π removal into a sequence of tokens. (3) Stemming converts
4 token words into their root form. Here, we use the Porter’s
Volume =
∫ π (tan x) dx
0
2
stemming algorithm [13] for stemming.

Mathematical MathML
2 2
By using the given identity sec x = 1 + tan x , Expression (Presentation Markup)
π <math>
π <mrow>
4

∫
<msup>
Volume = π (sec2 x − 1)dx = π [tan x − x]04 <mrow>
<mi>x</mi>
0 </mrow>
π 3π 2
x2 +1 <mrow>
= π (1 − )= = 7.40 units3 (Answer) <mn>2</mn>
4 4 </mrow>
</msup>
Figure 2. A mathematical document. <mo>+</mo>
<mn>1</mn>
Fig. 3 gives an example of the MathML representation </mrow>
for an expression. MathML is basically a well-formed XML </math>
Figure 3. An example MathML representation.
that describes different elements of a mathematical

584
 (4) Indicative Keyword Identification and Counting higher weightage to keywords that can be used to distinguish
identifies a set of indicative keywords (such as differentiate documents effectively. Next, the resultant vectors are then
and integrate). The keyword list is then used for identifying weighted using TFIDF (Term Frequency x Inverse
the indicative terms from the stemmed set of words. Here, Document Frequency) [14]. Finally, the resultant TFIDF
vectors are normalized into unit length.
the indicative keywords from the document text are
Fig. 5 shows an example on document preprocessing and
extracted and counted to obtain their frequency of transformation.
occurrences.
2) Mathematical Expression Preprocessing D. Model Generation
In this step, mathematical expressions are preprocessed Once the normalized vectors are created, they are fed into
for the extraction of semantic meanings contained in the the SOM for training. Before the training session begins, the
expressions. It extracts the semantics from the MathML weights of the neurons in the SOM grid are initialized with
representation of the mathematical expressions using regular random real numbers within the interval [0, 1]. In each
expressions. As such, we construct a list of mathematical training iteration, the entire set of normalized vectors is fed
into the SOM network. The winning neuron for a particular
regular expressions with their corresponding semantic
input vector is determined by finding the neuron which gives
meanings (represented using indicative keywords). The the smallest Euclidean distance between the input vector and
matching between a mathematical expression and a list of the neuron’s weight vector. Once the winning neuron is
mathematical regular expressions is illustrated in Figure 4. found, only the weights associated with the winning neuron
Then, the indicative keywords identified during regular and its neighbourhood is adjusted to adapt to the input
expression matching are converted into their root form for pattern. When the training session is completed, the weight
further processing. vectors of the SOM network and the cluster information are
After preprocessing the text and mathematical stored in the knowledge base which can then be used for
expressions, the stemmed sets of keywords with their document retrieval.
frequency of occurrences are counted and merged to form a
keyword vector. The resultant keyword vector is then passed Differentiate the following with respect to x : 7x2 + 4
to the Transformation step.
Feature Selection
7x2 + 4 Plain Text Math.
MathML Representation Expressions
in Presentation Markup
Differentiate the following
<math> with respect to x :
7 x2 + 4
<mrow>
<msqrt> Mathematical
Mathematical Regular Text
<mrow> Expression
Expressions Preprocessing
<mn>7</mn> List of Keywords Preprocessing
<msup> Matching Regular Indicative Extracted Index Index Keyword Extracted Index
<mrow> Expression Keyword Keyword 1 absolute Keyword
<mi>x</mi> <mi[^>]*>\p ... function differentiate 30 2 accelerate mroot 170
</mrow> <msup>\s*(?: … inverse … … … …
<mrow> … … 30 differentiate
<mn>2</mn> .
. <m(?:sqrt|root)>\s mroot … …
</mrow> *<mrow>.*?[x|y].* 170 mroot
</msup> .
?</mrow>\s*(?:<m … …
<mo>+</mo> n[^>]*>\d</mn>\s* 284 zero
<mn>4</mn> . )?</m(?:sqrt|root)
</mrow> . >
</msqrt> . … …
</mrow>
</math> Preprocessing
Results
Matching Result
Keyword Count 1 ... 30 … 170 … 284
mroot 1
polynomialpowertwo 1
0 ... 1 … 1 … 0
… … Keyword Vector
Transformation
Figure 4. Mathematical expression matching.
1 ... 30 … 170 … 284
0 ... 0.901 … 0.401 … 0
C. Transformation
Normalized Vector
Transformation converts the keyword vector into a
normalized vector for neural network training. First, a weight Figure 5. Document preprocessing and transformation.
multiplication is applied to re-compute the weight with
frequency counts in the keyword vector. The idea is to give

585
E. Retrieval
In document retrieval, user enters a query using the
retrieval interface, and in response, a set of documents
(together with the corresponding step-by-step solutions) is
retrieved that are most similar to the features of the user’s
query. First, plain text and mathematical expressions are
extracted as features from the user’s query which are done in
a similar way as the Feature Selection step. The extracted
features are then preprocessed using the same preprocessing
techniques adopted in the Preprocessing step. After this, the
retrieval step will go through Query Transformation, Cluster
Selection, and Document Retrieval and Ranking.
1) Query Transformation
The generated query vector is then transformed into a
normalized vector using the similar steps followed in the
Transformation step. However, one difference in
transforming the query vector is that, instead of weighting
query terms using TFIDF, it is weighted using qf × idf
[15], where qf is the frequency of occurrence of a term in
the query and idf is the inverse document frequency of the
query term in the document database collection. The reason
for using qf × idf is because a keyword that occurs more
frequently in the query is likely to be more important than
those which occur infrequently, and those keywords that
occur infrequently in the document collection are likely to
be more important than the frequent terms.
2) Cluster Selection
After query transformation, the generated normalized
query vector is used to find clusters which contain most
similar mathematical documents. To do this, the cluster
centroid vectors stored in the knowledge
G G base are retrieved,
and the Euclidean distance distG ( q , c ) is calculated between
the normalized query vector q and a cluster centroid vector
G
c using the following formula:
→ → n document retrieval approach based on Kohonen’s Self-
dist ( q , c ) = ∑ (q − c )
i =1
i i
2 Organizing Map that can help provide hints or solutions to
where n is the number of dimensions in the normalized
query vector (and the centroid vector) and q i and ci are
the weights of the ith element in the normalized vector and
the centroid vector respectively. The smaller the Euclidean
distance, the more likely those clusters will contain
documents similar to the user’s query. Because of that, and
to limit the number of retrieved results, the nearest 10
clusters are selected to retrieve documents from those
clusters.
3) Document Retrieval and Ranking
Once the clusters to be retrieved are determined, the
mathematical documents are then retrieved starting from the
nearest cluster. The retrieved documents are eventually
ranked based on the Euclidean distance between the
normalized query vector and the normalized document
vector. To aid in the ranking process, the normalized
document vector information stored in the knowledge base
586
is used. Finally, the ranked set of mathematical documents
is displayed to the user.
IV. EXPERIMENTAL RESULTS
We have conducted experiments to evaluate the retrieval
performance of the proposed approach. The performance is
measured based on the Kohonen’s Self-Organizing Map
clustering algorithm and compared it with two other popular
clustering algorithms on AHC and K-means. The data set
used for our evaluation consisted of 304 mathematical
documents which were extracted from typical problem
questions from G.C.E. O-Level Additional mathematics. We
used 30 training samples and 30 test samples to evaluate the
training and testing performance respectively.
The retrieval accuracy is measured using the metric on
Average Precision at Seen Relevant Documents [16]. This
metric favors the system to retrieve relevant documents
higher in ranking. It is a good indicator for the usability of a
retrieval engine, since users typically spend more time
examining top results than lower ranked ones. For each
query, we manually analyze the result set to evaluate the
relevance of the retrieved mathematical documents. In the
experiments, we only analyzed the first 20 documents ranked
in the result set in evaluating the retrieval accuracy. The
relevance was assessed using two categories: relevant or not
relevant.
The average retrieval precision obtained for training and
test samples for the three clustering algorithms is shown in
Table 1. The three clustering algorithms have achieved very
high training performance and the training data that are being
searched are mostly retrieved as the first ranked result. For
test data, the performance of SOM has achieved around 80%,
which has performed better than the other two algorithms.
V. CONCLUSION
In this paper, we have proposed a mathematcial
solve students’ mathematical problems by finding similar (or
exactly the same) documents to the user’s query problem.
The performance of the proposed approach has been
evaluated and good initial results have been obtained.
Currently, further experiments are being conducted with a
much larger set of training documents, and more regular
expressions are being written for semantic feature extraction
from various mathematical topics.
TABLE I. PERFORMANCE RESULTS
Clustering Training Testing
Technique Accuracy Accuracy
SOM 100% 79.70%
K-means 100% 78.80%
AHC 97% 77.13%
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[6] J. Misutka and L. Galambos, “Mathematical Extension of Full Text
Search Engine Indexer,” Proc. 3rd International Conference on
Information and Communication Technologies: From Theory to
Applications (ICTTA 08), April 2008, pp. 1-6.
[7] A. Youssef, “Roles of Math Search in Mathematics,” in Mathematical
Knowledge Management, MKM 2006, Lecture Notes in Computer
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[9] M. Nanni, “Speeding-Up Hierarchical Agglomerative Clustering in
Presence of Expensive Metrics,” in Advances in Knowledge
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[12] Mathematical Markup Language (MathML). Available at:
http://www.w3.org/TR/MathML2.
[13] M.F. Porter, “An Algorithm for Suffix Stripping,” Program, 14(3):
pp. 130-137, 1980.
[14] G. Salton, A. Wong, and C.S. Yang, “A Vector Space Model for
Automatic Indexing,” Communications of the ACM, Vol. 18, Issue
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[15] H. Turtle and W.B. Croft, “Evaluation of an Inference Network-
Based Retrieval Model,” in ACM Transactions on Information
Systems, Vol. 9, No. 3, 1991, pp. 187-222.
[16] R.B. Yates and B.R. Neto, Modern Information Retrieval, Addison-
Wesley Longman Publishing Co.., Inc., Boston, MA, 1999.
 2009 International Conference on Computer Engineering and Technology
Lossless Data Hiding Scheme Based on Adjacent Pixel Difference
Zhuo Li, Xiaoping Chen, Xuezeng Pan, Xianting Zeng
College of Computer Science
Zhejiang University
Hangzhou, China
lizhuo84@gmail.com
Abstract—This paper proposes a lossless data hiding scheme
with high embedding capacity, which can recover the original
image without any distortion after the hidden data have been
extracted. In this scheme, difference value of the adjacent
pixels is calculated first. After adjusting the difference slightly,
we can increase (decrease) it by 1 or maintain it intact to
embed information into the cover image. Experimental results
show that the average embedding capacity can be up to
0.91bpp (bit per pixel) while the PSNR stays larger than 30dB.
Compared with some existing known lossless data hiding
schemes, the proposed method provides higher embedding
capacity and lower image degradation. Moreover, the
presented algorithm is quite simple and the execution time is
rather short.
Keywords-component; data hiding; steganography; lossless;
pixel difference
I. INTRODUCTION
Data hiding (or steganography) is a commonly used
technique that hides secret information into digital media
imperceptibly, such as images, videos, audios, etc. The
information can serve as authentication codes, annotation, or
secret data depending on the purpose of the application itself.
In most cases of data hiding, the cover media will suffer
some permanent distortion due to embedded secret data and
cannot be reversed back to the original media. In some
applications, such as medical image system, law enforcement
and military imagery, it is desired to recover the original
media without any distortion after the hidden data have been
extracted because of the requirement of high-precision
nature. Therefore, a distortion-free, reversible, lossless data
hiding technique is required.
Recently, the number of lossless data hiding schemes
reported in the literatures has been adding up. Honsinger et
al. [1] in their patent used addition modulo 256 to embed the
hash value of the original image for authentication. By using
the addition modulo 256, the overflow/underflow is
prevented and reversibility is achieved. But the images
processed by their algorithm exists some annoying salt-and-
pepper noise. Fridrich et al. [2] compressed the least
* Project supported by the Major Science and Technology Special Project of
Zhejiang Province, China (No. 2007C11088) and the Science and Technology
Project of Zhejiang Province, China (No. 2008C21077)
978-0-7695-3521-0/09 $25.00 © 2009 IEEE 588
DOI 10.1109/ICCET.2009.40
significant bit (LSB) planes to obtain extra space for
embedding secret data. Xuan[3] proposed a scheme based on
integer wavelet transform (IWT) and created more space in
high frequency subbands for hiding data, which results in a
two-five times payload as large as that in [2]. Celik et al. [4]
improved Fridrich et al.’s technique by compressing the
quantization residuals of pixels instead of compressing LSB
planes to obtain more extra space for embedding secret data.
The compressed residuals and secret data are concatenated
and embedded into the cover image. Tian [5] proposed a
high capacity reversible data embedding technique by using
a difference expansion. Alattar [6] extended Tian’s scheme
by introducing the difference expansion of a vector to obtain
more extra space for embedding secret data. Ni et al. [7]
presented a reversible data hiding scheme. By utilizing the
maximum point and the minimum point of image histogram,
they shifted image histogram to obtain extra space for
embedding data. It is simple and novel, however, the
disadvantage of this method is that few image contains a
large number of pixels with equal pixel values, so the
embedding capacity is small.
In this paper, we enhance Ni et al.’s scheme using the
difference of adjacent pixels － between the first and the
second pixel as well as the second and the third pixel, and so
on － row-by-row or column-by-column to improve the
embedding performance. Experimental results show that the
payload and the image quality of stego-image are
dramatically improved. Among the eight commonly used
cover images, the average payload could even achieve
0.91bpp (bit per pixel) when the PSNR is larger than 30dB.
The rest of this paper is organized as follows. The
proposed scheme and its characteristics are presented in
Section 2. Experimental results are illustrated in Section 3,
and Section 4 concludes our work.
II. PROPOSED SCHEME
A. Adjacent pixel difference
Suppose that the cover image I is an 8-bit grayscale
image with size M×N, and denote I(i,j) as the pixel value at
location (i,j) in the original image. The adjacent pixel
difference can be expressed as
 r
d (i , j ) = I (i , j ) − I (i , j + 1)
in row direction, where 1 ≤ i ≤ M and 1 ≤ j ≤ N − 1 , and
c taken into account for different δ.
d (i , j ) = I (i , j ) − I (i + 1, j )
in column direction, where 1 ≤ i ≤ M − 1 and 1 ≤ j ≤ N .
For simplicity, in this paper, adjacent pixel difference and
difference are interchangeable.
Fig.1 shows the histograms of pixel value and difference
of Lena and Baboon. It is obviously to see that the maximum
number of the point with equal value in (b) and (d) is much
larger than that in (a) and (c), respectively. So, more space
will be obtained to embed data if utilizing these differences.
where 1 ≤ i ≤ M and 1 ≤ j ≤ N − 1 .We introduce a
threshold δ. The difference, whose absolute value equals to
δ, will be be used to embed secret data.
Before data embedding, we modify the difference to
obtain extra space. As Fig.2 illustrates, two cases will be
Case 1. δ=0
Differences with value of 0 are utilized to embed data.
All differences, whose value are greater than 0, are added by
1. And the others remain unchanged, As Fig.2(a) shows. This
process can be described as
⎧⎪ d (i , j ) +1,
' d (i , j )>0
d (i , j ) = ⎨ (2)
⎪⎩d (i , j ), otherwise
After the adjustment, we will find that there is no pixel
difference with value 1 in the whole image. Then check the
to-be-embedded bit b in the bit stream. If the to-be-
'
embedded bit is binary 1, d (i , j ) is added by 1. Otherwise,
'
d (i , j ) remains intact. In this way, data are embedded into
the cover image. This process is described as

' ⎧ '
* ⎪d (i, j)+1, d (i, j)=0 and b =1
d (i, j) = ⎨ (3)
⎪d ' (i, j),
⎩ otherwise

Case 2. δ>0
If δ>0, as the Fig.2(b) illustrates, the differences, whose
value are greater than δ, are added by 1, and those with value
less than -δ are subtracted by 1. The differences between -δ
and δ remain unchanged. It is described as formula (4).
Figure 1. Histograms of pixel value and difference of Lena and Baboon.
(a) histogram of Lena; (b) difference histogram of Lena; (c) histogram of ⎧ d (i , j ) +1, d (i , j ) >δ
Baboon; (d) difference histogram of Baboon. ' ⎪
d (i , j ) = ⎨ d (i , j )−1, d (i , j )<−δ (4)
⎪d (i , j ),
⎩ otherwise

Now, we can find there is no pixel difference with value

of δ+1 or -(δ+1) in the whole image. If the to-be-embedded
'
bit is binary 1, d (i , j ) with value of δ will increase 1, or,
'
decrease 1 if its value is equal to -δ. Otherwise d (i , j )
remains intact if the to-be-embedded bit is 0. Formula (5)
describes this process as following
Figure 2. Pixel difference adjustment. (a) δ=0; (b) δ>0.
⎧ d ' (i, j)+1, d ' (i, j)=δ and b =1
⎪
B. Embedding procedure * ⎪
d (i, j) = ⎨d' (i, j)−1, d' (i, j)=−δ and b=1 (5)
Without loss of generality, we will describe the ⎪
r ⎪ d ' (i, j),
embedding procedure using d (i , j ) . In simplicity, we otherwise
r ⎩
represent d (i , j ) as d (i , j ) in next sections. Certainly,
c
d (i , j ) can also be utilized to hide data. * *
Denote I as the stego image. I (i , j ) can be calculated
by
d (i , j ) = I (i , j ) − I (i , j + 1) (1)
* *
I (i , j ) = I (i , j + 1) + d (i , j ) (6)

589
where 1 ≤ i ≤ M and 1 ≤ j ≤ N − 1 . ⎧ * *
Note that the pixels in the last column will be used to ⎪ d (i, j)−1, d (i, j)>0
d(i, j) = ⎨ (12)
retrieve the hiding data and restore the original image, so ⎪d * (i, j), d * (i, j)≤0
⎩
they remain unchanged with no hidden data embedded as
formula (7) describes.
in the extraction procedure.
*
*
Repeat the process (9) ~ (11) for all I (i , j ) until j=1, all
I (i , j )= I (i , j ), j = N (7) hidden data are extracted and the original image is restored
without any distortion.
It is clearly observed from the embedding procedure that
the capacity of this method equals to the number of D. Prevention of overflow/underflow
difference with value of ±δ. Assume that there is no pixel As mentioned above, we may add (or subtract) 1 to the
with overflow and underflow in the embedding phase. In the original pixel difference in the embedding procedure. That
worst case, all pixels will be altered by 1. Hence, the mean is to say, the original pixel value I(i,j) could be changed to
square error (MSE) is at most equal to one, i.e., MSE=1. I(i,j)+1 or I(i,j)-1 after hiding data. Therefore, for the pixels
Therefore, the PSNR of the stego image can be calculated as whose value equal to 0 or 255 in an 8-bit grayscale cover
image, overflow or underflow will occur. In the proposed
255×255 scheme, we pre-process the cover images. At first, we
PSNR = 10 × log10 ( ) ≈ 48.13 (8) preserve the location and grayscale value of pixels which
MSE
equal to 0 or 255, and reset them to 1 or 254 respectively.
In short, the lower bound of the PSNR of the stego image Then compress this overhead information losslessly and
is about 48.13dB. append it to the hidden data. As a result, the actual
embedding capacity (pure payload) is less than or qula to
C. Data extraction the maximum size of embedded data. In this way, we can
Alien to the embedding sequence, hidden data will be prevent the overflow/underflow and restore the original
image losslessly.
acted reversely, from right to left in each row.
*
At first, let I (i , N ) = I (i , N ) . Then calculate the pixel III. EXPERIMENTAL RESULTS AND ANALYSIS
difference of stego image by
In order to evaluate the performance of our proposed
scheme, we use eight commonly used 8-bit grayscale
* *
d (i , j ) = I (i , j ) − I (i , j + 1) (9) images with size of 512×512 as the cover images, and a
pseudo random generator to generate hidden data. Fig.3
shows the test images. Peak-Signal-to-Noise-Ratio (PSNR)
where 1 ≤ i ≤ M and 1 ≤ j ≤ N − 1 . is used to evaluate the image quality.
Suppose that we used the pixel differences with value of Comparision with Ni et al.’s [7] scheme.
*
δ(δ>0) in the embedding procedure. If d (i , j ) equals to δ We use the difference of adjacent pixels in row direction
* r
or -δ, a bit 0 will be extracted; if d (i , j ) equals to δ+1 or – denoted as d (i , j ) mentioned in Section 2 and choose
(δ+1), a bit 1 will be extracted. The original pixel difference “δ=1” to embed hidden data. We re-implement Ni’s scheme
between location (i,j) and (i,j+1) can be calculated by on all the eight test images. Tab.1 shows the performance of
the proposed scheme on all test images. Experimental
⎧ d * (i, j)−1, d * (i, j)≥δ +1 results clearly reveal that the capacity and the stego image
⎪ quality of our proposed and Ni et al.’s scheme are both
⎪
d(i, j) = ⎨d * (i, j)+1, d * (i, j)≤−(δ +1) (10) better than Ni’s method.
⎪ The smoother the images are, the more centralized the
⎪ d * (i, j), −δ ≤ d * (i, j)≤δ
⎩ pixel differences are. As Fig.1(b)(d) illustrate, the number of
pixel difference equals to “±1” in smooth image Lena is
Then we can restore the original pixel value in the much larger than that in complex image Baboon. So, more
data will be hidden in Lena. Experimental results in Table1
location (i,j) by using formula (11) also prove this conclusion. The smooth images (such as
Lena, Airplane, Tiffany) have higher embedding capacity
*
I (i , j ) = I (i , j + 1) + d (i , j ) (11) than complex images (such as Baboon, Goldhill). Based on
this characteristic, our proposed scheme can provide better
performance in medical images.
Note that if we have used δ=0 to embed data, formula
(10) will be replaced by

590
 extra data. Hidden data are composed of embedded bits in
these two phases. In the extraction procedure, we can
distortion-free restore the temp image It using “δ=1” at first,
then use “δ=2” to restore the exact original image losslessly.
If the embedding procedure is called many times, a high
embed capacity can be achieved.
r
We propose a solution that use d (i, j ) to get pixel
differences first and “δ=6,5,4,3,2,1” sequentially to embed
c
data. After that, calculate pixel differences using d (i , j )
and use “δ=6,5,4,3,2,1” again to embed more data. In this
way, more data will be embedded into the cover image.
Fig.4 illustrates the performance of this proposed solution.
TABLE I. PERFORMANCE COMPARISON BETWEEN NI ET AL.’S AND
OUR SCHEME ON SOME COMMONLY USED IMAGES.

Ni[7] Proposed
Cover Image
Pure payload Pure payload
PSNR (dB) PSNR (dB)
(bits) (bits)
Lena 5,468 47.69 48,670 49.11

Airplane 15,388 48.29 60,500 49.54

Tiffany 10,153 46.10 58,770 49.42

Barbara 4,729 50.19 41,874 48.93

Baboon 5,499 45.94 15,002 48.40

Boat 10,546 49.14 50,290 49.12

Goldhill 5,178 46.08 33,006 48.74

Peppers 5,713 48.21 50,711 49.42

Figure 3. Test images. (a) Lena; (b) Airplane; (c) Tiffany; (d) Barbara; (e)
Baboon; (f) Boat; (g) Goldhill; (h) Peppers.

Extend to hide more data. Figure 4. Performance of the extended proposed scheme on all the eight
In fact, we can easily extend our proposed scheme to test images.
r c
hide more data by using both d (i , j ) and d (i , j )
We use bpp (bit per pixel) to measure the embedding
alternately. In addition, another δ can be utilized to hide bits.
capacity. From Fig.4 we can easily find that when the PSNR
Therefore, more data can be embedded into the original is above 30dB, our scheme can have the payload capacity
r
image. For example, we firstly use “δ=2” and d (i , j ) to more than 0.8bpp on all test images except Baboon.
embed data and obtain a temp image It. Then, for this temp Actually, the average capacity of our scheme is up to
c
image It we can use another “δ=1” and d (i , j ) to embed

591
0.91bpp when the PSNR stays above 30dB on all test IV. CONCLUSION
images. Embedding capacity and the quality of stego-image are
Comparison with some other existing schemes with two important factors for a lossless data hiding scheme. In
high capacity. this paper, we present a lossless data hiding scheme based
Xuan et al.’s [3] method and Celik et al.’s [4] method are on the difference of adjacent pixels. Experimental results
lossless data hiding schemes with high embedding capacity. demonstrate that our proposed scheme outperforms Ni et
Alattar [6] extended Tian et al.’s [5] method and obtained a al.’s [7] scheme in terms of the capacity and image quality.
higher capacity. Compared with all these three schemes, our Furthermore, our method can be easily extended to hide
proposed method is proved to provide better performance in more data. As this paper has shown, our proposed scheme
terms of payload capacity and image quality. performs high embedding capacity and low degradation.
Tab.2 describes the experiment result. The data of the
r
proposed scheme in Tab.2 are obtained by using d (i , j ) ,
c REFERENCES
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592
 Author Index - Vol 1
A.R, Ananya. ............................................. 267
Agarwal, Suneeta........................................ 536
Ahmad, R. Badlishah.................................. 573
Ahmad, R.B. .............................................. 108
Ahmed, Khandakar Entenam Unayes......... 205
Ai, Changsheng.......................................... 272
Akbar, Habibullah...................................... 343
Al-Fedaghi, Sabah........................................ 16
Aljunid, S.A. .............................................. 573
Allison, Brendan......................................... 223
An, Hong.................................................... 119
Anaraki, Firouz B. ..................................... 400
Arora, Prabhjyot......................................... 238
Aung, May Thu.......................................... 549
Azadeh, A. ................................................. 505
Azadeh, Ali................................................... 42
Bagheri-Soulla, A.A. ................................. 523
Bai, Chengzhe............................................ 303
Baolin, Yin................................................. 195
Bayat, Mahdi M. ........................................ 166
Bera, Debasish............................................ 145
Bhilare, D.S. .............................................. 365
Bian, Weiwei.............................................. 334
Bie, Libo..................................................... 200
Bikas, Md. Abu Naser................................ 371
Biswal, B.B. ................................................. 32
Bu, Shuyang............................................... 424
Cao, Shukun............................................... 272
Chand, Laxmi............................................. 560
Chander, Vinay............................................. 91
Chandrasekaran, K. ................................... 441
Chang, Chaowen.................................. 37, 312
Chen, Honghui.............................................. 47
593
Chen, Houquan........................................... 210
Chen, Junjie................................................ 218
Chen, Liangji.............................................. 565
Chen, Qian.................................................. 303
Chen, Shu................................................... 295
Chen, Xiaoping........................................... 588
Cheng, Jian-hua.......................................... 510
Chien, Yu-Hung......................................... 157
Choudhury, B.B. .......................................... 32
Cong, Ming................................................. 119
Cui, Herui........................................... 347, 395
Ding, Xiangqian......................................... 321
Dong, Liying.............................................. 281
Dong, Yuzhen..................................... 481, 527
Duan, Jiang................................................. 462
Fadhil, Hilal Adnan.................................... 573
Fang, Mei.................................................... 449
Feng, Yuan................................................. 321
Fouladian, Majid........................................ 166
Fu, Baowei.................................................. 419
Gai, Jingfen.................................................. 23
Gao, Lijun................................................... 215
Gao, Ruizhen...................................... 481, 527
Gaurav, Shriprakash..................................... 91
Ghaznavi-Ghoushchi, M.B. ............... 515, 523
Ghaznavi-Goushchi, M.B. ......................... 428
Gong, Dawei................................................. 28
Gong, Faming............................................. 326
Gräser, Axel................................................ 223
Guo, Hongyan............................................ 432
Guo, Li........................................................ 182
Guo, Zhongwen.......................................... 578
Guojie, Jin.................................................. 195
 Author Index - Vol 1
Gupta, Daya................................................ 238
Gupta, Manoj.............................................. 560
Haleh, H. .................................................... 252
Han, Jingti.................................................. 383
Hao, Sun..................................................... 129
Hasan, Md. Khalad............................. 205, 371
Hatefi, S.M. ....................... 252, 390, 457, 505
Hatefi, Seyed-Morteza.................................. 42
He, Hongjiang............................................ 361
He, Ming..................................................... 172
He, Yulan.................................................... 490
He, Yunhui................................................. 262
He, Zheng................................................... 177
Hendessi, Faramarz.................................... 166
Hong, Feng................................................. 578
Hou, Hong.................................................. 281
Hu, Aiqun................................................... 172
Hu, Dan........................................................ 47
Huang, Ningfeng........................................ 452
Huang, Weili.............................................. 432
Huang, Weitong............................................ 64
Hui, Siu Cheung................................. 490, 583
Huimei, Yuan............................................. 129
Hussain, Altaf............................................. 371
Imran, Ale................................................... 186
Iranmanesh, H. .......................... 252, 390, 457
Issac, Biju................................................... 414
Jia, Hongru................................................. 436
Jiang, Wenchao.......................................... 383
Jiangfan, Ouyang........................................ 554
Jie, Shen...................................................... 352
Jin, Ying..................................................... 569
Jing, Huang................................................. 103
594
Jolai, F. .............................................. 390, 457
Jung, Keechul............................................. 466
K, Arun Ram. ............................................ 267
Kakhani, Manish.......................................... 91
Kang, Cui.................................................... 326
Karray, Fakhri.............................................. 81
Kaur, Inderjeet............................................ 238
Kaushik, B.K. ............................................ 560
Khalib, Zahereel Ishwar Abdul.................. 108
Khanwalkar, Nishant.................................... 91
Kheiri, S. .................................................... 515
Kong, Jun............................................ 419, 436
Kor, H. ............................... 252, 390, 457, 505
Kor, Hamrah................................................. 42
Kulkarni, M. .............................................. 238
Kumar, Preetham........................................ 124
Lee, Yunli................................................... 466
Lei, B. .......................................................... 55
Lei, Yingjie................................................. 471
Li, Changchun.............................................. 73
Li, Huiying................................................. 565
Li, Min........................................................ 232
Li, Ruihu..................................................... 471
Li, Shufen................................................... 142
Li, Wenxin.................................................... 11
Li, Xiaolu.................................................... 436
Li, Yu-shen................................................. 510
Li, Zhengbao.............................................. 578
Li, Zhuo...................................................... 588
Li, Zongmin................................................ 161
Liang, Jinzhao.............................................. 98
Lin, Han-Yu................................................ 330
Lin, Hu........................................................ 103
 Author Index - Vol 1
Liu, Chen.............................................. 37, 312
Liu, Heping................................................. 232
Liu, Huan-Wen........................................... 243
Liu, Lianzhong............................................. 23
Liu, Wenshuo............................................... 11
Liu, Xiaodong......................... 73, 88, 321, 500
Liu, Yanmin.................................................. 64
Liu, Yujie.................................................... 161
Liu, Zhihua................................................. 142
Liu, Zhijing................................... 60, 229, 531
Lu, Fei........................................................ 114
Lu, Wei-Ping.............................................. 243
Lu, Yao....................................................... 485
Lu, Yinghua................................................ 419
Luo, Na....................................................... 150
Luo, Xueshan................................................ 47
M, Ganesh Kumar. .................................... 267
Ma, Lintian................................................. 419
Mahapatra, R.N. .......................................... 32
Mallika, R. ................................................. 137
Mei, Xie...................................................... 182
Meiling, Yuan............................................. 554
Mekhabunchakij, Kittima........................... 357
Meng, Yadong.............................................. 73
Munim, Md. Fazle...................................... 291
Naing, Thinn Thu....................................... 316
Nakajima, Fumio........................................ 475
Nematbakhsh, Naser................................... 276
Oi, Hitoshi.................................................. 475
Pan, Xuezeng.............................................. 588
Ping, Liu..................................................... 103
Prabuwono, Anton Satria........................... 343
Prasad, Rajesh............................................ 536
595
Qadeer, Mohammed A. ............................. 186
Qi, Chunjie................................................. 405
Qi, Zhenyu.................................................... 28
Qi, Zhi........................................................ 436
Qiao, Xiangjie.............................................. 51
Qiu, Hangping............................................ 172
Qiu, Yuxia.................................................. 191
Quan, Tho Thanh........................................ 490
Quan-Xun, Ding......................................... 352
Rafiee, M. .......................................... 428, 515
Rahimi, Morteza......................................... 166
Rahman, Mostafijur.................................... 108
Ramani, A.K. ............................................. 365
Ramya, I.R. ................................................ 441
Rana, Md. Mashud..................................... 205
Reddy, Aravind............................................ 91
Ren, Wuling................................................ 134
Ren, Yongqing............................................ 119
S, Ananthanarayana V................................ 124
Samarasinghe, Sidath Harshanath.............. 583
Saravanan, V. ............................................ 137
Sen, Jaydip.................................................. 145
Seyfe, B. ............................................ 428, 515
Shafieinejad, Alireza.................................. 166
Shahriar, Tamim......................................... 205
Shang, Yanmin................................... 481, 527
Shen, Ji-hong.............................................. 379
Shi, Jun-yu.................................................. 510
Shi, Kangyun.............................................. 114
Shoeb, Abu Awal Md. ............................... 371
Shohag, Md. Al-mamun............................. 205
Song, Li...................................................... 272
Song, Yinchen............................................ 452
 Author Index - Vol 1
Soon, I.Y. ..................................................... 55
Štava, Martin.................................................. 3
Sugavanan, V. .............................................. 68
Sugiarto, Indar............................................ 223
Sun, Liang.................................................. 281
Sun, Li-hong............................................... 379
Syama, R. .................................................. 441
Tan, Runhua............................................... 114
Tanwani, Sanjay......................................... 365
Tapaswi, Shashikala..................................... 91
Tawsopar, Kattiya...................................... 357
Thakur, Kamal............................................ 238
Tian, He........................................................ 28
Tian, Yawei................................................ 308
Tumblin, Jack............................................. 462
Tun, Khin Nwe Ni...................................... 549
Tvrdík, Pavel.................................................. 3
Ueno, Haruki.............................................. 177
Ukil, Arijit.................................................. 145
Usino, Wendi.............................................. 343
Venkatesh, V. Prasanna................................ 68
Wan, Fangrong........................................... 281
Wang, Bo.................................................... 308
Wang, Fan.................................................. 409
Wang, Jianzhong........................................ 419
Wang, Jianzhou............................................ 98
Wang, Jing.......................................... 229, 531
Wang, Lirong.............................................. 334
Wang, Shuyan............................................ 419
Wang, Sujian.............................................. 334
Wang, Xiangdong....................................... 200
Wang, Yuanyuan.......................................... 98
Wang, Yuqiao....................................... 37, 312
596
Wang, Zhiliang............................................. 51
Wei, Ping.................................................... 210
Wen, Shen.................................................. 352
Wu, Fangjun....................................... 257, 338
Wu, Guo-qing............................................. 295
Wu, Haining............................................... 452
Wu, Tzong-Sun.......................................... 330
Xian, Yi...................................................... 495
Xiao, Jing.................................................... 295
Xiaobing, Song........................................... 554
Xie, Keming............................................... 191
Xie, Yinggang.............................................. 51
Xin, Fan...................................................... 352
Xin, Liu...................................................... 449
Xinying, Wang........................................... 449
Xiuxia, Liu.................................................. 554
Xu, Gang............................................ 247, 495
Xu, Jing........................................................ 28
Xu, Lei........................................................ 215
Xu, Luxiong.................................................. 64
Yan, Hao....................................................... 73
Yan, Huiqiang............................................ 114
Yang, Gang................................................. 485
Yang, Li.............................................. 347, 395
Yang, Miaomiao......................................... 405
Yang, Wenxian........................................... 247
Yao, Xiaolan............................................... 161
Yen, Chien-Cheng...................................... 157
Yu, Song..................................................... 129
Yu, Xiaohao.................................................. 47
Yuan, Fuyu................................................. 150
Yue, Jingxia................................................ 177
Yunqi, Lei................................................... 554
 Author Index - Vol 1
Zaman, Safaa................................................ 81
Zamanifar, Kamran.................................... 276
Zaw, Wint Thida......................................... 316
Ze, Xiangbo................................................ 272
Zeng, Jianqiu.............................................. 218
Zeng, Xianting............................................ 588
Zhai, Zhengang........................................... 485
Zhan, Zhixiong........................................... 210
Zhang, Fan.................................................. 286
Zhang, Hao................................................... 60
Zhang, Heng............................................... 272
Zhang, Hu................................................... 545
Zhang, Jihai................................................ 541
Zhang, Jing................................................. 569
Zhang, Jingjun.................................... 481, 527
Zhang, Jun.................................................. 119
Zhang, Li.................................................... 308
597
Zhang, Lu................................................... 215
Zhang, Wenyu.............................................. 98
Zhang, Yamin............................................. 361
Zhang, Yongqiang...................................... 424
Zhao, Canming........................................... 119
Zhao, Hong................................................. 485
Zhao, Jijun.................................................. 334
Zhao, Xuejun.............................................. 471
Zhao, Yu..................................................... 405
Zhicai, Juan................................................ 449
Zhou, Li...................................................... 409
Zhou, Pan.................................................... 134
Zhou, Yuchen............................................. 218
Zia, Fatima Binte........................................ 291
Zohali, Alireza............................................ 276
Zuo, Wanli.................................................. 150
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