NSSP 2013 Proceedings Soft

ACHARYA NAGARJUNA UNIVERSITY
NAGARJUNA NAGAR - 522 510, GUNTUR (DT), A.P., INDIA
Prof. K. Viyyanna Rao Phone: off: 0863-2346101, 2293238

M.Com., MBA, PhD., MIMA, LII Fax : 0863-2293378, Resi:0863-2233334
VICE-CHANCELOR e-mail: nu_vc@yahoo.co.in
Kodativrao@yahoo.com
website: http:/www.anu.ac.in
VICE CHANCELLORS MESSAGE
I am very happy to note that the University College of Engineering and Technology,
Acharya Nagarjuna University is organizing a International Conference on Navigational
Systems & Signal Processing Applications (NSSP-2013) during December 13th & 14th, 2013. I
understand that the above international conference will promote the areas of study, teaching
and research activities in field of technical education to the participants and budding
technocrats and they will prove their skills equipped with human and social values required to
provide integration of technical skills, social character, leadership, self confidence, creativity
and renovation for mankind and all round growth of an individual. I feel the importance of
troubleshooting in our day-to-day work and for preparing the future academic and
administrative action plan. The Research and Development plays a very crucial role in the
institutional developments. No one can deny the role of technology and It is needless to
emphasize that all round growth of the institution will be possible only when we focus our
attention on state of art teaching and research.
Prof. K. Viyyanna Rao

NAGARJUNA NAGAR - 522 510, GUNTUR(DT), A.P., INDIA
Prof. Y. P. Rama Subbaiah Phone: off: : 0863 - (O)23462293238

LL.M., Ph.D., Fax : 0863-2293378
RECTOR Resi: 0863-2353164
e-mail: rectoranu@yahoo.co.in
RECTORS MESSAGE
I am very happy to note that the University College of Engineering and Technology,
Acharya Nagarjuna University is organizing a International Conference on Navigational
Systems & Signal Processing Applications (NSSP-2013) during December 13th & 14th, 2013.
The conference is concerned with the latest in theoretical, mathematical and scientific
developments in Communications Engineering as well as applications of established techniques
to new domains. In the modern world, this type of conference is very relevant to meet the
emerging fields. I believe that this conference will provide an International Technical forum for
experts and researchers from both the academia and industry to meet and exchange new ideas
and present their findings of ongoing research in various disciplines of Electronics &
Communication Engineering. I am confident that the research papers of the international
conference provide an opportunity to rapidly disseminate research results that are of timely
interest to the engineering fraternity. I warmly welcome all the participants for the fruitful
technical sessions. Hearty congratulations to the organizers and best wishes for the success of
the International conference.
Prof. Y. P. Rama Subbaiah.

Prof. R.R.L. Kantam Phone: off: 0863-2346102

M.Sc., PhD., Fax : 0863-2293320
REGISTRAR cell: +91-9441597927
e-mail: registraranu@yahoo.co.in
REGISTRARS MESSAGE
I Congratulate the University College of Engineering and Technology, for Organizing an

International Conference on Navigational Systems & Signal Processing Applications (NSSP-
2013) during December 13th & 14th, 2013. I am sure that the discussions and proceedings of
this conference will bring to light many issues regarding Navigational Systems & Signal
Processing Applications to achieve some more milestones in the years to come.
My Good wishes to all the participants and I wish the Seminar a grand success.
Prof. R. R. L. Kantam
26th Nov 2013
MESSAGE
I am happy to know that the Electronics and Communication Engineering

Department of the University College of Engineering & Technology, Acharya
Nagarjuna University is organizing a two day International Conference on
Navigational Systems and Signal Processing Applications during 13th-14th
December, 2013.
The deliberations in the areas of Signal processing techniques,

Navigational aids, Cellular communications, Radar Systems and Image
Processing Techniques will definitely bring out the state-of-the-art technologies and the
significance of Navigational services and its vital need towards countrys surveillance.
I wish the conference a grand success.
(G. SATHEESH REDDY)

GOVERNMENT OF INDIA, MINISTRY OF DEFENCE
Defence Research and Development Organization
SURANJAN PAL CENTER FOR HIGH ENERGY SYSTEMS
OS & Scientist H AND SCIENCES (CHESS)
Director P O: VIGNYANA KANCHA
HYDERABAD- 500 069
Tele: 040 2430 2870 Fax: 040 2430 2873
Date: 09 Dec. 2013
It gives me immense pleasure to know that Acharya Nagarjuna University college of

Engineering & Technology, Guntur, in technical cooperation with IEEE, Hyderabad Section is
organizing a 2 day international conference on Navigational Systems & Signal processing
applications (NSSP-2013) during 13-14 Dec 2013.
The conference focuses on an activity which is of great National importance. Precision

Inertial Navigation technology is one of the most crucial technologies that find widespread
applications in both civil as well as defence sectors. India is a member of the elite group of five
nations possessing the inertial sensor technology. India is indigenously producing various
inertial sensors like RLGs and FOGs as well as Accelerometers and is rapidly progressing towards
self reliance in Navigation technologies for naval, airborne and space applications.
The themes identified for discussions during the conference are very relevant to the
field of Navigation and Signal processing. I am also confident that this conference will prove to
be an excellent platform for the scientists, engineers and research scholars from various R&D
organizations, Academia and Industry to interact and exchange their concepts and ideas during
the event.
I am certain that the conference will be amply beneficial to the delegates and
participants since most of the emerging techniques and technologies in the field of Navigation
are proposed to be addressed adequately.
I wish this conference a grand success.
Sd/- (SURANJAN PAL)

UNIVERSITY COLLEGE OF ENGINEERING & TECHNOLOGY
Prof. P. Siddaiah Phone: off: 0863-2346251

M. Tech., PhD., Resi:0866-2586165
DEAN & Convener -NSSP 2013 Fax : 0863-2346246
cell: 9490049565
e-mail: Siddaiah_p@yahoo.com
DEANS MESSAGE
FOREWORD
I greet and welcome the participants to our institution and to this conference with the
hope that, together we will be able to churn out new ideas and suggestions in the area of
Navigational Systems & Signal Processing Applications I do believe that the conference will
bear fruitful results and lay firm ground for future developments. I hope that each participant
will go back with a feeling of academic satisfaction by being actively involved in the discussions
and deliberations.
At first i would like to thank University authorities for their unstinting coordination and
cooperation. With regard to the conference, i would like to convey special thanks to the
Honorable Vice-Chancellor of this University, Prof. K. Viyyanna Rao, for this inspiration and
continuous encouragement to organize this International Conference Successfully. I thank
Rector, Prof. Y. P. Rama Subbaiah and Registrar, Prof. R. R. L. Kantam for their continuous
support to conduct the conference fruitful.
I take this opportunity to particularly thank Sri.Avinash Chander SA to RM, to be the

chief guest for this inaugural function and to deliver the key note address.
I also thank Prof.U.R.Rao, former ISRO chairman an outstanding Scientist with a vision to
be acting as a conference chair and for his guidance at each and every step, to bring the
conference on the stage. I would like to thank Prof. K. Giridhar, IIT, Madras, an Eminent
Academician to be with us as a Technical program Chair.
I would like to extend my heartfelt gratitude to Sri.G. Satheesh Reddy , Director RCI for
his immense support to conduct this conference in a grand way. I also thank Prof.U.B. Desai,
Director IITH ,Prof.Rameswar Rao ,Vice Chancellor of JNTUH. Sri.N. Prahalad Rao, Director ISRO,
Prof Raveendran Parmeswaran , University of malaya and Dr.V.Seshagiri Rao, Associate
Director, SHAR ISRO, Sri. P. Raghavendra Rao, Associate Director, CHES, Sr. G. J. Das, Former
Scientist ISRO, Sri. Jayaram Scientist F , ISTRACT, Dr. R.Sreehari Rao ,former chief controller,
DRDO for their continues support and acceptance to be the Guests of Honour for this
international conference.
I thank IEEE for readily agreeing to be the Technical sponsor of this International
Conference. I would like to extend my thanks Dr. M. Lakshmi Narayana, IEEE for Readily
agreeing to be the technical sponsor. I thank all the sponsors UGC, DRDO New Delhi, ISRO,
Effectronics PvtLtd, Vijaya Entuple technologies, AP State Council of Higher Education, DLD
Logistics,Banglore, Accord-SoftSystems etc. for their financial support to the Conference.
Finally, I thank the advisory, organizing and Technical committees for their commendable job in
realizing this goal.
A total of 300 paper proposals were received from all over the world, in the form of "full
papers". After extensive screening, 90 papers were selected for presentation at the Conference.
These were divided into 15 sessions and sessions would run parallel. We have also included
invited tasks from experts within the country and abroad. I hope that will certainly bring in a
standard and originality to the Conference.
Finally, I take this opportunity to thank all the authors and invited speakers profusely for
contributing and participating in the conference.
I welcome all the participants to imbibe the spirit of the originality and technical content
of the Conference.
I wish the Conference a grand success and I hope this will be a trend setter for future
Conferences.
Dr. P. Siddaiah
CONVENER
NSSP-2013
UNIVERSITY COLLEGE OF ENGINEERING & TECHNOLOGY
Prof. E. Sreenivasa Reddy Phone: off: 0863-2346252

B.Tech(ECE), M.S (EC), M.Tech (CSE), PhD (CSE) Resi:0863-2210484
MIEEE, MISTE, MCS, MBA Fax : 0863-2346246
PRINCIPAL cell: 7893111985
e-mail: edara_67@yahoo.com
esreddy67@gmail.com
PRINCIPALS MESSAGE
In this global scenario we are all aware that Indias secured future lies in making India a
complete knowledge nation. Keeping that in view, we are organizing an International
Conference on Navigational Systems & Signal Processing Applications (NSSP-2013) during
December 13th & 14th 2013. To strengthen India in the field of Science and Technology. The
International Conference also aims to bring in the unexplored areas of research in different
aspects of Navigational Systems & Signal Processing Applications in RADAR, Satellite imaging
and Medicine.
I wish the seminar a grand success.
Prof. E. Sreenivasa Reddy

SESSION-I : VLSI & FUZZY LOGIC
S.No Paper Title and Author Page No
Software Designing and Implementation of VLSI Multipliers
1. 1
Kavita, Umesh Goyal
Real Time Implementation of DSP Based Fuzzy Logic Controller for
2. Speed Control of DC Motor 7
Ms.Madhavi Mallam , M. Bhanu Sri, Dr.A.GuruvaReddy
Implementation of Spartan 3E FPGA based Multi Serial to Ethernet
3. Gateway for DAS using VHDL 13
A Swetha, E. Purushotham Reddy
Design and Implementation of FPGA based Wi-Fi MAC Transmitter
4. 20
G. Padma Ratna, S Naga kishore Bhavanam, K. L. Bhavani
Goertzel Algorithm based DTMF Detection Using FPGA
5. 29
S Nagakishore Bhavanam, Dr. P. Siddaiah, Dr. P. Ramana Reddy
Implementation of digital Communication Laboratory on FPGA
6. 34
Molabanti praveen kumar, T S R ksishna prasad, M .Vijaya kumar
SESSION-II : ANTENNAS-I
Design of Non-Uniform Circular Arrays for Side lobe Reduction Using
1. the Method of Real Coded Genetic Algorithm 39
Ms.M. Nirmala
Effect Of Dielectric Superstrate On The Characteristics Of Square
2. Patch Antenna 44
V. Saidulu, K. Srinivasa Rao, P.V.D. Somasekhar Rao
Analysis On Flexural Vibration Characteristics of Hemispherical Shell
3. 52
Shailesh kumar,Anil badiger, Kishore chandra das, Runa banerjee
Effect of dielectric superstrate on the characteristics of Rectangular
4. Patch antenna 57
V. Saidulu, K.Srinivasa Rao, P.V.D.Somasekhar Rao
Design of Printed Dipole antenna Integrated with Resonator and
5. Defected Ground Structure (DGS) for improving noise rejection 65
Takeshore Khumanthem, C.Sairam, S.D.Ahirwar and M.Santha
Design of Dual Frequency Circular Patch Antenna Using HFSS
6. 70
Akhila John. Davuluri, Dr.P.Siddaiah
SESSION-III : SATELLITE COMMUNICATIONS-I

Open Loop Tracking Technique for Weak Signals in GPS Radio
Occultation Receiver
1. 74
Namit Rarotra, Hemanth Kumar Reddy N, Chitra Ramamurthy, Yateendra
Mehta
Acquisition of Modernized GPS L2C signal
2. Kishore Babu N, Hemanth Kumar Reddy, Chitra Ramamurthy, Yateendra 78
Mehta
Acquisition Techniques of Modernized GPS L1C BOC(1,1) Signal
3. Kishore Babu N, Mukesh Kumar Gupta, Hemanth Kumar Reddy, Chitra 82
Ramamurthy
TEC Response Due to Geomagnetic Storm at Low Latitude GPS Station
4. 87
N.S.M.P.Latha Devi, D.VenkataRatnam, S.Venkatesh
Implementation of CIC Filter based DDC for Extraction of Delta DOR
5. Signals 91
K R Yogesh Prasad, Priyanka Das, Jyothi B R, Chitra Ramamurthy
Relative Characterization of Steered Timescales for Traceability
6. Anu Arora, Aakanksha Avnish Bhardwajan, Suresh Dakkumalla, A. Amala 97
Agnes, T. Subramanya Ganesh, V. Jayaraman, L. Srinivasan
SESSION-IV: WIRELESS SENSOR NETWORKS-I

Reconstruction Algorithms for Compressive Sensing
1. 102
T.J.V.Subrahmanyeswara Rao, M.Venu Gopala Rao
Design of a Readout Circuit for MEMS Based Seismometer
2. V Thamarai, Etika Agarwal, Gaurav Dutta Saxena, Kalpana Arvind, A S 108
Laxmi Prasad, Dr G Nagendra Rao
A Survey on Poly House Cultivation Using Wireless Sensor Networks
3. 113
Prathiba Jonnala
Performance of Cooperative Wideband Spectrum Sensing and
4. Localization Using Radio Frequency Sensor Networks 117
P.Dhana Lakshmi, K.Murali
Wireless Multi-Flow Sensor Water Consumption Monitoring System
5. using Zigbee for Gated Communities 124
G.S.R.Satyanarayana, K.Vijaya Kumar
Opportunistic Spectrum Access for Prioritized Secondary Users in
6. Cognitive Radio Networks 128
K.Annapurna, V. Harshavardhan, Dr.B.SeetaRamanjaneyulu
SESSION-V: IMAGE PROCESSING-I

FPGA Implementation of Image Filtering Applications Using XGS
1. 133
N.Vani, N. Usha Rani
Morphological Image Segmentation Using marker-Controlled
2. Watershed Transformation 137
Dibyendu Ghoshal, Pinaki Pratim Acharjya
Study Of Watershed Approach For Morphological Based Gradient
3. Image Acquisition 144
Dibyendu Ghoshal, Pinaki Pratim Acharjya
Image Bin Based Full-Reference Color Image Quality Measure Using
4. SM-DCT 149
D.V.N. Sukanya, D. Nagesh Babu, N. Swathi
Filtering Respiration and Low Frequency Artifacts in Impedance
5. Cardiography 153
Dr. K. Chandra Bhushana Rao , Ms.Madhavi Mallam , Dr.A.GuruvaReddy
IRIS Pattern Matching and Recognition for Eunuch in India by the
6. Method of Cumulative Sum Based Change Analysis 158
Parthsarathi De, Dr. Dibyendu Ghoshal, Parimita malla
Digital Implementation of Geometric Moments for Image Processing
7. 161
Ahlad Kumar, Raveendran Paramesran, Chern Loon Lim
SESSION-VI : NAVIGATIONAL SYSTEMS & RADARS-I
Development of BOC/Interplex Modulator with FPGA & Novel RF
1. Design for Regional Navigation Satellite System 166
Rahul Gupta, Sohan M. Patel, Dipti R. Patel, T.V.S. Ram
Navigation Filter for Real Time Orbit Determination of Leo Spacecrafts
2. Namit Rarotra, Hemanth Kumar Reddy N, Chitra Ramamurthy, Yateendra 170
Mehta
IRNSS One Way Measurement Simulation
3. Chandrasekhar M.V, Kalasagarvarma.K, Manamohan Kamat, 175
Satyanarayana.G, Nirmala.S, S.C. Rathnakara, A.S. Ganeshan
FPGA Based Design of NSGU to achieve Timing Delay Bias and Time
4. synchronization in nanoseconds for Navigation Satellites 182
Durga Digdarsini, Gopichand Khot, Sanjay D. Mehta, T.V.S Ram
IRIS Signal Dispensation and Cataloging by Means of Compound
Gabor Filter
5. 185
Prof. P.V. Rama Raju, R.DivyaSree, P.Lakshmi, Y.L.Prannetha and
Mr. CH. Ganesh
In Orbit Clock Performance of IRNSS RAFS
6. Kaitha Rajaiah, Umesh Swami, Manamohan Kamat, Nirmala S, S C 190
Ratnakara, A S Ganeshan
SESSION-VII : VLSI & FUZZY LOGIC-II

Embedded Extended Visual Cryptography Schemes
1. 194
Prof. M Rama Bai, C.Raghunandhan Reddy
FPGA Implementation of Low Power, Area Efficient Radix 8 Multiplier
2. using Verilog HDL 197
V V S Vijaya Krishna, T.Pitchaiah, Sk. Sadulla, P V Sridevi
Implementation of stereo vision system using Blackfin Processor
3. 201
N.Prasad, M.Rajakumari, P.Sitaramanjaneyulu, T.Venkata rao
Implementation Of Energy Efficient And Low Complex FIR Filters
4. With Reconfigurability Usage 206
R. Vijayalaksmi, P. Radhika Ramya, Y Sudha Vani
Handwritten Digit Recognition with MLP Neural Network by Using
5. Back Propagation Training with Tunneling Algorithm 209
P.P.S.Subhashini, Dr. M.Satya Sairam, Dr.D. srinivasarao
SESSION-VIII: SIGNAL PROCESSING

Signal Processing Schemes for Interferometric Fiber Optic Gyroscope
1. Kalpana Arvind, D.Praveena, M.Gayatri, M.N.Srinivas, Lekshmi.S.Rajan, 217
V.V.Lakshmipathy, A S Laxmi Prasad
Design of Optimal Digital FIR Low Pass Differentiator by using PSO
2. 222
Venkatappareddy.P, M.Subramanyeswara Rao
Design and Simulation of Pipelined FFT Processor using FPGA
3. 225
N.Amarnath Reddy, D. Srinivasa Rao
Development of an Algorithm for 8 bit WTM
4. 230
Sravanthi.kantamaneni, Dr.V.V.K.D.V.Prasad,Veera Vasantha Rao.Battula
FPGA Implementation of Least Mean Square Algorithm for Speech
5. Enhancement 235
Ravi Bolimera, M. Paul Sandeep
Design Of Energy Efficient Logic Using Adiabatic Technique
6. 239
K B V Babu, B.I. Neelgar
Performance Analysis of Linear Phase Symmetric FIR QMF Bank
7. 243
Qusay Zalil Kadhim, B.Bhaskara Rao, K.S.Rajasekhar
SESSION-IX: COMMUNICATION SYSTEMS-I

PAPR Reduction And Channel Estimation In MIMO-OFDM Systems
1. Using Adaptive Active Constellation Extension And DWT 248
N.Renuka, Ch.Loknadh, Murali Krishna
Adaptive Channel Equalizer For Wireless Communication Systems
2. 252
S.Dileep kumar, G B S R Naidu, V Jagan Naveen
Generation of Sequences with Good Correlation Properties Using
3. Modified Particle Swarm Optimization 257
S.Srinivasa Rao, Dr.P.Siddaiah ,K. Raghu
Hartley transform-based block FSLMS Algorithm for active noise
4. Control 261
T. Srinivasa rao
High Bit-Rate X-Band QPSK Modulator Realization
5. and Analysis 266
Jyothi B R, K R Yogesh Prasad, Priyanka Das, Chitra Ramamurthy
SESSION-X : WIRELESS SENSOR NETWORKS-II

On the positioning of low-pass interpolator for channel estimation in
1. OFDM systems 265
Pallaviram Sure, Chandra Mohan Bhuma
Taxonomy Of Privacy Preserving Techniques In Wireless Sensor
2. Networks: A Survey 276
G.Sumalatha,N.ZareenaM.Parimala
Variance based multiservice DBA scheme for GPON Access
3. Networks 283
D Khasim Hussain, and R manivasakan
FPGA Based Excitation Generation for Navigation Sensors
4. Deepthy C K, Rajesh George P, Raghunath K P, Mary Jermila M, Aparna 288
Vishnu N,B. VenkataRamana, Dr. P P Mohanlal
SESSION-XI : IMAGE PROCESSING
An Approach to Morphological Image Segmentation For Noisy Digital
1. Image 291
Dibyendu Ghoshal, Pinaki Pratim Acharjya, Uddalak Chatterjee
Wavelet Transform Based Texture Features For: Content Based Video
2. Retrieval 296
Mrs.G.Prathibha, Mrs. K. Chaitanya
Medical Image segmentation using Rough Sets with optimized centroids
3. 299
N.Anupama, Dr. E. Srinivasa Reddy, Dr. Srinivas Kumar
Audio Authentication and Copyright Protection Using DWT and Arnold
4. Transform 306
S Suresh Kumar, N.V.Lalitha
Visual Enhancement for better quality of Images and Displays using
5. skeletonization under Low-Backlight 311
Sirisha.M, M.Padma Sailaja, P.V.Subbaiah
An Optical Flow Approach to Robust Face Recognition Under
6. Expression Variations 316
Mr. V. Muralikrishna, Dr. M. Madhavi Latha, Dr. M. Sathya Sai Ram
A Joint Encryption and Watermarking System
7. 323
B. Santosh Kumar, T. Geetamma
Development of Embedded Vision Enhancement System using
8. Real Time Image Fusion 329
Sri Swathi Nimmagadda, Mr. Kanakaraju K
SESSION-XII: NAVIGATIONAL SYSTEMS & RADAR-II

Pseudo Random Noise Code Selection Criteria for GNSS
1. 334
G. Sai Anirudh, G.Shanmukh
Performance Assessment of IRNSS Signals:BPSK(1) and BOC(5,2)
2. 343
T. Tashi1, G. Ramesh, L. Mruthunjaya
Sizing Design of Laser Retro Reflector Payload and ILRS Network
Planning Analysis for IRNSS
3. 349
Anu Arora, T. Subramanya Ganesh,V. Jayaraman, L. Srinivasan, B. S.
Chandrasekhar
An Adaptive Filter Design For Side Lobe Reduction In Pulse
4. Compression Radar System 354
Sadhu Sadhana, D.Tirumala rao
Performance Evaluation of Non-Uniform PRF Radar Signals Using
5. MOGA 358
B Jithendra Kumar Reddy, G Manmadha Rao
Laser Assisted Vision Based Tracking for Gimbal Control
6. 362
Srikanth S S, Mr K Kanakaraju
A Novel Architecture for Real Time Ultra-Tight Integration of
Miniaturized GPS-GLONASS-GAGAN L1 Receiver and MEMS
7. 368
Inertial Sensors
Gundra Satheesh Reddy, Manjit Kumar
Unique NAV data Design for Optimal TTFF in IRNSS Receiver
8. 374
Dr. Vyasaraj Guru Rao, Sashidharan MAA, Sumadhva B Pappu
SESSION-XIII: COMMUNICATION SYSTEMS-II
Inter-Carrier Interference Reduction in OFDM System
1. 379
Velamala.suryanarayana, vadaravu jagan naveen
Successive Interference Cancelation Based Detection Algorithm for MIMO-
2. OFDM Systems 383
Panchumarthi V Bhargavi, K.V.S.N.Raju, Buddaraju Revathi
Bit Error Rate Performance of MIMO- CDMA System for Wireless
3. Communications by Using Space Time Block Codes 389
T.Nageswara Rao, M.Sandeep
Haar Wavelet-Based BPSK OFDMMIMO System Null Channels with
4. Reduced PAPR 392
RatnaKumari Bitra, P.Subba Rao
Power Control Factor Selection in Uplink OFDMA Cellular Networks
5. 397
Suman Kumar and K. Giridhar
SESSION-XIV: ANTENNAS
Electronically Steerable Planer Phased Array Antenna using HFSS
1. 401
Vasujadevi M, Dr. P. Siddaiah, P.Sai Vinay Kumar
Analysis and Performance Comparison of DOA Algorithms for Continuous
Tracking And Quality Enhancement Of Desired Signal In Wireless Services
2. 406
With Smart Antenna Technology
M.Siva Ganga Prasad, KVR Chowdary, P.Siddaiah, P.Sai Vinay Kumar
Design and Analysis of C-Band Rectangular Microstrip Patch Antenna
3. 414
B. Sekharbabu , BV. Subba Rao , I. Kullayamma, Dr.R.V.S. Satyanarayana
The Dielectric Loss Tangent Effect on Radiation Characteristics of Co-
4. axially Feed Rectangular Patch Antenna: Design & Analysis 418
Ch. Raja, K. Praveen Kumar, Dr. M. Madhavi Latha, Dr.E.G.Rajan
SESSION-XV: IMAGE PROCESSING-II

A video watermarking technique using Discrete Cosine Transform
1. 423
Dokuparthi Jagadeesh , Dr.K.Gauthami
A Novel Approach for Denoising of Medical Images corrupted with
2. Poisson Noise 428
V. Naga Prudhvi Raj,Dr. T. Venkateswarlu
An Efficient Data Mining Tool for Medical Image Diagnosis Using soft
3. computing Technique 434
G. madhavi
Authentication and Self-Repair Capability for Fixing Tampered Image
4. Data 441
V.Balaji , Anusha Begum , Prof E.Sreenivasa Reddy
Comparative Analysis of Block Matching Algorithms for Motion
5. Estimation 445
V.S.N.Kumar Devaraju, B. Prasanthi
Spatial domain image enhancement for roi Using local parameterized
6. hybrid model 453
I.Suneetha, Dr.T.Venkateswarlu
Content Based Video Retrieval By Integrating Shape And MotionFeatures
7. 456
GS Naveen Kumar, Dr. VSK Reddy, Dr. S.Srinivas Kumar
International Conference on Navigational Systems & Signal Processing Applications, December 13th & 14th,2013
University College of engineering & Technology, Acharya Nagarjuna University, Nagarjuna Nagar-522510,A.P.,India
Software Designing and Implementation of

VLSI Multipliers
Kavita1, Umesh Goyal2
Electronics and Electrical Communications Department
PEC University of Technology
Chandigarh, India
Umesh.goyal330@gmail.com
AbstractThis paper presents the design of an array one bit of the multiplier. These partial products are shifted
multiplier, Wallace Tree multiplier and Booth in accordance to their bit orders and are then added
multiplier. Due to the advancements of technology in together [3]. The addition operation is performed with the
the area of signal processing, graphics, image and video Carry Propagate Adder. Number of adders required for the
processing etc. multipliers play an important role in purpose of multiplication is N-1 where N is the multiplier
these fields. Due to the numerous applications of length. The Basic multiplication process is shown in Fig. 1.
multipliers in different areas, the research is going in
this direction to achieve highly efficient, fast, low cost
multipliers. This paper presents the software design of
an array multiplier, Wallace Tree Multiplier and Booth
Multiplier with the help of flow charts. These
multipliers are implemented using Xilinx ISE Design
suite 14.2 and thus compared with each other based on
different criteria like number of look up tables used,
number of input output blocks used and others.
I. INTRODUCTION
Due to the advancement in technology for fast floating

point co-processors, digital signal processing chips, and
audio, video processors, there is a need for high speed, low
Figure 1. Array Multiplication
power, and area-efficient multipliers. Current architectures
of multipliers range from small, low-performance
multipliers, to large, high-performance Even though the method of multiplication is simple the
multipliers.Conventional linear array multipliers achieve addition is performed serially as well as in parallel. Carry
high performance than shift and add multipliers in a propagate adders are generally replaced with the Carry
regular structure, but require large amounts of silicon [1]. Save Adders in order to improve chip size and delay, in
Tree multipliers achieve even higher performance than which every carry signal and sum signal obtained at each
array stage is passed to the adders of the next stage. Final result
multipliers but the tree interconnection is complex and les is obtaine in a final adder by using any one of the fast
regular, thus making them even larger than linear arrays adder (basically ripple carry adder). In array multiplication
Ideally, a multiplier should have speed benefits, small size we add as many partial products as there are multiplier bits
and low power consumption. This paper describes the present. Array multiplier can be categorized according to
architecture of an Array, Wallace Tree and Booth multiplie the carry propagation into two basic structures. These two
and also software design of all these multipliers and at last structures are Ripple-Carry Array (RCA) and Carry-Save
comparing these multipliers on the basis of number of Array (CSA).
LUT used, number of input output blocks used that tells
the percentage utilization of the resources [2]. 1) Ripple Carry Array Multiplier- A ripple carry array
multiplier performs the multiplication using shift and add
II. ARRAY MULTIPLIER multiplication algorithm. Fig. 2 shows the basic structure
of ripple carry array multiplier that how the multiplication
Array Multiplier is known for its simplest and the most operation is performed and the carry is propagarted from
basic regular structure. Multiplication operation performed one stage to another. Carry is propagated in the form of
by array multiplier is simply add and shift. Partial products ripple therefore it is named as ripple carry array multiplier.
are generated by the multiplication of the multiplicand by The maximum delay generated in ripple carry array
1
multiplier is from LSB of the input to the MSB of the but the carry input in Full Adder becomes one of the input
product. The delay generated is 2*n. in CSA and the carry output from each bit becomes one of
the output along with the sum in CSA rather than being
moved to the next MSB [4]. It takes three numbers (i, j, k)
as input to add together and gives carry (c) and sum (s) as
output. In carry save adder, the The carry and sum is
brought to the last stage and are then added together by
ordinary addition. Carry Save Adders are used to obtain
partial products in integer or bit wise multiplication, whic
allows these adders to be used in those architectures where
a tree of carry save adders are needed and hence producing
fast partial product terms. Carry bits and the last partial
products are added by the normal adder to give the final
result of multiplication. For this last stage usually a carry
select adder or a carry look ahead adder is used for optimal
performance of the multiplier. By using carry save adder,
carry propagation in the adder is avoided due to which one
Figure 2. Basic Structure of Ripple Carry Array Multiplier gate delay is reduced because the gate delay of adder is
equal to the latency of one addition.
2) Carry Save Array Multiplier- A carry save array The multiplier implementation basically involves three
multiplier uses a carry save adder to add three or more steps as shown in Fig. 4. They are Partial-Product
binary numbers of n-bits. In ripple carry array multiplier generator, Wallace Tree structure and ordinary adder.
there was only one output but in carry save-array
multiplier there are two outputs as shown in Fig. 3. One is
partial sum bits and other is carry bits. These two bits are
then added together to produce the final result.
Figure 3. Carry Save Adder
III. WALLACE TREE MULTIPLIER
Wallace established the fast process for multiplication

of two numbers. C.S. Wallace in 1964 observed that a Figure 4. Block Diagram of Wallace Tree Multiplier
structure can be made in which addition can be performed
in parallel which results in lesser delay. Wallace Conventional array multipliers achieve comparatively
introduced parallel addition of partial product bits using a good performance but they require large area of silicon,
tree of Carry Save Adders. This tree of Carry Save Adders unlike the add-shift algorithms, which require less hardwar
is known as Wallace Tree. Wallace tree is the hardware and exhibit poor performance [5]. The Booth multiplier
implementation of digital circuit which multiplies two makes use of Booth encoding algorithm in order to reduce
numbers ver efficiently. the number of partial products by considering two bits of
the multiplier at a time, thereby achieving a speed
In order to multiply two numbers by Wallace method, advantage over other multiplier architectures. This
partial product matrix is reduced to two row matrix with algorithm is valid for both signed and unsigned numbers.
the use of Carry Save Adders. remaining two rows are then Based on radix-2 computation booth multiplier accepts the
added using fast carry propagate adder in order to obtain number in 2's complement form [6]. Booths algorithm
the final result. Wallace Tree architecture uses Carry Save reduce the number of partial products
Adder for fast multiplication to achieve the final product. to half due to which the speed increases by the reduction of
the total number of partial product sums [7]. Booth
CSA performs the addition of n numbers in less time as Multiplication algorithm is a multiplication algorithm that
compared to simple addition. CSA is similar to Full Adder multiplies two signed binary numbers in twos
2
complement form. Let us illustrate the booth algorithm TABLE II. STEP 2 IN BOOTH ALGORITHM
with the following example that multiplies (3)10 x (-6)10.
1) Making the Booth table- From the two numbers pick the
number with the largest difference between a series of
consecutive numbers and make it a multiplier. i.e., 0011-
from 0 to 0 no change, 0 to 1 one change, 1 to 1 no
change, and so there is only one change in this one. 1010--
from 1 to 0 one change, 0 to 1 another change, 1 to 0 one
more change, so there are total three change on this one.
Hence, multiplication of 3 x (6), where 310 (0011)2 is the
multiplicand and (-6)10 (1010)2 is the multiplier.
Let X = 1010 (multiplier)
Let Y = 0011 (multiplicand)
Take the 2s complement of Y and call it Y.
-Y = 1101
Load the X value in the table. Load 0 for X-1 value it
should be the previous first least significant bit of X. Load
0 in U and V rows which will have the product of X and Y
at the end of operation. Make four rows for each cycle
because we are multiplying four bits numbers [8].
TABLE I. STEP 1 IN BOOTH ALGORITHM
2) Booth Algorithm:-
Booth algorithm requires examination of the multiplier
bits, and shifting of the partial products [9]. Before the
process of shifting, the multiplicand may get added to the
partial product, get subtracted or left unchanged in
accordance to the following rules: Look at the first least The four cycles are finished, so the answer is obtained in
significant bits of the multiplier X,and the previous least the last rows of U and V which is: 111011102,
significant bits of the multiplier X -1.
V. IMPLEMENTATION OF MULTIPLIERS
0 0 Shift only
1 1 Shift only This section of the paper covers the implementation
0 1 Add Y to U and then shift of
1 0 Subtract Y from U and then shift or add (-Y) to multipliers with the help of flow chart.
U and shift
1) Array Multiplier
Take U & V together and shift arithmetic right shift which a) Method 1 to implement array multiplier :- The steps to
preserves the sign bit of 2s complement number such that carry out an array multiplication are written as below.
positive number remains positive and a negative number 1. Initiate P=00000000, A= 00001001, B= 1101 and n=4
remains negative. Shift X circular right shift because this 2. Check for if B (0) =1 or not. Here in this case B (0)=1 so
will prevent from the use two registers for the X value. P= P + A then P becomes 00001001 and move to next
step. If B (0) 1 then directly move to next step.
3
3. Shift A left by one bit. So, A becomes 00010010

4. Shift B right by one bit. So, B becomes 110 2) Wallace Tree Multiplier:- As already expalined earlier
5. Reduce n by 1. So, n becomes 4-1=3 that wallace tree architecture makes use of carry save adder
6. Now check whether n=0 or not. If n 0 then repeat the to obtain the final result. So we will first show how to
same procedure from step 2 till n becomes 0. After implement carry save adder and then the architecture of
carrying out all the steps we obtain the result in P as wallace tree using carry save adder.
the product of A and B. In this example P comes out
to be 01110101.
Figure 7. Flow Chart to obtain carry save adder components

Figure 5. Flow Chart of an Array multiplier
3) Booth Multiplier:-
b) Method 2 :-
Figure 6. Flow chart of method 2 of array multiplier
4
Figure 8. Flow Chart of Booth Multiplier
VI. RESULTS
The results of multipliers are obtained using Xilinx ISE

Design Suite 14.2.
1) Array Multiplier (8x8 Multiplication) :-
Figure 9. 8x8 multiplication using Array Multiplier
2) Wallace Tree Multiplier:-
Figure 10. Carry Save Adder
5
[2] Lakshmanan, Masuri Othman, Mohamad Alauddin Mohd.Ali,

High Performance Parallel Multiplier using Wallace-Booth
Algorithm, IEEE, 2002.
[3] Norhuzaimin Julai, Yee Ming Fatt,Fast Processing
Multiplier, IEEE 2006 International RF And Microwave
Conference Proceedings, September 12 - 14, 2006, Putrajaya,
Malaysia.
[4] Rizalafande Che Ismail, Razaidi Hussin, High Performance
Complex Number Multiplier Using Booth-Wallace Algorithm,
IEEE, ICSE2006 Proc. 2006, Kuala Lumpur, Malaysia.
[5] Shiann-Rong Kuang, Jiun-Ping Wang, Design of Power-
Efficient Configurable Booth Multiplier, IEEE Transactions on
Figure 11. Wallace Tree Multiplier Circuits and SystemsI: Regular Papers, Vol. 57, No. 3, March
3) Booth Multiplier:- 2010.
[6] Yuan-Ho Chen, Chung-Yi Li and Tsin-Yuan Chang, Area-
Effective and Power-Efficient Fixed-Width Booth Multipliers
Using Generalized Probabilistic Estimation Bias, IEEE Journal on
Emerging and Selected Topics in Circuits and Systems, Vol. 1, No.
3, September 2011.
[7] Jagadeshwar Rao M , Sanjay Dubey, A High Speed and Area
Efficient Booth Recoded Wallace Tree Multiplier for fast
Arithmetic Circuits, IEEE 2012 Asia Pacific Conference on
Postgraduate Research in Microelectronics & Electronics
Figure 12. Booth Multiplier (PRIMEASIA).
[8] Nishat Bano, VLSI Design of Low Power Booth Multiplier,
VII. COMPARISON BETWEEN MULTIPLIERS International Journal of Scientific & Engineering Research,
Volume3, Issue 2, February -2012.
The array multiplier, Wallace Tree multiplier and [9] Wen-Chang Yeh, Chein-Wei Jen, High-Speed Booth Encoded
Parallel Multiplier Design, IEEE Transactions on Computers,
Booth Multiplier are compared based on number of Vol. 49, No. 7, July 2000.
Look up tables used, number of occupied slices and
number of input output blocks used. The percentage
utilization of resources by Wallace Tree Multiplier is
least as compared to other multiplier. This is shown
as below.
REFERENCES
[1] Shugang Wei, Shuangching Chen, Kensuke Shimizu, Fast

Modular Multiplication Using Booth Recoding Based On Signed-
Digit Number Arithmetic, IEEE, 2002.
6
Real Time Implementation of DSP Based

Fuzzy Logic Controller for Speed Control of
DC Motor
Ms.Madhavi Mallam1 , M. Bhanu Sri 2, Dr.A.GuruvaReddy 3
1,3
Department of Electronics & Communications
DVR & Dr.HS MIC College of Technology
Kanchikacherla, Krishna District, India
2
Student, M.Tech, Digital Electronics & Communications Systems
1
madhavi.418@gmail.com
2
bhanusri.mididodla@gmail.com
3
guruvareddy78@gmail.com
Abstract The main objective of this paper is inference system was presented to control a steam
controlling the speed of DC motor when the load is engine and boiler combination by linguistic rules.
varying. Various methods of speed control of speed Fuzzy logic is expressed by means of if-then rules
control of DC motor is available. This paper presents with the human language. In the design of a fuzzy
design and implementation of Fuzzy Logic in the speed logic controller, the mathematical model is not
control of DC motor. Fuzzy logic has found high necessary. Thus, the fuzzy logic controller owns good
applications as a speed control techniques because of its robustness. Fuzzy controller has been widely used in
ability to take into account vague and uncertainties . industry for its easy realization. However, the rules
this paper presents a MATLAB simulink model for and the membership functions of a fuzzy logic
speed control of DC motor using fuzzy logic. in this controller are constructed by expert experience or
paper TMS320F28335 DSP is used as the controller knowledge database.
necessary signal conditioning components are used to
ensure high processing speed and precision in the II. MODELING OF DC MOTOR
overall control system. The implemented system has a
A common actuator in control systems is the DC
motor. It directly provides rotary motion and, coupled
fast response with small overshoot and zero steady state
with wheels or drums and cables, can provide
error compared to conventional PI controller.
translational motion. The electric circuit of the
I. INTRODUCTION armature and the free-body diagram of the rotor are
shown in the following fig.1:
In spite of the development of power electronics
resources, the direct current machine became more
and more useful. Now a days their uses isnt limited
in the car applications (electrics vehicle), in
applications of weak power using battery system
(motor of toy) or for the electric traction in the multi-
machine systems too.
The speed of DC motor can be adjusted to a great
extent as to provide controllability easy and high
performance. The controllers of the speed that are
conceived for goal to control the speed of DC motor Fig. 1.Seperatly Exited DC Motor
to execute one variety of tasks, is of several For this example, we will assume that the input of
conventional and numeric controller types, the the system is the voltage source (V) applied to the
controllers can be: PID Controller, Fuzzy Logic motor's armature, while the output is the rotational
Controller; or the combination between them Fuzzy- speed of the shaft d(theta)/dt. The rotor and shaft are
Genetic Algorithm, Fuzzy-Neural Networks, Fuzzy- assumed to be rigid. We further assume a viscous
Ants Colony, Fuzzy-Swarm (Swarm). friction model, that is, the friction torque is
Fuzzy theory was first proposed and investigated proportional to shaft angular velocity.
by Prof. Zadeh in 1965. The Mamdani fuzzy
7
The physical parameters for our example are:

Parameter Value
E s Eb s
R-resistance 1
I s s a
6
L-inductance 29.79 mH
a
L R a a
T s sJ . s B. s
J-moment of inertia 0.01 kg.m2
Kt -torque constant 0.052 Nm/A m m m
Kb-electromotive force constant 0.1 V/rad/s
B-viscous
T s
friction coefficient 0.004
N.m/rad/s
In general, the torque generated by a DC motor is s sJ B
m
m
7
proportional to the armature current and the strength
of the magnetic field. In this example we will assume
that the magnetic field is constant and, therefore, T s K .i s
m T a
8
that the motor torque is proportional to only the
armature current i by a constant factor Kt as shown in
the equation below. This is referred to as an E s k . s
b b m
9
armature-controlled motor. The angular acceleration is equal to 1 / J multiplied
Here, the differential equation of armature circuit is- by the sum of two terms (one positive, one negative).
di t t
Similarly, the derivative of current is equal to
E t R I t L 1
a 1/ L multiplied by the sum of three terms (one
a a a E a b positive, two negative).The final design should look
d t like the example shown in the figure below. The
following fig.2. shows the modeling of DC motor,
The Torque equation is- which is obtained from the above mathematical
analysis using MATLAB/Simulink.
d t
T t J
m
m
B. m t 2
d t
The torque developed by motor is proportional to the

product of the armature current and field current i.e.
T t k i i
m f f a
3
Where , Kf is the constant. In armature-controlled D.C
motor the field current (if) is kept constant i.e.
T m
K T .i a 4 Fig. 2. Modeling of DC Motor
III. FUZZY LOGIC CONTROLLER

Where, KT=Kf.if is torque constant. The back e.m.f. of
motor is proportional to the speed i.e. A. Block diagram and its description
E t K . 5 Fuzzy logic control is a control algorithm based on

b b m a linguistic control strategy, which is derived from
expert knowledge into an automatic control strategy.
Fuzzy logic control doesn't need any difficult
Where, Kb is back e.m.f. constant. In order to create mathematical calculation like the others control
the block diagram of system initial conditions are system. While the others control system use difficult
zero and Laplace transform is implemented to the mathematical calculation to provide a model of the
equation. i.e. controlled plant, it only uses simple mathematical
calculation to simulate the expert knowledge.
Although it doesn't need any difficult mathematical
E s R .l s
a a a
S L .l s E s
a a b calculation, but it can give good performance in a
8
control system. Thus, it can be one of the best The fuzzy logic controller for speed controller for
available answers today for a broad class of DC motor was implemented in MATLAB/Simulink.
challenging controls problems. A fuzzy logic control Block diagram show the simulation model.
usually consists of the following:
Fuzzification: This process converts or transforms
the measured inputs called crisp values, into the fuzzy
linguistic values used by the fuzzy reasoning
mechanism.
Knowledge Base: A collection of the expert control
rules (knowledge) needed to achieve the control goal.
Fuzzy Reasoning Mechanism: This process will
perform fuzzy logic operations and result the control
action according to the fuzzy inputs.
Defuzzification unit: This process converts the result
of fuzzy reasoning mechanism into the required crisp
Fig. 5.Modeling of Fuzzy Logic Controller
value
The above fuzzy logic controller modeling has two
subsystems. One is DC motor, which shows fig.2.
And another is IGBT switch, shown in the following
fig.6.
Fig. 3.Basic Block Diagram For FLC

Fig. 6. Modeling for IGBT Switch
D. Fuzzy Controller Design

B. Design procedure for fuzzy logic controller
The Fuzzy Logic Controller is an seven step

procedure to control the speed of a DC motor. Those
steps are mentioned in the following fig.4.
The most important things in fuzzy logic control

system designs are the process design of membership
functions for inputs, outputs and the process design
of fuzzy if-then rule knowledge base [2]. They are
very important in fuzzy logic control. The basic
Fig. 4. Flow chart for FLC structure of Fuzzy Logic Controller is given in Fig.5.
For the D.C. drive, speed error (e) and change in
C. Simulink Model for FLC speed error (ce) are taken as the two inputs for the
9
fuzzy controller and one output (z) . For this, a two-

member as well as a five-member rule base is
devised. The rule base for five membership functions
is shown in TableI.
E. Rule View
F. Surface View
In this paper as mentioned in table1,it consist of two

input variables(e & ce), one out variable (z), and five
linguistic variables {NB,NS,ZE,PS,PB} described in
TABLE II.so that here we can have 25 (no.of Fig. 9.Surface View of FLC
linguistic variables ^ no.of input variables = 5^2 =
25) rules totally. As mentioned in fig.5,we can plot
the membership functions for both inputs and
outputs. Here TSK(Takagi Sugino Kang) method was
used to plot the membership function.
10
IV. SIMULATION RESULTS
V. CONCLUSION
A fuzzy logic based controller for a D.C.

servomotor has been studied. The design of the fuzzy
logic controller has been explained and the
performance was evaluated by simulation. The results
show significant improvement in speed .The above
results shows the comparison between speed of the dc
motor before applied to the Fuzzy Logic Controller,
which is shows in the above fig.2. and speed of the dc
motor after applied to the Fuzzy Logic Controller,
shown in fig.5. The fig.10. shows the results for the
simulink block diagram of fig.2. and fig.11. shows
the results for the simulink block diagram of fig.5. By
comparing fig.10 & 11, fig.11. has the better speed,
current and torque response. So we can say FLC is
an better speed controller.
ACKNOWLEDGMENT
Firstly, Author would like to pay Special thanks
to Department of Science & Technology (DST),
Ministry of Science and Technology, Government of
India, New Delhi, and thanks to Faculty and
Management of MIC College of Technology,
11
Kanchikacharla, and also to the government of India, Devineni Venkataramana & Dr.HimaShekar MIC College
for encouragement. of Technology, Kanchikacherla ,A.P, India.
REFERENCES Dr. A. GuruvaReddy is working as Professor, Dept. of

ECE, DVR & Dr.HS MIC College of Technology,
[1] www.mathworks.in/help/fuzzy logic controller.html. Kanchikacharla, Krishna (D). He received his B.E (ECE)
[2] Dipraj, Dr. A. K. Pandey, Speed Control Of Dc. degree from the University of Mysore, Karnataka, M.E
Servo Motor By Fuzzy Controller International (Communication Systems) from the Anna University and
Journal Of Scientific & Technology Research Volume Ph.D from JNTU Ananthapur. His area of interests is Image
1, Issue 8, September 2012 Processing and Signal Processing
[3] G.Sakthivel, T.S.Anandhi, S.P.Natarjan. Real Time
Implementation Of DSP Based Fuzzy Logic Controller
For Speed Control Of BLDC Motor International
journal of computer applications (0975-8887) volume
10- no.8, November 2010.
[4] S.R. Khuntia, K.B. Mohanty, S. Panda and C. Ardil, A
Comparative Study Of P-I, I-P, Fuzzy And Neuro-
Fuzzy Controllers For Speed Control Of DC Motor
Drive, International Journal of Electrical and
Computer Engineering 5:5 2010
[5] TMSC320F2812 Users Guide, Texas
Instruments,Inc., 2008.
[6] Dr. Shereen F. Abd-Alkarim, Application Of Fuzzy
Logic In Servo Motor, Al-Khwarizmi Engineering
Journal, Vol.3, No.2, pp8 -16 (2007).
[7] Jun-UK chu,In-Hyuk Moon,Gi-won choi,Jei-chenong
Ryu,Mu-seony Mun, Design Of BLDC Motor
Controller For Electric Power Whell Chair,
Mechatronics, 2004,ICM04,Proceedings of the IEEE
international
[8] I.J. Nagrath & M. Gopal, "Control Systems
Engineering,"New Age Publication 2003.
[9] SIMULINK User's Guide, The Math Works Inc.
1992.
[10] P.Pillay and R.krrishnan. Modelling , Simulation And
Analysis Of A Permanent Magnet Brushless Dc Motor
Drive .IEEE trans.Ind Applicant., Vol26,pp124-
129,2002 conference.
[11] Bohdan Butkiewicz About Robustness of Fuzzy
Logic PD and PID Controller under Changes of
Reasoning Methods , Institute of Electronic Systems,
Warsaw University of Technology , 2002
[12] H. Ying, "Fuzzy Control and Modeling, Analytical
Foundations and Applications", Institute of Electrical
and Electronic Engineers Inc.
USA, 2000
AUTHORS BIOGRAPHY
Mrs. M. Madhavi is working as Associate Professor, Dept.

of ECE, DVR & Dr.HS MIC College of Technology,
Kanchikacharla, Krishna (D). She received her B.Tech,
from JNTU Hyderabad and M.Tech (DECS) from
Jawaharlal Nehru Technological University, Hyderabad in
2008. Her areas of interests is Signal Processing and
communication.
M. Bhanu Sri received B.Tech degree in Electronics And

Communication Engineering from Nimra College of
Engineering and Technology, Jupudi, A.P, India in
2011.She is currently pursuing M.tech in Digital
Electronics and Communication Systems (DECS) in
12
Implementation of Spartan 3E FPGA based

Multi Serial to Ethernet Gateway
for DAS using VHDL
A Swetha 1, E. Purushotham Reddy 2
Associate Professor, Guru Nanak Institute of Technology, Hyderabad, Andhra Pradesh, India
Assistant Professor, Guru Nanak Institute of Technology, Hyderabad, Andhra Pradesh, India
E-mail:swetha.ece.gnit@gmail.com
Abstract:-Data acquisition (DA) is the process of UAV Firstly, this paper introduced the structure and
sampling signals that measure real world physical working principle of system. Then the design of
conditions and converting the resulting samples into system's hardware and software programming were
digital numeric values that can be manipulated by a described, as well analyzed configuration of every
computer. The communication between them is
module and communication of serial ports Ethernet
standard asynchronous serial form by typical serial
control card, which have complicated cable wiring, connectivity is achieved with small Ethernet adapter
frequently interruption of CPU, which greatly reduces (serial to Ethernet) UART to Sensors that convert
the CPUs efficiency and impacts the systems real-time physical parameters to electrical signals[3]. Data
processing. To resolve these problems, a method of acquisition is the process of sampling signals that
multi-serials to Ethernet Gateway based on the field measure real world physical conditions and
programmable gate array (FPGA) with network converting the resulting samples into digital numeric
interface chip is implemented in this project. values that can be manipulated by a computer.
Finally, the timing simulation of two serial ports of
The Gateway will send data as Ethernet frame
UART was shown in Chip scope modules compared
format after receiving serial data, indirectly achieves
multi-serials communication, simplifies cabinet wiring with traditional methods, it possesses the characters
and improves CPUs efficiency. In this project we use of high-integration, low-power, and flexibility [5] ,
WIZ net W5100 Hardwired TCP/IP Embedded [6].
Ethernet Controller chip will be interfaced to FPGA to
provide Ethernet interface. Various digital signals will II. ARCHITECTURE DESIGN
be captured by FPGA and will be sent on a serial line.
The FPGA implements all the necessary logic to read
the data from these sensors and store the data in it. On
FPGA logic also will be implemented to read data from
multiple numbers of serial ports. The WIZ Net module
takes data from serial port and sends to PC in Ethernet
form. In PC application will be developed to read data
from Ethernet. The code will be developed in VHDL.
ModelSim Xilinx Edition (MXE) and Xilinx ISE will

be used for simulation and synthesis respectively. The
Xilinx Chip scope tool will be used to test the FPGA Fig. 1 Block diagram of multiport serial to Ethernet gateway
inside results while the logic running on FPGA. The
Xilinx Spartan 3 Family FPGA development board will
be used in this project. The above figure can represents the Block
diagram of multiport serial to Ethernet gateway. It
Keywords:- Data Acquisition, CPU, TCP/IP, Ethernet, consists of various modules like FPGA, UART,
PC, Sampling, WIZ net W5100, Gateway, Xilinx, ADC, Ethernet, etc.
FPGA, VHDL.
A. ADC Module:
I. INTRODUCTION
FPGAs are well suited for serial Analog to Digital
This paper describes multi-serials to Ethernet (A/D) converters. This is mainly because serial
Gateway [2] for data acquisition systems based on interface consumes less communication lines, while
field programmable gate array (FPGA). +ADC the FPGA is fast enough to the all accommodate the
Converter communicates in UART mode [3]. In order high speed serial data. The ADCS7476MSPS is a
to achieve multi serials communication between high speed, low power12-bit A/D converter. A/D
13
converter is a high speed serial interface that As shown in above figure Receiver section is a
interfaces easily to FPGAs. The A/D interface combination of a baud generator and receiver section
adapter (AD1_PMOD) is implemented within the module, each module has been explained below in
FPGA. Inside the FPGA, this adapter facilitates detail [3].
parallel data acquisition. Sampling is initiated at the
rising edge of a clock applied at the line sample. The Baud generation section is a clock divider circuit,
timing diagram of the communication protocol FPGA board clock runs at 50MHz, but UART
obtained with Modelsim is illustrated in Fig.2. transfers data at predefined standards that had to be
maintained, in present system is designed for a rate of
9600 bits/sec (i.e. 50x106 is scaled down for 9600). It
generates 9600 pluses per sec; this implies the speed
of UART is 9600 bits per sec [3]. Another clock is
with a 16 times faster than required, and it is given to
the receiver section, so that the data will not be
corrupted [4].
Fig. 2 PMOD connection diagrams
D. Receiver Section:
B. UART Block:
UART receiver handles reception of data from
The UART is a serial interface with a frame RS232 port. Main functions of receiver block are to
format of start bit of active low 0at beginning of convert the serial data to parallel data, and check the
frame and 8 bit of information with a stop bit of correctness of data from parity and store the
active high1 signal at the end The operation of received data. UART receiver state machine is shown
UART is controlled by Clock signal which is fed in Figure 5. The receiver is in IDLE state by default.
from external crystal [15]. When the serial data pin goes low, indicating the start
bit, the state machine enters DATA0 state. The data is
received; one bit at a time from LSB to MSB in states
DATA0 to DATA7. If parity is enabled, the state
machine checks the parity bit received against the
parity obtained from received data. If the data
received is fine, the data_Rx (data_Rx_done) bit is
set to 1 and the receiver goes back to IDLE state
again.
Fig.3 UART Module
C. UART Receiver:
Fig.5 UART receiver state machine
E. Enable Generator:
This section receives baud clock signal as an

Fig.4 UART block diagram enable in signal as a input signal and gives enable out
14
signal as a output signal. And this enable_ out signal format form, and that data is transmitted to the
will be given as a input signal to the transmitter destination system in the form of Ethernet packet
section as a enable input signal. This signal is used to .By Team Tera Pro software we observed the
synchronize the transmitter section, when the data is transmitted data in the output PC.[7]
to transfer is defined by this signal [18].
Fig.6 Enable Generator block diagram
F. Transmitter Section:
Fig.8 Ethernet module
The transmitter block is responsible for the
transmission of serial data from UART. It takes 8-bit F. RS-232:
data from the receiver section in parallel and send
data in serial form. In Telecommunications RS232 (Recommended
Standard 232) is the traditional name for a series of
Data is inserted between start and stop bits. An standards for serial binary single-ended data and
optional parity bit also may be used for error control signals connecting between a DTE (Data
detection. State machine for transmitter is shown in Terminal Equipment) and a DCE (Data Circuit-
Figure 7. Transmitter stays in IDLE state unless terminating Equipment). It is commonly used in
transmit enable (tx_enable) is made as 1. The data computer serial ports. The standard defines the
transmission starts with tx_enable = 1. As mandated electrical characteristics and timing of signals, the
by the protocol, a 0 is transmitted to indicate start of meaning of signals, and the physical size and pin out
transmission or start bit. This is done in START state. of connectors.
Then data bits 0 to 7 are transmitted in states DATA0
to DATA7. If parity is enabled in configure ration
register, the data is attached with parity in PARITY
state. Then transmitter enters STOP state and sends
1 as stop bit. This indicates the completion of
transmission. Then the transmitter enters the IDLE
state and waits for next data transmission.[18]
Fig.9 Telecommunications RS232
In RS-232, user data is sent as a time-series of

bits. Both synchronous and asynchronous
transmissions are supported by the standard. In
addition to the data circuits, the standard defines a
number of control circuits used to manage the
connection between the DTE and DCE. Each data or
control circuit only operates in one direction that is,
signalling from a DTE to the attached DCE or the
reverse. Since transmit data and receive data are
separate circuits, the interface can operate in a full
duplex manner, supporting concurrent data flow in
both directions. The standard does not define
character framing within the data stream, or character
Fig.7 UART transmitter state machine. encoding.
Power Supply Voltages: supplied +5 V and -5 V
and labelled them as "RS-232 compatible.
G. Ethernet Module:
G. JTAG USB Cable:
The Ethernet module takes the input from the Joint Test Action Group (JTAG) is the common
transmitter assigned port of the FPGA in a serial name for what was later standardized as the IEEE
15
1149.1 Standard Test Access Port and Boundary-

Scan Architecture.
Fig. ADC logic symbol
Fig.10 JTAG USB Cable
H. W5100 Hardwired TCP/IP Embedded Ethernet

Controller chip:
In this project we use WIZ net W5100 Hardwired

TCP/IP Embedded Ethernet Controller chip will be
interfaced to FPGA to provide Ethernet interface is a
full-featured, single-chip, Internet-enabled 10/100
Ethernet controller designed for embedded
applications where ease of integration, stability, Fig.12 Simulation Wave Form of ADC
performance, area and system cost control are
required. The W5100 has been designed to facilitate
easy implementation of Internet connectivity without Waveform Description:
an OS hard wired TCP/IP stack and integrated
Ethernet MAC & PHY. Hardwired TCP/IP stack Rst is used to reset module or clear previous data,
supports TCP, UDP, IPv4, ICMP, ARP, IGMP and clk is used for the synchronization, when the raising
PPPoE which has been proven in various applications edge of clk is 1 then count refers as a counter, go on
for several years. 16Kbytes internal buffer is included counting from 0 to 15. If count is 0 then data
for data transmission. present in shift resisters (signals) will be forced on to
the outputs pdata1, if count is above 3 then inputs
Various digital signals will be captured by FPGA will be forced on to shift registers (signals) from
and will be sent on a serial line. The F on FPGA logic LSB to MSB till the count 15 by shifting 1 bit for
also will be implemented to read data from multiple each count. CS will 1 for the count of 15. Pclk
numbers of serial ports. The WIZ Net module takes will be 1 for 0 to 8 counts and 0 for the rest of
data from serial port and sends to PC in Ethernet counts i.e it is clk /16.
form. In PC application will be developed to read
data from Ethernet. PGA implements all the
necessary logic to read the data from these sensors
and store the data in IT internal storage.
Fig. UART Transmitter logic symbol
Fig.11 WIZ net W5100 Hardwired TCP/IP Embedded Ethernet

Controller chip.
III. SIMULATION RESULTS
For the simulation we used Modelsim Simulator

Tool for Functional Verification. Simulation results
and the corresponding descriptions are mentioned
below with logic symbol:
Fig. 13 Simulation Wave form of UART transmitter.
16
Waveform Description: Data is ready on the data _rx bus to transmit to the
next block transmitter section this will in the active
Transmitter section consists of clk , high(1) state. Else in active low state (0).
BDclk_Posedge, reset , data_in , data_ready as a
input and outputs are Send_data , tx . when rst equals
to 1, the data _in set to all zeros and present state set
to S0 (PS = S0).when clk and BDclk is active and on
raising edge and when data ready is one (1) the data
is transmitted serially from one state to next state.
When tx and send equals 1 ,s0 set to 1 ,s1 set to
zero as start bit and next eight states the data is
transmitted and the tenth state (s10) is set to 1 to
represents the end of data transmission. i.e stop
bit.s10 and s0 bit will one until the next data
transmission is identified.
Fig.15 Simulation Wave form of Top module
Waveform Description:
The inputs to the system are ADC input (S_data)

and three UART inputs i.e. rxd1, rxd2, rxd3. These
inputs are selected by the selection signal. According
to selection signal the input is received by the UART
and it transmitted to the Ethernet module which acts
Fig. UART logic symbol Receiver as a transmitter section and which can be seen outputs
on the pc. Whenever the clock is active one of the
input data signal is selected. And whenever the
clk_16 and clk_9600 is active, the received data from
any of the input module is taken and that data is
received when Rxd signal is 1. We will all the input
signals from the URAT module and it transmitted to
the data collection module. In data collection module,
the input data from the all the modules is collected
and it placed in RAM and addressed accordingly.
From 0 to 511 addresses the ADC data is stored and
from 512 to 1023 the uart1 data is stored and from
1024 to 1535 the uart2 data is stored and from 1536
Fig.14 Simulation Wave Form of UART Receiver to 2047 the uart3 data is stored. From the data
collection module the data is transmitted serially to
Waveform Description: Ethernet module. From that module the data received
serially as s0 start bit and 8 bits of transmitted data
Receiver module consist of inputs Reset, clk_16, and s10 stop bit.
clock, RxD and outputs Data_rx, BYTE Enable. if
reset is 1 then data_rx and byte enable is zero. Else Screen shot for ADC Out Put :
input data rxd is collected bit by bit and detects start
bit at the starting of data and stop bit at end of data.
After processing data it will be reflected back on to
data _rx.
Fig. Top module logic symbol

Fig.16 simulation wave form ADC
17
Screen shot for UART Transmitter: system in real time. FPGAs flexible programming
features also allow further upgrade for system. This
design could be used in the domain of net control and
information management system. It has a certain
value for application. Low cost, as the multi card
solution can come in single FPGA card with small
modules around it, Low power in comparison with
multi card solution, Lot of scope for adding
additional functionality on FPGA.
ACKNOWLEDGMENT
We would like to thank the Organization of Guru

Fig.17 simulation wave form UART Transmitter Nanank Institute of Technology, Hyderabad, for the
support to use R&D Laboratories.
Screen shot for UART Receiver:
REFERENCES
[1] Simon S. Young (2001). Computerized Data Acquisitio and

Analysis for the Life Sciences. Cambridge University Press.
ISBN 0-521-56570-7
[2] Tong YAO, Yonghong YU, Lu Ding, Fpga based fo
implementation of Multi-Serials to Ethernet Gateway, 978-
1-4244- 5874-5/10/$26.00 2010 IEEE
[3] Yu Zhang, Yugui Hu, Kuixi Yin, Jilian Li, Implementation
of Multiserials Expand Based on FPGA, Electronic
instrument, vol. 32, n. 1, 2009, pp. 233-236.
[4] Yonghong Hu, Lu Ding, Design and Realization of
Multifunctional Gateway Based on SingleChip, 2009 2nd
international Congress on Image and Signal Processing,
Fig.18 simulation wave form UART Receiver Tianjin, China, October 2009, pp. 3709-3712.
[5] WWW.LM35 series are precision integrated-circuit
Device Utilization Summary: temperature sensors data sheet.
[6] Ethernet h10032.www1.hp.com/ctg/Manual/bpe50027.pdf
[7] http://www.renuelectronics.com/pdfs/letter/GWY-610.pdf
Selected Device : 3s500efg320-5 Ethernet gateway protocol Converter
Number of Slices : 290 out of 4656 6% [8] Reimer, Jeremy (July 25, 2007). "New Ethernet standard: not
Number of Slice Flip Flops: 320 out of 9312 3% 40Gbps, 100, but both". Ars Technica. Retrieved December
Number of 4 input LUTs : 445 out of 9312 4% 17, 2011
[9] Denton J. Dailey (2004), Programming Logic Fundamentals
Number of IOs : 34 Using Xilinx ISE and CPLDs, Pretince Hall, 203 pages.
Number of bonded IOBs : 34 out of 232 14% introduction to PLDs using Xilinx ISE.
Number of BRAMs : 1 out of 20 5% [10] Google Patent Search, "Re-programmable PLA". Retrieved
Number of GCLKs : 3 out of 24 12% February 5, 2009.
[11] Williams, Royce. "Working with Xilinx FPGAs on
FreeBSD". FreeBSD. Retrieved 8 May 2012.
IV. CONCLUSION [12] McConnel, Toni. EETimes. "ESC - Xilinx All Programmable
System on a Chip combines best of serial and parallel
processing." April 28, 2010. Retrieved February 14, 2011
In this paper we have discussed the importance [13] Tim Erjavec, White Paper, "introducing the Xilinx Targeted
Design Platform: Fulfilling the Programmable Imperative."
Data acquisition is the process of sampling signals February 2, 2009. Retrieved February 2, 2009
that measure real world physical conditions and [14] Wilson, Richard, Electronics Weekly. "Xilinx acquires ESL
converting the resulting samples into digital numeric firm to make FPGAs easier to use." January 31, 2011.
values that can be manipulated by a computer. Data Retrieved February 15, 2011.
[15] Shouqian Yu, Lili Yi, Weihai Chen, Zhaojin Wen,
acquisition systems (abbreviated with the acronym Implementation of a Multi-channel UART Controller Based
DAS or DAQ) typically convert analog waveforms on FIFO Technique and FPGA
into digital values for processing. The components of [16] Adam Osborne, An introduction to Microcomputers Volume
data acquisition systems include. The communication 1: Basic Concepts, Osborne-McGraw Hill Berkeley
California USA, 1980ISBN 0-931988-34-9 pp. 116-126
between monitor computer and port devices, improve [17] Zilog Product specification Z8440/1/2/4, Z84C40/1/2/3/4.
the efficiency of CPU, and ensure the processing of Serial input/output controller". 090529 zilog.com
18
[18] "FAQ: The 16550A UART & TurboCom drivers

1994". 090529 cs.utk.edu
[19] bill.herrin.us - Hayes ESP 8-port Enhanced Serial Port
Manual, 2004-03-02
[20] Maxim MAX232 datasheet
Author Biography
Smt. A. Swetha obtained M. Tech

Degree from Satyabhama University,
Chennai, India. She is currently
working as Associate Professor,
Depatment of Electronics and
Communication Engineering in Guru
Nanak Institute of Technology,
Hyderabad, India. Her interesting
fields are VLSI System Design,
Cellular and Mobile Communications, and EMFT.
Mr. E.Purushotham reddy, obtained

B.Tech degree in Electronics and
communication Engineering from
NITS, JNTUH University, M. Tech
Degree from Guru Nanak Institute of
Technology, JNT University,
Hyderabad. . He is currently working
as Assistant Professor, Dept. of ECE
in Guru Nanak Institute of Technology, JNT University,
Hyderabad, India. His interesting fields are Signal
Processing, Communications, VLSI System Design.
19
Design and Implementation of FPGA based Wi-

Fi MAC Transmitter
G. Padma Ratna1, S Naga kishore Bhavanam2, K. L. Bhavani3
Assistant Professor,
University College of Engineering & Technology,
Acharya Nagarjuna University, Guntur
Abstract WiFi (Wireless Fidelity) is a wireless MAC layer provides a CSMA/CA (Carrier Sense
technology based on a cellular architecture where system Multiple Access/Collision Avoidance). The optional
is subdivided into cells. WiFi is the wireless way to handle PCF based on the contention free service is suitable
networking. It is also known as 802.11 networking and for delay sensitive traffic. Due to technology
wireless networking. The big advantage of WiFi is its
advancement in the 21St Century, wireless
simplicity. You can connect computers anywhere in your
home or office without the need for wires. The computers Communication had been most popular choices of
connect to the network using radio signals, and computers communication. More and more people are turning to
can be up to 100 feet or so apart. wireless due to the convenience of mobility. An
802.11 LAN is based on a cellular architecture where
WiFi transmitter implements the Basic Service set (BSS) the system is subdivided in to cells, where each cell
and controlled by the base station called access point. [called Basic Service Set or BSS] is controlled by a
These access points are connected with distribution Base Station [called Access point, or in short
system. The transmitter is responsible for transmission AP].Even though that a wireless LAN may be formed
and generation of various entities like Preamble includes
by a single cell, with a Single Access Point[ can also
synchronization and Start frame delimiter (SFD), PLCP
Header used to decode the frame consists of PLCP_PDU work without an Access Point], most installations
Length word, PLCP Signaling Field and Header Error will be formed by several cells, where the Access
Check Field and Mac Data Frame. But compared with Points are connected through some kind of back
wired network, wireless communication is having low bone[ called Distribution System or DS],typically
speed, less range, less reliable, cost effective and less Ethernet, and in some cases wireless itself.
security. We can design and implement some protocol
standards to overcome the disadvantages of wireless II. IEEE 802.11 STANDARD
communication. In this project we are designed and
implemented MAC (Medium Access Control) layer
functions such as authentication, association, probe Due to technology advancement in the 21st
request and cyclic redundancy check as hardware to Century, wireless Communication had been most
increase speed and security of WiFi WLAN popular choice of communication. More and more
communication. This Design coding, Simulation, Logic people are turning to wireless due to the convenience
Synthesis and Implementation is done using Xilinx ISE. of mobility. An 802.11 LAN is based on a cellular
The designed hardware was tested by simulating in the
architecture where the system is subdivided in to
Model Sim 6d and was successfully implemented in the
SPARTAN3E FPGA board using Xilinx ISE as a cells, where each cell [called Basic Service Set or
synthesis tool BSS] is controlled by a Base Station [called Access
point, or in short AP]. Even though that a wireless
Keywords:- WiFi, IEEE 802.11, BSS, SFD, PLCP, MAC, LAN may be formed by a single cell, with a Single
WLAN, Xilinx ISE. Access Point [can also work without an Access
Point], most installations will be formed by several
cells, where the Access Points are connected through
I. INTRODUCTION some kind of back bone [called Distribution System
or DS], typically Ethernet, and in some cases wireless
In recent year, the popularity of WLAN (Wireless it self. The whole interconnected wireless LAN
Local Area Network) has generated much interest on including the different cells, their respective Access
improvement and performance analysis of the Points and the Distribution System, is seen to the
IEEE802.11 protocol. The IEEE 802.11 protocol upper layers of the OSI model, as a single 802
defines the MAC (Medium Access Control) and the network and is called in the standard as Extended
physical layer functions of WLANs. The WLAN Service Set [ESS]. The standard also defines the
MAC protocol employs two medium access methods concept of a portal, a portal is a device that
for packet transmission which are the DCF interconnects an 802.11 and another 802 LAN [3].
(Distributed Coordination Function) and the PCF However, all is not prefect in the WLAN world.
(Point Coordination Function) .We focus on the Offering nominal bit rates of 11Mbps [802.11b] and
20
54Mbps (802.11a and 802.11g) the effective MAC LAYER Frame Format:
throughputs are actually much lower-owing to packet
collisions, protocol overhead, and interference in the The basic format for all packets sent by the MAC
increasingly congested unlicensed bands at 2.4GHz layer. Some actual packets do not actually contain all
and 5GHz. Furthermore, operation in these bands of the fields.
entails a strict regulatory transmit power constraint,
thus limiting range and even bit rates beyond a
certain distance [5].
III. MAC LAYER
The layer of MAC specifies two different

mechanisms: distributed coordination function (DCF)
with contention scheme and point coordination
function
(PCF) with contention free or polling scheme to Figure.3 MAC Frame Format
perform real-time applications. At present DCF is the
dominant MAC mechanism implemented by IEEE However, all fields are present in all data packets. Up
802.11compliant products. The 802.11 protocol to four addresses are needed because it is sometimes
covers the MAC and physical layer, the standard necessary to identify the address of the access point
currently defines a single MAC which interacts with used by the transmitter or receiver.
three physical layers. MAC layer acts as an
intermediate stage between Data link layer and Thus, if two wireless LAN users are sending packets
physical layer. Its primary responsibility is to provide to one another but each is using a different access
a reliable mechanism for exchanging transacting point, the 802.11 MAC addresses of both access
packets [data, control and management] on the points and both clients will be present in the four
communication channel through physical layer. MAC address fields.
layer performs the following functions:
IV. DESIGN OBJECTIVES
(i) Generation of various MAC frames.
(ii) 32-bit CRC for payload data. 4.1 Block Diagram:
(iii) FIFO buffer interface for receiver.
(iv) Parallelizing data by bit to byte converter.
MAC receiver controller state machine

implementation.
Fig.2 MAC architecture Medium access Control layer
performs function like:
(i) On transmission, assemble data into a frame.

(ii) On reception, disassemble frame and perform
address recognition and error detection.
Figure.2 Architecture of MAC Layer
21
enabled in the probe state that activates the probe

block only making all the other blocks inactive, this
makes only probe block to get the data_in data in
bytes to pass the data to the multiplexer block, so
with the remaining frames get transmitted ,with
authentication, association and RTS, data frames ,but
each frames are the request frames and so as soon the
request frame is transmitted the response frame is
received that is processed with the receiver block of
WLAN MAC and give the signal of no error for the
transmitted frames as prob_reply, asso_reply,
Figure 4.2 Data Transmission Process of Finite State Machine auth_reply, rts_reply, and data_reply, for their
respective request frame transmission in the MAC
4.2 Block Diagram Explanation transmitter block.
CLKGENERATION BLOCK: The clock generation So each and every frame transmission in transmitter
block is used to generate 1MHz clock frequency from of WLAN MAC is transferred only if the previous
50MHz that is available from the board. This frames response is received correctly. HEADER
generated clock is easy to use to the backoff block BLOCK: The header block defines the type of
and difs counter blocks which are sub multiples of frame that is transmitted, the contents of the frame
1MHz clock frequency. has
type, subtype values that define which frame body to
DIFS BLOCK: The DIFS counter block is operated transmit, and also the add1 recipient address of 6
by the main control machine which initiates the bytes
counter, the estimated period for the DIFS with and the transmitter address which is MAC address of
respect to PHY layer considering 802.11b is 50 micro transmitting node, duration of 2 bytes.
seconds, so the counter is made work with 50 as
counter under clk_1MHz clock frequency to obtain a PROBE BLOCK: The probe block has the contents
delay of 50 micro seconds. As soon as the block gets as
completed it intimates the control machine for further SSID 32 bytes and supported data rate and extended
back off operation. data rates of 8 bytes that are need to transmit in a
sequence.
BACKOFF BLOCK: This block is operated only
when the phy_idle signal that is provided by the AUTHENTICATION FRAME BLOCK: The
PHY layer becomes logic 0 before the DIFS gets authentication block transmits the frame body of the
completed. The start of backoff is initiated by control auth frame byte by byte. The content algorithm no,
machine, and initially the random value is taken as authentication transaction number status code and
30, that is defined by number of time slots. In this challenge text, that are send in sequence of bytes
case the slot time is taken as 20 micro seconds. through the block to MPDU block to PHY layer.
ASSOCIATION FRAME BLOCK: The association
CONTROL BLOCK: Back off state is an important frame
operation in wireless LAN MAC, that decides which block capability information, listen interval ,SSID of
node to access the channel, this is executed if the 32
phy_ilde becomes low within DIFS counter period bytes, supported rates are transmitted in a sequence of
else not, in our case the state transition has moved byte by byte to MPDU block.
DIFS to backoff state to test the states transition
operation. MPDU BLOCK: The MPDU block is a multiplexer
block that pass the bytes of information from each
The further states are for the transfer of the header frame block to the physical layer only one at a time,
frames and probe frames , auth frames association depending on the selection signals that are generated
frames and rts and data frames ,that are processed in from the control state machine.
sequence, by enabling the header block with
header_En so that the data_in byte of input that get 4.3 Finite State Machine
transfers framing it as the header frame through the
MPDU multiplexer block to the physical layer, very The below figure can represents the Functional
similar to the header block is the probe block is Diagram of Finite State Machine.
22
This clk1_MHz is used for the DIFS, SIFS, slot time

periods. 25 pulses for positive and 25 for negative
counter design
5.2 DIFS BLOCK:
Discrete Inter Frame Space is nothing but the delay of

50 micro seconds used while transmitting the next
frame or packet. Here it simply uses a counter of
count value 49 with the clock frequency of 1M Hz
which suggests that after the end of counter a delay of
50 micro seconds is generated. In the same way SIFS
is also generated with the count value of 9 i.e. 10
micro seconds of delay is generated.
Figure 4.3 Functional Diagram of Finite State Machine Figure 5.2 DIFS block diagram
IV. DESIGN IMPLEMENTATION After the end of DIFS and SIFS the outputs are given
as 1 to DIFS end and SIFS end.
Different Modules in the Project with Explanations:
5.1 CLOCK GENERATION BLOCK: 5.3 BACKOFF BLOCK:
Back off algorithm is used to select a node from

number of nodes. To do this every node is allocated
with a random number from 0 to 1023 number. This
random number is made to decrease on all the nodes
with a slot time of 20. Micro seconds, To generate
this a counter with count value 20 and clock 1 MHz is
used. The random number on the nodes is made to
decrease only when the channel state, back off start
Figure 5.1 Clk generation block diagram and slot end values are 1.When the random number
on any node becomes zero then it is allowed to access
Normally the frequency of the clock is 50M Hz in channel and back off end is given as 1 but when
Spartan 3e FPGA board. As the frequency is very transmission fail occurs i.e. when two nodes gets
high, as we need 50 micro seconds for DIFS and 10 random number 0 at same time then the random
micro seconds for SIFS as defined by IEEE 802.11 number is doubled and the algorithm is implemented
group, we have to step down the frequency to 1MHz again. If the channel state is not idle the back off
so that we can easily obtain 1 microsecond delay. To hault is 1.The node whose back off end becomes 1 is
get this we use a counter of count value 25 which allowed to access the channel.
means for every 25 clock cycles there is a shift in the
sign of the required wave form, in this way we 5.4 HEADER BLOCK:
generate 1MHz frequency.
COUNT VALUE = (available

frequency/required)
= (50MHz available from
FPGA
board)/ (required 1MHz)
= 50
Figure 5.4 Header block diagram
23
This header block will transmit the header frame

format in a sequence of bytes. Add1 is the recipient
address and control frame is a 2 byte length frame
which gives the information of which type of frame is
transmitted. Header en is a pin that comes from
control state machine for enabling the header block.
Machs input port signal will give information
whether the header is for management or data. If 1
it is for data and if it is 0 it is for management. The
bytes of data that is framed for the frame sequence is Figure 5.6 Probe Request block diagram
sent out from the header out. After completion of
back off or difs state, now the transmission of header 5.7 AUTHENTICATION BLOCK:
frame is controlled from control state machine by
enabling the header enable as 1. The sequence of
the header frame format is sent step by step. First
frame control, duration and so on. In this way the
header frame is transmitted through header out. This
header out is the input for MPDU multiplexer block
that will finally send the 8 bit information of the
header to the physical layer.
5.5 FRAME ENABLE BLOCK:

Figure 5.7 Authentication block diagram
This block has inputs association request,
authentication Data in and CRC out are the input data for this block
request, data request, probe request and RTS request. and auth_req and header_end are the enable for this
This block gives the values for type and sub type for block from control state mach and herder block.
the type of the frame which is given as input to frame Authentication frame is the 8 bit output port with
control block. which is used for header frame format bytes of transmission in a sequence according to the
formation. state mach implemented in this block. Auth_end is
the final signal that is made 1 after complete frame
transmission. Crc_stp_auth is made 1 only after the
auth frame body is completely transmitted. When
crc_stp is enabled, the CRC data is given to the CRC
out input port of this block from the CRC generation
block.
5.8 ASSOCIATION BLOCK:

Figure 5.5 Frame Enable block diagram
5.6 PROBE REQUEST BLOCK:
This block is enabled only when header_end and

probe_req is logic 1. In this block we implement a
state
machine for transmission of contents of probe frame.
The contents of probe frame are SSID with 32 bytes,
supported rates and extended rates with 8 bytes. That
makes up to 48 bytes of transmission for frame Figure 5.8 Association block diagram
format
through the probe frame output port and crc_stp_prb For this block CRC_OUT and DATA_IN are the
is a output that gives information of CRC for information inputs and HEADER_END and
transmission of CRC followed by probe frame data. ASSO_REQ are the enables. ASSO_FRM is the
Prob_end gives the end of the complete probe frame. output port that gives the sequence of the association
That would give the information to the state machine. frame format that gives information to the
24
multiplexer block. ASSO_END is the output port that the crc for bytes of information and the crc_stp is
gives the information of the completion of the used for stopping the crc calculation after the last
association frame to the control state block. byte is transmitted. The crc_out is 32bit of value that
CRC_STP_ASS is the output port that gives is shifted bytes in 4 times into the frame block to the
information for the CRC gen block to give the CRC MPDU block.
data to the ASSO_FRAME. Control state machine
has been implemented for the sequence of association 5.11 DATA TRANSMISSION BLOCK:
frame transmission. Each and every content is
specified will resemble a state that will transmit the
contents through the output 8 bit port. After the final
transmission of data AUTH_END is enabled that
gives the info to the control state machine.
5.9 RTS BLOCK:

Figure 5.11 Data block diagram
Data_msdu is the input port of 8 bit. Data_en is from

control state mach as enable to this block data_msdu
is
the output port which is the input to the mux block.
Crc_stp is the output signal to the crc gen block.
Date_end is the output port to the control state
Figure 5.9 RTS block diagram machine.((THIS block has input as data_msdu_in
which is raw data that is from llc layer. This is of
The contents of the rts frame are transmitted. In each length 0 to 2311 bytes. We receive a data of max of
state the contents of the rts frame are frame control, 2312 bytes of info in this block ))and we shift the
duration, TX addr, RX addr, fcs. The contents of the bytes of info to the data_msdu output port by
frame format are transmitted by designing a counter. designing a simple shift counter. After the data frame
Data contents are transmitted using a shift counter. is transmitted, crc is sent after it by enabling the
Each and every state has a byte shift counter. These crc_stp as 1 to the crcgen block.
are transmitted in a sequence of bytes through the
output port and crc_stp is enabled after complete rts 5.12 MPDU BLOCK:
frame transmission so that the crc gen block send the
32 bit calculated crc in data from of 4 bytes to
transmit crc followed by crc to this block as input. This mpdu block is a mux that transmits the different
rts_end is made 1 after the crc of the rts is frames like header, probe, asso, auth, rts and data
transmitted. it is given to the control state machine frames in a sequence depending on the selection
for further frame transmission. signals for the mux. the selection signals are
asso_req, atuh_req, data_en, header_en, probe_req,
rts_en. Mpdu is the output port from the top e\level
5.10 CRC GENERATION BLOCK: chip to the physical layer. Each input is passed
through the output mpdu port based on the selection
line. For ex, if the probe request is enabled probe
frame input info is passed thru the mpdu block.
Similarly all the frames are passed thru the mpdu
block.
Figure 5.10 Clk_generation block diagram
This is crc block generation ,this generates the crc 32

for the each bytes of information that is transmitted
right from the header and probe, assoc, auth, rts, data
frames, the management, control ,data frames have a
crc transmitted at the end of the frame body, the
crc_en is a enable signal that will start calculating
Figure 5.12 MSDU block diagram
25
5.13 FRAME CONTROL BLOCK:
This Block has inputs as type and subtype and output

as
control frame of 2 bytes length. This control frame
port is the input for the header block which is used
for first two bytes of transmission of header. Header
frame is dependent on type and subtype values. For
ex probe has
0d00 and for asso 0B00 etc.We can see that it s a
combinational block(use it every where clk is
there)using type enabled subtype as input, control Figure 6.1 Project Resources Utilized on FPGA Result
frame as output. This block just concatenates the type
and subtype with the remaining contents of the frame 6.2 Synthesis Report of Project
control format.
Figure 5.13 Frame Control block diagram Figure 6.2 RTL Schematic-1 of Top Module
5.15 TOP LEVEL:
Figure 6.3 RTL Schematic-2 of Top Module- Top Module Consists

Figure 5.14 Data Frame Block of
The data block is gives the raw data that is Blocks Named as input module, Transmitter Module and Output
transmitted by the LLC layer called LLCPDU ,also modules.
called the msdu data that could be max of 2312 bytes,
that is needed to fit into the frame body of the data
frame, for transmission. The above block shows the
inputs and outputs for this block, the inputs are
data_msdu_in of byte length that is transferred to the
output mpdu based on the enable data_en that is
enabled from the control state machine.
VI. EXPERIMENTAL RESULTS
6.1 Resources Utilized on FPGA Result Figure 6.4 RTL Schematic of input Module
26
Figure 6.10 Simulation result for Back off
Figure 6.5 RTL Schematic of Transmitter Module
Figure 6.11 Simulation result for Probe Request
Figure 6.6 RTL Schematic of Output Module
Figure 6.12 Simulation result for Authentication Request
Figure 6.7 Technology view of Top module
6.3 Simulation Results
Figure 6.13 Simulation result for Association
Figure 6.8 Simulation result for clock
Figure 6.9 Simulation result for DIFS Figure 6.14 Simulation result for RTS
27
VII. CONCLUSION
In this project work, the MAC layer has been

developed using VHDL language. The code is
designed to be implemented on Xilinx Spartan -3E
family FPGAs. To implement WiFi MAC
Transmitter, Three modules named as input or data
module, Transmitter module and Output module are
Figure 6.15 Simulation result for CRC designed. In this Project, I achieved more Speed and
Security for Data transmission because hardware
processing offers much greater speed than a software
implementation. Successful simulation results for
each module have been obtained.
ACKNOWLEDGMENT
Figure 6.16 Simulation result for Frame Enable We would like to thank the Organizations of
Acharya Nagarjuna University, Guntur, for support to
use R&D Laboratories.
REFERENCES
[1] W.L.Pang, K.W.Chew, F.Choong, E.S.Teoh, "VHDL Modeling

Of The IEEE 802.1 Lb DCFMAC", Proceedings Of The 6th
WSEAS International Conference On Instrumentation,
Measurement Circuits And Systems, Hang Zhou, China, April 15 -
Figure 6.17 Simulation result for Frame Control 17, 2007.
[2] Broadband Wireless Access, Institution of Engineering and
Technology Seminar, 6th Dec 2006,London, UK.
[3] Pabolo Brenner, "A Technical Tutorial on the IEE802.11
Protocol".
[4] Shambhu Upadhyaya, Wireless Network Security,"Lecture 5,
University at Buffalo, the State University of New York.
[5] Sebastien Roy, The Design of an FPGA Based MIMO
Transceiver for Wi-Fi,"DSP Magazine, May 2006.
[6] Prof. David Brooks, "Computer Science 141Computing
Hardware", fall 2007, Harvard University.
Figure 6.18Simulation result for Header
[7] Janak. H.Patel, Introduction to VHDL 1 and 2, Dept of
Electrical and Computer Engineering, University Of Illinois at
Urbana-
Campaign.
[8] Ahmed Abu-Hajar, Digital Design Using VHDL Digitavid, Inc,
San Jose, CA.
[9] K.Masselos, N.S.Voros. "Implementation Of Wireless
Communications System On FPGABased Platforms", EURASIP
Journal On Embedded Systems, Volume 2007, Article ID 12192.
Figure 6.19Simulation result for Data block [10] A.M.Bhavikattil, Dr.Subhash.Kulkarni 2009 IEEE
International
Advance Computing Conference (IACC-2009) Patiala, India, 6-7
March 2009.
[11] A.M.Bhavikattil, Dr.Subhash.Kulkarni 2009 IEEE
International
Advance Computing Conference (IACC-2009) Patiala, India, 6-7
March 2009.
Figure 6.20 Simulation result for MPDU block
28
Goertzel Algorithm based DTMF Detection

Using FPGA
S Nagakishore Bhavanam1 Dr. P. Siddaiah2 Dr. P. Ramana Reddy3
Research Scholar, Professor & DEAN Associate Professor of ECE
JNTUA University College of Engineering & Technology JNTUA
Ananthapuram, A.P.,INDIA Acharya Nagarjuna University, Guntur, A.P.,INDIA Ananthapuram, A.P.,INDIA
Abstract Dual-Tone Multi-Frequency (DTMF) is a frequencies. These frequencies should be generated

signaling standard in telecom applications that and detected according to the Consultative
produces two tones simultaneously for each key press. Committee for International Telegraph and
The DTMF tone detection is very crucial block in Telephone (CCITT) Recommendation. The DTMF
several telecom based embedded systems. As the system uses eight different frequency signals
current generation embedded systems are looking for transmitted in pairs to represent sixteen different
key feature of low power, the DTMF detection numbers, symbols and letters. The Fig.1 shows how
algorithm also must be implemented with low power
the frequencies are organized:
schemes. The DTMF detection can be done with FFT
based technique which is power consuming type and it
requires more hardware. In this project the FPGA
based DTMF detection with very low power and low
area by Goertzel algorithm is implementing using
VHDL. Since DTMF based applications dont require
high speed tone detection the resource sharing approach
can be used.
Keywords:-DTMF detection, Goertzel Algorithm,

resource sharing Approach, FPGA.
I. INTRODUCTION
Dual Tone Multi-Frequency (DTMF) is a method

for instructing a telephone switching system of the Fig.1: Keypad having low and high frequencies
telephone number to be dialled, or to issue commands
The frequencies used were chosen to prevent any
to switching systems or related telephony equipment.
harmonics from being incorrectly detected by the
The DTMF dialling system traces its roots to a
receiver as some other DTMF frequency [1].
technique AT&T developed in the 1950s called MF
(Multi-Frequency) which was deployed within the
AT&T telephone network to direct calls between II. DTMF GENERATION AND DETECTION
switching facilities using in-band signalling. In the
early 1960s, a derivative technique was offered by This project is of DTMF detection by Goertzel
AT&T through its Bell System telephone companies algorithm. The first touch tone telephone installation
as a "modern" way for network customers to place
was in 1963. DTMF signalling uses voice-band tones
calls. In AT&Ts Compatibility Bulletin No. 105, to send address signals and other digital information
AT&T described the product as "a method for from pushbutton telephones and other devices such as
pushbutton signalling from customer stations using modems and fax machines. Analog DTMF detection
the voice transmission path." The consumer product is done using band-pass filter banks with centre
was marketed by AT&T under the registered trade
frequencies at the DTMF signal frequencies. Digital
name Touch-Tone. Other vendors of compatible
detection of DTMF is done by several algorithms like
telephone equipment called this same system "Tone" goertzel, notch filter etc [2].
dialling or "DTMF".
DTMF signalling has many applications such as The transmitter of a DTMF signal simultaneously
telephone dialling, data entry, credit checking, and sends one frequency from the high-group and one
voice mail system control. A DTMF signal Consists frequency from the low-group. This pair of signals
of two superimposed sinusoidal waveforms with represents the digit or symbol shown at the
frequencies chosen from a set of eight standardized intersection of row and column in the Fig. 1.
29
For example, sending 1209Hz and 770Hz indicates including the extreme case of dial-tone being
that the "4" digit is being sent. The below figure sent by the central office at 0 dBm (which may
represents the basic DTMF Communication System. occur in extremely long loops). Above 600Hz,
any other signals detected by the receiver must
be at least -6 dB below the low-group frequency
signal strength for correct digit detection.
The values shown are those stated by AT&T in
Compatibility Bulletin 105. For compatibility with
ANSI T1.401-1988, the minimum inter-digit interval
Fig. 2: DTMF communication system shall be 45msec, the minimum pulse duration shall be
50msec, and the minimum duty cycle for ANSI-
At the transmitter, the maximum signal strength compliance shall be 100msec.
of a pair of tones must not exceed +1 dBm, and the
minimum strength is -10.5 dBm for the low - group Methods for Generating DTMF tones:
frequencies and
-8.5 dBm for the high-group frequencies. When the DTMF generation can be done by different
intentionally not sending DTMF tones (including the methods for analog and digital systems.
inter-digit interval), any leakage of these tones must Use special IC for generating DTMF tones.
not exceed -55 dBm. The frequencies generated by Modems and telephones use this method.
the transmitter must be nominally within 1.5% of the Generate DTMF tones using soundcard FM
stated values and the receiver must not accept signals synthesizer chip.
that deviate more than 3.5% from the stated Load sine wave sample to wavetable soundcard
values[8],[9]. memory. Play that sample using two instrument
channels at different frequencies.
The receiver is responsible for performing several Sample all DTMF tone combinations headed and
checks on the incoming signal before accepting the playback those samples as needed. 8 kHz at 8 bit
incoming signal as a DTMF digit: resolution is enough for that.
Generate the sample data which is played back
Energy from a low-group frequency and a high- using software.
group frequency must be detected. Our project is concentrating on the look-up table
Energy from all other low-group and all other method, because it is the most generic way to do
high-group frequencies must be absent or less the DTMF generation. You can use this method
than -55dBm. with every sound card which can play back
The energy from the single low-group and single samples and it is as well suitable for DSP
high-group frequency must persist for at least implementations also [5].
40msec*.
There must have been an inter-digit interval of at III. GENERATING DTMF
least 40msec* in which there is no energy
detected at any of the DTMF frequencies. The
minimum duty cycle (tone interval and inter-digit The another methods for the generation of the
interval) is 85msec*. DTMF tones will be
The receiver should receive the DTMF digits Polynomial approximation
with a signal strength of at least -25 dBm and no Look-up table
more than 0 dBm. Recursive oscillator
The energy strength of the high-group frequency
Dual tone generation can be done with 2 sine
must be -8 dB to +4 dB relative to the energy
wave sources connected in parallel. Different method
strength of the low-group frequency as measured
can be used for such implementation:
at the receiver. This uneven transmission level is
DTMF signal must meet certain duration and spacing
known as the "twist", and some receiving
requirements:
equipment may not correctly receive signals
where the "twist" is not implemented correctly. 10 Digits are sent per second.
Nearly all modern DTMF decoders receive Sampling is done via a codec at 8Khz.
DTMF digits correctly despite twist errors. Each tone duration must be > 40msec and a
The receiver must correctly detect and decode spacing of 50ms Minimum between two
DTMF despite the presence of dial-tone, digits is required.
30
DTMF generator refers to any electronic device or other auxiliary telephonic devices that can decode
capable of generating dual tone multi-frequency signals produced by a DTMF generator. Telephones
(DTMF) signals used for dialling touch-tone equipped with DTMF-based touch-tone dialling allow
telephones. The DTMF dialling system operates by consumers to access their account balances at banks,
configuring a keypad in a three column by four row as well as retrieve other information from
grid representing the numbers 0 through 9 and other organizations that provide a telephonic interface
special characters such as the # and * keys. The touch based on the DTMF system.
pad grid is arrayed by assigning a specific high
frequency for each column and a designated low A computer equipped with a sound card and digital
frequency for each row [3]. When a user presses a audio software is capable of acting as a
digit on the keypad, a dual tone is generated that DTMF generator by producing the dual tone multi-
represents the intersection of the two distinct frequencies used on touch-tone telephone keypads
frequencies assigned for each column and row on the [3],[4].
grid. For example, when the "2" digit is pressed on
the keypad, a unique audio sinusoidal tone is emitted The software program interprets the phone digits
by mixing a pure high frequency of 1,336 Hz with a and then generates the appropriate audio files that
pure low frequency of 697 Hz. The DTMF signal conform to the standardized DTMF paired frequency
tone generated by the keypad is then transmitted to tones. The dialling process can be activated either by
the local office of the phone company where it is a mouse click or through keyboard commands. This
translated or decoded into the appropriate allows for rapid automated computer dialling of
corresponding digits which allow the designated outgoing telephone numbers and is used by many
phone number to be dialled. Therefore the tones in businesses [4].
keypad are generated by taking one of the frequency
from the row selector and the other frequency from IV. GOERTZEL ALGORITHM
the column selector. These two frequencies combine
and give the dual tone at output. The Goertzel algorithm is a digital signal
processing (DSP) technique for identifying frequency
The block diagram of DTMF tone generator will components of a signal, published by Dr. Gerald
be the first commercial application of Goertzel in 1958. While the general Fast Fourier
a DTMF generator was the introduction of the touch- transform (FFT) algorithm computes evenly across
tone pad dialling system, first offered by AT&T to its the bandwidth of the incoming signal, the Goertzel
customers in 1963. The touch-tone dialling system algorithm looks at specific, predetermined
was demonstrably faster and more convenient than frequencies, Some applications require only a few
the mechanically based rotary dial pulse system, DFT frequencies. One example is frequency-shift
which it supplanted. After its introduction, touch-tone keying (FSK) demodulation, in which typically two
became the standard dialling system for telephones in frequencies are used to transmit binary data; another
the United States and was eventually adopted example is DTMF, or touch-tone telephone dialing, in
worldwide [13]. which a detection circuit must constantly monitor the
line for two simultaneous frequencies indicating that
a telephone button is depressed [15].
Goertzel algorithm reduces the number of real-
valued multiplications by almost a factor of two
relative to direct computation via the DFT equation
[6].
For a length of N, the Goertzel's series is
Fig. 3: DTMF tone generator
Touch-tone dialling permits activation and

communication with DTMF-compliant equipment. The below figure represents the block diagram of
This type of dialling allows users to control Goertzels Algorithm. Goertzel algorithm is second
answering machines remotely and to activate and order recursive algorithm. Only needs two real
interface with much call routing, voice mail systems, multiplications and three real additions [6], [7].
31
Frequency Detection block

Magnitude / index estimator
Frequency to digit look-up table
Hex Keypad gives input to the module. It is an
external component. The signal from column is taken
as input. Row and display are the output signals. By
using PC we analyze the corresponding results. The
below figure can represents the Goertzel Algorithm
blocks as frequency detector.
Fig.4: Block diagram of Goertzels Algorithm
TABLE-I : ITU STANDADS FOR DTMF SIGNALS
Characteristic Type Specification
Frequency Operational 1.5% ofHz

Tolerance Non-Operational 1.5% ofHz
Signal Operational 40 ms min.
duration Non-Operational 23 ms max.
Pause duration Fig.6: Goertzel Algorithm Blocks as frequency detector
Signal 40 ms min.
Signal
exceptions 10 ms max.
interruption
VI. IMPLEMENTATION
Forward 8dB
Twist
Reverse 4 dB
Since the telephone industry has preset the
15 dB min.
Signal-to-noise sampling frequency to 8 kHz and the DTMF
Signal
ratio frequencies to 697,770, 852, 941, 1209, 1336, 1477,
Strength -26 dBm
Signal Power and 1633 Hz, the filter length must be large enough to
min.
find the desired k value that corresponds to the
DTMF frequencies. Therefore, there is a trade off to
V. ARCHITECTURE DESIGN be considered between the computation burden and
better resolution. For this application FPGA based
DTMF detection can be done by a resource sharing
The below figure can represents the DTMF approach to get low power consumption and less
Detection General Module. hardware. For simulation & synthesis we use Xilinx
ISE, for functional verification we use Modelsim, for
on-chip verification we can use Xilinx Chip scope.
VII. CONCLUSIONS
The FPGA technology is increasing its

applications in communication technologies. In this
paper a robust implementation of a DTMF decoder
was presented. There is plethora of algorithms for
DTMF decoding. Still, most of them require
Fig.5: DTMF Detection (General Module) significant computational power as well as memory,
to be executed. The Goertzel algorithm gives an
alternative to the latter problem by implementing an
The module DTMF detection consists of IIR filter for the centred DTMF frequencies to be
detected instead of examining the whole spectrum.
Hex key pad The advantage of our implementation is the
DTMF test signal generator significant reduction in FPGA resource usage which
Additive white Gaussian noise lets us use very small and cheap devices.
32
ACKNOWLEDGMENT 15. M. K. Ravishankar and K. V. S. Hani, "Performance Analysis

of Goertzel's Algorithm based Dual-Tone Mlultifrequency
(DTMF) Detection Schemes", Technical Report,
We would like to thank the Organizations of ePrints@iisc (India), August 2004.
Acharya Nagarjuna University, Guntur and JNT
University, Ananthapuram, for support to use R&D AUTHOR BIOGRAPHY
Laboratories.
Mr. S NagaKishore Bhavanam obtained
REFERENCES B.Tech degree in Electronics and
communication Engineering from
S.V.V.S.N Engineering College, Acharya
1. C. Marven, General-Purpose Tone Decoding and DTMF
Nagarjuna University, Guntur in 2008, M.
Detection, in Theory, Algorithms, And Implementations,
Digital Signal Processing Applications with the TMS320 Tech Degree from Auroras Technological
Family, Vol. 2,Literature number SPRA016, Texas & Research Institute, JNT University, Hyderabad in 2010. He
Instruments (1990). is presently pursuing his Ph.D degree from JNTUA,
Anantapuram in the area of Signal Processing &
2. J. G. Proakis and D. G. Manolakis, Digital Signal Processing:
Communications. He is currently working as Assistant
Principles, Algorithms, and Applications, 2nd ed.,
Macmillan, New York, NY (1992).
Professor, Dept. of ECE in University College of Engineering
and Technology, Acharya Nagarjuna University, Guntur, India.
3. G. L. Smith, Dual-Tone Multifrequency Receiver Using the His interesting fields are Signal Processing, Communications,
WE DSP16 Digital Signal Processor, AT&T Application VLSI and EMFT. He is the student member of IEEE.
Note.
e-mail: kishore.ece.anu@gmail.com
4. P. Mock, Add DTMF Generation and Decoding to DSP-mP
Designs, in Theory, Algorithms, and Implementations,
Digital Signal Processing Applications with the TMS320 Dr. P Siddaiah obtained B.Tech degree in
Family, Vol. 1, Literature number SPRA012A, Texas Electronics and communication
Instruments (1989).
Engineering from JNTUA college of
5. Y. C. Huang, DSP Techniques for Telecommunication engineering in 1988. He received his
Systems, Masters thesis, Northern Illinois University M.Tech degree from SV University
(August 1995). Tirupathi. He did his Ph.D program in
6. Kamal Shaterian, Hossein gharaee DTMF detection with JNTU Hyderabad. He is the Chief Investigator for several
Goertzel Algorithm using FPGA, a resource sharing outstanding Projects sponsored by Defense Organizations,
approach IEEE 2010, pp.196-199. AICTE, UGC & ISRO. He is currently working as Professor &
7. Ge Jinzhao, Zhang Lulin, Qian Yumei, A new signal DEAN, Department of ECE in University College of
detection method based on Goertzel Engineering and Technology, Acharya Nagarjuna University,
algorithm,Communication Technology, volume 9, pp16-18, Guntur, India. He has taught a wide variety of courses for UG
2002. & PG students and guided several projects. Several members
8. Texas Instruments. Modified Goertzel algorithm for DTMF successfully completed their Ph.D under his guidance. Several
using the MS320C80, application report spra066. 1996. members pursuing their Ph.D degree. He has published several
9. Smith, G. L. Dual-Tone Multi-frequency Receiver Using papers in National & International Journals & Conferences. He
the WEDSP16 Digital Signal processor. AT&T application is the life member of FIETE, IE & MISTE.
note.
10. M. D. Felder, J. C. Mason, and B. L. Evans, Efficient dual-
tone Multifrequency detection using the non-uniform discrete
Fourier transform, IEEE Signal Process. Lett., vol. 5, no. 7, Dr P.Ramana Reddy, has completed M.E.
pp. 160163, Jul. 1998. (Electronics) from SGGSCET, Nanded,
11. Sun, welson, and neuendorffer " FPGA Pipeline Synthesis Maharashtra and Ph.D from
Design Exploration Using Module Selection and Resource JNTUniversity, Anantapur in the field of
Sharing" IEEE transactions on computer-aided design of Image Processing. Presently he is working
integrated circuits and systems, 2007, IEEE Trans. on as Associate Professor, ECE Department,
Computer Aided Design of integrated Circuits and Systems
JNTUA College of Engineering,
12. Sun, Hua. "Throughput Constrained and Area optimized
Anantapur. He has 18 years of teaching experience. He has
Dataflow Synthesis for FPGAs" A dissertation submitted to
the faculty of Brigham young university in partial fulfillment published 40 papers in national/ international
of the requirements for the degree of doctor of philosophy. journals/conferences. His research interests include Signal,
Brigham young university, april 2008. Image, Video and VLSI Signal processing. He has guided
13. G. Goertzel, An algorithm for the evaluation of finite around 50 M.Tech Projects and guiding 09 Ph.D. Scholars.
trigonometric series, Amer. Math. Monthly, vol. 65, no. 1, He is Fellow of IEI and IETE, life member of ISTE
pp. 3435, Jan. 1958.
14. M. Popovi, Efficient Decoding of Digital DTMF and R2
Tone Signalization, Facta Univ. Ser., Elec. Energ., Vol. 16,
No. 3, December 2003, pp. 389-399.
33
Implementation of digital Communication

Laboratory on FPGA
Molabanti praveen kumar1, T S R ksishna prasad2, M .Vijaya kumar3
M.tech student1, associate professor2, assistant professor3
molabanti.praveenkumar@gmail.com1,ad2prasad@gmail.com2,vij4b7@gmail.com3
ECE Department, Gudlavalleru engineering college, Gudlavalleru
Abstract I. IMPLEMENTATION
This paper presents a method to describe all modulation A. Binary Phase Shift Keying (BPSK)
techniques (like: BPSK, ASK, FSK, QPSK, QAM) on Binary Phase Shift Keying (BPSK) [1], [4], [6], [7]
Field Programmable Gate Array (FPGA) development demonstrates better performance than ASK and FSK.
board which is widely available and inexpensive. To PSK can be expanded to M-array scheme, employing
develop the system blocks Simulink environment and
multiple phases and Amplitudes as different states.
system generator version 13.1 are used under MATLAB
version 7.11 (R2010B). To achieve simulation and Filtering can be employed to avoid spectral
synthesis of Spartan 3 FPGA tools from Xilinx ISE 13.1 spreading.
are used. Very High Speed integrated circuit hardware
description language (VHDL) is used for describing the
hardware in system understanding language. Digital to
Analog converter is used to interface both FPGA and
CRO which is used to visualize the analog output of the Fig. 1. BPSK modulation.
digitally modulated signal. If the received data is logic 1, then the modulated
KeywordsAmplitude Shift Keying (ASK),Binary Phase Shift signal has no phase shift, if the received data is logic
Keying (BPSK),Frequency Shift Keying (FSK), Quadrature 0, then the modulated signal has 1800 phase shift as
Phase Shift Keying (QPSK), Digital Communication, Field shown in figure 1[1], [6].
Programmable Gate Array (FPGA), Receivers, Transmitters.
1. Simulink block diagram for BPSK using system
I. INTRODUCTION generator
The main objective of the paper is to implement all

the modulation techniques that are being used in
digital communication, all the modulation techniques
are Contained in a single configuration file that is
loaded into the Field Programmable Gate Array
(FPGA), no FPGA reconfiguration is necessary in
order to change the modulation, but rather than only a
user command. In the existing system [1], the Binary
phase shift keying (BPSK) modulation technique is
only discussed.
Fig. 2. BPSK modulator in the system generator.
In the proposed system we can include all possible
In the Simulink/system generator environment, we
modulation techniques along with BPSK. The
have to configure each block with their corresponding
developed FPGA board can also be used in courses
parameters to get the accurate output as shown in
for digital design, computer architecture and
figure 3 [1], [4].
embedded system, the flash memories on the board
are loaded with data for use in channel emulation, 2. Scope result for BPSK
and the proposed laboratory can be implemented on
Xilinx families of FPGA, like Spartan 3, FPGAs.
Compared with the software simulation tool such a
LABVIEW, which is quite expensive, the hardware
implementation of communication laboratory is very
less expensive.
Fig. 3. The wave forms on the scope.
34
C. Frequency Shift Keying (FSK)

B. Amplitude Shift Keying(ASK) Frequency Shift Keying (FSK) [2], [5] carries the
Amplitude Shift Keying ASK [2], [6] in the context information signal by representing the transmitter
of digital communications is a modulation process alphabet with M symbols using carriers with M
which imparts to a sinusoidal two or more discrete discrete frequencies. In general it is called M-array
amplitude levels 1. These are related to the number of FSK. When M is 2 then it is called Binary-FSK with
levels adopted by the digital message. two carriers usually termed as the mark and space
For a binary message sequence there are two levels, frequencies. The descriptive waveform example for
one of which is typically zero. Thus the modulated Binary-FSK is given in the figure 7.
waveform consists of bursts of a sinusoidal signal [2],
[6].Figure 4 illustrates an ASK signal (lower),
together with the binary sequence which initiated it
(upper).
Fig. 7. FSK modulation.
In FSK, two carriers f1, f2 are used whose

frequencies will differentiate between logic 1 and
logic 0 as shown in figure 7. Single carrier can also
Fig. 4. An ASK modulation and the data. be used for optimized system.
If the received data is logic 1, then the modulated 1. Simulink block diagram for FSK using
signal appears to be as carrier itself else if the system generator
received data is logic 0, then no signal will be there in
its corresponding time period as shown in figure 6.
1. Simulink block diagram for ASK using
system generator
Fig. 8. FSK modulator in the system generator

figure 8 [13], [4].
2. Scope result for FSK
Fig. 5. ASK modulator in the system generator.

figure 5 [2], [6].
2. Scope result for ASK
D. QPSK (Quadrature Phase Shift Keying)

Quadrature means the signal shifts among phase
states that are separated by 90 degrees, The signal
shifts in increments of 90 degrees from 45 to 135, -
Fig. 6. The wave forms on the scope. 45 (315), or -135 (225)
35
multiplexer is
1. QPSK Constellation Used, if the given data is shown in table 2[3].
2.
Carrier
Data transmitted Carrier phase S. Action BPSK ASK FSK QPSK
amplitude
o N
00 225 1.0
01 135o 1.0 1 Control bit 00 01 10 11
10 315o 1.0 2 Informatio 180,0 No change Carries 225,135,
11 45o 1.0 n phase amp f1,f2 315,45
,zero amp phase
Table 1. QPSK phases. 3 Parameter Phase Amplitude Frequency Quadrature
phase
135 o o
45
4 Carrier one one two Two
Table 2. Control Unit.
1.Simulink blockdiagram for control unit
225 315o
o
Fig. 10. QPSK constellation.
3. Simulink block diagram for QPSK using

systemgen
Fig. 11. QPSK Modulator in the system

generator. Fig. 13. Control unit in the system generator.
To get the accurate output as shown in figure 12, the To get the accurate output as shown in figure 13, the
parameters for each block must be configured parameters for each block must be configured
precisely in the Simulink/system generator precisely in the Simulink/system generator
environment [2], [5]. environment [3], [5].
3. Scope result for Quadrature Phase Shift 2.scope result for control unit
Keying(QPSK)

A. Control unit
In control unit , for controlling given data, the Fig. 14. The wave forms on the scope.
36
III. DESIGNFLOW In the complete setup shows the inter connections

between personal computer, Field programmable gate
array, cathode ray oscilloscope and regulated power
supply.
Fig. 16. The setup with Spartan 3 starter kit.

IV. SOFTWARE AND HARDWARE
V. RESULT ANALYSIS
A. software
This project uses MATLAB/ Simulink environment 1. Simulated result for BPSK
particularly Simulink library to connect all the blocks
to generate different modulations. Control unit in that
simulation library selects different modulations
according to data bits we have given.
System generator tools [13] converts corresponding

block diagram into VHDL code.
The ISE system edition [9] from Xilinx is a front-end

FPGA design solution that offers HDL synthesis and
2. Simulated result for ASK
simulation, implementation and used for interfacing
programs, user constraint files.
B. hardware
The complete lab measurement setup used for
realizing all kind of modulators is illustrated in
figures 15, 16. Some of the resources use there:
Spartan 3 starter kit board [8], [10], Digital to Analog
converter (NIFCO7A) with PQ28 package, Function
generator, Regulated power supply from Aligent and 3. Simulated result for FSK
Cathode Ray Oscilloscope.
C. Complete setup
Fig. 15. Complete Setup.
37
4. simulated result for QPSK
CONCLUSION AND FUTURE WORK

The work can be conclude that the implemented four
types of modulators in the Simulink environment,
like BPSK, QPSK, ASK, FSK using system generator
on FPGA. The will be like to extend my current work
by implementing all modulation techniques thus
whole digital laboratory can be done on a single kit.
REFERENCES
[1] Thotamesetty m Prasad, syed jahingir,Simulation and

implementation of a BPSK modulator on FPGA, ICECE- 16th
September 2012.
[2] F.Xiong, Digital Modulation Techniques, Artech House,UK,
2010.
[3] Swapan k samaddar, Atri sanyal,FPGA based generalized
architecture for Modulation and Demodulation techniques, JCT
journals, August 2012.
[4] Popescu, S. O.; Gontean, A.-S.; Budura, G., "Simulation and
implementation of a BPSK modulator on FPGA," Applied
Computational Intelligence and Informatics (SACI), 2011 6th
IEEE International Symposium on, vol., no., pp.459, 463, 19-21
May 2011.
[5] Linn, Y., "An Ultra Low Cost Wireless Communications
Laboratory for Education and Research," Education, IEEE
Transactions on, vol.55, no.2, pp.169, 179, May 2012
[6] Lopez-Villegas, J.M.; Macias-Montero, J.G.; Osorio, J.A.;
Cabanillas, J.; Vidal, N.; Samitier, J., "BPSK to ASK signal
conversion using injection-locked oscillators-part II: experiment,"
Microwave Theory and Techniques, IEEE Transactions on ,
vol.54, no.1, pp.226,234, Jan. 2006.
[7] Kikkert, C.J.; Blackburn, C., "Demodulating binary phase shift
keyed signals using programmable logic devices," Signal
Processing and Its Applications, 1999. ISSPA '99. Proceedings of
the Fifth International Symposium on, vol.2, no., pp.689, 692,
vol.2, 1999
[8] Spartan 3 FPGA Starter Kit board. User guide. Xilinx. 2011.
[9] ISE 13.1 Quick Start Tutorial, Xilinx, 2010.
[10] Spartan-3 FPGA Family Data Sheet. Xilinx . 2010.
[11] S.T.Karris, Introduction to Simulink with Engineering
Applications Orchard Publications, USA, 2006.
[12]http://www.ece.unm.edu/xup/docs/collaboration/BPSK_Mo-
dulator.pdf
[13] System Generator for DSP. Getting Started Guide. Xilinx.
2010.
[14]Banerjee, P.; Haldar, M.;Nayak, D.; Anderson, R.; Uribe, J.R.,
"Overview of a compiler for synthesizing MATLAB programs
onto FPGAs," Very Large Scale Integration (VLSI) Systems, IEEE
Transactions on , vol.12, no.3, pp.312,324, March 2004.
38
Design of Non-Uniform Circular Arrays for

Side lobe Reduction Using the Method Of Real
Coded Genetic Algorithm
Ms.M.Nirmala
Assistant Professor, Department of ECE, ANITS Engineering College, Visakhapatnam
Abstract - This paper is dealth with design problem other systems. In this Real Coded Genetic Algorithm
of non-uniform circular antenna arrays for maximal (RCGA) techniques is applied to design of non-
side lobe level reduction. This problem is modeled as uniform circular antenna arrays.The method of Real
a simple optimization problem. In this case, the Coded Genetic Algorithm (RCGA) is used to
antenna array[1] design problem consists of finding determine an optimum a set of weights that provide a
weights that provide a radiation pattern with maximal radiation pattern with maximal side lobe level
side lobe level reduction. Due to the great variety of reduction[4].
parameters involved, optimization techniques[11]
such as Real Coded Genetic Algorithm (RCGA)[5] II. ARRAY FACTOR ANALYSIS
are very appropriate tools to search for the best
antenna models. Genetic Algorithm techniques are a. Introduction to Real Coded Genetic
becoming widely used to solve electromagnetic Algorithm (RCGA)
problems due to their robustness, wide range of
applications and readiness in their implementation. The aim of antenna array geometry synthesis is to
The effectiveness of Real Coded Genetic Algorithm assess the physical setup of the array that produces
(RCGA) for the design of non-uniform circular arrays the radiation pattern that is nearest to the desired
is shown by means of experimental results. pattern. There are many problems in electrical
Experimental results reveal that design of non- engineering for which analytical solutions do not
uniform circular antenna array using the method of exist. In fact, most practical electromagnetic problems
Real Coded Genetic Algorithm (RCGA) provides a are nonlinear and non-differentiable. These problems
considerable side lobe level reduction with respect to require engineers and scientists to employ numerical
the uniform case. The purpose and contribution of methods. These methods provide no insight into the
this project is to present a model of problem that global optimum solution of the problem. Thus, the
includes design of non-uniform circular antenna numerical solvers resort to marching parameters up
arrays for side lobe level reduction using the method and down until a global optimum is obtained. These
of Real Coded Genetic Algorithm (RCGA). brute force methods are time consuming and
unintuitive. Real Coded Genetic Algorithm (RCGA)
I. INTRODUCTION stochastically evolve a population towards a solution
using the concepts of survival of the fittest. These
The design of circular antenna arrays[12] finds algorithms allow the global optimum, which is the
application in areas as mobile and wireless goal of all optimization techniques, to be attained far
communications systems. Generally speaking, the faster than the traditional brute-force techniques.
problem of designing antenna arrays is characterized
by different and conflicting requirements (beam b. Sequence of operation in Real Coded Genetic
width, side lobe level, directivity, noise sensitivity, Algorithm (RCGA)
robustness) to be satisfied. In this project a design 1. Generate an Initial Population
criterion is considered to evaluate the performance of Randomly generated quantized genes
circular array for minimum side lobe level[4]. (binary)
Each gene contains the genetic code of
Real Coded Genetic Algorithm (RCGA) techniques each individual
have been fairly successful at designing linear 2. Evaluate Fitness (or Objective) Function,
antenna arrays. However, array configurations in f(x)
which the elements are placed in a circular ring are of quality of individuals in the population
great interest. They have applications in radio
direction finding, air and space navigation, radar, and
39
Use numerical solver (i.e. NEC, Wire) implemented instead of a common weighted roulette
to calculate properties such as wheel selection.
Directivity, Bandwidth, etc.. The function update population assigns ranks to
3. Select the best members of the population individuals in the population generated by the union
based on Fitness of parents and children. This is in order to hold the
There are many methods to select the best individuals in each generation. The explanation
best members of the procedures involved at each step of this
4. Crossover or mate the individuals in the algorithm in detail.
population
This occurs with a probability pcross
(.6 < pcross < .9 )
Choose two members, swap their
genetic codes to create two new
members.
5. Mutate the individuals
This occurs rarely with a probability
pmut (0 .001 < pmut < 0.01)
This allows changes in the population
to not only slow the convergence rate,
but to also allow unique members to
appear.
6. Evaluate Fitness again
7. If the population has converged to a Fig.1: Geometry and notations used for non-uniform
superior race, end the process, otherwise circular antenna array
loop back and continue
The individual representations as well as the
c. Problem Statement crossover and mutation operators are explained in the
following subsections.
Design of a Non Uniform circular array that provide a
III. ARRAY FACTOR
radiation pattern with maximal side lobe level[10]
reduction.Real Coded Genetic Algorithm is applied to
Consider a circular antenna array of N antenna
design of Non-Uniform circular antenna arrays.The
elements uniformly spaced with a distance d on a
method of Real Coded Genetic Algorithm is used to
circle of radius a in the xy plane as shown in below
determine an optimum a set of amplitude excitation
figure 1. If the N elements in the circular antenna
weights to provide a radiation pattern with maximal
array are taken to be isotropic sources, the radiation
side lobe level reduction. Initially Uniform circular
pattern of this array can be described by its array
radiation pattern is generated for number of elements
factor.
N=10 and the sidelobe level for that is calculated.
The array factor for the circular array in the xy plane
Design a Non-Uniform circular for N=10 and 20 in figure 1 is given by[2]
with maximum SLL for different set of amplitudes.
These different set of element amplitude
[ ( ) ]
excitations[3] are generated by using Real Coded ( , )=
Genetic Algorithm. The main purpose is to design a
low side lobe radiation pattern for non-uniform
circular antenna arrays. For this purpose, a Where
population-based stochastic procedure is used In = Excitation of nth element
denominated Real Coded Genetic Algorithm. I chose N= no.of elements in circular ring
this algorithm for its easiness of implementation. The
k= phase constant
procedure for used Real Coded Genetic Algorithm is

described as follows. a= where d =
The function generate initial population randomly and Then ka =
uniformly generates a set of individuals. The main
idea in classify individuals is to rank the individuals n =
according to their fitness values. A selection scheme Where n = 1,2,.N
combining fitness ranking and elitist selection is n = ka cos ( 0 )
40
I = [I1, I2,., IN], In represents the excitation of parameters[8], after a trial and error procedure, were
the nth element of the array, k=2/ is the phase set as follows:
constant, is the angle of incidence of a plane wave, maximum number of Generations G = 200
is the signal wavelength and 0 is the maximum Chromosome size c = 80
radiation angle. Gene size b = N
Now need to formulate the objective function we
Where N =10 and 20.
want to minimize. Let us introduce first a couple of
definitions for the radiation pattern[1]. Let 0 be the Crossover probability pc = 1.0 and
angle where global maximum is attained in = [-, Mutation probability pm = 0.1
]. Let msl be the angle where the maximum side lobe
is attained.Based on the above considerations the
objective function[2][7] can be written as
F 1= |AF(msl, I)| / | AF(0, I)|
Where
msl = The angle where the maximum side
lobe is attained
0 = The angle where global maximum is
attained
IV. ALGORITHM AND FLOWCHART
1. Generated the initial population with

Chromosome size of c and gene size of
b[6].
2. Calculated the SLL for each Chromosome
and assign rank to each chromosome
according to SLL value. Update the initial
population according to rank.
3. Apply crossover and mutation to the updated
population with crossover propobility pc of 1
and mutation probability pm of 0.1.
4. Update the population after applying
crossover and mutation with chromosome
size of c.
5. Calculated the SLL for each Chromosome
and assign the rank to each chromosome
according to SLL value. Generate new
population according to rank.
6. Repeat the step 2 until no of generation reach
the maximum value Gmax.
7. Consider the SLL at Gmaxth generation and
1st Chromosome in new generation which
gives the required fitness and amplitude
distribution respectively.
8. Plot the Radiation pattern for amplitude
distribution in step 7.
Flow chart is given in figure 2.
V. EXPERIMENTAL SETUPS
In this experimental setup non uniform distribution

with uniform distance is considered. the radiation
patterns of the circular arrays with main lobe steered
to 0 = 0 degrees are considered. Several experiments
were carried out with number of antenna elements b Fig.2: Flow chart representation of Real Coded
(N = 10,20). In this the Genetic Algorithm Genetic Algorithm
41
VI. RESULTS FOR APPLICATION OF REAL 1
CODED GENETIC ALGORITHM TO A 0.9
CIRCULAR ARRAY 0.8
0.7
0.6
In the present work, a uniform circular array[9] is
Amplitude
0.5
considered and its field pattern is computed 0.4
numerically and represented in figure 3. To minimize 0.3
the sidelobe levels of Radiation pattern of a circular 0.2
0.1
array, the amplitude excitation coefficients are
0
optimized by using a Real Coded Genetic Algorithm. 1 2 3 4 5
Element number
6 7 8 9 10
Real Coded Genetic Algorithm is applied to a circular Fig.5:Amplitude distribution for N=10 using
array for different number of array elements for N=10 Real Coded Genetic Algorithm
and 20.
The amplitude excitation coefficients are synthesized The amplitude excitation coefficients and their
by using Real Coded Genetic Algorithm. By using respective Radiation patterns are presented in figure 5
the amplitude excitation coefficients, patterns are and figure 7. Computations are carried out for both
numerically computed. small and large arrays and the computed amplitude
excitation coefficients are presented in Table 1. The
1 reduced SL Level and Beamwidths are also presented
0.9 in the table.
0.8 0
0.7
-5
0.6
Amplitude
0.5
-10
Array factor (AF)in dB
0.4
0.3 -15
0.2
-20
0.1
0
1 2 3 4 5 6 7 8 9 10 -25
Element number
-30
-150 -140 -130 -120 -110 -100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 10 20 30 40 50 60 70 80 90 100 110 120 130 140 150
theta(in degrees)
Fig.3: Amplitude Distribution for N=10
Fig.6: Radiation pattern of circular array for
N=10 using Real Coded Genetic Algorithm
0
-5
-10 0.9
0.8
-15 0.7
0.6
-20
0.5
Amplitude
0.4
-25
0.3
0.2
-30
-150 -140 -130 -120 -110 -100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 10 20 30 40 50 60 70 80 90 100 110 120 130 140 150
theta(in degrees) 0.1
Fig.4: Radiation pattern of circular array for 0

0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20
Element number
N=10
Fig.7: Amplitude distribution for
N=20 using Real Coded GeneticAlgorithm
42
REFERENCES
0
[1]. G.S.N.Raju, Antennas and Wave Propagation, Pearson Edu,
-5
2005.
[2]. Marco A. Panduro, Aldo L. Mendez, Rene Dominguez,
-10 Gerardo Romero Design of non-uniform circular antenna
arrays for side lobe reduction using the method of genetic

-15 algorithmsInt. J. Electron. Commun. (AE) 60 (2006) 713
717
-20
[3]. Ares-Pena FJ, Rodriguez-Gonzalez JA, Villanueva-Lopez E,
Rengarajan SR. Genetic algorithms in the design and
-25
optimization of antenna array patterns. IEEE Trans Antennas
Propag 1999;47:50610.
-30
-150 -140 -130 -120 -110 -100 -90 -80 -70 -60 -50 -40 -30 -20 -10 0 10 20 30 40 50 60 70 80 90 100 110 120 130 140 150
Fig.8: Radiation pattern of circular array for N=20 theta(in degrees)
[4]. Bray MG, Werner DH, Boeringer DW, Machuga DW.
Optimization of thinned aperiodic linear phased arrays using
using Real Coded Genetic Algorithm genetic algorithms to reduce grating lobes during scanning.
IEEE Trans Antennas Propag 2002;50:173242.
Table 1: Amplitude excitation coefficients for various [5]. Haupt R. Thinned arrays using genetic algorithms. IEEE
no.of elements, their SLL, FNBW and HPBW using Trans Antennas Propag 1994;42: 9939.
Real Coded Genetic Algorithm(RCGA). [6]. Yan KK, Lu Y. Sidelobe reduction in array-pattern synthesis
using genetic algorithm. IEEE Trans Antennas Propag
1997;45:111722.
[7]. Panduro MA, Covarrubias DH, Brizuela CA, Marante FR. A
multi-objective approach in the linear antenna array design.
AEU Int J Electron Commun 2005;59:20512.
[8]. Lommi A, Massa A, Storti E, Trucco A. Sidelobe reduction
in sparse linear arrays by genetic algorithms. Microwave Opt
Technol Lett 2002;32:1946.
[9]. Lopez P, Rodrguez JA, Ares F, Moreno E. Low sidelobe
patterns from linear and planar arrays with uniform
excitations except for phases of a small number of elements.
Electron Lett 2001;37:14957.
[10].nd Stutzman WL, Thiele GA. Antenna theory and design.
2 ed., Chichester, New York, Brisbane, Singapore,
Toronto: Wiley; 1998.
[11]. Golberg DE. Genetic algorithms in search, optimization,
and machine learning. Massachusetts: Addison-Wesley;
1989.
[12]. Constantine A.Balanis Antenna Theory analysis ans
design. 3rd ed., Wiley India, 2005.
VII. CONCLUSION
In the present work, an attempt is made to design a

nonuniform circular antenna arrays for maximal side
lobe level reduction. The well-known method of Real
Coded Genetic Algorithm (RCGA) is proposed as the
solution for this design problem. The method of
genetic algorithms efficiently computes the design of
non-uniform circular antenna arrays to generate a
radiation pattern with maximal side lobe level
reduction.
Experimental results reveal that design of non-
uniform circular antenna arrays using the method of
Real Coded Genetic Algorithm (RCGA) provides a
considerable side lobe level reduction with respect to
the uniform case. It is evident from results, that
sidelobe level (SLL)[10][12] and Beamwidths are
reduced with increase of the no.of elements in array.
43
Effect Of Dielectric Superstrate On The

Characteristics Of Square Patch Antenna
1* 2
V. Saidulu , K.Srinivasa Rao , P.V.D.Somasekhar
3
Rao
1
Department of Electronics and Communication, MGIT,Hyderabad, AP, India
2
Department of Electronics and Communication, VIF, Hyderabad, AP, India
3
Department of Electronics and Communication, JNTU,Hyderabad, AP, India
Abstract: A dielectric superstrate or radome is above a light weight, low profile, conformable to planar and
square patchmicrostrip antenna has a remarkable effect non-planar surfaces and easy to fabricate. The
on its characteristics such as gain, bandwidth, antenna is suitable for high speed vehicles,
beamwidth, radiation pattern and resonant frequency. aircrafts, space crafts and missiles because of Low
This paper experimentally investigates the effect of a
profile and conformal nature of characteristics [2].
dielectric superstrtes with and without on the
performance characteristics of coaxial probe fed square Microstrip antenna has inherent limitation of narrow
patchmicrostrip antenna with varying the various bandwidth. So, superstrate(radome) is used
thickness of dielectric superstrates on gain, bandwidth, on a microstrip antenna as a cover to protect the
beamwidth, and radiation pattern and resonance antenna from external environmental conditions like
frequency. The antenna designed frequency is temperature, pressure etc. When microstrip antenna
2.4GHz(ISM band) and fabricated on Arlondiclad 880 covered with a dielectric superstrate(radome) its
substrate, whose dielectric constant () is 2.2, loss properties like resonance frequency, gain,
tangent (tan ) is 0.0009 and the thickness (h) is 1.6mm bandwidth and beam width are changed which may
and the 1 same substrate can be used as dielectric
seriously degrading the antenna performance[1-4].
superstrat whose dielectric constant () 2.2 ,at
various thickness such as 0.2mm, 0.5mm, 0.8mm,
By choosing the thickness of the substrate
1.0mm, 1.3mm, 1.5mm, 2.2mm, 2.4 and 3.2mm on the superstrate layer, a very large gain can be achieved
patch antenna with same specification. Experimental [5-9]. Coaxial probe fed squaremicrostrip
measured result observed that when the dielectric antenna characterstics have been investigated using
superstrate is above the substrate, the antenna resonant High Frequency Structure Simulator (HFSS) softer
frequency is altered and shifted towards the lower side and measured experimentally. When microstrip
of frequency of operation from 2.4GHz to 2.33 GHz, antennas are covered with protective dielectric
while other parameters have slight variation in their superstrates, are subjected to icing conditions, or
values. The gain of single patch antenna without come into contact with plasma, the resonant
dielectric superstrate is 4.8 dB and the microstrip patch
with dielectric superstre is varied from 0.47dB to
frequency is altered and shifted to lower sides,
3.43dB based on the thickness of the superstrate. The causing detuning which may seriously degrading the
bandwidth of microstrip single patch without dielectric antenna performance. As the bandwidth and gain of
superstrate is 4% and the microstrip patch with microstrip antennas is inherently low, typically of
dielectric superstrte is 1.0% to 2.0% based upon the the order of 1-2% [1]. It is important to determine
thickness of the dielectric superstrtes. The beamwidth the effect of dielectric superstrate on the resonant
for single patch without 2 dielectric superstrte in E- frequency of the microstrip antennas and in order
Plane is 108.160 and in H-Plane is 90.20. The beamwdth introduce appropriate corrections in the design of
with dielectric superstre in E-Plane is 95.40to 105.330 microstrip antennas. In this paper experimentally
and in H-Plane is 74.860 to 83.610. The value of VSWR
is from 1.38 to 2.80 based upon the thickness of the
measured the effect of dielectric superstrates on the
dielectric superstrates,as dielctricsuperstrat thickness antenna parameters such as bandwidth, beamwidth,
increases, VSWR increases, return loss increases, gain gain, resonant frequency and radiation pattern with
and bandwidth decreases. varying variousthickness of dielectric superstrates
Keywords: Square patch microstrip antenna, dielectric etc.
superstrate, VSWR, Gain, Beam- width, Bandwidth,
resonant frequency etc. II. ANTENNA SPECIFICATIONS:
I. INTRODUCTION: The geometry of a probe fed

squarepatch microstripantenna is shown in Fig1. The
Square microstrip antenna consists of radiating antenna under investigation is a patch of width (W)
patch on the one side of the substrate having the is 33.6mm, length (L) is 33.6 mm fabricated on
ground plane on other side. The major advantages are Arlondiclad 880 dielectric substrate, whose
44
dielectric constant( 1 ) is 2.2, loss tangent(tan )

is 0.0009, thickness(h) is 1.6mm and Substrate
dimension is 100mm100mm. The dielectric
superstrate is Arlon diclad 880 dielectric substrate,
whose dielectric costant ( 2 ) is 2.2, loss tangent (tan ) is
0.0009, thickness (h) is 1.6mm and substrate
dimension is 100mm100mm.
The antenna center frequency is 2.4GHz(ISM

band) and correspondingfeed location is X=0 and Fig1. Square microstrip patch antenna geometry
Y=10.0 is shown in Fig1. The ground plane is
perfectly conducting. Antenna geometry, material
properties and boundary conditions are considered
and simulated using HFSS. The variation of
superstrate thickness above the substrate studied
experimentally for various parameters such as
Bandwidth, Gain, Beam width, resonant frequency
radiation pattern etc.
III. SELECTION OF SUBSTRATE AND Fig 2.Microstrip antenna with superstrategeometry

SUPERSTRATES:
Suitable dielectric substrate of appropriate thickness

and loss tangent is chosen for designing the square
patchmicrostrip patch antenna. A thicker substrate is
mechanically strong with improved impedance
bandwidth and gain [10]. However it also
increases weight and surface wave losses. The
( )
dielectric constant is play an important role
similar to that of the thickness of the substrate. A Fig 3. Geometrical structure of square patch antenna

low value of for the substrate will be increase the
fringing field of the patch and thus the radiated
power.
(tan )
A high loss tangent increases the dielectric
loss and therefore reduce the antenna
performance. The substrate and superstrate
parameters chosen for designing the patch antenna as
follows: The substrate material is ArlonDiclad 880
dielectric constant ( 1 ) is
is substrate and whose Fig 4. Square patch antenna with superstrate
2.2, loss tangent (tan )
is 0.0009 and thickness
(h) of the material is 1.6mm. The superstrate
materials as same as substrate, dielectric
constant is 2 = 2.2and whose substrate dimension
is
100mm100mm and same specification.
The low dielectric constant materials increase
efficiency, bandwidth and better for radiation. The
dielectric constant of the substrate and superstrateis
shown in table 1 and table 2 and also substrate
and superstrate material is shown in Fig 5. Structure of square patch with dimension.
Fig 8.
45
IV. DESIGNING OF SQUARE MICROSTRIP

PATCH ANTENNAS:
TRANSMISSION- LINE MODEL:

The designing of the square microstrip patch antenna,
the transmission-line model and cavity model is used,
because easy to analysis and good physical
insight. Using the cavity model. Basically the
transmission line model represents the microstrip
antenna by two slots, separated by a low
impedance transmission line of length L.
FRINGING EFFECTS:
The q factor is referred to as the fringing factor
The amount of fringing is a function of the (length reduction factor). As the substrate height
dimensions of the p atch and the h e i g h t of the increases, fringing also increases and leads to larger
s ub str a te . Since fo r microstrip antennas L/H >>1, separations between the radiating edges and lower
fringing is reduced; however, it must be taken into resonant frequencies.
account because it influences the resonant
frequency of the antenna. The effective dielectric DESIGN:
1 < <
constant has values in the range of most
applications where the dielectric constant of the
greater than the unity ( 1), the
substrate is much
value of will
be closer to the value of the actual
dielectric constant
of the substrate.
CONDUCTANCE:
Each radiating slot is represented by a parallel

quivalent
admittance Y (with conductance G and susceptance
EFFECTIVE LENGTH, RESONANT FREQUENCY, B).
AND EFFECTIVE WIDTH:
Because of the fringing effects, electrically the
patch of the microstrip antenna greater than its
physical dimensions. The dimensions of the patch
along its length have been extended on each end by a
distance, which is a function of the effective
dielectric constant
For the dominant 010 mode, the resonant

frequency of the microstrip antenna is a function of
its length. Usually it is given by
46
RESONANT INPUT RESISTANCE:

The total admittance at slot 1 (input admittance) is
obtained by transferring the admittance of slot 2 from
the output terminals to input terminals using the
admittance transformation equation of transmission
lines.
or
Where the plus (+) sign is used for modes with odd
(ant
symmetric) resonant voltage distribution beneath the
patch and between the slots while the minus (-) sign V. MICROSTRIP ANTENNAS COVERED WITH A
is used for modes with even (symmetric) resonant DIELECTRIC SUPERSTRATES:
voltage
distribution. The mutual conductance is defined, in When microstrip line (antenna) is covered with the
terms of the far- zone fields, as dielectric superstrate or radome the characteristics of
antenna parameters change as a function of the
dielectric superstratelayer. The properties of a
Where 1 is the electric field radiated by slot 1, 2 is microstrip antenna covered with dielectric superstrate
the magnetic field radiated by slot 2 , 0 is the layer have been studied theoretical formulationusing
voltage across slot, and the integration is performed the transmission line and cavity model. The resonant
over a sphere of large radius. It can be shown that 12 frequency of a microstrip antenna covered with
can be calculated dielectric superstrate layer can be determined when
the effective dielectric constant of the structure is
47
known. The change of the resonant frequency by

placing the dielectric superstrate has been calculated
using the following the expression.
Fig 8. Structure of dielectric substrate and suprstrate

materials
VII. EXPERIMENTAL MEASUREMENTS:
The geometrical structure under consideration is

shown in Fig1. A square patch antenna, designed
patch width(W)= 33.6mm and length(L)= 33.6mm
was fabricated on thick dielectric substrate whose
dielectric constant 2.2, loss tangent is 0.0009,
VI. DESIGN OF MICROSTRIP ANTENNA thickness is 1.6mm and the dielectric superstrate is
WITH DIELECTRIC SUPERSTRATE: same substrate pecification. The patch was fed
through probe of 50 cable. The location of feed
The microstrip antenna can be designed using with probe had been found theoretically and chosen as
and x=0, y=10.0mm. Then the patch was covered with
without dielectric superstrates. The antenna dielectric superstrate material such as ArlonDiclad
dimension 880 whose dielectric constant () is 2.2, loss tangent
such as width(W) is 33.6mm and length(L) is (tan ) is 0.0009 and thickness (h) is 1.6mm.
33.6mm and feed point location is X=0 and Y=10.0.
The antenna The impedance characteristics were measured by
designed frequency is 2.4GHz(ISM band) and means of HP 8510B network analyzer. The radiation
abricated on Arlondiclad 880 substrate whose pattern measurements were performed in the
substrate dimension is 100mm100mm and whose anechoic chamber by the use of automatic antenna
dielectric constant() is 2.2, loss tangent(tan ) is analyzer. The measured results areshown in tables 3,
0.0009 and the thickness (h) of the material is 1.6mm. 4, and 5. The measured far field radiation patterns
The superstrate material can be used as same as the and VSWR plots for various thickness such as
dielectric substrate in the design of squaremicrostrip 0.2mm, 0.5mm, 0.8mm, 1.0mm, 1.3mm, 2.2mm,
patch antenna. We observed that when the dielectric 2.4mm and 3.2mm is shown in Fig 9 to Fig 22.
supeastrate is placed above the substrate the antenna
parameter will be changed. The resonance frequency
is shifted towards the lower side of frequency of
operation from 2.4GHz to 2.33 GHz(ISM band),
while other parameters have slight variation in their
values with the thickness of the superstrates. The
antenna designed prototype patch with feed point
location is shown in Fig 7 and square patch
microstrip antenna covered with is shown in Fig 6.
Fig7 . Fabricated Porto type square patch with feed

point location
48
Fig 11. Far field radiation pattern with and without

radome at 3.2mm in horizontal polarization
49
IX. MEASURED RESULTS AND

DISCUSSION:
In order to verify the simulation results for the square

patchmicrostrip antenna with and without dielectric
superstrates. The antenna designed frequency is
altered and shifted to lower side from 2.4GHz to
2.33GHz, when placing the dielectric superstrate on
the substrate materials. The experimental carried out
the varying the various thickness of the dielectric
superstrate on the microstrip patch antennas. We
observed experimentally the resonating frequency is
shifting to lower side 2.40 GHz to 2.33GHz. The
value of VSWR without dielectric superstrate is 1.6
50
and with superstrate is 2.80, the gain valve without gain 3.43dB and 0.2mm thicknes is best for higher
dielectric superstate is 4.8dB and with dielectric BW is 2%.
superstrate is varied according to thickness from
0.47dB to 3.43dB, the bandwidth value without REFERENCES:
dielectric superstrate is 4.0% and with dielectric
superstrate is 1.0%2.0%.We observed experimental 1. IE3D Manual, Zeland software Inc.,Fremount, USA, 1999
measured data in tables 3, 4 and 5 and also gain, 2. I J Bhal and P Bhartia, Microstrip antennas, Artech
radiation pattern in vertical and horizontal plane and house, 1980.
3. R.Shavit,Dielectric cover effect on Rectangular Microstrip
VSWR plot with and without superstrate is shown in Antennas array. IEEE Trans. Antennas propagat.,Vol 40,. PP.992-
Fig 9 to Fig 22.The discussion we conclude that in 995, Avg.1992.
the following points. 4. Inder ,Prakash and Stuchly, Design of Microstrip Antennas
1. Variation of VSWR with different dielectric covered with a Dielectric Layer. IEEE Trans. Antennas Propagate.
Vol.AP-30.No.2, Mar 1992.
superstrate (radome) thickness, as dielctricsuperstrate 5. O.M.Ramahi and Y.T.LO,Superstrate effect on the Resonant
thickness increases, VSWR increases. frequency of Microstrip Antennas, Microwave Opt.Technol. Lett.
2. Variation of Gain at different dielectric Vol.5, PP.254-257, June 1992.
superstrate(radome) thickness as dielectric superstrate 6. A.Bhattacharyya and T. Tralman, Effects of Dielectric
Superstrate on patch Antennas, Electron Lett., Vol.24,PP.356-
thickness increase, the gain decreases. 358, Mar 1998.
3. The antenna Beamwith in E-Plane increases from 7. Patil V.P, Kharade A.R Enhancement of directivity of RMSA
95.20 degree to 105.33dregree and the antenna Beam using multilayer structure, IJERD,79-84, 2012.
width in H-Plane increases from 74.86 degree to 8. Patil V.P, Kharadea.r Enhancement of Gain of RMSA using
multilayer structure, IOSRJECE,2278-34,2012.
83.61 degrees. 9. M..Younssi,A.Jaoujal Study of MSA with and without
4. The return loss first increases with increasing superstrate for Terahz frequency, ISSR Journal, 2013.
thickness of dielectric superstrates and then decrases. 10. L. Yousefi, H. Atta High gain patch antenna loaded with high
5. The bandwidth of the microstrip antennas also chr. Impedance superstrate, vol.10, 858-861, 2011.
11. S.D.Gupta, A. Singh, Design and analysis of multidielectric
increases with increasing thickness of dielectric layer MSA with varying superstrate layer chracterstics, IJAET,
superstrate for low dielectric constant materials, and vol.3, pp. 55-68, 2012.
decreases for high dielectric constant of the materials. 12. R.K Y adav, R.L.Yadave Effect on performance char. Of
rectangular patch antenna with varying height of dielectric cover,
IJPCSC, vol.2, no.1, ISSN: 0976-268X.
13. H.Attia, L.Yousefi and O.M.Ramahi Analytical model for
X. CONCLUSION: calculating the radiation fields of MSA with artificial
maganeticsuperstrates: Theory and experiment. IEEE Tranc.
The design of square patchmicrostrip antennas with Antennas and wave progation, vol.59,2011.
14. M..Younssi,A.Jaoujal Study of MSA with and without
dielectric superstrates have been presented. The superstrate for Terahz frequency, ISSR Journal, 2013
resonant frequency of a microstrip covered with
dielectric superstrates can be predicted accurately if
the effective dielectric constant of the structure is
known. The effective dielectric constant can be
calculated using transmission line and cavity model.
The experimentally results shows that the variation of
VSWR with different dielectric superstrate(radome)
thickness, as dielectric superstrate thickness
increases, VSWR increases. The variation of the
antenna gain at different dielectric superstrate
thickness as dielectric superstrate thickness increases,
the gain decreases. The bandwidth of the microstrip
antennas can also increases with increasing thickness
of the dielectric superstrate for low dielectric constant
materials, and decreases for high dielectric constant
of the substrate materials. Initially the return loss
increases with increasing thickness of dielectric
superstrates and then decreases. The antenna
beamwidth in E-Plane increases from 95.2 degree
to105.33 degree and beamwidth in H- Plane increase
from 95.20 degree to 105.33 degree and among
various thickness 2.2mm thicknes is best for higher
51
Analysis On Flexural Vibration Characteristics

of Hemispherical Shell
Shailesh kumar,Anil badiger, Kishore chandra das, Runa banerjee,
INS, Research Centre Imarat
Hyderabad, India
Shailesh.Kumar@rcilab.in, KishoreChandra.Das@rcilab.in
Abstract Mechanical gyros based on hemispherical right and left in the northern and southern
structure and working on the principle of Coriolis effect hemispheres, respectively. The effect was first
play a vital role in the navigation systems for very long observed by Gaspard-Gustave Coriolis, a French
range missions and deep space probes The precessional scientist who in 1835 described the forces that arise
phenomenon and flexural vibration characteristics of from the motion of an object in a rotating reference
hemispherical resonator shell for such a navigation frame [2].
application are investigated in this paper. Thin shell
theory and virtual displacement principle are employed Hemispherical Resonator Gyro (HRG), as a kind
to model hemispherical shell. The influence of shells of Coriolis Vibratory Gyros (CVG), is the only CVG
geometry parameters on its natural frequency and that meets the inertial grade performance. HRG is
precessional rate is analyzed. Depending on ABAQUS- solid state gyroscope whose sensing property is based
finite element analyzer, the influence of shells geometry on standing vibration precession. It has features of
parameters on each free frequency is analyzed for high accuracy, long life span, inherent high reliability,
optimal Q-factor. natural radiation hardness and small size.HRG is one
type of vibrating gyros that can work as an angle
KeywordsHemispherical Shell; flexural vibration; q- sensor as well as an angular rate sensor depending on
factor; resonant frequency.
its mode of operation. The two operation modes are
I. INTRODUCTION Whole Angle (WA) mode and Force to rebalance
mode (FTR). In WA mode, the standing wave is
Gyroscope is one of the most important sensors in allowed to precess freely with respect to the Coriolis
navigation applications. It is applicable to several forces due to external rotation. In FTR mode the
areas such as aircraft, satellite, and oil drilling- standing wave is fixed at a specific location by
exploration and so on. However, classical gyros are applying the rebalance force. The angular rate can be
composed of many mechanical moving parts which estimated by the force applied to fix standing wave. In
can cause the operation failure. In that sense, vibratory general, WA mode has relatively larger dynamic
gyroscope is relatively effective sensor since it has no range with relatively low accuracy while FTR mode
moving parts which mean that it has longer operation has shorter dynamics range with high measurement
time compared to classical mechanical gyros. accuracy [3].
Working principle of the vibratory gyros is based on
the Coriolis Effect. If the external rotation of the I. MODEL OF RESONATOR
platform occurs while the vibratory gyro is at its
resonance, this external rotation causes the precession
Dynamic Precession Analysis
of the gyros. By measuring the precession angle, one
can measure the amount of external rotation.
The Hemispherical Resonator Gyro, HRG, has its
notional roots back in the late 19th century when
G.H.Bryan had his seminal paper, On the beats in the
vibrations of a revolving cylinder or bell published in
the proceedings of the Cambridge Philosophical
Society in 1892 [1]. In this paper he mathematically
described the nature of the beats that are heard as a
result of rotating a vibration shell about its symmetric
(or cylindrical) axis. In a vibrating shell such as a
wineglass, rotation of the wineglass about its stem will
cause the nodes of vibration on the rim of shell to
move at precession rate slower than shell itself. The
physics of HRG is based on the forces arising from The basic dynamics of shell for the second
Coriolis acceleration. This is the same force that resonant mode, n=2, half period is shown in
drives direction of winds and ocean currents to the fig1.Assuming the shell rotates at an angular velocity
52

, analyzing a small element at a point A: its
amplitude and nullify the quadrature of vibration
without creating drift. The undesirable generated drift
transport velocity is V1 and Coriolis acceleration is depends on two characteristics:

Wk 2 V1 . For an element m, if it receives a The modulus of the forces needed to
maintain the vibration and nullify the
Coriolis acceleration Wk , it can generate an inertia quadrature which are defined by the physical
characteristics of the resonator (damping and
force Fk with same magnitude but in opposite anisofrequency) and directly linked with the
direction. For other elements the elastic deformations precision of manufacturing process.
are similar. The Coriolis acceleration and inertia force The errors generated by electronics used to
are shown in fig.1. Due to the inertia force caused by
estimate the position of vibration and to
, shadow region is compressed, while blank area is apply the forces to the resonator.
extended [4]. The result is precessional phenomenon: The displacement vector of an arbitary point on
the vibration wave is not standing anymore, but a the shell is given by:

negative relative angular velocity with . ua v b wc
(1)
Co-ordinates and Equations of Vibration Mode:
A model for a thin axis-symmetrical Where:
hemispherical shell is shown in fig 2. It has two Y in
u, v,w are the magnitude of the displacement
orthogonal modes of vibration that are coupled by the generatrix, ring, and radial directions,
Coriolis forces. The basic equation of the system is respectively[8].
X
the following:
fx a , b , c are the unit vectors in corresponding
x 2. x . x . x x2 . x 2. .. y directions.
m
By detailed analysis of inertia force F, it can be
fy derived that: when resonator rotates at an angular
y 2. y . y . y . y 2. .. x
2

y
m velocity of , x yz , if 2 2 ,
/ t 2 , the influence of original stress can
In this equation, x and y stands for the be neglected, the force caused by angular acceleration
position of the vibration in each of the two modal doesnt affect the vibration mode.
direction, x and y are the pulsation of each mode, The vibration of a rotating hemispherical resonator
x and y are the damping of each mode, f x and which is assumed to be isotropic can be formulated as:
u ( , , t ) u ( ) cos n ( ) cos t
f y are the forces applied to each mode, m is the
v ( , , t ) v ( ) sin n ( ) cos t
modal mass, is the rotation rate and is a w ( , , t ) w ( ) cos n ( ) cos t
coefficient called Bryan Coefficient. (2)
The solution of such a system is an ellipse of
amplitude a, angle , phase and quadrature b. Where:
is the precession angle,
t

0
Pdt , P K x
K is precession coefficient.
n is number of ring waves
is resonator shell frequency.
In above equation we can see that Coriolis
forces are homogeneous to a damping force making The kinetic energy of the shell can be expressed as:
the angle of the vibration to rotate. This is the / 2 2
1 2
K r h ( u v 2 w 2 ) sin d d
2
physical phenomenon that allow to measure
rotational rates. 2 0 0
In order to reach high performance with HRG (3)

vibrating gyro, the main challenge is to sustain the
53
According to theory of the elasticity and Kirchhoff 0 (8)

hypothesis of thin shells, the strain energy in a body
during elastic deformation is Substituting Equations (7) & Equations (8) in
Equations (2), Equation (2) can be expressed as:
U 1
2 (
v
e e ) dV (4)

For Isotropic materials, u C sin tan n cos n sin n t ,
2
E
( e e )
1 2 v C sin tan n sin n sin n t ,
(5) 2
E
( e e )
1 2 w C ( n cos ) tan n cos n sin n t
E E 2
( ) (9)
1 2 2 (1 )
Where E is Youngs modulus and is the Poisson Substituting Equations (9) in Equations (3) &
Equations (4), the expressions for Kinetic& Potential
ratio. The total strains at an arbitrary point in the shell
energy are given by:
can be represented as:
/2

e z
(n cos)
1 2
Kmax n C 2r 2 h 2
2 sin2 sin tan2n d;
(6) 2 2
e z 0
/2
2 z C h E 2 2
2 3

U max n (n 1)2 sin3 tan2n d
6(1 )r 2 0
2
Where (10)
, are normal and shear strains in the III.ANALYTIC SOLUTION
middle surface, For elements on the shell, the energies include:
virtual energy W T due to total elastic force, Virtual
, are middle surface changes in curvature;
energy We due to Coriolis force due to angular

is a middle surface twist and z measure the velocity , virtual energy T due to inertia force.
distance of the arbitrary point from the corresponding According to the virtual displacement principle,
point o the middle surface along and varies over the
thickness. For the case of a hemispherical shell, the WT W e T 0
middle surface strain and curvature changes are [7]:
ResonatorFrequency
1
(w u / ), At resonance: K.Emax=P.Emax
r
1 1
( v / ) u.c. tan w), The resonator natural frequency can be expressed
r sin as:
1 1
( u / v / v.c. tan), /2

r sin
n(n 1)h E
2 sin3 tan2n d
2

1 n 0
2 (u / 2w / 2 ), r2 3(1 ) / 2

r
0 n cos 2
2sin2
tan2n
d
1 1 2 2
2 ( w / 2 v / cos w / u cos)
r sin sin (11)
1 2w w v For n=2 vibration mode of the resonator, the
( c tan u / sin v cos)
r sin
2
natural frequency can be expressed as:
1
I
2
2h 3E
(7) 2
r2 (1 ) J
As for free vibration of clamped free hemispherical (12)
shell, under the condition of paucity displacement, the
Lord Rayleigh inextensional condition is satisfied, so Where I and J are relative to the shape of
the normal stress and shear stress will be hemispherical shell are defined as follows:
approximately reduced into zero, which is:
54
/2
If the stem radius R2decreases, each order of
I sin 3 tan 2 n d ;
2
frequencies decreases. On the other hand if
0 the stem radius increases each order of
/2

(2 cos )
frequencies increases, but the distance
J 2
2 sin 2 sin tan 2 n d between neighbours are shortened.
0
2
(13)
IV.FINITE ELEMENT ANALYSIS
Modal Analysis
ABAQUS-Finite Element Analyzer is utilized for
analysis of Hemispherical Shell as per the geometry
parameters of the resonator are shown in fig.3.
By analyzing above results, it can be concluded

that the geometry parameters of the resonator have
great influence on frequency of resonator. It is key to
choose the geometry parameters for effective
working of hemispherical resonator.
CONCLUSION
Fig.4 Illustration of shells modes shapes
The resonator achieves different vibrating shapes This paper investigates the vibrational and
on a series of frequncies. As the number of the ring precessional mode of the hemispherical shell.
waves n increases, the frequncies increases too, it will Utilizing the virtual displacement principle the
be difficult to excite the resonator. As the number of kinetic equation of hemispherical shell is analyzed.
ring waves increases, vibrating shape become The influence of shells geometry parameters on its
complex, detection of signal will be more complex. natural frequency and precession rate are analyzed as
analytic solution of model. Utilizing ABAQUS-Finite
Geometry parameters influence on frequencies Element Analyzer, the influence of shells geometry
By changing the geometry parameters, several finite parameters on each free frequency is investigated [5].
element models are studied and analyzed utilizing The analysis in this paper may be useful in the
Abaqus development of Hemispherical Shell for optimal Q-
When the radius of shell R1 gets smaller, factor and in the analysis of source of error in
each series of frequencies become higher. performance evaluation.
The shell thickness has notable influence on ACKNOWLEDGMENT
each vibration mode. As the shell grows
thicker the n=2 mode frequencies increasebut
the distance between neighbours are The authors are grateful to the Research Centre
shortened. Imarat of Defence Research and Development
55
Organization for necessary support and infrastructure

facilities for carrying out the development work.
REFERENCES
[1] G. H. Bryan, On the beatsin the vibrations of arevolving

cylinder or bell,Proceedings of the Cambridge Philosphical
Society,vol.VII,1892.
[2] G.G.Coriolis, Memoire sur les equations du movement
relative des systems de corps(On the equation of relative
motion of a systems of bodies), J.Ec.Polytech,15,142-
154,1835.
[3] D.M.Rozelle, The Hemispherical Resonator : From wine
glass to planets AAS 09-176.
[4] A.Mathews,F.J.Rybak, Comparision of Hemispherical
Resonator Gyro and Optical Gyro, IEE AES
Magazine,vol.5,pp.40-45,1992.
[5] D.D.Lynch, Vibration-Induced Drift in the Hemispherical
Resonator Gyro, Proc.Inst. of NAV, 23-25June1987,
Dayton, Ohio.
[6] E.J. Loper and D.D.Lynch, Projected System Performance
Based on Recent HRG Test Results, Proc.IEEE/AIAA 5 th
Digital Avionics System Conference,Seatle,WA,October
31,1983.
[7] Chan-Shin Chou, Modal Precession of a Hemispherical
Shell Gyro Excited by Electrostatic Field, International
Journal of Applied Electromagnetics & Mechanics,1383-
5416,October 1,1999.
[8] E.A.Izmailov,M.M.Kolesnic,A.M.Osipov, Hemispherical
Resonator Gyro Technology Problems and Possible ways of
their solutions,RTO MP-43.pp.3-6,May 1999.
56
Effect of dielectric superstrate on the

characteristics of Rectangular patch antenna
V. Saidulu1*, K.Srinivasa Rao2, P.V.D.Somasekhar Rao3
1
Department of Electronics and Communication, MGIT,Hyderabad, AP, India
2
Department of Electronics and Communication, VIF, Hyderabad, AP, India
3
Department of Electronics and Communication, JNTU,Hyderabad, AP, India
Abstract A dielectric superstrate(radome) layer I. INTRODUCTION

above a rectangular microstrip patch antenna has
remarkable effects on its characteristics such as gain, Microstrip antenna consists of radiating patch on the
bandwidth, beamwidth, radiation pattern and resonant one side of the substrate having the ground plane on
frequency. This paper experimentally investigates the
other side. The major advantages are light weight,
effect of a dielectric superstrtes with and without on the
performance characteristics of coaxial probe fed low profile, conformable to planar and non-planar
rectangular microstrip antenna with varying the surfaces and easy to fabricate. The antenna is suitable
various thicknessof dielectric superstrateas, 0.2mm, for high speed vehicles, aircrafts, space crafts and
0.5mm, 0.8mm, 1.0mm, 1.3mm, 1.5mm, 2.2mm,2.4mm missiles because of low profile and conformal nature
and 3.2mm on gain, bandwidth, beamwidth, and of characteristics [2].Microstrip antenna has inherent
radiation pattern and resonance frequency. limitation of narrow bandwidth. So,
superstrate(radome) is used on a microstrip antenna
The antenna designed frequency is 2.4GHz(ISM band) as a cover to protect the antenna from external
and fabricated on Arlondiclad 880 substrate, whose
environmental conditions like temperature, pressure
dielectric constant() is 2.2, loss tangent(tan ) is 0.0009
and the thickness (h) of the material is 1.6mm and the etc. When microstrip antenna covered with a
same substrate can be used as dielectric superstrates at dielectric superstrate(radome) its properties like
various thickness on the rectangular patch. resonance frequency, gain, bandwidth and beamwidth
Experimental measdured result observed that when the are changed wich may seriously degrading the
dielectric superstrate material is above the antenna performance[1-4]. By choosing the thickness
substrate,the antenna resonant frequency is altered and of the substrate superstrate layer, a very large gain
shifted towards the lower side of frequency of operation can be achived[5-9]. Coaxial probe fed circular
from 2.4GHz to 2.33 GHz, while othe parameters have microstrip antenna characterstics have been
slight variation in their values. The gain of single patch
investigated using HFSS softere and measured
antenna without dielectric super
experimentally.
strate is 7.3 dB and the microstrip patch with dielectric The variation of some selected antenna characteristics
superstre is from 3.3dB to 6.0dB. The bandwidth of has been studied using with and without dielectric
microstrip single patch without dielectric superstrate is superstrate. When microstrip antennas are covered
2% and the microstrip patch with dielectric superstrte
with protective dielectric superstrates, are subjected
is from 2.% to 5.% based upon the thickness of the
dielectric superstrtes. The beamwidth for single patch to icing conditions, or come into contact with plasma,
without dielectric superstrte in E-Plane is 88.360 and in the resonant frequency is altered and shifted to lower
H-Plane is 90.20. The beamwdth with dielectric sides, causing detuning which may seriously
superstre in E-Plane is 84.200 to 90.940 and in H-Plane is degrading the antenna performance. As the
67.910 to 77.630. The value of VSWR is from 1.6935 bandwidth and gain of microstrip antennas is
to 2.2373 based upon the thickness of the dielectric inherently low, typically of the order of 1-2%[1]. It is
superstrates,asdielctricsuperstrate thickness important to determine the effect of dielectric
increases, VSWR increases, return loss increases, superstrate on the resonant frequency of the
gain and bandwidth decreases. The among varioue microstrip antennas and in order introduce
thickness at 1.3mm thicknes has given higher gain appropriate corrections in the design of microstrip
6.0dB and higher BW 5%. antennas. In this paper experimentally measured the
effect of dielectric superstrates on the antenna
IndexTerm Rectangular microstrip antenna, parameters such as bandwidth, beam width, gain,
dielectric superstrate, VSWR, Gain, Beamwidth, resonant frequency radiation pattern etc.at various
Bandwidth, resonant frequency etc. thickness.
57
important role similar to that of the thickness of the

substrate. A low value of and for the substrate
and superstrate will be increase the fringing field of
the patch and thus the radiated power in increase
efficiency and better for radiation. A high loss
tangent (tan ) increases the dielectric loss and
therefore reduce the antenna performance.
IV. THEORITICAL FORMULATION:

Fig 1. Geometry of rectangular microstrip antenna
DESIGNING OF RECTANGULAR PATCH
The desiging of the rectangular microstrip patch

antenna,the transmission-line model and cavity model
is used, because easy to analysis and good physical
insight. Using the cavity model. Basically the
transmission line model represents the microstrip
antenna by two slots, separated by a low impedance
transmission line of length L.
Fig 2.Geometry of microstrip antenna with dielectric FRINGING EFFECTS:

superstrate.
The amount of fringing is a function of the
II. ANTENNA SPECIFICATIONS: dimentions of the patch and the height of the
substrate. Since for microstrip antennasL/h 1,
The geometry of a probe fed rectangular microstrip fringing is reduced; however, it must be taken into
patch antenna is shown in Fig1. The antenna under account because it influences the resonant frequency
investigation is a patch of width(W) is 49.4mm, of the antenna.
length(L) is 40.3mm fabricated on Arlondiclad 880
dielectric substrate, whose dielectric constant() is The effective dielectric constant has values in the
2.2, loss tangent(tan ) is 0.0009, thickness(h) is range of 1 <reff< for most applications where the
1.6mm and substrate dimension is 1 dielectric constant of the substrate is much greater
100mm100mm. The dielectric superstrate is than the unity ( 1), the value of reff will be closer
Arlondiclad 880 dielectric substrate, whose dielectric to the value of the actual dielectric constant of the
constant () is 2.2, loss tangent (tan ) is 0.0009, substrate.
thickness (h) is 1.6mm and substrate dimension is
100mm100mm. The antenna center frequency is
2.4GHz(ISM band) and corresponding feed location
is X=0 and Y=10.5 is shown in Fig1. The ground
plane is perfectly conducting. Antenna geometry,
material properties and boundary conditions are EFFECTIVE LENGTH, RESONANT
considered and simulated using HFSS. The variation FREQUENCY, AND EFFECTIVE WIDTH:
of superstrate thickness above the substrate studied
experimentally for various parameters such as Because of the fringing effects, electrically the patch
Bandwidth, Gain, Beamwidth, Resonant frequency, of the microstrip antenna greater than its physical
Radiation pattern etc. dimensions.
III. SELECTION OF SUBSTRATE
SUPERSTRATE:
Since the length of the patch has been extended by
Suitable dielectric substrate and superstrate of
on each side, the effective length of the patch is
appropriate thickness and loss tangent is chosen for
now
designing the rectangular microstrip patch antenna. A
thicker substrate and superstrateis mechanically
strong with improved impedance bandwidth and gain For the dominant mode, the resonant
010
[6]. However it also increases weight and surface frequency of the microstrip antenna is a function of
wave losses. The dielectric constant () is play an its length. Usually it is given by
58
The q factor is referred to as the fringing factor

(length reduction factor). As the substrate height
increases, fringing also increases and leads to larger
separations between the radiating edges and lower
resonant frequencies.
RESONANT INPUT RESISTANCE:
DESIGN:
The total admittance at slot 1 (input admittance) is
1. For an efficient radiator, a practical width that obtained by transferring the admittance of slot 2 from
leads to good radiation efficiencies is the output terminals to input terminals using the
admittance transformation equation of transmission
lines
2 = 2 + 2 = 1 - 1 (15)
Or
G 2 = G 1; B 2 = - B 1
Therefore the total resonant input admittance is real
and is given by
= 1+ 2=2 1 (16)
Since the total input admittance is real, the resonant
CONDUCTANCE: input impedance is also real, or
Each radiating slot is represented by a parallel

equivalent
admittance Y (with conductance G and susceptance
B). Theequivalent admittance of slot 1, based on an
infinitely wide,uniform slot.
1= 1+ 1
Where the plus (+) sign is used for modes with odd
Where for a slot of finite width W
(antisymmetric) resonant voltage distribution beneath
the patch and between the slots while the minus (-
)sign is used for modes with even (symmetric)
resonant voltage distribution. The mutual
conductance is defined, in terms of the far- zone
Since slot 2 is identical to slot 1, its equivalent fields, as
admittance
Y2 = Y1, G2 = G1, B2 = B1 (11). The conductance of a
single slot can also be obtained by using the field Where 1 is the electric field radiated by slot 1, 2 is
expression derived by the cavity model. In general, the magnetic field radiated by slot 2 , 0 is the
the conductance is defined as voltage across slot, and the integration is performed
59
over a sphere of large radius. It can be shown that V. MICROSTRIP ANTENNAS COVERED
12 can be calculated WITH A DIELECTRIC SUPERSTRATES:
When microstrip line (antenna) is covered with the

dielectric superstrate or radome is shown in Fig 4.
The technique can be used effectively to match the The characteristic impedance, phase velocity, losses,
patch antenna using a microstrip line feed whose and Q factor of the line change as a function of the
characteristic impedance is given dielectric superstrate layer. The properties of a
microstrip antenna covered with dielectric superstrate
layer have been studied with the transmission line
and cavity model. The resonant frequency of a
microstrip antenna covered with dielectric superstrate
layer can be determined when the effective dielectric
constant of the structure is known. The change of the
resonant frequency by placing the dielectric
superstrate has been calculated using the following
the expression.
Where,
= Effective dielectric constant with dielectric
superstrte
0 =Effective dielectric constant without dielectric
superstrate
RESONANT FREQUENCY: = Change in dielectric constant due to dielectric
superstrate
For the rectangular patch antenna is shown in Fig1, =Fractional change in resonance frequency
the resonant frequency f, can be accurately =Resonce frequency
Fig 3.Geometry of rectangular patch antenna
Where all the dimension of the microstrip patch

antenna above are defined and the choice of the width
is very much important. The smaller values of W the
result lead to higher order modes. The optimum value
of W is
given
Fig 4. Geometry of rectangular patch with dielectric
Superstrate
60
VI. DESIGN OF MICROSTRIP ANTENNA

WITH DIELECTRIC SUPERSTRATE:
The microstrip antenna can be designed using with

the substrate and with and without dielectric
superstrates. The antenna dimension such as
width(W) is 49.4mm and length(L) is 40.3mm and
feed point location is X=0 and Y=10.5. The antenna
designed frequency is 2.4GHz(ISM band) and
fabricated on Arlondiclad 880 substrate whose
substrate dimension is 100mm100mm and whose
dielectric constant() is 2.2, loss tangent(tan ) is Fig 7. Structure of dielectric superstrate with various
0.0009 and the thickness (h) of the material is 1.6mm. thickness used in the measurement
The superstrate material can be used as same as the
dielectric substrate in the design of rectangular VII. EXPERIMENTAL MEASUREMENTS:
microstrip patch antenna. We observed that when the
dielectric supeastrate is placed above the substrate the The geometrical structure under consideration is
antenna parameter will be changed. The resonance shown in Fig1. A rectangular designed patch
frequency is shifted towards the lower side of width(W)= 49.4mm and length(L)= 40.3mm was
frequency of operation from 2.4GHz to 2.33 fabricated on thick dielectric substrate whose
GHz(ISM band), while other parameters have slight dielectric constant 2.2, loss tangent is 0.0009,
variation in their values with the thickness of the thickness is 1.6mm and the dielectric superstrate is
superstrates. The prototype of antenna with feed point same substrate specification. The patch was fed
location and superstrate, substrate material is shown through probe of 50 cable. The location of feed
in Fig 5. To Fig 7. probe had been found theoretically and chosen as
x=0, y= 10.5mm. Then the patch was covered with
dielectric superstrate material such as ArlonDiclad
880 whose dielectric constant () is 2.2, loss tangent
(tan ) is 0.0009 and thickness (h) is 1.6mm. The
impedance characteristics were measured by means
of HP 8510B network analyzer. The radiation pattern
measurements were performed in the anechoic
chamber by the use of automatic antenna analyzer.
VII. EXPERIMENTAL ANALYSIS

Fig 5. Fabricated protye of rectangular patch antenna
Fig 6. Structure of dielectric substrate
Fig 8. VSWR plot at 0.2mm thickness
61
Fig 9.VSWR plot at 05mm thickness
Fig 14..Farfield radiation pattern with and without

radome
Fig 10.VSWR plot at 1.0mm thickness at 1.3mm thickness in horizontal polarization

radome at 2.4mm in horizontal polarization
Fig16. Far field radiation pattern with andwithout

radome at 3.2mm in horizontalpolarization
62
SUPERSTRATE (SINGLE PATCH)

CHARACTERISTICS
OF MICROSTRIP PATCH
TABLE 5: EXPERIMENTAL DATA FOR THE

Fig 18. Far field radiation pattern with and without EFFECT OF DIELECTRIC SUPERSTRATE ON
radome at 1.0mm thickness in horizontal polarization THE CHARACTRISTICS OF MICROSTRIP

radome at 0.2mm thickness in horizontal polarization
TABLE 6: EXPERIMENTAL DATA FOR THE

TABLE 1: SPECIFICATION OF DIELECTRIC
EFFECT OF DIELECTRIC ON THE
SUBSTRATE MATERIAL USED IN THE
CHARACTRISTICS OF MICROSTRIP PATCH
DESIGING THE RECTANGULAR PATCH
TABLE 2: SPECIFICATION OF DIELECTRIC

SUPERSTRATE MATERIAL USED IN THE
DESIGING THE RECTANGULAR PATCH
TABLE 3: CALUCLATED DIMENSIONS OF

THE RECTANGULAR MICROSTRIP PATCH
VIII. MEASURED RESULTS AND

DISCUSSION:
TABLE 4: EXPERIMENTAL DATA FOR WITH
OUT
63
In order to verify the experimental results for the thickness increases, the gain decreases. The
rectangular microstrip antenna with and without bandwidth of the microstrip antennas can also
dielectric superstrates. The antenna designed increases with increasing thickness of the dielectric
frequency 2.4GHz fabricated on Arlondiclad 880 superstrate for low dielectric constant materials, and
substrate and the substrate dimension is 100 decreases for high dielectric constant of the substrate
100 , whose dielectric constant(= 2.2), loss materials. Initially the return loss increases with
tangent(tan )is 0.0009, increasing thickness of dielectric superstrates and
thickness(h) is 1.6mm. The dielectric superstrate is then decreases. The antenna beamwidth in E-Plane
same as the substrate whose dielectric constant is 2.2, increases from 84.20 degree to 9.94 degree and
loss tangent is 0.0009; thickness is 1.6mm.We beamwidth in H-Plane is increases from 67.91 degree
observed the resonating frequency is shifting to lower to 77.63 degree.The amongvariouethickness at
side 2.40 GHz to 2.33GHz, while other parameters 1.3mm thicknes will give higher gain 6.0dBand
have slight variation in their values. The gain of higher BW 5%.
single patch antenna without dielectric superstrate is
7.3 dB and the microstrip patch with dielectric XI. REFERENCES:
superstre is 3.3dB to 6.0dB. The bandwidth of
microstrip single patch without dielectric superstrate 1. IE3D Manual, Zeland software Inc.,Fremount, USA,1999
2. I J Bhal and P Bhartia, Microstrip antennas, Artech house,
is 2% and the microstrip patch with dielectric 1980.
superstrte is 2. % to 5.0% based upon the thickness of 3. R.Shavit,Dielectric cover effect on Rectangular Microstrip
the dielectric superstrtes. The beamwidth for single Antennas array. IEEE Trans. Antennas propagat.,Vol 40,. PP.992-
patch without dielectricsuperstrte in E-Plane is 88.360 995,Avg.1992.
4. Inder ,Prakash and Stuchly, Design of Microstrip Antennas
and in H-Plane is 90.20 . The beamwdth with covered with a Dielectric Layer. IEEE Trans. Antennas Propagate.
dielectric superstre in E-Plane is 84.200 to 9.940 and Vol.AP-30.No.2,Mar 1992.
in H-Plane is 67.910 to 77.630. The value of VSWR is 5. O.M.Ramahi and Y.T.LO, Superstrate effect on the Resonant
from 1.6935 to 2.2373 based upon the thickness of frequency of Microstrip Antennas, Microwave Opt.Technol. Lett.
Vol.5, PP.254257,June 1992.
the dielectric superstrates. The variation of VSWR 6. A.Bhattacharyya and T. Tralman, Effects of Dielectric
with different dielectric superstrate thickness, as Superstrate on patch Antennas, Electron Lett., Vol.24,PP.356-
dielectric superstrate thickness increases, VSWR 358, Mar 1998.
increases. The variation of gain at different dielectric
superstrate thickness, as dielectric superstrate
thickness increases, gain decreses. The beamwidth in
E-Plane decreases from 90.94 degree to 90.50 degree
and in H-Plane increases from 70.71 degree to 71.80
degree. The bandwidth of microstrip antenna is
increases with increasing thickness of dielectric
superstrates for low dielectric constant and decreases
for high dielectric constant.The experimental
measured results is shown in table 4, 5 and 6 and
corresponding measured VSWR and radiation pattern
plot for various thickness is shown in Fig 8 to Fig 19.
X. CONCLUSION:
The design of rectangular microstrip antennas with

and without dielectric superstrates has been
presented. The resonant frequency of a microstrip
covered with dielectric superstrates can be predicted
accurately if the effective dielectric constant of the
structure is known. The effective dielectric constant
can be calculated using transmission line and cavity
model. The experimentally results shows that the
variation of VSWR with different dielectric
superstrate (radome) thickness, as dielectric
superstrate thickness increases, VSWR increases. The
variation of the antenna gain at different dielectric
superstrate thickness as dielectric superstrate
64
Design of Printed Dipole antenna Integrated

with Resonator and Defected Ground
Structure (DGS) for
improving noise rejection
Takeshore Khumanthem*, C.Sairam, S.D.Ahirwar and M.Santha
Defence Electronics Research Laboratory,
Hyderabad-500005, India
E-mail: takeshore@gmail.com
Abstract:- In this paper design of printed dipole outside its band. Lower spectra to ISM band is the
antenna integrated with ring resonator and DGS is GSM-1800 and above it is the UWB band. To further
presented along with measured results. The ring improve the fidelity of data transfer rate, an antenna
resonator bandpass filter is coupled with DGS to can be designed to rejects the above unwanted bands.
provide wide band rejection characteristics. This
Recently, a numbers of research papers and literature
antenna shows high signal rejection outside the required
frequency band of 2.4GHz to 2.5 GHz (~ ISM band), on antenna with band rejection characteristics are
thereby improving the noise rejection in the unwanted published. To achieve band rejection, they either used
band. The measured VSWR of the antenna is less than different shaped slots or filters [1][2][3].In this paper
1.5:1 over the band. The antenna shows good radiation a printed dipole antenna in ISM band (2.4-2.5 GHz)
efficiency and omni-directionality. The antenna is is designed. This antenna is coupled with ring
simulated using CST microwave studio and then resonator and DGS to achieve wide band signal
practically realized on RT duriod substrate. Thus, the rejection outside the desired frequency band without
proposed antenna finds applications in trans-receive degrading its overall radiation efficiency and omni-
modes in todays wireless arena where high signal
directionality.
selectivity is in demand.
Keywords: Printed dipole, ISM band, ring resonator, II. DESIGN APPROACH
DGS, noise rejection.
A printed dipole antenna with center frequency
I. INTRODUCTION 2.45 GHz was designed. The length of the dipole is
taken less than /2 in order to cancel the inductive
Printed antennas are emerging as preferred choice reactance. The antenna is integrated with printed
among the different antenna designs for various balun as shown in figure 1[4]. The overall dimension
applications. Owing to their low profile, low cost, of the antenna is length x width = 108mm x 66mm
ease of fabrication and compatibility with the respectively. The balun and ring resonator are printed
microwave integrated circuit technology, these on front side of the substrate and the DGS and printed
antennas are in high demand from the system dipole are printed on back side. The substrate is RT
designers point of view. They can be implementing duriod with r = 2.2 and thickness = 30 mil. The ring
on the same PCB as the circuitry with little or no bandpass filter with wide band rejection
additional cost. For a wireless communication system characteristics as proposed by Chul Soo Kim and et.al
which requires an omni-directional radiation patterns, [5] is integrated with the printed antenna as shown in
as in a broadcasting system or a communication figure 1.
system, a printed monopole or dipole is a good
choice.
The conventional printed dipole antennas are
narrow band in nature with impedance bandwidth of
hardly 5-10% of its centre frequency. However, they
show wide radiation pattern and gain bandwidth.
Sometimes, it is required to have narrow gain
bandwidth antenna in order to minimize the
interferences from the unwanted signals. The antenna
for high data transfer rate in ISM band will be
required to have high band rejection characteristics
65
Most of the conventional BPF's are designed with B. Ring resonator with DGS
parallel coupled lines [6][7]. However, these filters
have a limitation in realization due to, Figure 4 shows the ring resonator coupled with
i) Large size because of /2 resonator. DGS as proposed by Chul Soo Kim and et.al [5]. The
ii) Very narrow gap between the coupled lines is same is integrated with the proposed printed dipole
required to antenna to achieved wide band signal rejection
get strong coupling. outside ISM band. From the figure 5 & 6 it is seen
On the other hand, the ring resonator can give that both the simulated and measured VSWR is less
compact size, narrow bandwidth and low radiation than 1.3:1 over the band. Figure 7 shows the insertion
loss [8]. But ring resonator has a passband at every loss of the filter and is less than 1 dB over the band.
harmonic frequency. These pass bands can be These results show high band rejection outside the
suppressed by using DGS along with resonator. By required band. Overall, there is good agreement
properly designing the DGS, the coupling value between the simulated and measured results.
controlled and good impedance matching over the
desired band is achieved.
III. RESULTS AND DISCUSSION
The proposed antenna is designed using EM

simulation software, CST microwave studio. Both the
simulated and measured results are compared and
presented in this section.
A. Printed dipole antenna
A printed dipole antenna with centre frequency

2.45 GHz was designed as shown in figure 2. Figure
3 shows the simulated and measured VSWR plot of
printed dipole antenna. From the figure it is seen that
both the simulated and measured VSWR is less than
2:1 over the band and they are in good agreement.
66
Figure 10 shows the VSWR plot of printed dipole

antenna with and without band rejection filter. From
the figure it is seen that using filter, high band
rejection outside the required band was achieved.
C. Printed dipole with Ring resonator and DGS
Figure 8 shows the Printed dipole antenna

integrated with ring resonator and DGS. They are
printed on a single substrate. Here, the ring resonator
along with DGS serves as narrow bandpass filter with
low insertion loss. The proposed integrated antenna is
excited at P as shown in figure.
D. Current distribution
For understanding the behavior of the antenna, the

simulated current distribution of the proposed
antenna at different frequencies is shown in figure 11.
Both the simulated and measured VSWR are

compared and plotted as shown in figure 9. From the
figure it is seen that the final measured VSWR of the
integrated antenna is less than 1.5:1 over the band
2.4-2.5 GHz. While high reflection coefficient is
obtain outside the required band which is one of the
desired characteristics of the proposed antenna.
67
Figure 12(a): Measured Elevation plane patterns of Printed dipole

antenna with ring resonator and DGS
Figure 12(b) shows azimuth plane patterns at 2

GHz, 2.45 GHz and 3 GHz. From the figure it is seen
that the antenna shows good omni directionality in
the desired frequency band.
Figure 11: Simulated current distribution of the proposed antenna

at a) 2 GHz b) 2.45 GHz and c) 3 GHz
From the above figures it is seen that at 2 GHz and

3 GHz which are outside the ISM band, the current
distribution over the surface is weak and only a
fraction of current could reach the antenna. This
results in high attenuation in out of the pass band.
However, at 2.45 GHz which is a desired frequency
band, high density of current can be seen flowing up
to the radiating elements. This will result in good
radiation efficiency and gain of the proposed antenna.
E. Radiation patterns measurement
The antenna far field radiation patterns are Figure 12(b): Measured Azimuth plane patterns of Printed dipole
measured inside a rectangular Anechoic chamber antenna with ring resonator and DGS
using PNA based antenna measurement system.
Figure 12(a) shows the measured elevation patterns
of the proposed antenna at 2GHz, 2.45 GHz and 3
GHz. From the figure it is seen that there is
reasonable gain variation within desired and
undesired frequency bands. This envisages the noise
rejection capability of the proposed antenna
Figure 13: Computed gain of Printed dipole antenna with and

without filter
The gain comparison of the proposed printed

antenna with and without wide band rejection filter is
shown in figure 13. From the figure it is seen that the
printed antenna with filter shows high signal rejection
of more than 17dB at 2 GHz and 15 dB at 3GHz
respectively. The gain variation of proposed antenna
68
with and without filter in the passband is expected

from insertion loss of the filter as well as the
polarization loss that arises due to radiation from the
filter section.
IV. CONCLUSIONS
The design of printed dipole antenna integrated

with wide band rejection filter is discussed along with
simulated and measured results. Using ring resonator
with DGS, wide band rejection was accomplished.
This antenna shows good VSWR, gain and radiation
patterns over the frequency band 2.4-2.5 GHz. Thus,
the proposed printed dipole antenna is envisaged to
have wide range of applications in wireless
communication systems which require minimum
interferences from the existing wireless technologies
such as IEEE 802.11, WiMax, UWB etc.
ACKNOWLEDGEMENT
Authors would like to thank S.P.Dash, Director

DLRL and M. Balachary, Sc.G, Additional
Director, DLRL for their constant encouragement,
motivation and support during the above work.
REFERENCES
[1] Movahedinia R and Azarmanesh M.N, " Ultra wideband band

notch printed monopole antenna," IET Microwaves, Antennas &
Propagation, Vol. No. 4, Issue No. 12, pp. 2179-2186, 2010.
[2] Yangwoon Roh, Kyungho Chung and Jaehoon Choi, "Design
of a microstrip-fed ultra wideband monopole antenna having band
rejection characteristic," IEEE's Antennas and Propagation Society
International Symposium 2005, Vol. No. 2B, pp.556-559.
[3] Pintos, J.F Chambelin P, Louzir A and Rialet D, "Low cost
UWB printed dipole antenna with filtering feature, "IEEE's
Antennas and Propagation Society International Symposium 2008,
pp.1-4.
[4] Michishita, N, Arai, H, Nakano, M, Satoh, T and Matsuoka
T,"FDTD analysisfor printed dipole antenna with balun,"
Microwave Conference, 2000, pp.739 - 742.
[5] Chul Soo Kim et.al, " A design of a ring bandpass filters with
wide rejection band using DGS and spur-line coupling structures,"
IEEE Microwave Symposium Digest, MTT-S 2005.
[6] Moradian M and Oraizi H, "Optimum design of microstrip
parallel coupled line band pass filters for multi-spurious pass band
suppression," IET Microwaves, Antennas & Propagation, Vol.
No.1, Issue No. 2, pp.488-495, 2007.
[7] Kim B.S, Lee J.W and Song M.S, " An implementation of
harmonic-suppression microstrip filters with periodic grooves,"
IEEE's Microwave and Wireless components Letters, Vol. No.14,
Issue No. 9, pp.413-415.
[8] de Dios Ruiz J and Hinojosa J," Double-sided open split ring
resonator for compact microstrip band pass filter design," IET
Microwaves, Antennas & Propagation, Vol. No.6, Issue No. 8,
pp.846-853, 2011.
69
Design of Dual Frequency Circular Patch

Antenna Using HFSS
1 2.
.Akhila John. Davuluri, Dr.P.Siddaiah.
1
Assistant Professor, Department of Electronics and Communication Engineering, University College of
Engineering and Technology, Acharya Nagarjuna University, Guntur, AndhraPradesh.
2
Dean & Professor, Department of Electronics and Communication Engineering, University College of
Engineering and Technology, Acharya Nagarjuna University, Guntur, AndhraPradesh.
Email:akhiladavuluri @gmail.com
Abstract This paper presents a circular patch The circular microstrip antenna is a metal disk of
microstrip antenna operate in dual band (1.28 GHz
radius a and has a driving point location at r which
and 2.67GHz). The proposed Circular patch antenna
will be in light weight, flexible, slim and compact unit makes an angle with the x axis. The thickness of the
compare with current antenna used in dual band. The substrate is h, where h<<o , Which has a relative
paper also presents the detail steps of designing the dielectric constant of r [2].The resonant frequency fnm,
circular patch antenna and the simulated results. for each TM mode of a circular microstrip antenna is
HFSS software is used to compute the return loss,3D given by
polarplot,3D rectangular plot,3D gain (LHCP), ,3D
gain (RHCP) and radiation pattern of the antenna.
Keywords- Circular patch, microstrip antenna,

Return loss, Bandwidth. (1.3)
I. where Anm is the mth zero of the derivative of the
INTRODUCTION Bessel function of order n. The constant c is the
Microstrip antennas are attractive due to their light speed of light in free space and aeff is the
effective radius of the patch. A list of the first four
weight, conformability and low cost. These antennas
Bessel function zeros used with the above
can be integrated with printed strip-line feed resonant frequency equation are presented in
networks and active devices [2]. This is a relatively Table 1-1.
new area of antenna engineering. An analysis of the
circular
Anm TMnm
microstrip antenna, which is very useful for
engineering purposes. The electric field under the 1.84118 1,1
circular microstrip antenna is described by: 3.05424 2,1
3.83171 0,2
4.20119 3,1
(1.1)
Table 1-1. First four Bessel function zeros used with equation.
The magnetic field components are described as
The modes for a circular microstrip antenna were

introduced as TMnm, where n is related to and m
is related to r. aeff is the effective radius of the
circular patch, which is given by
(1.2)
(1.4)
(1.5)
Figure 1.1 Circular microstrip antenna geometry where a is the physical radius of the antenna.
70
Equation (4) and equation (5) can be combined to

produce: 3. RESULTS
The return loss, 3D polar plot, 3D rectangular

plot,3D gain (LHCP), 3D gain (RHCP) and
radiation pattern are obtained using HFSS 14.0.At
(1.6) the frequency of 1.28 GHz the return loss is-
The form of equation (1.6) is 13.6316 dB and at 2.67 GHz the return loss is -
22.26dB.
A. Return loss
The initial approximation for the radius a0 to begin the
iteration is
(1.7)
2. DESIGN AND ANALYSIS OF ANTENNA
In this paper the microstrip antenna have been

modeled, designed and simulated. The analysis of dual
Figure 3.1. Plot between return loss and frequency
band microstrip patch antenna for 1.28 GHz and
2.67GHz frequency and the designing has been done
using HFSS software.
B. 3D Polar plot
Figure 3.2. 3D Polar plot

Figure 2.1. Microstrip Circular patch antenna
C. 3D Rectangular Plot
In the typical design procedure of the Microstrip
antenna, the desired resonant frequency, thickness
and dielectric constant of the substrate are selected
initially. In this design of Circular Microstrip Patch
antenna, FR4 dielectric material ( r=4.4) with
dielectric loss tangent of 0.02 is selected as the
substrate with 1.6 mm height. Then, a patch antenna
that operates at the specified operating frequency. f0 =
1.28GHz and 2.67 GHz can be designed by using the
above said equations. TM11 mode is selected to design
the radius of circular patch. The units of h must be
in mm and fr in GHz. Figure 3.3. 3D Rectangular Plot
71
D.3D Polar plot-Gain (LHCP) Figure 3.5. Radiation Pattern
. 4.CONCLUSION
After the whole analysis the characteristics of

proposed antenna enhanced at many parameters.
Obtained dual band at 1.28GHz and 2.67 GHz
frequencies, this antenna is perfect for
applications such as radar communication,
military communication. Band width is also
acceptable for both bands.
ACKNOWLEDGMENTS
Extending our grateful thanks to the authorities of

Acharya Nagarjuna University for their support
Figure 3.4. 3D Polar Plot(Left Handed circular Polarization) and encouragement to write this paper.
D .3D Polar plot-Gain (RHCP) REFERENCES
[1] T.F.Lai, Wan Nor Liza Mahadi, Norhayati Soin,

Circular Patch Microstrip Array Antenna for KU-band
World Academy of Science, Engineering and Technology 48
2008.
[2] C.A. Balanis, Antenna Theory, 2nd Ed., John wily &
York.1982.
[3] Veerendra Singh Jadaun, Pavan Kumar Sharma, Ashish
Duvey, Design A Microstrip Patch Antenna of Single Band
For 1.8GHz, international conference e-manthan 2012, 6-7
april 2012, Jhansi.
[4] C.A. Balanis, Advanced Engineering Electromagnetic,
John Wiley & sons, New york, 1989.
[5] Pavan Kumar Sharma, Hemant Kumar Gupta, Veerendra
Singh Jadaun, Ritesh Kumar Tambulkar, Design of Sierpinski
Microstrip Patch Antenna for Wireless Power Transmission
for ISM Band (2.45GHz), international conference e-manthan
2012, 6-7 April 2012, Jhansi.
[6] Arun Singh Kirar, Veerendra Singh Jadaun, Pavan Kumar
Figure 3.5. 3D Polar Plot(Right Handed circular Polarization)
Sharma Design a Circular Microstrip Patch Antenna for Dual
Band, International Journal of Electronics Communication
and Computer Technology (IJECCT) Volume 3 Issue 2
F. Radiation Pattern (March 2013).
[7] B.J.Kwaha, O.N Inyang & P. Amalu, THE CIRCULAR
MICROSTRIP PATCH ANTENNA DESIGN AND
IMPLEMENTATION. IJRRAS 8 (1) July 2011.
[8] Modern Antenna Design, Second Edition, By Thomas A.
MilliganCopyright 2005 John Wiley & Sons, Inc.
[9] D.G. Fang ,Antenna Theory and Microstrip Antennas .
[10] John R. Ojha and Marc Peters Germany "Patch Antennas and
Microstrip Lines". ISBN 978-953-7619-67-1
72
About Authors:
Dr. P Siddaiah obtained
B.Tech degree in Electronics
and communication Engineering
from JNTUA college of
engineering in 1988. He
received his M.Tech degree
from SV University Tirupathi.
He did his Ph.D program in
JNTU Hyderabad. He is the Chief Investigator for
several outstanding Projects sponsored by Defense
Organizations, AICTE, UGC & ISRO. He is currently
working as Professor & DEAN, Department of ECE in
University College of Engineering and Technology,
Acharya Nagarjuna University, Guntur, India. He has
taught a wide variety of courses for UG & PG students
and guided several projects. Several members
successfully completed their Ph.D under his guidance.
Several members pursuing their Ph.D degree. He has
published several papers in National & International
Journals & Conferences. He is the life member of
FIETE, IE & MISTE.
Ms.Akhila John Davuluri obtained B.Tech degree in

Electronics and Communication Engineering from , JNT
University 2005 and
M. Tech Degree in Automotive Electronics from VIT
University, Vellore 2008.She is currently working as
Assistant Professor, Dept. of ECE in University College
of Engineering and Technology, Acharya Nagarjuna
University, Guntur, India .Area of interest are
Communications and Signal Processing.
73
Open Loop Tracking Technique for Weak

Signals in GPS Radio Occultation Receiver
Namit Rarotra, Hemanth Kumar Reddy N, Chitra Ramamurthy, Yateendra Mehta
Communication Systems Group, ISRO Satellite Centre, Bangalore namit@isac.gov.in
Abstract: user anytime, anywhere, in any weather. GPS

The paper discusses the open loop tracking technique consists of three parts: the space segment, the control
used to track very weak signals during the Radio segment, and the user segment. The Space segment
occultation events. A low earth orbiting satellite equipped consists of a constellation of more than 24 satellites
with a GPS receiver can receive the signals from GPS orbiting at an altitude of approximately 20,200
satellites below the horizon. The signals coming from
km. There are six orbits with a minimum of 4
below horizon GPS satellites travel through earths
atmosphere and have very low signal strength and large satellites in each orbit. These orbits are inclined at
and random variation in signal phase. Tracking loops an angle of 55 degrees and separated by 60 degrees.
employing carrier PLL or Costas loop will drop more All satellites continuously transmit GPS signals at
occasionally leading to poor performance and improper L1 (1575MHz), L2 (1227MHz) and L5 (1176MHz)
outputs or no output during the radio occultation events. frequencies.
To overcome this problem, an open loop tracking
technique is employed which is able to track the GPS
signals well below the horizon.
INTRODUCTION:
GPS Radio Occultation (RO) technique tracks the

signals of GPS satellites close to earths horizon.
These signals travels through earths atmosphere and
because of atmospheric delay, these signals degrade
the positioning capability of the GPS receiver and
hence are not used in solution computation. A
typical Global Positioning System (GPS) receiver
employs carrier PLL or Costas loop to track the GPS Fig.1: GPS Constellation
signals. These loops provide phase and amplitude of a
signal as an output. Random phase accelerations of The Control Segment (CS) provides command,
the RO signal can be significantly larger than control, and maintenance services to the space
allowed for stable operation of the PLL in a segment and ensures that the satellites are working
generic GPS receiver. This technique is optimal properly.
for single tone signals and provides unstable results
The user segment consists of receivers that receives
for multi-tone RO signals. It may also results in loss
of lower tropospheric data especially during rising and processes the GPS signals and calculates the
occultation. An alternative to this is Open loop position, velocity and time of user. It works on the
(OL) tracking technique which provides the phase principle of triangulation and a minimum of four
and raw sampling of complex RO signals and satellites are required to precisely calculate a 3D
allows the tracking of both rising and setting solution that is (x,y,z,t).
occultations. It calls for the implementation of
predicted Doppler model, low pass filtering and GPS BASED SYSTEM IN LEO:
calculation of excess phase. The main difference
between the OL tracking and PLL tracking is that
LEO satellites 'orbit around the earth at an altitude
there is no feedback path between the received signal
of around 500-3000Km. Below 3000Km altitude,
and the extracted phase.
signals from 10 or more satellites are typically
received by an upward looking antenna from GPS
GLOBAL POSITION SYSTEM
constellation with nearly uniform power distribution.
The receiver onboard receives GPS signals and
The Global Positioning System is a navigation processes it to give the position and velocity
system that provides the exact position and time of vectors of the LEO satellite.
74
light (c = 299,792,458m/s). The receiver estimates

the pseudorange. The GPS satellite position xj , yj
and zj can be calculated by using the demodulated
navigation data ie the ephemeris data received from
GPS satellites. Four equations are needed to solve
for four variables x u , yu , zu and tu. Thus
a minimum of four visible GPS satellites are
required to solve for LEO spacecraft position
vectors.
RADIO OCCULTATION
Fig.2: GPS System in LEO

An occultation is an event which occurs when one
This receiver is high dynamics GPS receiver that celestial body moves into a position which
has ability to track the signals with large doppler obscures another body from the view of an
shifts. The maximum Doppler shift for a receiver in observer. During a GPS RO, a GPS satellite is
LEO satellite can be computed as occulted by the Earth as the satellite (transmitter) is
setting or rising on the local horizon relative to an
observer (receiver).
G is gravitational constant and M is mass of Fig.3: GPS Radio Occultation

earth. c is velocity of light, rleo is distance of LEO In the occultation model depicted in Figure 3, the
satellite to earth centre and rgps is distance of gps relative motion between the receiver in low earth
satellite to earth centre. Typically for lower orbit (LEO) and transmitter cause the
altitude satellites, the maximum Doppler shift electromagnetic ray to propagate through
comes close to 45KHz. So the GPS receiver for successively lower or higher, vertical layers of the
LEO satellite is designed to track signals upto atmosphere. The atmosphere is modelled as a
50KHz. spherically symmetric medium with each vertical
layer having a unique density, creating a vertical
GPS MEASUREMENTS density gradient. Assuming the signal propagates as a
single ray, the signal follows Snells law of
GPS receiver is used to estimate the position and refractivity when passing through a density
velocity vectors of the spacecraft. The gradient. The propagation through the atmosphere
will introduce a further phase delay compared to
measurements are in the form of pseudorange the one seen in free space propagation. Also the
which is the apparent distance between the tracked signal will not propagate in a straight line but rather
GPS satellite and the user LEO spacecraft. The user be bend. As the density decreases the signal delay
distance from the tracked GPS satellite is and bending will decrease accordingly. If the
proportional to the change in carrier frequency position of the GPS and the LEO satellite is known
relative to the user. This in turn is a function of the phase delay measured by the LEO satellite
contains information about the atmosphere. The
satellite position, velocity, precise carrier frequency
signal transmitted by GPS has a frequency of L1 =
and time which can be obtained from the satellite 1575.42MHz and L2 = 1227 MHz. Both
ephemeris. The pseudo-range (j) can be obtained frequencies delay differently as the pass through
by the atmosphere. The principle of radio occultation
measurement is to measure the change in the delay
measured by the LEO as the LEO sets behind the
limb of the Earth relative to the GPS satellite. The
where (xj , yj , zj ) are the coordinates of jth amplitude of the received signal in principle
GPS satellite, (xu , yu , zu ) are the coordinates contains the same information as the phase but is less
of user LEO spacecraft, tu is an offset of receiver precise. The radio occultation measurement can be
clock from the system time, and c is a speed of
75
inverted to give information on the density, pressure and one of the filters to make such a second order
and temperature of the atmosphere. A typical data PLL is
processing chain for Radio Occultation is shown in
fig. 4.
Fig.4: Radio Occultation data processing chain

In contrast, OL tracking does not rely on any
TRACKING feedback process. The reference signal is guessed
on the basis of knowledge of the orbits, receiver
In conventional tracking or a closed-loop (CL) clock drift, and an estimate of the atmospheric
tracking, the receiver uses a phase locked loop (PLL) Doppler shift and delay. The code tracking loop
and delay locked loop (DLL) to adjust the frequency accomplished using DLL is shown in fig
and code of the reference signal to match the
frequency and code of reference signal respectively
on the basis of previous measurements. This process
is very reliable when there is sufficient signal-to-
noise ratio (SNR) and the signal dynamics are not too
high. A basic PLL is shown in fig. 5.
Fig.6: DLL used for code tracking

The IF is first down-converted by the carrier
frequency to the baseband in I and Q components.
Fig.5: Basic PLL used in tracking The I and Q are multiplied by early, late and
The phase comparator measures the phase prompt PRN codes to get the correlation values.
difference between the input signal i(t) and the The values are summed and averaged for 20ms. It
output of voltage controlled oscillator (VCO) f(t). is then fed to early/Late discriminator which
The output (t) is given to a low pass filter which determines whether there should be a phase change
filters out the high frequency components and in locally generated prompt PRN code.
limits the noise in the loop. The input voltage Vo to In Open Loop (OL) tracking, there is no feedback
VCO controls its output frequency which can be path between the output and the input IF signal.
expressed as The reference signal is guessed on the basis of
knowledge of the orbits, receiver clock drift, and an
Where 1 is the centre angular frequency of VCO, K estimate of the atmospheric Doppler shift and delay.
is the gain of VCO and u (t) is the unit step function. The observables in case of OL tracking are path
The transfer function H(s) of the loop is defined as phase and the excess phase at both the frequencies
i.e. L1 (1575.42 MHz) and L2 (1227.6MHz). Fig.
shows how the open loop tracking is implemented.
k 1 is the gain of phase comparator and F(s) is a
filter function. Generally the phase locked loop In OL tracking, a precise Doppler model is
used in GPS receivers is a second order function of s implemented which estimates the Doppler between
76
the LEO satellite and the occulting GPS satellite. It in Doppler calculated from the model as compared to
estimates the Doppler using the GPS ephemeris and the Doppler seen by the spacecraft. Data to track
the LEO spacecrafts position and velocity vectors. using open loop tracking is generatedin matlab and
noise has been added to this data.Open loop tracking
The Doppler equation can be written as
simulations have been carriedout on this data and the
phase of output signal hasbeen calculated. This is
shown in fig. 8.
Fig 7: Open Loop tracking implementation

Fig. 8: Output phase with noise in data
Where ftx is the transmitted frequency, Vleo and Vgps
The results are for data with 15 Hz offset in carrier
are the LEO and GPS satellites velocities. These
i.e. 15 Hz Doppler mismatch and having a noise of -
velocity vectors of LEO spacecraft can be obtained
10dB in (a), -14dB in (b), -15dB in (c), and -20dB in
from POD (generic GPS) receiver. GPS ephemeris or
(d).
almanac is used to calculate the position and velocity
vectors of GPS receiver. The Doppler model
CONCLUSION
incorporates the Doppler shift due to clock oscillator
dynamics, LEO satellite dynamics and the GPS Open loop tracking technique helps in tracking weak
satellite dynamics. The Doppler shift observed by a signals that pass through the earths atmosphere.
LEO satellite, when GPS signal is transmitted at L1 These signals are having large random phase
frequency is shown in fig. 8. variations and very low SNR that leads to the loss of
lock of traditional PLL based tracking. Fairly
accurate Doppler model implementation helps in
removing the feedback loop.
REFERENCES
[1] David Hobbs, Preben Bohn PRECISE ORBIT

DETERMINATION FOR LOW EARTH ORBIT SATELLITES,
Fig.8: Doppler shift observed by LEO at L1 Journal of Theoretical and Applied Information Technology, vol.5,
With the presence of the atmosphere, an extra phase no. 15, pp 95-101.
delay is introduced to the measurement due to the [2] FUNDAMENTALS OF GLOBAL POSITIONING SYSTEM
RECEIVERS: A SOFTWARE APPROACH, by James Bao and
refraction and bending. The difference between the Yen Tsui, John Wiley & Sons, INC.
measured phase delay and the vacuum phase delay [3] Alex Aaen Birklykke, HIGH DYNAMIC GPS SIGNAL
is the excess phase delay. The GPS RO bending angle ACQUISITION - A CASE STUDY IN GPS RECIEVERS ON
can be derived from this excess phase delay. As the NANOSATELLITES IN LEO, AALBORG UNIVERSITY,
Spring 2010.
code tracking is maintained, so code tracking loop [4] Sergey V. Sokolovskiy, TRACKING TROPOSPHERIC
can be retained same as in CL tracking shown in fig RADIO OCCULTATION SIGNALS FROM LOW EARTH
6. The carrier tracking loop is implemented without ORBIT Radio Science, Volume 36, Number 3, pages 438-498,
feedback loop and the Doppler model is used to May/June 2001
[5] Hui Shao, Xiaolei Zou and George A. Hajj, TEST OF A
control the carrier NCO in OL tracking. This is NON-LOCAL EXCESS PHASE DELAY OPERATOR FOR GPS
shown in fig. 7. Simulations have been performed. RADIO OCCULTATION DATA ASSIMILATION Journal of
Doppler model has been checked with the GPS Applied Remote Sensing, Vol. 3, 033508 (12 February 2009)
simulator simulating the spacecraft scenarios. [6] C. O. Ao, G. A. Hajj, T. K. Meehan, D. Dong, B. A. Iijima, A.
J. Mannucci, and E. R. Kursinski, RISING AND SETTING GPS
Calculations of Doppler were performed on below OCCULTATIONS BY USE OF OPENLOOP TRACKING
horizon GPS satellites. There was very less deviation JOURNAL OF GEOPHYSICAL RESEARCH, VOL. 114, 2009
77
Acquisition of Modernized GPS L2C signal

1 2 3 4
Kishore Babu N , Hemanth Kumar Reddy , Chitra Ramamurthy , Yateendra Mehta
Communication Systems Group, ISRO Satellite Centre
Old Airport road, Bangalore-560017, India
1 kishoren@isac.gov.in, 2 hemanthr@isac.gov.in, 3 chitra@isac.gov.in, 4 ymehta@isac.gov.in
Abstract-
The modernization of the GPS led to availability of tracking loops for tracking code and carrier
additional GPS signals with more features. L2C, the signals and further demodulation of Navigation
second civilian GPS signal is going to become data of GPS satellites.
available in full constellation. L2C signals have longer Different methods for acquiring CM code like parallel
ranging codes which leads to larger search space and code search acquisition, parallel code search
increase in computational complexity. L2C signals make acquisition with reduced points and parallel
use of two PRN sequences CM (Civil Moderate) and CL
frequency search acquisition methods are used and
(Civil Long) of 20ms and 1.5sec duration. In this paper
compared.
two local replica codes, NRZ and RZ CODES are presented.
The code phase and carrier Doppler are estimated based on
the parallel code space search and parallel frequency II. L2C SIGNAL STRUCTURE
search space acquisition.
The L2C signal (1227 MHz) is composed of two
Keywords- GPS acquisition, Time domain ranging codes, namely L2 CM (Civil Moderate) and
correlation, FFT, IFFT, down sampling. L2 CL (Long).The L2 CM code is 20 ms long and
contains 10230 chips, while the L2 CL code has a
I. INTRODUCTION period of 1.5 s, containing 767250 chips. The CM code
is modulo-2 added to data, and the resultant sequence
The current civil GPS signal consists of a single of chips is time multiplexed with the CL code on
frequency transmission L1 (C/A). The Global a chip-by- chip basis. The individual CM and CL
Positioning System (GPS) modernization program codes are clocked at 511.5 kHz, while the
deploys new Block IIR-M satellites, equipped to composite L2C code frequency is 1.023 MHz. Code
initiate the transmission of the new civilian signal
boundaries of CM and CL are aligned, and each
L2 civil (L2C). One of the most commonly
CL period contains exactly 75 CM periods.
referenced limitations with GPS is the vulnerability
of the L1 signal to Radio Frequency Interference
(RFI) either intentional or unintentional. With its
novel code structure and compact data format, the L2C
signal can offer advantages like indoor positioning,
ionosphere error elimination, and improved
tracking performance.
The purpose of acquisition is to identify all satellites

visible to the user. The two important parameters
estimated are the Carrier Doppler and the Code Phase. Fig.1. L2C Signal structure
As the carrier and code signals are affected by the
relative motion of the satellite with respect to the user,
causing a Doppler effect. The Doppler frequency shift
depends upon the velocity of the GPS satellites and
user velocity. This can be 5 kHz for a stationary
receiver on Earth and as high as 50 KHz for a receiver
on Low Earth Orbiting (LEO) Satellites.
The code phase denotes the point in the current data

block where the L2C code begins. If a data block Fig.2. L2C code generator
of 20ms is examined, the data includes an entire
L2C code and thus one beginning of a L2C The CM and CL codes are generated using same
code. These measurement values are used by block clocked at 511.5Kbps. Each pattern is initiated
78
and reset with a specified initial states as described is multiplied with the Fourier transform of the PRN
for specific satellites. The CM codes are reset after code. The result of the multiplication is transformed
10230 chips, 20 ms long and CL code is 1.5s long into the time domain by an inverse Fourier transform.
and reset after 767250 chips. The absolute value of the output of the inverse Fourier
transform represents the correlation between the input
and the PRN code.
III. LOCAL L2C CODES
For efficient signal acquisition, two local replica codes

have been designed named Non Return Zero (NRZ
CM) and Return Zero (RZ CM). NRZ CM replica
code is designed based on chip wise (CW)
correlation strategy. Here the received L2C code
samples are accumulated across each chip period, and
the correlation is then performed at the L2C code Fig.3. Parallel code space search acquisition
chipping rate, reducing both the code search space and
The above operation is performed by varying the
the computational complexity significantly. The
locally generated oscillator frequency in steps of 500Hz
detection performance of CW strategy is assessed as
until the correlation with the input signal is achieved.
equivalent to that of the full-rate correlation,
B. Parallel code space search acquisition with
irrespective of the phase of the received L2C code. The reduced points
NRZ CM codes allow code search at larger steps This method uses much less number of computations
without losing the acquisition sensitivity. compared to parallel code search acquisition. This
technique uses a down sampling method. The total
RZ CM replica code consists of CM code alone and spectrum, after carrier multiplication, is brought down
zeros in place of the CL code. Since L2 CL has a to chip rate. The data samples are accumulated (pre-
longer code length; it is normally not used for correlation filtering and down sampling) due to which
acquisition process. The RZ codes have a sharp incoming samples in 1 chip are averaged to one sample
triangular acquisition peak. value. This leads to high computational efficiency and
small correlation loss.
IV. ACQUISITION METHODOLOGIES
Acquisition identifies all satellites visible to the user

and determines two properties of the signal: code phase
and carrier Doppler. Basically two types of
acquisition are followed namely serial search and
parallel search acquisition. The serial search
acquisition method is a very time-consuming
procedure to search sequentially through all possible Fig.4. Parallel code space search acquisition with
reduced points
values of the two parameters, namely frequency and
code phase. If any of the two parameters could be
C. Parallel Frequency space search acquisition
eliminated from the search procedure or if possible
implemented in parallel, the performance of the
procedure would increase significantly.
A. Parallel code space search acquisition

This algorithm estimates the cross correlation between
the input and the local code with shifted code phase.
The incoming signal is multiplied by a locally
generated carrier signal. Multiplication with the signal
generates the In-phase (I) signal, and multiplication Fig.5. Parallel frequency space search acquisition
with a 90 phase-shifted version of the signal generates
the Quadra-phase (Q) signal. The I and Q signals are In this algorithm the input signal is multiplied with the
combined to form a complex input signal X(n) = I(n) + j locally generated NRZ or RZ CM signal, and then the
Q(n) to the DFT function. The generated PRN code is signal is multiplied with a locally generated carrier,
transformed into the frequency domain and the result is which generates I and Q outputs. These outputs are
complex conjugated. The Fourier transform of the input down sampled to twice the maximum Doppler
79
frequency (100 KHz in case of receiver on a LEO (pre-correlation filtering and down sampling), so that
satellite) to get higher (finer) frequency resolution. incoming samples in 1 chip are averaged to one sample
Finally the Fourier transform is taken and correlated value. This leads to high computational efficiency and a
output is viewed. This is repeated until incoming signal small correlation loss. Simulation for the correlation
code phase and the locally generated code are in same values with locally generated RZ and NRZ CM codes
phase. This algorithm reduces the number of FFT with Doppler frequency of 0 Hz to 50 have been carried
computations as compared to the other algorithms. out.
V. SIMULATION RESULTS
The acquisition methods are carried out in MATLAB

2012 simulation software. The CL code chip rate is
1.023MHZ, the sampling frequency is 20.46MHZ and
the number of samples considered is 20*20460 (20 ms
of data).
A. Parallel code space search acquisition

The NRZ local code allows code search at longer steps
without losing the acquisition sensitivity (One-chip
duration), where as RZ codes allow a sharper transition.
The correlation peak for NRZ is 0.5069 and for RZ is
Fig.8. RZ correlation for various Doppler Carriers from 0 to 50 Hz
0.5.
Fig.6. correlation value of received code with NRZ

code Fig.9. NRZ correlation for various Doppler carrier from 0 to 50 Hz
Fig.7. correlation value of received code with RZ code

Fig.10. Comparison of correlation values between NRZ
B. Parallel code space search acquisition with and RZ local codes for various Doppler frequencies
reduced points
Here the code phase between the incoming and locally C. Parallel frequency space search acquisition
generated code is varied and the locally generated Correlation values are plotted for locally generated
carrier frequencies are varied between 0 to 50Hz codes with varying IF carrier frequency from 0 to 50
Doppler. Then 20 samples are accumulated and added Hz. Secondly IF carrier and locally generated carrier are
80
set at same frequency and by varying the code phase VI. CONCLUSION
correlation values are plotted.
The L2C code is a Time Multiplexing of two separately
generated codes (L2 CM and L2 CL). CL code is not
used for acquisition based on the NRZ and RZ local
codes. The acquisition based on circular correlation
FFT, parallel frequency space search and parallel code
space search acquisition are performed for various
Doppler carrier and code phase and the FFT
computations and acquisition time are analyzed. The
results show that the parallel frequency space search
acquisition provides less FFT computation and is very
efficient.
REFERENCES
Fig.11. Comparison of correlation values between NRZ [1] A Software Defined GPS and Galileo Receivers A Single
and RZ local codes for various Doppler frequencies Frequency Approach - Kai Borre, Dennis M. Akos,Nicolaj
Bertelsen,Peter Rinder,Sren Holdt Jenson.
[2] GPS directorate and system engineering and integration interface
specification IS-GPS-200F -21 SEP-2011.
[3] A SANA U. QAISAR, ANDREW G. DEMPSTER, Senior
Member IEEE University of New South Wales Australia Assessment
of the GPS L2C Code Structure for Efficient Signal Acquisition
IEEE TRANSACTIONS ON AEROSPACE AND ELECTRONIC
SYSTEMS VOL. 48, NO. 2 APRIL 2012.
[4] TUNG HAI TA,Hanoi University of Science and
Technology,Vietnam, SANA U. QAISAR, ANDREW G.
DEMPSTER, Senior Member, IEEE, University of New South Wales,
Australia, FABIO DOVIS, Politecnico di Torino, Italy ,Partial
Differential Postcorrelation Processing for GPS L2C Signal
Acquisition, IEEE TRANSACTIONS ON AEROSPACE AND
ELECTRONIC SYSTEMS VOL. 48, NO. 2 APRIL 2012.
[5] S.Mariappan, Swarna Ravindra Babu, Sasi Bhusana
Rao,Acquisition and tracking strategies of Modernised GPS L2C
Fig.12. correlation values with varying the code phase in steps of 5 signal,IEEE- International Conference on Recent Trends in
samples (Doppler is zero) Information Technology, ICRTIT 2011.
[6] Yanhong Kou, Xingyun Zhou, Beihang University, China,Yu
Morton, Miami University,Lei Zhang, Beijing Institute of
Technology/Miami University ,Processing GPS L2C Signals under
Ionospheric Scintillations.
Fig.13. correlation values with varying the code phase

in steps of 5 samples (Doppler is 25 Hz)
81
Acquisition Techniques of Modernized GPS

L1C BOC(1,1) Signal
Kishore Babu N1, Mukesh Kumar Gupta2, Hemanth Kumar Reddy3, Chitra Ramamurthy4
Communication Systems Group, ISRO Satellite Centre
Old Airport road, Bangalore-560017, India
1
kishoren@isac.gov.in, 2mukesh@isac.gov.in, 3hemanthr@isac.gov.in, 4chitra@isac.gov.in
Abstract- The Global Positioning System is undergoing the relative motion of the satellite with respect to the
continual modernization, providing continuous user, causing a Doppler effect. The Doppler
improvements to users worldwide. The development of frequency shift depends upon the velocity of the
the next generation of satellites, called GPS III. An GPS satellites and user velocity. This can be 5 kHz
integral part of the GPS III capabilities being for a stationary receiver on Earth and as high as 50
developed is a new civil signal, called L1C, which KHz for a receiver on Low Earth Orbiting (LEO)
will be transmitted on the L1 carrier frequency in
addition to current signals. The development of L1C
Satellites. The code phase denotes the point in the
signal represents a new stage in international GNSS. current data block where the L1C code begins. If a
The L1C signal is being designed for interoperability data block of 10ms is examined, the data includes
between GPS and Galileos Open Service Signal and an entire L1C code and thus one beginning of
also Japans Quazi-Zenith Satellite System (QZSS). The a L1C code. These measurement values are
L1C signal has many advanced features and employs used by tracking loops for tracking code and
the BOC modulation scheme. This paper describes the carrier signals and further demodulation of
features of the L1C signal and different methods for the Navigation data of GPS satellites.
acquisition of the L1C signal. The methods used for Different methods for acquiring like Fishman & al.,
unambiguous acquisition of the BOC signal include
BPSK- like methods (Fishman & al. method and Martin
Martin & al., and Sub Carrier Phase Cancellation
& al. method) and the Sub Carrier Phase Method are used and compared.
Cancellation (SCPC) method.
Keywords- GPS acquisition, Time domain correlation, II. L I C SIGNAL STRUCTURE
sub carrier phase cancellation, single side band, binary
phase shift keying. The L1C (1575.42 MHz) signal consists of two main
components, one denoted L1CP to represent a pilot
I. INTRODUCTION signal, without any data message and the data
message L1CD. The pilot and data components are
The Global Positioning System is undergoing transmitted using ranging codes. The pilot
continual modernization, providing on-going component is further modulated by an overlay code
improvements for users worldwide. The next which is unique to each satellite. The ranging codes
generation of satellites, called GPS III includes a new used to spread the pilot and data components are
civil signal, called L1C, which will be transmitted on also separate and unique to each satellite. Length
the L1 carrier frequency in addition to current 10230 sequences are used to spread both the pilot
signals. The L1C signal will be available with the and data components of L1C, with a repetition
first Block III launch, scheduled for 2014. L1C signal period of 10 ms at the 1.023MHz chip rate.
has advanced features like
Pilot component is sent along with the data. The pilot Different spreading sequences are used for each
component is useful for accurate acquisition. The satellite and for each pilot and data component. The
power split is unequal with 75% in the pilot spreading codes corresponding to the ith satellite are
component and 25% in the data component. The L1C denoted as L1CPi(t) and L1CDi(t). In addition, the
signal employs advanced Weil based spreading codes. pilot component is also modulated by an overlay
Advanced forward error control coding: the FEC code of duration 18s at 100 bps with length 1800.
coding provides an improvement in data The overlay code is unique to each satellite. The
demodulation threshold of more than 7 dB compared overlay code corresponding to PRN signal number
with no FEC, such as the C/A code signal. Data i is denoted as L1COi(t). The spreading codes
symbol interleaving to combat fading. are based on Weil sequences which in turn are
The purpose of acquisition is to identify the visible derived from the length 10223 Legendre sequence.
satellites to the user. The two important parameters The modulation used for L1C signal is binary offset
estimated are the Carrier Doppler and the Code carrier (BOC). A BOC signal comprises of the data
Phase. As the carrier and code signals are affected by message, a PRN spreading code, square sub carrier
82
and a sinusoidal carrier. The BOC modulated signal signals. This paper describes the acquisition of data
is generated by the multiplication of these component signal which is BOC(1,1). As BOC(1,1)
components in the time domain. Both the spreading signal auto correlation function (ACF) has multiple
code and the sub carrier are binary-valued. Thus, the peaks, which cautions the user to acquire the main
BOC modulated signal may be viewed as a BPSK peak rather to side peaks for effective initialization
signal multiplied with a square sub carrier. of tracking loops. Several techniques can be used to
Multiplication by the sub carrier causes the spectrum eliminate the ambiguity of the ACF envelope. Two
of the BPSK signal to split into two parts with main- such methods which can be used to acquire BOC
lobes on either side of the carrier frequency. For this signal include BPSK-like technique and the Sub
reason, the BOC modulation is also referred to as Carrier Phase Cancellation (SCPC) technique.
split spectrum modulation. A. Fishman & al. BPSK Method
The BOC signal has two parameters: the sub carrier The main idea behind these BPSK-like methods is
frequency fsc the chip rate of the spreading code that the BOC-modulated signal can be obtained as
fc. The sub carrier frequency and the chip rate are the sum of two BPSK-modulated signals, located at
chosen to as multiples of a reference frequency, positive and negative sub-carrier frequencies. The
such that: effect of sub-carrier modulation can be removed
fsc = f0 and fc = f0, Where, the reference by using a pair of single- sideband correlators. We
frequency f0=1.023MHz. may have a single-side band (SSB) receiver, where
either the negative or the positive of the sidebands
The BOC signal is thus represented as BOC (, ).
correlators is used, or a dual-side band (DSB)
The modulation for the L1C data component is
receiver, when both bands are combined non-
strictly BOC(1,1). The L1C pilot component
coherently. Due to filtering and correlation losses,
uses a time multiplexed combination of
the BPSK-like methods bring some degradation in
BOC(6,1) and BOC(1,1) known as TMBOC. L1C
the signal level. A SSB BPSK-like method induces
signal from one satellite is
at least 3 dB degradation in SNR. If DSB processing
is used, this loss can be partially compensated,
excepting anyway the squaring losses in non-
coherent integration. Compared to coherent
processing of both sidebands, non-coherent
processing loses about 0.5 dB of SNR.
Where
the total signal power is denoted as C(Watts) which
includes any antenna gain and receiver
implementation losses, dP (t) and dD(t) are the series
of pilot overlay code and data bits, cP (t) and cD(t)
are the spreading codes for the pilot and data
components, gP(t) and gD(t) are the periodic
repetition (repeating every spreading code chip) of
the spreading symbols (also known as the
subcarrier) for the data and pilot components, and
fd are the unknown delay and Doppler frequency Figure 1: Principle of Fishman method
of the signal, the signal is at an intermediate In this method, the receiver selects only the main
frequency fIF (Hertz), the unknown phase term is lobes of the BOC-modulated received signal and the
(radians), which is the phase difference reference code. The main lobe of one of the
between the received signal and the locally sidebands (upper or lower) of BOC-modulated
generated signal used for down conversion, received signal is selected (via filtering) in each
And n(t) is additive white Gaussian noise with channel as shown in Figure1. Both sidebands are
power spectral density N0/2. translated to the baseband by multiplication with
complex exponential. In each channel, the resulting
III. ACQUISITION METHODOLOGIES signal is correlated with the reference code, having
the tentative delay and the tentative Doppler
Acquisition identifies the visible satellites to the user frequency fD. Hence, the DSB Fishman & al.
and determines two properties of the signal: code approach needs two sideband selection filters. The
phase and carrier Doppler. Either the pilot or data outputs of the two channels are combined non-
component can be used for acquisition GNSS coherently. This method has rather high complexity
83
due to the fact that two complex filters are needed search and transition because there is only one main
for DSB processing. Instead of the DSB statistic, energy peak. The advantage of the technique is an
only one of the channels may be retained for increased simplicity of the implementation, and
lower complexity. This gives the SSB statistic. above all, the least effect of the filtering.
However, as mentioned earlier, in this case the
SNR degrades by at least 3dB. The correlation
function between the received signal and reference
code, on each sideband, will be unambiguous and
will resemble the ACF of a BPSK- modulated
signal. However, the shape of resulting ACF is not
exactly the one of a BPSK modulated signal,
since there are information losses due to selection of
main lobes.
B. Martin & al.BPSK Method Figure 3: Correlating the filtered signal with the local
Here, a single centered filter is used to select the BPSK Signal
main lobes and the secondary lobes between
them. For this reason, this method has lower
implementation complexity as compared to the C. Sub Carrier Phase Cancellation (SCPC)
Fishman & al. method. This method uses only one method
centered filter with a bandwidth including the two
principal lobes of the spectrum and the secondary The Sub Carrier Phase Cancellation technique
lobes between the two principal lobes. (SCPC), allows ones to make a non ambiguous
Two correlation channels are generated: right acquisition of a BOC signal. Figure4 present the
channel where the filtered signal situated at fIF+fsc general principle. The idea here is to get rid of the
sub carrier signal as it is done for the carrier signal.
and correlated with a BPSK signal and left channel
In addition to the local in phase and quadrature
where the filtered signal situated at fIF-fsc and
carrier signals, an in phase and a quadrature local
correlated with a BPSK signal. The principle of this sub carrier signals have to be generated. Thus, two
method is illustrated in Figure2. The correlation correlation channels are generated here. On one
outputs of the two channels are combined. channel, the received filtered signal is correlated
with the local BOC signal in sub carrier phase, and
on the other one the received filtered signal is
correlated with the local BOC signal in sub
carrier quadrature. When these two correlation
channels are combined, an ACF similar to the BPSK
one is obtained.
Figure 2: Principle of Martin method
The correlation function is unambiguous. The

correlation function presents correlation losses
(~3dB) and has not exactly the same shape as the
BPSK(y) one. This is due to the mismatch between
the received code (BOC) and the local code (shifted
BPSK) as seen in Figure3. In addition, the sub- Figure 4: Principle of SCPC method
carrier, which modulates the BPSK spreading code,
has a square wave shape instead of sine wave. IV. S I M U L A T I O N RESULTS
Due to unsymmetrical spectrum, considering only The acquisition methods are carried out in MATLAB
one lateral main lobe, the correlation function is simulation software. BOC(1,1) signal is generated by
complex. The consequence is a slight coupling modulating the PRN code onto the square wave
between the carrier phase and code delay measures. sub carrier. In this case the code rate fc = fref =
However this feature is not critical for the energy 1.023 Mcps and the sub carrier frequency is fsc = fref
84
= 1.023 MHz. Figure 5 shows the spectrum of the

BOC signal before modulation onto the carrier.
Figure 5: Correlating the filtered signal with the Local

BPSK Signal
Figure 7: Correlating peak Vs Doppler
It is observed that the signal has low energy at the
central frequency. The energy peaks at the sub
carrier frequency fsc = 1.023 MHz. For all the
following simulations, the IF frequency of the BOC
(1,1) signal is assumed to be 8 MHz. The sampling
rate is chosen to be 40 MHz. The Figure6 shows the
reconstructed autocorrelation function of the BOC
(1, 1) signal using the 3 methods as compared with
the ACF of the original signal shown in blue. It is
seen that all three methods remove the ambiguity
of the original ACF giving rise to a single peak. Of
the three methods, the SCPC method is seen to result
in the highest correlation value at 0th lag coinciding Figure 8: Correlating peak plot of Fishman & al.
with the peak value of ACF of the original BOC (1, method
1) signal.
Figure 9: Correlating peak plot of Martin & al.

method
Figure 6: Auto Correlation Function with different

Methods
The Figure7 shows the peak of the ACF versus
the Doppler shift for the three acquisition methods.
The BOC signal was generated using zero code

phase and 5KHz Doppler frequency. Figures 8,9,10.
Provides acquisition plots of the BOC signal in
the presence of noise (SNR = -30dB) using all
three methods. It is seen that the peak occurs at the
correct values of code phase = 0 chips and Doppler
shift = 5 KHz.
Figure 10: Correlating peak plot of SCPC method
85
[6] Burian, A.; Lohan, E.S.; Renfors, M., "BPSK-like

Methods for Hybrid-Search Acquisition of Galileo Signals,"
Communications,
2006. ICC '06. IEEE International Conference on , vol.11, no.,
pp.5211,5216, June 2006.
[7] Martin, N., Leblond, V., Guillotel, G., Heiries, V.,
"BOC(x,y) Signal Acquisition Techniques and Performances,"
Proceedings of the 16th International Technical Meeting of the
Satellite Division of The Institute of Navigation (ION GPS/GNSS
2003), Portland, OR, September 2003, pp. 188-198.
Figure 11: Correlating peak plot of SCPC method

with -40 dB SNR
V. CONCLUSION
Of the three methods, the peak is highest for the
SCPC method. From the above simulations it is
seen that with respect to the correlation function and
the energy search, the SCPC outperforms the other
two methods considerably.
REFERENCES
[1] Navstar GPS Space Segment/User Segment L1C Interface,

IS-GPS-800A.
[2] Betz, J., Blanco, M.A., Cahn, C.R., Dafesh, P.A., Hegarty,
C.J., Hudnut, K.W., Kasemsri, V., Keegan, R., Kovach, K.,
Lenahan, L.S., Ma, H.H., Rushanan, J., Rushanan, J.J., Sklar, D.,
Stansell, T.A., Wang, C.C., Yi, S.K., "Description of the L1C
Signal," Proceedings of the 19th International Technical
Meeting of the Satellite Division of The Institute of Navigation
(ION GNSS 2006), Fort Worth, TX, September 2006, pp. 2080-
2091.
[3] A Software-Defined GPS and Galileo Receiver: A
Single- Frequency Approach, By Dennis M. Akos, Nicolaj
Bertelsen
[4] J. Betz, Binary Offset Carrier Modulations for
Radionavigation, Journal of the Institute of Navigation, vol. 48,
pp. 227246, 2001.
[5] Zaixiu Yang; Zhigang Huang; Shengqun Geng,
"Unambiguous Acquisition Performance Analysis of BOC(m,n)
Signal," Information Engineering and Computer Science, 2009.
ICIECS 2009. International Conference on , vol., no., pp.1,4, 19-20
Dec. 2009.
86
TEC Response Due to Geomagnetic Storm At

Low Latitude GPS Station
N.S.M.P.Latha Devi, D.VenkataRatnam, S.Venkatesh
K L University (KoneruLakshmaiah Education Foundation) Vaddeswaram, Andhra Pradesh, India.

Email:lathansmp@gmail.com
Email :dvratnam@kluniversity.in
Email: swargam.venkatesh@gmail.com
Abstract- The responses of the ionosphere TEC due tracking performance (Das Gupta et al., 2004, Dubey
to the Geomagnetic storm have been investigated by et al., 2006, Skone, S., 2000)(3),(4)&(5). Since every
using the Global Positioning System (GPS) data geo magnetic storm has unique character and
during 29 June, 2013 at low latitude station, K L forecasting the response of ionospheric parameters
University, Vaddeswaram,(Geographic latitude like total electron content (TEC) is one of the major
16.440N, Geographic longitude 80.620E),India. The issues in space weather studies. The geomagnetic
variations of VTEC during this storm are analyzed, storms drastically affect the TEC of GPS signals,
significant deviations in VTEC during the main especially of low latitude region such as India and
phase of the storms are observed. During June Brazil etc,. In this context, the response of the
storm,TEC depeltions are observed. The depletion in ionosphere over low latitudes needs to
VTEC is due to the thermospheric composition investigated.Therefore there is a necessity of
changes; however disturbance dynamo electric field incorporate a provision in the ionospheric models for
also plays an important role. The GPS VTEC values detecting the ionospheric anomalies and magnetic
compared with VTEC derived from IRI-2007 & IRI storms (Ho et al., 1996) (6).
2012 models. It is observed that IRI models could
not predict these variations during storm days. It The severity of geomagnetic storm is usually
has been found that the development of more experiences its geomagnetic storm can be explained
accurate TEC models would be needed. So, it is in terms of Dst (Disturbance storm time) index as
necessary to modify the IRI models to get better well as Kp index. The Dst index is a measure of geo
accuracy. magnetic activity used to assess the severity of
magnetic storms. It is based on average value of
Key words: GPS TEC, IRI TEC, low-latitude horizontal component of earths magnetic field
Indian station, geomagnetic storm. measured hourly at four near geo magnetic
observatories. Use of Dst as index of storm strength
I. Introduction because the strength of the Earths surface magnetic
field at low latitudes is inversely proportional to the
The sun is the major source that controls the near energy content of the ring current, which increases
earth surface and drives its weather and climate. during geomagnetic storms. Generally, geomagnetic
Coronal mass ejections from the sun interact with storms can be classified according to different Dst
solar wind and the interplanetary magnetic field index levels: weak; -50nT Dst -30nT; moderate; -
(IMF) during their propagation disrupts the solar 100nT Dst -50nT and intense Dst -100nT. The
wind. Fast CMEs eventually drive shocks on the Kp index is computed on a three hour basis and
earths magnetosphere and trigger geo magnetic represents the overall level of planetary geo magnetic
storms. The magnetospheric storm is a global electro field disturbance. It is derived from ground based
dynamic disturbance in the magnetosphere- magnetic field measurements ranges from 0-9. A
ionosphere. Recently, the Global positioning system typical quiet day will have Kp values of 0-2.
(GPS) has been broadly used on scientific research According to NOAA scale, storms are classified into
because of its global and on time characteristics. strong (Kp = 7), Severe (Kp=8) and extreme (Kp=9).
During geomagnetic storms, the ionospheric From Dst and Kp indices, it is observed that the
disturbances cause data loss in satellite strong strom is occurred on 29th June, 2013.
communication links and loss of signal for GPS
satellite (Das Gupta et al., 2004, Chen Wu et al., In this paper the variations of VTEC measured by
2008)(1), (2). This can also reduce the availability of monitoring GPS L1 and L2 signals across low
satellites for GPS navigation, which may affect the latitude station, Vaddeswaram, India is investigated.
positional accuracy, navigational accuracy and GPS Measured VTEC were then compared with the TEC
87
predicted by the IRI-2007 and IRI-2012 models to model for all phases of the solar cycle. More
assess its predictability and observation. information about IRI can be found on the IRI
homepage at http://IRI.gsfc.nasa.gov/. The
II. GPS TEC data corresponding IRI TEC predictions are calculated
from IRI 2007 and IRI 2012 models. To predict
The GPS TEC data used in this study were recorded the IRI TEC from IRI 2007 and IRI 2012 we can
by the GPS receiver (Model: GPS 6 Station) which is access the
installed at KL University, siteathttp://ccmn.gsfc.nasa.gov/model/web/models/iri
VaddeswaramduringJanuary to May, 2013. The slant _vtmo.php/which has more recently become available
TEC (or STEC) from a satellite to a receiver can be as an interactive web interface accessible from the
obtained from the difference between the pseudo IRI home page.
ranges (P1 & P2) or the difference between the
carrier phases (L1 and L2) of the two frequencies IV. TEC estimation methodology
(Blewitt, 1990) (7) i.e.,
The dual frequency GPS data provides both code and
( ) carrier phase measurements on the two GPS
= ( ) (
+ +
) frequencies and navigation data parameters due to all
the visible satellites. In the estimation of TEC and
Or instrumental biases, several data processing steps
including extraction of desired data, smoothing,
2( 1 2) computation of satellite position, receiver position,
= + +
( 1 2)2 ( 1 1 2 2) slant factor and mean longitude of sun is to be carried
out. These data processing steps are implemented in
MATLAB code.
Where, k relate to the ionosphere refraction, is
80.62m3/s2; 1 & 2 are the wavelengths V. Results and discussions
corresponding to f1 &f2 respectively; r& r are the
differential code biases corresponding to the pseudo Major Geomagnetic storms originated from solar
ranges (P1 & P2) and the Carrier phases L1 & L2 bursts can cause strong disturbances in the geo space
respectively; s& s are the inter-frequency biases environment which ultimately affects the
corresponding to the pseudo ranges (P1 & P2) and the performance of GPS system. During geomagnetic
Carrier phases (L1 & L2) respectively. storm period, the ionospheric disturbances cause loss
VTEC in electron/m2 can be calculated as of signal for GPS satellite and data loss in satellite
(Ma and Maruyama, 2003) (8)as communication links. Usually these storms associated
VTEC = STEC *(cos X), with increased electron densities in the lower
Where the zenith angle X is expressed as X = arc sin ionosphere. GPS technology requires better space
(RE Cos /RE + h). weather support to compensate the ionospheric
Where is the elevation angle of the satellite, R E is induced errors. Hence there is a necessity to
the mean radius of the earth and h is the height of understand and model the effects of ionospheric delay
ionospheric layer, which is assumed to be 350 km. on radio system during the storm period.According to
previous studies, middle and low latitude ionosphere
II. IRI model can be affected with an Equatorial anomaly expanded
III. poleward and become pronounced during an storm
The IRI model is empirical model of the ionosphere events (Tanaka 1986, Abdu 1997)(14),(15). Migoya
by a working group that is jointly sponsored by the et al (2009)(16) reported that large values of TEC at
committee on Space Research (COSPAR) and the low latitudes during geo magnetic storms in
International union of Radio Science (URSI) (Bilitza correspondence with well developed crest of TEC.
and Rawer, 1993)(9). The model provides spatial and Das Gupta et al., (2007)(17) reported that the
temporal TEC representations. depletion in TEC are quiet frequent even under
magnetically quiet condition in the equatorial
The IRI model is continually upgraded as new data ionosphere particularly during equinoctial months of
and new modeling approaches become available and solar maximum period. Shagimuratov et al.,reported
this process results in several major milestone that the response of TEC to geo magnetic storm
editions of IRI (Rawer et al., 1978a,b, 1981; Bilitza& consists of effects of both enhancement and depletion
Rawer, 1996)(10),(11),(12), & (13) that progress (positive and negative disturbances). Dashora et
from a set of tables for typical conditions to a global al.,(2009) (18) studied the TEC response to the geo
88
magnetic storm of 15th May 2005 at Udaipur a station The performance of VTEC obtained from GPS
near the crest of equatorial ionization anomaly and derived TEC are compared with IRI model data in
Hyderabad a low latitude station using GPSTEC. order to determine their accuracy when storms takes
They found that the large enhancement (or place. During the storm day the maximum TEC
decrement) in VTEC is attributed to prompt obtained from IRI-2007 model is around 75 TECU.
penetration of electric fields and Travelling IRI model over estimates the GPS TEC value during
Ionospheric Disturbances. Bagia mala et al., (2009) the storm period and IRI model fials to predict
(19) studied the 24th August 2005 geo magnetic storm variability in the TEC value during geomagnetic
for Rajkot station using GPSTEC and found that the storm period.
enhancement to VTEC is due to prompt penetration
electric fields during storm day and thermospheric Conclusions
composition changes and disturbance dynamo electric
fields on the recovery day. In this paper, Analysis of TEC depletions during
29 June, 2013 Storm adverse space weather conditions is investigated.
The main feature of 29th June 2013 storm under
Fig.1 represents the variation of GPSTEC, and TEC investigation was a negative VTEC effect. The VTEC
derived from IRI-2012 and IRI-2007 models with Dst depression amounted up to 4.68 TECU. GPS TEC
index during storm period (26-07-2013 to 30-07- data results reveals that IRI models are
2013). The sun spot number during this period is overestimatedwhen compared with the GPS TEC
52.5. During pre storm day i.e., on 28th June, 2013 at measurements made with a ground-based GPS
00:00LT VTEC reaches to 2.33TECU, at 12:00LT receiver during storm days observation period.
VTEC attains a value 23.83 TECU with Dst index - Between these two models IRI-2012 model shows
13nT and at 22:00LT Dst index reaches to -72nT and better results than IRI-2007 model. Based on these
VTEC reaches to 8.3TECU during initial phase of the observations, it is necessary to modify the IRI models
storm. During storm day i.e., 29th June, 2013 at to get better accuracy.
00:00LT Dst index reaches to -96nT and VTEC
attains 2.472 TECU with Dst index 7nT. At 08:00LT Acknowledgement
Dst index reaches to -98nT and VTEC reaches to
4.68TECU during Main phase of the storm. After The above work has been carried out under the
10:00LT Dst index slowly reaches to -87nT and TEC project entitled Development of Ionospheric
reaches to 18.95 TECU and at 20:00LT Dst index Forecasting models for Satellite based Navigation
attains -60nT and VTEC reaches to 4.5 TECU during systems over the low latitude stations sponsored by
recovery phase. From fig.1, it is observed that there is Department of Science and Technology, New Delhi,
depletion in TEC during this storm day. India, vide sanction letter No: SR/FTP/ETA-
0029/2012, dated:08.05.12.
References
1. Das Gupta A, Ray S, Paul A, Banerjee P & Bose A,

Errors in position-fixing by GPS in environment of
strong scintillation in the Indian zone, Radio Sci
(USA), 39 (2004) RSIS30.
2. Chen Wu, Gao S & Hu congwei, Effect of
ionospheric disturbances on GPS observations in low
latitude area, GPS Solut (USA), 12(1) (2008) 33.
3. Das Gupta A, Ray S, Paul A, Banerjee P & Bose A,
Errors in position-fixing by GPS in environment of
strong scintillation in the Indian zone, Radio Sci
(USA), 39 (2004) RSIS30.
4. Dubey S, Wahi R, & Gwal A K, Ionospheric effects
on GPS positioning, Adv Space Res (UK), 38 (2006)
2478.
Fig.1 : Variation of GPS TEC, IRI-2012 TEC, and IRI-2007 TEC 5. Skone S, Impact of ionospheric scintillation on
withDst index during Geo magnetic storm period (26-07-2013 to SBAS performance, ION GPS 2000 (Salt Lake city,
30-07-2013). Utah), 2000.
6. Ho, C.M., A.J. Mannucci, U.J. Lindqwister, X. Pi,
and B.T. Tsurutani, Global IonosphericTEC
Variations During January. 10, 1997 Storm, Geo
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Physical research letters, Vol.25, Issue 14, Pages

2589-2592, 1998.
7. Blewitt G., An automatic editing algorithm for GPS
data, Geo Physical research letters, Vol.17, No. 3,
199-202, 1990.
8. Ma, G., Maruyama, T., Derivation of TEC and
estimation of instrumental biases from GEONET in
Japan, Ann. Geophys., 21, 2083-2093,2003.
9. Bilitza D, Rawer K., Bossy.L and Gulyaeva
,International Reference Ionosphere-Past, Present,
Future, Av.Spae Res.13, 3-23,1993.
10. Rawer k, Bilitza D, Ramakrishnan S, Sheikh M,
International Reference Ionosphere, International
Union of Radio Science (URSI), Brussels, Belgium,
1978a.
11. Rawer k, Bilitza D, Ramakrishnan S, Sheikh M,
Intentions and buildup of the International Reference
Ionosphere, in : Operational Modeling of the
Aerospace Propagation Environment, AGARD Conf.
Proc. Vol.238,pp.6.1-6.10,1978b.
12. Rawer, K., Lincoln, J.V., Conkright, R.O.,
International Reference Ionosphere-IRI 79, World
Data Center A for Solar Terrestial Physics, Report
UAG-82, Boulder, Colorado, 1981.
13. Rawer K, Bilitza D, Mahajan K, and Mitra A, Low
and equatorial latitudes in the International Reference
Ionosphere, Advances in Space Research, Volume
18, Number 6, 1996.
14. Tanaka, T.: Low-latitude ionospheric disturbances:
Results for March 22, 1979, and their general
characteristics, Geophys. Res. Lett., 13(13), 1399
1402, 1986.
15. Abdu M A, Major phenomena of the equatorial
ionosphere thermosphere system under disturbed
conditions, J Atmos. Sol-TerrPhys (UK) , 59 (1997)
1505.
16. Migoya-Orve Y.O., S. M. Radicella& P. Coisson,
Low latitude ionospheric effects of major geo
magnetic storms observed using TOPEX TEC data,
Ann. Geophys., 27, 3133-3139, 2009.
17. DasGupta A, Paul A & A. Das, Ionospheric total
electron content (TEC) studies with GPS in the
equatorial region, Indian Journal of Radio & Space
Physics, Vol. 36, Pp 278-292, 2007.
18. Dashora N, Sharma S, Dabas R S, Alex S &Pandey
R, Large enhancement in low latitude total electron
content during 15 May 2005 geo magnetic storm in
Indian zone, Ann. Geophys (France), 27,(2009),
1803.
19. Bagiya M S, Joshi H P, Iyer K N, Agarwal M,
Ravindran S &Pathan B M, TEC variations during
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Ann. Geophys(France), 27 (2009) 1047.
90
Implementation of CIC Filter based DDC for

Extraction of Delta DOR Signals
K R Yogesh Prasad*, Priyanka Das, Jyothi B R, Chitra Ramamurthy
AbstractReceiver architecture for recovery of Delta Hence, when the measurements are differenced the
Differential One-way Ranging (DDOR) signals has errors that are common to both the measurements get
been described. Characteristics of Cascaded Integrator eliminated resulting in higher accuracies as compared
Comb (CIC) filters have been analyzed. Extraction of Delta to plain Differential One- way Ranging (DOR).
DOR tones from the spacecraft downlink signal has been
demonstrated using Cascaded Integrator Comb (CIC) based
Digital Down Converter (DDC). The accuracies are good enough for satellite orbit
determination and navigation.
Index TermsDelta Differential One-way Ranging The concept of DDOR is graphically represented in
(DDOR), Cascaded Integrator Comb (CIC) filters, Digital Fig. 1. The line through the earth joining the two
Down Converter (DDC), I-Q signal processing. ground stations participating in the DDOR
measurements is known as the baseline. If B is the
length of the baseline and is the angle subtended
I. INTRODUCTION
by the spacecraft with respect to the baseline, then the
differential delay experienced by the signal in reaching
DELTA-DOR is a Very Long Baseline the two ground stations is given by the formula:
Interferometry (VLBI) technique that can be used in = B Cos() / c, where c is the velocity of light in
conjunction w i t h Doppler and ranging data to free space. Once is obtained from measurements, the
improve spacecraft navigation by more efficiently value of can be obtained from the above formula.
determining spacecraft angular position in the plane of
sky. An application of VLBI is spacecraft navigation in
space missions where delay measurements of a
spacecraft radio signal are compared against similar
delay measurements of angularly nearby quasar radio
signals. In the case where the spacecraft measurements
are obtained from the phases of tones emitted from the
spacecraft, first detected separately at each station, and
then differenced, this application of VLBI is known
as Delta Differential One-Way Ranging (DDOR or
DOR). The data produced during a DDOR
measurement session are complementary to Doppler and
ranging data.
One-way range for a single station is determined by

extracting the phases of two or more signals emitted by
the spacecraft. DOR observables are formed by Fig.1 Concept of DDOR
subtracting the one-way range measurements generated
at the two stations. High frequency sinusoidal tones, Section II of this paper discusses the Receiver
referred to as Delta DOR tones, modulating the architecture for applications such as recovery of DDOR
spacecraft downlink carrier at S-band, X-band or Ka- tones from the spacecraft downlink signal. Section III
describes CIC filter and discusses its frequency
band are used for making the range measurements.
response characteristics. Section IV describes the
Similar measurements are made by receiving signal
realization of a Digital Down Converter (DDC)
from a quasar that is angularly close to the spacecraft. utilizing Cascaded Integrator Comb (CIC) filter.
Even though data acquisition and processing are not MATLAB simulations are shown as a proof-of-concept
identical for the spacecraft and quasar, both types of for the CIC filter based DDC. Practical results obtained
measurements can be interpreted as delay by hardware implementation of the DDC scheme are
measurements and they have similar information presented in Section V. Conclusions are presented in
content and similar sensitivity to sources of error. Section VI.
91
II. RECEIVER ARCHITECTURE FOR DDOR discrete-time domain is carried out as sample- to-sample
multiplication. Heterodyning results in the signal
The block schematic of a typical ground segment for centered at IF being down converted to baseband
DDOR operations is shown in Fig.2. frequency. The heterodyning process also yields an up-
converted signal at double the IF frequency. This signal
is not of interest and will be filtered. The output of filter
contains signal of interest at baseband. This signal,
being a narrow band signal centered at zero frequency,
need not be sampled at the rate at which the original IF
signal was sampled. Hence, decimation can be
carried out so that subsequent discrete-time signal
processing can be carried out at a much lower sampling
rate. When I-Q processing is carried out, the minimum
sampling rate required for each of the two streams after
decimation can be as low as the bandwidth of the
original signal.
Fig. 2 Block diagram of Ground Segment for DDOR

Operations
The downlink signal from the spacecraft is

amplified by LNA and is typically subjected to two
stages of Down- Conversion. The output of the second
down-converter is digitized using an Analog to Digital Fig.3 Block diagram of a typical Digital Down
Converter (ADC). The sampling frequency of the ADC Converter
should be at least twice the maximum frequency For applications like these, a Cascaded Integrator
component present in the signal being digitized. The Comb filter may be used after the heterodyning
digitized signal is passed through a Digital Down stage. This filter performs filtering as well as
Converter where the signal is brought down to decimation as required by the DDC without the
baseband. This composite signal, which contains the requirement of multipliers for its implementation. As
upper and lower sidebands generated by Phase multipliers are resource intensive and performance
Modulation of DDOR tones onto downlink carrier, is limiting devices in several cases, CIC filter offers
passed through filters to separate out the sidebands. unique advantages when used in DDCs. Block
Each side band is a sinusoidal signal of different schematic of a DDC based on CIC filters is shown in
frequency. These sinusoids recovered at the two ground Fig. 4.
stations are correlated to obtain the phase difference.
This measurement provides the differential phase only
as a modulo of 2. Hence each of these
differential phases can differ by integral multiple of 2.
Four such signals forming the upper and lower
sidebands of two DDOR tones are used to resolve the
integral phase ambiguity of the received signals.
Block diagram of a Digital Down Converter is shown in
Fig.3. IF signal received from the analog front-end of a
receiver is fed to an Analog-to-Digital Converter
(ADC). The sampling frequency of the ADC is chosen
to be greater than twice the highest frequency Fig.4 Block diagram of a CIC Filter based DDC
component present in the analog signal. The digitized IF
signal is duplicated to form In-phase (I) and Quadrature III. CASCADED INTEGRATOR COMB FILTERS
(Q) signal streams that are fed to I and Q channels of
the DDC. An a priori knowledge of the IF signal is The bandwidth of a typical signal at the input of a
utilized for down conversion of IF signal to baseband. receiver is quite narrow compared to the IF frequency
A Numerically Controlled Oscillator (NCO) generating around which it is centered. When such a signal is
quadrature sinusoidal signals having frequency equal to subjected to low-pass sampling using a high
that of the IF signal forms part of the DDC. I and Q frequency ADC and is subsequently down-converted
streams of the digitized IF signal are heterodyned by the to baseband, as described in the earlier section,
quadrature sinusoids of the NCO. Heterodyning in results in the original narrow band signal being
92
centered around zero frequency. Hence, the sampling Fig.5 Block diagram of a 3 stage decimating CIC Filter
rate at this stage will be much higher than the Nyquist
criterion that needs to be satisfied. Reducing the The transfer function of a CIC filter at fs is given by the
sampling rate (Decimation) can lead to ease of discrete- equation:
time signal processing in the subsequent stages. The
ratio of IF low-pass sampling frequency and bandwidth
of the desired signal being very high, the process of
decimation can involve large rate change factors. Large
The above equation shows that even though a CIC has
rate changes require high order narrow band filters
integrators in it, which by themselves have an infinite
requiring fast multipliers. This, in several cases, can be
impulse response, a CIC filter is equivalent to N FIR
the bottleneck of the discrete-time system.
filters, each having a rectangular impulse response. As
Hogenauer devised an efficient way of performing
all the coefficients of these FIR filters are unity, and
decimation and interpolation using a flexible, ultiplier-
hence symmetric, CIC filter exhibits linear phase
free, linear phase filter suitable for hardware
response and constant group delay. For CIC decimators,
implementation that can handle arbitrary and large rate
the gain G at the output of the final comb section is
changes [1]. These are known as Cascaded Integrator
given by the equation:
Comb filters or CIC filters for short. CIC filters
G = (RM)N
performing decimation operation are only considered in
When two's complement arithmetic is used, the number
this paper as the requirement for interpolation does not
of bits required for the last comb due to bit growth can
arise for the application being considered. CIC filters
be calculated from the above equation. If B is the
consist of N cascaded stages of Integrators followed
number of input bits, then the number of output bits,
by Decimator and N cascaded stages of Comb
Bout, is given by:
sections as shown in Fig. 5 [2]. An integrator is a
single-pole IIR filter with a unity feedback coefficient
and is described by the equation: Bout bits are needed for each integrator and comb stage.
y[n] = y[n -1] + x[n] The input needs to be sign extended to B bits, but
A comb filter running at the high sampling rate, f, for a LSB's can either be truncated or rounded at later stages.
rate change of R is an odd symmetric FIR filter Frequency response of CIC filter for different
described by the equation: decimation factors and different number of stages are
y[n] = x[n] -x[n - RM] shown in Figs. 6 11.
In this equation, M is a design parameter called the
differential delay. M can be any positive integer, but it
is usually limited to 1 or 2.
A CIC filter can be built by cascading N integrator
sections together with N comb sections. However, this
filter can be made simpler by introducing a rate changer
before the Comb sections. The discrete-time output in
such a case would be given by the following equation at
the slower sampling rate:
fs2 = fs1/R
y[n] = x[n] - x[n - M]
By implementing the filter in this fashion the second
half of the filter following the decimator is operated at a Fig.6 Magnitude and Phase response of CIC filter
lower frequency. Also, the number of delay elements (R=8, N=1, M=1)
required for realizing the Comb section is reduced.
More importantly, the integrator and comb structures
are made independent of the rate change factor R.
Hence a CIC filter can be designed to have a
programmable rate change factor while retaining the
same filtering structure.
A CIC decimator consists of N Cascaded Integrator
stages clocked at f, followed by a Decimator of factor
R, followed by N Cascaded Comb stages running at fs2
= fs1/R as shown in Fig. 5.
Fig.7 Magnitude and Phase response of CIC filter
93
(R=8, N=2, M=1) (R=32, N=3, M=1)
IV. REALIZATION OF CIC FILTER BASED DDC

FOR RECOVERY OF DDOR TONE
The block diagram of CIC filter based DDC has already
been shown in Fig. 4. In this section, the results of
MATLAB simulation for CIC filter based DDC are
presented. The downlink signal from the spacecraft has
been assumed to be a sinusoidal carrier Phase
Modulated ( = 0.8) by a sinusoidal DDOR tone. This
signal is down-converted to IF and digitized. The
frequencies of the IF signal and modulating signal are
assumed to be 3.125 MHz (fc) and 195.3125 KHz (f)
respectively for demonstration of concept.
Fig.8 Magnitude and Phase response of CIC filter When PM modulated signal is heterodyned by sine and
(R=8, N=3, M=1) cosine signals at frequency equal to that of its carrier
frequency, the following analysis predicts the output of
the two heterodyne operations.
Fig.9 Magnitude and Phase response of CIC filter

(R=32, N=1, M=1)
From the above equations it may be noted that the

heterodyne operation to baseband results in fm, 3fm,
components in one case and 2fm, 4fm,
6fm,components in the other. Further analysis and
simulations have been carried out for the case where fm,
3fm, 5fm, components are obtained as the signal of
Fig.10 Magnitude and Phase response of CIC filter interest (DDOR tone) is at frequency f Fig. 12
(R=32, N=2, M=1) represents the spectrum of the PM signal centered at
3.125 MHz. Spectrum of the signal obtained by
heterodyning the PM signal with the NCO generating
3.125 MHz sinusoidal signal is shown in Fig. 13. The
output of the Integrator (1 stage) when the heterodyned
signal is fed is shown in Fig. 14. Fig. 15 shows the
output of Decimator for decimation factor R of 8. The
decimated output when fed to the Comb section (1
stage) results in signal having the spectrum as shown in
Fig. 16.
It can be seen from this figure that the PM modulating

tone of 195.3125 KHz (f) has been recovered at this
Fig.11 Magnitude and Phase response of CIC filter stage. All other frequency components are of much
94
lower power and are insignificant. The time domain

waveform of the recovered DDOR tone is shown in Fig.
17.
Fig.12 Spectrum of PM Signal containing fm, 2fm, Fig.16 Spectrum of Comb Output. 3fm component is
components centered around IF=3.125 MHz more than 30 dB down as compared to fm component.
Fig.13 Spectrum of Heterodyned Output containing fm,

3fm, 5fm, components centered around 0 Hz and
6.25MHz Fig.17 Time domain plot of the recovered DDOR tone
of frequency fm
V. RESULTS OF HARDWARE
IMPLEMENTATION OF CIC FILTER BASED
DDC
CIC filter based DDC was implemented in hardware
using Xilinx Virtex-5 130T device. Parameters chosen
for practical demonstration were same as those used for
MATLABsimulation. Fig. 18 represents the spectrum
of the PM signal centered at 3.125 MHz. Sampling
frequency of ADC is 12.5MHz. Spectrum of the signal
obtained by heterodyning the PM signal with the NCO
Fig.14 Spectrum of Integrator Output generating 3.125 MHz sinusoidal signal is shown in
Fig. 19. The output of the Integrator (1 stage)when the
heterodyned signal is fed is shown in Fig. 20. Fig.
21shows the output of Decimator for decimation factor
R of 8.The decimated output when fed to the Comb
section (1 stage)results in signal having the spectrum as
shown in Fig. 22. It can be seen from this figure that the
PM modulating tone of195.3125 KHz (f) has been
recovered at this stage. All otherfrequency components
are of much lower power and are insignificant.
Fig. 23 shows the recovered PM modulating tone
fordifferent test conditions. Sampling frequency of
Fig.15 Spectrum of Decimator Output ADC was chosen to be 50 MHz. Sinusoid of frequency
95
48.8 KHz was chosen to Phase Modulate carrier at

3.125 MHz. Decimation factor of 32 was chosen.
Fig.22 Spectrum of Comb Output showing recovered

Fig.18 Spectrum of PM Signal containing fm, 2fm, DDOR tone of frequency 195.3125 KHz.
components centered around IF=3.125 MHz (N=M=1;R=8)
Fig.19 Spectrum of Heterodyned Output containing fm,

3fm, 5fm, components centered around 0 Hz
Fig.23 Spectrum of Comb Output showing recovered

DDOR tone of frequency 48.8 KHz. (N=M=1;R=32)
VI. CONCLUSIONS
Receiver architecture for DDOR tone recovery has been

described. CIC filter characteristics have been imulated.
Recovery of DDOR tones using CIC filter based DDC
has been simulated and successfully demonstrated for
different decimation factors.
Fig.20 Spectrum of Integrator Output REFERENCES

[1] E. B. Hogenauer, An economical class of digital filters for
decimation and interpolation, IEEE Transactions on Acoustics,
Speech and Signal Processing, ASSP-29(2):155-162, 1981.
[2] Matthew P. Donadio, CIC Filter Introduction, 18 July 2000, For
Free Publication by Iowegian.
Fig.21 Spectrum of Decimator Output
96
Relative Characterization of Steered

Timescales for Traceability
Anu Arora1, Aakanksha Avnish Bhardwajan2, Suresh Dakkumalla3, A. Amala Agnes4
T. Subramanya Ganesh5, V. Jayaraman6, L. Srinivasan7
Navigation Systems Area,
ISRO TelemetryTracking and Command Network (ISTRAC)
Indian Space Research Organisation
Bangalore, India
1
anuarora@istrac.org
2
aakanksha_bhardwajan@istrac.org
3
dsuresh@istrac.org
4
amala@istrac.org
5
ganesht@istrac.org
6
jayaram@istrac.org
7
lsvasan@istrac.org
AbstractOne of the most important ground elements steering towards UTC, time transferbetween IRNWT
of a GNSS system is the Precise Timing Facility (PTF). and a standard metrological lab is carried out.
The timescale generated by the Precise Timing Facility Generally GNSS All-in-view (AV) technique is used
(PTF) is usually derived from an ensemble of Active for the time transfer for high accuracy. The offset of
Hydrogen Maser Clocks and High performance Cesium
the metrological lab with respect to UTCis obtained
Clocks.The time generated at the PTF needs to be
synchronized and steered towards Co-ordinated from Circular-T, a monthly bulletin published by the
Universal Time (UTC) for catering to various user Bureau of International Weigths and Measures
requirements and for traceability of GNSS time. For a (BIPM). Accordingly the relative offset of the GNSS
GNSS PTF, traceability to UTC is achieved by PTF with respect to UTC is worked out as follows
performing time transfers with national or international 6 offset values corresponding to UTC-
metrological time labs. For GNSS applications, the UTC(Met Lab) at 5 days intervals is taken from
offset of the PTF with respect to UTC or any other Circular-T provided by BIPM for the preceding
standard referenceneeds to be made available to the month. There values usually correspond to
user through satellite broadcast. Generally , the offset of
Modified Julian Dates (MJDs) ending with 4
the PTF with the national metrological lab and/or
UTC is provided in terms of bias (a0), drift (a1) and and 9.
drift rate (a2). This paper deals with the study carried The GNSS AV exercise provides the difference
out for mathematically characterizingaPTF between PTF and Met lab. This is derived as a
timescalewith respect to UTC and other metrological five day average spread 2.5 days on either side
standards. The applicability of a quadratic model for of the day with MJD ending either 4 or 9.
relative characterization of two steered timescales is The offset corresponding to UTC-UTC(PTF) is
discussed along with the results achieved. The optimum then be computed as a combination of the
number of days/data samples required to minimize the above two and can be described as follows:
prediction error in the offset value has also been
recommended. This work was carried out for Indian
Regional Navigation Satellite System(IRNSS). UTC-UTC (PTF) =
[UTC-UTC (Met Lab)] (Derived from Circular-T)
KeywordsPrecise Timing Facility, Traceability, Bias, [UTC (PTF)-UTC (Met Lab)] (Derived from
Drift, Drift Rate and Steering AV)
The offset of the PTF with respect to metrological
I. INTRODUCTION and UTC needs to expressedin terms of bias, drift and
drift rate and broadcast to the user through the
The timescale generated at the PTF is an ensemble of navigation data. This helps the GNSS user in
Active Hydrogen Maser Clocks and Cesium Clocks achieving traceability of GNSS time to international
which is being generated continuously to be used as a and metrological standards.The following sections
reference time scale for the GNSS. To achieve describe the details of the analysis carried out for
traceability towards UTC, the PTF is continuously characterszingthe GNSS PTF with respect to
steered to the reference timescale i.e. UTC. For metrological lab and UTC in terms of bias, drift and
97
drift rate. The results of the same have been provided obtained for the end user. Hence a quadratic model is
in subsequent sections. used for the relative characterizing.
CHARACTERIZATION OF STEERED TIMESCALES
The timescales generated at GNSS PTF andat the

metrological labsare generally steered towards UTC
and therefore the timescales may not express a linear
or a quadratic behaviour unlike unsteered
clocks.Therefore, the objective of the analysis was to
study and find out
The applicability of the quadratic model for

the relative characterization of steered
timescales
The optimum number of data (offset)
samples required to minimize the prediction
error.
Fig. 2 Quadratic fit to UTC-UTC (IRNWT)
B. Applicability of a quadratic model
It is seen that a linear or a quadratic model is

sufficient to characterize the bias, drift and drift rate
pertaining the phase differences between two free
running clocks say AHM or Cesium. Figure 1 below
illustrates the relative phase differences obtained
between two free running Active Hydrogen Masers.
Fig. 3 Cubic fit to UTC-UTC (IRNWT)
Fig. 1 Phase differences between two non-steered

clocks
The timescale generated at the GNSS PTF and the

metrological labs are continuously steered towards
UTC. Figures 2 to 7 reflectsa typical phase difference
profile between two steered timescales being
modelled with increasing order of polynomials.
For the purposes of easein modelling, and simplicity

of the user algorithm , a quadratic model is
considered for characterizing the two steered
timescales. Also it is seen that by increasing the
degree of the polynomial, no great advantage is
Fig. 4 a 4th order curve fit to UTC-UTC (IRNWT)
98
C. Estimation of bias, drift and drift rate for UTC

(GNSS PTF) - UTC(Met Lab)
The offset of GNSS PTF with respect to UTC (Met

Lab) is computed using the all-in-view technique on
thetime transfer measurements obtained from the
GNSS receivers located at both timing laboratories.
An analysis was carried out to estimate bias (a0), drift

(a1) and drift rate (a2) with an objective to find out the
optimum number of required samples which will
minimize the error in the future estimate. A second
order polynomial was fit in a least square sense to the
observations available and compared with the truth
data.In order to arrive at an optimum number of
samples that needs to be used for arriving at the best
Fig. 5 a 5th order curve fit to UTC-UTC (IRNWT) estimate, 4 different lengths of data sets were
analysed viz. 5, 10, 15 and 20 sample data set where
each sample represents the average of a day. Figure 8,
9, 10 and 11 below represents the quadratic fit
applied on the samples withdata sets of size 5, 10, 15
and 20 respectively.
Fig. 6 A 6th order curve fit to UTC-UTC (IRNWT)
Fig. 8 Quadratic fit applied of 5 samples
Fig. 7 A 7th order curve fit to UTC-UTC (IRNWT) Fig. 9 Quadratic fit applied of 10 samples
99
D. Estimation of bias, drift and drift rate for

UTC- UTC (IRNWT)
GNSS satellitesgenerally broadcast the time offset of

GNSS PTF with respect to UTC so that user can
compute the traceability to UTC. These time offsets
are required to be available in terms of bias, drift and
drift rate. An analysis was carried out to arrive at the
minimum number of samples required to compute
bias (b0), drift (b1) and drift rate (b2) such that the
future estimate of the same can be made by the user
with minimum prediction error.
A second order least square fit was applied to the
Fig. 10 Quadratic fit applied of 15 samples observations [UTC-UTC (GNSS PTF)] where each
sample is a5 day average and is made available on
MJD 4 or 9. The offset for a given day was predicted
using extrapolation and the so estimated value was
compared to the truth value obtained for that day to
compute the prediction error. To estimate b0, b1 and
b2, data set of different sizes with 10 samples (2
months), 20 samples (4 months) and 30 samples (6
months) were studied. Figure 12, 13 and 14 below
illustrates the quadratic behaviour of the offsets
computed using 10, 20 and 30 samplesrespectively.
The exercise was repeated many times using the data

spread over a year. Each time the prediction error was
found out by comparing the predicted value to the
truth. The RMS of the prediction errors was estimated
and considered as a figure of merit for sizing the data
set. Table 1 below presents the RMS of the prediction
error obtained for each size of data set considered.
Table 1: RMS of the prediction errors obtained for
different size of data sets
RMS of Prediction Errors

Size of Data Set
(ns)
5 Samples (Days) 4.79
It can be observed from above table that the offset

estimated using a data set of 10 samples was found to
be minimum and hence is receommended for use.
100
The exercise was repeated multiple times and each

time the prediction error was found by comparing the
predicted value to the truth. The RMS of the
prediction errors over all the experiments was
estimated and used as a figure of merit for that data
set. Table 2 below provides the RMS of the
prediction error obtained for the various sizes of data
sets that were considered.
Table 2: RMS of the prediction errors obtained for

different size of data sets
RMS of Prediction Errors

Size of Data Set
(ns)
10 Samples (~2
6.07
Months)
20 Samples (~4
4.75
Months)
30 Samples (~6
4.63
Months)
CONCLUSION
The behaviour of two steered clocks have been

analysed and results are provided. The above work
was carried out for IRNSS project for characterizing
IRNSS Network time with UTC. A study has been
carried out for estimating the future value of the
offsets UTC-UTC (IRNWT) in terms of bias, drift
and drift rate.A quadratic model was used for
characterszing the timescales. An optimumquantum
ofdata required to predict offsets has been worked out
and proposed and the results are presented.
101
Reconstruction Algorithms for Compressive

Sensing
T.J.V.Subrahmanyeswara Rao1, M.Venu Gopala Rao2
Research Scholar, Department of ECE, KL University, Guntur Dist, A.P, India.
Professor, Department of ECE, KL University, Guntur Dist, A.P, India.
tjvsrao@kluniversity.in, mvgr03@kluniversity.in
Abstract -- Compressive sensing is a technique of

sampling signals at a sub-Nyquist rate and reconstruct
them with small no of measurements, provided that the
original signal is sparse in a known basis. One of the
breakthroughs in CS is that linear programming methods
are used to efficiently reconstruct the signal with high
accuracy. Since then, matching pursuit algorithms and
Iterative thresholding algorithms have been proposed
which are faster or superior in terms of reconstruction
rate . Two of the Matching pursuit algorithms such as
Orthogonal matching pursuit (OMP) and Compressive
Sampling Matching Pursuit (CoSaMP) and one of the
Iterative Thresholding algorithms such as Modified
Frame Reconstruction (MFR) have been proposed.
Simulation results shows that the CoSaMP performs
better when the measurements are less than one fourth of
the signal length, similarly the MFR with least squares
performs well when the measurements are more than one
fourth of the signal length.
I.INTRODUCTION TO COMPRESSIVE
SENSING
Let f (t) be a signal with bandwidth B > 0. Then the

Shannon-Nyquist sampling theorem says that to
perfectly reconstruct this signal from its samples we
need to sample it at the Nyquist rate, equal to twice
the bandwidth of the signal, i.e. 2B. It can be shown II.METHODOLOGY
that by using compressed sens ing techniques a s
ignicantly lo wer s a mpling rate can be used, but 2.1Design of Measurement matrix and RIP
still effect perfect reconstruction for a large class of
signals. Consider a real valued, n ite length, one - The goal of compressive sensing is to design the
dimensional, discrete time signal x R . A higher and a reconstruction algorithm so that for ks parse
dimensional signal, such as an image, is treated as a signals only a small number of measurements are
one dimensional vector, by writing down its elements required, i.e. mk . In the designing of measurement
in someorder. The claim of compressive sensing is
that from m measurements where mn, it is possible matrix , it should not destroy any information
to perfectly reconstruct the original signal x where contained in the original signal x. However since
the measurements are chosen at random. R and m<n this is not possible in general as
the observations equation y is under-determined,
making the problem of solving for z or x ill-posed.
By restricting ourselves to k-sparse signals we can do

signicantly better. If the pos ition of the k non zero
entries of z were known a priori, i.e. if we were given
102
III.RECONSTRUCTION ALGORITHMS
3.1 L1 Mini mization Algorithm
103
Matching pursuit is a class of iterative algorithms that

decomposes a signal into a linear expansion of
functions that form a dictionary. At each iteration of
the algorithm, matching pursuit chooses dictionary
elements in a greedy fashion that best approximate
the signal.
3.2.1 Orthogonal Matching Pursuit (OMP)

Algorithm
Orthogonal matching pursuit (OMP) is an

improvement on matching pursuit. The principle is
the same, at every iteration an element is picked from
the dictionary that best approximates the residual.
However rather than simply taking the scalar product
of the residual and the new dictionary element to get
the coefficient weight, we t the original function to
all the already selected dictionary elements via least
squares or projecting the function orthogonally onto
all selected dictionary atoms, hence the term
orthogonal matching pursuit.
The running time of the OMP algorithm is dominated

3.2 Matchi ng Pursuit Algorithms by step 4 of above Algorithm, nding the maximally
correlated column of with the residual. By
104
3.3 Modi fied Frame Reconstructi on(MFR)

Algorithm
3.2.2 Compressive Sampling Matching
Pursuit(CoSaMP)Algorithm
An extension to orthogonal matching

pursuitalgorithms is the CoSaMP (Compressive
Sampling Matching Pursuit) algorithm. The basis of
the algorithm isOMP but CoSaMP can be shown to
have tighter bounds on its convergence and
performance.
Let m n be a measurement matrix thatobeys
the restricted isometry property of order 2s obeying Unlike for a basis, this representation is not
. Let Y=x+e be a measurement of the necessarily unique, as the x are not required to be
signal with error of arbitrary noise. Then for agiven linearly independent. Modied Fra me Recons
precis ion para meter , the CoSaMP algor ith m truction algorithm or MFR is motivated by the frame
produces an s-sparse vector that s atis es reconstruction algorithm.
105
This algorithm generalizes the iterative hard 3.3.2 MFR wi th Least Squares
thresholding (IHT) algorithm. The algorithm consists
of two parts, an update and a thresholding step. The MFR algorithm can be modified by stealing a
(a) Update: An update of the measurements is leaf from the orthogonal matching pursuit class of
performed as algorithms. Here the frame reconstruction algorithm
is used to provide column selection, by choosing the
columns of that correspond to the entries of x that
where y is the vector of measurements, is the have largest magnitude and then solve the least
measurement matrix and r is a control parameter squares problem on this set of columns. (k) Hence the
which is referred to as the step-length. idea here is to keep running the frame step until the
(b) Thresholding: The second part of the algorithm set of columns we select changes, and then perform
is the thresholding procedure where we generate the least squares. Convergence of this new algorithm is
next signicantly quicker than the plain MFR a lgorith m.
solution

Here we simply threshold the output of the frame step
producing an s-sparse approximation.
106
CoSaMP algorithm performs well when the

measurements are less than one forth of signalength,
and MFR algorithm is well when the measurements
are more than one forth of signal length. The
reconstruction rate can be improved bycombining
suitable algorithms with Model Based Compressed
Sensing. The principle of MBCS is that
thereconstruction process is enhanced by assuming
that the sparse coefficients have some extra
underlying structure related to their positions.
REFERENCES
[1] Cands, E.J.J. Romberg, and T. Tao, Robust uncertainty
Figure 4.1: Simulated results for several algorithms principles: exact signal reconstruction from highly incomplete
using matrix sizes with n = 400 and m = 50, 100, 150, frequency information. Information Theory, IEEE Transactions
and 200. on
52(2), 489509, 2006 Feb
[2] Cands, E.J.J. Romberg, and T. Tao, Stable signal recovery
from incomplete and inaccurate measurements. Communications
on Pure and Applied Mathematics 59(8), 12071223, 2006.
[3] Cands, E. J. and T. Tao, Decoding by linear programming.
Information Theory, IEEE Transactions on 51(12), 42034215,
2005 Dec.
[4] [2] D. Donoho, Comp ressed sensing, IEEE Transact ions on
Information Theory, vol. 52, no. 4, pp. 12891306, Apr.2006.
[5] R.G. Baraniuk, M . Davenp ort , R. DeVore, and M .B. Wakin,
A simple proof of the restricted isometry principle for random
matrices , Constructive Approximation, 2007.
[6] Cands.E. and J. Romberg. Sparsity and incoherence in
compressive sampling. Inverse Problems 23(3), 969985, 2007.
[7] Joel A. Tropp , Anna C. Gilbert , Signal Recovery From
Random Measurements Via Orthogonal Mat ching Pursuit . IEEE
transactions on information theory, vol.53, no.12, december 2007.
[8] D. Needell And J. A. Tropp , Cosamp : Iterative Signal
Recovery From Incomplete And Inaccurate
Samples,arXiv:0803.2392v2 [math.NA] 17 Apr 2008.
[9] Graeme Pop e, Modied Frame Reconst ruct ion Algorit hm
for
Comp ressive Sensin g, arXiv :0906.1079v1 [cs.IT ] 5 Jun 2009.
[10] R. G. Baraniuk, V. Cevher, M . F. Duarte, and C. Hegde,
Model-based comp ressive sensing, 200 . Prep rint . Availab le at
http://dsp.rice.edu/cs.
[11] Mallat, S. G. and Z. Zhang , Matching pursuits with
timefrequency dictionaries. Signal Processing, IEEE Transactions
on
41(12), 33973415, 1993, Dec.
Figure 4.2: Simulated results for several algorithms

using matrix sizes with n = 800 and m = 100, 200,
300, and 400.
V. CONCLUSIONS AND FUTURE SCOPE
In this paper various reconstruction algorithms

forCompressive Sensing were studied. The work has
been started from designing the measurement matrix.
First theraditional minimization algorithm was
discussed andater extended to far superior Matching
pursuit and Iterative thresholding algorithms. It was
concluded from thesimulation results that the
107
Design of a Readout Circuit for MEMS Based

Seismometer
V Thamarai, Etika Agarwal, Gaurav Dutta Saxena, Kalpana Arvind, A S Laxmi Prasad, Dr G Nagendra Rao
Laboratory for Electro-Optics Systems, (LEOS)
1st cross, 1st stage, Peenya Industrial Estate,
Bangalore-560058
thamarai@leos.gov.in
Abstract: ISROs Chandrayaan-2 lander carries a

output comparable to the conventional seismometers.
seismometer payload for measurement of lunar seismic But the main advantages of using such a sensor for
signals. It is proposed to use the in-house fabricated measurement of seismic signals lie in their small size,
Micro Electro Mechanical Systems (MEMS) based weight and reduced power consumption.
capacitance accelerometer sensor as seismometer for
measurement of lunar seismic signals. This paper reports This paper reports on the design of a high sensitive
on the design of a front end circuit for detection of lunar front end circuit, the various capacitance sensing
seismic signals. This paper also details the requirements methods and a comparison between the different
of the front end amplifier, the various capacitance sensing schemes for detection of seismic signals along with
methods tried out and brings out a comparison between
the different schemes along with their merits and
their merits and demerits. The paper also details the
demerits. criticalities and design requirements of the front end
amplifier for the given application.
I. INTRODUCTION II. SENSOR STRUCTURE
Micro Electro Mechanical Systems (MEMS) Bulk micro machined accelerometers developed by
technology enables fabrication of miniature, low LEOS, ISRO are based on capacitance detection
power and high performance sensors. The scheme. The schematic of the MEMS accelerometer
significance of this sensor is the fact that by sensor structure is as shown in Fig.1.
integrating the output signal, an accelerometer can
also provide a measure of velocity and position. The
in-house fabricated capacitive MEMS based
accelerometer consists of two fixed electrodes and a
seismic mass acting as a movable electrode. The three
electrodes together form a differential capacitor
assembly. A change in acceleration causes the
seismic mass to move and causes a differential
change in capacitance. It is proposed to use this
accelerometer as seismometer for measurement of
lunar seismic signals by integration of the Figure.1 Typical capacitive MEMS accelerometer
acceleration output. The sensitivity of the in-house sensor structure
fabricated sensor is around 6-8pF/g. The sensor is
proposed to be packaged separately and interfaced The accelerometer consists of a spring-mass
with the discrete readout electronics in printed circuit system also called as seismic mass acting as a
board form. Detection of lunar seismic signals movable electrode in a differential capacitor
involve measurement of very low magnitude and low assembly. Seismic mass is inter-digitized with fixed
frequency acceleration signals [1] leading to electrodes on both sides of a capacitor assembly.
measurement of low capacitances of the order of 1 These capacitors also called as the nominal
ppm of the nominal capacitance (few atto farads (10- capacitance are equal where there is no acceleration.
18
F) change in a nominal capacitance of 10pF).
Conventional seismometers are generally bulky and A small change in the relative position of these
have a very large mass resulting in an instrument electrodes due to an external acceleration, changes
noise well below the seismic noise. MEMS based the distance between the proof mass electrodes and
sensors due their low weight have relatively larger the fixed electrodes which can be transduced as a
self noise and also require integration to give an change in capacitance as given in Equation (1)
108
of 90Hz. The maximum measurable input range of

(1) the IC is fixed as 4pF which corresponds to a
resolution of 0.24 fF (10-15F). The IC can not be
Where N is the number of fingers per side, 0 is directly used for measurement of seismic signals as
the permittivity of space, h is structural thickness, this requires measurement of change in sensor
Loverlap is the overlap length of fingers, d1 is the gap capacitances of the order of aF (10-18F). The IC does
spacing and x is the proof mass displacement. This not have any provision for adding gain to shift the
capacitance change is converted in to an equivalent dynamic range to resolve low level signals and hence
voltage change through suitable electronics. The can not be directly used for the given application.
sensitivity of the in-house fabricated MEMS sensor is Capacitance to Voltage Converter
around 6-8 pF/g.
One of the proposed configurations for a pre-
III. DESIGN CONSIDERATIONS amplifier is a capacitance to voltage converter
(CVC). Fig.2 shows the basic configuration of a
The key concern while designing MEMS based capacitance to voltage converter consisting of sensor
system for sensing low strength seismic signals is the capacitor at the input of a low noise high impedance
systems ability to resolve vibrations from noise. amplifier connected in inverting configuration with
System noise sets a limit to the level of detection. capacitor and switch in the feedback.
System noise is due to both the electronic noise and
the mechanical noise of suspension. Since the sensor
is proposed to be packaged separately and interfaced Cf
with the discrete readout electronics in printed circuit
board form, care is required in the printed circuit
board layout and interfacing of sensor to electronics
to minimize the noise due to external interference. Vexc
Sensor -
The design of the front end amplifier has to ensure +
minimal intrinsic noise to keep system noise below
the seismic noise in its range of operation. Capacitive
sensors are basically high impedance sensors. This
combined with the low noise requirement makes pre
amplifier selection critical for the given application. Figure.2 Basic configuration of a Capacitance to
The pre amplifier should avoid any loading of the Voltage converter
sensor output and hence requires FET amplifiers at
the front end. The high excitation frequency Switch provides path for the dc bias currents and
requirement of the capacitive sensor calls for a high discharges the feedback capacitor, thus avoiding the
bandwidth front end amplifier. High bandwidth saturation of the amplifier. The sensor is excited by
combined with the large voltage noise of the FET applying high frequency square wave signal. The
amplifier limits the noise performance of the system. signal output of the capacitance to voltage converter
is directly proportional to the excitation voltage and
IV. DIFFERENT SENSING sensor capacitance and inversely proportional to the
METHODOLOGIES feedback capacitor as given by equation (2)
Many methods are reported for capacitance to

voltage conversion. Most methods are either bulky (2)
are not capable of handling very small signals
expected from seismic vibrations [3]. This paper Where Vexc is the excitation voltage, Cin is the
details the various methods which are easy to realize sensor capacitor and Cf is the feedback capacitor.
with discrete components. Capacitance to Voltage converter can be configured
in two different ways depending on the method of
Capacitance to Digital Converter IC connecting sensor to the capacitance to voltage
converter..
The usability of a commercially available capacitance Capacitance to Voltage converter with sensor in
to digital converter IC was initially explored for differential configuration:
detection of low sensor capacitance changes.
Capacitance to digital converter IC with 15 ENOB is In this method, the fixed electrodes of the sensor
commercially available with a maximum update rate are connected to excitation signals and the movable
109
electrode is connected to the capacitance to voltage tapped from sensor fixed electrodes. The output from
converter. The capacitance to voltage converter is each of the Capacitance to voltage converter is
configured to give an output proportional to the proportional to the individual capacitor values as
change in capacitance C as given in the equation (3) given in the equation (4) and not the difference as in
the previous scheme and requires further processing
to obtain an output proportional to the difference in
capacitance value.
(3)
Where Vexc is the excitation voltage, C is the
change in the sensor capacitane value and Cf is the
feedback capacitor. (4)
MEMS accelerometer sensor being differential in Where Vexc is the excitation voltage, C0 is the
nature, the excitation signal can be applied to top and sensor dead capacitance value and Cf is the feedback
bottom fixed electrodes of the sensor and the sensor capacitor.
movable electrode can be connected to the
capacitance to voltage converter. Fig. 3 shows the Cf
capacitance to voltage converter with sensor
connected in differential configuration.
-
+
Vexc Cf Vex
c
Cf
-
+ -
+
Vexc
Figure.4 Capacitance to Voltage converter with
sensor in single ended configuration
Figure.3 Capacitance to Voltage converter with
sensor in differential configuration The main advantage of this scheme is that this
scheme requires a single excitation source and being
For fixed excitation voltage and sensor sensitivity, a differential scheme any common mode noise or
gain of capacitance to voltage converter is limited by offset can be cancelled in further differential
the feedback capacitor used. Smaller the feedback processing. The main drawback of this scheme is that
capacitor higher is the gain. But, the change in sensor capacitance to voltage converters sensitivity is
capacitance is of the order of few aF and hence the limited as it is not possible to increase the gain of the
feedback capacitance has to be comparable in value circuit by increasing the excitation voltage or
to the change in sensor capacitance to have an reducing the Cf as this can cause saturation of the
appreciable output from the capacitance to voltage front end capacitance to voltage converter.
converter. Operational amplifiers internal
capacitance, PCB stray capacitance and availability The other option is to reduce the sensor dead
of small capacitance values in discrete form pose a capacitance value. But achieving a higher sensitivity
limit on Cf and hence the achievable gain. sensor with a low dead capacitor value is a
challenging task from sensor fabrication point of
Capacitance to Voltage converter with sensor in view. Another problem associated with this circuit is
single ended configuration: that the imbalance in the sensor capacitance due to
fabrication tolerances can not be balanced by adding
In this method, the capacitance to voltage converter balancing capacitors in parallel.
is configured to give an output proportional to the Buffer
sensor nominal capacitance. In this mode of
operation, two separate Capacitance to voltage Another simple configuration for converting the
converters are used for sensing of top and bottom change in sensor capacitance values to equivalent
sensor capacitors as shown in Fig 4. Excitation is fed voltage is by providing an Operational Amplifier
to the movable electrode of the sensor and output is buffer at the front end as shown in Fig 5.
110
Vex
c
-
+ R
Vexc 1
+
-
R
+ Vex
- 2 R
c
3
Figure.5 Opamp Buffer in the front end

Figure.6 Voltage Amplifier in the front end
Buffer prevents the loading of the sensor output by
providing the high input impedance required by the IV. SIMULATION AND RESULTS
capacitive sensor but does not provide any gain for
the signal input. The output of the buffer is Simulation of the full system consisting of the
proportional to the applied excitation voltage and the capacitive MEMS sensor, different front end sensing
ratio of sensor differential capacitance to total schemes and further processing blocks was done in P-
capacitance of the sensor as given by the equation (5) Spice. The complete behaviour of the system in terms
of linearity, sensitivity and SNR were analyzed.
Theoretical noise calculations were carried out for all
the above configurations. The configuration with
(5) Voltage amplifier at the front end was giving the best
signal to noise ratio of all the schemes. This front end
Where Vexc is the excitation voltage, C is the with the further processing stages and mathematical
sensor differential capacitance and C1 and C2 are the model of the sensor was simulated as a full system in
sensor dead capacitance values. Low sensitivity of P-spice. Fig 7 shows the linearity plot of the full
this configuration combined with the higher buffer system as a function of output voltage verses input
bandwidth results in a low level signal submerged in acceleration.
noise.
Voltage amplifier
This configuration is similar to the buffer

configuration except that a non-inverting band width
limited voltage amplifier is used as a pre-amplifier
for providing the gain as shown in Fig. 6. Output of
this amplifier is proportional to the ratio of sensor
differential capacitance to total capacitance of the
sensor multiplied by the gain of the voltage amplifier
and is as given by the equation (6) Figure.7 Linearity plot of the full system
VI. CONCLUSION
(6) The main consideration in the design of a MEMS

based system for sensing low strength seismic signals
Where Vexc is the excitation voltage, C is the is the systems ability to resolve vibrations from
sensor differential capacitance and C1 and C2 are the noise. It is not true that any small signal can be
sensor dead capacitance values. Use of non-inverting resolved by arbitrarily providing high amplification
amplifier in the front end increases the signal strength by cascading of gain stages. The noise of the system
and limitation of bandwidth increases the signal to poses a limit on the smallest signal that can be
noise ratio by reduction of the noise of the front end sensed. If a signal is so small that it is masked the
circuit. noise in the amplifier, it may become impossible to
recover the signal by amplification. Mechanical noise
of the sensor structure can be reduced by increasing
111
the seismic mass and by vacuum packaging of the

sensor. Low interference or external noise is achieved
by proper layout and packaging. Low intrinsic
electronic noise is achieved by proper electronics
design and by selection of suitable electronic
components. Various low noise front end circuits
were simulated in P-spice and the various intrinsic
noise sources including voltage, current noise of the
amplifiers and Johnson noise of resistors in the
system were calculated theoretically. The full circuit
is proposed to be fabricated in printed circuit form.
The test results of the circuit interfaced with the
MEMS sensor will be compared with the simulation
results to know the systems sensitivity.
Acknowledgement
The authors thank Sri. J A Kamalakar, Director

LEOS and Dr C L Nagendra, Associate Director,
LEOS for their encouragement, constant support and
guidance throughout the work. The authors also thank
MEMS Division, LEOS for their constant support in
providing MEMS accelerometer sensor for testing.
REFERENCES
1. ILN Final report NASA Jan2009

2. Joost C. Lotters, Wouter Olthuis, Peter H. Veltink
and Piet Bergveld, A Sensitive Differential
Capacitance to Voltage Converter for Sensor
Applications, IEEE Transactions on Instrumentation
and Measurement, Vol. 48, No. 1, February 1999
3. A.H.M. Zahirul Alam, Nurul Arfah, Sheroz Khan and
Md. Rafiqul Islam, Design of Capacitance to Voltage
Converter for Capacitive Sensor Transducer,
American Journal of Applied Sciences 7 (10): 1353-
1357, 2010
4. Paul Lee, Low noise Amplifier selection guide for
Optimal Noise Performance Analog Devices
Application Note
5. Ali Heidary and Gerard C M Meijer, An integrated
interface circuit with a capacitance-to-voltage
converter as front-end for grounded capacitive
sensors, 2009 Meas. Sci. Technol. 20 015202
6. W.Marshall Leach, Jr., Senior Member, IEEE,
Fundamentals of Low Noise Analog Circuit Design
112
A Survey on Poly House Cultivation Using

Wireless Sensor Networks
Prathiba Jonnala
Abstract This papers focus is primarily on good solution. WSN provides advantage in cost, size,
providing an overview of automated polyhouse. This complexity, power etc. Environmental monitoring
survey is mainly concentrated on the study of using wireless sensortechnology has become
automated polyhouse with regard to the sensors that
more and more important in todays world. WSN
are needed to be used for effective monitoring of
various climatic parameters that exist in a polyhouse carved a niche for itself among the galaxy of
environment, which in turn produce accurate results technologies and has become the mostly used one.
while saving energy and to develop a low cost One of its applications lies in agriculture also. WSN
automated polyhouse for commercial use. This review is used for monitoring the above said parameters that
also gives a projection on different wireless exist in the polyhouse via the sensors.
technologies that can be deployed for developing
an automated polyhouse and emphasize on
system integration and controlling various parameters.
The communication between the sensors unit and
control unit can be achieved using different wireless
sensor network technologies. The sensors unit collects
the data related to the environment in the polyhouse
using different sensors located in different parts of the
polyhouse and communicates with the control unit via
WSN protocols.
I. INTRODUCTION Fig. 1. Polyhouse [2]
The different wireless sensor network technologies

POLYHOUSE is a protective environment made up help in order to maintain the successful
of polythene sheets of different thickness used communication between the sensors unit and the
for growing plants as shown in Fig. 1. Polyhouse control unit. The information about the inside
provides a controlled environment for growing environment of polyhouse is gathered by the sensors
different varieties of plants. The growth of the unit which is situated in various locations of the
plant inside the polyhouse is independent of the polyhouseand that information is shared with the
season and location. Polyhouse protects the plant control unit using WSN protocols.
from unusual weather fluctuations and various This paper is organized as follows. Section II gives
plant diseases. Plants require suitable environment an overview on different climatic parameters and
for its proper growth and productivity. The different sensors that can be used. Section III deals
polyhouse provides optimum conditions for the about different WSN protocols that can be used.
plants to grow, which in turn increases the crop Section IV provides the overview of system
yield. In automated polyhouse the crop production Integration. Section V presents the conclusion of
is solely dependent on the controlled polyhouse this paper.
environment. Continuous m o n i t o r i n g a n d
c o n t r o l l i n g o f different climatic parameters II. PARAMETERS AND SENSORS
helps in the proper growth of
the plant and results in a high yield [1]. The Automated polyhouse system is a complex
Some of the important climatic parameters that system in which any change in any one of the
require attention are temperature, humidity, soil parameter affects the other parameters and influences
moisture and light intensity inside the polyhouse. In the plant growth and crop production. As discussed
order to maintain proper health conditions of a plant earlier continuous monitoring and controlling of
these conditions are needed to be monitored on a different climatic parameters helps in the proper
regular basis. The monitoring of the above said growth of plant and results in high yield. Various
parameters, need a large number of sensors and long parameters that can be monitored and controlled in
wires which makes the system complex and the polyhouse environment are Temperature,
expensive. To avoid the complexities in the Humidity, Soil Moisture content, Light Intensity and
system wireless sensor networks are found to be a carbon dioxide levels present in the air in the
113
polyhouse [3]. The crop growth inside the

automated polyhouse mainly depends up on
environment/climate inside the polyhouse and
soil conditions like moisture content in the soil. So,
in order to maintain proper growth of the plant and
increase the crop yield the above said parameters
needs proper monitoring and controlling. The
monitoring of all these parameters requires different
sensors. The growth of the plant mainly depends up Humidity is the presence of moisture content in the
on the surrounding environment in which the plant is air.
The amount of water vapor content in the air in a
grown. These parameters play a vital role in the
polyhouse influences the plant growth and its
plant growth and productivity [4]-[5]. Each
productivity. Humidity measurement determines the
parameter has its own effect on the plant and its amount of water vapor present in the air. Humidity
growth. The proper understanding of these effects on monitoring and controlling is of paramount
the plant growth will help in the detection and importance in an automated polyhouse for plant
prevention of different serious problems. protection. The raise in the humidity levels in the
A. Temperature Sensors polyhouse results in many fungal diseases like
Septoria leaf spot, Late Blight, Leaf mold,
Anthracnose etc. To prevent fungal diseases like
these continuous monitoring and controlling of
Humidity should be done. The different Humidity
sensors available in the market are Resistive
Humidity Sensor, Capacitive Humidity Sensor shown
in Fig. 3 and Thermal Conductivity Humidity Sensor.
The comparison on different attributes is reflected in
the Table 2.
Fig. 3. Capacitive Humidity Sensor [10]
Fig. 2. IC Temperature Sensor [9]

C. Light Intensity Sensors
There are different types of Temperature sensors
Sunlight plays an important role in the plant growth.
available in the market. Some of the widely used
It
temperature sensors are Platinum RTD,
helps in the process of photosynthesis. The 400 nm
Thermocouple, Thermometer, Thermistor, Infrared
700 nm wave length range is called
Thermometer, IC Temperature Sensors. Each sensor
photosynthetically Active Radiation. The abnormal
has its own advantages and disadvantages based on
changes in the light intensity results in diseases such
the application it is used for. The abnormal changes in
as Botrytis, Fusarium etc [6]-[8]. Various light
the temperature results in Chilling Injury, Freezing
sensors are available in the market that can be used
Injury, Pollen development and Mineral Nutrition
for monitoring the light in a polyhouse to prevent the
[6]-[8]. In order to avoid the above said problems
occurrence of such diseases. Some of the types of
Temperature need to be monitored and controlled.
light sensors that can be used in polyhouse are LDR,
Here is the comparison on different attributes of all
Photometric Sensor, IR Sensor and Quantum Sensor.
these sensors which help in giving the basic idea
Quantum Sensor shown in Fig. 4 is widely used in
about which sensor to choose. Thermocouple and
agriculture and environmental Industries. Here is the
RTD are best suited for Industrial applications where
comparison on different attributes of all these
as Thermistor and IC Temperature Sensor as shown
sensors.
in Fig. 2 are best suited for agriculture applications.
B. Humidity Sensors
114
III. WIRELESS SENSOR NETWORK
Now a days environmental monitoring using

wireless sensors technology has become more
important. A wireless sensor network consists of a
base station, number of wireless sensor nodes and a
control station. Wireless sensor nodes collect the
information regarding the environment in the
polyhouse. Base station analyzes and process the
information received from each sensor node. The
control unit takes the required control action based on
the collected data from the sensors and controls the
polyhouse environment [1]. There are many wireless
technologies available in the market for this
Fig. 4. Quantum Sensor [11] application which provides flexibility at low cost and
less power consumption. Some of the wireless
technologies are Bluetooth, RFID, NFC, Wi-Fi,
WiMax and Zigbee etc [4]-[5]. Out of the above
mentioned technologies Zigbee, Wi-Fi and Bluetooth
technologies are widely used in agriculture
applications [13]. Based on the survey conducted
Zigbee is proved to be the preferred wireless
technology because of its uniqueness in comparison
Moisture content is the amount of water present in the
to the other technologies.
soil. Surface soil moisture is the water that is present
in the upper 10 cm of soil, where as root zone soil
IV. SYSTEM INTEGRATION
moisture is the water content that is present in the
upper 200 cm of soil. Using Soil moisture sensor,
A system is needed to be designed before the
only the moisture in the top few centimeters can be
integration. Requirements of the system need to be
detected.
addressed first. The system should be able to monitor
more number of parameters with less power
The abnormalities in the soil moisture content results
consumption. The system consists of wireless sensor
in diseases such as Scabies etc [6][8]. Soil moisture
nodes which can be used to collect the environmental
content plays a vital role in the plat growth and crop
data in the polyhouse. The system should allow the
yield. Frequency domain sensor, Neutron moisture
communication between the base station and the
gauge shown in Fig. 5 and Time domain
various control unit modules that are placed at
reflectrometry are some of the sensors that can be
various parts in the polyhouse. The system should
used to gauge the soil moisture content. Here is the
have the flexibility for easy relocation. The base
comparison of various
station should be able to collect the information from
attributes of soil moisture.
the sensors and should be capable of analyzing and
processing the data collected. The wireless Network
protocol that is selected will enable the
communication between the base station and the
control unit. Hence, the control unit takes the
necessary action to control the unusual conditions in
the polyhouse.
V. CONCLUSION
This survey paper provided brief information

regarding
automated polyhouse like various parameters that
need to be monitored and controlled and different
wireless sensor network technologies that can be
Fig. 5. Neutron Gauge Meter [12]
used. The abnormalities in temperature, light
115
intensity, soil moisture and humidity should be [7] Fungal Diseases, http://www.ag.arizona.edu/.
[8] George. N.Agrios, Plant Pathology, Elsevier Science, January
monitored and controlled to prevent the polyhouse
2005, pages: 256-257.
plants from various diseases. The data about the [9] Texas instruments IC Temperature sensor,
environment in the polyhouse can be periodically http://uk.farnell.com/texas-instruments/tmp102aidrlt/ic-
collected from various sensors that are located in the temperaturesensor/dp/1652477
[10] Honeywell sensing hch-1000 series product sheet,
polyhouse and can be controlled by taking required
http://www.digikey.com/product-
control action by the control unit. The future scope of search/en/sensorstransducers/moisture-sensors-humidity/1966708
this survey lies in the [11] LI-COR LI-192 Underwater PAR Sensor,
implementation. http://www.fondriest.com/li-cor-li-192-underwater-par-sensor.htm
[12] Hanson, B., and G. Dickey. "When measuring soil water
content, field practices affect neutron moisture meter accuracy."
REFERENCES California Agriculture 47, no. 6 (1993): 29-31.
[13] Sengunthar Gayatri R, Ekata Mehul, A SURVEY ON
[1] Quan, Vu Minh, Gourab Sen Gupta, and Subhas GREENHOUSE AUTOMATION SYSTEMS, In International
Mukhopadhyay. Journal of Engineering & Science Research (IJESR), Feb 2013,
"Review of sensors for greenhouse climate monitoring." In Sensors Vol.
Applications Symposium (SAS), 2011 IEEE, pp. 112-118. IEEE, 3, no. 5, pp: 2338-2343.
2011.
[2]Polyhouseconstructionmodel,
http://agribusinessblog.wordpress.com/tag/polyhouse/
[3] C. Arun, K. Lakshmi Sudha. Agricultural Management using
Wireless Sensor Networks - A Survey. In 2nd International
Conference on Environment Science and Biotechnology, 2012
ICESB,
pp. 76-80. DOI: 10.7763/IPCBEE. 2012. V48.
[4] Park, Dae-Heon, Beom-Jin Kang, Kyung-Ryong Cho, Chang-
Sun
Shin, Sung-Eon Cho, Jang-Woo Park, and Won-Mo Yang. "A
study on greenhouse automatic control system based on wireless
sensor network." Wireless Personal Communications 56, no. 1
(2011): 117130.
[5] Kang, BeomJin, DaeHeon Park, KyungRyung Cho, ChangSun
Shin, SungEon Cho, and JangWoo Park. "A study on the
greenhouse auto control system based on wireless sensor network."
In Security Technology, 2008. SECTECH'08. International
Conference on, pp. 41-44. IEEE, 2008.
[6] Agro meteorology-Temperature, Relative Humidity,
http://www.agritech.tnau.ac.in/.
116
Performance of Cooperative Wideband

Spectrum Sensing and Localization Using
Radio Frequency Sensor Networks
P.Dhana Lakshmi
Asst.Professor in ECE
ANU College of Engineering &Technology
Guntur
dhanalakshmimails@gmail.com
K.Murali
Asst.Professor in ECE
Vijaya Institute of Technology for Women
Enikepadu, Vijayawada
kalipindimurali@gmail.com
Abstract-Spectrum sensing and localization are two reasoning capability to automatically disable its radio
important tasks in electronic warfare, signal frequency functions.
intelligence and cognitive radios. In cognitive radios,
unlicensed users implement spectrum sensing[1] and 2. Problem Formulation
localization not to interfere with licensed users.
Cooperation in spectrum sensing and localization In this section, we first present the general model for
improves the signal detection and position estimation channel sensing, and then review the energy detection
performance under fading, shadowing or noisy scheme, radiometry and cyclo stationary feature
conditions. detection and analyze the relation between the
This paper implements spectrum sensing and probability of detection, probability of miss detection
localization tasks using a radio frequency sensor [2]and probability of false alarm and three-bit hard
network and analyze the performance of this combination scheme is superior to the traditional hard
implementation through simulation. combination schemes in terms of false alarm
reduction.
Key words: spectrum sensing, cognitive radio,
sensor network, EIRP, localization scheme. 3. System Model and Background
1. Introduction 3.1 General Model for spectrum sensing
The terms "Cognitive Radio" and "Dynamic The goal of spectrum sensing is to determine if a
Spectrum Access" arise from recent technological licensed band is not currently being used by its
and regulatory trends in the area of wireless primary owner. This in turn may be formulated as a
communication. A "radio" is any device that can binary hypothesis testing problem
receive and transmit radio frequency signals, Two conditions are possible under hypothesis test
everything from cell phones to WiFi-enabled 1. When the primary user is not active, the received
computers. Recent advances in radio technology now signal at the secondary user can be represented as
allow radios to be "agile" in the sense that they can y n u n 1
reconfigure themselves to dynamically receive and
transmit on a wide variety of channels using a variety Where y n is the signal received by the
of communication modes.
secondary user and u n is noise.
A cognitive radio is a radio that has some ability to
represent and reason about goals, policies, features of 2. When the primary user is active, the received
the external world, aspects of its internal state, etc., signal is given by
that relate to radio communication. For example a
y n h n s n u n 2
cognitive cell-phone, upon boarding an aircraft, could
use its policy knowledge, context awareness, and
117
Where s n the signal is transmitted by the primary (2) Probability of false alarm, f , defines at the
users and received by secondary users over a channel hypothesis 0 , the probability of the sensing
h n . When the channel is non-fading, h n is algorithm claiming the presence of the primary
signal. That is, if we decide 1 , but is true, it is
constant. It is assumed that noise samples u n are
called a false alarm error. In Fig. 2 (b), the PDFs
independently and identically distributed (i.i.d.) with exceeding the threshold under the hypothesis 0 are
zero mean and variance u n u based on
2 2
defined as f .
the received signal [4]
3.2 Theory of Hypothesis Testing 3.3 Cogntive Channel Model
The goal of spectrum sensing is to make a decision, As a preliminary, we set up the model for signal
i.e. to choose between 0 and 1 , based on the detection. Consider a time continuous signal t .
received signal [5]. We wish to express the time-continuous signal in a
discrete vector representation over a finite time
The probability density function (PDF) under each interval. In general, the signal can be expressed by a
hypothesis is shown in Fig. 1(a) and in Fig. 1(b), basis expansion
where the threshold value for each hypothesis is
denoted as. t xii t 3
i 1
Where i t basis functions then the signal are

t can be represented by the vector
x1 , x1 ,......x

Assume that y is a received vector of length N, that

Fig. 1 (a) Theory of hypothesis testing; probability of consists of a signal plus
detection and probability of missed detection noise. That is
yxw 4
First we detect whether there is a signal present or not
we provide the separation between 0 and 1
y w 0 : signal is absent
5
y x w:1 : signal is present
The optimal Neyman-Pearson test is to compare the

Fig. 1 (b) Theory of hypothesis testing; probability of log-likelihood ratio to threshold i.e.
false alarm
y
1 1
e log
Generally, two probabilities are of interest for
(6)
y
indicating the performance of a sensing algorithm.
(1) Probability of detection d , defines the 0 0
probability of the sensing algorithm having detected This ratio depends on the distribution of the signal to
the presence of the primary signal at the hypothesis be detected
1 .Thus, in Fig. 1(a), under the hypothesis, the Note: No cyclic prefix c 0 (energy detection) so
PDFs bigger than the threshold value u is defined as that there is no structure in the signal that can be used
the detection probability. The PDFs smaller than the
threshold u is defined as probability of missed then
detection, m .
Under 0 ,
y C 0, 2
0
118
Under 1 ,
y
1
C 0, n2 s2 samples. The energy detection should be carried out
over all logical channels defined by the CR network.
Assuming that the channel is time-invariant during
Therefore log-likelihood ratio is
the sensing process, the energy detection on the given
1 y
2
channel is performed by accumulating the energy of
y exp 2 2

n s
samples and comparing it with the predefined
n2 s2
log 1 log 7 threshold, to decide whether signal is present or not
y y
2
1 [7].
0 exp 2

2
n n
5. Minimizing Sensing Time for Detection
By removing all constraints that are independent of
the received vector y, we obtain the optimal Neyman- In order to minimize the sensing time an algorithm
Pearson test has been proposed, whose flow chart is shown in
1 2 Figure 3. According to this algorithm there are three
e y yi e 8
1
2
possible states for the output of each detection
technique i.e. Low L, Medium M and High H. If
i 0 0
its output is H; it indicates the presence of primary
In this case the energy detector is also known as user, if its output is L; it means that primary user is
radiometer [6] not present. If the output is M then detection
4. Local Spectrum Sensing technique is not sure about the presence or absence of
The performance of a given spectrum sensing scheme primary user[8] The received signal is first passed
is fundamentally limited by the radio propagation through Matched Filter, as it takes least time for
channel. Typically, the effects of a radio channel can sensing among all three mentioned techniques. If
be divided into three main parts: path loss, small- output is H or L then its fine that we concluded
scale fading, and large-scale fading (shadowing). about presence or absence of primary user.
Path loss effects are incorporated in the received SNR
at a cognitive radio terminal. Small-scale fading
causes rapid, random variations in the signal strength
at the CR receiver and is modeled by Rayleigh fading
in this paper. Shadowing is the slow variation of
received signal power as the cognitive radio moves in
and out of the shadow of large structures like
mountains, buildings, and so forth. Shadowing is
often modeled as a lognormal distributed random
process that varies around a local mean given by the
path loss and with the standard deviation dB which
depends on the environment
Figure 3 Algorithm for minimizing sensing time for

detection
If output is M then we have to go for some other

technique. As Energy detection technique takes less
Fig 2: The considered scenario sensing time compared to Cyclostationary feature
detection. Then signal is passed through energy
The local spectrum sensing is accomplished by the detection filter and if its output is L or H then
energy detection [7]. An energy detector is again there is no need to go for another detection
implemented at each secondary user by calculating a technique. Finally if energy detectors output is M
decision metric out of all samples and antennas used. then go for Cyclostationary feature detection. If its
The purpose of energy detection is to make a correct output is H or M then we said that primary user is
decision between two hypotheses after observing present, otherwise primary user is not present.
119
6.Proposed Methods of Spectrum Sensing problems Radiometry becomes prone to false

detection.
6.1 Radiometry The hypothesis model for transmitter detection can be
defined as:
If the receiver does have prior knowledge of n(t ) 0
the primary signal then for the non-coherent method x(t )
Radiometry is the optimal way for spectrum sensing h * s(t ) n(t ) 1
below. In order to measure the energy of the received
signal the output signal of band pass filter with (10)
bandwidth W is squared and integrated over the Here x (t ) is the signal received by the unlicensed
observation interval T. Finally the output of the user, s (t ) is the signal transmitted by the licensed
integrator is compared with a threshold to detect
weather the primary or licensed user is present or not. transmitter, n(t ) is the noise introduced by AWGN
It can also be computed in frequency domain by and h is the channel gain. 0 is the null hypothesis
averaging bins of a Fast Fourier Transform[9].
when there is no primary signal and 1 indicates the
In this the processing gain is proportional to FFT size presence of primary signal [8]The probability of
N and the averaging time T. Increase in the size of detection d and false alarm f are given as follows
FFT improves the frequency resolution which is
helpful in detecting narrowband signals. Also if we d Y 1 Qm ( 2 , )
reduce the averaging time it improves the SNR by (11)
reducing the noise power.
f Y 0 =

m,
2
m
(12)
(9)
The number of samples required to optimally detect
the incoming signal is O(1/SNR2)
Figure 4 Radiometry using Welch

Periodogram method
The problems in using Radiometry is the threshold

that is used for detecting primary signal is]\rone to
unknown changes in noise levels. If the noise level is
changed adaptively still the presence of in-band
interference can cause poor detection of signal
energy. If the channel is not flat it is not possible to
set the threshold with respect to the notches caused
by its frequency selectivity. Since the Radiometry is
only concerned with the energy of the incoming Figure 5: Flow chart for Implementationof
signal it does not differentiate between noise and Radiometry
interference. In these equation is SNR, is threshold. A low
detection of probability will result in absence of
In context to cognitive radios interference and noise primary signal and a high would result in inefficient
should be treated differently because of the presence spectrum utilization[10]. Besides other limitations
of unlicensed and licensed users. Due to these Radiometry cannot be used for spread spectrum
120
signals and hence more sophisticated algorithms overlapping features in power spectral density are
should be employed for signal detection. non overlapping features in cyclic spectrum. Hence
different types of modulated signals that have
6.2 Cyclo stationary Feature Detection identical power spectral density can have different
cyclic spectrum. Because of all these properties cyclo
An alternative method for the detection of primary stationary feature detector can perform better than
signals is Cyclo stationary Feature Detection in which energy detector in discriminating against noise.
modulated signals are coupled with sine wave However it is computationally complex and requires
carriers, pulse trains, repeated spreading, hopping significantly large observation time. For more
sequences, or cyclic prefixes. This results in built-in efficient detection, the enhanced feature detection
periodicity. These modulated signals are scheme combined with cyclic spectrum analysis and
characterized as cyclo stationary because their mean pattern recognition based on neural networks is
and autocorrelation exhibit periodicity. proposed.
This periodicity is introduced in the signal format at

the receiver so as to exploit it for parameter
estimation such as carrier phase, timing or direction
of arrival. These features are detected by analyzing a
spectral correlation function. The main advantage of
this function is that it differentiates the noise from the
modulated signal energy. This is due to the fact that
noise is a wide-sense stationary signal with no
correlation however modulated signals are
cyclostationary due 15 to embedded redundancy of
signal periodicity. Analogous to autocorrelation
function spectral correlation function
(SCF)canbedefinedas:
t 1
sx f limT limT 1t 2t XT t, f 2 XT t, f 2 dt
2T
Figure 7 Flow chart for the implementation of Cyclo

(13) stationary Feature Detection
where the finite time Fourier transform is given by:
7 RECEIVED SIGNAL STRENGTH (RSS) -
X t, v x u e j 2 vu du
t T BASED LOCALIZATION
2
t T
2 (14) As mentioned earlier some of the RSS-based
localization techniques require knowledge of the
EIRP of the transmitter emitting the signal of interest.
This can be a limitation in using this technique for
systems in which the EIRP of the signal of interest
source is unknown, such as the scenario shown in
Figure 8.
Figure 6: Implementation of Cyclo stationarity
Feature Detector
Spectral correlation function is also known as cyclic

spectrum. While power spectral density (PSD) is a
real valued one dimensional transform, SCF is a
complex valued two dimensional transform. The
parameter is called the cycle frequency. If = 0
then SCF gives the PSD of the signal. Because of the
inherent spectral redundancy signal selectivity
becomes possible. Analysis of signal in this domain
retains its phase and frequency information related to Figure 8: A Planar Network Configuration for RSS-
timing parameters of modulated signals. Due to this, based Localization
121
This section explains the RSS-based localization 10

0
Probability of Eneregy dection under AWGN channel
approach used in the proposed sensor network based Simulation

Theory
cooperative wideband spectrum sensing and 10
-1 Arrpoximation
localization scheme[11].
-2
Pm
10
This localization approach does not require the EIRP

of the transmitter emitting the signal of interest. 10
-3
Figure 9.2 shows the network configuration proposed

. 10
-4
-4
-3 -2 -1 0
10 10 10 10 10
Pf
In this network configuration, it is implicitly assumed Fig 10: Perfect energy detection under
that the radiation patterns of the transmitter antennas AWGN channel
are azimuthally omni-directional. In this scheme at 1
x 10
-11
least four nodes, whose positions are known, must 0.5

measure the RSS[12] values from the signal of
0
interest source. Since every node sends its RSS
-0.5
measurement value and its position information to all
other nodes, localization procedures can be applied -1
PD
on any node. -1.5
-2
8. Results : -2.5
-3
The proposed sensor network based cooperative 0 10 20 30 40 50
SNRreq (dB)
60 70 80 90 100
wideband spectrum sensing and localization scheme

is appropriate for RF sensor networks since it senses Figure:11 Thersold level of cyclo stationary
a wide spectrum band in an energy efficient manner
by providing resilience to fading, shadowing, and
noise. Energy efficiency comes from the usage of x 10
-11 clostationary Feature Detection in Cognitive Rardio Network
4
MRSS and the proposed three-bit hard combination
scheme. 3.5
3
In particular, redundant exhaustive sensing on empty
bands is avoided with MRSS, and less overhead in 2.5
PD
collaboration with respect to the soft combination is

2
provided by three-bit hard combination. RSS-based
localization provides the location and EIRP of the 1.5
source of interest. Resilience to fading, shadowing, 1

and noise is due to the cooperation of the nodes.
0.5
0 10 20 30 40 50 60 70 80 90 100
-11
SNReq
x 10 Signa Received in Cogntive Radio Environment
1.2
Figure:12 cyclo stationary feature Detection in
Cognitive Radio Network
1
Wideband Radiometer, T1=96, W1=30, PFA=0.01

1
0.8
SNR level ---->
0.9
0.8
0.6
0.7
0.6
0.4
0.5
PD
0.4
0.2
0.3
0.2
0
0 10 20 30 40 50 60 70 80 90 100 0.1
No of Users --->
0
-10 -8 -6 -4 -2 0 2 4 6 8 10
Fig 9:Signal Received in Cogntive Radio SNRreq (dB)
Environment
Figure:13 Wide Band Radiometry
122
1
networks. An RF sensor network can be deployed
0.9
over an area of interest to detect the signals in the air,
to determine the frequencies of these signals, and to
0.8
estimate the positions and effective isotropic radiated
0.7
powers (EIRPs) of the transmitters emitting these
0.6
signals. Maximizing the signal detection performance
0.5
PD
and the accuracy of position and power estimations

0.4
with a minimal computation complexity is an
0.3
additional objective.
0.2
0.1 Future work:

0
-10 -8 -6 -4 -2 0 2 4 6 8 10
SNRreq (dB) In future we would like to explore other types of
Figure:14 channelized Radiometry feature detection and evaluate their performance
completely with energy detection. In band sensing of
wireless microphones should be another subject of
future work.
Aknowledgements:
The authors would like to thank the authorities of
Acharya Nagarjuna University, for providing me the
equipment and all the necessary information for
submission of this paper.
References:
1. S. Haykin, Cognitive radio: brain- empowered wireless

communications,IEEE journal on Selected Areas in
Communications, vol. 23, no. 2, pp. 201220, 2005.
2. H. Hannes and P. L. Kenneth, Eds., VANET: Vehicular
Figure:15 Detection percentage versus number of Applications and Inter-Networking Technologies, John Wiley
&Sons, New York, NY, USA, 2010.
Nodes for three different SNR Values 3. A. Sahai, N. Hoven, and R. Tandra, Some fundamental limits on
cognitive radio, in Proceedings of the 42nd Allerton Conference
Numerical results shows that radiometry on Communication, Control, and Computing, 2004.
4. H. Rasheed, N. Rajatheva, and F. Haroon, Spectrum sensing
technique has better performance compared with non with energy detection under shadow-fading condition, in
cooperative one and employing radiometry can Proceedings of the 5th IEEE International Symposium on
improve probability of detection and non cooperative Wireless Pervasive Computing (ISWPC 10), pp. 104109, May
2010.
detection at different SNR values.Cooperative 5. S. I. Gradshteyn and M. I. Ryzhik, Table of Integrals, Series, and
technique is more effective Products, Academic Press, 7th edition, 2007.
when received SNR in cognitive radio users is low 6. J. Ma, G. Zhao, and Y. Li, Soft combination and detection for
cooperative spectrum sensing in cognitive radio networks, IEEE
due to fading and shadowing.Non cooperative Transactions on Wireless Communications, vol. 7, no. 11, pp.
technique achieves the same detection probability 45024507, 2008.
values as cooperative technique as received When 7. H. Urkowitz, Energy detection of unknown deterministic
signals, Proceedings of the IEEE, vol. 55, pp. 523531, 1967.
received SNR is greater than 10dB ,Furthermore a 8. F. F.Digham, M. S. Alouini, and M.K. Simon, On the energy
minimum of 15 collaborated users relatively in CR detection of unknown signals over fading channels, IEEE
system can achiveve optimal value of detection Transactions on Communications, vol. 55, no. 1, pp. 2124,
2007.
probability .However ,it depends on the thersold 9. A. Ghasemi and E. S. Sousa, Collaborative spectrum sensing
value used in signal detection . Results of the for opportunistic access in fading environments, in Proceedings
cooperative spectrum sensing part of the proposed of the 1st IEEE International Symposium on New Frontiers in
Dynamic Spectrum Access Networks (DySPAN 05), pp. 131
scheme showed that using rectangular window in 136, November 2005.
MRSS provides the highest detection margin. The 10. J. G. Proakis and M. Salehi, Digital Communications, McGraw
positions of the nodes participating in the sensing Hill, New York, NY, USA, 5th edition, 2008.
11. S.P.Hearth and N.Rajatheva, analysis of equal gain combining in
process and the applied decision criterion affect the energy detection for cognitive radio over Nakagami channels in
detection performance. Proceedings of the IEEE Global Telecommunications Conference
(GLOBECOM 08), pp. 1-5, November 2008.
12. R.Tandra and A.Sahai, Fundamental limits on detection in low
9. Conclusions: SNR under noise uncertainty in Proc.of IEEE international
This paper focused on spectrum sensing and conference on wireless networks, communication and mobile
localization with radio frequency (RF) sensor computing ,June 13-16 2005, vol . 1, pp. 464-469.
123
Wireless Multi-Flow Sensor Water

Consumption Monitoring System using Zigbee
for Gated Communities
G.S.R.Satyanarayana and K.Vijaya Kumar School of Electronics, Vignan Uniersit
gsrsatyanarayana@gmail.com, vijkudari@gmail.com
Abstract - This paper presents the design of various parameters can be monitored and stored in
water consumption monitoring system for gated the Windows PC. A current popular and
communities using wireless sensor network. The widely used technology, Zigbee based wireless
application of flow-sensor for a water usage
monitoring system is composed of a number of
sensor network is used in this work as it is able to
flow-sensor nodes with Zigbee network that can be provide flexibility, low cost implementation and
deployed for continuous monitoring purpose. The reliability. Zigbee is a communication standard for
water consumption monitoring system will monitor the use in the wireless sensor network defined by the
flow of water, detects abnormal flow if any and Zigbee Alliance that adopting the IEEE
warns the occupant by alarming. Therefore, the use 802.15.4 standard for its reliable communication.
of the monitoring system will prevent the wastage of
The fig.1 shows the block diagram of overall
water and the cost of destruction to the property
that can be caused when the flow of water is not system with sensor unit at the individual and
properly controlled. In this paper, the fundamental monitoring units at base station.
design of the system is proposed and tested practically
using Zigbee technology[1]. The developed system is
cost-effective and allows easy customization. Results of
water consumption measurement system are also
presented.
Keywords - Flow-Sensor, Water Consumption Monitoring
System, Zigbee, Gated Community
I. INTRODUCTION
During the past few years, gated communities have
The sensor unit at the pump is optional to monitor the
increased rapidly in the Indian municipality and
incoming water level. Each individual Sensor and
metropolitan cities. Water is used by the people in
Monitoring units are connected with EEPROM
the gated communities for a variety of domestic uses
memory to store timely day wise results. The
such as washing, bathing and flushing toilets etc.
monitoring unit collects the data from individual
Usage of water by the gated communities in a proper
sensor units and stores all the information in the data
way is important in order to reduce the wastage of
base of the PC.
water. However, it is a challenging task for the
community to continuously monitor the consumption
II. HARDWARE DESIGN
of water and any leakage in the water pipes.
Therefore to control the wastage of water, a A. Sensor Unit
continuous monitoring system[3] is needed. For this
reason, it is important to have a monitoring system The Fig.2 shows the block diagram of the sensor
with the characteristics of autonomous and unit with a LCD display, Key pad, Real Time Clock,
reliability. The use of such monitoring system EEPROM memory and a Zigbee module connected
will reduce the reliance on man power and reducing with a programmable P89C51 microcontroller. A
the cost. This work focuses on the use of multiple sensor unit[4] is basically consists of flow-sensor
water flow- sensors working together as a device to used to detect the predetermined parameters like
check the consumption, leakage, level and flow of consumption, leakage, level and flow of water. All
water in the individual of the gated communities. The the sensor units use battery for their operation. The
flow-sensor unit in the individual of the gated P89C51 microcontroller in the sensor node is
community is able to continuously monitor these programmed such a way that the pulses sensed by the
parameters from the water pipes. Then, the sensed sensor are counted using the counter. Then, the same
data from the sensor unit will be transmitted information is converted to number of liters and is
wirelessly to the base monitoring unit where sent to the monitoring unit wirelessly by Zigbee
124
communication. The Fig.3 shows a sensor hardware monitoring unit[6] consists of Zigbee receiver
unit connected with a SF800 flow-sensor and a module programmed such a way that it receives the
Xbee module. data sent from the sensor units which are called end
devices wirelessly. For every 30 minutes, the
monitoring unit stores all the collected data from
various sensor units in the EEPROM memory. The
duration is maintained with reference to the Real
Time Clock connected to the unit. If there is any
abnormal flow through the water pipe, then it is
detected by the sensor unit and the same is informed
to the monitoring unit.
The monitoring unit identifies the sensor unit, from

where the abnormal flow is detected and alarms the
user to find the destruction of the pipe.
The SF800 is a flow sensor used in this
water consumption monitoring system for monitoring
the several parameters. The function of the flow-
sensor is when flow formers at the inlet of the
cylinder generate a controlled and constant swirl that
causes the rotor to float in the flow stream of the
metered fluid. This free rotation is almost without
friction and proportionate to the fluid throughput. The
output of the flow sensor generates electrical pulses
which are calculated by the T1 counter of P89C51
microcontroller in the sensor node. The number of
pulses per unit time will be used to identify the As the coordinator needs to receive data
consumption of the water, and the pulse duration will continuously from various end devices, it is normally
decide the existence of water flow in the connected mains powered. The data received by the monitor
sensor node pipe. unit from a sensor node is sent to the Personal
Computer (PC) using the RS232 protocol and the
Everyday water consumption is updated in the received data is stored in the data base of the
EEPROM memory of the sensor unit for every 30 Windows PC.
minutes; and the updated information is sent to the
monitoring unit where all the data from various C. Zigbee Communication Module:
sensor nodes are stored. The sensor node indicates
the destruction of the pipe when there is an abnormal In this work the wireless sensor unit and the
flow of water from the flow-sensor. The abnormal monitoring unit connected wtih XBee (Zigbee)
flow in the pipe alarms the occupant by ringing the data transmitting and receiving devices[4] form
buzzer. There is an optional sensor unit at the Maxstream Inc. The XBee RF module is an IEEE
tank intimates the level of inflow water in the tank 802.15.4 standard [1] and supports the needs of low-
to the monitoring unit using Zigbee module. cost, low-power wireless sensor networks[5] targeted
B. Monitoring Unit towards automation and remote control
applications. The Zigbee in the sensor unit is
The Fig.5 shows the monitoring hardware unit connected with P89C51 microcontroller to digitize
with ARM9 programmable device, Xbee module and and transmit the data to the monitoring unit; and the
RS232 to connect with the windows PC. The ARM9 in the monitoring unit is with the task of
125
signal reception from the sensor unit and sending the III. RESULTS
collected data to the PC through RS232 which can
be seen on the monitor. The monitoring unit and the The tested network is connected in star topology with
sensor u n i t s a r e p o w e r e d w i t h a three sensor units and with a monitoring unit. Each
battery. The Z i g b e e compliant wireless senor node is placed at 12 meters apart from the
devices are able to transmit 10-75 meters, depending monitoring unit. The monitoring unit is connected to
on the RF environment and the power output the PC through RS232 to show the sensed data on
consumption required for a given application which the system monitor in a readable format. The
will operate in the unlicensed RF range at 2.4GHz following Figure.6 shows the consumption of water
Global. The XBee module operating data rate is in liters from different sensor units on the screen of
250kbps at 2.4GHz. The Zigbee network [2] can be Windows PC.
configured to operate in a variety of different ways From the Fig.6 it can be easily observed that flat
to suit the application and environment with no.201 consumes more water than the other flats.
supported topologies like star, cluster tree and Furthermore, flat no.102 consumes least amount of
mesh. The use of high power Wireless Sensor water. The Fig.7 also shows water consumption at
Network is suitable for activities in industries three senor nodes 101,102 and 201 on their
involving large area monitoring. individual LCDs respectively. Hence, the results
shows that the proposed system working properly.
D. Working of the System:
The following algorithms indicate the functioning of
the sensor and monitoring units. The proposed sensor
unit will send the data to the monitoring unit for
every 30 minutes. This timing can be adjusted
with different duration depending on the need.
The RTC at every unit will be set with common
clocking. The sensor node at the water tank is
optional which can be used to identify the inflow
of the water in to the water tank, So that we can find
the water consumption with respect to the water in
the tank.
A. Functioning of the Sensor Unit

1. Initialize RTC, Memory and Zigbee.
2. Start counting the pulses received from the
flow- sensor.
3. If time is not matched then wait for time
duration else store the count value in to EEPROM
memory; transmit the same count to the monitoring
unit.
4. If transmit is forced from monitoring unit then IV. CONCLUSION
send information to monitoring unit; else repeat
from step 2. The proposed wireless multi-flow sensor water
consumption monitoring system for gated
B. Functioning of the Monitoring unit communities using Zigbee is having efficiency,
coordination, cost effective and adjustability. One of
1. Initialize RTC, Memory and Zigbee. the advantages of this proposed work is the easy
2. If transmit or receive forced then send installation where the monitoring unit can be placed
commands to sensor unit else wait for receive. at the local residence close to the end devices.
3. If the received count is abnormal then ring The monitoring task can be done by any person with
buzzer; else go to next step. minimal training at the beginning of the system
4. Store the received data in EEPROM memory. installation.
5. Repeat from step 2.
The scope of this proposed work is suitable for
The above algorithms A and B are used to activities in residential apartments, industries
program the P89C51 at the sensor node and involving large area monitoring such as
ARM9 at the monitoring node respectively. manufacturing, constructing, mining etc.
126
REFERENCES
[1] Zulhani Rasin and Mohd Rizal Abdullah Water Quality
Monitoring System Using Zigbee Based Wireless Sensor
Network International Journal of Engineering & Technology
IJET Vol: 9 No: 10.
[2] Gopala Krishna Moorthy .K, Dr.C.Yaashuwanth,
Venkatesh.K AWireless Remote Monitoring Of Agriculture
Using Zigbee (IJEIT) Volume 2, Issue 8, February 2013.
[3] K.Vairamani, K. Arun Venkatesh and N.
Mathivanan.Design and Development of Zigbee Based
Instantaneous Flat-plate Collector Efficiency Measurement
System MEASUREMENT SCIENCE REVIEW, Volume 11,
No. 2, 2011.
[4] Junguo ZHANG, Wenbin LI, Ning HAN, Jiangming KAN
Forest Fire detection system based on a Zigbee wireless sensor
Network Front. For China 2008, 3(3): 369374, DOI
10.1007/s11461-008- 0054-3.
[5] Peng Jiang, Hongbo Xia , Zhiye He and Zheming Wang
Design of a Water Environment Monitoring System Based on
Wireless Sensor Networks Sensors 2009, 9, 6411-6434;
doi:10.3390/s90806411.
[6] C. Arun and K. Lakshmi Sudha Agricultural
Management using Wireless Sensor Networks - A Survey,
Singapore, 2nd International Conference on Environment Science
and Biotechnology DOI: 10.7763 /IPCBEE. 2012. V48. 15.
127
Opportunistic Spectrum Access for Prioritized

Secondary Users in Cognitive Radio Networks
K.Annapurna, V. Harshavardhan, Dr.B.SeetaRamanjaneyulu
K.Annapurna, Asst. Professor, Dept. of ECE, vignans university,
EMAIL ID-anu_jannu85@rediffmail.com,
Harshavardhan velagapudi, PG student, Dept. of ECE, vignans university,
Phone no(+91-9052125252) ,EMAIL ID- harsh.velagapudi@gmail.com,
Dr.B.Seeta Ramanjaneyulu, Professor & HOD, Dept of ECE, vignans university EMAIL ID ramanbs@gmail.com.
Abstract Cognitive radio is a promising technology to ii)Spectrum Decision: Users can be allocate a
utilize the spectrum efficiently as radio frequency channel, based on spectrum availability. This
spectrum is precious and scarce. The main intension of allocation not only depends on availability of
cognitive radios is whenever licensed users are not in
use opportunistic access of spectrum is carried out.
spectrum but also based on internal policies.
However in practical scenario some applications like iii)Spectrum Sharing: If there are multiple CR users
medical, military have more importance than the others. trying to access the spectrum, there should be
Based on this paper proposes a cognitive radio network,
which contains primary users and three types of coordination to prevent collision.
prioritized cognitive users. Arrival times and conference iv)Spectrum Mobility: CR users are regarded as
times of all the users are generated by using exponential
random distribution. Based on these priorities channel
visitors to the spectrum. Hence, if the specific portion
allocation is done. Blocking probability for the cognitive of the spectrum in use is required by a primary user,
users is calculated. This work is carried out with the the communication must be continued in another
help of MATLAB. vacant portion of the spectrum and it has to change its
properties according to the available channel
I. INTRODUCTION characteristics.
The life cycle of cognitive radio is shown in figure1.
Nowadays, people are very much interested in
Sensing all the radio frequency bands, taking a
wireless devices and technology like Wireless
decision, sharing of spectrum and transmission of the
Routers, Mobile Phones, MODEMs, and GPS....etc.
signal is the main aim of cognitive radio.
Increase in wireless devices demand for radio
frequency spectrum, which is limited. So, efficient
utilization of the spectrum is required. FCCs
Spectrum Policy Task Force which has determined
that, in some locations or at some times of a day,
about 70 % of the allocated spectrum may not be in
use [1].Measurement [2] reveals that spectrum
utilization is often heavy in unlicensed bands while
low in TV bands or medium in some cellular bands.
These observations of actual spectrum usage made
the researchers to think of opportunistic spectrum
access. Cognitive radio is one of the solutions to the
above mentioned problem. Cognitive Radio offers a
solution by utilizing the spectrum holes that represent
the potential opportunities for non-interfering use of
spectrum which requires four major steps[3].
i)Spectrum Sensing: Unused portion of spectrum can Figure1: Cognitive Radio Life Cycle
be allocated to a cognitive radio user. Therefore
available spectrum bands should be monitored by the Sensing the entire spectrum is to be done for every
CR user.CR user needs to capture their information instant of time every channel is to be sensed and
and detect the spectrum holes. Various spectrum finding out which channels are vacant depending on
sensing techniques which, in general, could be user characteristics. Once spectrum holes are
classified as (1)energy-based sensing,(2) identified, cognitive radio users opportunistically
cyclostationary feature-based sensing,(3)matched utilize these holes for communication without
filter-based sensing and (4)other sensing techniques. causing any interference to primary users. But in this
128
scenario decision making plays a major role for interference as possible. Ultra Wideband (UWB)
different applications so prioritizing secondary users technologies are using this kind of spectrum sharing.
and allocating the channels to secondary users [6]
depending upon the priority. The rest of this paper Underlay Spectrum Sharing: The spectrum is shared
is organized as follows. Section II deals with in one of the following three ways:
available spectrum sharing techniques, section III i)Opportunistic: -The spectrum is shared among the
explains our proposed work, section IV contains the SUs whenever the PUs are not using it. [6]
simulation results, and sectionV describes ii)Cooperative: -The spectrum is allocated after
conclusion and future scope. negotiation with the PUs. [6]
iii)Mixed: - The spectrum is accessed in either a
II SPECTRUM SHARING cooperative or opportunistic way. [6]
Basically there are 2 types of spectrum sharing
The sharing of spectrum among cognitive users techniques. One is based on game theory and the
is divided into three main classes. This other is on genetic algorithm. Game theory based
classification is based on the assumptions of the cognitive channel assignment theory has originally
architecture, the way in which the spectrum is been proposed by the mathematician John von
allocated and spectrum access techniques. [4] Neumann to study human behavior [7]. It is an
First Classification: The first classification which is interdisciplinary research area where mathematics,
based on the architecture is illustrated as follows: social and behavioral sciences are brought together.
Centralized spectrum sharing: In this technique there Game Theory is one of the tools that more and more
exists a central entity that controls the allocation of researchers in this field try to apply to radio resource
spectrum and channels. All the users in the management. Significant work has been done by
cognitive network forward their measurements James Neel in applying Game Theory to the radio
towards the central entity that will make decisions resource management of cognitive radio systems
about the spectrum allocation. [4] [8],[9],[10],[11],[12].
Distributed Spectrum Sharing: Distributed spectrum Second one is a genetic algorithm which is an
sharing technique is implemented in the case where optimization technique that uses a number of bio-
there cannot be defined any infrastructure of the inspired evolutionary concepts like inheritance,
network. In this scheme each individual user selection, mutation and crossover. A random solution
makes decisions about the spectrum and channel is usually generated at the beginning of optimization.
allocation. [4] Then at each generation, the fitness of the solution is
evaluated by a predefined fitness function, and the
Second Classification: The second classification that solution will be modified accordingly. The algorithm
is based on the spectrum access behavior is is terminated when a satisfactory fitness level is
described as follows achieved or a maximum number of generations are
Cooperative spectrum sharing: The measurements of reached. Applications of a genetic algorithm can also
interference made by the individual cognitive users be applied to distributed optimization problems. A
are shared among all users. This information genetic algorithm based frequency allocation
collected from all the users is used for the approach for distributed cognitive radio networks is
allocation of the spectrum. Almost all of the proposed by Si Chen and Alexander M. Wyglinski in
centralized spectrums sharing techniques are [13]. A fitness function is developed to intelligently
cooperative. [4] Security is the major issue in this allocate frequency bands for subcarriers in a Non-
technique. [5] Contiguous Orthogonal frequency Division
Non-cooperative Spectrum Sharing: In non- Multiplexing (NC-OFDM) system.
cooperative spectrum sharing the cognitive users are
making decisions based on local observations. They III. PROPOSED WORK
do not cooperate with each other a n d ar e m a k i n g
d eci sio n s i n d e p e n d e n t l y f r o m others. The Compared to the above mentioned techniques, our
ad vantage o f this approach is that control traffic proposed work is simple and efficient. Considering
is mitigated to a large extent and the channel number of channels is equal to the number of primary
allocation behavior is also simplified. users, as primary users have own their individual
license. Keeping in mind that some applications are
Third Classification: Finally, the third technique is more important than the others secondary users are
based on how the spectrum is accessed: classified into three priorities secondary users1 (su1),
Overlay spectrum sharing: In this type of sharing the secondary users2 (su2) and secondary users3 (su3)
SUs are accessing the spectrum at the same time respectively. Figure 2 shows how the channel is
the PUs are accessing it, but with as little occupied by primary and secondary users
129
Non-blocking state: A secondary user is considered to Now search for medium priority secondary users
be in this state if it is completely serviced without arrival and conference time if any of the medium
being interrupted by a primary user on that channel. priority secondary users arrival time and conference
time are present within the time interval T and also
Dropping state: When the primary user of a channel
find if any of the primary user or high priority
returns, the secondary user utilizing that channel
secondary user is using that channel at that instant if
should vacate. If there are no more free channels
primary user or high priority secondary user is not
available then it is semi-serviced and its call is
using the channel allocate the channel for medium
dropped.
priority secondary user if primary user or high
Forced termination state or Transition state: This is priority secondary user is using the channel then
the state during which the secondary user is shifting block the medium priority secondary user.
its channel due to the return of the licensed user into
the previous channel. In this case there are free Searching for low priority secondary users arrival
channels to shift to and so the secondary user and conference time if any of the low priority
performs a spectrum hand-off. secondary users arrival time and conference time are
present within the time interval T and also find if any
of the primary user or high priority secondary user
or the medium priority secondary user is using that
channel at that instant. if primary user or high
priority secondary user or the medium priority
secondary user is not using the channel allocate the
channel for low priority secondary user if primary
user or high priority secondary user or the medium
priority secondary user is using the channel then
block the low priority user. MATLAB software
which is a computationally intensive environment is
used for simulate of the proposed work.
Figure 2: Random access in five channels.
ALGORITHM
Blocking state: When all the channels are occupied

by either primary users or secondary users, then an 1. Consider N number of channels, primary
incoming secondary user does not have any users, su1s,su2s and su3s.
opportunity for communication and it is considered to 2. Define arrival and conference times of
be completely blocked. primary users and secondary users based on
Primary user can use the channel at any instant of random distributions.
time. So, the mean arrivals and conference times of 3. Assign the channels to primary users.
primary users (N users) can be generated by using 4. Continuously sense the channel and
exponential random distribution. Secondary users whenever an ith channel is vacant, allot to ith
continuously sense the channels and whenever find a Secondary user based on requirement and
vacant channel they allot it to secondary users based
priority.
on priority.
If a high priority secondary user arrives at
For a particular number of channels for a particular the time of channel being utilised by low
period of time (T) if any of the primary users arrival priority secondary user then block that low
time and conference time is present within the time priority secondary user and offer the service
interval T allocate the channel. to the high priority secondary user.
Now search for high priority secondary users
5. This process is done for various numbers of
arrival and conference time if any of the secondary
users arrival time and conference time are present channels and various mean arrival times.
within the time interval T and also find if any of the 6. For each case, blocking probability of the
primary user is using that channel at that instant if three types of secondary users is calculated.
primary user is not using the channel allocate the Where, blocking probability is defined as
channel for high priority secondary user if primary the ratio of blocked secondary users to the
user is using the channel then block the high priority
total number of secondary users of that
user.
category.
130
Figure5: Blocking probability for 5 channels

From figure 5 it is clear that blocking probability is
Figure3: Channel occupancy reduced in 5 channels compared to 3 channel.
From figure 6 blocking probability is much reduced
Channel occupancy by primary users, su1s, su2s and comparing from figure 4 & 5. With the increase in
su3s with respect to time for 7 channels is shown in number of channels blocking probability is getting
figure 3. Red is used for primary users, green for reduced. With the decrease in blocking probability
su1s, yellow for su2s and blue for su3s. From figure1 the efficient utilization of spectrum can be achieved.
it is clear that su1 is blocked in channels 6, su2 is
blocked in 5 & 6 and su3 is blocked in 1,2 &4. So
more priority users were less blocked.
Blocking probability of su1, su2s and su3s with

respect to mean arrival timings of the primary and
secondary users is shown in figure 3.Green, red and
yellow are used to indicate the blocking probabilities
of su1, su2 and su3 respectively. It can be observed
that blocking probability of su1 is less compared to
su2s and su3s. In this way service can be provide to
real time world where some may be having
emergency tasks.
The blocking probability of prioritized secondary

users for 3 channels is shown in figure 4. Due to Figure6: Blocking probability for 7 channels
random distribution the blocking probability of the
prioritized secondary users varies at some mean V. CONCLUSION AND FUTURE SCOPE
arrivals.
We proposed a novel approach to opportunistic
access of the spectrum to prioritized secondary users.
This enables efficient utilization of spectrum by
knowing importance of the applications, instead of
assigning blindly to each application. It can be
extended to reduce the blocking by searching for
other vacant channels when the corresponding
channel is not free or wait for the channel to complete
the task of primary users or more priority secondary
users.
R EFERENCES
[1] Spectrum Policy Task Force report, Technical report 02-135,
Federal Communications commission, Nov. 2002.
Figure4: Blocking probability for 3 channels [2] M. McHenry, Report on spectrum occupancy measurements,
http://www.sharedspectrum.com/?section=nsf summary.
131
[3] I.F Akyildiz, Won-Yeol Lee, M,C Vuran, S. Mohanty, "A

survey on spectrum management in cognitive radio networks," in
IEEE communications magazine , vol.46, no.4, pp.40-48, April
2008
[4] I.F. Akyildiz,W.Y. Lee and M.C. Vuran,Next
generation/dynamic spectrum access/cognitive radio wireless
networks: A survey, IEEE Transaction on Computer
Networks, vol. 50, no. 13, pp. 2127-2159, May, 2006.
[5] Honggang Zhang, Xiaofei Zhou and Tao Chen, Cognitive
Radio Networks, edited by Yang Xiao and Fei Hu, CRC Press,
2008.
[6] C. Ianculescu, A. Mudra., Cognitive Radio and
Dynamic Spectrum Sharing, in Proceeding of the SDR 05
Technical Conference and Product Exposition, 2005.
[7] M. J. Osborne, An Introduction to Game Theory: Oxford
University Press, 2004.
[8]C. K. Tan, "Game theoretic approach for channel assignment
and power control with no internal -regret learning in wireless ad-
hoc networks,"IEEE Communications, vol. 2, pp. 1159-1169,
2008.
[9]J. O. Neel, "The Role of Game Theory in the Analysis of
Software Radio Networks, presented at the SDR Forum Technical
Conference, 2002.
[10]J. O. Neel and J. Reed, "Game models for cognitive radio
algorithm analysis," presented at the Software Define Radio Forum
Technical
Conference 2004.
[11]J. O. Neel, "Game theoretic analysis of a network of cognitive
radios," presented at the 45th Midwest Symposium on Circuits and
Systems Aug. 2002.
[12]J. O. Neel, "Game Theory in the Analysis and Design of
Cognitive Radio Networks," presented at the DYSPAN, 2007.
[13] D. E. Goldberg, Genetic Algorithms in Search, Optimization,
and Machine Learning: Addison-Wesley Professional, 1989.
132
FPGA IMPLEMENTATION OF IMAGE FILTERING

APPLICATIONS USING XGS
N.Vani, N. Usha Rani
Abstract : Now a days image processing plays a key produce synthesis in FPGA, HDL simulation and
role in many areas like defence, medical and radar development tools. The tool retains the hierarchy of
signal processing etc. Image processing algorithms Simulink when it is converted into VHDL.
can be implemented using PC or dedicated H/W. But
it takes large computational time. The speed of The shape of objects in images is described many
implementation can be reduced using FPGA which parameters like the edges, depth discontinuity,
supports parallelism. In this paper Image filtering surface orientation discontinuity, reflectance
operations are explained using Xilinx System discontinuity, illumination discontinuity, and noise
Generator(XSG) and SIMULINK. The filtering is level in the images.
performed using Sobel and Median filters on CT and
MRI images. The results are validated using PSNR II. XILINX SYSTEM GENERATOR
and MSE.
Keywords: Digital Image processing, Matlab, Xilinx System Generator (XSG) is an integrated
Xilinx System Generator, FPGA. design Environment (IDE) for FPGAs within the ISE
13.4 development suite, which uses Simulink, as a
I. INTRODUCTION development environment and is presented in the
form of model based design. It has an integrated
The handling of digital images has become a subject design flow, to move directly to the Bit stream file (*.
of widespread interest in different areas such as legal, bit) from Simulink design environment which is
medical, technological applications and many others. necessary for programming the FPGA.One of the
There are many areas where the image processing most important features of XSG is that it possesses
helps to analyze, infer and make decisions. The main abstraction arithmetic that is working with
objective of image processing is to improve the representation in fixed point with a precision
quality of the images for human interpretation or the arbitrary, including quantization and overflow. XSG
perception of the machines independently. FPGAs are can only perform simulations as a fixed point double
increasingly used in modern imaging applications precision type.
like image filtering, medical imaging, image
compression, and wireless communication. The XSG automatically generates VHDL/Verilog code
various applications where image filtering operations and a draft of the ISE model being developed. It also
applied are noise removal, enhancing edges and generates hierarchical VHDL/Verilog synthesis, floor
contours, blurring and so on. plan and mapping hardware. In addition to this it also
generates a user constraint file (UCF), simulation and
The main aim of this paper is to implement image testbench and test
filtering algorithms in H/W like Xilinx Virtex6. The vectors.
need to process the image in real time using dedicated
H/W or PC is time consuming, leads to this XSG was created primarily to deal with complex
implementation in hardware level, which offers Digital Signal Processing (DSP) applications, but it
parallelism, and thus significantly reduces the also deals with implementation of many images
processing time. It uses XGS, a tool with graphical processing application. The blocks in XSG operate
interface under the Matlab Simulink, based blocks with Boolean values or arbitrary values in fixed point
which makes it very easy to handle with respect to type, for a better approach in hardware
other software for hardware description. implementation. In contrast Simulink works with
numbers of double-precision floating point. The
In addition to offering computation speed this tools connection between XSG blocks and Simulink blocks
works at graphical simulation level. This paper are the gateway blocks. The Fig.1 shows the broad
presents architecture development of image flow design of XSG.As previously mentioned, XSG
processing application in Simulink which is a is configured to program the FPGA. The synthesis
bookstore called Blocks Xilinx. They are mapped and implementation of the program are done
architectures, entities, ports and attributes, whose subsequently.
script file is to
133
Figure .2 Image Read operations using in XSG
3.2) Image Pre-processing Block sets:

The model based design used for image pre-
processing is shown in Fig.3. The blocks utilized here
are discussed below. Input images which could be
color or grayscale are provided as input to the File
block. A color space conversion block converts RGB
to grayscale image and this data which is in 2D is to
be converted to 1D for further processing. Frame
conversion block sets output signal100 to frame
based data and provided to unbuffer block which
converts this frame to scalar samples output at a
higher sampling rated.
Figure .1 Design flow in Xilinx System Generator

using matlab
Figure 3: Image Pre-Processing

III. PROPOSED MODEL
2.3) Edge detection using XSG Block sets:
The model proposed to detect edges using sobel and
median is as shown in fig.2 proposed model uses The model based design using Xilinx Block
Simulink and Xilinx block Sets. The model includes sets for processing the input image for edge
four steps. They are Detection is shown in Fig.4 (a), 4(b).
1. Read an Image
2. Image Pre-processing
3. Image Edge Detection
4. Image Post-Processing
Figure 4(a): Sobel Edge Detection an CT & MRI
Images
3.1) Read an Image:
The fundamental part of the work is reading
and writing image into Xilinx system generator
through Matlab platform. Figure 2 and 3 shows
reading of image using XSG. All the image
processing is done between FPGA boundaries known
as gateway in and gateway out. Image is read into
Simulink from the directory through gateway in
whereas gateway out the image output as per the Figure 4(b): Median Filtering an CT & MRI Images
block specified under Xilinx FPGA. Figure 4: Image Filtering Using Sobel and Median Filtering
XSG
134
Xilinx fixed point type conversion is made possible

by Gateway In block. To perform the edge detection In this paper the technique is applied on CT and MRI
a convolution operation of the input image with a images respectively as shown in Fig.7 and 8.The
filter mask is to be performed for which a n-tap MAC corresponding noisy images are also shown in Fig. 9
FIR filter block is used provided with nine and 10.
programmable coefficients. This is followed by
certain arithmetic blocks to merge all the processed
data.
2.4) Image post-processing Block sets:
The image post-processing blocks which are

used to convert the image output back to floating
point type is shown in Fig.5,
Fig.7 MRI Image Fig.8 CT Image
Fig. 5 Image Post-Processing
For post-processing it uses a Buffer block which Fig.9 Salt & Pepper and Guassian noise
converts scalar samples to frame output at lower
sampling rate, followed by a 1D to 2D (matrix)
format signal block, finally a sink is used to display
the output image back in the monitor, utilizing the
Simulink block sets.
IV. IMPLEMENTATION Fig.10 Salt and Pepper and Gaussian Noisy on CT

Images
The image processing architecture explained is
converted into HDL with the help of XSG at each The algorithm is applied on noisy images
level. If the simulated and synthesized output is individually. The corresponding processed images are
satisfactory then it is implemented into FPGA shown in Fig. 9 and 10. The processed images are
through JTAG. The process is also shown in Fig 6. validated using the parameters like Mean Square
Error(MSE) and Signal to Noise Ratio (SNR).
5.RESULTS AND CONCLUSIONS
Fig. 6 System Generator Parameters
135
Table.1. Mean Square Error and Signal to Noise

Ratio value for MRI and CT Image
The results are shown in Fig.11 and 12.The

processed image quality is appeared to be same as the
input image. The algorithm is removing Gaussian and
salt-pepper noise perfectly. The performance is also
shown in table.1.
Conclusion
This work addresses the filtering techniques

using XSG and Simulink. Simulink model is
proposed for reading images and edge detection using
sobel and median filters for MRI and CT images and
are valilidated using PSNR and MSE values. The
proposed framework is applicable for any application
by generating simulink models using respective
blocks and hence FPGA implementation can be done
for better performance. It is observed that MSE
between input and output is high in H/W than S/W. It
is due to noise expected is high. But SNR is
Comparable in H/W and S/W. Algorithm uses less
time in FPGA than dedicated hardware.
REFERENCES
1. Model-based Design With Simulink,HDL Coder,Xilinx

System Generator For DSP User Guidance in Math Works
User Guidance Manual.www.mathworks.com .
2. Xilinx System Generator User Guidance manual in Xilinxs
Forum, http://www.xilinx.com/tools/dsp.htm
3. Allin, S.C., M. Vignesh and A. Kandaswamy, An efficient
FPGA implementation of MRI image filtering and tumor
characterization using xilinx system generator, Int. J. VLSI
Design. Commun. Syst.(VLSICS),2011.
4. VIRTEX 6 DSP FPGA Family
Datasheet,2012,http://www.xilinxs.com/.
5. Alba M. Snchez G., Ricardo Alvarez G., Sully Snchez G.;
FCC and FCE BUAP, Architecture for filtering images using
Xilinx System Generator. Int.J. Mathematics and Computers
in Simulation.
6. Image/Video Processing on Xilinx FPGA Tutorial by
Mostofa Noor.
136
Morphological Image Segmentation Using

marker-Controlled Watershed
Transformation
Dibyendu Ghoshal
Associate Professor, Department of ECE
National Institute of Technology
Agartala, Tripura, Pin - 799055
tukumw@gmail.com
Pinaki Pratim Acharjya

Assistant Professor, Department of CSE
Bengal Institute of Technology and Management
Santiniketan, West Bengal, Pin - 731236
ppacharjya@gmail.com
Abstract - In recent days, the Morphological image segmentation [12-17] approach has found wide
segmentation is an important problem and widely in applications in various types of images which are not
use in different fields of computer vision and image only confined in the optical spectrum but also in
processing. To extend the concept of segmentation, other frequency ranges like infrared, microwave and
an ordering of the data is required. In recent days, millimeter wave based imaging, viz radar imaging,
there are several methods in image segmentation. The satellite imaging. The main advantage of the
focus of this research paper is a morphological image watershed algorithm is that it can guarantee closed
segmentation tool called marker controlled watershed contours, and there is a good match between the
segmentation with Laplacian of gaussian spatial high boundaries of regions in the result and those that
pass filter. The efficiency of the results demonstrates appear in the original image. Though watershed
that combining of these two strategies has been very algorithm a very good method for image
helpful for image segmentation and for computer segmentation, but this technique suffers from over
vision. segmentation problem [8-10]. Applying of watershed
segmentation algorithm directly on the gradient
images produce high over segmentation problem. The
Keywords:Image segmentation, , Laplacian of
problems caused by these factors can be serious
Gaussian, markers, watershed algorithm. enough to render the result useless.
1. INTRODUCTION An approach used to control over segmentation is

based on the concept of controlled marker. At the
very concept of the watershed segmentation are
Digital color images are formed by objects with a normally undergone through obtaining a gradient
high variability in shapes, sizes and intensities [1]. image because it helps to eliminate the major
They are highly textured and have ahigh level of variations of the intensity of neighboring pixels.
noise, high contrast and great spatial resolution. These gradient images are acquired by proposed by
Mathematical morphology [2-4] is a nonlinear branch F. Mayer [5] by using standard high pass spatial
of the signal processing field and concerns the filters [6-7] like Sobel, LoG, Prewitt etc. All though a
application of set theory concepts to image analysis. large number of research papers are found in various
Morphology refers to the study of shapes and journals, no study based on the modified LoG spatial
structures from a general scientific perspective. filter in marker controlled watershed segmentation is
Watershed algorithm is a powerful morphological available in published or online literature. In
tool to segment textured images into regions of morphological image processing the selection of edge
interest. This transform is adaptable to different types detecting spatial filters foe generating gradients is a
of images and capable to distinguish extremely key factor. In the present study a marker controlled
complex objects. Watershed based image watershed segmentation technique has been carried
137
out with the Laplacian and Gaussian (LOG) filtering The mathematical formulation of watershed
approach. An approach with modified LoG spatial algorithm is sated below.
filter having 9x9 mask have been used for generating
gradient images. The LOG filtering mask although is 3. MARKER CONTROLLED WATERSHED
a 5x5 matrix as proposed by Marr-Hildreth [11], the SEGMENTATION
mask can be extended dimensionally on the basis of
the need of the image processing situation. The
Though watershed algorithm a very good method
proposed gradient kernel is capable of removing the
for image segmentation, but this technique suffers
noise much better than the standard LoG spatial filter
from over segmentation problem. An approach used
and marker controlled watershed segmentation is
to control over segmentation is based on the concept
used to controlled over segmentation. Comparative
of controlled marker. The concept of markers is a
analysis also has been done in this research paper.
good approach to control over segmentation. The
The Gradient images obtained using conventional
markers are connected component of an image.
LoG filter are compared with Gradient images of
There are internal markers and external markers
proposed or modified Log spatial filter having 9x9
where i n t e r n a l markers a r e a s s o c i a t e d
mask. The final segmented results are also compared.
w i t h o b j e c t of interest and external markers are
The results obtained show the good performance of
associated with the background. This is a very useful
this approach.
technique for improvisation of watershed transform
to overcome the over segmentation problem. In this
This paper is divided into a choice of sections. In
technique, the set of the catchment basins of the
section 2 introduces the watershed algorithm. Section
grayscale image function f with values in [l ,l ]
3 describes the marker controlled watershed
is equal to the set Y obtained after the following
segmentation. Section 4 gives a brief description on
Laplacian of Gaussian spatial filter. Section 5 recursion: Y = T (f) , where T is the
presents the proposed or modified Laplacian of threshold set at level l
Gaussian mask. The proposed approach has been
discussed in section 6. Experimental results are = , ( ) , 1
discussed in section 7 and we nish this paper with (2)
some concluding remarks with section 8.
Where, MIN1 is the union of all regional minima at
2. WATERSHED ALGORITHM altitude l
And ( )= ( )
(3)
Where, =
{ | [1, ]{ }: ( , )< ( , )}
(4)
Figure 1: Watershed segmentation-local minima of gray ( , )= ( , )

level yield catchment basins, local maxima define the (5)
watershed lines.
( , ) represents the geodesic distance between a
Watershed algorithm is a tool for morphological
and b within A. Let us note in the set of markers we
image segmentation. A gray scale image can be
can define a function r as:
interpreted as the topographic image of landscape.

This is accomplished with the image intensity as an r(p) =
( )
altitude. Using the features of these images, the
(7)
technique of digital image processing called
Watershed Transform. It consists in placing a water Where p represents pixel coordinates and
source in each regional minimum or catchment basins denotes a new value dedicated for initial markers.
(CB), to flood the relief from sources, and build Now the recursion definition will be:
barriers when different sources are meeting. The = ( )
resulting set of barriers constitutes a watershed by
flooding, i.e., the set of pixels along which the gray = ( )
( ), 1
levels changes sharply gives rise to a watershed edge. (8)
138
4. LAPLACIAN OF GAUSSIAN SPATIAL FILTER which constitute the fuzzy set, there will be no gross
deviation from real situation and the sum of the
The Laplacian of Gaussian filter (LOG) plays a very weights inside the mask 0.47 which is sufficient to
important role in image segmentation. It is a help enhance a contour pixel.
convolution filter that is used to detect edges of
different objects. This filter first applies a Gaussian
blur, then applies the Laplacian filter and finally
checks for zero crossings, i.e. when the resulting
value goes from negative to positive or from negative
to positive.
The main objective of this filter is to highlight edges
different objects. As a input the LOG operator takes a
single gray level image and produces another binary
image as output. A 5x5 mask LOG filter has been
shown in Figure 3.
Figure 3: Proposed 9x9 Laplacian and Gaussian

filter.
7. PROPOSED APPROACH
Watershed segmentation is an effective method for
image segmentation. The flowchart of the present
algorithm is stated below. This approach is based on
the concept of LoG spatial filter with markers. In first
Figure 2: A conventional 5x5 LOG surround filter. step of present algorithm three real life color images
has been chosen and accordingly converted into a
gray scale image in second step. The gray scale
5. PROPOSED OR MODIFIED LAPLACIAN OF image also called as black and white image only
GAUSSIAN FILTER contains the intensity information of an image
varying from black at the weakest intensity to white
A 9x9 mask has been proposed in the present work. at the strongest. In third step the gradient image is
There are two annular regions in this mask, i.e., the computed using proposed or modified LoG spatial
inner being excitatory and the outer being inhibitory filter having 9x9 mask in present study. The gradient
in nature which surrounds the central pixel where the magnitude image has high pixel values along object
positive weights (excitatory) are assigned for inner edges and low pixel values everywhere else.
one and negative weights (inhibitory) are assigned for
outer one. The influence of a point in the surround on Read the color image
the cell respond depends on its distance from the
centre and the nature of variation approximately Convert the image into gray
follows a Gaussian profile and also the excitatory scale
influence is weaker than the inhibitory influence [18],
[1].
Obtain gradient image using
After large number of trials using masks of larger Proposed 7x7 LoG mask
dimensions, the size of the optimal mask was
obtained and found to have 9x9 in dimensions. There
is no loss of generality if the weights of individual Enumerate internal markers
pixel in the masks are changed slightly, keeping in
the mind basic requirement of being isotropic and Enumerate external
ability to emulate HVS. The mask has been shown in markers
Figure 4. Where, by sampling a Gaussian profile of
standards deviation 1.4 the weights in the outer
inhibitory region of the optimal mask were found. If Apply watershed algorithm
anybody slightly alters the pixel values in the mask Figure 4: Flow diagram of the proposed approach.
139
number of such segmented regions are found to be

Next step of present approach computes the markers. much more less than those obtained by conventional
In an image marker is a connected component. There 5x5 LOG filter and marker controlled watershed
are two types of markers in an image, internal segmentation, which are final segmented images
markers and external markers. The set of internal shown in figure 14, 15 and 17 . It is also observed
markers are associated with objects of interest or with that the watershed ridge in final segmented images
the foreground objects and set of external markers are (form figure 18, 19 and 20) lines are very prominent
associated with the background. These markers are in spite of the number of regions are large.
used to modify the gradient image which is obtained From the statistical point of view, the gradient images
by applying proposed special edge detecting filter. with conventional LoG spatial filter produces lesser
So, first internal markers are computed and then Entropy and MSE with higher PSNR then the
external markers are com computed and finally the gradient images generated using proposed 9x9 LoG
watershed transform is computed in the final step. In mask. Contrary, the final segmented images with
final segmented images watershed ridge lines are conventional LoG mask produces higher Entropy and
superimposed on the original image. MSE with lesser PSNR then the final segmented
images generated using proposed 9x9 mask.
7. EXPERIMENTAL RESULTS AND
DISCUSSION It is also observed that 9x9 LoG mask with marker
controlled watershed segmentation (proposed
approach) produce better results in respective of the
Three real life images have been chosen to carry out statistical analysis and human perception. Thus, in
the image segmentation using watershed method. The this research work, the proposed approach has been
images are fruits of 500x333 dimensions, flower of found to be most efficient in reducing various types
700x525 dimensions and stones of 400x300 of noises.
dimensions. The watershed lines have been
superimposed on the edge detected images. The
original images are shown in from figure 5 to figure7
The gradient images obtained by using conventional
LoG spatial filter having 5x5 mask are shown in from
figure 8 to figure 10 respectively and there
corresponding final segmented results using marker
controlled watershed segmentation are shown in
from figure 11 to figure 13 accordingly. The gradient
images obtained by 9x9 LoG spatial filter are shown Figure 5: Original image of fruits.
in from figure 14 to figure16 respectively and their
final segmented images have been shown in Figure
17 to Figure 29 accordingly. The statistical analyses
of the segmented images have been calculated in
respective of Entropy, PSNR and MSE and are
shown in table 1.
From the segmented images its experiential that the
gradient images for all three images using 5x5 LOG
filter have produced a dark shed images, i.e., the gray
level of the pixel have been lower down and also the
edges in the images are not very sharp. On the other
hand the gradient images obtained by 9x9 LOG filter Figure 6: Original image of flower.
for three figures have given rise to images of more
whitish appearance. In these cases the edges and
boundaries of the images are also seen to be more
prominent and clear with better contrast. It have
been observed form figure 18,19 and 20 which are
final segmented images obtained by the proposed or
modified LoG spatial filter having 9x9 mask and
marker controlled watershed segmentation (proposed
approach) that each and every segment of the final
watershed segmented form of the has occupied an
appreciable area and size in the image region and the Figure 7: Original image of stones.
140
Figure 8: Gradient image of fruits with conventional LoG Figure 12: Gradient image of flower with proposed 9x9
mask. LoG mask.
Figure 13: Gradient image of stones with proposed 9x9

LoG mask.
Figure 9: Gradient image of flower with conventional LoG
mask.
Figure 14: Marker controlled watershed segmented image

of fruits with conventional Log mask.
Figure 10: Gradient image of stones with conventional LoG
mask.
Figure 11: Gradient image of fruits with proposed 9x9 LoG Figure 15: Marker controlled watershed segmented image
mask. of flower with conventional Log mask.
141

of stones with conventional Log mask.
8. CONCLUSION
Figure 17: Marker controlled watershed segmented image In this research paper deals with the problem of
of fruits with proposed 9x9 Log mask. morphological image segmentation. segmentation,
for which the marker controlled watershed algorithm
has been one of the most widely accepted solutions.
An effective methodology for digital color image
segmentation has been publicized in this paper with
LoG spatial filter and marker controlled watershed
algorithm. The proposed methodology has been
successfully tested on three real life digital color
images. The results demonstrates that the
arrangement of modified LoG spatial filter which is
proposed in this paper with marker controlled
watershed algorithm has been very effective for
image segmentation and for computer vision. The
experimental results and statistical measurements
confirm the efficiency of the proposed methodology.
of flower with proposed 9x9 Log mask.
DEDICATION
One of the others (Dibyendu Ghoshal) dedicates the

entire study to the loveliest and loving memory of his
only one and younger sister Kumari Sumita Ghoshal
who herself was a gem of the scholars, a symbol of
wisdom and art, peerless beauty and simplicity,
unfathomable knowledge and generosity.
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16. A. N. Moga and M. Gabbouj, "Parallel image
component labeling with watershed transformation,"
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intelligence, vol. 19, pp. 441-450, 1997.
17. J. Roerdink and A. Meijster, "The watershed
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strategies," Mathematical Morphology, vol. 41, pp.
187-S28, 2000.
143
Study Of Watershed Approach For

Morphological Based Gradient Image
Acquisition
Dibyendu Ghoshal
Associate Professor, Department of ECE
National Institute of Technology
Agartala, Tripura, Pin - 799055
tukumw@gmail.com
Pinaki Pratim Acharjya

Assistant Professor, Department of CSE
Bengal Institute of Technology and Management
Santiniketan, West Bengal, Pin 731236
ppacharjya@gmail.com
Abstract - In recent days many digital image Prewitt, Roberts etc. In this research paper, the
segmentation technique have been found in published gradient image is generated using morphological
or online literature, among them morphological approach. The image smoothing [13-16] technique is
image segmentation methods have found wide also introduced with mathematical morphology.
applications. In this research paper an approach of This paper is divided into various sections. Section 2
image segmentation using mathematical morphology introduces the morphological gradient. In section 3
is propose . The proposed approach has been applied morphological smoothing is presented. A brief
to images with almost similar objects and the description on Watershed algorithm and its
segmentation results are found to be very efficient mathematical formulations are given in section 4.
and encouraging. Section 5 presents the proposed scheme or approach.
The experimental results are discussed in section 5
and finally conclusion is discussed in section 7.
Keywords- Image segmentation, morphological
gradient, morphological smoothing, watershed 2. MORPHOLOGICAL GRADIENT
algorithm.
Morphological gradient can be generated using
1. INTRODUCTION dilation and erosion. Dilation gives the original set
plus an extra boundary, the size and shape of the
Mathematical morphology [1-3] is a nonlinear branch boundary depends on the shape and size of the
of the signal processing field and concerns the structuring element. Erosion gives the points for
application of set theory concepts to image analysis. which the structuring element is contained in the
Morphology refers to the study of shapes and original set. The outer boundary of the original shape
structures from a general scientific perspective. Many is removed by erosion. The morphological gradient is
morphological techniques have been proposed in the generated by subtracting an eroded image from its
published or online literature [4-10] for the dilated version. The morphological gradient
segmentation of digital images. Morphological image
highlights sharp gray-level transitions in the input
segmentation methods, particularly those using a
image.
watershed algorithm [7-10], have found wide
= ( )
applications.
( ) (1)
In this research paper an effective approach based on
mathematical morphology is presented. In 3. MORPHOLOGICAL SMOOTHING
conventional approaches the watershed algorithm is
applied on the gradient images [10-13] for image One way to achieve smoothing is to perform a
segmentation purpose. The gradient images are morphological opening followed by a closing.
usually generated using spatial high pass filters [12- Opening smoothes the contour by removing thin
13] like, Sobel, Laplacian of Gaussian (LoG), bridges and eliminating thin protrusions.
144
= ( ) [ ] = ( 1) ( 2) . . . . . . ( )
(2) (7)
Closing also smoothes the contour, but by enforcing [ + 1] = ( 1) ( 2) . . . . . . ( )
bridges and closing small holes. (8)
If the algorithm keeps on increasing flooding level
= ( ) then Cn(Mi) and T[n] will either remain constant or
(3) increase. Algorithm initializes [ + 1] =
The boundary of opening with a circular structuring [ + 1], and then precedes recursively by
element corresponds to rolling a ball on the inside of assuming that at step n C [n - 1] has been
the set. The boundary of closing corresponds to constructed.
rolling a ball on the outside of the set. Let, G is a set of connected components in T[n] and
for each connected component g G[n], there
4. WATERSHED TRANSFORM possibilities will arise.
1. g C[n - 1] is empty.
In mathematical morphology watershed transform is 2. g C[n - 1] contains one connected component of
the technique which is commonly used in image C[n - 1].
segmentation. A gray level image can be interpreted 3. g C[n - 1] contains more than one connected
as the topographic image. It can be assumed as a component of C[n - 1].
landscape where the gray intensities of higher
magnitude corresponds local maxima, whereas those
5. PROPOSED APPROACH
of lesser value are local minima or catchment basins.
When a landscape or topographic relief is flooded
The proposed image segmentation approach here is
with water, water will be filled up at these starting
able to segment the digital images with minimum
local minima and at points where water coming from
drawbacks of over segmentation. The steps of the
different basins would meet and dams will be built.
proposed approach are shown in figure 2. In first
When the water level reaches the highest peak in the
step, a color image is chosen for the processing
landscape the process is stopped. As a result, the
purpose and converted into gray scale image in the
landscape is partitioned into regions or basins
second step.
separated by dams, called watershed ridge lines or
simply watersheds. The mathematical formulation of
Gray scale images, also very popularly known as
watershed algorithm is sated below.
black-and-white images are mainly combination of
two colors, black and white. It carries the intensity
Assume, Mi where i= 1 to n be the set of coordinates
information where, black have the low or weakest
points in the regional minima (catchment basins), of
intensity and white have the high or strongest
the image P(x,y) and C(Mi) be the coordinates points
intensity. In third step the gradient image generated
of catchment basins associated with the regional
using morphological approach with dilation and
minima Mi
erosion. The gray scale erosion is performed using
= {( , ) | ( , ) < }
flat structural elements. A square element is applied
(4)
for the erosion purpose in this proposed approach.
Where,
The morphological gradient is generated by
T[n] = set of points in P(x,y) which are lying below
subtracting an eroded image from its dilated version.
the plane p(x,y) = n
min, max = minimum or maximum gray level value.
Watershed algorithm produces over segmentation if it
n = stage of flooding varies from min + 1 to max + 1
directly applied on gradient images. So in fourth step,
Let Cn(M1) be the set of points in the catchment basin
morphological smoothing operation is done on the
associated with M1 that are flooded at stage n.
gradient image. To achieve smoothing, an opening
( 1) = { ( 1), [ ]} operation is performed followed by a closing.
(5) Opening smoothes the contour by removing thin
Where, bridges and eliminating thin protrusions. Closing also
1, ( , ) ( ) ( , ) [ ]
( )= smoothes the contour, but by enforcing bridges and
0, closing small holes.
(6)
C[n] is the union of flooded catchment basin portions
In next step watershed transform is done and the final
at the stage n.
segmented map is obtained in the sixth and final step.
Where,
145
Read the color image analysis. The statistical measurements are also
included and shoewn in table 1. They are analyzed in
respective of entropy, peak signal to noise ratio
Convert the image into (PSNR) and mean square error (MSE).
gray scale Experimental results presented in this paper are
obtained by using MATLAB.
Obtain gradient image
using morphological
approach
Morphological smoothing
Apply watershed algorithm
Final segmented map

(a)
Figure 2: Flow chart of the proposed methodology.
7. EXPERIMENTAL RESULTS AND
DISCUSSION
The results obtained by the application of proposed

approach on three real life images. The images are,
Bird of 500 x 400 dimensions, Elephant of 600 x
450 dimensions, Zebra of 500 x 375 dimensions
and accordingly shown in figure , 3(a), 3(b) and in
3(c). The gradient of the original images are has been
computed using morphological approach. It is (b)
observed from the gradient images that the edges of
the foreground objects are very sharp, prominent and
almost accurate in human vision.
The gradient images, those are generated from

original images are shown in figure, 4(a), 4(b) and in
4(c). Morphological smoothing operation is also
included in proposed approach, applied on gradient
images and shown in figure 5(a), 5(b) and in 5(c). For
comparative study analysis, first the watershed
algorithm is directly applied on the gradient images
for the segmentation purpose and the segmented
results are shown in 6(a), 6(b), and in 6(c). The (c)
watershed ridge lines are superimposed on the
Figure 3: Original images, (a) Bird, (b) Elephant, (c) zebra.
segmented images. And secondly, watershed
algorithm is applied on the morphologically
smoothed images and shown in figure, 7(a), 7(b), and
in 7(c).
Here the watershed ridge lines are also super imposed

on the segmented images. It is observed from the
segmented images of figure 6, where watershed is
directly applied on gradient images are extremely
over segmented. Contrary the images of figure 7,
acquired by applying watershed algorithm on the
morphologically smoothed images, are less over
segmented. Thus, the proposed approach is very
(a)
efficient and encouraging for image study and
146
(b) (c)
Figure 5: Gradient images after morphological smoothing
operation, (a) Bird, (b) Elephant, (c) zebra.
(c)
Figure 4: Gradient images obtained by morphological

approach, (a) Bird, (b) Elephant, (c) zebra.
(a)
(a)
(b)
(b)
(c)
147
Figure 6: Segmented images obtained by applying image segmentation techniques, mostly those using a
watershed algorithm directly on gradient images, (a) Bird, watershed transform, have found in various
(b) Elephant, (c) zebra. applications of image processing. The proposed
approach that is presented in this paper is fully
developed using mathematical morphology and has
been applied to images with almost similar objects
and the segmentation results are found to be very
efficient and encouraging.
DEDICATION
One of the others (Dibyendu Ghoshal) dedicates the

(a) entire study to the loveliest and loving memory of his
only one and younger sister Kumari Sumita Ghoshal
who herself was a gem of the scholars, a symbol of
wisdom and art, peerless beauty and simplicity,
unfathomable knowledge and generosity.
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148
Image Bin Based Full-Reference Color Image

Quality Measure Using SM-DCT
D.V.N.SUKANYA, Asst.Prof OF E.C.E, Vignan Nirula Institute Of Technology And Science for Women, sukanyadvn@gmail.com
D.NAGESH BABU, Asst.Prof OF CSE, ST.Anns College of Engineering And Technology, dasarinagesh@yahoo.co.in
N.SWATHI, Asst.Prof OF E.C.E, Vignan Nirula Institute Of Technology And Science for Women, swathi.nallapati@gmail.com
Abstract: In this paper, we present a new objective quality

measure for color images. In any frequency domain
transform, the coefficients in different frequency bands
have different magnitudes. The 2-dimensional Discrete
cosine Transform (DCT) separates a given image into
low, medium and high frequency ranges. After splitting
the image into 16 subband, compute the DCT and
standard deviation and form an array with the16 SD
values. Find the difference for the arrays formed by both
original and degraded images. The proposed measure is
defined as the mean of sixteen SD values. Correlation of
the subjective ratings and objective scores gives the
performance of the measure. A comparison with the peak
signal- to-noise ratio (PSNR), and two state-of-the art
metrics, Q and MSSIM, shows that our results correlate
better with the judgment of human observers.
I. INTRODUCTION
An important criterion used in the classification of

image quality measures is the type of information
needed to evaluate the distortion in degraded
images. Measures that require both the original
image and the distorted image called full
reference or non-blind methods, measures that do
not require the original image are called no- Two of the state-of-the-art image quality metrics are
reference or blind methods, and measures that the universal image quality index (Q) [9] and the
require both the distorted image and partial Structural Similarity Index (SSIM)[10]. The
information about the original image are called universal image quality index, Q, is defined as
reduced-reference methods.
Although no-reference measures are needed in some
applications in which the original image is not
available, they can be used to predict only a small
Where xi, yi, i = 1,...n , represent the original and
number of distortion types. In the current literature,
distorted signals, respectively
a few papers attempt to predict JPEG compression
artifacts [1,2,3,4], and others blurring and JPEG
2000 artifacts [5,6]. Reduced-reference measures are
between full-reference and no-reference measures;
[7] evaluates the quality of JPEG and JPEG2000
coded images whereas [8] provides assessment for
JPEG and JPEG2000 compressed images, images
distorted by white Gaussian noise, Gaussian blur,
and the transmission errors in JPEG2000 bit
streams. The applicability of full-reference measures The dynamic range of Q is [-1,1], with the best
is much wider. They can be used to estimate a value achieved when yi = xi, i = 1,2,,n. The index
spectrum of distortions that range from blurriness is computed for each window, leading to a quality
and blockiness to several types of noise. Recent map of the image. The overall quality index is the
examples of such measures are given in Table 1. average of all the Q values in the quality map:
149
version of 512x512 Lena. Table 2 shows the tools

and parameters for six degradation types, and five
degradation levels. Note that all of these
degradations were performed in the pixel domain.
In order to circumvent this problem, the quality

measure that has been generalized to the standard
objective quality assessment metric known as
Structural Similarity Index (SSIM), it is defined by
the formulae given below:
Q is a special case of SSIM that can be derived by

setting C1 and C2 to zero. As in the case of Q, the
overall image quality MSSIM is obtained by
computing the average of SSIM values over all
windows:
High quality print-outs of 30 distorted full color

images were subjectively evaluated by 14 observers.
II. NEW IMAGE QUALITY MEASURE: The printer was a Hewlett-Packard printer with
SM-DCT model number hp color Laser Jet 4600dn. The 8-
2/16x8-2/16 images were printed on 8.5x11
For a given frequency domain transform (e.g., DCT, white paper with the basis weight 20lb and
DWT and DFT), the coefficients in different brightness 84.
frequency bands have different magnitudes. The
process of separating the frequency bands using the The observers were chosen among the Graduate
DCT is well-defined. In two-dimensional DCT, students and instructors from the Department of
separates a given image into low, medium and high Computer and Information Science at Brooklyn
frequency ranges. The YUV color model is a linear College. About half of the observers were familiar
transformation between the gamma-corrected RGB with image Processing and the others only had
components that produces a luminance signal and a computer science background.
pair of chrominance signals. A common approach
employed in developing a quality measure for color They were asked to rate the images using a 50-
images is to use only the luminance signal. point scale in two ways: Within a given distortion
type (i.e., Rating of the 5 distorted images), and
Our proposed algorithm is as follows: across six distortion types (i.e., rating of the 6
distorted images for each distortion level). As the
1. Split the original image into four bands. Again proposed measure is not HVS-based, no viewing
subdivide each band into four sub bands. distance was imposed on the observers in the
2. Apply DCT and compute standard deviation for experiment. Grade 1 was assigned to the best
the sixteen sub bands. image, and grade 50 was assigned to the worst
3. Form an array for original image with those image.
sixteen SD values.
4. Perform the above two-steps for degraded image In the Video Quality Experts Group (VQEG) Phase
also. I and Phase II testing and validation, a nonlinear
5. Form an array for degraded image with those mapping was used between the objective model
sixteen SD values. outputs and subjective quality ratings [13]. The
6. Find the absolute values for the difference of the performance of each proponent model was evaluated
two arrays. after compensating for the nonlinearity. To establish
7. Compute the mean of the sixteen absolute values. a nonlinear mapping, we followed the same
The measure was applied to a full color, 24-bit procedure by fitting the logistic curve
150
We will compare the performance of SM-DCT,

PSNR,two state-of-the-art metrics Q and MSSIM
with MOS. Figures below shows the scatter plots for
the four measures using the luminance layer of Lena
for thirty images. Each marked point represents the
corresponding values in Tables 3 and 4.
SM-DCT
Below table 5. gives correlation values of subjective

rating with the objective scores of various
techniques. Table 5. correlation values
III. CONCLUSIONS
We presented a new color image quality measure

based on the DCT. As it does not incorporate a
Human Visual System (HVS) model, we do not use
any assumptions regarding the viewing distance. In
the experiments, a wide range of distortion
types was used. For each distortion type, five levels
of distortion were introduced. Although PSNR is
still widely used by researchers, our results indicate
151
that it is an unreliable measure, especially for 11. D. Van der Weken, M. Nachtegael and E. E. Kerre, A new
similarity measure for image processing, Journal of
correlation across distortion levels. The performance
Computational Methods in Sciences and Engineering, Vol.3,No. 2,
of SM- DCT has a perfect match with the quality pp. 209-222, 2003.
perceived by human observers. The Proposed 12. A. Beghdadi and B. Pesquet-Popescu, A new image distortion
measure is also superior to state-of-the-art metrics Q measure based on wavelet decomposition, 7th International
Symposium on Signal Processing and Its Applications, Paris,
and MSSIM. In future work, we will use more color
France, July 1-4, 2003.
images, and extend the measure for evaluating the 13. A. M. Rohaly, J. Libert, P. Corriveau, and A. Webster
quality of watermarked images and video sequences. (editors),Final Report from the Video Quality Experts Group
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Assessment,March 2000.
1. A. C. Bovik and S. Liu, DCT-domain blind measurement of
blocking artifacts in DCT-coded images, Proceedings of
International Conference on Acoustics, Speech, and Signal
Processing, Salt Lake City, UT, May 7-11, 2001.
2. Z. Wang, A. C. Bovik and B. L. Evans, Blind measurement of
blocking artifacts in images, Proceedings of IEEE 2000
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152
Filtering Respiration and Low Frequency

Artifacts in Impedance Cardiography
Dr. K. Chandra Bhushana Rao1 , Ms.Madhavi Mallam2 , Dr.A.GuruvaReddy3
1
Associate Professor, Dept of ECE
JNTUK-University College of Engineering, vizianagaram.
2
Research Scholar at JNTU K- Kakinada.
3
Department of Electronics & Communications
1
cbraokota@yahoo.com
2
madhavi.418@gmail.com
3
guruvareddy78@gmail.com
AbstractImpedance cardiography is a non-invasive ensemble averaging but this suppresses the beat-to-
technique for monitoring stroke volume, based on beat variations that occur during respiration (Permutt
sensing variation in the thoracic impedance, z(t), due to and wise, 1986; Wang et al., 1991) and cardiac
blood flow.In this paper, we present two additions to the arrhythmias. Wang et al. (1991) have shown that the
LMS algorithm in order to improve the beat by beat r-r interval changes during different phases of the
estimation of fiducial points (B and X) and the respiratory cycle. if the r-z interval (time from the r-
maximum value of the derivative in impedance
wave of the e c g to the peak of dz/dt) changes
cardiography signal: addition of a reference input and
scanning each beat more than once. On data synthesized proportionally, then the peak of dz/dt will occur at a
from real ICG which exhibits beat by beat changes and different time with respect to the peak of the r-wave,
baseline variations, we used statistical measures to show hence, ensemble averaging will smear the peak of
that the modified algorithms give accurate beat by beat dz/dto
estimation conversely to the original version of LMS.
When accurate determination of cardiac output is
crucial, impedance cardiography may be used in
conjunction with a standard technique to establish a
baseline reference, thereby permitting further analysis.
The ease and precision of this technique warrants its
more widespread use in the assessment of pacemaker
patients.
Keywords: Filtering, LMS algorithm ,Respiratory

artifact,
Impedance cardiography.
I. INTRODUCTION
IMPEDANCE cardiography is a noninvasive

technique for monitoring stroke volume, based on Fig. 1. Typical impedance cardiogram
sensing the changes in the electrical impedance of the
thorax, z(t), caused by variation in blood volume
during the cardiac cycle [1]-[16].Time derivative of Yamamoto et al. 1988 have proposed a digital
the thoracic impedance (-dz/dt) is known as narrow band pass filter centred around the heart rate
impedance cardiogram (ICG). The parameters with an attenuation of -2.0db at the respiratory
required for estimating the stroke volume, using frequency for the removal of artefacts. the use of a
Kubicek, Bernstein,or Sramek formulas [1], [2], [11] band pass filter means that the higher harmonics in
are left ventricular ejection time (Tlvet) and the cardiac signal will be attenuated.
maximum change in the impedance cardiogram, (-
dz/dt)max. The signal-processing technique developed by Eiken
Patterson (1978), Muzi et al. (1985) and and Segerhammar (1988) uses a time window
Miyamoto et al. (1981) have suggested the use of technique to identify the breathing component . their
153
technique is computationally cumbersome and slow. was used to eliminate the breathing component
because it allowed averaging over exactly two
The purpose of our research was to develop a complete breathing cycles and ten cardiac cycles
digital filter that can remove respiratory and locked in phase with the respiration cycle. For
movement artefact signals without attenuating the normals at rest and during exercise, the ratio of
cardiac signal or causing phase distortion. heartbeats to respiratory cycles is between 4 : 1 and 6
In Impedance Cardiography, there is partial : 1 (hildebrandtand Dauman, 1957; Almasi,1972).
overlap between z(t) and respiratory artifact spectra Asking a subject to breathe at a fixed ratio of 5 " 1
and statistical properties of both vary. hence non- heartbeats per respiratory cycle does not cause any
adaptive digital filter is not effective in removing significant stress or discomfort. A tone generator
respiratory artifact from z(t). spectral subtraction triggered by the ECG was turned on for two beats,
method [21] which involves estimating noise signalling inspiration, and off for three beats,
magnitude spectrum and subtracting it from the signalling expiration. Before designing the digital
contaminated spectrum, cannot be applicable here filter the power spectra of the dZ/dt signals were
because respiratory artifact spectrum cannot be examined. The first spike in the power spectrum
dynamically estimated using average or quantile corresponds to the breathing frequency and the higher
based estimation. However, an adaptive filter [22], spikes correspond to the cardiogenic components.
[23] can be employed to dynamically change its Examining the spectrum shows that a filter with a
transfer function to remove respiratory artifact. very narrow transition band is required. A higher-
Respiratory artifact is the variation in the sensed order filter (sixth order) was chosen because it would
thoracic impedance, caused primarily by change in be able to filter out breathing artefacts without
the dimension of thoracic cage during inhale and causing significant attenuation of the cardiac signal.
exhale phases of respiration. Hence air flow during A Butterworth filter was chosen because of its fiat
respiration is directly related to the respiratory frequency response in the pass band and the stop
artefact and can be used to provide a reference for band. A recursive filter was used instead of a finite
adaptive cancellation of respiratory artifact from the impulse response (FIR) filter to avoid distortions that
z(t) signal. the respiratory signal acquired from a can be introduced due to windowing. Another
respiratory sensor, simultaneously with z(t) signal, is advantage of an infinite impulse response (IIR) filter
taken as reference input, contaminated z(t) being the is the computational economy.
primary input. the objective is to adaptively remove The IIR filter designed had a nonlinear phase shift
the respiratory artefact from the z(t) signal without which could cause distortions in the waveform.
having to control the respiration. Advantage of this Back filter-ing was used to eliminate the distortions
method is that it does not require any control of that could be caused by the phase shift. In back
respiration. Respiratory signal can be acquired during filtering the data are passed through the filter twice,
exercise. Hence beat-by-beat stroke volume first in a normal manner, forward in time, and then
calculation is possible, even if respiratory artifact has in a reverse manner, backward in time, which
large variation. Also, beat-by-beat variation in stroke cancels the phase shift caused by the forward filter.
volume and Tlvet may provide additional diagnostic
information about cardiovascular functioning. It is to A Signal Processing
be noted that the technique suppresses additive
respiratory artifact. The stroke volume itself gets We acquired four signals, each sampled at
modulated by the respiratory cycle. This modulation 1000 Hz and processed in 1-min segments:
should remain unaffected by the proposed technique. respirometry, spirometry, dZ/dt, and Z. Because the
Z signal is coupled inductively to the output of the
II. METHOD Minnesota impedance device, this signal is subject to
low frequency filtering and distortion. Consequently,
The standard reference Z and dZ/dt signals used we derived a measure of Z ( Zd) by integrating the
to compare with the different filtered outputs signals dZ/dt signal (to reconstruct Z undistorted by this
was obtained by ensemble averaging ten complete inductive filtering). The Zd displayed a closer
cardiac cycles over two complete respiratory cycles. correspondence to direct spirometric measures of
Voluntary cardio respiratory synchronisation respiration than did the Z output of the Minnesota
(VCRS), developed by Almasi and Schmitt (1970), device and was used for the primary analyses.
was used to achieve a lock between the respiratory After decimating each signal to 250 Hz, we applied a
and the cardiac signal and time stationarily. With this point by-point integration to the dZ/dt signal and to
technique the breathing cycles are synchronised with the spirometry signal. All four signals respirometry
the cardiac cycles in a predetermined ratio. VCRS Z, Zd, and spirometry were low pass filtered (0.5
154
Hz), using an interpolated finite impulse response filter to shift the coefficients to match for the
filter, to remove cardiac signals and high frequency difference in the delay. Adaptive filter output r(n) is
noise. subtracted from contaminated input z(n) so that the
All signals were then linearly detrended and total output error e(n) is progressively minimized.
zero-meaned, and a power spectrum Fast Fourier Fig. 2.Fig. 2. Adaptive respiratory artifact canceller
Transform (FFT) was calculated for each minute to For adaptation, we use the least mean square (LMS)
decompose the signals in the frequency domain. This algorithm, also known as stochastic gradient
method provided descriptive characteristics of the algorithm [22], [23]. The LMS algorithm is used to
respiration, and served as the basis for weighting of control the digital FIR Filter.
transfer functions as described below. In addition, a
Hanning window was applied to each of the four C. Signal Analysis
signals, and transfer function analyses were used to
compare Zd, Z, and respirometry (outputs) to the The recorded thoracic impedance signal was low
criterion spirometry signal (input). Transfer function pass filtered to attenuate high frequency noise. A 2-
analyses were run on each 60-s epoch, and were point differentiation was used to obtain ICG
based on seven 15-s overlapping windows to waveform. The resulting ICG waveform exhibited
minimize problems of nonstationarity and to smooth strong presence of respiratory artifact, particularly in
the resulting functions (Oppenheim & Schafer, 1989). the post-exercise recordings. The adaptive
These analyses yielded estimates of the parameters of cancellation scheme of Fig. 2 was implemented for
coherence (similarly of waveform morphology), gain different combination of filter tap length and delay in
(amplitude ratio) and phase relations of the signals. the primary signal path. Based on maximum eigen
The transfer functions provide parameter estimates value of autocorrelation matrix of the input signal
over the entire range of frequencies within the z(n), = 0.0044 was used for adaptation. Most
analytical window (0.050.50 Hz), whereas the satisfactory results were obtained for tap length M
distribution of respiratory power was more restricted. 200 and sample delay 70. The filter coefficients
Because respiratory frequencies are not uniformly were found to get stabilized in approximately 500
distributed across the frequency spectrum, the samples.
transfer functions were weighted using the output Because peak dZ/dt is an important parameter in
power spectrum. This method of weighting provided the calculation of stroke volume, the percentage error
more meaningful summary statistics of these was cal-culated for this parameter. The peak height of
parameters, appropriately scaled by the actual the unfiltered ensemble averaged waveform was
frequency distribution of power (for general rationale taken as the standard signal. The same data were then
for the weighting procedure, see Porges et al., 1980). passed through the filter with cutoff frequencies
The value of each point of the transfer function was selected as a fraction of the heart rate. The percentage
multiplied by the relative amplitude of the power error was then calculated between the peaks of the
spectrum of the output signal at that frequency standard waveforms and the filtered data with various
(expressed as a proportion of total cut-off frequencies.
power). This calculation yielded an overall average
estimate of the transfer function parameter, The percentage peak dZ/dt error plotted against
appropriately weighted by the relative power of the different cutoff frequencies. The percentage peak
specific respiratory frequency contributions to that dZ/dt error for all conditions was less than 2 percent
estimate. for a cut off frequency of less than 50 per cent of the
heart rate.
B. Adaptive Cancellation Scheme
A schematic of adaptive respiratory artifact
canceller is shown in Fig. 2. Signal z(n) is the
thoracic impedance signal contaminated by
respiratory artifact from the thoracic impedance
sensor. Reference signal r(n) is obtained from the
respiratory sensor. The respiratory sensor used here
senses the airflow at nostrils, and hence the reference
signal has a delay with respect to the respiratory
artifact because of the movement of thorax cage. To
compensate for this, a delay of nd samples is
introduced in signal path of z(n). Number of taps in
the FIR filter should be large enough for the adaptive
155
III. RESULTS Fig. 4. Processing outputs for waveforms in Fig. 3:

(a) ICG from z(t), (b) z(t) after adaptive cancellation,
Fig. 3 shows the recorded z(n), ECG, PCG, and (c) ICG obtained after the removal of respiratory
respiration waveforms from a post-exercise recording artifact.
on one of the subjects. Fig. 4 shows the processed
outputs. First waveform in this figure is the ICG
obtained by 2-point differentiation of z(n). It is seen IV. CONCLUSION
to be contaminated by respiratory artifact, making it
difficult to detect Tlvet and ICG peaks appear to For attenuating respiratory artifact present in ICG
change from cycle to cycle. The second waveform in signal, LMS based adaptive respiratory artifact
Fig. 4 is the impedance waveform after adaptive canceller is implemented. This method uses signal
cancellation of the artifact. ICG obtained by from respiratory sensor as the reference. The best
differentiating this waveform shows almost no effect range for the cut off frequency appears to be from 30
of respiration, making it easy to detect the B and X to 50 per cent of the heart rate under all three
points and these are found to be consistent with conditions. In this range the maximum attenuation of
simultaneously acquired PCG. Values of ICG peaks the first cardiac component in the frequency spectrum
are found to be stable.For an estimation of was about 11 percent and the minimum attenuation in
effectiveness of the adaptive cancellation in removing the breathing component was 90 percent. Advantage
the respiratory artifact, spectra of the original ICG, of this method is that it does not require any control
the respiratory reference signal, and ICG obtained of respiration and can be used for cancellation of
from cleaned waveform were compared . respiratory artifact.
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Authors Biography
Dr. K. Chandra Bhushana Rao is working as Associate

Professor, Dept. of ECE, JNTUK-University College of
Engineering, VIZIANAGARAM. He received his B.E (ECE)
degree from the Andhra University, Visakhapatnam, M.E
(Communication Systems) from the Andhra University,
Visakhapatnam, and Ph.D from Andhra University,
Visakhapatnam. His area of interests is Microwave
Communications and Signal Processing.
Mrs. M. Madhavi is working as Associate Professor, Dept. of

ECE, DVR & Dr.HS MIC College of
Technology, Kanchikacharla, Krishna (D). She received her
B.Tech, from JNTU Hyderabad and M.Tech (DECS) from
Jawaharlal Nehru Technological University, Hyderabad in 2008.
Her areas of interests is Signal Processing and communication.
Dr. A. GuruvaReddy is working as Professor, Dept. of ECE,

DVR & Dr.HS MIC College of Technology, Kanchikacharla,
Krishna (D). He received his B.E (ECE) degree from the
University of Mysore, Karnataka, M.E (Communication Systems)
from the Anna University and Ph.D from JNTU Ananthapur. His
area of interests is Image Processing and Signal Processing
157
IRIS Pattern Matching and Recognition for

Eunuch in India by the Method of Cumulative
Sum Based Change Analysis
1. Parthsarathi De,Assistant Professor,CSE Department,NIT Agartala
2. Dr.Dibyendu Ghoshal,Associate Professor ECE Department NIT Agartala
3. ParamitaMalla, Faculty Associate, ICFAI University Tripura
Abstract-Iris recognition is considered as one of the

most preferable and efficient biometric techniques for
personal identification. This identification is achieved
in this work by using the concept that, the iris patterns
are statistically unique and suitable for biometric
measurements. In this study, a novel method of
recognition of these patterns of an Eunuch (Hijra) iris is
considered by using Cumulative sum based change
Analysis. It is capable of excellent pattern recognition Fig 1. Flow Diagram of Proposed Iris Recognition
properties as the iris texture is unique for every
person used for Recognition and experimental Step1. Image Acquisition
results shows that this iris recognition system produce
very low error rates. First of all different images of an eye of any Eunuch
are taken from different angles and among these
I. INTRODUCTION the best eye image is selected for iris segmentation
for further processing which would lead to iris
A eunuch is a man who (by the common definition recognition in a meaningful way.
of the term) may have been castrated, typically early
enough in his life for this change to have major Step2. Preprocessing
hormonal consequences. Less commonly, in An iris image contains some irrelevant parts such as
translations of ancient texts, "eunuch" may refer to a eyelids sclera and pupil. Also, the size of an iris may
man who is not castrated but who is impotent, vary depending on camera-to-eye distance,
celibate or otherwise not inclined to marry and illumination level and amount of reflections and/or
procreate. Most eunuchs who are castrated before scattering etc.
puberty are not sexual.
Hijra, a Hindi and Urdu term traditionally which may give rise to various types of noises
translated into English as "eunuch", actually refers to [1,2] normalized. The process of normalization
what modern Westerners would call male-to-female has beenstated in subsection (ii) of step 2.
transgender people and effeminate homosexuals 1) Iris Localization: In Iris localization step the
(although some of them reportedly identify as inner (iris/pupil) boundary and the outer
belonging to a third sex). The Indian government (iris/sclera) boundary in the original image of the
has been hiring hijras as tax collectors, which eye are detected. The inner boundary as well as
utilizes their special persuading skills. They will the outer one is modeled as circles [1-3].
sing loudly about the debt outside the homes of the 2) Iris Normalization: Next step is to normalize
defaulters until they are shamed into paying up. iris images to compensate for iris
In this paper we focus on the personal identification deformation. The aim here is to transform the
of Eunuch by Iris recognition method for census iris region so that it has fixed dimensions in
of all Eunuch in India so that the records can be order to allow comparisons so that two
tracked that will be helpful in different government photographs of the same iris under different
works. conditions will have characteristic features at the
Iris Recognition involves image acquisition, iris same spatial location [1-4].
segmentation, Iris image normalization, iris feature
extraction and verification [1]. We will focus our Daugmans rubber sheet model has been used in the
work only at Iris feature extraction which uses present study for the normalization of the isolated
Cumulative SUM based changed analysis. area of the image.
158
representative values of each cell regionwithin a

group
First calculate the average .Calculate cumulative sum

from 0:S0=0
Calculate the other cumulative sums by adding the
difference between current value and average
to previous sum [4]
Step 3. Feature Extraction and encoding
,
Only the significant features of the iris must be

encoded so that comparisons between templates
can be made easier .The feature representation
should have ample information to classify various
irises and be less sensitive to noises. The figure (5)
shows how the correlate region of the Iris has been
extracted [1,5-7]. Therefore, the original After Calculating cumulative sums Iris codes are
image needs be generated for each cells byobtaining MAX and MIN
values among Cumulative sum [4].
The differences between the points of Max and MIN

indexes of Si will determine the slope of the position
of the grey values so that it will be easier to identify
different regions of Iris codes. The technique
generates iris codes by analyzing change of grey
Fig.3. Extraction of Correlate region of Iris. Values of Iris Patterns .Upward slope of Cumulative
sum represents Iris Pattern may change from
To analyze the change of grey values of iris and darkness to brightness and downward slope
extracting the features from iris image is very represents the opposite.
important. Earlier Gabor and wavelet transforms
were used for feature extraction. In this paper
cumulative summing technique is used for extracting
of features from the Iris image. Cumulative sums are
very simple as far as calculation is concerned and
avoids lots of processing steps.
II. MAIN RESULTS

Matching
Feature Extraction using Cumulative sum:
The data images are a total of 800 images from 50
Normalized iris image is used for feature
individual Eunuchs in a set of 10 images per person
extraction.Overall Feature Extraction steps are as
(left and right eye).performance evaluation of the
follows:
proposed method was measured in terms of False
1) Normalized iris image is divided into basic cells
Rejection ratio and False Acceptance ratio
for calculating cumulative sums (One cell region is
(FAR).Comparison of the bit patterns generated is
m x n pixel size an average grey value is used as a
done to check if the two irises belong to the same
representative value of a basic cell region to
Eunuchs. Calculation of Hamming Distance (HD) is
calculate the cumulative sum)
done for this comparison. HD is a fractional measure
2) Basic Cell regions are grouped in both horizontal
of the number of bits disagreeing between two binary
and vertical direction in five basic regions.
patterns. Since this code comparison uses the iris
3) Calculate the Cumulative sum over each group
code data and the noisy mask bits, the modified form
4) Generate Iris feature code.
of the HD is given by:
Cumulative sums are calculated as follows:
Suppose that X1, X2,., X5 mean five Where, Xj and Yj are the two iris codes and N is the
159
number of bits in each template. Since an individual III. CONCLUSIONS

iris region contains features with high degrees of
freedom, each iris region will produce a bit-pattern An area in which the algorithm is needed is to be
which is independent to that produced by another continuously developed and improvements have been
iris, on the other hand, two iris codes produced outlined, and the potentials of utilizing Iris
from the same iris will be highly correlated [1,2]. Recognition
If two bits patterns are completely independent, were examined. Further biometric identification of
such as iris templates generated from different Eunuchs will be helpful to make a clear note of
irises, the Hamming distance between the two differentiating between Eunuchs and Normal Human
patterns should equal 0.5. This occurs because beings and also to make the census of Eunuchs.
independence implies the two bit patterns will be Moreover differentiating of two species and also
totally random, so there is 0.5 chance of setting any gender recognition will be possible in further studies.
bit to 1, and vice versa. Therefore, half of the bits
will agree and half will disagree between the two ACKNOWLEDGEMENT
patterns. If two patterns are derived from the same
iris, the Hamming distance between them will be The author would like to thank .professor P.K .Bose,
close to 0.0, since they are highly correlated and the Director NIT Agartala, Dr Ajay Chakraborty for their
bits should agree between the two iris codes [1-3, 5]. constant support. I am also thankful to one of my
Figure shows the Hamming distance for the same senior
eunuch in and different Eunuch in another. The X and lecturer of our Institute for his invaluable help.
Y axis indicates the Hamming distance and number of
data samples respectively. Recognition of the REFERENCES
proposed method is 99%when threshold was
26.Experimental results shows the proposed method [1] J. Daugman, How Iris Recognition Works, IEEE Trans.
Circuits and Systems for Video Technology, vol. 14, no. 1, pp.
is quite effective and efficient. The proposed feature
21-30,Jan. 2004.
extraction is done in MATLAB 10. [2] Libor Masek, Recognition of Human Iris Patterns for
Biometric Identification, School of Computer Science and Soft
Engineering, the University of Western Australia, 2003.
[3] R. Wildes, J. Asmuth, G. Green, S. Hsu, R. Kolczynski, J.
Matey, S. McBride. A system for automated iris
recognition.Proceedings IEEE Workshop on Applications of
Computer Vision, Sarasota, FL, pp. 121-128, 1994.
[4] EmmanvelRaj,mChirchi,Dr.R.D .kharadkar Biometric Iris
recognition for Persons Identification using cumulative sum
AlgorithmIJSERvol 3,Isuues 5.May 2012.
[5] A.K. Jain, A. Ross, and S. Prabhaker, An Introduction to
Biometric Recognition, IEEE Trans. Circuits and Systems for
Video Technology, vol. 14, no. 1, pp. 4-20, 2004.
[6] J. Daugman, New Methods in Iris Recognition, IEEE
Trans.System, Man, and CyberneticsPart B: Cybernetics, vol.
37, no.5, pp. 1167-1175, 2007.
[7] R. Wildes. Iris recognition: an emerging biometric
technology. Proceedings of the IEEE, Vol. 85, No. 9, 1997.
[8] Anil K. Jain, Arun Ross and SalilPrabhakar, An Introduction
to Biometric Recognition, IEEE Transactions on Circuits and
Systems for Video Technology, Vol. 14, No. 1, January 2004.
[9] R. Singh, M. Vatsa, A. Noore, Improving verification
accuracy by synthesis of locally enhanced biometric images and
deformable model, Signal Processing87(11) (2007) 2746-2764
[10]. Ya-Ping Huang, Si-Wei Luo, En- Yi Chen, An efficient
iris recognition system, International Conference on Machine
Learning and Cybernetics, pp. 450-454, 2002.
[11] Y. Zhu, T. Tan, and Y. Wang, Biometric personal
identification based on iris patterns, inProc. IEEE Int. Conf.
Pattern Recognition., 2000, pp. 28012804.
[12] L. Ma, T. Tan, Y. Wang, and D. Zhang, Personal
identification based on iris texture analysis, IEEE Trans. Pattern
Anal.Mach.Intell., vol. 25, no. 12, pp. 1519 1533,Dec. 2003.
160
Digital Implementation of Geometric

Moments for Image Processing
Ahlad Kumar, Raveendran Paramesran, Chern Loon Lim
AbstractGeometric moments have been widely used

to characterize an image. This paper describe a
hardware implementation of calculating 2nd and
3rd order geometric moments using feedback filter
architecture. The performance of the proposed
architectures is measured in terms of speed and
power dissipation. It has been shown that the
computational time to calculate the geometric
moments is reduced to (N 2 + ), where N xN is
the image size and is small overhead. The
hardware has been implemented in BSIM4.3.0 50nm
technology and simulated using Spice simulator with
a supply voltage of 1V.
Index TermsGeometric Moments, Digital

Accumulator, Flip Flops
161
162
163
164
work. The research has been carried out under UMRG

grant
(RP023/2012A).
ACKNOWLEDGEMENTS
We acknowledge the University of Malaya for funding

this
165
Development of BOC/Interplex Modulator with

FPGA & Novel RF Design for Regional
Navigation Satellite System
Rahul Gupta*, Sohan M. Patel, Dipti R. Patel, T.V.S. Ram
SATCOM & Navigation Payload Area
Space Applications Centre, Indian Space Research Organization rahulgupta@sac.isro.gov.in
Abstract- This paper describes the State-of-the art composite signal, one Interplex Signal is added.
modulation scheme designed and developed for the The modulator system generates the BOC (5,2)
RNSS (Regional Navigation Satellite System). It spreaded signals for RS service and then modulates
elaborates theory, design, implementation and these signals along with the OS signal.
important results of the modulator. The modulator unit There are three baseband navigation signals as
generates the composite modulated signal with following:
constant amplitude vectors which uses the BPSK
(Binary Phase Shift Keying) modulation and BOC
(Binary Offset Carrier) modulation to provide OS 1. OS Data Signal (sos(t))
(Open Services) for civilian purposes and RS 2. RS BOC Pilot Signal (srs_p(t))
(Restricted Services) for military applications 3. RS BOC Data Signal (srs_d(t)) The sub-carrier is
respectively using Interplex technique. Here sinusoidal defined as :
BOC (5, 2) modulation scheme is chosen for RS signal.
scx(t,) = sgn[sin(2fsc,xt + )] (1)
The first constraint that any new signal has to fulfill is it
should be efficiently share the allocated spectral bands where fsc = Subcarrier frequency;
with the already existing signals, without the = Subcarrier phase
degradation on the other signals to become too high. So
the new spreading modulation technique (BOC The IRNSS RS data and pilot BOC signals are
Modulation) is designed to meet the frequency sinBOC. Hence the subcarrier phase =0. The
compatibility which is vital for the coexistence of
complex envelope of composite signal with Interplex
different GNSS (Global Navigation Satellite System).
BOC spreading modulations are one way to accomplish signal I(t) is:
this. s(t) = [2 (sos(t) + srs_p(t)) + j(2. srs_d(t) I(t))]
Keywords- RNSS, BPSK, BOC, OS, RS, CASM, (2)
Interplex. The Interplex signal is generated from the navigation
signals to realize the constant envelope composite
I. INTRODUCTION signal.
RNSS is useful for the implementation of an

independent and secure regional navigation system
independent from GNSS. The introduction of the
BOC (Binary Offset carrier) modulation as
alternative to the BPSK modulation for the
Navigational signals has enhanced the navigation
performance with meeting the spectrum
compatibility. It offers improved performance and
the opportunity for spectrum sharing while retaining Phasor diagram achieved by addition of three
implementation simplicity. modulators is with non constant modulus
(Hexaphase). For good performance of the data link,
II. THEORITICAL CONCEPT the generated Hexaphase modulation is to be
modified in order to keep constant amplitude of each
The OS service will have BPSK modulation phasor. To maintain constant amplitude of each
whereas the RS service will be provided by using phasor to avoid the problems like AM- PM distortion
Binary Offset Carrier (BOC) modulation. An and other saturation effects at final stage Power
additional BOC Pilot signal is being provided for Amplifier, two types of techniques were studied.
the RS Service in order to aid in better acquisition 1. The co mb i n ed signal of the t h r e e
and performance. In order to maintain the mo d u l a t o r outputs is passed through a Hard
constant envelope characteristic of the Limiter.
166
2. An Interplex Modulator (CASM).

The three channel signals OS, RS & Pilot are
multiplexed using Coherent Adaptive Subcarrier
Modulation (CASM), which is multichannel
modulation scheme also known as Tricode
Hexaphase modulation (Interplex Modulation).
CASM is used to ensure that the signal transmitted
from the satellite has a constant power envelop, i.e.,
the total transmitted power does not vary over time.
Thus the transmitted information is not containing in
the signal amplitude and the transmitted signal
amplitude becomes less critical. This is a very
desirable property of the signal since it allows the
use of efficient class C like power amplifiers.
Fig.3 Block schematic of FPGA implemented DSP
III. DESIGN
These digital modulating signals are mixed with the
As per the above theoritical explanation, the RF carrier signals generated by local oscillator. A
implemented design uses FPGA & Quad mixers as noval approch was found for mixing the digital and
per the block diagram shown below in fig.2. RF signals. Here, digital signal is not converted into
the bipolar signal, thus avoiding the driver results in a
significant amount of reduction in the hardware size
and power requirement with improved system
reliability. So the digital signal is fed to mixer in
differential format and then mixed with the RF carrier
signal. The outputs of four such double balanced
mixers
are combined to generate the modulated signal at RF
frequency. The total power of composite signal is
distributed in I1, Q1, I2 and Q2 channels as:
I1 = (v2/3)2 = 22.22%
I2 = (v2/3)2 =22.22 %
Q1 = (2/3)2 = 44.44 %
Q2 =(1/3)2 = 11.11 %
The above power sharing can be used for distributing
power in OS and RS services in different proportions
as per the requirement, just by switching of the four
Here, the differential signal interface, using LVDS signals in different channels to be modulated as
(Low Voltage Differential Signal) devies, between shown in table1.
Navigation baseband signals generating unit &
Modulator is used to transfer the navigation signals
at data rate up to 10.23 Mbps.
FPGA has the provision of selecting baseband

signals from Main or Redundant units. It can also
switch among different power sharing of OS & RS
services for modulation. This switching is performed
using telecommands and corresponding telemetry
signals are also generated from.
The FPGA generates the BPSK modulating signals
for OS service; BOC (5, 2) RS & Pilot digital signals
for RS service; and Interplex signal as shown in
fig.1, for getting the constant envelope constellation
modulated signal. IV. RESULTS
The block schematic designed in Verilog HDL sub-
modules format and implemented in ACTEL FPGA The simulated results of MATLAB Simulink
is shown in fig.3. design are shown in fig.4.
167
Fig.4 (a) Simulated Spectrum with higher measuring Fig.6 (a) Measured Spectrum of in-house space-
frequency qualified modulator
Fig.4 (b) Simulated Constellation Plot Fig.6 (b) Measured Constellation of in-house
The above described modulator system was developed space-qualified modulator
developed and tested in-house as shown in fig.5. Measured Results o f d e s ig n e d mo d ulato r are
Measured spectrum and constellation of designed shown in table 2 along with the specifications.
hardware are shown in the fig.6.
V. CONCLUSION
Fig.5 In-house developed space qualified Modulator
with test setup The BOC/Interplex modulator has successfully been
developed and the results of ground testing are quite
satisfactory, meeting the simulated ones. This
modulation scheme using code spreading as BOC is
efficient for bandwidth utilization where the same
168
bandwidth can be used to transmit a number of

signals keeping at offset from center frequency.
Thus RNSS modulated signal can coexist with the
available GNSS navigation signals without affecting
those.
VI. ACKNOWLEDGEMENT
Authors will like to express their gratitude to Shri

D. K. Das Deputy Director, SNPA for his continuous
support and Shri Alak Banik, Group Director, ODCG
for his encouragement. Authors will also like to
thank the Project for providing the fund without
which this work would not be possible to meet the
target so successfully.
REFERENCES
[1] Betz J. W., et al. (2002), Candidate Design for an Additional

Civil Signal in GPS Spectral Bnads Proceedings of ION NTM
2002 28- 30 January, 2002 San Diego, CA.
[2] L. Ries, A Software Simulation Tool for GNSS2 BOC
Signal Analysis, ION GPS 2002, 24 27 September 2002,
Portland.
[3] Hein G. W. et al. (2005) A candidate for the GALILEO L1
OS Optimized Signal, Proceedings of ION GNSS 2005 13
16
September 2005, Long Beach, California, USA.
[4] Apostolos Georgiadis, Gain, Phase Imbalance, and Phase
Noise
Effects on Error Vector Magnitude, IEEE Trans. Veh. Technol.,
vol.
53, No 2, pp. 443-449, Nov. 1993.
169
Navigation Filter for Real Time Orbit

Determination of Leo Spacecrafts
Namit Rarotra1, Hemanth Kumar Reddy N2, Chitra Ramamurthy3, Yateendra Mehta4
Communication Systems Group, ISRO Satellite Centre, Bangalore
1
namit@isac.gov.in, 2hemanthr@isac.gov.in, 3chitra@isac.gov.in, 4ymehta@isac.gov.in
Abstract space segment, the control segment, and the user
The paper presents a Navigation filter for segment. The Space segment consists of a
precise orbit determination of Low Earth Orbiting
constellation of more than 24 satellites orbiting at an
Satellites that improves the three dimensional
positioning accuracy of on-orbit Navigation systems altitude of approximately 20,200 kilometers. There
used in all the ISRO satellites under IRS programme. are six orbits with a minimum of 4 satellites in each
Kalman filter based Navigation filter which uses Global orbit. These orbits are inclined at an angle of 55
Positioning System (GPS) can be used for real time degrees and separated by 60 degrees. All satellites
orbit determination of space-craft. The precise orbit is continuously transmit GPS signals at L1 (1575MHz),
calculated on the basis of observations made from GPS L2 (1227MHz) and L5 (1176MHz) frequencies.
signals as measured by LEO satellite on real time The Control Segment (CS) provides command,
basisin fully autonomous mode. Navigation filter is control, and maintenance services to the space
capable of performing real time filtering of the noisy
segment and ensures that the satellites are working
system variables corrupted by guassian white noise. The
filter is implemented in ARM7TDMI platform in an properly. It consists of a master control station
optimised manner so as to perform all the navigational (MCS), an alternate master control station, four
functions of GPS Receiver without any timing issues. dedicated ground antennas and six dedicated monitor
The navigation receiver is tested and performance stations.
evaluated.The filter estimates the optimum states of the The user segment consists of receivers that receives
system at a particular epoch and significantly improves and processes the GPS signals and calculates the
the positioning accuracy of the system. position, velocity and time of user. It works on the
principle of triangulation and a minimum of four
Keywords Kalman filter, GPS, Orbit determination, satellites are required to precisely calculate a 3D
solution that is (x,y,z,t).
I. INTRODUCTION
Filtering is desirable in communication

systems where signals are corrupted by noise and this
can be interpreted as uncertainty superimposed on the
fundamental trend or signal. This uncertainty may
arise in the dynamics of the system because of the
electrical or mechanical inaccuracies in sensor or
arithmetic resolution in computing elements or
rounding off error. Orbit determination refers to the
technique of finding the position and velocity vectors
of the space-craft orbiting around the earth. GPS can
be used for real time orbit determination of space
craft. Final output vectors i.e. estimated position Fig.1: GPS Constellation
vectors has uncertainty because of inaccuracies in
calculation of pseudo-ranges. These inaccuracies can II. GPS MEASUREMENTS
be thought of as gaussian. Kalman filter can be used
to estimate the optimum position vectors recursively. GPS receiver is used to estimate the position
and velocity vectors of the space craft. The
GLOBAL POSITION SYSTEM measurements are in the form of pseudo-range which
is the apparent distance between the tracked GPS
The Global Positioning System is a satellite and the user. The user distance from the
navigation system that provides the exact position tracked GPS satellite is proportional to the change in
and time of user anytime, anywhere and in all carrier frequency relative to the user. This in turn is a
weather conditions. GPS consists of three parts: the function of satellite position, velocity, precise carrier
170
frequency and time which can be obtained from the State equation:
satellite ephemeris. These measurements of pseudo- xk = Axk-1 + Buk + wk-1... (ii)
ranges are noisy and not accurate. The accuracy of Output measurement equation:
final position and velocity vectors depend upon the zk = Hxk + vk... (iii)
accuracy of the pseudo-ranges. The errors introduced
Where x n is the state of a discrete time controlled
in the pseudo-ranges can be assumed to be random
and Gaussian in nature. process to be estimated and z m is the
The observable in GPS orbit determination is the measurement. A relates the state at the previous
time step (K-1) to the statesat the current step (K), in
pseudo-ranges () which means the apparent
the absence of either a driving function or process
distance between tracked jth GPS satellite to user
noise. B relates the optional control input u l(if
LEO spacecraft. This pseudo-range (j) can be
available) to the state x. H in the measurement
obtained by
equation relates the state to the measurement z k. The
= x x + y y + z z random variables wk-1 and vkrepresent the process and
measurement noise respectively. They are assumed to
+ ( ) be independent, white, and with normal probability
where (xj , yj , zj ) are the coordinates of jth GPS distributions
satellite, (xu , yu , zu ) are the coordinates of user p(w) ~ N(0,Q)
LEO spacecraft, tu is an offset of receiver clock from p(v) ~ N(0,R)
the system time, and c is a speed of light (c = Where Q and R are process noise covariance and
299,792,458m/s). The receiver estimates the pseudo- measurement noise covariance respectively and these
range. The GPS satellite position can be calculated by might change at every time step.
using the demodulated navigation data received from Each of these quantities is vectors and therefore
GPS satellites. The value of coordinates xj,yj and zj contains more than one element. The vector x
can be computed from the ephemeris of that contains all of the information about the presentstate
particular satellite from the demodulated navigation of the system, but we cannot measure x directly.
data. Four equations are needed to solve for four Instead we measure y, which is a function of x that is
variables xu,yu, zuand tu. Thus a minimum of four corrupted by the noise z. We can use y to help us
visible GPS satellites are required to solve for LEO obtain an estimate of x, but we cannot necessarily
spacecraft position vectors. take the information from y at face value because it is
corrupted by noise.
III. KALMAN FILTER The state estimation in kalman filter is a linear
combination of priori state estimate and the weighted
Kalman filter is a data processing algorithm difference of the measurementZkobtained and the
that estimates the states of a linear system from predicted measurementH x K -.This difference(Zk - H x
indirect, inaccurate and uncertain observations. It is a -
K )is called as residual. The weighting factor is called
set of mathematical equations that implement a as a gain or the blending factor and can be obtained
predictor-corrector type estimator that is optimal. by
This means that it minimizes the mean square error of Kk = P K -HT (H P K - HT +R)-1
the estimated parameters. Kalman filter is recursive
in nature, so the new measurements are processed as This value of gain is such that it minimizes the
soon as they are available. This makes it capable of posterior error covariance P K. The Kalman filter can
performing real time filtering of the noisy system be thought of as a feedback control system. It
variables corrupted by guassian white noise. The estimates the process state at some time and then
primary objective of Kalman Filter is to estimate the obtains feedback in the form of noisy measurements.
states of a discrete-time controlled process that is As such, the equations for the Kalman filter fall into
governed by the linear stochastic difference equation. two groups:
The extended Kalman filter (EKF) is a nonlinear 1.) Time update equations predict the state and
version of the Kalman filter and is widely used for error covariance estimates at current time
position estimation in GPS receivers. The EKF is using state model to obtain the priori estimates
formulated with the assumption that the measurement for the next time step. The time update
noise is white. equations are responsible for projecting
forward in time the current state and error
MATHEMATICAL FORMULATION OF KALMAN FILTER covariance estimates to obtain the priori
estimates for the next time step.
A linear process can be described by the following Time Update Equations
two difference equations:
xk = Axk-1 + Buk ... (iv)
171
- Kalman filter are compared with that computed from

P K = A P K+1 AT + Q... (v) the conventional method.
2.) Measurement update equations to incorporate Dynamic model
the new measurement in the predicted The dynamic model implemented in the Kalman filter
estimates to obtain the final optimal estimates. can be represented by a set of first order stochastic
Measurement Update Equations differential equations. Let the system be described by
Kk = P K -HT (H P K - HT +R)-1... a linear partial differential equation
(vi)
xk = x K -+ Kk(Zk - H x K -)... (vii) = ( )+ ( )
P K = (I -KkH ) P K -... (viii) If x is represented by[x x x ], then
(vi)computes the gain, Kk, called Kalman Gain that is
dependent on the error covariance P K obtained in the 0 1 0
time update equations (v).(vii)calculates the final = 0 0 1 +
state estimates taking into account the measurement 0 0 0
obtained from the system. This is called as a
posteriori state estimate. In equation(viii), posteriori Where w is considered to be guassian white noise
with zero mean and auto-covariance Q. It is also
error covariance is computed using Kalman gain and
Priori error covariance estimates to use it in next time known as dynamic noise or process noise.
step. = ( )
The kalman filter estimation algorithm can be thought
of as a predictor-corrector algorithm for solving 00
numerical problems as shown below in Figure 2. 0 0
=

0 0 0
Where Qnniscovariance of wn which is noise
associated with nth state and n is number of states.
The state vector at current time ti can be expressed
by a Taylor series as
1
( )= ( ) + ( ) + ( )
Fig. 2: Kalman Filter as Predictor- Corrector algorithm.
2
+ . ( )
The values of Q and R significantly affect the ( )is state at previous time step .Using eq
performance of Kalman Filter. If Q is too large, error
(ix) and (x) we can say that ( ) is a linear
covariance P K will also be large which in turn makes
combination of initial state ( )
the Kalman Gain Kktoo high and the output estimates
( )= ( )
will have the tendency to follow the measurements
is state transition matrix which transforms the state
and the output will bounce too much and opposite if
at time ( ) to corresponding state at ( ).
Q is too small. Similarly if R is large, the gain Kk will
be too small and it will not take into account the
Measurement Model
incoming measurements as much as it should be. So
the value of Q and R has to be properly tuned based
The measurements are in the form of ranges from
on the model and sensor accuracies.
satellite to receiver. The range from each of the
MODELING
satellite to receiver is given by
TABLE I. The main aim is to implement
simple and fairly accurate algorithm to determine the = x x + y y + z z +
orbit of LEO satellites, in real time, with low
computational burden and less storage requirements, +
by using the raw navigation measurements provided
Where Vjis zero mean, uncorrelated guassian
by GPS receivers.
measurement noise associated with the range
The navigation filter for orbit determination is
implemented by using Kalman filtering of the raw measurement j. b is the unknown bias on each
navigation data from a GPS receiver. The states to be range measurement. As the range measurement is a
estimated are the position, velocity, bias and drift of nonlinear function, we use partial derivative
the GPS receiver clock on board the spacecraft. The technique to linearize the range measurements. The
conventional method, i.e. iterative method is used linearized range measurements as a function of state
find the position and velocity vectors solution of the will be given by
spacecraft. Position vectors computed from the = H x + v
172
where x is the state vector, v is the additive The algorithm is implemented on the hardware based
measurement noise; andH the pseudo-range on Xilinx FPGA and ARM7TDMI processor. It is a
measurement matrix and is obtained by linearizing part of the navigation solution computing the state
the pseudo-range equation with respect to the filter vectors every one second. Code to implement the
states algorithm was optimized tomeet all the timing
The linearized measurement matrix is given by constraintsand compute the filtered solution along
with the conventional method once in a second.
( ) ( ) ( )
= 1 0 0
Algorithm is tested by obtaining the raw
Measurement noise matrix can be given by measurements from the GPS simulator simulating the
orbit scenarios. Simulations were carried out on
0 0 dataset generated from high dynamics GPS simulator

0 0 that simulates the GPS constellation. It also simulates
= the highly dynamic orbits of the LEO spacecraft.
Pseudo-ranges were obtained from the GPS
0 0 0 simulator, along with the ephemeris of GPS satellites.
N is the number of satellites being tracked and is This high dynamics signal was given to the GPS
the measurement noise variance for the range receiver which processes it and provides the state
measurements ofith satellite. vectors. State vectors were obtained from both the
iterative method as well as from the Kalman filter
algorithm. State vectors were compared and
significant improvement in 3-dimensional accuracy is
observed.
V. IMPLEMENATION
15
The steps for implementation of Kalman Iterative
Filtered
filter are shown in fig3. 10
5
Error(m)
-5
-10
Fig. 3: Kalman Filter Algorithm

-15
0 1000 2000 3000 4000 5000 6000 7000 8000 9000
The predicted state estimate for the receiver can be Samples
given by Fig. 4: Error in x position

=
15
Where represents the states to be estimated at this Iterative
time step and represents the states at next time Filtered
10
step.
As soon as the pseudo-range measurementscj from
5
the receiver are available, the residual can be
computed. The filter residual (Z) is the difference
Error(m)
0
between the projected range j and the measured
ranges cj
-5
= +
-10
The residual Z can be used to estimate the states at
current time step. The predicted error covariance
-15
estimates are given by (v). The Kalman Gain Kkcan 0 1000 2000 3000 4000 5000 6000 7000 8000 9000
Samples
be computed using (vi) and error covariance P K
using (viii). Fig. 5: Error in y position
173
[6] B.L.Malleswari, I.V.MuraliKrishna, K.Lalkishore, M.Seetha,

Nagaratna, P. Hegde, THE ROLE OF KALMAN FILTER
100 IN THE MODELLING OF GPS ERRORS, Journal of
Iterative Theoretical and Applied Information Technology, vol.5, no.
80 Filtered 15, pp 95-101.
60
[7] Min H. Kao, Donald H. Eller, MULTICONFIGURATION
KALMAN FILTER DESIGN FOR HIGH-PERFORMANCE
40 GPS NAVIGATION, ieee Transaction on Automatic
Control, Vol. AC-28, No. 3, March 1983.
20
[8] S. Purivigraipong, SPACECRAFT ORBIT
Error(m)
0 DETERMINATION USING THE FILTERING

ESTIMATOR, Centre for Electronic Systems Design and
-20 Signal Processing,Department of Electronic Engineering,
-40
Mahanakorn University of Technology.
-60
-80
-100
0 1000 2000 3000 4000 5000 6000 7000 8000 9000
Samples
Fig. 6: Error in z position
Fig. 4 shows the error in estimation of state vector x.

As seen in fig. 4, a significant improvement is visible
in estimation accuracy. Similarly fig. 5 and fig. 6 also
shows the improvement in estimation accuracy of
state vector y and state vector z respectively.
Accuracy analysis carried out on the output vector
also conform the enhancement in positioning
accuracy of the navigationsystems.
Testing of the filter algorithm has been done using
the polar, equatorial and re-entry orbital trajectories
and results in all these conditions found to be
satisfactory.
V. CONCLUSON
Kalman filter model formulated in this paper

worked well with high dynamic orbital scenarios.
RMS errors in the final output vectors of Kalman
filter are considerably less than that of the iterative
(conventional) method.
REFERENCES
[1] NamitRarotra, Hemanth Kumar Reddy N, Mukesh Kumar

Gupta, Chitra Ramamurthy, Yateendra Mehta, ORBIT
DETERMINATION TECHNIQUES FOR LEO
SPACECRAFTS,NATIONAL SYMPOSIUM ON
INSTRUMENTAION (NSI 37), 2012.
[2] James Bao Yen Tsui, FUNDAMENTALS OF GLOBAL
POSITIONING SYSTEM RECEIVERS A SOFTWARE
APPROACH, John Wiley & Sons, Inc
[3] Greg Welch, Gary Bishop, AN INTRODUCTION TO THE
KALMAN FILTER, UNC-Chapel Hill, TR 95-041, July24,
2006.
[4] FUNDAMENTALS OF KALMAN FILTERING: A
PRACTICAL APPROACH, Second Edition,by Paul Zarchan
, AIAA, 2005.
[5] Peter S. Maybeck, STOCHASTIC MODEL,
ESTIMATION, AND CONTROL, VOL.1, Academic Press,
New York.
174
IRNSS One Way Measurement Simulation

Chandrasekhar M.V, Kalasagarvarma.K, Manamohan Kamat, Satyanarayana.G, Nirmala.S
S.C. Rathnakara, A.S. Ganeshan
Space Navigation Group, ISRO Satellite Center, Bangalore 560017, India
sekharmv@isac.gov.in,sagark@isac.gov.in, mvkamat@isac.gov.in,sathyag@isac.gov.in,nirmalas@isac.gov.in,
ratnakar@isac.gov.in, asganesh@isac.gov.in
Abstract IRNSS is an Indian Regional Navigation a) The space segment consists of three satellites
Satellite System, which relies on constellation of in GEO orbit and four satellites in GSO orbit
regional navigation satellites to provide navigation inclined at 29 to the equatorial plane. IRNSS
services to various users in India and a region extending
satellites have two types of payloads, navigation
about 1500 km around India with a user position
accuracy of less than 20 meters. This paper addresses and ranging payload. the navigation payload
how the IRNSS measurements have been simulated to will transmit the ranging codes being generated
test and validate the navigation software in the absence onboard with the navigation data uplinked from
of IRNSS satellites one way range measurements. This the ground stations.
simulation software will simulate pseudo range
measurements in meters and carrier phase b) The ground segment is responsible for
measurements in cycles. maintenance and operation of IRNSS constellation
which comprises of seventeen one way ranging
IRNSS one way range measurements on L5 stations (IRIMS) spread over Indian landmass and
(1176.450MHz) and S (2492.028MHz) frequencies for around as shown in Figure 2, four two way ranging
all possible line of sights between seven IRNSS satellites stations, nine laser ranging stations, one centre for
(i.e., IRNSS 1A to 1G) and identified IRIMS (IRNSS health monitoring, analysis and control of IRNSS
Range and Integrity Monitoring Station) which are satellites, one precise time generation centre for
distributed in and around Indian land mass. The
IRNSS system time maintenance, and one centre for
simulated range measurements includes all the possible
errors such as free space delay, atmospheric delays, IRNSS navigation parameters generation by
clock offsets, multipath, phase center offsets, noises etc navigation software.
as close to the real measurements in RINEX (Receiver
Independent Exchange Format). And the paper c) The IRNSS user segment comprises of single and
discusses motivation and importance of this work along dual frequency navigation operating on L5/S and L5
with different error models included in the simulated and S frequencies respectively.
range measurements.
Keywords - MEASUREMENTS SIMULATION, SIGNAL IRNSS one way range measurements are the
DELAY MODELS primary source for the generation of navigation
parameters. In the absence of real time IRNSS range
measurements there is a need of a simulator which
I.INTRODUCTION
simulates the IRNSS one way range measurements on
IRNSS envisages establishment of regional both L5 and S frequencies, this simulator is
navigational satellite system using a combination of developed and used to generate the range
3GEO (Geo Stationary Earth Orbit) and 4GSO (Geo measurements on both L5 and S frequencies.
Synchronous Orbit) Satellites as shown in Figure 1.
IRNSS will provide basically two types of services:
Standard Positioning Service (SPS) and Restricted
Service (RS). Both of these services will be provided
at two different frequencies one in the L5 band and
the other in S-band.
Thus, user receiver can be operated in single and /or
dual frequency operation mode. IRNSS is expected to
provide position accuracy better than 20 meters over
India and a region extending about 1500 km around
India. IRNSS consists of three segments namely the
space segment, the ground segment and the user
segment.
Figure 1: IRNSS Satellites constellation
175
propagation model. The propagation algorithm

predicts the satellite position in ECI (Earth-Centered
Inertial) frame. The predicted ephemerides are then
transformed to ECEF frame for simulation.
Figure 2: IRNSS Range and Integrity monitoring station (IRIMS)

location map
II. ARCHITECTURE OF SIMULATION SOFTWARE

This simulator simulates both code and carrier
phase measurements on both L5 and S frequencies of
IRNSS,
Code phase measurment is the apparent transit
time of the signal from a satellite to the receiver. The
pseudo range () in meters on L5 and S frequencies is
computed by adding the various delays to the
geometric range (g) [1], [2] as shown in the
following pseudo range equation.
= g + c (tu - tS) + FSD + TP + I + MP + noise + Hd +

PCO Figure 3: Architecture of simulation software
Carrier phase measurement is the difference Orbit Propagation Model

between the phase of the receiver generated carrier
signal and the received carrier signal from a satellite The orbit propagation algorithm aims to predict the
at the instant of the measurement, accounting for the satellite state vectors and state transition matrix for
various measurements errors, clock biases and initial future time, using estimated state vector as initial
phase offsets [1], [2]. Pseudo range carrier phase guess. The performances of the satellite orbits will be
measurement in cycles is shown following equation. affected by many perturbations such as the non-
= -1 [g + c (tu - tS) + FSD + TP - I + Mp + noise + central part of Earths gravitational potential,
Hd + PCO] + N gravitational effect of sun and moon, solid earth &
ocean tidal effects, solar radiation pressure, albedo
where tu is receiver clock bias, tS is the satellite earth radiation pressure and gravitational effects of
clock bias, FSD is Free space delay, TP is troposphere the other planets. The algorithm comprises of two
delay, I is Ionosphere delay, MP, noise and Mp, noise parts, first part is the force modeling which computes
are delays due to multipath and noises for code and the perturbing forces acting on the satellite and
carrier measurements respectively, Hd is hardware computes the partials of acceleration due to these
delay, PCO is the phase centre offsets of satellite and perturbations. While the second part handles with
receiver, c, N, are speed of light in vacuum, Integer integrating these accelerations and partials of
ambiguity, and wavelength of signals respectively. accelerations using 12th order Adam Bashforth-
Figure 3 shows the architecture and various error Moulton predictor corrector method to obtain the
models of one way measurement simulation position and velocity of the satellite.
algorithm which simulates IRNSS range
measurement data on both L5 and S frequencies. By having predicted IRNSS ephemeris in hand and
Simulation of one way range measurements for precise IRIMS coordinates, true range can be
required line of sights between IRIMS stations and computed, which is geometric distance between
IRNSS satellites which in turn requires IRNSS IRIMS and IRNSS satellite coordinates in ECEF
satellites position be known precisely and predicted (Earth-Centered, Earth-Fixed) frame is computed for
for future time, which was done by full force orbit every instant, taking the earths rotation during the
176
signal travel time into account. The basic equation for

computation of true range can be represented as Ionospheric model
follows: The ionosphere, extending from a height of about
= ( ) +( ) +( ) 50km to about 1000km above the earth surface, is a
region of ionized gases (free electrons and ions).The
Where the satellite coordinate vector (xs, ys, zs) is a delay of code and advance of carrier phase signals
vector function of the time, and the receiver due to ionosphere while propagating from satellite to
coordinate (xr, yr, zr) is a function of time t. user receiver is called ionospheric delay. The
simulator has provision to compute the ionospheric
Different delays and corrections like free space delay using the below mentioned two models:
delays, atmosphere delays, satellite and receiver Klobuchar Model [1], [4]
clock errors, multipath delay, code and carrier signal
noises and antenna phase centre corrections are Grid based Multi layer data fusion (MLDF) [5]
computed using various mathematical models and
summed up with geometric range to obtain both
code and carrier phase measurements on L5 and S
frequencies in standard RINEX format
III. MATHEMATICAL MODELS OF SIGNAL DELAYS
To simulate the code and carrier phase range
measurements, different mathematical models are
used to compute the delays which are shown in
pseudo range equation. These models are common for
code and carrier phase range measurements, which is
explained in detail below. Figure 5: Variation of Ionospheric Delay for L5 signal using
Free space delay Klobuchar model with IRNSS time for LOS between IRNSS
1Asatellite and Bangalore station
It is the delay of signal in free space, when
transmitted from satellite to a receiver that includes
the corrections for the relativistic, sagnac and orbit
eccentricity effects.
The delay is modeled using the available standard
algorithms
[3] And the characteristics of this phenomenon are
shown in figure 4.
Figure 6: Variation of Ionospheric Delay for S signal using

Klobuchar model with IRNSS time for LOS between IRNSS
1Asatellite and Bangalore station
Figure 4: Variation of free space delay with IRNSS time for

IRNSS 1A satellite
Atmospheric delays of signal

The signal from the satellites before reaching the
user receiver passes through the ionosphere and
troposphere layers of the atmosphere and suffers a
delay due to ionosphere depending upon the time of
the day, solar cycle activity and the delay experienced
due to troposphere depending upon the satellite Figure 7: Variation of Ionospheric Delay for L5 Signal using
MLDF grid Model with IRNSS time for LOS between IRNSS 1A
elevation angle. The delay magnitude patterns of
signal due to ionosphere and troposphere are shown
in figures 5, 6, 7, 8 and 9 respectively.
177
satellite and HASSAN station a clock offset with respect to IRNSS time resulting in
random behavior of clock. These satellite and receiver
clocks offset errors along with randomly occurring
phase jumps can be simulated using following two
models and the corresponding graphs are shown in
figures 10 and 11
1) Polynomial method:
Clock Error (t) can be modeled using
t = af0 + af1 (t - toc) + af2 ( t toc)2 +
Where,
Figure 8: Variation of Ionospheric Delay for S Signal using af0 = Clock bias, af1 = Clock drift
MLDF grid Model with IRNSS time for LOS between IRNSS 1A
satellite and HASSAN station af2 = Frequency drift or aging,
toc = Clock data reference time
2) Tropospheric models
t = Current time epoch, =clock noise
Troposphere stretches to about 16km above the
equator and about 9km above poles from earth 2) Realistic clock Errors
surface. The general empirical models of the IGS precise clocks data available in public
troposphere depend on the station height, elevation domain is used to generate clock offsets with a
angle, pressure, temperature and relative humidity in provision to add phase jumps and simulated
signal line of sight. The simulator would be able to noise.
choose any of the following tropospheric delay
models and it has provision to take real time
meteorological data or computed meteorological data
using mathematical models as input.
Saastamoinen model [1],[6]

SBAS model [7]
Hopfield model [1], [7]
Marini Murray model [8]
Figure 10: Variation of Satellite Clock offset in meters (Realistic
clock model) which includes satellite clock phase jump with
IRNSS time for IRNSS 1A satellite.
Figure 9: Variation of Tropospheric Delay (Saastamoinen model)

with elevation angle for LOS between IRNSS 1A and Bangalore
station.
Satellite and receiver clock errors

IRNSS satellite will be carrying three Rubidium-
Atomic Frequency Standards clocks which give
Figure 11: Variation of Receiver Clock offset in meters (Realistic
satellite signal transmission time and clock model) with IRNSS time for HASSAN station
rubidium/caesium clocks will be used in IRIMS
stations to know the signal acquisition time from Multipath Delay
IRNSS satellite. Both satellite and receiver clocks
suffer from systematic and random frequency Multipath delay is caused by the interference of a
variations, which are attributed due to presence of signal that has reached the receiver antenna by two or
noise, aging and other environmental conditions. more different paths [1]. The multipath effect differs
These variations get accumulated over time leading to significantly in magnitude for both code and carrier
178
for each line of sight are computed and used for

phase measurement (Emr & Emp) and this can be
simulation.
simulated using empirical formula [3], which depends
upon receiver environment characteristics. Satellite & Receiver antenna Phase Center Offsets
(PCO)
Emr = Abr + Ar Kenv Krec Kran cos (.E)
Satellite antenna offset (also known as satellite
Emp = Abp + Ap Kenv Krec Kran cos (.E) attitude error) refer to the separation between the
Where Ar and Ap are the amplitude of the multipath satellite centre of mass and its antenna phase centre
This satellite antenna offsets are calibrated at ground
effect, Abr and Abp are residual multipath bias terms, (before launch of the satellite) and the offsets vary
Krec is the receiver sensitivity factor, is the with orientation of the satellite in orbit. The
multipath frequency and E is the elevation angle of topocentric satellite antenna offsets have to be
the transmitter relative to the receiver. Kenv and Kran projected in radial direction and then applied to the
are receiver environment characteristic coefficient pseudoranges in the simulation.
and time-correlated Gaussian distribution noise Similarly at the receiver end, the separation
respectively. between the receiver reference point and its antenna
Figure 12 and 13 shows typical variations due to phase center is called receiver antenna phase offsets.
multipath for L5 code and L5 carrier signals Signals from individual satellites arrive at receiver
antenna from different directions and hence the
position of antenna phase centre depends on azimuth
and elevation of signal which is transmitting from
satellite. The position of antenna phase centre with
respect to elevation and azimuth of signal is
computed and corresponding corrections are made in
code and carrier pseudo range equations for both L5
and S frequencies.
Measurement noise
The code and the carrier phase measurements are
affected by measurement noise called receiver noise
Figure 12: Variation of multipath delay for L5 code signal with
[1]. Additive white Gaussian noise (Z) has been
IRNSS time for LOS between IRNSS 1A satellite and HASSAN simulated as measurement noise by making it as
station. elevation dependent. Code Noise can be 1m and
Carrier Phase Noise is .001m approximately
z = + 2log(s) Cos (2t)
s and t are random variables with uniform
distribution on [0, 1], is mean, and is assumed to be
0 for both code and carrier. is standard deviation
which is of the order 1-3 meter for code
measurements and millie meter level for carrier phase
measurements. Figures 14 and 15 and shows the
simulated measurement noise pattern for code and
carrier signals.
Figure 13: Variation of multipath delay for L5 carrier signal with
IRNSS time for LOS between IRNSS 1A satellite and HASSAN
station.
Hardware delays
Hardware delay is defined as the delay between the

signals radiated output of a specific satellite and the
output of that satellites on-board frequency source
and vice-versa for receiver, the delay consists of a
bias term and an uncertainty. The Channel delay bias
value will be measured and provided by the
manufacturer for each signal of satellite. Using those Figure 1 4: Variation of code noise with elevation of IRNSS 1A
specified values satellite and receiver channel delays satellite for a line of sight
179
developed algorithm, which adds integer number of

cycles, randomly to the carrier phase measurements
throughout simulated data with size and number of
times as prerequisite for the algorithm. Whenever
cycle slip occur integer ambiguity changes with
corresponding cycle slip size.
Outliers and Data gaps
Outliers are the abnormal behavior in the

measurements. Outliers and data gaps are computed
same as cycle slips. But these are randomly added to
Figure 15: Variation of carrier noise with elevation of IRNSS both code and carrier phase measurements throughout
1A satellite for a line of sight
the simulation.
Signal to noise ratio
IV. VALIDATION
Signal to noise ratio (SNR) is defined as ratio of
signal power to noise power, usually expressed in The individual errors delays simulated here are
decibels [1]. Signal to noise ratio is simulated as a validated with commercially available software
linear dependent of elevation for corresponding line simulators and IGS data available in public domain,
of sights between IRNSS satellites and IRIMSs in and the results were observed to be in good
dB-Hz and its variation is shown in Figures 16 and 17 agreement. The tropospheric and ionosphere delays
for L5 and S signals. The minimum signal to noise simulated were validated using CODE products from
ratio taken in this simulation is approximately IGS stations of Bangalore (IISC) and Hyderabad
55dB-Hz for L5 signal and 45dB-Hz for S signal. (HYDE). The ionosphere delay has also been
validated using GAGAN Ionosphere messages. Other
major errors such as receiver and satellite clock
delays have been computed using GPS precise
products.
V. CONCLUSION
IRNSS one way range simulation software has

been developed and this simulation software is
capable to generate one way range measurement data
with all possible delays and errors experiences by the
Figure 16: Variation of signal to noise ratio (SNR) with elevation navigation signal while transmitting from IRNSS
of IRNSS 1A satellite for L5 signal. satellite to IRIMS such as free space delay,
atmospheric delays, satellite and receiver clock errors,
multipath delay, Satellite and receiver antenna phase
center offsets, measurement noise, carrier cycle slips,
clock phase jumps, signal data gap, etc. to simulate
the measurement data as close to the realistic data.
One way range measurement data with all possible

delays and errors of signal while transmitting from
IRNSS satellites to IRIMS stations is simulated using
IRNSS satellites state vectors data generated by orbit
propagator and IRIMS coordinates in ECEF frame for
Figure 17: Variation of signal to noise ratio (SNR) with elevation both L5 and S frequencies and this data fed to data
of IRNSS 1A satellite for L5 signal. processing ,batch estimation, orbit propagation and
Cycle clips and integer ambiguity navigation parameters generation modules of IRNSS
Navigation software to test and validate the robustness
Cycle slip is a discontinuity or a jump in the carrier of IRNSS navigation software.
phase measurements [1], [9], by an integer number of
cycles, caused by temporary signal loss (carrier phase
lock). Cycle clips are computed using in house
180
ACKNOWLEDGMENT
The authors would like to thank Shri A.S.
Ganeshan, Program Director, Satellite Navigation
Program and Sri S.C. Rathnakara, Division Head,
Navigation Software Division for many useful
conversations regarding this project. The authors also
wish to acknowledge all the members of Space
Navigation Group, ISAC for their motivation and
help.
REFERENCES
[1] Global positioning system, signals, Measurements and

Performance by Prathap Misra & Per Enge.
[2] Global positioning system: Theory and Applications
Parkinson & Bradford W. James J.Spilker Jr
[3] Galileo system simulation Facility Algorithms and
Models
[4] J. A. Klobuchar, Ionosphere time delay algorithm for
single frequency GPS user,IEEE Transactions on
aerospace and electronic systems,vol.23,pp.325-
331.Mar.1986
[5] IRNSS Navigation software design document.
[6] Parkinson, B.W., Spilker, J.J., Axelrad, P., and Enge,
Per, Global Positioning System: Theory and
Applications, Vol. I, Americal Institute of Aeronautics
and Astronautics, pp. 534-535, 1996.
[7] Ueno, M., Hoshinoo, K., Matsunaga, K., Kawai, M.,
Nakao, H., Langely, R., Bisnath, S., Assessment of
Atmospheric Delay Correction Models for the
Japanese MSAS, ION GPS, Salt Lake City, UT, pp.
2341-2350, 2001, 11-14 September 2001.
[8] http://www.fs.wettzell.de/publ.
[9] Introduction to GPS: The global positioning
system by El-Rabbany, Ahmed.
181
FPGA Based Design of NSGU to achieve Timing

Delay Bias and Time synchronization in
nanoseconds for Navigation Satellites
Durga Digdarsini, Gopichand Khot, Sanjay D. Mehta, T.V.S Ram Space Applications Centre (SAC),
Indian Space Research Organization (ISRO) Ahmedabad, Gujarat, India,digdarsini@sac.isro.gov.in
Abstract This paper describes the novel scheme for code on 5.115MHz sub carrier. The major
design and development of Actel FPGA based functions of NSGU are to receive uplink NAV
Navigation Signal Generation Unit (NSGU) for the data, store it in onboard memory, extract data
navigation payload, whose key function involves storing, depending upon time of applicability and validity
formatting and processing of bulk uplinked Navigation period, add timing information and format to
(NAV) data to prepare base band NAV signal for construct NAV subframes, process these subframes
onboard modulators. There are two critical prerequisite to prepare baseband signals for onboard modulators
from NSGU which will help in achieving position
of L5 and S band respectively. NAV data
accuracy < 20m on ground. One of the requirements is
to get Timing Delay bias in Nano seconds for which a Processing in NSGU includes Forward Error
novel method is adopted for design of NSGU using Correction (FEC) encoding, interleaving of encoded
master clock of 10.23MHz to achieve Timing Delay bias symbols, encryption and spreading with onboard
in nano seconds. Second requirement is setting of generated codes for SPS and RS services and
onboard time with ground reference time in nano selection of different data patterns at the output of
seconds which is most crucial necessity for position NSGU. This paper is further outlined as follows.
accuracy. An innovative scheme is developed in which Section II discusses the basic architecture of NSGU.
time synchronization between onboard and ground can Section III discusses the FPGA design for NSGU for
be achieved in nano seconds. This paper describes the
processing of NAV subframe to achieve Timing
design methodology of NSGU in Actel FPGA for
realization of Timing Delay bias in nanoseconds and Delay bias in nanoseconds. Section IV illustrates the
procedure for time setting to achieve time generation of onboard time and method of time
synchronization in nano seconds. synchronization between onboard time with ground
Keywords: NSGU, SPS, RS, NAV, Timing Delay Bias, reference time. Section V concludes the paper.
TOWC, WN, FPGA.
II. BASIC ARCHITECTURE OF NSGU
I. INTRODUCTION
As shown in Fig.1 NSGU consists of three major
Satellite Navigation has revolutionized the navigation functional units. SUMMIT DXE device for receiving
world, opening new opportunities in an increasing uplink NAV data on MIL-1553B interface to
number of sectors that requires high precision TCTM, Micro controller as main controller and for
positioning and timing information [2][4]. Generally formatting of NAV data subframes which uses
any Navigation satellite system targets basically two external RAM and PROM for storage/archival of
types of services named as Standard Positioning NAV data and program memory for software. Actel
Services (SPS) and Restricted services (RS). In FPGA which is the target hardware for complete
Navigation payload the Atomic Frequency Standard processing of formatted NAV data, timing generation
(AFS) acts as a basic clock source for the payload and setting, Encryption of RS code and spreading of
configuration which in turn provides master clock processed NAV subframe with SPS and RS codes.
source of 10.23MHz for NSGU and other The uplinked NAV data received from onboard
subsystems of payload. The NAV data is up TCTM through MIL 1553B protocol are stored in
linked periodically from ground and received by SRAM. Micro controller reads NAV data from
NSGU through onboard Telecommand-Telemetry SRAM and formats into subframe as per signal
(TCTM) unit. This NAV data is stored in NSGU, structure document. NAV subframes are then
formatted with onboard time and processed to transferred to FPGA memory. The complete
prepare as base band modulating signal for L5 and S processing of NAV subframe is carried out in FPGA
band onboard modulators. Each of L5 and S band of NSGU and processed frames are provided as base
provides SPS service with BPSK modulation with band signals to L5 and S band modulators. Fig. 2
1.023Mbps Spreading code and RS service with shows Flight Model (FM) of NSGU which consists
BOC ( 5, 2 ) modulation o n 2.046Mbps spreading of the main and redundant units.
182
convolutional encoder and interleaved by a block

interleaver to construct a NAV subframe @ 50 sps.
NAV data subframes are spreaded with SPS code
and data component of encrypted RS code to
generate L5 and S band signals for respective
modulators [1].
B. Requirement of Timing Delay bias in ns

The hardware delay of signal from generation to
antenna phase center (radiating element in spacecraft)
is very important and critical parameter in satellite
navigation. Each satellites Navigation message
Fig. 1: Block Diagram of NSGU contains parameters describing timing bias. A user
receiver uses these parameters to compute the
clock correction for each observation. The Timing
Delay bias is the parameter to correct the bias
component of the group delay which needs to be
transmitted in navigation message. This parameter
is used to improve the timing and position accuracy
of the user. The delay between the 1st chip of S band
Fig. 2: FM package of NSGU SPS code, and the 1st chip of S band SPS spreaded
data is known as Timing Delay bias for NSGU
III. FPGA DESIGN FOR TIMING DELAY subsystem. So this should be achieved in
BIAS IN NANOSECONDS nanoseconds by design of NSGU to provide best
position and timing accuracy for the user.
The Navigation data is formatted in a bit stream
known as major frame of 2400 symbols, consisting C. Calculation of Timing Delay bias for NSGU
of four subframes each of 600 symbols and operates
@ 50sps at the base band level.
Fig. 4: Functional Simulation for Timing Delay Bias
The functional simulation result in Fig. 4 show the

timing delay bias for NSGU in nano seconds. The
total time taken from 1st chip of SPS code where,
till writing data into interleaver is calculated to be
less than 10ms delay (worst case). So there is
Fig. 3: Processing of subframe in FPGA time margin for reading data from interleaver as all
other modules are operated with higher clock rates
A. Processing of NAV data subframe than
50 Hz. This time margin is utilized for reducing
The overall functionalities in FPGA of NSGU for the Timing Delay bias from NSGU with advance
processing of NAV data is shown in Fig. 3. The generation of subframes. Time taken from 1st
entire design of FPGA is driven by single clock of
chip/epoch of S band SPS code to 1st chip of S band
10.23MHz. Synchronization is maintained between
SPS spreaded data is found to be in nanoseconds. So
the derived clock signals for the entire operation in
the Timing Delay Bias for the NSGU is found to be
the design for achieving code to code and code to
in nanoseconds as shown in Fig. 4.
data coherency. SPS code is generated @ 1.023MHz
and RS code data component and Pilot are generated
IV. TOWC and WN GENERATION AND
@ 2.046MHz respectively. Both components of RS
SETTING
codes are passed through encryption logic. Onboard
time is generated from epoch@1ms form SPS code.
The onboard time present in NAV subframe is
The NAV data subframe is FEC encoded by a Rate
the most important parameter for Navigation
183
purpose [3]. In Navigation payload this onboard uploaded from ground, so that the time taken by
time is presented by a 27 bits counter with LSB telecommand to reach onboard and other processing
17 bits as Time of week counter (TOWC) and MSB delays doesnt affect the scheme, because difference
10 bits as week number (WN). TOWC and WN are of onboard time and Ground Reference time always
generated in FPGA of NSGU. remain within clock drift limit. After this the
onboard and ground time are set with accuracy of
A. TOWC Generation nanoseconds.
S-SPS epoch@1ms is used for generation of
TOWC in FPGA of NSGU. A novel scheme
consisting of counters [1] is implemented for TOWC
generation, which helps in achieving time
synchronization between onboard and ground in
nano seconds
B. Requirement of Time setting

Onboard satellite time needs to be synchronized
with Ground Reference system time with an
accuracy of nano seconds. The Ground Reference
system time is generated by using the ensemble of
Hydrogen maser and cesium frequency standards.
The ensemble algorithm generates the Ground
Reference time scale paper clock which is steered to
Universal coordinated time (UTC). The onboard time
in Navigation payload is presented by WN and
TOWC. The Ground Reference system time
obtained through Satellite Range and Integrity VI. CONCLUSION
Monitoring Stations receiver can be represented
using Ground Week no (GWN) and Ground Time of A hardware efficient architecture of Actel FPGA
week (GTOW). At the switch on of Payload the based NSGU for Navigation Payload is presented in
onboard time i.e. WN and TOWC are initialized to this paper. Generation of navigation signal is the
zero and one respectively. Since ground control most vital part of any navigation payload. NSGU is
station is synchronized with UTC time, all designed indigenously for Navigation payload to
Navigation satellites constellations are required to meet all the functional and interface requirements.
have same reference t i m e . Difference in the The code-code coherency and code-data coherency
received signals from different Navigation is maintained throughout the design meeting the
satellites to the user receiver will have greater impact requirement of payload. One of the most crucial
in user position and time accuracy. To meet this requirements for NSGU is to get Timing Delay bias
requirement every satellite should have facility of in Nano seconds which is achieved in this design.
setting the onboard time with best possible accuracy This will help to obtain better position accuracy by
to minimize the time difference. the system. The other novelty of the design is to
achieve time synchronization between onboard and
C. Procedure for Time Setting ground time in nanoseconds. For this an innovative,
Setting of counters by uploading time differences simple and elegant scheme is implemented onboard
between Ground Reference time and onboard time so that the path delay uncertainty does not affect the
converted into different count values by time setting procedure. The design is optimized to
telecommand is implemented in FPGA of NSGU occupy only 37% of total cells of the target device.
as shown in Fig. 5. Time setting can done in three
REFERENCES
steps in which after 1st step synchronization is
achieved in seconds, after 2nd step less than mili [1] Durga Digdarsini, Rajat Arora, Vijay V. Kadam, Gopichand
Khot, Sanjay D. Mehta, T.V.S. RAM, Design Implementation and
second and after 3rd step synchronization is Realization of Navigation Signal Generation Unit, Journal of
Spacecraft Technologies, (JST) Bangalore, 2011.
achieved in nano seconds. The main challenge in [2] Hofmann-Wellenhof, GNSS, Global Navigation Satellite System,
time setting is to make the scheme independent of GPS, GLONASS, Galileo & more Springer Wein Newyork, 2007.
path delay from ground to onboard. To combat this, [3] ICD published from NAVSTAR GPS named IS_GPS_200D on
7th December, 2004.
the logic implemented onboard is to add/subtract the [4] Global Positioning System Theory & Applications Edited by.
difference of time in terms of count which is to be BRAD FORD W. Parkinson
184
IRIS Signal Dispensation and Cataloging by

Means of Compound Gabor Filter
Prof. P.V. Rama Raju, R.DivyaSree, P.Lakshmi, Y.L.Prannetha and Mr. CH. Ganesh
Dept. of Electronics & Communication Engineering, SRKR Engineering College, Bhimavaram 534 204, A.P.
(Affiliated to Andhra University, A.P., India) pvramaraju41@rediffmail.com, cell: 9010144688.
Abstract- A resourceful iris processing method is Voice verification

suggested. In the suggested method the iris segmentation,
normalization, feature extractions are based on the 1.1 Iris Introduction
surveillance that the pupil has lower intensity than the
iris, and the iris has lower intensity than the sclera. By
noticing the edge between the pupil and the iris and the The iris (irises) is a thin, colored ring, circular
edge between the iris and the sclera, the iris area can be structure in the eye, responsible for controlling
parted from pupil and sclera. Then the normalization the diameter and size of the pupils. The iris is called
technique is applied to reduce the dimensional
the Living password because of its unique, random
inconsistencies later encoding technique is applied to
extract the unique feature of iris image, and the resulting features. Its always with human being and cant be
one- dimensional (1-D) signals are quantized using phase stolen or faked. As such it makes an excellent
quantization. The phase quantization process produces a biometrics identifier and classifier.
number of bits of information at various resolution levels
over pixels on the normalized iris. The Hamming distance
is employed for classification, if the hamming distance
between two templates is less than the set threshold value
then two templates are declared to match or else they are
not matched. A case identification program does the
unique signal analysis and completes the pattern
classification. This program gives out its output in
graphical form which indicates individual identification
and gives a statement which makes an individual case
identification.
Keywords- Segmentation, Normalization, Feature 1.2 Eye Pupil
extraction, Phase quantization.
The pupil is a hole located in the center of the iris of
I. INTRODUCTION the eye that allows light to enter the retina. It
appears black because most of the light entering the
A dictionary defines image as a reproduction or pupil is absorbed by the tissues inside the eye. In
representation of the form of a person or thing. response to the amount of light entering the eye,
Digital image can be defined to be a numerical muscles attached to the iris expand or contract the
representation of an object to be sampled in an aperture at the center of the iris, known as the
equally spaced rectangular grid pattern, and pupil. The larger the pupil, the more light can enter.
quantized in equal intervals of gray-level as function
of two dimensions. The word is carrying out for the 1.3 Iris Recognition methodology
simpler access conrols to personal authentication
systems and it looks like biometrics may be the
answer. Hence human body can be used to uniquely
identify. Furthermore, biometrics is best defined as
measurable physiological and/or behavioral
characteristics that can be utilized to verify the
identity of an individual. They include the following:
Iris scanning
Facial recognition
Fingerprint verification
Hand geometry
Retinal scanning
Signature verification Figure 2: Flow diagram of the iris recognition steps.
185
To examine the recognition system, eye image 3.Templates of irises.

will be used as input. Iris recognition relies on the 4Assignment in Databank.
unique patterns of the human iris to identify or 5.Iris signal classification and identification.
verify the identity of an individual. Eye localization
and iris segmentation are sounds the same, These III. INTER CLASS SAMPLE DATABASE
processes are done by using Canny edge detection CLASSIFICATION
technique, circular Hough transform (CHT) a
standard computer vision algorithm, is commonly Case1: Test signal- S1036L07.jpg (Left Eye)
employed to deduce the radius and center coordinates This signal has been taken from the CASIA iris
of the pupil and iris regions.The iris and pupil image database collected by Institute of Automation,
boundaries can be approximated as concentric circles Chinese Academy of Sciences.
and the outer boundaries of iris are detected with the
help of center of pupil. For a particular circle the
change in intensity between normal pointing toward
center and away from center is measured. The
radius having highest change in intensity is
considered as outer boundary.
Once the iris region is successfully segmented from
an eye image, the next stage is to transform the iris
region so that it has fixed dimensions in order to Figure 3: Test signal left eye template.
allow comparisons. Iris normalization is done in After processing and analyzing the iris signal, the
order to make the image independent of the template of the database signal has been plotted
dimensions of the input image. After iris is as shown in above figure. In figure 3 the image is
normalized, the algorithm is used to encode the iris being tested S1036L07.jpg, name reveals that
data. This process extracts features from the seventh left eye sample image of a 36th person in the
normalized iris images and encodes it to generate iris database.
templates. The resultant graphs shown in section4.
For obtaining test parameter, Hamming distance
gives a measure of how many bits are the same
between two bit templates. Using the
Hamming distance of two bit patterns, a decision can
be made as to whether the two patterns were
generated from different irises or from the same one.
II. CASE STUDY AND RESULT ANALYSIS Figure 4: Matched eye template from the database.
Figure 4 show that the resultant matched template
After processing and analyzing the iris signal,
from the database. The Hamming distance
The template of the each database signal have been distribution and matching condition between the
plotted in Matlab as shown in corresponding figures. templates are shown below. The result is plotted in
The results are plotted in terms of scales (Hamming terms of scales (Hamming distance) on the y-axis
distance) on the y-axis and the number of images in and the number of images in the database (sample
the database (sample images) on x-axis. The images) on x-axis.
matched result at that point is represented by a blue
color bar with the threshold value. Red bar indicates In figure5, the blue bar at position25 indicates that
that unmatched condition. In this exposition two matching condition below the threshold value. Since
cases are considered under test. database contains the different persons data,
For every Image in the dataset the following values tested image is matched with the best compatible
have been analyzed: image. The image is being used for test is matched
1.Iris radius and its center coordinates. with processed left eye images in the database
2.Filter parameters. corresponds to the same person, the Hamming
distance between tested template and matched
186
template is taken as the optimum threshold horizontal black line, it will be differed for each
value for the same category. In the above example individual test. Hence test of above signal is done
the optimum threshold value is set programmatically successfully and the corresponding observations will
about 0.4, it will be differed for each individual test. be discussed in section 4.
Figure 7: Tested image template unmatched with the any of

templates in the database
Figure 5: Tested image template matched with first template in the
database
IV. EXPERIMENTAL RESULT ANALYSIS
Case2: Test signal- S1094R03.jpg (Right Eye)
4.1 F ilter parameters, Decidability, Template
In this case the signal has been taken unlike the
above case, the right as tested eye. size, Number of shifts, Number of filters
The best filter parameters were found through
experimentation with the CASIA data set to
compare performance. The decidability value
determines the optimum parameters; those are
wavelength, template size, number of shifts, number
filters.
Figure 6: Test right sample template. Where d' is decidability, mean of the intra-class
After processing and analyzing the iris signal, the distribution s and the mean of the inter-class
template of the database signal has been plotted distribution d and also the standard deviation of
as shown in above figure. In figure6 the image is the intra-class and inter-class distributions 2, and
being tested S1094R03.jpg, name reveals that third 2d respectively. The results of experiment, various
right eye sample image of a 94th person in the filter parameters to encode iris templates are
database. observed below table1.
Figure 7 show that the resultant Hamming distance
distribution and not-match condition between the
templates are shown below, all the bar lines
exceeds the threshold value line 0.4, hence there is
no match at any case, the result is plotted in terms of
scales (Hamming distance) on the y-axis and the
number of images in the database (sample images) on
x-axis.
The image is being used for test is a right eye not-

matched with any of processed right eye images in
the database corresponds to the same person, which
can be observed at bars on x-axis. In the above
example the optimum threshold value is set
programmatically about 0.4 Which is shown as
187
Figure 10: Mean of inter-class HD distribution Vs Number of shifts

of two images of same class.
Figure 8: Decidability Vs center wavelength using filter bandwidth
0.5
With reference to Figure10, the CASIA data set
One factor, which will significantly influence the
requires 8 shifts to reach its minimum
recognition rate, is iris template size. For the CASIA
optimum threshold. However, it is noted that 4-to-8
data set, the optimum template size was found to
shifts are enough to be able to account for most of
be 20x240 pixels.
the rotational inconsistencies in the CASIA database.
Figure8 shows that there exits an optimum center
wavelength 18 for the data set, which produces V. OBSERVATIONS FROM THE PLOTS
maximum decidability d = 6.2160. Furthermore,
experiment results show that encoding templates with 1. The template generated for the initial scales is
multiple filters does not produce better decidability multiplication of number of filters used to the
values, therefore the optimum number of filters is one
signal processing, order of pixel array.
and it produces a compact iris template.
2. The 3D plot for 20-by-240, color shift
The optimum s/f value was found to be 0.50, Higher indicates shift of bits from 0 to 1 or 1 to 0.
the decidability, greater the separation of same class 3. The decreasing scale number to 10-by-1000
and inter-class distributions, which allows for more shown, easy to compare the given signal for
accurate recognition. each processing value.
4. The bar graph is generated with matching metric
looks better and suitable for finding optimum
signal classification.
5. Blue bar indicates the matching condition, red bar
indicates unmatched condition, and the circle line is
the optimum threshold value on y-axis.
6. Instead of numerical observations comparison
of signal analysis using the Hamming distance
bar graph yields better and accurate Identification.
Figure 9: Inter-class comparison with eight shifts VI. CONCLUSION
As the number of shifts increases, the mean of the This exposition has presented an Inter-class iris
inter-class distribution will converge to an optimum recognition system, which was tested using set of
value below the statistical value 0.5. It can be seen in images as databases of greyscale eye images from
figure 9.
different. Firstly, an efficient iris processing method
is processed. In the processed method the iris
segmentation, normalization, feature extractions are
being done. Then the encoding technique is applied
to extract the unique feature of iris image, and
188
the resulting one-dimensional (1-D) signals are phase

quantized using different dissimilarity functions. The
phase quantization process produces a bitwise
template containing a number of bits of information
at various resolution levels. The Hamming distance
is employed for classification of iris templates. The
Matlab program gives out its output in graphical
form which indicates individual identification and
gives a statement which makes an individual case Prof.P.V. Rama Raju received his Masters Degree in Microwave
identification. and Optical Engineering at the M.K. University, Tamil Nadu,
India. He is a Professor at the Department of Electronics and
Communication Engineering S.R.K.R. Engineering College, AP,
VII. FUTURE SCOPE India. His research interests include biomedical-signal Processing,
signal processing, VLSI design and microwave anechoic
The system presented in this publication was able chambers design. He is author of several research studies published
to perform Inter-class classification. However there in national and international journals and conference proceedings.
R.DivyaSree, P.Lakshmi, Y.L.Prannetha are pursuing their U.G.
are still a number of issues which need to be
Degree in ECE, Dept. at S.R.K.R Engineering college,
addressed. First of all, Inter-class classification Bhimavaram, WGDt., AndhraPradesh, India.
system to be implemented accurately. The Mr CH. Ganesh has completed his Masters Degree in
automatic segmentation was not perfect, since it Communication Systems in ECE, Dept. at S.R.K.R Engineering
could not successfully segment the iris regions for college, Bhimavaram, WGDt., AndhraPradesh, India.
all of the eye images in the database. Another

extension to the system would be to interface it to an
iris acquisition camera.
REFERENCES
[1] http://www.csse.uwa.edu.au/~masekl01/
[2] Recognition of Human Iris Patterns for Biometric
Identification, Libor Masek The University of Western Australia,
2003
[3] http://en.wikipedia.org/wiki/Iris_(anatomy)
[4] Canny, John, "A Computational Approach to Edge Detection,"
IEEE Transactions on Pattern Analysis and Machine
Intelligence,Vol. PAMI-8, No. 6, 1986, pp. 679-698.
[5] "A Human Identification Technique Using Images of the Iris
and Wavelet Transform" W. W. Boles and B. Boashash
[6] "Iris Recognition: An Emerging Biometric Technology"
RICHARD P. WILDES, MEMBER, IEEE
[7] Optimized Daugmans Algorithm for Iris Localization Dr.
Mohamed A. Hebaishy National Authority for Remote
Sensing and Space Science Gozif Titp St., Elnozha Elgididah.
Egypt (11769), Cairo.
[8] Recognition of Human Iris patterns for biometric
identification, Libor Masek, Dr.Peter Kovesi, The University of
western Australis.
[9] Exploit modeling for categorization of electrocardiogram
signals through wavelet transform. prof.P.V.Ramaraju,
V.Malleswara Rao, Mr CH. Ganesh. 6th international multi
conference on intelligent system, IISN-2012, March 16-18,NIT
Klawad,Haryana,India.
[10] ECG signal processing and classification via heart rate
fortitude in the vicinity of continuous wavelet
transforms.prof.P.V.Rama raju, V.Malleswara rao and Mr
CH.Ganesh, IEEE conference, International journal ACNCN-
2012,AU college of engineering, Visakhapatnam.
189
In Orbit Clock Performance of IRNSS RAFS

Kaitha Rajaiah1, Umesh Swami2, Manamohan Kamat3, Nirmala S4, S C Ratnakara5, A S Ganeshan6
Space Navigation Group, ISRO Satellite Center, Bangalore, India
1krajaiah@isac.gov.in,2urswami@isac.gov.in, 3mvkamat@isac.gov.in, 4nirmalas@isac.gov.in,
5ratnakar@isac.gov.in, 6asganesh@isac.gov.in
Abstract: Accuracy and repeatability of IRNSS (Indian Earth Orbit) orbit and four satellites in GSO
Regional Navigation Satellite System) is ultimately (Geosynchronous orbit) orbit. All the IRNSS
determined by the inherent stability of the on-board satellites are intended carry 3 RAFS: one primary
clocks. In IRNSS, the users position is determined by and two redundant (hot redundant and cold
measuring the time of travel of the signals broadcasted redundant).
from satellites. Hence, any error on the measurement
Accuracy and repeatability of IRNSS will be
of time leads to an error on the users position.
Therefore the on-board clock behaviour has to be
ultimately determined by the inherent stability of the
continuously monitored and any malfunctioning has to on-board clocks. The variation in the clock
be noticed immediately to improve the capability of the frequency or phase with time directly affects the
timing system for accurate positioning of the user. accuracy of the navigation solution. The Overlapped
Each IRNSS satellite carries three Rubidium Atomic Allan variance (OAVAR) is a standard statistical
Frequency Standards (RAFS), one primary and tool to estimate clock stability which is a
two redundant clocks (Hot redundant & Cold fundamental quantity used to assess the performance
redundant). Allan variance is commonly used to of a clock and its impact on the performance of the
estimate the stability of IRNSS satellite clock. The
system that uses the clock [1,2]. In navigation
classical Allan variance cannot capture the non-
stationarity in the random signal and it will give the systems, a stable clock is predictable and yields a
average stationary behavior over the whole low user range error, resulting in better positioning.
period. The stability of IRNSS satellite clock can A real time computation of Allan variance has been
change with time due to several factors, such as implemented to study instantaneous clock stability
temperature, humidity, radiations, aging and sudden and thus detect the possible non-stationary behaviour
breakdowns. Hence, it is important to represent the and related clock anomalies.
time-varying stability of the clock. In the present paper the initial results of the IRNSS-
This paper provides the In-Orbit clock performance 1A clock performance are shown using conventional
of IRNSS RAFS using both conventional Overlapping
Overlapping Allan Variance (OAVAR) and the real
Allan Variance (OAVAR) and the newly proposed Real
time Allan Variance (RAVAR). RAVAR has been time Allan variance (RAVAR) computation.
implemented to study instantaneous clock stability and
thus detect the possible non- stationary behaviour and II. SATELLITE CLOCK BEHAVIOR
related clock anomalies such as data gaps, outliers,
phase jumps and frequency jumps. IRNSS payload A thorough understanding of the behaviour of the
records the phase difference between primary and satellite clocks is required in order to use the current
secondary RAFS and transfers it to ground station system optimally and to plan future improvements
through telemetry(TM). Along with the stability the in upcoming navigation satellites. Deterministic and
temperature of RAFS and attitude variations of the
stochastic perturbations cause the variation of
IRNSS-1A are also monitored.
clocks frequency stability and accuracy. The main
Keywords: Satellite clock, Allan variance, Atomic clock, deterministic behaviour in atomic clocks is the drift
Clock characterization.
that is a systematic variation of the frequency
generated by the clock. It may be due to intrinsic
changes in the clock itself, to the ageing or to
I. INTRODUCTION environmental factors. Generally, the satellite clock
behaviour is mainly analysed in terms of:
Indian Space Research Organization (ISRO) is
implementing the Indian Regional Navigation 1. Clock deterministic behaviour: Bias, Drift, and
Satellite System (IRNSS) to provide navigation Drift- Rate.
services to various users over the intended service 2. Clock stochastic behaviour: Frequency
area. It is an independent navigation system of India Stability.
designed to provide position accuracy better than Frequency stability indicates the degree to which an
20m for dual frequency users over India and the oscillating signal produces the same frequency for a
region extending about 1500 Km around India. The specified interval of time. Depending on this
space segment of IRNSS will consist of seven information, one can judge whether the clock is
satellites: three satellites in GEO (Geo-stationary behaving as per the specification or not. If it is not
190
meeting the specification, switching over of primary III. LIMITATIONS OF OVERLAPPED ALLAN
clock is done or satellite clock is physically adjusted. VARIANCE
The OAVAR is the widely used statistics to from a set of M frequency measurements for averaging
characterize the frequency stability of atomic clocks, time t=mt0, where m is the averaging factor and t0 is
but it has its limitations that it cannot capture anomalies the basic measurement interval, by the expression
and the nonstationarity in the random signal. It simply
indicates the average stationary behavior of the clock
for the given period. But the OAVAR fails to provide
the stability of the clocks under the following cases. Now in real time for each next coming data point, the
1. Data Gaps: During the IOT since various satellite first data point will be removed from the existing data
subsystems are tested, continuous availability of so that always same length of data will exist. The
telemetry data from onboard phase meter was not proposed algorithm follows as:
guaranteed, thus leading to data gaps. a) Outlier screening and removal of bad data.
2. Clock Anomalies: The anomalies such as phase b) Computation of frequency accuracy
jump, frequency jump and outliers cannot be detected c) Phase jump detection and computation of its
by OAVAR since they are averaged out in the process. magnitude.
d) Frequency jump detection and computation of its
The proposed algorithm (RAVAR) with real time clock magnitude.
stability technique is designed to provide the clock e) Computing OAVAR corresponding to first data
stability measure even with data having the above point
discussed problems. In order to show the effectiveness f) Computing OAVAR corresponding to last data
of proposed algorithm in detecting time-varying clock point
behaviour that are not often visible from the OAVAR, a g) Computing final OAVAR for currently existing
simulation of all clock related anomalies was done and data length by removing and adding the effect of first
the results are discussed in the later section. and last data points OAVAR respectively.
h) Take next coming data point and remove first data
IV. PROPOSED ALGORITHM FOR REAL-TIME point so that the data length is always same.
CLOCK For each new epoch all the above steps will be repeated.
CHARACTERIZATION So the output will give the real time stability of the
clock behaviour and also the real time information such
Phase meter will give the offsets between the on-board as time and magnitude about the clock anomalies.
primary and hot redundant clock. The proposed
algorithm is designed to work for both clock phase data V. RESULTS AND DISCUSSIONS
and frequency data.
To show the effectiveness of the proposed clock
stability technique in case of clock anomalies, a clock
phase data simulation has been done with anomalies
such as outliers, clock jumps and data gaps. The results
of the study are shown in the below section.
Simulated data:
The simulated frequency data with phase jumps and its
stability plots are as shown in figure 2 and figure 3. The
phase jumps can be observed from figure 3 as
jumps/spikes in the surface plot.
Fig.1 Block Diagram of RAVAR.
Initially one hour data will be accumulated and

OAVAR is computed. The OAVAR can be estimated Fig.2 Simulated frequency data with phase jumps.
191
conventional OAVAR and RAVAR along with its

specifications.
Fig.3 Stability plot for frequency data shown in Fig.2

Fig.6 Frequency data for RAFS2 RAFS1
Fig.4 shows the simulated frequency data with outliers
and frequency jumps. Fig.5 shows the corresponding
stability plot. From Fig.5 it is observed that the first two
spikes indicate the outliers and the next two ramp
shaped spikes indicate the frequency jumps.
Fig.7 OAVAR for RAFS2 RAFS1
Fig.4 Simulated frequency data with outliers and

frequency jumps. Fig.8 RAVAR for RAFS2 RAFS1
RAFS1-RAFS3 Results: The phase meter TM data for
02/08/2013 (24hrs) has been used to determine the
relative stability of the onboard clocks. Figure 9 shows
the frequency data along with its frequency accuracy
computed using the phase meter measurements. Figure
10 and Figure 11 shows the clock stability computed
using conventional OAVAR and RAVAR along with its
specifications.
Fig.5 stability plot for the frequency data shown in

Fig.4
IRNSS-1A RAFS performance: After switching on,

IRNSS-1A clock performance is been continuously
monitored using both conventional OAVAR and the
proposed RAVAR. Clock performance of various Fig.9 Frequency data for RAFS1 RAFS3
RAFS combinations have been analysed during IRNSS-
1A In Orbit tests. The results of the IRNSS-1A clock
performance is been shown in the following section.
RAFS2-RAFS1 Results: The phase meter TM data for
23/07/2013 (24hrs) has been used to determine the
relative stability of the onboard clocks. Figure 6 shows
the frequency data along with its frequency accuracy
computed using the phase meter measurements. Figure
7 and Figure 8 shows the clock stability computed using
192
Fig.10 OAVAR for RAFS1 RAFS3 motivation, help and their constructive comments on
the real time characterisation of on-board clocks.
REFERENCES
[1] Definitions of Physical Quantities for Fundamental Frequency and

Time Metrology, IEEE standard 1139, 1999.
[2] D. W. Allan, statistics of atomic frequency standards, Proc.
IEEE, vol. 54, no. 2, pp. 221230, 1966.
[3] Senior, K. L., Ray, J. R., and Beard, R. L.Characterization of
periodic variations in the GPS satellite Clocks.GPS Solutions, DOI
Fig.11 RAVAR for RAFS1 RAFS3 10.1007/s10291-0080089-9, Feb.2008.
[4] W.J. Riley Hand book of frequency stability analysis
VI. SUMMARY AND CONCLUSION
Since Allan variance gives the average stationary

behavior for the given period, it cannot capture the
anomalies and non-stationarity behaviour in the clock
signal. The method discussed in the manuscript
represents the instantaneous stability of an atomic clock
with respect to time and it will detect the anomalies
present in the clock behaviour. The proposed method
RAVAR will detect the outliers, phase jumps,
frequency jumps, data gaps, and periodic nature present
in the clock behaviour. In this paper the initial clock
stability results of IRNSS-1A have been shown. Both
the conventional OAVAR and the proposed RAVAR
techniques have been used to determine the clock
stability. The summary of stability computations using
both the techniques is tabulated with respect to sections
below.
Fig.12 RAFS2-RAFS1 summary as on 23/07/2013
Fig.13 RAFS1-RAFS3 summary as on 02/08/2013
It can be clearly seen that all the RAFS of the IRNSS-

1A have been performing within the specifications and
the results of proposed RAVAR technique is very close
to the OAVAR stability.
ACKNOWLEDGMENT
The authors wish to acknowledge all the members of

Space Navigation Group (SNG), ISAC for their
193
Embedded Extended Visual Cryptography

Schemes
Prof. M Rama Bai, Dept. of CSE, MGIT, Hyderabad, and C.Raghunandhan Reddy, Student, MGIT
Abstract Avisual cryptography scheme (VCS) is a kind (i.e., sharing secret image), have been found, for
of secret sharing scheme which allows the encoding of a example, authentication and identification,
secret image into shares distributed to participants. The watermarking and transmitting passwords etc.
beauty of such a scheme is that a set of qualified The associated secret sharing problem and its
participants is able to recover the secret image without physical properties such as contrast, pixel expansion,
any cryptographic knowledge and computation devices. and color were extensively studied by researchers
An extended visual cryptography scheme (EVCS) is a worldwide. For example, showed constructions of
kind of VCS which consists of meaningful shares threshold VCS with perfect reconstruction of the
(compared to the random shares of traditional VCS). In black pixels.
this paper, we propose a construction of EVCS which is
realized by embedding random shares into meaningful
covering shares, and we call it the embedded EVCS.
Experimental results compare some of the well-known
EVCSs proposed in recent years systematically, and show
that the proposed embedded EVCS has competitive visual
quality compared with many of the well-known EVCSs in
the literature. In addition, it has many specific advantages
against these well-known EVCSs, respectively.
Furthermore, Eisenet al. proposed a construction of
I. INTRODUCTION threshold VCS for specified whiteness levels of the
recovered pixels. The term of extended visual
The basic principle of the visual cryptography cryptography scheme (EVCS) was first introduced by
scheme (VCS) was first introduced by Naor and Naoret al. in, where a simple example of (2,2)-EVCS
Shamir. VCS is a kind of secret sharing scheme that was presented. In this paper, when we refer to a
focuses on sharing secret images. The idea of the corresponding VCS of an EVCS, we mean a
visual cryptography model proposed in is to split a traditional VCS that have the same access structure
secret image into two random shares (printed on with the EVCS.
transparencies) which separately reveals no
information about the secret image other than the size Generally, an EVCS takes a secret image and original
of the secret image. The secret image can be share images as inputs, and outputs shares that satisfy
reconstructed by stacking the two shares. The the following three conditions: 1) any qualified subset
underlying operation of this scheme is logical of shares can recover the secret image; 2) any
operation OR. In this paper, we call a VCS with forbidden subset of shares cannot obtain any
random shares the traditional VCS or simply the information of the secret image other than the size of
VCS. In general, a traditional VCS takes a secret the secret image; 3) all the shares are meaningful
images.
image as input, and outputs shares that satisfy two
conditions:
II. IMPLEMENTAION AND METHODOLOGY
1) any qualified subset of shares can recover the
secret image; 2) any forbidden subset of shares We implement the Visual Cryptography. We used
cannot obtain any information of the secret image LZW Data Compression algorithm. The LZW data
other than the size of the secret image. An example of compression algorithm is applied for the gray scale
traditional (2,2)-VCS can be found in Fig. 1, where, image here. As a pre-processing step, a dictionary is
generally speaking, a VCS means any out of shares prepared for the gray scale image. In this dictionary,
could recover the secret image. In the scheme of Fig.
1, shares (a) and (b) are distributed to two the string replaces characters with single quotes.
participants secretly, and each participant cannot get Calculations are done using dynamic Huffman
any information about the secret image, but after coding. In compression of greyscale image select the
stacking shares (a) and (b). information pixels. Then generate halftone shares
the secret image can be observed visually by the using error diffusion method. At last filter process is
participants. VCS has many special applications, for applied for the output gray scale images. Filtersare
example, transmitting military orders to soldiers who used to improve the quality of reconstructed image to
may have no cryptographic knowledge or minimize the noises for sharpening the input secret
computation devices in the battle field. Many other
applications of VCS, other than its original objective image.
194
ALGORITHM: Decoding:
Input: The c x d dithering matrix D and a pixel with The decoding algorithm works by reading a value
gray-level g in input image I. from the encoded input and outputting the
Output: The halftoned pattern at the position of the corresponding string from the initialized dictionary.
pixel At the same time it obtains the next value from the
input, and adds to the dictionary the concatenation of
For i=0 to c-1 do the string just output and the first character of the
For j=0 to d-1 to do string obtained by decoding the next input value. The
If g<=Dij then print a black pixel at position (i,j); decoder then proceeds to the next input value (which
Else print a white pixel at position (i,j); was already read in as the "next value" in the
previous pass) and repeats the process until there is
no more input, at which point the final input value is
decoded without any more additions to the dictionary.
In this way the decoder builds up a dictionary which
is identical to that used by the encoder, and uses it to
decode subsequent input values. Thus the full
dictionary does not need be sent with the encoded
data; just the initial dictionary containing the single-
character strings is sufficient (and is typically defined
beforehand within the encoder and decoder rather
than being explicitly sent with the encoded data.).
Creating Transparencies:
This scheme provides theoretically perfect secrecy.

Encoding: An attacker who obtains either the transparency
image or the screen image obtains no information at
A high level view of the encoding algorithm is shown all about the encoded image since a black-white
here: square on either image is equally likely to encode a
clear or dark square in the original image. Another
valuable property of visual cryptography is that we
1. Initialize the dictionary to contain all strings
can create the second layer after distributing the first
of length one.
layer to produce any image we want. Given a known
2. Find the longest string W in the dictionary
transparency image, we can select a screen image by
that matches the current input.
choosing the appropriate squares to produce the
3. Emit the dictionary index for W to output
desired image. One of the most obvious limitations of
and remove W from the input.
using visual cryptography in the past was the problem
4. Add W followed by the next symbol in the
of the decoded image containing an overall gray
input to the dictionary.
effect due to the leftover black sub pixel from
5. Go to Step 2. encoding. This occurred because the decoded image
is not an exact preproduction, but an expansion of the
A dictionary is initialized to contain the single- original, with extra black pixel. Black pixel in the
character strings corresponding to all the possible original document remains black pixel in the decoded
input characters (and nothing else except the clear version, but White pixel becomes gray. This resulted
and stop codes if they're being used). The algorithm in a lot of contrast to the entire image. The extra
works by scanning through the input string for black sub pixel in the image causes the image to
successively longer substrings until it finds one that is become distorted.
not in the dictionary. When such a string is found, the
index for the string less the last character (i.e., the D - Secret information, K - Number of shares
longest substring that is in the dictionary) is retrieved generated from D, share - piece of information.Divide
from the dictionary and sent to output, and the new data D into n pieces in such a way that D is easily
string (including the last character) is added to the reconstruct able from any k pieces, but even complete
dictionary with the next available code. The last input knowledge of any k-1 pieces reveals no information
character is then used as the next starting point to about D. Stacking two pixels (each consists of four
scan for substrings. sub-pixels) can occur for example the following two
195
cases: Secret sharing scheme is a method of sharing known schemes, such as the fact that it can deal with
secret information among a group of participants. In a gray-scale input images, has smaller pixel expansion,
secret sharing scheme, each participant gets a piece of is always unconditionally secure, does not require
secret information, called a share. When the allowed complementary share images, one participant only
coalitions of the participants pool their shares, they needs to carry one share, and can be applied for
can recover the shared secret; on the other hand, any general access structure. Furthermore, our
other subsets, namely non-allowed coalitions, cannot construction is flexible in the sense that there exist
recover the secret image by pooling their shares. In two trade-offs between the share pixel expansion and
the last decade, various secret sharing schemes were the visual quality of the shares and between the secret
proposed, but most of them need a lot of image pixel expansion and the visual quality of the
computations to decode the shared secret information. shares.
The basic 2 out of 2 visual cryptography model
consist of secret message encoded into two
transparencies, one transparency representing the REFERENCES
cipher text and the other acting as a secret key. Both
transparencies appear to be random dot when
inspected individually and provide no information [1] G. R. Blakley, Safeguarding cryptographic keys, in Proc.
National Computer Conf., 1979, vol. 48, pp. 313317.
about the original clear text. However, by carefully [2] T. H. Chen and D. S. Tsai, Owner-customer right
aligning the transparencies, the original secret protection mechanism using a watermarking scheme and a
message is reproduced. The actual decoding is watermarking protocol, Pattern Recognit., vol. 39, pp.
15301541, 2006
accomplished by the human visual system. The [3] M. Naor and A. Shamir, Visual cryptography, in Proc.
original is encrypted into 2 transparencies you need EUROCRYPT 94, Berlin, Germany, 1995, vol. 950, pp.
both transparencies to decode the message. 112, Springer-Verlag, LNCS.
[4] C. Blundo, A. De Bonis, and A. De Santis, Improved
schemes for visual cryptography, Designs, Codes and
Un-hiding Image from Transparency: Cryptography, vol. 24, pp. 255278, 2001.
[5] A. Shamir, How to share a secret, Commun.ACM, vol.
22, no. 11, pp. 612613, 1979.
The simplest form of visual cryptography separates [6] M. Naor and B. Pinkas, Visual authentication and
an image into two layers so that either layer by itself identification, in Proc. CRYPTO97, 1997, vol. 1294, pp.
322336, Springer-Verlag LNCS.
conveys no information, but when the layers are [7] N. K. Prakash and S. Govindaraju, Visual secret sharing
combined the image is revealed. One layer can be schemes for color images using halftoning, in Proc. Int.
printed on a transparency, and the other layer Conf. Computational Intelligence and Multimedia
Applications (ICCIMA 2007), 2007, vol. 3, pp. 174178.
displayed on a monitor. When the transparency is [8] P. Tuyls, T. Kevenaar, G. J. Schrijen, T. Staring, and M.
placed on top of the monitor and aligned correctly, Van Dijk, Security displays enabling secure
the image is revealed. For each image pixel, one of communications, in Proc. First Int. Conf. Pervasive
the two encoding options is randomly selected with Computing, Boppard Germany, Springer-Verlag Berlin
LNCS, 2004, vol. 2802, pp. 271284.
equal probability. Then, the appropriate colorings of [9] G. Ateniese, C. Blundo, A. De Santis, and D. R. Stinson,
the transparency and screen squares are determined Visual cryptography for general access structures, Inf.
based on the color of the pixel in the image. Computat., vol. 129, pp. 86106, 1996.
III. CONCLUSION
In this paper, we proposed a construction of EVCS

which was realized by embedding the random shares
into the meaningful covering shares. The shares of
the proposed scheme are meaningful images, and the
stacking of a qualified subset of shares will recover
the secret image visually. We show two methods to
generate the covering shares, and proved the
optimality on the black ratio of the threshold covering
subsets. We also proposed a method to improve the
visual quality of the share images. According to
comparisons with many of the well-known EVCS in
the literature the proposed embedded EVCS has
many specific advantages against different well-
196
FPGA Implementation of Low Power, Area

Efficient Radix 8 Multiplier using Verilog HDL
V V S Vijaya Krishna#1, T.Pitchaiah#2, Sk. Sadulla#3, P V Sridevi*4.
#
School of electronics, Vignan University, Vadlamudi, India
*
ECE Dept Andhra University, Visakhapatnam, India
Email: v.vijayakrishna@gmail.com;2sadulla09@gmail.com;3tp1977.ece@gmail.com;4pvs6_5@yahoo.co.in
1
Compressor and a SQRT CSLA with BEC at the last

Abstract The multiplier is an essential element of the stage to reduce the power and area.
digital signal processing such as filtering and
convolution. The design of an efficient multiplier in
terms of power, delay and area simultaneously, has
FPGA based machines are less time consuming,
become a very challenging problem. This paper flexible, programmable and reduces hardware as it is
presents a Low power, Area efficient multiplier which reprogrammable. Use of such machines saves time
uses a 4:2 Compressor and a Square Root Carry Select and cost.
Adder (SQRT CSLA) with a Binary to Excess-1 The conceptual model for the proposed multiplier
Converter (BEC) Block. This is compared with Radix 8 is implemented using Virtex 5 XC5VLX50T FPGA
booth multiplier with 4:2 Compresssor and Carry Look board. In Section 2, proposed multiplier architecture is
ahead Adder (CLA). The deign entry is done in Verilog elaborated and details of blocks are explained in
and Simulated using Xilinx ISE. It is then synthesized sections 3,4 5. All the evaluated results of the
using Xilinx XST and the generated bit stream file is proposed design with the required information of
implemented on Virtex 5 FPGA board. screenshots, hardware schematics and comparisons
are placed in section 6. And finally the Conclusion
Keywords-; Booth Encodin; CSLA; BEC; Verilog and future scope is covered in final section.
implementation; Xilinx ISE simulator; FPGA; VIRTEX 5
I. INTRODUCTION II. ARCHITECURE

With the rapid advances in multimedia and
communication systems, real-time signal processing
and large capacity data processing are increasingly The architecture of the proposed multiplier is
being demanded. A multiplier is one of the key shown in fig.1. It consists of partial product generator
hardware blocks in most digital and high performance using booth encoder, partial product reduction using
systems such as FIR filters, digital signal processor, compressor, and a SQRT CSLA with BEC at the last
microprocessors etc. With advances in technology, stage of Wallace tree.
many researchers have tried to design multipliers
which offer either of the following- high speed, low
power consumption, less area combination of them in
multipliers, thus making them compatible for various
high speed, low power, and compact VLSI
implementations. However, area and speed are two
conflicting constraints.
Multiplication operation involves generation of

partial products and their addition. The speed can be
increased by reducing the number of partial products,
adding the partial products using Wallace tree
technique with SQRT CSLA at the last stage. Figure 1. Block diagram for Proposed Multiplier
Efficiency of Parallel multipliers depends on type
booth recoding and reduction of partial products. This The Booth Encoder performs Radix-8 encoding of
paper describes a multiplier which uses Booth multiplier bits. Based on the multiplicand and the
recoding and Wallace tree compressors. The Proposed encoded multiplier, partial products are generated.
Multiplier uses radix 8 Booth algorithm and 4:2 The partial products are given to Wallace tree and
Compressor with SQRT CSLA at the last stage and is added accordingly. At the last stage, the results are
compared with a multiplier which uses CLA at the last added using a SQRT CSLA with BEC to get final
stage. The number of partial products is reduced to n/3 product.
in Radix8 algorithm. The Wallace tree uses
197
III. BOOTH ENCODER 1111 0
Recoding of binary numbers was first hinted by COMPRESSOR FOR PARTIAL PRODUCT REDUCTION
Booth four decades ago. MacSorley proposed a The Wallace tree method is used in high speed
modification of Booths algorithm a decade after. The designs in order to produce two rows of partial
modified Booths algorithm (radix-4 recoding) starts products that can be added in the last stage. The
by appending a zero to the right of X0 (multiplier critical path and number of adders reduced when
LSB). Triplets are taken beginning at position of X0 compared to the conventional parallel adders. Using
and continuing to the MSB with one bit overlapping the carry save adder, three partial product terms can
between adjacent triplets. If the number of bits in X be added at a time to form the carry and sum. The
(excluding X0) is odd, the sign (MSB) is extended one Sum signal is used by the full adder of next level. The
position to ensure that the last triplet contains 3 bits. carry signal is used by the adder involved in the
In every step we will get a signed digit that will generation of the next output bit, with a resulting an
multiply the multiplicand to generate a partial product overall delay proportional to logarithm of rows.
entering the Wallace reduction tree. The meaning of A multiplier consists of various stages of full
each triplet can be seen in Table 1. adders, each higher stage adds up to total delay of the
Table 1 Radix-4 Recoding system. In the first and second stage of Wallace tree
Triplet Value Signed digit value structure, the partial products do not depend upon any
000 0 other values other than the inputs obtained from the
AND array. However for the immediate higher stages,
001 +1 the final value depends upon the carry out of previous
010 +1 stage. This operation is repeated for further stages.
011 +2 Hence the major cause for the delay is propagation of
100 -2 the carry from previous stage to next stage.
101 -1
110 -1 In the proposed structure compressors are used in
place of full adders and final carry propagate stage is
111 0
replaced by a SQRT CSLA with a BEC circuit. The
first stage consists of a full adder. In the second stage
This recoding scheme applied to a parallel two full adders have been grouped and implemented
multiplier halves the number of partial products so the using a 4:2 compressor. The number of
multiplication time and the hardware requirements interconnections is taken care since they playa
decrease. This gain is possible at the expense of important role in the flow of carry from one stage to
somewhat more complex operations in every step. next stage in the tree. A 4: 2 compressor is shown in
Radix-8 recoding applies the same algorithm as the figure2.
radix-4, but now we take quartets of bits instead of
triplets. Each quartet is codified as a signed-digit
using the table 2.
Table 2 Radix-8 Recoding
Quartet Value Signed digit value
0000 0
0001 +1
0010 +1
0011 +2
0100 +2
0101 +3
0110 +3
0111 +4
1000 -4
1001 -3 Figure 2. 4:2 Compressor
1010 -3
1011 -2
1100 -2 In this compressor the outputs generated at each
stage are efficiently used by replacing the XOR blocks
1101 -1 with multiplexer blocks. The select bits to the
1110 -1 multiplexers are available much ahead of the inputs so
that the critical delay path is minimized.
198
Table 3 Function Table of the 4-bit BEC
IV. SQRT CSLA WITH BEC

The Final results obtained at the output of the
Wallace tree are added using SQRT CSLA with BEC
which reduces the power and area at the last stage.
This adder is similar to SQRT CSLA but one of the
BEC block which reduces area and there by power.
Fig 5 shows 16 bit SQRT CSLA with BEC. Here the
BEC is performed using simple XOR and AND gates
which reduces power and area.
Fig 3 illustrates the 4 bit BEC and it can be
extended to any bit. Table 3 indicates the function
table of 4bit BEC. Fig 4 illustrates the basic function
of the CSLA obtained by using BEC together with
MUX. One input of the 8*4 Mux gets as it input
(B3,B2, B1 and B0) and other input of the mux is the
BEC output. This produces the two possible partial
results in parallel and the mux is used to select either
the BEC output or the direct inputs according to the Figure 5. 16-bit SQRT CSLA
control signal carry in(Cin).
The Boolean expressions of the 4 bit BEC is listed as V. RESULTS
X0=NOT (B0) The Design entry of 16X16 bit multiplier using

X1=B0 XOR B1 radix-8 with 4:2 compressor and proposed architecture
X2=B2 XOR (B0 AND B1) are done using Verilog HDl and simulated using
X3=B3 XOR ((B0 AND B1) AND B2) Xilinx ISE. Then both designs are synthesized using
Xilinx 10.1 synthesis tool and implemented on Virtex
5 XC5VLX50T FPGA Board. Fig 6 shows the
simulation waveform of proposed multiplier.
Table 4 shows the implementation results report of

Proposed Multiplier. Table 5 shows the comparison of
both multipliers by considering the Summary of
synthesis report generated by Xilinx synthesis tool.
Figure 3. 4-bit BEC
Figure 6. Simulation Waveform of Proposed Multiplier
Figure 4. 4-bit BEC with 8:4 Mux
199
Table 4 Summary Report of Proposed Multiplier REFERENCES

Table 5
Slice Logic Utilization Used Available Utilization [1] S.F hsiao, M R jiang and J S yeh,Design of High speed low
power 3:2 counter and 4:2 compressor for fast
Number of Slice multipliers,Electron.lett,vol.34, no.4,Pp 341-343
Registers 520 28,800 1% [2] K.prasad and K.K parhi,Low Power 4:2 and 5:2
compressor, in proc of 35th asilomar Conf on
Signals,systems and computers, Vol1,2001,pp129-133
Number of Slice LUTs 739 28,800 2% [3] Lakshmanan, masuri Othaman and Mohammad Alauddin
Mohd,Ali, High Performance parallel multiplier using
Number of fully used Wallace-Booth Algorithm, Semi conductor Elecronics,
LUT-FF pairs 394 753 52% IEEE International Conference, Pp:433-436, Dec 2002.
[4] C S wallace,A suggestion for a fast multiplier, Electronic
Number of bonded computers, IEEE Transactions, vol13,pages 14-17,Feb 1964
IOBs 32 480 6% [5] T.G Noll,Carry save architectures for high speed digital
signal processing, Journal of VLSI Signal Processing, 1991,
Pp 121-140.
Table 6 Comparision of Multipliers [6] O.J Bedri, Carry Select Adder,IRE
Radix -8 Booth Trans.Electron.Comput..Pp340-344,1962
Radix -8 Booth
Logic Utilization Algorithm with [7] Steve Kilts, Advanced FPGA Design :
Algorithm with
( XC3S5000 fg1156 - 4:2 compressors
4:2 compressors Architecture,Implementation, and optimization, Wiley-IEEE
4) and CSLA with
and CLA
BEC
press, 2007.
Number of Slice
LUTs 855 739
Number of Slice
Registers 980 520
Number of bonded
IOBs 44 32
Total equivalent gate

count 2718 2054
JTAG Gate Count for

IOBs 630 559
Maximum
combinational path 12.35 13.65
delay (ns)
Total Power (uW) 123.5 112.4
VII. CONCLUSION
In this paper implementation and analysis of low
power area efficient Radix 8 multiplier with 4:2
compressor and SQRT CSLA using BEC is proposed.
The Proposed multiplier is compared with Radix 8
booth multiplier with 4:2 compressor and CLA. Both
designs are implemented on Virtex 5XC5VLX50T
FPGA board. The proposed multiplier shows more
reduction in power and area as compared to other
multiplier. The proposed design is to be used to design
a FIR filter and is to be implemented on a FPGA. In
the next stage physical design of FIR filter is realized
using Cadence Encounter Tool towards optimizing
area, power and delay.
200
Implementation of stereo vision system using

Blackfin Processor
N.Prasad1 M.Rajakumari2 P.Sitaramanjaneyulu3 T.Venkata rao4
Department of Electronics and Communication Engineering
Vignans Nirula Institute of Technology & Science for Women, Guntur, India
1
prasad.niz@gmail.com, 2josh.sri.raji@gmail.com, 3sriram128@gmail.com,4venkatarao.srp@gmail.com
Abstract Stereo vision is required in a field of 3D which determines the range of depth that can be
applications, like robotic navigation. This paper describes measured (the horopter). A difference between
a real-time stereo vision system that is required to support locations of the same object on the two images (the
high-level object based tasks in a tele-operated shift) is called disparity. The depth information (Z axis)
environment. Stereo vision is computationally expensive,
is derived from the disparity, found between two
due to having to find corresponding pixels. Correlation is
a fast, standard way to solve the correspondence problem. images as shown in Fig1.
This paper analyses the behaviour of correlation based
stereo to find ways to improve its quality while
maintaining its real-time suitability. The paper describes
a correlation based stereo matching algorithm and a DSP
based implementation. Results are given on real stereo
images. A comparison is made in terms of times they will
take by taking two different stereo image pairs with
windows of different dimensions by using black fin
processor.
Keywords: Stereo Matching, Computer Vision, 3D Scene
Reconstruction, Image Processing, DSP, Real Time, Robotic
Navigation.
I. INTRODUCTION
Vision is nothing but how we experience the world.

Computer vision is the science and technology of
machines that see. As a scientific discipline, computer
vision is concerned with the theory for building
artificial systems that obtain information from images.
The image data can take many forms, such as a video II. ALGORITHM
sequence, views from multiple cameras. The goal of
Computer vision is to process images acquired with
cameras in order to produce a representation of objects There is a wide range of stereo matching algorithms,
in the world. A stereo vision is a capability to sense the based on different approaches. The iterative methods for
depth of the observed scene. The human vision is example, give excellent quality, but are computationally
capable to see the depth and distinguish distances to the expansive. Such methods are not suitable for real time
different objects, because of the fact that humans have applications. The algorithm that used to find the
two eyes. The human eyes are located on the same disparity in our system is based on correlation[4]and is
horizontal line, and the brain gets almost the same shown in Fig.2.The correlation method is a fast way
picture from both eyes, but shifted horizontally one (compared to iterative methods) to perform the disparity
calculations. The method gives good quality, while
relative to other. Each object in the scene is shifted
retaining its real time performance. In order to make the
differently, depending on the physical distance to the
matching process to be effective, both input images
object. The brain extracts the depth information from must be preprocessed before the matching process.
those different shifts between different objects. A
computer stereo vision is based on two (or more) video At first stage, both images are resampled, to decrease
cameras, located on the same horizontal line. The the total number of pixels that will be processed later. At
distance between two viewpoints is called the baseline . a second stage, histogram equalization is applied to both
The baselines length affects on the range resolution, images to compensate for the lightness difference (DC
201
offset) between two cameras. At the third stage

performed stereo correspondence process between
preprocessed images. A stereo correspondence Stereo Matching Algorithms
procedure is done twice once with left image as
reference, and once with right. When one of the images There are different methods for performing stereo
is used as a reference, the other one is searched for the matching[3]. They can roughly be classified in two
matches. At the next step, a validation filter is applied. categories
The purpose of the validation filter is to invalidate false Feature-based: These algorithms extract features of
matches, caused by occlusions and low texture areas. interest from the images, such as edge segments or
The validation filter performs left/right consistency contours, and match them in two or more views. These
check, i.e. ensures that for each pixel, the same disparity methods are fast because only a small subset of the
is found with left reference and with right reference.
image pixels are used, but may fail if the chosen
primitives cannot be reliably found in the images;
furthermore they usually only yield very sparse depth
maps.
Area-based: In these approaches, the system attempts to

correlate the grey levels of image patches in the views
being considered, assuming that they present some
similarity. The resulting depth map can then be
interpolated. The underlying assumption appears to be a
valid one for relatively textured areas; however, it may
prove wrong at occlusion boundaries and within
featureless regions.
Corelation-based: This is also called Intensity based

stereo matching .Intensity-based stereo matching,
commonly known as the window-based method, is to
only match those regions in the images that are
``interesting'', for instance, regions that contain high
variation of intensity values in the horizontal, vertical,
and diagonal directions.. After the interesting regions
are detected, a simple correlation scheme is applied in
the matching process; a match is assigned to regions
that are highly correlated in the two images.
All these techniques have their strengths and
weaknesses and it is difficult to compare their merits
since few researchers work on similar data sets. It
appears that the correlation-based system has produced
the best results both in terms of precision and
reliability. This system achieves precisions in the order
of half a pixel in disparity, but, unfortunately, only
matches a very small proportion, typically less than 1%,
of the image points. The advantage of this intensity
Most of invalidated pixels will appear on occluded
profile matching is that a dense disparity map and,
areas. Since left image is used as a main reference
(right image is used as a reference only for validation consequently a dense depth (or range) map, is output.
filter), occlusions may appear only on the left side of Unfortunately, like all constrained optimization
the front objects. Because of this, all invalidated pixels problems, whether the system would converge to the
are classified as their left neighbors, keeping clear global minima is still an open problem.
edges between objects. Finally, a median filter is
applied, to filter out inconsistent matches, caused by a III.MATCHING PROCESS
low texture. An output image formation is a procedure
The matching process[2] is explained for the first case
that scales the image to the valid range, and upsamples
left frame is a reference image, while the right frame is
it to the original input resolution.
202
compared to the reference. The following pseudo code window does not overlap the depth discontinuity to the
describes the matching (stereo correspondence) process. same extent. Generally, the choice of the correlation
The support region is relatively small window defined window size is a trade off between increasing reliability
around the correlated pixel on the left (reference) in areas with constant depth and decreasing errors in
image. In this manner the correlation process takes into areas where depth changes.
account the neighbors of the pixel, not only the pixel
itself. The search window, defined on the right image, Optimization
is relatively large window. Its horizontal size is The correlation is performed using the SAD (Sum of
determined by the baseline between two cameras Absolute Differences) approach. The approach may be
(which affects the horopter). The vertical size may be 1 expressed mathematically by the following equation:
pixel for ideal stereo camera, or bigger for the practical
case. For each pixel inside the search window, a
correlation window (around it) is defined. The size of
the correlation window must be the same as the support
region size on the left (reference) image. Than, each M and N are corresponding to the vertical and
correlation window (inside the search window) is horizontal sizes of the correlation windows. The
correlated with the support region. The whole process measure of similarity is inversed, i.e. smaller SAD
must be performed for each pixel on the left (reference) value means better similarity. When support region is
image. two small, the correlation becomes ineffective, and the
quantity of mismatch errors increases. When using a
large support region, a lot of processing must be done
to accomplish the correlation task. To optimize the
correlation process, proposed the following
For each pixel in the left frame do
improvement.
begin
Set support region around the pixel (left frame) Data Reuse algorithm
Set search window in right frame
For each pixel in search window (right frame) When processing next pixel on the same line on the
begin reference image, a new support region (on the left
Set correlation window around the pixel image) and a new search window (on the right image)
Correlate support region with correlation window must be set. Since both windows are moving
end synchronously one pixel to the same direction
Find the best match (compared to previous point), only one
Calculate disparity single column of data has been changing inside them.
end All other data is still there, but moved to different
locations inside the windows. This fact is used to
Another problem with the correlation matching is that perform a fast correlation of the next search window, as
the selection of window size. Correlation works by only single column must be processed and added to the
using a usually fixed, rectangular window around the previous data. In this manner only for the first left pixel
pixel of interest in the first image. The window is on the left reference image the search window must be
correlated with a second window, which is moved over processed fully, while the rest of the pixels on the same
all possible positions in the second image. The possible line may be processed very fast using Data Reuse
positions are defined by the minimal allowed distance algorithm.
between the camera and an object, which gives the
Matching constraints
maximum disparity. The position where correlation has
the highest value determines the pixel in the second
image that corresponds to the pixel of interest. Bigger Stereo matching process is a very difficult search
correlation windows increase the reliability by procedure. In order to minimum false matches, some
averaging over a bigger area, thus reducing the effect of matching constraints must be imposed. Below is a list
noise. Consequently, this part of the window of the commonly used constraints.
introduces an error in the calculation. Furthermore, it
should be noted that the left part of the correlation Similarity: For the intensity-based approach, the
window in the left image is at least partly occluded in matching pixels must have similar intensity values (i.e.
the right image. The size of the occluded part depends differ lower than a specified threshold) or the matching
on the disparity difference and the size of the windows must be highly correlated. For the feature-
correlation window. The use of a smaller correlation based approach, the matching features must have
window reduces the problem, because a smaller similar attribute values.
203
memory to hold both images, one of them must be

Uniqueness: Almost always, a given pixel or feature located on the external SDRAM memory. The problem
from one image can match no more than one pixel or is that the access time to the external memory is
feature from the other image. The uniqueness constraint relatively long, and takes a few clock cycles to
may not be applicable to the line segment-based complete, while the internal memory works on a single
approach. This constraint can also fail if transparent clock cycle and may be effectively pipelined. In order
objects are present in the scene. Furthermore, given a to solve this issue, the following approach is used. The
pixel or feature m in one image, its ``corresponding'' search image is located in the internal memory space.
pixel or feature may be occluded in the other image. In For the reference image is allocated a relatively small
amount of internal memory, while the image itself is
this case, no match should be assigned to m.
located in external SDRAM memory. Before
Continuity: The cohesiveness of matters suggests that
processing, each block of image data is loaded from
the disparity of the matches should vary smoothly
SDRAM to internal memory. In such a way, the access
almost everywhere over the image. This constraint fails to each pixel data at the external memory is done only
at discontinuities of depth, for depth discontinuities during the loading, while during the matching process
cause an abrupt change in disparity. each pixel is accessed (frequently) at internal memory.
Open problems
V.RESULTS
The complexity of the correspondence problem depends
on the complexity of the scene. As described in the The following pictures show the experimental results
previous sections, there are constraints and schemes on Tsukuba stereo pair is shown in Fig.3
that can help reduce the number of false matches, but
many unsolved problems still exist in stereo matching.
Some of these problems are:
Occlusion: Given a feature m in the first image
projected by a scene point M, the ``corresponding''
feature of m may not exist in the second image because
from the second camera's viewpoint, M may be
occluded.
Discontinuity: This is related to occlusion. Most
matching algorithms use the continuity and/or
relaxation as a global constraint to reduce false
matches. This constraint works everywhere in the
images except at depth discontinuities.
Regularity and repetitively: Repetitive patterns in the
scene cause ambiguity in matching. Most existing
matching algorithms are still not robust enough to
handle this.
IV. IMPLEMENTATION
The algorithm was developed using MATLAB. First,

the synthetic stereo pairs were used as an input to the
algorithm. Secondly, a real world images captured from
two video cameras (stereo pair) were used as input. At
the next stage, after that was proved that the algorithm comparision:
as capable to find the disparity images on both synthetic
and a real world images, it has implemented on
hardware. The implementation is based on Analog Window size 384X288 640X480
Devices Black fin DSP, and is implemented by using
BF535 EZ-KIT LITE evaluation board. 3X3 1.12 sec 5.16 sec
Hardware optimization 7X7 2.05 sec 9.79 sec
The most computationally expansive part of the
algorithm is a correlation process. During the process The algorithm was able to complete processing on any
the core has to access to the preprocessed images data. stereo image pair. The performance result in terms of
Since the Black fin DSP doesnt have enough internal time for different window size is given in Table.A
204
VI.CONCLUSIONS
The above results show that the system can distinguish
between different objects in the observed scene and can
determine the relative distances to the objects. The
hardware implementation enables fast processing times
and acceptable quality, so the system may be used for a
real time applications, like robotic navigation.
REFERENCES
[1] Stereo Geometry, Kurt Konolige, SRI International, August,

1999
www.ai.sri.com/~konolige/svs/disparity.htm
[2] [Middlebury College, Stereo Vision Research Page
www.middlebury.edu/stereo
[3] D. Scharstein and R. Szeliski, A Taxonomy and Evaluation of
Dense Two-Frame Stereo Correspondence Algorithms, Intl J.
Computer Vision
www.breadnet.middlebury.edu/stereo/taxonomy- IJCV.pdf
[4] Real-Time Correlation-Based Stereo Vision with Reduced
Border Errors, Heiko Hirschmuller, Peter R. Innocent and Jon
Garibaldi, Centre for Computational Intelligence
www.cs.nott.ac.uk/~jmg/papers/ijcv-02.pdf
205
Implementation Of Energy Efficient And Low

Complex Fir Filters With Re-Configurability
Usage
R. Vijayalaksmi, P. Radhika Ramya, Y Sudha Vani
Vignan University,Guntur,
vijji.ravuri451@gmail.com, radhikaramya@gmail.com
compliments.Its use is so wide today because it does

ABSTRACT: not require the addition and subtraction circuitry to
Power decreasing, area optimizations are main examine the signs of the operands to determine
requirements of F.I.R filters employed in multiple whether to add or subtract. Twos compliment and
wireless communication systems. F.I.R. filter is used to ones complimentrepresentations are commonly used
find the response of finite (periodic) system under any since arithmetic units are simpler to design.Fig 1 shows
conditions. In this project, implementation of FIR filter Twos compliment and ones
is implemented; which possess less power consumption, complimentrepresentations.
low area occupancy by using Baugh-wooley algorithm.
FIR filter involves multiplications, additions, shifting Designing of FIR Filters: Most FIR filter design
operations, along with storing of yielded outputs.These methods are based on ideal filter approximation. The
methods include low power Baugh-wooley multiplier resulting filter approximates the ideal characteristic as
and parallel adder. So, dynamically reconfigurable the filter order increases, thus making the filter and its
filterscanbe efficiently implemented by using proposed implementation more complex. The filter design
algorithms for various D.S.P applications. process starts with specifications and requirements of
KEYWORDS: Baugh-wooley, Finite Impulse the desirable FIR filter. Which method is to be used in
Response, Reconfigurability, Linear-in-variant, Array the filter design process depends on the filter
multipliers, Periodic system, Optimization. specifications and implementation. This chapter
discusses the FIR filter design method using window
INTRODUCTION: Strength reduction yields to a functions. Each of the given methods has its advantages
decrease in hardware complexity by exploiting and disadvantages. Thus, it is very important to
substructure allotment and implies to less area and carefully choose the right method for FIR filter design.
power consumption in a VLSI ASIC implementation or Due to its simplicity and efficiency, the window method
less iteration periodin a programmable DSP is most commonly used method for designing filters.
implementation. Filters are signal conditioners. Each The sampling frequency method is easy to use, but
functions by accepting an input signal, blocking filters designed this way have small attenuation in the
prespecified frequency components, and passing the stopband. As we have mentioned above, the design
original signal minus those components to the output. process starts with the specification of desirable FIR
For example, a typical phone line acts as a filter that filter. The Lth-order LTI FIR filter is graphically
limits frequencies to a range considerably smaller than interpreted in Fig.1. It can be seen to consist of a
the range of frequencies human beings can hear. That's collection of a tapped delay line, adders, and
why listening to CD-quality music over the phone is not multipliers. One of the operands presented to each
as pleasing to the ear as listening to it directly. multiplier is an FIR coefficient, often referred to as a
Multiplication involves 2 basic operations: the tap weight for obvious reasons. Historically, the FIR
generation of the partial product and their accumulation filter is also known by the name transversal filter,
[5].Therefore, there are Possible ways to speed up the suggesting its tapped delay line structure. An FIR with
multiplication: reduces the complexity, and as a result constant coefficients is an LTI digital filter. The output
reduces the time needed to accumulate the partial of an FIR of order or length L, to an input timeseries
products .Both solutions can be applied simultaneously. x[n], is given by a finite version of the convolution sum
Baugh-WooleyTwos Compliment Signed Multiplier given in (1), namely:
:TwosCompliments is the most popular method in
L-1
representing signed integers in Computer sciences.It is
also an operation of negation(Converting positive to y[n]=x[n]*f[n]=f[k]x[n-k], (2)
negative numbers or vice versa) in computers which k=0
represent negative numbers using twos
206
They also correspond to the FIRs impulseresponse. For have three rows of full adders to add width to the
LTI systems it is sometimes more convenient to express layout. This resulted in a more rectangular and flat
in the z-domain with Y(z)=F(z)X(z) circuit, but allowed us to reduce drastically the distance
the signal had to travel between chips, and decrease the
where F(z) is the FIRs transfer function defined in the overall complexity of connections within the circuit.
zdomain by Dissecting the Timing:
When we take a closer look at the timing contribution
of
i).the partial-product generation,
ii) the reduction tree, and
iii) the final adder to the total delay for MB and BW
multipliers, we see that the delay that is gained by
having a smaller reduction tree for the MB
implementation is canceled by the large delay of the
MB recoding circuitry. Table I shows that the partial-
Folded Bit-Plane Architecture product generation of the MB multiplier is about two
Using a new assignment of folding sets, which is times as slow as the generation in the BW multiplier.
applied on the transformed DFG from [4], we obtain This comes as no surprise, since for BW there is only
folded Bit-Plane architecture in general form (Fig. 1). one 2-input AND -gate in the critical path, while for the
Input Data Entering Module (IDEM), denoted with MB multiplier the critical path goes through the
dashed lines in Fig. 1, provides input data for the folded encoder followed by the decoder. Taking gate fanout
architecture in accordance with the allocation table into consideration, the primary inputs of a BW
(Table 1). Sections S0, S1, , Sk-1 in Fig. 1. are multiplier need to drive only N minimum-sized 2-input
Processing Elements (PE) of the folded architecture. A ND gates, while for the MB multiplier the x primary
Each section is devoted to computations from the inputs first have to driveN / 2 encoders, which in turn
corresponding folding set. Sections are implemented as drive N decoders. The delay difference of the reduction
rows of basic cells, where the basic cell is comprised of tree is not as large as above.
AND gate and full adder.
Baugh-wooley multiplier: The Baugh Wooley RESULT:
multiplier is an interesting implementation of a
multiplier. It produces a relatively standard cell shape SIMULATION:
for easier manufacturing, while maintaining good
driving caharacteristics. Each full adder within the
multiplier performs a similar number of computations,
with the diagonal ripple carry propagating the input
signal a constant and similar number of times. Baugh-
Wooley Multiplier is used for both unsigned and signed
number multiplication. Signed Number operands which
are represented in 2s complemented form. Partial
Products are adjusted such th at negative sign move to
last step, which in turn maximize the regularity of the
multiplication array. Baugh-Wooley Multiplier operates
on signed operands with 2s complement representation
to make sure that the signs of all partial products are
positive. An issue encountered during the design and
layout of the multiplier involved the layout of the
required full adders. A single full adder circuit SYNTHESIS: RTL SCHEMATIC:
naturally lays out in a very wide (or tall) chip, which
creates problems when working toward smallest form
factor and efficiency of cost. We attempted a number of
creative solutions to work around this, including
stacking these chips to create a more square form. This,
however, produced long connections as the signal
wrapped around from the output of one full adder to the
input of another and introduced both losses and
capacitance. It was determined that a better solution
was to implement the multiplier in a single, linear
direction through four stages of full adders, but only
207
ADVANTAGES: [14] E. Borch, E. Tune, S. Manne, and J. Emer, Loose loops sink
chips, in Proc. of HPCA-8, February 2002, pp. 270281.
[15] J. A. Butts and G. S. Sohi, Use-based register caching with
Power optimizing FIR filters consumes less power and decoupled indexing, in Proceedings of the 31st annual international
occupies less area. High throughput FIR filters used symposiumon Computer architecture, 2004, pp. 302313.
for all industrila applications.Mulyi hopping is main [16] R. E. Kessler, The alpha 21264 microprocessor, IEEE Micro,
cretireia in now a days. So, this designed FIR filter vol. 19, no. 2, pp. 2436, March-April 1999.
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algorithms in this project are more effective and
efficient when compare with all other technologies. All
proposed algorithms in this project are supports all type
of applications and yields high throughputs with more
security.
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208
Handwritten Digit Recognition with MLP

Neural Network by Using Back Propagation
Training with
Tunneling Algorithm
PPS SUBHASHINI1,
Associate Prof ,ECE Dept.,
RVR & JC College of Engg., Guntur, k_shiva_111@yahoo.co.in
Dr. M.SATYA SAIRAM 2,

Prof & Head,Department of ECE,
Chalapathi Institute of
Engg.and Technology, Guntur, msatyasairam1981@gmail.com
Dr. D SRINIVASARAO 3
Prof & Head,
Department of ECE,
J N T U H, HYDERABAD, dsraoece@gmail.com
Abstract - Character Recognition has been an connectivity. The security systems, one different
active area of research in the past and due to its approach of dealing authentication is to use
diverse applications it continues to be a BIOMETRICS - the science of identifying or
challenging research topic. Methods based on verifying the identity of a person based on
multi layer perceptron (MLP) neural network physiological or behavioural characteristics.
provide attractive possibilities for solving pattern Physiological characteristics include fingerprints, iris,
classification problems. Several algorithms have hand geometry and facial image. The behavioural
been proposed for training the multi layer characteristics are actions carried out by a person in a
perceptron (MLP) neural network. We explore characteristic way and include recognition of
these techniques to design an optimal handwritten signature, machine printed characters, handwriting,
number recognition system. The main approach and voice [1].The character recognition is required
for handwritten number recognition is based on for different applications like postal address system,
segmentation. After segmentation the problem signature verification system and recognition of
gets reduced to the recognition of simple isolated characters from filled applications [2].
digits. We here adopt segmentation based
handwritten digit recognition where neural Character recognition is classified into two
networks are used to identify individual digits. In categories, off-line character recognition and on-line
this paper multi layer perceptron (MLP) neural character recognition. In off-line character
network using back propagation training with recognition the system accepts image as input from
tunnelling algorithm is proposed for handwritten the scanner [3]. It is more difficult than on-line
digit recognition application. The efficiency of the character recognition because of unavailability of
proposed method is tested on the handwritten contextual information and lack of prior knowledge
digits of different fonts and found that it is of the character. Noise will also exist in the images
successful in recognizing the digits. acquired in off-line character recognition. Machine
Printed character recognition comes under this
Index Terms - ANN, MLP network, Back category.
Propagation Training with tunneling , Zoning method On-line character recognition system accepts the
moment of pen from the hardware such as graphic
1. INTRODUCTION tablet, light pen and lot of information is available
during the input process such as current position,
The digital revolution has drastically changed our moments direction, start points, stop points and
perspective of the concept of communication and stroke orders [4-5].
209
For digitizing the handwritten document first the causal chain. With unsupervised learning it is
source material must be scanned using a scanner. possible to learn larger and more complex models
Older character recognition systems match these than with supervised learning. This is because in
images against stored bitmaps based on specific supervised learning one is trying to find the
fonts. The hit-or-miss results of such pattern- connection between two sets of observations.
recognition systems evaluate the performance of the Backpropagation method is the basis for training a
system accuracy. The current character recognition supervised neural network. The output is a real value
systems add the multiple algorithms of neural which lies between 0 and 1 based on the sigmoid
network technology. function [8].
This paper focuses on handwritten digits recognition 3. MLP NEURAL NETWORKS

using Multilayer Perceptron (MLP) Neural Network
based on back propagation training with tunneling Multi Layer Perceptron (MLP) is a type of artificial
algorithm. From the results it was observed that the network for applications to problems of supervised
proposed method has very high success rate in learning such as pattern classification. A feed-
recognising handwritten digits of different fonts. forward network has a layered structure. Each layer
consists of units which receive their input from units
2. ARTIFICIAL NEURAL NETWORKS from a layer directly below and send their output to
units in a layer directly above the unit. There are no
Neural network or Artificial Neural Network (ANN) connections within a layer. The inputs are fed into the
is a massively parallel distributed processor made up first layer of hidden units. No processing takes place
of simple processing units, which has a natural in input units. The training output values are vectors
propensity for storing experiential knowledge and of length equal to the number of classes. After
making it available for use. A neural network training, the network responds to a new pattern.
contains a large number of simple neuron like
processing elements and a large number of weighted This network consists of three layers, one input
connections encode the knowledge of a network. layer, number of hidden layers and one output layer
Though biologically inspired, many of the neural as shown in Fig. 1. The activation of a hidden units of
network models developed do not duplicate the layer of Nh,1 is a function f of the weighted inputs
operation of the human brain. [6] plus a bias .The output of the hidden units is
distributed over the next layer of Nh,2 hidden units,
The intelligence of a neural network emerges from until the last layer of hidden units, of which the
the collective behavior of neurons. Each neuron outputs are fed into a layer of N0 output units [9] .
performs only very limited operation. Even though
each individual neuron works slowly, they can still The output of each hidden neuron is then weighted
quickly find a solution by working in parallel. This and passed to the output layer. The outputs of the
fact can explain why humans can recognize a network consist of sums of the weighted hidden layer
visual scene faster than a digital computer Training neurons. The design of an MLP requires several
the Neural Networks can be accomplished in two decisions that include the number of hidden units in
ways. the hidden layer, values of the prototypes, the
functions used at the hidden units and the weights
supervised learning method applied between the hidden layer and the output layer
[10].
unsupervised learning method
The performance of an MLP network depends on the
In supervised learning, the model defines the effect number of inputs, the shape of the sigmoid function
one set of observations, called inputs, has on another at the hidden units and the method used for
set of observations, called outputs. In other words, the determining the network weights. MLP networks are
inputs are assumed to be at the beginning and outputs trained by selecting the weights randomly from the
at the end of the causal chain. The models can include training data.
mediating variables between the inputs and outputs
[7]. A. Working Principle
In unsupervised learning, all the observations are Considering the MLP Neural Network with one
assumed to be caused by latent variables, i.e. the hidden layer as shown in Fig.2 consisting number of
observations are assumed to be at the end of the input nodes I, the number of hidden nodes J and the
210
number of output nodes K . The I-dimensional input z When the MLP Neural network is used in
is passed directly to a hidden layer. Suppose there are classification, the hidden layer performs clustering
J neurons in the hidden layer, each of the J neurons in while the output layer performs classification. The
the hidden layer applies unipolar sigmoid activation hidden units apply Non linear sigmoid activation
function defined by using (1) function for the input patterns. The output layer
would linearly combine all the outputs of the hidden
( )= ( . )
layer. Each output node would then give an output
value, which represents the probability that the input
pattern falls under that class. Back-propagation can
(1)
be
applied to networks with any number of layers. In
this paper a feed-forward neural network with a
single layer of hidden units is used with a sigmoid
activation function for the units.
Input Layer Hidden Layer Output Layer
Fig.2 Multi-Layer Network with single Hidden

Fig.1 Multi-Layer Network with L number of hidden Layer
layers
4. BACKPROPAGATION TRAINING
Where >0 and net of a node is the summation of its
weighted inputs. Where J is number of nodes in the The error surface of a complex network is full of hills
hidden layer. Each of the K neurons in the Output and valleys. Because of the gradient descent, the
layer applies Generalized Sigmoid activation function network can get trapped in a local minimum when
defined by using (2) there is a much deeper minimum nearby.
Probabilistic methods can help to avoid this trap, but
they tend to be slow. Another suggested possibility is
( )= ( ) (2) to increase the number of hidden units. Although this
will work because of the higher dimensionality of the
error space, and the chance to get trapped is smaller,
Where K is the number of nodes in the output layer. it appears that there is some upper limit of the
The generalised sigmoid function is the general case number of hidden units which, when exceeded, again
of the sigmoid function with the ability to specify the results in the system being trapped in local minima
steepness of the function as well as an offset that [12].
should be taken into consideration.The use of the
The Back Propagation with tunneling is used to train
generalized sigmoid function introduces additional
the Designed MLP Neural Network which replaces
flexibility into the MLP model. Since the response of
the gradient-descent rule for MLP learning can find
each output neuron is tempered by the responses of
the global minimum from arbitrary initial choice in
all the output neurons, the competition actually
the weight space. The algorithm consists of two
fosters cooperation among the output neurons [11].
phases. The first phase is a local search that
211
implements the BP learning. The second phase 3) Error signal vectors and of both layers are
implements dynamic tunneling in the weight space computed. Dimension of Vector is (K X 1)
avoiding the local trap and thereby generates the and Dimension of is (J X 1). The error
point of next descent. Alternately, repeating the two signal terms of output layer is given by using
phases forms a training procedure that leads to the
global minimum. This algorithm is computationally =( )(1 ) , (6)
efficient.
= 1,2, .
The first phase in the Back Propagation tunneling The error signal terms of hidden layer is given by
training algorithm is Back-Propagation training using (7)
Algorithm. A two layer feed-forward network did not
present a solution to the problem of how to adjust the = 1 ,
weights from input to hidden units The solution is
that the errors for the units of the hidden layer are
= 1,2,
determined by back-propagating the errors of the
(7)
units of the output layer. The method is often called
the back-propagation learning rule. Back-propagation
4) The output layer weights are adjusted as
can also be considered as a generalization of the delta
= + , (8)
rule for non-linear activation functions layer
networks [13]. =
The error measure E is defined as the total quadratic 1,2,
error for pattern p at the output units. When a = 1,2, .
learning pattern is clamped, the activation values are
propagated to the output units, and the actual network 5) The hidden layer weights are adjusted by using
output is compared with the desired output values, we (9)
usually end up with an error in each of the output = + ,
units [14]. = 1,2,
(9)
Let I be the Number of input nodes, J be of Number = 1,2, .
of hidden nodes and K be the Number of output 6) Repeat the steps 1) to 5) for all the feature
nodes for the MLP neural network. Consider V be the vectors
weight vector for hidden layer and W be the weight
vector for output layer. The size of W matrix is K X 7) The training cycle is repeated for 1000 epochs.
J. The size of V matrix is JXI
5. TUNNELING ALGORITHM
A. The steps for the training cycle The Second phase in the Back Propagation tunneling
training algorithm is dynamic tunneling. A random
1) By applying the feature vectors one by one to the point W in the weight space is selected. For a new
input layer the output of hidden layer is point + , where is a perturbation related to W,
computed as if ( + ) ( ), the BP is applied until a local
= = 1,2, minimum is found; otherwise, the tunneling
(3) technique takes place until a point at a lower basin is
found as shown in the Fig 3. The algorithm
The function f1(.) is unipolar Sigmoid Function automatically enters the two phases alternately and
defined by using (1). Output of output layer is weights are modified according to their respective
computed by using (4) update rules. The tunneling is implemented by
solving the deferential Equation by using (10)
= ( ) = 1,2,
(4) ()
() ( ) /
= ( )
The function f2(.) is Generalized Sigmoid Function (10)
defined by using (2)
( )
Where is the learning rate, is the local
2) The error value is computed by using (5) () () ( ) ()
minima of , = +
= ( ) + , = 1,2, . ,
(11)
(5)
212
The equation is integrated for a fixed amount of time, This sequence is repeated until no additional image
with a small time-step . After every , E(W) is is found
()
computed with the new value of keeping the A. Thinning
emaining components of W unchanged. Thinning is the process of transforming a digital
pattern from one form to another with less thickness
Tunneling comes to a halt when E(W) E(W*), and while maintaining connectivity of the original
initiates the next gradient descent. If this condition of pattern[17]. It is used in several applications , one of
descent is not satisfied, then this process is repeated which is character recognition in the field of
( ) document analysis. In this contest the scanned image
with all the components of until the above
is converted into a binary image (I). It has a finite set
condition of descent is reached. If the above of points in the Euclidean plane where each point is
()
condition is not satisfied for all , then the last mapped into 0 or 1. The set of pixels in I that are
local minimum is the global minimum [8-14]. assigned the value 1 is called the foreground or the
pattern and denoted by P where as the background
6. PRE PROCESSING pixels are denoted by Pc. A good thinning algorithem
is one that ensures some basic desirable properties
The hand written digits document was first such as: maintaining connectivity of resulting
preprocessed before applying to the MLP skeletons, insensitivity to noise. Parallel
network.Preprocessing of hand written documents implementation of thinning algorithems are
consists three major steps [15]. First the document significantly faster than sequential ones. After
image was segmented and then the segmented digit is thinning process, the features of the image are
applied to thining alogorithm. obtained by feature extraction method.
B. Feature Extraction Using Zoning Method
A. Segmentation To train the MLP network with any image requires its
features.In this paper to extract the features Zoning
Character segmentaion is an opreation that seeks to method is used . In zoing method, the scanned image
decompose an image of a sequence of character in to is divided into NxM zones. From each zone, features
sub image of individual symbols . it is one of the are extracted to form feature vector [16]. The goal of
decision processes in a system for optical character zoning is to obtain the local characteristics instead of
recognition .Its decision, that a pattern isolated from global characteristics. Each sub-zone consists number
the image is that of a character or some of foreground pixels. A feature is considered as an
identificatiable unit), can be right or wrong . it is average of foreground pixel density.
wrong sufficently often to make a major
The scanned image is divided into 4-by-4 zones asa
shown in the Fig. 4. In addition, in each sub zone, the
foreground pixel values are taken and by averaging
all those pixel values, we can obtain the feature of
that sub zone. Such that the features of remaining sub
zones are obtained. For each digits 16 features are
obtained which is placed as feature vector of 1 x 16.
Fig.3 Error surface
contribution to the error rate of the system .The

classical approach to ocr and segmentation is the
intial step in a three procedure Fig. 4 Dividing image into zones
Given a starting point in a document image:
Find the next hand written numerical image. B. EXPERIMENTAL RESULTS
Extract distinguishing attributes of that
image. The results obtained by using the designed MLP
Find the member of a given symbol set those neural network based on Back Propagation training
attributes best match those of the input , and with tunneling algorithm for recognition of
output is identity
213
handwritten digits for the numbers 0 to 9 of 25 fonts

as shown in Fig .5 are presented in this section.
Fig. 7 Block Diagram of Handwritten digit

recognition
The performance of training method is explored by

averaging its performance over five trials, where each
trial consists of a random selection of training and
test data. The number of hidden units in the MLP
network is varied between 1 and 50. The number of
input nodes are 16 and the number of output nodes
Fig .5 Handwritten digits of 25 different fonts
are 10 for the designed MLP neural network
The result after segmentation of the document image
for digit 4 for 25 fonts is shown in Fig .6
The MLP network is trained with 13 different
.
handwritten fonts of each digit and is also tested with
12 other handwritten fonts which are different from
the trained fonts.
The Graph in the Fig.8 Shows Percentage of correct

classification for the applied test samples by varying
number of hidden units. It is observed that this
method is successful in recognizing the digits of all
these 25 handwritten scripts with correct
classification rate of 88 %.
90
Fig .6 Segmented image for digit 4 80
70
Thinned image was obtained for each segmented
% of correct classification
60
hand written digit and 16 features are extracted using
Zoning method.The features so obtained is kmown as 50
feature vector.The input training data set is a martrix
40
of dimensions 250 x 16 obtained by the combination
of feature vectors of total 250 hand written digits. 30
The designed MLP neural network is trained by using 20

backpropagation training with tunneling algorithm by
10
using the input training data set and target vector of
size 250 x10. The different steps involved in 0
0 5 10 15 20 25 30 35 40 45 50
recognition of the handwritten digits are illustrated in No.of Hidden Nodes
Fig. 6. The training algorithm initialized the weight Fig. 8 Percentage of Correct Classification Versus
matrix W to zero. The no. of hidden nodes
214
The Graph in the Fig. 9 shows the Mean square Error is found that this method requires less time for
versus number of Iterations for training the MLP training the MLP neural network and has good
neural network. success rate in recognizing the handwritten digits.
It was also observed that Mean Square Error

gradually reduces as numbers of Iterations are
increased.
Fig. 9 Mean square Error Versus No. of Epochs

The features of these handwritten digits are extracted
It was observed that Mean Square Error by using (5) using Zoning method. Each digit can be represented
gradually reduces from the value 1, as numbers of as 16 features.
iterations are increased to and is terminated when the
error function decreases by less than 0.0166. These features are known as feature vector . The 1 to
It is observed that this method is successful in 8 features of each segmented digit of one font
recognizing the digits of all these 25 handwritten obtained by using zoning method are mentioned in
fonts. The features of these handwritten digits are Table 1. These feature vectors are used to train the
extracted using Zoning method and tabulated in Table MLP neural network and they are applied as inputs to
1 and Table 2 for single font out of 25 different hand the input layer.
written fonts. Each digit can be represented as 16
features. The combination of all these features is Table 2: 9 to 16 Features of Handwritten Digits 0 to 9 for
called a feature vector. The feature vector is applied one font
to the designed Neural Network. If the digit is
recognized, then the output node corresponding to the
digit is 1 and the remaining is 0. The training
method here applied is supervised training method.
This requires the target vector that is the desired
output, which is tabulated in Table 3. It was observed
that the Mean square error gradually decreases as
numbers of iterations are increased
C. CONCLUSION
The MLP Neural Network using Back Propagation

training with Tunnelling algorithm is proposed for
recognition of handwritten digits of different fonts.
The proposed method is tested on handwritten digits The features of these handwritten digits are extracted
of 0 ,1,2,3,4,5,6,7,8, 9 of 25 different fonts each. The using Zoning method. Each digit can be represented
handwritten document is pre-processed with as 16 features. These features are known as feature
segmentation ,thinning methods. Then the features vector .
are extracted for the pre-processed image by using
Zoning method. The 9 to 16 features of each segmented digit of one
font obtained by using zoning method are mentioned
These features are used to train the MLP Neural in Table 2. These feature vectors are used to train the
Network with 16 input nodes , variable hidden nodes MLP neural network and they are applied as inputs to
and 10 output nodes one for each digit from 0 to 9 . It the input layer.
215
Table 3: Target vector for hand written digits for 0 to [10] Tom Mitchell, Artificial Neural Networks, in Machine
Learning, 1st ed. McGraw- Hill ,1997 pp.95-111
9
[11] O.Batsaikhan and Y.P. Singh, Mongolian Character
Recognition using Multilayer Perceptron, Proce edings of the
9th International
[12] Kambiz rahbar, Saeid rahati quchani, Muhammad rahbar
recognition persian handwritten digits using templates and
back-propagation network with adaptive learnings rate
proceedings of the 5th international conference on signal
processing, istanbul, turkey, may 27-29, 2006 (pp46-51)
[13] D.Y. Lee, Handwritten digit recognition using K nearest
neighbour, radial basis function and backpropogation neural
networks, Neural Computation, Vol. 3, pp. 440-449.
[14] A.M numan-al-mobin, mobarakol islam, kaustubhdhar, tajul
Islam, md. rezwan,m. Hossain backpropagation with vector
chaotic learning rate
[15] Jaekyu Ha & Robert M. Haralick Ihsin, T. Phillips,
For supervised learning at each instant of time when Document Page Decomposition by the Bounding-Box
the input is applied, the desired response of the Projection Technique, IEEE Transactions on Systems, Man,
network known as target vector is provided by the and Cybernetics, Vol.18, No.1, January 1995, pp.1118-1122.
[16] Oivind Due Trier, Anil K.Jain and Torfinn Text, Feature
teacher. The distance between the actual and desired Extraction methods for Character recognitionA Survey,
response serves as an error measure and is used to Pattern Recognition, Vol. 29, No. 4, 1996, pp.641662.
modify weights so that the error decreases. The [17] P. S. P. Wang and Y. Y. Zhang, A fast and flexible thinning
target vectors for one font is mentioned for digits 0 to algorithm, IEEE Transactions on Computers, Vol.38, No.5,
May 1989.
9 in Table 3. By using these target vector the training
of MLP neural network will terminate. for the
designed MLP network there are 10 output nodes and
16 input nodes ,to which the feature vector of 1 x 16
is applied ,for example for digit 0 ,the output of the
first node is 1 and the output of the remaining nodes
are 0.The target vector is 1 x 10 vector.
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216
Signal Processing Schemes for Interferometric

Fiber Optic Gyroscope
Kalpana Arvind, D.Praveena, M.Gayatri, M.N.Srinivas, Lekshmi.S.Rajan, V.V.Lakshmipathy, A S Laxmi Prasad
Laboratory for Electro-Optics Systems, (LEOS)
st st
1 cross, 1 stage, Peenya Industrial Estate, Bangalore-560058 kalpana@leos.gov.in
Abstract: This paper describes t he various anal og which is given by n L/C where n refractive index ;
signal processing schemes used for detecting the L length of the fiber ; C velocity of light.
rate inform at ion from square wave bias modulated
Interferometric Fiber Optic Gyro(IFOG).
Development of Single axis fiber optics gyro carried
out in LEOS in open loop and closed loop schem es
are described here. The design consideration,
tradeoffs are highlighted for the front end
electronics o f an all - digital closed loop approach.
Achieved results are also presented .
1. INTRODUCTION
Over the last three decades IFOG has progressed

from a
laboratory experiment to inertial guidance and
navigation specialist. A FOG provides extremely
precise rotational rate information, in part because of
its lack of cross-axis sensitivity to vibration,
acceleration, and shock. The operation of FOG is
based on the so called Sagnac Effect, which states
that two light beams, propagating in opposite
directions in a rotating frame experience an optical
path length difference which is proportional to the
rotation rate of the optical path with respect to the
inertial frame. Unlike the mechanical gyros the FOG
has no moving parts and doesn't rely on inertial
resistance to movement, light weight and long
working lifetime. Because of their intrinsic reliability,
FOG is used for high performance space applications.
Inertial grade gyro s having bias stability of
0.01deg/hr to 0.001 deg/hr is required for
cartographic applications. This paper describes the This modulation and demodulation technique yields a
development of FOG activities carried out at LEOS sine response with a very stable bias. If the
for spacecraft applications. modulation frequency is equal to the eigen frequency
(f = 1/2) bias error induced by the modulator defects
A. Basic Principle of FOG such as spurious amplitude modulation and non-
linearity are avoided. Detector output, when the
The basic scheme of the IFOG is illustrated in fig-1.
system is at rest is equal and when is rotated, the
It is a passive interferometer where the fiber optic
operating point is shifted from p/2 bias; hence the
coupler splits the light from the light source into a
output of system will be a square wave as shown in
two counter propagating beams, clockwise and
fig- 2. If the system is rotated in clockwise direction
counter clockwise to
the positive peak will be more than negative and vice
the fiber coil through an integrated optics chip for
versa. Demodulating this signal will give a phase
bias modulation to get high sensitivity. Interfered
proportional to rate.
signal is detected using a photo detector. The
In Open loop mode, demodulating the detected signal
minimum response time of the IFOG is the transit
gives the rotation rate. In Closed loop mode the
time through the coil demodulated signal is used to generate a
217
nonreciprocal phase shift within the interferometer to open loop system, even though the SLED
counteract the Sagnac phase shift and thereby temperature was controlled to 25C + 0.5C using
maintaining the detected built-in thermoelectric cooler.
output to zero.
3. CLOSED LOOP SCHEME
Closed loop scheme with Integrated Optic approach

using Digital Serrodyne scheme was implemented.
Fig.2 Square wave Bias Modulation All digital logics are implemented in Field
Programmable Gate Array (FPGA) with different
The demodulation can be done in the analog or digital ANALOG signal processing versions are made to
domain. In analog domain, phase sensitive detection improve the performance of gyro.
method was normally used. Digital domain is
preferred A. Version-1
because of low noise and drift of signals. In this paper Single axis 1KM PMF fiber wound on 139 X
various front end processing scheme adopted to h40mm bobbin in quadrupolar configuration is the
derive the rate information is described below. sensing element. Source and detector are same as
open loop scheme. Multifunction Integrated Optic
2. OPEN LOOP SCHEME
Chip (MIOC) with a Vvoltage of 2.2V is used as
phase modulator and digital serrodyne ramp for
Block diagram of the first model developed at LEOS
feedback. Instead of Analog demodulation, digital
is shown in fig.3 Polarisation Maintaining Fiber
demodulation technique was used, which is shown in
(PMF) of
fig.4
ength 500m wound on 120xh20 mm bobbin with a
Super Luminescent Diode (SLED) having
wavelength of 1300nm & 300W source power and a
phase modulator for biasing was assembled in all
fiber approach. Detector power was 4W. Phase
sensitive scheme was mplemented using discrete
Fig.4 Integrated Optic Approach
Lock-in amplifier scheme. Commandable gain for
Coarse / Fine mode Analog voltage was converted to
By increasing the length L and diameter D, sagnac
digital format using sample and hold and 12-bit
scale factor is improved and resulted in better
ADC. Parallel data from ADC was converted to serial
performance than the open loop scheme and the
data using shift registers. 12 bits Rate output was
results are shown in table-2. FOG Sensor with
monitored.
electronics is shown in fig.5 and the stability plot is
shown in fig.6
Fig.3 Open loop scheme

Package has successfully passed all the
environmentaltests. Scale factor variation was
observed during the thermo vacuum (-10C to Fig.5 FOG Sensor
+50C) testing which affected the performance of
218
In this scheme, instantaneous peak to peak noise is 30 as shown in fig.10. PINFET detector signal is
deg/hr and thermal sensitivity is also high. Upon converted to
analysis heat dissipating elements like SLED & DC by integrators and sample & hold for positive and
Processing electronics are integrated in a single box. negative bias modulation time and the analog
This problem is solved in the next version by demodulation is done by differential amplifier.
isolating the heat dissipating elements from the Amplified rate information is filtered by 4th order
sensor head and compact mechanical design for 4axes Bessel filter and converted to digital signal by ADC.
FOG version. Achieved results are shown in table-3. ARW plot,
Stability plots are shown in fig.11 & 12 respectively.
Dynamic range and resolution of this FOG is
improved, but instantaneous noise peak to peak is 20
deg/hr and drift is 0.2 deg/hr. Though heat dissipating
elements inside the SHA are isolated bias drift in
thermo vacuum cycling ( 0 to 50C) is of the order of
8 deg/hr/C, which is shown in fig.13 along with
SHA temperature..
Fig.10 Analog Processing Block diagram
B.Version-2 Table-3
In Sensor Head Assembly (SHA) non dissipating Closed loop scheme version-2 results
components like fiber and MIOC alone assembled
and all the heat dissipating elements are shifted to
Processing Electronics Module (PEM) as shown in
fig.7. High power SLED source is set to 1mW and
the detector power is 100 W. Both the source and
detector are changed to get better signal to noise
ratio. Coupler is also replaced with Circulator (CIR)
to avoid the light reflecting back to SLED. Bobbin
size is reduced to 110xh20mm and the assembled
single axis sensor is shown in fig.8 and 4-Axes SHA
mounting on a tetrahedral assembly is shown in fig.9
Fig.11 Stability plot of version-2
Fig.7 Single axis Block diagram
Fig.12 Allan Variance Plot of version-2

Problem analysis & Signal to Noise Ratio (SNR)
In this scheme to provide the gain, analog processing calculation are carried out for the circuit.
scheme is adopted and the detailed block diagram is
219
Commercially available off the shelf PINFET 1) Detector selection: Pin diode junction capacitance
detectors are having very high bandwidth and output decides the detector bandwidth and hence the noise.
saturation voltage is 2V only. PIN diode to be chosen carefully and layout
guidelines to be strictly followed not to increase the
capacitance.
Fig.13 Thermo Vacuum plot of version-2

Though source power of SLEDs are available with
>20mW, PINFET output is saturated, since current to Fig.15 Block diagram of version-3
voltage gain is fixed. From the SNR calculation
relative 2) Bias Voltage: The reverse bias voltage of
intensity noise contribution is high compared to shot photodiode can go up to 20 V as per the data sheet.
and thermal noise of the detector and amplifier noise Application of a
of reverse bias can greatly improve the speed of
PINFET (85mV) is also high due to its high response and linearity of the devices. Also higher bias
bandwidth, voltage reduces the junction capacitance and
which is shown in fig.14. increases the bandwidth. Therefore reverse bias was
applied. Instead of transimpedance amplifier scheme,
simple load resistor scheme was selected, so that
reverse biasing can be increased to -10V as shown in
fig.14. Voltage induced due to photocurrent cannot be
greater than the bias voltage of a photodiode in the
case of a simple load
resistor VBias > V= PR, where P-power falling on
the detector : responsivity of the detector & RL
load resistance. This scheme allowed to increase the
incident
Fig.14 PINFET Output Noise Waveform optical power to 100 times more than the previous
version on the detector, which improved the SNR
This noise is further amplified in the discrete gain drastically.
amplifiers used in the analog front end. Moreover 3) Buffer Amplifier: This was selected not to load the
ON resistance of switches used in the integrator detector and faithfully reproduce the rate information
circuits varies with temperature and supply voltage to
and amplifier offset also changes in temperature. All the next stage. Bandwidth of the amplifier also to be
these affected the overall performance of FOG and chosen not to increase the noise further. Buffer output
resulted in bias variation in temperature. New scheme waveform is shown in fig.16 for a rate input along
was developed to achieve the inertial grade with
performance of gyro, which is explained below: bias modulation to MIOC and the noise (35mV) is
less
3. Version-3 than half of the previous scheme
Instead of PINFET detector, PINDIODE is selected 4) AC coupled Gain Amplifier: AC coupling is
to change the gain and bandwidth of the detector. All introduced to avoid the DC (optical light due to bias
the other opto components and SHA is identical to modulation) and the AC signal due to rate alone to be
version-2 Front end preamplifier scheme is modified amplified. Gain of the amplifier to be selected in such
and digital demodulation was implemented. After a way that the spike width should not be wide enough
reviewing various schemes for the pindiode so that next stage ADC can sample as many samples
configuration and front end processing is as shown in within the transit time.
fig.15. Before making the layout, front end circuit 5)Digital Demodulation: Amplifier output signal is
component selection the below factors are converted directly into digital form by ADC and
considered. sampled at both the cycles and the difference which
is proportional to the rotation rate is done inside the
220
FPGA. Such digital demodulation is intrinsically free Fig.18 Allan Variance Plot of version-3
of any source of electronic long-term drift and analog
input offset of the A/D converter is cancelled out by
the subtraction of A & B samples as shown in fig.16.
With the above scheme achieved performance of the
FOG is shown in table-4.
Table-4
Closed loop scheme version-3 results
4. CONCLUSION
Signal processing schemes for detecting the rate
information for IFOG has taken a decade. For each
scheme layouts and PCBs are made and testing in
live environmental conditions are carried out.
Selection of precision components and up screening
of opto components for space use is also attempted.
Detecting the signal from noise is quite challenging
and winding of fiber also carried out in-house using
Fig.16 Buffer output of version-3 semi automatic winding machine. Signal integrity
issues are solved and 4axes FOG in tetrahedral
Inertial grade FOG performance is achieved by the configuration is in progress.
improved analog processing scheme and modulation
depth is also increased to 0.65instead of 0.5to ACKNOWLEDGEMENT
reduce the RIN noise. Stability plot and Allan
variance plot is shown in fig.17 & 18 respectively. The authors are thankful to Dr. G. Nagendra Rao,
Fig.19 shows the earth rate plot for the Bangalore Dy.Director, LEOS and Shri J.A.Kamalakar Director
latitude (14.59deg/hr) with an offset of 1.5deg/hr. LEOS for their guidance, constant encouragement
and support in the development of interferrometric
fiber optic gyroscope. Authors are also wish to
express their gratitude to Dr.T.K.Alex, Ex Director,
LEOS for his support and guidance in developing the
gyro.
REFERENCES
[1]. William K.Burns, Basic Principles of Fiber-Optic Gyroscope

Fig.17 Stability plot of version-3 in Optical Fiber Rotation Sensing, Academic Press Inc.,
Sandiego, CA
[2]. Eric Udd, Herve C.Lefevre, Kazuo Hotate: Closedloop fiber
optic gyros in Fiber Optic Gyros: 2oth Anniversary Conference,
VOL 2837, SPIE, Bellingham, Washington
[3]. H. Lefevre, Scale Factor Accuracy in The Fiber Optics
Gyroscope, Artech House, Norwood, MA
[4]. R.B.Smith. ed., Selected Papers on Fiber-Optic Gyroscopes,
SPIE Milestone Series, Vol.MS8, SPIE Optical Engineering Press,
Bellingham, Washington
[5]. IEEE Standard Specification format Guide and Test procedure
for Single axis Interferometric Fiber Optic Gyroscope IEEE STD
952-1997
[6]. H. C. Lefevre, J. P. Bettini, S. Vatoux, and M. Papuchon,
Progress in Optical Fiber Gyroscopes using Integrated Optics,
AGARD CPP-383, 9A/1l3,1985
221
[7]. W.Marshall Leach, Jr., Senior Member, IEEE, Fundamentals of Low Noise Analog Circuit Design
Design of Optimal Digital FIR Low Pass

Differentiator by using PSO
Venkatappareddy.P1 M.Subramanyeswara Rao2
Dept of ECE,Viganans University
Vadlamudi,Guntur,AP-522213
differentiators are used in many analog and digital
systems to take the derivatives of a signal.
Abstract-This paper presents design of optimal digital
FIR low pass differentiator by using Particle swarm
Optimization (PSO).PSO is purely random algoritm.In 2. PROBLEM FORMULATION
order to design an optimal digital FIR low pass differ
etiator we have considered TYPE-3 an Type-4 linear p An ideal differentiator has a frequency response that is
hse FIR filters specifications.Impulse response, freque linearly proportional to frequency. Similarly, an ideal
nc response and error plots are obtained by using thes differentiator is defined as one that has the frequency
e secifications for low pass differentiator. Also we hav response
e poved TYPE-4 filters are preferred than TYPE-3 filt Hd(j)= j - (2)
ersbecause of their optimal filter coefficients mean sq An ideal differentiator has an antisymmetric unit
uareerror is smaller. PSO technique as been applied i sample response hd(0)=0.
n many areas such as function optimization, fuzzy The amplitude response for the case of TYPE- 3 linear
systems etc. phase FIR filter is given as
A( ) k 1
( N 1)/2
Key words: PSO, Differentiator, pbest, gbest, mean
c(k )sin( k )
(3)
square error, FIR. N 1
k
1. INTRODUCTION Where c (k) = 2h ( 2 ) 1k(N-1)/2
TYPE-3 FIR filter has Antisymmetric unit sample
PSO Algorithm:
response ie.,
PSO is stochastic optimization technique developed by
h(n)=-h(N-1-n) where N is odd.
Eberhert and Kennedy in 1995. It has been developed
The amplitude response of the TYPE-4 liner phase FIR
based on bird flocking. In PSO the potential solutions,
filter is given by
called particles, fly through the problem space by
A( ) k 1 d(k )sin( k )
( N /2
following the current optimum particles. Each particle
keeps track of its coordinates in the problem space
N
which are associated with best solutions (fitness) it has d ( k ) 2h( n)
achieved so far. This value is called pbest. When a 2
particle takes all the population as its topological TYPE-4 FIR filter has Antisymmetric unit sample
neighbors. The best value is called global best (gbest). response ie.,
The particle swarm optimization concepts consist of, at h(n)=-h(N-1-n) where N is even .
each time step, changing the velocity of each particle The design of a linear phase FIR filter with least mean
towards its pbest location. PSO has no potential square error criterion, we find the filter coefficients a(k)
evolution operators such as crossover and mutation. such that error is minimized. Corresponding to the
FIR Filters: coefficients the filter coefficients are obtained as shown
An FIR filter of length N with input x(n) and by the equations.
output y(n) is described by the following difference equ The Least mean square design function for this design
ation is given as
N 1

k
W (i )[ A(i D(i )]
y (n )
k0
b (k ) x(n k ) = i 1 (4)
(1)FIR for type 3 FIR filter the amplitude response A(w) is a
filters are classified into four types based on symmetry function of a(k) to arrive at the minimum value of ,
and length of filter coefficients TYPE-1, TYPE-2, we set
TYPE-3 and TYPE-4. TYPE-3 and TYPE-4 filters are / a ( k ) 0 0kN
used for design of low pass differentiator. These
222
Which results in a set of (N+1) linear equations that can c 2 * r 2( gbestik 1 U ik 1 )

be solved for a(k).
the position among all partilcles.r1 and r2 are random
For the type3 the expression for mean square error is digits between [0,1].c1 and c2are acceleration
expressed as constants. In the design of FIR differentiators based on
N
the chebyeshev approximation criterion, the weighting
{W (i )[( c(k ) sin k ) D (i )]}2
k
i 1
n0
function W() is specified in this program as
(5) W()=1/ , 02fp.
A similar formation can be derived for the other three Step 7:
types of linear phase FIR filters. This Design approach
Update position of each individual particle as
can be used to design a linear phase FIR filter with
arbitrarily shaped desired response. Where D(w )is the
frequency response. U ik U ik 1 Vi k
Where i=1, 2....M.
3. PROPOSED PSO ALGORITHM Particle Swarm Step 8:
Optimization (PSO)3 algorithm is a population based Pbestik Gbest k
optimization algorithm. Its population is called a swarm Update and
and each individual is called a particle. Each particle Pbestik U ik
=
flies through the solution space to search for global
optimization solution. The implementation of PSO U ik Pbestik
if C( )<C( )
algorithm for optimizing the filter coefficients is given k k
as follows. ref (3). Pbest i U
i ) i Pbest k
= if C ( )>=C(
Step 1: Out of all particles which give min error gives the
Error function is to be minimized is expressed in Gbest.
equation Step 9:
Step 2: Repeat 5 to 9 for maximum no of times to get the least
Initial population (swarm) is generated where each possible error.
particle in the swarm is a solution vector containing
N=10elements, then initial population can be expressed 4. SIMULATION RESULTS
as
U i0 [U i01 , U i02 , U i03 , U i04 , U i05 ] When N=15(TYPE-3)
impulse response
0.05
U ij0 U 0j 0.04
Particle of particle is generated from uniform
0 0.03
U ij ,max
distribution [0 , ]. 0.02
Step 3: 0.01
Initial velocities of each particle are written as follows

h(n)
Vi 0 [Vi10 , Vi 02 , Vi 30 , Vi 04 , Vi 50 ] -0.01
i=1, 2 ,.....5
-0.02
0 0
V ij V j -0.03
Each elements of is selected a random digits for
example, between -0.04
x -0.05
0 2 4 6 8 10 12 14
[0, 0.1 i ,max ] n
Step 4: Fig.(1)
Set iteration count K=1.
Step 5:
Calculate Error value by using eq (5).
min[1 ,2,3,4,5 ]
And corresponding particle is the gbest is the particle
which leads to .
Step 6: Update velocities of each particle using
following relation
Pbestik
Where is the best previous position of the i th
particles, is particle which leads to
Vi k W *Vi k 1 c1* r1( pbestik 1 U ik 1 )
223
frequency response
1.2
0.8
0.6
0.4
gain in db
0.2
-0.2
-0.4
-0.6 Fig.(5)
0 0.2 0.4 0.6 0.8 1 1.2 1.4
error plot
normalized frequency 24
Fig.(2). 23.5
error plot
25.2
23
25.1
Mean square error

22.5
25
22
Mean square error
24.9
21.5
24.8
21
24.7
20.5
24.6 0 10 20 30 40 50 60 70 80 90 100
number of iterations
24.5
0 10 20 30 40 50 60
number of iterations
70 80 90 100 Fig.(6)
Mean Square error obtained for same specifications.
Fig.(3) N=16(TYPE-4)
TYPE MSE()
3 24.6
4 20.8
Table.(1)
5. CONCLUSIONS
The information mechanism in PSO is significantly
different. So the whole population moves like a one
group towards an optimal area. In PSO, only gbest
gives out the information to others. PSO converges to
the best solution quickly and gives optimal filter
coefficients. The amplitude response of TYPE-4 is
smoother than the TYPE-3 and mean square error is
Fig.(4) small in case of TYPE-4. So TYPE-4 filter is better
suited for design of optimal digital FIR low pass
differentiator than TYPE-3.
REFERENCES
[1]Youlian Zhu, Chng hung, Hybridoptimization design of FIR filters,

IEEE.int conference neural networks &signal processing
zhenjing,June 8~10, 2008.
[2] L.S.Titare and L.D. arya A particle swarm optimization for
improvement of voltage stability reactive power reserve management
a journal of the institute of engineers (India) Sep 2006.
[3] www.swarmintelligenc.org/tutorials.php
[4] Genetic search approach .A new learning algorithm for adaptive
IIR filters, IEEE. Int by Leung S.H, Chung.C.Y. Jan 1996.
224
[5] S.K.Mitra, Digital Signal Processing: A Computer-Based

Approach. New York, New York: McGraw-Hill, third ed., 2006.
[6] Digital signal processing principles,algorithm,
and applications by John G.Prokis, D.G.Manolakis , pearson
publications,fourth edition.
[7] Genetic search approach .A new learning algorithm for adaptive
IIR filters, IEEE.int Chung.C.Y., Jan 1996.
[8] N.J.Fliege,Multirate Digital Signal Processing. New York, New
York: John Wiley & Sons, 1994.
Design and Simulation of Pipelined FFT

Processor using FPGA
N.Amarnath Reddy, VLSI@ES-MTech, GMR institute of Technology, Rajam, INDIA,amar.nagu02@gmail.com
D Srinivasa Rao, Asst Prof, Dept. of ECE, GMR institute of Technology, , Rajam, INDIA, srinivasa.dasari@gmail.com
Abstract A parallel and pipelined Fast Fourier reduce the computational burden.
Transform (FFT) processor for use in the Orthogonal In the communication world of today more and more
Frequency division Multiplexer (OFDM) . Unlike being OFDM systems are brought on-line with an ever
stored in the traditional ROM, the twiddle factors in our increasing number of standards, applications, and
pipelined FFT processor can be accessed directly. A services, all with different requirements on the
novel simple address mapping scheme and the physical layer of the transceiver. The physical layer
modified radix 4 FFT also proposed. Finally, the
pipelined 64-point FFT processor can be completely
is most often, due to speed and power constraints,
implemented within 23.750ns. implemented as an ASIC and thus locked to one
Index Terms OFDM, radix- 4, FPGA. specific case. Hence, the number of required
implementations will grow fast. Due to this fact,
1. INTRODUCTION there will be an interest from the industrys point of
view to search for a flexible architecture that can be
Active radar systems are commonly used in military configured to function with several standards,
and commercial applications for the detection and applications, or services. A flexible architecture will
tracking of objects through a medium such as air. lead to reusage of design and implementation and
Most active radar systems work by transmitting a thus reduced cost. This project addresses both OFDM
signal pulse through the medium and receiving the in theory and the implementation aspects of a
reflected energy wave field as it scatters off objects flexible hardware solution for digital baseband
in the medium. By processing the received wave OFDM. This project a flexible FFT processor and a
field, an active radar system can determine an flexible OFDM transmitter are presented. The FFT
object's distance, velocity, and other features. A processor is a central part of an OFDM transceiver,
passive radar system can consist solely of an array of and has been fabricated both as a standalone chip and
receiving antennas as opposed to an active radar as part of an OFDM transmitter chip. The transmitter
system with a co-located transmitter and receiver. chip includes in addition to the FFT, a signal
Without a transmitter the passive radar system mapper and a cyclic prefix inserter. It is shown
relies on other sources of electromagnetic waves, that high flexibility is obtained in the transmitter
such as AM or FM radio waves, TV broadcast, with a reasonable amount of extra hardware.
nearby radar, cell towers, or a wideband waveform to Comprehensive measurements on power consumption
illuminate objects in the medium. Various signal are presented for both chips.
processing algorithms can be applied to the data from A scheme to reduce hardware and delay in an OFDM
the antenna array. One algorithm of interest is the transceiver is proposed in this project. The scheme
Direction of Arrival (DOA) of the incoming scattered shows that the data buffers in an OFDM transceiver,
waveform and therefore the object. A computationally due to a cyclic prefix and a bit reversing pipelined
efficient version of the DFT is known as the Fast FFT processor, can be significantly reduced with a
Fourier Transform (FFT) and is commonly used to cyclic suffix and a bidirectional FFT processor.
225
To keep signals in the frequency domain simplifies

2. BASIC THEORY OF OFDM some signal processing
operations, e.g. convolution in the time domain
becomes multiplication in the frequency domain. The
transformation into the time domain is done in order
to reduce the number of backend RF-oscillators and
demodulators [5]. The available bandwidth (B) in
frequency is split into N sub channels, one for each
Fig. 1. The digital baseband parts of an OFDM subcarrier, where each has a power spectrum shape of
transceiver. a
squared sine pulse. The individual power spectra,
The digital baseband parts of an OFDM transceiver after the IFFT in the transmitter, of a number of
[12], is shown in Figure 1. The basic idea of OFDM subcarriers are shown in Figure 4. Since the IFFT is a
is to divide the available spectrum into N orthogonal linear operation the sub-carriers can be separated
sub channels. In the mapper, data is converted to again with the FFT, even though there spectrum
signals located in the frequency domain where each overlap. This is also true after the signals have passed
sub channel is assigned one signal. The signals are a multipath channel, due to the cyclic extension. The
then transformed to the time domain with the property that each subcarrier is unaffected by the
inverse fast Fourier transform (IFFT). The FFT is an other subcarriers is
efficient method to implement the DFT algorithm, called orthogonally, hence the name OFDM.
based on a divide and conquer approach [13]. The
implementation of an FFT is discussed in Chapter 4 FFT
and from now on, only the FFT notation will be used. The first designed chip is an FFT processor. The FFT
The last digital part of the transmitter inserts a cyclic processor has a central position both in the OFDM
extension to remove the effects of intersymbol transmitter and receiver. The FFT is a
interference (ISI) and inter channel interference (ICI). computationally demanding operation that requires an
The FFT and the IFFT of length N, is defined by the ASIC implementation to reach high performance, i.e.
equations high throughput combined with low energy
consumption.
FFT Architecture
The FFT and IFFT Equation 2.1 and 2.2 has the
property that,
In the receiver the cyclic extension is discarded and

the transmitted signals are transformed back to the
frequencydomain with an FFT. The N received
signals points Y, can bewritten as
Thus, it is only necessary to discuss and implement
the FFT equation. To calculate the inverse transform,
the real and imaginary part of the input and output are
swapped. Since N is a power of two, scaling with 1/N
is the same as right shift the binary word Log (N)
bits. Even simpler, is to just remember that the binary
point has moved log2(N) bits to the left. Not
performing the bit shift until, if ever, it is necessary,
which depends on how the output from the IFFT will
be used.
where G is the FFT of g. To estimate g known data
are transmitted over the channel, these data are called
pilots. In the demapper the estimated signals X are Fig. 2. A radix-2 DIF butterfly (a) and a radix-2 DIT
converted back to data butterfly (b),
IFFT and FFT
The Inverse Fast Fourier Transform (IFFT) Where W is the twiddle factor. The FFT algorithm
transforms signals from frequency domain to time can be realized with a butterfly operation as the basic
domain and the FFT performs the reverse operation.
226
building block [23]. There are two types of butterfly butterflies but the number of complex multipliers are
operations, decimation in time (DIT) and decimation reduced as if it were a radix-4 [24]. Figure 3 shows
in frequency (DIF), both are shown in Figure 2. The an N = 64-point radix-22 DIF FFT. Each stage in a
difference between DIT and DIF lies in the position radix-22 FFT consists of two butterflies,
of the twiddle factor multiplication, which is either one trivial multiplier, i.e. multiplication with -j, and
performed before or after the subtraction and one
addition. Since the FFT is based on divide and complex multiplier that multiply data with the
conquer, it will be most efficient if the input sequence twiddle factor.
is of length N = rp, where N is called point, r is called
radix, and p is a positive integer. To compute an N- 3. FPGA IMPLEMENTATION
point FFT, p stages of butterflies are connected. An
example of a hardware mapped N = 64-point radix-2 The Field Programmable Gate Array is majorly used
DIF FFT is shown in Figure 3. Observe that the input for
data, x(n), occurs in consecutive order, whereas the generation ASIC ICs to the computations. They offer
output data, X(n), is rearranged. The order of the more speed in execution process. SO, for generation
rearranged data is called bit reversed order [23] . It is ASIC ICs FPGAs are majorly used. The 64 FFT
possible to reconfigure the FFT in Figure 3 so that the with radix 4 is simulated and synthesized as well as
input occurs in bit reversed order and the output implemented on the FPGA of below configuration.
occurs in natural order. The reconfiguration can be
thought of as vertically moving the horizontal data
paths in Figure 3, while the cross connections remain
fixed to respective path, until the output is in natural
order. Also notice that, if all arrows in Figure 3 is
turned around a DIT FFT is performed instead of a
DIF FFT. 4. RESULTS:
The RTL view of the butterfly structure obtained

after the simulation of the 64-point FFT block,
Decimation in time domain is shown next and also
the internal architecture of the butterfly block is
shown.
Fig. 3. A hardware mapped N =64-point radix-4 DIF

FFT algorithm
Fig. 5. RTL View Of A Butterfly Component Used

In 64-Point FFT
Fig. 4. A Simple Radix 4 DIF FFT algorithm.
When N is a power of 4, i.e. N =4p, a radix-4 FFT

can be used instead of a radix-2 FFT. With a radix-4
the computational complexity is reduced, i.e. the
number of complex multiplications is reduced
compared to a radix-2 FFT. The drawback with a
radix-4 is that the butterfly structure is more
complicated. Another, better type of radix-4 is the
radix22, there the butterflies still look like radix-2
227
Fig. 6. Internal Architecture of The Butterfly

Component
Fig. 11. Timing report on Hardware
Fig. 7. Simulation result of 64 FFT
Fig. 12. Result of 64 FFT on Hard ware

implementation
Fig. 8. Simulation result of 64 FFT
5. CONCLUSION
This project it is shown that a baseband ASIC can be

fast and at the same time flexible with a low power
consumption. The term fast do not refer to extreme
clock frequencies but to the fact that no part of the
designs needs more than one clock cycle to process a
sample once the pipe is filled. Hence, the design does
not need to be clocked any faster than the requested
Fig. 9. Synthesis report bandwidth and compared to modern CMOS
technology this is a low number, in the order of 5-100
MHz There are several advantages with a low clock
frequency, first
is possible use a low power/low speed cell library
with low static leakage current and secondly, it is
easier to create a clock tree. In addition, there will be
more margins to reduce supply voltage if the required
clock frequency is low. To have a low leakage power
is important in these designs, since flexibility is
achieved with independent modules, where the
Fig. 10. Timing Report Of 64FFT operation mode decides if a module should be used or
not. The unused modules are not clocked and hence
Hardware implementation was through system C only consume static leakage power.
coding andits results are as follows
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229
Development of an Algorithm for 8 bit WTM

1
Sravanthi.kantamaneni, Asst Professor, 2Dr.V.V.K.D.V.Prasad,Professor, 3 Veera Vasantha Rao.Battula,
Abstract-Binary Multipliers plays an important role by the multiplier on a FPGA (Field Programmable Gate
in digital circuits. There are many methods for Array) increases. Hence various algorithms have been
generating a Simple binary multiplication and some developed.
of them are like Ripple carry array multipliers, Row
adder tree multipliers, Partial product LUT in order to achieve lesser complexity in computation
multipliers, Wallace trees, Booth recoding etc.,. Our involving minimum calculation steps, which in turn can
project mainly concentrates on 8x8 Wallace tree reduce delay, power and area constraints of multipliers.
multiplier. It uses a famous Wallace tree structure
which is an implementation of an adder tree The Wallace tree has three computation steps:
designed for minimum propagation delay. Rather
than completely adding the partial products in pairs 1. Generation of Partial products multiplying each
like the ripple adder tree does, the Wallace tree bit of one binary input with every bit of the other
sums up all the bits of the same weights in a merged binary input. If each input has n-bits the result of
tree. Usually full adders are used, so that 3 equally this step will give us n2 number of binary bits
weighted bits are combined to produce two bits: one called Partial products distributed in n-rows and
(the carry) with weight of n+1 and the other (the 2n-columns. This step is very same as what we do
sum) with weight n. Each layer of the tree to multiply two numbers by hand.
therefore reduces the number of vectors by a factor 2. Reduction of partial products the partial products
of 3:2. are to be added according to their place values (or
weights) using half adders and full adders until
A conventional adder is used to combine these to only two rows of partial products are left.
obtain the final product. The benefits of the Wallace 3. Last stage addition remaining two rows will be
tree is that there are only O(logn) reduction layers, added using a conventional adder to get final result
and each layer has O(1) propagation delay. As of multiplication.
making the partial products is O(1) and the final
addition is O(logn), the multiplication is only
O(logn), not much slower than addition (however,
much more expensive in the gate count). Naively
adding partial products with regular adders would
require O(log2n) time. Our project is to develop 8 x 8
Wallace tree multiplier using VHDL and will be
simulated with the help of XILINX simulator and
verified on Spartan-3E FPGA circuit board.
INTRODUCTION
Recent advancements in mobile computing and

multimedia applications demand for high performance
and low-power consuming VLSI (very large scale
integrated circuit) Digital Signal Processing (DSP)
systems. One of the most important components of DSP
systems is a multiplier. Multiplication is basically shift
and add operation. Usually in a DSP system, multiplier
units consume large amount of power and cause most of
the delay compared to other units like adders.
Depending on size of the inputs (2 X 2 bit, 4 X 4, 8 X 8

etc.,) the number of steps a normal binary multiplier Wallace tree multiplier reduction stages for 8X8
takes to compute the product increases drastically. multiplication
Larger the steps of calculation larger will be the delay
as well as the power consumption. Also area occupied
230
least significant bits (LSBs) of A and B. LSB of B

say b0 will be AND with all the eight bits of A from
a0 to a7. This gives us a single row of partial products.
The second row of PPs will be generated when all
bits of A will be AND with b1. Similarly, every row
will be formed due to AND operations.
Rows of partial products of 8 bit multiplication
Every partial product of first row has b0. Every

second row element has b1. Similarly every partial
product of nth row will have the common AND input
bn-1. So, we can write mathematically the first row as
(a7 a6 a5 a4 a3 a2 a1)*b0. Any nth row can be written as
(a7 a6 a5 a4 a3 a2 a1)*bn-1.
We know that all the above bits are binary
digits i.e. either 0 or 1. Hence two possibilities exist.
If b0 = 0:
Then the first row (a7 a6 a5 a4 a3 a2 a1)*b0

will be equal to
(a7 a6 a5 a4 a3 a2 a1)*0 = (0 0 0 0 0 0 0 0)
If b0 =1:
FINAL RESULT BITS
Then the first row (a7 a6 a5 a4 a3 a2 a1)*b0
will be equal to
(a7 a6 a5 a4 a3 a2 a1)*1 = (a7 a6 a5 a4 a3 a2 a1) = A
Hence any row will be equal to the Multiplicand A or

it will be a row full of Zeros.
Keeping the above fact in mind we can use another

way to generate partial products for our need without
using n2 number of AND gates (n is the size of
inputs). This method is given below. Any nth row
will be 0 or A based on the value of common input
Improvements to the Algorithm:
bn-1 of that row.
First stage: Row n = 0, if bn-1=0
It is evident that every multiplication in Row n =A, if bn-1=1

Wallace tree algorithm is done in three logical stages.
They are the partial product generation, the reduction Consider the following example to comprehend this
stages and the last stage addition using a new logic in a better way.
conventional adder. If we recall the description for Let A= 11111111 and B=10011011. A is
Partial product generation, the first step, we are doing multiplicand and B is multiplier as usual.
it by use of AND gates. If two 8 bit numbers say A The rows of partial products for the multiplication A
and B are to be multiplied, the algorithm starts at the X B are:
231
Row 1 = b0 AND (11111111) = 1 AND (11111111) There is another sound drawback of

=11111111 =A representing variables (or signals with respect to
Row 2 = b1 AND (11111111) = 0 AND (11111111) VHDL coding) using normal alphabets like A, B, M
=11111111 =A or N etc. Let us assume that we have come across a
Row 3 = b2 AND (11111111) = 0 AND (11111111) signal N2 while verifying the design. We cannot
=00000000 =O readily identify which reduction stage this signal N2
Row 4 = b3 AND (11111111) = 1 AND (11111111) belongs to. We must go through the code once again
=11111111 =A form start and locate where N2 has its origin.
Row 5 = b4 AND (11111111) = 1 AND (11111111) Imagine a 32 bit multiplier which will have a very
=11111111 =A large number of such signals. To go through the code
Row 6 = b5 AND (11111111) = 0 AND (11111111) every other time to know about a signal, it is a
=00000000 =O tremendous burden for the designer. So, it is of high
Row 7 = b6 AND (11111111) = 0 AND (11111111) importance that we have a proper representation
=00000000 =O scheme for signals or variables. We must be able to
Row 8 = b7 AND (11111111) = 1 AND (11111111) identify from the name of a signal or variable several
=11111111 =A aspects. They are:
Or
1. The reduction stage to which it belongs
Simply we can write: 2. The column or the weight of the partial product
3. Whether it is a SUM bit or CARRY bit
Row 1 = A since b0= 1
Row 2 = O since b1= 0 Family Spartan 3E
Row 3 = O since b2= 0
Row 4 = A since b3= 1 Category General purpose
Row 5 = A since b4= 1
Row 6 = O since b5= 0 Device Type XC3S250E
Row 7 = O since b6= 0
Row 8 = A since b7= 1 Package Type PQ 208: pin Plastic
Quad Flat Pack
Advantage: (PQFP)
In case of previous method for a n-bit
multiplier, the first stage will require a total of n2 Speed Grade -5: High Performance
number of AND gates. But with the new method the
task of partial product generation will be done by just 4. If more than one sum and carry bits are present
n number of steps instead of n2 steps. It is important in the column, then position of that bit in the
to note that while describing the multiplier in VHDL column.
code, each of the n2 AND operations have to be
written manually. For a 32 bit- multiplier it requires
1024 steps to be written for simple AND operations.
Instead, with the new modification, only 32 steps are
to be written which will save a lot of energy and time
to the design engineer during development of the
code. So, we have adopted the latter method in
designing the WTM system.
Representation of signals:
As described in an algorithm, we have to use
different variables to indicate the partial products in
different levels of reduction stages. For example, we
used a, b in first stage, then P0, P1 in the next and Scheme of representation of signals
later S, C, M, N etc. in the figures of chapter 2. These To satisfy the above four requirements, we adopt the
variables are of our choice and we must make sure above representation scheme.
that PPs in different levels of reduction do not have
the same representation. It means that designer has to Ri reduction stage number i ; Eg: R1,
make a note of which variables he is using in what R2, R4 etc.
stage of reduction, clearly and without confusion. Sj sum bit of column j+1
232
Cj carry bit of column j+1Weight of 30 26.642

partial product = 2j WTM
k place of the signal in the column. 18.641
Delay(ns)
20 13.924
If there are 4 sum bits in the column k takes the
values of 1, 2, 3 and 4. CSHM
10
Always the sum bits are taken first and the carry bits
are taken next to sums. Reverse will also give the
0 Booth
same answer, but to avoid confusion sums are given 8*8 Multiplier Multiplier
first priority in any column. Let us consider a column
having 4 sums and 3 carries. Let all the bits belong to
column number 6 (j=5) of 3rd reduction stage. It will
be represented as follows.
Comparison of Multipliers in terms of Area:
R3S5_1
R3S5_2 200 146 169180
180
WTM
Area(LUT)
R3S5_3
R3S5_4
R3C5_1 CSHM
0
R3C5_2 8*8 Multiplier
R3C5_3
A column of signals of 3rd reduction stage, 5th column
Designing, Synthesis and Results of WTM for the
Spartan 3E family FPGA chip. Comparison of Multipliers in terms of PDP:
Simulated output of WTM

2 1.385 WTM
0.969
0.724
PDP
0 CSHM
8*8 Multiplier
Comparison of Multipliers in terms of ADP:
WTM CSHM Booth Multiplier

Maximum Combinational Path Delay:
0.6 0.271 0.2353
0.4 0.4795
ADP
Size of Maximum combinational 0.2

0
S. No 8*8 Multiplier
multiplier path delay (Nano seconds)
WTM 0.271
CSHM 0.2353
1 4 10.426
Booth Multiplier 0.4795
8*8 Multiplier
2 6 11.921
3 8 13.924
Comparison of Multipliers w.r.t Delay (ns):
233
Features of WTM:
Conclusion and Discussion:

Parameter Usage Available Pre
Layout It can be concluded that Wallace tree multiplier
Values is superior in all respects like Delay, Area and speed.
However array multiplier requires more power consumption
(or) Ratio
and gives optimum number of components required, but it
can provides a better delay .If we utilize this multiplier as a
83 2448 3% module of real time applications like FIR Filter
Number Of Slices 96 ,it can pump
up the Filtering action. Further the work can be extended for
146 4896 2% optimization of said multiplier to improve speed or to
Number of 4-input minimize the Power Consumption.
LUTs 178
References:
32 158 20%
Number Of Bonded
1. C. S. Wallace, A Suggestion for a Fast Multiplier,
Input IEEE Transactions on Electronic Computers, 32 February
1964, EC-13:1417.
32 158 20%
Number Of Bonded 2. Vijaya Prakash A. M, Dr. MGR, K. S. Gurumurthy,
Output A Novel VLSI Architecture for Low power 32 FIR Filter,
International Journal of Advanced Engineering &
13.924 Application, January 2011, PP 218 - 224.
Delay(ns) 3. Gary W. Bewick, Fast multiplication algorithms
and implementation, The Department Of Electrical
Slice Utilization 100 Engineering and The Committee on Graduate studies of
Ratio STANFORD UNIVERSITY, February 1994. PP 8 - 16.
4. J. Bhasker, AVHDL Primer, Third Edition, Pearson
BRAM Utilization 100 Education, 2007, PP-21 to 50, 88 to 101
Ratio 5. John F. Wakerly, Digital Design Principle and
Practices, fourth edition, Prentice Hall Pearson Education,
2009, PP 235-250, PP 786-795
6. http://en.wikipedia.org/wiki/Wallace_tree, http://en.wikipedia.org/wiki/FPGA.
7. Multiplication in FPGAs The performance FPGA DESIGN specialist.
234
FPGA Implementation of Least Mean Square

Algorithm for Speech Enhancement
1 2
Ravi Bolimera , M. Paul Sandeep
Abstract - The aim of this paper is to investigate the for a large area, it is sufficient to use fixed-point
speech signal enhancement using Adaptive filters. arithmetic for the adaptive filter to work well. The
Background noise elimination is very important in algorithm used to update the filter coefficient is
many applications like hearing aids, forensic the Least Mean Square (LMS) algorithm which is
applications, military, cellular environments etc. The
known for its simplification, low computational
adaptive filters has advantages like low processing
delay and better tracking of the trajectory of non- complexity, and better performance in different
stationary signals. If the evaluation of background running environments. In terms of the number of
noise is more slowly than the speech, i.e., noise signal iterations required for convergence, LMS
is more stationary than the speech signal, we can algorithm is seen to perform very well when
easily estimate the noise during the pauses in speech. compared to other algorithms used for
Otherwise it is more difficult to estimate the noise implementing adaptive filters.
which results in degradation of speech. In order to
enhance the quality a n d intelligibility of speech, here II. SPEECH ENHANCEMENT
we proposed a design based on LMS algorithm to
remove n o i s e from speech signal using VHDL based
on commercially available FPGAs. In this design the Speech enhancement is concerned with improving
LMS algorithm used as a noise canceller and the some perceptual aspect of speech that has been
reference sig nal was adaptively f i l t e r e d a n d d e g r a d e d b y additive noise. In most applications
subtracted from primary signal to obtain the the aim of speech enhancement is to improve the
estimated speech signal. quality and intelligibility of degraded speech. The
improvement in quality is highly desirable as it can
Keywords: LMS, Noise canceller, FPGA, VHDL listener fatigue particularly in situations where the
and Speech Enhancement. listener is exposed to high levels of noise for long
period of time. The need to enhance speech signals
I. INTRODUCTION
arises in many situations in which the speech signal
To model the relationship between two signals in
originates from a noise location or is affected b y
real time in an iterative manner, An Adaptive filter
noise location or is affected b y noise over a
is used which is a computational device. Adaptive
communication channel.
filters are often realized either as a set of program
Speech enhancement algorithms can therefore be
instructions running on an arithmetical processing
used to improve the quality of speech at the
device such as a DSP chip or as set of logic
receiving end. The solution to the general problem
operations implemented in a field programmable
of speech enhancement depends largely on the
gate array (FPGA) or in a custom VLSI integrated
application at hand, the characteristics of noise
circuits.
source or interference, the relationship of the noise
Adaptive filters, as a part of digital signal system,
to the clean signal.
have been widely used in communication industry,
as well as one of the applications such as adaptive Critical to the design of speech enhancement
noise cancellation. However, implementation of algorithms is knowledge of the range of speech
adaptive filter takes a great deal and becomes a and noise intensity levels in real-world scenarios.
very important role in digital system design. From that we estimate the range of signal-to-noise
Adaptive filters are usually implemented in DSP r atio [SNR] levels encountered in realistic
processor because they are capable of performing environment. This is important since speech
fast floating-point arithmetic operations. But now a enhancement algorithms need to be effective in
day in signal processing, FPGA becomes an suppressing the noise and improving speech quality
important competitor by providing a lot of facilities within that range of SNR level.
to designers in the market. In addition, FPGA is a
form of programmable logic that can offer III. ADAPTIVE FILTERS
flexibility for repetitive reconfiguration. Because
of slices organized as array of rows and columns Adaptive filters learn the statistics of their operating
in FPGA, a great deal of parallelism can be environment and continually adjust their parameters
explored. Although it is not efficient to use accordingly. This chapter presents the theory
floating-point arithmetic in FPGA due to its need of the algorithms needed to train the filters. In
235
practice, signals of interest often become error signal, forcing the adaptive filter output signal
contaminated by noise or other signals occup ying to approximate the reference signal in a statistical
the same band of frequency. When the signal of sense.
interest and the noise reside in separate
frequency bands, conventional linear filters are IV. LMS ALGORITHM
able to extract the desired signal . However, when
there is spectral overlap between the signal and The LMS algorithm is a stochastic gradient
noise, or the signal or interfering signals algorithm, which uses a fixed step-size parameter to
statistics change with time, fixed coefficient filters control the updates to the tap weights of a
are inappropriate. transversal filter, (Widrow and Hoff 1960). The
The goal of any filter is to extract useful algorithm aims to minimize t h e mean-square error,
information from noisy data. Whereas a normal the error being the difference in y(n) and yest(n). The
fixed filter is designed in advance with knowledge dependence of the mean-square error on the
of the statistics of both the signal and the unknown tap weights may be viewed as a
unwanted noise, the adaptive filter continuously multidimensional paraboloid referred to as the error
adjusts to a changing environment through the surface .The surface has a uniquely defined
use of recursive algorithms. This is useful when minimum point defining the tap weights for the
either the statistics of the signals are not known optimum Wiener solution. However, in the non
beforehand o f change with time. stationary environment t h i s error surface is
The adaptive filter has advantages like lower continuously changing, thus the LMS algorithm
processing delay and better tracking of the trajectory needs to be able to track the bottom of the surface.
of non-stationary signals [2]. These are essential The LMS algorithm aims to minimize a cost
characteristics in applications such as noise function,
estimation, echo cancellation, adaptive delay V (w(n)), at each time step n, by a suitable
estimation and channel equalization, where low choice of the weight vector w(n). The strategy is
delay and fast racking of time-varying to update the parameter estimate proportional t o
environments and time-varying processes are the instantaneous g r a d i e n t value, dV (w(n))/dw(n),
important objectives. The e x i s t e n c e of a so that:
reference signal which is hidden in the fixed-filter
approximation step, defines the performance w(n + 1) = w(n) dV (w(n))/dw(n) (2)
criterion. The general adaptive-filter configuration where is a small positive step size and the
is shown in fig 1. minus sign ensures that the parameter estimates
descend the error surface. The cost function, V
(w(n)), which minimizes the mean-square error,
results in the following recursive parameter
update equation
w(n + 1) = w(n) x(n) (y(n) yest (n)) (3)
The recursive relation for updating t h e tap
weight vector Equation (1) may be rewritten as
w(n + 1) = w(n) x(n)(y(n) xT(n)w(n)) (4)
which can be represented as filter output ,
estimation error and tap weight adaptation.
The basic configuration of an adaptive filter,
operating in the discrete-time domain k, is
illustrated in Figure 1. In such a scheme, the input
signal is denoted by x(n), the reference signal d(n)
represents the desired output signal (that usually The LMS algorithm requires only 2N + 1
includes some noise component), y(n) is the multiplications
output of the adaptive filter, and the error signal is and 2N additions per iteration for an N tap weight
defined as vector. Therefore it has a relatively simple
e(n) = d(n)y(n) (1) structure and the hardware is directly proportional
The error signal is used by the adaptation to the number of weights.
algorithm to update the adaptive filter coefficient The L M S a l g o r i t h m is b y far the most widely
vector w(n) according to some performance used algorithm in adaptive filtering for several
criterion. In general, the whole adaptation reasons. The main features that attracted the use of
process aims at minimizing some metric of the the LMS algorithm are low computational
236
complexity, proof of convergence in stationary embedded processor designs based on Xilinxs

environment, unbiased convergence in the mean to MicroBlaze. Genesys is compatible with all Xilinx
the Wiener solution, and stable behaviour when CAD tools, including ChipScope, EDK, and the free
implemented with finite-precision arithmetic. The WebPack. The Virtex5-LX50T is optimized for
convergence analysis of the LMS presented here high performance logic and offers: 7,200 slices,
utilizes the independence assumption. each containing four 6- input LUTs and eight flip-
flops,1.7Mbits of fast block RAM, 12 digital clock
V. FPGA IMPLEMENTATION managers, six phase-locked loops, 48 DSP slices
and 500MHz+ clock speeds.
FPGA Implementation of an Adaptive Filter
architecture using LMS algorithm. Adaptive filter A. Simulation Results
is commonly used for a wide range of applications A review of adaptive filters shows that the LMS
such as Echo cancellation, Prediction, Noise algorithm is still a popular choice for its stable
cancellation, Adaptive interference cancelling, performance and high speed capability. The other
system identification, Digital communication advantage of the LMS over other adaptive algorithm
Receiver, Radar signal processing, Equalizations of is its high convergence rate. The high-speed
communication channels, Biomedical signal capability and register rich architecture of the
enhancements, Navigational systems, Adaptive FPGA is ideal for implementing LMS.
antenna systems beam forming etc..
In this paper, adaptive filter architecture
implemented mainly application for noise
cancellation. In the noise cancellation the
architecture is taken in both sequential as well as
parallel. For implementing A d a p t i v e fi lt er
architecture, LMS algorithm is used because of low
computational complexity, simplicity and gives its
better performance in different running
environments.In all adaptive filter architecture try to
minimize error i.e. minimization of different
between the desired output and the real one for all
the input vectors. Nowadays, the use of Field
programmable gate arrays (FPGAs) is growing.
Field programmable gate arrays (FPGAs) are widely Fig 2. Xilinx ISE Simulation Results
used in many areas such as audio and video,
Digital signal processing, Image signal Processing,
Digital communication systems, mobile
communication system and many other embedded
applications, because of their high performance,
parallel processing ability and flexibility.
There are usually two ways to implement the LMS
algorithm, hardware implementation and software
implementation [6], [7]. The hardware
implementation of the algorithm in an FPGA has
good real-time ability, but requires large resources.
VI. HARDWARE IMPLEMENTATION
In this work, the ANC is implemented in the

Xilinx virtex-5 kit (Genesys) board. The Genesys Fig 3. Simulation Wave Forms
circuit board is a complete, ready-to-use digital
circuit development platform based on a Xilinx The Xilinx ISE 13.4 software package is used in
Virtex 5 LX50T. The large on-board collection this project to implement the VHDL program. The
of high-end peripherals, including Gbit Ethernet, system can be simulated to observe and verify its
HDMI Video, 64-bit DDR2 memory array, and behaviour and the results were showed in fig. 2 and
audio and USB ports make the Genesys board an fig. 3. The design is used for the implementation on
ideal host for complete digital systems, including the FPGA chip.
237
B. Chipscope Results
ChipScope provides you with a convenient software
based interface for controlling the integrated logic REFERENCES
analyzer, including setting in the triggering options
[1] S. Haykin, Adaptive Filter Theory, Prentice-Hall, third
and viewing the waveforms. Time required for
edition, 2002.
simulating co mplex design for all possible test [2] Roger Woods, FPGA-based Implementation of Signal
cases becomes prohibitively large and simulation Processing Systems John McAllister, Gaye Lightbody, Ying Yi.
approach fails. For rapid testing, such designs can be [3] U. Meyer Baese, Digital Signal Processing with Field
Programmable Gate Arrays, Springer, 2001.
loaded on to the target FPGAs and tested by
[4] Xilinx Inc., Block Adaptive Filter, Application Note XAPP
applying test inputs and directly observing their 055, Xilinx, San Jose, CA, January 9, 1997.
outputs. The ChipScope Pro tools integrate key [5] www.xilinx.com
logic analyzer and other test and measurement [6] P. Waldeck and N. Bergmann, Evaluating software and
hardware implementations of signal-processing tasks in an FPGA,
h a r d wa r e components with the target design
in Proc. IEEE International Conference on Field-Programmable
inside the FPGA. Computer based software tool Technology, Brisbane, Australia, Dec. 2004, pp. 299-302.
communicate with these hardware components and [7] Ravi Bolimera, T.Kranthi Kumar, M.Paul Sandeep, Speech
provide a designer robust logic analyser solution. Enhancement using Normalized Kernel Affine Projection
Algorithm, Proc. of the Intl. Conf. on Advances in Computer
Waveforms are showed in fig. 4.
Science and Electronics Engineering CSEE 2013, ISBN: 978-
981-07-5461-7
doi:10.3850/ 978-981-07-5461-7_42.
[8] A. Elhossini, S. Areibi and R. Dony, An FPGA
Implementation of the LMS Adaptive Filter for Audio Processing,
in Proc. IEEE International Conference on Reconfigurable
Computing and FPGAs, Sept. 2006, pp. 1-8.
[9] www.digilentinc.com/
Fig 4. Chipscope waveforms
C. Timing Summary
Speed Grade: -2
Minimum period: 23.452ns
(Maximum Frequency: 42.641MHz)
Minimum input arrival time before clock:
15.361ns
Maximum output required time after clock:
2.826ns
Maximum combinational path delay: 3.237ns
VIII. CONCLUSION
Based on LMS algorithm an adaptive equalizer is

designed and implemented using hardware
description language VHDL. The design is simulated
on Xilinx ISE 13.4 and a satisfactory result is
obtained.
238
Design of Energy Efficient Logic Using Adiabatic

Technique
K B V Babu, (M.Tech-VLSI), GMR institute of Technology, Rajam, INDIA, bvbabu.411@gmail.com
B I Neelgar Professor, Department of ECE, GMR institute of Technology, Rajam, INDIA,
Abstract: With the adiabatic switching approach, complexity. Adiabatic logic offers away to reuse the
the circuit energies are conserved rather than energy stored in the load capacitors rather than the
dissipated as heat. Depending on the application traditional way of discharging the load capacitors to
and the system requirements, this approach can the ground and wasting this energy.
be used to reduce the power dissipation of the This demonstrates the low power dissipation of
digital systems. This work focuses on the design of Adiabatic Logic by presenting the results of
a new full adder circuit, using adiabatic logic designs designing various design/ cell units employing
such as ECRL and PFAL are implemented in Micro- Adiabatic Logic circuit techniques. A family of full-
wind & DSCH. The efficiency of the circuits is explored custom conventional CMOS Logic and an Adiabatic
and compared for different adiabatic logics. With the Logic units for example, an inverter, a two input
help of adiabatic logic, the energy savings of upto NAND gate, a two-input NOR gate, a two-input
76% to 90% can be reached. Circuit XOR gate, a two-to-one multiplexer and a one-bit
simulations show that the adiabatic design units Full-Adder were designed in Mentor Graphics IC
can save energy by a factor of 10 at 50MHz and Design Architect using standard TSMC0.35 m
about 2 at 250MHz, as compared to the logically technology, laid out in Micro wind IC Station.
equivalent conventional CMOS implementation. All the circuit simulations has been done using
Index Terms-: Adiabatic, ECRL Adder, PFAL Adder, Full various schematics of the structures and post-layout
adder. simulations are also being done after they all have
been laid-out by considering all the basic design
I. INTRODUCTION rules and by running the LVS program. Finally, the
analysis of the average dynamic power dissipation
The main objective of this design is to provide new with respect to the frequency and the load
low power solutions for Very Large Scale Integration capacitance was done to show the amount of power
(VLSI) designers. Especially, this work focuses on dissipated by the two logic families.
the reduction of the power dissipation, which is
showing an ever-increasing growth with the scaling II. MOTIVATION
down of the technologies. Various techniques at the
different levels of the design process have been In the past few decades ago, the electronics industry
implemented to reduce the power dissipation at the has been experiencing an unprecedented spurt i n
circuit, architectural and system level. growth, thanks to the use of integrated circuits in
Furthermore, the number of gates per chip area is computing, telecommunications and consumer
constantly increasing, while the gate switching electronics. We have come a long way from the
energy does not decrease at the same rate, so the single transistor era in 1958 to the present day
power dissipation rises and heat removal becomes ULSI (Ultra Large Scale Integration) systems with
more difficult and expensive. Then, to limit the more than 50 million transistors in a single chip.
power dissipation, alternative solutions at each level The ever-growing number of transistors integrated on
of abstraction are propose. The dynamic power a chip and the increasing transistor switching speed
requirement of CMOS circuits is rapidly becoming a in recent decades has enabled great performance
major concern in the design of personal information improvement in computer systems by several orders
systems and large computers. of magnitude. Unfortunately, such phenomenal
In this work, a new CMOS logic family called performance improvements have been accompanied
ADIABATIC LOGIC, based on the adiabatic by an increase in power and energy dissipation of the
switching principle is presented. The term adiabatic systems. Higher power and energy dissipation in
comes from thermodynamics, used to describe a high performance systems require more expensive
process in which there is no exchange of heat with packaging and cooling technologies, increase cost,
the environment. The adiabatic logic structure and decrease system reliability. None the less, the
dramatically reduces the power dissipation. The level of on- chip integration and clock frequency will
adiabatic switching technique can achieve very continue to grow with increasing performance
low power dissipation, but at the expense of circuit demands, and the power and energy dissipation of
239
high-performance systems will be a critical design the conventional CMOS gates and adiabatic counter
constraint. For example, high-end microprocessors parts. To convert a conventional CMOS logic gate
in 2010 are predicted to employ billions of into an adiabatic gate, the pull-up and the pull-down
transistors at clock rates over 30GHz to achieve networks must be replaced with complementary
TIPS (Tera Instructions per seconds) performance transmission-gate(T-gate) networks. The T- gate
[1]. With this rate, high-end microprocessors power network implementing the pull-up function is used to
dissipation is projected to reach thousands of Watts. drive the true output of the adiabatic gate, while the
This thesis investigates one of the major sources of T-gate network implementing the pull-down
the power/energy dissipation and proposes and function drives the complementary output node.
evaluates the techniques to reduce the dissipation. Note that all the given inputs should also be
Digital CMOS integrated circuits have been the available in the complementary form. Both the
driving force behind VLSI for high performance networks in the adiabatic logic circuit are used to
computing and other applications, related to science charge-up as well as charge-down the output
and technology. The demand for digital CMOS capacitance, which ensures that the energy stored at
integrated circuits will continue to increase in the the output node can be retrieved by the power supply,
near future, due to its important salient features at the end of each cycle. To allow adiabatic
like low power, reliable performance and operation, the DC voltage source of the original
improvements in the processing technology. circuit must be replaced by a pulsed- power supply
The word ADIABATIC comes from a Greek word with the ramped voltage output.
that is used to describe thermodynamic processes
that exchange no energy with the environment and III. ADIABATIC LOGIC TYPES:
therefore, no energy loss in the form of dissipated
heat. In real-life computing, such ideal process Practical adiabatic families can be classified as either
cannot be achieved because of the presence of PARTIALLY ADIABATIC or FULLY
dissipative elements like resistances in a circuit. ADIABATIC [12]. In a PARTIALLY ADIABATIC
However, one can achieve very low energy CIRCUIT, some charge is allowed to be transferred
dissipation by slowing down the speed of to the ground, while in a FULLY
operation and only switching transistors under ADIABATICCIRCUIT, all the charge on the load
certain conditions. The signal energies stored in the capacitance is recovered by the power supply. Fully
circuit capacitances are recycled instead, of being adiabatic circuits face a lot of problems with respect
dissipated as heat. The adiabatic logic is also known to the operating speed and the inputs power clock
as ENERGY RECOVERY CMOS. synchronization.
It should be noted that the fully adiabatic operation
of the circuit is an ideal condition which may only be ECRL-Efficient Charge Recovery Logic:
approached asymptotically as the switching process Efficient ChargeRecovery Logic (ECRL) proposed
is slowed down. In most practical cases, the energy by Moon and Jeong [8], shown in Figure, uses cross-
dissipation associated with a charge transfer event is coupled PMOS transistors. It has the structure similar
usually composed of an adiabatic component and a to Cascode Voltage Switch Logic (CVSL) with
non-adiabatic component. Therefore, reducing all differential signaling. It consists of two cross-
the energy loss to zero may not possible, coupled transistors M1 and M2 and two NMOS
regardless of the switching speed. With the adiabatic transistors in the circuit.
switching approach, the circuit energies are An AC power supply pwr is used for ECRL gates, so
conserved rather than dissipated as heat. Depending a store covers and reuses the supplied energy. Both
out and/out are generated so that the power clock
on the application and the system requirements, this
generator can always drive a constant load
approach can sometimes be used to reduce the power
capacitance independent of the input signal. A more
dissipation of the digital systems.Here, the load
detailed description of ECRL can be found in .Full
capacitance is charged by a constant-current source
output swing is obtained because of the cross-
(instead of the constant-voltage source as in the
coupled PMOS transistors in both pre charge and
conventional CMOS circuits).
recoverphases. But due to the threshold voltage of
Here, R is the resistance of the PMOS network. A
the PMOS transistors, the circuits suffer from the
constant charging current corresponds to a linear
non-adiabatic loss both in the pre charge and recover
voltage ramp. Assume, the capacitor voltage VC is
phases. That is, to say, ECRL always pumps charge
zero initially. on the output with a full swing. However, as the
In the following, we will examine simple circuit voltage on the supply clock approaches to ck.So the
configurations which can be used for adiabatic recovery path to the supply clock is disconnected,
switching. shows a general circuit topology for thus, resulting in incomplete recovery. Vtp is the
240
threshold voltage of PMOS transistor. The amount of PMOSFETs as in ECRL logic, and that the
loss is given as functional blocks are in parallel with the
transmission PMOSFETs. Thus the equivalent
resistance is smaller when the capacitance needs to
Thus, from Equation , it can be inferred that the non- be charged. The energy dissipation by the CMOS
adiabatic energy loss is dependent on the load. Logic family and Adiabatic PFAL Logic family can
beSeen.
IV. POSITIVE FEEDBACK ADIABATIC
LOGIC
The partial energy recovery circuit structure

named Positive Feedback Adiabatic Logic
(PFAL)[15] has been used, since it shows the
lowest energy consumption if compared to other
similar families, and a good robustness against
technological parameter variations. It is a dual-rail
circuit with partial energy recovery. The general
schematic of the PFAL gate is shown in Figure4.3.
Figure3: PFAL Sum Circuit
The core of all the PFAL gates is an adiabatic
amplifier, a latch made by the two PMOS M1-M2
and two NMOS M3-M4, that avoids a logic level
degradation on the output nodes out and /out. The
two n-trees realize the logic functions. This logic
family also generates both positive and negative
outputs. The functional blocks are in parallel with
the PMOSFETs of the adiabatic amplifier and form
a transmission gate. The two n-trees realize the
logic functions. This logic family also generates both
positive and negative outputs.
Figure4: PFAL Carry Circuit
Figure2: The Basic Structure of the Adiabatic PFAL Figure5: ECRL SUM Circuit
Logic.
V. ADIABATIC FULL ADDER USING PFAL &

ECRL:
A partially adiabatic logic family PFAL one-bit Full

Adder block can be implemented as shown in the
Figure 5.23 ( for SUM block) and Figure 5.24
(for OUTPUT_CARRY) below, respectively.
The two major differences with respect to ECRL are

that the latch is made by two PMOSFETs and two
NMOSFETS, rather than by only two Figure6: ECRL Carry Circuit
241
The power dissipation results of general adder and REFERENCE

full adder using adiabatic is given below.
[1]. P.CHANDRAKASAN, S.SHENGAND, R.W.BRODERSEN,
Low Power CMOS Digital Design, IEEE Journal of Solid-state
Circuits, Vol.27, No.04, pp.473484, April 1999.
[2]. H.J.M.VEENDRICK, Short-circuit Dissipation of Static
CMOS Circuitry and its Impact on the Design of Buffer Circuits,
IEEE JSSC, pp. 468-473, August 1984.
[3]. J.M.RABAEY AND M.PEDRAM, Low Power Design
Methodologies, Kluwer Academic Publishers, 2002.
[4]. M. HOROWITZ, T. INDENNAUR, AND R. GONZALEZ,
Low Power Digital Design, Technical Digest IEEE Symposium
In this work,efficient charge recovery logic(ECRL) Low Power Electronics, San Diego, pp. 08-11, October 1994.
and positive feedback adiabatic logic(PFAL) with [5]. T.SAKURAIANDA.R.NEWTON, Alpha-Power Law
MOSET Model and its Applications to CMOS Inverter Delay and
sum and carry respective outputs are noticied as other Formulas, IEEE JSSC, vol.25, no. 02, pp. 584- 594,
below in tabular form October 1990.
[6]. W.C.ATHAS, J.G.KOLLER ,L.SVENSSON, An Energy -
Efficient CMOS Line Driver using Adiabatic Switching, Fourth
Great Lakes symposium on VLSI, California, March 2005.
[7]. T.INDERMAUER AND M.HOROWITZ, Evaluation of
Charge Recovery Circuits and Adiabatic Switching for Low Power
Design, Technical Digest IEEE Symposium Low Power
Electronics, San Diego, pp.102-103, October 2002.
[8]. Y.MOON AND D.K.JEONG, An Efficient Charge Recovery
Logic Circuit, IEEE JSSC, Vol. 31, No. 04, pp. 514-522, April
1996.
[9]. A. KAMER, J. S. DENKER, B. FLOWER, et al., 2N2D-
order Adiabatic Computation with 2N-2P and 2N-2N2P Logic
Circuits, In Proc. of the International
Symposium on Low Power design, Dana Point, pp. 191196, 1995.
[10]. A.BLOTTI AND R.SALETTI, Ultralow-Power Adiabatic
Circuit Semi-Custom Design, IEEE Transactions on VLSI
Systems, vol. 12, no. 11, pp. 1248-125, November 2004.
VI. CONCLUSION
The paper primarily was focused on the design of low

power CMOS cell structures, which is the main
contribution of this work. The design of low power
CMOS cell structures uses fully complementary
CMOS logic style and an adiabatic PFAL logic style.
The basic principle behind implementing various
design un its in the two logic styles is to compare
them with reference to the average power dissipated
by all of them.
A family of full-custom conventional CMOS Logic
and an Adiabatic Logic units were designed in
Mentor Graphics IC Design Architect using standard
TSMC 0.35m technology,layout them in Microwind
& Digital Schematic and the analysis of the average
dynamic power dissipation with respect to the
frequency and the load capacitance was done. It was
found that the adiabatic PFAL logic style is
advantageous in applications where power
reduction is of prime importance as in high
performance battery-portable digital systems running
on batteries such as note-book computers, cellular
phones and personal digital assistants.
242
Performance Analysis of Linear Phase

Symmetric FIR QMF Bank
Qusay Zalil Kadhim, B.Bhaskara Rao, K.S.Rajasekhar
Dept. Of ECE, Acharya Nagarjuna University, Guntur
Abstract:This paper explores the design and use of

resulting from decimation of sub band signals must
Quadrature-Mirror-Filters (QMF) with linear phase
be negligible. A particularly practical solution to the
Symmetric FIR filters .A QMF bank in which the
aliasing problem is to use Quadrature Mirror
analysis &synthesis filters are FIR is said to be an
Filter(QMF). This is a multirate digital filter structure
FIR QMF bank. Due to the non-ideal
characteristics of the analysis bank filters and
subsequent decimation, the sub band signals suffer that employs two decimators in the signal analysis
from three sources of error; namely aliasing, section and two interpolators in the signal synthesis
amplitude and phase distortion. The underlying section. The low pass and high pass filters in the
design principle is that by judicious selection of the analysis section have impulse responses h0(n) and
linear phase symmetric FIR QMF structures which h1(n) respectively. Similarly the low pass and high
comprise the synthesis bank filters at the receiver, pass filters contain in thesynthesis section have
all three errors can be cancelled, so leading to the impulse responses f0(n) and f1(n) respectively.
property of Perfect Reconstruction (PR) for the
entire system. In this Proposed system Objective The input signal x(n) is first filtered by the filters
function is formulated using perfect reconstruction H0(z) and H1(z). The typical frequency response
condition or Magnitude response condition of characteristics of the filters are shown in the Fig.2
prototype filter at a frequency (w=0.5) in ideal
conditions. Computer numerical simulations shows
that our Proposed system is simple, easy to
implement and linear in nature.
IndexTermsComponent,formatting,style,styling,insert.(key
words)
Keywords-Aliasing, APU, FIR, Filter bank, Quadrature-

Mirror-Filter, Linear phase, MPU, PR system.
I.INTRODUCTION Fig.2.The typical frequency response of H0(z) and

H1(z).
During last two decades Multirate systems and
Filter banks have traditionally played an important Since transition bandwidths of the filters in two-
role in subband coding of speech signals, image channel filter banks are usually larger than those in
processing, source coding, Compression for multichannel filter banks, lengths of the filters in
contemporary communication application and two-channel filter banks are usually shorter than
transmultiplexers [1]. those in multichannel filter banks. Moreover, as
only a single prototype filter is required for the
design of a quadrature mirror filter (QMF) bank and
all other filters are derived from the prototype filter,
the total number of filter coefficients required for
the design of a QMF bank is usually smaller than
those in general filter banks. Furthermore, as the
linear phase property of the filters guarantees no
phase distortion of the filter bank and the FIR
property of the filters guarantees the bounded input
Fig.1.Basic structure of QMF bank bounded output stability of the filter bank, two-
channel linear phase FIR QMF banks find many
Filter design is particularly important in achieving applications in image and video signal processing
good performance in sub band coding. Aliasing Unlike a multichannel QMF bank [2], [3], a two-
243
channel QMF bank could not achieve the exact ( ) = ( ) = ( ( ) + ( ))

perfect reconstruction with the prototype filter
(6)
having very good frequency selectivity [4].
The responses of synthesis filters
This paper presents a new method for the design ( )= ( ) ( )
of QMF bank by using linear phase symmetric FIR (7)
method. The outline of this correspondence is as ( )= ( ) ( )
follows. In section II the definition of aliasing, (8)
imaging and various distortions created by the QMF The reconstructed signal is
bank are reviewed. In section III the proposed two
channel linear phase symmetric FIR QMF bank ( )= ( )+ ( )
problem is discussed. In section IV the analysis of (9)
FIR QMF bank is performed. In section V computer ( )= ( ) ( )+ ( ) ( )
numerical simulations are illustrated. Finally, (10)
conclusions are redrawn in section VI.
= ( ) + ( ) ( ) + ( ( ) +
II. REVIEW OF ERRORS CREATED IN THE ( )) ( )
QMF BANK (11)
The decimated signals are encoded using one of

many possible coding techniques and resulting
signals are actually transmitted. The receiver = ( ) ( )+ ( ) ( )) ( ) +
reconstructs and approximated signal from the
( ) ( ) + ( ) ( )) ( )
encoded signals. The decoding error is the difference
between approximated signal and encoded signal that (12)
represents a nonlinear distortion. This is called sub
band quantization error. If the nonlinear distortion is We know that decimator interpolator is linear
ignored QMF bank still suffers from the three time variant systems, so QMF bank is LTV system,
fundamental errors, i.e. Aliasing distortion, phase now above equations can be written as
distortion and amplitude distortion. The aliasing
distortion is eliminated with the use of suitable ( ) = ( ) ( ) + ( ) ( )
synthesis filters. (13)
Basically, QMF bank is a two-channel filter Where T(z)= ( ) ( )+ ( ) ( )
bank, which consists of analysis filters, down- ( ) ( ) + ( ) ( )
A(z)=
samplers at transmission end, and up-samplers and
synthesis filters at receiving end [5]. The block
diagram of QMF bank is shown in Fig. 1. The The X(-z) originates because of decimation so we
analysis of errors created by the filter banks will be refer this term as the alias term or alias component or
discussed below: aliasing distortion.
The expression for the analysis filter responses The term T(z) is the distortion transfer function
and it can be represented in-terms of time domain as
( )= ( ) ( )
(1) y(n)= ( ) + (1) ( ) ( )
( )= ( ) ( ) (14)
(2) In order to obtain an alias free filter bank A(z) has
The z-Transforms of decimated signals are to be cancelled i.e A(z)=0 then
( )= + ( )= ( ) ( )
(15)
(3)
Then the reconstructed signal in equation (12) can
( )= + be represented as
(4) ( )= ( ) ( )+ ( ) ( )) ( )
Z-Transform of interpolated signal is
(16)
( ) = ( ) = ( ( ) + ( ))
(5)
244
Here aliasing is cancelled and T(z) suffers from ( ) = ( )+ ( )

LTV distortion &it is called as overall transfer (22)
function of alias free system.We know that the QMF ( ) = ( ) ( )
Banks are having analysis filters and synthesis (23)
filters. The analysis filters are related as ( )=
( ).Where ( ) is a good high pass filter and
( ) is a good low pass filter.For real coefficients as
time means ( )= ( ( )
) .To avoid
aliasing distortion by setting ( ) = ( ) Or
( ) = ( ) and ( ) = ( ) Or ( )=
( ).With filters chosen as above the distortion
function is
( ) = ( ( ) ( ))
(17)
IV. PERFORMANCE ANALYSIS OF
( ) = ( ( ) ( ))
PROPOSED SYSTEM
(18)
The distortion transfer function can be A QMF bank in which the analysis and synthesis
represented in frequency domain by putting = filters are FIR is said to be an FIR QMF bank.We
))
know that the FIR filters are having linear phase
( )= ( ) ( (
condition and symmetric conditions. If ( ) has
(19) linear phase, then T(z) is also linear phase, thereby
eliminating phase distortion. Let ( )=
Where ( ) represents amplitude distortion
( ) with ( ) is real.The linear phase
arg ( ( ) ) Represents phase distortion.
constraint requires ( ) = ( ) since
If a QMF bank is free from aliasing,
amplitude distortion and phase distortion is said to be ( ) has to be low pass filter, the only possibility is
a Perfect reconstruction filter [6]-[7] i.e equivalent to ( ) = ( ).
( )= . Where c=constant. From the equation With this choice ( )= / ( )
(15)
where ( ) is a real for all
( )= ( ) Using above equation we get
(20) ( )=
From the time domain representation of perfect ( )
( ) (1) ( ) (24)
reconstruction filter is given by
If N is even then the above expression reduces to
( )= ( ) zero at = , resulting in severe amplitude
(21)
The reconstructed signal y(n) is merely a scaled distortion. So we have to chose N to be odd so that
and delayed version of x(n). ( )= ( ) +
( )
III. DESIGN OF PROPOSED LINEAR PHASE ( ) (25)
FIR QMF BANK But ( )= ( ( )
) so the above
We know that the transfer functions of low
equation can be written as
pass and high pass analysis filters of a two channel
linear phase FIR QMF bank be H0(z) and H1(z) ( )= ( ) + ( )
respectively, and those of the synthesis filters of the
filter bank be F0(z) and F1(z) respectively. (26)
If this condition is not satisfied, then it is
Here H0(z) is a transfer function of the prototype amplitude distortion and termed as reconstruction
filter and impulse response of the prototype filter is error or peak reconstruction error (PRE) is given as
h0(n) . Let the poly phase components of H0(z) be
= 20 log ( ) +
E0(z2) and E1(z2) then poly phase type-I
representation of system[8] is given below. ( ) (27)
245
If the prototype filter is assumed to be ideal in the Figure 4(a) shows the normalized frequency
pass band and stop band, then there is a vsMagnitude plot for a LPF of analysis section. A
reconstruction error in the transition region, which range of -100 to 50 dB values are considered for
can be minimized for improving the performance Magnitude. The plot reflects the performance of the
ofthe filter. This can be achieved by the linear phase system, in terms of Magnitudeversus normalized
analysis filters having power complementary. frequency illustrated. Figure 4(b) shows the
normalized frequency vs Magnitude plot for a HPF of
V.SIMULATION RESULTS AND analysissection.A range of -100 to 200 dB values are
DISCUSSION considered for
Magnitude. The plot reflects the performance of the

The performance of the design filter is evaluated system, in terms of Magnitude versus normalized
in terms of peak reconstruction error and proto type frequency illustrated.
filter has been designed using window technique. The Figure 4(c) shows the normalized frequency vs
simulations are carried out using MATLAB. In order Magnitude plot for a LPF &HPF of analysissection.
to have a fair comparison the performance of the A range of -100 to 200 dB values are considered for
QMF using our method is compared with the design Magnitude. The plot reflects the performance of the
method of Near Perfect Reconstruction QMF. system, in terms of Magnitude versus normalized
(a)amplitude response of LPF H0(z) (b)amplitude response of HPFH1(z) frequency illustrated.Figure 4(d) shows the
50 200
normalized frequency vs Magnitude plot for
amplitude distortion of LPF&HPF of analysissection.
0 0
gain in db
gain in db
A range of -100 to 200 dB values are considered for

Magnitude. The plot reflects the performance of the
-50 -200 system, in terms of Magnitude versus normalized
frequency illustrated.
-100 -400 (a)amplitude response of LPF F0(z) (b)amplitude response of HPF F1(z)
0 0.5 1 0 0.5 1
50 200
normalised frequency normalised frequency
(c)amplitude response of both LPF&HPF (d)amplitude distortion function
200 1.005 0 0
gain in db
gain in db
0 1 -50 -200
Magnitude
Magnitude
-200 0.995 -100 -400

0 0.5 1 0 0.5 1
(c)amplitude response of both LPF &HPF (d)amplitude distortion function
-400 0.99 200 1.005
0 0.5 1 0 0.5 1
/ /
0 1
Magnitude
Magnitude
Fig.4.Magnitude response of the analysis filters

-200 0.995
(a)amplitude response of LPF F0(z) (b)amplitude response of HPF F1(z)
50 200
-400 0.99
0 0.5 1 0 0.5 1
0 0
gain in db
gain in db
/ /
-50 -200
Figure 5(a) shows the normalized frequency vs
-100 -400 Magnitude plot for a LPF of synthesis section. A
0 0.5 1 0 0.5 1
range of -100 to 50 dB values are considered for
(c)amplitude response of both LPF &HPF (d)amplitude distortion function Magnitude. The plot reflects the performance of the
200 1.005
system, in terms of Magnitude versus normalized
frequency illustrated. Figure 5(b) shows the
0 1
Magnitude
Magnitude
normalized frequency vs Magnitude plot for a HPF of

synthesissection.A range of -100 to 200 dB values are
-200 0.995
considered for Magnitude. The plot reflects the
performance of the system, in terms of Magnitude
-400 0.99
0 0.5 1 0 0.5 1 versus normalized frequency illustrated.Figure 5(c)
/ /
shows the normalized frequency vs Magnitude plot
Fig.5.Magnitude response of the synthesis filters for a LPF &HPF of synthesissection. A range of -100
246
to 200 dB values are considered for Magnitude. The Guntur, India.We would like to express our special
plot reflects the performance of the system, in terms thanks of gratitude to the dean of Acharya Nagarjuna
of Magnitude versus normalized frequency University college of engineering & technology
illustrated.Figure 5(d) shows the normalized Prof.P.Siddaiah for his valueble suggestions.
frequency vs Magnitude plot for amplitude distortion
of LPF&HPF of synthesissection. A range of -100 to REFERENCES
200 dB values are considered for Magnitude. The plot
reflects the performance of the system, in terms of [1]. Vaidyanathan, "Multirate Systems and Filter Banks", Prentice
Magnitude versus normalized frequency illustrated. Hall, 1990
[2] Y.-P. Lin and P. P. Vaidyanathan, Linear phase cosine
Figure 6 shows the normalized frequency vs modulated maximally decimated filter banks with perfect
Magnitude plot for a analysis section. A range of -1 reconstruction, in Proc. Int. Symp. Circuits Systems
(ISCAS), May 30Jun. 2, 1994, vol. 2, pp. 1720.
to 1.5 dB values is considered for Magnitude. The
[3] T. Q. Nguyen, Near-perfect-reconstruction pseudo-QMF
plot reflects the performance of the system, in terms banks, IEEE Trans. Signal Process., vol. 42, no. 1, pp. 65
of Peak Reconstruction error 76, Jan. 1994.
peak reconstruction error [4] C.-W.Kok, W.-C. Siu, and Y.-M. Law, Peak constrained least-
1.5
squares QMF banks, Signal Process., vol. 88, pp. 23632371,
2008.
1
[5] J. D. Johnston, A filter family designed for use in quadrature
mirror filter banks, in Proc. Int. Conf. Acoustics, Speech,
Signal Processing (ICASSP), Apr. 1980, vol. 5, pp. 291294.
0.5
Magnitude
[6] P. P. Vaidyanathaq "Quadrature Mirror Filter Banks, M-Band

Extensions and Perf&-ReconstructionTechniques," IEEE
0
ASSP Magazine, July 1987
[7] A. Croisier, D. Esteban, C. Galand, Perfect channel splitting
by use of interpolation /decimation/tree decomposition
-0.5
techniques, International conference on Information
Sciences and Systems, Patras, 1976.
[8].S. K. Mitra, Digital signal processing: A computer based
-1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1 approach,Tata McGraw Hill Edition, 2006.
/ [9].John G Proakis, Digital signal processing:,
Fig.6.Peak error response of analysis filters
VI. CONCLUSION
This correspondence proposes a modified method
for the design of a two channel linear phase
symmetric FIR QMF bank, so that a weighted sum of
the maximum amplitude distortion of the analysis
filter bank and synthesis filter bank are minimized.
The simulation results show the improved
performance with regard to the amplitude distortion
and complexity and also decreases the peak
reconstruction error. It is also observed that the
proposed method is efficiently implemented by using
poly phase type-I representation. If order of the filter
is N then we requires a total multiplications per
unit time (MPU) and additions per unit time
(APU).
ACKNOWLEDGMENTS
This work was carried out at Acharya Nagarjuna

University College Of Engineering & Technology,
247
PAPR Reduction And Channel Estimation In

MIMO-OFDM Systems Using Adaptive Active
Constellation Extension And DWT
N.Renuka1, Ch.Loknadh2, Murali Krishna
1
Associate Professor, Department of Electronics and Communication Engineering, RVR & JC College of Engineering,
Guntur, Andhra Pradesh, India
2
Assistant Professor, Department of Electronics and Communication Engineering, Mallineni Lakshmaiah Womens
Engg. College, Andhra Pradesh, India
channel is available at the transmitter, then the

ABSTRACT:
The major problems in the OFDM systems are peak to OFDM transmitter can adapt its signaling strategy to
average power ratio and spectral efficiency. One of the match the channel [8]. Due to the fact that OFDM
effective methods used for reducing peak to average uses a large collection of narrowly spaced sub
power ratio (PAPR) in multi-input multi-output channels, these adaptive strategies can approach the
orthogonal frequency division multiplexing (MIMO- ideal water pouring capacity of a frequency- selective
OFDM) systems is adaptive active constellation channel. OFDM is a block modulation scheme where
extension algorithm. Wavelets based systems provide
better spectral efficiency because of no cyclic prefix
a block of N information symbols is transmitted in
requirement, have very narrow side lobes and also parallel on N subcarriers [9]. The time duration of an
exhibit improved BER performance. Meanwhile, OFDM symbol is N times larger than that of a single-
channel estimation in wavelet based multicarrier carrier system [7]. In MIMO-OFDM wireless
systems still remains a big challenge. In this work an systems, independent OFDM signals are
investigation into the performance of discrete wavelet simultaneously transmitted from multiple-transmit
transform (DWT) based multicarrier system using zero
antennas to multiple-receive antennas. Therefore,
forcing equalization in time domain is presented. One of
the big advantages is that the side information does not similar to OFDM systems, MIMO-OFDM systems
need to be sent to the receiver. We also evaluating the still suffer an inherent drawback of a high peak-to-
existing PAPR reduction technologies namely partial average power ratio (PAPR) [2].
transmit sequence (PTS) and selective level mapping
(SLM) over the performance of adaptive active II. PAPR IN OFDM
constellation extension algorithm for analysis of PAPR.
The results produced were then compared to that of
conventional FFT-based channel estimation OFDM
Let X (0), X (1) . . . , X (N-1) represents the data
system. The proposed system shows superior BER sequence to be transmitted in an OFDM symbol with
performance. subcarriers. The baseband representation of the
OFDM symbol is given by:
Index Terms: OFDM, Peak to average power ratio N 1 2 nt
1

j
(PAPR), Active constellation extension (ACE), Adaptive, x (t ) X ( n )e N
,0 t T
BER, SNR, Channel estimation, discrete wavelet transform N n0
(DWT)
Where x (t) is OFDM symbol at time t, T is the
E. INTRODUCTION duration of the OFDM symbol [3]. The input
information symbols are assumed to be statistically
Orthogonal frequency division multiplexing (OFDM) independent and identically distributed. According to
is an attractive technique for wireless the central limit theorem, when N is large, both the
communications because it supports robust reliability real and imaginary parts of x (t) becomes Gaussian
and high data rate in the frequency selective fading distribution, each with zero mean and a variance of E
channel environments [3]. Moreover, multiple-input [| x (t) |2] / 2, the amplitude, or modulus, of OFDM
multiple-output (MIMO) is another attractive signal is given by
technology to improve the wireless system capacity. x t R e 2 { x t} I m 2
{ x t}
Therefore, the combination of MIMO and OFDM
(MIMO-OFDM) could exploit the spatial dimension The PAPR of the transmitted signal x(t) is given by
2 2
capability of a wireless communication system to max x(t) max x(t)
improve the wireless link performance and system PAPR(x(t)) 0t NT
0t NT
E x(t)
2 NT
1
capacity by employing multiple antennas at both the
x(t)
2
dt
transmitter and receiver ends. If knowledge of the NT 0
248
Since most practical systems deal with discrete-time by A and the number of iterations, denoted by i. In
signals, instead of reducing the continuous-time peak the first step, the iteration is taken as two i.e., i = 2
max |x(t)|, the maximum amplitude of LN samples of and the initial target clipping level is to be taken as A.
x(t) is reduced, where parameter L denotes the
oversampling factor [1]. To evaluate the PAPR The predetermined clipping level, denoted by A, is
performance from statistical point of view of the related to the target clipping ratio, and given as
OFDM signals, the complementary cdf (CCDF) is A 2
used to describe the probability of exceeding a given = 2
threshold PAPR0, i.e., CCDF(PAPR0) = Pr(PAPR > E { x n }
Where, Target
PAPR0) [4].
Clipping Ratio, A Predetermined Clipping Level, xn
Oversampled OFDM signal The clipping of the
III. PAPR REDUCTION TECHNIQUES
peak signal results to distortion of the original OFDM
signal [3]. The signal obtained after filtering the
Several techniques have been proposed to mitigate
the high PAPR of MIMO-OFDM signals. Clipping is
clipped signal
x (i+ 1 )
=x ,
(i)
+ c (i)
used to reduce the peak power by clipping the OFDM
signals to the threshold level. Selected mapping
(SLM) and partial transmit sequence (PTS) schemes Where,
(i)
c
Anti-Peak Signal at the ith iteration,
generate several alternative signal sequences convergence factor. The Convergence Factor (CF),
representing the same OFDM signal sequence and denoted by can be estimated by
select the one with the minimum PAPR among them. [ c (i)
, c (i)
]
Tone reservation (TR), tone injection (TI), and active
constellation extension (ACE) change constellation c (i)
, c (i)
points for some subcarriers to reduce the PAPR [3].
,Where, convergence Factor, Real Part, C(i)
Among various peak-to-average power ratio (PAPR)
reduction techniques, the active constellation
Peak Signal above the Pre-Determined Level, c (i)
extension (ACE) technique is attractive because ACE
allows the reduction of high-peak signals by ,
Anti-Peak Signal at the ith iteration, Complex
extending some modulation constellation points
Inner Part.
toward the outside of the constellation without any
loss of data rate [4]. The basic principle of clipping- The anti-peak signal at the ith iteration generated for
based ACE (CB-ACE) algorithms involves switching
between the time domain and the frequency domain.
the PAPR reduction, denoted by
(i)
, is given by c
However, the CB-ACE algorithms have a low
clipping ratio problem in that they cannot achieve the
minimum PAPR when the target clipping level is set
c ( i ) = T ( i ) c ( i ) Where,
(i)
Anti-Peak c
below an initially unknown optimum value. Our Signal at the ith iteration, T(i) Transfer Matrix at the
approach combines a clipping-based algorithm with ith iteration, C(i) Peak Signal above the Pre-
an adaptive clipping control, which allows us to find Determined Level. The transfer matrix at the ith
the optimal clipping level [4]. Simulation results iteration, denoted by T (i), used for generating the
show that our proposed algorithm can achieve the anti-peak signal is given
minimum PAPR regardless of the low target clipping
level.
T (i)
= Q * (i)
Q (i)
Where, T(i) Transfer
* (i)
IV. PROPOSED PAPR TECHNIQUE: Matrix at the ith iteration, Q Conjugate of
The main objective of the Adaptive Active (i)
Constellation Order,
Q
Constellation Order
Constellation Extension (Adaptive ACE) algorithm
for reducing the Peak-to-Average Power Ratio [4].
(PAPR) [11], is to control both the clipping level and The original Orthogonal Frequency Division
the convergence factor at each step and thereby Multiplexing (OFDM) signal, denoted by xn, is to be
minimize the peak power signal whichever is greater clipped in order to reduce the peak signals. The
than the initial target clipping level [4]. The Adaptive clipping signal is given by
j
Active Constellation Extension (Adaptive ACE) c (i)
n ( x (i)
n e n
,
x (i)
n
algorithm can be initialized by selecting the

x (i)

parameters namely the target clipping level, denoted n
249
(or larger than) the number of transmit antennas. The

Where,
c (in ) Clipping Sample, A Predetermined
simplest example of an optimal training matrix is to
select P as a (scaled) identity matrix of the same size
x (i) that the number of transmit antennas [8].
Clipping Level, n arg(- n ). The clipping level,
denoted by A, for the next iteration is given by
MMSE estimation:-If the channel and noise
A (i+ 1 ) = A (i) A (i+1) distributions are known, then this a priori information
, Where, A
can be exploited to decrease the estimation error. This
Next Iteration Level, A(i) Present Iteration Level ,
approach is known as Bayesian estimation and for
Rayleigh fading channels it exploits that vec (H) ~CN
Convergence Factor, A Gradient with respect
to A. The gradient with respect to the target clipping (0, R), vec (N) ~CN (0, S). The MMSE estimator is
A the Bayesian counterpart to the least-square estimator
ratio, denoted by ,is given and becomes
(i+ 1 ) Vec (HMMSE-estimate) = (R^1+ (P^TX1) ^H.S^-

c n 1(P^TX1)) ^-1(P^TX1) ^H.S^-1Vec(Y)
Where X denotes the Kronecker
n I 1( i ) I 3( i )
A product and the identity matrix I has the dimension of
the number of receive antennas [9]. The estimation
N p Mean Square Error (MSE) is tr (R^-1+ (P^T X I)
Where,

A Gradient with respect to A, N
^Hs^-1(P^T X I)) ^-1 and is minimized by a training
p matrix P that in general can only be derived through
Number of peak samples larger than A. numerical optimization. But there exist heuristic
solutions with good performance based on water
The Peak-to-Average Power Ratio by the Adaptive filling. As opposed to least-square estimation, the
Active Constellation Extension (Adaptive ACE) [8] , estimation error for spatially correlated channels can
algorithm is to be calculated for the Orthogonal be minimized even if N is smaller than the number of
Frequency Division Multiplexing (OFDM) signal transmit antennas. Thus, MMSE estimation can both
which is obtained after filtering the clipped signal [3]. decrease the estimation error and shorten the required
The clipped and filtered OFDM signal obtained by training sequence [7]. It needs however additionally
using Adaptive Active Constellation Extension the knowledge of the channel correlation
(Adaptive ACE) algorithm is to be transmitted via an matrix R and noise correlation matrix S. In absence
Additive White Gaussian Noise (AWGN) channel, in of an accurate knowledge of these correlation
order to calculate the Signal-to-Noise Ratio (SNR) matrices, robust choices need to be made to avoid
and Bit Error Rate (BER). MSE degradation.
V. CHANNEL ESTIMATION: MMLS Channel Estimation

LS algorithm is the simplest channel estimation.
It is assumed that the system has NT transmitter It is assumed that H^ls is the estimate of the channel
antennas and NR receiver antennas. The total number impulse response H .The LS estimate of the channel
of the subcarriers is N. At the sending end, the data in frequency domain on subcarrier k can be obtained
stream is modulated by inverse wave let transform as:
(IWT) [2] and a guard interval is added for every
OFDM symbol to eliminate ISI caused by multi-path H^ls(k) = Y(k)/X(k) = H(k)+(W(k) / X(k)); 0<= k
fading channel. The receiver performs opposite <=N-1 Where, H(k)- Channel at pilot subcarriers,
operations [7]. X(k)-Input at the kth pilot subcarrier, Y(k)-Output at
the kth pilot subcarrier [6]
Least-square estimation:-If the channel and noise
distributions are unknown, then the least- VI. RESULTS AND DISCUSSIONS:
square estimator (also known as the minimum-
variance unbiased estimator) is The Peak-to-Average Power Ratio (PAPR)
of the Orthogonal Frequency Division Multiplexing
where (OFDM) system is equal to 11.8 dB (approximately
denotes the conjugate transpose. The 12 dB) without using any algorithm i.e., by using the
estimation Mean Square Error (MSE) is proportional basic formula of PAPR for the original OFDM signal.
to Tr(PP^H)^-1 where denotes the trace. The The proposed method i.e., the Adaptive Active
error is minimized when PP^H is a scaled identity Constellation Extension (Adaptive ACE) algorithm
matrix. This can only be achieved when N is equal to reduced the Peak-to-Average Power Ratio (PAPR) to
250
4.4 dB for all the target clipping ratios, which says

that the low target clipping ratio problem is
eliminated completely.
In a noisy channel, the BER is often expressed as a

the process of reducing the PAPR by using the
function of the normalized carrier-to-noise
Clipping-Based Active Constellation Extension (CB-
ratio measure denoted Eb/N0, (energy per bit to noise
ACE), [8] Algorithm. The improved BER transmits
power spectral density ratio), or Es/N0 (energy per
the data via a transmission channel with fewer errors,
modulation symbol to noise spectral density).
while the minimized OBI reduces the effects caused
by clipping. Hence, by reducing the Peak-to-Average
Power Ratio (PAPR), the complexity of the Analog-
to-Digital Converter (ADC) and Digital-to-Analog
Converter (DAC) can be reduced. The reduced Peak-
to-Average Power Ratio (PAPR) also increases the
efficiency of the Power Amplifiers. In case of DWT
compressed signal transmission over OFDM channels
where binary channel state information is available at
the transmitter, but retransmission is not allowed [6] .
Where we are arranged in descending order of
priority and mapped over the channels starting with
the good ones. The coefficients with lower
importance level, which are likely mapped over the
bad channels, are discarded at the transmitter to save
power without significant loss of reception quality.
References:
Packets are distributed in time and frequency domain
as described before, but the packets which are [1]. B. S. Krongold and D. L. Jones,PAR reduction in OFDM via active
constellation extension, IEEE Trans. Broadcast., vol. 49, no. 3, pp. 258
transmitted back to back through same group of 4 sub 268, Sep. 2003.
channel are corrupted due to slowly time varying [2]. L. Wang and C. Tellambura, An adaptive-scaling algorithm for
OFDM PAR reduction using active constellation extension, in Proc. IEEE
nature of fading. We maintain coherence time to be Veh.
more than that of the packet transmission time Technology Conf., Sep. 2006, pp. 15.
through a sub channel, and the channel condition is [3]. E. Van der Ouderaa, J. Schoukens, and J. Renneboog, Peak factor
minimization using a time-frequency domain swapping algorithm, IEEE
fed back for each packet. We simulated block fading Trans. Instrum. Meas., vol. 37, no. 1, pp. 145147, Mar. 1988.
channel with number of sub-bands N = 4 and the [4]. M. Friese, On the degradation of OFDM-signals due to peak-clipping
in optimally predistorted power amplifiers, in Proc. IEEE Globecom, Nov.
coherence bandwidth equivalent to 16 subcarriers (M 1998, pp. 939944.
= 16). QAM is used as modulation scheme. Thus, 128 [5]. J.J. van der Beek, 0. Edfors, M. Sandell, S.K. Wilson, and P. 0.
Borpesson, "On Channel estimation in OFDM systems," Proc. VTC'95, pp.
2 bits per OFDM symbol is transmitted through a 815-819.
sub channel. [6]. V. Kumbasar, O. Kucur, Alamouti Coded Wavelet Based OFDM for
Multipath Fading Channels, Wireless Telecommunications Symposium,
2009.
VII. CONCLUSIONS: WTS 2009, pp. 1 5.
[7]. B. G. Negash and H. Nikookar. Wavelet-Based Multicarrier
Transmission Over Multipath Wireless Channels, IEE Electronics Letters,
The Active Constellation Extension (ACE) vol. 36, (2000) October, pp. 1787-1788.
[8]. K. Abdullah and Z. M. Hussain, Studies on DWT-OFDM and FFT-
Algorithm provides the minimum Peak-to-Average OFDM Systems, IEEE International Conference on Communication,
Power Ratio (PAPR), even when the initial target Computer and Power, (2009) February 15-18.
[9]. W. Saad, N. El-Fishawy, S. EL-Rabaie and M. Shokair, An Efficient
clipping ratio is set below the unknown optimum Technique for OFDM System Using Discrete Wavelet Transform,
clipping point. Hence, the proposed algorithm avoids Springer-Verlag Berlin Heidelberg, (2010), pp. 533541.
the problem of low clipping ratio, which is caused in
251
Adaptive Channel Equalizer For Wireless

Communication Systems
S.Dileep kumar, G B S R Naidu, V Jagan Naveen
symbol. This is referred as inter symbol interference
Abstract The Data rates and spectrum efficiency
of Wireless Mobile Communication have been (ISI). Other factors like thermal noise, impulse noise,
significantly improved over the last decade or so. and cross talk cause further distortions to the received
Recently, the advanced systems such as 3GPP
symbols. Signal processing techniques used at the
LTE and terrestrial digital TV broadcasting have
receiver to overcome these interferences, so as to
been sophisticatedly developed using OFDM and
restore the transmitted symbols and recover their
CDMA technology.
information, are referred to as channel equalization
In general, most mobile communication systems or simply equalization. In principle, if the
transmit bits of information in the radio space to characteristics of the channel are precisely known, then
receiver. The radio channels in mobile radio it is always possible to design a pair of transmitting
systems are usually multipath fading channels, and receiving filter that can minimize the effect of ISI
which cause inter symbol interference (ISI) in and the additive noise. However, in general the
the received signal. To remove ISI from the signal characteristics of channel are random in the sense that
there is a need of strong equalizer which required it is one of an ensemble of possible channels.
the knowledge on the channel impulse response. Therefore, the use of a fixed pair of transmitting and
(LMS)Least Mean Square, (RLS) Recursive Least
squares and (PSO) Particle swarm optimization receiving filter designed on the basis of average
algorithms are used to implement the adaptive channel characteristics, may not adequately reduce
channel equalizer. The results are measured in inter symbol interference. To overcome this problem
terms of mean square error (MSE) and bit error adaptive equalization is widely used, which provides
rate (BER) Vs the number of iterations. precise control over the time response of the channel.
Adaptive equalizers have therefore been playing a
Index Terms LMS, RLS, PSO, MSE, BER crucial role in the design of high-speed
communication systems. The data transmitted through
1. INTRODUCTION a band limited communication channel suffers from
linear, nonlinear and additive distortions. In order to
The growth in communication services during the reduce the effects of these distortions an equalizer
past five decades has been phenomenal. Satellite and is used at the receiver end. The function of the
fibre optic networks provide high-speed equalizer is to reconstruct the transmitted symbols by
communication services around the world. Currently, observing the received noisy signal[1,2,7]. Section
most of the wired line communication systems are 2 describes the system model,section3 explains the
being replaced by fibre optic cables which provide LMS, RLS, and PSO algorithms, section 4 and 5
extremely high bandwidth and make possible the explains the simulation results and conclusion.
transmission of a wide variety of information sources,
including voice, data, and video. With the 2. SYSTEM MODEL
unimaginable development of Internet technologies,
efficient high-speed data transmission techniques over
communication channels have become a necessity of
the day. As the rate of the data transmission increases
to fulfil the needs of the users, the channel introduces
distortions in data. One major cause of distortion is
Inter Symbol Interference (ISI). In digital
communication, the transmitted signals are generally
in the form of multilevel rectangular pulses. The
absolute bandwidth of multilevel rectangular pulses is
infinity. If these pulses passes through a band limited
communication channel, they will spread in time and
the pulse for each symbol maybe smeared into
adjacent time slot and interfere with the adjacent Fig.1:Block diagram of channel equalizationIn an
252
ideal communication channel, the received information approximately satisfied.

is identical to that transmitted. However, this is not the 3.ALGORITHM IMPLEMENTATIONS
case for real communication channels, where signal
distortions take place.
3.1 LEAST MEAN SQUARE
A channel can interfere with the transmitted data (LMS) ALGORITHM
through three types of distorting effects: power
degradation and fades, multi-path time dispersions Algorithm for LMS:
and background thermal noise. Equalization is the
process of recovering the data sequence from the (i) Read the data signal.
corrupted channel samples. A typical base band
transmission system is depicted in Fig.1, where an
(ii) Calculate length of the data signal.
equalizer is incorporated within the receiver. (iii)
Generate random noise and add it to data
Figure.1 shows a block diagram of a
communication system with an adaptive equalizer in
the receiver. If b (t) is the (v) Filter output is y(n)=w(n)x(n) Where x(n) is
noise data signal.
where f* (t) is the complex conjugate of f(t) ,
is the baseband noise at the input of the equalizer, and Error e(n)=d(n)-y(n) Where d(n) is part of
(vi)
0 denotes the convolution operation. If the
original data signal
impulseresponse of the equalizer is
, then the output of the equalizer is (vii) Filter coefficient updating:
(viii) Update filter coefficient and

where g(t) is the combined impulse response of minimize the error[4]
the transmitter, channel, RF/IF sections of the
receiver, and the equalizer. The complex baseband 3.2Recursive Least Squares (RLS) Algorithm
impulse response of a transversal filter equalizer is
given by
ALGORITHM FOR RLS
(i) Read the data signal.
(ii) Calculate length of the data signal.
where are the complex filter coefficients of the (iii) Generate random noise and add it to data signal.
equalizer. The desired output of the equalizer is x(t),
(iv) Apply RLS algorithm on noisy data signal
the original source data. , Then, in order
to force d (t) = x (t) in equation (2.2) must be equal (v) Initial weight vector w(n)=0
to
(vi) Error e(n)=d(n)-y(n) Where d(n) is part
of original data signal
The goal of equalization is to satisfy equation (vii) Filter coefficient updating h(n)=h(n-
(2.4). In the frequency domain, equation (2.4) can be
expressed as 1)+k(n)e*(n) (viii) Update filter coefficient and
minimize the error [4]
3.3PARTICLE SWARM OPTIMIZATION (PSO)
Particle swarm optimization algorithm:
In PSO a swarm consists of a set of volume-less

particles (a point) moving in a D dimensional search
response and linear phase response. For a time- space, each representing s potential solution. Each
varying channel, an adaptive equalizer is designed to particle flies in the search space with position and
track the channel variations so that equation (2.5) is velocity which are dynamically adjusted according to
253
its own as well as its companions flying experiences. i = 1,2 ....N , d = 1,2 ....D and N is the number of
Particles are assumed as tap weights in the adaptive particles . The symbols r1 and r2 represent random
channel equalizer. Initially tap weights are randomly numbers between 0 and 1. Similarly c1 and c2 denote
selected. The tap weights are continuously updated acceleration constants that pull each particle towards
by taking number of iterations. The continuous its best and global best positions. The acceleration
updation depends on position and velocity of the constants are usually taken as 2.05 for most
particles. Initial position error is maximum. For each applications.
iteration, position best(pbest) and global best(gbest) is
calculated. The concept of movement of particles of The inertia weight, W is employed to control the
PSO described in Fig. 2 impact of pervious history of velocities on the current
one in order for tradeoff between the global and local
exploitations. At early stage of optimization it is
desirable that the individual particles wonder though
the entire search space, without clustering around the
local optima. On the other a hand, during later stages
it is very important to enhance convergence
toward the global optima so as to find optimum
solution efficiently .
Large inertia weight enables the PSO to explore

locally. So a self-adaptive strategy is introduced such
that the value of w is decreased linearly as the
generation goes on increasing .The time-varying
inertia weight is given by
Fig 2. Procedure of particle swarm optimization
where k is the search space number, wi and w f
are the initial and final value of inertia weights taken
The ith particles is represented values 0.4 and 0.9 respectively. I is the maximum
by a vector: number of search or generation.
, the best previous position (the position giving
the best fitness value) of the ith particle is 4.SIMULATION RESULTS
recorded and represented as P = [pi1,pi2 4.1Simulation results for channel equalization
....pid..piD]. At each iteration, the global at using LMS
best particle in the swarm is represented by
P = [pg1,pg2 ....pgd ..pgD]. The rate of algorithm:
change of position of the ith particle is represented
as
V= [vi1,vi2 ....vid ..viD]. [3,5]
The maximum velocity and the range of particles
are given by
V max = [vmax1, vmax2 ......vmaxd .......vmaxD ]
and
X max = [xmax1, xmax2 .....xmaxd ......xmaxD ].
Each particle tries to modify its position using
following information
The current positions,
The current velocities,
The distance between
the current position
The distance between the current position
and the gbest.
The velocity and position of the dth element of
the ith particle at (k+1)th search from the
knowledge of previous search are modified as per the
following
Vid(k+1)=w(k)*vid(k)+c1*r1*(pid(k)- xid(k))+
c2*r2*(pgd(k +xid(k))
254
MSE is minimized by taking more number of iterations

using LMS algorithm.
Fig.4.2(b):Bit error rate plot using RLS

Fig.4.1(b):Bit error rate plot using LMS algorithm
algorithm
MSE, BER are reduced by increasing the number
BER is minimized by taking more number of of iterations when compare with LMS, RLS shows
iterations using LMS algorithm better performance in reducing MSE, BER.
Table.1:Tabular Form for different using LMS

Algorithm
Avg.MSE is depends on the value in the

RLS algorithm.
Avg.MSE is depends on the value of in the
LMS algorithm. 4.3 Simulation results for channel equalization
4.2.Simulation results for channel equalization using PSO algorithm:
using RLS algorithm
Fig.4.2(a): Error plot obtained using RLS .4.3(a): Error plot obtained using PSO algorithm
Algorithm
MSE is calculated using RLS algorithm for the MSE is reducing by increasing the number
forgetting factor is 0.998 of iterations.
255
Table 4:Tabular Form in comparison of all the three

algorithms for Channel Equalization:
.
5. CONCLUSION
The main problem that is being addressed in this

project is related the most important issues associated
Fig.4.3(b): Bit error rate plot using PSO with the wireless communication channels, namely
algorithm Inter-Symbol Interference (ISI) caused by multipath
nature. ISI is causing data errors and hence the main
BER is reduced to zero by increasing the number of obstacle in the way of reliable and fast wireless digital
iterations. communications. The common methods to eliminate or
sufficiently mitigate the effect of distortion caused by
ISI are collectively known as channel equalization.
Equalization techniques are of enormous importance in
the design of high data rate wireless systems. They can
combat for inter symbol interference even in mobile
fading Channel with high efficiency. In this project
performance of adaptive channel equalizer with three
PSO shows better performance than RLS. BER is algorithms used to reduce ISI is presented. The analysis
almost reduced to zero with the same number of showed that all the three algorithms are successful in
iterations taken in both the algorithms. reducing ISI to some extent. The Results for all the
three algorithm are compared with the help of three
4.4Comparison of three algorithms: parameters they are Mean Square Error (MSE), Bit
Error Rate (BER) and Time of Convergence (TOC).
It was concluded that the Particle Swarm
Optimization (PSO) offers two advantages than LMS
and RLS algorithms, first one in order to reduce the
MSE, however the second one to reduce the TOC.
REFERENCES
[1] MK Ozdemir, H Arslan, Channel estimation for wireless
OFDM systems. IEEE Commun. Surv. Tutor. 9(2),
1848 (2007)
[2] R Negi, J Cioffi, Pilot tone selection for channel estimation in a
mobile OFDM system. IEEE Trans. Consum. Electron. 44,
11221128 (1998)
[3] L.N.de Castro,J.Timmis,An Introduction to artificial immune
systems:A New Computational Intellegence Paradigm, springer-
verlag, 2002.
[4] B.Widrow and S.D.Stearns,Adaptive Signal
Processing,Second Edition,pearson,2002.
Fig.4.4.Comparison in mean square error plot [5] S.Haykins,Neural Networks:A Comprehensive
of all three algorithms Foundation,Second Edition, Pearson, 2002.
[6] Widrow, B, Walach, E., Adaptive Inverse Control: A Signal
Processing Approach, Wiley-IEEE Press, Nov. 2007.
[7] Qureshi, S.,Adaptive Equalization, IEEE
Communications Magazine, March 1992, pp. 916.
[8] Proakis, J.G., Adaptive Equalization for TDMA Digital Mobile
Radio, IEEE Transactions on Vehicular Technology, Volume 40, No. 2,
May 1991.M. Young, The Technical Writer's Handbook. Mill Valley, CA:
UniversityScience, 1989.
256
Generation of Sequences with Good Correlation

Properties Using Modified Particle Swarm
Optimization
S.Srinivasa Rao1, Dr.P.Siddaiah2 ,K. Raghu3
codes is equal to the length of the code.

Abstract Radar applications require sequences
Unfortunately, the Barker sequences for lengths
with individually peaky autocorrelation. Obtaining
greater than 13 have not been found. In view of the
such sequences is a combinatorial problem. Thus
fact that the discrimination is the figure of merit of
the problem of signal design referred to above is a
the code, many researchers have been start working
challenging problem for which many global
with different ways of design of binary sequences
optimization algorithms like genetic algorithm,
with discrimination greater than 13.
particle swarm optimization algorithm, simulated
annealing, tunneling algorithm were reported in
II.BINARY PHASE CODES
the literature. The paper aims at the design of
optimal set of Binary Sequences using Modified
The performance of a sequence is optimal, if it has
Particle Swarm Optimization Algorithm which
peaky main lobe energy for zero shifts and zero
makes use of Hamming Scan Algorithm for
sidelobe energy for other shifts. Usual criteria of
Mutation. The main advantage of HSA Algorithm
goodness of pulse- compression sequences are
is it enhances the search space of PSO, thereby
evaluated by different methods such as Merit
preventing the local optimum trapping of PSO.
factor, Discrimination factor, Energy efficiency
The synthesized Binary Sequences using MPSO
and Quality factor .
have better autocorrelation properties than
Let S = [x0, x1, x2, x3 ..xN-1] be a real
standard PSO.
sequence of length N, with values of 1 and -1.
KeywordsAuto-Correlation, Optimization,
Modified Particle Swarm Optimization (MPSO),
Discrimination Factor (D), Hamming Scan Where k=0,1,2, _ _ _,n-1 is its aperiodic auto
Algorithm (HSA). correlation
The Discrimination factor is defined as the ratio of
I. INTRODUCTION amplitude of main peak of the auto correlation
to the absolute maximum amplitude in the side
Pulse compression is a method of breaking the lobes.[9].
unwanted constraint between range and
resolution [6]. Pulse compression radar transmits a
frequency or phase modulated pulse, which is both
long (having good range characteristics) and
wideband (having good range resolution criterion).
Discrimination is used to measure weather a signal
The received echo signal is processed in a
is Good or Poor. This means that a code with high
matched filter that compresses long pulse to a
discrimination is a good code while a code with
short pulse that separates closely spaced targets. low discrimination is poor code. The discrimination
However, this separation is achieved at the cost of measures how the main lobe signal is different form
sidelobes, which hide weak targets. Therefore, in the peak side lobe level. Thus it gives the measure
radar applications the pulse compression sequences of goodness of the given coded waveform.
with high discrimination factor are desired to
achieve high range resolution. Range resolution is
the ability of the radar receiver to identify closely III. PSO ALGORITHM
spaced targets. The term Discrimination (D) is The Particle Swarm Optimization algorithm is a
defined as the ratio of main peak in the auto- novel population-based stochastic search algorithm
correlation to absolute maximum amplitude among and an alternative solution to the complex non-
side lobes, Moharir [3]. In this context, Barker [4] linear optimization problem. This algorithm is
proposed binary sequences with sidelobe levels either capable of optimizing a non-linear and
zero or unity. The discrimination of the Barker multidimensional problem which usually reaches
257
good solutions efficiently while requiring minimal connected swarm, meaning that all the particles share
parameterization. The PSO algorithm was first information, any particle knows what is the best
introduced by Dr. Kennedy and Dr. Eberhart in 1995 position ever visited by any particle in the swarm
and its basic idea was originally inspired by
simulation of the social behavior of animals such as The basic steps of MPSO algorithm for the
bird flocking, fish schooling and so on [1]. These design of binary sequences is as follows
roots in natural processes of swarms lead to the Step1: (Initialization): Generate initial particles by
categorization of the algorithm as one of Swarm randomly generating the position and velocity for
Intelligence and Artificial Life. It is based on the each particle.
natural process of group communication to share Step 2: Evaluate each particles cost (E), which is
individual knowledge when a group of birds or amplitude of main peak of the auto correlation
insects search food or migrate and so forth in a to the absolute maximum amplitude in the side
searching space, although all birds or insects do not lobes.
know where the best position is. But from the nature Step 3: For each particle, if the cost (E) is less than
of the social behavior, if any member can find its previous best (Pbest) fitness, update Pbest.
out a desirable path to go, the rest of the members Step 4: For each particle, if the cost (E) is less
will follow quickly. Researchers shown that although than the best one (Gbest) of all the particles, update
PSO finds solutions much faster than most of the Gbest.
contemporary search techniques like Evolutionary Step5: For each
and Genetic Algorithm, it usually do not improve the particle
quality of solutions as the number of iterations a) Generate a new particle according
increase and thus becomes a victim of premature to the following formulae
convergence resulting in a suboptimal solution. This
drawback of PSO is due to the lack of diversity,
which forces the swarm particles to converge to
the local optimum.
IV. HAMMING SCAN ALGORITHM

where V i [ t ] and V i [ t+1] are the previous
One of the methods for maintaining the diversity of th
the population is inclusion of the concept of and the current velocity of the i particle. Here P
th
mutation. The mutation is a term metaphorically used i [t] represents the position of the i particle. 0
for a change in an element in the sequence. In the W 1 is an inertia weight which determines how
present work we have used HSA for mutation. HSA much the previous velocity is preserved (chosen =
searches in the neighbourhood of the point in all 0.99). This signifies that the previous velocity is
directions. In HSA, each element of the sequence is almost preserved but not preserved completely to
mutated with all other possible elements in the avoid escaping from the optimum value, C 1 and C 2
sequence. For example the code element 1 of are the accelerating constants assigned a random
sequence X is mutated by -1, and cost for the value picked between 0 to1 from an uniform
mutated element is calculated. If the cost is reduced distribution, rand() picks a value from a uniform
after mutation then the new element is accepted else probability distribution.
the original element is retained. This process is
recursively applied to all elements in the
sequence. Thus, HSA performs search among all b) Generate a new particle P1 by
the Hamming-1 neighbours of the sequence and mutating the elements in P with HSA.
selects the one whose cost function value is minimum.
c) Compare P with P1 and select the one
V. WORKING OF MPSO ALGORITHM with low cost value.
In MPSO each particle keeps track of its Step6: If the stop criterion is satisfied, then stop, else
coordinates in the solution space which are associated go to Step 3.
with the best solution (fitness) that has achieved so
far by that particle. This value is called personal best, VI. IMPLEMENTATION OF MPSO
Pbest. Another best value that is tracked by the PSO ALGORITHM
is the best value obtained so far by any particle in
the neighbourhood of that particle. This value is Binary codes having good auto-correlation
called Gbest. In this algorithm we have a completely properties are synthesized by using Particle swarm
258
optimizationalgorithm through a MATLAB

program. A MATLAB program is designed to
perform autocorrelation on each sequence. The
output of the autocorrelation function gives peak
lobes and sidelobes.
Fig 1: Matlab result for 13-bit Binary Phase

Sequence
259
are widely used in radar and spread spectrum

communications for improving system performance.
The results shown in Table.1 indicates that the
proposed algorithm (MPSO) outperforms the
standard PSO. Unlike in genetic algorithms, in PSO,
there is no selection operation which increases the
speed and reduces complexity of the algorithm.
As the sequences length increases the genetic
algorithm consumes more time. Hence global
optimization techniques such as PSO algorithm are
used for the generation of sequences with good
correlation properties.
REFERENCES
[1] J.KENNEDY and R. EBERHART, Particle swarm

optimization, in Proc. IEEE Int. Conf. Neural logical
Networks, vol. IV, Perth, ustralia, 1995, pp. 1942
press.1948.
[2] J. M. Baden, and M. N. Cohen (1 990), Optimal peak
sidelobe filters for biphase pulse compression,Proceedings
of 1990 IEEE International Radar Conference, 1990, pp
249-252
[3] P.S.Moharir, Signal Design, Int J electron, vol 41pp381-398,
1976. [4] Barker, R.H. (1953) Group synchronizing of
binary digital systems.In W.Jackson (Ed.),Communication
Theory. London: Butterworth, 1953.
[5] Anna Dzvonkovskaya*, Hermann Rohling Long Binary
Phase Codes with Good Autocorrelation
Mach.Properties.ieeexplore.ieee.org/iel5/4577717/4585695/
045857 50. pdf?. .
[6] Merrill I Skolnik, Introduction to Radar Systems, 2nd
edition,2008.
[7] Nadav Levanon, Eli Mozeson, Radar Signals, 1.st Editon
Wilet- Interscience,2004.
[8] Xinmin Deng and Pingzhi Fan, New Binary Sequences
with Good aperiodic Autocorrelations Obtained by
Evolutionary Algorithm, IEEE Communications Letters,
Vol. 3, No. 10.
[9] Golay. M.J.E., The merit factor of long low autocorrelation
binary sequences, IEEE Trans. on Inform. Theory, IT-
28,1982, pp 543-
549.
[10] Changhe Li at.el, A Fast Particle Optimization Algorithm
with Cauchy Mutation and Natural Selection Strategy.
Proceedings of the 2nd international conference Advances
in computation and Intelligence 2007, Wuhan, China,
pp.334-343.
[11] Lindner, J. (1975) Binary sequences up to length 40
with best possible Autocorrelation functions. Letters,
11(1975), 507.
V. CONCLUSION
The objective in this paper was mainly to

demonstrate the signicance of the MPSO
algorithm in the generation of binary sequences
with good discrimination values. These sequences
260
Hartley transform-based block FSLMS

algorithm for active noise control
T.Srinivasa Rao
because of following reasons:
AbstractIn general, the noise generated from a 1) The propagation path may contain nonlinearity
dynamical system is nonlinear. The Filtered-S because of high air pressure [3].
least mean square (FSLMS) and Volterra
filtered-X least mean square (VFXLMS) algo- 2) Because of nonlinear elements such as
rithms are proved to be the most effective loudspeaker, sensors, actuators and amplifiers present
algorithms for suppressing nonlinear noise in the ANC system [4].
process. The time-domain implementation of 3) secondary path may have non minimum
these algorithms have a very high computational phase [3].
complexity because of its complex structure. This
computational complexity increases rapidly for
In any of those cases, the noise is nonlinear and
multichannel case . There is a strong need and
so, there is a need for efficient algorithm to
scope to reduce the complexities of these
suppress the nonlinear noise process. Das and
algorithms.
Panda [4] developed Filtered-S LMS (FSLMS)
In this paper, we proposed Hartley transform- algorithm by expanding non linearity of the
noise using trigonometric basis functions. Tan
based Block FSLMS (HBFSLMS-M) algorithm
and Jiang [3] proposed Volterra filtered-X LMS
for multichannel system. The fast Hartley
(VFXLMS) algorithm modeling the non linearity
transform (FHT) is used in the algorithm to
using Volterra series. These two algorithms are
reduce computational burden. The proposed
successful in mitigating nonlinear noise process.
HBFSLMS-M have less computational complexity
However it is proved that FSLMS algorithm
compared to the standard time- domain and
performs better than VFXLMS algorithm [4], [5].
frequency domain based FSLMS-M algorithms
The computational complexity of these two
with the matching performance. The detailed
algorithms is very high which increases with the
computational complexity analysis is done to
order of structure making the algorithm difficult to
verify the performance. Simula- tions are done to
operate in real time. Prashant Reddy et al [6], [7]
show the performance equivalence graphically.
proposed a method to reduce complexity of FSLMS
Single channel HBFSLMS (HBFSLMS-S)
and VFXLMS algorithms by re-utilizing a part
algorithm also studied and the corresponding
of the computation in calculating the next
simulations are presented.
sample.Vikram Kumar et al [8] implemented
FSLMS algorithm in the frequency-domain for the
I. INT RODUCT multichannel case which reduces the complexity
ION further.
Mehar et al [9] developed a faster method to compute

Living in a world without machines is impossible to discrete Hartley transform (DHT). Because of fast
imagine. We are always surrounded by machines
implementation of DHT, it is easier to compute
such as fans, air conditioners, transformers,
convolution sum in the Hartley-domain. The
engines, motors etc. All these machines make noise.
authors of [10] have proved that fast Hartley
The passive noise control techniques used to cancel
transform (FHT)-based FXLMS algorithm has less
these noises are neither effective nor practical [1],
[2]. Active noise control (ANC) is the only efficient computational complexity than the FFT-based
technique for reducing or canceling these types of FXLMS and time-domain FXLMS algorithms. In
noises, since they are cheap and adaptive. ANC this paper, we have developed FSLMS-M
works on the principle of destructive interference. It algorithm in the Hartley-domain using FHT. The
electronically generates an acoustic signal which computational analysis shows that the proposed
combines with noise and forms a noise free zone [1], HBFSLMS-M algorithm saves computations of
[2]. Filtered-X least mean square (FXLMS) algorithm the order of 98 % (when the block size, N
[2] is the most popular ANC algorithm for =1008) with respect to time-domain FSLMS-M
suppressing noise. When the noise is nonlinear it is algorithm. The organization of this paper is as
found to be ineffective [3].The noise is nonlinear follows: In Section II a brief review of FSLMS-M
261
algorithm is presented in the time-domain. Section

III covers the implementation of HBFSLMS-M
algorithm with the necessary mathematical
equations. Section IV discusses the detailed
computational analysis. Section V demonstrates
the simulation results of all the models considered.
Finally, conclusions are presented in Section V1.
The schematic diagram of a typical multichannel
ANC set up for the (I , J, K ) configuration is
shown in Fig. 1. In (I, J, K) tuple I , J , K
represents the number of reference mikes,
loudspeakers and error mikes respectively.
vectors for one reference input.
II. TIME -DOMAIN IMPL E ME NTAT

ION OF FSLMS-M Al GORIT HM
The filter-bank implementation of FSLMS

algorithm is shown in Fig. 2. In Fig. 2 the
functional link artificial neural network (FLANN)
nonlinearly expands the input noise xi
(n) using trigonometric basis as given below
262
263
264
265
High Bit-Rate X-Band QPSK

Modulator Realization and Analysis
Jyothi B R*, K R Yogesh Prasad, Priyanka Das, Chitra Ramamurthy
ISRO Satellite Centre, Vimanapura, Bangalore: 560017, India.
(*Corresponding author: jyothibr@isac.gov.in, phone: 91-080-25083338, fax: 91-080-25083343)
Fig. 1. In-phase and Quadrature carriers are

AbstractRealization of an X-band QPSK
Modulator supporting bit-rates up to 500 Mbps using
independently subjected to BPSK modulation by I and Q
I-Q modulator and I- Q up-converting mixer is data streams respectively and are then combined to yield
reported. Effect of skew between I and Q data streams a QPSK modulated signal. By choosing multi-level I and
on the spectrum of the modulated signal has been Q data streams, higher order modulations can be
analyzed. Asymmetry in the QPSK spectrum as a implemented
function of I-Q skew has been quantified. Concept of
I-Q mixer for frequency up-conversion or down-
conversion has been explained and implemented. Effect
of phase error in quadrature splitter of I-Q mixer on
the suppression of the unwanted frequency band has
been studied.
Index TermsQPSK Modulator, I-Q modulator, I-Q

mixer, Up-conversion of modulated signal.
I.
INTRODUC
TION
Fig.1 Block Schematic
PSK modulation is widely used in satellite of I-Q Modulator
data transmission taking into consideration the
spectral efficiency offered by the scheme at a given The data rate of a QPSK modulator is twice the
transmitted power. With the amount of data to be data rate of I and Q data streams. The
transmitted steadily going up, the demand on the data spectrum of a typical QPSK modulated signal
rates to be supported by the modulator is increasing. centered at 2.0 GHz modulated at data rate of
For Low Earth Orbit (LEO) satellites, the 500 Mbps is shown in Fig. 2. In this simulation,
frequency band allotted for payload data transmission PRBS data has been considered for modulation.
lies in the X-band. Realization of a high bit-rate (up to When an alternating sequence of 1 and 0 is
500 Mbps) X- band QPSK modulator and related considered for modulation, the spectrum of the
analysis is presented in this paper. Section II of the QPSK modulated signal would be as shown in Fig.
paper describes the operation of I-Q modulator and I-Q 3. Suppression of power in the carrier and
mixer. Generation of X-band QPSK modulator using symmetry in the amplitude of the frequency
these components is discussed and simulation results components on either side of the suppressed
are presented. Effect of I-Q data skew on QPSK carrier may be noted in this figure.
spectrum asymmetry is discussed in Section III.
Section IV discusses the effect of phase error
introduced in quadrature power divider on the I-Q
mixer output. Practical results of the realized X-band
QPSK modulator supporting 500 Mbps are presented
in Section V. Section VI summarizes the work done and
the results obtained.
II. I-Q MODULATOR AND I-Q MIXER BASED X-

BAND QPSK MODULATOR Fig.2 QPSK Spectrum (Carrier: 2 GHz; Data Rate:
500 Mbps; PRBS)
The block schematic of an I-Q modulator is shown in

Using I-Q modulator QPSK modulated signal can be
266
generated at a lower frequency, say, at 2 GHz as in the

case considered, and can be subsequently up-converted
to X-band using an I-Q mixer. Fig. 4 shows the block
schematic of an I-Q mixer. The Intermediate Frequency
(IF) and Local Oscillator (LO) signals are heterodyned
in the I-Q mixer as shown in the block schematic.
Fig.3 QPSK Spectrum (Carrier: 2GHz; Data Rate:

500Mbps; 1010 Data)
Fig.4 Block
Schematic of I-Q
Mixer
267
268
269
270
On the positioning of low-pass interpolator for

channel estimation in OFDM systems
Pallaviram Sure Chandra Mohan Bhuma
Research Scholar, ECE Dept. Professor and HOD, ECE Department
Acharya Nagarjuna University, Guntur Bapatla Engineering College, Bapatla
E-mail: pallaviram.sure@yahoo.com E-mail: chandrabhuma@gmail.com
AbstractIn pilot symbol aided channel estimation, LS estimated channel frequency response coefficients
the position of interpolator plays an important at the pilot positions are interpolated in a low-pass
role. In conventional channel estimator, which interpolator. Many types of interpolation schemes are
uses Least Squares (LS) estimation, initially LS discussed in literature [1],[2],[5],[6],[7]. Though low
estimates are obtained at pilot positions, then pass interpolation needs more hardware than linear,
interpolation and denoising operations are low-pass performance is much higher in terms of
performed in order. The most commonly used BER vs. Signal to Noise Ration (SNR) than the
interpolator is the low-pass interpolator. In this others. So it is common in conventional OFDM
paper, it is shown that OFDM systems, which use systems. The basic pilot arrangements are block and
denoising prior to interpolation, have poor system comb types, out of which comb type, suits for both
performance in terms of Bit Error Rate (BER) flat fading and frequency selective fading cases. In
compared to conventional systems which use comb type pilot arrangement, interpolation in an
denoising after interpolation. The theoretical OFDM symbol is required. Based on the
treatment as to why the positioning of the low- interpolation scheme used, the overall system
pass interpolator should be before the denoising performance changes.
block is framed. Further, the simulation results Denoising methods improve the accuracy of LS
obtained in various channel environments provide estimated, interpolated channel response [4], [8], [9].
strong evidence to the proposal. In [4], denoising based on truncation of estimated
channel impulse response is proposed. Thus it is
Keywords-OFDM; interpolation; denoise; understood that in conventional OFDM systems, low-
channel estimation; interpolation; LS; MMSE; pass interpolation is performed on the pilot position
LS channel estimates and then the obtained channel
I. INTRODUCTION AND RELATED WORK
frequency estimates are denoised to obtain the final
Channel estimation in OFDM systems is generally channel response. In literature, most of the channel
done either at the receiver side or at the transmitter estimation schemes based on LS, first use
side. If it is performed at the transmitter side, the interpolation and then denoising rather than the
information is fed back to the receiver so as to design reverse. In this paper, a detailed treatment on why
the equalizer. If the estimation is performed at the this should be the case is given. Simulation results
receiver, the hardware complexity slightly increases are also presented which verify this fact.
but the methods yields better performance and The rest of the paper is organized as follows. In
removes the necessity of feedback. This estimation at section II, the OFDM system is introduced along with
the receiver side is done in many ways, blind, semi- the notations used. The conventional channel
blind or pilot based. In this paper pilot symbol based estimation technique is discussed in section III. The
channel estimation is performed. theoretical analysis on the positioning of
In pilot based estimation, known pilot sequences interpolation filter, is given in section IV. In section
are inserted in the OFDM symbols, at known V, simulation results are shown in different channel
positions and are transmitted along with the data environments. Section VI concludes the paper.
symbols. At the receiver a channel estimation based
on either Least Squares (LS) or Minimum Mean II. OFDM SYSTEM AND SYMBOLS
Square Estimation (MMSE) is commonly used. LS The OFDM system is shown in figure (1). As it is
estimation is simple but it has poor performance understood from the block diagram, the baseband
compared to MMSE [1],[3],[7]. But, LS is more transmission in an OFDM system requires serial to
common than MMSE because MMSE needs prior parallel conversion. The modulation scheme used by
statistics about channel autocorrelation and noise, the mapper may be any of the digital modulation
which are not so easily available in practice. techniques like QAM, BPSK, QPSK, etc. The
271
parallel transmissions at a given time instant are said domain symbol. Using an IDFT block the symbol
to be one OFDM symbol at that time instant. A gets converted to time domain. In time domain,
length N OFDM symbol has N different subcarriers. Guard band namely the cyclic prefix is added and is
The OFDM symbol is considered to be a frequency then sent through the channel serially.
S (k ) s (n)
Serial to Pilot Cyclic Parallel to

Binary Parallel
data Mapper insertion IFFT Prefix Serial
conversion Block Insertion Conversion
Channel
AWGN
R (k ) r (n)
Parallel to Channel
Binary Estimation Cyclic Serial to
data Demapper Serial
and Symbol FFT Prefix Parallel
Conversion
Estimation Removal Conversion
Figure 1: OFDM system
The received OFDM symbol after removing the guard R(k )

band is given in (1). S (k ) , 0 k N 1 (3)
H (k )
r ( n) s ( n) h( n) g ( n) In (3), H ( k ) represents the estimated channel, R (k )
(1)
is the DFT of received symbol, S ( k ) is the estimate of the
In (1), h(n) indicates channel impulse response, g (n ) transmitted OFDM data symbol in parallel stream. The
estimated symbol is demapped to the binary data stream
indicates the Additive White Gaussian Noise (AWGN), after converting to serial data.
s (n) indicates the transmitted OFDM symbol.
III. CONVENTIONAL CHANNEL ESTIMATION
indicates convolution operation. The channel impulse
response is given as a multipath fading channel as shown The conventional channel estimator comprises of four
in (2) [1]. blocks as shown in Figure (2), which are explained as
G 1 j 2f DmTn follows. The pilot symbols are retrieved by the receiver
h(n) hm e N
( m ),0 n N 1 and are compared with the original pilot symbols
m0 transmitted. The receiver is aware of the sent pilot
(2) symbols and also the pilot positions. The pilot
In (2), G is the total multipath number. fDm is the arrangements most commonly used are comb-type and
maximum Doppler frequency of the mth path, is the block-type arrangements shown in Figure (3). After the
maximum delay spread and T is the sampling time.Also pilots are retrieved they are sent to the LS estimator.
m is the mth normalized path delay. The OFDM symbol H LS ( k ) H 1 ( k )
R (k )
r(n) is passed through a DFT block and the frequency Pilot LS OFDM OFDM
domain received symbol is obtained. On this symbol, DFT data
symbol estimation Symbols symbol
channel estimation is performed. The channel estimation Retrieval block Interpolation denoising
Block Block block
block is shown in Figure (2). The LS channel estimation
is given by (3).
Figure 2: Conventional Channel estimation
272
noisy because it comprises not only the actual channel

response H(k), but also has the added modulated noise
term D(k). To filter out this noise, H LS (k ) is passed
OFDM Subcarriers
through a denoising block.

H (k ) S (k ) G (k )
H LS ( k ) H (k ) D(k )
S (k )
(7)
Many denoising techniques as discussed in [4],[8],[9] can
be used to filter D(k) and obtain H(k). One such technique
Block-type Comb-type suggested in [4] is considered in this paper, which is
truncation. According to this method, let the sequence,
OFDM symbols
h(n) , 0 n N 1 ,be the interpolated channel
Figure 3: Pilot arrangements
impulse response. If the largest normalized multipath
The LS estimation is given in (4), where k indicates the delay of the channel is L, then the channel impulse
pilot position. response after L is made 0, as in (8).
R(k ) h( n), 0nL
H LS (k ) , k { pilot position} h1 ( n)
S (k ) 0, L 1 n N 1
(4) (8)
The channel frequency response coefficients of (4) are An equivalent representation of channel impulse response
passed through an interpolation block, which can be (8) is shown in (9) using the definition of a time window
represented as a filter. The interpolated channel frequency w(n). The window function w(n), is shown in (10).
response and impulse response is given by (5) and (6). In
(5) I(k) represents the interpolation filter frequency h1 ( n) h( n) w( n)
response i.e. transfer function. Corresponsingly, (9)
i(n)represents the impulse response of the interpolation 1, 0nL
filter. w(n)
H (k ) H LS (k ) I (k ) 0, L 1 n N 1
(10)
(5) The resulting sequence from this denoising block, i.e.
h(n) hLS (n) i (n) h1 (n) is the denoised channel impulse response. From this
(6) H ( k ) is obtained as the channel frequency response. If
1
Interpolation is of many types, linear, second order, cubic
this is used in (3), the transmitted OFDM symbol can be
etc. In this paper low-pass interpolation is treated. The
estimated.
process of lowpass interpolation [1], [2] is shown in
Figure (4). Initially zeros are uniformly inserted between IV. ANALYSIS OF INTERPOLATOR AND DENISING
the channel coefficients given by (4) such that the length BLOCKS
of the resulting sequence becomes N, i.e. length of one
OFDM symbol. This is passed through a lowpass filter, In this section, to justify the positions of the
such that the zeroes get modified but the original values interpolator and denoising blocks, two cases are
of (4) are still retained. This method is based on considered. The first case is the conventional case, where
minimization of Mean Square Error (MSE). Thus the interpolator output is passed through a denoising block. In
channel transfer function obtained in (5) has k ranging the second case, the denoising block output is passed
from 0 to N-1. through the interpolator.
Input data Output data
(length Np ) (length N>Np )
Case I: The output of the channel estimation when
Zero insertion block FIR low pass filter
interpolation is done first and then denoising is done, is
given by (9).
Figure 4: Low pass interpolation Case II: The output of the channel estimation when
denoising is done first and then interpolation, is analyzed
Note: The low pass interpolator in this paper, is below. The LS channel estimates are passed through the
represented as a filter, i(n) as shown in (6). denoising block whose impulse response is w(n). The
output of the denoising block is in (11).
Using DFT of (1) in (4), we obtain the LS channel
estimate as given in (7). As seen from (7), LS estimate is h2, LS ( n) hLS ( n) w( n) (11)
273
Comparing (18) and (20), which correspond to the final

From the sequence h2, LS (n) obtained by (11), H 2, LS (k )
channel transfer function estimates of case I and case II
is calculated and then interpolation is performed as shown respectively, two facts can be concluded:
in (12) to obtain H 2 ( k ) and correspondingly h2 ( n) as (a) H 1 ( ) and H 2 ( ) are equal only if A=1. But A=1
shown in (13). means V ( ) is no longer doing low pass interpolation.
H 2 (k ) H 2, LS (k ) I (k ) (12)
So, this case is not feasible which means H 1 ( ) and
h (n) h (n) i (n)
2 2 , LS (13) H 2 ( ) are never equal which also means that both the
In the following analysis, the channel estimation outputs channel estimates will never give same receiver
given for the two cases by (9) and (13) are compared. performance.
From the DFT of (9) and using (5), we get (14).
(b) The estimate H 2 ( ) given in (20) exists only in a
Similarly, using (12) and the DFT of (11) we get (15).
H 1 (k )
F h(n) w(n) part of the frequency axis,

, and so
H (k ) W (k ) (14)
A A

H LS (k ) I (k ) W (k ) h2 ( n) exists for all , because a band limited response
is never time limited. The actual channel response is time-
H 2 (k ) H 2, LS (k ) I (k ) limited because, the largest multipath is L. Thus the actual

F h (n) w(n) I (k ) (15) channel response is similar to h1 ( n) because it is time

LS
limited but not band limited.
H LS (k ) W (k ) I (k )
So, it can be concluded that theoretically H 1 ( ) gives
To make the analysis simple, two facts are assumed: First
is using DTFT instead of DFT for theoretical analysis and better receiver performance than H 2 ( ) and practically
second, assuming a low-pass type of interpolation. With
these assumptions, (14) is re-written as shown in (16),
H 1 ( k ) gives better receiver performance than H 2 (k ) .
This fact is further supported form the simulation results.
using I ( ) , which is the frequency response of the low
pass interpolation filter, given in (17). In (17), A is a V. SIMULATION RESULTS
positive value which is greater than 1 and is chosen based In this section, the conventional channel estimation
on cut-off frequency of low pass filter. Using (17) in scheme is implemented in three different channel
(16), we have (18). environments for two different cases, namely 1)

1 Interpolation first and then denoising 2) Denoising first
H 1 ( )
2 H

LS ( ) I ( )W ( )d (16) and then interpolation. The three different channel
environments are Advanced Television Technology
Center (ATTC), Digital Video Broadcasting for
1
I ( ) A A (17) Television (DVB-T), [1] and Brazil-E [2] channel. In all
the three channel environments, the conventional channel
0 otherwise estimation with case 1, showed superior performance than
case 2. The simulation parameters are shown in table (1).
/A
1
H 1 ( )
2 H
/ A
LS ( )W ( )d , Using comb type of pilot arrangement and lowpass
interpolation, the OFDM system performance results in
three channel environments are shown in figures (6),(7)
(18) and (8) respectively. In DVB-T the channel length is 50.
Similarly using the two assumptions in (15) we have So truncation is done by choosing L as 50. Similarly, in
(19) and using (17) in (19), we get (20). Brazil E channel, L is chosen 25. In ATTC, L is chosen as
128.

I ( )
2
TABLE I: OFDM System Simulation Parameters
H 2 ( ) H LS ( )W ( )d (19) Parameter Value
Mapper QPSK
Number of subcarriers (N) 1024
1
H 2 ( )
2 H

LS ( )W ( )d ,
A

A
Rate of pilot subcarriers
Guard interval (per OFDM symbol)
N/8
256
Channels: Static type ATTC, DVB-T, Brazil-E
(20)
274
VI. CONCLUSION
In this paper, a theoretical treatment on the positioning Denoising and then Interpolation
-0.2
of interpolation filter and denoising block in a 10 Interpolation and then denoising
conventional channel estimator is framed. It is shown that

interchanging the low-pass interpolator block and the
denoising block of a conventional channel estimator will -0.3
10
only lead to performance degradation. This fact is further
supported from the simulation results obtained from DVB-
BER
T, Brazil-E and ATTC channel environments. The
-0.4
theoretical proof is given which shows that low pass 10
interpolator should be before the denoising block. The
study in this paper is limited to static impulse responses of
various channels. But it can be extended to slow fading -0.5
10
channels as well as fast fading channels which give similar
results.
5 10 15 20 25
SNR in dB
-0.1
10 Denoising and then interpolation
Interpolation and then denoising
Figure 7: BER Performance of OFDM system in Brazil-E
channel environment
-0.2
10
-0.3
REFERENCES
10
BER
-0.4
[1] Sinem Coleri, Mustafa Ergen, Anuj Puri, Ahmad Bahai, Channel
10
estimation techniques based on pilot arrangement in OFDM
systems, IEEE transactions on broadcasting, vol 48,No. 3,Sep.
-0.5 2002, pp. 223-229
10
[2] J Carlos Augusto Rocha, Luciano Leonel Mendes, Adonias Costa
5 10 15 20 25 da Silveira, Performance Analysis of channel estimation schemes
SNR in dB for OFDM systems, Proc. Of International workshop on
Figure 5: BER Performance of OFDM system in ATTC channel telecommunications-IWT-2007, pp.32-36.
environment [3] Meng-Han Hsieh, Che-Ho Wei, Channel estimation for OFDM
systems based on comb type pilot arrangement in frequency
selective fading channels, IEEE Transactions on Consumer
Denoising and then interpolation
-0.2
Interpolation and then denoising
Electronics, Vol. 44, No. 1, February 1998, pp. 217-225.
10
[4] Jan-Jaap van de Beek, Ove Edfors, Magnus Sandell, Sarah Kate
Wilson, Per Ola Borjesson, On Channel Estimation in OFDM
systems, VTC, IEEE, 1995,pp. 815-819
-0.3
10 [5] Baoguo Yang, K.B. Letaief, Roger S. Cheng, Zhigang Cao,
Windowed DFT based pilot-symbol-aided Channel estimation for
BER
OFDM systems in multipath fading channels, IEEE Vehicular

-0.4
Tech. Conference, 2000, pp. 1480-1484.
10
[6] Ahmed N. H. Alnuaimy, Mahamod. Ismail, Mohd. A. M. Ali,
Successive Data Schism and Interpolation for wavelet channel
estimation in OFDM using Wavelet Denoising, ICACT February,
-0.5
10 2010, pp:661-665..
5 10 15 20 25 [7] Yong Soo Cho, Jaekwon Kim, Won Young Yang, Chung Gu
SNR in dB Kang, MIMO OFDM Wireless Communications with MATLAB
Figure 6: BER Performance of OFDM system in DVB-T Wiley2010.
channel environment
[8] Y Kang, K Kim and H.Park,Efficient DFT based Channel
estimation for OFDM systems on multipath channels,
Communications, IET, Vol. I, no. 2, pp. 197202, 2007.
[9] Y.S. Lee, H.C. Shin, and H.N.Kim, Channel estimation based on
a time domain threshold for OFDM systems, Broadcasting IEEE
transactions on, vol. 55, No. 3, pp. 656-662, 2009.
275
Taxonomy of Privacy Preserving Techniques

in Wireless Sensor Networks: A Survey
1 2 3
G.Sumalatha, N.Zareena, M.Parimala
1
Asst.Prof, Department of CSE, Vignan's LARA Institute Of Technology & Science, Vadlamudi Guntur Dist., A.P., India.
2
3
1 2 3
suma_rajitha@yahoo.co.in , zareenacse@gmail.com, parimala.cse@gmail.com
Abstract-This paper provides a survey of privacy- There are number of the applications in the WSN.
preserving techniques for WSNs. The wireless This includes [1]the military applications which
transmission required by a WSN, and the self- controls the monitoring, tracking and surveillance
organizing nature of its architecture, makes privacy of the borders. Other applications include the
protection for WSNs an especially challenging problem. environmental applications, health applications,
In particular, we review two main categories of privacy- home applications, commercial applications.WSN
preserving techniques for protecting content-oriented are capable to collect their data automatically with
and context-oriented privacy, respectively. Thus the help of the sensor devices. Even though there is
protection of the physical location is needed at data great benefit to the users some misuse them so
source and the data sink. Suppose we take the scenario
privacy is a concern there. If we take famous
of the panda hunter where in sensors are being
scenario of the panda and hunter [2], the hunters
deployed in the forest to monitor the endangered
can find physical location of the panda by
pandas. The adversary is quite efficient to monitor the
panda and capture the panda. We present taxonomy of
monitoring the traffic or using sound monitoring
various privacy preserving techniques against a local where in panda is tracked with help of the sound
eavesdropper is being presented. This paper also recognition.
compares the performances of all the privacy preserving
techniques. 1.2 Need for Privacy Preservation In Wireless
Sensor-Networks
Keywords:WSNs, Privacy Preserving Techniques,
Routing protocols, Traffic analysis.
The privacy preservation techniques of wired
networks are not useful in sensor networks,
1. INTRODUCTION firstly due to the fact that network being different
the set of problems are different and secondly
1.1WirelessSensorNetwor because many of the methods pose overheads which
ks are too burdensome for the sensor networks. The
shared wireless medium enhances the chances for
A wireless sensor network (WSN) is a heterogeneous the adversary to locate the origin of the
network composed of a large number of tiny low-cost radio transmission and thus facilitating the hop
devices, denoted as nodes (or motes), and one or few by hop trace back to the origin of the multi hop
general purpose computing devices referred to as communication. Launch of physical attacks and
base stations (or sinks). A general purpose of the node compromises by the adversary thus posing a
WSN is to monitor some physical phenomena (e.g., menace to the whole wireless sensor networks is
temperature, barometric pressure, light) inside an quiet evident due to the miniature size of the sensor
area of deployment. Nodes are equipped with a nodes and very nature of the wireless
communication unit (e.g., radio transceiver), communication environment.
processing unit, battery and sensor(s). Nodes are As we know a wireless sensor network is severely
constrained in processing power and energy, whereas constrained by various resources as computation,
the base stations have laptop capabilities and not storage, and wireless communication bandwidth
severely energy resources. The base stations usually and battery power. The adversary could monitor
act as Gateways between the WSN and other such activities of the sensor as the communication
networks (e.g., Internet). patterns to figure out the energy depletion or
resource usage in order to spot the most
276
vulnerable spots in the network and use them to

attack the network as a whole.
In the above mentioned example of noise and sound
monitoring system; adversary can use sensor data for
The unique challenges for privacy preservation in matching and being able to trace any person. Two main
WSN call for the development of effective privacy- approaches for data-oriented privacy preserving
preserving techniques. In this paper, we provide a techniques are data aggregation and private data
survey of existing privacy-preserving techniques in query.
WSNs. We review two main categories of privacy-
preserving techniques for protecting two types of 3.1.1 Data Aggregation
private information, data-oriented and context- oriented
privacy, respectively. In the category of data privacy, Protection to data can be guaranteed by secret keys
we mainly discuss how to enable the aggregation of between sensor nodes [8, 9]. In this approach an
sensed data without violating the privacy of the data intermediate node has to decrypt the received data,
being collected and guarantee the privacy of data then aggregate the data according to the
query initiated by users of the network. For context- corresponding aggregation function, and finally
encrypt the aggregated result before forwarding it. This
based privacy, we analyze the protection of location
approach requires lots of computation and due to this
privacy and temporal private information. In addition, the overhead increase. WSNs have limited power so
we build a table to compare different techniques in this approach is fairly expensive for WSNs. Cluster-
terms of their effectiveness in practical applications. based Private Data Aggregation (CPDA) is more
Last but not the least, we discuss some interesting and effective approach for data aggregation [10]. As Figure
challenging open issues on this topic, which are 2 shows, the first step of CPDA Query Server Q
expected to shed light on a fruitful direction of future triggers a query by HELLO message. A recipient of
research on privacy preservation in WSNs. HELLO message elects itself as a cluster leader
randomly. Some of the nodes become cluster leader, so
2. RELATED WORK they broadcast the HELLO message to their neighbors
then it decides to join one of the clusters by
broadcasting a JOIN message. As this procedure
Privacy issue is widely explored in the field of
goes on, multiple clusters are constructed. The second
database, networks, data mining and other field. Lots of step of CPDA is the intermediate aggregations within
techniques are proposed for privacy preservation such clusters with the help of encryption keys. Computation
as: Cryptographic security [5], K-anonymity [7], overhead reduces in this approach as compared to
Random perturbation technique [6]. These techniques previous approach but still high.
are use to protect data when it flows from one node to
other.
A large numbers of attacks are possible in WSN such as
Denial of Service attacks, The Sybil attacks, Traffic
Analysis attacks,Node Replication attacks, attacks
against Privacy, and Physical attacks. Lot of work has
done to overcome these attacks [4]. Our area of
interest is on attacks against Privacy. Privacy attacks
can be further classified into two broad categories data
oriented attacks and context oriented attacks. Figure 1
shows the types of privacy- preserving problems in (a) Query Server Q triggers a (b) A and X become
cluster query by HELLO message. A leader, so they
WSN.
broadcast the recipient of HELLO message HELLO
message to their elects itself as a cluster leader neighbors.
3. PRIVACY PRESERVING randomly.
TECHNIQUES
This section presents review two types of privacy
preserving techniques, data-oriented privacy and
context oriented privacy respectively.Initially we
discussed data-oriented techniques then followed by
context oriented privacy preserving techniques.
3.1Data-OrientedPrivacy PreservingTechniques (c) Node E receives multiple (d) Several clusters have been
HELLO messages, and then constructed and the
In data-oriented privacy preserving technique main aggregation randomly selects one to join. tree of cluster
focus is on data which is collected and send to the sink. leaders is formed.
277
3.1.2 Data Query

Other privacy problem is due to data query. A query
protection technique was proposed in [11]. Instead of
enlarging the query regions, this technique intends to
disconnect the mapping between a user identity and
the query issued by the user. With this technique, a
user is able to access data in a WSN after purchasing a
certain amount of tokens from the WSN owner Such a
token not only controls access to the sensed data, but
also hides the user identity and there by guarantees
query privacy. Other privacy problem is due to data (b)Mixing
query. A query protection technique was proposed in
[11]. Instead of enlarging the query regions, this
technique intends to disconnect the mapping between
a user identity and the query issued by the user. With
this technique, a user is able to access data in a WSN
after purchasing a certain amount of tokens from the
WSN owner. Such a token not only controls access to
the sensed data, but also hides the user identity and
thereby guarantees query privacy.
3.2 Context-oriented privacy protection techniques

(c)Aggregation
Although privacy of data can be achieved through Figure 3 Three steps in SMART
effective protection techniques, sensor events are so 3.2.1.1 Location privacy for data source
sensitive that it is needed to protect all information
surrounding these sensor events. The context-oriented Flooding [14] has been used to preserve data source
information includes information on source location, location. In this technique each node broadcasts data
sink location and timing of events. Adversary can packets to their neighbor node to forward that data
obtain this type of information by using traffic packet towards the sink. This technique required lots of
analysis techniques [12]. Context-oriented privacy is energy due to broadcasting of data packet. Probabilistic
summarized in the following two subsections. flooding [14] is proposed to overcome this problem.
This technique is more energy efficient. A node
3.2.1 Location privacy forwards a message with some probability so that only
a subset of nodes is involved in forwarding. Phantom
Location privacy plays an important role in WSN flooding [14] technique is also been proposed to reduce
such as, location of special sensor node, data source energy consumption. This technique is carried out
and the base station. The famous Panda-Hunter with two steps. In the first step a message takes
Problem [3] in WSN sensor nodes are used to locate random or directed walk to a random node in the
pandas in their local habitat. Adversary can find out network. Then in the second step, the message is
location of sensor node that monitors the panda, flooded by the phantom node into the network to reach
successfully localize and capture the panda. the base station. In comparison to previous two
Similarly, the location of base station is needed by techniques higher level of source location privacy can
adversary to mount different network attacks on the be achieved by Phantom flooding.
base station. Location privacy is further classified In order to improve the phantom flooding arguments
into two categories; privacy for location of data are proposed in [15, 16]. They argue that both pure
source and location of base station. random walk and directed random walk have major
drawbacks. Pure random walk may stick around the
source node, whereas in directed random walk, an
adversary can obtain sensitive information from
source node. Instead of random walk, they propose to
route a message through single or multiple randomly
selected intermediate nodes from the network.
However, the adversary is able to obtain approximate
location of the source node; similar approaches still
provide local source location privacy only.
(a) Scaling In order to achieve global network-wide source
278
location privacy Network Mixing Ring (NMR) [17] the inconsistency of the traffic pattern caused by
is proposed that extends the routing through a hotspots, and camouflage the source node within the
random intermediate node. Logical ring containing group of nodes forming the cloud. The fake packets
large number of nodes is formed around the base also enable the real source node to send the sensed
station to mix messages globally. The actual routing data anonymously to a fake source node selected
has three steps. In the first step message is routed from the clouds nodes to send to the Sink. Crypto-
through a random intermediate node to some node in graphic operations are used to change the packets
the ring, which provides local source location appearance at each hop to prevent packet correlation
privacy. In the second and make the source node indistinguishable because
step, the message circulates in the ring to achieve the the adversary cannot differentiate between the fake
global source location privacy. In the last step, the and real traffic, i.e., the clouds traffic pattern looks
message is simply routed from a random node in the random for the adversary. Moreover, tracing the
ring to the base station. Source location privacy packets back to the source node is nearly impossible
problem was also taken up in [15] to trap an because the real traffic is indistinguishable and the
adversary in a false cyclic path. Several cyclic paths real source node sends its packets through different
are pre-generated in the network which results in fake source nodes.
improved safety period. Once a real message crosses 3.2.1.2 Location privacy for base station
one of the cyclic paths, fake messages start to Since base station collects data from entire network
circulate along this path. Then, adversary which security for a base station is highly essential. If an
backtracks on real messages can be stuck onto the adversary is able to locate the base station, it can
cyclic path. The safety period is improved by this apply several attacks such as Denial of Service
technique while maintaining low latency. attacks [4]. Hence protecting the location of the base
station is extremely important.There are three basic
The idea of fake source and dummy traffic [19] are
traffic analysis techniques [23] for discovering a base
proposed to hide real messages and sources within a
station location: the rate monitoring attack, the time
dummy traffic and fake sources. Two techniques are
correlation attack and the content analysis attack. In the
proposed for above idea namely Short-lived and
rate monitoring attack, an adversary observes node
Persistent Fake Source techniques. Again, these message sending rate and moves closer to a node with
techniques cannot protect the source location against the highest rate. In the time correlation attack, an
a global adversary, because the fake sources are adversary monitors a correlation in sending times
activated by real messages. In this way first source between a node and its neighbors. The adversary tries
can be recognized as the real one by global to detect which node forwards the current message and
adversary. To achieve protection against global traces the path directly to a base station. In the content
adversary; techniques are proposed such as Periodic analysis attack, an adversary tries to acquire valuable
Collection [20] and the Source Simulation [20]. information from message headers and payloads.
Periodic Collection provides optimal source location
privacy. In this technique every sensor node To get protection against the rate monitoring attack,
periodically sends a dummy or real message to the every sensor node sends messages at a constant rate.
sink at pre-defined intervals thus the real messages A child node continuously transmits data packet until
are covered within the dummy ones and the network it is accepted by a parent node. If the child node has
traffic is totally independent from the sensed events. no packet to send, it injects a dummy packet. This
In second technique several fake event sources are technique guarantees constant sending rate over the
added in the network. These fake sources behavior is network at the same time it decreases the lifetime of
identical with the real sources to confuse the the WSN. Different encryption techniques can be
adversary. The biggest challenge of the Source used to protect the base station against content
Simulation technique is to precisely simulate the analysis but they are not adequate. These techniques
source behavior. A statistical framework for do not solve the problem of message identity, which
evaluation of source anonymity in WSNs was remains the same along the path and easy to track.
proposed in [22]. All the above techniques are based Hop-by-hop re-encryption technique [23] can be use
on the fake sources and the dummy traffic that to overcome this problem.
ensures source location privacy in the presence of Different techniques [24, 25] are proposed to counter
global adversary on the cost of huge communication the traffic analysis attacks. The rate monitoring attack
overhead or higher latency. Instead of gaining great can be partially prevented by the multiple parents
source location privacy these techniques cannot be routing technique since traffic spreads along multiple
paths. In this technique, every sensor node has
practically implemented over a long period of time. multiple parent nodes, which route messages to the
Finally, a cloud-based scheme was proposed [21] for base station. Packet is forwarded through any parent
efficiently protecting source nodes location privacy node which is randomly selected by its child nodes.
against Hotspot- Locating attack by creating a cloud
This technique can be extended by the controlled
with an irregular shape of fake traffic, to counteract
random walk. A node forwards a message to one of
279
its parent nodes with probability p.with probability1- entities in the network are indistinguishable to the
p the node forwards the message randomly to one of nodes around the sink with regard to communication
its neighbors including the parent nodes. This statistics. Arranging the location of k entities is
technique introduces delivery time delays, which are complex as it affects two conflicting goals: the
proportional to extra hops taken by the messages. routing energy cost and the achievable privacy level,
This technique is still error-prone to the time and both goals are determined by a no analytic
correlation attack. Therefore, Multi-parent routing function.Wemodel such a positioning problem as a
technique with fractal propagation is used to avoid nonlinearly constrained nonlinear optimization
errors. When a node observes that a neighbor problem. To tackle it, we design a generic-algorithm-
forwards a message to the base station, the node based quasi-optimal (GAQO) method that obtains
generates a fake message with some probability and quasioptimal solutions at quadratic time. The
forwards it to one of its neighbors. The main problem obtained solutions closely approximate the optima
with this technique is that it generates a large amount with increasing privacy requirements. Furthermore,
of traffic near the base station, because nodes near to solve k-anonymity sink-location problems more
the base station usually forward more messages. efficiently, we develop an artificial potential-based
quasi-optimal (APQO) method that is of linear time
All the above discussed techniques provide sink
complexity. Our extensive simulation results show
location privacy against the local adversary. A
that both algorithms can effectively find solutions
completely randomized [26] technique is used to
hiding the sink among a large number of network
protect against global adversary. In this technique, a
nodes.
data packet randomly travels through a network for a
predefined number of hops discarding the fact that it 4. COMPARISONS AND CHALLENGES
reaches the base station or not. To improve the
In this we compare all the privacy preserving
probability of data packet delivery, multiple copies of
techniques that have been reviewed in this paper. Table
the same data packet can be sent out by the source or
1 depicts the performance of privacy preservation
a history of already visited nodes can be appended to
techniques in WSNs. We evaluate their performances in
the message to prevent loops in the routing path. In
metrics: privacy, efficiency, location, delay time, and
addition, this technique provides protection against
an internal adversary, because no information on the power consumption, message overhead, scalability.
sink is included in a data packet. This technique may Privacy refers to the degree. The delay time includes
seem to be impractical for larger networks, because both the computation and communication time of data
the probability of delivery decreases rapidly with an transmission at the intermediate sensors. The power
increasing number of nodes. Another scheme based consumption measure focuses on the additional
on local flooding of source and greedy random- messages required for transmission (i.e., additional
walk[26] of sink is proposed to protect the location energy consumed) in the WSN. There are some open
privacy of mobile sinks in sensor networks. In this issues for future research. As the network behavior
scheme, sensor do not know any information about changes so sensor nodes should be designed
sink-location, data are forwarded by local flooding accordingly, along with it there can be improvement in
and stored at pass nodes in the network, the sink every technique with respect to overhead, power,
move in greedy random-walk to collect data from the accuracy.
local nodes occasionally, which prevents the attackers
from predicting their locations and movements. The Table 1: Comparison between different techniques
analysis shows that the scheme can provide location
privacy of mobile sinks effectively, while providing
satisfactory data collection services.
We also use location privacy routing protocol (LPR)
[27]that is easy to implement and provides path
diversity. Combining with fake packet injection, LPR
is able to minimize the traffic direction information
that an adversary can retrieve from eavesdropping.
By making the directions of both incoming and
outgoing traffic at a sensor node uniformly
distributed, the new defense system makes it very
hard for an adversary to perform analysis on locally
gathered information and infer the direction to which
the receiver locates.
To protect the sink-location privacy from a powerful
adversary with a global view, we propose to achieve
k-anonymity [28] in the network so that at least k
280
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Mobile-Sink-Location Privacy in Wireless Sensor Network by
Jianbo Yao Zunyi Normal College, Zunyi Guizhou 563002China
282
Variance based multiservice DBA

scheme for GPON Access Networks
D Khasim Hussain, and
R Manivasakan
AbstractGigabit Passive Optical Network terminologies are used interchangeably. MPCP

(GPON) is the flexible optical access network, that provides a mechanism to communicate OLT and
provides huge bandwidth and allows to play
ONU, while REPORT and GATE are the control
differentiated services (DiffServ). In this paper, we
implemented a hierarchical scheduling algorithm to messages used to convey the information between
share the upstream bandwidth. This consist of Inter OLT and ONU. ONUs send the REPORT messages or
ONU and Intra ONU scheduling, that utilizes status reports containing buffer sizes, while the OLT
bandwidth efficiently by eliminating light loaded sends the ONUs GATE messages or grants
penalty and providing Service Level Agreement
containing the granted time slots. The exchange of
(SLA), and provision of Quality of Service (QoS)
requirements by implementing Dynamic Bandwidth REPORT and GATE messages are shown in Fig 2.
Allocation (DBA) respectively. We proposed Variance OLT broadcasts GATE messages in the downstream
based Dynamic Weighted Fair Queueing (VBDWFQ) to the ONUs. The intended ONUs collects the GATE
and Hybrid based DWFQ (HBDWFQ) in Intra ONU messages, and in the upstream the ONUs transmits
scheduling that supports Proportional their data in the allocated time slots. The ONUs share
Differentiated Services. The simulation results of
VBDWFQ and HBDWFQ are compared interms of the common upstream transmission medium. Hence,
channel utilization and average queueing delay. the upstream is called Time Division Multiple
Access (TDMA)
I. INTRODUCTION
Passive Optical Networks (PONs) are widely

deployed to deliver high speed broadband
technologies for the majority of FTTx (Fiber to
the x) subscribers. A PON is a point- to-
multipoint optical network with a passive element,
splitter/combiner, in the signals path from source to
destination. Majority of the PONs are deployed in
tree topology, where central office equipment called
Optical Line Terminal (OLT) is connected to
multiple user-side optical devices called Op- tical
Network Unit (ONU) through a passive splitter. The exchange of REPORT and GATE messages are
Widely deployed PON architectures are Gigabit shown in Fig 2. OLT broadcasts GATE messages in
Passive Optical Net- work (GPON) and Ethernet the downstream to the ONUs. The intended ONUs
Passive Optical Network (EPON), standardized by collects the GATE messages, and in the upstream
ITU-T [1] and IEEE respectively. the ONUs transmits their data in the allocated
time slots. The ONUs share the common upstream
While the channel rates are 2.488 Gbps (GPON) transmission medium. Hence, the upstream is called
and 1Gbps (EPON), and covering the distances over Time Division Multiple Access (TDMA). OLT
25 kms. Typical GPON architecture is shown in distributes the upstream channel based on a
Fig1.Due to the increase of end user demand and bandwidth allocation algorithm. Due to the variable
advent of new applications, strict Quality of Service nature of end user traffic demand, employing a
(QoS) requirements are imposed. MAC layer offers Dynamic Bandwidth Allocation (DBA) algorithm is
these QoS requirements, and the MAC protocol much efficient to allocate the bandwidth
used in GPON network is Multi Point Control according to the instantanious requirements. Various
Protocol (MPCP) [2]. QoS is implemented in GPON bandwidth allocation algorithms are proposed in the
by introducing the concept called Transmission literature [3], [4], [5]. In this paper we are
Containers (T- CONTs), where the T-CONT implementing a hierarchical scheduling scheme [6],
represents a logical queue or class of service. which allocates the bandwidth in two steps. In the
Throughout this paper queue and T-CONT first step a transmission window is allocated to each
283
ONU, that is called Inter ONU scheduling. In the

second step the individual queue at the ONU is
scheduled within the granted transmission window,
that is called Intra ONU scheduling. Our Intra ONU
dynamic weighted fair queuing (DWFQ) schedular
supports Proportional Differentiation service models
for Variance[7] and Hybrid (Delay and Variance of
delay). The rest of the paper is organised as
follows. In section 2, we explain the Inter ONU
bandwidth allocation. In section
3, we present Variance, Hybrid and Delay based Intra
ONU DWFQ schedulars. In section 4, we discuss
Network Modeling and Performance Evaluation. In
section 5, a conclusion is drawn from our paper.
II. INTER ONU BANDWIDTH

ALLOCATION
OLT performs the Inter ONU scheduling among the

multiple ONUs, and it avoids the overlapping of ONU
transmissions by appropriately sharing the upstream
channel. OLT distributes a Scheduling Interval(SI ) Substitute excess in equation (1) to get the GATE
among ONUs in the upstream direction, and the (Gj ) of jthe heavy load ONU. The distribution of
cycle time is defined as the time difference between granted transmission window (Gj ) among service
the time instant at which the first ONU sends its classes of each ONU is discussed in Intra ONU
REPORT and the time instant at which the last ONU DWFQ scheduling
receives its GATE.
III. I NTRA ONU DWFQ S CHEDULING
Cycle time is shown in Fig2. As per the Service Level In this section we study Variance based DWFQ
Agreement (SLA) between the central office and (VBD- WFQ) and Hybrid based DWFQ
the end user, minimum guarenteed bandwidth is (HBDWFQ) algorithms. These algorithms are aimed
allocated to each ONU. Let N be the no of ONUs. at changing the queue weights dynamically based
Minimum bandwidth is calculated as follows. on the predicted arrivals and measured buffer
occupancy. This change of weights controlls the
average delays of queues by controlling the service
min is the minimum guarenteed amount of rates of the queues. Each diffserv enabled ONU
bandwidth given to each ONU. C is the channel supports three service classes:
capacity. We have seen in the MAC protocol, that
the OLT collects reports before sending the GATE ExpediatedForwarding(EF),AssuredForwarding(AF)
messages to the ONUs. In the current cycle the ONUs ad BestEffort(BE)serviceclasses.TCONT1,TCONT2
send their REPORTs to the OLT. OLT and
recalculates the bandwidth window to each ONU
TCONT3holdsEF,AFandBEservicesrespectively.
based on the current REPORTs.
Inorder
i to calculate the average
i
delays of packets of a
queue, we assume that the packets arrive at
discrete times and the packets that enter in time
interval (k , k) are considered to enter the queue
at the time k . is the cycle time that is shown
in Fig 2. Time interval from k to k is defined as
the nth cycle. In nth cycle DWFQ allocates a
service rate to class i at time k , is denoted as s
k

. Buffer occupancy at time k is bk .
284
Before the GATE reaches the ONUs, the ONUs

has to wait a certain amount of time, which is called
waiting time or measurement period . of a
particular ONU varies from cycle to cycle. Denote
the average arrival rate of the class-i traffic is i ,
and the estimated number of packets in waiting
time interval by i . In the current cycle the
REPORT of the class-i, bk + i k , of ONU j is
sent to OLT. OLT collects ItheseREPORTs,and in
thenextcylcetheInterONUschedulersendstheGATEm
essagestotheONUs.IntraONUschedul ng Is shown in
Fig.3.
service classes the service rates or weights changes

based on the VBDWFQ and HBDWFQ algorithms
respectively. Hence, these algorithms are called
Dynamic Bandwidth Allocation algorithms.
IV. N ETWORK M ODELING AND

P ERFORMANCEE VALUATION
285
Fig.5 shows the average queueing delay of EF,

AF and BE under VBDWFQ and HBDWFQ
algorithms. We can say that, for high or low queue
weights the corresponding queue delays are low or
high respectively. we can observe that the average
queueing delay trends shown in Fig.5 is evident from
the queue weights shown in Fig.4. Under VBDWFQ
algorithm the average queueing delays of AF and BE
increases rapidly from load 0.6, where as the EF class
performs stadily. Average packet delays of AF and
BE are decreased under HBDWFQ, when compared
to VBDWFQ. At high load conditions average packet
delay of EF class is decreased under VBDWFQ,
when compared to the HBDWFQ.
Fig.6 shows the utilization versus the network load.

the graph we can notice that the VBDWFQ weights Under HBDWFQ algorithm the utilization is
fluctuates widely compared to the HBDWFQ. At increased at high load conditions, when compared to
network load 0.9 the weights of high, medium and the VBDWFQ algorithm. From equation (4), the
low priority classes are higher, medium and lower bandwidth shared among the queues under VBDWFQ
respectively for VBDWFQ and HBDWFQ is based on the expected arrivals, that are measured by
algorithms. High priority class of VBDWFQ using maximum function. There exists error in
performs better when compared to the HBDWFQ, measuring the expected arrivals, and hence the
where as medium and low priority classes of bandwidth allocated to the particular queue is under
HBDWFQ performs better than the VBDWFQ. utilized. Thus VBDWFQ under utilizes the channel.
286
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287
FPGA Based Excitation Generation for

Navigation Sensors
Deepthy C K1, Rajesh George P, Raghunath K P, Mary Jermila M, Aparna Vishnu N,
B. VenkataRamana, Dr. P P Mohanlal.
ISRO Inertial Systems Unit,
Thiruvananthapuram-695013
Abstract This paper details the implementation of of a sinusoidal waveform. A sinusoid is synthesised
Field Programmable Gate Array (FPGA) based sine when the LUT containing digital data is scanned and
wave generation and three phase square wave delivered to the input of a digital to analog converter.
generation technique for the functioning of navigation
sensors. The Sine waveform of 6V and different This paper describes the implementation details
frequencies is required for excitation of rotor angle of Digital Synthesis technique on FPGA platform.
sensor and 3 phase power signal for the wheel rotation. This scheme makes stable waveform with
The Sine wave generation includes counter logic for an programmable frequency. The amplitude and
address generator to read the stored values from a
LUT. The sampled values are converted in to
frequency trimming requirements of conventional
continuous sinusoid using a digital to analog methods can be avoided here. In addition it
converter (DAC). The 3 Phase waveform based on provides good spectral purity. The output
Johnson counter logic is implemented in FPGA. This frequency can be easily changed by changing the
digital technique of wave form generation has many frequency of the reference clock or by
advantages over conventional scheme based on analog reprogramming the LUT.
design. The VHDL program is implemented on FPGA
chip ACTEL A1020. The simulation result of VHDL II. DEVICE SELECTION
is shown in the form of timing diagram obtained using
ModelSim software. The experimental result of
generation of excitation waveform for the sensors is also The design is realized with Actel FPGA A1020. A
presented. single FPGA services both three phase signal for
motor power and sine wave for the Pickoff
I. INTRODUCTION excitation. FPGA is clocked by a Crystal Oscillator,
which has a frequency stability of 50ppm over full
Navigation system use Dynamically Tuned MIL temperature range. Power on Reset circuit is
Gyroscope to acquire the angle/rate information. kept outside the FPGA. DAC AD7845 having
Operation of the gyroscope requires precise and high temperature stability is used to generate sine
stable sine power signal with frequency wave of required frequency from the FPGA output.
between12-17KHz and the voltage amplitude The DAC output goes to current buffer which also
of 6V for exciting the rotor angle measurement does the filtering to remove the sampling noise.
coil (pickoff coil). The spin motor of DTG is a
hysteresis synchronous motor, which is designed Switching amplifier is kept outside the FPGA to
to run between 7800 and 10200rpm and 3 phase provide necessary current drive for the three phase
motor power of 260-340Hz is required for the motor signal for the wheel excitation signal of DTG. The
operation. design utilizes 54% of logic I/O gates and I/O used is
32 out of 60.
Traditionally the sine signal has been generated
byWeinbridge Oscillator with output power stage and III. FUNCTIONAL BLOCK DIAGRAM OF FPGA
amplitudestabilization network. The motor power has
been generated by a digital counter circuit followed
by power switching. Themotor power generator and
Weinbridge Oscillators are freerunning and
harmonics of these signals can beat in
precisioncircuits and degrade navigation
performance. Theperformance of the analog circuits
is also subject to ageingand environmental changes.
digital synthesis is a Look-Up Table (LUT) based
technique to generate any arbitrary waveform using
stored digital samples. The LUT contains digital data
288
divides the precounter output by six. Johnson outputs

IV. THREE PHASE WHEEL POWER are used to command the power switching circuit are
GENERATION named as r-ph-out, y-ph-out and b-ph-out. The
inverted signals of r-ph-out, y-ph-out and b-ph-out
The 3 Phase waveform required for motor are also required for the operation of switching
exciting is generated using the FPGA. The rotor amplifier. Additionally a Phase delay of 20 S is
rpm for each DTG is experimentally determined introduced at the leading edge of each three phase
within 0.5rpm during its realization phase. command outputs and corresponding inverted outputs
Square wave supply is sufficient for the motor as before given to switching amplifier to avoid cross
the running current is free of higher harmonics. conduction.
Operation of the rotor at this rpm is essential to for
obtaining accuracy and stability and this is known V. PICK OFF EXCITATION (SINE WAVE
as the tuned speed of the DTG. The rpm of the GENERATION)
motor is related to the supply frequency by the
motor equation 120 f/p. To set the motor rpm the The DTG requires stable 6Vrms, sine wave
frequency of the 3 phase waveform is calculated excitation in a frequency range of 12-17KHz for
from the above equation. the operation of rotor angle measurement of
DTGs. As this block also uses the same clock as the
motor power generation a different programmable
pre-counter is required. This pre-counter is also
used to set the output frequency.
The pre-counter output clocks a three bit

up-down counter which generates the address
for the LUT. As the LUT is not possible. To
reduce the number of data to be 270 phase.
The address counter sequentially counts in up
and down directions to complete the sine wave.
The DAC AD7845 is used to generate sine
wave of required frequency from the FPGA
The motor power generation consists of the 3 output. DAC receives the codes from a LUT
power switching circuit which feeds the motor stored in FPGA. The frequency of are
and the control command for power switching written in to DAC. The output from the DAC is
which is part of the FPGA. Provision for a wide given to an analog buffer to obtain the necessary
range of frequencies within close separation of current drive. the requirement for a large LUT
0.1Hz is made by running the FPGA at a higher as the number of samples can be reduced
clock frequency and pre-counting with a without increasing the sampling noise.
programmable down counter. Nine bit Sampling rate has been selected as 12 times
Programmability is provided in the pre- counter the output frequency and six samples are stored
for setting the tuned speed in the desired range. in LUT.
The pre-counter output frequency is the 6th

multiple of the required motor excitation Programmable pre-counter generates an
frequency. This is accessible externally from the output frequency of 12 times the required
FPGA for verification purposes. Three phase signals excitation frequency from the crystal oscillator.
are generated using a 3bit Johnson counter and it The preload value is varied (by selection of
289
jumpers external to FPGA) to adjust the

sine wave frequency.
The LUT data is generated by sampling a sine

wave at 12 times its frequency and encoding in
offset binary format as required by the DAC. The
LUT data is reproduced in Table below:
The VHDL code was synthesised and fused in
A1020 FPGA. A Standalone tests on the FPGA
were conducted and all timing and margins are
ensured. Programmability of the device was also
verified. The device was used in system and all
tests including environmental tests were carried
out. Frequency stability of <0.05Hz is obtained in
both motor and pickoff excitation. Amplitude
stability of pickoff excitation is better than 1%.
CONCLUSION
Excitation signals required for the inertial

sensors were generated and meeting the
specifications. The timing for the design has
been validated and found meeting
requirements.. This excitation circuits are flight
tested and validated. The amplitude stability,
settability and low distortion makes this ideal
for the future navigation packages. With the
selection of suitable clock frequency and the
programming options, the resolution can be
increased further.
VI. SIMULATION AND TEST RESULTS
VHDL code for this module and its test benches

are implemented using Libero IDE and all test
cases are validated. Pre, Post synthesis
simulations were carried out and Timing margins
are verified. Phase relationship of 3 phase motor
waveform and timing delay between in phase
and out of phase component of RYB signals
are also verified. The Timing diagram of Sine
wave pickoff excitation and 3 phase motor drive
are shown below(Fig 7 and Fig 8)
290
An Approach to Morphological Image

Segmentation For Noisy Digital Image
Dibyendu Ghoshal, Pinaki Pratim Acharjya, Uddalak Chatterjee
Associate Professor Assistant Professor

Department of ECE Department of CSE
National Institute of Technology B.I.T.M.,Santiniketan
Agartala, Tripura, Pin - 799055 West Bengal, Pin 731236
tukumw@gmail.com ppacharjya@gmail.com
ABSTRACT: This paper introduces a very effective external disturbance [8]. If an image is being sent
approach of digital color image segmentation. In electronically from one place to another via satellite
recent days segmentation and edge detection of or wireless transmission or through networked
noisy images is a vast challenge for the computer cables, we may expect errors to occur in the
vision. In this paper Minimum Mean Square Error image signal. These errors will appear on the image
(Wiener) Filtering has been introduced for noise output in different ways depending on the type of
removal from the noisy images and watershed disturbance in the signal. Usually we know what
algorithm using distance transform is applied on the type of errors to expect and the type of noise on
noise free images for segmentation purpose. The the image, hence we investigate some of the
combination of these two strategies produces an standard noise for eliminating or reducing noise in
effective segmentation approach for noisy images. color image [9].Image Noise is classified as
Comparative study is also done in this paper. The Amplifier noise (Gaussian noise), Salt-and-pepper
experimental result ensures the effectiveness of the noise (Impulse noise),Shot noise, Quantization
proposed approach. noise (uniform noise),Film grain, on-isotropic
noise, Speckle noise (Multiplicative noise) and
Key words: Image segmentation, Wiener filtering, Periodic noise.
Watershed algorithm. To perform image segmentation and edge detection
tasks, there are many methods that incorporate region
INTRODUCTION growing and edge detection techniques [10-19] but no
such research work is done with the combination of
Any gray tone image can be considered as a Minimum Mean Square Error (Wiener) Filtering and
topographic surface. If we flood this surface from Morphological image segmentation (Watershed)
its minima and, if we prevent the merging of the algorithm. In this paper a combination of Wiener
waters coming from different sources, we partition filtering and watershed algorithm using distance
the image into two different sets: The catchment transform techniques for image segmentation and
basins and the watershed lines. If we apply the edge detection procedures were used. The wiener
watershed transformation to the image gradient, filtering method was applied to obtain noise free
the catchment basins should theoretically images. Then a watershed algorithm using distance
correspond to the homogeneous gray level regions transform technique is used to segmentation of
of this image. However, in practice, this images.
transform produces an important over-segmentation The structure of this work is the following: Section 3
[1-7] due to noise or local irregularities in the introduces Minimum Mean Square Error (Wiener)
gradient image [8-9]. Filtering. Section 4 is devoted to the segmentation
Images are often degraded by noises. Noise can occur process for edge detection using watershed algorithm
during image capture, transmission, etc. Noise using distance transform. Section 5 presents the
removal is an important task in image processing. In proposed scheme. The results are discussed in section
general the results of the noise removal have a 6 and we nish this paper with some concluding
strong influence on the quality of the image remarks with section 7.
processing and image segmentation technique. Noise
is the result of errors in the image acquisition process 2. WIENER FILTERING
that results in pixel values that do not reflect the true
intensities of the real scene. Noise reduction is the The inverse filtering is a restoration technique for
process of removing noise from a signal. Noise to be deconvolution, i.e., when the image is blurred by a
any degradation in the image signal caused by known lowpass filter, it is possible to recover the
291
image by inverse filtering or generalized inverse cascade implementation of inverse filtering and noise
filtering. However, inverse filtering is very sensitive smoothing:
to additive noise. The approach of reducing one
degradation at a time allows us to develop a =
restoration algorithm for each type of degradation and (4)
simply combine them. The Wiener filtering executes
an optimal tradeoff between inverse filtering and The disadvantage of this implementation is that when
noise smoothing. It removes the additive noise and the inverse filter is singular, we have to use the
inverts the blurring simultaneously. generalized inverse filtering. People also suggest the
The Wiener filtering is optimal in terms of the mean power spectrum of the original image can be
square error. In other words, it minimizes the overall estimated based on a model such as the model.
mean square error in the process of inverse filtering
and noise smoothing. The Wiener filtering is a linear 4. WATERSHED SEGMENTATION USING
estimation of the original image. The approach is THE DISTANCE TRANSFORM
based on a stochastic framework. The orthogonality
principle implies that the Wiener filter in Fourier
domain can be expressed as follows:
( , ) ( , )
( , )=| ( )| ( )
, , ( , )
(1)
Where ( , )+ ( , ) are respectively
power spectra of the original image and the additive
noise, and ( , ) is the blurring filter. It is easy to
see that the Wiener filter has two separate part, an
Fig. 1: Watershed segmentation-local minima of gray
inverse filtering part and a noise smoothing part. It
level yield catchment basins, local maxima define the
not only performs the deconvolution by inverse
watershed lines.
filtering (highpass filtering) but also removes the
noise with a compression operation (lowpass
The Watershed method or the watershed transform is
filtering).
an image segmentation technique based on gray-scale
mathematical morphology. It can be classified as a
To implement the Wiener filter in practice we have to
region-based segmentation approach where the main
estimate the power spectra of the original image and
idea behind this method comes from geography. If a
the additive noise. For white additive noise the power
landscape or topographic relief which is flooded by
spectrum is equal to the variance of the noise. To
water, the divide lines of the domains of attraction of
estimate the power spectrum of the original image
rain falling over the region forms watersheds.
many methods can be used. A direct estimate is the
periodogram estimate of the power spectrum
computed from the observation:
= [ ( , ) ( , ) ]
(2)
Where, ( , ) is the DFT of the observation. The

advantage of the estimate is that it can be
implemented very easily without worrying about the
singularity of the inverse filtering. Another estimate
which leads to a cascade implementation of the
inverse filtering and the noise smoothing is:
=
| |
(3)
which is a straightforward result of the fact: =

+ | | .The power spectrum can be
estimated directly from the observation using the Fig 2: (a) shows a binary image matrix, and (b) shows
periodogram estimate. This estimate results in a the corresponding distance transform.
292
An alternative approach is to imagine the landscape In initial stage four noisy color images are chosen and
being immersed in a lake, with holes pierced in local converted into gray scale or black and white image.
minima also called catchment basins (CB), will fill up In next step the noisy gray scale images are de noised
with water starting at these local minima, and, at using Minimum Mean Square Error (Wiener)
points where water coming from different basins Filtering. In fourth step Edge map using Sobel
would meet, dams are built. When the water level has operator has been calculated. In fifth step image
reached the highest peak in the landscape, the process complement is accrued. In next step distance
is stopped. As a result, the landscape is partitioned transform is applied on the image complement and
into regions or basins separated by dams, called finally watershed algorithm is applied to get the final
watershed lines or simply watersheds For digital segmented image map.
image segmentation, the distance transform method is
Noisy Color image
commonly used in conjunction with the watershed
transform. The distance transform is the distance
from every pixel to the nearest pixel of a binary Get greyscale image
image. In distance transform method every 1-valued
pixel has a distance transform value of 0 because its Image denoising using
closest nonzero pixel is itself. In below, figure 4(a) wiener filtering
shows a binary image matrix, and in figure 4(b) Edge map using Sobel
shows the corresponding distance transform. operator
5. PROPOSED APPROACH Image complement
In signal processing, the Wiener filter is a filter used Apply distance transform
to produce an estimate of a desired or target random
process by linear time-invariant filtering an observed Final segmented image using
noisy process, assuming known stationary signal and watershed algorithm
noise spectra, and additive noise. The Wiener filter
minimizes the mean square error between the Figure 3: Flow diagram of the proposed approach.
estimated random process and the desired process.
The goal of the Wiener filter is to filter out noise that 6. EXPERIMENTAL RESULTS AND
has corrupted a signal. This approach often produces DISCUSSION:
better results than linear filtering. The adaptive filter
is more selective than a comparable linear filter,
preserving edges and other high-frequency parts of an
image.
And, watershed algorithm is a tool for morphological

image segmentation. A gray scale image can be
interpreted as the topographic image of landscape.
This is accomplished (the image intensity) as an
altitude. Using the features of these images, the
technique of digital image processing called
Watershed Transform. For digital image
segmentation, the Watershed algorithm using
distance transform method is commonly used in
conjunction with the watershed transform. The
distance transform is the distance from every pixel to
the nearest pixel of a binary image. In distance
transform method every 1-valued pixel has a distance Figure 4: A collection of sample noisy images.
transform value of 0 because its closest nonzero pixel Combination of Wiener filtering and watershed
is itself. algorithm using distance transform techniques for
image segmentation and edge detection procedures
In our proposed approach these two techniques has were used. The wiener filtering method was applied
been combined and accordingly modified to get the to obtain noise free images. Then a watershed
image segmentation process more effective. The algorithm using distance transform technique is used
flowchart of the proposed scheme is given in below. to segmentation of images. Four real life noisy
293
images of Lena, Laure, ball and frits of 512x512, grayscale images which are shown in figure 5 and
318x554, 256 x 256 and 512x512 dimensions (Figure final segmented images which are obtained by
4) has been chosen to do the experimental work. The proposed approachapplyed on same grayscale noisy
gray scale images of originals are shown in figure 5. images and accordingly shown in figure 8. Image
It has been observed from the segmented images after entropy is a quantity which is used to describe the
using watershed algorithm directly on the noisy gray `business' of an image, i.e. the amount of information
scale images images that they are extremely over which must be coded for by a compression algorithm.
segmented (Figure 6). Image entropy is calculated with the formula:
=
(5)
Figure 5: Gray scale images of the originals.
Figure 7: Noise free images of originals using wiener

filtering.
Figure 6: Segmented images obtained by using

standard watershed algorithm.
To reduce the over segmentation a new approach of

watershed algorithm using distance transform is Figure 8: Final segmented images using proposed
applied with wiener filtering. The noise free images approach.
with wiener filtering are shown in figure 7. Final
segmented images are shown below in figure 8 using The Peak Signal to Noise Ratio (PSNR) is the
propped and better accuracy is achieved in detection value of the noisy image with respect to that of
of edges with lesser over segmentation. the original image. The value of PSNR and MSE
(Mean square Error) for the proposed method is
The statistical measurements with entropy, PSNR and found out experimentally. The PSNR and the Mean
MSE are also calculated for segmented images using Square Error of the retrieved image can be calculated
conventional watershed approach applied on noisy by using the equation number 2 and 3.
294
2. Luis Patino, Fuzzy relations applied to minimize over

( , ) = 10 10 segmentation in watershed algorithms, Pattern Recognition
( , )
(6) Letters, Volume 26, Issue 6, Pages 819-828, 1 May 2005.
( [ (, , ) ( , , )] 3. Mariela Azul Gonzalez, Gustavo Javier Meschino, Virginia
( , )= Laura Ballarin, Solving the over segmentation problem in
(7) applications of Watershed Transform, ournal of Biomedical
Graphics and Computing, , Vol. 3, No. 3, pp.29-40 , 2013.
Where Org is the original image, Img is the
filtered color image of size M. M, S is the 4. Pinaki Pratim Acharjya, Dibyendu Ghoshal, A Modified
Watershed Segmentation Algorithm using Distances
maximum possible intensity value (with m-bit Transform for Image Segmentation, International Journal of
integer values, S will be 2m-1). The results of the Computer Applications, Volume 52 No.12, pp. 47-50,
calculations for the proposed method are given in August 2012
Table I. 5. Mark Polak, Hong Zhang*, Minghong Pi, Mark Polak,
Hong Zhang *, Minghong Pi, Image and Vision Computing,
Vol. 27, pp. 12231227, 2009.
Table I- Statistical measurements.
6. S. Thilagamani, N.Shanthi, A Novel Recursive Clustering
Algorithm for Image Oversegmentation, European Journal
of Scientific Research, Vol.52, No.3, pp.430-436, 2011.
7. Peter Eggleston, Understanding Oversegmentation and
Region Merging, Vision Systems Design, December 1,
1998.
8. Alasdair Mcandrew.2004. Introduction to digital image
processing with MATLAB, Course Technology a part of
cengage learing.
9. Tom Melange, Mike Nachtegael, Etienne E. Kerre.2009.A
Fuzzy Filter for the Removal of Gaussian Noise in Colour
Image Sequences, 1474-1479.
10. S. Beucher and C. Lantuejoul, "Use of watersheds in contour
detection," 1979.
11. F. Meyer and S. Beucher, "Morphological segmentation,"
Journal of visual communication and image representation,
vol. 1, pp. 21-46, 1990
12. L. Vincent, Algorithmes morphologiques a base de files
d'attente et de lacets: extension aux graphes: Paris, 1990.
13. F. Meyer, "Topographic distance and watershed lines,"
7. CONCLUSION Signal Processing, vol. 38, pp. 113-125, 1994.
14. A. N. Moga and M. Gabbouj, "Parallel image component
This paper introduces the Minimum Mean Square labeling with watershed transformation," IEEE transactions
on pattern analysis and machine intelligence, vol. 19, pp.
Error (Wiener) Filtering with watershed algorithm 441-450, 1997.
using distance transform for image segmentation and 15. J. Roerdink and A. Meijster, "The watershed transform:
edge detection. The combined use of these two make Definitions, algorithms and parallelization strategies,"
the present approach efficient for image segmentation Mathematical Morphology, vol. 41, pp. 187-S28, 2000.
and edge detection in a variety of practical situations. 16. P. Soille, Morphological image analysis: principles and
The performance of the proposed approach is applications: Springer-Verlag New York, Inc. Secaucus, NJ,
USA, 1999.
compared with conventional watershed approach for
image segmentation task. The method performs best 17. J. Serra, Image analysis and mathematical morphology:
Academic Press, Inc. Orlando, FL, USA, 1983.
when the objects of interest in the image are well
18. J. Serra and L. Vincent, "An overview of morphological
defined, with strong edges and uniform background filtering," Circuits, Systems, and Signal Processing, vol. 11,
and in such cases, can be electively used as a pp. 47-108, 1992.
replacement of classical image segmentation 19. W. J. Niessen, K. L. Vincken, J. A. Weickert, and M. A.
algorithms. Viergever, "Nonlinear multiscale representations for image
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REFERENCES
1. Sheng Liu ; Zhejiang Univ., Hangzhou, China ; Rui Cao ;

Shengyong Chen ; Zichen Chen, Novel Quality Evaluating
Method for Over-Segmentation Approaches Using Real-
Time Boundary Information, Workshop on Digital Media
and Digital Content Management (DMDCM), pp. 252 256,
15-16 May 2011
295
Wavelet Transform Based Texture Features For:

Content Based Video Retrieval
Mrs.G.Prathibha Mrs. K. Chaitanya
Assistant Professor Assistant Professor
Acharya Nagarjuna University Acharya Nagarjuna University
The image function captured at any time t is
Abstract: the projection of the light distribution in the 3D scene
Communication, knowledge sharing and learning on to a 2D image plane. Let P(.) represent the camera
information in the global manner is done through projection operator so that the projected 2D position
worldwide networking. Multimedia databases and of the 3D point X is given by x=P(X). Let P-1(.)
digital library are rapidly increasing, so efficient denote the inverse projection operator so that X=P -
1
retrieval algorithms need to be developed. The (x) specifies the 3D position associated with a 2D
necessity of today's world is the retrieval of point x. Then the projected image is related to the 3D
information according to user's choice. Content image by
Based Video Retrieval System is based on retrieving ( P ( X ), t ) (X,t) or ( x, t ) (P-1(x),t)
a video sequence from large database. Textual
Annotations used in Google by user has very high
error rate. In this paper Video Retrieval is based on a The function ( x, t ) is what is known as a video
sample image. The focus of this paper is on the video signal[8].
processing aspects, in particular using texture
information of an image for video retrieval. A unique Architecture of CBVR System:
wavelet transforms based texture features for Content
Based Video Retrieval is presented.
The use of Tree Structured Wavelet is proposed for
extracting texture features such as mean and standard
deviation. The comparison of results is done using
Euclidean distance, Square chord distance and Chi-
Square and found that Squared-Chord distance is
better.
Introduction:
Video is the technology of recording, reproducing, or

broadcasting of moving visual images. The science of
searching for documents and information with in
documents is Information Retrieval. Content-Based
Video Retrieval is about searching of videos in large
database. Increased use of Multimedia has resulted in
paving the need for Content Based Video Retrieval
Systems. The Actual content of the image is Fig 1: Architecture of Video Retrieval System
analyzed in Content Based. The term content refers to
Content Based Video Retrieval system consists of
Color, Shape, Texture etc.
two parts: Video Collection and Video Retrieval. The
Video records the emitted and/or reflected light
user needs to have a large number of Video files from
intensity i.e. C(X, t, ) from the objects in the scene which a video needs to be retrieved based on the
that is observed by a viewing system[8]. In general input image. The design of architecture requires three
this intensity changes with both time and space. Let fundamental components a feature transformation,
the spectral absorption function of the camera be feature representation, and similarity function.
denoted by ac( ), then the light intensity distribution Video Collection: The features of the video are
in the 3D world that is visible to the camera is: computed and stored in a feature file. The new video
is registered in video data base.
(X,t)= C ( X , t. )ac ( ) Video Retrieval: Compute feature of a query image
0 and find similar videos from the database.
296
The main emphasis in video retrieval system is based The algorithm [1] of tree structured wavelet
on similarity function. The similarity function transform is as follows:
compares image with a video for approximate or i) Decompose the given image into four sub images
similar matching. It becomes a primary procedure in ii) The energy of each sub image is calculated
Video retrieval system.In the present system, the iii) If the energy of a sub image is significantly
feature file consists of energy parameters: mean and smaller, stop decomposing the sub image since it
standard deviation. The query feature file is used to contains less information
find the matching feature file from the database, iv) If the energy of the sub image is larger, apply the
computed by distance-measured techniques. The best above decomposition procedure.
matched videos are displayed.
Tree Structured Wavelet Transform:

Wavelet transforms are a mathematical means to
perform signal analysis when signal frequency varies
over time. For certain classes of signals and images, Fig2: Three Level DWT
wavelet analysis provides more precise information
about signal data than other signal analysis
techniques. Common applications of wavelet
transforms include speech and audio processing,
image and video processing, biomedical imaging, and
1-D and 2-D applications in communications and
geophysics. Wavelet transforms that use wavelet
frame representations, such as continuous, discrete,
nondecimated, and stationary wavelet transforms.
Wavelet transform decomposes a signal in to a set of
basis functions. These basis functions are called
wavelets. Wavelets are obtained from a single Fig3: Tree structured Wavelet Decomposition
prototype wavelet called mother wavelet by
translations and dilations Video Retrieval Procedure:
The CWT or continuous-time wavelet transform of
f(t) with respect to a wavelet (t) is defined as[7] : The video database is of MPEG-4 format. Size of

t b each video is limited to 10 seconds. Each of the video

1
f (t )
*
W(a,b)= dt is divided into images of size 512x512.
a a Algorithm:
Where a is the scaling parameter and b is the shifting 1. Take a video and divide in to frames
parameter. The parameters a and b are real and * 2. Apply DWT for each frame
denotes complex conjugation. Thus, the wavelet 3. Calculate energy of four sub bands using the
transform is a function of two variables. Both f(t) and formula[2]
(t) belong to L2(R), the set of square integrable 1 M N
functions, also called the set of energy signals. E= x(i, j )
MN i 1 j 1
The Discrete wavelet transform is just a sampled
version of CWT and its computation consumes 4. The sub band with dominant energy is further
significant amount of time and resources, depending decomposed by applying DWT
on the resolution required. The DWT which is based 5. After applying two level DWT take the highest
on sub-band coding is found to yield a fast energy sub band and calculate mean and standard
computation of Wavelet Transform. It is easy to deviation.
implement and reduces the computation time and 6. The above procedure is repeated for all the videos
resources required. in the database
The tree structured wavelet transform is a discrete 8. Decompose the query image by applying two level
wavelet transform which can be applied to output of DWT and then calculate mean and standard
any filter GLL,GLH,GHL or GHH. It is unnecessary and deviation.
expensive to decompose sub signals in each scale to 9. Calculate the difference between video images and
achieve full decomposition[1]. To avoid full query image using Euclidean distance, Manhattan
decomposition energy is considered as a criterion to distance and Chi Square distance.
decide whether a decomposition in needed for 10. The videos below threshold value are retrieved.
particular output.
297
Distance Measures:
Fig5: Distance measures of matched video
Given two feature vectors fx and fy from the same The video with minimum distance is retrieved and
transform, where fx is a query feature vector and fy is displayed as shown in fig.6
the feature vector from database, then the Squared-
chord distance between them is given as:
D( x, y) = n

i 1
2
fx i fy i
where n is the length of feature vector .
The Euclidean distance measure used for retrieval is
given by the formula[2]
n
fxi fyi
D (x, y) = 2 2
i 1
The Chi-Square distance is given by the formula Fig 6: Video retrieval
2
D (x, y) = n fx fy 1
i i Conclusion: Video Retrieval is a major area of
i 1 f f
i f y i
research which is the present need of hour. In most of
the day-to-day applications videos need to be
Experimental Results: retrieved based on the query. Video Retrieval is
The distance measures are calculated between the major challenging area where the retrieval need to be
image query and the video frames of each video using made very fast based on the input from the user.
different distance techniques. The video with Content Based Video Retrieval is
minimum distance is retrieved by using a threshold performed in which a video is retrieved based on a
value. The performance of the three distances is query image. The tree structured wavelet transform is
compared and it was found that squared-chord used to decompose the images. Different distance
distance is the best. The fig. 4 shows the distance measuring techniques are employed for retrieval.
measures of an unmatched video using Euclidean, Further work in this area can be done based on color,
Squared-Chord and Chi-Square. shape, histogram, edges e.t.c. The major drawback of
working with video database is the large number of
video frames which makes the runtime process more.
References:
[1] Tianhorng Chang and C-C, Jay Kuo, Senior Member,IEEE

"Texture Analysis and Classification with Tree-Structured Wavelet
Transform" IEEETransactions on Image Processing,VOL 2, No.4,
October1993
[2] Vaishali Pawar and Milind Mushriff,"Texture Features for
Image Retrieval using Wavelet Transforms" Proceedings of the
International Conference on Cognition and Recognition
Fig4: Distance measures of unmatched video [3] [1] R. Datta, D. Joshi, J. Li, and J. Z. WANG, Image
retrieval: Ideas,influences, and trends of the new age, ACM
The distance measures of the matched video are Computing Surveys, vol.40, no.2, 2008, pp.160.
shown in fig.5. The distances for the matched video
are minimum. The video with minimum distance is [4] Y. Chen and J. Z. Wang, A region-based fuzzy feature
retrieved matching approach to content-based image retrieval, IEEE Trans.
Pattern Anal. Mach. Intell., vol.24, no.9, 2002, pp.12521267.
[5] Y. Chen and J. Z. Wang, Image categorization by learning and

reasoning with regions, Journal of Machine Learning Research,
vol.5, 2004, pp.913-939.
[6] J. Li, J. Z. Wang, and G. Wiederhold, Irm: Integrated region

matching for image retrieval, in Proceedings of the ACM
International Conference on Multimedia, 2000.
[7] Raghuveer M.Rao, Ajit S. Bopardikar, " Wavelet Transforms,

Pearson Education
[8] Yao Zhang, Wang, Video Processing and Communications
BSP Publications, Page 6.
298
Medical Image segmentation using Rough Sets

with optimized centroids
Performance Evaluation of Rough k-means and Rough Fuzzy C-means
N.Anupama1,
Department of CSE,
Acharya Nagarjuna University,
Guntur, Andhra Pradesh, India.
namburianupama@gmail.com
S.Srinivas Kumar2
Department of ECE,
JNT University Kakinada, Kakinada,
samay_ssk2@yahoo.com
E.Sreenivasa Reddy3
Department of CSE,
edara_67@gmail.com
AbstractMedical image segmentation plays a vital correction of functional imaging data, and computer
role in image processing due to the catering needs of integrated surgery [1]. In this paper, Medical image
the medical images in automating, delineating segmentation & rough sets are briefly discussed in
anatomical structures and diagnosis. Very often the section II. Optimized centroid algorithms for Rough
medical images contain uncertain, vague, and k-means and Rough fuzzy C-means are presented in
incomplete data definition. The concepts of lower and section III. Experimental results are presented in
upper approximations of rough sets effectively handle section IV. Concluding remarks are given in section
this data. In this paper, rough sets based clustering is V.
analyzed with respect to K-means and Fuzzy C-
Medical Image segmentation using Rough sets
means algorithms for MRI images of BrainWeb
database and the centroids are optimized using local Medical image segmentation
maxima of histogram. The performance of proposed
There exist many approaches to image segmentation
Rough K-means(RKM) and Rough Fuzzy C-means
such as histogram based methods, edge detection,
(RFCM) are evaluated and compared with classic k-
region growing, split and merge methods, PDE based
means (KM) and fuzzy C-means (FCM).
methods and clustering methods[3].However, in this
Keywordssegmentation; k-means;C-means;Rough
paper clustering techniques are considered where
k-means; Rough fuzzy C-means.
each image pixels is assigned to a cluster such that all
I. INTRODUCTION members in the same cluster are similar. In Classical
K-means clustering, each object must be assigned to
Images acquired through the techniques like
exactly one cluster. On the other hand, fuzzy C-
Magnetic Resonance Imaging (MRI), Ultra sound
means allows object to belong to a cluster to a degree
imaging, Computed Tomography (CT), Positron
of memberships [2, 5, 12].Fuzzy C-means cannot
Emission Tomography (PET), Single-Photon
recover from noise and outliers. A possibilisty
Emission Computed Tomography (SPECT), and
clustering approach by Krishnapuram and Keller [6,
functional MRI (fMRI) are studied and analyzed by
7] was proposed which uses possibilistic type of
interception and perception of the radiologists [9].
membership function to describe the degree of
There is the need for segmentation algorithms for
membership. Possibilistic C-means generates
delineating of anatomical structures and other region
coincident clusters [11]. To handle the uncertainty
of interest for assisting and automating radiologists
and vagueness in images Rough set theory, developed
tasks. There exist numerous biomedical imaging
by Z Pawlak (1981) [10] gained lot of interest.
applications based on segmentation algorithms such
Lingras and West introduced a new clustering
as the extracting tissue volumes, diagnosis,
method, called rough C-means, which describes a
localization of pathology, study of anatomical
cluster by a prototype (center) and a pair of lower and
structure, treatment planning, partial volume
upper approximations [8].
299
Rough sets the brain images are processed to eliminate the

background of the images.Figure2.1 show the
Definition: Let U be the universe and an indiscernible
preprocessing of the images. Then find out the initial
relation A U U. Let X be a subset of U. Then the centroids of the image as per presented in section III
rough set over the approximation space < U, A > is A .Apply the clustering techniques as described in
given by < (X), A(X)>. section III B and III C.
A( X ) X; i Rough K-Means And Rough Fuzzy C-means
X i X
Optimization of centroids using historgram based
method. Initial centroid play a great role in the
A( X ) Xi; segmentation using clustering algorithms like kmeans
X i X and fuzzy for initial membership function. The main
idea is to find the peaks in the histogram and choose
The lower approximation A(X) of a set X is the set the corresponding gray levels for initial cluster
of all elements that definitely belong to X with centers of the K-means, Fuzzy C-means, Rough K-
respect to A. Means and Rough Fuzzy C means.
The upper approximation (X)of a set X is the set of
all objects which can be possibly belong to X with
respect to A.
The boundary region of a set X is the set of all
objects, which can belong to X or does not belong to
X with respect to A.
Set X is crisp, if the boundary region of X is empty.
Set X is rough, if the boundary region of X is
nonempty.
Steps in Algorithm:
Rough sets when applied to clustering ,Let A(Vj) and
1. Find the histogram of the image.
(Vj) be the lower and upper approximations of
cluster Vj, and boundary for cluster Vj is denoted by 2. Identify the local maximum of peaks in the
B(Vj) = { A(Vj) (Vj)}. Then the object Xi can histogram.
obeys the properties :
3. Choose the global maxima as first centroid and
An object Xi can belong to at most one lower
find other local maximas or corresponding gray levels
approximation.
and find the centroids.
An object Xi belonging to A(Vj) also belongs to
(Vj) These optimized centroids Vj are used as the initial
An object Xi is not part of any lower approximation cluster centroids for KM, FCM, RKM and RFKM to
then Xi belongs to two or more upper approximation. find the cluster distances. The Fuzzy membership
Roughsets for image segmentation. values initially can be calculated using the equation 1
based on these optimized centroid Vj using the below
equation.
1
U ij 2
c c
X i V j m1

j 1 k ! j X i Vk (1)
Rough K-Means
Figure 2.1 Phatom t1 weighted images Incorporating rough sets into K-Means is to add the
a)original b)background eliminated. concept of lower and upper approximation which acts
The process involves three step preprocessing, as an interval of rough sets for each cluster [7, 13].
choosing optimized centroids and then clustering the The image pixel that definitely belong to the cluster
image into segments. In preprocessing,as the are kept in lower approximations and the elements
BrainWeb dataset images are in MINC format which neither nor belong to the cluster are kept in the upper
are to be converted to MATLAB readable format approximations and further, calculations are done to
which is achieved by MIPAV software[17] and then
300
move the boundary region elements to clusters hence, Wlower is the weights for lower approximation and
reducing the no of computations. Wupper is the weights for Upper approximation.
Let dij be minimum similarity distance of an element Algorithm for Rough K-Means:
Xi to clusters, by dividing all other distances dik, if
Input: image data(X); number of cluster K, threshold,
the result is less than threshold then Xi belongs to
last_Cj(Image clusters), new_Cj(Image clusters),
A(Vj) and (Vj) and Xi cannot be a member of any
Wlower, Wupper, stop criterion (here =0.001,
lower approximation belongs to (Vj) such that dij is
threshold =2, Wlower=0.5 and Wupper=0.5).
minimum over the k clusters.
Output: new_Cj( segmented image data).
Let
Chose initial centroids as described in section III.A.
T = {j: dik/dij threshold and ij} Vj; 1 <= j<=k; for the k clusters.
(2) Calculate the similarity distance between the data
point Xi to all cluster means Vj. Update the last_Cj
Then we get: with the data points Xi which belong to same
If T then at least one other centroid is close to clusters.
the element. Let dij be the distance from Vj to Xi data point and
dik be the distance between the Xi and the Vk.
If T = then no other centroids are similarly close Determine the nearest centroid for Xi let it be dij.
to the element. Check for other centroid closer to the nearest centroid
if (T ) using equation 1:
if (T )
then Xi (Vj) and Xi (Vk),Xi T then Xi (Vj) and Xi (Vj),Xj T
else Xi (Vj) and Xi A(Vj) else Xi (Vj) and Xi A(Vj)
Compute new centroids Vj based on equation 2.
Calculate the similarity distance between the data
Hence, the centroid calculation is modified to include point Xi to all cluster means Vj. Update the new_Cj
the effects of lower as well as upper approximation. with the data points Xi which belong to same
The value of centroid is given by: clusters.
Repeat step 2 to 6 until |last_Cj-new_Cj<=stop
criteria|.
Stop.
P, if A(V j ) and A(V j ) A(V j ) ,
Rough Fuzzy C-means Rough K-means assign an
object definitely to a cluster based on the interval of
lower and upper approximation where as Rough
Q, if A(V j ) and A(V j ) A(V j ) Fuzzy C-means follows a membership function for
Vj
the degree of belonging of object to a cluster based
on the boundary of rough set. The membership value
Wlower P Wupper Q, if A(V j ) and A(V j ) A(V j ) for the objects belonging to the lower approximations
is 1 and the objects belonging to boundary region has

(3) a membership of [0,1]. Figure 3.1 presents the details.
Where Figure 3.1 shows the membership values at lower and
boundary approximation
x( A (V j ) A (V j ))
Xi
P
A(Vj ) A(V j )
(3.1)
x( A (V j ) A (V j ))
Xi
Q
A(V j ) A(V j )
(3.2) The membership of the objects present in the
boundary is defined by membership function [15].
301
Input: image data(Xi); number of cluster k, threshold,

Wlower, Wupper, stop criterion . (here =0.001,
1,if x j A(Vj ) threshold =0.5,Wlower=0.5 and Wupper=0.5)

1
U ij 2
,if x j A(V j ) A(V )
j
Output: segmented image data
c c
i
X V j
m 1

X V
Chose initial centroids as described in section III.A.

j 1 k ! j i k <= j <=k; for the k clusters.
(4)
Compute Uij using equation 1 for k clusters and Xi
The centers of the fuzzy C-means is calculated using objects.
the
Calculate the new centroid using equation 5.
2
U
c
1
xi v j
m
Vj ij
Update the last_Cj with the data points Xi which
n i 1 x j X belong to same clusters based on the Uij.
Determine the nearest centroid for Xi based on
(5) maximum Uij.
The rough fuzzy C-means partition the objects into Check for other centroid closer to the nearest centroid
clusters by minimizing the objective function. The using equation 7:
objective function is given by equation 5. if (T )
then Xi (Vj) and Xi (Vk),Xi T
else Xi (Vj) and Xi A(Vj)
P1 ,if A(Vj ) and A(Vj ) A(Vj ) , Calculate Uij using equation 4.
Compute the objective function JRF based on
J RF Q1 ,if A(Vj ) and A(Vj ) A(Vj ) equation 6.

Wlower P1 Wupper Q1 ,if A(Vj ) and A(Vj ) A(Vj ) Repeat step 3 to 8 until | last_JRF-new_JRF<= (stop
(6) criteria)|.
stop
where IV. EXPERIMENTAL RESULTS
c 2
p1 xi v j This section presents a set of experiments to validate

i 1 x jA ( x )
the performance of proposed algorithms. The
(6.1) experiment datasets are collected from the BrainWeb
2 database [16] for MRI image. The BrainWeb images
U
c
Q1
m
xi v j ij
are in MINC format which are to be converted to
i 1 x j A (V j ) A (V j )
MATLAB readable format which is achieved by
MIPAV software[17]. The BrainWeb contains
different anatomical structures ground truth images
(6.2) for T1 weighted images among which we consider
Let Uij be maximum membership, by dividing all white matter (WM), gray matter (GM) and
other memberships Uik, if the result is less than cerebrospinal fluid (CSF). The performance is
threshold then Xi belongs to A(Vj) and (Vj) and Xi evaluated on normal t1 weighted images with 1mm
cannot be a member of any lower approximation thickness of phantom, subject04, subject05,
belongs to (Vj) such that Uij is maximum over the subject08, subject20 from BrainWeb. The phantom,
k clusters. subject04, subject05 normal t1 weighted images are
presented in figure 4.1. The extracted WM, CSF, GM
T = {j: Uij /max (Uij) threshold and ij }. segmentation results for phantom image are presented
(7) in figure 4.2 and for subject 04 image are presented in
Then we get: figure 4.3.
If T then at least one other centroid is close to The evaluation of segmentation results are calculated
Xi. using the quantitative comparison with the ground
truth. A number of similarity coefficients methods [4]
If T = then no other centroids are close to the Xi.) are used to specify how well segmentation A
IF (T ) matches a referenced ground truth B.
( )
Algorithm for Rough Fuzzy C-means: Jaccard coefficient =
( )
302
The jaccard coefficient is above 70% means that the

segmentation result is good. Performance of the
algorithm can be also measured using other criteria
[14].The average results of phantom, subject04,
subject05, subject08, subject20 results are presented
in table 4.1 to table 4.5. Figure 4.3 shows the chart
for similarity index.
(g) (h) ` (i)
Sensitivity =
Specificity =
Precision =
Segmentation Accuracy=
The proposed algorithms are implemented in (j) (k) (l)

MATLAB 10.1 on 1.8 GHz Intel core 2 Duo
processor.
(m) (n) ` (o)

Figure 4.2.Segementation results for Subject04 T1 weighted image
(a) (b) (c) of 1mm thickness.(a) Ground truth WM of Subject04, (b) Ground
truth CSFof Subject04, (c) Ground truth GM of Subject04,(d) (e)
Figure 4.1. (a)Ground truth original phantom Image, (f) WM,CSF,GM exctracted using K-means,(g) (h) (i)
(b) Ground truth original Subject04 Image, (c) WM,CSF,GM exctracted using Rough K-means,(j) (k) (l)
WM,CSF,GM exctracted using Fuzzy C-means,(m) (n) (o)
Ground truth original Subject05 Image. WM,CSF,GM exctracted using rough Fuzzy c-means for
Subject04.
Figure 4.2.Segementation results for Phantom normal
T1 weighted image of 1mm thickness.(a) Ground
truth WM of Phantom, (b) Ground truth CSFof
Phantom, (c) Ground truth GM of Phantom, (d)(e)(f)
WM,CSF,GM exctracted using K-means,(g)(h)(i)
WM,CSF,GM exctracted using Rough K-
means,(j)(k)(l) WM,CSF,GM exctracted using Fuzzy
c-means,(m)(n)(o) WM,CSF,GM exctracted using
rough Fuzzy c-means. (a) (b) (c)
(a) (b) (c) (d) (e) ` (f)
(d) (e) ` (f) (g) (h) ` (i)
303
(j) (k) (l)
(m) (n) ` (o)
V.CONCLUSION REMARKS
In this work the segmentation frame work based on
Rough sets is proposed and the results show
significant improvement in accuracy for segmenting
complex MR images of brain.
The experimentation on MR images prove
the efficiency of proposed method compared to
classic models such as k-means and Fuzzy C-means.
The performance is evaluated by visual perception
and quantitative measurements of sensitivity,
specificity, precision, accuracy and similarity.
However, the results depend on the initial centroids
for K-means and initial membership values for fuzzy
c-means. The random initialization of centroid leads
to uncertain clusters and increase computation time.
In this work the optimization of the centroids and
member ship values using histogram local maxima
based method provide accuracy and closer proximity
of centroids to ground truth.
304
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Statistics and Probability, 1967, 281297.
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About Data, Dordrecht, The Netherlands: Kluwer,1991.
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methods in medical image segmentation 1." Annual review of
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means clustering: a case study." Proceedings of ALAR 2006
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305
Audio Authentication and Copyright Protection

Using DWT and Arnold Transform
S Suresh Kumar
(M.Tech-DECS)
GMR Institute of Technology
Rajam, INDIA
resh.sontyana@gmail.com
N.V.Lalitha
Department of ECE
GMR Institute of Technolog
Rajam, INDIA.
lalitha.nv@gmrit.org
Abstract-Digital Audio Watermarking involves the

conce- alment of data within a discrete audio file.
2. TRANSFORMS USED
Digital watermarking is identified as a partial solution
to related problems like illegal reproduction and A. DISCRETE COSINE TRANSFORM(DCT)
distribution of digital media. The technology of
embedding image data into the audio signal and The Discrete Cosine Transform is a technique for
additive audio watermarking algorithm based on one converting a signal into elementary frequency
dimensional discrete wavelet transform(DWT) and components. The most common DCT definition of a
discrete cosine transform(DCT) for the application of 1-D sequence of length N is
copyright protection. This algorithm is realized to
embed a binary image watermark into the audio signal
and improve the imperceptibility of water-marks. The
embedding information used is a binary image and
Arnold Transform is used for image encryption.
Different wavelet filters are considered and the
performance is evaluated. In addition, the performance
of the proposed algorithm is measured in terms of Peak
Signal to Noise Ratio(PSNR).The proposed scheme
achieved good robustness against most of the attacks
such as requantization, filtering, addition and
multiplication of noise. The extracted watermark image
quality is shown by
considering correlation coefficient (CC) value with a B.DISCRETE WAVELET TRANSFORM
suitable scaling parameter for embedding. (DWT)
INTRODUCTION The Wavelet Series is just a sampled version of

CWT and its computation may consume significant
The rapid development of the internet and the digital amount of time and resources, depending on the
information revolution cased significant changes in resolution required. The Discrete Wavelet Transform
the global society, ranging from the influence on the (DWT), which is based on sub-band coding is found
world economy to the way people nowadays to yield a fast computation of Wavelet Transform. It
communicate. Digitizing of multimedia data has is easy to implement and reduces the computation
enabled reliable, faster and efficient storage, transfer time and resources required.
and processing of digital data. It also leads to the
consequence of illegal production and redistribution The foundations of DWT go back to 1976 when
of digital content. Digital watermarking is identified techniques to decompose discrete time signals were
as a partial solution which allows content creator to devised. Similar work was done in speech signal
embed hidden data such as author or copyright coding which was named as sub-band coding. In
information into multimedia content. In general, all 1983, a technique similar to sub-band coding was
audio watermarking algorithms work by exploiting developed which was named pyramidal coding.
the perceptual property of Human Auditory System Later many improvements were made to these
(HAS).
306
coding schemes which resulted in efficient multi- TN = periodicity

resolution analysis schemes.
3. Watermark Embedding Algorithm
In CWT, the signals are analyzed using a set of basis
functions which relate to each other by simple The process of watermark embedding is shown in
scaling and translation. In the case of DWT, a time- Figure l.
scale representation of the digital signal is obtained
using digital filtering techniques. The signal to be Step 1: Sample the original audio signal at a sampling
analyzed is passed through filters with different rate (fs) and partition the sampled file into frames
cutoff frequencies at different scales. each having certain samples.
The 1-D wavelet transform is given by : Step 2: Perform DWT transformation on original
audio signal. This operation produces Two sub-
bands: A, D. The D represents the Details sub-band,
and A represents the approximation sub-band.
Step3: Apply Arnold Transform and DWT to
watermark image.
Wavelet transform decomposes a signal into a set of
basis functions. These basis functions are called Step 4: Embed the transformed watermark
wavelets. Wavelets are obtained from a single coefficients into the DWT transformed original audio
prototype signal.
wavelet y(t) called mother wavelet by dilations Step 5: Apply the inverse DWT operation to obtain
and each watermarked audio frame.
shifting:
a = scaling parameter
b = shifting parameter
C. ARNOLD TRANSFORM
The KxK binary watermark image W is transformed

into W' by Arnold transformation to lower the
autocorrelation coefficient of image and then the
confidentiality of watermark is strengthened. Arnold
transformation is periodic and when it is iterated
some times the original signal will be obtained. The
Arnold Fig.1 Watermark Embedding
transformation is given by 4. Watermark Extraction Algorithm
The process of watermark extraction is shown in

Figure 2.
Arnold transformation is periodic and when it is Step 1: Sample the watermarked audio signal at a
iterated some times the original signal will be sampling rate (f ) and partition the sampled file into
obtained. frames each having certain samples.
Step 2: Perform DWT transformation on

watermarked
audio signal.
Step 3: Obtain the DWT coefficients of the original

Where N = no of pixels audio signal.
307
Step 4: Subtract the coefficients of original signal

from
the watermarked signal coefficients.
Step 5: Apply IDWT to the values obtained in Step

4.
Step 6: Perform Arnold operation to the

valuesobtained
in step 5.
Figure 3b:original audio
6. Imperceptibility Test
Imperceptibility is related to the perceptual quality

of the embedded watermark image within the original
audio signal. To measure imperceptibility, we use
Peak Signal-to-Noise Ratio (PSNR) as an objective
measure, and a listening test as a subjective measure.
For subjective quality evaluation, a listening test was
performed with ten listeners to estimate the Mean
Opinion Score (MOS) grade of the watermarked
signals for four different signals.
5. Expermental Results Each listener was presented five times with the pairs
of original signal and the watermarked signal and
In the experiments a gray image of size 256x256 is asked to report whether any difference could be
used as watermark shown in Fig.3b & original audio detected between the two signals. The MOS criteria
signal in Fig.3a. are listed in Table 1. and the MOS values are
tabulated in Table2.
Figure 3a: watermark image
Experiments were conducted on four different classes

of signals such as speech, pop music, rock music and PSNR
instrumental as they differ in spectral properties. The
performance of the proposed technique is evaluated PSNR is defined using the following equation:
based on MOS criteria, PSNR and Pearson
Correlation. PSNR= 10log10(X2/MSE)
308
In the above equation, X is the maximum fluctuation

in
the images. MSE is the mean square error and
defined as in equation.
Where I and I are the watermark image and the

extracted watermark image respectively.
CORRELATION COEFFICIENT (CC)
Normally correlation coefficient is used to measure

the
quality of the image. The correlation coefficient is
given by:
Figure 4a:comparision of PSNR between DCT &
DWT
Where r = correlation coefficient

A= extracted image
B = original image
A bar and B bar are the means of A and B
respectively.
Figure 4b:Comparision of CC between DCT & DWT
Figure 4c: Comparision of PSNR for speech signal

with
different Multiplication factors
309
Extracted watermark images:

REFERENCES
[1] N.V Lalitha, G.Suresh, T.Prabhakar Audio Authentication

Using Arnold and Discrete Cosine Transform in IEEE
International Conference on Computing, Electronics and Electrical
Technologies ICCEET-2012 Organised by EGS Pillay College of
Engineering, Tamilnadu to be held on March 22-24, 2012.
[2] Robust Audio Watermarking in the Time Domain. Paraskevi
Bassia, Ioannis Pitas, Senior Member, IEEE, and Nikos Nikolaidis,
Associate Member, IEEE. IEEE transactions on multimedia, vol. 3,
no. 2, june 2001
[3] Nedeljko, Cvejic "Algorithms for Audio Watermarking and
SteganoGraphy",ISBN 951-427384 2, OULU University press.
[4] Kaengin.S, Airphaiboon.S, Patthoumvanh.S, New technique
for embedding watermark image into an audio signal.
Communication and Information Technology,(ISCIT)2009.
Figure 4d: extracted watermark images [5] The Discrete Cosine Transform(DCT): Theory
and Application, Syed Ali Khayam, De partment of
Comparision of extracted images with different Electrical & Computer Engineering, Michigan State
University, March 10th, 2003.
multiplicationfactor(a) [6] Liu Ping Feng, Liang Bin Zheng, Peng Cao,
"A DWT-DCT Based Blind Watermarking Algorithm
for Copyright Protection", IEEE Transactions, 2010.
[7] Vivekananda K. Bhat, I. Sengupta, and A.
Das, Audio Watermarking Based on Quantization in
Wavelet Domain, in ICISS 2008, LNCS 5352, pp. 235
-242, 2008.
[8] M. Steinebach and J. Dittmann,
Watermarking-based digital audio data
authentication, EURASIP J. Applied Signal Proc, vol.
2003, no. 10, pp. 10011015, 2003.
[9] Audio Authentication based on Distributed
Source Coding David Varodayan, Yao-Chung Lin and
Bernd Girod Information Systems Laboratory, Stanford
University.
[10] 2-Dimention Digital Audio Watermarking
Based on Integer Wavelet Transform Wang Xiangyangt
Cui Yongrui Zhao Hong School of computer and
information technique Liaoning Normal University,
Dalian, China
Figure 4e: Comparision of extracted images with

different multiplicationfactor(a)
7. Conclusion
The proposed scheme achieved good robustness

against most of the attacks such as requantization,
filtering, resampling and cropping. The extracted
watermark image quality shown by considering
correlation coefficient(CC) value with a suitable
scaling parameter and PSNR is calculated which
produced better result compared to DCT based
algorithm.
310
Visual Enhancement for better quality of Images

and Displays using skeletonization under
Low-Backlight
Sirisha.M 1, M.Padma Sailaja 2, P.V.Subbaiah 3
1
IEEE Member, sirishamalluri@ieee.org, Dept.Of ECE, DVR&DHS MIC College of Technology,
2
M .Tech Student, magantipadmasailaja@gmail.com, DVR&DHS MIC College of Technology,
3
Ph.D, pvs_ece2000@yahoo.co.in, Principal, Amruth Sai Institute of Science &Technology,
AbstractPropose a system which uses techniques that gradient image decomposition to separate the
provide visual quality of images and displays under image in to base layer and detail layer. After that
Low backlight conditions for saving the battery. This using median-preserving gamma correction output
method first proposes a morphological skeleton is levels of a monitor is altered. Using JND-based
a skeleton representation of a shape of an image, Boosting for detail . Enhancement of an image and
computed by means of operators. Then using the results show that this approach performs better
constrained L0 gradient image decomposition to
than the existing methods.
separate the image in to base layer and detail layer.
Using median-preserving gamma correction output II. PROPOSED SYSTEM
levels of a monitor is altered and using JND-based
Boosting for detail enhancement of an image will be
done and results show that this approach performs
better than the existing methods.
I. INTRODUCTION
Displays are the largest power consuming

components on a modern mobile device like Touch
screens and pad-like devices. The needs of displays
increase now a days. It is hard to save the battery life
of displays on the handheld devices. Doing the
literature survey on backlight scaled displays [5]
and also the power constrained contrast
techniques [6] and [7] for different syntax while
implementation of techniques. This method
proposed the concept of Backlight Scaling and also
enhancing the image signals for better visual
quality of displays and images for saving the
battery life. Using only 40% backlight more
battery power will be saved. To maintain image Fig.1. System Pipeline
quality under different lighting conditions is difficult
to portable multimedia devices. Depending on the Human visual system (HVS) , this
method proposes the algorithm that it will preserves
This paper proposes a method which uses techniques the perceptual quality of images displayed under low
that provide visual quality of images and displays back-light conditions. In this system the brightness of
under Low backlight conditions for saving the a pixel on the display is the product of the
battery. Studying different image enhancement transmittance and backlight illumination. Figure:1
techniques like color image enhancement shows the system pipeline. Visual enhancement work
technique[3] and JND based enhancement techniques for all types of backlight-scaled displays with better
[4].So finally making use of these concepts this image quality is proposed in this method. The H and
method first proposes a morphological skeleton, is S given directly to the output block. The V is having
a skeleton representation of a shape of an image, the illumination signal V(x,y) is taken for our system.
computed by means of operators. Then using
constrained L0
311
A. Skeletonization no.of non-zero gradient. Where

C x , y # x , y || B1 x , y 0
In this method after converting RGB to HSV which is
a gray scale image. A grayscale digital image is an
Where # denotes the numbers of pixels while
image in which the value of each pixel is a
denotes the difference operator. Since the L0-norm of
single sample, that is, it carries only
gradient represents the numbers of non-zero gradient,
intensity information. From this the illumination
it is generally counted on the positions containing
signal which is taken separately is used for the
strong edges. Based on this observation multiplying
skeletonization. Skeleton of a shape is a thin version
the Edge (x,y) to the C(x,y) [1]
of that shape that is equidistant to its boundaries. In
this the skeleton usually has the properties of the
shape,

C x, y # x, y Edge x, y . B1 x, y . 0 (5)
thosearesuchasits connectivity, topology, length, dire Here Edge(x,y) obtained by adopting the first order
ction, and the width. They contain all the information Gaussian (FDOG) edge detection on V1(x,y) with a
necessary to reconstruct the shape. Here to perform threshold. The thresholds of FDOG and edge map are
skeletonization operation on a grayscale image by set to 0.75 and 0.2, respectively. Introducing
converting it in to Binary image. Then Binary Edge(x,y) in to minimization can eliminate some
skeletons can be retrieved by using the MATLAB inappropriate and speed up the whole process.In this
function bwmorph with the operation set to skel paper gauss gradient is used as edge detection
and the parameter N, the number of operations to be technique. Gradient using first order derivative of
repeated, set to infinity, which means the operation Gaussian for edge detection. This function outputs
will be repeatedly executed until the image does not the gradient image using 2-D Gaussian kernel.
change any more.Skeletonization is also used to Determining the size of the kernel and generating 2-D
reduce all objects in an image to lines, without Gaussian kernel along both x and y directions.
changing the essential structure of the Applying 2-D filtering by introducing Edge(x,y)
image.figure:2(a) shows the input gray scale image process can eliminate in appropriate things. So by
and figure:2(b) shows the binary image and taking the absolute values of x and y in a gauss
figure:2(c) shows the skeleton of binary image. gradient this method can detect the edges or
boundaries of a signal. As it is tough to calculate the
(4) referring [2] to approximate the L0-norm of
gradient by introducing the two auxiliary variables of
ax(x,y) and ay(x,y) corresponding to the two variables
shown below here:
( B 1 ( x , y )) ( x ) and ( B 1 ( x , y )) ( y )
Fig. 2. The figures shows : (a) Input gray scale image, (b) Binary Now the L0 gradient minimization is represented as:
V1x, y B1x, y .Cx, y
image, 2
(c) Skeleton of binary image. min x, y
B1,ax ,ay
So finally after applying the skeletonization operation
to the illumination signal of V(x,y) it will get the B x, y 2
B1x, y
2
. ax x, y ay x, y
1
V1(x,y) which is used for further process. x y
(6)

D. Constrained L0 Gradient Image Decomposition
Where C(x,y) is represented in equation (5) and is
Now V1(x,y) is the signal after skeletonization
an automatic tuning parameter to control the
operation. So first this method Decompose the
similarity between variables ax(x,y) and ay(x,y) and
illumination signal V1(x,y) in to base layer B1(x,y)
their corresponding gradients. The parameters when
and detail layer D1(x,y) using the constrained L0
doing the L0 smoothing while getting the base layer
gradient minimization. The original L0 gradient
the smoothing weight is set to 0.01, while is set to
minimization is represented as follow
0.02 and is multiplied by 2 during iterations, kappa
Parameter that controls the rate, kappa = 2 is
|V1 ( x, y ) B1 ( x, y ) | .C ( x, y )
2
min ( 4) suggested for natural images. max has the fixed value
B ( x, y ) ( x, y ) of 1e5 [2].Equation (6) is solved by alternatively
minimizing B1, ax, ay. We fix one set of variables
In L0-gradient minimization C(x,y) represents the while obtain another set of variables.
312
gamma is not altered, the images on the screen would

1) Finding B1: be dark and difficult to see. By applying gamma
correction, the brightness and contrast of the display
Fixing ax(x,y) and ay(x,y) to obtain B1(x,y). This is by are enhanced, making the images appear brighter and
solving the quadratic cost function in a linear system: more natural looking. Here in this system it uses the
Vx, y B1x, y gamma correction to alter the output levels of the
1 2
min
B1 x, y signals applying the scaling to the input signal.
B1x, y B1x, y
2 2
ax x, y ay x, y 7
x y The concept of gamma can be applied to any
nonlinear relationship. For the power
2) Finding ax and ay: law relationship Vout = Vin, the curve on a loglog
plot is a straight line, with slope everywhere equal to
In this iteration, fixing B1(x,y) to obtain new ax (x,y) gamma. (Slope is represented here by
and ay(x,y) by solving the cost function: the derivative operator):
B1 x , y
2

min C x, y a x x, y d log(v out )
a x ,a y x (11)
d log(vin )
B1 x , y
2
a y x, y 8
y Based on the gamma represented in the median
preserving gamma correction is:
logmedianB1 x, y
Where the L0-norm of gradient can be modeled as:
(12)
logmedians.B1 x, y

C x, y Edge x, y .H ax x, y a y x, y (9)
This method adopt gamma correction and let the
median of the scaled brightness approximate to the
median of the original brightness this design forces
Where the H(|ax(x,y)|+|ay(x,y)|) is the binary function. the final scaled base layer to maintain the brightness
Returning 1 when (|ax(x,y)|+|ay(x,y)|) 0 and of the original full backlight.
returning 0 otherwise.By calculating (7) and (8) final Suppose if <1 the image is weighted toward higher
Base layer B1(x,y) is obtained from the L0-gradient (brighter) output values. If >1 the image is weighted
minimization.The illumination signal is V1(x,y) from toward lower (darker) output values. If = 1 the
the skeletonization operator. So the detail layer is transformation has no effect on the image.An
obtained by the difference between illumination underexposed photograph can be corrected using
signal V1(x,y) and the base layer B1(x,y). gamma correction with <1. An overexposed
photograph can be corrected using gamma correction
i.e. D1(x,y) = V1(x,y) B1(x,y) with the > 1.Using the equation (12) the value of
(10) =0.25 is obtained. By this if the < 1 the image is
weighted toward higher (brighter) output values. The
The detail layer which is useful for the JND for final scaled base layer will come out after applying
extracting the details and boosting the low intensity scaling of 40% backlight which is denoted as denoted
regions of the image. as Bs(x,y).
The final scaled base layer can be obtained:
E. Median-Preserving Gamma Correction

As this method compress the range of illumination to B ( x, y) J max
Bs ( x, y) s.W . 1 (13)
simulate low backlight that is applying the 40 % W
scaling, adopt a Gamma Correction - controls the
overall brightness of an image and is a setting that Where W is the white value (equal to 255 in 8-bits
determines how bright the output of the display will image or 1 in normalized images) and J max is the
be. Therefore, "gamma correction" is used to alter the maximum value of human JND. J max= + *W
output levels of a monitor. The gamma setting affects referring [2]. Where and are constants.
both the brightness and the contrast of the display. F. JND Based Detail Boosting
The reason gamma correction is used is because the
input signal, or voltage, sent to a monitor is not high Just-noticeable-difference (JND) is the smallest
enough to create a bright image. Therefore, if the stimulus for human vision to perceive the difference
313
between the operating pixel intensity and its

background intensity. Since the flowers are salient as
compared to the rest regions, the flowers are
enhanced more strongly by using our JND-Based
detail boosting.In addition to the new scaled Bs(x,y),
it also enhance the detail layer D1(x,y) to ease human
perception. This method proposes a detail boosting
method based on JND. Two-layer image
decomposition architecture based on the just
noticeable difference (JND) theory is proposed for
extracting the details, and a boosting scheme is
applied to the details that are in the low intensity
regions of the image. Modifying Jmax=+.*W for
the purpose of obtaining the JND is:
JNDx, y . Bs x, y (14)
The parameters of JND are is set to 4 and is set to

0.015 and Jmax is 0.02. The enhanced detail layer
DEN(x,y) is obtained from the JND-Based detail
Boosting by using
W JNDx, y
DEN x, y D1x, y . 15
W
The final enhanced V-signal can be obtained by

combining the Enhanced detail layer DEN(x,y) from
the JND-Based detail boosting and scaled base layer
Bs(x,y) from median-preserving gamma correction.
VS(x,y) = Bs(x,y) + DEN (x,y) (16)
Now the final enhanced V-Signal is denoted as

Vs(x,y)and the original Hue signal H(x,y) and
saturation signal S(x,y) of the input HSV image will
combined in the output block in order to get the Fig. 4. The results shows: (a) original image, full100% backlight:
s=1, (b) original image, linear-scaled to 40% backlight: s=0.4, (c)
output in the form of HSV. Finally the method our result using 40% backlight: s=0.4 with improved visual quality
converts the HSV output in to RGB image. of image and also display
III. RESULTS IV. CONCLUSION
The proposed method provides the visual quality

for both images and display of systems by visual
enhancement under low back-light conditions. So
for saving the battery life, this method propose the
concept of Backlight Scaling and also enhancing the
image signals for better visual quality of displays
and images. By using only 40% backlight also
more battery power is saved. This method not only
prolong the battery life but also enhance visual
quality. In this work, visual enhancement framework
for all kinds of backlight-scaled displays with better
image quality are proposed. The Experimental result
314
shows that the proposed system performs better than

the existing systems.
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1. Soo-Chang Pei, Fellow, IEEE, Chih-Tsung Shen, and Tzu-
Yen Lee Visual Enhancement Using Constrained L0
Gradient Image Decomposition for Low Backlight Displays
IEEE Signal Processing Letters, VOL. 19, NO. 12,
December 2012
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gradient minimization,ACM Trans Graph., vol. 30, no. 6,
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enhancement based on single-scale retinex with a JND-based
nonlinear filter, in IEEE Symp. Circuit and Syst., 2007, pp.
39483951.
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based enhancement of perceptibility for dim images, in
IEEE Conf. Image Process., 2008, pp. 17521755
5. P.-S. Tsai, C.-K. Liang, T.-H. Huang, and H. H. Chen,Image
enhancement for backlight-scaled TFT-LCD displays, IEEE
Trans. Circuits Syst. Video Technol., vol. 19, no. 4, pp. 574
583, 2009
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contrast enhancement for emissive displays based on
histogram equalization,IEEE Trans. Image Process., vol. 21,
no. 1, pp. 8093, Jan. 2012.
315
An Optical Flow Approach to Robust Face

Recognition Under Expression Variations
Mr. V. Muralikrishna1 Dr. M. Madhavi Latha2, Dr. M. Sathya Sai Ram3,
1
Dept. of ECE,KKR & KSR Institute of Technology & Sciences, Guntur.
2
Director of ITC & Professor of ECE, JNTUH, Hyderabad.
3
HOD, ECE, Chalapathi Instotute of Technology & Sciences, Guntur.
veerepalli75@gmail.com
AbstractFace recognition is one of the most intensively reduction algorithm or a robust classification
studied topics in computer vision and pattern technique may yield more accurate recognition
recognition, but few are focused on how to robustly results, they usually require multiple training images
recognize faces with expressions under the restriction of for each subject. How- ever, multiple training images
one single training sample per class. A constrained
optical flow algorithm, which combines the advantages
per subject may not be available in practice.
of the unambiguous correspondence of feature point Some authors have proposed different approaches to
labeling and the flexible representation of optical flow deal with facial expression variations in face
computation, has been developed for face recognition. These algorithms can be roughly
recognition from expressional face images. In this divided into two main categories: morphable model
paper, we propose an integrated face recognition system based and optical flow based. The basic idea in the
that is robust against facial expressions by
combining information from the computed intraperson former category is to warp images to similar
optical flow and the synthesized face image in a global face geometries as the ones used for training.
probabilistic framework. Our experimental results show The concept of separately modeling texture and
that the proposed system improves the accuracy of geometry information has been applied in active
face recognition from expressional face images. shape model and active appearance model
Index TermsConstrained optical flow, face (ASM/AMM) [3], [4]. Face geometry is defined
recognition. via a set of feature points in ASM, while face
texture can be warped to the mean shape in AAM.
INTRODUCTION Ramachandran et al. [15] presented preprocessing
FACE recognition has been studied for several steps to convert a smiling face to a neutral face. Li
decades. Comprehensive reviews of the related works et al. [9] applied a face mask for face geometry
can be found in [14]. Even though the 2-D face normalization and further calculated the
recognition methods have been actively studied in the eigenspaces for geometry and texture separately, but
past, there are still some inherent problems to be not all images can be well warped to a neutral
resolved for practical applications. It was shown that image because of the lack of texture in certain
the recognition rate can drop dramatically when the regions, like closed eyes. Moreover, linear warping
head pose and illumination variations are too large, was usually applied, which was not consistent to the
or when the face images involve expression nonlinear characteristics of facial expression
variations. Pose, illumination, and expression movements.
variations are three essential issues to be dealt with in The other category is to use optical flow to compute
the research of face recognition. To date, there was not the face warping transformation. Optical flow has
much research effort on overcoming the expression been used in the task of expression recognition [5].
variation problem in face recognition, though a number However, it is difficult to learn the local motion in
of algorithms have been proposed to overcome the the feature space to determine the expression
pose and illumination variation problems. change for each face, since different persons have
To improve the face recognition accuracy, researchers expressions in different motion styles. Martinez [12]
have applied different dimension reduction proposed a weighting method that independently
techniques, including principle component analysis weighs the local areas which are less sensitive to
(PCA) , linear discriminant analysis (LDA) [10], expressional changes. The intensity variations due to
independent component analysis (ICA) [1], expression may mislead the calculation of optical
discriminant common vector (DCV) [2], kernal-PCA, flow. A precise motion estimation method was
kernal-LDA, kernal-DCV [7], etc. In addition, proposed in [11], which can be further applied for
several learning techniques have been used to train expression recognition. However, the proposed
the classifiers for face recognition, such as SVM. motion estimation did not consider intensity
Although applying an appropriate dimension changes due to different expressions.
316
In this paper, we focus on the problem of face brightness constancy constraint, however, is not valid
recognition from a single 2-D face image with facial in many circumstances. Therefore, with the
expression. Note that this paper is not about facial generalized dynamic image model (GDIM)
expression recognition. For many practical face proposed by Negahdaripour and Yu [13], Teng et al.
recognition problem settings, like using a passport [16] generalized the optical flow constraint to
photo for face identification at custom security or
identifying a person from a photo on the ID card, it is where the subscript denotes the location, is the
infeasible to gather multiple training images for each concatenation vector of all the flow components an
subject, especially with different expressions. and all the where and denote the multiplier
Therefore, our goal is to solve the expressive face and offset factors of the scene brightness variation
recognition problem under the condition that the field,is the image intensity function, the subscripts ,
training database contains only neutral face images
, anddenote the spatiotemporal partial derivatives,
with one neutral face image per subject. In our
is a point in the spatiotemporal domain, and
previous work [8], we combined the advantages of
the above two approaches: the unambiguous is the motion vector at the point . They
correspondence of feature point labeling and the proposed to minimize the following discrete energy
flexible representation of optical flow computation. function, which is defined with an adaptive
A constrained optical flow algorithm was proposed, smoothness adjustment scheme to constrain both flow
which can deal with position movements and components ( and) and all the brightness
intensity changes at the same time when handling the variation multiplier and offset factors (and)
corresponding feature points. With our proposed constrained
optical flow algorithm, we can calculate the
expressional motions from each neutral faces in the
database to the input test image, and estimate the
likelihood of such a facial expression movement.
Using the optical flow information, neutral images in
the database can be further warped to faces with the
exact expression of input image. In this paper, we
propose to exploit the two different types of
information, i.e., the computed optical flow and the
synthesized image, to improve the accuracy of face
recognition. Experimental validation on the brightness variation multiplier and offset factors,
Binghamton University 3-D Face Expression (BU- and are the parameters controlling the degree of
3DFE) Database is given to show the improved smoothness in the motion and brightness fields, is the
performance of the proposed face recog nition system. set of all the discretized locations in the image
Since we do not attempt to solve the automatic facial domain, is the weight for the data constraint, and
land mark localization problem in this paper, the , , , and are the weights for the
facial land-mark points in our experiment are labeled corresponding components of the smooth constant.
manually. In our application, we regard a neutral face image
The remainder of this paper is organized as and an expressional face image as two adjacent time
follows. We briefly review the constrained optical instance.
flow computational technique in Section II. The
proposed expression-invariant face recognition However, the motion vectors of the facial feature
system is presented in Section III. Section IV gives points, which were used only as references for
the experimental results by applying the proposed interpolating computation in ICPCG, cannot be
expression-invariant face recognition algorithm. guaranteed to be consistent in the final converged
Finally, we conclude the paper in the last section. optical flow. In order to guarantee the computed
optical flow to be consistent with the motion vectors
II. CONSTRAINED OPTICAL FLOW at these corresponding feature points, we modify
COMPUTATION the unconstrained optimization problem in the
original formulation of the optical flow estimation to
The computational algorithms of traditional optical a constrained optimization problem [8] given as
flow cannot guarantee that the computed optical flow follows:
corresponds to the exact pixels in different images,
since the intensity variations due to expression may
mislead the computation of optical flow. The
317
where is the set of feature points and is the

specified optical flow vector at the feature point.
A modified ICPCG procedure was applied to solve
this constrained optimization problem and the details
are referred to [8]. By solving this optimization
problem with the displacements of the feature points
imposed as hard constraints, we can compute the
motion vectors at all pixels in the entire image for
different expressions or subjects.
III. PROPOSED FACE RECOGNITION SYSTEM
A. Basic Concept
We treat the expression-invariant face recognition
system as a probabilistic maximum a posteriori
(MAP) classification problem. To do this, we
formulate the problem as follows:
Fig. 1. Symbolizations of
operators
Furthermore, the prior probability for each candidate

is assumed equally probable, i.e., is constant for all
. The formulation can be simplified as
Fig. 2. Illustration of decomposing input optical flows
318
319
Fig. 7. (a) Face region selection. The 21 feature points on (b) a

neutral face image and (c) a surprised face image.
tip and the other six around the mouth region. With
the labeled points, the distance between the outer
corners of both eyes is used as the reference to
normalize face images (0.5, 0.5, 0.5, and 1.5 times to
left, right, top, and bottom, respectively), which is
depicted
Fig. 7.
B. Benchmark Test
Since our goal is to solve the expressive

facerecognition problem under the restriction of a
single training sample per class, the training database
in the benchmark contains only neutral face images
with one neutral face image per subject, i.e., 100
training face images for 100 classes in total. There
are 24 expression variant images for each subject, and
we used totally 2400 images for all 100 subjects for
testing. In the training phase for the original data, we
first use PCA to compute the low-dimensional vector
for all the 100 neutral images for all subjects. In the
training phase, we use all the neutral images of all
subjects in the training dataset for computing the
PCA subspace by preserving 95% energy, which
yields 51 eigenvectors. In the testing phase, the input
A. Preprocessing image is projected to the PCA subspace and classified
We manually labeled 21 feature points, including by the nearest neighbor classifier. The average
three points for each eyebrow and four points for each recognition rate is 56.88%, used as a benchmark,
eye, one at the nose and the recognition results for all different
expressions and levels are shown in Fig. 8(a). We can
see that the surprise and disgust expressions give the
worst results, and the result indicates that higher
facial expression levels lead to worse recognition
rates. This is consistent with the statement in [6] that
the correlation between an image and another
image is directly related to the Euclidean distance in
the original space, i.e., , if all images are
normalized to zero mean and unit variance. Fig. 8(b)
shows the result of direct subtraction without PCA
preprocessing, which is slightly better than Fig. 8(a).
Fig. 6. Sample images in BU-3DFEDB. The left-top We
most is the neutral face. The others are the face
images with angry, disgust, fear, happy, sad, and
surprise expressions in columns from left to right with
increasing levels in rows from top to bottom
320
images are shown in Fig. 10. We apply a mask,

shown in Fig. 10(b), and defined from the global
neutral face[Fig. 10(a)], to extract the region of
interest. Moreover, the region inside the mouth is dis-
carded, as illustrated in Fig. 10(e). Both the optical
flow and the grayscales of the synthesized image
within the mask are used in the face recognition
process. For an input image, as depicted in Fig.
10(d), we first position the corresponding mask [Fig.
10(e)] to obtain the masked image [Fig. 10(f)]. After
that,
Fig. 8. (a) Recognition rates from the original images

after PCA reduction under different expressions and
levels (with average recognition rate 56.88%). (b) Fig. 16. Recognition rates with different numbers of
Recognition rates from the original images by direct training data.
subtraction under different expressions and levels
(with average recognition rate 60.71%).
Fig. 9. Recognition rates from weighted optical flow

result proposed by Mar- tinez [12] (with average
recognition rate 67.36%).
C. Face Recognition With the Proposed System
In our proposed system, an extra training dataset is

needed for constructing the prior distribution for the
expression motion as described in Section III-B.
Among the BU-3DFE database,
34 subjects are randomly selected for intraperson
optical flow training, and the remaining 66 subjects
are used as the testing set. In (9), the dimensions of
intraperson optical flows are reduced by using PCA
with 99% energy preservation and all the Gaussians
are equally weighted.
As described in the previous section, we follow the

flowchart shown in Fig. 5. Some experimental Fig. 17. (a) Original normalized image using feature points
321
without noise; (b) normalized image using feature points with computational complexity of the proposed face
small noises; (c) size difference between normalized images with
and without noises. recognition approach.
Table I :Computational complexities for a testing

procedure. Note that c is the total number of
candidates
Fig. 18. Recognition rates using feature points and normalized

face images with noise and the integrated information (with
average recognition rate
the training data size is increased, while the REFERENCES
recognition rates of using only synthesized images
[1] M. S. Bartlett, J. R. Movellan, and T. J. Sejnowski, Face
are independent of the training data size.The recognition by independent component analysis, IEEE Trans.
performance of the proposed face recognition Neural Netw., vol.13, no. 6, pp. 14501464, Nov. 2002.
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Trans. Pattern Anal. Mach. Intell., vol. 27, no. 1, pp. 413, Jan.
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In this paper, we proposed a 2-D expression- common vectors for face recognition, in Proc. Int. Conf. Machine
Learning and Cybernetics, Guangzhou, China, Aug. 1821, 2005,
invariant face recognition system based on pp. 46054610.
integrating the optical flow information and image [8] C.-K. Hsieh, S.-H. Lai, and Y.-C. Chen, Expression-invariant
synthesis. Only one neutral image for each candidate face recognition with accurate optical flow, in Proc. PCM, Hong
subject is needed in our face recognition system. Kong, Dec. 1114, 2007.
[9] X. Li, G. Mori, and H. Zhang, Expression-invariant face
Two kinds of intraperson optical flow fields, recognition with expression classification, presented at the 3rd
and , were computed and Canadian Conf. Computer and Robot Vision, Jun. 2006.
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Trans. Pat-tern Anal. Mach. Intell., vol. 23, no. 2, pp. 228233,
and expressive image synthesis, respectively. Feb. 2001.
The proposed algorithm combines the face image [11] A. M. Martinez, Matching expression variant faces, Vis.
comparison and optical flow prior information in a Res., vol. 43, pp. 10471060, 2003.
probabilistic MAP framework. As shown from the [12] A. M. Martinez, Recognizing expression variant faces from a
singlesample image per class, presented at the IEEE Conf.
experimental results, the proposed face recognition Computer Vision and Pattern Recognition, Jun. 2003.
system significantly improves the accuracy of face [13] S. Negahdaripour, Revised definition of optical flow:
recognition on expressional face images. How- ever, Integration of radiometric and geometric cures for dynamic scene
the proposed integrated system is more analysis, IEEE Trans. Pattern Anal. Mach. Intell., vol. 20, no. 9,
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I. In the future, we will aim to reduce the
322
A Joint Encryption and Watermarking System

B. Santosh Kumar
Digital Electronics and Communication Systems
GMR Institute of Technology
Rajam, A.P.,India. Santuece18@gmail.com
T. Geetamma, Assistant Prof.,

,Dept. of Electronics & Communication Engg.,GMR Institute of Technology
Rajam, A.P., India. tgeetamma@yahoo.co.in
Abstract Multimedia and communication information is no longer protected and it becomes

technologies offers new means of sharing and remote hard to verify its integrity and its origin. From this
access to data. In any information systems data point of view, these cryptographic means, especially
confidentiality, authentication, integrity and encryption, rather appear as an a priori protection
nonrepudiation services are usually required. In this
mechanisms.
paper, a jointencryption/watermarking system is for
thepurpose of protecting images. It merges a chaotic
mapping, anencryption algorithm which is a block Watermarking has been proposed as a complementary
cipher algorithm(e.g.,AES) and a substitutive blind and mechanism to improve the security of aerospace
nonblind watermarkingalgorithms based on DWT & images. When it is applied to images,
DCT. watermarking modifies or modulates the image
pixels, gray-level values in an imperceptible way,
Key words; chaotic mapping AES - DWT & DCT. in order to encode or insert a message (i.e., the
watermark). Thus, it allows us to intimately associate
I. INTRODUCTION protection data with the information to be protected.
Watermarking can be used for verifying the
The rapid evolution of multimedia and reliability of an image by asserting its integrity and
communication technologies offers new means of its authenticity. For instance, in a transaction, patient
sharing and remote access to transmitted images. In name and physician identity can be inserted in the
particular, aerospace imaging is already called to image. As defined, watermarking is an a
play important roles in applications like satellite posteriori control mechanism as the image content
transmissions and so on. But at the same time, this is still available for interpretation while remaining
ease of transmission and sharing of data increases protected.
security issues.
Different approaches have been proposed in order to
1. confidentiality, which means that only benefit from the complementarily of these two
authorized users can access transmitted data mechanisms in terms of a priori/a posteriori
2. Availability, which guarantees access to protection, essentially in the context of copyright
transmitted information in the normal scheduled protection. Technically, two categories of methods
conditions of access and exercise can be distinguished according to the way
3. Reliability, which is based on a) integritya watermarking and encryption are merged.
proof that the information has not been altered or
modified by non-authorized persons; and b) 1) Joint decryption and watermarking, where
authenticationa proof of the information origins watermark embedding is conducted during the
and of its attachment tone receiving station can be decryption process.
used confidently by the physician.
2) Joint encryption and watermarking, where
In any information systems, data confidentiality, watermarking and encryption step processes are
integrity, and non-repudiation services are usually merged. In this case, the watermark can be extracted.
achieved by cryptographic means. ASTM E2660, a. In the spatial domain, i.e., after the
Standard Digital reference images for investment decryption process, or
steel casting for Aerospace Images, allows data b. In the encrypted domain, or c. In both domains.
encryption through the triple DES , the AES, as well The system we propose in this paper
as digitally signing a A S T M E 2 6 6 9 by making belongs to the second category. It merges a
use of the DSA. However, once decrypted or its substitutive watermarking algorithm, Chaotic
digital signature deleted or lost, one piece of Mapping followed by SIFT [2], and an
323
encryption algorithm which can be a stream cipher respectively.

algorithm (e.g., RC4 ) or a block cipher
algorithm (e.g., AES). Our objective is to give
access to embedded security attributes in the
encrypted and spatial domains for the purpose of
verifying the reliability of an image. The rest of
this paper is organized as follows. In Section II, we
independently present the watermarking and the
encryption algorithms we used, before introducing
their combination in Section III. We then detail
our implementation in Section IV. Section V
presents some experimental results.
II. ENCRYPTION AND WATERMARKING

PRIMITIVES
A.ENCRYPTION Some of the main advantages of this type
of algorithms are that they are simple and operate
Basically there exists two methods for encrypting the at a higher speed than block cipher algorithms.
image, Cryptographic and Mapping. The specificity of such stream cipher algorithm
resides in how the bit/byte key stream is
A.1.Cryptographic Primitives : generated by the PRNG. The RC4 PRNG is based on
two steps.
Basically, there exist two types of encryption
algorithms: block cipher algorithms and stream 1) Initialization, where a table of
cipher 256 bytes is filled by repeating the
algorithms. Block cipher algorithms, like the AES encryption key as often as necessary until
and the DES, operate on large blocks of plaintext, to fill this table.
whereas stream cipher algorithms, like the RC4 or the
SEAL, manipulate stream of bits/bytes of plaintext. 2) Byte key stream generation,
where the elements of the table are
RC4 Stream Cipher Algorithm: As described combined by applying permutations and
in Fig. 1, stream cipher algorithms combine the additions to generate the key stream.
bits/bytes of plaintext T = [t1 , . . . ti , . . . , tn ]
with a secret key stream of bits/bytes AES in CBC Mode of Operation:
K = [k1 , . . . ki , . . . , kn ] issued from a
pseudorandom number generator (PRNG),
through a XOR operation typically. The key
stream generation depends on one secret key Ke
, making stream cipher algorithms as part of
symmetric encryption techniques. Thus,
bits/bytes of cipher text C = [c1 , . . . ci , . . . ,
cn ] are usually defined
as block, the encrypted block, and the encryption
key, respectively.
ci = t i ki .
(1) Fig: 2. AES encryption in CBC mode
in this AES in the cipher block chaining (CBC)

Fig: 1. Encryption/Decryption process of stream mode of operation in order to be compliant with the
Cipher algorithm DICOM standard. The concept of mode of
operation refers to the manner in which blocks of
Encryption/decryption processes of a stream plaintext (sequence of bytes) are treated at the
cipher algorithm which secret key is Ke, ti, ci encryption stage (respectively, decryption stage). As
correspond to the plain text bits/bytes, the cipher depicted in Fig. 2, when the CBC mode is applied, a
text bits/bytes, and the secret key stream bits/bytes, plaintext block is combined, with the previous
324
cipher text block through a XOR operation transformed image as has been shown in [3].
before Therefore the transformed chaotic image is richer in
being encrypted with the AES. If we denote the frequency content, a desirable property in
encrypted version of a block Bi and the previous watermarking, as there are more suitable candidate
encrypted block, s thus given b coefficients for manipulation and information
embedding. Voyatzis and Pitas in [3] presented a
watermarking scheme based on a two dimensional
Where K is the encryption key. The reader may refer chaotic function, called toral auto orphism. This
fora complete description of the AES. AES cyclic chaotic function, each time applied on a square
Encryption in CBCmode, Bi, denote the plaintext image rearranges its pixels. After applied T times,
block, theencrypted block, andthe encryption key, where T is the period of the function, the pixels are
respectively. Iv is a random vectorinitialization found in their initial location. If (x, y) are the initial
coordinates of a pixel, the outcome coordinates of the
A.2.Mapping Primitives: chaotic function (x, y) are given by
In this mapping method of encryption B. WATERMARKING PRIMITIVES:

we have several mappings in Chaotic maps like
Arnold, Bernoulli, Bifurcation, Chebyshev, Mach- The Mixed transform Technique for Digital Image
Zehnder interferometer etc., proposed by Chien-Pen Chuang , Cheng-Hung
Arnold or Toral automorphism c- map: Liu , Yi-Tsai Liao , Huan-Wei Chi in [5] used to
protect intellectual property right of digital image. A
Chaotic systems are deterministic systems scale-invariant feature transform technique was
(predictable if you have enough information) employed for resisting geometric attacks and two
that are governed by non-linear dynamics. These dimensional bar code was used for its high capacity
systems show deterministic behavior which is very and fault tolerance to enhance digital watermarking
sensitive to its initial conditions, in a way that the capacity. Besides, the discrete wavelet transform
forthcoming results are uncorrelated and seem (DWT) and discrete cosine transform (DCT) were
random. One category of chaotic systems is chaotic joined to cope with noise problem and enhance
maps. A chaotic map, which can be considered a two perceptual transparency of watermarking image. The
dimension chaotic function, is also a tool that could algorithm performance was presented with a plurality
relocate the pixels of an image and break spatial of experimental results, including several attack
continuity. models such as lossy compression, scaling, blurring
and sharpening etc. The experimental results proved
the robustness of this mixed transformation technique
on protecting intellectual property right of digital
image.
The embedded watermarking algorithm adopted

can be divided into five stages:
Stage 1: With scale-invariant feature transforms to
extract feature information, so as to solve
synchronization problem of watermarking.
Stage 2: In discrete wavelet transform domain, with
noise visibility function ,to estimate watermarking
embedded robustness.
Stage 3: In discrete cosine transform domain, in
accordance with JPEG quantization table, to select
embedded position for the watermark.
Stage 4: By Toral Auto orphism to disturb
watermarking sequence for embedding.
Stage 5: According to estimated robustness of
watermark, to embed the watermark onto frequency
domain coefficient of original image.
The watermark embedding can be explained with the
If we transform the resulting chaotic image in the following schematic
frequency domain, the significant coefficients are
highly increased in comparison with the respective
325
and thus the last stage if encryption and decryption is

cleared. Now the resulted image is dewatermarked
using the blind watermarking technique which results
the encrypted watermark. This watermark is
decrypted using the cmap technique Arnold thus the
original watermark is obtained.
Fig:4. System Architecture

III. PROPOSED JOINT ENCRYPTION AND
WATERMARKING TECHNIQUE
The purpose of our system is to verify the reliability IV. IMPLEMENTATION OF JOINT ENCRYPTION AND
of the image within the special domain as well as the WATERMARKING METHOD
encrypted domain. The watermark is passes through
the c-map for protection of the original watermark Earlier, our implementation works with AES and
through which the encryption process is done and chaotic-maps encryption methods and the blind and
then the watermark is placed in the original aerospace non-blind watermarking methods. The methods and
image to be transmitted. This encrypted watermark their implementation are stated as follows.
image is again subjected to the encryption through Choatic map Implementation:
the AES method for the extra protection. Now this
encrypted
watermarked image is allowed for the watermarking Of all the chaotic maps we prefer the Arnold chaotic
mapping or toral auto orphism for the present
followed by the decryption which result the image
encryption. In this toral auto orphism the actual
that can be transmitted. Thus the encryption and
watermark is applied a toral transform say Arnold
watermarking can be joined together which gives
transform by using the formula
justification for the title joint encryption and
watermarking.
The dual encrypted and watermarked image is

taken/received and encrypted first then the path for
the
detection of the watermark. Now the watermark
detection is applied to the encrypted image. This
detected image is obtained when the non-blind
watermarking is used so the watermark detected In this mixed transform watermarking
image is decrypted using the AES decryption method technique the process consists of four steps
326
basically given as in[4].

1) Estimating Watermarking Embedded
robustness
2) Selecting watermark embedding
positions
3) Disturbing watermark
4) Embedding watermark
In this estimating watermarking embedded robustness

the actual image will be tested for the amount of
watermark to be embedded, by dividing the actual
image into blocks of size 8X8 an then calculating the
NVF (Noise Visibility Function) which shows local
texture change at a particular image position. Which
can be obtained by knowing the local variance of
the particular sub block. Non Blind Watermark Embedding algorithm:
The watermark embedding position is obtained by In this non-blind watermarking algorithm the actual
using the quantization tables as shown below: image pixels are added by the watermark pixels by
multiplying them with a constant factor say a. The
implementation can be given as
V. PERFORMANCE EVALUATION AND

DISCUSSION
Experiments are conducted on the aerospace images

and thusthe reliability is verified
Image Distortion:
Now this blocks are applied through the toral auto
orphism for the security (developed by scholars of
We are using Peak Signal to Noise Ratio (PSNR) in
G.Yoatis and I.Pitas) before watermarking
order to measure the distortion between the image (I)
embedding this is done using the formula..
and itswatermarked and deciphered version (I)wd
The formulae to calculate the distortion is given as
follows
AES encryption implementation :
This encryption method is implemented

mainly using 4 steps which are shown as follows.
Thus applying these steps on an image the encryption

methods are carried out and thus the image will get
encrypted.These steps are followed based on the Where L corresponds to the length of the pixels of
round around key given by the AES method. Image (I) and d corresponds to its depth. Our choice
relies on the fact that the algorithm we proposed in
Section IV introduces on average the same image
327
distortion in each block, thus spreading it over the of

Applied Electronics 2008 Int. Conference, Pilsen, Czech
whole image. Furthermore, it does not take advantage
Republic, 10-11 September
of a psycho visual model which is helpful to adapt the [3] Applications of toral automorphisms in image
watermark amplitude locally into the image, making watermarking G.Voyatzis and I.Pitas.
at the same time the PSNR not appropriate. Even [4] A Robust Digital Watermarking with Mixed Transform
Technique for Digital Image Chien-Pen Chuang , Cheng-
though there exist some models for natural images,
Hung Liu , Yi-Tsai Liao , Huan-Wei Chi Proceedings of the
none of them have been proved adapted for aerospace International multiConference of Engineers and Computer
imaging yet. scientists 2012 vol I IMECS 2012 march 14-16 Hong Kong
[5] G. Coatrieux, H. Matre, B. Sankur, Y. Rolland, and R.
Collorec, Relevance of watermarking in medical imaging, in
The method seems to be more robust against
Proc. IEEE EMBS Int. Conf. Inf. Technol. Appl. Biomed.,
geometrical attacks (due to wavelets features) but no 2000,pp.25
explicit measurements for compression or filtering 0255.
attacks are mentioned. The method of this paper is [6] G. Coatrieux, C. Le Guillou, J.-M. Cauvin, and C. Roux,
Reversible watermarking for knowledge digest embedding and
outperformed, as it merely withstands JPEG reliability control in medical images, IEEE Trans. Inf. Technol.
compression down to quality 20 (with PSNR = 56.34 Biomed., vol. 13, no. 2, pp. 158165, Mar. 2009.
dB in [1]) opposed to JPEG compression with quality [7] U. Rajendra Acharya, D. Acharya, P. Subbanna Bhat, and U.
factor 5 and PSNR 64.34 dB presented in this work C. Niranjan, Compact storage of medical images with patient
information, IEEE Trans. Inf. Technol. Biomed., vol.5, no. 4,
Image Correlation : pp. 320323, Dec.2001.
[8] W. Pan, G. Coatrieux, N. Cuppens-Boulahia, F. Cuppens,
The robustness of the watermark depends on the and C. Roux, Medical image integrity control combining
digital signature and lossless watermarking, Data Privacy
strength parameter (strength) used during the Manage. Autonom. Spontaneous Sec., vol. 5939/2010, pp. 153
embedding procedure. As the strength increases 162, 2010.
the robustness of the scheme raises respectively.
Nevertheless, the quality of the watermarked image
is conversely proportional to strength parameter and
robustness.
The correlation method is based on the fact that the

sequence generated is normally distributed with
zero mean and unit variance. The correlation factor
varies depending on the starting coefficient and
the number of coefficients affected (N).
In our experiments the acceptable PSNR threshold

(PSNTthres) was set to 60, while the correlation
threshold (corthres) which determines whether an
image is watermarked or not was set to 4. Strength
parameter for the embedding procedure was set to
0,5 and a series of images were used for
experimentation. For most images the results
were similar, therefore due to limited space, only the
results regarding Lena are presented in this work,
for comparisons with other methods. Achieving
correlation factor of 15.60 dB with a PSNR value
of 60.00 dB.
REFERENCES
[1] A Joint Encryption/Watermarking System for Verifying
the
Reliability of Medical Images Dalel Bouslimi, Member, IEEE,
Gouenou Coatrieux, Member, IEEE, Michel Cozic, - IEEE
TRANSACTIONS ON INFORMATION TECHNOLOGY IN
BIOMEDICINE, VOL. 16, NO. 5, SEPTEMBER 2012
[2] Chaotic-Correlation Based Watermarking Scheme for
Still
Images E.Chrysochos V.Fotopoulus M.Xenos Pro ceedings
328
Development of Embedded Vision

Enhancement
System using Real Time Image Fusion
Sri Swathi Nimmagadda
Design Engineer, Design and Development Department, System Controls Technology Solutions Pvt Ltd, Bangalore,
India
nimmagaddasriswathi@gmail.com
Mr. Kanakaraju K
Technical Director,
System Controls Technology Solutions Pvt Ltd, Bangalore, India
design@system-controls.com
AbstractVision enhancement using multi-view image A. Contribution

fusion technique has diversified applications like
surveillance systems wherein fusing information from In this paper the design and development of a vision
camera captured at different viewpoints is used to enhancement system for surveillance applications
monitor activities in crowded areas. Thus there is using real time image fusion is proposed. The
need to develop robust real time system that developed algorithm captures video from moving
performs automated image registration and fusion. camera, where frames are aligned to same coordinate
This paper deals with the design and development of system using feature based image registration
a vision enhancement system for surveillance technique and fuses the registered frames using
applications using real time image fusion. The DWT based maximum selection image fusion. The
developed algorithm captures video from moving developed feature based image registration algorithm
camera, where frames are aligned to same coordinate obtains the corners
system using feature based image registration
technique. Then the registered and reference frames in the reference and unregistered frames, calculates the
are fused using DWT based maximum selection corner matching point pairs and estimates the
image fusion. After validating image registration and geometric transformation matrix with all distortion
fusion algorithm with MATLAB, the enhancement parameters such as rotation, translation and
system is developed by making a target processor scaling factor. Using obtained estimated
specific image registration and fusion Simulink transformation matrix, the geometric transformation is
model and porting on DM642EVM DSP processor with done to unregistered frame to obtain the registered
the help of Embedded Coder and CCStudio. The frame. The reference and registered frames are
developed vision enhancement system gives 30 fps and fused using DWT based maximum selection image
it is jitter free. The execution time of Un-Optimised fusion. Image registration and fusion algorithm is
implemented real time image fusion algorithm on developed and validated using MATLAB.
DM642 processor gives 770 ms. The algorithm is
Optimised further to give average execution time of B.Organ
740 ms in
This paper is organized as follows: Section II deals
. INTRODUCTION with related work. Section III provides the
development of proposed vision enhancement
Vision enhancement using Image Fusion plays system. Section IV proposes the results and
an important role in diversified domains of research. performance analysis. Section V contains conclusions
Those fields include mainly medical imaging, followed by future work
microscopic imaging, remote sensing, computer vision,
image processing, robotics, geosciences and robot . RELATED
vision. By obtaining information from one or more
WORK
cameras and fusing one can monitor activities in
crowded areas, buildings etc where surveillance is
required as a preventive measure. The development The already existing image registration techniques
of Vision enhancement system using Real time broadly classified in to area based and feature based
image Fusion requires robust image registration and methods as in [1]. A computational vision
image fusion algorithms to be implemented approach is presented for the estimation of 2-D
successfully.
329
translation, rotation, and scale from two partially A. Problem Statement

overlapping images as in [2]. The approach results in Vision Enhancement using Image Fusion has gained
a fast method that produces good results even when its importance in various domains and in
large rotation and translation have occurred between surveillance applications particularly. The goal of
the two frames and the images are devoid of image fusion is to integrate multi-view information
significant features. An illuminant direction into one new image containing information, the
estimation method is first used to obtain an initial quality of which cannot be achieved otherwise and
estimation of camera rotation. A small number of to provide better visual understanding of certain
feature points are then located, using a Gabor phenomenon. But most of the existing systems are not
wavelet model for detecting local curvature capable of providing this type of information
discontinuities. An initial estimate of scale and successfully. Also frequent problem arises when
translation is obtained by pairwise matching of the images taken, by sensor from different viewpoints or
feature points detected from both frames. Finally, at different times, need to be compared and fused.
hierarchical feature matching is performed to obtain The images need to be aligned with one another so that
an accurate estimate of translation, rotation and scale differences can be detected. All of these problems, and
[2]. A method for automatic image registration many related variations, are solved by developing
which is characterized by its insensitivity to scaling, robust real time system that performs automated image
rotation, and intensity changes is described [3]. This registration and fusion.
method is based on similarity assessment of the
structures in the Landsat and Spot images, and on a
B. System Top Level Design Overview
check of their spatial arrangement. Pairs of structures
that correspond to each other provide sets of control The top level design diagram shown in Figure 1 shows
points to geometric mapping functions. This sub blocks such as image acquisition that grabs video
scheme [3] is feasible for remote sensing application. frames from CCD camera, image pre-processing that
would perform color space conversions and image
Image Fusion is process of combining two or more registration that would align all the captured frames
images/frames in to a single image/frame for as reference frame and finally image fusion. All
better information and for situational awareness in real these sub blocks are implemented at algorithm level
on the DSP video processor. Then the final
time surveillance applications. In real time systems for
enhanced fused video is displayed on LCD monitor.
performing image fusion initially frames are to be
registered to same coordinate system accurately. A
critical review of some of the image fusion techniques
for image fusion like, primitive fusion (Averaging
Method, Select Maximum, and Select Minimum),
Discrete Wavelet transform based fusion, Principal
component analysis (PCA) based fusion etc is
provided in [4].Spatial domain provide high spatial
resolution. But spatial domain have image blurring
problem. The Wavelet transforms is the very good
technique for the image fusion provide a high quality
spectral content [4]. Wavelet transforms provide a
frame work in which an image is decomposed, with C. Real Time Image Fusion Algorithm Implementation
each level corresponding to a coarser resolution band
[5]. The wavelet sharpened images have a very good
spectral quality. Fourier transform provides
information regarding frequency. Short time Fourier
transforms gives only constant resolution. So wavelet
transform is preferred over Fourier transform and short
time Fourier transforms since it provides multi
resolution as in [5],[6].
III.DEVELOPMENT OF VISION ENHANCEMENT

SYSTEM
This section provides all the details of development

of vision enhancement system using real time image
fusion. Initially it gives the problem statement, top level
design, algorithm implementation, DSP hardware
implementation and porting details.
330
reference and unregistered frames with

As per the top level design the algorithms are
parameters like rotation, translation and scale and a
developed for image acquisition, image pre-
transform matrix is generated with those estimated
processing, image registration and fusion, validated
parameters. Then finally basing on that transform
using MATLAB. Figure 2 gives the algorithm
estimation matrix geometric transformation is done to
implementation flow for image acquisition and image
align both frames to single coordinate system. Initially
pre-processing. Initially a video object is created, then
the registered and reference frames from image
camera source is selected, then properties like
registration subsystem are taken as input to this
adapter name, device ID and video format are set. A
fusion subsystem. Then one level two dimension
stream of frames is previewed, finally the required
discrete wavelet transform is applied to both reference
number of video frames is selected and video object
and registered frames obtained from image registration
after acquiring frames is removed from memory.
subsystem. The wavelet transform is obtained by
Later these acquired frames are fed to image pre-
decomposition using row wise and column wise
processing subsystem.
operation. That is independent wavelet decomposition
of the two frames is performed until level one to
get approximation (LL) and detail (LH, HL, HH)
coefficients. Pixel based algorithm for
approximations which involves fusion based on taking
the maximum valued pixels from approximations of
source frames is applied. Then fusion decision is
made to select with maximum pixel values thus
gets a new concatenated fusion matrix. The final
fused frames can be reconstructed using inverse
wavelet transform.
D. DSP Hardware Implementation
1) Selection of suitable DSP Video processor:

2) This section provides the justification for
selecting suitable digital signal processor
required for implementing the validated
real time image fusion algorithm. The
processor requirements for this application
as follows:
3) Video Encoder for capturing video
source as
4) Vid
io
IN
5) Video Decoder for displaying video output
6) Thus the Digital Media Processor

TMS320DM642 (Version 3), which
belongs to the DaVinci family of Texas
Instruments C6000 series is selected.
2) Hardware Implementation Approach
Figure 3 gives the complete flow of the 2) Hardware Implementation Approach

implemented image registration and fusion
algorithms. Initially two frames are read and one The approach followed for implementing Real time
first frame is taken as reference frame whereas the image fusion is with Simulink, but however it is not
next frame is considered as unregistered frame. possible to load Simulink model directly to
Then corners detection of both frames is performed. If DM642EVM video processor.
required number of corners are detected their corner
feature descriptors are extracted. Next corner points A Simulink model which is set with target
from both the frames are matched and locations of (DM642EVM V3) is developed. Figure 4 represents
corresponding points are retrieved. Transform the interfacing between MATLAB, CCS and
estimation is done to find the distortion between the
331
DM642EVM DSP. Thus the designed Simulink checked accordingly. Later the incremental building of
model facilitates to generate C code automatically the Simulink model is done. Finally the code is
corresponding to the desired problem with the generated and downloaded as a project including all
help of embedded target and Embedded Coder board specific header files and source files. This
facility provided in MATLAB/Simulink. However project in CCStudio is build and a.out file generated
the DSP development software, Code Composer is loaded in to DM642 DSP chip and the project is
Studio can accept either C or assembly code to run for results. The complete porting procedure of
generate output (.out) file, which can be load on DSP developed Simulink model on to the DM642 EVM DSP
chip. So Code composer studio version 3.3 used to processor to make is given in Figure 6.
generate .out file. This file now can be load into our
DM642EVM processor. Thus the Simulink model
developed for standalone application can be run on the
processor continuously without fail making it a real
time product. In addition this approach has an
advantage of verification of the developed model once
if the system Fails.
3) Simulink Model for Real Time Image Fusion
The validated real time image fusion algorithm with

image registration and fusion is ported on to the DSP
video processor by developing functional equivalent
Simulink model. Also the selected target preference
block (DM642 EVM) is added to the Simulink model
for dumping procedure to start. Figure 5 gives the
developed Simulink model.
Fig. 6. Overall porting procedure
5) Hardware Setup
Figure 7 gives hardware setup snapshot of

development process
Fig. 5. Real time image fusion Simulink Model
4) Hardware Porting Procedure

It initially starts with the development of DM642
EVM board specific Simulink model along with its
target preference block from Simulink with embedded
coder (Simulink coder).Then it is run on the simulator.
If any errors found the configuration parameters are
332
RESULTS C. Performance Analysis

Table 2 provides the performance analysis by
comparing the average execution time of optimized
The developed real time image fusion algorithm and unoptimized video registration and fusion code on
shown in design and development section is tested with the DM642 Platform.
designed test cases that are shown in testing chapter
D. Critical Analysis
for various sets of inputs as static input images,
recorded video file frames and live video frames. The developed system is jitter free as the
priority of video processing task spawned is set
as 15 to give the highest priority and no other
A. MATLAB Results of Static
high priority tasks are present for preemption
Images
The average execution time of un-optimized
The developed MATLAB code is tested with various Real time image fusion algorithm
designed test cases on static images and the various implemented on the DSP processor is 770
stages of implementation gives step by step result
milli seconds
which are shown as in Figure 8.
The average execution time of optimized
real time image registration and fusion
algorithm implemented on DM642 Video
Processor is 740 milli seconds
CONCLUSION
(a) (b) (c) The feature based image registration and DWT
(d) based image fusion techniques can be used for
developing automated and robust real time vision
Fig. 8. (a) Reference image, (b) Unregistered image, (c) Registered
Image, (d) Fused image
enhancement systems for moderate lighting conditions.
B. MATLAB Results of Recorded video The developed system is implemented using single
frames camera thus this system can be extended to multiple
sensors and dissimilar fixed or moving at different
The developed MATLAB code is tested with various locations.
designed test cases on frames from recorded video file ACKNOWLEDGMENT
and the various stages of implementation gives step
by step result which are shown as in Figure 9 that This work was supported by System Controls Technology
gives the results of reference frame, unregistered frame, Solutions Pvt Limited, Bangalore. I extend my
color composite of reference and unregistered frames, thanks to Mr. K. Kanakaraju, Technical Director and Mr.
registered frame and finally fused frame. Naveen K. S, Senior Design Engineer at System
Controls Technology Solutions Pvt. Ltd. for their support
during the Work.
REFERENCES
[1] W. Pratt , Digital Image Processing. New York: Wiley, 1993.
(a) (b) (c) (d) [2] Q. Zheng and R. Chellappa, A computational vision
(e) approaches to image registration, IEEE Trans on Image
Processing, Vol.2, No.3, pp.
311326, July 1993.
Fig. 9. (a) Reference frame, (b) Unregistered frame, (c) color
composite of [3] A. Ventura, A. Rampini and R. Schettini, Image
Reference and Unregistered frames, (d) Registered frame, (e) Registration by Recognition of Corresponding Structures, IEEE
Fused frame Trans on Geosciences and Remote sensing, Vol.28, No.3, May
Table 1 gives the calculated Image Quality metric 1990.
values of the obtained final fusedRecorded
output. video [4] S. Deepak Kumar, M.P. Parsai, Different Image Fusion
Image Static Images Techniques A Critical Review, International Journal of
Quality frames(dwt based max Modern Engineering Research, Vol. 2, No.5, pp. 4298-4301,
Measures sel fusion) Sep-Oct-2012.
PSNR(dB) +24.56534 +24.24678 [5] Deepali A.Godse, Dattatraya S. Bormane Wavelet based
MSE 229.05804 228.91583 image fusion using pixel based maximum selection rule,
RMSE 15.13466 15.11166 International Journal of Engineering Science and Technology,
Entropy 6.89873 6.81177 Vol. 3 No. 7 July 2011, ISSN :
of Fused 0975-5462
output [6] Susmitha Vekkot, Pancham Shukla, A Novel Architecture for
0.80632 0.88562 Wavelet based Image Fusion, World Academy of Science,
Universal
Engineering and Technology 57, 2009.
IQI
[7] DM642EVM DSP Video Processor Technical Reference
EME 11.18711 11.10636
Manual, Rev.B, Spectrumdigital, INC. 2004.
SNR(dB) 0.46595 0.64738
333
Pseudo Random Noise Code Selection Criteria

for GNSS
G. Sai Anirudh1, G.Shanmukh2
Sir M Visvesvaraya Institute of Technology, Bangalore1, ANU College of Engineering & Technology, Guntur2
E-mail ID: anil_2118@yahoo.com, gollapudi.shanmukh@gmail.com
Abstract -Global Navigation Satellite System (GNSS) coded modulation characteristics uniquely qualify SS
is a specific term for any satellite navigation system that for navigation from the standpoints of both range
provides autonomous geo-spatial positioning with global measurements and direct fading. DS-SS is the most
coverage.Some of the GNSS are already in service, but preferred for ranging.
some of them still under development. These systems
rely on the code division multiple access (CDMA) in A. Properties of Spreading Codes: The code used for
which Pseudo noise (PN) sequences are generated and Spreading should possess some properties such as
widely used in GNSS. This paper analyses the spread 1. It must appear deterministic.
spectrum codes and verifying some interesting 2. It must appear random to a listener.
properties of PN sequence: Balance Property, Auto 3. The correlation of two different codes (Cross -
Correlation, and Cross Correlation through which the Correlation) should be small.
criteria of selecting PN Sequence out of various 4. The correlation of a code with a time-delayed
sequences is done.
version of itself (auto-correlation) should be equal
to 0 for any time delay other than zero in order to
I. INTRODUCTION
reject multi-path interference.
GNSS is used to determine the position of a 5. The code must have a long period.
receiver on land, at sea, or in space by means of
constellation of multiple artificial satellites. The past B. Choice of Code:
decade has seen a rapid development of
severalGNSSs. Some of them are already in service In BPSK transmission, bandwidth is a direct
(e.g. GPS by USA). However, some GNSSs still function of the code chip rate. Code repetition rate is
under planning or partial operational phase (e.g., simply given as
Compass by China and Galileo by European Union).
There are two types of GNSS, one is Global RRep=Clock rate(chips/sec)/Code length
Navigation and second is Regional Navigation.In this (chips)(1)
paper we overview the GNSS systems like GPS, This repetition rate determines the line spacing in the
Beidou-I (open service). The Section II talks about RF output spectrum. Hence, it is an important
the various spreading codes used for the Spread consideration in a system design. Another important
spectrum communication. Section III gives the criterion is that the period of the code must be very
knowledge of the PN sequence properties. Section IV large. Table I lists various code lengths for a 1-Mcps-
makes you understand about the code selection chip rate. Other consideration should be that the
criteria and the results we obtained. This paper ends choices of code rate and code length determine the
with Section VI which presents you our conclusion relationship of the repetition rate to the information
on this paper baseband and use of the system for ranging. The code
II. SPREADING CODES used in a direct sequence system should be adjusted
The method of Spread Spectrum communication for repetition rate and length such that noise does not
involves the spreading of the signal spectrum to make pass into the demodulator especially under jammed
its appearance noise-like. The signal will thus be conditions. A properly chosen code can not only
hidden and any intentional jammer would not be able improve resolution but also can ease range
to detect it. The method usually employed for measurement problem. if the chip rate is chosen in
spreading the signal spectrum is to multiply it with such a way that an integral number of code offset can
spreading sequences.Some ways to spread the be used to measure range. Clock rates that are
bandwidth of the signal: Frequency Hopping,Time integral multiple of 161,875Hz-produce code at rates
Hopping, Direct sequence (DS),Hybrid system.Due that are integrally related to the speed of propagation
to the LPI, AJ characteristics, the most popular SS are of RF signal.
in fields where security is an important consideration.
Also, multiple access enables many users to operate The relationship may be derived as follows:Speed of
at the same time without much interference.The light in vacuum is c = 2.997926 x 108 m/s, 1 nautical
334
mile = 1.852 x 103 meter, The clock rate

(wavelength = 1nmi) = c/R=2.997926 x 108/1.852 x 2) Gold Sequence: Gold sequences are
103 = 161,875 Hz constructed by the XOR of a preferred pair of m-
The clock rate for 1 statute mile resolution would be sequence with the same clocking. They have well
186,333 Hz. Table 1, list the rates and sequence defined cross correlation properties and only simple
length that should be used. circuitry is needed to generate large number of
unique codes show below in figure 2
Among the biggest problems in code generation are
(1) finding feedback logic that gives desired code
length and (2) checking the code one a sequence
generator has been constructed to ensure that it is
operating properly. Tables of feedback connections
and irreducible polynomials have been generated.
TABLE I Fig 2.Double Shift Relaization

CODE SEQUENCE PERIODSFORVARIOUS M-SEQUENCES
3) Kasami Codes: Kasami codes have popular
LENGTHS,1-MCPS RATE
use in 3G wireless scheme. They can be classified as
(1) large Kasami set or (2) small Kasami set. Small
kasami set has family size (M) = 2n/2 and period = N
= 2n-1. Their maxium cross-correlation is 2n/2-1.Large
kasami set contains both gold sequence and small set
of kasami sequences as subset.Its period is N=2n-
1.Maximum cross-correlation is 2(n+2)/2.
4) JPL Ranging Codes:The JPL refers to Jet

Propulsion Laboratories. The JPL Ranging codes are
constructed by modulo-2 addition of two as more
maximal linear sequences whose lengths are
relatively prime to one another. Often a clock term is
also modulo-2 added to the other sequence (the clock
is short maximal sequence in itself.
There are several advantages to such technique.
C. Popular Codes: Very long codes useful for unambiguous
1) Maximal Length Sequence or PN ranging over long ranges are available
Sequence:The pseudorandom noise (PN) sequence These long codes are generated by relatively
are a series of 1s and 0s which lack of any definite small number of shift register stages
pattern and consists of a deterministic sequence of Synchronization of a receiver can be
pulse that will repeat after its period. In a truly accomplished by separate operations on the
random sequence the bit pattern never repeats. A component codes
pseudo random binary sequence is a semi-random
sequence in the sense that it appears random within 5) Hadamard-Walsh Code:WalshHadamard
the sequence length, fulfilling the needs of code is an error-correcting code that is used for error
randomness, but the entire sequence repeats detection and correction when transmitting messages
indefinitely. The heart of PN sequence generator is over very noisy or unreliable channels. Some of the
usually made up of linear feedback shift registers important points are:
(LFSR). It generates a maximal-length sequence of Walsh codes do not have the properties of m-
length N = 2n 1 elements.Because of their good sequences regarding cross correlation.
autocorrelation two similar PN sequences can easily The codes do not have a single, narrow
be phase synchronized, even when one of them is autocorrelation peak. In fact the non-peak auto-
corrupted by noise. The below figure 1 represents the correlation is dependent on the actual code word,
LFSR to generate PN Sequence but it's never zero.
Orthogonality is the most important property of
Hadamard-Walsh codes. Because of this, the
cross-correlation between any two Hadamard-
Fig 1. Single Shift Realization
335
Walsh codes of the same set (matrix) is zero, Mathematically, the correlation between two
when system is perfectly synchronized. sequences x (k) and y (k) as a function of the time
Walsh codes are not maximal length or PN type delay m is expressed as
codes for spread spectrum. Although the members
R(m)xy= ( ) ( + ) (4)
of the set are orthogonal, they do not give any
spreading. The correlation equation for the digital bit sequence
Walsh code spreading can be used if all users of can thus be written as
the same channel are synchronized in time,
because the cross-correlation between different R (m) = total number of 1s / total number of bits
shifts of Walsh codes is not zero. (5)
6) Barker and Williams Sequence:An important These three properties make PN sequences efficient
sequence designed to achieve the minimum aperiodic for speechencryption. However, particularly due to
correlation values when l0 is known as barker the third property, adjacent bits correlation becomes
sequence. These sequences are used primarily for considerably less, thereby making the PN sequences
synchronization of data streams, such as frame more effective to be used in systems like CDMA.
synchronization. A barker sequence is characterised Therefore, useful PN sequences must have very good
by the fact that its aperiodic auto correlation function auto-correlation and cross-correlation properties as
satisfies wellas maintaining some randomness properties.
|Ca(l) | 1 for l 0..(2)
Welch bound: Designing codes optimized for any of
7)Bent Sequence:Bent sequences are nonlinear the potential application is practically impossible,
binary sequences that achieve Welchs bound on using code centric metric is more appropriate. This is
cross and auto correlation. The period of the sequence the reason why the Welch bound has gained
is 2m-1 with m the number of stage in the shift importance in recent years as suitable metric for
register. This value m must be multiple of 4 (n= 0 evaluating PRN codes.The Welch bound is the
mod 4).The number of sequences in the set is 2m/2 theoretical minimum of the maximum value of cross-
and the auto correlation function has 3 values and correlation that can be obtained for a given code
max=2m/2+1. length L within a set of M codes [3].
The Welch bound for a set of K sequences with
III. PROPERTIESOF PN-SEQUENCE eachsequence of length N (N K) is defined as
A PN sequence is a bit stream of 1s and 0s (6)

occurring randomly, with some unique properties. such a bound is no longer achievable when N >
The sequence serves as a reference pattern with K(K+1)=2 for real cases. Note that, in the sequel,
known random characteristics for the analysis, sometimes we shall represent binary sequences using
optimization and performance measurement of zeros and ones and in other cases +1s and -1s. The
communication channels and systems. appropriate mapping is thatthe zeros are mapped to
A. Balance Property +1s and ones are mapped to -1s.
In each period of a maximum length sequence, the
number of 1s is always one more than the number of IV. PN SEQUENCECRITERIAFOR GNSS
0s. A) Global Positioning System (GPS):
B. AutocorrelationProperty: The autocorrelation The GPS signal for Standard Positioning Service
function r(i) of any PN sequence of length N is (SPS) consists of the Navigational Data stream at
given by 50bpsmixed with the C/A code, modulated using a
carrierof 1575.42MHz (L-band). The C/A has a
1 =0 chipping rateof 1.023MHz with a code period of
r(i)= (3)
1|| 1. 1.023ms. The bitsobtained after spreading the data
with the PN sequence isrepresented in NRZ format,
C. Correlation Property:Correlation is a measure of which is then used for BPSKmodulation of a
similarity between two sequences. When the two sinusoidal carrier of 1575.42MHz. So,the GPS-SPS
sequences compared are different it is the cross signal for the ith satellite can be representedas
correlation and when they are the same it is the s ( , t) = A(C (t)D (t)) cos( ( , t) (7)
,
autocorrelation.
Where,
336
A: The carrier amplitude They are verified with the welch bound condition
Ci(t) : C/A code for the ith satellite given by
Di(t) : Navigational data for the ith satellite
: L1 carrier with frequency of 1575.42MHz {Cross Correlation maximum peak value} x {Code
Length} Code Length
Gold codes are constructed by taking the modulo-2
sum of two maximal length sequences of length 2n-1,
although the two sequences are required to be related
to one another in a particular way. An explanation for
the theory of Gold codes is beyond thescope of this
thesis, although an excellent summary may be found
in the Appendix of(Spilker, 1996).
In the case of the C/A codes, the pair of maximal-
length sequences are constructed using generating
polynomials G1 and G2, where G1 and G2 are given
by
G1= 1+X3+X10
G2= 1+X2+X3+X6+X8+X9+X10
With the G1 and G2 generators reset to all 1s after
generation of 1023 pseudo-randombinary chips. The
Gold code is constructed as the modulo-2 sum of the
G1 sequenceand a delayed version of the G2 Fig 3. C/A code Generator of GPS
sequence, with each satellite number allocated TABLE II
adifferent delay quantity. As there are 1025 different OBTAINED AUTO CORRELATION VALUES
sequences in the family of codes,but only 513 codes Auto Correlation Off peak value in
are balanced codes where the number of 1s and 0s Sl Satellite
Off-peak dB
differ by 1, theassignment of codes to satellites is no Number
maximum value (10*log10(value))
from the set of balanced codes.In the case of GPS,
two different taps of the 10-bit G2 register are added
1 1 78 18.920
thereby allowing a maximum of 45 different codes to
be generated.Alternative approaches that are more
widely used are to reset the G2 register to aspecified 2 2 85 19.294
initial value rather than all 1s, or to insert a
programmable delay that delaysthe reset of G2 until a 3 3 81 19.084
specified number of chips in the sequence have
elapsed. Theadvantage of these methods is that not 4 4 80 19.030
only do they permit all of the 1023 codes in thefamily
to be generated, but they also offer a simpler 5 5 72 18.573
implementation3. In particular,they also permit
generation of the codes used by the WAAS and 6 6 92 19.637
EGNOS satellite basedaugmentation systems
(SBAS),Figure 3 illustrates the implementation based 7 7 68 18.325
on LFSR
8 8 83 19.190
All the 37 Satellite vehicles have different code phase
assignments so as to generate the Pseudo random 9 9 71 18.512
noise codes. From the generated PN sequence the
various properties mentioned in Section III are
10 10 72 18.573
performed and they are verified. The balanced
property is verified for all the satellites.
11 11 83 19.190
The Auto correlation values obtained for the
generated PRN code is tabulated along with the
waveforms generated. The Cross Correlation is 12 12 83 19.190
generated and tabulated along with the waveforms.
337
13 13 70 18.450 Cross Off peak Non

Correlatio value in normalise
Sl Satellite
14 14 79 18.976 n dB d cross
no number
maximum (10*log10( correlatio
15 15 66 18.195 peak value value)) n values
16 16 71 18.512 1 1&2(Max) 0.2659 -5.75 272.01
17 17 82 19.138 2 1&16(Min) 0.2649 -5.76 270.99
18 18 91 19.590 3 2&1(Max) 0.2659 -5.75 272.01
19 19 78 18.920 4 2&22(Min) 0.2620 -5.81 268.02
5 3&4(Max) 0.2659 -5.75 272.01

20 20 71 18.512
6 3&16(Min) 0.2620 -5.81 268.02
21 21 77 18.864
7 4&6(Max) 0.2659 -5.75 272.01
22 22 75 18.750
8 4&19(Min) 0.2620 -5.81 268.02
23 23 87 19.395
9 5&8(Max) 0.2659 -5.75 272.01
24 24 75 18.750
10 5&18(Min) 0.2620 -5.81 268.02
25 25 83 19.190
11 6&10(Max) 0.2659 -5.75 272.01
26 26 84 19.242
12 6&16(Min) 0.2610 -5.83 267.00
27 27 70 18.450
13 7&10(Max) 0.2659 -5.75 272.01
28 28 74 18.692
14 7&26(Min) 0.2610 -5.83 267.00
29 29 74 18.692
15 8&10(Max) 0.2659 -5.75 272.01
30 30 76 18.808
16 8&21(Min) 0.2630 -5.80 269.04
31 31 73 18.633
17 9&10(Max) 0.2659 -5.75 272.01
32 32 78 18.920
18 9&17(Min) 0.2620 -5.81 268.02
33 33 78 18.920
19 10&11(Max) 0.2659 -5.75 272.01
34 34 79 18.976
20 10&15(Min) 0.2639 -5.78 269.96
35 35 75 18.750
21 11&22(Max) 0.2659 -5.75 272.01
36 36 85 19.294
22 11&28(Min) 0.2620 -5.81 268.02
37 37 79 18.976
23 12&13(Max) 0.2659 -5.75 272.01
TABLE III 24 12&17(Min) 0.2630 -5.81 269.04

OBTAINED CROSS CORRELATION VALUES
338
25 13&18(Max) 0.2659 -5.75 272.01 52 26&33(Min) 0.2620 -5.81 268.02
26 13&28(Min) 0.2610 -5.83 267.00 53 27&30(Max) 0.2659 -5.75 272.01
27 14&15(Max) 0.2659 -5.75 272.01 54 27&28(Min) 0.2630 -5.80 269.04
28 14&16(Min) 0.2630 -5.80 269.04 55 28&29(Max) 0.2659 -5.75 272.01
29 15&19(Max) 0.2659 -5.75 272.01 56 28&31(Min) 0.2620 -5.81 268.02
30 15&18(Min) 0.2600 -5.85 265.98 57 29&30(Max) 0.2659 -5.75 272.01
31 16&23(Max) 0.2659 -5.75 272.01 58 29&32(Min) 0.2630 -5.80 269.04
32 16&17(Min) 0.2620 -5.81 268.02 59 30&31(Max) 0.2659 -5.75 272.01
33 17&18(Max) 0.2659 -5.75 272.01 60 30&32(Min) 0.2649 -5.76 270.99
34 17&22(Min) 0.2630 -5.80 269.04 61 31&34(Max) 0.2659 -5.75 272.01
35 18&21(Max) 0.2659 -5.75 272.01 62 31&33(Min) 0.2620 -5.81 268.02
36 18&26(Min) 0.2630 -5.80 269.04 63 32&34(Max) 0.2649 -5.76 270.99
37 19&20(Max) 0.2659 -5.75 272.01 64 32&33(Min) 0.2610 -5.83 267.00
38 19&27(Min) 0.2639 -5.78 269.96 65 33&35(Max) 0.2659 -5.75 272.01
39 20&22(Max) 0.2659 -5.75 272.01 66 33&34(Min) 0.2649 -5.76 270.99
40 20&27(Min) 0.2639 -5.78 269.96 67 34&37(Max) 0.5005 3.00 512.01
41 21&29(Max) 0.2659 -5.75 272.01 68 34&36(Min) 0.2659 -5.75 272.01
42 21&27(Min) 0.2620 -5.81 268.02 69 35&37(Max) 0.2659 -5.75 272.01
43 22&23(Max) 0.2659 -5.75 272.01 70 35&36(Min) 0.2649 -5.76 270.99
44 22&25(Min) 0.2620 -5.81 268.02 71 36&37(Max) 0.2639 -5.78 269.96
45 23&24(Max) 0.2659 -5.75 272.01 As mentioned above from the condition {Cross
Correlation maximum peak value} x {Code Length}
46 23&27(Min) 0.2630 -5.80 269.04 Code Length (which is 34.684) is must satisfy for
all satellites Therefore Code Length =1023 chips;
47 24&26(Max) 0.2650 -5.75 272.09 Correlation Max Peak Value * Code length=
272.0157
48 24&31(Min) 0.2639 -5.78 269.96 (272.0157 45.23237) Satisfies Welch bound
49 25&26(Max) 0.2659 -5.75 272.01
50 25&27(Min) 0.2630 -5.80 269.04
51 26&29(Max) 0.2659 -5.75 272.01

SV-2 SV-12
339
five Geostationary Earth Orbit (GEO) satellites,

twenty-seven Medium Earth Orbit (MEO) satellites
and three Inclined Geosynchronous Satellite Orbit
(IGSO) satellites. The GEO satellites are operating in
SV-17 SV-25 orbit at an altitude of 35,786 kilometres and
positioned at 58.75E, 80E, 110.5E, 140E and
160E respectively. The MEO satellites are operating
in orbit at an altitude of 21,528 kilometres and an
inclination of 55 to the equatorial plane. The IGSO
satellites are operating in orbit at an altitude of
35,786 kilometres and an inclination of 55 to the
equatorial plane. By the end of 2012, there are five
GEO, four MEO and five IGSO BeiDounavigation
SV-32 satellites in orbit. The nominal frequency of B1I
signal is 1561.098 MHz.The transmitted signal is
Fig 5.Autocorrelation Waveforms for various SVs modulated by Quadrature Phase Shift Keying
(QPSK). The signal multiplexing mode is Code
Division Multiple Access (CDMA). The chip rate of
SV-1 & SV-7 the B1I ranging code is 2.046 Mcps, and the length is
2046 chips. The B1I ranging code (hereinafter
referred to as CB1I) is a balanced Gold code
truncated with the last one chip. The Gold code is
generated by means of Modulo-2 addition of G1 and
SV-10 & SV-19 G2 sequences which are respectively derived from
two 11-bit linear shift registers [14].
The generator polynomials for G1 and G2 are as
follows:
G1(X)=1+X+X7+X8+X9+X10+X11
SV-15 & SV-23
G2(X)=1+X+X2+X3+X4+X5+X8+X9+X11
The initial phase of generator polynomials are G1:

01010101010; G2: 01010101010
SV-32 & SV-36
Fig.6:Cross-correlation Waveforms for various SVs Fig 7. C/A Code Generator for Beidou-1
B) Beidou-1(Regional Navigation System):BeiDou The different phase shift of G2 sequence is

Navigation Satellite System is called BeiDou System accomplished by respective tapping in the shift
for short, with the abbreviation as BDS. When fully register generating G2 sequence. By means of
deployed, the space constellation of BDS consists of Modulo-2 addition of G2 with different phase shift
340
and G1, a ranging code is generated for each Satellit No. Cross Correlation
satellite. e of maximum peak
(Max) x (Min) x
Combi Co value
(Code (Code
TABLE IV nations mbi
Length) Length)
nati
AUTO CORRELATION RESULTS THROUGHMAT LAB ons
Max Min
Satellite Auto 1 37 36 0.2581 0.2571 528.0726 526.0266
Off peak
number Correlation
value in dB 2 37 35 0.2581 0.2571 528.0726 526.0266
Sl No and Off-peak
(10*log10(val
Signal maximum 3 37 34 0.2581 0.2571 528.0726 526.0266
ue`))
Type value 4 37 33 0.2581 0.2576 528.0726 527.0496
1 1 114 20.56904851
5 37 32 0.2581 0.2576 528.0726 527.0496
2 2 102 20.08600172
3 3 100 20 6 37 31 0.2581 0.2571 528.0726 526.0266
4 4 99 19.95635195 7 37 30 0.2581 0.2571 528.0726 526.0266
5 5 89 19.49390007 8 37 29 0.2581 0.2571 528.0726 526.0266
6 6 104 20.17033339
9 37 28 0.2581 0.2576 528.0726 527.0496
7 7 112 20.49218023
8 8 105 20.21189299 10 37 27 0.2581 0.2576 528.0726 527.0496
9 9 95 19.77723605 11 37 26 0.2581 0.2571 528.0726 526.0266
10 10 95 19.77723605 12 37 25 0.2581 0.2576 528.0726 527.0496
11 11 96 19.82271233
13 37 24 0.2581 0.2571 528.0726 526.0266
12 12 112 20.49218023
13 13 89 19.49390007 14 37 23 0.2581 0.2566 528.0726 525.0036
14 14 93 19.68482949 15 37 22 0.2581 0.2576 528.0726 527.0496
15 15 2047 33.11117843 16 37 21 0.2581 0.2576 528.0726 527.0496
16 16 95 19.77723605
17 37 20 0.2581 0.2571 528.0726 526.0266
17 17 96 19.82271233
18 18 100 20 18 37 19 0.2581 0.2576 528.0726 527.0496
19 19 97 19.86771734 19 37 18 0.2581 0.2576 528.0726 527.0496
20 20 96 19.82271233 20 37 17 0.2581 0.2571 528.0726 526.0266
21 21 116 20.64457989 21 37 16 0.2581 0.2576 528.0726 527.0496
22 22 104 20.17033339
23 23 96 19.82271233 22 37 15 0.2581 0.2566 528.0726 525.0036
24 24 94 19.73127854 23 37 14 0.2581 0.2571 528.0726 526.0266
25 25 99 19.95635195 24 37 13 0.2581 0.2571 528.0726 526.0266
26 26 102 20.08600172 25 37 12 0.2581 0.2571 528.0726 526.0266
27 27 93 19.68482949
28 28 93 19.68482949 26 37 11 0.2581 0.2576 528.0726 527.0496
29 29 96 19.82271233 27 37 10 0.2581 0.2576 528.0726 527.0496
30 30 98 19.91226076 28 37 9 0.2581 0.2576 528.0726 527.0496
31 31 94 19.73127854 29 37 8 0.2581 0.2576 528.0726 527.0496
32 32 108 20.33423755
30 37 7 0.2581 0.2576 528.0726 527.0496
33 33 95 19.77723605
34 34 115 20.6069784 31 37 6 0.2581 0.2571 528.0726 526.0266
35 35 99 19.95635195 32 37 5 0.2581 0.2576 528.0726 527.0496
36 36 106 20.25305865 33 37 4 0.2581 0.2576 528.0726 527.0496
37 37 102 20.08600172
34 37 3 0.2581 0.2581 528.0726 528.0726
TABLE V 35 37 2 0.2581 0.2576 528.0726 527.0496
CROSS CORRELATION RESULTS THROUGHMAT 36 37 1 0.2581 0.2581 528.0726 528.0726
LAB
As mentioned above from the condition {Cross
Correlation maximum peak value} x {Code Length}
341
Code Length (which is 45.23237) is must satisfy

for all satellites
Code Length =2046 chips;
Correlation Max Peak Value * Code length=
528.0726
(528.0726 45.23237) Satisfies Welch bound.
SV-1 Auto-Correlation
SV-14 Auto-Correlation
2500 2500
2000 2000
1500 1500
1000 1000
500 500
0 0
Fig 9. Cross Correlation Waveforms for various SVs
-500 -500
0 500 1000 1500 2000 2500 3000 3500 4000 4500 0 500 1000 1500 2000 2500 3000 3500 4000 4500
SV-23 SV-29
V. CONCLUSION
Auto-Correlation Auto-Correlation
2500 2500
2000 2000
1500 1500 Out of all the codes the PN sequence is an ideal test
1000 1000
signal for GNSS, as it simulates the random
500 500
characteristics of a digital signal and can be easily
0 0
-500 -500
generated. If the PN sequence is chosen to minimize
0 500 1000 1500 2000 2500 3000 3500 4000 4500 0 500 1000 1500 2000 2500 3000 3500 4000 4500
the cross correlation then the effect of jamming signal

SV-35 is minimized.
Auto-Correlation
2500
2000 VI. REFERENCES

1500 [1] B. Hofmann-Wellenhof, H. Lichtenegger, and E. Wasle, GNSS
Global Navigation Satellite Systems: GPS, GLONASS, Galileo,
1000
and more. Springer; 1 edition, December 2007.
500 [2] Haykin, S Communication Systems. 4th Edition. New York:
John Wiley and Sons, 2001.
0
[3] Official U.S. Government information about the Global
-500
Positioning System (GPS) and related topics, November 2011.
0 500 1000 1500 2000 2500 3000 3500 4000 4500
[Online]. Available:
[4] http://www.gps.gov/
Fig 8.Autocorrelation Waveforms for various SVs [5] N. Harper, Server-Side GPS and Assisted-GPS in Java, ser.
Artech House GNSS Technologies and Applications. Artech
House, 2009.
[6] ISSN 2249-6343 (JCTEE) - Analysis of Different Pseudo
SV-3 & SV-9 Noise Sequences by AlkaSawlikar, Manisha Sharma
[7] ACEEE Int. J. on Communications, Vol. 03, No. 03, Nov 2012-
Analysis and Simulation of Pseudo Ranging Noise codes for
Geo-Stationary Satellites and its Doppler Effect
[8] Global Journal of Mathematical Sciences: Theory and Practical.
ISSN 0974-3200 Volume 4, Number 2 (2012), pp. 121
132.Correlation of Pseudorandom Binary Sequences from de
SV-14 & SV-18 Bruijn Graphs
[9] International Global Navigation Satellite Systems Society
IGNSS Symposium 2009-Doppler Collision and cross
correlation interference in GPS receivers.
[10] R.N. Mutagi,Pseudo noise sequences tor engineers
[11] amonn Patrick Glennon,Cross Correlation Mitigation for C/A
Code GPS Receivers
SV-21 & SV-19 [12] International Journal of Engineering Research and Applications
(IJERA) ISSN: 2248-9622,Vol. 2, Issue 3, May-Jun 2012,
pp.2298-2304 -Simulation and Analysis of Binary Offset Carrier
Modulation for Modernized Global Navigation Satellite Systems
[13] ICD-GPS-200C (10th Oct 1993)
SV-21& SV-26 [14] BeiDou Navigation Satellite SystemSignal In SpaceInterface
Control Document - Open Service Signal B1I (Version 1.0)
[15] GPS Standard Positioning Service Signal Specifications,2nd
Edition,2nd June 1995.
342
Performance Assessment of IRNSS Signals:

BPSK(1) and BOC(5,2)
T. Tashi1, G. Ramesh2, L. Mruthunjaya3
IRNSS Project, Satellite Navigation Program, ISRO Satellite Centre, Bangalore-560017, India
tashi@isac.gov.in
gudivada@isac.gov.in
mithun@isac.gov.in
Abstract - A Global Navigation Satellite System (GNSS)

signals are transmitted in L1, L2 and L5-bands.
COMPASS and IRNSS signals are in S-band as well.
IRNSS will transmit two types of signals in L5 & S
band at centre frequency of 1176.45 MHz and
2492.028 MHz respectively. It provides two basic
services such as Standard Positioning Service (SPS)
for common users and Restricted Service (RS) for
special authorized users with modulation scheme of
BPSK(1) and BOC(5,2) respectively.
This paper describes the theoretical signal performance

assessment of BPSK(1) and BOC(5,2). The various
parameters such as Power Spectrum Density,Spectral
Separation Coefficient, Correlation functions, Root II. IRNSS
Mean Square (RMS) bandwidth, Code Tracking jitter, FREQUENCY BAND
Multipath performance etc has been considered for
The IRNSS SPS and RS services are
the performance assessment of IRNSS signals.
transmitted on L5 (1164-1215 MHz) and S
Keywords - Navigation, Spectrum, Signals, Satellite, (2483.5-2500 MHz) bands. The IRNSS carrier
Frequency frequencies and the bandwidths of transmission for
the SPS & RS services are shown in Table-II.
I. INTRODUCTION
Indian Space Research Organization (ISRO) is

implementing the Indian Regional Navigation Satellite
System (IRNSS) to provide navigation services to the
various users. IRNSS is an independent regional
navigation satellite system of India.
It is designed to provide position accuracy better than III. IRNSS SIGNAL MODULATION SCHEME
20m over India and the region extending about 1500
Kms around India. IRNSS system consists of Space Standard Positioning Service
Segment, Ground Segment & User Segment. IRNSS The Standard Positioning Service (SPS) signal is
will transmit two types of signals in L5 & S band. BPSK(1) modulated on L5 and S bands. The
navigation data at data rate of 50 sps (1/2 rate FEC
L5 band centre frequency is 1176.45 MHz and S encoded) is modulo 2 added to PRN code chipped at
band centre frequency is 2492.028 MHz. Both L5 and 1.023 Mcps identified for SPS service. The CDMA
S band consists of two downlinks. IRNSS provides modulated code, modulates the L5 and S carriers at
two basic services such as Standard Positioning 1176.45MHz and 2492.028 MHz respectively.
Service (SPS) for common users and Restricted
Service (RS) for special authorized users. The SPS Restricted Service
and RS signal are BPSK(1) and BOC(5,2) The Restricted Service (RS) is mainly for the
respectively modulated on L5 and S bands. authorized users. The RS signal is transmitted on L5
343
and S bands using Binary Offset Coding (BOC) IV. SIGNAL PERFORMANCE ASSESSMENT
modulation. It has two channels 1) Data channel and
2) Pilot or data less channel. In the data channel the
The performance assessment of IRNSS signals (L5
navigation data of 25 bps is modulo2 added with
& S) with modulation scheme of BPSK(1) and
designated PRN code chipped at 2.046 Mcps. The
BOC(5,2) has been carried out. The parameters
CDMA bit stream modulates the L5 and S carriers
considered for this analysis are Power Spectrum
using BOC(5,2). The pilot channel is transmitted
Density (PSD), Spectral Separation Coefficient
using Primary code and Secondary code without data
(SSC), Correlation function, Root Mean Square
modulation. The primary codes are chipped at 2.046
(RMS) bandwidth, Code Tracking jitter, Multipath
Mcps. The pilot carrier is in phase quadrature with the
performance etc. The above functionalities are
data channel.
simulated in the in-house developed MATLAB
codes.
Interplex Modulation
A. Power Spectral Density
IRNSS modulation scheme is shown in figure-1.
Each carrier is modulated by three signals namely
The Power Spectral Density (PSD) describes the
BPSK (1), data channel BOC(5,2) and Pilot channel
distribution of power within the signal with regard to
BOC(5,2). These when passed through power
frequency. The PSD for BPSK signal is given by
amplifier or TWTA operated at saturation will
produce a non constant envelope. Hence
additional fourth signal namely interplex signal is
added in order to have a constant envelope at the
output of TWTA RF spectrum.
A BOC signal may be viewed as being the product of
a
BPSK-R signal with a square wave sub-carrier, it is
represented by BOC(n,m), where n is the sub-
carrier
frequency (fs) in multiples of 1.023 MHz and m is
chip rate (fc) in multiples of 1.023 Mcps. The energy
of the signal is allocated around sub-carrier
frequency, so the maximum of the power spectrum is
shifted with respect to the centre frequency. PSD for
BOC(n,m) is given as
The BOC(5,2) modulation is composed of 5.115

MHz subcarrier frequency and 2.046 Mcps code
rate. The PSD for BPSK(1) and BOC(5,2) are
plotted in figure-3.A comparison is made
between PSD of BPSK(1) andBOC(5,2) with
other GNSS signals in L-5 (1176.45 MHz) band
such as BSPK(10)-Galileo-E5a, BPSK(10)-GPS
as shown in figure-4
344
SSC indicates theextent to which signals interfere

with each other. The smalleris the SSC of a
modulation with itself the greater is themodulation
resistance to multiple access interference withother
signals having the same modulation and thus better a
modulation can be used for many signals at different
receivepower levels. The normalized base band PSD
of BOCmodulation signal is
Over transmit bandwidth t , Gs(f) is the normalized

PSD and r is the receiver front end bandwidth.
Smaller SSC indicates the ability of a modulation to
provide enough processing gain against multiple
access interference from similar signals so that it
can support higher data rates. The smaller is the SSC
of the main signal, lesser it interferes with other
signal under consideration. More signals can be
accommodated at higher power levels without
interfering with reception of other GNSS signals.
The SSC values for IRNSS signals with other GNSS

signals in L-5 band and GlobalStar in S-band is given
in Table-III. It is observed that the IRNSS signals
BPSK(1) and BOC(5,2) has good spectral isolation
with BPSK(10) of Galileo-E5p / GPS- L5 and
GlobalStar in S-band as shown in Table-III
The fraction of power of various signals in the 24

MHz bandwidth is shown in fig.5. It is seen that
BPSK(1) and BOC(5,2) signals has more than 90%
of its energy within 3MHz and 16.5 MHz
bandwidth respectively as desired particularly for S- C. Correlation Function
band.
The auto-correlation function (ACF) is the most
B. Spectral Separation Coefficient important properties of any navigation signals. This
correlation property drives the performance of the
Spectral Separation Coefficient (SSC) quantifies the receiver in signal acquisition and tracking. For a
degree to which the interference having a given PSD given signal s(t), the ACF, R is expressed as s(
degrades theeffective C/No of a receiver channel. The
345
D. Root Mean Square (RMS) Bandwidth
Root Mean Squared (RMS) bandwidth of the signal is

a measure of the sharpness of the correlation peak.
RMS bandwidth () for a band limited signal is given
as
The auto-correlation function normalized to a chip is

calculated for various GNSS signals as shown in fig.-
6&7 The frequency-squared term in the RMS bandwidth
indicates that even very small amounts of high
frequency content in the signal can enable more
accurate code tracking. Intuitively, these high
frequency components produce sharper edges and
more distinct zero crossings in the waveform and
gives more accurate code tracking performance.
Signals with larger RMS bandwidths offer the
potential for more accurate code tracking and better
multipath rejection capability.
The rms at 10Hz loop bandwidth, 20 MHz signal
bandwidth has been calculated for BOC(5,2) and
BPSK(1), it is plotted in figure-8r. It is observed that
BOC(5,2) has higher RMS bandwidth value than
BPSK(1) for front-end bandwidth more than 4 MHz.
It indicates BOC(5,2) performance is better than
BPSK(1). The BOC(5,2) receiver should be
It is observed that the autocorrelation of a BPSK has operated for front-end bandwidth of higher than 12
a triangular shape in the interval [-Tc to Tc]. The MHz, ideal operation range would be 12-28 MHz and
BOC modulated signals has a narrower correlation 33-28MHz.
function around the origin but with side peaks. The
performance of a system is related to the ability of
identifying the main peak of the correlation function.
The BOC signal can potentially give better accuracy
due to sharper correlation function. The sharper the
main peak of the auto-correlation function better is
the performance of signal in terms of code tracking
jitter, multipath mitigation and signal acquisition. But
due to the presence of the side peaks the
improvement is trade-off with the complexity of the
receiver design.
From figure-9, it is observed that IRNSS RS signal

BOC(5,2) has values similar to GPS BPSK(10)-
PPS signal. However, it is lower than other GNSS
new authorized signal such as BOC(10,5) of GPS
military code (L1M, L2M), Galileo PRS signal
BOCcos(10,5) - (E6p), BOCcos(15,2.5) - (E1p) and
Compass authorized signal BOC(14,2).
346
end BW > 20 MHz.

LB reduces for BOC(4,2), BOC(5,2) and
BPSK(10) as we increase the receiver front-end
BW greater than 10 MHz.
LB reduces for BOC(10,5), BOCcos(10,5),
BOC(12,2), BOC(14,2) and BOCcos(15,2.5) for
receiver front-end bandwidth greater than 20 MHz.
These signals have much lower LB as compared to
other signals.
E. Tracking Jitter
Performance
The code tracking error performance of a given

signal shows its performance in terms of signal
acquisition and tracking as well as multipath
rejection capability. Lower the code tracking jitter
better is the signal performance. There are many
different designs for discriminators and tracking
loops for BPSK and BOC signals, code tracking
error may have different effects on each. However, a
lower bound on the code tracking error has been
developed that is independent of code tracking F. Multipath Performance
circuit design, yet is a tight bound in the sense that Multipath (MP) error is the most important
it provides reasonably accurate predictions of code unavoidable source of error contributing to the
tracking performance for well-designed tracking User Equivalent Range Error (UERE). The presence
circuits. For first-order performance analysis, model of MP signals provokes tracking error in the DLL
the receiver filter by an ideal rectangular transfer which is a code discriminator that utilizes early-
function with complex bandwidth r. Let the early- minus-late correlators and thus causing ranging
late spacing be (s), the correlation integration time errors.The MP tracking performance depends
within the receiver by T(s), and BL(Hz) be the one- onvarious signal and receiver parameters such as
sided noise equivalent bandwidth of the code modulation scheme, type of Pseudo Random Noise
tracking loop in Hz. The variance of the code (PRN) code, pre-correlation bandwidth and filter
tracking error for non-coherent early-late processing characteristics, chip spacing between correlators in
(NELP) given by; the DLL loop, type of discriminator used for tracking.
It also depends on relative amplitude of MP signal,
number of MP signals and MP delay.
The Multipath error of IRNSS RS signal for the case

of LOS and a single MP signal are analysed. The MP
has amplitude that is equal to 0.5 with respect to the
LOS. The MP delay is varied from 0 to in meters
with respect to the LOS. The phase of the MP is
For a small loop bandwidth, a lower bound (Cramer- taken equal to 0 and 180 with respect to the LOS
Raolower bound) on code tracking error in white (These values correspond to the maximum MP
noise for fixedfront-end bandwidth is independent of tracking error) and an infinite bandwidth filter in the
discriminator spacingand is given by receiver has been considered. Mulipath performance
It is observed that of IRNSS BOC(5,2) as compare to other various
LB for BPSK(1), BOC(1,1) & CBOC/TMBOC GNSS signals also plotted in figure-11.
(6,1,1/11) does not improve for receiver front-
347
[5] Elliott D. Kaplan, Christopher J. Hegarty, Understanding GPS

Principles and Applications, Second Edition
[6] B. W. Parkinson and J. J. Spilker , Global Positioning System:
Theory and Applications , Vol. 1, AIAA, 1996.
[7] Misra, P and Enge, P., Global Positioning System, Ganga-
Jamuna press , New York, 2011.
CONCLUSIONS
IRNSS signals BPSK(1) and BOC(5,2) contains

more than 90% its power within 3 MHz and 16.5
MHz bandwidth respectively. BPSK(1) and
BOC(5,2) has good spectral separation with
various GNSS signals in L5 and S-band.
BOC(5,2) has similar RMS bandwidth as

BPSK(10) of GPS PPS signal. It indicates that
IRNSS RS signals performance in terms of tracking
jitter, multipath etc is comparable with GPS PPS
signal.
Higher code rate and sub-carrier frequency
leads to better signal performance.
BOC signal produces multiple correlation
peaks which call for advanced receiver design
technology.
ACKNOWLEDGMENT
We express our sincere gratitude to Dr. S.K

ShivaKumar, Director, ISRO Satellite centre and
Shri. A.S Ganesahn, Programme Director, Satellite
Navigation Program, ISRO Satellite Centre,
Bangalore for their encouragement, guidance. We are
also thankful to all our colleagues in the project for
their encouragement and cooperation.
REFERENCES
[1] J. W. Betz, Design and Performance of Code Tracking for the

GPS M Code Signal, Proc. ION GPS00, September 2000.
[2] J. W. Betz, Major David B. Goldstein, Candidate Designs for
an
Additional Civil Signal in GPS Spectral Bands.
[3] Maurizio Fantino, Gianluca Marucco, Paolo Mulassano, Marco
Pini , Performance Analysis of MBOC, AltBOC and BOC
Modulations in Terms of Multipath Effects on the Carrier Tracking
Loop within GNSS Receivers.
[4] Paul Blunt, R. Weiler, Stephen Hodgart, Demonstration of
BOC(15, 2.5) acquisition and tracking with a prototype hardware
receiver
348
Sizing Design of Laser Retro Reflector Payload

and ILRS Network Planning Analysis for IRNSS
Anu Arora1, T. Subramanya Ganesh2
V. Jayaraman3, L. Srinivasan4, B. S. Chandrasekhar5
Navigation Systems Area (NSA)
ISRO Telemetry Tracking and Command Network (ISTRAC)
Indian Space Research Organisation (ISRO)
Bangalore, India
1
anuarora@istrac.org
2
ganesht@istrac.org
3
jayaram@istrac.org
4
lsvasan@istrac.org
5
director@istrac.gov.in
AbstractIndian Regional Navigation Satellite System (PNT) to the user with an accuracy of better than 20m
(IRNSS) is ISROs initiative to build an independent within the IRNSS service volume.
satellite navigation system based on a constellation of
GEO and GSO satellites. The IRNSS is a regional IRNSS space segment is formed by a constellation of
navigational system providing position accuracy better 7 satellites, which will broadcast ranging signals along
than 20m in and around the Indian region. IRNSS will with time. The satellites will be placed in three
have three main segments, namely the space segment, different orbital planes; 3 satellites in the Geo-
the ground segment and the user segment. The ground
stationary orbit (GEO) and 4 satellites in two Geo-
segment consists of a network of one-way ranging
stations and a network of two-way CDMA ranging
synchronous orbit (GSO) with an inclination of 29.
stations a precise timing facility along with control For enabling precise orbit determination of the IRNSS
centers for navigation and satellite control. It is also satellites, one-way CDMA ranging data from IRIMS,
planned to carry out Laser ranging of IRNSS satellites two-way CDMA ranging data from IRCDR and laser
on a campaign basis for calibration. In view of this a ranging data from ILRS is planned to be used.
suitable laser ranging payload was designed for IRNSS Satellite laser ranging technique, using a pulsed laser
satellites. This paper addresses the design and a telescope, measures the range of the retro-
considerations of retro reflectors for IRNSS, link reflectors from the ground with an achievable
margin analysis that was carried out and the strategy
accuracy of few centimeters. By fitting range
that is planned for ILRS network selection to achieve
the mission goals.
measurements from multiple laser ranging stations,
the orbit of the satellite can be determined precisely to
Nomenclature an accuracy of few meters. Laser tracking may be
used periodically to calibrate the IRNSS orbit
CDMA: Code Division Multiple Access determined by the other techniques.
ECI : Earth Centered Inertial
GEO : Geostationary Equatorial Orbit A suitable laser array was designed and fitted on the
GSO : Geosynchronous orbit earth-viewing surface of IRNSS satellites. This paper
ILRS : International Laser Ranging Service brings out the analysis that was carried out for sizing
IRNSS : Indian Regional Navigation Satellite System the corner cube and the laser array respectively and
LARA : LASER Retro reflector Array the link margin study with respect to ILRS stations.
NPE : Number of Photo Electrons This work is supported by mathematical modeling and
NNE : Number of Noise Electrons analysis.
SLR : Satellite Laser Ranging
SNR : Signal to Noise Ratio II.OBJECTIVES
The objective of the mathematical model is to study
the following
I. INTRODUCTION
1. Sizing of the individual retro-reflector cube
Indian Regional Navigation Satellite System (IRNSS)
for IRNSS
is ISROs initiative to build an independent satellite
2. Sizing of the Laser Retro Reflector Array
navigation system based on a constellation of 3 GEO
(LARA) for IRNSS
and 4 GSO satellites. The objectives of IRNSS are to
3. Computation of received power at SLR
provide accurate Position, Navigation and Time
station in terms of NPE, NNE and SNR
349
4.Sensitivity Analysis of various parameters This vector is also the direction of the laser beam
like vector in the inertial frame. Orbit Reference Frame is
Station-Satellite Geometry a coordinate system represented about the
Cube/Array Dimensions
SLR Uplink power
instantaneous position of the spacecraft with XO
Transmitter Efficiency etc towards the earth center and Z 0 towards the
A model for the above needs was developed in
MATLAB and it includes the following negative orbit normal while YO completes the right-
1. Satellite Orbit Propagator handed triad. They are computed as given below.
2. Satellite Attitude Profile Model
3. Link Margin Calculator X O = unit vector along the -ve spacecraft position
vector = - SCPOSVEC _ UNIT
III.MATHEMATICAL FORMULATION
Z0 = - SCPOSVEC_UNIT x
The aim of the modeling is to develop the required SCVELVEC_ UNIT
mathematical model to compute
The angle of incidence of the laser beam and YO = Z 0 x X O
X Y Z is
from a given station on the laser array
mounted on the IRNSS satellite The spacecraft body reference frame b b b
The number of photoelectrons that will defined about the spacecraft center of mass such that
arrive at the SLR for a transmitted pulse this frame is coincident with yaw, roll and pitch axes
of the body. When attitude bias is not introduced,
Modeling and Analysis ensures a flawless approach spacecraft body frame coincides with orbit reference
in operations and also cautions about possible frame. In the presence of attitude biases like the yaw
steering bias , the body frame is computed from the
X Y Z as
deviations in the future. The development of
mathematical model requires spacecraft position and knowledge of the orbit reference frame o o o
velocity vectors for every instant T.
given below [3]
T: year, month, day, hour, min, sec, msec

[Xsc Ysc Zsc]: Spacecraft Position Vector =
SCPOSVEC

X sc Ysc Z sc : Spacecraft Velocity Vector =
SCVELVEC
[Xstn Ystn Zstn]: Station Position Vector =
STNPOSVEC
Fig-1 depicts the geometry involved.

The line of sight vector for a given station and
satellite combination is then calculated as follows
STN 2 SCVEC = SCPOSVEC - STNPOSVEC

X Y Z = A X Y Z
b b b o o o
STN 2SCVEC _ Unit = unit ( STN 2SCVEC ) Where the transformation matrix A is given by
350
cos e12(1cos) e1e2(1cos) e3 sin e1e3(1cos) e2 sin

Ae1e2(1cos) e3 sin cos e22(1cos) e2e3(1cos) e1sin
ee (1cos) e sin e e (1cos) e sin cos e2(1cos)
13 2 23 1 3
where, e1, e2 and e3 are the direction cosines of the

rotation axis and
e 12 + e 22 + e 32 = 1
Euler angle and Euler axis method of attitude
representation is suitable here as the rotation takes
place about the body axis which is an arbitrary axis in
the Earth Centered Inertial Frame and is given in [3]. The optical cross section mostly depends on the
number of the retro reflectors used in the array as
The incident angle of the laser beam is defined as the well as the dimension of the individual retro reflector
angle between the normal to the laser retro reflector cube and its orientation in the satellite. In IRNSS
array and STN 2 SCVEC transformed to the body satellites it is planned to have a linear array mounted
on the earth-viewing panel such that the normal of the
frame. The ECI to Body transformation is done
retro reflector array is parallel to the yaw axis of the
through the direction cosine matrix formed with the
satellite. The optical cross section of a retro reflector
unit vectors of the body axes defined in the ECI
is defined [1] as
frame. Let Y , R, P be the unit vectors representing
the three axes of the spacecraft body. These vectors
cc = Acc(4/ ) = Acc (4Acc /2 )
are referenced in the ECI frame. It is clear that
where,
specifying the components of Y , R, P along the
three axes of the coordinate frame will fix the is the retro reflector reflectivity
orientation completely. This requires nine parameters
Acc = Rcc2 is the light collecting area of the retro
which can be regarded as the elements of a 3X3
matrix A, called the attitude matrix. is the effective solid angle occupied by the far field
diffraction pattern of the retro reflector
Y1 Y 2 Y 3
4/ is the on axis retro reflector gain
A= R1 R 2 R3
= 2/4, where is optical wave length of the laser
P1 P 2 P3 beam
However the above computation is valid only when
Where Y1, Y2, Y3 are the components of the Yaw there is no angle of incidence between the direction
Vector of the spacecraft in the ECI frame. Therefore of the incoming laser beam and the normal of the
any vector can be mapped from the ECI frame to the retro reflector. If the incoming laser beam is off
spacecraft body frame using the transformation normal then the effective retro reflector area is
matrix A reduced by the factor [1]
VBody = A.VECI (inc) = 2/ ( sin -1 - 2 tan ref ) cos inc
Thus the incident angle of the laser beam which is where,
nothing but the station-to satellite vector can be
computed as (inc) is the efficiency factor
inc is the angle of incidence
= cos 1 ( X B STN 2SCVEC _ UNIT )
and ref is the angle of refraction
where
X B is the Yaw axis of the spacecraft or the which can be determined by Snells law.
normal to the retro reflector array. Fig-2 depicts the
The quantities ref and of the previous equation are
geometry involved.
given by [1]
ref = sin -1(sin inc/n)
where n is cube index of refraction
351
= 1-2tan2 ref The total number of noise electrons is computed as

the root sum square of two components namely
The effective cross section is then given as [1]
optical background noise and detector dark current
eff = 2(inc) cc noise.
n noise is the number of noise electrons

A. Link Margin Analysis
Q is the quantum efficiency
The radar link equation is used to estimate the tgate is the ranging gate width
number of the photoelectrons that will be available at N is the background spectral radiance
the SLR for detection. If the number of received is the optical filter bandwidth
photoelectrons is higher than the threshold of the is the solid angle field of view in steradians
detector then a positive link margin is said to exist AT is the effective area of the telescope
and it is feasible to carry out laser ranging on the
r is the receiver coefficient
satellite.
IF is the IF filter coefficient
1) Number of Photo Electrons h is Plancks constant
is the frequency of operation
Number of Photo Electrons is given by [1] as
Dd is the diameter of the photo cathode
Ep ( x0 ) AT
n pe Q t a c c a r IF G id is the dark current density
h t trb R R2
2 2 e is the electron charge
For any station in the ILRS network each of the
where,
above parameters are well quantified and published
npe is the number of photo electrons that is received and can be looked up from [1]. These values were
by the SLR taken for computation of the number of noise
Q is the quantum efficiency electrons
Ep is the energy of the laser pulse
h is Plancks constant 3) S/N Ratio
is the frequency of operation
t is the transmission coefficient of the transmitter
S/N is given in [2] as
a is the atmospheric transmission coefficient
c is the cloud transmission coefficient n pe
is the optical cross section
S/N
nnoise
t is the beam divergence of the transmitter
trb is the turbulence coefficient
R is the range of the satellite IV.RESULTS AND DISCUSSION
r is the receiver coefficient
An analysis for the number of photo electron returns
IF is the IF filter coefficient
for a retro reflector array was carried out for the 7
AT is the effective area of the telescope
satellites and the various ILRS stations. Some sample
G is the aperture fitting parameter
results for the various options for number of cubes for
R is slant range to the satellite from ILRS station
a retro reflector array are shown below in Table 1.
The range of the satellite, the incident geometry and
The profile of the Number of Photo Electrons for a
the resultant optical cross section of the IRNSS
GEO satellite from one of the ILRS station have been
satellites are computed from the model. For any
presented in Figure 3 and the profile for a GSO
station in the ILRS network, the other parameters are
satellite from the ILRS station have been presented in
well quantified and published. Explanation and the
Figure 4.
value of the above parameters can be looked up from
[1].These values were taken for computation of the
An example of LARA Sizing Analysis
number of photoelectrons.
2) Electron Number of Noise
Electron number of noise is given by the governing
equation given in [2] as
Q N r AT r IF 2 id
2 2
nnoise t gate

e
D
h
d
4
352
3. Laser ranging has been carried out for the

first IRNSS satellite and the results are
encouraging.
REFERENCES
[1] Millimeter Accuracy Satellite Laser Ranging: A Review

John J. Degnan
[2] Automated tracking for advanced satellite laser ranging
system, Jan F. McGarry, Paul Titterton, Brion P. Conklin and
Peter J. Dunn
[3] Spacecraft Attitude Determination and Control by
JamesR.Wertz
A. ILRS Network Planning

Based on the NPE return analysis, a network was
proposed for ranging for each of the 7 IRNSS
satellites. The factor for consideration is that the NPE
return is more than or equal to 1.
CONCLUSION
A model has been developed for computing the laser
link margins for IRNSS satellites and the sizing of
the individual cube and the array have been arrived
at. Sensitivity analysis with respect to the cube
dimensions, array size and station characteristics have
also been performed. The following are the
conclusions.
1. A LARA of 40 cubes with each cube having
an approximate diameter of 38 mm was sized
for providing an adequate link margin for
IRNSS Laser ranging.
2. Based on the link margin analysis, a network
of ILRS stations was also proposed for
IRNSS.
353
An Adaptive Filter Design For Side Lobe

Reduction In Pulse Compression Radar System
Sadhu Sadhana, D.Tirumala rao
Dept. of ECE GMR institute of Technology Rajam, INDIA tirumalarao.d@gmrit.org
AbstractPulse compression techniques are widely used coefficients vary from code to code, and change as a
and active research topic in radar systems. There were function of code and filter length. There are several
few demerits like masking with the existing methods. techniques used to generate a mismatched filter to
In order to overcome these limitations , in this project minimize side lobes.
a R-G filter is used to suppress the sidelobes of the
radar coming out from the matched filter and BLMS
algorithm is used to calculate the filter coefficients. There are two methods which have been usually used
A weighting function is utilized to shape the sidelobe for side lobe reduction. Designing a mismatched
energy in an iterative manner that will yield more filter directly from codes. Employing an additional
sidelobe reduction. Comparison of the (R-G) filter using weighting network after the matched filter.
BLMS algorithm and the matched filter shows that at
the expense of insignificant loss in signal-to-noise ratio One possibility is conventionally called the (R-
(LSNR), adequate mainlobe-to-peak-sidelobe ratio G) filters which were introduced by Rihaczek and
(MSR) can be achieved. Golden [3]. The (R-G) filters have advantages of
simple filter structure and relatively suitable
I. INTRODUCTION performance.
In this work an adaptive (R-G) filter is used and
High range-resolution is important for many radar the Block LMS algorithm is used to calculate the
applications. Pulse compression technique is filter unknown coefficients. The filter yields a
employed to achieve high range resolution with satisfactory mean square sidelobe level (MSSL) and
enough low power[1]. Radar range resolution an adequate MSR.
depends on the Bandwidth of the received signal.
Bandwidth of a pulse is inversely proportional to the II. DESIGN ISSUES
pulse duration. So, short pulses are better for
range resolution. But
large Bandwidth can increase system complexity

and interference. A long pulse can have same
Bandwidth of short pulse if it is modulated in phase. Fig. 1. Filter structure in general
The modulated long pulse is compressed by the The overall filter structure is shown in Fig. 1. The (R-
matched filter. To modulate the long pulse various G) filter[4] in Fig. 1 is employed to suppress the
types of encodings are used. sidelobe level. Consider the input waveform at the
receiver front end x(t):
In Biphase coding, Changes in phase can be used to
increase the signal bandwidth of a long pulse. A long
pulse is divided into some sub pulses each of same
width. An increase in Bandwidth is achieved by
changing the phase of each sub pulse. The phase of
the
transmitted signal alternates between 0 and 180
degrees according to the binary sequences[2]. When
a target echo signal passes a matched filter, the filter
output consists of a spike-like main lobe and some
noise-like side lobes. These side lobes can mask the
main lobe of weak target echo signals. To prevent
this problem, there is a need to design a binary code
whose autocorrelation function mainlobeto-peak-
sidelobe ratio (MSR) is maximized for a given code
length. Furthermore, a side lobe reduction filter is
used to achieve an adequate MSR. The filter
354
The continuous part i.e., u(a, b) essentially Hence, the transfer function of sidelobe reduction
determines the output pulse shape, whereas the filter is suggested as the following parameterized
discrete part i.e., op reveals the relative output form
amplitude. Therefore, they can be evaluated
separately [5] and the designing can be performed in
discrete domain. The true output of matched filter can
be also be written as
where p t (t) is the autocorrelation function of the p

(t).
The second term in (7) is due to sidelobes, so this true
correlation function (r(t)) may not have some
required properties for a given application. Therefore
one method is try to synthesize the objective
correlation function d(t) by resorting to a filtering
step H(f) after the matched filter step: r
355
case, performances of the suggested filter and the

matched filter are compared for a point target. The
second case has several targets distributed in range.
In this case the accuracy of the matched filter is
expected degrade significantly because of a masking
problem. The Dopplers effect is evaluated in the
third case. Furthermore, the filter is compared with
some other filters in individual cases.
A. Point Target:
The waveform used for this case is the length N = 40
minimum peak sidelobe (MPS) biphase coded signal
which is5A5C3BC444 in hex representation. By
considering a point target scenario, we compare the
matched filter with the (R-G-2) filter using BLMS
algorithm. Fig. 4 shows the outputs of standard
matched filter (SMF) and (R-G-2)BLMS filter.
Fig. 4. Output of SMF and (R-G-2)BLMS filter for 40 bit

binary code.
B. Doppler effect:
Fig. 5. shows the MSR results in which
Doppler frequency varies from 0 to 100 kHz .
As we can see in Fig. 5, the sensitivity of the
(R-G) filter using Block LMS algorithm is
raised by increasing the stages. However, the
MSR results of the (R-G)BLMS filters are much
better than the matched filter and (R-G-3)WF in
the range 0 to 70 kHz.
IV. PERFORMANCE
EVALUATION
To demonstrate the acceptability of the
filters performance, the (R-G) filter is compared Fig. 5. MSR results of matched filter and one, two (R-G)WF
with the matched filter for three cases. In the first filters
356
C. Comparison with other Filters: conference on aerospace and electronic systems

10.1109/TAES.2012.6129640, January 2012
In Table I, the performance(MSR) of the [5] Wang, W. C. and Scholtz, R. A. Sequence design for IIR
matched filter, (R- G-2) filter using BLMS inverse filter pulse compression. IEEE Transactions on Aerospace
algorithm and (R-G-2) Filter using wiener and Electronic Systems, 31 (Apr. 1995), 670684.
filtering, weighted wiener filtering, least squares [6] J.Jebastine,Dr.B.Sheela Rani,Implementation of block least
mean square adaptive algorithm for effective noise cancellation in
and linear programming methods are compared. speech signal, IJECIE,Research vol.1,Issue.1 (2011), 1-11.
CONCLUSION
A m-stage adaptive filter is designed for pulse

compression radar systems. This filter yields a
remarkable reduction in sidelobe levels. The number
of stages is chosen based on the required value of
MSR. The filter coefficients were obtained by using
several algorithms. one of the best from the previous
works is wiener filtering, which gives the MSR
value of 33.25dB for the 2nd stage (R-G) filter. In
this work, by using BLMS algorithm, more side
lobe reduction can be achieved and MSR value is
raised to 40.05dB.
The weighting function in an iterative algorithm

may be modified in order to have more reduction in
the sidelobe level. Such filters can be used in
tracking applications or in target recognition
systems.
References
[1] Skolnik, M. I. Introduction to Radar Systems (3rd ed.). New

York:
McGraw-Hill, 2001, 339369.
[2] Skolnik, M. I. Radar Handbook (2nd ed.). McGraw-Hill, 1990,
1015.
[3] Rihaczek, A. W. and Golden, R. M. Range sidelobe
suppression for Barker codes. IEEE Transactions on Aerospace
and Electronic Systems, AES-7 (Nov. 1971), 10871092.
[4] Akbaripour,A.; Bastani,M.H. Range Sidelobe Reduction Filter
Design for Binary Coded Pulse compression system international
357
Performance Evaluation of Non-Uniform PRF

Radar Signals Using MOGA
B Jithendra Kumar Reddy, G Manmadha Rao,
Dept. of ECE GMR institute of Technology Rajam, INDI
Abstract In this paper a novel method for solving the optimization of each objective. Many, or even most, real
multifunction optimization problems proposes. Radar engineering problems actually do have multiple-
is mainly used to detect the targets in different objectives, i.e., minimize cost, maximize performance,
environments by analyzing the echo signal from the maximize reliability, etc. These are difcult but realistic
target. One method of distinguishing multiple-time- problems. GA are a popular meta-heuristic that is
around echoes from unambiguous echoes is to operate particularly well-suited for this class of problems.
with a varying pulse repetition frequency. The different
pulse repetition frequencies also allow eliminating
Traditional GA are customized to accommodate multi-
the blind speeds in finding the moving targets objective problems by using specialized tness functions
information. The use of more than one PRF offers and introducing methods to promote solution diversity.
additional flexibility in the
design of MTI radars. The detection level of the target The general approach to multiple-objective optimization
can be decided by the performance factors called merit is to determine an entire Pareto optimal solution set or a
factor and discrimination factors. The performance representative subset. A Pareto optimal set is a set of
factors are varying with the variation of PRI sequence. solutions that are non-dominated with respect to each
Hence we are generating the non-uniform PRI sequence other. While moving from one Pareto solution to
to maximize the both performance factors using multi another, there is always a certain amount of sacrice in
objective genetic algorithm optimization technique.
one objective(s) to achieve a certain amount of gain in
the other(s).
Index Terms Merit Factor, Discrimination Factor,
MOGA.
II. MULTI-OBJECTIVE OPTIMIZATION
I. INTRODUCTION FORMULATION
Consider a decision-maker who wishes to
The objective of this paper is to an optimize non- optimize K objectives such that the objectives are
uniform PRI parameters of Radar signals by multiple- non-commensurable and the decision-maker has no
objective optimization methods using genetic algorithms clear preference of the objectives relative to each
(GA). For multiple-objective problems, the objectives other. Without loss of generality, all objectives are
are generally conicting, preventing simultaneous of the minimization type--a minimization type
optimization of each objective. Many, or even most, real objective can be converted to a maximization type
engineering problems actually do have multiple- by multiplying negative one. A minimization multi-
objectives, i.e., minimize cost, maximize performance, objective decision problem with K objectives is
maximize reliability, etc. These are difcult but realistic dened as follows:
problems. GA are a popular meta-heuristic that is
particularly well-suited for this class of problems. Given an n-dimensional decision
Traditional GA is customized to accommodate multi-
variable vector x {x1 ,...., xn } in the solution
objective problems by using specialized tness functions
and introducing methods to promote solution diversity. space x, find avector x* that minimizes a given
The general approach to multiple-objective optimization set of K
is to determine an entire Pareto optimal solution set or a
representative subset. A Pareto optimal set is a set of
solutions that are non-dominated with respect to each
other. While moving from one Pareto solution to
another, there is always a certain amount of sacrice in
one objective(s) to achieve a certain amount of gain in
the other(s). The objective of this paper is to an In many real-life problems, objectives under
optimize the parameters of Radar signals by multiple- consideration conict with each other. Hence,
objective optimization methods using genetic algorithms optimizing x with respect to a single objective often
(GA). For multiple-objective problems, the objectives results in unacceptable results with respect to the
are generally conicting, preventing simultaneous other objectives. Therefore, perfect multi- objective
358
solution that simultaneously optimizes each of GA may exploit structures of good solutions with
objective function is almost impossible. A respect to different objectives to create new non-
reasonable solution to a multi- objective problem is dominated solutions in unexplored parts of the Pareto
to investigate a set of solutions, each of which front. In addition, most multi-objective GA do not
satises the objectives at an acceptable level require the user to prioritize, scale, or weigh
without being dominated by any other solution. If objectives.
all objective functions are for minimization, a
feasible Therefore, GA have been the most popular heuristic
approach to multi-objective design and optimization
problems. Jones et all reported that 90% of the
approaches to multi objective optimization aimed to
approximate the true Pareto front for the underlying
is said to be a pareto optimal if it is not problem. A majority of these used a meta-heuristic
dominated by any other solution in the solution technique, and 70% of all met heuristics
space. A Pareto optimal solution cannot be approaches were based on evolutionary approaches.
improved with respect to any objective without
worsening at least one other objective. The set of all
IV. DESIGN ISSUES AND COMPONENTS OF MULTI-
feasible non-dominated solutions in X is referred OBJECTIVE
to as the Pareto optimal set, and for a given Pareto GA:
optimal set; the corresponding objective function
values in the objective space are called the Pareto Assign no. of chromosomes(n),
front. For many problems, the number of Pareto
Assign no. of bits(b).
optimal solutions is enormous (perhaps innite).
Assign no. of variables to optimize(m).
Initialization of population:
The ultimate goal of a multi-objective optimization
Generate population by using a random function
algorithm is to identify solutions in the Pareto
and this random function values multiply with the
optimal set. However, identifying the entire Pareto
R max and Rmin values such that the generated
optimal set, for many multi-objective problems, is
random values become with in specified range.
practically impossible due to its size. In addition, for
r= rand(n,m*b)-0.5;
many problems, especially for combinatorial
P=(Rmin)+((Rmax-
optimization problems, proof solution optimality is
Rmin)/1023)*r;
computationally infeasible. Therefore, a practical
The generated population acts as
approach to multi-objective optimization to
chromosomes.
investigate a set ofsolutions (the best-known Pareto
Calculate the fitness values i.e., merit factor
set) that represent the Paretooptimal set as well as
and discrimination factor,
possible. With these concerns in mind, a
Assign the rank based on fitness values.
multi-objective optimization approach should achieve
Calculate crowding distance
the following three conflicting goals.
The best-known Pareto front should capture the

whole spectrum of the Pareto front. This
requires investigating solutions at the extreme ends
of the objective function space.
III. MULTI-OBJECTIVE GA
Being a population-based approach, GA are well

suited solve multi-objective optimization problems. A
generic single- objective GA can be modied to nd
a set of multiple non- dominated solutions in a Selection is performing to chosen the
single run. The ability of GA to simultaneously individual population for crossover operation.
search different regions of solution space makes it
Crossover operation is done by combining
possible to nd a diverse set of solutions for difcult
two parents to produce Childs
problems with non-convex, discontinuous, and
multi-modal solutions spaces. The crossover operator
359
The process of entire operation gives the following flow

chart diagram, it specify the step by step procedure in the
algorithm.
V. RESULTS:
Pareto front of optimized PCW signal:
360
VI. CONCLUSION
This paper consists non uniform PRI pcw radar signal .as
the non-uniform PRI avoids the blind speeds. Using multii
objective genetic algorithm merit factor and discrimination
factors have been set as two object functions using MOGA
approach better results have been obtained for PSLR,
ISLR, with compared to conventional uniform and
nonuniform(ascending, descending, and random) PRI
strategies
REFERENCES
[1] K. Deb, S. Agrawal, A. Pratap, and T. Meyarivan, Afast elitist non-

dominated sorting genetic algorithm for multi-objective optimization:
NSGA-II, in Proc Parallel Problem Solving from Nature VI, 2000, pp.
849858.
[2] Osycza and S. Kundu, A new method to solve generalized
multicriteria optimization problems using the simple genetic algorithm,
Struct. Optimiz., no. 10, pp. 9499, 1995.
[3] A New Optimal Design Method of Rotor Slot of Three-Phase
Squirrel Cage Induction Motor for NEMA Class D Speed-Torque
Characteristic Using Multi-Objective Optimization Algorithm: IEEE
Trans.2012. Dianhai Zhang, Chang Soon Park, and Chang Seop Koh
[4] K. Rasheed and H. Hirsh, Informed operators: Speeding up genetic-
algorithmbased design optimization using reduced models, in Proc.
Genetic and Evolutionary Computation Conf. (GECCO2000), 2000, pp.
628635.
[5] C. M. Fonseca and P. J. Fleming, Genetic algorithms for
multiobjective optimization: Formulation, discussion and generalization,
in Proc. 5th Int. Conf. Genetic Algorithms, S. Forrest, Ed. San Mateo,
CA: Morgan Kauffman, 1993, pp. 416423.
[6] D. E. Goldberg, Genetic Algorithms in Search, Optimization and
Machine Learning. Reading, MA: Addison Wesley, 1989
[7] Multi-objective optimization using genetic algorithms: Abdullah
Konak, David W.Coit, Alice E.Smith
[8] Multi-Objective Optimization by Genetic Algorithms: A review by
Hisashi Tamaki, Hajinme Kita, Shigenobu Kobayashi
[9] An introduction to Genitic algorithm : PP 24-63 by Melanie
Mitchell.
[10] Manmadha Rao.G, Raja Rajeswari - RADAR target detection
using LFM chirp-Proceedings of the international conference on
systematics, Cybermatics and informaticsICSCI-09,Hyderabad.
[11] Introduction to Radar Systems by Merrill I. Skolnik,3rd Edition,
McGraw-Hill publications-2006,Pages456-487.
[12] Goldberg, D. E. (1989), Genetic Algorithms in search, optimization
and machine learning, Addison-Wesley, Massachusetts.
361
Laser Assisted Vision Based Tracking for

Gimbal Control
SRIKANTH S S, Mr K Kanakaraju
Design Engineer, Technical director
System Controls Technology Solutions Pvt Ltd Bangalore, India Srikanth.shapur@gmail.com
AbstractIn recent years interest on vision based calculate the position of interested target through
tracking systems for applications like navigation, sensory information obtained through a variety of
surveillance, robotics, augmented reality, automotive, sensors such as position, velocity, and vision sensor.
etc., has grown. Vision based tracking techniques face The information obtained is processed through on
the challenge of detecting and tracking the moving board system or in ground station to give the target
object for extracting accurate target position
location, position co-ordinates, etc. Earlier seperate
measurements under dark conditions. The problem
becomes more complex to track a moving object using a
ground computer was used for processing the
camera while keeping the object within the image. This information obtained through sensors. This system
work presents the development of a vision based real depends heavily on the communication between the
time tracking system with the use of a laser point platform and the ground computer and a
and gimbal mounted camera. A better measure of a communication failure will result in tracking system
targets position is obtained through laser because it is failure. The solution for this type of tracking system
capable of determining the object at long range and is on-board visual tracking for platform (gimbal)
helps the vision based tracking system to track the stabilization[1].This onboard tracking system
objects under dark conditions. Stabilization unit of the
eliminates the dependency on the communication with
gimbal mounted camera is used to keep the object
within the image by tracking the laser point and
the ground station and makes the system less prone to
controlling the gimbal position. The algorithm for failure. Tracking and targeting a target with visual
tracking the laser needs detection of laser point and servo control simply requires recognizing the target
calculation of X and Y co-ordinates of the laser point. within the field of view, capture this target,
Segmenetation and intensity based algorithm is used for following it, and finally giving the order to fire on
detection and centroid tracking algorithm is used to the target. This actually requires an optical sensor to
calculate the co-ordinates of laser point. The video based obtain the field of view, a signal processing to
detection and tracking algorithm was prototyped and recognize the target, a control system for the target
validated using MATLAB with a USB connected camera
tracking loop[3]. Compared to the traditional vision-
mounted on the gimbal system interfaced to the PC
using RS232. The Atmega128 microcontroller on
based tracking system, as a new kind of
board the gimbal system is programmed for receiving measurement instrument, the laser range scanner has
the co-ordinate data and controlling the motors. The received the increasing attention for solving tracking
validated algorithm is ported to TMS320DM642 DSP problems in recent years[4]. To keep the target within
and evaluated for its real-time performance. The the field of view, stabilized platform approach is
embedded DSP based system was able to track a laser employed. The stabilized platform is a two axis pan-
point for camera stabilisation in real time at 30 fps and tilt platform which recieves sensor error signals for
720x480 resolution. stabilization.
1 INTRODUCTION Specific objects can be recognized relatively easily
using vision-based techniques, but these techniques
experience difficulties in extracting accurate target
Object tracking is an important field that as attracted position measurements. A better measure of a targets
lot of interest in computer vision. Tracking has vast position is obtained through laser. Laser is capable of
range of applications in different areas such as determining distances to objects at long range,
surveillance, traffic monitoring, gaming, gesture with high accuracy.The paper aims at developing a
recognition, augmented reality, defence, human laser assisted vision based tracking system with
activity, robotics, industrial automation, etc. Object stabilization unit to keep the target within the field
tracking can be defined as the process of of view.
segmenting an object of interest from a video scene
and track its motion and orientation in order to
extract useful information. The information In this paper, video based detection and tracking
obtained can be used for activity control. Visual algorithm was developed. For detection of laser point,
target tracking is to identify the location and to segmentation and intensity algorithm was used
362
and for tracking centroid algorithm was employed. Image acquisition setup: It consists of webcam, video
The developed algorithm was prototyped and camera with suitable interface connected to processor
validated using MATLAB with USB connected or personal computer.
camera mounted on the gimbal system interfaced to Image analysis: The tools which are necessary for
the PC using RS232. The co-ordinates calculated analyzing the content in the video or image captured
are transmitted to the gimbal system for controlling for detecting and tracking the laser point.Processor:
its azimuth and elevation motors. The Atmega128 The Digital signal processor(TMS320DM642) with
micro-controller on board the gimbal system is a dedicated video processing unit is used for
programmed for receiving the coordinate data and processing the video for detecting and tracking the
controlling the motors. The validated algorithm is laser point.
ported to TMS320DM642 DSP and evaluated for its
real-time performance. Gimbal Control: After obtaining the co-ordinate
pixel values, the mechanical action is taken for
2 DESIGN OF LASER ASSISTED VISUAL controlling the motors of a gimbal using serial port
TRACKING SYSTEM of a PC.
A detailed description of each module and the 2.3 Detection and Tracking Algorithm
algorithm for the real-time visual tracking system will
be given followed by the experimental results and Processing algorithm is used for the extracted frame
discussion on the performance of the system to detect and track the laser point. The Flowchart for
algorithm is shown in Figure 2.
2.1 Problem Statement
The problem of detecting and tracking the object

and extracting the position information is a challenge
for vision based tracking systems under dark
conditions. Tracking a moving object and keeping the
object within the field of view of the camera is
another problem of vision based tracking systems. To
overcome these problems, the laser and stabilisation
unit are used in vision based tracking systems. The
proposed laser assisted vision based tracking
system is developed to calculate the co-ordinates of
laser point which is necessary to control the gimbal
system
2.2 System Architecture

The simplified block diagram of the system is
shown in Figure 1.
The processing of image for tracking laser point

includes detection and calculating the co-ordinates.
Segmentation technique is used for detection and co-
ordinates of laser point are calculated for tracking.
The vision based tracking system The extracted RGB image from the video is further
for gimbal stabilization consists of the following resized to resolution of 640x640. Resized image is
Blocks segmented with the use of morphological
363
operations to detect the laser point in the image.

After performing the morphological operations, the 2.4.1Simulik
image is binary with values 1's and 0's. The red Model
laser point in the image is one and other pixels
are zero. Each pixel is compared to the value of red The developed MATLAB algorithm is again
intensity. If the pixel value is greater than or equal to developed using simulink, which is a tool for
red laser point pixel, then laser point is detected. modelling, simulating and analyzing multi-domain
Once the red laser point is detected, next step is to dynamic systems. Simulink model is developed by
calculate the co-ordinates and track the laser point choosing the DSP blocks of DM642, image
The co-ordinate calculation is necessary for tracking processing blocks and embedded function block. The
the laser point. Before calculating the co-ordinates, simulink model is developed by configuring
the resolution of 4 edges of the image are defined the specific target preference block and DSP/BIOS.
After the detection, the respective row and column The simulink model is shown in Figure 4.
value i.e. is obtained for the laser point. The
obtained row and column value is compared with
resolution of 4 edges and after comparison, the 4
edges of images are updated with new values and
from these values, the X and Y co-ordinate values
are calculated.
Laser X = leftmost+Rightmost/2
Laser Y = Topmost+Bottomost/2
The algorithm was developed and validated using

MATLAB. The algorithm is again modelled in
SIMULINK for DSP implementation. 2.4.2 Code Generation
2.4 DSP Implementation
The designed image processing algorithm is

implemented on digital signal processor(DSP). The
TMS320DM642 evaluation module is used for
implementation. The DM642 EVM is a
development board that enables evaluation of and
design with the DM642 Digital Media Processor. It is
designed to work with Texas Instruments Code
Composer Studio software tools connected through a
JTAG emulator.
The design flow for implementing the algorithm in

DSP platform is shown in Figure 3.The algorithm
implementation on DSP is divided into two stages ,
they are Implementing the algorithm on DSP requires board
specific C code. The C code can be generated
Developing simulink model automatically from the developed simulink model
Code generation and dumping with target preference block TI C6000 DSP
DM642EVM. The model is made to build
incrementally and C code is generated with all header
files required for the board. Code Composer Studio
version 3.3 IDE accepts that downloaded code and
generates an executable (binary) file i.e. a.out file is
generated which is code required for board. The a.out
file is made to dump on board through JTAG. Once
the file is dumped on board, the CCD camera
364
connected to video in port of the board captures live The serial communication is used to transmit X and
video where the laser point. The video is processed to Y coordinate values from the PC to the micro-
give necessary co-ordinates for tracking the laser controller in the Gimbal system for controlling the
point. The co-ordinates from the processor is azimuth and elevation motor. The PC with Matlab is
transmitted serially to micro-controller in the Gimbal interfaced to Gimbal system through RS232 cable. In
to keep the target within the field of view . the PC, serial com port 3 is intialized with the baudrate
of 9600 and databits of 8. After intialization, the serial
2.5 Gimbal Control and Processing Unit com port is opened. Once the com port is opened, the
X and Y co-ordinate values are sent to controller in
The camera is focused on background where the form of packet as show in Figure 5.
laser is illuminated. The image with laser point is
captured and processed to give the X and Y co-
ordinate values. The X and Y co-ordinate values from
PC or DSP is sent to controller in the Gimbal system
through serial port i.e. RS232. The gimbal was
controlled from two ways,
1. The processing results from the algorithm 2.7 Hardware Implementation
validated in MATLAB
The software design implemented in PC and processor
2. The processing results from the DSP is interfaced to gimbal system. The obtained X and Y
coordinates of the laser point are fed to controller via
RS232. Based on these values, the motors of the
gimbal system take necessary action to keep the
camera focussed to the laser point.
Here the work was done on algorithm development for

controlling the motor to keep the camera focussed and
the developed algorithm was implemented on AVR
studios.
In this design, the image focused is divided into
different bandwidths for the motor control. Based on The hardware design of gimbal system consists of two
this bandwidths the motor is controlled in clock-wise independent motion drive systems mounted one over
and anti-clockwise. The different bandwidths are the other, azimuth axis consist of a Servo Actuator
shown in Figure 5. coupled with encoder and gear and elevation axis
consist of an individual BLDC motor coupled to a set
The resolution 640x640 is divided by 10. this is done
of gear trains and an incremental encoder for position
because the transmission of data is limited to 8 bits
feedback.
and makes the operation reliable. The resolution 0 to
64 in X axis is divided into different bandwidths for
motor control. The motor speed is controlled in Two Axis Surveillance Pedestal assemblies contain
clockwise and anti-clockwise direction based on the two L clamp arrangement, which is used for
different bandwidths as shown in Figure4. mounting the CCD/IR camera on the scanner and
laser payload. It is shown in Figure
For Gimbal control, ATMEGA128 micro-controller is

used.The control and reception algorithm for
controller is developed in AVR studios 4. Based on
the reception X and Y values, the gimbal is controlled.
2.6 Communication Protocol and Packet Figure 9: Hardware Implementation
Description
365
3 RESULTS AND DISCUSSION RESULTS
The implementation and analysis of developed The TMS320DM642 DSP is chosen for
algorithm has been compared for its computation time implementation. For this, a simulink model is
on various platforms like MATLAB, TMS320DM642 developed using specific DSP blockset. The captured
DSP. Initially the algorithm was tested with dark video is processed in TMS320DM642 DSP and the
background with red laser source. The algorithhm was tracking result is displayed on TFT LCD monitor. The
able to detect and track the laser spot. The tracking DSP setup and results are shown in Figure 13.
result is represented with the intersection of two red
lines as shown in Figure10.
Figure 13: DSP results of tracked laser spot

Figure 10: Track Result with Dark The implementation of vision based tracking
Background algorithm has been carried out in Matlab and
The algorithm was tested in lighting conditions and TMS320DM642 DSP. Table 1 gives the timing
was able to detect and track the laser spot. It is shown comparison for implementation of vision based
in Figure 11 tracking system on DSP as compared with Matlab.
The computation time taken by the TMS320DM642
to track the laser point is less compared to the
MATLAB.
Table 1: Timing comparison of vision based

tracking system on various
platforms
4.CONCLUSIONS
Figure 11: Track result with Lighting
Conditions
Vision based tracking system for tracking the laser
point has been implemented for gimbal control to
The tracking algorithm was modelled in simulink and
keep the target within the field of view. The
was tested with different backgrounds. The algorithm
developed algorithm was tested on MATLAB and
was able to track the laser spot and was giving exact
DSP for its functionality. The co-ordinates calculated
X and Y co-ordinates of laser spot to the gimbal for
from the tracking algorithm was serially transmitted
control. This is shown in Figure12.
to Gimbal mounted camera for controlling and
keeping the object within the field of view. Laser
pointing mechanism can be employed to enhance the
detection and tracking of moving objects under low
light conditions and with low resolution cameras.
The tracking algorithm has been developed for
static camera movements. In future, the work can
be extended to track moving object with dynamic
camera movements . The algorithm will be improved
Figure 12: Simulink to adapt with different surrounding conditions.
366
ACKNOWLEDGMENTS
This work was supported by System Controls
Technologies Pvt Limited. I wish to thank our
guidesMr. K. Kanakaraju, Technical Director,
System Controls Technology Solutions Pvt and Mr.
Naveen K. S, Senior Design Engineer at System
Controls Technology Solutions Pvt. Ltd. for their
support and co-operation during the project.
REFERENCES
[1] A.Quadir, J.Neubert, W.Semake, On Board Visual Tracking

with Unmanned Aircraft System(UAS), Infotech Aerospace
Conference, March 2012
[2] Mokbel, H., Ying, L., Roshdy, A. 'Design Optimization of
the Inner Gimbal for Dual Axis Inertially Stabilized Platform
Using Finite Element Modal Analysis'
[3] Shiao, Y. S. and Road, S. (2001) 'Design and Implementation
of Real-Time Tracking System Based on Vision Servo
Control'. Tamkang Journal of Science and Engineering, 4(1), 45-
58
[4] Papanikolopolous, N., and Kanade, T. (1993) 'Visual Tracking
of Moving Target by a Camera Mounted on a Robot' IEEE
Transactions on Robotics and Automation, 9(1)15-23
[5] Song, X., Zhao, H. 'Fusion of Laser and Vision for
Multiple Targets Tracking via On-line Learning'
[6] Yilmaz,A., Javed,O., and Shah,M. (2006).' Object
tracking: A survey'. ACM Computing Surveys, 38(4)16-22
367
A Novel Architecture for Real Time Ultra-Tight

Integration of Miniaturized GPS-GLONASS-
GAGAN L1 Receiver and
MEMS Inertial Sensors
Gundra Satheesh Reddy# , Manjit Kumar $
#
Outstanding Scientist & Director, Research Center Imarat, DRDO, Hyderabad
$
Scientist, Research Center Imarat, DRDO, Hyderabad
Abstract- In the last few years the work towards the receiver in a single package [5,8,9]. The basic idea
integration of the Inertial Navigation System (INS) with of this system is to assure an optimal usage of the
Global Navigation Satellite System (GNSS) is being GPS+GAGAN and GLONASS measurements in
actively carried out by multiple R&D groups across the
two operating modes. A nineteen-state Extended
globe. Presently the investigations are going on to
evaluate new integration schemes using MEMS inertial Kalman Filter implemented and tested under Ultra
sensors with GNSS sensors towards the possible tightly coupled scheme to improve both INS and
development of new generation low cost, small size and GNSS performance on many fronts as discussed in
low weight system by targeting multiple navigation the end of this paper [2,4,5]. Further analysis on this
applications in the industry. To meet the prime scheme is in progress.
requirement of this class of work in various defence
navigation applications, a novel architecture for Ultra
Tightly integrated navigation scheme for MEMS INS II. MEMS INERTIAL SENSORS
with miniaturized GPS-GLONASS-GAGAN receiver TECHNOLOGY
has been realized. The partial work in this paper is
focused on the design and development of PLL Doppler The micro electromechanical system based on the
aid to Scalar based Receiver tracking loop for achieving
micro electronics technology is an integration of
more accurate navigation solution, improved tracking
ability in dynamics and enhanced anti-jamming
silicon micro technology, LIGA technology and
performance of the combined system. precision mechanical technology. It is also an
intelligent combination system of micro sensors,
Keywords: MEMS, GoM, INS, ECEF, UTC, PLL etc micro actuators, signal conditioning and control
circuits, interfaces, communication and power
supply with the modern information technology fig
I. INTRODUCTION
1[5]. Micro inertial devices, consisting of micro
accelerometers and micro gyroscopes, have the
Combined MEMS and GNSS/INS optimum character of smaller dimensions, lower costs and
solutions offers low power, low cost and light available integration on one chip over the traditional
weight navigation sensors for autonomous vehicle devices.
navigation. INS in itself is a self-contained device
which operates independently of any external signals
or inputs, providing a complete set of navigation
parameters, including position velocity and attitude,
with a high data rate [1]. However, one of the main
drawbacks of INS especially with MEMS class
sensors when operated in standalone mode leads
rapid growth of systematic errors with time [5]. In
contrast to INS short-term positioning accuracy,
satellite based GNSS navigation techniques can
offer relatively consistent accuracy if sufficient
GNSS signal can be tracked during the entire course
of navigation [8].
This paper describes the design of a integrated
navigation systems for short and medium long range
navigation applications, which comprises of MEMS
Fig.1Basic MEMS Structure
IMU and a miniaturized multi ASICs based G3oM
368
a) Micro accelerometer proof mass is in the middle of opposite electrodes.

The outputs of two electrodes are equal. If the proof
The micro accelerometer, a very important
mass is not in the middle of opposite electrodes, it
kind of micro inertial device, is used to measure the
means that certain acceleration is existed. The
acceleration of carriers. The velocity and disposition
accelerometer information can be got by detecting the
information could be achieved by integrating the
outputs of two opposite electrodes differentially.
relative accelerometer. Its structure includes a
cantilever beam with a fixed end and a cantilevered This accelerometer has the character of high
end on a proof mass[10]. The principle is shown in sensitivity, excellent temperature performance, low
Fig 2. noise, and simple structure and easy to form the high
Whether the proof mass moves or not in the y precision force-balance device. The sensitivity and
direction indicates if there is an accelerometer resolution of capacitive accelerometer, however, has
produced in the corresponding direction. The moving large dependency on the area of capacitive plate.
displacement and related accelerometer could be With the decreasing of accelerometer size, the
obtained with various testing methods. sensitivity and resolution of unit have a limited level
which holds back the further integration and
micromation. The principle of tunneling current
accelerometer is based on the tunneling effect to test
the movement displacement of
mass. Because of the sensitivity of tunneling current
accelerometer can up to10-7g, the strong 1 / f noise
of the accelerometer limits the application in the low
frequency range. There are other testing disposition
methods such as optical, thermal, electromagnetic,
pezo-resistive, and so on.
Fig.2Principle of MEMS Accelerometer

The kinds of micro machined accelerometer include b) Micro gyroscopes
piezo-resistive accelerometer, capacitive
accelerometer, A gyroscope is a device which measures the
rotating rate or whole angle displacement. Micro
machined gyroscopes typically use vibrating
structures because of the difficulty of
micromachining rotating parts with sufficient mass to
be useful. Its current structure consists of quartz and
silicon vibrating beam. The quality factors of quartz
material structure are high. The performance of
gyroscope is excellent and is early commercialized.
Vibrating gyroscopes include tuning forks, vibrating
beam, vibrating shells, and others. Fig 2 shows the
principle of tuning forks used generally [5,10].
The principle of vibrating gyroscope is based

Fig.3 Principle of MEMS Gyro on the Coriolis Effect. If there is rotational about
the X axis, the Coriolis force will occur in the Z axis
resonant accelerometer, thermal accelerometer and when the centre masses vibrate in the same direction
tunneling current accelerometer. The principle of along the Y axis. Coriolis force is given by:
piezo-resistive accelerometer is to gain the
accelerometer information by placing a piezo- F c 2 m v
resistive resistance in the cantilevered beam. The
main features of this accelerometer are simple in where v is the vibrating velocity of vibrator,
structure and easy in fabrication, but larger in
temperature drift. The temperature compensation is the rotational angular velocity,
circuit is used to operate normally. The principle of
capacitive accelerometer lies in a proof mass between
a pair of opposite electrodes. If no acceleration, the
369
Fc and M c are Coriolis force and Coriolis

force momentum separately.Other vibrating
gyroscope has the similarity with tuning forks. The
difference lies in the vibrating device. The electro-
static, electromagnetic and piezoelectric vibrating
devices are the currently used.
III. G3oM (GPS-GLONASS-GAGAN ON Fig.5 G3oM

MODULE)
The complete design supports Fast Time to First Fix
The block schematic of the G3oM Receiver is shown technology (<42 s) in real time without any almanac
in Figure 4. The receiver consists of multiple RF storage on board. The Fast Acquisition of Signals
Down-converter blocks with one each for GPS and from multiple GNSS constellations requires a
GLONASS constellations. The main functions of the multitude of correlators for signal search in frequency
Down-converters are to amplify, translate and digitize and time. Further, during Dynamics platforms
the signals from GPS and GLONASS bands to experience larger Doppler in satellite signals
intermediate frequency (IF). The outputs from the obligating a wider search in frequency domain.
Down-converter are fed to the respective correlators.
The enormous magnitude of signal processing can be
suitably performed by a dedicated hardware with
highly parallel correlator architecture. With
programmable devices such as FPGAs, it would be
difficult to meet the form factor and power
requirements. Sequential execution of instructions in
todays CPUs and DSPs would not meet the real-time
requirements of the system.
The module supports four GAGAN channels in L

band, which basically which helps in improved
navigation solution b correcting the ino, tropo errors
from the GPS measurements. Any point in time the
G3oM can supports 5 parallel navigation solutions;
GPS, GLONASS, GPS+GAGAN, GPS+GLONASS,
GPS+GLONASS+GAGAN and their respective raw
measurements to integration with INS[1].
Fig.4Functional Block Diagram of G3oM
The Digital Correlators de-spread the signal; work

out the correlation values and generate measurements
through a standard interface out to a processing unit.
The processing core receives measurements from the
respective Correlators and computes the navigation
solution and communicates with the host [7,8].
architecture. With programmable devices such as
The G3oM is a dual antenna, 28 channels GPS- FPGAs, it would be difficult to meet the form factor
GLONASS-GAGAN miniaturized receiver module and power requirements. Sequential execution of
designed and developed towards low and moderate instructions in todays CPUs and DSPs would not
dynamics including UVAs. As shown in the fig.4 it is meet the real-time requirements of the system. The
designed around the customized multi System on module supports four GAGAN channels in L band,
Chips for GPS RF front-end, GLOANSS RF-front which basically which helps in improved navigation
end, GPS digital base band and GLONASS digital solution by correcting the ino, tropo errors from the
base and in addition to a FPGA, Memory, Power GPS measurements. Any point in time the G3oM
Supply components in the form factor of 32mm x can supports 5 parallel navigation solutions; GPS,
32mm. GLONASS, GPS+GAGAN, GPS+GLONASS,
370
GPS+GLONASS+GAGAN and their respective raw user; the other from the receiver clock drift. The
measurements to integration with INS[1]. Doppler aided PLL can be modeled as a control
system, as shown in fig.6. In order to facilitate the
IV. ARCHITECTURE OF ULTRA TIGHTLY analysis of the phase errors induced by Doppler
GNSS-INS WITH AN AID TO RECEIVERS aiding error, the true Doppler, Fd(s), is removed from
SCALAR BASED TRACKING LOOP the block diagram of Doppler-sided PLL, resulting in
In this architecture the PLL of the GNSS receiver is figure NN, the block diagram of Doppler aiding error
aided with the Doppler shift calculated from the propagation. In this figure, phase error
navigation solution of the GNSS-INS system ( and r(s)= r(s)- Fd(s)/s 3.2
satellite information) as indicated in fig.6, the noise
bandwidth of the loop filter of the PLLs can be and Doppler aiding error
narrowed more than in a GNSS-only case. Since the Fd(s) = Fdc(s) - Fd(s) 3.3
loop noise bandwidth cannot be made arbitrarily
small and is limited by the navigation solution error are inputs and
of the GNSS-INS and the receivers oscillator errors,
(s)=(s)- Fd(s)/s 3.4
methods of choosing the noise bandwidth for the PLL
loop filter are investigated in order to improve signal is output, where r(s) is PLL input phase, (s) is
tracking, signal reacquisition and overall navigation output phase, Fdc(s) is the calculated Doppler aiding.
solution performance. Specifically, the bandwidth can The phase error induced by Doppler aiding error is
be selected according to the performance of the obtained from fig .7 as follow
integration system and the quality of the receivers
Total Phase Error in Doppler aided PLL
oscillator. By selecting the adaptive loop filter(ALF)
for Doppler aided PLL loop the ALF noise The total phase error (1-sigma) of the Doppler-aided
bandwidth can be adjusted according to the PLL can be expressed as
performance of the integration system, the quality of
PLL =( 2tPLL+ 2v+ Arx+ Asx+ GM )+ RR / 3 3.6
the receivers oscillator, and the satellites information.
With the ALF, both the navigation performance and Where tPLL , v, GM, and RR are the thermal
the PLL tracking ability of the UT GNSS-INS can be noise, oscillator vibration phase jitter, Doppler aiding
improved [4]. GM phase jitter, and Doppler aiding RR phase error
respectively and Arx and Asx are respectively the
receiver and satellite Allan phase jitters. The RR
phase error is considered as a dynamic stress error
(much like the dynamic stress error in a standard
tracking loop), so it appears outside the square root in
equation 3.6 [4,5]. In equation 3.6 the thermal noise,
oscillator Allan phase jitter oscillator vibration phase
jitter, and Doppler siding error-induced phase error
are considered to be independent. Furthermore, the
Doppler aiding GM phase jitter and Doppler aiding
It is noted in the fig. 6 that the tight integration box phase error are assumed to be independent to simplify
contains an extended Kalman Filter to integrate both the analysis of Doppler aiding error- induced phase
the GNSS error. It is noted that equation 3.6 is applicable to any
Doppler aided PLL. Furthermore, if the parameters
and INS data. The Doppler aiding rate for the for calculating the total phase error with equation 3.6
tracking loop is equal to the INS measurement rate are known, according to the PLL tracking threshold
i.e. the output rate of the Klamn filter for Doppler PLL 15 (degree) (Ward et al 2006), the PLL is
aiding calculation. The Doppler aiding value is considered to be locked.
calculated from as:
The minimum noise bandwidth of Doppler-aided
fd=-fT ( (V-vu )./c) - fT tu +d 3.1 PLL can be obtained with equation 3.6. Since PLL is
Where fd is in Hz, V is the satellite velocity vector, vu a nonlinear function of the noise bandwidth, as shown
is the user velocity vector, is the user-to-satellite in equation 3.6 there are no analytical solution for the
line of sight unit vector, fT is the transmitted signal noise bandwidth. Therefore, a search strategy is
frequency, tu is receiver clock drift rate and d is needed to obtain the minimum noise bandwidth
noise.It consists of two parts: one comes from the solution. If the noise bandwidth solution is obtained
relative motion of the satellite with respect to the in real time and applied to the PLL loop filter
371
simultaneously, an daptive loop filter with varying

noise bandwidth can be obtained[4,5].
V. TEST PERFORMANCE ANALYSIS

As indicated in the dynamic trials of the UTC in the
INS loop the performance of the GNSS receiver and
hybrid are compared against each other in the fig.8
and fig.9. It has been conceived in the results that the
performance of the EKF in the position and velocity
loop is quite stable and the error in the ECEF position
and velocity is < 12m and 0.01 m/s respectively. The
significant improvement is noticed in the hybrid
velocity under the condition of PLL loop of the
Receiver is assisted by the INS Doppler shift
experienced due the dynamics of the vehicle.
The final implementation Doppler shift loop in the
Scalar tracking loop based receiver will support as an
adaptive FLL assisted PLL Doppler input from INS.
The tracking loop filter of the receiver channel will
have wide BW as long as no valid Doppler input is
available and narrow BW as an when the INS assist
the loop with valid Doppler information( a,w ).
The analysis has been carried on the data received

when the GNSS PLL loop is closed to very low
Doppler shift under live antenna for the
improvement in the velocity of the receiver and fused
velocity. As shown in the fig.10 the receiver velocity
has improved to the accuracy of 0.004 m/s as long as
the PLL loop of the receiver is assisted in due course
Fig11. Simulated Dynamic Doppler Shift Aiding
of test run during no change in acceleration. Another
from INS to Receiver PLL Loop
test we have carried out using GNSS and INS
simulator for the Doppler aiding ( INS accuracy < VI. CONCLUSION
0.5m/s) during dynamics as shown the fig.11 in the
constant acceleration zone the GNSS receiver
The UTC scheme using MEMS and GNSS sensors
velocity accuracy is one order better i.e 0.006 m/s.
has been successfully implemented and studied to
improve the INS Hybrid velocity during constant
acceleration phase of the vehicle. The INS Doppler
aid to receivers FLL assisted PLL loop improved the
velocity accuracy one order higher than otherwise.
Further investigation on this scheme is in progress.
REFERENCES
[1] Oleg S Salychev, Inertial systems in Navigation and Geophysics,
Bauman MSTU Press MOSCOW, 1998
[2] George M.Siouris, An Engg. Approach to Optimal Control and
Estimation Theory, Air force Institute of Technology, Wright-
Patterson AFB,Ohio.
[3] G.Sateesh Reddy, Manjit Kumar Integrated INS & GPS
Navigation approach using Kalman Filter GNSS-2007,NERTU,
Osmania University, Hyderabad, October 25-26, 2007.
372
[4] Debo Sun, Ultra Tight GPS/Reduced IMU for land Navigation
UCGE Reports Number 20305, March 2010 ,University of
Calagry, Alberta,
[5] Kedog Wang,Young Li, Chirs Rizos, The Feasibility of MEMS
inertial sensors for deep integration of GPS and INS Natinal
Technical Meet U.S ION, Fort Worth, Texas,12-14 June 2011
,2889-2895
[6] Brown, Robert Grover, Hwang,Patrick Y.C, Introduction to
Random Signals and Applied Kalman Filtering.
[7] Kaplan,ElliottD,Ed., Understanding GPS Principles and
Applications,Artech House, Norwood,MA,1996.
[8] M.S Grewal,Lawrence R. Weill, Angus P. Andrews, Global
Positioning Systems, Inertial Navigation, and Integration, john
Wiely& Sons,ISBN 0-471-20071-9
[9] G.Sateesh Reddy, Manjit Kumar INS- GPS-GLONASS Navigation
fusion scheme for high dynamics guided projectilesCo-
ordinates,pg . 35-42 August, 2012.
[10] A. Soloview, S Gunawardena, F Graas, Deeply integrated
GPS/Low-cost IMU for low CNR signal processing. Concept
description and in-flight demonstration Navigation: Journal of
the Institute of Navigation, 2008, 55(1): 1-13.
373
Unique NAV data Design for Optimal TTFF in

IRNSS Receiver
Dr. Vyasaraj Guru Rao
Sr. Manager, GNSS Technology Accord Software & Systems Pvt. Ltd. Bangalore, India
vyas@accord-soft.com
Sashidharan MAA & Sumadhva B Pappu

Engineer, Accord Software & Systems Pvt Ltd. Bangalore
sashidharan.m@accord-soft.com&sumadhva.bp@accord-soft.com
Further, it generalizes the scheme from a GLOBAL

Abstract: India is planning to deploy an autonomous
regional satellite based navigation system to cover its perspective as well.
territorial footprint and surrounding areas. The
purpose of this system as any other existing GNSS is to
cater to the needs of both specific users (Precision II. DESIGN DETAIL OF THE PROPOSED NAV
Service (PS)) and civilian users (Standard Positioning DATA
Service (SPS)). The overall constellation consists of
seven satellites, of which three will be in geostationary This section explains the proposed NAV data in detail
and four in geosynchronous orbits. This system will be
to reduce Teph(worst case time required to collect
used for surveying, telecommunication, transport,
identifying disaster locations and public safety, along ephemeris parameters of a satellite) with an objective
with a host of other applications [1]. Time To First to improve TTFF taking the Three Subframe
Fix (TTFF) is an important parameter used to assess a Architecture (henceforth named as Three Subframe
receivers performance. As a part of this research, Normal(TSN)) as proposed in [2] as a reference.
attempt is made to understand this component in detail
- its measure in various types of receivers and To establish the theoretically achievable TTFF limit
associated limitations. Subsequently, an optimal NAV (from transmission perspective) in a single frequency
data transmission method is deduced for single/dual of operation for IRNSS an assumption made here is
frequency application. A hardware simulator developed that the UTC and Ionosphere terms are only
to generate this signal is explained in detail. To test the transmitted as a part of secondary NAV data as in
proposed signal, new receiver software is developed. GPS L1. With this input, and extending the previous
Finally, the test methodology and the results are derivation as in [2], this section deduces the optimal
elaborated. NAV data structure.
I. INTRODUCTION
The Indian Regional Navigation Satellite System

(IRNSS) is in the early stages of signal design [1].
This provides ample opportunities for learning from
the existing operational GNSS to be incorporated into
IRNSS. In addition, it serves as a platform to adopt
new system level improvements proposed by Figure 1: Three Subframe Architecture
academia and industry. The possible areas that may
be explored during the signal design stages are Considering the argument that the almanac is
sensitivity improvements, jamming margins, redundant for IRNSSfrom a system perspective, UTC
robustness towards spoofing, multipath related and Ionosphere parameters are transmitted as a part of
improvements, TTFF etc. to name a few. TTFF is an a subframe 3, staggered satellite wise in fixed
important receiver specification that serves as a slots.The details of the scheme are as follows: The
yardstick to compare receivers from different first two subframes transmit ephemeris and clock
manufacturers. Work till date has focused on parameters of satellites as in a TSN method.
minimizing the TTFF for IRNSS [5]. However, this Following this, the third subframe data of one of the
research extends the above work and deduces the most satellite transmits UTC and Ionosphere terms, which
optimal scheme to obtain the best TTFF in IRNSS. appear once in 18 s from the constellation, as shown
374
inFigure 2 One additional design consideration is the Ionosphere terms are collected in 18 s. In addition, the
change in the subframe slotting following UTC and PDOP issue highlighted above with 7 satellites
Ionosphere transmission. For example, for SV1 after disappears in a global scenario with an assumption of
T0 + 18 s, instead of transmitting subframe 2 for SV1 a minimum of 8 or more satellite available at any
commensurate with the other satellites, subframe 1 is point on the globe.Finally, the UTC and Ionosphere
transmitted. This ensures that the maximum time terms shall be straddled in multiple satellites to ensure
required to collect ephemeris (from the satellite) is users across the globe obtain it from at least one
restricted to 18 s. The merit of this scheme is satellite.
Tephreduces to 12 s with Ionosphere and UTC
parameters transmitted periodically every 18 s. The
proposed scheme of Three Subframe Fixed (TSF) is as
shown inFigure 2.
One limitation with this approach is for 6 s (for
example, T0 + 12 to T0 + 18,) ephemeris from six
satellites, as opposed to all seven, is available (Figure
2).For 6 s (till T0 + 30 s), the Position Dilution of
Precision (PDOP measure of satellite geometry [3]
will be relatively high as shown inFigure 3.
Figure 3:PDOP variation over Indian

subcontinent with all satellites and exclusion of
GEO-1.
To improve the TTFF of the restricted service, this
approach drops the secondary data completely on
both frequencies and employs the proposed method
(Figure 2) of NAV transmission. Optimized TTFF
shall be achieved extending the conceptson both
Figure 2:Proposed NAV data architecture with services/frequencies with necessary design
TSF transmission to enhance Tephin IRNSS. modifications as follows: The MT-3 message
The PDOP variation as a function of a single satellite inFigure 2, is replaced by MT (n, m) as shown
dropped (7 satellites in all, 3 geostationary and 4 inFigure 4, where n is the message types supported
geosynchronous) is analyzed over the Indian as shown in Table1 and m can vary from 1 to 14,
subcontinent. The maximum PDOP is observed with which signifies specific transmission slots in either
the exclusion of GEO-1 as shown in Figure 3. It is frequency. Slots 1 to 7 correspond to 1- 7 satellite id
very clear that even with 6 satellites the PDOP is in L5 and 8-14 to that of 1-7 of S1 respectively.
relatively good for most of the Indian subcontinent. Table1: Proposed message types with their
periodicities as a part of third subframe in TSN
With TSF NAV data streaming, the theoretically
architecture.
best possible TTFF is achieved for any GNSS
system and in addition, the requisite secondary Message Description Satellite Rate (s)
parameters are collected in minimal time. Type Id
Further, assuming global extension of IRNSS, the 00 Reserved -- ---
scheme is still valid w.r.t TTFF which is explained as 01 Almanac- In all (T0 +
follows: With M satellites in IRNSS global IONO satellites 18) * N+,
constellation, a single frequency receiver is assumed staggered where N
to have M parallel channels. This ensures a dedicated = 1, 3
channel for each satellite and obviates the need for 33599
visibility (thus almanac) computation and reduces the
signal search time. With this arrangement, Teph is 02 Almanac- In all (T0 +
effectively reduced to 12 s, and the UTC and UTC satellites 18) * N,
375
Message Description Satellite Rate (s)

The NAV data symbols are staggered and
Type Id
transmitted from the satellite. The staggering of a
staggered where N subframe is accomplished as shown in Figure 4b).
= 1, 3 The objective of this approach is to ensure that 1200
33599 symbols (2 subframe effective data) are transmitted
03 Iono-UTC One (T0 + in 6 s from either service/frequency. For example, at
satellite, 18) * N, T0, the SPS channels on either frequency are
staggered where N initialized with SF 1 and SF 2 first portion of their
= 2, 4 subframes respectively. The complement portions of
33600 these subframes are assigned to the restricted service
04 63 Any User satellites, (T0 + on either of their frequencies. For T 0 + 3 to T0 + 6 s,
Data staggered 18) * N, the above sequence is reversed across service and
(cyclic) where N frequencies.
= 2, 4 On either frequency of both services, the NAV data
33600 is transmitted as shown in Figure 4a) where the
+
- Time at completion of 3rd subframe carrier diversity of SPS and restricted services are
met individually.
With Figure 2as a reference, message slotting as
shown in The receiver operation in a single frequency mode is
Figure 5is adopted on either services. unaffected by this method of NAV data signal
transmission.
Figure 4: Proposed dual-frequency NAV data signal

design based on TSF.
Sequencing of textual messages similar to SPS
is adopted in restricted services as well.
The message slotting shown in
Figure 5is generic w.r.t a particular message
transmission. One possible approach is that the
UTC/IONO message be transmitted
alternatively on either frequency to ensure that
single frequency users obtain the ionosphere
corrections faster. Figure 5:Timing diagram illustrating the
transmission slots of various messages in either
Following the above design constraints, the frequencies of the SPS/RES service proposed NAV
architecture shown in Figure 4is designed. For more data design for dual frequency applications.
details refer [6]. The advantage with this design is that in real-time
For simplicity, the MT message (Figure 2) is (directly from satellites), the receivers cold start
indicated as SF 1 and SF 2 in Figure 4with SF1 / TTFF performance is equal to the HOT start
SF2 carrying the MT message corresponding to its performance (data stored in receivers memory) of
respective slot is as shown in GPS or the AGPS ephemeris assistance [4].
Figure 5. The OCM method enhances the primary
NAV data transmission and thus facilitates faster Test Methodology
TTFF in a receiver supporting diverse modes of
operation. The signal design methodology followed To validate the performance of the proposed
in OCM (Figure 4) is explained as follows: Optimal Civilian Mode (OCM) method [6], the
376
simulator and receiver hardware used in the [2]with and the ionosphere is dropped. This is under the
the modified versions of the software developed are assumption that the receivers are operating in dual
used. The software change in the simulator is frequency mode and thus, with measurements, will
according to the flowchart specified in Figure 6. be able to estimate directly the ionosphere effect.
With the simulator and the receiver software in
OCM configurations, the receiver data was logged in
GGVISION (a graphical tool) to profile the T eph
component. Three runs were conducted to establish
the results in different configurations as follows:
1. First, to profile SPS only mode of operation
2. Second, restricted direct (hybrid, only in
OCM)
The results of the proposed scheme are as shown in
Figure 8.
Figure 6: Signal generation methodology to derive

the proposed OCM signals.
Figure 8:Altitude plot illustrating the TTFF

performance with the OCMmethod of NAV data
signal design.
ACKNOWLEDGEMENTS
The first author would like to thank Accord

Software & Systems Pvt. Ltd for the partial financial
assistance towards his doctoral studies. The contents
expressed as part of the paper are the views of the
authors and are not binding on any organization.
Figure 7:High level receiver design to derive the
benefits of OCM NAV data signal design. REFERENCES
To derive the benefits of the proposed signal, the
Satellite Data Base Manager module is modified 1. Rao, V. G., G. Lachapelle, and S.B. Vijaykumar (2011)
"Analysis of IRNSS over Indian Sub-continent," in
w.r.t the processing shown in Figure 7. The Proceedings of the International Technical Meeting, , San
algorithm assumes that the contents are available Diego, The Institute of Navigation, pp. 13
from the correlators from all four paths (SPS/RES 2. Rao, V. G., G. Lachapelle, and S. B. Pappu (2012)
L5/S1). The processing on the individual channels is "Proposed NAV Data Signal Design for Optimal TTFF in a
as shown inFigure 7. To derive the benefits, in Single Frequency IRNSS Receiver," in Proceedings of the
GNSS-2012, , Nashville, TN, The Institute of Navigatio
accordance to the OCM methods of signal
3. Petovello, M. (2010) "GNSS Solutions: GLONASS inter
transmission, satellite wise data processing is frequency biases and ambiguity resolution," InsideGNSS
performed after 300 symbols are collected. 4. Hegarty, C.J, A.J Van Dierendonck, D. Bobyn, M.Tran, and
As shown inFigure 7, the second synch pattern T.Kim, J, Suppressing of Pulsed Interference through
search (across frequency/service) ensures optimal Blanking Proceedings of the IAIN World Congress, San
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processing and facilitates easy extraction of
5. Rao, V. G., G. Lachapelle, and M. Sashidharan (2011)
straddled data across frequencies/services to "Proposed LOS Signal Design for IRNSS to Reduce TTFF
generate the NAV data. in a Single Frequency Receiver," in Proceedings of the
results GNSS Signals 2011 Workshop, , Toulosse
6. Vyasaraj Guru Rao (2013) Proposed LOS Fast TTFF
The results presented only depict the Teph component Signal Design for IRNSS, Doctoral Thesis, Department of
377
Geomatics Engineering, Calgary, Albert.
378
Inter-Carrier Interference Reduction in

OFDM System
Velamala.suryanarayana, vadaravu jagan naveen
GMR Institute of Technology, Rajam, AP, India
AbstractThe implementation of OFDM system In an OFDM system the input bit stream is
suffers from the effect of phase noise generated by multiplexed into N symbol streams each with symbol
the local oscillator which disturbs the period T and each symbol is modulate with sub-
orthogonality among sub carriers and causes inter carriers.A serial to parallel converter groups the
carrier interference (ICI).The ICI may also suffers stream of input bits from the source encoder into
due to insufficient cyclic prefix at the transmitter groups of bits. Where M is the alphabet size
Self cancellation is the method is suggested for of digital modulation scheme
reducing the ICI. The frequency offsets between
the transmitter and receiver is measured in terms A typical discrete-time baseband OFDM transceiver
of carrier to interference ratio (CIR) and bit error system. A serial to parallel converter groups the
rate (BER). stream of input bits from the source encoder into
Index Terms Orthogonality, self cancellation, Inter groups of log2M bits. Where M is the alphabet size
Carrier Interference (ICI), Carrier to Interference Ratio of the digital modulation scheme employed on each
(CIR). sub-carrier a total of N such symbols Xm are
created. The N symbols are mapped to bins of an
1. INTRODUCTION IFFT. The IFFT bins corresponding to the
orthogonal sub-carriers in the OFDM system.
OFDM is emerging as the preferred modulation
scheme in the modern high data rates wireless Therefore OFDM symbol can be expressed as
communication systems [1, 2.]. OFDM is a special
case of MC (Multicarrier modulation).MC is the
concept of splitting a signal into a no. of signals,
modulating each new signal to several frequency
channels and combining the data received on the
multiple channels at the receiver. OFDM has been
used in many communication systems such as The Digital to Analog converter creates an analog
WLAN (wireless LAN), DVB (Digital video time- Domain signal which is transmitted through the
broadcasting), etc. However one of the major channel. At the receiver, the signal is converted back
problems in OFDM is its vulnerability to to a discrete N point sequence y(n), corresponding to
frequency offset which leads to loss of orhogonality each sub-carrier. The discrete signal is demodulated
resulting into ICI [3, 4]. For suppressing ICI there is using an N-point FFT operation at the receiver
self cancellation, (SC) [5, 6, and 7] techniques in
OFDM. Self- cancellation is a two stage technique The demodulated symbol stream is given by
that uses predefined weighting coefficients to reduce
ICI for OFDM systems [8,9, 10, and 11].
2. OFDM SYSTEM DESCRIPTION

Where w (m) corresponds to the FFT of the
samples of w(n), which is the additive white
Gaussian noise (AWGN) introduced in the channel.
The high speed data rates for OFDM are
accomplished by the simultaneous transmission of
data at a lower rate on each of the orthogonal sub-
carriers. Because of low data rate transmission
distortion in the received signal induced by
multipath delay in the channel is not as significant
as compared to single carrier high data rate systems.
Multipath distortion can also cause inter
Fig.1.Baseband OFDM transceiver symbol interference (ISI) where adjacent symbols
system overlap with each other. The ISI is prevented in
379
OFDM by the insertion of a cyclic prefix between all subcarrier indices
successive OFDM symbols. This cyclic prefix is
discarded at the receiver to cancel out ISI. The complex ICI Coefficient S (l-k) are plotted for
all subcarrier indices
3. ANALYSIS OF INTER-CARRIER
INTERFERENCE
The main disadvantage of OFDM is its susceptibility

to small differences in frequency at the transmitter and Fig.3.ICI coefficients for N=16 carriers.
receiver, normally referred to as frequency offset. This
frequency offset can be caused by Doppler shift due to
relative motion between the transmitter and receiver or 4. ICI SELF-CANCELLATION
by difference between the frequencies of the local SCHEME
oscillators at the transmitter and receiver.
The main idea is to modulate the input data symbol
The frequency offset is modeled as a multiplicative on to a group of sub-carriers with predefined
factor introduced in the channel The received signal is coefficients such that the generated ICI signals within
given by that group cancel each other hence the name self-
cancellation.
Where N is the total number of subcarriers, X(k)

is the transmitted symbol for the kth subcarrier ,nk is
the FFT of w(n),and S(l-k) are the complex
coefficients for the ICI components in the received
signal . The ICI components are the interfering signals
transmitted on sub-carriers other than the kth sub-carrier.
The complex coefficients are given by
To analyze the effect of ICI on the received signal.

We The complex ICI Coefficient S (l-k) are plotted for
380
381
REFERENCES
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art,Experimental Study and Comparison of OFDM
th
Transmission Techniques, 5 international OFDM-
Workshop in Hamburg, September 2000.
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Strength for OFDM Inside Buildings, IEEE TRANSACTIONS
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VOL.49,NO.11, NOVEMBER 2001.
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channels, IEEE TRANSACTIONS ON WIRELESS
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IEEE COMMUNICATIONS LETTERS,VOL.6,NO 8
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382
Successive Interference Cancelation Based

Detection Algorithm for MIMO-OFDM Systems
Panchumarthi V Bhargavi, K.V.S.N.Raju, Buddaraju Revathi
Department of ECE, SRKR Engineering College, Bhimavaram, AP, India
bhagi.ece5@gmail.com
Abstract When the data is transmitted at high bit cells that share the same frequency channel, leading
rates, over mobile radio channels, the channel impulse to distortion of the desired signal and also low system
response can extend over many symbol periods, which performance. Therefore, wireless system must be
lead to Inter-Symbol Interference (ISI). Orthogonal designed to mitigate fading and interference, to
Frequency Division Multiplexing (OFDM) is one of the
guarantee a reliable communication. A successful
promising technologies to mitigate the ISI. In an OFDM
signal the bandwidth is divided into many narrow sub- method to improve reliable communication over a
channels which are transmitted in parallel. Each sub- wireless link is to use multiple antennas and another
channel is typically chosen narrow enough to eliminate promising scheme is using orthogonal frequency
the effect of delay spread. Analysis of the performance division multiplexing (OFDM), which can effectively
of 2x2 MIMO-OFDM system with Successive mitigate inter symbol interference (ISI) induced by
Interference Cancellation (SIC) and 64-QAM wireless multipath fading channels..
modulation for higher data rate has been carried out In wireless communications, it is well known that the
assuming flat fading Rayleigh channel. Combining channel capacity can linearly increase with the
different linear detection techniques (ZF, MMSE) and
number of antennas (provided that the numbers of
nonlinear detection techniques (SIC, OSIC) improves
the quality of the received signal in high interference transmit and receive antennas are the same) [1], [2].
environment. 64-QAM constellation transfers more bits Thus, to increase the channel capacity, the transmitter
per symbol and the number of bits per symbol impacts and receiver can be equipped with multiple antennas,
the accuracy of OSIC. Using Concatenating ZF or and the resulting channel becomes a multiple-input
MMSE with OSIC equalization, better cancellation gain multiple-output (MIMO) channel.
can be achieved. In this paper, MMSE-OSIC has been The growing demand for multimedia services in
presented, whose performance is better in high wireless communications has developed methods to
interference environment when compared to the earlier increase system capacity and reliability. In multiple-
detection schemes like ZF, ZF-SIC, ZF-OSIC, MMSE,
inputmultiple-output(MIMO) systems, capacity
MMSE-SIC etc,.
increase is brought about by using the spatial
Index terms: Successive Interference Cancellation (SIC), multiplexing mode, which offers capacity
Orthogonal Frequency Division Multiplexing (OFDM), proportional to the number of parallel transmit
Multiple-Input Multiple-Output (MIMO), Quadrature streams that can be created (i.e., the minimum
Amplitude Modulation (QAM), Inter symbol number of transmit and receive antennas) [3][6].
interference (ISI). There are MIMO receivers for spatial multiplexing
schemes, such as the Maximum-Likelihood (ML)
I. INTRODUCTION receiver, the linear receiver, the Successive
Interference Cancellation (SIC) receiver, etc. [3]. The
Wireless communication is highly challenging due to ML receiver is an optimal receiver, but it is difficult
complex time varying propagation medium. If we to implement due to high complexity arising from
consider a wireless link with one transmitter and one exhaustive searches over all candidate vector
receiver the transmitted signal that is launched into symbols.
wireless environment arrives at the receiver along a On the other hand, linear receivers, such as zero-
number of diverse paths referred to as multipath. forcing (ZF) or minimum mean square error (MMSE)
These paths occur from scattering and rejection of receivers, have low decoding complexity, but
radiated energy from objects (buildings, hills, trees detection performance decreases in proportion to the
etc.) and each path a different time-varying delay, number of transmit antennas. Therefore, there has
angle of arrival and signal amplitude. As a been a study on a practical nonlinear receiver,
consequence the received signal can vary as a namely, the ordered SIC (OSIC) receiver, which
function of frequency, time and space. These successively decodes data streams through nulling
variations are referred as fading and cause detoriation and canceling [6].
of the system quality. Furthermore wireless channel Although the OSIC receiver requires higher
suffer from co-channel interference (CCI) from other complexity than the linear receiver, it outperforms
383
linear receivers, and its performance is improved due where r is the received vector of size NrX1, H is
to the selection diversity arising from canceling the the channel matrix of size NrXNt s is the
detected signal out of the received signal [3]. SIC transmitted vector of size NtX1, and n is the noise
algorithm relies on a sequential detection of the vector of size NrX1. Each noise element is typically
received block. At each step, one symbol is detected modeled as independent identically distributed white
before being subtracted from the received block. This Gaussian noise [5], [6] with variance Nt/(2*SNR) [3].
introduces successively freedom degrees which To prevent correlation due to spacing they are
enable to reduce the noise/interference influence for typically spaced at least _c/2 where _c is the
the next users to be detected and therefore increases wavelength of the carrier frequency [5]. The second
the reliability of the decision process. reason correlation can occur is due to lack of
multipath components. It is for this reason that rich
Wireless communication networks design has the multipath is desirable in MIMO system. The
limitation on spectrum resources is an important multipath effect can be interpreted by each receive
restriction for achieving high bit rate transmissions. antenna being in a different channel. For this reason
The use of M-ary Quadrature Amplitude Modulation the rank of MIMO channel is defined as the number
(M-QAM) is considered an attractive technique to of independent equations offered. It is important to
overcome this restriction due to its high spectral note that
efficiency, and it has been studied and proposed for Rank (H) min (N r, Nt) (2)
wireless systems by several authors [8], [9]. And the maximum number of streams that a MIMO
system can support is upper bounded by min (Nr, Nt).
II .SYSTEM MODEL Current MIMO system includes MISO and SIMO
system that uses MIMO technique to improve the
MIMO systems are composed of three main performance of wireless system can be divided into
elements, namely the transmitter (Tx), the channel two kinds. One is spatial multiplexing which provides
(H), and the receiver (Rx). In this paper, Nt is denoted a linear capacity gain in relation to the number of
as the number of antenna element at the transmitter transmitting antenna and the other is spatial diversity
and Nr is denoted as the number of elements at the schemes which can reduce the BER and improve the
receiver. Fig1depicts the Physical MIMO channel. reliability of the wireless link.
The channel with Nr outputs and Nt inputs is denoted
as Nr x Nt matrix. A. Spatial Multiplexing
The transmission of multiple data stream over more

than one antenna is called spatial multiplexing. In this
technology multiple antennas are used at both the
ends to increase the bit rate in wireless radio link
without additional power or bandwidth consumption.
It offers a linear increase in spectral efficiency with
the number of antennas. There are two types which
have to be taken into account. The first is the V-
BLAST (Vertical Bell Laboratories Layered Space-
Time) which transmit spatial un-coded data streams
without any consideration in equalizing the signal at
Fig.1. NR X NT MIMO system the receiver. In V-BLAST algorithm, instead of
decoding all transmitted signal at the same time, we
first decode the strongest signal, then subtract this
strongest signal from the received signal and proceed
to decode the strongest signal of the remaining
transmit signal and so on. Other algorithms include
1. Zero Forcing (ZF) algorithm.
2. Minimum Mean square error (MMSE) algorithm
3. Maximum Likelihood (ML) receiver algorithm.
Where each entry hi,j denotes the attenuation and The ML receiver algorithm can yield the best
phase shift(transfer function) between the jth performance. The second one is realized by space
transmitter and the ith receiver. The MIMO signal time codes. In contrast to V-BLAST space time codes
model is described as deliver orthogonal & thereby independent data
r = Hs + n (1) stream.
384
B. Diversity Schemes randomly varying complex number hj, i. As the

To improve the link reliability we are using diversity channel under consideration is a Rayleigh channel,
schemes. Spatial diversity improves the signal quality the real & imaginary parts of hj,i are Gaussian
and achieves higher signal to noise ratio at the distributed having mean hj,i =0 variance 2hj,i = 1/2.
receiver side. Two kinds of spatial diversities are 4. The channel experienced between each transmit to
considered, Transmitter diversity and Receiver the receive antenna is independent and randomly
diversity. There are two famous space time coding varying in time.
schemes. Space time block code (STBC) and Space 5. On the receive antenna, the noise n has the
time trellis code (STTC). Gaussian probability density function with
y (n) (1 / 2 2 )e ( x u )
2
/ 2n2
II.PRAPOSED DETECTION ALGORITHM (3)
FOR MIMO SYSTEMS
The co-channel interference is one of the major =0 and 2 = N0/2.
limitations in cellular telephone network. In the case 6. The channel (hj, i) is known at the receiver.
of cellular network such as 3G or beyond 3G (4G),
the co-channel interference is caused by the A. Suboptimal Matched Filtering:
frequency reuse. Our main idea is to reject the co- 1. Zero forcing (ZF) equalizer:
channel interference in MIMO-OFDM cellular Let us now try to understand the math for extracting
systems. To eliminate the inter symbol interference the two symbols which interfered with each other. In
(ISI) occurs in highly interference channel ZF and the first time slot, the received signal on the first
MMSE equalization techniques are used. receive antenna is,
x
22 MIMO channel y1 h1,1 x1 h1, 2 x 2 n1 h1,1 h1, 2 1 n1
x2
In a 22 MIMO channel, probable usage of the (4)
available 2 transmit antennas can be as follows: The received signal on the second antenna is,
1. Consider that we have a transmission sequence, for
x
example {x1, x2, x3. xn}
2. In normal transmission, we will be sending x1 in
y 2 h2,1 x1 h2, 2 x 2 n2 h2,1 h2, 2 1 n2
the first time slot, x2 in the second time slot, x3 and so
x2
on. (5)
3. However, as we now have 2 transmit antennas, we y1, y2 are the received symbol on the 1st and 2nd
may group the symbols into groups of two. In the antenna respectively,a
first time slot, we send x1 and x2 from the first and h1,1 is the channel from 1st transmit antenna to 1st
second antenna. In second time slot, we send x3 and receive antenna,
x4 from the first and second antenna, send x5 and x6 h1,2 is the channel from 2nd transmit antenna to 1st
in the third time slot and so on. receive antenna,
4. Notice that as we are grouping two symbols and h2,1 is the channel from 1st transmit antenna to 2nd
sending them in one time slot, we need only n/2 time receive antenna,
slots to complete the transmission data rate is h2,2 is the channel from 2nd transmit antenna to 2nd
doubled. receive antenna,
5. This forms the simple explanation of a probable x1, x2 are the transmitted symbols and
MIMO transmission scheme with 2 transmit antennas n1, n2 is the noise on 1st, 2nd receiver antennas.
and 2 receive antennas. We assume that the receiver knows h1,1, h1,2, h2,1 and
h2,2. The receiver also knows y1 and y2.The unknowns
Other Assumptions: are x1 and x2. We have two equations and two
unknowns. For convenience, the above equation can
1. The channel is flat fading In simple terms, it be represented in matrix notation as follows:
means that the multipath channel has only one tap.
So, the convolution operation reduces to a simple y1 h1,1 h1, 2 x1 n1
multiplication. y h h2, 2 x n
2. The channel experience by each transmit antenna is 2= 2,1 2 + 2 (6)
independent from the channel experienced by other Equivalently, y = Hx+n (7)
transmit antennas. To solve for x, we know that we need to find a matrix
3. For the ith transmit antenna to jth receive antenna, W which satisfies WH = I. The Zero Forcing (ZF)
each transmitted symbol gets multiplied by a linear detector for meeting this constraint is given by,
385
Square Error (MMSE) linear detector for meeting this

W = (HHH)-1 HH (8) constraint is given by
Using the Zero Forcing (ZF) equalization, the W=[HHH+NoI]-1HH (13)

receiver can obtain an estimate of the two transmitted
symbols, x^1 andx^2 Using Minimum Mean Square Error
(MMSE) equalization, the receiver can obtain an
x1 y1 estimate of the two transmitted symbols x1, x2, i.e.
x y
2 = (HHH)-1 HH 2 (9) x1 y1
2. Minimum Mean Square Error (MMSE) x y
Equalizer: 2 = (HHH+N0I)-1HH 2 (14)
Let us now try to understand the math for extracting
the two symbols which interfered with each other. In B. Optimal Matched Filtering with Successive
the first time slot, the received signal on the first Interference Cancellation:
receive antenna is,
In Successive Interference Cancellation (SIC), the
x receiver arbitrarily takes one of the estimated
y1 h1,1 x1 h1, 2 x 2 n1 h1,1 h1, 2 1 n1 symbols for example the symbol transmitted in the
x2 second spatial dimension x^2, and subtract its effect
(10) from the received symbol y1 and y2. Once the effect
of x^2 is removed, the new channel becomes a one
The received signal on the second antenna is, transmit antenna, 2 receive antenna case and can be
optimally equalized by Maximal Ratio Combining
(MRC).
x
y 2 h2,1 x1 h2, 2 x 2 n2 h2,1 h2, 2 1 n2 Using the Zero Forcing (ZF) equalization approach
x2 described above, the receiver can obtain an estimate
(11) of the two transmitted symbols x1, x2, i.e..
Where
y1, y2 are the received symbol on the 1st and 2nd
antenna respectively, x1 y1
h1,1 is the channel from 1st transmit antenna to 1st x y
receive antenna,
2 = (HHH)-1HH 2 (15)
h1,2 is the channel from 2nd transmit antenna to 1st
Take one of the estimated symbols (for example 2 )
x
receive antenna,
and subtract its effect from the received vector y1
h2,1 is the channel from 1st transmit antenna to 2nd
and y2, i.e.
receive antenna,
h2,2 is the channel from 2nd transmit antenna to 2nd
receive antenna, r1 y1 h1, 2 x 2 h1,1 x1 n1
x1, x2 are the transmitted symbols and r y h x h x n
n1, n2 is the noise on 1st, 2nd receiver antennas. 2= 2 2, 2 2
= 2,1 1 2
(16)
We assume that the receiver knows h1,1, h1,2, h2,1
and h2,2. The receiver also knows y1 and y2.The Expressing in matrix notation,
unknowns are x1 and x2. We have two equations and
two unknowns. For convenience, the above equation
r1 h1,1 n1
can be represented in matrix notation as follows:
r h n
2= 2,1 x1 + 2 (17)
y1 h1,1 h1, 2 x1 n1
y h h2, 2 x n r = hx1 +n
2= 2,1 2 + 2 (12)
The above equation is same as equation obtained for
Equivalently, y = Hx+n receive diversity case. The equalized symbol is,
To solve for x, we know that we need to find a matrix W1=hkr/hkh (18)

W which satisfies WH = I. The Minimum Mean
386
C. SIC with Optimal Ordering: 64-QAM. Channel estimation as well as

synchronization is assumed to be ideal.
When SIC is used the system performance is affected
by the order S in which the layers are detected. That
follows from the previous considerations. If, for
example, in a 22 MIMO-OFDM system layer 1 will
be detected first, the post-detected SNR will differ
from that one, achieved by detecting the layer 2 first.
So the layer that leads to the highest post-detection
SNR should be detected first for maximizing the
systems performance. The proof can be found in [10].
Performing successive interference cancellation
(SIC) with optimal ordering ensures that the
reliability of the symbol which is decoded first is
guaranteed to have a lower error probability than the
other symbol. This results in lowering the chances of
incorrect decisions resulting in erroneous interference
cancellation. Hence gives lower error rate than simple
successive interference cancellation. Fig. 2. 2X2 MIMO-OFDM System, ZF equalizer,
However, we can have more intelligence in choosing BER vs Eb/No for BPSK modulation with and
without SIC
whether we should subtract the effect of
x1 first or
x2 first. To make that decision, let us find out the Fig. 2 shows the simulation result for a 2x2 MIMO-
OFDM system with a ZF-equalizer for the case of
transmit symbol (after multiplication with the pure equalization and the combination with
channel) which came at higher power at the receiver. successive interference cancellation. SIC is simulated
with random and optimal order for BPSK
The received power at both the antennas Modulation.
corresponding to the transmitted symbol x1 is,
2 2
Px1 h1,1 h2,1
(19)
The received power at both the antennas

corresponding to the transmitted symbol x2 is,
2 2
Px 2 h1, 2 h2, 2
(20)
If Px1 Px2 then the receiver decides to remove
the effect of
x1 from the received vector y and y
1 2,
x
and then re-estimate 2 . Else if Px 1 Px2 the
x
receiver decides to subtract effect of 2 from the
received vector y1 and y2, and then re-estimate
x1 .
x x
Once the effect of either 1 or 2 is removed, the
new channel becomes a one transmit antenna, 2
receive antenna case and the symbol on the other
spatial dimension can be optimally equalized Fig. 3. 2X2 MIMO-OFDM System, ZF equalizer,
by Maximal Ratio Combining (MRC). BER vs Eb/No for 64-QAM modulation with and
without SIC
II.SIMULATION RESULTS
Fig 3 shows the simulation result for a 2x2 MIMO-
In all simulation results the 2x2 MIMO-OFDM OFDM system with a ZF-equalizer for the case of
system in Rayleigh fading channel is taken and the pure equalization and the combination with
two modulation schemes are applied i.e. BPSK and Successive Interference Cancellation. SIC is
387
simulated with random and optimal order for 64- MIMO-OFDM systems are gaining much more
QAM Modulation to improve the transmission rate. attention in wireless communication research due to
their potential to increase considerable capacity in
mobile cellular communication.
MMSE-SIC-SORTING detector for 64-QAM MIMO
system performance is better than ZF, ZF-SIC, ZF-
SIC-SORTING, MMSE and MMSE-SIC. The MIMO
system performance is improved especially by using
higher order modulation scheme such as QAM than
BPSK modulation.
IV.FUTURE SCOPE
MMSE-SIC-SORTING detection performance may

be improved with the aid of the recent advances in
Artificial Intelligence (AI). Using the AI, a range of
other problem solving methods Neuro-travelling
Fig. 4. 2X2 MIMO-OFDM System, MMSE particle swarm optimizer has also emerged which
equalizer, BER vs Eb/No for BPSK modulation with improves the MIMO performance by employing
and without SIC. feedback.
Fig 4 shows the simulation result for a 2x2 MIMO- REFERENCES
OFDM system with a MMSE-equalizer for the case
of pure equalization and the combination with [1] I. E. Telatar, Capacity of multiple-antenna Gaussian
Successive Interference Cancellation. SIC is channels, Eur. Trans. Telecommun., vol. 10, no. 6, pp. 585
simulated with random and optimal order for BPSK 595, Nov/Dec. 1999.
Modulation. [2] G. J. Foschini and M. J. Gans, On limits of wireless
communications in a fading environment when using
multiple antennas, Wirel. Pers. Commun., vol. 6, no. 3, pp.
311335, Mar. 1998.
[3] A. Paulraj, R. Nabar, and D. Gore, Introduction to Space
Time Wireless Communications, 1st ed. Cambridge, U.K.:
Cambridge Univ. Press, 2003
[4] F. R. Farrokhi, G. J. Foschini, A. Lozano, and R. A.
Valenzuela, Link-optimal spacetime processing with
multiple transmit and receive antennas, IEEE Commun.
Lett., vol. 5, no. 3, pp. 8587, Mar. 2001.
[5] P. W. Wolniansky, G. J. Foschini, G. D. Golden, and R. A.
Valenzuela, V-BLAST: An architecture for realizing very
high data rates over the rich-scattering wireless channel, in
Proc. ISSSE, Pisa, Italy, Sep. 29, 1998, pp. 295300. Invited
paper.
[6] R. W. Heath, Jr., M. Airy, and A. J. Paulraj, Multiuser
diversity for MIMO wireless systems with linear receivers,
in Proc. 35th Asilomar Conf. Signals Syst. Comput., Nov.
Fig. 5. 2X2 MIMO-OFDM System, MMSE 2001, vol. 2, pp. 11941199.
equalizer, BER vs Eb/No for QAM modulation with [7] W.T. Webb and L. Hanzo, Modern Quadrature Amplitude
and without SIC Modulation Principles and Applications for Fixed and
Fig 5 shows the simulation result for a 2x2 MIMO- Wireless Channels. New York: IEEE Press, 1994. 275
OFDM system with a MMSE-equalizer for the case [8] A. Goldsmith and S. G. Chua, Variable-rate variable power
of pure equalization and the combination with M-QAM for fading channels, IEEE Trans. Commun., vol.
45, no. 10, pp. 12181230, Oct. 1997.
Successive Interference Cancellation. SIC is
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single-carrier frequency-domain MMSE equalization, in
QAM Modulation. For MMSE-equalization the gain Proc London Comm. Symposium, 2002, pp. 237240.
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is additionally used for calculating the equalizer. channels, AT&T Technical Journal, vol. 64, no. 6, pp. 1147
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A.Valenzuela, V-blast: An architecture for realizing very
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Proc of ISSSE-98, Pisa, Italy, Sept. 1998
388
Bit Error Rate Performance of MIMO- CDMA

System for Wireless Communications by Using
Space Time Block Codes
T.Nageswara Rao, M.Sandeep
Dept. Of ECE ANURAG Engineering College, KODAD Nalgonda, Andhra Pradesh, India nagu.iitkgp17@gmail.com
Abstract.In this paper we propose a new method II.CONVENTIONAL MIMO-CDMA

for Designed MIMO-CDMA with STBC (Space-
Time Block Code) to Increase the high data rate Third generation cellular systems employ direct
and high capacity in wireless communication sequence code-division multiple-access (DSCDMA) as
system. CDMA (Code Division Multiple access) the channel access scheme. In conventional MIMO-
CDMA systems, unique codes are assigned per user.
plays an important role in modern wireless
In conventional MIMO- CDMA, each user use a
communication systems. MIMO refers to links with
different spreading code for each transmitting
multiple antennas at the transmitter and receiver antenna. In conventional MIMO-CDMA system
side. CDMA with MIMO is a very promising the spreading waveforms are fixed.
technique beyond 3G and 4G wireless
communications. The BER performance of MIMO- Wmi =Cmi (t-nT) ---------------------------- (1)
CDMA system depends on its spreading strategy Where Wmi is the spreading waveform for i th data
stream of user m and Cmi is the orthogonal spreading
Index TermsCDMA, STBC, MIMO, BER sequence. Each antenna uses one spreading waveform,
for Nt =4 antennas system 4 spreading waveforms
I.INTRODUCTION are required per user. At the receiver side, the
received data is given bank of matched filters.
Based on the matched filters decision, the transmitted
CDMA is an underlying technology used in 3G and 2G. data is estimated
Wireless systems with multiple transmit and receive The block diagram of a MIMO-CDMA transmitter
antennas systems are an important part of discussions on and receiver as shown in Figure(1)
future wireless communication systems. In rich scattering
environments, these systems offer large capacity
gains[1].
In MIMO-CDMA to improve the bit error rate

performance parity bit selected spreading sequences
are introduced
initially [2], in this linear block coding technique is used.
In that Parity bits are used to select the spreading
sequence from a set of orthogonal spreading
sequences. The information bits are then spread by this
spreading sequence. At the receiver, the spreading code
employed is determined from the outputs of the filters
that are matched to each spreading sequence. The
information bits are then determined from the output
of the specific matched filter. Error correction is then
performed by assuming that the parity bits are correct. Fig. 1. Block diagram of MIMO-
So compared to conventional system BER is improved, CDMA system
but some sort of self interference occurred due to
summing of all transmitted signals. III.MIMO-CDMA WITH T-DESIGNS
In this paper, we design the spreading code MIMO-CDMA with Parity bit selected spreading is
permutations based on Space-Time Block Codes discussed in [2]. In parity bit selected spreading
(STBC).Compared to the results presented in [2], the sequence technique, based on the calculated parity bits
STBC-based design can improve the bit error rate the spreading waveform is Selected for each antenna. In
(BER) performance over the flat fading channel this approach, for each user 2(n- k) spreading sequences
without increasing the system complexity[3] will be assigned. Where n-k is the number of parity bits
389
of the code. The system MIMO-CDMA with t-design The above expression finds the squared Euclidean
permutation spreading is similar to parity bit selected distance between the received vector and all possible
spreading. remaining vectors and the vector which gives the
Now M is a set of all possible message vectors, contains smallest distance that is the transmitted message.
16 elements. M is divided into 8 possible cosets. The
MIMO- CDMA system is designed with Nt=Nr=4
antennas, and N=8. On signaling interval, the message to V.RESULT ANALASIS
be transmitted is {m1, m2,mNt}.The cosets are
M1={0000 1111},M2={00011110}, M3={0010 MIMO-CDMA system with 4 transmit and 4 receive
1101}, M4={0011 1100} M5={01001011}, antennas are considered in this proposed technique
M6={0101 1010}, M7={0110 1001}, and existing techniques .Simulation results for BER
M8={01111000}. And the spreading sequences are (bit error rate) performances are presented for 4x4
assigned to the antennas as shown in TABLE I. for antenna system. From Fig.2 we see that MIMO-CDMA
example, if the message comes from M1, then spreading system employing STBC permutation spreading giving
waveforms for four antennas the better BER performance as compared to t-designs
are c1, c3, c5 and c7correspondingly that is shown in the method, and conventional spreading also taken as
first row of the TABLE I.The received vector at the reference. And here the proposed technique
output of matched filters of the system is outperforms the conventional and t-designs. From
TABLE I, we can see that in t-design permutation
IV. SPACE-TIME BLOCK CODE APPROACH spreading table, there is no code symmetry between
FOR BIT ERROR PROBABILITY ANALYSIS different cosets, that leads to some degrees of freedom
are lost, when MLD (maximum likelihood detection) is
The MIMO-CDMA system employing STBC performed at the receiver. So that cause to slight
permutation spreading [4] is shown in Fig 1. The increment of BER. But the system with STBC spreading,
input bit stream is converted into Nt parallel data the permutation spreading table provides code symmetry
streams. And these bits are between different cosets, so that gives better BER
given to spreading sequence selector that selects Nt performance.
spreading waveforms, then these data bits are modulated
with BPSK modulation, then each bit is spread by Fig. 2. BER performance for conventional-designs and
using spreading STBC for 4x1 and 4x4 antenna system.
waveform. On a given signaling interval Nt spreading
waveforms will be selected from N orthogonal spreading
waveforms. And at the receiver side, the output of
each
antenna is given to a bank of matched filters, and
according to matched filter decision, the transmitted data
is estimated.
The spreading assignment is done based on 8x8
STBC
matrixes. And the matrix [6] is given as:
Here the columns of 1,5,8, and 6 of above matrix shown

are assigned to columns 1,2,3,4 of TABLE II.
The received vector r = ub+n ,
Where r = [ 11 12 18 21 41 48 , ,..., ,..., ,..., ,...r r r r r , r
]
and
n= [ 11 12 18 21 41 48 n ,n ,..., n ,..., n ,..., n ,..., n ]
Where ub is the received data vector that depends on
the transmitted data vector without noise. and b =
[b1,b2,b3,b4] in case of 4x4 antenna system.
For example if m = [0, 0, 0, 1] then b= [-1,-1,-1, 1] and
Maximum likely hood detection is performed to detect

the data. And squared Euclidean distance is found
between the received vector and all other possible
received vectors and the value which is minimum will
be taken as the transmitted message.
390
CONCLUSIONS
Code division multiple access technology, has been

widely used in mobile communications, because of its
high spectrumefficiency, low transmit power. The T-
designs Permutation Spreading method does not provide
code symmetry so some sorts of degrees of freedom are
lost in the squared Euclideandistance between different
messages. The lack of code symmetry increases the
BER. The STBC technique create dependency
between the data streams and provide
performance improvement compared to conventional
spreading. And code symmetry also provides by using
space-time block codes. Spacetime block coding is a
simple and method for transmission using multiple
transmit antennas in wireless communication systems.
REFERENCES
[1] C. DAmours, Parity bit selected spreading sequences: a block

coding approach to spread spectrum, IEEE Commun. Lett., vol. 9,
pp. 16-18, Jan. 2005.
[2] C. DAmours and J.-Y. Chouinard, Parity bit selected and
permutation spreading for CDMA/MIMO systems, in Proc. IEEE
Veh. Tech. Conf., pp. 1475-1479, Apr. 2007.
[3] V. Tarokh, H. Jafarkhani, and A. R. Calderbank, Space- time
block codes from orthogonal designs, IEEE Trans. Inf. Theory,
vol. 45, pp. 1456- 1467, July 1999.
[4] J. G. Proakis, Digital Communications, 4th. ed. New York:
McGraw-Hill, 2001.
391
Haar Wavelet-Based BPSK OFDMMIMO

System Null Channels with Reduced PAPR
RatnaKumari Bitra, P.Subba Rao
Dept. of ECE
SRKR Engineering College
ratna.bitra.ece@gmail.com, patsrao@rediffmail.com
Abstract In Multi-Input Multi-Output (MIMO) based

passed through a nonlinear power amplifier at the
systems, with the increasing demand for data rate and transmitter end. Clipping degrades the bit-error-rate
reliability in Wireless communications and devices, (BER) performance and causes spectral spreading. One
several issues become very important like bandwidth way to solve this problem is to force the amplifier to
efficiency, quality of service and radio coverage. The work
subcarriers already occupied by the primary users
cannot be used by the secondary users. This leads to in its linear region [6]. In high speed digital wireless
possibly non-contiguous positions of the available applications, however, the inter-symbol interference
subcarriers for the secondary users. The conventional (ISI) channel may have spectral nulls, which may
pilot design methods are no longer effective for such degrade the performance of the existing OFDM systems
systems. This paper proposes a novel BPSK OFDM because the Fourier transformation of the ISI channels
MIMO system based on Haar wavelet transformation. needs to be inverted for each subcarrier at the OFDM
The Haar wavelet transformation operates receiver [7]. Hence, PAPR and spectral null channels
decomposition over the data symbol sequence after need to be properly handled in implementation of
binary-to- complex mapping shows that half of the data OFDM systems [8], which are crucial for the
symbols are zeros and the rest are either2 or 2. performance of areal system.
Then, we have the PAPR reduced by10 log10 2 3 dB
at most, compared with the conventional OFDM - The Wavelet-OFDM system was widely studied in [9-
MIMO system. We also proposed novel decoding
10]. However, Wavelet-OFDM just substitutes the
algorithm for the proposed OFDM - MIMO system to
DFT and IDFT with DWT and IDWT, respectively.
show robustness to spectral null channels, and
The Wavelet-OFDM was
derive the bit error rate (BER) performance in theory
from modulation. Finally, With the MIMO concept we studied to show robustness to Doppler [9]. In this
improve the bit rate and BER of the overall system paper, we propose a novel Haar wavelet-based
.Simulation results show that the pilot index sequences BPSKOFDM system. Since the data symbol produced
obtained by the proposed method exhibit significantly by BPSK modulator is either 1 or1, the Haar wavelet
better performance than those obtained by existing transformation operates decomposition over the data
pilot design methods. symbol sequence after binary-to- complex mapping
shows that half of the information symbols
are zeros and the rest are either 2 or 2.
I. Then, the proposed BPSK-OFDM has the PAPR
INTRODUCTION reduced by 10 log10 2 3 dB at most, compared
with the conventional OFDM system. We also
Wireless digital communication is rapidly expanding, propose a novel decoding algorithm for our proposed
resulting in a demand for wireless systems that are OFDM system to show robustness to spectral null
reliable and have a high spectral efficiency. channels. Simulations and theory analysis are
Orthogonal frequency division multiplexing (OFDM) conducted to illustrate our analysis correctness.
has been considered as a promising candidate to
achieve high rate data transmission in a mobile Hence exploiting the spatial diversity of the
environment. Recently, OFDM has become the wireless environment by using Multiple Input Multiple
technique of choice for several popular Output (MIMO) expands the system capacity (higher
broadband applications, such as asymmetric digital data throughput) and it combat the faded channel
subscriber line(ADSL) modems, digital audio effectively. It has been shown in the literature [10] that
broadcasting (DAB) [1], digital video broadcasting the capacity of a single transmits antennas system
(DVB)[2,3] and wireless local area networks increases logarithmically with the number of receive
(WLAN) systems (IEEE 802.11a [4], IEEE 802.11g[5]). antennas. Meanwhile, the capacity of a MIMO
system with equal number of transmitters and
However, due to the large number of subcarriers receivers increases linearly. Implementing MIMO
used, OFDM systems have a large dynamic signal range leads to improvement in the data transmission
with a very high peak-to-average power ratio (PAPR). reliability (very low BER). MIMO is the most
As a result, the OFDM signal will be clipped when effective technique to accomplish reliable
392
communication through the wireless channel where

the receiver is provided with independently faded
replicas of the transmitted signal. These advantages
are achievable without any increase of the transmitted the Haar wavelet transformation. The Haar wavelet
power or expansion of the bandwidth. Several transformation unit in Fig. 1 operates decomposition
independent signals are transmitted from several over the input data symbol sequence after binary-to-
antennas.In this paper, we assume no frequency offset complex mapping. We denote the input data symbol
and channel impulse response (CIR) is given in our as x(n) = [x(0), x(1), . . . , x(N 1)]T , and the
OFDM MIMO system like our previous work [11 decomposition result is x_(n) =_x_(0), x_(1), . . . ,
13]. The impact of ICI on OFDM system performance x_(N 1)_T . Since the input N by 1 vector sequence
is widely discussed in [14, 15]. Since the ICI exists in x(n) is produced by BPSK baseband modulator,
the real application system, we will jointly consider each component of x(n) is either 1 or 1. The detail
different digital modulations and wavelets to analysis process of Haar wavelet decomposition over x(n) =
the ICI impact and theoretical BER performance in our [x(0), x(1), . . . , x(N 1)].Si
next research work. The CIR is usually estimated
using feedback information, and different estimators
(i.e., ML, MMSE) are proposed for CIR estimation.
The rest of the paper is organized as follows. In Sect.
2,(a) we present the Haar wavelet-based BPSK
OFDM system,(b) we study the PAPR performance
of the proposed OFDM system and (c) we present
a novel decoding algorithm and derive the BER
performance of our proposed OFDM MIMO system.
In Sect. 3, the theoretical and simulation results are nce the Haar wavelet transformation is added in
presented to illustrate our theory analysis correctness. conventional OFDM system, the computational
complexity will increase. However, the Haar wavelet
transformation is simply to operate at the transmitter.
II. SYSTEM DESIGN MODEL
(A) Haar Wavelet-Based BPSKOFDM System
In this section, we propose a novel Haar-wavelet based

BPSKOFDM system. It is formulated as follows, which
is the goal of this section. The oldest and most basic
wavelet system is named Haar wavelet that is a group of
square waves with magnitude of 1 in the interval[0,1) .
In other words, the Haar functions are defined on the
interval [0, 1) as Fig. 1 TheHaar wavelet-based BPSK OFDM system.
(B) PAPR Performance
393
antenna r at time t. nr(t) is the Additive White Gaussian

Noise. For mobile communications, the channel tap
coefficients are random variables. In case the wireless
channel varies very slowly, the tap coefficients remain
constant for each frame of data. For Rayleigh fading
channels, the channel tap coefficients are
modelled as complex Gaussian random variables which
have zero mean. The different channel taps are assumed
to be independent. The average channel gains for
different paths are determined from the power Delay
profile of the wireless channel. In this work we assume
that the channel tap powers decays exponentially. Hence
we use the exponential power delay profile. If the
MIMO-OFDM system has Nc subcarriers and the fading
coefficients are spatially
uncorrelated and that the fading coefficients remain

constant during one OFDM symbol. Then the
N is the number of subcarriers in OFDM system. transmitted signal over M antennas can be represented
The peak power of the Haar wavelet-based BPSK by a matrix XOFDM with dimensions NcxM. A symbol
OFDM system is reduced by half, compared with the transmitted at subcarrier n on transmit antenna is xi (n).
conventional OFDM system. So Haar wavelet-based
BPSKOFDM systems peak- average power ratio
(PAPR) is reduced by 10 log10 2 =3dBatmost,
compared with the conventional OFDM system.
Hence, the proposed Haar wavelet-based BPSKOFDM
system is able to over come the drawback of
conventional OFDM system, with lower peak-average
power ratio Next, we present our empirical study
characterizing the PAPR benefits via cumulative
distribution functions, assuming N
=1,024subcarriers.
(C) MIMO model
The MIMO-based CR system under consideration is

shown in Fig.1. The pilots are designed according to the
result of the spectrum sensing. After subcarrier
assignment where the subcarriers occupied by the
primary users are deactivated, pilot symbols are inserted
and the data are modulated on the remaining activated
subcarriers. We employ the MIMO concept in our
simulation platform because it has been proven that
MIMO can achieve a major breakthrough in providing
reliable wireless communication links. This reliability is
in the context of the channel estimation in our case. With
the MIMO concept we improve the bit rate and BER of
the overall system. MIMO is capable of this
improvement, because of the property of multiple
transmission multiple reception. This property is a form
of spatial diversity. This diversity is the most effective
technique to accomplish reliable communication over
the wireless channel and combating with fading,
because it provides the receiver with multiple copies of
the transmitted signal. Those multiple copies are
independently faded. If at least one copy of the
transmitted signal is received correctly, we will have
the transmitte.Where si(t) is the transmitted signal from
antenna i at time t, h(k) i,r(t) is the channel coefficient
for the kth path from transmit antenna i to receive
394
395
CONCLUSIONS
In this paper, we proposed a novel Haar wavelet-based

BPSKOFDM - MIMO system. Since the Haar wavelet
transformation is used in our proposed OFDM system,
half of the information symbols are zeros and the rest are
either2 or 2 in each OFDM symbol. The simulation
results and theory analysis illustrate the proposed system
has two advantages compared with conventional OFDM
system: (1) reduces the PAPR by 3dB at most (2) shows
robustness to spectral null channels, improving BER
performance 3dB at most. Analysis also shows that our
proposed OFDM system does not increase too much
computational complexity at the transmitter. It has been
shown that the MIMO concept is a good means to do
that But then another challenge faces us. The pilot
patterns must be redesigned to be suitable for
implementation in MIMO OFDM. We enhance the
overall system performance by implementing the MIMO
concept. The channel estimation in MIMO becomes a
third challenge. We solve this challenge by making the
designed pilot patterns suitable for MIMO.
396
Power Control Factor Selection in Uplink

OFDMA Cellular Networks
Suman Kumar and K. Giridhar
Dept. of Electrical Engineering
ndian Institute of Technology Madras
Chennai, 600036, India
Email: {suman, giri}@tenet.res.in
AbstractUplink power control plays a key role on and closed loop power control has been studied in
the performance of uplink cellular network. In this [2], [3]. It has been shown in [4] that the
work, the power control factor1 ( [0, 1]) is evaluated fractional path loss compensation factor with
based on three parameters namely: average transmit closed loop power control can greatly improve the
power, coverage probability and average rate. In other system performance. The impact of fractional
words, we evaluate power control factor such that
average transmit power should be low, coverage
power control on the SINR and interference
probability of cell-edge users should be high and also distribution has been studied in [5] and also a sub-
average rate over all the uplink users should be high. optimal configuration is proposed for the fractional
We show through numerical studies that the power power control.
control factor should be close to 0.5 in order to achieve
an acceptable trade-off between these three It has been shown in [6] that fractional path loss
parameters. compensation is advantageous than the full path loss
compensation in terms of cell-edge capacity and
Power control is an important consideration for also battery life time. A modified fractional power
the uplink cellular networks. It has two modes of control utilizing the path loss difference between
operation: closed loop and open loop power control serving cell and strongest interfering cell is
[1]. In closed loop power control, the base station proposed in [7] to improve the cell-edge bitrate and
(BS) compare the received Signal-to-noise- plus- overall spectral efficiency. Recently, authors of [8],
Interference-ratio (SINR) to the desired target [9] proposed an analytical approach to this
SINR. If the received SINR is lesser than the problem and they have provided the insight for
desired target SINR a transmit power control choosing the fractional power control. However,
command is transmitted to the MS to increases although insightful design guidelines are provided,
the transmit power. Otherwise, transmit power they have not provided the specific parameter value
control command is transmitted to decrease the which is at the end, the interest of cellular operator.
transmit power. On the other hand in open loop
power control power there is no feedback path. In
this work, an open loop power control is
considered.
Power control factor ( [0, 1]) controls the power

transmitted by mobile stations (MS). A lower
power control factor
allows the cell-centre users (users close to BS) to
transmit higher power which result higher rate but it
also provides higher interference power for the cell-
edge users (users close to BS). On the other
hand, a higher power control factor allows the
cell-centre users to transmit lower power which
result lower rate but it provides lower interferer
power for the cell-edge users. Therefore, how
should a cellular network operator should choose
Fig. 1: Hexagonal structure of 2-tier macrocell.
the power control factor. This is the question we Interference for 0th cell in FR1 system is contributed
attempt to answer in this work. form all the neighbouring 18 cells, while in a FR3
system it is contributed only from the shaded cells.
The difference between the performance of pure
open loop power control and combined open loop In this work, we evaluate the power control factor
397
based on the three parameters namely: average by

transmit power, coverage probability and average
rate. We show that the power control factor should
be close to 0.5. Since at power control factor =
0.5, average transmit power is low, coverage
probability of cell-edge users are moderate and also
the average rate is moderate.
Coverage probability is the probability that the
II. S YSTEM MODEL measured SINR at the BS of the MS is greater
than the target SINR T . It is defined as
We consider the uplink macrocell network having
hexagonal shape with inter cell site distance 2R
as shown in Fig. 1. The user is assumed to be
uniformly distributed. A path loss model l(x) = The average rate is calculated based upon the
kxk is considered, where 2 is path loss Shannon capacity limit, R = E[ln(1 + (r))].
exponent. Similar to [9], it is assumed that all the We start the discussion with the average
MS uses distance-proportion fractional power normalized uplink transmit power by the MS. Fig. 2
control factor of the form d , where [0, 1] shows the variation in the average normalized
denotes the power control factor. The parameter power transmitted by the MS versus power control
controls the transmit power. = 1 corresponds to factor using the expression given in (2). In can
an equal received power from all the MS, and at the be seen that at = 0, all the MS will transmit
other end, = 0 corresponds to an equal transmit with equal power and hence the average normalized
power The received SINR at the tagged BS which transmitted power is 1.
is located at origin from the nearest MS is at the
distance r is On the other hand, for = 0, the received power
at the BS will be equal and hence the average
power transmitted by MS is lowest and it is 0.34
The distance between the MS to its serving BS

(BS at the origin) is denoted by r, and the
distances between interfering MSs to their
respective serving BSs are denoted by ri . The
distance between an interfering MS to the serving
BS at the origin is denoted by di . We denote
the set of interfering MSs by . Channel fading
gain from tagged BS and ith interfering MS are
denoted by g and hi , respectively, which are
independent and identically exponentially
distributed with unit mean (corresponding to
Rayleigh fading).
III. S ELECTION OF POWER CONTROL

FACTOR
One important observation can be made is as
In this section, we evaluate power control factor follows: as increases, the rate of decreasing
based on the three parameters namely: average 3
average transmitted power decreases. For example,
transmit power, coverage probability, and average
as goes from 0 to 0.1, average power decreases by
rate. where Pt (r) is the transmit power of MS at
16% but at the other extreme, as goes from 0.9 to
distance r from the BS, and fR (r) is the
probability that MS is at distance r from the BS. 1, average power decreases by only 6%.
The probability density function (PDF) of r (and
also ri ), i.e., fR (r) is given by, The average
normalized transmit power used by the MS is given
398
Fig. 5 plots the average rate for the FR1 and FR3
with respect to power control factor for different
path loss exponents. Firstly, it can be observed
that as power control factor increases average rate
decreases for both the reuse systems and both the
path loss exponents. Secondly, it can be observed
that as power control factor increase from 0 to 0.25
average rate does not decrease significantly
(especially for FR3 system).
By observing Fig. 2, 3, 4 and 5 and behaviour of
these plots as discussed before, one can conclude
that at power control factor = 0.5, the average
transmitted power is decreased by 2, the coverage
probability of cell-edge users increases significantly
and also the average rate does not decrease
significantly and hence the power control factor
should be chosen close to 0.5.
We have one another way to evaluate the power
control factor. We develop a cost function J which
take care of all three parameter: average rate, edge
coverage probability and average transmitted power.
J = a(average rate) + b(edge coverge probability)

+c(average transmitted power).
Here a > 0, b > 0 and c < 0 are weight

parameter corresponding to average rate, edge
coverage probability and average transmitted
power, respectively. Now, we need to maximize
this to evaluate the power control factor. Fig. 6
plots the cost function J versus power control factor
for three different sets of weight parameter. it can
be observe that cost function is maximum at =
0.5. Although, for some sets of parameter cost
function will not maximum at = 0.5, most of
the acceptable sets of parameter cost function will
be maximum at = 0.5.
399
Fig. 4: Coverage probability of FR3 with respect to

distance from the BS for different values of power control
factor. Here = 3 and T = 0dB is assumed.
V. CONCLUSION
In this paper, we have evaluated the uplink power

control factor such that average transmit power
should be low, coverage probability of cell-edge
users should be high and also average rate should
be high. It turns out that power control factor should
be close to 0.5. The natural extension of this work
could be to evaluate the uplink power control factor
in presence of inter-cell interference coordination
scheme, i.e. fractional frequency reuse and soft
frequency reuse [10].
REFERENCES
400
Electronically Steerable Planer Phased Array

Antenna using HFSS
Vasujadevi M 1
Assistant Professor,
Chalapathi Institute of Engineering & Technology,
Guntur, Andhra Pradesh,India
Dr. P. Siddaiah 2
Professor & DEAN, IEEE member
University College of Engineering & Technology,
P. Sai Vinay Kumar

Student, VR Siddhartha Engineering College, Vijayawada
Abstract- A planar phased-array antenna has been need for any mechanical rotation. These unique
constructing from a 15x15 square grid of z-directed capabilities have found phased arrays a broad range
monopoles with a length of 0.475 , element spacing of of applications since the advent of this technology.
0.29 , average directivity of 20.0 dBi across all scan Phased arrays have been traditionally used in military
angles, an average H-plane HPBW of 37 degrees, an
applications for several decades. Recent growth in
average E-plane HPBW of 23 degrees, and an efficiency
of 99.6%. Two current distributions are tested: uniform civilian radar-based sensors and communication
and binomial. The binomial current distribution is systems has drawn increasing interest in utilizing
selected to minimize the side lobe level. For this phased array technology for commercial applications.
implementation we are using HFSS 14.0. Phased array antennas are common in
communications and radar and offer the benefit of
Keywords- Phased array, Electronically steerable, far-field beam shaping and steering for specific, agile
Current Distribution, HFSS. operational conditions. They are especially useful in
modern adaptive radar systems where there is a trend
I. INTRODUCTION toward active phased arrays and more advanced space
time adaptive signal processing. In phased arrays all
In antenna theory, a phased array is an array of the antenna elements are excited simultaneously and
antennas in which the relative phases of the the main beam of the array is steered by applying a
respective signals feeding the antennas are varied in progressive phase shift across the array aperture.
such a way that the effective radiation pattern of the
array is reinforced in a desired direction and
suppressed in undesired directions. An antenna array II. PHASED ARRAY ARCHITECTURE
is a group of multiple active antennas coupled to a
common source or load to produce a directive The phased array antenna has an aperture that is
radiation pattern. Usually, the spatial relationship of assembled from a great many similar radiating
the individual antennas also contributes to the elements, such as slots or dipoles, each element being
directivity of the antenna array. Use of the term individually controlled in phase and amplitude.
"active antennas" is intended to describe elements
whose energy output is modified due to the presence
of a source of energy in the element (other than the
mere signal energy which passes through the circuit)
or an element in which the energy output from a
source of energy is controlled by the signal input.
Phased array antenna is a multiple-antenna

system in which the radiation pattern can be
reinforced in a particular direction and suppressed in
undesired directions. The direction of phased array
radiation can be electronically steered obviating the Fig. Phased array antenna for radio astronomy.
401
(courtesy from skatelescope.org) antennas, the beam will experience a time delay equal
Accurately predictable radiation patterns and beam to Eqn. 1 in reaching successive antennas.
pointing directions can be achieved. A phased array is = 2 d sin ()/ (1)
an array antenna whose beam direction or radiation
pattern is controlled primarily by the relative phases Here, is the wavelength of the signals. Hence, if
of the excitation coefficients of the radiating the incident beam is a sinusoid at frequency with
elements. Physically it is composed of a group of amplitude of A, the signals received by each of the
individual elements that are arranged in a linear or antennas can be written as Eqn 2.
two dimensional (typically planar) spatial
configurations. Si = A e-jn (2)
The plane wave incident at an angle upon the
The first phased array antennas were developed phased array experiences a linear delay progression at
by the Nobel laureate Karl Ferdinand Braun in the the successive antenna elements. Therefore, the
beginning of the 20th century. He demonstrated the variable delay circuits must be set to a similar but
possibility to enhance antenna transmission in a with reverse delay progression to compensate for the
specific direction. Quite a number of applications of delay of the signal arrived at the antenna elements. In
phased array transmissions have been developed, linear arrays, variable time delays are designed to
ranging from radar systems with marine vessels to provide uniform phase progression across the array.
airplane navigation systems in which antennas are Therefore, the signal in each channel at the output of
placed in matrix configurations. The technique has the variable delay block can be written as Eqn. 3.
been refined considerably so as to allow for multi

functional solid state radar systems in which ships Si' = A e-jn e-jn (3)
and airplanes can be detected and in which
communication between moving objects can be In this equation, denotes the difference in phase
realized in an efficient manner. Other applications of shift provided by two successive variable time delay
phased array techniques can be found in radio blocks. Therefore, array factor which is equal to the
astronomy, your mobile phone and with military sum of all the signals normalized to the signal at one
applications. path can be written as Eqn. 4.
N
A. Phased Array Principle F = e-jn (-) (4)
n=1
According to Eqn. 1, 4, the peak of the array factor

occurs at an incident angle which can be determined
by Eqn. 5.
2 d sin ()/ = (or) =arc sin (/2d)
At this incident angle, which is called scan angle,

the linear delay progression experienced by the wave
arriving at the successive antennas is perfectly
compensated with the time delays incorporated at
each path resulting in all the signals being coherently
combined at the output of the receive array. The array
factor can also be shown as Eqn.6.
Fig. 1 Block Diagram of N-element Phased Array
The block diagram of an N-element phased array

(6)
is shown in Fig. 1 N identical antennas are equally
spaced by a distance d along an axis. Separate At the scan angle , the array factor has a
variable time delays are incorporated at each signal maximum value of N2. For other angles of incident,
path to control the phases of the signals before the array factor
combining all the signals together at the output. will be lower than this value indicating spatial
A plane-wave beam is assumed to be incident selectivity of phased array. In addition to the spatial
upon the antenna array at an angle of to the normal filtering, one of the main capabilities of phased array
direction. Because of the spacing between the is that the peak gain of the array, according to Eqn.5,
402
can be controlled by electronically tuning the variable that meets the directive angle requirements is
time delays eliminating the need for any mechanical required. Therefore, we consider an arrangement of
rotation of antenna array. It should be also noted that concentric circles, a 2dimentional rectangular planar
the benefits of using phased array is increased as the array, and a 3dimentional placement of dipoles as
number of elements in the phased array is increased. possible element arrangements. We were not able to
As mentioned before, the maximum value of array consider the concentric circles or 3dimensional
factor N2 is directly proportional to the number of arrangement because the required mathematical
array elements. Furthermore, the beam width of the analysis involved is lengthy and our design time is
array can be decreased by increasing the number of limited. In addition, 3 dimensional arrays add extra
array elements in order to enhance the spatial mechanical complexity to the design, and a goal of
selectivity of phased array. this project is to trade mechanical complexity for
electrical complexity.
B. Classification of Phased Arrays
Therefore, only the 2dimensional square array is
(1) Scanning methods considered. Two dimensional rectangular arrays are
(2) Radiator feed methods much easier to analyze. Although we expect to lose
(3) Positioning of radiators in the array some degree of accuracy azimuthally, a rectangular
array seemed like an effective trade off between
The main scanning methods are phase scanning design complexity and performance.
and frequency scanning. It is the phase scanned
arrays that are referred to as phased arrays. From the B. Element Spacing
viewpoint of feed methods, arrays are divided into
constrained feed arrays and space fed arrays, the In general, increasing element spacing allows for
latter taking the form of reflect arrays or transmission finer beam widths, but element spacing greater than
arrays. With regard to element positioning, phased halfwavelength results in undesired grating lobes.
arrays are divided into uniformly spaced and These grating lobes are shown for a separation of 0.8
unequally spaced arrays. They provide the radar with wavelengths along the x and yaxis elements: We
flexibility and adaptation to the assigned task, ability chose a uniform element spacing of 0.29 wavelengths
to change beam position in space almost such that the beam width is moderately fine, does not
instantaneously (electronic scanning), generation of produce grating lobes, and allows sufficient number
very high powers from many sources distributed of elements to fit on the circular mounting structure.
across the aperture, high directivity and power gain
possibility of synthesizing any desired radiation This element spacing determines the total number of
pattern (including formation of pattern nulls in the array elements due to restrictions in the maximum
directions of undesired interference sources) allowable array area. Since the maximum length of
capability of combining search, track, and recognition the square array is 0.707 the allowable diameter (50
functions when operating in multiple target and inches), the number of elements allowed with a 0.29
severe interference environments (including wavelength element separation equals the following:
jamming), enhanced target throughput capability, and
compatibility with digital computers and digital N or M max=
signal processing algorithms.
III. ANALYSIS N or M max =15.48 = 15

This antenna design attempts to find the optimal
combination of several properties: element C. Element Type
distribution, element type, element spacing,
efficiency, current distribution, and phase ZDirected monopoles were chosen over patch
requirements. Ansoft HFSS 14.0 simulation is going antennas for improved efficiency. The monopoles are
to be used to demonstrate the effectiveness of the zdirected to exploit the blind spot in the = 0o
design. direction, which falls outside of the design
requirements (minimum =10o). A halfwave dipole
A. Element Arrangement, 2- Dimensional Square is used in the simulation because this represents a
Array realistic length, when compared to the ideal and short
dipoles at 125MHz, and because larger structures are
Element arrangement is the first factor we more resistant to changes in characteristics produced
consider in this design. An array factor expression by rust and chemical erosion.
403
In addition, the input impedance of such a dipole

can be made real (70) by shorten ing the dipole V. RESULTS AND DISCUSSIONS
slightly to achieve resonance. Since our array consists
of erect monopoles positioned directly normal to a We simulate the following change in variables for
PEC ground plane, the length of our array elements our rectangular array design:
are only a quarterwavelength (60cm). The actual
length is slightly less than 60cm to achieve resonance 1. Rectangular Array of size N x M
when considering the extra effective length due to 2. Arbitrary Element Spacing Dx and Dy
fringe field effects. The input impedance is half of 3. Uniform or Binomial current distribution
that for the equivalent dipole, or 35. The expected 4. 0.5 Dipoles or Isotropic Sources to simulate
directivity is twice that for a halfwave dipole in free a pure Array Factor
space or 5.16dB. 5. Phase distance x & y or beam direction o
& o.
D. Current Distribution
HFSS 14.0 Simulation displays the following data:
In order to achieve the lowest side lobe levels
1. Normalized 3d mesh grid plot of the Total
(SLL) possible, we decided to use a binomial array Radiated Pattern
amplitude distribution, which completely eliminates
2. HPBWE (Present State, Max, and
the side lobes. This has the effect of decreasing
Simulation Average)
directivity and halfpower angle in the E and
3. HPBWH(Present State, Max, and Simulation
Hplanes. The uniform pattern produces unacceptable Average)
SLL when the beam is positioned 10 degrees from the 4. Directivity (Assuming Dipole, Present State,
zaxis. The uniform and binomial distributions are Min, and Simulation Average.
shown using HFSS.
Given: N = M = 15, Dx = Dy = 29.
IV. HFSS
Binomial Current Distribution:
HFSS is a commercial finite element
method solver for electromagnetic structures
HPBW_E_MAX:56.1728o :
from Ansys. The acronym originally stood
HPBW_H_MAX:27.6654o :
for high frequency structural simulator. It is one of
D_MIN:58.6124
several commercial tools used for antenna design,
and the design of complex RF electronic circuit HPBW_E_AVG:37.0722 o
:
elements including filters, transmission lines, and HPBW_H_AVG:23.7516 o :
packaging. D_AVG:101.6584
There are six main steps to creating and solving a proper Since the beam widths were so high we decided
HFSS simulation.
to see what kind of beam widths we could have
They are: expected with uniform current distribution, despite its
1. Create model/geometry large side lobes.
2. Assign boundaries
3. Assign excitations Uniform Current Distribution:
4. Set up the solution
5. Solve HPBW_E_MAX:34.153o :
6. Post-process the results HPBW_H_MAX:11.5014o
HPBW_E_AVG:19.9463o:
HPBW_H_AVG:10.1628 o
VI. CONCLUSION
Phased array, capable of providing a directional

beam that can be electronically steered, can
significantly enhance the performance of sensors and
communications systems. The directional beam of a
phased array also allows for a more efficient power
Figure: Steps creating and solving a proper HFSS simulation management. In addition, the spatial filtering nature
404
of the phased array systems alleviate the problem of [14] R. J. Mailloux, Phased array antenna handbook. Artech
House, Boston, 1994.
multipath fading and co-channel interference by
[15] A. K. Bhattacharyya, Phased Array Antennas. Wiley-
suppressing signals emanating from undesirable Interscience, New York,2006.
directions. Unfortunately, side lobe levels on the [16] A. Ishimaru, R. Coe, G. Miller, and W. Geren, Finite
uniform distribution are above the 20dBi periodic structure approach to large scanning array
problems, Antennas and Propagation, IEEETransactions
requirement. on, vol. 33, no. 11, pp. 12131220, Nov 1985.
[17] A. Skrivervik and J. Mosig, Analysis of nite phase arrays
The uniform distribution experiences significantly of microstrip patches, Antennas and Propagation, IEEE
reduced beam width than the binomial distribution; Transactions on, vol. 41, no. 8,pp. 11051114, Aug 1993.
[18] A. Neto, S. Maci, G. Vecchi, and M. Sabbadini, A truncated
however, directivity for each case exceeds the
Floquet wave diraction method for the full wave analysis of
specified requirement (20dBi). By using Dolph large phased arrays. I. Basic principles and 2-D cases,
Chebyshev current distribution we can reduces Antennas and Propagation, IEEE Transactions on, vol. 48,
HPBW while maintaining -20 dBi side lobes relative no. 4, pp. 594600, Apr 2000.
to the main beam. [19] C. Craeye, A. Thuis, and D. Schaubert, An e cient MoM
formulation for nite-by-innite arrays of two-dimensional
antennas arranged in a threedimensional structure, Antennas
ACKNOWLEDGMENT and Propagation, IEEE Transactions on, vol. 52, no. 1, pp.
271282, Jan. 2004.
[20] D. M. Pozar, General relations for a phased array of printed
We would like to thank the Organization of antennas derived from innite current sheets, Antennas and
Acharya Nagarjuna University, Guntur, special Propagation, IEEE Transactions on, vol. 33, no. 5, pp. 498
504, May 1985.
thanks to the Dr. P. Sidaiah garu, for support to use
R&D Laboratories .
AUTHOR BIOGRAPHY
REFERENCES
Smt. M. Vasujadevi obtained B.Tech
degree in Electronics and
[1] Danial Ehyaie, Novel Approaches to the Design of Phased communication Engineering from
Array Antennas, Doctor of Philosophy(Electrical S.V.V.S.N Engineering College,
Engineering) thesis, University of Michigan,2011 Acharya Nagarjuna University, Guntur
[2] Kyle Howen , Andrew Huard,Design of Planer Phased in 2008, M. Tech Degree from BSIT,
array antenna, unpublished. JNT University, Hyderabad in 2011.
[3] R. J. Mailloux and I. Books, Phased array antenna handbook:
Artech House,2005.
She is currently working as Assistant Professor, Dept. of
[4] R. J. Mailloux, "Electronically scanned arrays," Synthesis ECE in Chalapathi Institute of Engineering and
Lectures on Antennas, vol. 2, pp. 1-82, 2007. Technology, affiliated to Acharya Nagarjuna University,
[5] Joseph Spradley, A Volumetric Electrically Scanned Two- Guntur, India. Her interesting fields are Antennas, Signal
Dimensional Microwave Antenna Array, IRE National Processing, Communications, VLSI Design.
Convention Record, Part I - Antennas and Propagation; e-mail: vasujadevibhavanam@gmail.com
Microwaves, New York: The Institute of Radio Engineers,
1958, 204-212.
[6] P. D. Trinh, S. Yegnanarayanan, F. Coppinger and B. Jalali
Silicon-on-Insulator (SOI) Phased-Array Wavelength
Dr. P Siddaiah obtained B.Tech degree
Multi/Demultiplexer with Extremely Low-Polarization in Electronics and communication
Sensitivity, IEEE Photonics Technology Letters, Vol. 9, No. Engineering from JNTUA college of
7, July 1997 engineering in 1988. He received his
[7] "YIG-sphere-based phase shifter for X-band phased array M.Tech degree from SV University
applications". Scholarworks. Tirupathi. He did his Ph.D program in
[8] Worlds Most Complex Silicon Phased Array Chip JNTU Hyderabad. He is the Chief
Developed at UC San Diego in UCSD News (reviewed 02. Investigator for several outstanding Projects sponsored by
November 2007)
[9] A. Balanis, Antenna theory, 3rd ed. Wiley-Interscience, New
Defense Organizations, AICTE, UGC & ISRO. He is
York, 2005. currently working as Professor & DEAN, Department of
[10] P. Patel, D. Kant, E. Wal, and A. van Ardene, Phased array ECE in University College of Engineering and Technology,
antennas demon- strator as a radio telescope - embrace, Acharya Nagarjuna University, Guntur, India. He has
Antennas and Propagation Society International Symposium, taught a wide variety of courses for UG & PG students and
2008. AP-S 2008. IEEE, pp. 14, July 2008. guided several projects. Several members successfully
[11] S. Hay, J. OSullivan, J. Kot, C. Granet, A. Grancea, A. completed their Ph.D under his guidance. Several members
Forsyth, and D. Hayman, Focal plane array development for pursuing their Ph.D degree. He has published several
ASKAP (Australian SKA pathnder), Antennas and
Propagation, 2007. EuCAP 2007. The Second
papers in National & International Journals & Conferences.
EuropeanConference on, pp. 15, Nov. 2007. He is the life member of FIETE, IE & MISTE.
[12] N. Amitay, V. Galindo, and C. P. Wu, Theory and analysis of
phased arrayantennas. Wiley-Interscience, New York, 1972.
[13] R. J. Mailloux, Phased array antenna handbook. Artech
House, Boston, 1994.
405
Analysis and Performance Comparison of DOA

Algorithms for Continuous Tracking And Quality
Enhancement Of Desired Signal In Wireless
Services With Smart Antenna Technology
M.Siva Ganga Prasad1, KVR Chowdary1, P.Siddaiah2, P.Sai Vinay Kumar3
1: Faculty at Sir CR Reddy college of Engg, 2: Dean, ANU, 3: Student
Corresponding e-mail id: msivagangaprasad@gmail.com
Abstract - Employment of powerful Direction-of-Arrival The very purpose of DOA estimation is to use the
(DOA) estimation and beam-forming algorithms with data received by the array to estimate the direction of
smart antenna systems enables continuous tracking of arrival of the signal. The results of DOA estimation
desired signal irrespective of the signal environment
are then used by the array to design the adaptive
there by, enhancing the quality of wireless services. In
this paper emphasis is laid mainly on the performance beam-former [4], which is used to maximize the
of the various powerful high-resolution subspace power radiated towards users, and to suppress
algorithms like MUSIC, root-MUSIC and another fast interference. As a result, we can infer that a
DOA estimation algorithm making use of pseudo- successful design of an adaptive array depends highly
covariance matrix, taking into the consideration, the on the performance of the DOA algorithm. Direction
number of snap-shots required to build an array of arrival algorithms are usually complex and their
correlation matrix. The aim is to compare their performance depends on many parameters such as
performances by varying the number of input signal number of mobile users and spatial distribution, the
snap-shots and to find the best algorithm that gives
number of array elements and their spacing and the
optimum performance with reduced number of snap-
shots from the results obtained. Their performance in number of signals samples.
cases of high signal correlation and rapidly changing
signal DOAs is also discussed. In this paper, the performance comparison of two
DOA Estimation algorithms is done based on number
Keywords - array correlation matrix, Direction-of- of snap-shots required for each of them to give a
Arrival (DOA) estimation, MUSIC, pseudo-covariance good resolution. The number of snap-shots required
matrix, root-MUSIC, subspace algorithms. determines the effectiveness of the system to rapidly
update DOAs of the signals and enabling the
I. INTRODUCTION continuous tracking of the targets. In the following
sections, the algorithms considered for comparison
In wireless communication system, the design,
are analyzed in detail and the results obtained are
orientation and performance of the antenna decides
discussed.
the quality of the wireless services. And so, the
performance of the antenna systems deployed for II.MUSIC and root-MUSIC DOA ESTIMATION
such purposes is of prime importance. Hence the ALGORITHMS
Omni-directional[1] were replaced by the smart (Sub-Space based Methods)
antenna array systems [2] to reduce multipath and co-
channel interference and to offer preferential gain for In Subspace based method [5], the observed
the signals of served users by rejecting the signals covariance matrix is decomposed into two orthogonal
that interfere with those of desired ones. This enables spaces: signal subspace and noise subspace. The
a higher capacity, in addition to spectral efficiency DOA estimation is calculated from any one of the
and frequency reuse [3]. This is achieved by focusing subspaces. The subspace based DOA estimation
the radiation only in the desired direction and algorithms MUSIC root-MUSIC and ESPRIT
adjusting itself to changing traffic conditions or provide high resolution, more accuracy and are not
signal environments. Smart antennas employ a set of limited to physical size of array aperture. The steering
radiating elements arranged in the form of an array vectors corresponding to the directional sources are
(ULA, in this case). The signals from these elements orthogonal to the noise subspace. As the noise
are combined to form a movable or switchable beam subspace is orthogonal to the signal subspace, these
pattern that follows the desired user. steering vectors are contained in the signal subspace.
406
The noise subspace is spanned by the eigenvectors

associated with the smaller eigen-values of the
(2.4)
correlation matrix, and the signal subspace is spanned
by the eigenvectors associated with its larger eigen- However, when the signal sources are coherent or
values. as the noise variances vary, the resolution of
MUSIC diminishes. Also when the DOAs of signals
A. MUSIC Algorithm change rapidly, the performance of the MUSIC
algorithm is severely degraded. To overcome this,
MUSIC stands for MUltiple SIgnal Classification, one many array input data snapshots are required to
of the high resolution subspace DOA algorithms, obtain a good performance.
which gives an estimate of a number of arrived
signals, hence their direction of arrival. Estimation of B. Root-MUSIC Algorithm
DOA is performed from either signal or noise
subspace[6], assuming that noise in each channel is The MUSIC algorithm in general can apply to any
highly uncorrelated. This makes the covariance arbitrary array regardless of the position of the array
matrix diagonal. elements. Root-MUSIC implies that the MUSIC
algorithm is reduced to finding roots of a polynomial
If the number of signals impinging on M element as opposed to merely plotting the pseudospectrum or
a r r a y is D, the number of signal eigenvalues and searching for peaks in the pseudospectrum [7].
eigenvectors is D and number of noise eigenvalues Barabell simplified the MUSIC algorithm for the
and eigenvectors is M-D. The array correlation case where the antenna is a ULA. Recalling that the
matrix with uncorrelated noise and equal variances is MUSIC pseudospectrum is given by
then given by
(2.1) (2.5)
The denominator expression can be simplified by
Where defining the matrix which is Hermitian.
is an M x D array steering matrix, and This leads to the root-MUSIC expression
(2.2)
is D x D source correlation matrix. (2.6)
Rxx has D eigenvectors associated with signals and In case of a Uniform Linear Array (ULA), the
M D eigenvectors associated with the noise. We element of the array steering vector is given by:
can then construct the M x (M-D) subspace spanned
by the noise eigenvectors such that
where m=1, 2, .M
(2.3)
The denominator argument of the root-MUSIC
Noise eigen-vectors are orthogonal to steering vectors expression can be written as
at the angles of arrival .
Because of this orthogonality condition, the
Euclidean distance
for each and every arrival angle. The noise subspace
eigenvectors are orthogonal to array steering
vectors at the angles of arrivals
and placing this distance
expression in the denominator gives sharp peaks at
the angles of arrival. The MUSIC Pseudo-spectrum (2.7)
is given as:
407
Where is the sum of the diagonal elements of

along the diagonal such that The MUSIC algorithm can also estimate the DOAs of
signals with adjacent incidence angles. However,
many array input data snapshots are required to obtain
a good performance. In addition, the DOAs of the
(2.8) signals must not be changed while the input data is
being received to acquire the covariance matrix, plus
The matrix has off-diagonal sums such that the incidence signals must be uncorrelated. Thus,
for . Thus the sum of off-diagonal when signals are correlated or the DOAs of signals
elements is always less than the sum of the main change rapidly, the performance of the MUSIC
diagonal elements. In addition, . algorithm is severely degraded. It is also impossible
to estimate the DOAs of signals quickly, because the
For a 6 x 6 matrix we have 11 diagonals ranging from MUSIC algorithm cannot perform a DOA estimation
diagonal numbers l = -5, -4, 0, 4, 5. The lower left until it has received all the snapshots of the array for
diagonal is represented by l= -5 where the upper right the covariance matrix.
diagonal is represented by l=5. The coefficients
are calculated by In contrast, a direct data domain adaptive (DDDA)
beam-former uses a single snapshot to acquire a
and so on. pseudo-covariance matrix [8], which performs similar
function to a covariance matrix. A beam-former
The above equation can be simplified to the form of a creates a beam pattern that allows the look direction
polynomial whose coefficients are , thus signal to pass, while removing the rest of the signals.
As such, for the rapid identification of the DOAs of
incidence signals to array systems based on less
sample data, even under coherent signal conditions,
this paper presents a DOA estimation algorithm that
(2.9)
uses a pseudo-covariance matrix based on a single
where snapshot. The bearing response and the directional
spectrum are then obtained using the pseudo-
The roots of D (z) that lie closest to the unit circle covariance matrix.
correspond to the poles of the MUSIC The proposed DOA estimation algorithm operates
pseudospectrum. Thus, this technique is called root- based on the following two steps: First, rough
MUSIC. The above polynomial is of order 2(M-1) incidence angle ranges for the signals are obtained by
and thus has roots of . Each root using the bearing response. Second, the exact
can complex and using polar notation can be written incidence angles are obtained by combining the
as bearing response and directional spectrum. The
proposed algorithm only uses a few snapshots to
estimate the DOAs of signals, whereas the MUSIC
where i = 1, 2, 2(M-1) algorithm requires hundreds of snapshots, the
proposed algorithm can estimate the DOAs of signals
where arg( ) is the phase angle of . with rapidly changing incidence angles. Moreover,
the DOAs from coherent signals can be discriminated
Exact zeros in D(z) exist when the root magnitudes from different DOAs by taking advantage of the
. We can calculate the Angle of pseudo-covariance matrix.
Arrival(AOA) by comparing to Direct Data Domain Adaptive (DDDA) Beam-

to get forming Technique
Consider a uniformly spaced linear array with N

(2.10) array elements. Assume that L narro w band
signals impinge o n array fro m L directio ns
III. FAST DOA ESTIMATION ALGORITHM USING .T he angles i=1, 2, L. are
PSEUDO-COVARIANCE MATRIX defined from the broadside direction of the array.
408
The signal vector X (K) impinging on the array at

time is K defined as
(3.1)
(3.4)
It performs a similar function to a covariance matrix,

where superscript T denotes a transpose.A() is a where M is equal to (N+1)/2. The degree of freedom
matrix consisting of steering vectors, s(K) is the (DOF) of the array is reduced to M because the size
signal vector consisting of L different incidence of a pseudo-covariance matrix is only M x M in spite
signals and n(K) is the white noise vector generated of an original DOF of N-1. When the look direction
at each array element with a zero mean and variance of the beam-former is can be expressed as:
of .
A(),S(K) and n(K) can be expressed as:

3.5)
(3.2a) where and d are the wave-length of the carrier and

inter element spacing, respectively.
(3.2b)
The look constraint generator transposes into
(3.2c)
respectively, where i=1,2,3..L, are the

steering vectors of coming from
(3.6)
th
The i steering vector is defined as
The pseudo-covariance matrix X (k) is the signal
received at the antenna elements, and it is assumed
that is the incidence signal from the look
(3.3) direction , where is the signal strength.
The received signal X (k) includes the desired signal,

interferences, and thermal noise, when the desired
signal is . Therefore, [X (k) - ] only
includes thermal noise and interferences. The vector
v for minimizing the noise power is defined to satisfy
[X (k) - ][v]= [0] (3.7)
The above equation can be expressed as a generalized

eigen value equation given by
X (k) v = v (3.8)
The eigenvalue-decomposition produces a diagonal

matrix consisting of generalized eigenvalues and full
matrix whose columns are the corresponding
Fig.1 DDDA Beam-former Structure eigenvectors, where the smallest eigenvalue and v is
is an Mx1 eigenvector corresponding to. The
The Fig.1 shows a DDDA beam-former using a eigenvector is defined as a noise subspace. So, the
pseudo-covariance matrix. The pseudo-covariance eigenvector v is orthogonal to other steering vectors
matrix generator builds X (k) as
409
except for the steering vector for the look direction In the proposed algorithm, there is no signal
. cancellation problem, even though the signals are
correlated. The output power of array is
The constraint condition is determined as
(3.9)
(3.13)
Where
Where K is the number of snapshots required to
observe the output power. If a signal impinges
and ( is the steering vector for the look on the array within a small level of deviation from the
direction with Mx1 dimension. look direction, the bearing response can be
given by
The weight vector can be obtained from the eigen
vector v and the steering vector as (3.14)
The bearing response only has the power of

(3.10) added noise. If there is no signal impinging on the
array within a small level of deviation from the look
When the array is steered to the look direction
direction, the bearing response only includes
the output of the array y (k) is defined as
the noise power. Therefore, the bearing
responsearound an incidence signal has a much
higher power than that from a noise direction. To
(3.11) locate the possible angles of incidence, the algorithm
must search for peaks that are higher than a preset
The output power can be minimized by the weight threshold. This threshold is chosen based on the
vector whose pattern nulls are formed based signal and noise power levels.To get more accurate
on the directions of the signals except for the DOAs of incidence signals, the directional spectrum
incidence signal from the look direction. may be used. First of all, the angle with the smallest
bearing response should be searched from
The array should satisfy L M to discriminate all the bearing response as
incidence signals. If the look direction is equal to
the incidence angle, the weight vector can be (3.15)
obtained from the eigenvalue-decomposition of the
pseudocovariance matrix. Since the weight vector The corresponding weight vector w (
, which has a unit gain in the look direction, can be obtained by performing the DDDA
has pattern nulls at the incidence angles for the rest of beamforming. Next, the normalized directional
the signals. spectrum can be taken as
The array output y (k) is
+ (3.16)
) Because the weight vector w ( is

(3.12) orthogonal to the steering vectors of the incidence
signals, the directional spectrum has the peaks at the
Where and are correlated to each DOAs of the incident signals. So, accurate DOAs can
other. be obtained by using both the bearing response and
the directional spectrum. First, rough DOA ranges
can be obtained from the bearing response. Second,
peak DOAs can be considered as accurate DOAs if
410
peaks in the directional spectrum are within rough

DOA ranges acquired from the bearing response.
IV.RESULTS AND DISCUSSIONS
Here, the results are obtained for above algorithms

taking six array elements. The y-axis and the x-axis
represent the spectral power P() and the range of
angles respectively.
(a) The roots of polynomial plotted on z- plane
Fig.2 DOA estimation of MUSIC algorithm for

uncorrelated signals incoming from and for
100 snapshots.
(b) DOA estimation algorithm for uncorrelated signals
incoming from and
Fig.4 Root-MUSIC algorithm for 100 snapshots
Fig.3 DOA estimation of MUSIC algorithm for

uncorrelated signals incoming from and for 5
snapshots
If we observe Fig.2 and Fig.3, we can see sharp peaks

at the angles and in the case of 100 snap
shots but this cannot be seen in the case of 5
snapshots. The resolution is very poor in case of 5 (a) The roots of polynomial plotted on z-plane
snap-shots.
411
Fig.7 DOA estimation using the proposed algorithm for

uncorrelated signals incoming from and for 5
snapshots
(b) DOA estimation for uncorrelated signals incoming

Here, in Fig.6 and Fig.7 we observe the same
from and for resolution for 5 snap-shots and for 100 snap-shots.
Hence, the proposed algorithm is superior to MUSIC,
Fig.5 Root-MUSIC algorithm for 5 snapshots
root-MUSIC algorithms in its performance and speed
If we observe the above results this is similar to of DOA Estimation with reduction in the number of
required signal snap-shots.
music algorithm. We got sharp peaks in case of 100
snapshots. From this we can estimate the DOA
CONCLUSIONS
clearly which is not possible in case of 5 snapshots.
The above results clearly show that the proposed
In root-MUSIC, we have an advantage of reduced algorithm which makes use of pseudo-covariance
complexity over MUSIC that, we can find the DOA matrix gives fine resolution even for 5 snap-shots
of the signal simply by finding the locations of the which is not possible in case of MUSIC and root-
roots of polynomial rather than plotting the spectrum MUSIC algorithms. The reduction in the number of
searching for peaks in it. snap-shots required reduces the delay in scanning of
the input data and processing of it. The time required
Fig. 4(a) and Fig.5(a) give the location of roots on z- for acquiring say 5 snap-shots is very less thereby
plane and the Fig.4(b) and Fig 5(b) give the peaks at enabling the system to adapt to the rapidly changing
angles of arrival corresponding to the location of the or correlated interference environments. The time
roots of polynomial for 100 and 5 snap-shots each taken for the DOA estimation will be very less giving
respectively. scope for the system for continuous updating of
signal DOAs and signal tracking in rapidly changing
signal environments.
REFERENCES
[1] K. D. Prasad, Antenna and Wave Propoation, Third Edition,

Satya Prakashan, 1996.
[2] Liberti, J., and T.Rappaport, Smart Antennas for Wireless
Communications : IS-95 and Third Generation CDMA
Applications, Printice Hall, New York, 1999.
[3] Constantine A. Balanis, Panayiotis, Introduction to Smart
Antennas (Synthesis Lectures on Antennas), Series edition 2007
by Morgan &Claypool.
[4] M. Chryssomallis, Smart antennas IEEE antennas and
propagation magazine Vol.42 No 3 pp 129-138, June 2000
[5] A. Paulraj, B. Ottersten, R. Roy, A. Swindlehurst, G. Xu, and
T. Kailath, Subspace Methods for Direction of Arrival Estimation.
Amsterdam:North-Holland, 1993, vol. 10,ch. 16, pp. 693739.
Fig.6 DOA estimation using the proposed algorithm for [6] Lal Chand Godara, Smart antennas, CRC Press, 2004.
uncorrelated signals incoming from and for [7] F. Gross, Smart Antennas for Wireless Communication,
McGraw-Hill, September 14, 2005.
100 snapshots
412
[8] J. T. Kim, S. H. Moon, D. S. Han, and M. J. Cho, Fast DOA

estimation algorithm using pseudocovariance matrix, IEEE Trans.
Antennas Propagat., vol. 53, no. 4, pp. 13461351, Apr. 2005. .
[9] T. S. Rappaport, Smart Antennas Adaptive Arrays, Algorithms,
Wireless Position Locations Selected Readings. Piscataway, NJ:
IEEE, 1998.
[10] Brookner, E., Trends in Array Radars for the 1980s and
Beyond, IEEE Antenna and Propagation Society Newsletter,
April 1984.
About Authors:-
Dr. M. Siva Ganga Prasad was born in Vijayawada, A.P., India

on April 22, 1974. He received bachelors degree in engineering
from M V S R Engineering College and M.Tech in VLSI from
Bharat Institute of Higher Education & Research, Chennai. He
received his PhD degree electronics and communication
engineering from JNTU, Kakinada in the area of Wireless
Communications and Antennas. He has published papers in various
National and International Journals, Conferences and is currently
working as a Professor & HOD, Dept of ECE, in Sir CR Reddy
college of Engg, Eluru.
Mr. K.V.R Chowdary was born in Marrivada, Krishna Dist,

A.P., India on August 26, 1960. He received bachelors degree in
engineering from V.R Siddartha Engineering College and M.Tech
in Adavanced Electronics from J.N.T.U, Hyderabad. He has
published papers in various National Conferences and is currently
working as an Associate Professor, Dept of ECE, in Sir CR Reddy
college of Engg, Eluru.
Dr. P Siddaiah obtained B.Tech degree in Electronics and

communication Engineering from JNTUA college of engineering
in 1988. He received his M.Tech degree from SV University
Tirupathi. He did his Ph.D program in JNTU Hyderabad. He is the
Chief Investigator for several outstanding Projects sponsored by
Defense Organizations, AICTE, UGC & ISRO. He is currently
working as Professor & DEAN, Department of ECE in University
College of Engineering and Technology, Acharya Nagarjuna
University, Guntur, India. He has taught a wide variety of courses
for UG & PG students and guided several projects. Several
members successfully completed their Ph.D under his guidance.
Several members pursuing their Ph.D degree. He has published
several papers in National & International Journals & Conferences.
He is the life member of FIETE, IE & MISTE.
P.Sai Vinay kumar obtained his B.Tech from KL University and

he is pursuing M.Tech. He has published one International Journal
paper.
413
Design and Analysis of C-Band Rectangular

Microstrip Patch Antenna
B. Sekharbabu 1, BV. Subba Rao 2, I. Kullayamma 3& Dr . R.V.S. Satyanarayana 4
ECE Department, Sri Venkateswara University, Tirupati.
Abstract - The development in communication systems In this paper co-axial feed rectangular microstrip
requires the development of low cost, minimal weight and antenna is designed for Radar applications and is
low profile antennas that are capable of maintaining expected to operate within 5.45GHz
high performance over a wide spectrum of 5.85GHz frequency span.
frequencies. This technological trend has focused much
effort into the design of a Microstrip patch antenna. This
paper deals with design aspects of Rectangular
Microstrip patch Antenna with co-axial feed and its
properties. In this paper simulations are carried out on
C- band Microstrip patch antenna that suits for Radar
applications. With the adjusted parameters it
exhibits maximum 6.88 dBi Gain and VSWR=1.3 at the
centre frequency 5.65GHz. Desired patch antenna design
was simulated by IE3D Software
I. INTRODUCTION
Antennas play a very important role in the field of

Wireless communications. Some of them are Fig 1. Rectangular Microstrip
parabolic reflectors, patch antennas, slot antennas, Patch
and folded dipole antennas with each type having
their own properties and usage. Microstrip antenna
A. How patch Antennas
technology began its rapid development in the
work
late1970s. By the early 1980s basic microstrip
antenna elements and arrays were fairly well establish Like all antennas, patch antennas work
in term of design and modeling. In the last decades because currents in a conductive surface induce
printed antennas have been largely studied due to electric fields in the surrounding space. There are
their advantages over other radiating systems, which two important factors about how a patch
include: light weightiness, reduced size, low cost, antenna compares to other antennas.
conformability and the ease of integration with active
Device A Microstrip Patch antenna having Radiating
patch on one side of a dielectric substrate and ground
plane on the other side is shown in Figure 1.
The patch is generally made of conducting material

such as copper or gold. The radiating patch and the
feed lines are usually photo etched on the dielectric
substrate. Microstrip patch antennas radiate primarily
because of the fringing fields between the patch edge
and the ground plane. Therefore, the antenna can be
fed by a variety of methods. These Methods can be
classified into two categories- contacting and non-
contacting. In the contacting method, the RF power is Fig 2. Rectangular Microstrip Patch Radiating
fed directly to the radiating patch using a connecting Geometry
element such as a microstrip line or probe feed In Primarily antennas are most commonly
the non-contacting scheme, electromagnetic field compared to dipole antennas because they are
coupling is done to transfer power between the the simplest to analyze which is also true for
microstrip line and the radiating patch this includes patch antennas. In a patch antenna the edge of the
proximity feeding and aperture feeding. antenna Acts like a dipole antenna, inducing an
414
electric field perpendicular to the radiating edge constructive and deconstructive interference effect
(this is shown in The Figure 2). While the in the far field.
radiating edges are perpendicular to the E field
plane the size of the patch is also very D. Co-axial Feed
important due to the fact that a mode will be set
The Coaxial feed or probe feed is a very common
up in the cavity beneath the patch which
technique used for feeding Microstrip patch
determines the shape of the currents and the
antennas. As seen from Figure 3 the inner conductor
shape of the effective radiators.
of the coaxial connector extends through the
dielectric and is soldered to the radiating patch,
B. Uses of Patch Antennas
while the outer conductor is connected to the
ground plane The main advantage of this type of
Patch antennas are becoming more common because
feeding scheme is that the feed can be placed at any
more complicated designs are continually being
desired location inside the patch in order to match
created allowing patch antennas to enter new
with its input impedance. This feed method is easy
niches of technology. The most common use of a
to fabricate and has low spurious radiation.
patch antenna today is in the cell phone. Patch
However, a major disadvantage is that it provides
antennas are cheaper to manufacture and can provide
narrow bandwidth and is difficult to model since a
Equal or better gain performance as compared to the
hole has to be drilled in the substrate and the
standard dipole and monopole antennas traditionally
connector outside the ground plane.
utilized in cell phones in the past. Also patch
antennas, being integrated into the PWB, can very
easily be created with active structures which create
opportunities for active and smart antennas in small
packages. This means that as time goes on, it is
feasible more and more antennas will utilize the
patch antenna methodology because very high
directional gains are achievable with the added
capability of being steerable while still in a small
package. Patch antennas are also utilized in many
other locations such as airplanes and space
applications because of their very light weight and
small size. It is also quite easy to create a patch
antenna which can handle harsh environments
because of their simple structure. Future work in
patch antennas will include smart active antenna
designs as well as multiple dimensional arrays, which
utilize the multi-layer PWB technology to create Fig 3. Rectangular Microstrip Patch Antenna with
advanced microstrip antennas. Co-axial feed
C. Important factors for patch antenna design 2. DESCRIPTION OF THE ANTENNA

GEOMETRY:
The main factors in the design of a patch antenna
are the geometry, respective size of the patch, and The geometry of single element antenna is shown in
the location of the feed. As the patch gets the figure 3. The main objective of this work is
larger in comparison to the wavelength of to design Co-axial feed Rectangular microstrip
the exciting waveform there will be higher order Patch antenna to resonate at 5.65GHz. The electrical
modes excited within the patch antenna cavity properties of substrate include substrate thickness
h, and dielectric constant r these parameters
causing interesting Results. The higher order
modes can cause multiple areas along the same
edge of the antenna to take on different current determines the size of the antenna. The dielectric
densities due to the standing waves in the patch. loss tangent of the substrate should be low (<0.002)
This means that one edge of the antenna can change in Order to ensure high performance and good
from a single dipole effect to mimise the pattern of efficiency There are numerous substrates that can be
multiple collinear antennas. This will give results used for the design of microstrip antennas with
much like a single dipole that has a length of dielectric constant in the range 2.2 < r < 12
multiple wavelengths. In effect you get a Thick substrates are most desirable for antenna
415
performance whose dielectric constants is in the on Zeland program Manager. The Zeland
lower end of the ranges because they provide better window will be opened then click on
efficiency and larger bandwidth but at the cost of file and select MGRID.
larger element size. RT/Duroid 5880 is used the
purpose of selecting this particular substrate is New window will be opened and in
because; it possesses lowest electrical loss tangent that click on insert the substrates symbol
and has a better thermal conductivity that are very and insert the dielectric constant, Height
much needed for aerospace application antennas. and thickness of the substrate.
A. Physical Parameters of the Patch Antenna
After entering all the details go to
The antenna parameters of this antenna can be Entity then click on rectangle a
calculated by the transmission line method (Balanis, dimensions window will appears then
2005), as exemplified below. insert our length and width of the patch
L=17mm and W=21mm.
After entering patch length and width go

to Entity then click probe feed to patch by
trail and error calculate feed location(-7,0) .
After entering patch length and width

feed Location Save the design and assign
port to patch.
After entering port then go for the

Simulation and assign range of frequency
and meshing frequency and also select the
graph parameters which are needed to
calculate for the antenna.
3. SIMULATION RESULTS
Where is the wave length, fr is the resonant

frequency, L and W are the length and width of the
patch element respectively and r is the
dielectric constant
B. Theoretical calculations of Microstrip Patch

Antenna
The length and width of the microstrip patch are

calculated using the equations 1&2. So, the
Fig 4. Rectangular Microstrip Patch antenna
theoretical calculations for a Rectangular patch are
with co-axial feed
The given frequency: 5.45 GHz to 5.85GHz
Resonant frequency fr=5.65GHz
In this designing RT Duroid 5880 with thickness

h=1.6mm is used so the dielectric constant of this
substrate is
r
=2.2 Therefore the length and width of the patch
are W =21mm L=17mm
C. Steps to design an antenna in IE3D:
Click on Start Go to Programs then click Fig 5. VSWR of the Patch antenna
416
4. CONCLUSIONS
The simulation results obtained during this

investigation demonstrate a promising new approach
to the design of antenna with High Gain. The method
described provides a means of obtaining linearly
polarized antenna having 6.88 dBi Gain and and
76Azimuth 3dB Beam width, 90Elevation 3dB
beam width and VSWR=1.3 at the centre frequency
5.65GHz.with the given specifications Simulation is
carried out in the frequency band (5.45-5.85GHz).
The gain pattern throughout the band is calculated.
Fig 6.Current distribution in the patch Thus, the proposed antenna is a very attractive
solution for small portable devices and for spatial
diversity systems where compact antennas that can
operate with a small ground plane.
5 .FUTURE SCOPE
C band is a name given to certain portions of the

electromagnetic spectrum, including wavelengths of
microwaves that are used for long-distance radio
telecommunications. The IEEE C-band (4 GHz to
8 GHz) - and its slight variations - contains frequency
ranges that are used for many satellite
communications transmissions, some Wi-Fi devices,
some cordless telephones, and some weather radar
Fig 7. 3D Geometry view of the the patch systems. For satellite communications, the
microwave frequencies of the C-band perform better
under adverse weather conditions in comparison with
Ku band (11.2 GHz to 14.5 GHz) microwave
frequencies.
REFERENCES
rd
[1]. Antenna Engineering Handbook part 1 &2 (3
edition) by Richard C. Johnson.
[2]. Constantine. A Balanis, Antenna Theory and Design,
Second edition, 1997, John Wiley Publications.
[3]. Antenna and Wave propagation by K.D. Prasad
[4]. J.D Krauss and Ronald J Marhefka, Antennas for all
applications, 2000,Third Edition
Fig 8. Frequency properties of the patch [5]. Microwave Engineering by D.M.Pozar
[6]. CAD of Microstrip Antennas for Wireless
Applications by Robert A.Sainati
[7]. Handbook of Microstrip Antennas, Volume-1 by
J R James & P S Hall.
[8]. Radar Handbook, Second Edition by Merrilli. Skolnik
[9]. IE3D Users manual , Release 9, Zeland software.
[10]. Microstrip antenna using MStrip 40 Lab
Manual by Mr. Martin Leung.
[11] I.Govardhani , M.Venkata Narayana Published paper in
International Journal of Computer Science & Communication
Networks,Vol 2(1), 375-380, ISSN:2249-5789 on Rectangular
Patch Array Antenna with Liquid Crystal Substrate for Ka and Q
Band Applications
[12] Dr.K.S.N Murthy, Venkata Raviteja.K, I.Govardhani,
M.Venkata Narayana Published a paper on Multi-band Ladder-
shape Microstrip Patch Antenna IJSER Volume 3, Issue 3.
Fig 9. Gain vs Frequency display
417
The Dielectric Loss Tangent Effect on Radiation

Characteristics of Co-axially Feed Rectangular
Patch Antenna: Design & Analysis
Ch. Raja1, K. Praveen Kumar2, Dr. M. Madhavi Latha3, Dr.E.G.Rajan4
1
Research Scholar, Dept of ECE, JNTU, Hyderabad, A.P, India, chraja@mgit.ac.in
2
Research Scholar, Dept of ECE, JNTU, Hyderabad, A.P, India,
3
Director, Innovative Technology Center, JNTU, Hyderabad, A.P, India, mlmakkena@yahoo.com
4
Director, Pentagram Research Center, Hyderabad, A.P, India,
Abstract - Selection of proper substrate material is prime

important task in microstrip patch antenna design.
Because of the limitations of micro strip antenna such
as low gain, low efficiency and high return loss can
In microwave engineering the materials are defined
overcome by selecting proper substrate materials,
because permittivity of substrate is critical parameter in with complex permittivity
controlling band width, efficiency, and radiation pattern
of patch antenna. But the substrate materials have two
basic properties such as dielectric constant and loss
tangent. In this Present paper comprehensive study of
various dielectric materials of same permittivity at
different loss tangent and its effect on radiation
characteristics of rectangular patch antenna are for loss less materials there is no loss tangent
investigated. Results such as resonance frequency, (Tan=0) then the permittivity is real (= r0)[8]. The
bandwidth, gain, return loss, input and impedance are structure of rectangular micro strip antenna in its
presented. simplest form consisting of sandwich of two
Keywords - Dielectric constant, patch, loss tangent.
conducting layers separated by single thin dielectric
substrate is considered, where lower conductor
I.INTRODUCTION
function as ground plane and upper conductor
function as radiator. Larger ground plane gives better
Generally a dielectric substrate is defined by its two
performance but makes the antenna size bigger. This
prime parameters one is its permittivity (It describes
structure is excited with coaxial feed. At resonate
the materials with high polarizability) and another is
frequency of 1.7616GHz. Present paper different
loss tangent (It explains the dissipation of
dielectric materials of same permittivity with
electromagnetic energy), =r0(1-JTan). Loss
different loss tangent were considered. Height of
tangent includes dielectric damping loss and
substrate 1.56mm taken common for all simulations.
conductivity loss of material and it is a frequency
dependant.
II. MICROSTRIP PATCH ANTENNA DESIGN
For a material with conductor loss and dielectric
damping loss can be given as [7]
Fig 1: Rectangular Patch Antenna

The imaginary part of above equation explains The structure of rectangular patch antenna consists of
loss of material. Including dielectric damping loss dielectric substrate backed with optically plannar
reflecting metal ground plane, on other side consists
, and the conductive loss .
of radiating element.
418
The length of patch (L) is about g/2 (g is effective III.CO-AXIAL PROBE FEED
wave length) and substrate height(h) is of order of
g/20.Due to very small space between radiating This is a common feed technique here outer
element and ground plane main power is radiated conductor is connected to ground plane and the inner
towards broad side. The fringing fields effectively conductor of co-axial connector is extends through
increase the length(L) of patch need to be accounted dielectric and soldered to patch. Inner conductor of
in determine resonance frequency[1]. co-axial cable transfers the power from strip line to
microstrip antenna from slot in the ground plane.
Placing of feed position is important in order to have
best matching with input impedance. Here fees is
applied at (0mm, 9.2mm). It provides narrow
bandwidth performance and it is difficult to design
for thick substrate.[1]
IV. DIELECTRIC MATERIALS
Fig 2: Electrical Field Lines(Side View) Present paper we have considered different dielectric
materials with same permittivity and different loss
The most commonly used design equations of tangent are listed bellow.
antenna
a) Effectivedielectricconstant TABLE I
DIELECTRIC LOSS TANGENT
Material Name Permittivity Loss
Tangent
b) Lengthextensionis
Rubber_hard 3 0
Roger RO3003 3 0.0013
Arlon AD300A 3 0.002
Neltec NH 9300 3 0.0023
c) EffectiveLength Rogers Ultralam 3 0.003
1300
d) Actuallengthofpatch
V. NUMERICAL DESIGN
e) Patchwidth
The proposed co-axial feed rectangular micro strip
patch antenna operating at 1.7616GHz having
physical dimensions of radiating patch length
f) Groundplanedimensions L=47.4mm along Y-axis, width W=56.5mm along X-
axis (here width of patch is maintained higher than
length because of wider operating band width). The
co-axial feed is provided at 0mm, 9.2mm along X, Y-
axis respectively. The substrate height is 1.56mm.
Fig 3: Physical & Effective Lengths of Patch
419
Fig 6 : Return loss
TABLE II
RETURN LOSS
Fig 4: Coaxial feed micro strip antenna Substrate Return loss Operating
Frequency(GHz)
Rubber_hard -18.1618 1.7616
VI. SIMULATION SETUP Roger RO3003 -20.9417 1.7616
Arlon AD300A -22.5992 1.7616
Current proposal of coaxial feed rectangular micro
Neltec NH 9300 -23.3269 1.7616
strip antenna. And results of return loss, gain and
Rogers Ultralam -24.8394 1.7616
band width were presented. The Ansoft HFSS
1300
software is utilized here for execution of
One thing we can say that as the loss tangent of
dielectric material increases the return loss value also
decreasing. Indicates more amount of power is
forwarded and very less amount of power is reflected
back.
B. Bandwidth:
It can be defined as the range of frequencies over
which gain is constant. In software simulation this
value is taken at intersecting point s of -10db line and
return loss curve.
TABLE III
BAND WIDTH
Fig 5: Simulated Antenna Model
Substrate Band width Operating
(MHz) Frequency(GHz)
VII. RESULTS & DISCUSSION
Rubber_hard 22.5 1.7616
A. Return loss Roger RO3003 22.5 1.7616
Arlon AD300A 22.5 1.7616
According to maximum power transfer theorem, Neltec NH 9300 22.5 1.7616
maximum amount of power will be transferred when Rogers Ultralam 22.5 1.7616
there is a perfect matching between input and output. 1300
If load is mismatched the whole power is not
delivered to load and there is a return of power that is The loss tangent of substrate does not effect the
called loss, since this loss is returned hence is called operating bandwidth of antenna.
return loss is -20log. Where is reflection C.Impedance:
coefficient. The response of magnitude of S11 verses
frequency curve clearly explains return loss Here only one impedance diagram is shown which
indicates all the substrate materials have same
420
impedance, means loss tangent is not effected the Accepted 0.009 0.009 0.009 0.009 0.009
impedance. Power 7394 8149 8422 8515 867
Incident 0.009 0.009 0.009 0.009 0.009
Power 9012 9012 9012 9012 9012
Radiation 1.002 0.930 0.895 0.881 0.849
Efficiency 3 23 53 39 91
FBR 26.57 26.57 26.57 26.57 26.43
3 7 2
By careful observation of antenna parameters given

in above table, the antenna parameters (such as gain,
directivity and radiation efficiency) are decreasing as
the loss tangent of substrate material is increasing.
IX. CONCLUSIONS
The dielectric materials provide mechanical strength

to patch antenna design. To have effective radiation
characteristics it is better to consider a dielectric
material with a low loss tangent. But loss tangent will
not affect the operating band width.
Fig 7: Impedance
ACKNOWLEDGMENT
D. Gain
The Authors want to thank management of
Gain explains figure of merit of antenna which Mahatma Gandhi Institute of Technology,
combines antennas directivity and electrical Hyderabad, for their support and encouragement
efficiency. Since antenna is a passive device the gain
cannot be measured directly. Gain is a measure of REFERENCES
power radiated per unit surface area by the test
[1] Constantine A. Balanis; Antenna Theory, Analysis and
antenna in a given direction at an arbitrary distance, Design, John Wiley & Sons Inc. 2ndedition. 1997.
to hypothetical isotropic antenna.
[2] Y.T. Lo. and S.W. Lee, editors, Antenna Handbook Theory,
VIII. ANTENNA Parameters Applications and Design, Van Nostrand Reinhold Company,
New York, 1988.
TABLE IV [3] Stutzman Warren L. and Thiele Antennas and propagation

ANTENNA PARAMETERS Magazine, vol.52, Feb 2010)
Parameter/ Rubb Roge Arlon Nelte Roge [4] Broadband Microstrip Antennas, Girish Kumar and K. P.
Ray, Artech House, 2002.
Substrate er r AD3 c NH rs
Hard RO3 00A 9300 Ultim [5] Daniel H. Schaubert, A review of Some Microstrip Antenna
003 a Characteristics Microstrip Antennas - The Analysis and
1300 Design of Microstrip Antennas and Arrays, edited by David
M. Pozar, Daniel H. Schaubert, John Wiley & Sons, Inc.,
Max U 0.004 0.003 0.003 0.003 0.003 1995, ISBN 0-7803-1078-0.
0633 7989 6672 6126 4885
(w/sr (w/sr [6] A Derneryd, Linearly Polarized Microstrip Antennas, IEEE
Trans. Antennas and Propagation, AP-24, pp. 846-851, 1976
) )
Peak 5.231 5.228 5.228 5.228 5.227 [7] D.M.Pozar Microwave Engineering, 3rd ed. NewYork, John
Directivity 8 5 4 6 Wily & Sons, 1998.
Peak Gain 5.242 4.864 4.682 4.608 4.443
[8] J.A.Kong, Electromagnetic wave theory., EMW Publishing,
8 3 2 2008, pp 268-272.
Peak 5.157 4.821 4.654 4.585 4.427
Realized 1 6 4 1 6
Gain
Radiated 0.009 0.009 0.008 0.008 0.008
Power 7641 1301 814 683 386
421
422
A video watermarking technique using

Discrete Cosine Transform
Dokuparthi Jagadeesh , Dr.K.Gauthami
Abstract: In this paper, we present a video requires the use of the original image in the extraction
watermarking algorithm to resist the video attacks and process then it is called Non-Blind watermarking
noise attacks.. Each video is divided in to frames. Each technique. The watermarking scheme requires the
frame is subdivided into non-overlapping 8*8 blocks.
watermark data and/or the parameters used to embed
Each block is transformed 2D-DCT. After performing
the same transformation to all the blocks and frames, the data, then it is called Semi-Blind watermarking.
1D DCT is applied along time axis. Now, the
quantization index modulation is used to embed the In this paper, we present an invisible watermarking
secret logo. Embedding is performed to DCT technique. The watermarking logo is embedding is
intermediate frequency coefficients of each block. The transform domain of the transmitted video. There are
DCT intermediate coefficients are adjusted by using a so many transformation techniques (DCT, DWT,
proposed algorithm that can resist the attack DFT, DHT etc). We use a discrete cosine
transform and quantization technique. The DCT gives
Index Terms Discrete cosine Transform, SQM, good PSNR.
Resistance to attacks
II. DCT
1. INTRODUCTION The Discrete Cosine Transform is similar to the

Fourier Transform in that it transforms a signal from
the spatial or time domain to the frequency domain.
Internet is the major source of exchange the
Just as the Fourier Transform can be completed in a
multimedia information. Watremarking is the
smaller amount of calculations by use of the FFT, the
powerful technique to prevent the unauthorised
complexity of calculations needed to use the DCT
duplication. In this technique, owner information (in
the form of logo) is embedded to the original video also can be reduced. A method called the Fast Cosine
signal. Transform, or FCT, can be used when N = 2q, where
N is the number of vectors to be transformed and q is
Digital watermarking is the process of protecting an element of the integers, and the complexity is
the illegal copying of data. Steganography and reduced, just as with the FFT, from N2 calculations to
watermarking are important techniques to hide a number of calculations on the order of Nlog2N.
important information in an undetectable and/or The FFT is actually the foundation for calculating the
irremovable way in audio,image and video data. FCT.
Steganography and watermarking are main parts of
the fast developing area of information hiding. so The 2D discrete cosine transform (DCT) represents
many video watermarking algorithms are developed. an image as a sum of sinusoids of varying
But the issue is to resist the noise attack, filtering magnitudes and frequencies. The DCT has the
attacks. Our proposed algorithm resists the filtering
property that, for a typical image, most of the visually
noise and compression attacks and preserves the
quality of the video. significant information about the image is
concentrated in few coefficients. For this reason, the
There are two types of watermarking i) Visible DCT is used in JPEG compression. DCT is at the
watermarking ii) Invisible watermarking. heart of the international standard lossy image
compression algorithm known as JPEG (Joint
Based on the algorithm used, watermarking is
classified in to three types. i) Blind watermarking Photographic Experts Group). There are two types of
ii)Semi-Blind watermarking iii)Non-blind DCT domain watermarking techniques. One is Global
watermarking. If the watermarking scheme does not DCT watermarking, other is Block based DCT
require the original image or any other data in watermarking. In this paper block based DCT
extraction process, then it is called Blind watermarking is used. In this method, a 2D-DCT and
watermarking Technique. The watermarking scheme 1D DCT is applied to the frames. Each frame is
423
divided in to non- overlapping 8*8 blocks. 2D-DCT (2 y 1)v

is applied to each block and the same is repeated to cos (Eq 4)
each frame in a video using the following the 2N
equations.
For x, y = 0, 1, 2 N-1, where is given as
Mathematical Representation
The 1-D discrete cosine transform (DCT) is defined 1
foru 0
as N
(u )
2
foru 1,2,...... N 1
N 1
(2 x 1)u
C (u ) (u ) f ( x ) cos ( Eq 1)
N
x 0 2N
1
forv 0
N
For u = 0, 1, 2 N-1, similarly, the inverse DCT is (v )
defined as 2
forv 1,2,......N 1
N
N 1
(2 x 1)u
f ( x ) (u )C (u ) cos (Eq Quantization Index Modulation
u 0 2N
2) In this paper, Quantization is performed after
dividing with standard quantization matrix. Two
For x = 0, 1, 2 N-1. In both the equations (2.17) different quantizes are used. One for embedding the
watermark bit 1; other is for embedding 0.Here,
and (2.18), is the embedding is performed in the compressed
domain to resist the attacks.
1
foru 0 IV PROPOSED SCHEME
N
(u )
Pseudo-3D DCT Technique:
2
foru 1,2,......N 1
N The DCT transformation is taken twice here. The
first-performed DCT is a 2-D transformation applied
to the image plane. The second-performed DCT can
The corresponding 2-D DCT pair is be regarded as a 1-D transformation. Based on this
relationship, we regard the proposed embedding
method as a pseudo-3-D DCT. The main advantage
N 1 N 1 ( 2 x 1)u of the pseudo-3-D DCT is that it can reduce the
f ( x, y ) cos
C (u, v ) (u ) ( v ) 2N computational cost more than the traditional 3-D
x 0 y 0 DCT does.
(2 y 1)v
cos (EQ 3)
2N
For u, v = 0, 1, 2 N-1 and the values C (u, v) are

called the DCT coefficients of f(x, y).
N 1 N 1
( 2 x 1)u
f ( x, y ) (u ) ( v )C (u, v ) cos Fig .1 Illustration of the Proposed DCT Method
u 0 v 0 2N
424
A video is divided into group of frames. Watermark extraction:

Each frame in a RGB video is divided in to
luminance and chroma components. The watermark 1) Divide the watermarked video in to a
logo is embedded in luminance component. In this, number of frames and converted RGB to
the transformation is performed in two steps. In the YCbCr.
first step, each frame is divided in to non overlapping 2) Apply the 2D-DCT and 1D-DCT to as
8*8 blocks. Each block is transformed in to DCT
mentioned above in the embedding process.
domain by using 2D discrete cosine transform
equation. In the second step, one dimensional discrete (i.e. steps 2 to 7 of embedding process).
cosine transform is applied on time axis. i.e. The 3) if ((A_ex(5,5,1)))>=A_ex(4,4,1)
corresponding pixels of each frame is transformed w_e(k)=0;else
using one dimensional DCT. w_e(k)=1;end
Watermarking embedding and extracting:
Watermarking embedding:
1) Divide the video into frames

2) Convert the each frame from RGB to Y Cb
Cr color space.
3) Each frame is divided into 8*8 blocks
4) Apply 2D-DCT to each block. Repeat this
to all the frames.
5) Apply 1D-DCT to (1, 1) coefficient of first
block of each frame.
6) Perform the same to (1, 2) coefficient of
first block of each frame.
7) Repeat the same till (8, 8) coefficient is
reached.
8) Repeat the steps 4, 5 and 6 till all the blocks
are finished.
9) Now, each block is dividing with standard
quantization matrix.
10) One bit of watermark is to be embedded in Fig .2 Block diagrams of Embedding Technique
each block.
Choose any two mid frequency coefficient
resultant matrix and modify the
coefficients according to the following
condition.
if w(k)==1 A (4, 4, 1) = (A(4, 4, 1) + k);
end
if w(k)==0 A (4, 4, 1) = (A (4, 4, 1) - k); Fig. 3 Public Channel
End
11) Multiply the modified blocks with standard
quantization matrix.
12) Apply the inverse 1D IDCT along the time
axis.
13) Perform the block based inverse 2D IDCT
to each frame.
14) Convert the each frame from Y Cb Cr to
RGB color space.
15) Write the all the frames in to the AVI
format.
Fig.4 Block diagram of Extraction Technique
425
Resultant 8*8 matrix after dividing pseudo 3D-

DCT coefficients with standard quantization matrix Fig. 7 Watermarked Video
Performance analysis
Experimental Results:
The experiment is performed on a video of

size 240*320 and the number of frames are 50.The
logo size is 32*32.The controlling parameter k value
0.9 is selected. The experiment is performed in Fig. 8 Extracted Logo (0.9978)
MATLAB 2012b.The PSNR of proposed method is
47.05db. Extracted Logos after applying Attacks:
Table 1
S.No Type of Attack NCC
1. Salt & Pepper Noise 0.97

(Density=0.01)
Salt & Pepper Noise(d=0.01)
2. Video resize 0.85
3. Frame Removal 0.98
4. Frame Averaging 0.99
5. Gaussian Noise 0.983

(mean=0,variance=0.001)
Video resize
Fig. 5 Original Watermark logo
Frame removal
Fig. 6 Original video Frame average
426
[9] Multidimensional signal,image, and video processing and

coding john w. woods rensselaer polytechnic institute troy, new
York.
About Authors:-
D.Jagadeesh currently working as a Assistant professor in

Vignans Nirula institute of Technology and science for women,
Gaussian noise (m=0, variance=0.001) Palakaluru, Guntur. He completed M.Tech in Bapatla engineering
college, Bapatla. He completed B.Tech in Vignans Engineering
College, vadlamudi. He has three years of teaching experience.
CONCLUSIONS Dr.K.Gauthami currently working as a professor and ECE HOD

in Vignans Nirula institute of Technology and science for women,
The proposed method gives good results for noise Palakaluru, Guntur. She received Ph.D from Acharya Nagarjuna
attacks and filtering attacks. For compression attacks, University in 2011. She has 18 years of teaching experience.
the extracted logo is slightly disturbed, but it is easy
to recognize. This scheme is not robust against
MPEG compression attacks. In future, the algorithm
can be improved to resist these attacks.
REFERENCES
[1] A Video Watermarking Technique Based on Pseudo-3-D DCT

and Quantization Index Modulation Hui-Yu Huang, Member,
IEEE, Cheng-Han Yang, and Wen-Hsing Hsu IEEE transactions
on information forensics and security, vol. 5, no. 4, December
2010
[2] A MPEG2 Video Watermarking Algorithm Based on DCT

Domain Lu Jianfeng; Yang Zhenhua;Yang Fan; Li li Institute of
Computer Graphics and Image Processing Hangzhou Dianzi
University Hangzhou, China 978-0-7695-4413-7/11 $26.00
2011 IEEE Computer society
[3] Multipurpose Watermarking for Image Authentication and

Protection; Chun-Shien Lu, Hong-Yuan Mark Liao ieee
transactions on image processing, vol. 10, no. 10, october 2001
[4] Ye Dengpan, Dai Yuewei, Wang Zhiquan, A Survey of Video

Watermarking Computer Engineering and Applications. 2005,
41(1),pp. 14 18
[5] Hartung F, Girod B. Digital watermarking of MPEG-2 coded

video in the bit stream domain [A]. In: IEEE International
Conference on Acoustic, Speech, and Signal Processing, Munich,
Germany, 1997. 2621~2624.
[6] ZHENG Zhen-dong, WANG Pei, LI Li, A Hybrid Video

Watermarking Scheme Based on Motion Vectors and DCT,
Journal of Image and Graphics.2009, 14(12), pp.2631-2634.
[7] LIU Hong, LIU QUAN. Novel MPEG-2 Compression

Domain-base Video Digital Watermarking Algorithm, Journal of
Wuhan University of Technology.2008, 30(7), pp.134-137.
[8] Errata for VIDEO PROCESSING AND COMMUNICATIONS

Yao Wang, Joern Ostermann, and Ya-Qin Zhang.
427
A Novel Approach for Denoising of Medical

Images corrupted with Poisson Noise
V. Naga Prudhvi Raj, Dr.T.Venkateswarlu
AbstractX-ray imaging and Nuclear Imaging (PET, very important to remove the noise from these
SPECT) are helpful in medical diagnosis. The images images.
acquired from these modalities are suffering from In medical Image processing applications image
data dependent noise named as Poisson noise. The
presence of noise will hide the diagnostic details in
denoising is the important preprocessing task before
the images. So the images corrupted with the noise performing image analysis tasks such as image
is not suitable for automatic analysis by the segmentation, image feature extraction, image
computers. Many algorithms in the literature are registration, storage and image retrieval etc. The
using Variance Stabilization Transforms (VST) to noise present in the images will lead to wrong
convert the Poisson noise into Gaussian noise and diagnois and create problems in image guided
then apply the denoising algorithms developed for surgery. So the signal to noise ratio (SNR) of the
Gaussian noise on the transformed image. This method
is suitable when the image is having high photon count images must be improved before performing image
i.e. the noise level is low. In the low photon count analysis tasks. The major challenge in designing the
situations i.e. when the noise level is high, this denoising algorithms is removing the noise
approximation will not hold good. In this paper we are effectively without removing the image features
proposing a novel approach based on Blockwise Non- such as edges, lines etc. from the images [1].
Local (NL-) means filter, Variance Stabilization
Transform and Dual Tree Comlex Wavelet Transform
(DTCWT) for images corrupted with poisson noise.
The noise may be additive or multiplicative
Quantitative validation of the proposed method was depending on the modality used for medical
carried out using several quality metrics. The results image acquisition. The noise due to electronic
show that the proposed filter performed well than the components in the acquisition hardware will be
recently proposed filters based on Discrete Wavelet modeled with Gaussian noise which independent
Transform, NL-means filters. The proposed filter
removes noise effectively while preserving fine
of data, the data dependent noise such as quantum
structures such as edges, lines etc. even for very noisy noise in X-ray imaging is modeled with Poisson
cases. distribution, the speckle noise in ultrasound imaging
is modeled with Rayleigh distribution and the noise
Index TermsDualtree Complex Wavelet in MRI is modeled with Rician distribution. Here in
Transform, NL- means Filtering,Variance Stabilization this paper we are attempting to denoise the images
Transform,Universal Quality Index (UQI), Structural corrupted with quantum noise.
Similarity Index (SSIM)
II. POISSON NOISE
I.INTRODUCTION
The noise in X-ray imaging and Nuclear Imaging
Medical imaging became an integral part of medical (PET, SPECT) is modeled with Poisson noise. X-ray
diagnosis in present days. Various medical imaging photons incident on a receptor surface in a
modalities are developed for various applications
random pattern. We cannot force them to be evenly
since last few decades. These modalities are used, to
distributed over the receptor surface. One area of
acquire the images of the anatomical structures within
the receptor surface may receive more photons
the body to be examined, without opening the body.
than another area, even when both the areas are
X-rays, Computed Tomography, Ultrasound, Magnetic
resonance Imaging and Nuclear imaging are the exposed to the same average x-ray intensity. In all
popular modalities at present to diagnose the various medical imaging procedures using gamma or x-ray
diseases. However these modalities are suffering with photons most of the image noise is produced by the
a big problem called noise. Every modality is random behaviour of the photons that are distributed
suffering from noise in image acquisition and within the image. This is generally designated
transmission stage such as Quantum noise in X- quantum noise. Each individual photon is a quantum
rays and Nuclear imaging, speckle noise in ultrasound (specific quantity) of energy. It is the quantum
imaging, Rician noise in Magnetic resonance structure of an x-ray beam that creates quantum
imaging etc. The noise present in the images will noise.
degrade the contrast of the image and creates A Poisson model assume that each pixel x of an
problems in the diagnostic phase. So denoising is image f (x) is drawn from a Poisson distribution of
428
parameter = f0 (x) where f0 is the original

image to recover. The Poisson density Is given as In statistics the variance stabilization transforms are
used to convert the distributions with variable
variance into distribution with constant variance.
These transforms are useful to convert the Poisson
distribution and Rician distributions into Gaussian
distribution with constant variance. The noise in the
X-ray imaging and nuclear medical imaging such
as PET and SPECT is modeled with Poisson
distribution because its variance depends on the
mean value and this makes difficult the denoising
task. Here in this work we want to use the
variance stabilization transforms as a preprocessing
stage for the denoising filter to make the Poisson
distribution into Gaussian distribution with known
variance and then we perform the denoising
operation.
The Fig.1. gives the Poisson distribution for various After the preprocessing we will apply the inverse
values of as the increases the Poisson distribution
turns towards the Gaussian distribution variance stabilization transform to get back the
original distributions. In this work we are using
The denoising problem was attempted by many Anscombe transforms. The mathematical
researchers from many directions such as representation of the transform is given below. x
spatial filtering using local statistics, Transform is the input image pixel, Z is the transformed
domain approaches (Multiscale & Geometrical pixel and I is the inverse transform of the
transforms) [2], statistical modeling of transformed pixel.
images(Gaussian Scale Mixtures, Hidden Morkov
models etc.), Scale space theory and dictionary based IV. NON LOCAL MEANS FILTERING
approaches [3],[4]. Recently mathematical image
processing techniques such as Partial differential The nonlocal means filtering is also considered
equations (PDEs), Calculus of variations, Kernel as the extension of the fundamental neighborhood
Regression and patch based algorithms became
filter (Yaroslavsky filter) where the pixel intensity is
effective in solving the denoising problem [5], [6].
decided by averaging the intensities of its
The spatial filtering approaches were easy to
neighborhood [1],[16]. Here in this method the
implement but tend to remove image features along
pixels which are having similar neighborhood are
with noise. The image structures were preserved by
the algorithms based on Wavelet Transforms, PDE found and averaged in deciding the intensity of a
based methods such as anisotropic diffusion [7], Total pixel. Normally similarity between the pixels is
Variation problem of Poisson denoising is found by observing their intensities, if the two
transformed into a classical Gaussian denoising pixels having the same intensity we can say that
problem by applying a VST, a large variety of they are similar otherwise they are different.
standard denoising approaches can be used. In the Here in the nonlocal means algorithm the similarity
case of very low SNR images, the two major is found by considering the neighborhoods of the
problems are: VST robustness in low count situations pixels. Two pixels are similar when the
and the efficiency of the selected denoising filter neighborhoods of the two pixels are having same
under very noisy conditions. intensities. So we can define the neighborhood of
pixel i is a set of pixels j whose
III. VARIANCE STABILIZATION neighborhoods are similar to the neighborhood of
TRANSFORM i. Fig.2. shows the similarity windows in the image
429
Let is the area of an image I , x is the The weight can be defined as

location inside ,u(x) and v(x) are the clean and
observed noisy image value at the location x
respectively. Then the nonlocal algorithm is given as
follows
Z (i) is the normalization constant to ensure that

w(i, j) =1 and h is the smoothing parameter. As
the main drawback ofthe NL-means filter is its
computational time for multidimensional data we
are using the block wise approach proposed in [6] to
speed up the computation. This approach will
restore the entire pixels in the patch instead of
restoring the pixel by pixel. The technical details
and detailed explanation of this approach can be
Fig. 2. Self similarity in an Image (Fundamental found in [17].
to Patch based Algorithms)
V. DUALTREE COMPLEX WAVELET
Where Ga is a Gaussian function with standard TRANSFORM
deviationa and h is a filtering parameter
Since past two decades the DWT (Discrete wavelet
Trans- form) was used widely in the image
processing tasks by replacing the sinusoidal basis
In the discrete case the non-local means algorithm functions (Fourier Transform)which were infinetely
is given as oscillating with the locally oscillating basis
functions. Due to this feature the wavelets are good
in local- ising the discontinuties in the image.
However these critically decimated DWT based
methods suffer from four shortcomings
Where the weight w(i, j) depends on the distance 1) Oscillations around singularities 2) Shift variance
between observed gray level vectors at points i and 3) Alias- ing 4) Lack of directionality. Detailed
j . The distance can be calculated as explanation about the above drawbacks can be found
in. To overcome this drawbacks a redundant wavelet
transforms were developed recently. In this paper
we are using the 3D dual tree complex wavelet
transform (DTCWT)in our image fusion
Ni is the neighborhood of the pixel i which is
framework. Fig.3. presents the analysis section of
the squared window around the pixel i. Every pixel
the 3D Dual tree complex wavelet transform.
in the image will have their own neighborhood. The
size of the neighborhood can be varied depending
VI. PROPOSED FILTERING ALGORITHM
on the application. Now we can define N Si the
neighborhood system of a pixel i . It is a set of
The blockdiagram of the proposed method is given
similar neighborhoods to the neighborhoods of i
in Fig.4. In the proposed method first, a copy of
(Ni ). The similarity between the neighborhoods is
the noisy image is pre-filtered using a median
calculated by calculating the difference in the
filter. Median filtering was chosen for its
intensity gray levels in the neighborhoods as shown
robustness, its low computational burden and
in the above eq.(8). The pixels
because it works directly on the Poisson noise,
i and j are said to be similar if v(Ni ) is similar before VST. In this fashion, the instability of VST
to v(Nj ) under low SNR conditions can be sidestepped.
430
The original noisy image and the median

filtered image are then transformed with the
Anscombe Transform (AT) which is fast, simple
and does not require any parameter estimation.
Afterwards, local noise levels are estimated within
the stabilized noisy image by using the MAD
estimator in the dual tree complex wavelet transform
domain. This approach is widely used for
Gaussian noise estimation due to its robustness The
core of the proposed approach is the Non Local
Means filtering. In the proposed method, we
used the estimation of patch similarity over the
median filtered image, instead of using the noisy Fig. 4. Proposed Denoising
input image. By reducing the noise, we provide a method
more robust patch comparison that enables a better
restoration. It is important to note that the weights VII. IMAGE QUALITY
estimated from the median filtered image, are used METRICS
to average the noisy input image. The median
filtered image is not used in other steps. Next, we To evaluate the quality of the image processing
estimated the local noise level by using the algorithms there are several metric sproposed in the
multiscale transform such as DTCWT and Median literature. The metrics are classied as pixel
Absolute Deviation(MAD) estimator. The estimated difference based measures, correlation based
local noise level is used as the smoothing parameter measures, edge based measures, spectral distance
of the optimized Non local means filter for each measures, context based measures and Human visual
voxel and thus optimally controls the amount of system based measures. Here we are comparing our
denoising over the image. Finally, an optimal in- denoising algorithms using a group of metrics drawn
verse Anscombe transform is used to obtain the from the above class and performance of the
final denoised image, without intensity bias under algorithms was observed.
very low SNR conditions.
A. PSNR (Peak Signal to Noise Ratio)
PSNR is the peak signal-to-noise ratio in decibels

(dB). The PSNR is only meaningful for data
encoded in terms of bits per sample, or bits per
pixel. For example, an image with 8 bits per pixel
contains integers from 0 to 255.
Where B represents bits per sample and MSE

(Mean Squared error) is the mean square error
between f(x,y) and an approximation f(x, y) is the
squared norm of difference divided by number of
elements in the signal.
MSE and RMSE measures the difference between

the original and distorted sequences. PSNR
Fig. 3. 3D Dual Tree complex Wavelet measures the fidelity i.e how close a sequence is
Transform Decomposition similar to an original one.
431
B.Universal image quality index (UQI) words, the measuring of image quality based on an
initial uncompressed or distortion-free image as
It is a measure used to find the image reference. SSIM is designed to improve on traditional
distortion. It is mathematically defined by making methods like peak signal-to-noise ratio (PSNR)
the image distortion relative to the reference image and mean squared error (MSE), which have proved to
as a combination of three factors: Loss of be inconsistent with human eye perception.
correlation, Luminance distortion and contrast
distortion. This measure was developed by Alan
C.Bovik. If two images f (x, y) and g(x, y) are
considered as a matrices with M column and N
rows containing pixel values f (x, y) and g(x, y)
respectively the universal image quality index Q
may be calculated as a product of three
components
VIII. RESULTS & DISCUSSIONS
The performance of the proposed filter is carried

out with the DWT filter, Non-Local means Filter
and Block matching 3D(BM3D) filter. The scores of
the metrics for all the filters were tabulated in the
Table II and the denoised images of the filters are
shown in the Fig.4, Fig.5 and Fig.6.
It is observed that the there is a considerable

improvement in the Peak signal to noise ratio
which indicates the efficiency of filters while
performing denoising. The another measure structural
The first component is the correlation coefficient similarity is used to verify the structure preserving
which measures the degree of correlation of two nature of the denoising filters.
images. and its dynamic range is [-1,1]. The best
value 1 is obtained when the images are linearly
related. Even if the two images are linearly related, If SSIM value is close to zero it indicates that there is
there still might be relative distortions between them, a considerable loss in the structural information. That
which are evaluated in the second and third is the filter is removing the structural details such as
components. The second component measures how edges, lines, etc. while performing the denoising. If
close the mean luminance is between images with SSIM value is close to one, it indicates the structure
a range [0,1]. It equals 1 if and only if f = g. f preserving nature of the denoising filter. The results
and g can be viewed as estimate of the contrast of show that the proposed algorithm is removing more
f and g. So The third component measures the amount of noise than the existing filters while
contrasts of the images, the range for this component preserving the underlying structures in the images.
is [0,1]. The best value 1 is achieved if and only if f
= g . The range of values for Q is [-1,1]. The best
value1 is achieved if and only if the images are
identical.
c. Structural similarity (SSIM) :
Structural similarity (SSIM) index is a method for

measuring the similarity between two images [18].
The SSIM index is a full reference metric, in other
432
means algorithm.
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algorithms, with a new one, Multiscale Modeling & Simulation,
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This proposed procedure improves the denoising and x-ray images using variance stabilization and patch based
performance compared to the NL-means filter. The algorithms, The International Journal of Multimedia & Its
estimation of patch similarity over the median Applications (IJMA), vol. 4, no. 6, pp. 5371, 2012.
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filtered image, instead of using the noisy input image
Multiscale variance-stabilizing transform for mixed-poisson-
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433
An Efficient Data Mining Tool for Medical Image

Diagnosis Using soft computing Technique
G.Madhavi
Assosiate professor , dept of ECE, MGIT,Hyderabad
Abstract: . In medical and health care areas, because of such as scans, different types of X-rays, MRI images
regulations and the availability of computers, a large and other radiological techniques. With all this
amount of data is becoming available .Data mining is a effort done in the research field, there is still a lot of
relative important objective to acquire knowledge from
large amounts of data. . On one side, practitioners are
place for improvements and the medical image
expected to use all this data in their work but, in a short processing is a domain in continuous expansion.
time it is difficult to process such a large amount of data Data mining is the non trivial extraction of implicit
by humans to make diagnosis, prognosis and treatment previously unknown and potentially
schedules. Evaluateing data mining tools in health care useful information about data[1]. Data mining
and medical applications to make timely and accurate technology provides a user- oriented approach to
decision is the aim of this work . The conventional method novel and hidden patterns in the data.
in medicine for brain MR images classification and tumor
detection is by human inspection. Operator-assisted The discovered knowledge can be used by the
classification methods are impractical for large amounts healthcare administrators to improve the quality of
of data and are also non-reproducible. MR images service. The discovered knowledge can also be used
also always contain a noise due to operator performance by the medical practitioners to reduce the number of
which can lead to serious errors in classification. The use adverse drug effect, to suggest less expensive
of artificial intelligent techniques , such as neural therapeutically equivalent alternatives.
networks, fuzzy logic, neuro fuzzy have play important
role in this field. In this paper we are going to consider
image registration technique as soft computing technique. II. DISCOVERY OF KNOWLEDGE FROM
Here we will consider geometry of the image and process MEDICAL DATABASES
the images. The dominant features which we got from
registration scheme are given to neural network for Data mining is an important aspect for acquiring
robust training and appropriate result analysis on medical knowledge . In Information now a days it has
images. attrected great deal of interest . [13]. Knowledge
Key Word: Treatment schedules, health care application, discovery process consists of an iterative sequence
MR images, tumor detection, soft computing technique, of, data selection data cleaning, data integration,
registration scheme data mining pattern recognition and knowledge
presentation. In particulars, data mining may
I. INTRODUCTION accomplish class description, association,
classification, clustering, prediction and time series
Automated classification and detection of tumors in analysis.
different medical images demands high accuracy
since it deals with human life. Also, computer
assistance is highly sought in medical institutions
due to the fact that it could improve the results of
humans in such a domain where the false negative
cases must be at a very low rate. It has been proven
that double reading of medical images could lead to
better tumor detection. But the cost incurred in
double reading is very high; therefore good software
to assist humans in medical institutions is of great
interest nowadays. Different approaches that can
produce medical images must be studied. Also, the
technique that produces those Fig. 1: Data extracting cycle
images is very important in order to know what Data mining in contrast to traditional data analysis is
to apply to a certain medical image in order to get discovery driven. Data mining is a young
better results. A lot of methods have been proposed interdisciplinary field closely connected to data
in the literature for CT (Computed Tomography), warehousing, statistics, machine learning, neural
434
networks and inductive logic programming. Data a number of basic problems concerning document
mining provides automatic pattern recognition and retrieval, data mining in the databases, and
attempts to uncover patterns in data that are difficult visualization of high-dimensional data.
to detect with traditional statistical III.PROBLEM STATEMENT
methods. Without data mining it is difficult to
realize the full potential of data collected within The conventional method in medicine for brain MR
healthcare organization as data under analysis is images classification and tumor detection is human
massive, highly dimensional, distributed and inspection. Operator-assisted classification methods
uncertain. are impractical for large amounts of data and are
also non-reproducible. MR images also always
Massive healthcare data needs to be converted contain a noise caused by operator performance
into information and knowledge, which can help which can lead to serious inaccuracies classification.
control, cost and maintains high quality of patient The MR images data is by nature, a huge, complex
care. Healthcare data includes Patient centric data and cognitive process. Accurate diagnosis of MR
and Aggregate data. images data is not an easy task and is always time
Recent progress in data storage and acquisition has consuming. In some extreme scenario, diagnosis
resulted in a growing number of enormous with wrong result and delay in delivery of a correct
databases. The information contained in these diagnosis decision could occur due to the
databases can be extremely interesting and useful; complexity and cognitive process of which it is
however the amount is too large for humans involved. The use of artificial intelligent techniques
to process manually. These databases store for instance, neural networks, fuzzy logic, neuro
information covering every aspect of human fuzzy has shown great potential in this field. With
activity. In the business world, the data bases are the involvement of soft computing, the pattern
created for marketing, investment, manufacturing, matching, classification and detection of algorithms
customers, products or transactions, and such which have direct applications in many medical
information can be stored both for accounting and problems have become much easier to be
analysis purposes. The domain of scientific research implemented and diagnosed. Hence, this project
is another area which makes heavy use of databases tries to find out how the neuro registration system or
to store information about, for example patients, the hybrid combination of neural network and
patient histories, surveys, medical registration techniques can be applied in
investigations. Medical databases contain data in a the classification and detection purposes,
variety of formats: images in the form of X-rays or particularly on the medical field.
scans, textual information to describe details of
diseases, medical histories, psychology reports, IV.DATA MINING
medical articles or various signals like EKG, ECG,
EEG, etc. This data does not need to be located on Data mining is a new discipline within the field of
the same system. It may be distributed amongst information retrieval that extracts interesting and
various disparate computers depending on the useful material from large data sets. In the broader
source of the data. Such heterogeneous data make sense, data mining is defined as part of knowledge
the process of information retrieval an even more discovery in databases (KDD) and draws on the
complex process. There is a substantial need for fields of statistics, machine learning,
signal/image processing and data mining tools pattern recognition and database management. The
in medical information processing systems and in general approach of KDD process is presented on
telemedicine that are flexible, effective and used figure 2. The KDD process involves the following
friendly. Telemedicine is defined as: The steps:
investigation, monitoring and management of 1. Selecting the target data (focusing on the part of
patients and education of patients and staff the data on which to make discoveries). The data
using systems which allow ready access to expert may consist from different type of information.
advice and patient information no matter of where 2. Data cleaning and preprocessing includes
the patient or relevant information is located. Such removal of noise, outliers, missing data (if they are
systems serve a dual role: they can assist medical not an object of the discovery).
professionals in medical research and in the clinic 3. Transforming data includes data reduction and
medical diagnosis and secondly, they can be used as projection in order to find and investigate useful
information sources for patients. Telemedicine faces features and reduce the data feature space
435
dimensionality. pixels, where each word or pixel is here understood

4. Performing the data mining task, e.g., as a feature. Currently, processing abilities are often
classification, regression, clustering and as well the not able to handle such high dimensional data,
outlier detection and imputation of missing data, mostly due to numerical difficulties in processing,
if they were not removed in data cleaning process. requirements in storage and transmission within a
Data mining is performed in order to extract patterns reasonable time. To reduce the computational time,
and relationships in the data. it is common practice to project the data onto a
smaller, latent space. Moreover, such a space is
often beneficial for further investigation due to
noise reduction and desired feature extraction
properties. Smaller dimensions are
also advantageous when visualizing and analyzing
the data. Thus, in order to extract desirable
information, dimensionality reduction methods are
often applied. This task is non-trivial. The goal is to
determine the coordinate system where the mapping
will create low-dimensional compact representation
of the data whilst maximizing the information
contained within the sample.
VI.DESIGN METHODOLOGY
Knowledge discovery is the process of automatically

generating information formalized in a form
understandable to humans
Fig 2: The steps of the KDD process.
In the first step the data is extracted from the source

location, then the preprocessing is performed and
the data is transformed to the form suitable for
further processing. In the next step the data mining
is applied, for example clustering, classification,
regression etc. The final part of KDD process is a
meaningful interpretation is the obtained results.
5. Interpretation and visualization includes
interpretation of mined patterns, visualization of Fig3: Knowledge discovery is the process of
models, patterns and data. Typically, data mining automatically generating information
algorithms assume that the data is already stored
in main memory; therefore no database extraction STEP 1:- The first step is to predefine our mission or
methods are investigated in this work. However, in a goal before discovering knowledge. We also have
order to facilitate the extraction of the required to point out from which database we can obtain the
information from the databases, the issue of large knowledge.
data sets and the preprocessing and STEP2:- Consider a case where we have millions of
required transformation algorithms is addressed. data points. We have to select a subset of the
database to perform the required knowledge
V.DIMENSIONALITY REDUCTION AND discovery steps. Selection is the process of selecting
FEATURE SELECTION the right data from the database on which the tools
in data mining can be used to extract information,
When processing large databases, one faces two knowledge and pattern from the provided raw data.
major obstacles: numerous samples and high STEP3:- Data preprocessing and data cleaning. In
dimensionality of the feature space. For example, this step we try to eliminate noise that is present in
the documents are represented by several thousands the data. Noise can be defined as some form of error
of words; images are composed of millions of within the data. Some of the tools used here can be
436
used for filling missing values and elimination of categories: area- and feature based methods [8]
duplicates in the database. [11]. Area-based methods, a.k.a. template matching,
STEP 4:- Transformation of data in this step can be combine the feature-detection and feature-matching
defined as decreasing the dimensionality of the data steps. These methods first define a window in the
that is sent for data mining. Usually there are cases input image as the template and then search for a
where there are a high number of attributes in the corresponding template in the reference image. The
database for a particular case. With the reduction centers of matched image templates are used as
of dimensionality we increase the efficiency of the CPs. There are many ways to find a matching
data-mining step with respect to the accuracy and template, and one of the most popular criteria is
time utilization. cross correlation. Commonly used correlation-based
STEP 5:- The data mining step is the major step in methods include normalized cross
data KDD. This is when the cleaned and correlation (NCC), sequential similarity detection,
preprocessed data is sent into the intelligent and Fourier methods [8], [12]. Template-matching
algorithms for classification, clustering, similarity approaches using correlation based criteria, because
search within the data, and so on. Here we chose the of their ease of use and high efficiency, A
algorithms that are suitable for discovering patterns disadvantage of the correlation-based approach,
in the data. Some of the algorithms provide however, is that it only works well if the two images
better accuracy in terms of knowledge discovery have rather similar gray-level
than others. Thus selecting the right algorithms can characteristics. Therefore, the approach is not suited
be crucial at this point for co registration of images from different sensors.
STEP 6:- Interpretation. In this step the mined data For multi sensor images, the use of mutual
is presented to the end user in a human-viewable information which relies on statistical
format. This involves data visualization, which the dependences between the two images is proposed
user interprets and understands the discovered [10]. Feature-based registration methods rely on
knowledge obtained by the algorithms. salient structures in the images to be matched and
can be based on regions, lines, or points. Two
VII.IMAGE REGISTRATION METHODS critical steps in feature-based methods are feature
extraction and feature matching. There are
In most studies, multi angle image registration relies many kinds of feature-extraction algorithms, and
on the manual selection of homologue points in among them, point based feature-detection methods
each image [1], carried out by a skilled operator receive the most attention. Commonly used point-
familiar with the study area. However, the manual based feature-detection algorithms include the
selection of CPs is tiring, time consuming, prone to Harris operator, the Gabor filter or Gabor wavelet,
inconsistency, and may result in low and the scale-invariant feature transform (SIFT)
accuracy. Manual selection is also impractical in [11], [13][15]. Detected points are expected to be
situations where a large number of images need to distinct and ideally evenly spread over the image.
be registered. Thus, there are pressing needs for After features have been detected in the pair of
automatic image registration methods. A typical images, they need to be matched. Important feature
registration method can be divided into four steps. matching methods include invariant descriptor and
1) Feature detection: Detect features of salient and spatial relation and relaxation methods [8]. Most of
distinctive objects in the input image (the image to these methods are equipped with filtering criteria to
be registered) and the reference image. reject possible outliers. Generally speaking, area-
2) Feature matching: Establish the correspondence based methods attain higher accuracies than feature-
between features in the input image and reference based methods [14] and are particularly suited for
image. the registration of images acquired from the
3) Transformation model estimation: Set up a same sensor. However, it is difficult to apply area-
transformation model and estimate the model based methods directly on multi angle imagery.
parameters by the use of matched CPs. Geometric distortion is usually high for images
4) Image re sampling: Transform the input image acquired at large view angles. Hence,
according to the transformation model and the rigid transformation models such as affine and
chosen re sampling function [8]. projective transforms are not suitable. A viable
Many registration methods have been proposed. alternative is a non rigid registration model.
Depending on whether feature detection [step 1)] is However, for a non rigid model to work well,
involved, methods can be broadly classified into two an adequate number of quality CPs and a good
437
estimation of the model parameters are needed. developed to cope with real-world situations. They
Furthermore, because CPs are selected by first are mainly considered in the situations such as ill-
defining a window in the input image and then defined and noisy natural data.
searching for a match in the reference image,
one faces the problem of having to deal with a very Under these conditions, ANN computing methods
large search space (in the reference image). In order are more effective and economical than the
to solve this problem, a coarse-to-fine hierarchical traditional computation methods. In order to deal
strategy is usually adopted [14]. Using a pyramid with non-stationary data, the properties of the ANNs
approach, candidate matching locations are first should be made adaptive. In this case, the
identified from the images that have been performance of the system should improve with use,
downscaled to coarser resolutions. Then, the and the system should be able to capture and store
accuracy of the matched locations is gradually information, or to perform learning.
improved by moving up to finer
resolutions. Compared with area-based approaches, In addition, ANNs should be able to perform
feature-based methods are more robust to variation generalization, that is, able to deal with subsets of
in view angles and are therefore more suited for the problem domain that are not yet to be
multi angle image registration [16]. The main encountered. All these properties must be
disadvantage of feature-based methods is that appreciated and inherited in the context of ANNs.
the number of detected CPs may sometimes be low, Without these properties, ANNs are similar to mere
which will cause problems in the estimation of a look-up tables.
complex non rigid transformation such as a TPS or a
piecewise linear model. To tackle the IX.RESULTS
aforementioned problems, we propose an
automatic registration method that integrates the
merits of both area- and feature-based methods.
The automatic registration routine was specifically

designed to be incorporated in a super
resolution image reconstruction approach The
proposed method involves a two-stage CP detection
scheme, where candidate CPs are first selected with
a SIFT method and, then, in a second step, with a
template-matching method using NCC as a criterion.
It also incorporates a hierarchical approach to refine Intensity Maped image
the set of selected CPs. Outliers in the candidate
SIFT CPs are discarded by the ambiguity criterion
and robust estimation of the projective
transformation model.
VIII.ARTIFICIAL NEURAL NETWORKS
ANNs are computational networks that attempt to

simulate, in a gross manner, the networks of nerve
cell of human or animal biological central nervous
system. Two important aspects of ANNs are
Extracted Region without outer Skull Region
(Graupe, 1997): a. it allows the use of very simple
computational operations to solve
complex, mathematically ill-defined, non-linear and
stochastic problems, and b. it has a self-organizing
features and learning ability, allowing it to solve
for a wide range of problems. These aspects are very
similar to the ability of the human brain to resolve
simple problems, such as movement and vision.
Several computational formalisms of ANNs are
438
Thresholded Binarized image Intensity Maped image
Morphologically operated Dilated image
Extracted Region without outer Skull Region
Region filled Image
Thresholded Binarized image
Extracted Region
Morphologically operated Dilated image
Histogram equalized image
439
Region filled Image [9] Ramiro Castellanos and Sunanda Mitra, Segmentation of
magnetic resonance images using a neuro-fuzzy algorithm. IEEE
Symposium on Computer-Based Medical Systems, 2000.
[10] K. Johnson, A. C. Rhodes, I. Zavorin, and J. Le Moigne,
Mutual information as a similarity measure for remote sensing
image registration, in Proc. SPIE Aerosense, Geo-Spatial Image
Data Exploitation II, Orlando, FL, Apr. 2001, vol. 4383, pp. 51
61.
[11] Y. Bentoutou, N. Taleb, K. Kpalma, and J. Ronsin, An
automatic image registration for applications in remote sensing,
IEEE Trans. Geosci. Remote Sens., vol. 43, no. 9, pp. 21272137,
Sep. 2005.
[12] J. P. Lewis, Fast template matching, in Proc. Vis. Interface,
1995, pp. 120123.
[13] L. Yu, D. Zhang, and E. J. Holden, A fast and fully
automatic registration approach based on point features for multi-
source remote-sensing images, Comput. Geosci, vol. 34, no. 7,
pp. 838848, Jul. 2008.
[14] G. Hong and Y. Zhang, Wavelet-based image registration
technique for high resolution remote sensing images, Comput.
X.CONCLUSION Geosci, vol. 34, no. 12, pp. 17081720, Dec. 2008.
Medical image classification is an important

thing in medicine. It allows for biological structures
to be isolated non-invasively. Whether it is for
diagnostic purposes or practically applied in image
guided surgeries, image classification has many
forms and uses. Unfortunately, currently there is no
classification strategy that can accommodate all its
applications. The system that has been designed is
used for classifying and detecting tumors in brain MR
images. The system involves two major modules, one
that performs classification and the other one that
Mrs. G. Madhavi obtained her B.tech degree in Electronics
detects the tumors from the images. Neuro
and communication Engineering from Velagapudi
registration procedure is suitable and feasible to be Ramakrishna Siddhartha engineering college, Vijayawada
used as supportive tools for medical problems. The in 1998 and M.Tech degree in Digital systems and
combination of both, neural network and image computer electronics from JNTUH in 2006.Presently she
registration or known as neuro registration model has is working as an Associate professor in the Dept of ECE in
been tested and evaluated for classification and Mahatma Gandhi Institute of Technology, Gandipet,
detection of MR images. Hyderabad. Mrs Madhavi has more than 13 years of
teaching experience .she is pursuing her PhD from Acharya
REFERENCES Nagarjuna University, Guntur in Image processing Area
[1] L.I. Kuncheva, F. Steimann, Fuzzy diagnosis, Artificial She is the student member of IEEE. Her areas of interests
Intelligence in Medicine, vol. 16, pp. 121-128, 1999. are Image processing, Nueral networks, Digital Design,
[2] J.-S.R. Jang, ANFIS: Adaptive-network-based fuzzy inference Microwaves.
system, IEEE Transactions on Systems, Man, and Cybernetics,
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[3] S.Y. Belal, A.F.G. Taktak, A.J. Nevill, S.A. Spencer, D. Roden,
S. Bevan, Automatic detection of distorted plethysmogram pulses
in neonates and pediatric patients using an adaptive-network-
based fuzzy inference system, Artificial Intelligence in Medicine,
vol. 24, pp. 149-165, 2002.
[4] I. Virant-Klun, J. Virant, Fuzzy logic alternative for analysis
in the biomedical sciences, Computers and Biomedical Research,
vol. 32, pp. 305- 321, 1999.
[5] Joseph P. Hornak, the Basics of MRI, 1996-2008.
[6] Kasabov, N.: Foundations of Neural Networks, Fuzzy Systems
and Knowledge Engineering. MIT Press, Cambridge, 1996.
[7] Rami J. Oweis and Muna J.Sunna,A Combined Neuro Fuzzy
Approach for Classifying Image Pixels in Medical
Applications. Journal of Electrical Engineering, Vol.56, No.5-6,
2005.
[8] N.Benamrane, A. Aribi, L.Kraoula, Fuzzy Neural Networks
and Genetic Algorithms for Medical Images Interpretation. IEEE
Proceedings of the Geometric Modeling and Imaging, 2006.
440
Authentication and Self-Repair Capability for

Fixing Tampered Image Data
V.Balaji 1, Anusha Begum 2, Prof E.Sreenivasa Reddy3
1
M.Tech Assistant professor, Computer Science and Engineering, Acharaya Nagarjuna University College of
Engineering and Technology, Guntur, Andhra Pradesh, INDIA, v12.balaji@gmail.com
2
B.Tech, Computer Science and Engineering, Andhra Pradesh, INDIA,
3
professor, Department of Computer Science and Engineering, Acharaya Nagarjuna University College of Engineering
and Technology, Guntur, Andhra Pradesh, INDIA, edara_67@gmail.com
ABSTRACT: line arts, etc., as main contents, are often digitized

The purpose of this paper is to provide the intensity into grayscale images with two major gray values,
and the authentication of digital images. We can one being of the background (including mainly blank
protect the document images even if the image has spaces) and the other of the foreground (including
been altered or destroying a particular content. These mainly texts). It is noted that such images, although
document images are digitalized into the gray scale gray valued in nature, look like binary.
which is digital format. These gray images provide
the different levels of confidentiality by transforming In general, the image authentication problem is
gray images, applying byte run length code and difficult for a binary document image because of its
Shamirs secret sharing on this encoded data. Then simple binary nature that leads to perceptible changes
they are different system like existing system in after authentication signals are embedded in the
which Lee et al has using an edge line similarity image pixels. Such changes will arouse possible
measure to select flippable pixels to reduce the suspicions from attackers. A good solution to such
distortion. In the proposed system we can fix the binary image authentication should thus take into
tampered image data and provides alpha channel for account not only the security issue of preventing
the stego-image. This process is very useful for the image tampering but also the necessity of keeping the
securing the data and also that the retrieving the visual quality of the resulting image. Here an
tampered data. authentication method that deals with binary-like
grayscale document images instead of pure binary
ones and simultaneously solves the problems of
Keywords: Image authentication, Radon transform,
Stego-image, Tampering image, binarization, Alpha image tampering detection and visual quality keeping.
channel
In this proposed method, a PNG image is created
1. INTRODUCTION from a binary-type grayscale document image with an
alpha channel plane. The original image may be
Digital image is a form for preserving important thought as a grayscale channel plane of the PNG
information. However, with the fast advance of digital image.
technologies, it is easy to make visually imperceptible
modifications to the contents of digital images. How 1.1 Purpose
to ensure the integrity and the authenticity of a digital Image data protection and image-based
image is thus a challenge. It is desirable to design authentication techniques offer efficient solutions for
effective methods to solve this kind of image controlling how private data and images are made
authentication problem [1][3], particularly for available only to selected people. Essential to the
images of documents whose security must be design of systems used to manage images that contain
protected. It is also hoped that, if part of a document sensitive data - such as medical records, financial
image is verified to have been illicitly altered, the transactions, and electronic voting systems - the
destroyed content can be repaired. Such image methods presented are useful to counter traditional
content authentication and self-repair capabilities are encryption techniques, which do not scale well and
useful for the security protection of digital documents are less efficient when applied directly to image files.
in many fields, such as important certificates, signed
documents, scanned checks, circuit diagrams, art 1.2 Scope
drawings, design drafts, last will and testaments, and If the image is transformed to frequency
so on.Document images, which include texts, tables, domain, then quantization is required and also
441
processing time is required for the conversion. Due to authentication codes are accordingly computed and
quantization the retransformation may not result in inserted into selected random pixels.
the original image. So we are using spatial domain In Tzeng and Tsais method [8], randomly
transformation. generated authentication codes are embedded into
The given gray image is transformed using image blocks for use in image authentication, and a
radon transform. Radon is a spatial domain transform, so-called code holder is used to reduce image
where the pixels intensity will be oriented in a distortion resulting from data embedding.
particular angle. Since the transformed image is Lee et al. [9] proposed a Hamming-code-
different from the original image the confidentiality based data embedding method that flips one pixel in
will be maintained. each binary image block for embedding a watermark,
This gives the first level of confidentiality to yielding small distortions and low false negative rates.
the secret image. Then the byte run length code is Lee et al. [10] improved the method later by using an
applied to get the compressed transformed image. edge line similarity measure to select flippable pixels
This gives second level of confidentiality. Then apply for the purpose of reducing the distortion.
Shamirs secret sharing on this encoded data to get n
number of shares. This gives third level of Disadvantages:
confidentiality. With water marking only we are performing
security.
1.4 Model Diagram Pixels neighboring factor is too less here,
that means with association among pixels.
Working with document images is not
generating stego-images with desired quality.
3. Proposed System
In proposed system, a method for the
authentication of document images with an additional
self-repair capability for fixing tampered image data
is proposed. The input cover image is assumed to be a
binary-like grayscale with two major gray values.
After the proposed method is applied, the cover image
is transformed into a stego-image in the Portable
Network Graphics (PNG) format with an additional
Fig. 1.4 Model Diagram alpha channel for transmission on networks or
archiving in databases. The stego-image, when
2. Existing System received or retrieved, may be verified by the proposed
Several methods for binary image method for its authenticity. Integrity modifications of
authentication have been proposed in the past. Wu the stego-image can be detected by the method at the
and Liu [4] manipulated the so-called flippable pixels block level and repaired at the pixel level. In case the
to create specific relationships to embed data for alpha channel is totally removed from the stego-
authentication and annotation of binary images. image, the entire resulting image is regarded as
Yang and Kot [5] proposed a two-layer inauthentic, meaning that the fidelity check of the
binary image authentication method in which one image fails. The proposed method is based on the so-
layer is used for checking the image fidelity and the called -threshold secret sharing scheme proposed by
other for checking image integrity. In the method, a Shamir [11] in which a secret message is transformed
connectivity-preserving transition criterion for into shares for keeping by participants, and when of
determining the flippability of a pixel is used for the shares, not necessarily all of them, are collected,
embedding the cryptographic signature and the block the secret message can be losslessly recovered. Such a
identifier. Later, Yang and Kot [6] proposed a pattern- secret sharing scheme is useful for reducing the risk
based data hiding method for binary image of incidental partial data loss.
authentication in which three transition criteria are Conventionally, the concepts of secret
used to determine the flippabilities of pixels in each sharing and data hiding for image authentication
block, and the watermark is adaptively embedded into are two irrelevant issues in the domain of information
embeddable blocks to deal with the uneven security. However, in the proposed method, we
embeddability condition in the host image. In the combine them together to develop a new image
method proposed a set of pseudorandom pixels in a authentication technique. The secret sharing scheme
binary or halftone image are chosen and cleared, and is used in the developed technique not only to carry
442
authentication signals and image content data but also III. Generation, transmit of stego image
to help repair tampered data through the use of shares.
Here we are generating the stego-image for
Advantages: binarization to receiver. The stego-image, when
Loss less data is able to be transmitted received or retrieved, may be verified by the proposed
through the network. method for its authenticity. Integrity modifications of
Here authenticity operation is done with the stego-image can be detected by the method at the
stego image. block level and repaired at the pixel level. In case the
Repair operation is done successfully here alpha channel is totally removed from the stego-
and generates PNG Image image, the entire resulting image is regarded as
dynamically. inauthentic, meaning that the fidelity check of the
image fails.
Four Modules Description
IV. Binarization and verify image at receiver side
I. Gray scale image binarization from
sender side
After performing the binarization at the receiver side,
II. Appending alpha channel phase
the image is to be filtered from the alpha channel.
III. Generation , transmit of stego image
After stego image generation if there is no authentic
IV. Binarization and verify image at
process repair the tampered image blocks then
receiver side
remove the alpha channel. If the authentication is
success directly receive the PNG Image at the
I. Grayscale image binarization from sender side
receiver side.
If an image has had its background
normalized to a constant value, it can be binarized by
V.DISCUSSIONS
a global value chosen to be less than the background
by an appropriate amount. It must be less because the
In addition to being capable of data repairing and
background cannot be made exactly constant,
being blind in nature (requiring no overhead other
especially in situations where it was changing rapidly
than the stego-image), the proposed method has
or where there is bleed through from text or image on
several other merits, which are described in the
the other side of the page. Here we are performing
following.
binarization on an input document image.
5.1 Providing pixel-level repairs of tampered
II. Appending alpha channel phase
image parts: As long as two untampered partial
Appending the alpha channel to the binarized
shares can be collected, a tampered block can be
image at the time the pixels in the binarized is
repaired at the pixel level by the proposed method.
performed the associativity the neighbouring pixels.
This yields a better repair effect for texts in images
Alpha channel is an additional channel that can be
because text characters or letters are smaller in size
added to an image. It contains transparency
with many curved strokes and need finer pixel-level
information about the image, and depending on the
repairs when tampered with.
type of alpha it can contain various levels of
transparency. The alpha channel basically controls the
5.2 Having higher possibility to survive image
transparency of all the other channels.
content attacks: By skillfully combining the Shamir
scheme, the authentication signal generation, and the
By adding the alpha channel to an image you control
random embedding of multiple shares, the proposed
the transparency of the red channel, green channel
method can survive malicious attacks of common
and the blue channel. There are two types of alpha
content modifications, such as superimposition,
channels, pre-multiplied alpha and straight alpha.
painting, etc., as will be demonstrated by
Which alpha type you use depends on the software
experimental results subsequently described.
being used.
5.3 Making use of a new type of image channel for
data hiding: Different from common types of
images, a PNG image has the extra alpha channel
plane that is normally used to produce transparency to
the image. It is differently utilized by the proposed
method for the first time as a carrier with a large
space for hiding share data. As a comparison, many
Fig.5.1 Appending Alpha channel phase other methods use LSBs as the carriers of hidden data.
443
5.4 Causing no distortion to the input image: Hierarchical watermarking for secure image authentication
with localization, IEEE Trans. Image Process., vol. 11, no. 6,
Conventional image authentication methods that
pp. 585595, Jun. 2002.
usually embed authentication signals into the cover [3] Z. M. Lu, D. G. Xu, and S. H. Sun, Multipurpose image
image itself will unavoidably cause destruction to the watermarking algorithm based on multistage vector
image content to a certain extent. Different from such quantization, IEEE Trans. Image Process., vol. 14, no. 6, pp.
822831, Jun. 2005.
methods, the proposed method utilizes the pixels
[4] M. Wu and B. Liu, Data hiding in binary images for
values of the alpha channel for the purpose of image authentication and annotation, IEEE Trans. Multimedia, vol.
authentication and data repairing, leaving the original 6, no. 4, pp. 528538, Aug. 2004.
image (i.e., the grayscale channel) untouched and thus [5] H. Yang and A. C. Kot, Binary image authentication with
tampering localization by embedding cryptographic signature
causing no distortion to it. The alpha channel plane
and block identi-fier, IEEE Signal Process. Lett., vol. 13, no.
may be removed after the authentication process to 12, pp. 741744, Dec. 2006.
get the original image. [6] H. Yang and A. C. Kot, Pattern-based data hiding for binary
images authentication by connectivity-preserving, IEEE
Trans. Multimedia, vol. 9, no. 3, pp. 475486, Apr. 2007.
6. CONCLUSION [7] H. Y. Kim and A. A?f, Secure authentication watermarking
for halftone and binary images, Int. J. Imag. Syst. Technol.,
A new blind image authentication method with a data vol. 14, no. 4, pp. 147152, 2004.
repair capability for binary-like grayscale document [8] C. H. Tzeng and W. H. Tsai, A new approach to
authentication of binary images for multimedia
images based on secret sharing has been proposed.
communication with distortion reduc-tion and security
Both the generated authentication signal and the enhancement, IEEE Commun. Lett., vol. 7, no. 9, pp. 443
content of a block have been transformed into partial 445, Sep. 2003.
shares by the Shamir method, which have been then [9] Y. Lee, J. Hur, H. Kim, Y. Park, and H. Yoon, A new
binary image au-thentication scheme with small distortion and
distributed in a well-designed manner into an alpha
low false negative rates, IEICE Trans. Commun., vol. E90-
channel plane to create a stego-image in the PNG B, no. 11, pp. 32593262, Nov. 2007.
format. The undesired opaque effect visible in the [10] Y. Lee, H. Kim, and Y. Park, A new data hiding scheme for
stego-image coming from embedding the partial binary image authentication with small image distortion, Inf.
Sci., vol. 179, no. 22, pp. 38663884, Nov. 2009.
shares has been eliminated by mapping the share
[11] A. Shamir, How to share a secret, Commun. ACM, vol. 22,
values into a small range of alpha channel values near no. 11, pp. 612613, Nov. 1979.
their maximum transparency value of 255. [12] C. C. Lin and W. H. Tsai, Secret image sharing with
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7. FUTURE ENHANCEMENT
[13] W. H. Tsai, Moment-preserving thresholding: A new
approach, Comput. Vis. Graph. Image Process., vol. 29, no.
In the process of image block authentication, a block 3, pp. 377393, Mar. 1985.
in the stego-image has been regarded as having been
tampered with if the computed authentication signal
does not match that extracted from corresponding
partial shares in the alpha channel plane. For the self-
repairing of the content of a tampered block, the
reverse Shamir scheme has been used to compute the
original content of the block from any two
untampered shares. Measures for enhancing the
security of the data embedded in the alpha channel
plane have been also proposed. Experimental results
have been shown to prove the effectiveness of the
proposed method. Future studies may be directed to
choices of other block sizes and related parameters
(prime number, coefficients for secret sharing,
number of authentication signal bits, etc.) to improve
data repair effects. Applications of the proposed
method to the authentication and the repairing of
attacked color images may be also tried.
REFERENCES
[1] C. S. Lu and H. Y. M. Liao, Multipurpose watermarking for

image authentication and protection, IEEE Trans. Image
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444
Comparative Analysis of Block Matching

Algorithms for Motion Estimation
V.S.N.Kumar Devaraju, #1, B. Prasanthi *2
#
Sr.Asst.Professor,Electronics and Communication Engineering, *Sr.Asst.Professor,Computer Science Engineering,
Mahatma Gandhi Institute Of Technology
Gandipet, Hyderabad-Andhra Pradesh ,India
1
dvsnkumar@mgit.ac.in 2biraliprashanthi@mgit.ac.in
Abstract: Block-matching motion estimation is an 2. BLOCK DIAGRAM

important part for any motion compensated video coding
standards such as MPEG and H.26x. In this case, video
frames are divided into a set of non- overlapped macro
blocks; each target macro block of the current frame is
compared with the search area in the previous frame for
the purpose of finding the best matching macro block.
The displacement of the best-matched macro block within
the search window from the previous frame is the motion
vector of the target macro block. This paper is a review of
the block matching algorithms used for motion estimation
in video compression. It implements and compares 7
different types of block matching algorithms that range
from the very basic Exhaustive Search to the recent fast
adaptive algorithms like Adaptive Rood Pattern Search.
Fig 1: MPEG/H.26x video compression process flow.
The algorithms that are evaluated in this paper are widely
accepted by the video compressing community and have
been used in implementing various standards, ranging blocks of image from the previous frame. The motion
from MPEG1 / H.261 to MPEG4 / H.263. The paper also vectors for blocks used for motion estimation are
presents a very brief introduction to the entire flow of transmitted, as well as the difference of the
video compression.
compensated image with the current frame is also
JPEG encoded and sent. The encoded image that is
Keywords: Block-matching, MPEG, Video Compression sent is then decoded at the encoder and used as a
reference frame for the subsequent frames. The
1. INTRODUCTION
decoder reverses the process and creates a full frame.
The whole idea behind motion estimation based video
compression is to save on bits by sending JPEG
Video technology promises to be the key for the encoded difference images which inherently have less
transmission of motion video. A number of video energy and can be highly compressed as compared to
compression techniques and standards have been sending a full frame that is JPEG encoded. Motion
introduced in the past few years, particularly the JPEG, where all frames are JPEG encoded, achieves
MPEG-1 and MPEG-2 for interactive multimedia and anything between 10:1 to 15:1 compression ratio,
for digital NTSC and HDTV applications, and whereas MPEG can achieve a compression ratio of
H.261/H.263forvideo telecommunications. These 30:1 and is also useful at 100:1 ratio [2] [3] [4]. It
techniques use motion estimation techniques to should be noted that the first frame is always sent
reduce the amount of data that is stored and full, and so are some other frames that might occur at
transmitted for each frame. some regular interval (like every 6th frame). The
standards do not specify this and this might change
The basic flow of the entire compression- with every video being sent based on the dynamics of
decompression process is largely the same and is the video. The most computationally expensive and
depicted in Fig 1. The encoding side estimates the resource hungry operation in the entire compression
motion in the current frame with respect to a previous process is motion estimation. Hence, this field has
frame. A motion compensated image for the current seen the highest activity and research interest in the
frame is then created that is built of past two decades.
445
3. CLASSIFICATION OF ALGORITHMS
where N is the side of the macro bock, Cij and Rij are
the pixels being compared in current macro block and
reference macro block, respectively.
Fig 2: Classification
Frequency domain algorithms: In these

algorithms, the algorithm is applied on the
transformed coefficients. The different
algorithms/techniques used are phase Fig 3: Block matching a macro block of side 16
correlation, matching in wavelet and DCT pixels and a search parameter p of size 7 pixels.
domain.
Time domain algorithms: These comprise of The PSNR is most commonly used as a measure of
matching algorithms and gradient-based quality of reconstruction in image compression etc. It
algorithms. is most easily defined via the mean squared error
Block matching algorithms: match all or (MSE) which for two mxn monochrome images I and
some pixels in the current block with the K where one of the images is considered a noisy
block in the search area of reference frame approximation of the other. Peak-Signal-to-Noise-
based on some cost function. Ratio (PSNR) given by equation (iii) characterizes
Feature based algorithms: match the meta the motion compensated image that is created by
information/data of the current block with using motion vectors and macro clocks from the
that of the block in reference frame. reference frame.
4. SET UP PROCESS
peak to peak value of original data 2
PSNR 10 log10
MSE
The idea behind block matching is to divide the
current frame into a matrix of macro blocks that are
then compared with corresponding block and its 5. DIFFERENT ALGORITHMS
adjacent neighbors in the previous frame to create a
vector that stipulates the movement of a macro block
5.1 EXHAUSTIVE SEARCH (ES)
from one location to another in the previous frame.
Usually the macro block is taken as a square of side
16 pixels, and the search parameter p is 7 pixels.
The idea is represented in Fig 3. The matching of one

macro block with another is based on the output of a
cost function. The macro block that results in the
least cost is the one that matches the closest to current
block. There are various cost functions, of which the
most popular and less computationally expensive is
Mean Absolute Difference (MAD) given by equation
(i). Another cost function is Mean Squared Error
(MSE) given by equation (ii).
Fig 4. Exhaustive Search
446
This algorithm calculates the cost function at each algorithm is as follows. As mentioned above, NTSS
possible location in the search window. As a result of differs from TSS by full search assuming a center
which it finds the best possible match and gives the biased checking point pattern in the first step and
highest PSNR amongst any block matching NTSS incorporating a half-way stop technique for
algorithm. The obvious disadvantage to ES is that the stationary or quasi stationary blocks. Details are as
larger the search window gets the more computations follows.
it requires.
1. In addition to the original checking points
5.2 THREE STEP SEARCH (TSS) used in TSS, eight extra points are added,
which are the eight extra neighbors of the
It is a block based search technique and uses a search window (center-bias).
maximum of three steps. It is mainly used for Motion 2. A half-way stop technique is used for
Estimation in low bit rate video compression stationary and quasi stationary block in
applications such as videophone and video order to fast identify and then estimate the
conferencing. motions for these blocks.
a. If the minimum MBD in the first
step occurs at the search window
The TSS algorithm is a simple and effective method
enter, stop search (first-step-stop).
used for Motion Estimation. When compared with
Exhaustive Block Matching Algorithm (EBMA) it
has a saving factor of greater than a 100. It has a
fixed number of three search steps and a maximum
and minimum number of search points of 25. The
disadvantage of this technique is that it uses a
uniformly allocated checking point pattern in the first
step which makes it inefficient for the estimation of
small motion.
1. An initial step size is picked. Eight blocks at

a distance of step size from the center (One
center point and eight points on the
boundary of the search square) are compared
in the first step and thereafter only 8 points Fig 5. Flow chart for NTSS algorithms
are searched.
2. At the start of a new step the search point b. If the minimum MBD point in the
center is moved from the best matching first step is one of the eight
point from the previous step. Step size is neighbors of the window center, the
reduced by half after each step. search in the second step will be
3. At the end of the search the step size is one performed only for eight
pixel. neighboring points of the minimum
and then stop search (second-step-
5.3 NEW THREE STEP SEARCH (NTSS) stop).
NTSS algorithm is a modification of the well-known For the search window , in the first step the N3SS
TSS algorithm. Since most real world motion algorithm utilizes a 3x3 grid search pattern at the
sequences are smooth and gentle, detecting small center in addition to the larger 9x9 grid in the 3SS.
motions becomes an important issue. One solution to
this could be to make the search adaptable to the If the minimum BDM point is found at the center of
uncertainty and motion scale. This is achieved by the search window, the search will stop. If the
employing a center-biased checking point pattern in minimum BDM point is one of the eight points on the
the first step. 3x3 grid, only additional five or three points will be
checked, otherwise the search window center will be
Flow chart moved to the winning point and the following
procedure is the same as in 3SS. The details are as
The block diagram shown in Fig. 5 shows the steps follows.
involved in the NTSS algorithm. The NTSS
447
Step 1 If MAD (A) MAD (B) and MAD (A) < MAD (C),
II is selected;
Totally points are checked including the central nine
points on the 3x3 grid and the eight neighboring If MAD (A) < MAD (B) and MAD (A) < MAD (C),
pointsonthe 9x9grid.Iftheminimum BDM point is the III is selected;
search window center, the search will be terminated;
otherwise go to Step 2. If MAD (A) < MAD (B) and MAD (A) MAD (C),
IV is selected.
Step 2
If one of the central eight neighboring points on the

3x3 grid is found to be the minimum in the first step,
go to Step 3; otherwise go to Step 4.
Step 3
Move the small 3x3 search window so that the

window center is the winning point found in Step 1.
Search additional five or three points according to the
location of the previous winning point, then the
search will stop.
Step 4 Fig6. Search patterns corresponding to each selected

quadrant: (a) Shows all quadrants (b) quadrant I is
selected (c) quadrant II is selected (d) quadrant III is
Reduce the large 9x9 search window size by half and
selected (e) quadrant IV is selected
move the center to the minimum BDM point in Step
1, follow the searching process of Step 2 and Step 3
in 3SS (refer to [2]). 5.5 FOUR STEP SEARCH (4SS)
5.4 SIMPLE AND EFFICIENT SEARCH (SES) With a search pattern of +/-7, the FSS algorithm
utilizes a center-biased search pattern with nine
checking points on a 5 X 5 window in the first step,
The main idea behind the algorithm is that for a
instead of a 9 X 9 window as in the TSS. The center
unimodal surface there cannot be two minimums in
of the window is then shifted to the point with the
opposite directions and hence the 8 point fixed
minimum BDM. In addition to that, FSS reduces the
pattern search of TSS can be changed to incorporate
worst case search points from 33 to 27. Similar to
this and save on computations
NTSS, 4SS [6] also employs center biased searching
and has a halfway stop provision. 4SS sets a fixed
Consider the nth step. pattern size of S = 2 for the first step, no matter what
the search parameter p value is. Thus it looks at 9
In the rst phase, we compute the MADs of A, B, locations in a 5x5 window.
and C, where A is the center location, B and C are the
locations s(n) pixels away from A along the The FSS algorithm can be summarized as follows
horizontal and vertical directions, respectively.
Furthermore, we denote MAD (X) as the MAD of
1. A minimum BDM point if found from a nine
location X and label four quadrants I, II, III, and IV
checking point pattern on a 5 X 5 window
as shown.
located at the center of the 15 X 15
searching. If the minimum BDM is found at
The rule for determining a search quadrant can be the center of the search window, then go to
described as follows. step 4 else, go to step 2.
2. The search window size is maintained 5 X 5.
If MAD (A) MAD (B) and MAD (A) MAD (C), I However search pattern will depend on the
is selected; position of the previous minimum BDM. a If
the previous minimum BDM point is located
448
at the comer of pervious search window, performance that is close to that of full search with a
five additional checking points are used. significant speedup.
a. If the previous minimum BDM The DS algorithm is summarized as follows.

point is located at the middle of the
horizontal or the vertical axis of the 1. The initial LDSP is centered at the origin of the
previous search window, three search window, and the 9 checking points of LDSP
additional checking points are used. are tested. If the MBD point calculated is located at
b. If the minimum BDM is found at the center position, go to Step 3; otherwise, go to
the center of the search window, go Step 2.
to step 4 else, go to step 3.
2. The MBD point found in the previous search step
3. The searching pattern strategy is the same as is re-positioned as the center point to form a new
step 2, but it will finally go to step 4. LDSP. If the new MBD point obtained is located at
4. The search window is reduced to 3 X 3 and the center position, go to Step 3; otherwise,
the direction of the overall motion vector is recursively repeat this step.
considered as the minimum BDM point
among these nine searching points. 3. Switch the search pattern from LDSP to SDSP.
The MBD point found in this step is the final solution
The Flow chart of FSS algorithm is as shown in Fig. of the motion vector which points to the best
7. matching block.
5.6 DIAMOND SEARCH (DS) 5.7 ADAPTIVE ROOD PATTERN SEARCH

(ARPS)
DS algorithm is a fast block matching ME algorithm
based on the study of motion vector distribution from ARPS [9] algorithm makes use of the fact that the
several commonly used test sequences while reducing general motion in a frame is usually coherent, i.e. if
the complexity of the Motion Estimation. Compared the macro blocks around the current macro block
with the NTSS algorithm DS requires 22% less moved in a particular direction then there is a high
computation while achieving performance close to probability that the current macro block will also
NTSS. have a similar motion vector.
This algorithm uses the motion vector of the macro

block to its immediate left to predict its own motion
vector. An example is shown in Fig.
Fig 7: Flow chart for Four Step Search

Fig 8. Adaptive Root Pattern: The predicted motion
vector is (3,-2) , and the step size S = Max( |3|, |-2|) =
This algorithm performs better than FSS in terms of
3.
the required number of search points. The result of
applying these algorithms shows a promising
449
6. SIMULATION & RESULTS
The general algorithm used in this process is as

follows.
Step 1: Select a video sequence on which the block

matching algorithms are to be applied.
Step 2: Take the selected video and convert it into
frames. The frames are usually of JPEG format.
Step 3: The JPEG frames are to be converted to raster
files. The raster or bitmap format is used as it has
built in functions in MATLAB.
Step 4: The frames are then divided into macro
blocks. Usually the macro block is taken as a square
of side 16 pixels.
Step 5: A reference block in previous frame is taken
in order to compare it with macro block.
Step 6: The macro block is then compared with Fig 11: Video 1: Comparison of PSNR values
reference block of previous frame and the cost
function at all points is calculated.
Step 7: If a point with least cost exists, go to step 8.
Otherwise, vary the reference block and go to step 6.
Step 8: Encode and transmit the frame.
Fig 12: Sample Frame of Video 2
Fig 9: Sample Frame of Video-1
Fig 13: Video 2:Comparison of no. of computations
Fig 10: Video 1:Comparison of no. of computations Fig 14: Video 2: Comparison of PSNR values
450
algorithms, with their comparative study at the end.

Three more, very recent, block matching algorithms
are studied in the end as part of literature review. Of
the various algorithms studied or simulated, ARPS
turns out to be the best block matching algorithm.
This can be inferred by looking at the comparison of
computations and PSNR values of the three videos
presented by the tables below.
Technique Video 1 Video 2 Video 3
(Caltrain) (Static (Dynamic

background) background)
Fig 15: Sample Frame of Video 3
Exhaustive
Search
207.41 202.0485 202.0485
Three Step
Search
23.8000 24.0939 23.9121
New
Three Step 26.0855 29.9879 29.5212
Search
Simple
and
Efficient
Search 16.1873 17 17
Four-Step
Fig 16: Video3: Comparison of no. of computations Search
19.9055 24.5909 23.9121
Diamond
18.8418
Search
28.7636 27.5242
Adaptive
Rood
Pattern
Search 10.3127 14.3364 15.5303
Fig 18:Comparison of computations for Video1,2

&3
Fig 17: Video 3: Comparison of PSNR values Video 2 Video 3

Video 1
Techniqu
7.CONCLUSION e (Static (Dynamic
(Caltrain
backgroun backgroun
)
d) d)
Block matching techniques are the most popular and
efficient of the various motion estimation techniques. Exhaustiv
This paper first describes the motion compensation e Search
based video compression in brief. It then illustrates 29.6157 32.1256 25.3361
and simulates 7 of the most popular block matching
451
Three [8] Chun-Ho Cheung, and Lai-Man Po, A Novel Small Cross-
Diamond Search Algorithm for Fast Video Coding and Video
Step Conferencing Applications, Proc. IEEE ICIP, September 2002.
Search 27.2403 31.4614 24.1004
New [9] Chun-Ho Cheung, and Lai-Man Po, A Novel Cross-Diamond
Three Search Algorithm for Fast Block Motion Estimation, IEEE Trans.
Step Circuits And Systems For Video Technology, vol 12., no. 12, pp.
Search 28.9353 31.9846 23.8800 1168-1177, December 2002.
Simple
and
Efficient
Search 27.011 30.8501 23.5710
Four-Step
Search
29.3966 31.6058 24.1317
Diamond
Search
29.4049 31.8737 23.9882
Adaptive
Rood
Pattern
Search 29.3871 31.8748 24.5474
Fig 19: Comparison of PSNR values for Video 1, 2 &

3
REFERENCES
[1] Z. Ahmed, et al., "Mean Predictive Block Matching (MPBM)

for fast block-matching motion estimation," presented at the 3rd
European Workshop on Visual Information Processing (EUVIP),
Paris, France, 2011.
[2] Borko Furht, Joshua Greenberg, Raymond Westwater, Motion

Estimation Algorithms For Video Compression. Massachusetts:
Kluwer Academic Publishers, 1997. Ch. 2 & 3.
[3] M. Ghanbari, Video Coding, An Introduction to Standard

Codecs, London: The Institute of Electrical Engineers, 1999. Ch.2,
5, 6, 7 & 8
[4] Iain E. G. Richardson, Video Codec Design, West Sussex: John

Wiley & Sons Ltd., 2002, Ch. 4, 5, & 6.
[5] C. W. Lam, L. M. Po and C. H. Cheung, "A New Cross-

Diamond Search Algorithm for Fast Block Matching Motion
Estimation", Proceeding of 2003 IEEE International Conference on
Neural Networks and Signal Processing, pp. 1262-1265, Dec.
2003, Nanjing, China.
[6] Shan Zhu, and Kai-Kuang Ma, A New Diamond Search

Algorithm for Fast Block-Matching Motion Estimation, IEEE
Trans. Image Processing, vol 9, no. 2, pp. 287-290, February 2000.
[7] Yao Nie, and Kai-Kuang Ma, Adaptive Rood Pattern Search
for Fast Block-Matching Motion Estimation, IEEE Trans. Image
Processing, vol 11, no. 12, pp. 1442-1448, December 2002.
452
Spatial domain image enhancement for roi

Using local parameterized hybrid model
1
I.Suneetha and 2Dr.T.Venkateswarlu
Abstract- Images play a vital role to provide information Sections II and III cover related work done about
to the viewers in every field i.e. medical images for linear PGI model and nonlinear PAR model.
doctors, forensic images for police investigation, text
images for readers etc. In the process of image
II.LINEAR PGI MODEL
acquisition, contrast of an image becomes poor because of
lighting, weather, distance, or equipment used for image
capture. Noise corrupts the images during sensing with Relation between the input image and output
malfunctioning cameras, storing in faulty memory image in a linear PGI model is
locations or sending through a noisy channel. Sometimes 0x<
g(x, y) = G f(x, y) + I
quality of the image in some areas may be corrupted by 0y<
poor contrast or unwanted noise or by both. This paper
proposes a method for image enhancement for ROI
through contrast improvement or noise suppression or by
both by using a Local Parameterized Hybrid Model in
spatial domain. The proposed method provides good
results subjectively as well as objectively for both gray
scale and true color images. The proposed method is
better, faster, and also useful for interactive image
processing applications as it provides various
enhancement images for an image.
Key Words-Local Parameterized Gradient Intercept

(LPHI), Region of Interest (ROI), Mean Square Error
(MSE), Peak Signal to Noise Ratio (PSNR), and Digital
Image Processing(DIP).
I.INTRODUCTION
Spatial domain image enhancement improves digital

image quality without knowing the source of
degradation by direct modification of pixels and
provides visually acceptable images for human
viewers and/or automated image processing
techniques. We reviewed enhancement techniques for
gray scale images in spatial domain and implemented
them using MATLAB [1]. These techniques have
been extended successfully to true color images [2].
Image enhancement through noise suppression can be
done using a Nonlinear Parameterized Adaptive
Recursive (PAR) model [3]. Image enhancement
through contrast improvement can be done by using a
Linear Parameterized Gradient Intercept (PGI) model
[4]. Linear and nonlinear models work well when an
image is corrupted by either poor contrast or
unwanted noise, but fails when corrupted by both. Fig. 1: (a) Man image, darken image, THE image,
This paper proposes a method for image enhancement AHE image and PGI image (b) Their Histograms
through contrast improvement or noise suppression or
both using a Local where G is Gradient and I is Interception of the
transformation. G and I values can be zero, positive,
or negative. When G and/or I values are varied for
Parameterized Hybrid model in spatial domain. The improving the image contrast, above transformation
type of noise considered is salt and pepper noise. becomes simple linear or nonlinear but not
453
exponential or logarithmic as in traditional point IV.PROPOSED METHOD

processing methods and does not require PDF
calculations as in histogram processing operations. PGI model improves contrast with smallest
PGI model works well for a gray scale image and mean square error and low computational time. PAR
results are much more pronounced for true color model suppresses noise with highest PSNR and low
image by preserving maximum color details. Results computational time. Hence linear PGI model and
indicate that mean square error (MSE) and nonlinear PAR model can be combinable to enhance
Computational time (tc) of PGI method is smaller an image to that is corrupted by both poor contrast
when compared to Traditional Histogram and unwanted noise. As we are combining a linear
Equalization (THE) and Adaptive Histogram and nonlinear model, the resultant model can be
Equalization (AHE) methods. named as Local Parameterized Hybrid Model (
LPHM) in which G and/or I, A and/or R are varied.
Table 1: MSE and tc for Man image The proposed LPHM has smallest MSE and
highest PSNR at low computational cost in spatial
domain. The following are the steps involved in PHM
algorithm simulation.
1. Consider a good contrast and noiseless

image i(x,y).
2. fc(x,y) is a corrupted image of i(x,y).
III.NONLINEAR PAR MODEL 3. Divide fc(x,y) into two sub images
Based on ROI:f1(x, y) and f2(x, y).
The relation between input image and output
4. If f1(x, y) is corrupted by unwanted noise go
image for nonlinear PAR model is to step 5, or go to step11
5. For f1(x, y), select values of A1and R1.
g(x,y) = imed[fn(x,y)]
6. Pad all sides of the image with noisy image
with (A1-1)/2 zeros to get fp1(x, y)
where imed means Intentional median filter that
7. If 0<fp1(x, y) < L-1, go to next pixel.
performs filtering to noisy pixels intentionally. Let A
Perform imed filtering to get gp1(x, y).
be the window size that is adaptive and R be the
8. If gp1(x, y) is noisy, vary A1 and/or R1.
Recursive order. A and/or R can be varied. Results
9. If A1 varies go to step 10 otherwise go to step
indicate that PAR method has small tc and high
11.
PSNR when compared to TMF, RMF, and AMF.
10. Remove the padded zeros in gp1(x,y)
to get denoisy image g1(x,y).
Table 2: PSNR and tc for Man image
11. If f1(x, y) is corrupted by poor contrast go to
step12, or go to step14
12. Multiply f1(x,y) with G1 and add with I1 to
get g1(x,y).
13. If g1(x,y) is not good in contrast change G1
and/or I1, then go to step5
Get good contrast image g1(x,y).
14. If f1(x, y) is corrupted by both unwanted
noise and poor contrast go to steps 6 to 10
and then 12 , and 13 to get enhanced image
g1(x,y)
15. Repeat the steps 5 to 14 for f2(x, y) to get
enhanced image g2(x,y)
16. Get the enhanced output image g(x,y) from
g1(x,y) and g2(x,y).
V. RESULTS
The LPHM performance can be verified by not only

by visual inspection of the resultant images but also
by evaluating the mean square error and Peak Signal
to Noise Ratio in decibels (PSNR) [5-7]. Decrement
454
in mse and Increment in PSNR for cameraman image VI.CONCLUSIONS

are 1.25x10-9 and 25dB respectively. Decrement in Spatial Domain Image enhancement using Local
mseR, mseG, mseB and Increment in PSNRR, PSNRG, Parameterized Hybrid Model has been successfully
PSNRB for Lena image are 1.23x10-9 ,1.28x10- implemented using MATLAB. This paper considers
9
,1.24x10-9 and 22dB, 24dB, 21dB respectively The gray scale and true color images from different fields.
subjective results and objective results are shown in Choice of A and R depends on noise intensity where
the following figures. as choice of G and I depend on amount of poor
contrast. As proposed algorithm is a faster and better,
1 LPHM can be used as a tool for Photo editing
= 20
software like Photoshop or any existing image
processing software by attaching two sliding bars for
1 A and R that suppresses noise and two sliding bars
= [ ( , ) ( , )] for G and I that improves contrast. The LPHM model
can be used for suppressing high level salt and pepper
noise or other types of noises with slight changes in
algorithm. Future scope will be the development of
LPHM for binary images derived from gray scale and
bicolor from true color images when they are
corrupted differently in various regions.
REFERENCES
[1] Ms. I.Suneetha and Dr.T.Venkateswarlu,
Enhancement Techniques for Gray scale Images in
Spatial Domain, International Journal of Emerging
Technology and Advanced Engineering, website:
www.ijetae.com(ISSN 2250-2459) Volume 2, Issue 4,
April 2012, pp.13-20.
[2] Ms. I.Suneetha and Dr.T.Venkateswarlu,
Enhancement Techniques for True Color Images in
Spatial Domain, International Journal of Computer
Science & Technology (IJCST), Website:
www.ijcst.com(ISSN 0976-8491) Volume 3, Issue 2,
Version 5, April - June 2012, pp. 814-820.
[3] Ms. I.Suneetha and Dr.T.Venkateswarlu, Image
Enhancement Through Noise Suppression Using
Nonlinear Parameterized Adaptive Recursive Model,
International Journal of Engineering Research and
Applications (IJERA), Website: www.ijera.com (ISSN
2248-9622), Volume 2, Issue 4, July-August 2012, pp.
1129-1136.
[4] Ms. I.Suneetha and Dr.T.Venkateswarlu, Image
Visual inspection of subjective results Enhancement Through Contrast Improvement Using
indicates that, the Local Parametric Hybrid Model Parameterized Gradient Intercept Model, ARPN Journal
works very well by enhancing gray scale and true of Engineering and Applied Sciences (ARPN-JEAS),
color images that are corrupted by decreased contrast Website:www.arpnjournals.com (ISSN 1819-6608),
Volume 7, No. 8, August 2012.
and unwanted noise. Visual inspection of objective [5] J Rafael C Gonzalez, Richard E. Woods, and Steven L.
results shows that, MSE was decreased and PSNR Eddins, Digital Image Processing Using MATLAB
was increased for the gray scale images and also for (Second Edition, Gates mark Publishing, 2009).
the R, G, and B components of true color images. The [6] J. Y. im, L. S. Kim, S. H Hwang, An advanced Contrast
Enhancement Using Partially Overlapped SubBlock
limitation in the proposed model is small decrement Histogram Equalization, IEEE Transactions on Circuits
in mse and small improvement in PSNR for and Systems for Videc Technology, Vol. 11, No. 4,
enhancing gray scale and true color images that are pp.475-484,2001.
corrupted by increased contrast and unwanted noise. [7] Prof. A. Senthilrajan, Dr. E. Ramaraj, High Density
Impulse Noise Removal in Color Images Using Region of
The reason is, while increasing contrast for getting Interest Median Controlled Adaptive Recursive Weighted
simulation results some pixel values reach maximum Median Filter, Proceedings of the International
value which are then treated as salt during denoising. MultiConferenceof Engineers and Computer Scientists
Therefore resultant images are not having very good (IMECS), Vol. II, March 17-19, 2010, Hong Kong.
contrast. This problem can be overcome by slight
change in LPHM algorithm.
455
Content Based Video Retrieval By Integrating

Shape And Motion Features
GS Naveen Kumar1, VSK Reddy2, S.Srinivas Kumar
1
Research Scholar, JNTUK, Kakinada, India.
2
Professor & Principal, MRCET, Hyderabad, India.
3
Professor and Director R&D, JNTUK, Kakinada
e-mail: gsrinivasanaveen@gmail.com
Abstract-With large amount of multimedia information organized as follows: In Section 2, we remind in

connected to the global information network, effective detail the proposed overall system. Section 3 gives
and efficient video retrieval from large video repositories the Shape representation and CSS matching follows.
has become an imminent research issue. Since visual Section 4 gives the Trajectory representation. Section
media requires large amounts of storage and processing, 5 gives the experimental results. Section 6 concludes
there is a need to efficiently index, store, and retrieve the the paper.
visual information from multimedia database. Content-
based search and retrieval of video data has become
a challenging and important issue. In this paper, we II.PROPOSED OVERALL SYSTEM
present novel methods for content based video retrieval by
integrating shape and motion features. We have used Our system integrates two important features: shape
Curvature scale space for shape representation and and motion of the video object to retrieve video shots
Polynomial curve fitting for trajectory representation and containing video objects with similar shape and
retrieval. The shape representation is invariant to motion properties. Figure 2.1 shows the overall
rotation, scaling, translation, and robust with respect to flowchart.
noise. Trajectory matching incorporates size dissimilarity,
velocity dissimilarity and visual distance, for retrieval.
The cost of matching two video objects is based on shape
and motion features, to retrieve similar video shots. We
have tested our system on standard synthetic databases.
We have also tested our system on real world databases.
Experimental results have proved the superiority of our
proposed method.
I.INTRODUCTION
Along with the rapid development of information

technology, more and more multimedia information
is available [1], especially for video data. Then a new
problem arises: how to get the interested video from
the vast amount of video data. Keywords based
traditional retrieval method cannot work efficiently
anymore. There is an urgent need for effective video
retrieval method [2]. Therefore, Content Based Video
Retrieval (CBVR) has become a hot research topic in
Figure 2.1. Overall system for Content-based video
recent years. Content-based Video Retrieval (CBVR)
retrieval
systems appear like a natural extension (or merge) of
Content-based Image Retrieval (CBIR) systems. The preprocessing step needed to extract features
Color, size, shape, texture and motion were used as from the video shot is explained below. In our
features for video retrieval in VideoQ [3]. Shape and system, background subtraction has been used to
motion are important features in content based video segment foreground moving objects from the video.
retrieval. Gaussian mixture model is used to model the
In our system Curvature Scale Space has been used background as described in [8]. After extracting
for representing shapes [4] and Polynomial Curve foreground objects any representative frame is
fitting for representing motion trajectory [5]. Our chosen, from which the shape of video object is
system integrates both these features to retrieve extracted and represented. Centroid of the foreground
similar video objects. The rest of the paper is objects are used to extract the motion trajectory of the
456
moving video object. Figure 2.2 shows an example of 5. If we determine the locations of curvature zero
a frame from a video shot and its corresponding crossings of every during evolution. We can
extracted shape contour. display the resulting points in (u, ) plane.
Where u is normalized arc length and is width
or standard deviation of the Gaussian kernel.
6. The process starts with =1 & at each level is
increased by , chosen as 0.1. As increases,
repeat steps 3 to 5, then shrinks and becomes
smoother & the number of curvature zero
crossing points on it decreases.
7. Finally when is sufficiently high will be a
convex curve with no curvature zero crossing
Figure 2.2. (a) Sample frame from a video shot (b) points.
Extracted shape contour from frame (a). 8. The result of this process can be represented as a
binary image called the CSS image.
After feature extraction, the shape and motion 9. We find the peaks of both branches of a contour
trajectory of the video object are represented using in the CSS image and consider the midpoint of
two different techniques standardized in MPEG-7: the line segment joining the pair as a maximum
Curvature Scale Space for shapes and Polynomial of the CSS image. The maxima of CSS image is
Curve fitting for motion trajectory. The shape and indicates shape representation. In our system, if a
motion features are stored in the video database. For maximum is less than 1/6 of the largest
a query video, shape and motion features are maximum of the same CSS image, it is
extracted from the query. The represented query considered as noise or small ripples.
shape is matched with the shape feature of a model
from the database using CSS based matching. The CSS image
features of the object in the query are matched against
the features of the stored objects in the database. The The CSS representation is robust with respect to
features of the object in the query are matched against scale, noise and change in orientation. A rotation of
the features of the stored objects in the database. The the object usually causes a circular shift on its
video shots corresponding to the video objects are representation, which is easily determined during the
rank ordered and retrieved according to their matching process (compare Fig.3.1a and b). Note that
computed mathcosts. the effect of a change in the starting point is also the
same. Due to arc length normalization, scaling does
III.SHAPE REPRESENTATION AND MATCHING not change the representation, and as Fig. 3.1 shows,
noise may create some small contours on the CSS
Algorithm for Shape Representation image, but the main contours and therefore the
corresponding maxima remain unaffected.
1. Every object contour is represented by x and y Compactness is another aspect of the CSS
coordinates of its boundary points representation. A shape is represented by about less
2. To normalize the arc length, the boundary is than ten pairs of integer values which can be
resample and represented by 200 equally distant determined without any ambiguity.
points.
3. The curve is then smoothed by convolution of
contour with a Gaussian function. The smoothed
curve is called , where denotes the width of
the Gaussian kernel g(u, ) here u is the
normalized arc length, is the width of the
Gaussian kernel
g(u, ) = e

(1)
4. To locations of curvature zero crossings on
are determined at different levels of scaling using
Fig 3.1: a) A boundary and its CSS image. b) Change
( ,) ( ,) ( ,) ( ,)
k(u, ) = in orientation causes a circular shift in CSS image. c)
( ( ,) ( ,) )
Noise creates small contours in CSS
(2)
457
The matching algorithm which compares two sets of 5. To expand a node, select the largest scale image
representations and assigns a match value as the curve CSS maximum (which is not in the first list)
measure of similarity between the shapes is also and apply that nodes shift parameter to map that
simple and fast. Another property of the CSS image maximum to the model CSS image. Locate the
is that it retains the local properties of the shape. nearest model curve CSS maximum (which is not in
Every contour of the CSS image corresponds to a the second list). If the two maxima are in a reasonable
concavity or a convexity of the shape. A local horizontal distance (0.2 of the maximum possible
deformation of the shape mainly causes a change in distance), define the cost of the match as the straight-
the corresponding contour of the CSS image. Using line distance between the two maxima. Otherwise,
this property, one can include more local information define the height of the image curve CSS maximum
about the shape in the CSS image. For example, it is as the cost of the match. If there are no more image
possible to use the average curvature of a segment curve CSS maxima left, define the cost of match as
together with the maximum of its CSS contour. the height of the highest model curve CSS maximum
Alternatively, one can find the corresponding corners not in the nodes second list. Likewise, if there is no
of each pair of curvature zero-crossings. In shape more models curve CSS maxima left; define the cost
similarity retrieval, the task is not to accurately of match as the height of the selected image curve
recognize the input among the existing models, but it maximum. Note that these cases may occur when the
is to find the most similar models to the input and number of image maxima is different from the
rank them in terms of a similarity measure. The CSS number of model maxima. Add the match cost to the
representation is a reliable tool to handle this task. node cost. Update the two lists associated with the
node.
CSS matching 6. Select the lowest cost node. If there are no more
model or image curve CSS maxima that remain
The complete matching algorithm which compares unmatched within that node, then return that node as
the two sets of maxima, one from the image and the the lowest cost node. Otherwise, go to step 5 and
other from the model is as follows. expand the lowest cost node.
1. Create a node consisting of the largest scale 7. Reverse the place of the image and the model and
maximum of the image and the largest scale repeat steps 1 to 6 to find the lowest cost node in this
maximum of the model. Initialize the cost of this case.
node to the absolute difference of -coordinates of 8. Consider the lowest node as the final matching cost
the image and the model. Compute a CSS shift between the image and the model. Using this
parameter for each node: algorithm and considering its amendment, the system
= Um Ui ; where U is the horizontal coordinate of associates a matching value to every candidate and
a maximum, and i and m refer to image and model, then displays the n best matched as its output, where
respectively. This parameter is used to compensate n has already been selected by the user.
the effect of different start points or change in
orientation. IV.TRAJECTORY REPRESENTATION
2. If there are more than one maximum in the model
which have a -coordinate close (within 80%) to the The motion trajectory of a video object is extracted
largest scale maximum of the image, create extra from the video shot by finding the centroid of the
nodes consisting of the largest scale maximum of the moving object. A Polynomial curve is fitted for every
image and that respective additional maximum of the sub trajectory [7]. The sequence of steps involved for
model. Also create the same nodes for the second trajectory representation and matching are briefly
largest scale maximum of the image and the explained below.
respective maxima of the model. Initialize the cost Algorithm for Trajectory Representation
and compute the CSS shift parameter for each node
accordingly. 1. A sampling method is first used to detect all
3. Create two lists for each node obtained in steps 1 key points with high curvatures from the
and 2. The first list will contain the image curve trajectory for capturing its geometrical features.
maxima and the second list will contain the model 2. High curvature points are points on the
curve maxima matched within that node at any point trajectory whose angle () is less than a certain
of the matching procedure. Initialize the first and threshold (T). Angle () at a key point is
second list of each node by the corresponding computed by selecting two specified points,
maxima determined in the first two steps. which lie on either side within a certain range
4. Expand each node created in steps 1 and 2 using of distance (dmin and dmax) from the keypoint.
the procedure described in step 5. 3. As we start from the initial key point of the
trajectory, whenever there is a clockwise
458
direction change followed by an anticlockwise into two sub trajectories (c) Key points regenerated
direction change or vice versa, the trajectory is (d) Polynomial curve fitted on the input trajectory.
divided into sub trajectories at that key point.
4. To improve the accuracy of curve fitting, if These key points are used as control points for curve
distance between any two control points are fitting. To improve the accuracy of curve fitting, if
greater than some threshold (> Td) in a sub distance between any two control points are greater
trajectory, their middle point is selected as a than some threshold (> Td) in a sub trajectory, their
new control point. middle point is selected as a new control point. A
5. A polynomial curve is fitted through the polynomial of second order is fitted through the
control points for every sub trajectory. control points for every sub trajectory as shown in
V.IMPLEMENTATION OF POLYNOMIAL dotted lines in Fig. 4d. The coefficients of the
CURVE FITTING polynomials along with their temporal occurrences
are stored for every sub trajectory.
The motion trajectory of a video object is extracted
from the video shot by finding the centroid of the Trajectory matching between the query (q) and model
moving object. Polynomial curve fitting is used for (m) in the database is done based on their position,
trajectory representation as standardized in MPEG-7 velocity and size differences. For similarity
[6]. Trajectory is divided into sub trajectories based comparison, the sampled data points from fitted
on the key points detected as described in [7]. A polynomial curve are normalized to maintain scale
Polynomial curve is fitted for every sub trajectory. and shift invariance. Position difference (Cp) is
The sequence of steps involved for trajectory calculated from the Euclidean distance between the
representation and matching are briefly explained positions of query and model trajectories as:
below. A sampling method is first used to detect all
key points with high curvatures from the trajectory Trajectory matching
for capturing its geometrical features. High curvature
points are points on the trajectory whose angle () is Trajectory matching between the query (q) and model
less than a certain threshold (T). Angle () at a key (m) in the database is done based on their position,
point is computed by selecting two specified points, velocity and size differences.
which lie on either side within a certain range of
distance (dmin and dmax) from the keypoint. The Position difference (Cp) is calculated from the
high curvature points are further eliminated for better Euclidean distance between the positions of query
curve fitting such that two key points lie far from and model trajectories as:
each other (> dmin). The direction change at the key N 1
1
points is detected to divide the trajectory into sub Cp
N
{( x
K 0
q (k ) xm (k )) 2 ( y q (k ) ym (k )) 2 }
trajectories. Clockwise or anticlockwise direction
change is determined using the two adjacent key (1)
points which lie on both sides of the key point. As we Similarly, velocity difference (Cv) is given by,
N 1
1
{(v (k ) v
start from the initial key point of the trajectory,
whenever there is a clockwise direction change Cv x
q
x
m (k )) 2 (vqy (k ) vmy (k )) 2 }
N K 0
followed by an anticlockwise direction change or vice
versa, the trajectory is divided into sub trajectories at (2)
that key point. This segmentation is done for accurate Where vx and vy are the x and y components of the
representation and easier partial trajectory matching. velocity vector and N is the number of sampled
Figure 4.1 shows an example trajectory. High points. Size difference (Csz) is given by,
curvature points are marked on the input trajectory in 2 | size(q ) size(m) |
Csz
Fig. 4.1 a. The segmented sub trajectories with their size(q ) size(m)
key points are shown in Fig. 4.1 b.
(3)
Where size (q) = max (max (xq) min (xq), max (yq)
min (yq)). The three distances are integrated to give
the dissimilarity between the trajectories as follows:
Ct = wt1 Cp + wt2 Cv + wt3 Csz
(4)
Figure 4.1. (a) Input trajectory with key (high
curvature) points marked (b) Trajectory segmented
459
Experimental Results for Video Retrieval Performance Measurements
Integration of shape and motion features is done to The precision-recall graph averaged over 200 queries
retrieve video shots with similar shape and motion. for the three methods is shown in Fig. 5.3. The
The matchcosts obtained from CSS matching for precision-recall graph of our method outperforms the
shape (Cs) and trajectory matching (Ct) are other two methods based on only shape [6] or
normalized and summed up with appropriate weights trajectory [7].
as follows:
Precision: In the field of video retrieval, precision is
C = ws *Cs + wt * Ct the fraction of video that are relevant to the search. A
(5) good retrieval system should only retrieve relevant
items.
ws and wt are weights assigned for shape and
trajectory respectively.
The synthetic database used for our experiments

consists of 200 shapes from MPEG7-B and 200
trajectories. The shape and trajectory pair database Recall: Recall in video retrieval is the fraction of the
consists of 20 classes with 10 shape-trajectory pairs documents that are relevant to the query that are
per class. We have evaluated the performance of our successfully retrieved. A good retrieval system
system using Precision vs. Recall ratio. We have should retrieve as many relevant items as possible.
compared our system with (i) only shape (CSS) [5]
and (ii) only motion (Polynomial Curve fitting) [7]
used as a feature for video retrieval for the same
database.
Shape Feature Extraction
In the input video after extraction of object contours

then extracted shape representation features using
curvature scale space. In the below Figure 5.1 (a) is
indicates smoothing the contour with convolution of
various Gaussian kernels. Figure 5.1(b) is CSS image (a)
plot between arc length (200 samples) Vs Gaussian
standard deviation.
Figure 5.1 : boundary and its CSS image for different (b)
values
Figure 5.2: a) Query video b) Retrieved videos
460
[6] N. Dimitrova and F. Golshani, Motion recovery for video

content classification, ACM Transactions on Information
Systems, vol. 14, no. 13, pp. 408439, 1995.
[7] S. Jeanin and A. Divakaran, MPEG-7 Visual Motion
Descriptors, IEEE Transactions on Circuits and Systems for
Video Technology, vol. 11, no. 6, pp. 720724, June 2011.
[8] P. KaewTraKulPong and R. Bowden, An Improved
Adaptive Background Mixture model for real-time tracking
with shadow detection, in 2nd European Workshop on
Advanced Video Based Survey.
Figure 5.3. Performance of Precision-recall
We have also tested our system on real world

databases consisting of 250 real video shots. The
system performs well for real world databases.
Retrieved results will depend on the database chosen.
VI.CONCLUSION AND FUTURE SCOPE
We have proposed a system for content based video

retrieval based on shape and motion. Curvature scale
space has been used for shape representation and
Polynomial Curve fitting for trajectory
representation. The two features are integrated by
combining their matchcosts. We have tested our
system on synthetic and
real world databases. Experimental results have

shown good performance. This work will be extended
to develop retrieval applications based on high level
semantic query combined with multiple features.
Enlarging our database and testing more versatile
data.
REFERENCES
[1] Weiming hu, Nianhua Xie, Li Li, Xianglin Zeng, and

Stephen Maybank, " A Survey on Visual Content-Based
Video Indexing and Retrieval," IEEE transactions on
systems, man, and cybernetics-part c: applications and
reviews 2011, pp. 1-23
[2] M. S. Lew, N. Sebe, C. Djeraba, and R. Jain, Content-based
multimedia information retrieval: State of the art and
challenges, ACM Trans. Multimedia Comput., Commun.
Appl., vol. 2, no. 1, pp. 119, Feb. 2006.
[3] S.F.Chang and et.al, A fully automated content-based video
search engine supporting spatiotemporal queries, IEEE
Transactions on Circuits and Systems for Video Technology,
vol. 8, no. 5, pp. 602615, September 2008.
[4] L. J. Latecki, R. Lakamper, and U. Eckhardt, Shape
descriptors for non-rigid shapes with a single closed
contour, in IEEE Conference on Computer Vision and
Pattern Recognition, 2009, pp. 424429.
[5] F. Mokhtarian and M. Bober, Curvature Scale Space
Representation: Theory, Applications and MPEG-7
Standardization. The Netherlands: Kluwer Academic
Publishers, 2006.
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ACHARYA NAGARJUNA UNIVERSITY

NAGARJUNA NAGAR - 522 510, GUNTUR (DT), A.P., INDIA

Prof. K. Viyyanna Rao Phone: off: 0863-2346101, 2293238

VICE CHANCELLORS MESSAGE

Prof. K. Viyyanna Rao

Prof. Y. P. Rama Subbaiah Phone: off: : 0863 - (O)23462293238

Prof. Y. P. Rama Subbaiah.

Prof. R.R.L. Kantam Phone: off: 0863-2346102

I Congratulate the University College of Engineering and Technology, for Organizing an

I am happy to know that the Electronics and Communication Engineering

The deliberations in the areas of Signal processing techniques,

I wish the conference a grand success.

(G. SATHEESH REDDY)

Date: 09 Dec. 2013

It gives me immense pleasure to know that Acharya Nagarjuna University college of

The conference focuses on an activity which is of great National importance. Precision

I wish this conference a grand success.

Sd/- (SURANJAN PAL)

Prof. P. Siddaiah Phone: off: 0863-2346251

I take this opportunity to particularly thank Sri.Avinash Chander SA to RM, to be the

Prof. E. Sreenivasa Reddy Phone: off: 0863-2346252

I wish the seminar a grand success.

Prof. E. Sreenivasa Reddy

SESSION-III : SATELLITE COMMUNICATIONS-I

SESSION-IV: WIRELESS SENSOR NETWORKS-I

SESSION-V: IMAGE PROCESSING-I

SESSION-VII : VLSI & FUZZY LOGIC-II

SESSION-VIII: SIGNAL PROCESSING

SESSION-IX: COMMUNICATION SYSTEMS-I

SESSION-X : WIRELESS SENSOR NETWORKS-II

SESSION-XII: NAVIGATIONAL SYSTEMS & RADAR-II

SESSION-XV: IMAGE PROCESSING-II

Software Designing and Implementation of

Due to the advancement in technology for fast floating

Figure 3. Carry Save Adder

III. WALLACE TREE MULTIPLIER

Wallace established the fast process for multiplication

TABLE I. STEP 1 IN BOOTH ALGORITHM

3. Shift A left by one bit. So, A becomes 00010010

Figure 7. Flow Chart to obtain carry save adder components

Figure 6. Flow chart of method 2 of array multiplier

Figure 8. Flow Chart of Booth Multiplier

The results of multipliers are obtained using Xilinx ISE

1) Array Multiplier (8x8 Multiplication) :-

Figure 9. 8x8 multiplication using Array Multiplier

2) Wallace Tree Multiplier:-

Figure 10. Carry Save Adder

[2] Lakshmanan, Masuri Othman, Mohamad Alauddin Mohd.Ali,

[1] Shugang Wei, Shuangching Chen, Kensuke Shimizu, Fast

Real Time Implementation of DSP Based

The physical parameters for our example are:

The torque developed by motor is proportional to the

III. FUZZY LOGIC CONTROLLER

E t K . 5 Fuzzy logic control is a control algorithm based on

Fig. 3.Basic Block Diagram For FLC

D. Fuzzy Controller Design

The Fuzzy Logic Controller is an seven step

The most important things in fuzzy logic control

fuzzy controller and one output (z) . For this, a two-

In this paper as mentioned in table1,it consist of two

IV. SIMULATION RESULTS

A fuzzy logic based controller for a D.C.

REFERENCES Dr. A. GuruvaReddy is working as Professor, Dept. of