ADAPTIVE WIENER FILTERING APPROACH FOR SPEECH

ENHANCEMENT

M. A. Abd El-Fattah*, M. I. Dessouky , S. M. Diab and F. E. Abd El-samie #

Department of Electronics and Electrical communications, Faculty of Electronic Engineering
Menoufia University, Menouf, Egypt
E-mails: * maro_zizo2010@yahoo.com , # fathi_sayed@yahoo.com

ABSTRACT
This paper proposes the application of the Wiener filter in an adaptive manner in
speech enhancement. The proposed adaptive Wiener filter depends on the adaptation of the
filter transfer function from sample to sample based on the speech signal statistics(mean
and variance). The adaptive Wiener filter is implemented in time domain rather than in
frequency domain to accommodate for the varying nature of the speech signal. The
proposed method is compared to the traditional Wiener filter and spectral subtraction
methods and the results reveal its superiority.

Keywords: Speech Enhancement, Spectral Subtraction, Adaptive Wiener Filter

1 INTRODUCTION

Speech enhancement is one of the most
important topics in speech signal processing.
Several techniques have been proposed for this
purpose like the spectral subtraction approach, the
signal subspace approach, adaptive noise canceling
and the iterative Wiener filter[1-5] . The
performances of these techniques depend on
quality and intelligibility of the processed speech
signal. The improvement of the speech signal-to-
noise ratio (SNR) is the target of most techniques.
Spectral subtraction is the earliest method for
enhancing speech degraded by additive noise[1].
This technique estimates the spectrum of the clean
(noise-free) signal by the subtraction of the
estimated noise magnitude spectrum from the noisy
signal magnitude spectrum while keeping the phase
spectrum of the noisy signal. The drawback of this
technique is the residual noise.
Another technique is a signal subspace
approach [3]. It is used for enhancing a speech
signal degraded by uncorrelated additive noise or
colored noise [6,7]. The idea of this algorithm is
based on the fact that the vector space of the noisy
signal can be decomposed into a signal plus noise
subspace and an orthogonal noise subspace.
Processing is performed on the vectors in the signal
plus noise subspace only, while the noise subspace
is removed first. Decomposition of the vector space
of the noisy signal is performed by applying an
eigenvalue or singular value decomposition or by
applying the Karhunen-Loeve transform (KLT)[8].
Mi. et. al. have proposed the signal / noise KLT
based approach for colored noise removal[9]. The
idea of this approach is that noisy speech frames
are classified into speech-dominated frames and
noise-dominated frames. In the speech-dominated
frames, the signal KLT matrix is used and in the
noise-dominated frames, the noise KLT matrix is
used.
In this paper, we present a new technique to
improve the signal-to-noise ratio in the enhanced
speech signal by using an adaptive implementation
of the Wiener filter. This implementation is
performed in time domain to accommodate for the
varying nature of the signal.
The paper is organized as follows: in section
II, a review of the spectral subtraction technique is
presented. In section III, the traditional Wiener
filter in frequency domain is revisited. Section IV,
proposes the adaptive Wiener filtering approach for
speech enhancement. In section V, a comparative
study between the proposed adaptive Wiener filter,
the Wiener filter in frequency domain and the
spectral subtraction approach is presented.

UbiCC Journal - Volume 3 1

 2 SPECTRAL SUBTRACTION
Spectral subtraction can be categorized as a
non-parametric approach, which simply needs an
estimate of the noise spectrum. It is assume that
there is an estimate of the noise spectrum that is
typically estimated during periods of speaker
silence. Let x(n) be a noisy speech signal :
x ( n) = s ( n) + v ( n)
where s(n) is the clean (the noise-free) signal, and
v(n) is the white gaussian noise. Assume that the
noise and the clean signals are uncorrelated. By
applying the spectral subtraction approach that
estimates the short term magnitude spectrum of the
noise-free signal S (ω ) by subtraction of the
estimated noise magnitude spectrum Vˆ (ω ) from
the noisy signal magnitude spectrum X (ω ) . It is
sufficient to use the noisy signal phase spectrum as
an estimate of the clean speech phase
spectrum,[10]:
Sˆ(ω) = ( X (ω) − Nˆ (ω) ) exp(j∠X (ω))
The estimated time-domain speech signal is
obtained as the inverse Fourier transform of
Sˆ (ω ) .
Another way to recover a clean signal s(n)
from the noisy signal x(n) using the spectral
subtraction approach is performed by assuming
that there is an the estimate of the power spectrum
of the noise Pv (ω ) , that is obtained by averaging
over multiple frames of a known noise segment.
An estimate of the clean signal short-time squared
magnitude spectrum can be obtained as follow [8]:
⎧X(ω) 2 −Pˆv(ω), if X(ω) 2 −Pˆv(ω) ≥0
2 ⎪
Sˆ(ω) = ⎨
⎪ 0, otherwise
⎩
It is possible combine this magnitude spectrum
estimate with the measured phase and then get the
Short Time Fourier Transform (STFT) estimate as
follows:
j∠X (ω )
Sˆ (ω ) = Sˆ (ω ) e
UbiCC Journal - Volume 3
A noise-free signal estimate can then be obtained
with the inverse Fourier transform. This noise
reduction method is a specific case of the general
technique given by Weiss, et al. and extended by
Berouti , et al.[2,12].
The spectral subtraction approach can be
viewed as a filtering operation where high SNR
regions of the measured spectrum are attenuated
(1) less than low SNR regions. This formulation can be
given in terms of the SNR defined as:
2
X (ω )
SNR = (5)
Pˆ v (ω )
Thus, equation (3) can be rewritten as:
2 2
Sˆ (ω ) = X (ω ) − Pˆ v (ω )
−1 (6)
2⎡ 1 ⎤
≈ X (ω ) 1 +
⎢⎣ SNR ⎥⎦
An important property of noise suppression
using spectral subtraction is that the attenuation
(2) characteristics change with the length of the
analysis window. A common problem for using
spectral subtraction is the musicality that results
from the rapid coming and going of waves over
successive frames [13].
3 WIENER FILTER IN FREQUNCY
DOMAIN
The Wiener filter is a popular technique that
has been used in many signal enhancement
methods. The basic principle of the Wiener filter is
to obtain a clean signal from that corrupted by
additive noise. It is required estimate an optimal
filter for the noisy input speech by minimizing the
Mean Square Error (MSE) between the desired
signal s(n) and the estimated signal sˆ( n) . The
frequency domain solution to this optimization
(3)
problem is given by[13]:
Ps(ω)
H(ω) = (7)
Ps(ω) + Pv(ω)
where Ps (ω ) and Pv (ω ) are the power spectral
densities of the clean and the noise signals,
respectively. This formula can be derived
considering the signal s and the noise signal v as
(4)
2
 uncorrelated and stationary signals. The signal-to- Pv (ω ) =σv2
noise ratio is defined by[13]: (10)

Ps (ω ) Consider a small segment of the speech

SNR = (8) signal in which the signal x(n) is assumed to be
Pˆ v (ω ) stationary, The signal x(n) can be modeled by:

This definition can be incorporated to the Wiener x(n) = mx + σxw(n) (11)

filter equation as follows:
where mx and σx are the local mean and standard
deviation of x(n). w(n) is a unit variance noise.
−1
⎡ 1 ⎤ Within this small segment of speech, the
H ( ω ) = ⎢1 + (9)
SNR ⎥⎦
Wiener filter transfer function can be approximated
⎣ by:

The drawback of the Wiener filter is the fixed

Ps (ω ) σs
2

frequency response at all frequencies and the H (ω ) = =

Ps (ω ) + Pv (ω ) σs + σv
2 2
requirement to estimate the power spectral density
of the clean signal and noise prior to filtering. (12)
From Eq.(12), because H (ω ) is constant over the
4 THE PROPOSED ADAPTIVE WIENER
small segment of speech, the impulse response of
FILTER
the Wiener filter can be obtained by:
This section presents and adaptive
implementation of the Wiener filter which benefits
σs
2
from the varying local statistics of the speech h( n) = δ ( n) (13)
signal. A block diagram of the proposed approach σs + σv
2 2

is illustrated in Fig. (1). In this approach, the

estimated speech signal mean mx and variance From Eq.(13), the enhanced speech sˆ( n) within
this local segment can be expressed as:
σx 2
are exploited.

σs
2
A priori knowledge sˆ(n) = mx + ( x(n) - mx ) ∗ δ ( n)
σs + σv
2 2

Space-
σs
2
Degraded speech variant Enhanced
x(n) speech
= mx + ( x(n) − mx )
σs + σv
2 2
h(n)
signal sˆ( n) (14)
If it is assumed that mx and σs are updated at
each sample, we can say:

Measure of
σs (n) ( x(n) − mx(n))
2

Local speech sˆ(n) = mx (n) +

σs (n) + σv
2 2
A priori statistics
knowledge (15)

In Eq.(15), the local mean mx(n) and

Figure 1: Typical adaptive speech enhancement system
for additive noise reduction
( x(n) − mx (n)) are modified separately from
segment to segment and then the results are
combined. If σs is much larger than σv the
2 2

It is assumed that the additive noise v(n) is

output signal sˆ( n) is assumed to be primarily due
of zero mean and has a white nature with variance
to x(n) and the input signal x(n) is not attenuated. If
of σv .Thus, the power spectrum Pv (ω ) can be
2

σs is smaller than σv , the filtering effect is

2 2

approximated by:
performed.

UbiCC Journal - Volume 3 3

 Notice that mx is identical to ms when In the first experiment , all the above-
mentioned algorithms are carried out on the Handle
mv is zero. So, we can estimate mx (n) in Eq.(15) signal with different SNRs and the output PSNR
from x(n) by:
results are shown in Fig. (2). The same experiment
n+M
is repeated for the Laughter and Gong signals and
1
mˆ s (n) = mˆ x (n) = ∑ x(k ) the results are shown in Figs.(3) and (4),
respectively.
(2 M +1) k =n−M
From these figures, it is clear that the proposed
(16) adaptive Wiener filter approach has the best
performance for different SNRs. The adaptive
where ( 2 M + 1) is the number of samples in the Wiener filter approach gives about 3-5 dB
improvement at different values of SNR. The non-
short segment used in the estimation. linearity between input SNR and output PSNR is
due to the adaptive nature of the filter.
To measure the local signal statistics in
the system of Figure 1, the algorithm developed
uses the signal variance σs . The specific method
2

used to designing the space-variant h(n) is given by

(17.b).
Since σx = σs + σv may be estimated
2 2 2
80
from x(n) by:
70

⎧σˆx (n) − σˆv , if σˆx (n) > σˆv

2 2 2 2

σˆs (n) = ⎨
2 60
O u tp u t P S N R (d B )

⎩0, otherwise
50

(17.a)
Where 40

30
n+ M
1
σˆx (n) =
2

(2 M + 1)
∑ ( x(k ) − mˆ (n))
k =n−M
x
2
20
Spectral Subtraction
(17.b)
10 Wiener Filter
By this proposed method, we guarantee that Adaptive Wiener Filter
the filter transfer function is adapted from sample
to sample based on the speech signal statistics. 0
-10 -5 0 5 10 15 20 25 30 35
Input SNR (dB)
5 EXPERIMENTAL RESULTS

For evaluation purposes, we use different Figure 2: PSNR results for white noise
speech signals like the handel, laughter and gong case at-10 dB to +35 dB SNR levels for Handle signal
signals. White Gaussian noise is added to each
speech signal with different SNRs. The different
speech enhancement algorithms such as the
spectral subtraction method, the Weiner filter in
frequency domain and the proposed adaptive
Wiener filter are carried out on the noisy speech
signals. The peak signal to noise ratio (PSNR)
results for each enhancement algorithm are
compared.

UbiCC Journal - Volume 3 4

 reveal that the best performance is that of the
60 proposed adaptive Wiener filter.

1 1
50

A m p lit u d e

A m p lit u d e
0 0

40 -1 -1
O u tp u t P S N R (d B )

0 2000 4000 6000 8000 0 2000 4000 6000 8000

(a) (b)

30 1 1

A m p lit u d e

A m p lit u d e
0 0
20
-1 -1
0 2000 4000 6000 8000 0 2000 4000 6000 8000
Spectral Subtraction (c) (d)
10 Wiener Filter 1

A m p lit u d e
Adaptive Wiener Filter 0
0
-10 -5 0 5 10 15 20 25 30 35 -1
0 2000 4000 6000 8000
Input SNR (dB) (e) Time(msec)

Figure 3: PSNR results for white noise case at -10 dB

to +35 dB SNR levels for Laughter signal Figure 5: Time domain results of the Handel sig. at
SNR = +5dB (a) original sig. (b) noisy sig. (c) spectral
subtraction. (d) Wiener filtering. (e) adaptive Wiener
filtering.
80

70

Amplitude (dB)
Amplitude (dB)

0 0
60 -20 -20
-40
O u tp u t P S N R (d B )

50 -40
0 1000 2000 3000 4000 0 1000 2000 3000 4000
(a) (b)
40
Amplitude (dB)
Amplitude (dB)

0 0
30 -20
-20
20 Spectral Subtraction -40 -40
0 1000 2000 3000 4000 0 1000 2000 3000 4000
Wiener Filter (c) (d)
10
Adaptive Wiener Filter
Amplitude (dB)

0
0
-10 -5 0 5 10 15 20 25 30 35 -20
Input SNR (dB) -40
0 1000 2000 3000 4000
Figure 4: PSNR results for white noise case at -10 dB (e) Freq.(Hz)
to +35 dB SNR levels for Gong signal

The results of the different enhancement Figure 6:The spectrum of the Handel sig. in Fig.(5) (a)
original sig. (b) noisy sig. (c) spectral subtraction. (d)
algorithms for the handle signal with SNRs of 5,
Wiener filtering. (e) adaptive Wiener filtering.
10,15 and 20 dB in the both time and frequency
domain are given in Figs. (5) to (12). These results

UbiCC Journal - Volume 3 5

 A m p lit u d e

A m p lit u d e
1 1

A m p lit u d e

A m p lit u d e
1 1
0 0
0 0
-1 -1
0 2000 4000 6000 8000 0 2000 4000 6000 8000
-1 -1
(a) (b) 0 2000 4000 6000 8000 0 2000 4000 6000 8000
A m p lit u d e

1 A m p lit u d e 1 (a) (b)

A m p lit u d e

A m p lit u d e
1 1
0 0
0 0
-1 -1
0 2000 4000 6000 8000 0 2000 4000 6000 8000 -1 -1
(c) (d) 0 2000 4000 6000 8000 0 2000 4000 6000 8000
A m p lit u d e

(c) (d)

A m p lit u d e
1
1
0 0

-1 -1
0 2000 4000 6000 8000 0 2000 4000 6000 8000
(e) Time (msec) (e) Time(msec)

Figure 7: Time domain results of the Handel sig. at

SNR = 10 dB (a) original sig. (b) noisy sig. (c) spectral Figure 9: Time domain results of the Handel sig. at
subtraction. (d) Wiener filtering. (e) adaptive Wiener SNR = 15 dB (a) original sig. (b) noisy sig. (c) spectral
filtering. subtraction. (d) Wiener filtering. (e) adaptive Wiener
filtering.
Amplitude (dB) Amplitude (dB)
Amplitude (dB) Amplitude (dB)

Amplitude (dB) Amplitude (dB)

Amplitude (dB) Amplitude (dB)

0 0 0 0
-20 -20 -20
-20
-40 -40
-40 -40
0 1000 2000 3000 4000 0 1000 2000 3000 4000 0 1000 2000 3000 4000 0 1000 2000 3000 4000
(a) (b) (a) (b)
0 0 0 0
-20 -20 -20 -20
-40 -40
0 1000 2000 3000 4000 -40 -40
0 1000 2000 3000 4000 0 1000 2000 3000 4000 0 1000 2000 3000 4000
(c)
Amplitude (dB)

(d) (c)
Amplitude (dB)

(d)
0
0
-20
-20
-40
-40
0 1000 2000 3000 4000 0 1000 2000 3000 4000
(e)Freq. (Hz) (e)Freq. (Hz)

Figure 8: The spectrum of the Handel sig. in Fig.(7) Figure 10: The spectrum of the Handel sig. in Fig.(9)
(a) original sig. (b) noisy sig. (c) spectral subtraction. (d) (a) original sig. (b) noisy sig. (c) spectral subtraction. (d)
Wiener filtering. (e) adaptive Wiener filtering. Wiener filtering. (e) adaptive Wiener filtering.

UbiCC Journal - Volume 3 6

 6 CONCLUSION
1 1

A m p lit u d e

A m p lit u d e
An adaptive Wiener filter approach for
0 0 speech enhancement is proposed in this papaper.
This approach depends on the adaptation of the
-1 -1
0 2000 4000 6000 8000 0 2000 4000 6000 8000 filter transfer function from sample to sample
(a) (b) based on the speech signal statistics(mean and
variance). This results indicates that the proposed
1 1 approach provides the best SNR improvement
A m p lit u d e

A m p lit u d e
0 0 among the spectral subtraction approach and the
traditional Wiener filter approach in frequency
-1 -1 domain. The results also indicate that the proposed
0 2000 4000 6000 8000 0 2000 4000 6000 8000 approach can treat musical noise better than the
(c) (d) spectral subtraction approach and it can avoid the
1 drawbacks of Wiener filter in frequency domain .
A m p lit u d e

0 REFERENCES

-1 [1] S. F. Boll: Suppression of acoustic noise in

0 2000 4000 6000 8000 speech using spectral subtraction, IEEE Trans.
(e) Time(msec) Acoust., Speech, Signal Processing, vol. ASSP-27,.
pp. 113-120 (1979).
Figure 11: Time domain results of the Handel sig. at [2] M. Berouti, R. Schwartz, and J. Makhoul:
SNR = 20 dB (a) original sig. (b) noisy sig. (c) spectral Enhancement of speech corrupted by acoustic
subtraction. (d) Wiener filtering. (e) adaptive Wiener noise, Proc. IEEE Int. Conf. Acoust., Speech
filtering. Signal Processing, pp. 208-211 (1979).
Amplitude (dB)
Amplitude (dB)

0
[3] Y. Ephriam and H. L. Van Trees: A signal
0
subspace approach for speech enhancement, in
-20 -20 Proc. International Conference on Acoustic,
-40 Speech and Signal Processing, vol. II, Detroit,
-40
0 1000 2000 3000 4000 0 1000 2000 3000 4000 MI, U.S.A., pp. 355-358, May (1993).
(a) (b) [4] Simon Haykin: Adaptive Filter Theory,
Prentice-Hall, ISBN 0-13-322760-X, (1996).
Amplitude (dB)
Amplitude (dB)

0 0 [5] J. S. Lim and A. V. Oppenheim.: All-pole

-20 -20 Modelling of Degraded Speech, IEEE Trans.
Acoust., Speech, Signal Processing, ASSP-26,
-40 -40 June (1978).
0 1000 2000 3000 4000 0 1000 2000 3000 4000
[6] Y. Ephraim and H. L. Van Trees, A spectrally-
(c) (d)
based signal subspace approach for speech
Amplitude (dB)

0 enhancement, in IEEE ICASSP, pp. 804-807

(1995).
-20 [7] Y. Hu and P. Loizou: A subspace approach
-40 for enhancing speech corrupted by colored noise,
0 1000 2000 3000 4000 in Proc. International Conference on
(e)Freq. (Hz) Acoustics, Speech and Signal Processing, vol. I,
Orlando, FL, U.S.A., pp. 573-576, May (2002).
Figure 12: The spectrum of the Handel sig. in Fig.(11) [8] A. Rezayee and S. Gazor: An adaptive KLT
(a) original sig. (b) noisy sig. (c) spectral subtraction. (d) approach for speech enhancement, IEEE Trans.
Wiener filtering. (e) adaptive Wiener filtering. Speech Audio Processing, vol. 9, pp. 87-95
Feb. (2001).
[9] U. Mittal and N. Phamdo: Signal/noise KLT
based approach for enhancing speech degraded by
colored noise, IEEE Trans. Speech Audio
Processing, vol. 8, NO. 2, pp. 159-167,(2000).
[10] John R. Deller, John G. Proakis, and John H.
L. Hansen. Discrete- Time Processing of Speech

UbiCC Journal - Volume 3 7

 Signals. Prentice-Hall, ISBN 0-02-328301-7
(1997).
[11] S. F. Boll: Suppression of Acoustic Noise in
Speech Using Spectral Sub- traction. IEEE Trans.
Acoustics, Speech, and Signal Processing. vol.
ASSP-29. no. 2, pp. 113-120, April (1979).
[12] M. R. Weiss, E. Aschkenasy, and T. W.
Parsons: Processing Speech Signal to Attenuate
Interference, in Proc. IEEE Symp. Speech
Recognition, pp. 292-293, April (1974).
[13] J. S. Lim and A. V. Oppenheim:
Enhancement and band width compression of
Noisy speech, Proc. of the IEEE, vol. 67, No..12,
pp. 1586-1604, Dec. (1979).

UbiCC Journal - Volume 3 8

 FREQUENCY SELECTIVITY PARAMETERS ON
MULTI-CARRIER WIDEBAND WIRELESS SIGNALS

Víctor M Hinostroza, José Mireles and Humberto Ochoa

Institute of Engineering and Technology, University of Ciudad Juárez
Valle del tigris # 3247,Ciudad Juárez Chihuahua México C. P. 32306
vhinostr@uacj.mx, jmireles@uacj.mx, hochoa@uacj.mx

ABSTRACT.
This work is a study of the effects of frequency selectivity on multi-carrier wideband signals in three different
environments; indoors, outdoor to indoor and outdoors. The investigation was made using measurements carried
out with a sounder with a 300 MHz bandwidth. The main part of this work is related to evaluate the contribution
of several parameters; frequency selective fading, coherence bandwidth and delay spread on the frequency
selectivity of the channel. A description of the sounder parameters and the sounded environments are given. The
300 MHz bandwidth is divided in segments of 60 kHz to perform the evaluation of frequency selective fading.
Sub channels of 20 MHz for OFDM systems and 5 MHz for WCDMA were evaluated. Figures are provided for
a number of bands, parameters and locations in the three environments. It is also shown the variation of the
signal level due to frequency selective fading. The practical assumptions about the coherence bandwidth and
delay spread are reviewed and a comparison is made with actual measurements. Statistical analysis was
performed over some of the results.

Keywords. Coherence bandwidth, frequency correlation, frequency selective fading and multi-carrier
modulation
.
I. INTRODUCTION.
To simulate and evaluate the performance of a
wireless mobile system a good channel model is
needed. Mobile communication systems are using
larger bandwidths and higher frequencies and these
characteristics impose new challenges on channel
estimation. The channel models that have been
developed for the mobile systems in use may not be
applicable anymore. To validate that the old
models can be used for future systems or to design
new models, it is necessary to answer the question
about how the same parameters performs at higher
bandwidths? Also, we have to be able to measure
and validate some parameters and compare them to
well known practical assumptions. Measurements
for analysis of the fading statistics at common
frequencies have been performed before, but they
have been performed at small bandwidths, it is
necessary to update the models with higher
bandwidths.
As the data rate (the bandwidth) increases the
communication limitations come from the Inter
Symbol Interference (ISI) due to the dispersive
characteristics of the wireless communications
channel. The dispersive channel characteristics
arise from the different propagation paths, i.e.
multipath, between the receiver and the transmitter.
This dispersion could be measured, if we could
measure the channel impulse response (CIR). As a
general rule the effects of ISI on the transmission
errors is negligible if the delay spread is
significantly shorter than the duration of the
transmitted symbol. Due to the expected increase in
demand of higher data rates, wideband multi-
carrier systems such as; OFDM and WCDMA are
expected to be technologies of choice [1], [12] and
[14]. This is because these two technologies can
provide both; high data rates and an acceptable
level of quality of service. However, these systems
need first to address better the problem regarding
channel prediction or estimation, because this
condition is the main boundary for higher data
rates. The study of correlation of the mobile radio
channel in frequency and time domains has helped
to understand the problem of channel estimation.
One of this work objectives is to evaluate
frequency selective fading (FSF) in several
environments. This work begins with the results of

UbiCC Journal - Volume 3 9

measurements made with a sounder that uses the E{H (t1 ; f1 ) * H (t2 ; f 2 )} =
chirp technique for sounding. ∞ ∞

∫ ∫ E{h(t ;τ ) * h(t ;τ )}e

− 2πf1τ 1 + j 2πf 2τ 2
1 1 2 2 e dτ 1dτ 2
− ∞− ∞
Multipath fading channels are usually classified
into flat fading and frequency selective fading (2)
according to their coherence bandwidth relative to
the one of the transmitted signal. Coherence By considering the channel to have uncorrelated
bandwidth is defined as the range of frequencies scattering (US) and to be wide sense stationary
over which two frequency components remain in a (WSS), the subscript for τ is eliminated and f1 and
strong amplitude correlation. Physically, it defines f2 can be replaced by f + ∆f and t1 and t2 replaced
the range of frequencies over which the channel by t + ∆t, then:
∞
can be considered “flat”. The analytic issue of
coherence bandwidth was first studied by Jakes [1] R H (∆t ; ∆f ) = ∫R h (∆t ;τ )e − j 2π∆fτ dτ
where by assuming homogeneous scattering, his −∞
work revealed that the coherence bandwidth of a (3)
wireless channel is inversely proportional to its In (3) RH and Rh represents the correlation of
root-mean-square (rms) delay spread. The same random variations in the channels transfer function
issue was subsequently studied by various authors and its impulse response respectively. If there are
[4], [8], [9], [10]. Since many practical channel US, then ∆t is 0 then:
environments can significantly deviate from the
homogeneous assumption, various measurements
were conducted to determine multipath delay
Rh (0;τ ) = E h (0;τ ) { 2
}= E {h(τ ) } 2

profiles and coherence bandwidths [19], [20], [21], (4)

[22], aiming to obtain a more general formula for
coherence bandwidth. In this work the variations of substituting into (3) gives:
this formula are reviewed and compared with

actual results and a comparison is provided.
The rest of this document is structured as follow; in
part II the theoretical foundations of the channel
impulse response frequency selective fading and
coherence bandwidth are reviewed. Also in this
part, the characteristics of the three environments
sounded are described. In part III, the frequency
selective fading evaluation and analysis are
presented. Plots of the dependency of fading deep
and frequency separation of two specific points in
the response are studied. At part IV, data about the
relationship between delay spread and coherence
bandwidth are provided. At the end in part V,
conclusions and future work are mentioned.
II. MATHEMATICAL BACKGROUND
2.1 The wideband channel model.
The radio propagation channel is normally
represented in terms of a time-varying linear filter,
with complex low-pass impulse response, h(t, τ). Its
time-varying low-pass transfer function is [4] [6]
[8] [10]:
∞ The coherence bandwidth of a fading channel is
∫ h(t;τ )e τ dτ
− j 2πf
H (t , f ) =
−∞
(1)
Where τ represents delay, using (1) the frequency
correlation function for the channel can be written
as:
∫ E {h(τ ) }e
∞
2 − j 2π∆fτ
R H ( ∆f ) = dτ
−∞
(5)
where
{
E h (τ )
2
}
is the average Power Delay
Profile PDP of the channel. So, under the above
conditions, RH is the Fourier transform of the
average PDP.
2.2 Coherence bandwidth.
The multipath effect of the channel, the arrival of
different signals in different time delays causes the
statistical properties of two signals of different
frequencies to become independent if the frequency
separation is large enough. The maximum
frequency separation for which the signals are still
strongly correlated is called coherence bandwidth
(Bc). Besides to contribute to the understanding of
the channel, the coherence bandwidth is useful in
evaluating the performance and limitations of
different modulations and diversity models.
probed by sending two sinusoids, separated in
frequency by ∆f = f1- f2 Hz, through the channel.
The coherence bandwidth is defined as ∆f, over
which the cross correlation coefficient between r1
and r2 is greater than a preset threshold, say, η0=
0.9. Namely:

UbiCC Journal - Volume 3 10

 Cov( r1, r 2) ⎛2 λ ⎞ π
C r1,r 2 = = η0 (1 + λ ) E ⎜⎜ ⎟−
var(r1) var(r 2)
⎝ 1 + λ ⎟⎠ 2
(6) ρ ( s, τ ) =
π
2−
Then, using (2) 2
J (ω τ )
2
∞ ∞ = λ2 = 0 2m 2
R( s, τ ) = r1r 2 = ∫ 1+ s σ
0
∫ r1r 2 p(r1, r 2)dr dr
0
1 2
(12)
(7)
It is possible to see in this expression that the
Where p(r1,r2) is
correlation decreases with frequency separation.
This formula has been substituted by several
2π 2π
practical expressions some of them are the
p( r1, r 2) = ∫ ∫ p(r1, r 2, θ , θ
0 0
1 2 )dθ1dθ 2 following [4], [8], [9], [10].

(8) 1 1
BC =0.9 = (13) BC =0.5 = (14)
50σ rms 5σ rms
r1r 2 ⎡ r + r ⎤ ⎛ r1r 2 λ ⎞ 2 2
= exp⎢− I ⎜
2 ⎥ 0⎜
1
⎟
2 ⎟
2
1 1
µ (1 − λ )
2 2
⎣ 2 µ (1 − λ ) ⎦ ⎝ µ 1 − λ ⎠ BC =0.9 = (15) BC = (16)
8σ mean 2πσ rms

Where I0(x) is the modified Bessel function of zero In general

order. Then, substituting (8) in (7) and integrating k
BC = (17)
π 1 1 σ rms
R( s,τ ) = b0 F ( − ,− ;1; λ2 )
2 2 2
(9) It will be shown, comparing with practical
measurements that none of these expressions are
this may also be expressed as accurate and it is difficult to obtain a
comprehensive expression for all environments.
π⎛ λ2 ⎞
R( s, τ ) = b0 ⎜⎜1 + ⎟⎟ (10) 2.3 Sounder systems characteristics and
2 ⎝ 4⎠ environment description.

The sounder system used to make the

R ( s, τ ) − r1 r 2 measurements of this work was developed at
ρ ( s, τ ) =
[r 1
2
− r1
2
][ r
2
2
− r2
2
] UMIST in Manchester UK and is described in [2]
and [3]. This sounder uses the FMCW or chirp
technique. The generated chirp consists of a
linearly frequency modulated signal with a
⎛2 λ ⎞ bandwidth of 300 MHz and a carrier frequency of
R( s,τ ) = b0 (1 + λ ) E ⎜⎜ ⎟
⎟ (11) 2.35 GHz. The chirp repetition frequency is 100
⎝1+ λ ⎠ Hertz, which allows having 50-Hertz Doppler
range measurements. The receiver has the same
architecture than the transmitter. But in the
Where E(x) is the complete elliptic integral of the receiver, the generated chirp is not transmitted but
second kind. The expansion of the hyper geometric mixed with the incoming signal from the antenna,
function gives a good approximation to (9). After which are the multi-path components of the
several reductions and considerations, the transmitted chirp. This mixing allows having the
correlation coefficient becomes multi-path components at low frequencies, these
low frequencies can be sampled, digitized and
stored in a computer to perform the required
analysis.

The three environments where the measurements

took place were the following: 1) Indoors, in

UbiCC Journal - Volume 3 11

different floors around a building in an eight stories
building. Each floor form a rectangle with four
long corridors; two 66 m long and two 86 m long,
the width of the corridors is 3 m, the total covered
area was 912 square meters and the ceiling height
is 5 meters. The transmitter was static and the
receiver was moved around the corridors and in
different floors, measurements were taken at
specific distances in each corridor. 2) From a
building to different building. These two buildings
are eight stories high, they have about the same
high and they are separated by about 200 meters.
The transmitter was located in the top of one of the
buildings and the receiver was moved around
specific locations inside the second building in all
eight different floors. 3) An urban environment
around the city center, a commercial area with
several high rising buildings. Each location in each
environment was sampled during one second and
100 impulse responses were stored for this specific
location. When the measurements were taken,
every location was sampled with 100 impulse
responses, an impulse response (IR) was taken on
that specific location every 10 milliseconds.
III. FREQUENCY SELECTIVITY
PARAMETERS
To carry out the calculation of frequency selective
fading, each processed IR was split in samples of
60 kHz, which was the minimum sampled
frequency, each 300 MHz bandwidth was sampled
5000 times every 10 milliseconds, this means that a
sample was taken every 2 microseconds or every
60 kHz. Each IR was averaged over the complete
second, meaning that every frequency was
averaged 100 times for each IR. The level of the
frequency response of each 60 kHz segment was
calculated and recorded
In each environment the channel transfer function
for about 50 different locations were calculated and
recorded. Each transfer function has 5000 sampled
frequencies, i.e. 5000 segments for each location.
Every sample represents the signal level on that
segment, relative to the maximum over the
complete 300 MHz bandwidth. The outdoors
environment has a bandwidth of 120 MHz, the
indoors and indoor to outdoor environments has
300 MHz bandwidth.
The fading characteristics in each of the
environments were calculated. For each of the
locations the fading characteristics that were
calculated were; average delay spread, RMS delay
spread, coherence bandwidth, channel transfer
function and frequency correlation. Using the
channel transfer function for each IR, specific
fading characteristics for sub channels of 5 and 20
UbiCC Journal - Volume 3
MHz were calculated. Figure 1 shows a typical
channel transfer function. In figure 2 are shown the
fading characteristics for a 20 MHz sub channel, in
this figure there are 15 different lines, each one
corresponds to a 20 MHz sub channel in a 300
MHz bandwidth. To get this figure, the following
has been done; first, the fading information of the
complete 300 MHz bandwidth was divided in 15
segments of 20 MHz each. Then, each point on that
segment, that represents a 60 kHz sample, was
measured and the result was compared to the next
sample, then the next sample was compared and so
on up to the complete 20 MHz bandwidth was
compared. After that, a second 20 MHz sub
channel was applied the same procedure and so on
up to the end of the 300 MHz bandwidth. To form
figure 3, the same procedure was followed, but in
this case the sub channel bandwidth was of 5 MHz,
then this figure has 60 different lines which come
from the 300 MHz total bandwidth.
Figure 2 shows that the maximum fading deep
within the 20 MHz sub channel is lower than 14 dB
in all the bandwidth. On the other hand, in the 5
MHz bandwidth the maximum fading deep was of
18 dB. It is possible to see in figure 2, that most of
the lines follows a pattern, which means that the
fading in all sub channels is about the same. In
figure 2, most of the lines stay below 5 dB and only
a few lines go higher than 6 dB; this could mean
that deep fading in this bandwidth is rare.
Figure 1. Typical channel transfer function.
Figure 3 shows the calculations of fading
characteristics for the second environment with 15
different 20 MHz sub channels, this figure shows
that there are more dispersion of the lines, which
means there are more deep fading in the responses
of the IR. Since this environment is the
measurement of the propagation for the penetration
of the signal in different floors in a building, higher
delay spread, (time dispersion) than the former
environment was expected and therefore more
fading.
12
Another way to look at the statistics of the fading is
to calculate the CDF of this parameter. To make
the calculations of these CDF’s figures, the mean
of all locations in the involved environment were
used. Figure 4 shows the CDF of the building-to-
building environment for both, the 20 and 5 MHz
bandwidths, this figure shows that the fading deep
for a 20 MHz sub channel is below 7 dB for 90%
of the time. On the other hand, for the 5 MHz sub
channel, the fades are below 5 dB for 90% of the
time.
correlation coefficient, the coherence bandwidth
(Bc) is lower than 10 MHz most of the locations.
This is corroborated in figure 6, this figure shows
the average Bc for all locations in the indoors
environment. Figure 7, shows the RMS delay
spread for all locations for the same environment.
Quick calculations comparing figure 11 results and
expression (13) show that, few calculated values of
the versions of expression (13) match with the
measured values of figure 7.

Figure 2 Indoor to outdoor fading Characteristics

for a 20 MHz sub channel Figure 4. Fading CDF for indoor to outdoor
for a 5 MHz sub channel

Figure 3. Outdoor to indoor fading Characteristics

For a 5 MHz sub channel
Figure 5. Coherence bandwidth for indoors
IV. COHERENCE BANDWIDTH
EVALUATION.

Figure 5 shows the frequency correlation of all

locations in the indoors environment. To make this
figure the following was done; first the PDP of all
locations was calculated. Then a Fourier transform
was performed on the PDP, which gave us the
frequency correlation for all locations. Then the
frequency correlation for each location was plotted
in figure 5. On this figure, the thick and dashed line
is the line for the maximum coherence bandwidth,
when the transmitter and receiver are connected
directly. In figure 5, one can see that at 0.9

UbiCC Journal - Volume 3 13

Figure 6. Average of coherence bandwidth for
indoors

Figure 8 shows the frequency correlation for the

outdoor to indoor environment, this figure shows
that in this environment the Bc at frequency
correlation of 0.9 is higher than the indoor
environment, although the delay spread is not
different is both environments. Figure 9, shows the
average Bc for the outdoors to indoors
environment, one can see in this figure, that the
coherence bandwidth is higher than the indoor
environment, which was an expected result, but the
difference is higher than expected. In indoors the
coherence bandwidth is not bigger than 20 MHz in Figure 8. Coherence bandwidth for outdoor to
average. In the other hand, in the outdoor to indoor, indoor
the average is about 100 MHz, here is relation of 5
to 1. The difference in RMS delay spread is 100 nS
versus 200 nS, there is a relation of 2 to 1.

Figure 9. Average of coherence bandwidth for

indoor to outdoor

Figure 7. RMS delay spread for indoors
Figure 11 shows the frequency correlation for the
outdoors environment. Figure 12, shows the Bc at
frequency correlation of 0.9. In this case the Bc can
not be compared to the Bc for the other two
environments, since in this environment a lower
bandwidth is evaluated, 120 MHz instead of 300
MHz. Despite this difference and observing
figures 11 and 12, Bc is not significantly lower
even when we have higher distances and higher
delay spread. In outdoors the Bc is not bigger than 2
MHz in average. In the other hand, the RMS delay
spread is 1.5 µS in average.
Table 1, shows the comparisons of Bc for the
three environments with the different versions of
expressions 13 -16 and measured results. This
table shows that the values of the expressions are
always lower than the measured results, which
induce to conclude that the expressions were
underestimated, at least in these environments.
Moreover, it is possible to conclude that these
expressions were deduced with not enough
measured results. Also, table 1 show that the
relationship between delay spread and coherence
bandwidth, not necessarily is a single constant.

UbiCC Journal - Volume 3 14

 Figure 11. Coherence bandwidth for outdoors

Figure 10. RMS delay spread for outdoors to

indoors
Figure 12. Average of coherence bandwidth for
outdoors
V. CONCLUSIONS.

In this work the results of analysis of frequency

Table 1. Coherence bandwidth calculations
selective fading on two indoor and one outdoor
environment have been presented. The three Value from Indoors Outdoors Outdoors
environments analyzed demonstrate that the fading to indoors
is within specific limits, these results could help to (13) 400 kHz 200 kHz 30 kHz
the designers of adaptive receivers to estimate the (14) 4 MHz 2 MHz 300 kHz
channel more accurately. The division of the (15) 3.3 MHz 2.5 MHz 250 kHz
channel impulse bandwidth in segments of 20 and (16) 3.2 MHz 1.6 MHz 212 kHz
5 MHz bandwidths, allow the calculation of fading Measured0.9 5.3 MHz 12 MHz 300 kHz
in the bandwidth of interest for OFDM and Measured0.5 19 MHz 72 MHz 5.6 MHz
WCDMA transmission. Plots of the frequency
selective fading will help for this assessment. The
analysis of coherence bandwidth show that the
expressions accepted in the literature for its
calculation are not accurate and the accepted direct
relationship between delay spread and coherence
bandwidth is not simple. Also, additional work is
require on try to determine how much the
combined effect of Doppler spread, time variability
and frequency offsets affects the transmission on
multi-carrier signals as the ones on OFDM y
CDMA

Figure 13. RMS delay spread for outdoors

References.

1. Jakes W. C., Microwave mobile

communications, (Wiley, 1974).
2. Aurelian B, Gessler F, Queseth O, Stridh
R, Unbehaun M, Wu J, Zander J, Flament

UbiCC Journal - Volume 3 15

 M. “4th-Generation Wireless 16. Namgoong N, and Lehnert J., “
Infrastructures: Scenarios and Research Performance of DS/SSMA Systems in
Challenges”, IEEE Personal Frequency Selective Fading”, IEEE
Communications Magazine, 8(6), 25-31, Transaction on Wireless communication,
Dec 2001. April 2002, Vol. 1, No. 2, pp. 236-244.
3. Salous S, Hinostroza V,” Bi-dynamic 17. TA0 X., et. al., “ Channel Modeling of
indoor measurements with high resolution Layered
sounder”, 5th. International Symposium Space-Time Code Under Frequency
on wireless multimedia Communications, Selective fading Channel”, Proceedings
Honolulu Hawaii USA, October 2002. of ICCT2003, May 2003, Berlin Germany.
4. Golkap H., “ Characterization of UMTS 18. Shayevitz O. and Feder M.,” Universal
FDD channels ”, PhD Thesis, Department Decoding for Frequency Selective
of Electrical Engineering and Electronics, Fading”, IEEE Transactions on
UMIST, UK 2002 Information Theory, August 2005, Vol.
5. Lee W. C. Y., Mobile Communication 51 , N0. 8, pp. 2770-2790.
Engineering, (McGraw-Hill, 1998). 19. Sánchez M and García M, RMS Delay and
6. Bello P.A., “Characterization of randomly Coherence Bandwidth Measurements in
time-variant linear channels”, IEEE Indoor Radio Channels in the UHF Band,
Transactions on Communications Systems, IEEE Transactions on Vehicular
December 1963, pp. 360-393. Technology, vol. 50, no. 2, march 2001
7. Hehn T., Schober R.m and Gerstacker W., 20. Jia-Chin Lin, Frequency Offset
“Optimized Delay Diversity for Frequency Acquisition Based on Subcarrier
Selective Fading Channels”, IEEE Differential Detection for OFDM
Transaction on Wireless communications, Communications on Doubly-Selective
September 2005, Vol. 4, No. 5, pp. 2289- Fading Channels,
2298. 21. Yoo D. and Stark W. E., Characterization
of WSSUS Channels: Normalized Mean
8. Hashemi H., “The indoor radio Square Covariance, IEEE Transactions
propagation channel”, IEEE Proceedings, on Wireless Communications, vol. 4, no.
Vol. 81, No. 81, July 1993, pp. 943-967. 4, july 2005.
9. Lee W. C. Y., Mobile Communication
Engineering, (McGraw-Hill, 1999)
10. Rappaport T. S., Wireless
communications, (Prentice-Hall, 2002, 2nd
ed.)
11. Parsons J. D., The mobile radio
propagation channel, (Wiley, 2000).
12. Morelli M., Sanguinetti L. and Mengali
U., “Channel Estimation for Adaptive
Frequency Domain Equalization ”, IEEE
Transaction on Wireless communication,
September 2005, Vol. 4, No. 5, pp. 2508-
2518.
13. Salous S., and Hinostroza V., “ Bi-
dynamic UHF channel sounder for Indoor
environments “, IEE ICAP 2001, pp. 583-
587
14. Biglieri E., Proakis J. and Shamai S.,
“Fading Channels: Information Theoretic
and Communications Aspects”, IEEE
Transactions on Information Theory,
October 1998, Vol. 44 , No. 6, pp. 2619-
2692.
15. Al-Dhahir N., “Single Carrier Frequency
Domain Equalization for Space-Time
Blok-Coded Transmission over Frequency
Selective Fading Channels”, IEEE
Communications Letters, July 2001, Vol.
5, No. 7, pp. 304-306.

UbiCC Journal - Volume 3 16

 Continuous Reverse Nearest Neighbor Search

Lien-Fa Lin*, Chao-Chun Chen

Department of Computer Science and Information
Engineering National Cheng-Kung University, Tainan, Taiwan, R.O.C.
Department of Information Communication Southern Taiwan
University of Technology, Tainan, Taiwan, R.O.C.

lienfa@cc.kyu.edu.tw,chencc@mail.stut.edu.tw

ABSTRACT
The query service for the location of an object is called Location Based Services
(LBSs), and Reverse Nearest Neighbor (RNN) queries are one of them. RNN queries
have diversified applications, such as decision support system, market decision,
query of database document, and biological information. Studies of RNN in the past,
however, focused on inquirers in immobile status without consideration of
continuous demand for RNN queries in moving conditions. In the environment of
wireless network, users often remain in moving conditions, and sending a query
command while moving is a natural behavior. Availability of such service therefore
becomes very important; we refer to this type of issue as Continuous Reverse
Nearest Neighbor (CRNN) queries. Because an inquirer’s location changes
according to time, RNN queries will return different results according to different
locations. For a CRNN query, executing RNN search for every point of time during a
continuous query period will require a tremendously large price to pay. In this work,
an efficient algorithm is designed to provide precise results of a CRNN query in just
one execution. In addition, a large amount of experiments were conducted to verify
the above-mentioned method, of which results of the experiments showed significant
enhancement in efficiency.

Keywords: Location Based Services, Location-Dependent Query, Continuous

Query, Reverse Nearest Neighbor Query, Continuous Reverse Nearest Neighbor
Query

1 INTRODUCTION Location-Dependent Query (LDQ), of which

As wireless network communications and mobile
device technology develop vigorously and
positioning technology matures gradually, LBS is
becoming a key development in the industrial as well
as academic circles [2, 5, 13, 21, 26, 27]. According
to the report of “IT Roadmap to a Geospatial Future”
[6], LBSs will embrace pervasive computing and
transform mass advertising media, marketing, and
different societal facets in the upcoming decade.
Despite the fact that LBSs have been existing in the
traditional calculation environment (such as Yahoo!
Local), its greatest development potential lies in the
domain of mobile computing that provides freedom
of mobility and access to information anywhere
possible.
LBSs shall become an indispensable application
in mobile network as its required technology has
matured and 3G wireless communication
infrastructure is expected to be deployed everywhere.
The query that answers to LBSs is referred to as
applications include Range Query, Nearest Neighbor
(NN) query, K-Nearest Neighbor (KNN) query, and
Reverse Nearest Neighbor (RNN) query.
There are plenty of studies about NN [14, 22, 26],
KNN [4, 9, 14, 23, 25], CNN [17, 3, 12, 20], and
CKNN [17, 20] queries, and issues pertaining to
Reverse Nearest Neighbor (RNN) Query [10, 11, 16,
18, 19, 22, 24] have been receiving attention in recent
years. RNN query means finding a collection of
nearest neighbor objects for S, a given collection of
objects, with q, a given query object. Practical
examples of RNN query are provided in [10]. If a
bank is planning to open a new branch, and its clients
prefer a branch on a nearest possible location, then
such new branch should be established on a location
where the distance to the majority of its clients is
shorter than that of other banks. Taxi cabs selecting
passengers is another good example. If a taxi cab uses
wireless devices to find out the location of its
customer, then RNN queries will be far more
advantageous than NN queries from the aspect of

UbiCC Journal - Volume 3 17

competition. Figure 1 illustrates that Customer c is
the nearest neighbor for Taxi a, but that does not
necessarily mean Taxi a can capture Customer c
because Taxi b is even closer to Customer c. On the
contrary, the best option for Taxi a should be
Customer d because Taxi a is the nearest neighbor for
Customer d. That is, d is the RNN for a, and a may
reach d faster than any other taxi. This is an example
of CRNN query for that the query object, the taxi,
changes location according to time. Mobile users will
be mobile in a wireless environment, and that is why
the continuous query is an important issue in the
wireless environment.
As far as the knowledge available to the
researchers is concerned, there is not yet any
researcher working on this issue. Because an inquirer
changes location constantly according to time,
changes of location will cause RNN queries to return
different results. For a CRNN query, executing RNN
search for every point of time during a continuous
query period will require a tremendously large price
to pay. The larger the number of query objects and
the shorter the time segment are, the longer the
calculation time will be.
In addition, due to the continuance nature of
time, defining the appropriate time segment for RNN
search will be a concern; if the interval between RNN
searches is too short, then more CRNN queries need
to be executed to complete the query, and vice versa.
If a RNN search is repeated over a longer period of
time to reduce the number of execution, the RNN
query result for the whole time segment will lose
accuracy due to insufficient frequency of sampling.
In this paper, a more efficient algorithm is
designed to replace processing of each and every
point of time for RNN search; just one execution of
CRNN query is all it takes to properly define the
segment for the query time that a user is interested in,
and find out the segments that share the same answer
and the RNN for each of the intervals.
Other than that, an index is also used to filter out
unnecessary objects to reduce search space and
improve CRNN search efficiency. The experiment
result suggests that using index provides efficiency
20 times better than not using index when the number
of objects is 1000.
This Study provides major contribution in three
ways:
z This Study pioneers into continuous query
processing methods opposite to static
query regarding RNN issues.
z A CRNN search algorithm is proposed;
just one execution will return all CRNN
results.
z The proposed method allows the index
which was only applicable to finding RNN
for a single query point to support CRNN
query to improve CRNN search efficiency.
The structure of the other sections in this work:
UbiCC Journal - Volume 3
Related works about RNN search are introduced in
Section 2. Concerned issues are defined and
assumptions made are described in Section 3. The
proposed CRNN search algorithm is introduced in
Section 4 The experiment environment and
evaluation parameters for experimental efficacy are
described in Section 5. In the end, a conclusion and
future study directions are provided in Section 6.
Figure 1: Example of RNN query.
2 RELATED WORK
RNN search concerns about finding q, a query
that is the NN for some objects. Related works of
study about RNN search are introduced and
summarized in this section:
z Index methods that support RNN search
The number of objects can be infinite; if one must
first find out the distance from query q to each object
for identifying the RNN for query q, then the
efficiency may be unacceptably low due to
overwhelmingly large computation cost. To
accelerate processing speed, most of studies adopt the
index methods. Major index methods are introduced
in this section.
z RNN search of different types
RNN searches in different scenarios are described
and categorized according to static and moving
situations of query q and the objects.
2.1 Index Methods for RNN Query
RNN search concerns about finding q, a query
that is the NN for some objects, and it is necessary to
find out the distance between query q and each object,
or the distance from the coordinate of query q to the
coordinate of an object. For a given q, not every
object is its RNN, and these objects which can not be
RNN may be practically left out of consideration to
reduce the number of objects to be taken into
consideration and accelerate processing speed for
RNN search. Many studies were dedicated to the
designing of an effective indexing structure for
coordinates of an object. The most famous ones are
R-Tree proposed by [8] and Rdnn-Tree proposed by
18
[10]. These two index methods are described below.
2.1.1R-Tree
R-Tree is an index structure developed in early
years for spatial database and was used by [10] to
accelerate RNN search processing. All objects are
grouped and then placed on leaf nodes according to
the closeness of their coordinates. That is, objects at
similar coordinates are put in one group. Next, each
group of objects is contained in a smallest possible
rectangle, which is called Minimum Bounding
Rectangle (MBR). Next, MBRs are grouped in
clusters, which are contained inside a larger MBR
until all objects are contained in the same MBR.
What is stored on an internal node of a R-Tree is an
MBR, in which all nodes underneath are contained,
and the root of the R-Tree contains all objects. The
size and range of an MBR is defined by its lower left
coordinate (Ml，Md) and upper right coordinate (Mr，
Mu). Figure 2 is an example of R-Tree. From a to l,
there are total 12 objects; (a ， b ， c ， d) belong to
MBR b1, and (e，f，g) belong to MBR b2. MBR b1
and b2 belong to MBR B1, and MBR R contains all
objects..
Figure 2: Example of R-tree Indexing
2.1.2 Rdnn-Tree
Rdnn-tree (R-tree containing Distance of Nearest
Neighbors) [22] improves the method of [10]. The
author proposes a single index structure (Rdnn-tree)
to provide solutions for NN queries and RNN queries
at the same time. Rdnn-tree differs from standard R-
tree structure by storing extra information about
nearest neighbor of the points in each node.
Information of (ptid，dnn) is stored on the leaf node
of Rdnn-tree, as shown in Figure 3. ptid means an
object of which the data concentrate on the dimension,
denoted as d, and dnn means the distance from such
object to its NN. Information of (ptr ， Rect ，
MaxDnn) is stored on a non-leaf node, where ptr
points to the address of a child node, Rect contains
the MBR of all child nodes subordinate to this node,
and MaxDnn means the maximum value of dnn of all
objects in the child trees subordinate to this node. The
maximum distance from any object contained in
UbiCC Journal - Volume 3
these child trees to its NN will not exceed MaxDnn.
Figure 3. Data structure of Rdnn-tree
2.2 Categories of Rnn queries
Depending on the static or moving status of
query q and the query objects, related studies can be
summarized into 4 categories.
1. If query q and the query objects are both static,
then this category is called static query vs. static
objects.
2. If query q is moving and the query objects are
static, then this category is called moving query vs.
static objects.
3. If query q is static and the query objects are
moving, then this category is called static query vs.
moving objects.
4. If both query q and the query objects are moving,
then this category is called moving query vs. moving
objects.
2.2.1 Static query vs. static objects
The scenario that both query q and query objects
are static is first discussed because the query and
query objects are immobile and are therefore easier
for processing than other scenarios. The method
proposed in [10] is now introduced. For static
database, the author adopts a special R-tree, called
RNN-tree, for answering RNN queries. For static
database that requires being frequently updated, the
author proposes a combined use of NN-tree and
RNN-tree. NN of every object is stored in the RNN-
tree, and what are stored in the NN-tree are the
objects themselves and their respective collections of
NN. The author uses every object as the center of a
circle, of which the radius is the distance from the
object to its NN, to make a circle, and then examines
every circle that contains query q to find out the
answers of RNN queries. Such method, however, is
very inefficient for dynamic database because the
structures of NN-tree and RNN-tree must be changed
whenever the database is updated. In [22], the method
proposed by [10] is therefore improved. The author
proposes a single index structure, Rdnn-tree, for
answering NN queries and RNN queries at the same
19
time. It differs from normal R-tree; it separately
stores the information of NN of every object (i.e.
Distance of Nearest Neighbor), and NN of every
object must be calculated in advance.
2.2.2 Static query vs. moving objects
Studies mentioned above primarily assume a
monochromatic situation that all objects, including
query q and query objects, are of the same type. In
[18], the researcher addresses this type of issues in a
bichromatic situation that objects are divided into two
different types; one is inquirer, and the other is query
object. NN and range query techniques are used in
this Paper to handle RNN issues.
2.2.3 Moving query
This subsection discusses the situation when an
inquirer is no longer static but changes his or her
location according to time, and the query object can
be either static or moving. That is, two categories of
query: moving query with static objects and moving
query with moving objects, are involved. Because the
inquirer is moving, these two categories of query will
return different results for the identical RNN search
at different points of time. This type of issue is
obviously more complicate than the issues previously
discussed. As far as the knowledge available to the
researcher is concerned, no related study has ever
discussed about the issues of these two categories. In
this Study, solutions for a moving query with static
objects are pursued.
3 Problem Formulation
CRNN query concerns about a period of
continuing time where adjacent points in such period
of time may have the identical RNN. That is, a period
of time may have the same RNN unless the query q
has moved beyond this period of time. Please refer to
Figure 4. When a user executes CRNN query q, the
time segment of the continuous query is [Ps，Pe], and
the query objects are {a，b，c，d}. If time point P1
can be identified, and any given point of time in the
time segment from Ps to P1, or [Ps，P1], has the same
RNN result, then one-time execution of RNN search
is all it needs for the time segment of [Ps ，P1]. If
points of time, P2, P3, and P4, are also identified, and
any given point of time in the time segment of [P1，
P2] has the identical RNN result, while any given
point of time in the time segment of [P2，P3] has the
identical result, and any given point of time in the
time segment of [P3，P4] has the identical result, then
the entire CRNN query needs only one-time
execution of RNN search at each time segment.
For processing CRNN query, it is not necessary
to execute RNN search for every point of time and
return the RNN of every point of time back to query
UbiCC Journal - Volume 3
q; instead, i segments of time, such as segment1,
segment2 …segmenti, that have the same result, are
first identified among the entire CRNN query time
period. RNN result of each segment of time, such as
RNN1, RNN2, …, is calculated separately, and the
result is returned in the format of (q ，
[segment1])={RNN1 result} ， … ， (q ，
[segmenti])={RNNi result} back to the inquirer.
Figure 4. Example of a CRNN query
Based on the description above, CRNN query,
one issue that this Study concerns, may be stated as
below:
Given:
A collection of static objects S={O1，O2，…，On}
A query point q, its current position (q.x，q.y), and
moving velocity (v.x，v.y)
A continuous query time [Ps ， Pe]; where Ps
represents the coordinate of the point of time when
the query begins, and Pe represents the coordinate of
the point of time when the query ends
Find:
The RNNs of q between any two adjacent points of
time of {P1 ，P2 ，…，Pi} within [Ps ，Pe] remain
constant.
Such that:
RNN(q，[Ps ，P1]) = {RNN1}，RNN(q，[P1 ，P2])
={RNN2}，…， RNN(q，[Pi ，Pe]) = {RNNi}，
where [Pi，Pe] ⊆ [Ps，Pe]，{RNN1} ∈ {O1，O2，…，
On}.
Under the assumptions:
1. The moving direction of query q is fixed.
2. All query objects are static.
As described above, two adjacent points of time
may share the same RNN, or a segment of time has
only one RNN unless query q moves to another
segment of time that has a different RNN. The CRNN
search algorithm proposed in this Study uses exactly
this concept. First, the points of time that produce
different RNN results within a query time are
identified. These points of time divide the query time
into several segments that have different RNN results,
and then the RNN results are identified for each of
the segments. The detailed algorithm of CRNN
Search is explained in the next section.
4. CRNN Search Algorithm
The detailed procedure of CRNN Search
algorithm is introduced in this section. CRNN Search
20
algorithm is divided into two steps.
Step 1: Finding segment points of CRNNq
Points of time that produce different RNN
results are identified. Based on these points of time,
CRNN query is divided into several time segments
that require execution of RNN search. The RNN
result for any given point of time within one segment
will remain constant, and different segments have
different RNN results.
Step 2: Calculating RNN result of each segment
Separately calculate the RNN results for each of
the segments that have been divided in the previous
step.
The entire procedure for processing CRNN
Search is illustrated in Figure 5. On top of the
necessary query objects and continuous query (query
path), it is divided into two steps: finding segment
points of CRNNq and calculating RNN result of each
segment; each of the steps is described below:
As illustrated in Figure 6, if the NN of object a
is b, and a circle is made using ab as the radius with
a as the center point, then the distance from query q
to a must be shorter than the distance from a to its
NN, or object b, as long as query q falls within this
circle. Therefore, during the period of time when
query q remains within this circle, RNNs of object a
must include a, unless query q moves out of this
circle. Because the moving direction of query q is
assumed to be fixed, CRNN query will form a query
line (qline) from its beginning to its end. The point to
which this CRNN query begins to leave this circle is
the intersection S of this circle and the query line
formed by CRNN query. Before intersection S, the
result of RNN query must include object a; beyond
intersection S, the result of RNN query will not
include object a; the RNN results will be different.
This intersection is referred to as a segment point.
This explains why the intersection of the circle
with NN as its radius and the query line is the point
of time where RNN query produces different results.
Making a circle by using an object itself as the center
and the distance to its NN as the radius will enable all
of the intersections of the circle and the query line of
CRNN query to cut CRNN query into several time
segments that have different results of RNN query.

Figure 5: Flow chart of CRNN query

processing

4.1 Finding segment points of CRNN

What CRNN query pursues is a period of
continuous time; the moving distance of query
objects is very short among some adjacent points of
time for the query, thus possibly resulting in the same
RNN result. That is, the entire period of continuous
query is divided into several segments, and the RNN
results in each segment are the same. If these points
of time share the same RNN result, then it is not
necessary to execute RNN search for each of the
points of time; one-time calculation is enough.
Therefore, CRNN query does not require executing
RNN search for all points of time. Instead, points of
time that share the same RNN result are grouped into
time segments, and one-time RNN search is executed
for each of the segments. RNN of query q is a
collection of the objects of which the NN is query q.
If the distance, or N, is realized in advance, then
these objects are the RNN for query q when the
distances from query q to the objects are shorter than
the distances from the objects to their respective NN.
Figure 6. Finding segment point of CRNN search
Figure 7 illustrates the time segmentation
process described above. For object a, b, and c, their
respective NNs are identified first: NN(a)=b,
NN(b)=a, and NN(c)=b. Next, use each object as the
center of a circle, and the distance to its respective
NN as the radius to make circles of a, b, and c. Then,
intersections of the circles and qlines, Ps, P1, P2, P3,
P4, and Pe , are sorted according to time, and every
two intersection points define a time segment. The
entire CRNN query is cut into five time segments, [Ps，
P1] , [ P1， P2] , [P2，P3] , [P3，P4] , and [P4，Pe].
Every segment has a unique RNN query result.

UbiCC Journal - Volume 3 21


Figure 7: Segmenting of the CRNN query
4.2 Calculating RNN result of each segment
In the previous section, intersections of qlines
and the circles with the distances between the objects
and their respective NNs as the radiuses are defined.
With these intersections, CRNN query is cut into
several time segments. The next step is to find RNNs
for each of the time segments. Because the distances
from query objects to their respective NNs are used
as the radiuses to make circles which are coded by
the objects’ numbers, if a segment falls within a
certain circle, then the resulting RNN of this time
segment for the CRNN query is the object collection
represented by such circle. This is illustrated in
Figure 8. First, intersections of qlines that represent
the CRNN query and the circles of the objects are
sorted by time; every two intersection points define a
time segment, and there are five segments, [Ps，P1] ,
[ P1 ， P2] , [P2 ， P3] , [P3 ， P4] , and [P4 ， Pe].
Segment [Ps ， P1] is contained only by circle a,
therefore: RNN(q，[Ps ，P1]) ={a}. Next, examine
segment [Ps，P1]; this segment is contained by circle
a and circle b. Therefore: RNN(q， [Ps，P1]) = {a，
b}. If this process is repeated, then the obtained
results will be RNN(q ， [P2 ， P3]) = {a ， b ， c},
RNN(q，[P3，P4]) ={b，c}, and RNN(q，[P4，P5])
={c}.
UbiCC Journal - Volume 3
Figure 8: Calculating RNN result of each segment.
1. CRNN Algorithm with Index
Not every object will be an answer in the
processing of CRNN query. To improve RNN query
efficiency, it is preferred that the objects that can not
be answers are filtered out in advance to greatly
reduce search space for CRNN query, size of data
that requires CRNN query, and consequently,
computation cost. The process that further improves
CRNN query efficiency dramatically is referred to as
pruning process. Figure 9 illustrates the flowchart of
CRNN query processing with a pruning process
added.
Figure 9. Flow chart of CRNN query with index
Step 2 and 3 are identical to Step 1 and 2 in
CRNN search algorithm, which have been described
in the previous sections, and they will not be
reiterated again here. For step 1, the pruning process,
an index structure for Rdnn-tree is designed to
effectively execute the pruning process. The three
steps of CRNN query with index are illustrated in
Figure 9. The pruning process is described below. For
every internal node of Rdnn-tree, the distance from
query q to its node will be computed for every
separation, and the distance is denoted as D(q，Rect).
If D(q，Rect) of a node is larger than MaxDnn of the
node, then all the objects beneath it will not be
considered because the distance from query q to Rect
Node will be equal to or larger than the distances
from query q to all the objects underneath Rect node.
When the distance from query q to Rect node is
longer than MaxDnn, it is impossible that query q is
closer to its NN than any other object underneath
Rect node, and no object underneath can be the RNN
result for query q. On the contrary, if D(q，Rect)
equals the MaxDnn of such node, then the distances
from some objects underneath Rect node to their
respective NNs are shorter than the distance from
query q to Rect. That is, some objects are the RNN
results for query q. The examination continues along
22
the branch all the way to the lead node. All entries
underneath such leaf node are recorded as the
candidate objects for RNN query result. The
collection of these candidate objects is referred to as
RNNCanSet, which means the possible results for
RNN query must exist within this collection, and the
objects outside of RNNCanSet can not possibly be
RNN query results. All that are needed to be
considered when finding segment point of CRNNq of
CRNN search algorithm are the objects inside
RNNCanSet. This will greatly reduce the quantity of
objects needed to be handled and enhance CRNN
search algorithm efficiency.
Figure 3 explains the pruning process. It begins
with root node R. Because D(q，R) ≦MaxDnn of R,
child nodes of B1 and B2 must be examined. Because
the MaxDnn of MBR B1 ≦D(q，B1), all child nodes
underneath B1 can be pruned. Next, D(q ，
B2)≦MaxDnn of B2, so child nodes b3 and b4 of B2
must be examined. D(q，b3) is equal to or smaller
than the MaxDnn of b3, which is also a leaf node;
Algorithm 1: Pruning Algorithm.
and comparison of experiment results.
5. Performance Study
To evaluate the improvement which the method
proposed in this Study has made in CRNN query
efficiency, some experiments are designed, and this
section provides descriptions of experiment
environments, experimental parameters and settings,
UbiCC Journal - Volume 3
therefore, h and i are placed inside RNNCanSet. Next,
b4 is examined. MaxDnn of b4 is equal to or smaller
than D(q ， b4); therefore, b4 can be pruned. The
entire pruning process then ends.
However, the CRNN query to be processed is
not a RNN query of a single query point; therefore,
the pruning process in [22] can not be directly used.
To ensure that no possible RNN result is deleted, the
criteria of pruning is changed from the condition that
D(q，Rect), the distance from query point to Rect,
must be longer than MaxDnn to the condition that
MinD(q，Rect)>MaxDnn, where MinD(qline，Rect)
represents the minimum distance from qline to Rect
node. The reason why the shortest distance is selected
is that if the minimum distance from the entire qline
to Rect node is larger than MaxDnn, then the distance
from any given point of time on the qline to Rect
node must be longer than MaxDnn. Therefore, all the
objects underneath Rect node can not be RNN for
qline, and pruning is out of consideration. Details of
the pruning algorithm are exhibited in Algorithm 1:
5.1 Experiment Settings
The coordinates of the objects disperse in an
experiment environment of [0 ， 1]×[0 ， 1] plane.
Because distribution density of the objects may
influence efficiency, it should be taken into
23
consideration in the experiment. In the experiment,
three different types of distribution are used in the
generation of objects’ coordinates. The three different
types of distribution are Uniform distribution,
Gaussian distribution, and Zipf distribution. In
uniform distribution, the objects are evenly
distributed on the plane, as shown in Figure 10(a). In
Gaussian distribution, most of the objects concentrate
on the center of the plane, as shown in Figure 10(b).
In Zipf distribution, most of the objects will distribute
at the extreme left and extreme bottom of the plane.
In the experiment, skew factor is set at 0.8, as shown
in Figure 10(c). In addition, 30 queries are generated
randomly in a [0.4 ， 0.6]×[0.4 ， 0.6] plane by
1 1
0.8
0.6
Y axis
0.4
0.2
0 0
0 0.2 0.4 0.6 0.8
X axis
Figure 10 Data sets of experiment evaluation
In addition to the object distribution described
above, the influences that the amount of query time
(qline) and the number of objects may impose on
efficiency are also considered. Three data sets of
Uniform, Gaussian, and Zipf are considered in object
distribution. The amount of query time (qline)
changes from query length 1 to query length 10. The
number of objects changes from 1K to 10K.
Parameters and settings used in the experiment are
listed in Table 1.
Table 1: Parameter settings of experiment
Parameter Description
distribution Data distribution
interval Time interval of
Query
object-no Number of Data
Objects
5.2 Compared Algorithms and Performance
Metrics
The most intuitive method for finding RNN is
looking for the NN of every object. If the number of
query objects is N, then time Complexity is O(n2).
Next, determine which objects’ NNs are query points.
If the NNs are the query points, then the objects will
be the RNNs for the query points. The required time
complexity for the RNN algorithm is O(n3).
However, the most intuitive method for finding
UbiCC Journal - Volume 3
referring to [15], and the velocity vector of each
query falls between [-0.01 ， 0.01]. Because the
influence of different types of object distribution on
efficiency is concerned in this experiment, the queries
are generated as close to the center of the plane as
possible. Having executed 30 queries, the average
cost of executing one CRNN search is used in
determining which method is more favorable. As to
the program coding of Rdnn-tree in the CRNN search
algorithm, R*-tree code of GIST[7] is used in
perfecting Rdnn-tree to make it match with the
requirement of this experiment.
1
0.8 0.8
0.6 0.6
Y axis
Y axis
0.4 0.4
0.2 0.2
0 0
0.2 0.4 0.6 0.8 1
1 0 0.2 0.4 0.6 0.8 1
X axis X axis
CRNN query is executing RNN algorithm for every
point of time which is continuous, and it is
impossible to calculate the required count of
execution. Therefore, the CRNN query time must be
segmented before the total execution time required
for CRNN query may be calculated. The more the
time is segmented, the more executions of RNN are
required. If a period of time is segmented into m
segments, then time complexity will be O(m×n3), and
if time is not adequately segmented, then the RNN
result may be erroneous. These make it an inefficient
CRNN search algorithm, and it will not be compared
Settings in this experiment. Efficiency of two methods is
Uniform, compared in this experiment: one uses Rdnn-tree as
Gaussian, Zipf the index, and the other uses no index. To evaluate
1, 2, 5, 8, 10 these two methods, comparison of the time required
for one CRNN search execution can be used, and this
1, 10. 30, 50 , comparison is referred to as total cost in this Study.
100(k)
5.4 Performance Results and Discussion
Based on the changes of metrics (distribution,
interval, and object-no), different types of
experiments have been conducted. Results are
summarized by object-no and query interval in the
next section.
5.4.1 The effect of object-no parameter
First, the fixed query interval is set at 5. The
influence imposed on efficiency by object-no
24
parameter, which is the number of objects, under comparison of influence from object distribution on
different types of object distribution, will be efficiency clearly suggests Zipf distribution offers the
discussed. The experiment result is shown in Figure best efficiency, followed by Uniform distribution,
11. X axle represents the total cost of time required and Gaussian distribution offers the worst. This result
for executing one CRNN search, and Y axle can be explained as such: because Zipf distribution is
represents the number of query objects. Total cost located at the far left and the lowest bottom, most of
increases as the number of query objects increases. the data will be pruned, and the number of objects
In addition, when the number of objects is 1K, that are included in RNNCanSet without being
the efficiency of the CRNN search that uses Rdnn- pruned is very small, offering the lowest total cost.
tree is about 300 seconds, and that of the CRNN On the opposite, data in Gaussian distribution
search using no Rdnn-tree index is about 15 seconds; concentrate in the center of the plane and very few
about 20 times faster. It is obvious that pruning some data can be pruned, allowing many objects to remain,
unnecessary objects by adopting Rdnn-tree as the and causing a large RNNCanSet, thus resulting in the
index to reduce CRNN search space provides much highest total cost.
higher efficiency than not adopting Rdnn-tree. The
7
10
7
10 107

CRNN without index 6 6 CRNN without index

106 10 CRNN without index 10
CRNN with index CRNN with index
CRNN with index
5 5 5
10 10 10

Total time (sec.

Total time (sec.

Total time (sec.

)
)

104 104 104

103
103 103

2 2 2
10 10 10

10 10 10

1 1 1
1K 10K 30K 50K 100K 1K 10K 30K 50K 100K 1K 10K 30K 50K 100K
object-no object-no object-no

Figure 11: Influences on different types of data distribution from changing object-no

5.4.2 The effect of query interval parameter distance of MinD(qline ， Rect) decreases, causing
pruning efficiency to reduce. On the contrary, when
This section focuses on the influence from the the interval increases, the number of time
length of query interval on each method under segmentation by CRNN query increases.
different types of object distribution. Results of the Consequently, the number of RNN searches for every
experiment are shown in Figure 12. Generally segment increases, and total cost of CRNN query
speaking, when the query interval is lengthened, the increases as well.
105 105 10
5

104 104 10
4

CRNN without index CRNN without index

CRNN with index CRNN with index
Total Time (sec.

Total Time (sec.

Total time (sec.
)

)
)

3
10 103 103 CRNN without index
CRNN with index

102 10
2
10
2

10 10 10

1 1 1
1 2 5 8 10 1 2 5 8 10 1 2 5 8 10
Query Interval Query Interval Query Interval

Figure12: Effect of query interval parameter for different data distribution

6. Conclusions and Future Works Study also prove the efficiency of the proposed
method. As wireless communication and mobile
An efficient CRNN search algorithm is proposed device technology become mature, more and more
in this Study. Such algorithm requires only one users access information from wireless information
execution to find out RNN results from all continuous systems through mobile devices. To process requests
RNN searches. The diversified experiments in this from more and more mobile users, data dissemination

UbiCC Journal - Volume 3 25

through broadcast is an effective solution for
scalability. The future goal of this Study is extending
the issues of CRNN search to the wireless
broadcasting environment.
7. References
[1] AmitSingh,H.F. and Tosun,A.S. (2003) High
dimensional reverse nearest neighbor queries.
Proceedings of the 20th International
Conference on Information and Knowledge
Management (CIKM’03), NewOrleans, LA,
USA, pp.91–98.
[2] Barbara,D. (1999) Mobile computing and
databases-a survey. IEEE Transactions on
Knowledge and Data Engineering, 11, 108–117.
[3] Benetis,R.,Jensen, C.S.,Karciauskas,G., and
Saltenis,S. (2002) Nearest neighbor and reverse
nearest neighbor queries for moving objects.
International Database Engineering and
Applications Symposium, Canada, July17-19,
pp.44–53.
[4] Chaudhuri,S. and Gravano,L. (1999) Evaluating
top-k selection queries. Proceedings of the 25th
IEEE International Conference on Very Large
Data Bases, pp.397–410.
[5] Civilis,A., Jensen,C.S., and Pakalnis,S. (2005)
Techniques for efficient road-network-based
tracking of moving objects. IEEE Transactions
on Knowledge and Data Engineering, 17, 698–
712.
[6] Computer Science and Telecommunication
Board.IT Roadmap to a geospatial future, the
national academies press,2003.
[7] http://gist.cs.berkeley.edu/.
[8] Guttman,A. (1984) R-trees:A dynamic index
structure for spatial searching. Proceedings of
the 1984 ACM SIGMOD international
conference on Management of data, pp.47–57.
[9] Hjaltason,G.R. and Samet,H. (1999) Distance
browsing in spatial data bases. ACM
Transactions on Database Systems (TODS), 24,
265–318.
[10] Korn,F. and Muthukrishnan,S. (2000) Influence
sets based on reverse nearest neighbor queries.
Proceedings of the 2000 ACM SIGMOD
International Conference on Management of
Data, Dallas, Texas, USA, May16-18, pp.201–
212.
[11] Korn,F.,Muthukrishnan, S.,and Srivastava.,D.
(2002) Reverse nearest neighbor aggregates
over data streams. Proceedings of the
International Conference on Very Large
DataBases (VLDB’02), Hong Kong, China,
August, pp.91–98.
[12] Korn,F., Sidiropoulos, N.,Faloutsos, C.,Siegel,E.,
and Protopapas,Z. (1996) Fast nearest neighbor
search in medical image database. In
Proceedings of the 22th International
Conference on Very Large Data Bases
UbiCC Journal - Volume 3
(CLDB’96), pp. 215–226.
[13] Lee,D.L., chien Lee, W.,Xu,J., and Zheng, B.
(2002) Data management in location dependent
services. IEEE Pervasive Computing,1, 65–72.
[14] Roussopoulos,N., Kelley,S. ,and Vincent,
F.(1995) Nearest neighbor queries.
Proceedings of ACM Sigmod International
Conference on Management of Data , Illinois,
USA, June, pp.71–79.
[15] Ross,S.(2000) Introduction to Probability and
Statistics for Engineers and Scientists.
[16] Stanoi,I., Agrawal,D., and Abbadi,A.E. (2000)
Reverse nearest neighbor queries for dynamic
databases. ACM SIGMOD Workshop on
Research Issues in Data Mining and
Knowledge Discovery, pp.44–53.
[17] Song,Z. and Roussopoulos,N. (2001) K-nearest
neighbor search for moving query point.
Proceedings of 7th International Symposium on
Advances in Spatial and Temporal Databases,
LNCS2121, RedondoBeach, CA, USA, July12-
15, pp.79–96.
[18] Stanoi,I.,Riedewald, M.,Agrawal,D., and
Abbadi,A.E. (2001) Discovery of influence sets
in frequently updated databases. Proceedings of
the 27th VLDB Conference, Roma, Italy, pp.99–
108.
[18] Tao,Y., Papadias,D., and Lian,X. (2004) Reverse
knn search in arbitrary dimensionality.
Proceedings of 30th Very Large Data Bases,
Toronto, Canada, August29-September3,
pp.279–290.
[20]Tao,Y., Papadias, D., and Shen,Q. (2002)
Continuous nearest neighbor search.
International Conference on Very Large Data
Bases, Hong Kong, China, August 20-23,
pp.279–290.
[21] Xu,J., Zheng,B., Lee,W.-C.,, and Lee,D.L. (2003)
Energy efficient index for energy query
location-dependent data in mobile
environments. In Proceedings of the 19th IEEE
International Conference on Data Engineering
(ICDE’03), Bangalore, India, March, pp.239–
250.
[22] Yang,C. and Lin,K.-I. (2001) An index structure
for efficient reverse nearest neighbor queries.
Proceedings of the 17th International
Conference on Data Engineering, pp.485–492.
[23] Yiu,M.L., Papadias,D., Manoulis,N., and Tao,Y.
(2005) Reverse nearest neighbors in large
graphs. Proceedings of 21st IEEE International
Conference on Data Engineering (ICDE),
Tokyo, Japan, April5-8, pp.186–187.
[24] Yu,C.,Ooi,B.C., Tan,K.-L., and Jagadish,H.V.
(2001) Indexing the distance: An efficient
method to knn processing. Proceedings of the
27th VLDB Conference, Roma, Italy, pp. 421–
430.
[25] Zheng,B.,Lee, W.-C., and Lee,D.L. (2003)
26
 Search k nearest neighbors on air. Proceedings
of the 4th International Conferenceon Mobile
Data Management, Melbourne, Australia,
January, pp.181–195.
[26] Zheng,B., Xu,J., chien Lee, W., and Lee,D.L.
(2004) Energy conserving air indexes for
nearest neighbor search. Proceedings of the 9th
International Conference on Extending
Database Technology (EDBT’04), Heraklion,
UbiCC Journal - Volume 3
Crete, Greece,March, pp.48–66.
[27] Zhang,J., Zhu,M., Papadias,D., Tao,Y., and
Lee,D.L. (2003) Location-based spatial queries.
In Proceedings of the 2003 ACM SIGMOD
international conference on Management of
data, SanDiego, California, USA, June9-12,
pp.443–454.
27
 REDUCTION OF INTERCARRIER INTERFERENCE IN OFDM
SYSTEMS
R.Kumar
* Dept. of Electronics and Comm. Engg., SRM University, Chennai, India-603203
rkumar68@gmail.com
ABSTRACT
Orthogonal Frequency Division Multiplexing
(OFDM) is a promising technique for the broadband wireless
communication system. However, a special problem in OFDM
is its vulnerability to frequency offset errors due to which the
orthogonality is destroyed that result in Intercarrier
Interference (ICI). ICI causes power leakage among
subcarriers thus degrading the system performance. This
paper will investigate the effectiveness of Maximum-
Likelihood Estimation (MLE), Extended Kalman Filtering
(EKF) and Self-Cancellation (SC) technique for mitigation of
ICI in OFDM systems. Numerical simulations of the ICI
mitigation schemes will be performed and their performance
will be evaluated and compared in terms of bit error rate
(BER), bandwidth efficiency and computational complexity.
Keywords: Orthogonal Frequency Division Multiplexing
(OFDM), Intercarrier Interference (ICI), Carrier Frequency
Offset (CFO), Carrier to Interference Ratio (CIR), Maximum
Likelihood (ML), Extended Kalman Filtering (EKF).
1. Introduction
Orthogonal frequency division multiplexing (OFDM),
because of its resistance to multipath fading, has attracted
increasing interest in recent years as a suitable modulation
scheme for commercial high-speed broadband wireless
communication systems. OFDM can provide large data rates
with sufficient robustness to radio channel impairments. It is
very easy to implement with the help of Fast Fourier
Transform and Inverse Fast Fourier Transform for
demodulation and modulation respectively [1].
It is a special case of multi-carrier modulation in
which a large number of orthogonal, overlapping, narrow band
sub-channels or subcarriers, transmitted in parallel, divide the
available transmission bandwidth [2]. The separation of the
subcarriers is theoretically minimal such that there is a very
compact spectral utilization. These subcarriers have different
frequencies and they are orthogonal to each other [3]. Since
the bandwidth is narrower, each sub channel requires a longer
symbol period. Due to the increased symbol duration, the ISI
over each channel is reduced.
However, a major problem in OFDM is its
vulnerability to frequency offset errors between the
transmitted and received signals, which may be caused by
Doppler shift in the channel or by the difference between the
transmitter and receiver local oscillator frequencies [4]. In
such situations, the orthogonality of the carriers is no longer
maintained, which results in Intercarrier Interference (ICI). ICI
results from the other sub-channels in the same data block of
the same user. ICI problem would become more complicated
when the multipath fading is present [5]. If ICI is not properly
compensated it results in power leakage among the
subcarriers, thus degrading the system performance.
UbiCC Journal - Volume 3
Dr. S.Malarvizhi
In [6], ICI self-cancellation of the data-conversion
method was proposed to cancel the ICI caused by frequency
offset in the OFDM system. In [7], ICI self-cancellation of the
data-conjugate method was proposed to minimize the ICI
caused by frequency offset and it could reduce the peak
average to power ratio (PAPR) than the data-conversion
method. In [8], self ICI cancellation method which maps the
data to be transmitted onto adjacent pairs of subcarriers has
been described. But this method is less bandwidth efficient. In
[9], the joint Maximum Likelihood symbol-time and carrier
frequency offset (CFO) estimator in OFDM systems has been
developed. In this paper, only carrier frequency offset (CFO)
is estimated and is cancelled at the receiver. In addition,
statistical approaches have also been explored to estimate and
cancel ICI [10].
Organization: This paper is organized as follows: In
section 2, the standard OFDM system has been described. In
section 3, the ICI mitigation schemes such as Self-
Cancellation (SC), Maximum Likelihood Estimation (MLE)
and Extended Kalman Filtering (EKF) methods have been
described. In section 4, simulations and results for the three
methods has been shown and are compared in terms of
bandwidth efficiency, bit error rate (BER) performance.
Section 5 concludes the paper and inference has been given.
2. System Description
The block diagram of standard OFDM system is given
in figure 1. In an OFDM system, the input data stream is
converted into N parallel data streams each with symbol
period Ts through a serial-to-parallel Port. When the parallel
symbol streams are generated, each stream would be
modulated and carried over at different center frequencies.
The sub-carriers are spaced by 1/NTs in frequency, thus they
are orthogonal over the interval (0, Ts). Then, the N symbols
are mapped to bins of an inverse fast Fourier transform
(IFFT). These IFFT [11] bins correspond to the orthogonal
sub-carriers in the OFDM symbol. Therefore, the OFDM
symbol can be expressed as
1 N −1
x ( n) =
N
∑X e
m =0
m
j 2πnm / N
(1)
where the Xm’s are the base band symbols on each
sub-carrier. The digital-to-analog (D/A) converter then creates
an analog time-domain signal which is transmitted through the
channel.
At the receiver, the signal is converted back to a
discrete N point sequence y(n), corresponding to each sub-
carrier. This discrete signal is demodulated using an N-point
Fast Fourier Transform (FFT) operation at the receiver.
S/P IFFT P/S D/A
Channel
w(n) 28
 Figure 2: Comparison between | S ' ' (l-k)|,
|S ' (l-k)| and |S (l-k)|
Figure 1: OFDM System Model It is seen from figure 2 that |S ' (l-k)| << |S (l-k)| for
most of the l-k values. Hence, the ICI components are much
The demodulated symbol stream is given by: smaller Also, the total number of interference signals is halved
N −1 in as opposed to since only the even subcarriers are involved
Y (m) = ∑ y (n)e − j 2πnm / N + w(m) (2) in the summation.
n=0
N-1 3.1.2 ICI Canceling Demodulation
where w (m) corresponds to the FFT of the samples of w ICI modulation introduces redundancy in the
(n), which is the Additive White Gaussian Noise (AWGN) received signal since each pair of subcarriers transmit only one
introduced in the channel. data symbol. This redundancy can be exploited to improve the
system power performance, while it surely decreases the
3. ICI Mitigation Schemes bandwidth efficiency. To take advantage of this
redundancy, the received signal at the (k + 1) th subcarrier,
3.1 Self-Cancellation (SC) Scheme where k is even, is subtracted from the kth subcarrier. This is
In this scheme, data is mapped onto group of
expressed mathematically as
subcarriers with predefined coefficients. This results in
Y '' (k) = Y '(k) -Y ' (k+1)
cancellation of the component of ICI within that group due to N−2
the linear variation in weighting coefficients, hence the name
self- cancellation. The complex ICI coefficients S (l-k) are
= ∑X (l)[−S(l − k −1) + 2S(l − k) − S(l + k +1)]+ n − n
l =0, 2,4,..
k k +1

given by
(7)
Sin(π (l + ε − k )) (3) Subsequently, the ICI coefficients for this received signal
Sin(l − k ) = exp( jπ (1 − 1 / N )(l + ε − k ))
NSin(π (l + ε − k ) / N ) becomes
S '' (l-k) = – S (l-k-1) + 2S (l-k) – S (l-k+1) (8)
3.1.1 ICI Canceling Modulation When compared to the two previous ICI coefficients
The ICI self-cancellation scheme requires that the |S (l-k)| for the standard OFDM system and |S'(l-k)| for the ICI
transmitted signals be constrained such that X (1) = - X (0), X canceling modulation, |S''(l-k)| has the smallest ICI
(3) = - X (2) …X (N-1) = - X (N-2).The received signal on coefficients, for the majority of l-k values, followed by
subcarriers k and k + 1 to be written as |S' (l-k)| and |S (l-k)|. This is shown in Figure 2 for N = 64 and
N −2 ε = 0.5. The combined modulation and demodulation method
Y ' (k ) = ∑ X (l )[S (l − k ) − S (l + 1 − k )] + n
l = 0 , 2 , 4 , 6 ,..
k (4) is called the ICI self-cancellation scheme. The reduction of the
ICI signal levels in the ICI self-cancellation scheme leads to a
N −2 higher CIR. The theoretical CIR is given by
Y ' (k + 1) = ∑ X (l )[S (l − k − 1) − S (l − k )] + n k +1
− S (−1) + 2 S (0) − S (1)
2
l = 0 , 2 , 4 , 6 ,..
CIR = 2
(9)
(5) N −1
where nk and nk+1 is the noise added to it.
And the ICI coefficient S ' (l-k) is denoted as
∑ − S (l − 1) + 2S (l ) − S (l + 1)
l = 2 , 4 , 6 ,..
S '(l-k) = S (l-k) – S (l+1-k) (6)
As mentioned previously, the redundancy in this
scheme reduces the bandwidth efficiency by half. There is a
tradeoff between bandwidth and power tradeoff in the ICI self-
cancellation scheme.
3.2 Maximum Likelihood Estimation
The second method for frequency offset correction
in OFDM systems was suggested by Moose in [12]. In this
approach, the frequency offset is first statistically estimated
using a maximum likelihood algorithm and then cancelled at
the receiver. This technique involves the replication of an
OFDM symbol before transmission and comparison of the

UbiCC Journal - Volume 3 29

 phases of each of the subcarriers between the successive z(n) is linearly related to d(n). Hence the normalized
symbols. frequency offset ε (n) can be estimated in a recursive
When an OFDM symbol of sequence length N is procedure similar to the discrete Kalman filter. As linear
replicated, the receiver receives, in the absence of noise, the approximation is involved in the derivation, the filter is called
2N point sequence i.e., {r (n)} given by the extended Kalman filter (EKF). The EKF provides a
1 K trajectory of estimation for ε(n). The error in each update
r ( n) =
N
∑ X ( k ) H ( k )e
k =− K
j 2πn ( k +ε ) / N
(10) decreases and the estimate becomes closer to the ideal value
during iterations.
where {X(k)} are the 2K+1 complex modulation
values used to modulate 2K+1 subcarriers, 4.2 ICI Cancellation
The first set of N symbols are demodulated using an There are two stages in the EKF scheme to mitigate
N-point FFT to yield the sequence R1(k), and the second set is the ICI effect: the offset estimation scheme and the offset
demodulated using another N-point FFT to yield the sequence correction scheme.
R2(k). The frequency offset is the phase difference between R1
(k) and R2 (k), that is 4.2.1 Offset Estimation Scheme
To estimate the quantity ε (n) using an EKF in each
R2 (k) = R1 (k) ej2πε (11) OFDM frame, the state equation is built as
ε(n) = ε (n-1) (23)
Adding the AWGN yields i.e., in this case we are estimating an unknown constant ε. This
Y1 (k) = R1 (k) + W1 (k) (12) constant is distorted by a non-stationary process x(n), an
Y2 (k) = R1 (k) ej2πε + W2 (k) observation of which is the preamble symbols preceding the
k = 0, 1 ...N – 1 data symbols in the frame. The observation equation is
The maximum likelihood estimate of the normalized
frequency offset is given by: y(n) = x(n) e j2 π n ε(n) / N + w(n) (24)
⎧ K ⎫

∑ Im Y (k )Y * (k )
⎪ ⎪
⎪ ⎪
⎪
2 1
⎪ where y(n) denotes the received preamble symbols
∧ 1
tan − 1
⎪ ⎪
distorted in the channel, w(n) the AWGN, and x(n) the IFFT
ε=
⎪ ⎪
⎪ k =− K ⎪
⎨ ⎬ (13) of the preambles X(k) that are transmitted, which are known at
2π ⎪ K ⎪

∑ Re Y (k )Y * (k )
⎪ ⎪
⎪
⎪
⎪
⎪
the receiver. Assume there are Np preambles preceding the
⎪ 2 1 ⎪
⎪⎩ k =− K ⎪⎭ data symbols in each frame are used as a training sequence
This maximum likelihood estimate is a conditionally and the variance σ2 of the AWGN w(n) is stationary.
unbiased estimate of the frequency offset and was computed
using the received data. Once the frequency offset is known, 4.2.2 Offset Correction Scheme
the ICI distortion in the data symbols is reduced by The ICI distortion in the data symbols x(n) that
multiplying the received symbols with a complex conjugate of follow the training sequence can then be mitigated by
the frequency shift and applying the FFT, multiplying the received data symbols y(n) with a complex
X (n) = FFT {y (n) e-j2π nε / N} (14) conjugate of the estimated frequency offset and applying FFT,
i.e.
3.3 Extended Kalman Filtering xˆ(n) = FFT{ y(n) e -j 2 π n ε(n) / N} (25)
A state space model of the discrete Kalman filter is As the estimation of the frequency offset by the EKF
defined as scheme is pretty efficient and accurate, it is expected that the
z(n) = a(n) d(n) + v(n) (15) performance will be mainly influenced by the variation of the
In this model, the observation z(n) has a linear AWGN.
relationship with the desired value d(n). By using the discrete
Kalman filter, d(n) can be recursively estimated based on the 4.3 Algorithm
observation of z(n) and the updated estimation in each 1. Initialize the estimate εˆ(0) and corresponding state
recursion is optimum in the minimum mean square sense. error P(0)
The received symbols in OFDM System are 2. Compute the H(n), the derivative of y(n) with respect to
y(n) = x(n) ej 2 π n ε(n) / N + w(n) (16) ε(n) at εˆ(n-1) the estimate obtained in the previous
where y(n) the received symbol and x(n) is the FFT iteration.
of transmitted symbol. It is obvious that the observation y(n) is 3. Compute the time-varying Kalman gain K(n) using the
in a nonlinear relationship with the desired value ε(n), i.e error variance p (n-1), H(n), and σ2
y(n) = f(ε(n)) + w(n) (17) 4. Compute the estimate yˆ(n) using x(n) and εˆ(n-1) i.e.
where f(ε(n)) = x(n) ej 2 π n ε(n) / N (18) based on the observations up to time n-1, compute the
In order to estimate ε(n) efficiently in computation, error between the true observation y(n) and yˆ(n)
we build an approximate linear relationship using the first- 5. Update the estimate εˆ(n) by adding the K(n)-weighted
order Taylor’s expansion: error between the observation y(n) and yˆ(n) to the
y(n)≈f(εˆ(n-1))+f'(εˆ(n-1))[ε(n)-εˆ(n-1)]+w(n) (19) previous estimate εˆ(n-1)
6. Compute the state error P(n) with the Kalman gain K(n),
where εˆ(n-1) is the estimate of ε(n-1). H(n), and the previous error P(n-1).
To Define 7. If n is less than Np, increment n by 1 and go to step 2;
z(n) = y(n) – f(εˆ(n-1) (20)
d(n) = ε(n) - εˆ(n-1) (21) otherwise stop.
and the following relationship It is observed that the actual errors of the estimation εˆ(n) from
z(n) = f'(ε(n-1)) d(n) + w(n) (22) the ideal value ε(n) are computed in each step and are used for
adjustment of estimation in the next step.

UbiCC Journal - Volume 3 30

 4. SIMULATIONS AND RESULTS
In order to compare the ICI cancellation schemes,
BER curves were used to evaluate the performance of each
scheme. For the simulations in this project, MATLAB was
employed. The simulations were performed using an AWGN
channel.
Table 1: Simulation Parameters
PARAMETERS VALUES
Number of carriers (N) 1705
Modulation (M) BPSK
Frequency offset ε [0.25,0.5,0.75]
No. of OFDM symbols 100
Bits per OFDM symbol N*log2(M) Figure 5: BER performance with ICI
Cancellation for ε=0.75
Eb-No 1:20
the effect of this residual ICI increases for larger offset values.
IFFT size 2048 However, ML method has an increased BER performance and
proves to be efficient than SC method.

Figure 3: BER performance with ICI
Cancellation for ε=0.25
Figure 3 shows that for small frequency offset
values, ML and SC methods have a similar performance.
However, ML method has a lower bit error rate for increasing
values of Eb/No.
5. CONCLUSION
It is observed from the figures that Extended
Kalman filter method indicates that for very small frequency
offset, it does not perform very well, as it hardly improves
BER. However, for high frequency offset the Kalman filter
does perform extremely well. Important advantage of EKF
method is that it does not reduce bandwidth efficiency as in
self cancellation method because the frequency offset can be
estimated from the preamble of the data sequence in each
OFDM frame.
Self cancellation does not require very complex
hardware or software for implementation. However, it is not
bandwidth efficient as there is a redundancy of 2 for each
carrier. The ML method also introduces the same level of
redundancy but provides better BER performance, since it
accurately estimates the frequency offset. EKF
implementation is more complex than the ML method but
provides better BER performance.
Further work can be done by extending the concept
of self-ICI cancellation and by performing simulations to
investigate the performance of these ICI cancellation schemes
in multipath fading channels.

Figure 4: BER performance with ICI
Cancellation for ε=0.5
Figure 4 illustrates that for frequency offset value of
0.5, BER increases for both the methods but ML method
maintains a lower bit error rate than SC.EKF is better than SC
method.
In figure 5, for frequency offset value of 0.75, self-
cancellation method has a BER similar to standard OFDM
system since the self-cancellation technique does not
completely cancel the ICI from adjacent sub-carriers and
6. REFERENCEs
[1] Ramjee Prasad, “OFDM for wireless communication
system”,Artech House,2004.
[2]S.Weinstein and P.Ebert, ‘Data transmission by
frequency-division multiplexing using the discrete Fourier
transform,’ IEEE Trans. Commun.,vol.19, pp. 628-634, Oct.
1971.
[3] L.J. Cimini, “Analysis and Simulation of a Digital Mobile
Channel Using Orthogonal Frequency Division Multiplexing”,
IEEE Transactions on Communication. no.7 July 1985.
[4] Russell, M.; Stuber, G.L.; “Interchannel interference
analysis of OFDM in a mobile environment”, Vehicular
Technology Conference, 1995 IEEE 45th, vol. 2, pp. 820 –
824,.Jul. 1995
[5] X.Cai, G.B.Giannakis,”Bounding performance and
suppressing intercarrier interference in wireless mobile
OFDM”, IEEE Transaction on communications, vol.51, pp.
2047-2056, no.12, Dec.2003.

UbiCC Journal - Volume 3 31

 [6] J. Armstrong, “Analysis of new and existing methods of
reducing intercarrier interference due to carrier frequency
offset in OFDM,” IEEE Transactions on Communications,
vol. 47, no. 3, pp. 365 – 369, March 1999.
[7] Y. Fu, S. G. Kang, and C. C. KO, “A new scheme for
PAPR reduction in OFDM systems with ICI self-
cancellation,” in Proc. VTC 2002- Fall, 2002 IEEE 56th
Vehicular Technology Conf., vol. 3, pp 1418–1421, Sep. 2002.
[8] Y.Zhao and S. Häggman, “Intercarrier interference self-
cancellation scheme for OFDM mobile communication
systems,” IEEE Transactions on Communications, vol. 49, no.
7, pp. 1185 – 1191, July 2001.
[9] J.-J. van de Beek, M. Sandell, and P.O. Borjesson, “ML
estimation of time and frequency offset in OFDM systems,”
IEEE Trans. Signal Process., 45, pp.1800–1805, July 1997.
[10] Tiejun (Ronald) Wang, John G. Proakis, and James R.
Zeidler “Techniques for suppression of intercarrier
interference in ofdm systems”. Wireless Communications and
Networking Conference, 2005 IEEE Volume 1, Issue, 13-17
pp: 39 - 44 Vol. 1, March 2005.
[11] William H.Tranter, K.Sam Shanmugam, Theodore
S.Rappaport, Kurt L.Kosbar, “Principles of Communication
system simulation with wireless application”, Pearson
Education, 2004.
[12] P.H. Moose, “A technique for orthogonal frequency
division multiplexing frequency offset Correction,” IEEE
Trans. Commun., 42, pp.2908–2914, October 1994

UbiCC Journal - Volume 3 32

 A NEW SIGNALLING PROTOCOL FOR SEAMLESS ROAMING
IN HETEROGENEOUS WIRELESS SYSTEMS

Azita Laily Yusof, Mahamod Ismail, Norbahiah Misran

Dept of Electrical, Electronic & System Engineering,
Universiti Kebangsaan Malaysia,
43600 UKM Bangi, Selangor,
Malaysia.
Tel.: +60389216122, Fax : +60389216146
Email: laily012001@yahoo.com, {mahamod, bahiah}@eng.ukm.

ABSTRACT

The world is undergoing a major telecommunications revolution that will provide

ubiquitous communication access to citizens, wherever they are. Seamless
roaming across different wireless networks which has different types of services
and quality of service guarantees has becomes a major topic for the past several
years in the research area. With the integration of different technologies, the
signaling protocol of mobility management must be designed to support seamless
roaming for both intra and interdomain system. In this paper, we designed a
simplified system architecture, called enhanced system architecture evolution
(eSAE) to support mobility between multiple heterogeneous wireless system.
eSAE contains fewer network nodes and is reduced to only the enhanced node B
(eNB) and access gateway (aGW) that comprise Mobility Management Entity
(MME) and User Plane Entity (UPE). We designed a signaling protocol for the
location registration due for intersystem roaming in next generation wireless
systems. Performance analysis has been carried out and based on this proposed
architecture, it is shown that this enhancement can reduce the signaling cost and
latency of location registration.

Keywords: Seamless roaming; Handoff latency; Intra and interdomain system;

Heterogeneous wireless system

1 INTRODUCTION network to another. This protocol needs to request

In the next generation wireless systems, it is
expected that the population of the mobile users will
be increased with the development of various
applications in the seamless global . Mobile users
can have different services that suits their need and
can move freely between different wireless systems.
However, different wireless system will have
different environments, interworking and integration.
This scenario has becomes challenges for the
researcher to support intra and intersystem mobility
for providing continuous wireless services to mobile
users in the next generation heterogeneous wireless
networks.
There has been many proposals to integrate
different wireless systems. In [1], the mobility
gateway location register (GLR) has been developed
to support the intersystem roaming. The GLR
converts signaling and data formats from one
location registration after it receives signals from the
new system and this cause high overhead of
signaling cost and processing time. It also causes the
triangular call routing problem because the call for
roaming mobile in the same network need to route to
the previous network before delivered to the new
network. Boundary location register (BLR) [2] was
designed in order to solve this problem. In this
protocol, the home location register (HLR) is not
involved in location registration unless the mobile
goes through into another system. So the incoming
calls of intersystem roaming mobiles are delivered to
them directly. However, this approach is not scalable
in the sense that one BLR gateway is needed for each
pair of adjacent networks when integrating multiple
networks.
In [3], they proposed a distributed gateway
foreign agent (GFA) where each foreign agent [FA]
can function dynamically either as an FA or GFA.

UbiCC Journal - Volume 3 33

There is no fixed regional network boundary and according to its changing mobility and packet arrival
mobile decides to perform the home location update pattern. However, this
scheme increases the requirement of the processing among different network operators. The architecture
capability on each mobility agent and mobile shown in figure 1, where the NIA functions as a
terminals. The hierarchical Intersystem Mobility trusted third party for authentication dialogs between
Agent (HIMA) [4] was proposed where it acts as an the foreign agent and home network. The working
anchor point to forward data as the user moves from principle of this third party architecture is as follows.
one network to another. The HIMAs are placed at the When a mobile user requests services from an
gateway routers or anchor routers for mobile users foreign network (FN) and the FN determines that it
with high roaming profiles. However, the scheme of has no SLA with the user’s HN provider, it forwards
address administrative issues and service level the request to NIA to authenticate the user. Then,
agreements across different wireless network and NIA talks to the user’s HN provider and mediates
service providers is not analyzed in this paper. between the FN and HN for authentication message
In [5], the author introduced an architecture exchanges. Once the user is authenticated, NIA also
called ubiquitous Mobile Communications (AMC) to creates security associations/keys required between
integrate multiple heterogeneous systems. AMC different network entities. At the end of the proposed
eliminates the need for direct SLA among service security procedures, the HN and FN will be mutually
providers by using a third party, Network authenticated, and will have session keys for secured
Interoperating Agent (NIA). In this paper, they use data transfer. They integrate the authentication and
distributed and hybrid scheme as a network selection. Mobile IP registration processes as defined in [5].
However, because the decision making is
implemented in the mobiles, so the system
information has to be broadcasted to the mobiles
periodically by the handoff management module,
resulting in a great update cost of the system.
Moreover, the existing protocol does not consider the
determination of the NIA’s number required for
global integration. Low complexity, centralized
network selection scheme [6] has been proposed to
overcome the shortcomings of NIA. The proposed
scheme eliminated the update cost whereby this
scheme will only be invoked by changes in end
users’ service requirements, beginning of a new
application, or ending of an existing application.
In this paper, we propose a simplified
network architecture, eSAE to support the low
latency system. The network is simplified and reduce
to only the Base Station called enhanced Node B
(eNB) and access gateway (aGW) that consists of
Mobility Management Entity (MME) and User Plane
Entity (UPE). The system uses all Internet Protocol
(IP) network where all services are via packet switch
domain only. In this proposed architecture, we
design a signaling protocol for authentication and
authorization.
The rest of this paper is organized as
follows. First we describe the existing system
architecture and the signaling protocol called AMC.
Then we present our proposed simplified architecture
followed by the authentication and authorization
information flow in eSAE. We discuss the simulation
results and finally the conclusion.

2 CURRENT AMC PROTOCOL

AMC integrates heterogeneous wireless systems

using a third party, called Network Interoperating
Agent (NIA) which eliminates the need for SLAs

UbiCC Journal - Volume 3 34

 NIA
FN HN

AAAL

HLR
AU

AAAH
FA

HA
UE

Figure 1 : The architecture for AMC

3 THE PROPOSED ARCHITECTURE equipment (UE) context, generate

Figure 2 shows our proposed architecture for the
next generation wireless systems. eSAE will have
two types of network elements supporting the user
and control planes.
• The first is the enhanced base station, so
called enhanced node B (eNB). This
enhanced base station provides air interface
and performs radio resource management
for the access system.
• The second is the access gateway (aGW).
The aGW provides termination of the bearer.
It also acts as a mobility anchor point for
the user plane. It implements key logical
functions including Mobility Management
Entity (MME) to manage/store user
temporary identities, UE authentication and
authorization and mobility management and
User Plane entity (UPE) to manage/store
UE context and packet routing/forwarding,
initiation of paging.
Comparing the functional breakdown with existing
architecture:
• Radio Network elements functions, such as
Radio Network Controller (RNC), are
distributed between the aGW and the eNB.
• Core Network elements functions, such as
SGSN and GGSN or PDSN (Packet Data
Serving Node) and routers are distributed
mostly towards the aGW.

UbiCC Journal - Volume 3 35

 aGW
(MME/UPE)
(E/UPE)
eNB eNB
Figure 2 : The proposed mobility management architecture for next generation all-IP-based wireless systems
3.1 Authentication and Authorization
The working principle of this architecture is as
follows. When a mobile user requests service from a
FN and the FN determines that it has no SLA with
user's home service subscriber (HSS), it forwards the
request to aGW to authenticate the user. Then, aGW
talks to user's HSS and mediates between FN and
HSS for authentication message exchanges. Once the
user is authenticated, aGW also creates security
associations/keys required between different network
entities. Finally the HSS and FN will be mutually
authenticated, and will have session keys for secured
data transfer.
The authentication and Mobile IP
registration processes are integrated in the proposed
architecture using the procedures defined in [7].
IEEE 802.1x port access control standard [8] is used
for end-to-end mutual authentication between a UE
UbiCC Journal - Volume 3
aGW
(MME/UPE)
ME/UPE)
eNB eNB
and its HSS. IEEE 802.1x uses a special frame
format known as Extensible Authentication Protocol
(EAP) over LAN (EAPOL) for transportation of
authentication messages between a UE and an access
point (AP). EAP [9] over RADIUS [10] or Diameter
[11] is used for the transportation of authentication
messages between other entities. When the UE
roams into a FN, the authentication and MIP
registration are carried out as described below. Here,
EAP-SIM [12] is used to illustrate the authentication
process. Note that any other authentication schemes,
e.g. EAP-AKA [13], EAP-SKE [14], EAP-TLS [15]
etc. can also be used. Figure 3 shows the location
registration procedure.
36
 UE eNB aGW AAAH AuC Inter AS HSS
(MME/UPE) Anchor

1. Network Discovery and

Access System Selection

2. Attach Request [c1 + c2 ]

[c1 + c2 ] 3. Authentication [c3 + c4]

4. Attach Reply [c1 + c2 ]

5. Register MME [c3 + c4 + c5 + c6 ]

6. Confirm Registration [c3 + c4 + c5 + c6 ]

7. Selection of Intersystem
Mobility Anchor GW

[c1 + c2 ] 8. User Plane Route Configuration [c3 + c4 + c5]

9. Configure IP Bearer QoS [c7]

10. Attach Accept [c1 + c2 ]

Figure 3 : The authentication and authorization signaling messages

1. The UE discovers new access system and
performs access system and network selection.
2. The UE sends an attach request, MIP
Registration Request including Mobile-AAA
Authentication extension (as defined in [16]) to
the aGW. The UE also includes a SIM Key
Request extension [19] and a Network Access
Identifier (NAI) [18], e.g. UE@relam, in its MIP
Registration Request. The SIM Key Request
extension contains a random number
(NONCE_UE) picked up by the UE, which is
used for new authentication key generation as
discussed later in this section.
3. When the aGW receives the MIP Registration
Request and finds the Mobile-AAA
Authentication extension, it learns that the UE is
a roaming user. Based on the NAI in the MIP
Registration Request, the aGW recognizes that
the operator does not have direct SLA with the
UE's HN and forwards the MIP Registration
Request to the Home AAAH server (AAAH).
Once the AAAH receives the MIP Registration
Request containing the SIM Key Request
extension, first it verifies the Mobile-AAA
authentication extension. If the authentication is
successful, it contacts the home authentication
center (AuC) of the UE and obtains n number of
triplets (RAND, SRES, Kc), where RAND
denotes a random number, SRES denotes the
response and Kc is the key used for encryption.
Then it forwards a copy of these triplets to aGW.
When aGW receives n triplets it derives a
UE_AAAH key (KUE_AAAH) and calculates
message authentication code (MAC) for the
RANDs (MAC_RAND) using [19]
KUE_AAAH = h(n * Kc│NONCE_UE) and
MAC_RAND = PRF(KUE_AAAH, α) (1)

UbiCC Journal - Volume 3 37

 where α is n*RAND│key lifetime; and h() and
PRF() denotes a one-way hash function and a
keyed pseudo-random function, respectively.
Then, aGW sends the RANDs, MAC_RAND
and SIM Key Reply extension to UE. The UE
derives the corresponding SRES and Kc values
using its SIM card and the received RANDs. It
also calculates (KUE__AAAH) and MAC_RAND
using (20). It validates the authenticity of
RANDs by comparing the calculated
MAC_RAND with the received MAC_RAND.
Thus, confirming that the RANDs are generated
by its HN. If the MAC_RAND is valid, the UE
calculates a MAC for its SRES values using [19]
MAC_SRES = PRF(KUE _AAAH, n * SRES)
(2)
The MAC_SRES is used by aGW to know if the
SRES values are fresh and authentic. The UE
also generates security association keys;
(KUE_eNB) for the eNB and (KUE_HSS) for the HSS
using [19]
KUE_eNB = PRF(KUE _AAAH, AddeNB) and
KUE_HSS = PRF(KUE_AAAH, AddHSS) (3)
where AddeNB and AddHSS are the IP address of
eNB and HSS, respectively. These keys are used
to authenticate subsequent Mobile IP
registrations until the key lifetime expires.
4. Now, the UE resends MIP Registration Request
message to the eNB containing SRES extension
[19] and Mobile-AAA Authentication extension.
When eNB detects the presence of Mobile-AAA
Authentication extension, it forwards the MIP
Registration Request message to aGW. aGW
calculates MAC_SRES and compares that with
the received MAC_SRES. If valid, it forwards
the MIP Registration Request message to the
AAAH. After successful authentication AAAH
forwards the MIP Registration Request
containing KUE_HSS (calculated using (4)) to the
HSS.
KUE_HSS = PRF(KUE_AAAH, AddeNB, AddHSS ) (4)
5. The HSS confirms the registration of the new
aGW. Subscription data authorising the Default
IP Access Bearer are transferred. Information
for policy and charging control of the Default IP
Access Bearer is sent to the aGW.
6. An Inters AS Anchor is selected. The IP address
configuration is determined by user preferences
received from the UE, by subscription data, or
by HPLMN or VPLMN policies.
7. The Inter AS Anchor configures the IP layer
UbiCC Journal - Volume 3
with the determined user IP address. The user
plane is established and the default policy and
charging rules are applied. The user plane
establishment is initiated by the aGW.
8. The aGW provides the Evolved RAN with QoS
configurations for the Default IP Access Bearer,
e.g. the upper limits for transmission data rates.
9. The aGW accepts the UE's network attachment
and allocates a temporary identity to the UE.
Also the determined user IP address is
transferred. aGW calculates UE-eNB security
key, KUE_eNB, and forwards the MIP Registration
Reply (containing KUE_eNB and the Kc keys) to
eNB. eNB extracts KUE_eNB and the Kc keys and
send a MIP Registration Reply to the UE. The
Kc keys are used for secure data transfer
between the UE and eNB providing
confidentiality and integrity to the data traffic.
4 PERFORMANCE ANALYSIS of eSAE
In this section, we analyze the performance of
signaling cost and latency of location registration due
to intersystem roaming. The costs for location
registration are associated with the traffic of
messages between the entities and the accessing cost
of databases. To compare the total of signaling cost
between the proposed and existing architecture, we
assume the following parameters :
Table 1 : Simulation parameters
p transmission cost of messages between
the UE and the eNB
α transmission cost of messages between
the eNB and the aGW
β transmission cost of messages between
the aGW and the HSS
c1 transmission cost of messages between
the UE and the eNB
c2 transmission cost of messages between
the eNB and the aGW
c3 transmission cost of messages between
the aGW and the AAAH
c4 transmission cost of messages between
the AAAH and the AUC
c5 transmission cost of messages between
the AUC and the IASA
c6 transmission cost of messages between
the IASA and the HSS
c7 transmission cost of messages between
the eNB and the aGW
38
 We assume that a mobile keeps the same
1.6
mobility pattern when it moves into another system.
Further, we assume that the updating, deletion and 1.4
retrieval in the database have the same cost, a. We

Latency of location registration

calculate the total signaling of location registration 1.2

which is the sum of the transmission cost and the

cost associated with database access. Then we 1

calculate the latency of location registration where 0.8

we assume the average processing time of each
database access is 1/μ and the average waiting time is 0.6
w. So the latency for location registration is the total
time including waiting time in queue and the 0.4
NIA
processing time. eSAE

Figure 4 shows the comparison of total 0.2

0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5

signaling cost as a function of intersystem roaming Probability of intersystem roaming

probability. As we can see from the graph, the total

signaling cost increases as the intersystem roaming Figure 5: Latency of location registration
probability increases,. We can also observe that the
total signaling cost of the eSAE protocol is much
lower than the NIA protocol. It is seen that as
compared to the NIA protocol, the eSAE protocol
yields significantly improved because of the
simplified architecture. The NIA protocol has to 5 Conclusion
access more databases compared to the eSAE
protocol. Similar to the case of total signaling cost, In this paper, we introduced a new
the latency of location registration increases with the signaling protocol for mobility management
increases of the intersystem roaming probability. which is based on the enhancement of the SAE
Figure 5 shows the result obtained. Therefore, eSAE architecture. We proposed the detailed procedure
protocol reduces the total signaling cost and latency of location registration for the eSAE protocol.
of location registration so that it is more suitable for This protocol is specifically developed to decrease
an intersystem roaming environment. the latency of the NIA protocol. To summarize the
comparison of eSAE and NIA protocol, we
measured the signaling cost of location
16 registration. Moreover, we evaluated the latency
of the location registration, which is composed of
14
waiting time and processing time at a specific
12 database. The results show that the eSAE protocol is
Total signalling cost

able to reduce the signaling cost and latency of

10 location registration for the mobile’s moving across
different networks.
8

6 4 REFERENCES

4
NIA
[1] ETSI TS 129 120 V3.0.0, “Universal mobile
eSAE telecommunications systems (UMTS); mobile
2
0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5 application part (MAP) specification for
Probability of intersystem roaming
gateway location register (GLR)”, 3GPP/ETSI
2000, 2000-2003.
Figure 4 : Total cost of location registration [2] I.F. Akyildiz, W. Wang, “A new signaling
protocol for intersystem roaming in next
generation wireless systems”, IEEE Journal on
Selected Area in Communications, vol.19, no.
10, Oct. 2001, pp. 2040-2052.
[3] I.F. Akyildiz, W. Wang, “A novel distributed
dynamic location management scheme for
minimizing signaling costs in mobile IP”, IEEE
Transactions on Mobile Computing, vol. 1, No 3,
July 2002, pp. 163-175.
[4] N. Shenoy, “A framework for seamless roaming
across heterogeneous next generation wireless

UbiCC Journal - Volume 3 39

 networks”, Journal on ACM Wireless Networks.
[5] I.F. Akyildiz, S. Mohanty, J. Xie, “A ubiquitous
mobile communication architecture for next-
generation heterogeneous wireless systems”,
IEEE Communications Magazine, vol. 43, no. 6,
pp. 29-36, June 2005.
[6] H. Jia, Z. Zhang, P. Cheng, H. Chen, A. Li,
“ Study on network selection for next generation
heterogeneous wireless networks”, in Proc.
IEEE International Symposium on Personal,
Indoor and Mobile radio Communications”,
2006.
[7] Glass, S., Hiller, T., Jacobs, S., and Perkins, C.,
“Mobile IP authentication, authorization, and
accounting requirements,” RFC 2977, IETF,
2000.
[8] “IEEE Standard for Local and metropolitan area
networks - Port-Based Network Access
Control.” IEEE Std 802.1X-2001.
[9] Blunk, L. and Vollbrecht, J., “PPP Extensible
Authentication Protocol (EAP),” RFC 2284,
IETF, 1998.
[10]Rigney, C. and et al, “Remote Authentication
Dial In User Service (RADIUS),” RFC 2865,
IETF, 2000.
[11]Calhoun, P. R., “Diameter Mobile IPv4
application,” Internet Draft, draft-ietf-aaa-
diameter-mobile ip 16.txt, work in progress,
2004.
[12]Haverinen, H. and Salowey, J., “EAP SIM
authentication,” Internet Draft, draft-haverinen-
pppest-eap-sim-16.txt, work in progress, 2004.
[13]Arkko, J. and Haverinen, H., “EAP AKA
Authentication,” Internet Draft, draft-arkko-
pppest-eap-aka-09. txt, work in progress, 2003.
[14] Salgarelli, L., “EAP SKE authentication and key
exchange protocol,” Internet Draft, draft-
salgarelli-pppext-eap-ske-03.txt, work in
progress, May 2003.
[15]Aboba, B. and Simon, D., “PPP EAP TLS
Authentication Protocol,” RFC 2716, IETF,
1999
[16]Aboba, B. and Simon, D., “PPP EAP TLS
Authentication Protocol,” RFC 2716, IETF,
1999
[17]Haverinen, H., Asokan, N., and Maattanen, T.,
“Authentication and key generation for Mobile
IP using GSM authentication and roaming,” in
Proc. IEEE ICC (ICC'01), pp. 2453{2457.
[18]Calhoun, P. and Perkins, C., “Mobile IP network
access identi¯er extension for IPv4,” RFC 2290,
IETF, 2000.
[19] Haverinen, H., Asokan, N., and Maattanen, T.,
“Authentication and key generation for Mobile
IP using GSM authentication and roaming,” in
Proc. IEEE ICC (ICC'01), pp. 2453{2457.
[20]“3GPP System to WLAN Interworking:
Functional and Architectural De¯ni-tion.” Tech.
rep. 3GPP TR 23.934 v0.3.0. 3GPP.

UbiCC Journal - Volume 3 40

 Artificial Neural Network application in Parameter Optimization of Rectangular Microstrip
Patch Antenna
1 2
R.Malmathanraj , S.Thamarai Selvi
1
Lecturer /ECE National Institute of Technology, Tiruchirapalli malmathan@gmail.com
2
Professor and Head /Information Technology, Madras Institute of Technology, Anna University, Chennai.
stselvi@annauniv.edu

Abstract - Printed microstrip antennas and simple rectangular patch fed at the center of
arrays are known to have limitations in terms radiating wall. A microstrip patch antenna is a
of bandwidth and efficiency, all imposed by the radiating patch on one side of a dielectric
very presence of the dielectric substrate. The substrate, which has a ground plane on the
paper deals with the design of a probe fed and underside. The EM waves fringe off the top
edge fed rectangular microstrip patch antenna patch into the substrate, reflecting off the
with the basic parameters W,h,L,ε r,fo to ground plane and radiates out into the air.
achieve better bandwidth and directivity with Radiation occurs mostly due to the fringing
efficient radiation pattern and Gain. The field between the patch and ground. The
analytical results for various possible radiation efficiency of the patch antenna
dimensions and different dielectric values were depends largely on the substrate permittivity
calculated for achieving bandwidth and (εr) of the dielectric[2]. The basic geometry of
directivity without any structural complexities the microstrip patch is shown in fig (1)
.The analytical results were tested by
simulating with basic design software
PCAAD,MSTRIP40. To obtain an optimum
value for the design parameters of the
microstrip antenna Support Vector Machines
(SVM), Generalised Regularisation Neural
Network (GRNN) and Back Propagation
Network (BPN) were implemented to train the
network to attain optimized values to yield
wide bandwidth and better directivity with
high Gain. The application of artificial neural
network ensures an optimum design
methodology for microstrip antenna design
which is revealed when comparing the results
with analytical methods and the results of the Figure 1.Microstrip Patch Antenna Geometry
simulation softwares.

1. Introduction
Microstrip patch antennas have been
attractive due to their conformal properties.
Mathematical modeling of the basic microstrip
radiator was initially carried out by the Ideally, a thick dielectric is preferred
application of transmission-line analogies to for broadband purposes. Small values of width
W of patch result in low antenna efficiencies

UbiCC Journal - Volume 3 41

while large W values lead to higher order various complex structures adopted for the
modes. Substrate thickness should be chosen as enhancement of bandwidth, Directivity and
large as possible to maximize bandwidth and Gain. The size of the probe is selected as 0.2
efficiency, but not so large as to risk surface- mm and the various feeding positions were
wave excitation. The patch length is considered for the calculation[5]. The
determined by condition for resonance. This dimensions of the patch antenna along with the
occurs when the input impedance is purely substrate permittivity and the probe position is
real. The bandwidth of the patch is defined as varied for different operating frequencies and
the frequency range over which it is matched the numerical results were arrived using the
with that feed line within specified limits. In basic design formulas of the microstrip patch
other words, the frequency range over which listed below,
the antenna will perform satisfactorily. This
means the channels have larger usable The width of the Microstrip patch antenna is
frequency range and thus results in increased given by
transmission. The bandwidth of an antenna is
c
usually defined by the acceptable standing W= -(1)
wave ratio (SWR) value over the concerned 
r 1
2 f0
frequency range[3,4]. Dimensions of the top 2
patch were calculated to get the required Effective dielectric constant is given by
bandwidth and the impedance matching.
The advantages of microstrip antenna 
1
1 r 1  h 2
is that they are low-cost, conformable, reff = r  1 12  -(2)

lightweight and low profile, while both linear 2 2  W
and circular polarization easily achieved.
Disadvantages of microstrip antenna include Effective length ( Leff ) is given by
such as a narrow bandwidth, a low gain (~6
C
dB) and polarization purity is hard to achieve. Leff  -(3)
Several methods were reported in literature to 2 f 0 reff
improve impedance bandwidth including
employing wide band impedance matching, Length extension ( L ) is given by
stacked patches and utilizing thicker
substrates[9].  0.3
reff
W
 0.264

L 0.412h h  -(4)

2.Design Methodology reff 0.258W 

 0.8 
h 
2.1 Basic Rectangular Microstrip
The major design task of this paper is
optimization of the dimensions of the probe
fed rectangular microstrip patch antenna. The Actual length of patch ( L ) is given by
simplest approach is adopted to demonstrate
how effectively the Artificial neural network L= Leff 2L -(5)
can be used to train and optimize the various
parameters involved in the design of microstrip Ground plane dimensions ( Lg and W g) is given
patch antenna. This work concentrates only on by
the basic geometry of the microstrip ignoring Lg = 6h +L -(6)

UbiCC Journal - Volume 3 42

 Wg = 6h + W -(7)
Height of the substrate is given by
0.3C
h
2fo r
-(8)
The bandwidth of a rectangular patch is given
by
BW 3.77  
r 1
/ 2r  
W / L
h / 0  --(9)
Where f 0 is the resonant frequency,  r is the
relative substrate permittivity, C is the speed of
light 3 108 m/s.
[A][ L1 , L2 , r, H1 , X,Y, Xf Yf]T=[G,BW,S 11,PL]T
G=f(L1 , L2 , r, H1 , X,Y, Xf , Yf);
BW=f(L1 , L2 , r, H1 , X,Y, Xf , Yf);
S11=f(L1 , L2 , r, H1 , X,Y, Xf , Yf);
PL=f(L1 , L2 , 
r, H 1 , X,Y, Xf , Y f );
In the neural network design the inputs are L1 ,
L2 , r, H1 , X,Y, Xf , Yf and the output is
G,BW,S 11,PL.
An artificial neural network is an
information processing paradigm. That is
inspired by the way biological nervous
systems, such as the brain, process
information. The key element of this paradigm
is the novel structure of the information
processing system .It is composed of a large
number of highly interconnected processing
elements (neurons), working in union to solve
specific problems. Artificial neural network is
like a normal human, it learns by
examples.[2,6]
A neural network with feedback is an
adequate representation of the information
UbiCC Journal - Volume 3
2.2 Artificial Neural Network Design
for Rectangular Microstrip Patch Antenna
.
A microstrip rectangular patch
antenna (Fig. 1) can be viewed as a matrix
with X variables and four unknown the bs,
such that
AX=b
The unknown are the resonant frequency (RF),
bandwidth (BW), gain(G), and polarization
(PL). The variables X, are the patch length
(L1), patch width(L2), substrate height (H 1) ,
substrate relative permittivity (r) and the
feeding positions(Xf) and (Yf)
processing structure of rectangular microstrip
antennas, where the input neuron units are (L
,W, εr ,h ,P ,fo )and the output units are (BW ,D
,G ,RP ). The learning paradigm on the
microstrip is supervised learning, where the
mapping function between the inputs and
outputs is the matrix A. The inputs are
weighted and the effect that each input has at
decision making is dependent on the weight of
the particular input. The weight of an input is a
number that when multiplied with the input
gives the weighted input. Their calculation is
based on the method of moments. These
weighted inputs then generate the unknowns.
Those unknowns are then compared to stored
information that gives the desired bandwidth, -
directivity along with radiation pattern and
gain. The gain is expected to be greater than
3 db and polarization should be linear or
circular [1,2].
A good paradigm of supervised
learning that is of interest to microstrip
antenna designer is error correcting learning
that is minimization of error between the
desired and computed values. In this learning
paradigm, the set of weights that minimizes the
error between the teaching input and the
43
weighted inputs is obtained. Neural networks
are general function approximators. One
important characteristic is that they can learn
any Input-Output (IO) mapping by using the
information contained in a given input-output
data set without needing a structure definition
of the IO-mapping. The type of network,
parameter settings, number of hidden neurons,
and the connectivity of the neural network
define the structure of the approximated IO-
mapping. The only additional information a
neural network needs, besides the input-output
data, is the definition of the input and output
parameters, the relevant parameters which
span the IO-mapping. The ideal use of neural
networks is antenna model parameters
optimization. Without knowing the IO-
mapping structure of the model, the neural
network can learn to mimic the IO-mapping
[7,8].
In the IO-mapping problem a feed-
forward neural network is used because the
antenna model is a static mapping (except at
the very moment the design failure occurs).
Also the ’tangent sigmoidal’, e.g. ’tansig’,
activation function will be used in the hidden
layer of the network because of two reasons.
One, in the antenna design, IO-mapping is a
smooth mapping with little to none
discontinuities. The ’tansig’ is also a smooth
function with the capability of approximating a
discontinuity (by squashing the shape with
respect to the input axis). The second reason
for choosing the ’tansig’ function is that this
function is a very ’general’ function. After a
failure, the antenna design IO-mapping has
changed into an unknown form. Therefore an
activation function, which can be used to
mimic almost all shapes, is more preferable
since in that case all (unknown) IO-mapping
can be approximated by the neural network.
Neural networks are based on the
human brain and its enormous capability of
learning and adapting. Over decades, people
have been trying to model the human brain
UbiCC Journal - Volume 3
mathematically. The structure of the human
brain and the learning process is known, but
the main difference between both networks is
the effciency. The human brain is capable of
recognizing a familiar face in approximately
100-200 ms, where conventional computers
can take hours or days to fulfill less complex
tasks. The biological neural network is still
much faster then the artificial neural network
(ANN or NN), but the capabilities of the
neural networks are promising. In this section,
the general structure of a neural network will
be explained. First the concept of neurons is
treated with a special attention to the several
activation functions. The possible
combinations of several neurons in layers is
handled in the discussion about the
architecture of networks. Some example
networks are used to illustrate the effect of the
parameters of the networks. The primary
element in a neural network is the neuron, an
information-processing unit. A mathematical
model of an artificial neuron in given in Fig.
(2). The structure is similar to the human
neuron; more information about the
functioning of the human neural network. The
elements xi on the left side are the input signals
to the neuron k. These inputs are multiplied by
the corresponding weights wki and summed
together with the bias bk. The results of the
summation vk is passed through an activation
function (vk) producing the output yk.
The mathematics of the neuron given in
Fig. (2) can start with the weighted inputs
p
v k   wki x i -(10)
i 1
The output can be written as
yk k (vk bk ) -(11)
A weighted bias can be included by adding an
extra term in the first equation.
p
v   w xi -(12)
k ki
i 0
44
Where the bias is changed to a fixed input of 1
and with a weight of wk0 . The shape of the
output is then only depending on the activation
function of vk.
y k k (v k ) -(13)

The type of activation function has a large

influence on the output of the neuron as can be
seen from the equation (12). In a signal flow
diagram, a neuron can be represented as shown
in Fig. (3).

x0=1

x1 wk,0

wk,1

x2 wk,2 vk (k) yk

.
.
.
xn wk,n

Figure 2. Mathematical
definition of a neural network

X1
wk, bk
1

wk,
X2 2 Σ
.
UbiCC Journal - Volume 3 . 45
.
 Fig 3. Signal flow diagram
neuron
A neural network is a directed graph
consisting of nodes with interconnecting
synaptic and activation links, and is
characterized by four properties:
1. Each neuron is represented by a set of linear
synaptic links, an externally applied bias, and a
possibly nonlinear activation link. The bias is
represented by a synaptic link connected to an
input fixed at +1.
2. The synaptic links of a neuron weight their
respective input signals.
3. The weighted sum of the input signals
defines the induced local field of the neuron in
question.
4. The activation link squashes the induced
local field of the neuron to produce an output.
3. Architecture Selection
A variety of Neural network
architectures are available to process the data
from the input data set files. A multi layer
Backpropagation Network architecture,
Generalized Regularization neural network
(GRNN) and Support vector machines (SVM)
were used for training because of its ability to
generalize well when applied to a wide variety
of applications and also for the ability to have
better regression.
UbiCC Journal - Volume 3
of a
3.1 Learning
As the neural network software read
the training set, the network learns the data
patterns in the training set. Learning
subprograms differ depending on the
architecture selected. As training progressed,
statistical graphs furnished by the neural net
software, provided a means to monitor training
progress.
Numerical historical data and repetitive
examples in which the solution is already
known are required to train a neural network.
While the relationship between variables may
not be known, network results can be
improved by the addition of more variables.
Data may need different representation, for
example if data has a very large value range,
logarithms or other data transformations or
conversions may be necessary.
3.2 Generalised Regularisation
Neural Network (GRNN)
The GRNN is based on the Nadaraya –
Watson Kernel regression. GRNN’s feature
fast training times can model non linear
functions and have been shown to perform
well in noisy environments given enough data.
The primary advantage of the GRNN is the
speed at which the network can be trained.
Training a GRNN is performed in one pass of
the training data through the network, the
46
 training data values are copied to become the
weight vectors between layers. The
architecture of the GRNN is shown in the
figure 4, it has four layers input pattern,
summation and output, with weighted
connections Wij between the input and pattern
layer and

Pattern Units Summation Layer Output Units

Input
S
S
fo D

S
L

W S
BW

ε
r
D

P S- Summation Unit
D- Division Unit

fo

Figure 4.Architecture of GRNN Network

Input

UbiCC Journal - Volume 3 47


3.3 Support Vector Machines
Traditionally neural networks have
been used for classification, which is based on
Empirical Risk Minimization (ERM). SVM
was developed by Vapnik and had become
popular tools for data mining. The formulation
embodies the Structural Risk Minimization
(SRM), which is superior to empirical risk
minimization. SRM minimizes the upper
bound on expected risk as supposed to ERM
that minimizes the error on training data. So,
SVM generalizes much better. There are many
linear classifiers that can separate data, but
SVM only maximizes the margin i.e. the
distance between it and the nearest data point
in each class.
We have N training data {(x 1,y1),
(x2,y2),….. (xN,yN)} Where xi Є Rd and yi Є
{+1,-1}. It needs to be classified using a linear
hyper plane classifier
f(x) =sgn (w.x - b) -(14)
This hyper plane will have maximum distance
between each class. This hyper plane
H : y =w . x – b = 0 and two hyper planes
parallel to it
H1 : y =w . x – b= +1 -(15)
H2 : y =w . x – b = -1 -(16)
With no data points between H1 and H2, and
distance between H1 and H2 maximized.
Some training point will lie on the hyper plane
H1 and H2, they are called support vector
machines because they define the separating
plane and the other training points can be
removed or moved provided they don’t cross
the planes H1 and H2. The distance between
hyper plane H1 and H2 is 2/ || w||. To
maximize the distance between the two data
UbiCC Journal - Volume 3
sets or between H1 and H2 we need to
minimize ||w|| with the condition that there are
no data points between H1 and H2
w . x –b ≥+1 for yi = +1 -(17)
w . x –b ≤-1 for yi = -1 -(18)
Combining the above two equations,
yi ( w . x – b ) >= 1 -(19)
So the problem of maximizing the
distance between hyper plane H1 and H2 is
formulated as min ½ wT w subject to
yi ( w . x – b ) ≥1 -(20)
This is a convex quadratic problem in w, b in a
convex set. The solution is found by solving
using lagrangian method by introducing
lagrangian multipliers. It is easier to solve
using lagrangian dual equation given by
LD = Σi αi - Σi Σj αi αj yi yj xi · xj -(21)
The significance of the above equation is that
the training input vectors appear only as dot
product. So when the data is not linearly
separable it is required to transform the data
into a higher dimensional. This causes
complex calculations in neural networks but in
SVM as data appear only as a dot product all
calculation can be carried explicitly in low
dimension if a kernel function exists for
LD = Σi αi - Σi Σj αi αj yi yj Φ(xi) · Φ(xj) -(22)
as Φ(xi ) · Φ(xj ) = K(xi , xj )
Where K is the kernel function. This is
equivalent as the dot product in high
dimension is equal to kernel function in input
space. The common kernel function used is
Gaussian kernel,
K (xi , xj) = e - || xi – xj || 2 / σ2 -(23)
48

Mercers condition determines whether
a function g(x) can be used as a kernel or not,
∫g(x)2 dx should be finite.
4.Design Implementation and Result.
The dimensions of the rectangular
patch were selected in a trial and error basis
considering the constraints of the design in
selecting the values. The different geometrical
parameters were designed analytically and the
bandwidth given in equation (9) was used to
calculate the value for the selected dimensions.
The parameters were used to construct the
structure using the simulation software. The
bandwidth and directivity along with the gain
and radiation pattern of the design were
obtained. The parameters of the patch
equations (1-9) with the feed position for a
resonant frequency are fed as input to the
networks. The impedance bandwidth and
directivity was taken as the output of the
network. The analytical data values are given
as input to train the network to obtain an
optimized geometry for the probe fed
microstrip antenna .The wide range of
parameters was used to provide the optimum
result and the training steps were increased to
obtain the accuracy.
The validity of the network was tested
by comparing the analytical results obtained
from the basic formulas for a given set of input
values. The same parameters were used to
construct a probe fed rectangular patch using
simulation software shown in figure (7) and
the output radiation pattern was obtained as
shown in figure (8). The current pattern of the
designed antenna is also plotted which shows
the even distribution due to proper impedance
matching of the probe feed. The same values
were trained using the three networks
Backpropagation Network architecture (BPN),
Generalized Regularization neural network
UbiCC Journal - Volume 3
(GRNN) and Support vector machines (SVM)
whose results were in good agreement with the
analytical as well as the designed structure
output shown in Table (1). The input output
relations were also checked for the
experimental results. The Backpropagation
Network architecture achieves the antenna
parameter optimization with maximum time
for convergence. The GRNN and the SVM
neural network achieves optimization with
quicker learning time as shown in Fig 11 and
12. In this research analysis for antenna
parameter optimization the GRNN neural
network produced the accurate result with
comparatively minimum time for convergence.
The computational time was very less in terms
of seconds with high accuracy as shown in Fig
13. The optimized parameters obtained using
the training neural networks achieved high
impedance bandwidth of 7.8%, directivity
7.73db without side lobes and offered high
gain 8.67 dbi and radiation efficiency 100%
was attained. The results were comparatively
better when compared with the results from
analytical analysis and simulation analysis for
Microstrip patch antenna using PCAAD and
MSTRIP40.
To train the SVM parameters
[alpha,b] = trainlssvm ({X, Y, type, gam, sig2,
kernel, preprocess})
Outputs
alpha matrix with support values of SVM
b vector with bias term(s) of SVM
Inputs
49
Fig 5. Antenna output model using MATLAB
Software .

Fig 6 Antenna output model using MATLAB

Software .

UbiCC Journal - Volume 3 50

Model Trained object oriented representation Gam Regularization parameter
of the SVM model sig2 Kernel parameter (bandwidth in the
Model Object oriented representation of the case of the 'RBF_kernel')
SVM model kernel Kernel type default 'RBF_kernel'
X matrix with the inputs of the Xt inputs of the test data
training data preprocess preprocess
Y vector with the outputs of the
training data Plotting the graph in SVM
Type function estimation plotlssvm({X,Y,type,gam,sig2,'RBF_kernel','p
Gam Regularization parameter reprocess'},{alpha,b});
sig2 Kernel parameter (bandwidth in the
case of the 'RBF_kernel') Inputs
X matrix with the inputs of the training
kernel Kernel type (by default 'RBF_kernel') data
preprocess preprocess'(*) or 'original' Y vector with the outputs of the training
data
Simulating the SVM Type function estimation
Yt=simlssvm({X,Y,type,gam,sig2,'RBF_kerne Gam Regularization parameter
l','preprocess'},Xt); sig2 Kernel parameter (bandwidth in
the case of the 'RBF_kernel')
Outputs kernel Kernel type (by default
Yt matrix with predicted output of test 'RBF_kernel')
data preprocess preprocess
alpha support values obtained from
Inputs training
X matrix with the inputs of the training b Bias term obtained from
data training
Y vector with the outputs of the training 5.Conclusion
The radiation pattern of the designed
data
antenna presented in this paper figure (7)
Type function estimation clearly depicted that it is a wideband antenna
with high directivity, gain with radiation

UbiCC Journal - Volume 3 51

 efficiency. The major attraction of this antenna
is size reduction along with bandwidth and
directivity made it most suitable for satellite
communication, commercial applications. The
size reduction and operating frequency make it
suitable
for mobile communication. The parameter
optimization using the networks is the major
attraction of this paper, which highlights the
simplicity, accuracy and reduction in
computational time for the designers of
interest.

Figure 7.Structure of Probe Fed Microstrip Rectangular

Patch Antenna
Figure 8.Current Distribution
Pattern

Figure 9. Radiation Pattern of the

Optimized Patch Antenna
Figure
10. Plot Showing the Learning Trial of
Back Propagation Network

Fig 1 1. Plot to show the time for convergence

for SVM neural network.

UbiCC Journal - Volume 3 52

 Fig 13. Plot to show the weight surface of SVM

Fig 12. Plot to show the time for convergence for

GRNN neural network.

Figure 15. Output of the Optimized output of the Fig 14. Plot to show the radiation pattern
rectangular patch antenna using MSTRIP using PCAAD

UbiCC Journal - Volume 3 53

References

1.Dipak K.Neog, Shyam S.Pattnaik, C.Panda, Swapna Devi, Bonomali Khuntia, and Malaya Dutta, “Design of a Wideband Microstrip
Antenna and the use of Artificial Neural Networks in Parameter Calculation”, IEEE Antennas and Propagation Magazine, Vol.47,
No.3, June 2005,pp.60-65.

2. Inder J.Bahl, Prakash Bhartla and Stanislaw S. Stuchly, “ Design of Microstrip Antennas Covered with a Dielectric Layer”, IEEE
Transactions on Antennas and Propagation, Vol. AP-30, No. 2, MARCH 1982, pp. 314-318.

3. Kin-Lu Wong and Yi-Fang Lin, “Small broadband rectangular microstrip antenna with chip-resistor loading”, ELECTRONICS
LETTERS, 1 l September 1997,Vol. 33 No. 79,pp.1593, 1594.

4. S.Lebbar, Z.Guennoun, M.Drissi, and F.Riouch, “A Compact and Broadband Antenna Design Using a Geometrical- Methodology-
Based Artificial Neural Network”, IEEE Antennas and Propagation Magazine, Vol.48, No.2, April 2006,pp.146-154.

5. C. L. Mak, K. M. Luk, Senior Member, IEEE, K. F. Lee, Fellow, IEEE, and Y. L. Chow, “Experimental Study of a Microstrip Patch
Antenna with an L-Shaped Probe,” IEEE Transactions on Antennas and Propagation, VOL. 48, NO. 5, MAY 2000,pp.777-783.

6. R.K.Mishra and Patnaik, “Designing Rectangular Patch Antenna Using the Neurospectral Method”, IEEE Transactions on Antennas
and Propagation,AP-51,8 August 2003,pp.1914-1921.

7. S.S.Pattnaik, D.C.Panda and S.Devi, “Input Impedance of Rectangular Microstrip Patch Antenna Using Artificial Neural Networks”,
Microwave and Optical Technology Letters,32,5,5 March 2002,pp.381-383.

8. S.S.Pattnaik, D.C.Panda and S.Devi, “Radiation Resistance of Coax-Fed Rectangular Microstrip Patch Antenna Using Artificial
Neural Networks”, Microwave and Optical Technology Letters, 34,1,5 July 2002,pp.51-53.

9. D.M.Pozar, “Microstrip Patch Antennas,” in L.C.Godara (ed), Handbook of Antennas in Wireless Communications, New York, CRC
Press, 2001,Chapter 6.

10.Ye Bin Hu Gu Yu , “The analyze and improve TCP performance using a DSR route protocol based on signal strength”, IEEE
Wireless Communications, Networking and Mobile Computing, pp. 846 – 849, 2005.

11. Dongkyun Kim , Hanseok Bae, Jeomki Song, “Analysis of the interaction between TCP variants and routing protocols in
MANETs”, IEEE Parallel Processing, ICPP 2005 Workshops, pp 380-386, 2005.

12. Prabakaran, M. Mahasenan, A. , “Analysis and enhancement of TCP performance over an IEEE 802.11 multi-hop wireless network:
single session case”, IEEE International Conference on Personal Wireless Communications, pp-29- 33, 2005

13. Caihong Kai Yuzhong Chen Nenghai Yu, “An Improvement Scheme Applied to TCP Protocol in Mobile Ad Hoc Networks”, IEEE
International Conference on Mobile Technology, Applications and Systems, pp.1-6, 2005

UbiCC Journal - Volume 3 54

 PERFORMANCE OF TRANSMISSION SPACED SELECTION
DIVERSITY IN DS-CDMA SYSTEMS

Mona Shokair*, Maher Aziz**, Mohamed Nasr**

*Faculty of Electronic Engineering, Menoufia University, Egypt.
** Faculty of Engineering, Tanta, Tanta University, Egypt.
maher_luka@yahoo.com

ABSTRACT
The performance of transmission spaced selection diversity (SD) placed at base
station (BS) in DS-CDMA system remains insufficiently clear. This performance
will be evaluated by considering the effect of space distance between antennas
and the maximum Doppler frequency (fd) on bit error rate (BER) performance under
optimum conditions which are not clarified until now. Moreover, analysis of this
system is presented under the effect of Rayleigh fading.

Keywords: DS-CDMA, transmission spaced selection diversity, Rayleigh fading.

1 INTRODUCTION receiving or transmitting array is greater than a

The demand for many radio services is
increasing. New techniques are required to
improve spectrum utilization to satisfy that demand
without increasing the radio frequency spectrum
that is used. One technique in a digital cellular
system is the use of spread spectrum Code Division
Multiple Access (CDMA) technology [1]. Another
technique is diversity system. Cooperation between
a CDMA system and diversity system has also been
studied in [6]. Actually the main purpose of
diversity system is mitigating the multipath fading
which has negative effect on the quality of
transmission of mobile radio communication.
There are classifications of diversity system.
One view of classifications is transmission and
reception diversity. Other classifications are
frequency diversity, polarization diversity, spaced
diversity, time diversity and angle diversity. All
these classifications are presented in detail in [2].
To combine diversity branches, many combing
techniques are explained in detail in [1] and [2],
including space Selection Diversity SD. In SD one
of the M antenna branches that provides the highest
Signal-to-Noise Ratio (SNR) is selected for data
recovery.
The success of diversity techniques depends on
the degree to which the signals on the different
branches are uncorrelated. This requires that the
spacing between the antenna elements in the
certain minimum distance.
In this paper, the performance of transmission
selection spaced diversity under the effects of
spacing distance between antennas and the
maximum Doppler frequency will be studied under
using optimum conditions. These effects are not
clarified until now.
The organization of this paper is made as
follows: Sect. 2 introduces the analysis of the
system under Rayleigh fading. Computer
simulation conditions are done in Sect. 3. Results
are presented in Sect. 4. Conclusions are achieved
in Sect. 5.
2 ANALYSIS OF THE SYSTEM OVER
RAYLEIGH FADING
To get expressions for both SNR and BER
values, we consider a two – branch diversity system
at BS with correlated fading channels.
The received signal from each branch of the system
can be modeled as [3]
rk(t)=Rkejαkejψm(t) + nk(t) k=1,2 (1)
Where ψm(t) is the transmitted signal, Rk is a
Rayleigh – distributed amplitude factor, αk is a
uniformly distributed phase factor, and nk(t) is zero
– mean Additive White Gaussian Noise (AWGN).
The received signal can be described by:

UbiCC Journal - Volume 3 55

 rk(t)=[ Xk + jYk ] ejψm(t) + nk(t) k=1,2 (2) Eq. (9) and Eq. (10), a new SNR is defined for each
uncorrelated signal:
Where X1, X2, Y1, and Y2 are all Gaussian random
2
variables with zero mean and variance σ . Γ3 = (1 + ρ ) Γ (11)
The expectation can be expressed as, Γ4 = (1 - ρ ) Γ (12)

E [XiYk] = 0 i = 1, 2; k=1, 2 (3) Where Γ is the SNR of the original correlated

signals.
Now the BER values for a two-branch selective
E[X1X2] = E [Y1Y2] = ρσ2 (4)
diversity system can be calculated from the
following expression [3],
Where ρ is the correlation coefficient between the
fading channels.
Another assumption is that the channel statistics 1 ⎡ Γ3 Γ4 Γ3Γ4 ⎤
are independent of the AWGN variables, which are BER= ⎢1− − + ⎥
2 ⎣ Γ3 +1 Γ4 +1 Γ4Γ3 +Γ3 +Γ4 ⎦
also uncorrelated with each other, therefore
(13)
E [n1n2] = E[niXk] = E[niYk] = 0
i=1,2 ; k = 1,2 (5) 3 COMPUTER SIMULATION CONDITIONS

Ref. [3] introduces a transformation matrix T to
transform the correlated received signals r1(t) and
r2(t) into two new uncorrelated signals r3(t) and
r4(t) therefore,
DS-CDMA system with three antennas at the
BS and one antenna at the MS is assumed. Fig. 1
shows propagation model at the BS. Table 1 shows
simulation parameters.
Incident waves

⎡ r3 (t ) ⎤ ⎡ r1 (t ) ⎤
⎢r (t )⎥ = T ⎢r (t )⎥ (6)
⎣4 ⎦ ⎣2 ⎦
φ

⎡ 2 2⎤ θ
⎢ ⎥ #3
Where T = ⎢ 2 2 ⎥ #2 #1
(7)
⎢− 2 2⎥ d/λ
⎢⎣ 2 2 ⎥⎦
Figure1: Linear array and propagation model at
The two new received signals can be expressed as, BS.

rk(t) = [ Xk + jYk ] ejψm(t) + nk(t) Table 1: Simulation parameters

k=3 , 4 (8)
Modulation QPSK
Demodulation Coherent detection
By writing out the expressions of X3, X4, Y3,
Symbol rate 30 Ksps
and Y4, it can be seen that they are functions of
Spreading code Walsh code
Gaussian random variables, therefore they are also
Gaussian random variables; in addition they are Spreading factor 128
mutually independent. Thus,
To model the Rayleigh fading, we consider a set
of 8 plane waves that are transmitted in random
E[X23] =E [Y23] = (1+ρ) σ 2
(9)
direction within the range of φ degrees at the BS
[4]. The value of φ will be determined in the next
E[X24] = E [Y24] = (1- ρ) σ2 (10) section. Each of the plane waves has constant
amplitude and takes the random initial phase
Also, n3 and n4 are functions of AWGN distributed from 0 to 2π. The Doppler frequency is
random variables, and they have the same noise uniformly distributed from +fd to –fd ( fd : is the
power, and are uncorrelated with the new channel maximum Doppler frequency). The 8 incident plane
statistics. waves arrive in random direction from 0 to 2π at the
If the noise power at each receiver for the MS. QPSK is assumed with coherent detection. A
original correlated signals is the same, then, from square root raised cosine filtering with a roll-off

UbiCC Journal - Volume 3 56

 factor α of 0.5 is employed. A symbol rate of 30 1.0E-01
ksps is assumed. The spreading code is Walsh code
with spreading factor of 128. The Rayleigh fading M=2

BER
channels were disturbed by AWGN. 1.0E-02
M=1
The Performance of the diversity system
depends on correlation between antenna elements.
1.0E-03
The correlation is determined by antenna elements 80 100 120 140 160 180 200
spacing, angle spread of incident waves φ and
fd (Hz)
direction of arrival θ [5]. Thus, we have to optimize
these values to get better BER performance.
Figure 4: Maximum Doppler frequencies (fd ) vs.
BER for Eb/N0=10 dB
4 COMPUTER SIMULATION RESULTS

1.0E+00 rho=0
Fig. 2 shows the effect of arrival angle, θ, of the rho=0.1
signal on BER performance at Eb/N0=10dB. From 1.0E-01 rho=0.5
this figure, it can be concluded that changing the rho=0.9
value of θ gives slightly small effect. Therefore, we 1.0E-02
rho=1.0

BER
use in our simulation the value 300 of θ. 1.0E-03

1.0E-04
1.0E-02

1.0E-05
0 5 10 15 20
BER

SNR dB

Figure 5: Effect of branch correlation on BER

1.0E-03
0 45 90 135
performance of SD with two –branch diversity
cita in degrees (theoretical)

Figure 2: Arrival angle of the signal θ vs. BER. 1.0E-02

The effect of angle spread of incident waves φ 1.0E-03

is presented in Fig. 3. From this figure, we select
BER

the value of 120 which gives better BER

1.0E-04
performance.

1.0E-05
1.0E-02 0 2 4 6 8
d/λ
1.0E-03
BER

Figure 6: Normalized distance (d/λ) vs. BER

1.0E-04
We use these values on the following paragraphs.
1.0E-05
6 8 10 12 14 16 18 20
Fig. 5 shows the results of the theoretical BER
fay in degrees
in Eq.13 for two-branch diversity system with
different values of the correlation coefficient ρ.
Figure 3: Angle spread of incident waves φ vs. From this Figure it can be concluded that as the
BER correlation coefficient increases the BER
performance decrease. Also, as the coefficient
The effect of fd on BER performance is shown approaches 1, one of the diversity branches is
on Fig. 4. As fd increases, due to the increase in the effectively removed; this leads to lose the
speed of the Mobile, BER performance will advantage gained from antenna diversity. On the
degrade. This degrading is due to rapid changes in other hand, reducing values ρ correspond to an
channel characteristics. The lowest value of fd that increase in the spatial separation between antennas.
gives better BER is 90 Hz. For this reason we have to look for the optimum
antenna separation that yields better BER
performance. Simulations were performed where
the ratio d / λ was varied between 0.1 and 8. The
results are indicated in Fig. 6. It is clear that as the
ratio is increased, the BER performance is better.
When d / λ is 6, we already have optimal BER

UbiCC Journal - Volume 3 57

 results. Also, increasing d / λ beyond 6 does not due to diversity gain. This gain comes from
have any noticeable benefits. uncorrelated diversity branches. Moreover
increasing the maximum Doppler frequency
1.0E-01 degrades the BER performance due to the rapid
changes of channel characteristics. Moreover the
1.0E-02
M=1 analysis of this system is explained under Rayleigh
1.0E-03 M=2/0.5λ fading.
BER

M=3/0.5λ
1.0E-04
M=2/5.25λ
6 REFERENCES
1.0E-05 M=3/5.25λ

1.0E-06 [1] M. K. Simon and M. S. Alouini : Digital

0 5 10 15 20 Communication over Fading Channels: A Unified
Eb/N0
Approach to Performance Analysis. Wiley Series in
Telecommunications and Signal Processing. New
Figure 7: BER vs. Eb/N0 for fd =90 Hz York: Wiley- Interscience, 2000.
d/λ=0.5, 5.25 M=1, 2, and 3 [2] W. C. Jakes : Microwave Mobile
Communications. New York: John Wiley & Sons,
Fig. 7 displays the BER of transmit diversity for Inc.1974
different numbers of antennas, M=1, 2, and 3, and [3] L. Fang, G. Bi, and A. C. Kot, "New Method of
different values of antenna separation d / λ at the Performance Analysis for Diversity Reception with
base station. It is clear that increasing M and d / λ Correlated Rayleigh-fading Signals," IEEE
have a positive effect on the BER performance. Transactions on Vehicular Technology, September
Numerically, the amount of improvement in Eb/N0 2000, vol. 49, pp. 1807 – 1812.
for M = 2, and 3 is 4dB and 6dB with respect to [4] C. X. Wang and M. Patzold: Methods of
M=1 at d / λ=0.5 and BER=10E-4, respectively. Generating Multiple Uncorrelated Rayleigh Fading
Also, more improvement has been got at d / λ=5.25. Processes. IEEE Vech. Tech. Conf. 2003_spring.
It is increased to be 12dB and 14dB for M=2 and 3 [5] S. Kosono, and S. Sakagumi, "Correlation
with respect to M=1 and BER of 10E-4. Coefficient on Base Station Diversity for Land
Mobile Communication Systems", IEICE Trans.,
Comm., Vol. J 70-B No.4 1987, April, pp. 476-482
1.0E+00 [6] Qiang Zhao, New Results on Selection
theoritical
1.0E-01 simulation
Diversity Over Fading Channels, Thesis, December
27, 2002
BER

1.0E-02

1.0E-03

1.0E-04
0 2 4 6 8 10
Eb/N0 dB

Figure 8: Comparison of simulated and theoretical

results

Figure 8 compares the simulated and theoretical

results of BER for M=2. It shows that the
simulation results are a close match to the BER in
Eq. (13). The small difference is due to optimizing
the simulation parameters.

5 CONCLUSIONS

In this paper, the performance of transmission

selection spaced diversity in DS-CDMA system
was studied. This performance is not clarified until
now under the effect of changing the space distance
between antennas at BS and the maximum Doppler
frequency by using the optimum conditions. The
results show that increasing the space distance
between antennas gives better BER performance

UbiCC Journal - Volume 3 58

 WEB-BASED DECISION SUPPORT SYSTEMS
AS KNOWLEDGE REPOSITORIES FOR
KNOWLEDGE MANAGEMENT SYSTEMS

Yuri Boreisha
Minnesota State University Moorhead, USA
Boreisha@mnstate.edu

Oksana Myronovych
North Dakota State University, USA
Oksana.Myronovych@ndsu.nodak.edu

ABSTRACT
Problem solving and learning processes conducted on the basis of contemporary Web-
based DSS provide for development and enhancement of knowledge management
systems. Knowledge objects form the foundation of the conceptual approach to the
knowledge management based on the contemporary Internet technologies and
knowledge accumulated in DSS.

Keywords: knowledge management systems, decision support systems.

1 INTRODUCTION popularity, particularly with businesses.

Knowledge management (KM) has become an
important theme as managers realize that much of
their firm’s value depends on ability to create and
manage knowledge. To transform information into
knowledge a firm must use additional resources to
discover patterns, rules, and context where the
knowledge works [1-3].
Knowledge that is not shared and applied to the
practical problems does not add business value.
Today people can share their knowledge in three
primary ways. Organizational information systems
(IS) that store, manage, and deliver documents are
called content management systems (CMS). With the
arrival of modern communications technology,
people can share their knowledge via collaborating
knowledge management systems (KMS). In addition
to content management and collaboration, the
knowledge can be shared via expert systems.
Comprehensive discussion of important dimensions
of knowledge, the knowledge management value
chain, and types of KMS can be found in [2, 3].
Web 2.0 companies use the Web as a platform
to create collaborative, community-based sites (e.g.,
social networking sites, blogs, wikis, etc.). The Web
has now become an application, development,
delivery, and execution platform [4].
Software as a Service (SaaS) - application
software that runs on a Web server rather than being
installed on the client computer – has gained
Collaborating on projects with co-workers across the
world is easier, since information is stored on a Web
server instead of on a single desktop.
Rich Internet Applications (RIAs) are Web
applications that offer the responsiveness, “rich”
features and functionality approaching that of
desktop applications. RIAs are the result of today’s
more advanced technologies (such as Ajax) that
allow greater responsiveness and advanced GUIs.
Web services have emerged and, in the process,
have inspired the creation of many Web 2.0
businesses. Web services allow you to incorporate
functionality from existing applications and Web
sites into your own applications quickly and easily.
Web 2.0 companies use “data mining” to extract
as much meaning as they can from XHTML-encoded
pages. XHTML-encoded content does not explicitly
convey meaning, but XML-encoded content does. So
if we can encode in XML (and derivative
technologies) much or all of the content on the Web,
we’ll take a great leap forward towards realizing the
Semantic Web.
Many people consider the Semantic Web to be
the next generation in Web development, one that
helps to realize the full potential of the Web – the
“Web of meaning”. Though Web 2.0 applications are
finding meaning in the content, the Semantic Web
(heavily depended on XML and XML-based
technologies) will attempt to make those meaning
clear to computers as well as humans [5].

UbiCC Journal - Volume 3 59

 These trends in the Web Science – the new
science of decentralized information systems –
provide for new opportunities in the KM.
In this paper we consider contemporary
Decision Support Systems (DSS) as knowledge
repositories that can be expanded to KMS using the
Web 2.0 software development technologies and
tools. This paper is based on a series of previous
authors’ publications [6-11].
2 KNOWLEDGE MANAGEMENT AND
DECISION SUPPORT SYSTEMS
The AI representation principle states that once a
problem is described using an appropriate
representation, the problem is almost solved. Well-
known knowledge representation techniques include
rule-based systems, semantic nets and frame systems
[12].
KM refers to the set of business processes
developed in an organization to create, store, transfer
and apply knowledge. KM increases the ability of the
organization to learn from its environment and to
incorporate knowledge into business processes. There
are three major categories of KMS: enterprise-wide
KMS, knowledge work systems (KWS), and
intelligent techniques [2, 3].
Enterprise-wide KMS are general purpose,
integrated, firm-wide efforts to collect, store,
disseminate, and use digital content and knowledge.
Such systems provide databases and tools for
organizing and storing structured and unstructured
documents and other knowledge objects, directories
and tools for locating employees with experience in a
particular area, and increasingly, Web-based tools for
collaboration and communication.
KWS (such as computer-aided design,
visualization, and virtual reality systems) are
specialized systems built for engineers, scientists, and
other knowledge workers charged with discovering
and creating new knowledge for a company.
Diverse group of intelligent techniques (such as
data mining, neural networks, expert systems, case-
based reasoning, fuzzy logic, genetic algorithms, and
intelligent agents) have different objectives, from a
focus on discovering knowledge (data mining and
neural networks), to distilling knowledge in the form
of rules for a computer program (expert systems and
fuzzy logic), to discovering optimal solutions for
problems (genetic algorithms).
It is said that effective KM is 80% managerial
and organizational, and 20% technology. One of the
first challenges that firms face when building
knowledge repositories of any kind is the problem of
identifying the correct categories to use when
classifying documents. Firms are increasingly using a
UbiCC Journal - Volume 3
combination of internally developed taxonomies and
search engine techniques.
Organizations acquire knowledge in a number
of ways, depending on the type of knowledge they
seek. Once the corresponding documents, patters, and
expert rules are discovered they must be stored so
they can be retrieved and used. Knowledge storage
generally involves databases, document management
systems, expert systems, etc. To provide a return on
investment, knowledge should become a systematic
part of the organizational problem solving process.
Ultimately, new knowledge should be built into a
firm’s business processes and key application
systems.
KMS and related knowledge repositories should
facilitate the problem solving process (Figure 1).
During the process of solving problems managers
engage into decision making, the act of selecting
from alternative problem solutions.
The different levels in an organization
(strategic, management, and operational) have
different decision-making requirements. Decisions
can be structured, semi-structured or unstructured.
The structured decisions are clustered at the
operational level of the organization, and
unstructured decisions at the strategic level.
Management information systems (MIS)
provide information on firm performance to help
managers monitor and control the business, often in
the form of fixed regularly scheduled reports based
on data summarized from the firm’s transaction
processing systems (TPS). MIS support structured
decisions and some semi-structured decisions.
DSS combine data, sophisticated analytical
models and tools, and user-friendly software into a
single powerful system that can support semi-
structured and unstructured decision making [3, 13,
14].
The main components of the DSS are the DSS
database, the user interface, and the DSS software
system (Figure 2). The DSS database is a collection
of current data from a number of applications and
groups. Alternatively, the DSS database may be a
data warehouse that integrates the enterprise data
sources and maintains historical data.
The DSS user interface permits easy interactions
between users of the system and the DSS software
tools. Many DSS today have Web interfaces to take
advantages of graphics displays, interactivity, and
ease of use.
The DSS software system contains the software
tools that are used for data analysis. It may contain
various OLAP tools, data mining tools, or a
collection of mathematical and analytical models that
easily can be made accessible to the DSS users.
60
 Problem

Alternative
Standards solutions
(Desired state) Problem
(DSS)
solver
(Manager)
Information Constraints
(Current state)

Solution

Figure 1: Elements of the problem solving process.

The dialog manager is also in charge for the
information visualization. Finally, access to the
Internet, networks, and other computer-based systems
permits the DSS to tie into other powerful systems,
including the TPS or function-specific subsystems.
There are many kinds of DSS. The first generic
type of DSS is a Data-Driven DSS. These systems
include file drawer and management reporting
systems, data warehousing and analysis systems,
Executive Information Systems and Spatial DSS.
Data-Driven DSS emphasize access to and
manipulation of large databases of structured data
and especially a time-series of internal company data
and sometimes external data. Relational databases
accessed by query and retrieval tools provide an
elementary level of functionality. Data warehouse
systems that allow the manipulation of data by
computerized tools tailored to a specific task and
setting or by more general tools and operations
provided additional functionality. Data-Driven DSS
with Online Analytical Processing (OLAP) provide
the highest level of functionality and decision support
that is linked to analysis of large collections of
historical data.

Internal External
Data Data
DSS
Database/
Data Warehouse

DSS Software System

Models
OLAP Tools
Data Mining Tools

User Interface
(Dialog Manager)

Users

Figure 2: Main components of the DSS.

UbiCC Journal - Volume 3 61

 A second category, Model-Driven DSS, includes
systems that use accounting and financial models,
representational models, and optimization models,
and optimization models. Model-Driven DSS
emphasize access to and manipulation of a model.
Simple statistical and analytical tools provide an
elementary level of functionality. Some OLAP
systems that allow complex analysis of data may be
classified as hybrid DSS providing modeling, data
retrieval, and data summarization functionality.
Model-Driven DSS use data and parameters provided
by decision-makers to aid them in analyzing a
situation, but they are not usually data intensive.
Very large databases are usually not needed for
Model-driven DSS.
Knowledge-Driven DSS or Expert Systems can
suggest or recommend actions to managers. These
DSS are human-computer systems with specialized
problem-solving expertise. The expertise consists of
knowledge about a particular domain, understanding
of problems within that domain, and skills at solving
some of these problems (AI algorithms and solutions
can be used). A related concept is data mining. It
refers to a class of analytical applications that search
for hidden patterns in a database. Data mining is the
process of sifting through large amounts of data to
produce data content relationships. Tools used for
building Knowledge-Driven DSS are sometimes
called Intelligent Decision Support methods.
Relational
Database
Relational
DBMS
Report
Writing Mathematical
Models Groupware
Software
Periodic Outputs from Outputs
and mathematical from
special models groupware
reports
Figure 3: A DSS model that incorporates GDS, OLAP, and AI.
UbiCC Journal - Volume 3
Document-Driven DSS are evolving to help
mangers retrieve and manage unstructured documents
and Web pages. A Document-Driven DSS integrates
a variety of storage and processing technologies to
provide complete document retrieval and analysis.
WWW provides access to large document databases
including databases of hypertext documents, images,
sounds and video. Examples of documents that would
be accessed by Document-Driven DSS are policies
and procedures, product specifications, catalogs, and
corporate historical documents, including minutes of
meetings, corporate records, and important
correspondence. Search engines are powerful
decision-aiding tools associated with Document-
Driven DSS.
Group DSS (GDSS) came first, but now a
broader category of Communications-Driven DSS or
groupware can be identified. These DSS includes
communication, collaboration and related decision
support technologies. These are hybrid DSS that
emphasize both the use of communications and
decision models to facilitate the solution of problems
by decision-makers working together as a group.
Groupware supports electronic communication,
scheduling, document sharing, and other group
productivity and decision support enhancing
activities.
A DSS model that incorporates Group Decision
Support, OLAP, and AI is shown on Figure 3.
Knowledge Multidimensional
Database Database
Inference Multidimensional
Engine DBMS
Solutions Outputs
and from
explanations OLAP
62
 DSS facilitate the decision-making. Decision
making is an integrated part of the overall problem
solving process. KMS should facilitate the problem
solving process. In the next section we are going to
discuss how Web-enabled DSS can be integrated into
contemporary KMS.
3 WEB-ENABLED DECISION SUPPORT
SYSTEMS
All types of DSS can be deployed using Web
technologies and can become Web-based DSS.
Managers increasingly have Web access to data
warehouses and analytical tools. To discuss the
recent trends in this area the latest achievements in
the three-layer design, Rich Internet Applications
(RIA), and Web services should be taken into
account.
Three-layer design is an effective approach to
development robust and easy maintainable systems.
The corresponding architecture is appropriate for
systems that need to support multiple user interfaces.
Contemporary Web applications are three-layer
applications.
The most common set of layers includes the
following:
Data layer that manages stored data, usually in
one or more databases.
Business logic (domain) layer that implements
the rules and procedures of the business
processing.
View layer that accepts input and formats and
displays processing results.
RIA have two key attributes – performance and
rich GUI. RIA performance comes from Ajax
(Asynchronous JavaScript and XML), which uses
client-side scripting to make Web applications more
responsive by separating client-side user interaction
and server communication, and running them in
parallel. Various ways to develop Ajax applications
are discussed in [5].
Web services promote software portability and
reusability in applications that operate over the
Internet. Web service is a transition to service-
oriented, component-based, distributed applications.
Web services are applications implemented as Web-
based components with well-defined interfaces,
which offer certain functionality to clients via the
Internet. Once deployed, Web services can be
discovered, used/reused by consumers (clients, other
services or applications) as building blocks via open
industry-standard protocols. Web service architecture
is built on open standards and vendor-neutral
specifications. Services can be implemented in any
programming language, deployed and then executed
on any operating system or software platform.

Internal External
Data Data
DSS
Database/
Data Warehouse

Web Services
provide access to
DSS Software System

Ajax-Enabled Applications
implement
Dialog Manager

Internet Users

Figure 4: Web-enabled DSS.

The service-oriented architecture (SOA) provides consisting of different software components working
the theoretical model for all Web services. The model together. Consuming Web services is based on open
behind Web services is a loosely coupled architecture, standards managed by broad consortia (e.g., World

UbiCC Journal - Volume 3 63

Wide Web Consortium, Organization for the
Advancement of Structured Information Standards,
Web Services Interoperability Organization).
What makes Web services different from ordinary
Web sites is the type of interaction that they can
provide. Most of the enthusiasm surrounding Web
services is based on the promise of interoperability.
Every software application in the world can potentially
talk to every other software application. This
communication can take place across the old
boundaries of location, operating system, language,
protocol, and so on.
Three-layer architecture maps well on the
structure of main components of the DSS (see Figure
DSS
Database/
Data Warehouse
Web Service
Ajax-Enabled Application
Figure 5: Structure of a knowledge object.
Web-enabled DSS provide for expandable
collections of the knowledge objects that constitute the
knowledge repository of the corresponding KMS. From
this point of view the knowledge objects can be
considered as a knowledge representation technique.
4 PROBLEM SOLVING AND LEARNING
AI distinguishes two general kinds of learning. The
first kind is based on coupling new information to
previously acquired knowledge. Typical examples
include learning by analyzing differences, by managing
multiple models, by explaining experience, and by
correcting mistakes. The second kind is based on
digging useful regularity out of data; a practice often
refers as data mining. Typical examples include
learning by recording cases, by building identification
trees, by training neural nets, by training perceptrons,
by training approximation nets, and by simulation
evolution (e.g. genetic algorithms).
Expert systems primarily capture the tacit
knowledge of individual experts, but organizations also
have collective knowledge and expertise that they have
UbiCC Journal - Volume 3
2). RIA provide for efficient implementation of the
Dialog Manager GUI for DSS. Web services allow
incorporating functionality from existing applications
and due to this providing for access to the DSS
Software System through the SOA. The components of
the Web-enabled DSS are shown on Figure 4.
We can call a group of the following related
components a knowledge object (Figure 5). Discussed
techniques allow to create new Web services (based on
the existing ones and contemporary DSS software
systems), and Ajax-enabled application interacting with
these Web services. So we can talk about creation and
modification of the knowledge objects.
built up over the years. This organizational knowledge
can be captured and stored using case-based reasoning
(CBR). In CBR description of the past experiences of
human specialists, represented as cases, are stored in a
database for the later retrieval when the user encounters
a new case with similar parameters. The system
searches for stored cases with problem characteristic
similar to the new one, finds the closest fit, and applies
the solution of the old case to the new case. Successful
solutions are tagged to the new case and both are stored
together with the other cases in the knowledge base.
Unsuccessful solutions are also appended to the case
database along with explanations as why the solutions
did not work.
Problem-based learning (PBL) is (along with active
learning and cooperative/collaborative learning) one of
the most important developments in contemporary
higher education. PBL is based on the assumption that
human beings evolved as individuals who are motivated
to solve problems, and that problem solvers will seek
and learn whatever knowledge is needed for successful
problem solving. PBL is a typical example of an
64
application of the first type of learning in higher Combining the main ideas of CBR and PBL the
education [11]. following problem solving and learning process can be
depicted as it’s shown on Figure 6.

User describes User learns

the problem about the knowledge objects
that facilitate
the problem solving

System searches
Repository of Repository of
knowledge objects for knowledge objects
the suitable ones (based on a
Web-enabled DSS)

System asks user

additional questions
to narrow search

System stores
System finds the problem description
the closest fit and and the knowledge
provides access to object in the repository
knowledge objects

New knowledge object

is created to
better fit the problem

Figure 6: Problem solving and learning with knowledge objects.

5 CONCLUSIONS 6 REFERENCES

Knowledge is a complex phenomenon, and there are
many aspects to the process of managing knowledge.
Knowledge-based core competencies of firms are key
organizational assets. Knowing how to do things
effectively and efficiently in ways that other
organizations cannot duplicate is a primary source of
profit and competitive advantage that cannot be
purchased easily by competitors in the marketplace.
This paper discusses Web-enabled DSS, related
knowledge repositories, and KMS that facilitate the
problem solving and learning. The knowledge objects
approach to the knowledge representation allows
considering contemporary DSS as integrated parts of
the corresponding KMS.
[1] V. Supyuenyong, N. Islam: Knowledge
Management Architecture: Building Blocks and
Their Relationships, Technology Management for
the Global Future, Vol. 3, pp. 1210-1219 (2006).
[2] K.C. Laudon, J.P. Laudon: Management
Information Systems. Managing the Digital Farm,
Prentice Hall, pp. 428-508 (2006).
[3] R. McLeod, G. Schell: Management Information
Systems, 10th Edition, Prentice Hall, pp. 250-274
(2006).
[4] P.J. Deitel, H.M. Deitel: Internet and World Wide
Web. How to Program, 4th Edition, Prentice Hall,
pp. 50-117 (2008).
[5] T. Berners-Lee, et al: A Framework for Web
Science, Foundations and Trends in Web Science,
Vol. 1, No 1, pp. 1-130 (2006).

UbiCC Journal - Volume 3 65

[6] Y. Boreisha, O. Myronovych: Web-Based Decision
Support Systems in Knowledge Management and
Education, Proceedings of the 2007 International
Conference on Information and Knowledge
Engineering, IKE’07, June 25-28, Las Vegas,
USA, pp. 11-17 (2007).
[7] Y. Boreisha, O. Myronovych: Web Services-Based
Virtual Data Warehouse as an Integration and ETL
Tool, Proceedings of the 2005 International
Symposium on Web Services and Applications,
ISWS’05, June 27-30, Las Vegas, USA, pp. 52-58
(2005).
[8] Y. Boreisha, O. Myronovych: Data-Driven Web
Sites, WSEAS Transactions on Computers, Vol. 2,
No 1, pp. 79-83 (2003).
[9] Y. Boreisha: Database Integration Over the Web,
Proceedings of the International Conference on
Internet Computing, IC’02, June 24-27, Las Vegas,
USA, pp. 1088-1093 (2002).
[10] Y. Boreisha: Internet-Based Data
Warehousing, Proceedings of SPIE Internet-Based
Enterprise Integration and Management, Vol. 4566,
pp. 102-108 (2001).
UbiCC Journal - Volume 3
[11] Y. Boreisha, O. Myronovych: Knowledge
Navigation and Evolutionary Prototyping in E-
Learning Systems, Proceedings of the E-Learn
2005 World Conference on E-Learning in
Corporate, Government, Healthcare, and Higher
Education, October 24-28, Vancouver, Canada, pp.
552-559 (2005).
[12] P.H. Winston: Artificial Intelligence, Addison-
Wesley, pp. 15-228 (1992).
[13] S. French, M. Turoff: Decision Support
Systems, Communications of the ACM, Vol. 50,
No 3, pp. 39-40 (2007).
[14] Chien-Chih Yu: A Web-Based Consumer-
Oriented Intelligent Decision Support System for
Personalized E-Services, ACM International
Conference Proceeding Series, Vol. 60, pp. 429-
437 (2004).
66
 Simulating social interaction scenarios in an office.
∗ ∗ †
Michele Bezzi Robin Groenevelt Frederick Schlereth
October 15, 2007

Abstract extremely hard even in well structured environments,

such as an office. The main issue is the complexity of
Work team coordination is becoming a major chal- social human behavior due to its high variability, its
lenge in the contemporary complex working environ- dependency on external constraints such as temporal,
ments. Coordination process takes place through spatial context (e.g., environment layout) and task
direct interaction and explicit communication, but context (e.g., personal list of activities and goals).
it takes also advantage of informal social network A successful model should therefore incorporate all
within team members. Consequently, in order to de- these aspects, and, to be realistic, parameters have
velop realistic model of team coordination, we need to be set using experimental data.
to measure and model such interactions in real world On the positive side, recent sensor technologies
environments. We present an agent-based model for provide us an unprecedented recording of informa-
simulating people movement in a workspace, which tion from the physical world. In previous studies, we
may be used as tool for developing and testing social investigated the social patterns during some typical
relationship models. We demonstrate the model by office activities [2], using data from a sensor network
simulating office life in one of our laboratories and located in one of our laboratories [15, 14]. Collect-
comparing the results to actual measurements ob- ing long-term and reliable data using this pervasive
tained with a sensor network. environment is a long process and may raise privacy
issues. Consequently, working with a real life environ-
ment does not allow us to efficiently test the impact
1 Introduction of changes in the environment (e.g., impact of some
Large corporations are often organized in functional space rearrangements on group dynamics).
teams. The objective of team work is to achieve a The aim of the paper is to introduce an agent-based
common goal by integrating and coordinating indi- model for simulating a workspace with movements of
vidual capabilities. In this framework, social interac- people and face-to-face contact between individuals.
tions play a major role, and—although many commu- This model can be used as tool for investigating the
nication media are nowadays available—-face-to-face dynamics of social interactions, for which the results
interactions are still highly important [1, 5]. Accord- can be fed by and/or validated against actual mea-
ingly, theoretical models of how people interact in a surements obtained with a sensor network. In par-
certain environment can be useful to shed some light ticular, this allows assessment of these measures un-
on the mechanisms underlying the collective behavior der different conditions, such as assessing the impact
of teams and business units. However, finding realis- of a physical change in the environment, the effects
tic mathematical descriptions of social interactions is of team building exercises, the arrival of a new em-
ployee, or changes in layout of the teams.
∗ Accenture Technology Labs, 449, route des Cretes, Sophia

Antipolis, France The important reason for being able to simulate so-
† Chalmers University, Goteborg, Sweden cial encounters is that it allows us to study the effect

UbiCC Journal - Volume 3 67

of (changes in) the environment on the social behav-
ior of people. There are many questions for which,
to the best of our knowledge, little or few quantita-
tive studies exist. For example, how well and quickly
does a new employee get integrated into the working
society under a variety of scenarios? These scenarios
could include: having people working in open space
offices instead of in cubicles, having team meeting in
various locations, the location of a coffee machine,
the effect of being at the far end of building. Do
people get more social connections when teams are
mixed so that it forces people to walk around more?
We see our simulator as a step towards quantitatively
answering these kinds of questions, in the lack of real-
world measurements.
The sketch of the paper is the following: in Sec-
tion 2 we briefly summarize the main features to
model and the actual sensor network. In Section 3
we describe the probabilistic model underlying our
model. Social behavior, derived through numerical
simulations, are presented in Section 4. Finally, con-
clusions are drawn in the last section.
2 Modeling Office Activities
We chose an office environment as a test setting for
two reasons. First of all, quantitative evaluations of
various office activities have important practical ap-
plications (e.g., assessing the quality of space organi-
zation in the office, estimating connections amongst
different people/departments, safety and security).
Secondly, a video-camera infrastructure which col-
lects data on peoples movements and presence was
readily available in one of our offices and the data
thus collected is accessible to us [15, 14]. This last
experimental environment is composed of an office
floor at Accenture Technology Labs in Chicago. The
floor is equipped with a network consisting of 30 video
cameras, 90 infrared tag readers, and a biometric sta-
tion for fingerprint reading.
The first step was the fusion of this raw-sensor data
into a higher-level description of peoples movements
inside the office. Identification and tracking of the
people was performed using a Bayesian network. In
short (see [14] for details), the office space was di-
UbiCC Journal - Volume 3
vided into 50 locations, each of them the size of ap-
proximately a room. This allowed us to remove the
variability of paths inside a room while still main-
taining enough information about the movements of
people. Each sensor detects signals of people in its
sensory field. For each person and location the signals
were merged together to build the current probabilis-
tic evidence of finding a certain person in a specific
location, after which this information was integrated
with the current belief of the system (derived from
previous observations). The result was a sequence of
matrices, one for each time step, where the probabil-
ity of finding a person in each location is reported.
In the second step, starting from these matrices,
we derived the most likely paths for each tracked in-
dividual; these data were then analyzed to find fre-
quent patterns and appropriate statistical quantities
to describe long term activities. Extracted recurrent
patterns were identified later exploiting local seman-
tics (e.g., meetings usually take place in the meet-
ing room) as well as context-based knowledge (e.g.,
matching movement patterns with the information
available from electronic calendars). The data acqui-
sition system is currently still under development, so
we had too little data available to find meaningful
long-term recurrent patterns. Nonetheless, to give a
glimpse of the kind of statistical analysis we are inter-
ested in, we analyzed a limited data set showing, for
example, that functional teams, such as research and
development groups, tend to be strongly intercon-
nected inside the group, but loosely connected across
different groups. Results of this analysis are reported
in Ref. [2].
3 Numerical Simulations
In this section we present a model for simulating
movements of people in an office setting analogous
to the workspace described above. In fact, data col-
lection in a real environment is a long process and
it may generate privacy concern. Therefore, to freely
test our algorithms and hypothesis, we built an agent-
based simulator of movements of people an office. As
in the real-life setting, the office map was divided into
50 locations, each of them the size of a room (see
68

Figure 1: Snapshot of the simulation (at time 11
a.m.). Numbers indicate locations. Note that a meet-
ing is taking place in the North-East corner room.
Fig. 1). In total there were 30 people (agents). In
its simplest version, each agent had a set of possible
destinations in the office floor, with different proba-
bilities derived from the collected data and from our
knowledge of their office life. At each time step, each
agent decides to stay in the current location with
a certain probability (usually large if it is in his or
her own office) or to move to a destination sampled
from a distribution of destinations. In this last case,
the agent starts moving according to a specific path,
usually the shortest one, with possible random fluc-
tuations. An agent also has a personal schedule in
which specific tasks are listed (e.g., meetings, lunch,
coffee) with a corresponding time and probability of
performing that action. This schedule was derived
from samples of employees electronic calendars and
then integrated with context knowledge, such as typi-
cal arrival, lunch, and departure times. Furthermore,
in case two or more agents cross paths in the same
location, the probability of staying was increased by
a quantity, ∆p, specific for each agent. This proba-
bility mimics the fact that random encounters may
result in short conversations. Its numerical value was
derived from real data whenever available and using
context knowledge in the other cases. Fig. 1 shows
UbiCC Journal - Volume 3
a snapshot of the simulation (at time 11 a.m.). The
output of the agent-based system consisted of a tem-
poral sequence of matrices, which report the location
of each agent for each time step, with the same for-
mat as for the sensor network. This allowed us to
use the same analysis tools for both the agent-based
model and for the real-life data collected. Despite its
simplicity, this model showed a visual agreement with
the trajectories observed in the real environment. We
used this model to study the evolution of social in-
teractions.
4 Social Network Analysis
Social network analysis provides a powerful tool for
assessing patterns of relationships in informal net-
works [5, 3]. The nodes in the network represent the
people and the links represent the interactions be-
tween the nodes. Social network theory has a long
history [11], but has only recently been able to take
full advantage of the large use of digital communi-
cations; the properties of such networks have been
extensively studied using data from emails [6, 9] and
instant-messaging [16]. In the first study an individ-
ual’s emailing history is analyzed and his connections
are automatically generated and displayed as a graph.
Typical analysis include: the number of connections
and frequency of contacts, the diameter and clique-
ness (i.e., degree of local clusters) of the network,
the time evolution of the network, and identifying
the most-connected nodes. The distribution of con-
nections in social networks has often been shown to
follow a power law, i.e., the number of nodes with
connectivity k falls as:
n(k) ∝ k α
where a is a negative constant, usually somewhere
between 1 and 4. This leads to a scale free network in
which there are many nodes with few connections as
well as the existence of highly connected hubs, which
foster network cohesion and connections between dis-
tant nodes, even in very large networks. Emails or
instant-messaging logfiles provide a large source of
data about social relationships, and they give inter-
esting results and potential applications [17, 7], but
69
 1
10
n(k)

0
10 0 1
10 10
k
Figure 3: Social network as extracted from movement
Figure 2: The degree of connectivity k of a node plot- data from one day of simulation. Black and white cir-
ted against the frequency of nodes with degree k on a cles indicate researchers and developers, respectively.
log-log scale. Each point represents data points from
one numerical simulations over a 7 period.
ple from different teams are loosely connected. Re-
sults vary across different runs but the a two-clusters
they do not consider physical interactions and face- structure was already present. Similar results were
to-face communications that are at the basis of hu- obtained by analyzing the tracking data from the
man behaviors. In this study, we focused on this last real-life sensor network (see Fig. 2 in Ref. [2]).
feature, we estimate social relationships from pat- We simulated one week of activity and measured
terns of collocation in the workplace. This approach the properties of the resulting social network. Fig. 2
will be integrated with data collected from electronic shows the degree of connectivity k versus the fre-
communications in future studies, to better specify quency of nodes with degree k for one simulation.
the structure of the network and to investigate the In general, the observed distributions do no follow a
(possible) different topologies of electronic and phys- power law (straight line in the log-log plots). This is
ical social networks. probably due to the limited sampling size: there are
We inferred the structure of the social network in few agents and a short duration of the simulation.
the office by simulating the movement of a group of In fact, due to the small size of the environment,
people for long periods and considering a simple prox- this frequency distribution converges to a delta af-
imity rule: two individuals share a link if they spend ter 6 − 8 weeks of simulations, at this time every
enough time in the vicinity of one another. In addi- agent is directly connected to everybody else. Fur-
tion, we added to the system some context specific ther investigations and more experimental data are
rules, e.g., we excluded the entrance hall. This sim- clearly required to fully characterize the topology of
ple rule can lead to a number of false positives, e.g., this network, and to assess whether the structure of
two individuals may share the same location with- the social network in a real world physical space dif-
out interacting. However, we expect that in the long fers from those measured with email or chat log files,
run and with a large number of users it provides a where spatial extension and physical constraints are
gross estimation of global structure of the network of not taken in account.
interactions and of its evolution in time. Extending the period of simulation to 4 weeks, we
Fig. 3 illustrates the social network amongst two observed the network becomes fully connected after
departments (Research and Development) after one 9 working days (on average), even if the clusters cor-
day of simulation; it shows, for example, that peo- responding to the different teams are still present at

UbiCC Journal - Volume 3 70

the end of the simulation. This suggests that in small
environments people get connected in rather short
amounts of time. To check this hypothesis, we simu-
lated the arrival of a new employee in the office and
measured over the time the number of hops needed
to connect him to all the other people in the office
(shortest average path length). Fig. 4 shows the aver-
age number of hops (links) needed for this new joiner
to connect to any other employees in the office (tri-
angles indicate the average over 50 simulations and
bars correspond to the standard deviation). After
one month the new employee had directly interacted
with all the people in the office, i.e., Fig. 4 black
triangles ≃ 1 at day 30. Excluding formal meetings
from this dataset, we can estimate the contribution of
random encounters (square dots in Fig. 4). Random
encounters contribute largely to the increase the con-
nectivity stressing the relevance of informal contacts
to establish a personal social network. Indeed consid-
ering random encounters only, the network becomes
connected after 13 days (on average) and after 30 the
new joiner is, almost (1.3 hops on average, Fig. 4),
connected to all the others.
Our current experimental setup does not permit
long recordings so we were not able to compare the
simulation results to experimental data.
5 Conclusions
Social interactions are highly important in collective
activity, such as goal-oriented work teams. In par-
ticular, despite the fact that many communication
media are accessible, face-to-face interactions still
constitute one of the preferred media for informa-
tion transmission [1] and contribute to increase the
cohesion within groups. Furthermore, it has been
shown [10, 12] that the actual physical context, such
as the design of the environment and physical loca-
tions of agents, can considerably impact the human
agent coordination.
Accordingly, suitable measures of social interac-
tions in real environments are needed to develop ab-
stract model of team functioning. We previously de-
veloped a prototype pervasive environment allowing
the measuring of face-to-face interactions inside one
UbiCC Journal - Volume 3
of our laboratory during normal office hours. In this
paper we presented an agent-based model for mod-
eling peoples movements and social interactions in
the same setting. This simulator uses a set of simple
rules which reproduces a persons trajectories inside
the office, and provides a cheap and flexible tool to
develop and test pervasive environment and human
interaction models. In particular, we investigated the
social interactions taking place during normal work
days.
The paper does not present a complete model for
modeling the dynamics of interactions as we did not
consider, for example, digital communication media,
and—more importantly—we disregarded the content
of the interactions. Still, the results of this prelimi-
nary study show that it is technically possible to an-
alyze the spatial influence of the environment on the
behavior of the people and relevant numbers concern-
ing face-to-face interactions in real-environment can
easily be generated. This allows for important input
for collective human behavior modeling, as well as
practical implications to evaluate the implementation
of certain measures such as office design, team build-
ing efforts, efficient information transmission, and the
correct integration of new joiners. The next step will
be to validate these against real-life data from our ex-
perimental setup, and to possibly extend it to larger
(and richer) environments. To this scope, privacy is
clearly a major concern. Possible solutions include
users controlling the personal data released, limiting
the data a single party can access, data anonymiza-
tion, and following accepted ethical guidelines. In
applications where real-time is not a requirement (as
in our case for identifying social networks), the users
could have full control over the data released, e.g., re-
ceiving a weekly e-mail with the summary of events;
and deciding which of them to disclose for the analy-
sis. Even more important is finding a reasonable equi-
librium point in the trade-off between privacy and
benefits. In other words, users need to be provided a
clear and tangible return for their privacy investment
for gaining acceptance.
Lastly, even an analysis in some specific cases (re-
search laboratories, conferences, public events [8, 4,
13]) will hopefully increase our—at the moment very
limited—quantitative knowledge on social interac-
71
 2.5
2
Hops
1.5
1 cal co-presence at events into digital social net-
0.5
5 10 15 20 25 30
Days
Figure 4: Triangles indicate the average number of
hops to connect one worker with all the workforce
(average shortest path). In the x-axis the number
of days are reported, starting from day 8. Previous
days are not shown because the network is not fully
connected. Vertical bars indicate standard deviations
taken over 50 simulations. Square dots indicate the
same quantity considering random encounters only.
Standard deviations for random encounters are not
shown for clarity.
tions and their effects on collective behaviors.
Acknowledgments
[9]
Frederick Schlereth contributed to this study dur-
ing his internship at Accenture Technology Labs in
Sophia Antipolis. We thank Valery Petrushin and
Gang Wei for providing tracking data obtained from
Accenture Technology Labs in Chicago. [10]
References
[11] S. Milgram. The small world problem. Psychol-
[1] T. Allen. Architecture and Communication
Among Product Development Engineers. MIT
Press, Cambridge, MA, 1997. [12] A. Omicini, A. Ricci, M. Viroli, C. Castel-
[2] M. Bezzi and R. Groenevelt. Towards under-
standing and modeling office daily life. In 2nd
International Workshop on Exploiting Context
UbiCC Journal - Volume 3
Histories in Smart Environments , also eprint
arXiv: 0706.1926, 2006.
[3] K. Chan and J. Liebowitz. The synergy of so-
cial network analysis and knowledge mapping:
a case study. International Journal of Manage-
ment and Decision Making, 7(1):19–35, 2006.
[4] S. Counts and J. Geraci. Incorporating physi-
working. In CHI ’05: CHI ’05 extended abstracts
on Human factors in computing systems, pages
1308–1311, New York, NY, USA, 2005. ACM
Press.
[5] R. Cross and A. Parker. The Hidden Power of
Social Networks: Understanding how Work Re-
ally Gets Done in Organizations. Harvard Busi-
ness School Press, 2004.
[6] H. Ebel, L.-I. Mielsch, and S. Bornholdt. Scale-
free topology of e-mail networks. Phys. Rev. E,
66(3):035103, Sep 2002.
[7] R. Guimera, B. Uzzi, J. Spiro, and L. A. N. Ama-
ral. Team Assembly Mechanisms Determine Col-
laboration Network Structure and Team Perfor-
mance. Science, 308(5722):697–702, 2005.
[8] Q. Jones and S. A. Grandhi. P3 systems:
Putting the place back into social networks.
IEEE Internet Computing, 9(5):38–46, 2005.
H. Kautz, B. Selman, and M. Shah. Referral
web: combining social networks and collabo-
rative filtering. Commun. ACM, 40(3):63–65,
1997.
D. Kirsh. Distributed Cognition, Coordination
and Environment Design. Proceedings of the Eu-
ropean conference on Cognitive Science, pages
1–11, 1999.
ogy Today, 2(1):60–67, 1967.
franchi, and L. Tummolini. Coordination Arti-
facts: Environment-Based Coordination for In-
telligent Agents. Proceedings of the Third In-
ternational Joint Conference on Autonomous
72
 Agents and Multiagent Systems-Volume 1, pages
286–293, 2004.

[13] A. Pentland, T. Choudhury, N. Eagle, and

P. Singh. Human dynamics: computation for
organizations. Pattern Recogn. Lett., 26(4):503–
511, 2005.

[14] V. Petrushin, R. Ghani, and A. Gershman. A

Bayesian Framework for Robust Reasoning from
Sensor Networks. AAAI Spring Symposium on
AI Technologies for Homeland Security March,
pages 21–23, 2005.
[15] V. Petrushin, G. Wei, R. Ghani, and A. Ger-
shman. Multiple sensor integration for indoor
surveillance. Proceedings of the 6th international
workshop on Multimedia data mining: mining
integrated media and complex data, pages 53–60,
2005.

[16] R. D. Smith. Instant messaging as a scale-

free network. Arxiv preprint cond-mat/0206378,
2002.

[17] F. Wu, B. Huberman, L. Adamic, and J. Tyler.

Information flow in social groups. Physica
A: Statistical Mechanics and its Applications,
337(1-2):327–335, 2004.

UbiCC Journal - Volume 3 73

 NEW STOP & WAIT ARQ PROTOCOL

Nitin Jain, Rishi Asthana & Manuj Darbari

Uttar Pradesh Technical University, Lucknow
nitinjain_22@rediffmail.com, asthana_rishi@yahoo.com, manujuma@rediffmail.com

ABSTRACT
In all types of data communication systems, errors may occur. Therefore error
control is necessary for reliable data communication. Error control involves both
error detection and error correction. Previously error detection can be done by
Cyclic Redundancy Check (CRC) codes and error correction can be performed by
retransmitting the corrupted data block popularly known as Automatic Repeat
Request (ARQ). But CRC codes can only detect errors after the entire block of
data has been received and processed. In this work we use a new and “continuous”
technique for error detection namely, Continuous Error Detection (CED). The
“continuous” nature of error detection comes from using arithmetic coding. This
CED technique improves the overall performance of communication systems
because it can detect errors while the data block is being processed. We focus only
on ARQ based transmission systems. We will show have the proposed CED
technique can improve the throughput of ARQ systems by up to 15%.

Keywords: Cyclic redundancy check codes, arithmetic coding, automatic repeat request.

1 INTRODUCTION integration of this novel paradigm into popular,

When we talk about any type of data
communication system, we concern only on its
reliability. In all types of data communication
systems, errors may occur. Error control is the only
way out for avoiding this problem. It comes by
detecting the error in first step and then correctin g it
in another step. For error detection we had CRC
codes and for error correction we use to retransmit
the corrupted data which is popularly known as ARQ.
Although efficient, CRC’s can detect errors only
after an entire block of data has been received and
processed. An error detection scheme that is
“continuous” can detect errors while the block is
being processed. Thus, it can enhance the overall
performance of the communication systems.
In this paper, we use this type of new and
continuous method for detecting the errors. The new
method of error detection is based on the arithmetic
coder, and allows for an efficient way to detect errors
continuously in the bit-stream by investing a
controlled amount of redundancy in the arithmetic
coding operation. During our research, we became
aware that the idea of continuous error detection
based on arithmetic coding had been tackled earlier
by Boyd et al. [1], albeit with little system
performance analysis, or exposition of its utility in
communication systems. In this paper, we not only
undertake a more rigorous analysis of this paradigm,
quantifying the underlying tradeoffs involved in the
process, but also establish impressive gains in system
performance attainable through sophisticated
powerful transmission scenarios such as ARQ.
Upon applying this method of error detection to
stop-and-wait ARQ, gains in throughput were
achieved over conventional ARQ schemes at all bit
error probabilities. Result shows that the throughput
of new stop -and-wait ARQ protocol i.e. with CED is
approximately 15% enhanced than the throughput of
the conventional stop -and-wait ARQ protocol i.e.
with CRC.
The rest of the paper is organized as follows.
The basic idea behind the continuous error detection
is introduced in Section 2. Section 3 presents an
application of CED for ARQ transmission where it
provides significant throughput gains over
conventional CRC-based schemes. We conclude in
Section 4. References are given in Section 5.
2 IDEA BEHIND CED
To understand the error detection scheme, an
understanding of how the arithmetic coder works is
necessary. We assume that the reader is familiar with
arithmetic coding and refer readers that are
unfamiliar with arithmetic coding to [2].
Arithmetic coding is a method of data
compression in which data strings are mapped to
code strings which represent the probabilities of the
corresponding data strings. The method in which this
mapping is achieved requires a model which
specifies the assumptions that are made about the
source data. A simple example of a model is the
memoryless model where the current symbol being

UbiCC Journal - Volume 3 74

 encoded is independent of the previously encoded
symbols. Another simple example is the first-order
Markov model, where the current symbol being
encoded is dependent only on the previously encoded
symbol. For simplicity, we will examine the
memoryless model, keeping in mind that the analysis
generalizes to more sophisticated models. Thus,
encoding and decoding via the arithmetic coder
function by repetitively partitioning subint ervals
within the unit interval [0, 1) according to the
probabilities of the data symbols.
The basic idea is simple and consisting of
adding a “forbidden” symbol that is never encoded
by the arithmetic coder, but nonetheless occupies a
nonzero measure on the set [0, 1), then upon
decoding, if an error occurs, this “forbidden” symbol
is guaranteed to be decoded due to the loss of
synchronization. The amount of time it takes to
decode the “forbidden” symbol after the occurrence
of an error is inversely related to the amount of
redundancy added through introducing the
“forbidden” symbol. T his allows for control of the
number of bits we suspect need to b e retransmitted.
In fact, we can guarantee to a specified accuracy, that
errors will be localized to the previous m bits (where
m is a function of the amount of redundancy added)
upon decoding the “forbidden” symbol. This is
useful in an ARQ setting, becau se as soon as the
error is detected, we have a statistical assurance as to
how many bits need to be retransmitted to ensure that
the bit in error will be retransmitted.
3 APPLICATION OF CED
Simulations were run using a binary symmetric
channel at various bit-error probabilities. Several ten
kilobit packets were sent at each bit-error
probability, and the resulting throughput was
calculated. As a measure of performance, we
compared our method of ARQ i.e. with CED to the
conventional methods of ARQ i.e. wit h CRC. The
conventional methods of ARQ function by dividing
the data into packets and then attaching CRC’s [3] to
each packet. Upon detection of an error in the
conventional ARQ method, a retransmission of the
entire block is requested. To simulate a fair
comparison for our method versus conventional
ARQ methods, we used the optimal packet size for
each bit-error probability tested using conventional
ARQ. The optimal packet size was calculated by
differentiating the throughput equation for
conventional ARQ (details can be found in [4]) with
respect to the packet size, and solving for the packet
size which maximizes throughput. The resulting
throughputs are shown in Fig. 1. and we see that the
new method of ARQ outperforms conventional ARQ
methods at all bit-error probabilities.
UbiCC Journal - Volume 3
Figure 1: Throughput comparison curves for new
stop-and-wait ARQ protocol i.e. with CED (upper
curve in red color) versus conventional stop -and-wait
ARQ protocol i.e. with CRC (lower curve in blue
color ).
4 CONCLUSION
In this paper we have introduced a new method
of error detection for common ARQ protocols. We
analytically characterized the tradeoff of added
redundancy versus error -detection capability and
formulated a method for incorporating this new error
detection “device” into an ARQ type scenario.
We would also like to mention here that CED
can be put to good use to improve throughput
performance of transport protocols like TCP over
heterogeneous networks, where early detection of an
error can result in a potentially greater number of
retransmits, thereby increasing the probability of
successful reception over a fading channel. This is
currently being verified. The goal of this work is to
present the benefits that communication systems can
derive from using CED for throughput enhancement.
5 REFERENCES
[1] C. Boyd, J. Cleary, S. Irvine, I. Rinsma-
Melchert, and I. Witten, “Integrating error
detection into arithmetic coding,” IEEE Trans.
Commun., vol. 45, pp. 1–3, Jan. 1997.
[2] G. Langdon, “An introduction to arithmetic
coding,” IBM J. Res. Develop. , vol. 28, pp. 135–
149, Mar. 1984.
[3] T. Ramabadran and S. Gaitonde, “A tutorial on
crc computations,” IEEE Micro, vol. 45, pp. 62–
74, Aug. 1988.
[4] M. Schwartz, Telecommunication Networks:
Protocols, Modeling and Analysis. Reading,
MA: Addison-Wesley, 1987.
75
 Cohesive Modeling, Analysis and Simulation for Spoken Urdu Language
Numbers with Fourier Descriptors and Neural Networks

Engr. S K Hasnain, Member IET, Assistant Professor,

Pakistan Navy Engineering College, Karachi
National University of Engineering & Technology, Pakistan
hasnain@pnec.edu.pk

Dr. Azam Beg, Member IEEE, Assistant Professor

College of Information Technology,
UAE University Al-Ain, United Arab Emirate
abeg@uaeu.ac.ae

ABSTRACT
This unified research describes spoken Urdu numbers investigative analysis from ‘siffar’ (zero) to
‘nau’ (nine) for making a concrete foundation in recognition of Urdu language. Sound samples
from multiple speakers were utilized to extract different features using Fourier descriptor and
Neural networks. Initial processing of data, i.e. normalizing and time-slicing was done using a
combination of Simulink in MATLAB. Afterwards, the MATLAB tool box commands were used for
calculation of Fourier descriptions and correlations. The correlation allowed comparison of the
same words spoken by the same and different speakers.

The analysis presented in this paper laid a foundation step in exploring Urdu language in
developing an Urdu speech recognition system. In this paper the speech recognition feed-forward
neural network models in Matlab were developed. The models and algorithm exhibited high
training and testing accuracies. Our major goal work involves in the future use of TI
TMS320C6000 DSK series or linear predictive coding. Such a system can be potentially utilized in
implementation of a voice-driven help setup in different systems. Such as multi media, voice
controlled tele-customer services.

Keywords: Spoken number, Fourier, Correlation, Feature extraction, Feed-forward neural networks, Learning rate

Many researchers have worked in this regard. Some

1. INTRODUCTION
Automatic speech recognition has been an active
research topic for more than four decades. With the
advent of digital computing and signal processing,
the problem of speech recognition was clearly posed
and thoroughly studied. These developments were
complemented with an increased awareness of the
advantages of conversational systems. The range of
the possible applications is wide and includes:
voice-controlled appliances, fully featured speech-
to-text software, automation of operator-assisted
services, and voice recognition aids for the
handicapped [1]. The speech recognition problem
has sometimes been treated as a speech-to-text
conversion problem.
commercial software is also available in the market
for speech recognition, but mainly in English and
other European languages. While the humans are
speaking, the formats vary depending on the position
of jaws, tongue and other parts of the vocal tract.
Two related key factors are: bandwidth of each
format, and format membership in a known
bandwidth. The vowel for all human beings tends to
be similar [2].
Sounds are mainly categorized into these groups:
voiced sounds (e.g., vowels and nasals), unvoiced
sounds (e.g., fricatives), and stop-consonants (e.g.,
plosives). The speech starts in lungs but is actually
formed when the air passes through larynx and vocal
tracts. Depending on the status of vocal fold in larynx
the sound can be grouped into: voiced sound that is

UbiCC Journal - Volume 3 76

time-periodic in nature and harmonic in frequency;
the unvoiced is more noise-like. Speech modeling
can be divided into two types of coding: waveform
and source. In the beginning, the researchers tried to
mimic the sounds as it is, meaning, waveform
coding. This method tries to retain the original
waveform using quantization and redundancy. An
alternative approach makes use of breaking the sound
up into individual components which are later
modeled, separately. This method of utilizing
parameters is referred to as source coding. Different
characteristics of speech can be used to identify the
spoken words, the gender of the speaker, and/or the
identity of the speaker. Two important features of
speech are pitch and formant frequencies:
(a) Pitch is a significant distinguishing factor
among male and female speakers. The frequency of
vibration of vocal folds determines the pitch, for
example, 300 times per second oscillation of the
folds results in 300 Hz pitch. Harmonics (integer
multiples of fundamental frequency) are also created
while the air passes through the folds. The age also
affects the pitch. Just before puberty, the pitch is
around 250 Hz. For adult males, the average pitch is
60 to 120 Hz, and for females, it is 120 to 200 Hz.
(b) The vocal tract, consisting of oral cavity, nasal
cavity, velum, epiglottis, and tongue, modulates the
pitch harmonics created by the pitch generator. The
modulations depend on the diameter and length of
the cavities. These reverberations are called formant
frequencies (or resonances). The harmonics closer to
the formant frequencies get larger while others are
attenuated. While the humans are speaking, the
formants vary depending on the positions of the
tongue, jaw, velum and other parts of the vocal tract.
Two related key factors are: bandwidth of each
formant, and formant’s membership in a known
bandwidth. The vowels for all human beings tend to
be similar. Each vowel uttered by a person generates
different formants. So we can say that the vocal tract
is a variable filter, whose inputs are (1) the pitch, and
(2) the pitch harmonics. The output of the filter is the
gain or the attenuation of the harmonics falling in
different formant frequencies. The filter is called
variable filter model. The transfer function for the
filter is determined by the formant frequencies [3].
Correlation exists between objects, phenomena, or
signals and occurs in such a way that it cannot be by
chance alone. Unconsciously, the correlation is used
in everyday life. When one looks at a person, car or
house, one’s brain tries to match the incoming image
with hundreds (or thousands) of images that are
UbiCC Journal - Volume 3
already stored in memory [4]. We based our current
work on the premise that same word spoken by
different speakers is correlated in frequency domain.
In the speech recognition research literature, no
work has been reported on Urdu speech processing.
So we consider our work to be the first such attempt
in this direction. The analysis has been limited to
number recognition. The process involves extraction
of some distinct characteristics of individual words
by utilizing discrete (Fourier) transforms and their
correlations. The system is speaker-independent and
is moderately tolerant to background noise.
2. REVIEW OF DISCRETE
TRANSFORMATION & ITS MATLAB
IMPLEMENTATION
Discrete Fourier transform (DFT) is itself a sequence
rather than a function of continuous variable and it
corresponds to equally spaced frequency samples of
discrete time Fourier transform of a signal. Fourier
series representation of the periodic sequence
corresponds to discrete Fourier transform of finite
length sequence. So we can say that DFT is used for
transforming discrete time sequence x(n) of finite
length into discrete frequency sequence X[k] of finite
length. This means that by using DFT, the discrete
time sequence x(n) is transformed into corresponding
discrete frequency sequence X[k][4]-[5]..
DFT is a function of complex frequency. Usually the
data sequence being transformed is real. A waveform
is sampled at regular time intervals T to produce the
sample sequence of N sample values, where n is the
sample number from n=0 to n=N-1.
{x(nT)} = x(0), x(T ),...,x[(N − 1)T ]
The data values x(nT) will be real only when
representing the values of a time series such as a
voltage waveform. The DFT of x(nT) is then
defined as the sequence
X[(N − 1)ω] in the frequency
{X [kω]} = X(0), X(ϖ),........
domain, where ω is the first harmonic frequency
given by ω = 2π / NT.
Thus X [ k ω ] has real and imaginary components
in general, so that for the kth harmonic
X ( k ) = R ( k ) + j I( k )
X (k ) = [R 2
(k ) + I 2 (k ) ]
1/ 2
(2.1)
77
and
X [k] has the associated phase angle
φ ( k ) = tan −1 [I ( k ) / R( k )] (2.2)
where X[k] is understood to represent X[k]. These
equations are therefore analogous to those for the
Fourier transform. Note that N real data values (in
time domain) transform to N complex DFT values (in
frequency domain). The DFT values, X[k], are given
by:
N−1
FD [x (nT)] = ∑x(nT)e− jkωnT, k = 0,1,...,N − 1 (2.3)
n=0
where ω= 2π / NT and FD denotes the DFT.
N −1
(2.4)
X [k ] = ∑ x (n T ) e
n =0
− jk 2 π n / N
The Fast Fourier transform (FFT) eliminates most of
the repeated complex products in DFT. In C version
of signal processing algorithm, there are several
different routines for real and complex versions of
the DFT and FFT. When these routines are coded
into the MATLAB language, they are very slow
compared with the MATLAB fft routine, which are
coded much more efficiently. Furthermore, the
MATLAB routines are flexible and may be used to
transform real or complex vector of arbitrary length.
They meet the requirements of nearly all signal
processing applications; consequently, in this paper,
the fft routines are preferred over all discrete
transform operations. MATLAB’s fft routine
produces a one-dimensional DFT using the FFT
algorithm; that is when [XK] is a real sequence, fft
produces the complex DFT sequence [Xm]. In
MATLAB, the length N of the vector [XK] may not
be given. Thus both of the following are legal
expressions:
X=fft(x) (2.5)
X=fft(x, N)
The first expression in (2.5) produces a DFT with
same number of elements as in [XK], regardless of
whether [XK] is real or complex. In the usual case
where [XK] is real and length N, the last N/2 complex
elements of the DFT are conjugates of the first N/2
elements in the reverse order, in accordance with
(2.4). In the unusual case where [XK] is complex, the
DFT consists of N independent complex elements.
For example, the results of the following commands
UbiCC Journal - Volume 3
with N=4 can be easily verified using definition in
(2.6).
FFT computing time
=
1
log 2 N (2.6)
DFT computing time 2 N
The results of the following commands with N=4 can
be for example easily verified with [5]:
x=[1 0 0 1];
X=fft(x)
In this example, the DFT components [Xm]
=[2, 1-j, 0, 1+j] are found from (2.4). The second
expression in (2.6) specifies the value of N in (2.4),
which effectively overrides any previous
specification of the length of the vector x, thus the
following commands produce the same result:
x=[1 0 0 1 3];
X=fft(x,4)
The DFT, x= [Xm] has length = 4 is the same as in
previous example.
x=[1 1];
X=fft(x, 4)
[Xm] =[ 2, 1-j, 0, 1+j]
The result here is same because, when N is greater
than the length of x; X is the DFT of a vector
consisting of x extended with zeros on the right, from
the length of x to N. (The length of vector x itself is
not increased in this process). The MATLAB library
also includes a two dimensional fft routine called fft2.
The routine computes two-dimensional FFT of any
matrix, whose element may be, for example, samples
(pixel values) of a two dimensional image.
Usually, some recognition occurs when the incoming
images bear a strong correlation with an image in
memory that “best” corresponds to fit or is most
similar to it. A similar approach is used in this
investigation, to measure the similarity between two
signals. This process is known as autocorrelation if
the two signals are exactly same and as cross-
correlation if the two signals are different. Since
correlation measures the similarity between two
signals, it is quite useful in identifying a signal by
comparing it with a set of known reference signals.
The reference signal that results in the lowest value
of the correlation with the unknown signals is most
likely the identity of the unknown object [9].
78

Correlation involves shifting, multiplication and
addition (accumulation). The cross-correlation
function (CCF) is a measure of the similarities or
shared properties between two signals. Application
of CCF includes cross spectral density, detection and
recovery of signals buried in noise, for example the
detection return signals, pattern, and delay
measurement. The general formula for cross-
correlation rxY(n) between two data sequences x(n)
and y(n) each containing N data might therefore be
written as:
N −1 (2.7)
rxy = ∑x
n =0
(n) y (n)
The autocorrelation function (ACF) involves only
one signal and provides information about the
structure of the signal or its behaviour in the time
domain. It is special form of CCF and is used in
similar applications. It is particularly useful in
identifying hidden properties.
N −1 (2.8)
∑ x ( n) x ( n)
rxx =
n =0
3. FEEDFORWARD VS RECURRENT
NETWORKS
Neural networks have proven to be a power tool for
solving problem of prediction, classification and
pattern recognition. [12]-[17].
Fig.1 Neural network model for feedforward multilayer
Perceptron for analysis
Neural network architecture can be divided into two
principal types: recurrent and non-recurrent
networks. An important sub-class of non-recurrent
NN consists of architectures in which cells are
UbiCC Journal - Volume 3
organized into layers, and only unidirectional
connections are permitted between adjacent layers.
This is known as a feed-forward multi layer
Perceptron (MLP) architecture. This architecture is
shown in Fig 1.
4. DATA ACQUISITION AND PROCESSING
The system presented in this model was limited to
processing of individual Urdu numerals (0 to 9). The
data was acquired by speaking the numerals into a
microphone connected to MS-Windows-XP based
PC. We recorded the data for fifteen speakers who
uttered the same number set (0 to 9), specifically,
siffar, aik, do, teen, chaar, paanch, chay, saat, aath,
and nau. Each sound sample was curtailed to 0.9
minute.
FDATool
From Wave File
s1_w1.wav Out
(22050Hz/1Ch/16b)
One Digital To Wave
S-Function
Filter Design1 Device
yout
Signal To
Workspace
|FFT| 2 yout1
Magnitude Signal To
FFT Workspace1
Fig.2 Simulink model for data of numbers extraction
for analysis
One of the obvious methods of speech data
acquisition is to have a person speak into an audio
device such as microphone or telephone. This act of
speaking produces a sound pressure wave that forms
an acoustic signal. The microphone or telephone
receives the acoustic signal and converts it into an
analog signal that can be understood by an
electronic system. Finally, in order to store the
analog signal on a computer, it must be converted to
a digital signal [6],[8].
The data in this paper is acquired by speaking Urdu
numbers into a microphone connected to MS-
Windows-XP based PC. The data is saved into
‘.wav’ format files. The sound files are processed
after passing through a (Simulink) filter, and are
saved for further analysis. We recorded the data for
fifteen speakers who spoke the same number set, i.e.
zero to nine.
79
 In general, the digitized speech waveform has a high
FDATool dynamic range, and can suffer from additive noise.
From Wave File yout2
s10_w1.wav Out So first, a Simulink model was used to extract and
(22050Hz/1Ch/16b)
Signal To analyze the acquired data as shown in Fig.3.
Workspace2
Digital
Another Simulink model was developed for
One performing analysis such as standard deviation,
S-Function Filter Design1
To Wave mean, median, autocorrelation, magnitude of FFT
|FFT|2
Device2
From Wave File and data matrix correlation. We also tried a few
s10_w2.wav Out Magnitude Freq Autocorr 0.00000000000000 other statistical techniques, however, most of them
A
(22050Hz/1Ch/16b) FFT2 Vector LPC
failed to provide us any useful insight into the data
Scope2
Two Autocorrelation Display5 characteristics. We would also like to mention that
S-Function LPC we had started our experiments by using Simulink,
From Wave File but found this GUI-based tool to be somewhat
s10_w3.wav Out Time limited because we did not find it easy to create
Horiz Cat
(22050Hz/1Ch/16b)
|FFT|2
To
yout1
Vector multiple models containing variations among them.
Frame Scope This iterative and variable-nature of models
Three
Magnitude Buffer Frame Conversion Matrix Signal To eventually led us to MATLAB’s (text-based) .m
From Wave File FFT1 Concatenation Workspace1 files. We created these files semi-automatically by
s10_w4.wav Out
0.0000000
using a PERL-language script; the script was
(22050Hz/1Ch/16b) developed specifically for this purpose. Three main
Four Mean Display data pre-processing steps were required before the
data could be used for analysis:
Waterfall
To U( : )
From Wave File Frame Scope
4.1 Pre-Emphasis
s10_w5.wav Out
(22050Hz/1Ch/16b) Frame Conversion1 Convert 2-D to 1-D Waterfall2
By pre-emphasis [7], we imply the application of a
Five normalization technique, which is performed by
0.000000000000000e+000
dividing the speech data vector by its highest
From Wave File magnitude.
s10_w6.wav Out Display1
Median
(22050Hz/1Ch/16b) Matrix
Viewer
4.2 Data Length Adjustment
Six
RMS 0.0000000 Matrix FFT execution time depends on exact number of the
From Wave File samples (N) in the data sequence [XK], and the
Viewer
s10_w7.wav Out Display2
RMS execution time is minimal and proportional to
(22050Hz/1Ch/16b) Column 0.0000000
Sum
N*log2(N), where N is a power of two. Therefore, it
Seven is often useful to choose the data length equal to a
0.0000000 Matrix Display6
power of two [10].
From Wave File Sum
s10_w8.wav Out Display3
Standard 4.3 Endpoint Detection
(22050Hz/1Ch/16b) Deviation
Eight CONV 0.0000000 The goal of endpoint detection is to isolate the word
to be detected from the background noise. It is
From Wave File
s10_w9.wav Out Convolution Display4 necessary to trim the word utterance to its tightest
(22050Hz/1Ch/16b) limits, in order to avoid errors in the modeling of
subsequent utterances of the same word. As we can
Nine
see from the upper part of Fig. 4, a threshold has
been applied at both ends of the waveform. The
From Wave File
s10_w10.wav Out front threshold is normalized to a value that all the
(22050Hz/1Ch/16b) spoken numbers trim to a maximum value. These
values were obtained after observing the behavior of
Ten the waveform and noise in a particular environment.
We can see the difference in frequency
Fig.3 Extended Simulink model for analysis of Urdu characteristics of the words aik (one) to paanch
spoken numbers

UbiCC Journal - Volume 3 80

(five) in Fig 4–8 respectively.

4.4 Windowing

Speech signal analysis also involves application of a

window with a time less than the complete signal.
The window first starts with beginning of the signal
and then shifted until it reaches the end. Each
application of the window to the part of the speech
signal results in a spectral vector.
Fig.4 Correlation of the spoken Urdu number
4.5 Frame Blocking aik (one)

Since the vocal tract moves mechanically slow,

speech can be assumed to be a random process with
slowly varying properties. Hence the speech is
divided into overlapping frames of 100 ms. The
speech signal is assumed to be stationary over each
frame and this property will prove useful in further
operations [18].

4.6 Fourier Transform

The MATLAB algorithm for the two dimensional Fig.5 Correlation of the spoken Urdu number
FFT routine is as follows [9]: dau (two)

fft2(x) =fft(fft(x), ‘);

Thus the two dimensional FFT is computed by first

computing the FFT of x, that is, the FFT of each
column of x, and then computing the FFT of each
row of the result. Note that as the application of fft2
command produced even symmetric data, we only
show lower half of the frequency spectrum in our
graphs.

4.7 Correlation Fig.6 Correlation of the spoken Urdu number teen

(three)
Calculations for correlation coefficients of different
speakers were performed [2]-[3]. As expected, the
cross-correlation of the same speaker for the same
word did come out to be 1. The correlation matrix of
a spoken number was generated in a three-
dimensional form for generating different
simulations and graphs.

Figures 4 - 8 show the combined correlation

extracted for fifteen different speakers for the Urdu
number siffar (zero) to paanch (five), that
differentiate the number spoken by different speaker
for detailed analysis. Fig.7 Correlation of the spoken Urdu number
char (four)

UbiCC Journal - Volume 3 81

 SPEAKER: s1 w2
700

600

500

400

300

200

100

0
0 50 100 150 200 250 300

Fig 11. FFT magnitude spectrum for spoken

Fig.8 Correlation of the spoken Urdu number Urdu number dau (two)
paanch (five)

Figures 9 - 14 show the FFT magnitude spectrum 700

SPEAKER: s1 w3

extracted for fifteen different speakers for the Urdu

600
number siffar (zero) to paanch (five), that again
differentiate the number spoken by different speaker 500

for detailed analysis. 400

300

SPEAKER: s1 w0
800 200

700
100

600
0
0 50 100 150 200 250 300
500

400 Fig.12 FFT magnitude spectrum for spoken

300
Urdu number teen (three)
200

100 SPEAKER: s1 w4
400

0 350
0 50 100 150 200 250 300

300

Fig.9 FFT magnitude spectrum for the spoken 250

Urdu number siffar (zero) 200

150

100

SPEAKER: s1 w1 50

500
0
0 50 100 150 200 250 300
450

400
Fig.13 FFT magnitude spectrum for spoken
350
Urdu number char (four)
300

250

200
SPEAKER: s1 w5
500
150
450

100 400

50 350

300
0
0 50 100 150 200 250 300
250

200

Fig.10 FFT magnitude spectrum for spoken 150

Urdu number aik (one) 100

50

0
0 50 100 150 200 250 300

Fig.14 FFT magnitude spectrum for spoken

Urdu number paanch (five)

UbiCC Journal - Volume 3 82

Figures 15-19 show the combined magnitude 900
SPEAKER ONE: s1 w4

spectrum of correlation extracted for fifteen different 800

speakers for the Urdu number siffar (zero) to paanch 700

(five), shown here for brevity which differentiate the 600

number spoken by different speaker for detailed 500

analysis, while Fig 20 shows surface graph of the 400

300
number aik (one). 200

100

0
0 2000 4000 6000 8000 10000 12000
SPEAKER ONE: s1 w0
1800

1600
Fig.18 Magnitude spectrum of the correlation of the
1400
spoken Urdu number spoken chaar(four) by speaker-
1
1200

1000

800
SPEAKER ONE: s1 w5
1200
600

400 1000

200
800

0
0 2000 4000 6000 8000 10000 12000
600

Fig.15 The waveform of the correlation of the spoken 400

Urdu number siffar (zero) by speaker-1

200

0
0 2000 4000 6000 8000 10000 12000
SPEAKER TWO: s2 w0
600
Fig.19 Magnitude spectrum of the correlation of the
500 spoken Urdu number spoken paanch (five) by speaker-
1
400

300
SPEAKERS 15 WORD 1 surface

200

100 0.8

0.6

0
0 2000 4000 6000 8000 10000 12000 0.4

0.2

Fig.16 The waveform of the correlation of the spoken 0

15
Urdu number siffar (zero) by speaker-2 10
15
10
5
5
0 0

SPEAKER THREE:s3 w0
1800 Fig.20 The surface plot of the correlation of the
1600 spoken Urdu numbers spoken aik (one) by speaker-15
1400

1200

1000
4.8 Creating a Network
800

600

400
To create a feedforward neural network (MLP)
200
object in an interactive way to use the command
0
nntool to open the Neural Network Manager. We can
0 2000 4000 6000 8000 10000 12000
then import or define data and train our network in
Fig.17 Magnitude spectrum of the spoken Urdu number the GUI or export our network to the command line
siffar (zero) by speaker-3 for training [5].

net = newff(MxMx, [S1 S2 . . . SK], {TF1 TF2 . . .

TFK}, BTF, BLF, PF)

UbiCC Journal - Volume 3 83

%where MnMx (p − 1 × 2) matrix of min and max This section describes the system workflow for
values for x input vectors. creating, initializing, training and running the
%Si Size of ith layer, for K layers. network on test data. The following sub sections
%TFi Transfer function of ith layer, default = tansig. demonstrate the Word Network to demonstrate the
%BTF Backpropagation network training function, system workflow of the demo program.
default = traingdx.
%BLF Backpropagation weight/bias learning The first step is to create and initialize the Neural
function, default = learngdm. Network and set up the network configuration.
%PF Performance function, default = mse.
net=newff(minmax_of_p,[64, 32, 32, 1;],
4.9 Network Structure and Parameters {'tansig','tansig','tansig','purelin'},'traingd');
// this is input layer.
%Neural Network object: Layer = 64
... // this is hidden layer no 1.
%subobject structures: Layer = 35
inputs: {1x1 cell} of inputs // this is hidden layer no. 2
layers: {2x1 cell} of layers Layer = 35
outputs: {1x2 cell} containing 1 output // this is output layer.
targets: {1x2 cell} containing 1 target Layer = 1
%biases: {2x1 cell} containing 2 biases
inputWeights: {2x1 cell} containing 1 input weight
layerWeights: {2x2 cell} containing 1 layer weight 4.12 Data Structure Training
...
Data training is an array list of a structure which is
%weight and bias values:
presented. Each of the initial four string values are
IW: {2x1 cell} containing 1 input weight matrix
part of the input vector provided to the network. The
LW: {2x2 cell} containing 1 layer weight matrix
last string variable represents the target output for the
b: {2x1 cell} containing 2 bias vectors
network in the code fragmented below:
4.10 Network Training / Testing
net.trainParam.show = 50;
% x-axis of graph is 50 values apart
In incremental training the weights and biases of the
net.trainParam.lr = 0.01;
network are updated each time an input is presented
% learning rate
to the network. The cell array format is most
net.trainParam.epochs = 6000;
convenient to be used for networks with multiple
net.trainParam.goal = 0.01;
inputs and outputs, and allows sequences of inputs to
% tolerance
be presented. The matrix format can be used if only
net=train(net,p,t);
one time step is to be simulated, so that the sequences
of inputs cannot be described. There are two basic
4.13 Setting and Loading Training Data
functions available for the incremental training:
learngd and learngdm. Refer to the manual for
Following code loads up the default training data into
details.
the training data list. The default training data
includes hard-coded input and output patterns for a
% learning rate
number of different homonyms word scenarios along
net.trainParam.lr = 0.01;
with their corresponding suggested output.
net.trainParam.epochs = 6000;
% tolerance
% Minimum array size has been kept 1e10
net.trainParam.goal = 0.01;
net=train(net,p,t);
min_arr_size = 1e10;
%test the network with
y_start = sim(net,p);
% Read data from .wav files and calculate
figure(200),plot(t,'g');
magnitudes from FFTs
% 15 speakers have spoken the same word
4.11 System Workflow
pronounced as "siffar"
s1_w0 = wavread('s1_w0.wav');

UbiCC Journal - Volume 3 84

s1_w0_norm = s1_w0/max(s1_w0);
s1_w0_fft0 = abs(fft2(s1_w0_norm));
min_arr_size =min(min_arr_size,
size(s1_w0_norm));
% Minimum array size has been kept 1e10
min_arr_size = 1e10;
% Read data from .wav files and calculate
magnitudes from FFTs
% 15 speakers have spoken the same word
pronounced as "siffar"
s1_w0 = wavread('s1_w0.wav');
s1_w0_norm = s1_w0/max(s1_w0);
s1_w0_fft0 = abs(fft2(s1_w0_norm));
min_arr_size= min(min_arr_size,
size(s1_w0_norm));
…………..
s15_w0 = wavread('s15_w9.wav');
s15_w0_norm = s2_w0/max(s15_w9);
s15_w0_fft0 = abs(fft2(s15_w9_norm));
min_arr_size=min(min_arr_size,
size(s15_w9_norm))
4.14 Network Training
Before using the Neural Networks in the demo
program to detect and classify correct word structure
as well as able to find out the alternate word for the
incorrectly recognized word (by the Speech
Recognition engine), we need to train the Word
Network with the training data. The train network
uses the object to train the word frequency and
network 6000 and 1500 times respectively. The
following code snippet shows how the word network
gets trained.
cutoff = 64;
for i=1:150;
for j=3:cutoff+2;
ptrans(i,j-2) = log(alldat(i,j));
end
end
p = transpose(ptrans); % inputs
t = transpose(alldat(:,2)); % target array
for i = 1:cutoff
minmax_of_p(i,1) = min(p(i,:));
end
for i = 1:cutoff
minmax_of_p(i,2) = max(p(i,:));
end
UbiCC Journal - Volume 3
In the above code, notice that the first line in the
function is the training of the word structure network,
which the Word Network uses to first detect the
problem in the network and then try to identify the
alternate word for the incorrect word in word engine.
4.15 Pattern Detecting
Once the network is trained, it can be used to detect
the correct output for the given input vector. The
following functions take the input pattern as an array
and use the function to retrieve the output from the
network.
% Creating new array 'alldat (:,:)' for NN use ...
alldat(1,1) = 1;
alldat(1,2) = 1;
alldat(1,3:258) = s1_w1_fft0(1:256);
alldat(2,1) = 1;
alldat(2,2) = 2;
alldat(2,3:258) = s1_w2_fft0(1:256);
alldat(3,1) = 1;
alldat(3,2) = 3;
alldat(3,3:258) = s1_w3_fft0(1:256);
4.16 Setting and Loading Test Data
The demo program is capable of loading the required
input for the neural network from the wave file. The
wave file that contains the input sentences to be
detected by the neural network, however should be in
a specific format so that the program can correctly
load and feed the input data to the neural network
and generate the correct output.
load 'neural_network.mat'
% The data of 15 speakers spoken in Urdu from
siffar (zero) to nau (nine).
% Minimum array size has been kept 1e10
min_arr_size = 1e10;
% Read data from .wav files and calculate
magnitudes from FFTs
% 15 speakers have spoken the same word
pronounced as "siffar"
s1_w0 = wavread('s1_w0.wav');
s1_w0_norm = s1_w0/max(s1_w0);
s1_w0_fft0 = abs(fft2(s1_w0_norm));
min_arr_size= min(min_arr_size,
size(s1_w0_norm));
85
% Creating new array 'alldat(:,:)' for NN use ...
The effect of learning rate (α) on training is shown in
alldat(1,1) = 1; Fig.25-38. In our experiments, a larger value of α
alldat(1,2) = 1; took lesser time to train, although convergence to the
alldat(1,3:258) = s1_w1_fft0(1:256); steady state value was noisy. It shows that there
alldat(2,1) = 1; exists a trade off between learning rate (α), network
alldat(2,2) = 2; parameter setting and the epoch for which the
alldat(2,3:258) = s1_w2_fft0(1:256); algorithm was used.

alldat(3,1) = 1; We observed that Fourier descriptor feature was

alldat(3,2) = 3; independent of the spoken numbers, with the
alldat(3,3:258) = s1_w3_fft0(1:256); combination of Fourier transform and correlation
technique commands used in MATLAB, a high
% min_arr_size = min_arr_size/2; accuracy recognition system can be realized.
Recorded data was used in Simulink model for
min_arr_size = 11000; introductory analysis. The feedforward neural network
was trained for different learning rates, goal and epochs.
% 15 speakers have spoken the same word It was found that there is a trade off between learning
pronounced as "siffar". rate, goal and epochs. The network was trained and
% To find min the array length so all arrays are tested as shown in Fig 21-30.
clipped to it.
2
Performance is 0.0089228, Goal is 0.01
10
s1_w0_fft(1:min_arr_size,1)=s1_w0_fft0(1:min_arr_
size, 1); 1
10
s2_w0_fft(1:min_arr_size,1)=s2_w0_fft0(1:min_arr_
size, 1);
Training-Blue Goal-Black

0
10
s3_w0_fft(1:min_arr_size,1)=s3_w0_fft0(1:min_arr_
size, 1); -1

s4_w0_fft(1:min_arr_size,1)=s4_w0_fft0(1:min_arr_ 10

size, 1);
-2
10

% testing the network

y_start = sim(net,p);
-3
10
0 1 2 3 4 5 6 7 8

figure(200),plot(t,'g'); 8 Epochs

hold on; Fig.21 Training on Class 5with 12 speakers

plot(y_start,'b'); Layers (64,35, 35,1)
legend('actual','predicted'); Learning rate 0.01
hold off Goal 0.01

For the above model we can examine structure of the 3

10
Performance is 0.00942263, Goal is 0.01

created network nnet1 and its all initial parameters.

Typing in nnet1 on the command line gives the 2
10

structure of the top level neural network object. The

Training-Blue Goal-Black

extract of the structure is as follows: 10

0
10
5. ANALYSIS & RESULTS
-1
10

When we compared the frequency content of the

same word by different speakers, we found striking
-2
10

similarities among them. This helped us to get more -3

10
confidence in our initial hypothesis that a single word 0 1 2 3 4 5
10 Epochs
6 7 8 9 10

uttered by a diverse set of speakers would exhibit

similar characteristics. Additionally, Fig.23 shows Fig.22 Training on number 6 with 12 speakers
surface graph for the correlation of frequency content Layers (64,35, 35,1)
among different speakers, for words aik (one). Learning rate 0.01

UbiCC Journal - Volume 3 86

 Goal 0.01

2
Performance is 0.00987841, Goal is 0.01
10

2
Performance is 0.00979687, Goal is 0.01
1
10 10

Training-Blue Goal-Black
1 0
10 10
Training-Blue Goal-Black

0 -1
10 10

-1 -2
10 10

-2 -3
10 10
0 500 1000 1500 2000
2459 Epochs

-3
10
0 10 20 30 40 50 60
Fig.26 Training on 15 speakers
65 Epochs
Layers (64,35, 35,1)
Fig.23 Training on number 7 with 12 speakers Learning rate 0.01
Layers (64,35, 35,1) Goal 0.01
Learning rate 0.01
Goal 0.01 2
10
Performance is 0.00994686, Goal is 0.01

2
Performance is 0.00912022, Goal is 0.01
1
10 10
Training-Blue Goal-Black

1 0
10 10
Training-Blue Goal-Black

0 -1
10 10

-1 -2
10 10

-2 -3
10 10
0 500 1000 1500 2000
2285 Epochs

-3
10
0 1 2 3 4 5 6 7 8 9 10 11
Fig.27 Training on 12 speakers
11 Epochs
Layers (64,35, 35,1)
Fig.24 Training on number 8 with 12 speakers Learning rate 0.01
Layers (64,35, 35,1) Goal 0.01
Learning rate 0.01
Goal 0.01 2
10
Performance is 0.00955099, Goal is 0.01

2
Performance is 0.00967774, Goal is 0.01
1
10 10
Training-Blue Goal-Black

1 0
10 10
Training-Blue Goal-Black

0 -1
10 10

-1 -2
10 10

-2 -3
10 10
0 500 1000 1500
1569 Epochs

-3
10
0 2 4 6 8 10 12
Fig.28 Training on 10 speakers
12 Epochs
Layers (64,35, 35,1)
Fig.25 Training on number 9 with 12 speakers Learning rate 0.01
Layers (64,35, 35,1) Goal 0.01

UbiCC Journal - Volume 3 87

 7 64, 10, 10, 10 64.44%
8 64, 15, 15, 10 69.63%
Performance is 0.0098974, Goal is 0.01
10
2
9 64, 20, 20, 10 58.52%
10 64, 25, 20, 10 71.85%
1
10
11 64, 25, 25, 10 71.85%
Training-Blue Goal-Black

10
0
12 64, 30, 30, 1 72.59%
13 64, 35, 35, 1 94%
10
-1
Table 1. Training of Neural Network with
different number of speakers
-2
10

The feedforword neural network was trained for

10
-3

0 200 400 600 800 1000 1200

number of speakers with same layers, learning rate,
1246 Epochs
goal and epochs. It was found that there is trade off
Fig.29 Training on 7 speakers between neural network, learning rate, goal and
Layers (64,35, 35,1) epochs as shown in table-2.
Learning rate 0.01
Goal 0.01 S No of Number of Accurac
No Training Testing y Rate %
speakers Speaker
10
2
Performance is 0.00996304, Goal is 0.01
1 15 5 94
2 12 4 95
10
1 3 10 3 96.7
4 7 2 95
Training-Blue Goal-Black

10
0 5
10
-1
-2
10
10
-3
0 200 400
1243 Epochs
Fig.30 Training on 5 speakers
Layers (64,35, 35,1)
Learning rate 0.01
Goal 0.01
We created and tested the networks in different
configurations, especially the hidden layer size. The
following table shows the learning accuracy with
some of the networks. A maximum accuracy of 94%
was achieved with double hidden layers network (64,
35, 35 1). Therefore double-hidden layers should
suffice for application as shown in table 1.
S No
Neuron count
1 64, 10, 10
2 64, 20, 10
3 64, 30, 10
4 64, 40, 10
5 64, 60, 10
6 64, 80, 10
5 2 100
Table 2. Training of Neural Network with
different number of speakers showing accuracy
rate.
Relationship between network layers and training
data can be formulated from the effect of learning
600
800 1000 1200
rate (α) on training is shown in Fig 21-30. In
experiments, a large value of α took lesser time to
train, although convergence to the steady state value
was noisy. It shows that there exists a trade off
between layers of the network (network parameter
setting), learning rate (α), and the epoch for which
the algorithm was used.
6. CONCLUSION
In this paper, we presented recognition analysis of Urdu
numbers siffar to nau (one to nine), which is totally an
unique idea. The data was acquired in moderate noisy
environment by word utterances of 15 different
speakers. FFT algorithm was used in MATLAB to
Learning
analyze the data. As expected, we found high
accuracy
correlation among frequency contents of the same
81.48% word, when spoken by many different speakers.
81.48%
72.59% We have investigated creation of neural network
83.70% models for automatically recognizing individual Urdu
62.96% numbers to be specific. The feedforward neural
45.93% network was trained for different learning rates;
combined and individually for different goals and

UbiCC Journal - Volume 3 88

epochs. It was found that there is a trade off between
learning rate, goal and epochs. It means all values to be
adjusted at a level of learning rate, parameter goals and
epochs. We are still in further development stage of the
system using TI TMS 320C6713. This recognition
system could be of many potential applications, for
example, voice-driven menu selection in a telephone-
based customer service in Urdu speaking countries such
as Pakistan/India.
7. REFERENCES
[1] S K Hasnain, Azam Beg and Samiullah Awan,
“Frequency Analysis of Urdu Spoken Numbers
Using MATLAB and Simulink” Journal of
Science & Technology PAF KIET ISSN 1994-
862x, Karachi, Dec. 2007.
[2] D. O’Shaughnessy. Speech Communication:
Human and Machine. Addison Wesley
Publishing Co., 1987.
[3] T. Parsons. Voice and Speech Processing.
McGraw-Hill College Div., Inc, 1986.
[4] S K Hasnain, Pervez Akhter, “Digital Signal
Processing, Theory and Worked Examples,”
2007.
[5] MATLAB User’s Guide. Mathworks Inc., 2006.
[6] DSP Blockset (For use with Simulink) User’s
Guide Mathworks Inc., 2007.
[7] Samuel D Stearns, Ruth A David, “Signal
Processing Algorithms in MATLAB,” Prentice
Hall, 1996.
[8] “TMS320C6713 DSK User’s Guide,” Texas
Instruments Inc., 2005.
[9] G. C. M. Fant. Acoustic Theory of Speech
Production. Mouton, Gravenhage, 1960.
[10] S K Hasnain, Aisha Tahir, “Digital Signal
Processing Laboratory Workbook”, 2006.
[11] S. Varho. New Linear Predictive Systems for
Digital Speech Processing. PhD dissertation,
Helsinki University of Technology, Finland,
2001.
[12] A. Beg, W. Ibrahim. An Online Tool for
Teaching Design Trade-offs in Computer
Architecture. In Proc. International Conference
UbiCC Journal - Volume 3
on Engineering Education, Coimbra, Portugal,
September 3-7, 2007.
[13] A. Beg. Predicting Processor Performance with a
Machine Learnt Model. IEEE International
Midwest Symposium on Circuits and Systems,
MWSCAS/NEWCAS 2007, Montreal, Canada,
August 5-8, 2007, pp. 1098-1101.
[14] A. Beg, P.W.C. Prasad, S.M.N.A. Senanayake.
Learning Monte Carlo Data for Circuit Path
Length. In Proc. International Conference on
Computers, Communications & Control
Technologies, CCCT 2007, Orlando, Florida,
July 12-14, 2007.
[15]A. Beg, P.W.C. Prasad, M. Arshad, S K Hasnain.
Using Recurrent Neural Networks for Circuit
Complexity Modeling", In Proc. IEEE INMIC
Conference, Islamabad, Pakistan, December 23-
24, 2006, pp. 194-197.
[16] S K Hasnain, Nighat Jamil, “Implementation
of Digital Signal Processing real time Concepts
Using Code Composer Studio 3.1, TI DSK
TMS 320C6713 and DSP Simulink Blocksets,”
IC-4 conference, Pakistan Navy Engineering
College, Karachi, Nov. 2007
[17] M M El Choubassi, H E El Khoury, C E Jabra
Alagha, J A Skaf, M A AL Alaoui, “Arabic
Speech Recognition Using Recurrent Neural
Networks,” Symp. Signal Processing & Info.
Tech., 2003, ISSPIT 2003, Dec. 2003, pp. 543-
547.
[18] M A Al-Alaoui, R Mouci, M M Mansour, R
Ferzli, “A Cloning Approach to Classifier
Training,” IEEE Trans. Systems, Man and
Cybernetics – Part A: Systems and Humans, vol.
32, no. 6, pp. 746-752, 2002.
89