Neuro-Genetic based Speech Processing for
Promoting Global Cyber Security using
Steganography technique
Tilendra Shishir Sinha, and Gautam Sanyal
Abstract—The goal of Cryptography is to provide
confidentiality and privacy by encrypting the information. If any
unauthorized human body does cryptanalysis then this goal may
be defeated. In this paper the authors have attempted an
algorithm called STEG_NGBSP (Steganography/Neuro-Genetic
based Speech Processing) using a technique called
Steganography that inherits some features of Cryptography in
such a way, that the information is encrypted as well as hidden.
The methodology adopted here is the concept of null ciphers,
which are extracted from the speech signal of a Bengali speaker
using Neuro – Genetic approach. The Speech signal is processed
and then veiled with an image. Every image is encapsulated with
some information, which may be decrypted at the receivers end
using the master image – voiced frame, in which the speech data
is covered by another image. The proposed algorithm is unique
in handling the security measures from Steganalysis, thus
promoting global cyber security and bridging the gap between
the computer and the user with more security through Natural
Language Processing.
Index Terms — Cryptography, Global Cyber Security,
Principal Component Analysis, Steganography, Steganalysis,
Wavelet.
I. INTRODUCTION
C RYPTOGRAPHY [1] is the art of writing secret
information in such a way that no one can understand
except the intended viewer. It provides privacy, integrity and
authenticity. If the methodology is known then it is very
difficult to secure the data from the hackers. These hackers
can be categorized into two groups, first type is the passive
hacker, who just tries to read the information, and the second
type is the active hacker, who not only tries to read the
message but also modifies the data and may forward it to its
destination. In cryptography, the data cannot be hidden. This
is possible by adopting a new technology called
Steganography. Steganography [2], [3] is a Greek word,
meaning “Covered Writing”. Through this technology, the
secret message is hidden in another message, in such a way
Tilendra Shishir Sinha is with the Department of Computer Science and
Engineering, as Research Scholar at NIT, Durgapur, West Bengal, India (e-
mail: tssinha2006@yahoo.co.in).
Dr. Gautam Sanyal is with the Department of MCA & CSE as HOD, NIT,
Durgapur, West Bengal, India (e-mail: gs_comp@nitdgp.ac.in).
1-4244-0370-7/06/$20.00 ©2006 IEEE
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that no one can decode the message unless the master image –
voiced frame is found. The format of this master image –
voiced frame is shown in Fig. 1.
FrmNo FrmSeq ImgparBG ImgparFG Vpara
Fig. 1. Master image – voiced file format required for the successful
Steganalysis at the Receiver’s end.
There are five fields, FrmNo, FrmSeq, ImgparBG, ImgparFG
and Vpara. The FrmNo field is used for keeping the track of
the frame number, which acts as a primary key. The field
FrmSeq, which acts as a secondary key, maintains the frame
sequence. The field ImgparBG holds the details of the
background image and ImgparFG holds the details of
foreground image. The field Vpara holds the various
parameters extracted after speech processing. The format of
this field is shown in Fig. 2.
A B C D E F G H I J K L M
Fig. 2. Processed Speech Parameters format using GSAMTSS Algorithm.
Where A : lower bound pitch; B : actual pitch; C : upper
bound pitch, D : lower bound amplitude, E : actual amplitude;
F : upper bound amplitude; G : auto-correlation coefficient; H
: power spectral density; I : eigen vector; J : eigen value; K :
adaptive vector Quantization coefficient; L : wavelet
coefficient; M : equivalent Bengali text. The general literature
survey in this area of research has been briefly expressed in
the subsequent paragraphs.
Most of the researchers organize the data in a more
compressed way by adopting various methods, but the work
done by Hongwei et al [4], describes the proper usage of
Vector Quantization using Genetic Algorithm [11], [12]. The
work has been carried out for an efficient image processing
using digital signal processing. Further work has been carried
out by Fernando Bacaa et al [5], to divide the outputs into
zones, here only five zones have been created and each zone
parameters are aggregated in the network model and the
resultant is fed as an input for the best-fit search. This work
has been further carried out by, Gautam Sanyal et al [6], for
the recognition of the speech and the speaker using
GSAMTSS algorithm. The algorithm interprets up to certain
number of words and this has been verified with tested data.
Further the technique adopted by Neil F Johnson [7], is the
null cipher, which discusses that a complete plain text is sent
and the cipher text is generated by extracting the specified
character from each of the words of plain text. From this, new
sentences are generated, forming null ciphers. The work
carried out by Yasser Ghanbari et al [8], describes the
computation of wavelet coefficient, which has been adopted
by the present authors for the betterment of the results and
comparisons with the existing algorithms. The comparison has
been attempted with respect to the performance measures like
Accuracy and Precision with the work of C. Orasan et al [15]
and Hang Li et al [16].
The paper has been organized in the following manner,
section II proposes problem formulation and solution
methodology, section III describes the results and discussions,
section IV gives the concluding remarks and further work and
finally section V incorporates all the references been made for
the completion of this work.
II. PROBLEM FORMULATION AND SOLUTION
METHODOLOGY
Promoting global cyber security is gradually becoming a
social responsibility of computer professionals. Serious
research is underway in this area, but maximum work is being
done using cryptography technique with Digital Signature [9],
[10] in which RSA algorithm is used but with unhidden
encrypted data. With the sole aim of promoting global cyber
security, a dubious system has been created, such that the
encrypted data is cleverly hidden from the hacker. This unique
system has been shown in Fig. 3.
Processed Speech
Image (BG) Image (FG)
Hiding Mechanism
Sandwiched Processed Speech Data between two images
Master Data Extraction Images are
Cropped
Mechanism
Speech and Voice Recognition at Receivers end
Fig. 3. Block diagram for Steganalysis.
The system involves utterance of Bengali words at the
sender’s end and these words are processed using the
methodology adopted by Gautam Sanyal et al [6]. This
produces a processed speech data, which is then sandwiched
between two digital images just like butter or jam is
sandwiched between two bread slices. The details of this
processed speech data is kept in a master image – voiced
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frame, which is encapsulated within the images. This master
image – voiced frame is sent to its destination as per
requirement. If this frame is not received positively then the
speech processed at the sender’s end will not be recognized at
the receiver’s end. Hence this master – voiced frame is
playing the key role in this proposed work. The solution
methodology is discussed in the subsequent subsection.
A. Proposed Solution Methodology
Adopting the proposed algorithm can summarize the solution
of the present work with some mathematical analysis. The
mathematical analysis starts with the extraction of various
speech parameters which includes wavelet coefficient, ranges
of pitch and amplitude values as a Genetic number, power
spectral density, auto-correlation coefficient value, eigen
value and eigen vector after performing principal component
analysis and Adaptive Vector Quantization coefficient. The
compression of speech data is to be done by employing
Discrete Cosine transform and Karhunen – Loeve Transform
[13], [14]. Using these parameters an Artificial Neural
Network model is created with an arrangement of word maps.
This strategy of forming a word class is called Part-of-speech
(POS). Incorporating word classes into Network model yields
good smoothing or hopefully meaningful generation of
sentences in Bengali Language (Indian Language). The
proposed algorithm has been compared with the algorithm as
devised by Hang Li et al [15], as per the size of the test data.
Further measures are computed using the concepts adopted by
Constantin Orasan et al [16]. The complexity of the proposed
algorithm is also being computed.
B. Mathematical Analysis
Based on the assumption that the original spectral is additive
with noise. To compute the approximate shape of the wavelet
(i.e., Any real valued function of time possessing some
structure), in a noisy signal and also to estimate its time of
occurrence, two methods are available, first one is a simple
structural analysis and the second one is the template
matching technique. For the detection of wavelets in noisy
signal, assume a class of wavelets, Si(t), I = 0,2,…..N-1, all
having some common structure. Based on this assumption that
noise is additive, then the corrupted signal is modeled by
following equation -:
x(n) = s(n) + G d(n)
Where s(n) is the clean speech signal, d(n) is the noise and G
is the term for signal-to-noise ratio control. To de-noise this
speech signal, wavelet transform is to be applied. Let the
mother wavelet or basic wavelet be ψ(t), which is given by,
ψ(t) = exp (j2πft – t2/2) (1)
Now as per the definition of Continuous Wavelet transform
CWT(a,τ), yields,
CWT(a, τ)=(1/ a ) ∫ x(t ) ψ{(t-τ)/a}dt
The parameters in the above equation may be discretized to
yield the Discrete Parameter Wavelet transform, DPWT(m, n),
 m Now finding the covariance of equation (8), it yields, the
, τ = n τ0 a
m
by substituting a = a . Thus it yields,
0 0 corresponding Eigen vector
DPWT (m, n) = 2-m/2 ∑ k
x(k ) ψ(2-mk – n) (2) P = cov( p n ) (9)
Where ‘m’ and ‘n’ are the integers, a0 and τ0 are the sampling and P . M i = λi . M i (10)
intervals for ‘a’ and ‘τ’, x(k) is the speech signal. Equation (2)
has been obtained by substituting a0 = 2 and τ0 = 1. This gives where λi are the corresponding Eigen values.
Now the resultant parameters are taken as weights and then
the wavelet coefficient.
formed a large class of Artificial Neural Model called
Now assume that a speech signal has been sampled at regular
Perceptron. The arrangement is done in the form of word
time interval ‘T’ to produce a sample sequence {x (nT)},
maps, with averaged weights of each word during learning
which is denoted as,
period. The matching is to be done in such a way that the
{x (nT) } = x(0), x(T), x(2T),…….,x[(N-1)T] of N sample
values, where ‘n’ is a sample number from n = 0 to n = N-1. output response must be achieved as per the input weights. If
After employing proper Discrete Fourier Transformation the result is not achieved then it should be updated or
(DFT) method, it yields to the equation of the form, searched next averaged weight from the word map.
N−1 Using the above-discussed mathematical parameters, the
X(k)=FD[x(nT)]= ∑x(nT) exp (− jk(2π / N)n)
n=0
(3) speech is processed and stored in a template, after selecting a
proper view port on the background image. This image is to
where k = 0, 1, 2, ……..,N-1 be blurred using Gaussian Low pass filtering, whose transfer
Let WN = exp (-j2π/N) function, is given by,
Thus equation (3) becomes, GLP(u, v) = exp(-(D2(u, v) / 2σ2)) (11)
N −1 Where ‘σ’ is the Standard deviation and D(u, v) is the
X(k) = FD[x(nT)] = ∑ x(nT )W
n =0
kn
N (4) distance from the point (u, v) to the center of the filter.
For De-blurring the image, Gaussian High Pass filtering has
Now to find the power spectral density, it yields, been used, whose transfer function, yields,
N −1 GHP(u, v) = 1 – GLP(u, v) (12)
1
E[x(nT)] =
N
∑ x(nT )
n =0
Mirroring of the foreground image has been done with the
help of transformation matrix, shown below,
where ‘E’ denotes expectation. The variance is, 1 0 
Var[x(nT)] = E{[x(nT) – x’(nT)]2} [x’ y’] = [x y]   (13)
The Auto-Covariance of the signal is given by, 0 − 1
Cxx(g) = E{[x(nT) – x’(nT)][x(nT) – x’(nT)]} (5) Considering two measures, Accuracy and Precision has been
Thus the Power Spectrum Density is, derived to assess the performance of the system, which may
N −1
(6) be formulated as,
PE ( f ) = ∑C xx ( m )W ( m ) exp( − j 2π fm )
m =0 Correctly Re cognized words
where Cxx(m) is the Auto-covariance function with Accuracy = (14)
‘m’ sample and W(m) is the Blackman window function with
Total number of words
‘m’ sample. TPR
Now to perform the data compression, employ Discrete Precision = (15)
Cosine Transform (DCT), which yields only Real Values, and
TPR + FPR
can be easily employed using algorithms, Hence, Where TPR = True positive recognition and FPR = False
N −1
 k 2π nT  positive recognition.
Xc(k) = Re[X(k)] = ∑ x ( nT ) cos 
n=0 N


(7)
C. Proposed Algorithm
Further data compression is to be done using Karhunen- STEG-NGBSP Algorithm {Starts}
Loeve (KL) method, which is widely used for non-linear Step 1. Employ GSAMTSS [6] algorithm for processing the
signals. Using this method, principal component values i.e.,
Speech.
Eigen Values and the corresponding Eigen Vectors, can be
Step 2. Store the parameters in a template and append in the
computed for the proper recognition of speech and the
master – voiced frame file
speaker. Thus, equation (3) yields, a pattern vector p n can be Step 3. Find the size of the template, say, TS
represented by another vector q n of lower dimension by a Step 4. Read the image and find its size, say GS.
Step 5. Select a view port of size TS, over the image, only
linear transformation. when TS < GS otherwise select another image and
Thus p n = [M ] qn (8) do step 1.
where [M ] = [X (k )] for k = 0 to N-1 from equation (3). Step 6. Employ Gaussian Low pass filtering algorithm to
blur the image
and q n = min([M]), such that q n > 0 Step 7. Superimpose one image over the other.

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 Step 8. Read the incoming image frame.
Step 9. Check for the master image – voiced frame.
Step 10. If found then do step 11 else do step 17. TABLE II
SPOKEN WORDS ARE /TAAKA/ /JOMA/ /KORE/ /DAO/
Step 11. Take the Mirror Reflection of the image
Important Lower Bound Actual Bound Upper Bound
Step 12. Clear the View port parameters
Step 13. Employ now Gaussian High pass filtering Amplitude -0.213701 0.0370962 0.172668
algorithm to de-blur the image. Pitch -32.11065 -9.458762 30.77158
Step 14. Read the template parameters again RXX 0.0 1.51249e+010 1.0
PSD 0.0 5.24789e+009 1.0
Step 15. Employ GSAMTSS [6] algorithm for the Evector 0.0 4.34587e-005 1.0
recognition of the spoken speech. Evalue 0.0 1.65892e-005 1.0
Step 16. Display the desired message in Bengali Font. PDF 0.0 4.14651e+012 1.0
Step 17. Display the error and stop. AVQC 0.0 0.0548569 1.0
WLC 0.0 0.09486995 1.0
STEG-NGBSP Algorithm {Ends}
An original image, which is acting as a background image, is
D. Computation of Complexity selected as shown in Fig. 4a, and its histogram is shown in
Let us assume that ‘n’ is the average length of a word. So for Fig. 4b. The Processed speech data is then entered in the
a single word matching the time complexity of the proposed selected view port and then it is blurred using Gaussian Low
algorithm is almost or roughly linear to the size of the pass filtering and the histogram is plotted, as shown in Fig. 4c
vocabulary, O(n log n). But when three words are present in a and Fig. 4d respectively. Again the original image, which is
particular sentence, then the time complexity of the proposed now acting as a foreground image, is then superimposed over
algorithm is O(n2). In a worst case, the time complexity for the blurred image, as shown in Fig. 4e and its histogram is
the proposed algorithm expected to be F * O(n log n), where shown in Fig. 4f., and next the mirroring has been done
‘F’ is the size of the complete phrase or sentence. during recognition phase at the receiver’s end after employing
Gaussian High pass filtering for de-blurring the image
III. RESULTS AND DISCUSSIONS received.
The Algorithm has been tested over a standalone machine,
and the result is remarkable. The threshold value of
recognizing the speech is kept as 5% tolerance and this is
verified from the trained data stored in the master image –
voiced frame file. Some of the vocabularies extracted are
shown in Table I and Table II.
TABLE I
SPOKEN WORDS ARE /TUMI/ /KE/
Important Lower Bound Actual Bound Upper Bound
parameters
Amplitude -0.063701 0.0375432 0.101245
Pitch -41.6458 -8.67741 30.2124 Fig. 4a. Original Image used as Background.
RXX 0.0 1.08268e+010 1.0
PSD 0.0 3.7574e+009 1.0
Evector 0.0 4.33129e-006 1.0
Evalue 0.0 7.14436e-007 1.0
PDF 0.0 2.54233e+012 1.0
AVQC 0.0 0.0470479 1.0
WLC 0.0 0.0987955 1.0
Where the symbols are illustrated as -:
RXX Auto-correlation coefficient;
PSD Power Spectral density
Evector Eigen Vector
Evalue Eigen Value Fig. 4b. Histogram of Fig. 4a.
PDF Probability Density Function
AVQC Adaptive Vector Quantization coefficient
WLC Wavelet Coefficient

Fig. 4c. After performing Gaussian Low pass filtering.

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 TABLE IV
PERFORMANCE MEASURES (PRECISION)
Size of Test Li’s Precision Our Method’s Precision
Data (in KB)
1 0.802 0.825
2 0.799 0.815
3 0.793 0.796
4 0.786 0.786
5 0.784 0.789
8 0.791 0.799
Fig. 4d. Histogram of Fig. 4c.

Performance Measures (Precision)

0.84

Precision Values
0.82 Li's precision

0.8 Our Method's

0.78 Precision
Fig. 4e. Another image similar to background image is superimposed by
0.76
another image as foreground image like sandwich of images.
1 2 3 4 5 8
Size of the Test data (KB)

Fig. 5b. Graphical Representation of Precision Measures.

Then at the receiver’s end, the speech will be processed for

the recognition and the result will be obtained with the actual
speech spoken at the sender’s end in text form, here in
Bengali font, as shown in Fig. 6.
Fig. 4f. Histogram of Fig. 4e.

The performance measures for Accuracy and Precision are

compared with the method adopted by Constantine Orasan et
al [15] and Hang Li et al [16], with the data shown in Table
III and Table IV and the corresponding graphical
representations are shown in Fig. 5a. and Fig. 5b. respectively.

TABLE III
PERFORMANCE MEASURES (ACCURACY)
Size of Test Li’s Accuracy Our Method’s Accuracy
Data (in KB)
1 0.88 0.98
2 0.91 0.95
3 0.95 0.97
4 0.93 0.95 Fig. 6. The desired output waveforms and the Bengali Font /tumi/ /ke/
5 0.91 0.95
8 0.95 0.96 Here the Bengali words are used and the equivalent English
meanings are given below -:
Performance Measures (Accuracy)
In Bengal, /tumi/ /ke/ ?
In English, “Who are you?”
Accuracy Values

1 Li's Accuracy In Bengali, /taaka/ /joma/ /kore/ /dao/

0.95 In English, “Deposit the Amount.”
0.9
Our Method's In Bengali, /aami/ /khabar/ /khabo/
0.85
Accuracy In English, “I will take food.”
0.8
1 2 3 4 5 8
Size of the Test data (KB)

Fig. 5a. Graphical Representation of Accuracy Measures.

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 IV. CONCLUDING REMARKS AND FURTHER WORK [16] Hang Li and Naoki Abe, ’Word Clustering and Disambiguation Based
on Co-occurrence data,’ Proceedings of the 18th International
An attempt has been made successfully to verify the Conference on Computational Linguistics and the 36th Annual Meeting
technology called Steganography, using Bengali words and of Association for Computational Linguistics, 749-755, 1998.
phrases with the Bengali font displayed in the result, as shown
in Fig. 6. The result has been tested in a standalone machine. VI. BIOGRAPHIES
However, this approach is limited to Bengali words
commonly used in daily conversation. The work may be Tilendra Shishir Sinha was born in July 1968 in
Dhanbad district, Jharkhand, India. He is graduated
further carried out for more complex Bengali words and from Nagpur University, Nagpur, Maharashtra
phrases for more secured data transmission promoting global State, India, in Computer Technology Engineering
cyber security. in the year 1992 and later did his post graduation
The Accuracy and the Precision measures for the performance from National Institute of Technology,
Jamshedpur, Jharkhand, India. Now he is pursuing
of the system is computed and compared, and the result of the his Ph.D in Computer Science and Engineering
proposed algorithm is satisfactory. from NIT, Durgapur, West Bengal, India.

V. REFERENCES
Gautam Sanyal was born in August 1954 in
Periodicals: Burdwan district, West Bengal, India. He has
[1] David Clark, “Encryption Advances to Meet Internet Challenges,” IEEE received his B.E and M.Tech degree from REC,
Transactions on Computer Communication, pp. 20-24, August, 2004. Durgapur, now, National Institute of Technology,
[2] 2nd Lt. James Caldwell, “U. S. Air Force, Steganography, Software Durgapur. He has received his Ph.D (Engg.) from
Technology Support Center,” June 2003. [Online] Available: Jadavpur University, Kolkata, West Bengal, India,
http://www.stsc.hill.af.mil/crostalk/2003/06/caldwell.html, 12th in the area of Robot Vision. He is presently
November, 2005. working as an HOD (CSE & MCA) at NIT,
[3] Clair, Bryan, “Steganography: How to Send a Secret Message”, 8th Durgapur. His area of interest includes Soft
October, 2001. [Online] Available : Computing, Networking and NLP.
http://www.strangehorizons.com/2001/20011008/steganography.shtml,
12th November, 2005.
[4] Hongwei Sun, Kwok-Yan Lam, Siu-Leung Chung, Weiming Dong,
Ming Gu, Jiaguang Sun,’ Efficient Vector Quantization using Genetic
Algorithm, ‘ Neural – Computing and Applications (2005) 14 : 203 –
211, DOI 10.1007/s00521-004-0413-4, Springer – Verlag London
Limited.
[5] Fernando Bacao, Victor Lobo, Marco Painbo, ‘Applying Genetic
Algorithms to Zone design, ‘ Soft Computing(2005) 9 :341 – 348, DOI
10.1007/s00500-004-0413-4, Springer – Verlag London Limited.
[6] Tilendra Shishir Sinha, Gautam Sayal and Abhijit Mukherji,’Some
Aspects of Modelling and Simulation for the Recognition of Speech and
Speaker of Bengali Language using proposed GSAMTSS Algorithm,’
International Journal of Systemics, Cybernetics and Informatics, April,
2006, pp. no. 69 – 75.
[7] Neil F. Johnson, ‘History and Steganography, ‘ Johnson and Johnson
Consultants, LLC, 1995 – 2003, [Online] Available :
http://www.jjtc.com/stegdoc/sec202.html, 12th November, 2005.
[8] Yasser Ghanbari and Mohammad Reza Karami,’Spectral Subtraction in
the Wavelet Domain for Speech Enhancement’,International Journal of
Softwares and Information Technologies, Vol 1, No.1, August, 2004, pp.
no. 26 – 29.
[9] Yu Lei, Deren Chen, Zhongding Jiang, ’Peer Generating Digital
Signatures on Mobile Devices,’ IEEE Transactions on Mobile
Computing, pp. 532, March, 2004.
[10] Scott Cambell,’Supporting Digital Signatures in Mobile Environments,’
IEEE Transactions on Mobile Computing, pp. 238, June, 2003.

Books:
[11] David E. Goldberg, ’Genetic Algorithms in search Optimization &
Machine Learning,’ Pearson Education Asia, ISBN 81-7808-130-X, pp.
185-213.
[12] Douglas O’ Shaughnessy, ’Speech Communication Human and
Machine,’ Universities Press, Second Edition, ISBN 81-7371-374-X.
[13] J.-S. R. Jang, C.-T. Sun, E. Mizutani, ‘Neuro-Fuzzy and Soft
Computing, - A Computational approach to Learning and Machine
Intelligence,’ First Edition, PHI, ISBN 81-297-0324-6.
[14] Mohammad H. Hassoun, ‘Fundamentals of Artificial Neural Networks,’
Eastern Economy Edition, PHI, ISBN 81-203-1356-9.

Papers from Conference Proceedings (Published):

[15] Constantine Orasan and Richard Evans, 2001, ‘Learning to identify
animate references,’ In Proceedings of Learning (CoNLL-2001), pages
129 – 136, Toulouse, France, July 6 –7, 2001.

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