(SPEECH ENCRYPTION AND DECRYPTION)

.
ABSTRACT
Digital signal is a numerical representation of the analog signal. It may be easier and
more cost effective to process these signals in the digital world. One of the main
advantages of digital signal is that it can be stored and manipulated easily. To do this
several processes are in use. Of these, Digital Signal Processor plays a leading
role.This is used in various applications like instrumentation measurement,
communications , audio video processing ,graphics , image enhancement ,
navigation , radars ,control!robotics, machine vision ,guidance etc.
The ob"ective is to encrypt the speech signal. That is to convert the speech signal into
an unknown form and then the scrambled speech signal is transmitted. The person
who know, how the speech signal is encrypted can decrypt #i.e.,converting the
scrambled speech signal into original signal$ the signal. Thus security of sending
speech message is done.
%or encryption and decryption of the signal, we develop a software program in
assembly language programming techni&ues of 'DSP ()*). In this paper, encryption
and decryption of the speech signal is done using the Digital Signal Processor 'DSP
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In recent trends, cell phone, voice chat, etc., is the main topic of the communication.
9ut it is not possible to speak secretly through them. Our ob"ective is to speak
secretly, i.e., to maintain security of certain important information. So for that we
are going for a technology using digital signal processor.
Our paper also deals with storing and retrieval of speech signal using digital signal
processor.
:e have used 'DSP processor ()*). This is )7 bit fi;ed point processor.
<ere the voice signals is ac&uired and then digitally coded and stored in the memory of
processor. /sing %%T #%ast %ourier Transform$ and I%%T #Inverse %ast %ourier
Transform$ techni&ues, speech signal is scrambled. Then it is decoded to obtain the
same speech signal and now the original speech is confused. Then it is transmitted
using transmitter. The coding and decoding is done with +OD-+.
In the receiver side, if the receiver knows how the speech signal is scrambled, then the
receiver would descramble the scrambled signal and get the original speech signal. The
programming is done on the assembly language of the 'DSP ()*) processor
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In general signal carries information, and the ob"ective of signal processing is to
e;tract this information.
Signal processing is the method of e;tracting information from the signal. Thus signal
processing is concerned with representing in mathematical terms and e;tracting the
information by carrying out algorithmic operations on the signal.
'nalog signal are real world signals that you and I e;perience everyday! sound, light,
temperature, pressure. ' digital signal is a numerical representation of the analog
signal. :e can convert the analog signal into digital signal through our analog to
digital conversion process, process the signals, and if needed, bring the signal back out
to analog world through the digital to analog converter.
The basic e;planation of DSP is the processing of signal by digital systems, to improve
the signal. The improvement may be clearer sound, sharper images, or faster data. The
digital systems are software and hardware. In digital processing of signal consists of a
number of mathematical operations as specified by a software program.
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' digital signal processor is a type of microprocessor that is incredibly fast and
powerful. They are specifically designed to handle digital processing tasks. 's a broad
generali@ation, traditional microprocessors, such as the Pentium, primarily directed at
data manipulation. Similarly, DSPs are designed to perform the mathematical
calculations needed in digital signal processing.
' DSP is uni&ue because it processes data in real time. To start we need to distinguish
between offline and real time processing.
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In this processing, the entire input signal resides in the computer at the time of
manipulation, for e;ample, in medical imaging such as tomography, the data is
ac&uired while the patient is inside the machine, but the image reconstruction may be
delayed until a later time. The key point is all the information is simultaneously
available to the processing program.
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Signal is produced at the same time that the input signal is being ac&uired. This real
time capability makes the DSP perfect for applications that cannot tolerate any delays.
%or e;ample, did you ever talk on a cell phone where two people couldnAt talk at onceB
6ou had to wait until the other person finished talking. If you both spoke
simultaneously, the signal will be cut. 6ou couldnAt hear the other person. :ith todayAs
digital cell phones, which use DSP, you can talk normally. The DSP processors inside
cell phones process sounds so rapidly you hear them as &uickly as you can speak in
real time.
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To calculate the output sample, we must have access to a certain number of the most
recent samples from the input. 1et aC,a),D,a8 be the value of the * most recent
samples from the input signal. These * samples must be stored in memory and
;EnF,;En!)F,D,;En!8F continually updated as new samples are ac&uired.
The best way to manage the stored samples is circular buffering.


+ircular buffers are used to store the most
recent values of a continually updated
signal. The fig shows how an eight sample
circular buffer might appear at some instant
of time.
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%igure illustrates an eight sample circular buffer. +onsider that we have placed this
circular buffer in eight consecutive memory locations, (CC3) to (CC3*. %ig shows how
the eight samples from the input might be stored at one particular instant of time.
The idea of circular buffering is that the end of this linear array is connected to its
beginningH memory location (CC3) is viewed as being ne;t to (CC3*, "ust as (CC33 is
ne;t to (CC35. 6ou keep track of the array by a pointer #a variable whose value is an
address$ that indicates where the most recent sample resides. %or instance in the fig the
pointer contains the address (CC33, while it contains (CC35 after the arrival of ne;t
sample. :hen a new sample is ac&uired, it replaces the oldest sample in the array, and
the pointer is moved one address ahead.
+ircular buffers are efficient because only one value needs to be changed when a new
sample is ac&uired. %our parameters are needed to manage the circular buffer.
There must be a pointer that indicates the start of the circular buffer
in memory #in this e;ample, (CC3)$.
There must be a pointer indicating the end of the array #e.g. (CC3*$
or a variable that holds its length #e.g. *$.
The step si@e of the memory addressing must be one. -;ample=
address (CC32 contains one sampleH address (CC33 contains the ne;t sample and so on.
The above three values define the si@e and configuration of the circular buffer, and will
not change during the program operation.
The pointer to the most recent sample must be modified as each new
sample is ac&uired.
In other words, there must be program logic that controls how this fourth value is
updated based on the value of the first three values. :hile this logic is &uite simple, it
must be very fast. This is the whole point of this discussionH DSPs must be optimi@ed at
managing circular buffers to achieve the highest possible e;ecution speed.
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Speech signal
scrambled signal

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Scrambled signal Speech
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The Discrete %ourier Transform #D%T$ of a discrete time signal ;#n$ is a finite
duration discrete fre&uency se&uence. D%T is obtained by sampling one period of the
%ourier Transform G #$ of the signal ; #n$ at a finite number of fre&uency points.
The %T of a discrete time signal is a continuous function of and so canAt be
processed by digital system. <ence we go for D%T, which converts continuous function
of into a discrete one, allowing us to perform fre&uency analysis on a digital
computer.
D%T is important here because
It allows us to determine fre&uency content of a signal, to
perform spectral analysis.
To perform filtering operations in fre&uency domain.
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It is an algorithm for computing D%T with reduced number of calculations. The
computational efficiency is achieved by a divide and con&uer approach, which is based
on the decomposition of ,!point D%T into successively smaller D%TAs. In an ,!point
se&uence if ,Ir
m
, then the se&uence can be decimated into r!point se&uences. <ere JrA
is called the radi;. If rI( it is called radi;!( %%T. The decimation can be done in two
ways namely Decimation In Time #DIT$ and Decimation In %re&uency #DI%$. <ere we
use .adi;!( DIT %%T algorithm.
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The ID%T of the se&uence is computed using the I%%T. The %%T serves the
evaluation of both direct and inverse D%TAs.
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Scrambling means make confusion. 'fter doing %%T in assembly language program in
'DSP!()*), the signal is converted into spikes and it is in fre&uency domain. The
signal spikes are stored in the memory. /sing the circular buffers, the spikes positions
are interchanged. ,ow the original signal is scrambled. This is called scrambling.
'gain if the spikes are placed into its original position, then we get original signal.
This is called descrambling and is also done with the use of circular buffer.
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)$ %irst the speech signal #analog signal$ is converted into digital signal by passing
through the +OD-+. ,ow the signal is in time domain and it is sent to the 'DSP ()*)
processor.
($ 9y running )(*!point %%T algorithm using computer #system is connected to the
processor$ on 'DSP ()*), the signal which is in time domain is converted into
fre&uency domain. ,ow the signal is sampled into )(* samples and it is stored in the
processor memory.
2$ 9y doing assembly language program on 'DSP ()*), first 73 samples are displaced
into ne;t 73 samples position and vice versa. This is called scrambling.
3$ ,ow the original signal is scrambled and by doing I%%T algorithm on 'DSP ()*),
the scrambled signal, which is in fre&uency domain, is converted into time domain.
5$9y +OD-+ scrambled signal which is a digital signal is converted into analog signal
and then it is transmitted.
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.eceived signal #scrambled signal$ is converted into digital signal by +OD-+ and it is
sent to the processor.
9y )(*!point %%T algorithm, the signal in time domain is converted into fre&uency
domain as done in the encryption part.
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($ 9y doing assembly language program on 'DSP ()*), the displaced 73
samples are placed to its original position. This is called descrambling.
2$ ,ow the scrambled signal is descrambled and we get the original signal.
3$ 9y doing I%%T, the original signal, which is in fre&uency domain, is
converted into time domain and by +OD-+ we get original analog signal #speech
signal$.
The assembly language program was e;ecuted for the speech signal and
the output was checked. -ncryption and Decryption is highly reali@ed.
:e have stored the speech signal in the memory in encoded form. :e
decode the same to get the analog output. %or coding and decoding, +OD-+ is used.
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:e can use this techni&ue in military for sending secret messages. :e can
use this techni&ue wherever we want security. :hat we have done is storing the speech
signal for a period of few milli seconds. This can be increased if we e;tend the memory
si@e and select that through a decoding switch.
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San"it >. ?itra, KDigital signal processingL
'lan 4.Oppenheiim and 'llan :illsky, KSignals and SystemsL
P..amesbabu , KDigital signal processingL
K'DSP!()*G DSP Instruction Set .eferenceL
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