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<article xmlns:xlink="http://www.w3.org/1999/xlink">
  <front>
    <journal-meta />
    <article-meta>
      <title-group>
        <article-title>Method of speech signal scrambling based on matched wavelet filters ⋆</article-title>
      </title-group>
      <contrib-group>
        <contrib contrib-type="author">
          <string-name>Oleksandr Lavrynenko</string-name>
          <email>oleksandrlavrynenko@gmail.com</email>
          <xref ref-type="aff" rid="aff0">0</xref>
          <xref ref-type="aff" rid="aff1">1</xref>
        </contrib>
        <aff id="aff0">
          <label>0</label>
          <institution>CPITS-II 2024: Workshop on Cybersecurity Providing in Information and Telecommunication Systems II</institution>
        </aff>
        <aff id="aff1">
          <label>1</label>
          <institution>National Aviation University</institution>
          ,
          <addr-line>1 Lubomyr Huzar ave., 03058 Kyiv</addr-line>
          ,
          <country country="UA">Ukraine</country>
        </aff>
      </contrib-group>
      <fpage>229</fpage>
      <lpage>235</lpage>
      <abstract>
        <p>In this research study, a method of speech information protection using digital wavelet filter banks is proposed. An inverse scheme of single-level discrete wavelet transform is used to build the protection system. It includes digital synthesis and analysis filter banks. The filters used are synthesized using a key sequence. The key identifies the sender and receiver of the information and is used only in the filter synthesis stage. Also presented is a method for synthesizing matched wavelet filters satisfying the property of orthogonality of the wavelet basis, the presence of zero moments. The important requirements of the filters are the conditions of complete signal recovery and elimination of overlapping spectra. The results of the research show the effectiveness of synthesized wavelet filters in solving the problem of information protection. A speech protection algorithm is developed, which uses matched wavelet filters at the stage of building a bank of analysis-synthesis filters matched to the key. The algorithm has a simple implementation, and fast algorithms of digital signal processing (convolution, decimation, interpolation), allowing encrypting of the signal in real-time. The proposed algorithm is noise-resistant and can be used in channels with intensive interference. The algorithm is robust to time delays and hiccups, as well as distortions in the communication channel.</p>
      </abstract>
      <kwd-group>
        <kwd>eol&gt;speech signal</kwd>
        <kwd>wavelet transform</kwd>
        <kwd>packet wavelet transform</kwd>
        <kwd>matched wavelet filter</kwd>
        <kwd>speech scrambling</kwd>
        <kwd>speech information protection</kwd>
        <kwd>speech intelligibility</kwd>
        <kwd>masking noise 1</kwd>
      </kwd-group>
    </article-meta>
  </front>
  <body>
    <sec id="sec-1">
      <title>1. Introduction</title>
      <p>
        Information protection is an integral part of communication
systems. Nowadays, more and more attention is paid to the
protection of speech information, which is associated with the
growth of speech communication in the modern information
environment [
        <xref ref-type="bibr" rid="ref1">1</xref>
        ].
      </p>
      <p>
        With the development of digital communication in radio
engineering, gaming methods, and cryptographic
algorithms have become widespread. Initially, the analog
speech signal is converted into digital form [
        <xref ref-type="bibr" rid="ref2">2</xref>
        ]. An
encryption algorithm is applied to the coefficients or signal
parameters obtained after encoding [
        <xref ref-type="bibr" rid="ref3">3</xref>
        ]. Such systems have
a high level of protection and require computational
resources. Under interference conditions such algorithms
do not work efficiently. A wide range of tasks requires
algorithms that are applicable in the presence of sufficiently
strong interference [
        <xref ref-type="bibr" rid="ref4">4</xref>
        ].
      </p>
      <p>
        Along with mathematical methods of speech
information protection, methods using digital signal
processing (DSP) algorithms are widely demanded.
Scrambling algorithms using fast linear orthogonal
transforms (fast Fourier transform, fast wavelet transform)
and discrete filter banks are of considerable interest [
        <xref ref-type="bibr" rid="ref5">5</xref>
        ]. As
a rule, they are based on manipulations with spectral
coefficients of linear transformation of signals. Such
algorithms, when scrambling, cause a relatively small
change in the signal bandwidth and very low residual
intelligibility of the signal in the communication channel
[
        <xref ref-type="bibr" rid="ref6">6</xref>
        ]. When using fast transformations increases the degree
of information closure, but also increases the computational
complexity of the processing algorithm, there is a delay in
the signal [
        <xref ref-type="bibr" rid="ref7 ref8">7, 8</xref>
        ]. Orthogonal scramblers are not deprived of
the common disadvantages of scramblers and introduce
distortions in the recovered speech signal determined by the
dispersion in the channel and synchronization error [
        <xref ref-type="bibr" rid="ref9">9</xref>
        ].
Thus, the problem of developing new fast algorithms for the
protection of speech information operating under noise
conditions is urgent.
      </p>
    </sec>
    <sec id="sec-2">
      <title>2. Literature review and problem statement</title>
      <p>
        Ukraine adheres to the following classification by the level
of complexity of devices: maskers (simple), dynamic
scramblers (medium complexity), and encryptors (high
complexity) [
        <xref ref-type="bibr" rid="ref10">10</xref>
        ]. The maskers providing the tactical level of
information protection include spectrum inverters, and
static scramblers [
        <xref ref-type="bibr" rid="ref11">11</xref>
        ]. The proposed speech masking
algorithm also belongs to this class.
      </p>
      <p>
        Scramblers using filter banks are widely used among
tactical closure systems [
        <xref ref-type="bibr" rid="ref12">12</xref>
        ]. In general, the traditional
scheme contains M-channel analysis-synthesis filter banks,
and forward and backward permutation blocks (Fig. 1) [
        <xref ref-type="bibr" rid="ref13">13</xref>
        ].
0000-0002-7738-161X (O. Lavrynenko)
© 2024 Copyright for this paper by its authors. Use permitted under
Creative Commons License Attribution 4.0 International (CC BY 4.0).
Mixing of signal segments according to a certain
unlike other masking methods, the proposed method of
permutation rule takes place in the block  . Reverse
permutation occurs at the input of the decoder in the block
      </p>
      <p>
        . The permutation rule is the key to the system. They
have very low residual intelligibility of the scrambled signal
in the communication channel, but introduce time delay and
distortion in the reconstructed signal [
        <xref ref-type="bibr" rid="ref14">14</xref>
        ]. For its class,
speech signal scrambling based on matched wavelet filters
has a high degree of information closure, high quality of
reconstructed speech, and a sufficiently large number of
keys.
x(n)
      </p>
      <p>H0(z)
H1(z)</p>
      <p>HM-1(z)
.
.
.</p>
      <p>.
.
.</p>
      <p>↓M
↓M
↓M</p>
      <sec id="sec-2-1">
        <title>Encoder</title>
        <p>rule, requires radical technical solutions affecting the design
of devices. When modernizing radios, the reliability of the
protection system is important its simple design, and the
insignificance of material costs.</p>
        <p>P(z)</p>
        <p>P(z)-1
↑M
↑M
↑M</p>
      </sec>
      <sec id="sec-2-2">
        <title>Decoder</title>
      </sec>
    </sec>
    <sec id="sec-3">
      <title>3. Proposed method</title>
      <p>
        In this section, a speech information protection algorithm
using digital wavelet filter banks is proposed [
        <xref ref-type="bibr" rid="ref16">16</xref>
        ]. To build
the protection system, an inverse scheme of single-level
discrete wavelet transform is used. It includes digital
synthesis and analysis filter banks (Fig. 2). The filters used
are synthesized using a key sequence. The key identifies the
sender and receiver of information and is used only at the
filter synthesis stage [
        <xref ref-type="bibr" rid="ref17">17</xref>
        ].
      </p>
      <p>H'
G'
↓2
↓2
3
4</p>
      <sec id="sec-3-1">
        <title>Reconstructed noise</title>
      </sec>
      <sec id="sec-3-2">
        <title>Reconstructed signal</title>
      </sec>
      <sec id="sec-3-3">
        <title>Masking</title>
        <p>noise 1</p>
      </sec>
      <sec id="sec-3-4">
        <title>Useful</title>
        <p>signal 2
↑2
↑2
H
G</p>
      </sec>
      <sec id="sec-3-5">
        <title>Communication channel</title>
      </sec>
      <sec id="sec-3-6">
        <title>Synthesis Bank</title>
      </sec>
      <sec id="sec-3-7">
        <title>Analysis Bank</title>
        <p>filters. They must satisfy the conditions imposed on wavelet
filters: orthogonality of the wavelet basis, and presence of
zero moments.</p>
        <p>
          The zero moments of the frequency response of the
approximating filter  can be introduced a priori to solve
the synthesis problem. If the filter has zero moments, the
expression for  ( ) can be written in the form:  ( ) =
1 +   ( ), where  ( ) is some function [
          <xref ref-type="bibr" rid="ref23">23</xref>
          ].
        </p>
        <p>Let us present the developed method of synthesizing the
MWF. The theory of MWF is developed from the following
problem. It is required to construct for  ( ) a set of
orthogonal quadrature-mirror wavelet filters in such a way
that at its wavelet decomposition, the output of the detailing
filter is zero, i.e. all detailing coefficients of the wavelet
domain should be equal to zero (Fig. 3).</p>
        <sec id="sec-3-7-1">
          <title>HMWF</title>
        </sec>
        <sec id="sec-3-7-2">
          <title>GMWF</title>
          <p>
            ↓2
↓2
a(n)
{0}
The procedure of wavelet transform of the signal  ( ) in
the frequency domain can be written in the following form:
 ( ) ( ) +  ( +  ) ( +  ) =  ( ) (1)
 ( ) ( ) +  ( +  ) ( +  ) =  ( ),
where  ( ),  ( ),  ( ) are the Fourier images of the
sequence  ( ), interpolated approximating and detailing
wavelet transform coefficients, respectively, and  ( ) and
 ( ) are the frequency response (FR) of the decomposition
filters [
            <xref ref-type="bibr" rid="ref24">24</xref>
            ].
          </p>
          <p>In order to prevent elision, we can assume that the
relationship between the  and  filters is set to be fair for
the QWF:
 ( ) = 
⋅  ∗( +  ).</p>
          <p>(2)</p>
          <p>This will immediately satisfy one of the constraints placed
on filters:</p>
          <p>( +  ) ⋅  ( ) +  ( +  ) ⋅  ( ) = 0,
where  ( ),  ( ) are the FRs of the corresponding
recovery filters, with  ( ) =  ∗( ) and  ( ) =  ∗( ).</p>
          <p>Another important property, the orthogonality
property of the wavelet basis for filters in the frequency
domain is written in the form:</p>
          <p>| ( )| + | ( +  )| = 2. (3)</p>
          <p>Solving the system (1) under the assumption
that ( ) = 0, and using the relation (2), we get
 ( ) = | ( )(| )⋅| ∗(( ) )| .</p>
          <p>The filter  ( ) is almost built, it remains to find the
condition on  ( ). This can be done using (3), given that
 ( ) =  ( +  ). As a result, we obtain</p>
          <p>| ( )| = 2 ⋅ (| ( )| + | ( +  )| ).</p>
          <p>Let  ( ) be a real analytic function, then</p>
          <p>( ) = √2 ⋅ | ( )| + | ( +  )| .</p>
          <p>The final result is represented as:</p>
          <p>( ) = | ( √)| ⋅ |∗( ( ) )| . (4)</p>
          <p>
            As a result of solving the problem, digital wavelet filters
matched to the input sequence have been found. Such filters
are called matched wavelet filters [
            <xref ref-type="bibr" rid="ref25">25</xref>
            ] since their impulse
response is formed taking into account the properties of the
processed signal. In our case, it is a key sequence. Thus, the
information about the key is embedded in the filters
themselves.
          </p>
          <p>Thus, we can present a speech protection algorithm that
relies on the dual use of masking noise. The difference
between the proposed system and its first variant is the
addition of masking noise to the signal at the input (before
the transmultiplexer) and the inverse transformation at the
output. For this purpose, in addition to the transmultiplexer,
including expanders and compressors of the sampling
frequency, analysis, and synthesis filters, adders are
introduced into the system. Fig. 4 shows a detailed block
diagram of the second variant of the scheme.</p>
          <p>H'
G'
↓2
↓2</p>
          <p>Reconstructed</p>
          <p>noise
3
4
‒</p>
          <p>
            Reconstructed
signal
A generalized speech information protection scheme can
also be proposed to close the conversations of several users.
It is known [
            <xref ref-type="bibr" rid="ref27">27</xref>
            ] that wavelet decomposition over both
subbands yields a complete balanced tree (Fig. 5). If the
initial block of wavelet filters is orthogonal, then the scheme
corresponding to any level of the full tree decomposition is
orthogonal. Such a scheme as a whole, as well as its separate
block, has the property of accurate signal recovery. An
inverse scheme consisting of separate blocks can also be
constructed for the full wavelet tree [
            <xref ref-type="bibr" rid="ref28">28</xref>
            ].
          </p>
          <p>Masking
noise 1
Useful
signal 2
+
↑2
↑2</p>
          <p>H
G</p>
          <p>Communication</p>
          <p>
            channel
Synthesis Bank
Analysis Bank
The masking noise [
            <xref ref-type="bibr" rid="ref26">26</xref>
            ] is fed to input 1, then to the
sampling rate expander and filter-interpolator  . At the
same time, it is also fed to the adder. The useful signal is fed
to input 2, mixed in the adder with the higher power
masking noise. The mixture then goes to the sampling rate
expander and filter interpolator  . The signal images
transformed by the filters are mixed to form a noise-like
mixture. The reconstruction of the signal in the analysis
bank is done in reverse order. The system has the property
of accurate recovery. The key point is the uniqueness of the
analysis and synthesis filter banks. Wavelet filters are
synthesized in the previously described way.
          </p>
          <p>H'</p>
          <p>G'</p>
        </sec>
      </sec>
    </sec>
    <sec id="sec-4">
      <title>4. Results and discussion</title>
      <p>
        The research on the system of speech information
protection using MWF was carried out on speech signals.
The system was analyzed for non-recursive (FIR filters) and
recursive (IIR filters) systems. The case of operation of the
protection algorithm in the conditions of application of the
ITU-T G.711 standard for coding signals in the channel with
8, 16, and 32 bits is considered [
        <xref ref-type="bibr" rid="ref31">31</xref>
        ]. The operating parameter
of the system is the masking noise power. To analyze the
influence of masking noise, the parameter  is the ratio of
signal and masking noise power in dB is introduced. To
study the noise immunity of the system, the parameter  is
the ratio of signal and external noise power in the
communication channel in dB was introduced:

= 10 ⋅ 
;  = 10 ⋅ 
.
      </p>
      <p>Estimates of speech parameters used in the paper are
discussed.</p>
      <p>PESQ (Perceptual Evaluation of Speech Quality) is used
to automatically evaluate the quality of speech transmitted
in
telecommunication
environments.</p>
      <p>To
obtain
the
evaluation, the source signal and the signal at the system
output are compared. The evaluation is graded on the MOS
scale (mean opinion score, ITU-T recommendation P.800),
which covers the range from 1 (poor) to 5 (excellent). The
acceptable quality of the reconstructed signal corresponds
to a PESQ score greater than 2.5 points.</p>
      <p>Expert evaluation
of speech
intelligibility 
is
introduced to determine speech intelligibility in the channel
and at the system output. Based on the results of listening,
the experts evaluate the intelligibility of the signal. The
traditional 5-point scale was used, where the best sound
quality corresponds to the highest score. At one point of the
scale, the
useful signal is completely
unintelligible.</p>
      <p>Acceptable quality of the recovered signal corresponds to
the assessment</p>
      <p>
        &gt; 3 [
        <xref ref-type="bibr" rid="ref32">32</xref>
        ]. The  score agrees well with the
PESQ speech quality score.
      </p>
      <p>It is considered what requirements the protection
system should meet.</p>
      <p>The main purpose of the system is reliable closure of
speech information with the possibility of its full recovery.
Based on the purpose, for the proposed speech protection
system the recovered signal should have good intelligibility.
The signal in the channel on the contrary should be
completely unintelligible. Such conditions are fulfilled for a
certain interval of values of the parameter  . The lower
limit of the parameter  is determined from the conditions
when distortions of the reconstructed signal become
unacceptable for perception. It is estimated by the PESQ
criterion and the expert evaluation of speech intelligibility
 . The upper limit is determined from the conditions when
the useful signal in the channel becomes completely
unintelligible and is based on the evaluation of  . The
masking noise power satisfying these conditions is selected
from the interval formed by the intersection of the shaded
regions in Figs. 7 and 8. The value of the PESQ score, which
is chosen to be greater than 2.5, is also considered.
Thus, the selection interval for the parameter  is:
(a) −34 &lt; , дБ &lt; −15 for FIR filters.
(b) −20 &lt; , дБ &lt; −15 for IIR filters.</p>
      <p>The noise immunity of the system is considered as the
performance of the system under information distortion in
the presence of noise. Generalized, as an external noise is
used AWGN, modeling data distortion. Based on the values
of  &gt; 2.5 points (Fig. 9), it follows that acceptable
quality of the transmitted signal is achievable at  &gt; 25 dB.</p>
      <p>It is established that the algorithm is robust to external
noise. The dependence of the signal-to-noise ratio (SNR) at
the system output on the SNR in the channel is linear.</p>
      <p>The resilience of the system, i.e., the extent to which it
is secure against the tampering of the contents of the
negotiation, is the most important and challenging issue.</p>
      <p>The technical unrealizability of the cracking system is
confirmed by the results of a direct search of key
combinations, which did not yield any results for a limited
time interval (a week). The search was conducted on a test
computer (Windows 11 operating system; 11th Gen Intel (R)
Core (TM) i7-11370H 3.3 GHz processor; 16 GB memory)
and the algorithm was simplified. As in most modern
defense systems, the system persistence is determined by
the amount of key information. In this work, we used keys
with dimensionality  from 150 to 250 bits (20 samples).
Accordingly, there are 2 variants of the key sequence.
There is a dependence on the key length in the system
(Fig. 10). With its increase the quality of the reconstructed
signal deteriorates from excellent to good (at 150 samples
and further). This behavior of the system can be explained
by the accumulation of errors as a result of convolution with
a filter with a long impulse response. It is acceptable to use
a key with a dimensionality of 20–30 samples.</p>
      <p>The physical unrealizability of the hacking system is
based on the fact that to date there are no systems that allow
to distinguish the signal against the background of noise at
the values of the parameter  used in this work. The
application of known methods of noise suppression did not
give a positive result. It is technically very difficult to isolate
a useful signal.
To ensure reliable protection of information, the key is
chosen to be noise-like and can be generated using a
pseudo-random number generator. When the key length
increases, the frequency response of filters becomes more
complicated, and the degree of information closure
increases. It is found that it is inefficient to use very long
keys because the algorithm performance decreases as a
result of long convolutions. The method of key distribution
is a separate non-trivial task. Most often participants agree
on the key to be used beforehand.</p>
    </sec>
    <sec id="sec-5">
      <title>5. Conclusions</title>
      <p>Based on the research conducted on the speech information
protection system, the following results are obtained in this
paper:
(1) A method for synthesizing matched wavelet
filters satisfying the property of orthogonality of
the wavelet basis, the presence of zero moments is
presented. The important requirements on the
filters are the conditions of complete signal
recovery and elimination of overlapping spectra.
The results of the studies show the effectiveness
of synthesized wavelet filters in solving the
problem of information protection.
(2) A speech protection algorithm is developed that
uses MWF at the stage of building a bank of
analysis-synthesis filters consistent with the key.
The algorithm has a simple implementation, and
fast algorithms of digital signal processing
(convolution, decimation, interpolation), allowing
encrypting of the signal in real-time. The
proposed algorithm is noise-resistant and can be
used in channels with intense interference, with
the value of the parameter  &gt; 25 dB. The
algorithm is robust to time delays and hiccups,
distortions in the communication channel.
(3) Operating parameters of the protection system
are obtained, satisfying the high degree of system
closure and acceptable quality of the recovered
signal. Operating parameters are calculated based
on PESQ estimation, and expert estimation  .
Such estimations allow us to qualitatively describe
the processes occurring in the speech protection
system. The parameter setting interval  : −34 &lt;
 , дБ &lt; −15 for FIR filters and −20 &lt;  , дБ &lt;
−15 for IIR filters is obtained.
(4) The system is analyzed for FIR and IIR filters. The
quality of the decoded signal is lower when using
FIR filters, because of the error amplification in
recursive systems.
(5) The case of operation of the protection algorithm
in conditions of application of ITU-T G.711
standard for signal coding is considered. The
results are obtained taking into account the
quantization of the signal in channels 8, 16, and 32
bits. It is found that 8 and 16-bit quantization of
encrypted signal in the channel provides
necessary conditions for correct protected
transmission of speech information. There is no
need to increase the number of quantization levels
(for example, up to 32 bits).
(6) The estimation of the degree of speech
information closure, determined mainly by the
ratio of masking noise and useful signal levels, is
carried out. From the analysis of signal
spectrograms in the channel, it follows that it is
very difficult to distinguish a useful signal from
the mixture. The amount of key information,
which determines the number of tests required for
key selection, has been estimated. In this work, we
used keys of dimension  from 150 to 250 bits, and
the number of key combinations is 2 . Direct
search of key combinations for the allotted time
did not give results. The information can be
decrypted only by knowing the key on the
receiving side with which it was encrypted.</p>
      <p>Future research requires analyzing the system behavior
depending on the type of masking noise, i.e., investigating
noise, structural, and combined interference. To find out
which noise interference of white noise type, a mixture of
white and pink noise, and structural interference of “speech
chorus” type are the most suitable for the proposed
information protection system.</p>
    </sec>
  </body>
  <back>
    <ref-list>
      <ref id="ref1">
        <mixed-citation>
          [1]
          <string-name>
            <given-names>S. B.</given-names>
            <surname>Sadkhan</surname>
          </string-name>
          ,
          <string-name>
            <given-names>A.</given-names>
            <surname>Salah</surname>
          </string-name>
          ,
          <article-title>The Trade-off between Security and Quality using Permutation and Substitution Techniques in Speech Scrambling System</article-title>
          ,
          <source>in: International Conference of Computer and Applied Sciences (CAS)</source>
          (
          <year>2019</year>
          )
          <fpage>244</fpage>
          -
          <lpage>249</lpage>
          . doi:
          <volume>10</volume>
          .1109/CAS47993.
          <year>2019</year>
          .
          <volume>9075489</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref2">
        <mixed-citation>
          [2]
          <string-name>
            <given-names>L.</given-names>
            <surname>Kriuchkova</surname>
          </string-name>
          , et al.,
          <article-title>Experimental Research of the Parameters of Danger and Protective Signals Attached to High-Frequency Imposition, in: Cybersecurity Providing in Information and Telecommunication Systems II</article-title>
          , vol.
          <volume>3550</volume>
          (
          <year>2023</year>
          )
          <fpage>261</fpage>
          -
          <lpage>268</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref3">
        <mixed-citation>
          [3]
          <string-name>
            <given-names>Y.</given-names>
            <surname>Huang</surname>
          </string-name>
          ,
          <string-name>
            <given-names>Y.</given-names>
            <surname>Wang</surname>
          </string-name>
          <article-title>, Multi-Format Speech Perception Hashing based on Time-Frequency Parameter Fusion of Energy Zero Ratio and Frequency Band Variance</article-title>
          ,
          <source>in: 3rd International Conference on Electronic Information Technology and Computer Engineering</source>
          (
          <year>2019</year>
          )
          <fpage>243</fpage>
          -
          <lpage>251</lpage>
          . doi:
          <volume>10</volume>
          .1109/EITCE47263.
          <year>2019</year>
          .
          <volume>9094822</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref4">
        <mixed-citation>
          [4]
          <string-name>
            <given-names>D.</given-names>
            <surname>Abdulrida</surname>
          </string-name>
          ,
          <string-name>
            <given-names>N. A.</given-names>
            <surname>Abbas</surname>
          </string-name>
          ,
          <source>Speech Descrambling Based on Chaotic Parameter Estimation, in: 1st Babylon International Conference on Information Technology and Science (BICITS)</source>
          (
          <year>2021</year>
          )
          <fpage>33</fpage>
          -
          <lpage>38</lpage>
          . doi:
          <volume>10</volume>
          .1109/BICITS51482.
          <year>2021</year>
          .
          <volume>9509926</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref5">
        <mixed-citation>
          [5]
          <string-name>
            <given-names>A. M.</given-names>
            <surname>Raheema</surname>
          </string-name>
          ,
          <string-name>
            <given-names>S. B.</given-names>
            <surname>Sadkhan</surname>
          </string-name>
          ,
          <string-name>
            <given-names>S. M.</given-names>
            <surname>Abdul Sattar</surname>
          </string-name>
          ,
          <source>Performance Evaluation of Voice Encryption Techniques Based on Modified Chaotic Systems, 6th International Engineering Conference Sustainable Technology and Development (IEC)</source>
          (
          <year>2020</year>
          )
          <fpage>135</fpage>
          -
          <lpage>140</lpage>
          . doi:
          <volume>10</volume>
          .1109/IEC49899.
          <year>2020</year>
          .
          <volume>9122933</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref6">
        <mixed-citation>
          [6]
          <string-name>
            <given-names>N.</given-names>
            <surname>Hayati</surname>
          </string-name>
          , et al.,
          <string-name>
            <surname>End-</surname>
          </string-name>
          to-
          <source>End Voice Encryption Based on Multiple Circular Chaotic Permutation, in: 2nd International Conference on Communication Engineering and Technology (ICCET)</source>
          (
          <year>2019</year>
          )
          <fpage>101</fpage>
          -
          <lpage>106</lpage>
          . doi:
          <volume>10</volume>
          .1109/ICCET.
          <year>2019</year>
          .
          <volume>8726890</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref7">
        <mixed-citation>
          [7]
          <string-name>
            <given-names>O.</given-names>
            <surname>Romanovskyi</surname>
          </string-name>
          , et al.,
          <article-title>Prototyping Methodology of End-to-End Speech Analytics Software</article-title>
          ,
          <source>in: 4th International Workshop on Modern Machine Learning Technologies and Data Science</source>
          , vol.
          <volume>3312</volume>
          (
          <year>2022</year>
          )
          <fpage>76</fpage>
          -
          <lpage>86</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref8">
        <mixed-citation>
          [8]
          <string-name>
            <given-names>I.</given-names>
            <surname>Iosifov</surname>
          </string-name>
          , et al.,
          <source>Transferability Evaluation of Speech Emotion Recognition Between Different Languages, Advances in Computer Science for Engineering and Education</source>
          <volume>134</volume>
          (
          <year>2022</year>
          )
          <fpage>413</fpage>
          -
          <lpage>426</lpage>
          . doi:
          <volume>10</volume>
          .1007/978-3-
          <fpage>031</fpage>
          - 04812-8_
          <fpage>35</fpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref9">
        <mixed-citation>
          [9]
          <string-name>
            <given-names>G.</given-names>
            <surname>Konakhovych</surname>
          </string-name>
          , et al.,
          <source>Method of Reliability Increasing Based on Spare Parts Optimization for Telecommunication Equipment, Lecture Notes in Networks and Systems</source>
          <volume>992</volume>
          (
          <year>2024</year>
          )
          <fpage>296</fpage>
          -
          <lpage>309</lpage>
          . doi:
          <volume>10</volume>
          .1007/978-3-
          <fpage>031</fpage>
          -60196-5_
          <fpage>22</fpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref10">
        <mixed-citation>
          [10]
          <string-name>
            <given-names>H.</given-names>
            <surname>Ye</surname>
          </string-name>
          , et al.,
          <source>A Voice Encryption Method Based on Complex Bao Chaos System, International Conference on Computing, Communication, Perception and Quantum Technology (CCPQT)</source>
          (
          <year>2022</year>
          )
          <fpage>268</fpage>
          -
          <lpage>273</lpage>
          . doi:
          <volume>10</volume>
          .1109/CCPQT56151.
          <year>2022</year>
          .
          <volume>00053</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref11">
        <mixed-citation>
          [11]
          <string-name>
            <given-names>V.</given-names>
            <surname>Kuzmin</surname>
          </string-name>
          , et al.,
          <source>Method for Correcting the Mathematical Model in Case of Empirical Data Asymmetry, Lecture Notes in Networks and Systems</source>
          <volume>657</volume>
          (
          <year>2023</year>
          )
          <fpage>249</fpage>
          -
          <lpage>260</lpage>
          . doi:
          <volume>10</volume>
          .1007/978-3-
          <fpage>031</fpage>
          -36201-9_
          <fpage>21</fpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref12">
        <mixed-citation>
          [12]
          <string-name>
            <given-names>K. P.</given-names>
            <surname>Pushpavathi</surname>
          </string-name>
          ,
          <string-name>
            <given-names>B.</given-names>
            <surname>Kanmani</surname>
          </string-name>
          ,
          <article-title>FIR Filter Design using Wavelet Coefficients</article-title>
          ,
          <source>International Conference on Wireless Communications Signal Processing and Networking (WiSPNET)</source>
          (
          <year>2019</year>
          )
          <fpage>410</fpage>
          -
          <lpage>415</lpage>
          . doi:
          <volume>10</volume>
          .1109/WiSPNET45539.
          <year>2019</year>
          .
          <volume>9032718</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref13">
        <mixed-citation>
          [13]
          <string-name>
            <given-names>S. C.</given-names>
            <surname>Venkateswarlu</surname>
          </string-name>
          ,
          <string-name>
            <given-names>N. U.</given-names>
            <surname>Kumar</surname>
          </string-name>
          ,
          <string-name>
            <given-names>A.</given-names>
            <surname>Karthik</surname>
          </string-name>
          ,
          <article-title>Speech Enhancement using Recursive Least Square based on Real-Time Adaptive Filtering Algorithm</article-title>
          , in: 6th International Conference for Convergence in
          <source>Technology (I2CT)</source>
          (
          <year>2021</year>
          )
          <fpage>1</fpage>
          -
          <lpage>4</lpage>
          . doi:
          <volume>10</volume>
          .1109/I2CT51068.
          <year>2021</year>
          .
          <volume>9417929</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref14">
        <mixed-citation>
          [14]
          <string-name>
            <given-names>O.</given-names>
            <surname>Holubnychyi</surname>
          </string-name>
          ,
          <string-name>
            <given-names>O.</given-names>
            <surname>Lavrynenko</surname>
          </string-name>
          , D. Bakhtiiarov, WellAdapted to Bounded
          <source>Norms Predictive Model for Aviation Sensor Systems, Lecture Notes in Networks and Systems</source>
          <volume>736</volume>
          (
          <year>2023</year>
          )
          <fpage>179</fpage>
          -
          <lpage>193</lpage>
          . doi:
          <volume>10</volume>
          .1007/978-3-
          <fpage>031</fpage>
          -38082-2_
          <fpage>14</fpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref15">
        <mixed-citation>
          [15]
          <string-name>
            <given-names>M.</given-names>
            <surname>Joorabchi</surname>
          </string-name>
          ,
          <string-name>
            <given-names>S.</given-names>
            <surname>Ghorshi</surname>
          </string-name>
          ,
          <string-name>
            <given-names>Y.</given-names>
            <surname>Naderahmadian</surname>
          </string-name>
          ,
          <source>Speech Denoising Based on Wavelet Transform and Wiener Filtering, in: 8th International Conference on Frontiers of Signal Processing (ICFSP)</source>
          (
          <year>2023</year>
          )
          <fpage>43</fpage>
          -
          <lpage>46</lpage>
          . doi:
          <volume>10</volume>
          .1109/ICFSP59764.
          <year>2023</year>
          .
          <volume>10372899</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref16">
        <mixed-citation>
          [16]
          <string-name>
            <given-names>A.</given-names>
            <surname>Saleh</surname>
          </string-name>
          ,
          <string-name>
            <given-names>S. B.</given-names>
            <surname>Sadhkan</surname>
          </string-name>
          ,
          <source>A Proposed Speech Scrambling based on Haar Transform and Permutation, 2nd International Conference on Engineering Technology and its Applications</source>
          (
          <year>2019</year>
          )
          <fpage>31</fpage>
          -
          <lpage>36</lpage>
          . doi:
          <volume>10</volume>
          .1109/IICETA47481.
          <year>2019</year>
          .
          <volume>9013013</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref17">
        <mixed-citation>
          [17]
          <string-name>
            <given-names>O.</given-names>
            <surname>Lavrynenko</surname>
          </string-name>
          , et al.,
          <article-title>A Method for Extracting the Semantic Features of Speech Signal Recognition Based on Empirical Wavelet Transform</article-title>
          ,
          <source>Radioelectronic and Computer Systems</source>
          <volume>3</volume>
          (
          <issue>107</issue>
          ) (
          <year>2023</year>
          )
          <fpage>101</fpage>
          -
          <lpage>124</lpage>
          . doi:
          <volume>10</volume>
          .32620/reks.
          <year>2023</year>
          .
          <volume>3</volume>
          .09.
        </mixed-citation>
      </ref>
      <ref id="ref18">
        <mixed-citation>
          [18]
          <string-name>
            <given-names>I. K.</given-names>
            <surname>Alak</surname>
          </string-name>
          ,
          <string-name>
            <given-names>S.</given-names>
            <surname>Ozaydin</surname>
          </string-name>
          ,
          <article-title>Speech Denoising with Maximal Overlap Discrete Wavelet Transform</article-title>
          ,
          <source>International Conference on Electrical and Computing Technologies and Applications (ICECTA)</source>
          (
          <year>2022</year>
          )
          <fpage>27</fpage>
          -
          <lpage>30</lpage>
          . doi:
          <volume>10</volume>
          .1109/ICECTA57148.
          <year>2022</year>
          .
          <volume>9990250</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref19">
        <mixed-citation>
          [19]
          <string-name>
            <given-names>M.</given-names>
            <surname>Chandni</surname>
          </string-name>
          ,
          <string-name>
            <given-names>D.</given-names>
            <surname>Govind</surname>
          </string-name>
          ,
          <article-title>Effectiveness of Wavelet Synchrosqueezed Transform for Improved Epoch Estimation from Telephonic Speech Signals Using Zero Frequency Filtering</article-title>
          ,
          <source>in: 18th India Council International Conference (INDICON)</source>
          (
          <year>2021</year>
          )
          <fpage>1</fpage>
          -
          <lpage>5</lpage>
          . doi:
          <volume>10</volume>
          .1109/INDICON52576.
          <year>2021</year>
          .
          <volume>9691652</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref20">
        <mixed-citation>
          [20]
          <string-name>
            <given-names>O.</given-names>
            <surname>Lavrynenko</surname>
          </string-name>
          , et al.,
          <article-title>A Wavelet-Based Steganographic Method for Text Hiding in an Audio Signal</article-title>
          ,
          <source>Sensors</source>
          <volume>22</volume>
          (
          <issue>15</issue>
          ) (
          <year>2022</year>
          )
          <article-title>5832</article-title>
          . doi:
          <volume>10</volume>
          .3390/s22155832.
        </mixed-citation>
      </ref>
      <ref id="ref21">
        <mixed-citation>
          [21]
          <string-name>
            <given-names>G.</given-names>
            <surname>Yang</surname>
          </string-name>
          ,
          <string-name>
            <given-names>Y.</given-names>
            <surname>Song</surname>
          </string-name>
          ,
          <string-name>
            <given-names>J.</given-names>
            <surname>Du</surname>
          </string-name>
          ,
          <source>Speech Signal Denoising Algorithm and Simulation Based on Wavelet Threshold, in: 4th International Conference on Natural Language Processing (ICNLP)</source>
          (
          <year>2022</year>
          )
          <fpage>304</fpage>
          -
          <lpage>309</lpage>
          . doi:
          <volume>10</volume>
          .1109/ICNLP55136.
          <year>2022</year>
          .
          <volume>00055</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref22">
        <mixed-citation>
          [22]
          <string-name>
            <given-names>S. B.</given-names>
            <surname>Sadkhab</surname>
          </string-name>
          ,
          <string-name>
            <given-names>A. M.</given-names>
            <surname>Raheema</surname>
          </string-name>
          ,
          <string-name>
            <given-names>S. M.</given-names>
            <surname>Abdul Sattar</surname>
          </string-name>
          ,
          <article-title>Design and Implementation Voice Scrambling Model Based on Hybrid Chaotic Signals</article-title>
          , in: International Conference of Computer and
          <source>Applied Sciences (CAS)</source>
          (
          <year>2019</year>
          )
          <fpage>193</fpage>
          -
          <lpage>198</lpage>
          . doi:
          <volume>10</volume>
          .1109/CAS47993.
          <year>2019</year>
          .
          <volume>9075626</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref23">
        <mixed-citation>
          [23]
          <string-name>
            <given-names>O.</given-names>
            <surname>Yu. Lavrynenko</surname>
          </string-name>
          , et al.,
          <article-title>Application of Daubechies Wavelet Analysis in Problems of Acoustic Detection of UAVs</article-title>
          , in: 6th Workshop for Young Scientists in Computer Science &amp; Software Engineering, vol.
          <volume>3662</volume>
          (
          <year>2024</year>
          )
          <fpage>125</fpage>
          -
          <lpage>143</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref24">
        <mixed-citation>
          [24]
          <string-name>
            <given-names>G.</given-names>
            <surname>Singh</surname>
          </string-name>
          , et al.,
          <article-title>Novel Architecture for Lifting Discrete Wavelet Packet Transform with Arbitrary Tree Structure</article-title>
          ,
          <source>Transactions on Very Large Scale Integration Systems</source>
          <volume>29</volume>
          (
          <issue>7</issue>
          ) (
          <year>2021</year>
          )
          <fpage>1490</fpage>
          -
          <lpage>1494</lpage>
          . doi:
          <volume>10</volume>
          .1109/TVLSI.
          <year>2021</year>
          .
          <volume>3079989</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref25">
        <mixed-citation>
          [25]
          <string-name>
            <given-names>C.</given-names>
            <surname>Zhang</surname>
          </string-name>
          , et al.,
          <source>Research on Extracting Algorithm of Speech Eigenvalue based on Wavelet Packet Transform and Gammatone Filter, in: 3rd Information Technology, Networking, Electronic and Automation Control Conference (ITNEC)</source>
          (
          <year>2019</year>
          )
          <fpage>165</fpage>
          -
          <lpage>169</lpage>
          . doi:
          <volume>10</volume>
          .1109/ITNEC.
          <year>2019</year>
          .
          <volume>8729292</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref26">
        <mixed-citation>
          [26]
          <string-name>
            <given-names>J.</given-names>
            <surname>Guo</surname>
          </string-name>
          , et al.,
          <source>Modeling and Simulation of Power Grid Voltage Harmonic Detection Method based on the Improved Wavelet Packet Transform, Chinese Control Conference (CCC)</source>
          (
          <year>2021</year>
          )
          <fpage>6716</fpage>
          -
          <lpage>6721</lpage>
          . doi:
          <volume>10</volume>
          .23919/CCC52363.
          <year>2021</year>
          .
          <volume>9549731</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref27">
        <mixed-citation>
          [27]
          <string-name>
            <given-names>D.</given-names>
            <surname>Bakhtiiarov</surname>
          </string-name>
          , et al.,
          <source>Method of Binary Detection of Small Unmanned Aerial Vehicles, in: Cybersecurity Providing in Information and Telecommunication Systems</source>
          , vol.
          <volume>3654</volume>
          (
          <year>2024</year>
          )
          <fpage>312</fpage>
          -
          <lpage>321</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref28">
        <mixed-citation>
          [28]
          <string-name>
            <given-names>C.</given-names>
            <surname>Zhang</surname>
          </string-name>
          , et al.,
          <source>Research on Extracting Algorithm of Speech Eigenvalue Based on Wavelet Packet Transform and Gammatone Filter, in: 3rd Information Technology, Networking, Electronic and Automation Control Conference (ITNEC)</source>
          (
          <year>2019</year>
          )
          <fpage>165</fpage>
          -
          <lpage>169</lpage>
          . doi:
          <volume>10</volume>
          .1109/ITNEC.
          <year>2019</year>
          .
          <volume>8729292</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref29">
        <mixed-citation>
          [29]
          <string-name>
            <given-names>O.</given-names>
            <surname>Lavrynenko</surname>
          </string-name>
          , et al.,
          <article-title>Method of Remote Biometric Identification of a Person by Voice based on Wavelet Packet Transform</article-title>
          ,
          <source>in: Cybersecurity Providing in Information and Telecommunication Systems</source>
          , vol.
          <volume>3654</volume>
          (
          <year>2024</year>
          )
          <fpage>150</fpage>
          -
          <lpage>162</lpage>
          .
        </mixed-citation>
      </ref>
      <ref id="ref30">
        <mixed-citation>
          [30]
          <string-name>
            <given-names>D.</given-names>
            <surname>Sun</surname>
          </string-name>
          , et al.,
          <source>Damage Degree Assessment Based on Lamb Wave and Wavelet Packet Transform, Chinese Control and Decision Conference (CCDC)</source>
          (
          <year>2019</year>
          )
          <fpage>3179</fpage>
          -
          <lpage>3184</lpage>
          . doi:
          <volume>10</volume>
          .1109/CCDC.
          <year>2019</year>
          .
          <volume>8833396</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref31">
        <mixed-citation>
          [31]
          <string-name>
            <given-names>A.</given-names>
            <surname>Dutt</surname>
          </string-name>
          ,
          <string-name>
            <given-names>P.</given-names>
            <surname>Gader</surname>
          </string-name>
          ,
          <article-title>Wavelet Multiresolution Analysis Based Speech Emotion Recognition System Using 1D CNN LSTM Networks</article-title>
          ,
          <source>IEEE/ACM Transactions on Audio, Speech, and Language Processing</source>
          <volume>31</volume>
          (
          <year>2023</year>
          )
          <fpage>2043</fpage>
          -
          <lpage>2054</lpage>
          . doi:
          <volume>10</volume>
          .1109/TASLP.
          <year>2023</year>
          .
          <volume>3277291</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref32">
        <mixed-citation>
          [32]
          <string-name>
            <given-names>O.</given-names>
            <surname>Lavrynenko</surname>
          </string-name>
          , et al.,
          <article-title>Remote Voice User Verification System for Access to</article-title>
          IoT
          <source>Services Based on 5G Technologies, in: 12th International Conference on Intelligent Data Acquisition and Advanced Computing Systems: Technology and Applications (IDAACS)</source>
          (
          <year>2023</year>
          )
          <fpage>1042</fpage>
          -
          <lpage>1048</lpage>
          . doi:
          <volume>10</volume>
          .1109/ IDAACS58523.
          <year>2023</year>
          .
          <volume>10348955</volume>
          .
        </mixed-citation>
      </ref>
    </ref-list>
  </back>
</article>