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<article xmlns:xlink="http://www.w3.org/1999/xlink">
  <front>
    <journal-meta />
    <article-meta>
      <title-group>
        <article-title>Study of the system of the main functions of Schauder as a means of presenting and compressing sound information for wireless sensor networks</article-title>
      </title-group>
      <contrib-group>
        <contrib contrib-type="author">
          <string-name>Mykola Meleshko</string-name>
          <email>mykola.meleshko@npp.nau.edu.ua</email>
          <xref ref-type="aff" rid="aff1">1</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Vadym Rakytskyi</string-name>
          <xref ref-type="aff" rid="aff1">1</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Andriy Dudnik</string-name>
          <email>a.s.dudnik@gmail.com</email>
          <xref ref-type="aff" rid="aff0">0</xref>
          <xref ref-type="aff" rid="aff3">3</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Andriy Fesenko</string-name>
          <email>aafesenko88@gmail.com</email>
          <xref ref-type="aff" rid="aff2">2</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Vitalijus Cernej</string-name>
          <email>cernejvitalijus@gmail.com</email>
          <xref ref-type="aff" rid="aff0">0</xref>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Vira Mykolaichuk</string-name>
          <xref ref-type="aff" rid="aff3">3</xref>
        </contrib>
        <aff id="aff0">
          <label>0</label>
          <institution>Interregional Academy of Personnel Management</institution>
          ,
          <addr-line>Frometivska Str., 2, Kyiv, 03039</addr-line>
          ,
          <country country="UA">Ukraine</country>
        </aff>
        <aff id="aff1">
          <label>1</label>
          <institution>Scientific Cyber Security Association of Ukraine</institution>
          ,
          <addr-line>Mykhaila Dontsia Str., 2A, Kyiv, 03161</addr-line>
          ,
          <country country="UA">Ukraine</country>
        </aff>
        <aff id="aff2">
          <label>2</label>
          <institution>State Scientific and Research Institute of Cybersecurity Technologies and Information Protection</institution>
          ,
          <addr-line>Maksym Zalizniak Str., 3/6, Kyiv, 03142</addr-line>
          ,
          <country country="UA">Ukraine</country>
        </aff>
        <aff id="aff3">
          <label>3</label>
          <institution>Taras Shevchenko National University of Kyiv</institution>
          ,
          <addr-line>Volodymyrska Str., 60, Kyiv, 03022</addr-line>
          ,
          <country country="UA">Ukraine</country>
        </aff>
      </contrib-group>
      <abstract>
        <p>An analysis of the technical possibilities of using the Schauder basis function system for presenting and compressing information, for example, network data, is provided. Advantages compared to other bases have been revealed. On the basis of experimental studies, recommendations are provided for their use for digital processing of audio information at a representative level in computer systems and networks. From our point of view, digital systems that use the representation of sound signals by a system of basic functions have great potential. Further considerations will be directed to justifying the expediency of their use in engineering developments. At the same time, computer modeling is an efective tool for finding rational methods of building digital signal processing equipment. Preliminary modeling of signal analysis and synthesis processes allows at the research stage to determine the complexity of hardware solutions, possible parameters of digital processing (accuracy of transmission and reproduction, degree of elimination of redundancy (compression), speed, and others).</p>
      </abstract>
      <kwd-group>
        <kwd>eol&gt;system of basic functions of Schauder</kwd>
        <kwd>presentation and compression of information</kwd>
        <kwd>digital processing of speech signals</kwd>
      </kwd-group>
    </article-meta>
  </front>
  <body>
    <sec id="sec-1">
      <title>1. Introduction</title>
      <p>
        A characteristic feature of modernity is the sharp excess of the growth of the flow of information over
the expansion of data transmission channels [
        <xref ref-type="bibr" rid="ref1 ref2 ref3">1, 2, 3</xref>
        ]. Therefore, the search for rational methods and
technical means of information processing is essential [
        <xref ref-type="bibr" rid="ref4 ref5 ref6 ref7">4, 5, 6, 7</xref>
        ]. One of the stages of processing is
information compression. At the modern level, digital transmission systems have become widespread,
the relevance of which is, first of all, connected with the rapid development of information computer
technologies that provide transformation, storage and transmission of information, as well as modeling
of presentation and processing processes signals, systems and devices. Among the important areas of
application of digital methods are the presentation and processing of broadcast signals, audio information
in the broad sense [
        <xref ref-type="bibr" rid="ref10 ref8 ref9">8, 9, 10</xref>
        ].
      </p>
      <p>
        The ever-growing volume of useful information obtained from sources of various physical origins
requires the search for efective preprocessing methods at the stage of presentation, storage, and
determination of channel transmission methods [
        <xref ref-type="bibr" rid="ref11 ref12">11, 12, 13</xref>
        ].
      </p>
      <p>The most important task of this problem is to reduce the redundancy while maintaining the desired
transmission quality [14, 15, 16]. Traditional experiments have become methods of reducing the
redundancy of sound signals, which involve coding of acoustic oscillations or analysis/synthesis based
on hardware and software tools [17, 18, 19]. Despite the fact that a significant amount of research
has been conducted in the field of digital processing of sound signals, especially speech signals, there
are still many unsolved problems [20, 21, 22]. One of the reserves of improving systems for reducing
the redundancy of sound signals is the search for new methods of presentation and processing using
programming computing tools of computer technologies [23, 24, 25].</p>
    </sec>
    <sec id="sec-2">
      <title>2. Review of previous studies, literary sources and research methods</title>
      <p>In 1927, Schauder proposed a system of linearly independent basis functions. The use of Schauder
functions in digital signal processing systems is the main topic of research. The distribution in a row
according to the system of basic functions is generally presented as:</p>
      <p>() = ∑︁   (),</p>
      <p>=0
where () is the realization of the signal as a random process;  () – system of basic functions; 
are the coeficients of the schedule.</p>
      <p>From our point of view, it is appropriate to present the Schauder function in the form [26]:
where () is a system of Haar functions defined on the orthogonality interval as follows:
() =</p>
      <p>1() ,
⎧⎪1,
⎪
⎪
⎪
⎪⎪⎪⎪ 1 ∫︁ 
⎪⎨⎪  0</p>
      <p>+1 ∫︁ 
⎪⎪ 2 2
⎪
⎪
⎪⎪⎪⎪  0
⎪⎩⎪0,
() = () =
 = 0,  ∈ [0,  ],
 = 1,  ∈ [0,  ],
() () ,  = 2, 3, . . . ,  ∈ [0,  ],</p>
      <p>∈/ [0,  ],
1 () = 1,  [0,  ] ,
⎪⎪⎧⎪2 2−1 ,  ∈
⎪
⎪
⎪
⎪
⎨−2 2−1 ,  ∈
⎪
⎪
⎪
⎪
⎪⎪⎩⎪0,  ∈
︂[ ( − 1)</p>
      <p>2−1
︂[ ( − 0.5)</p>
      <p>2−1
︂[ ( − 1)
2−1
( − 0.5) ]︂
2−1
 ]︂
, 2−1
 ]︂
, 2−1
,
where  = 2−1 + ;  = 1, 2, 3, . . .;  = 1, 2, . . . , 2−1 — the number of the Haar function
group — sets the duration of the zero value of the Haar function, which is equal to  /2−1 , and its
amplitude, which is equal to 2(−1)/2 . The index  determines the position of a non-zero value on
the segment [0,  ]. One group includes Haar functions with the same duration of a non-zero value.
The relationship between single and double numbering of functions is as follows:  = 1, 2, 3, . . . ,  ;
 = 1, 2, 3, . . . , 2−1 ;  = +2−1 . Figure 1 presents graphical images of the Haar (a) and Schauder (b)
functions, which are locally defined on the interval [0,  ].</p>
      <p>(a)
(b)
Figure 1: Graphic representation of Haar (a) and Schauder (b) functions.</p>
      <p>The formulas for calculating the expansion coeficients for
 = 1, 2, 3 have the following form
[26, 27, 28]:
using the values of the realization of the signal () at the points ∆ t are given [0,  ].</p>
      <p>Analyzing the expression for calculating coeficients:
 = 
︂(  − 0.5
2−1</p>
      <p>︂)
 −
2</p>
      <p>1 [︂ (︂  − 1
2−1 
︂)
+ 
︁(</p>
      <p>2−1 
︁) ]︂
value of ( ) respectively for 2, 4, 8, . . . , 2 ; coeficients
where  = 2−1 +;  = 1, 2, . . . , 2−1 ;  = 1, 2, . . . ,  ;  = 2 ∆ t ; ∆ t
is the signal quantization
interval over time, it can be established that for  = 0, 2, . . . , 2 there is a regular repeatability, the
same () values are used many times to calculate other coeficients.</p>
      <p>So, for example, the value of (0) is used to calculate the coeficients 2, 3, 5, 9, . . . , 2(−1) +1,
4, 5, 9, . . . , 7·2 −3 +1, and
1, 4, 8, . . . , 7·2 −3 .</p>
      <p>During the research, it was established that the values of () at the points  = 2−
+  (where
 = 1, 2, 3, . . .,  = 0, 1, . . . , 2) are used for calculation of a series of coeficients with serial numbers
 = 2 + . When using this approach, the calculation time can be significantly reduced. Without taking
into account the regularity, each time when calculating the next coeficient, it would be necessary to
recalculate the coordinates of the () values on the segment [0,  ].</p>
    </sec>
    <sec id="sec-3">
      <title>3. The method of operation of the audio codec according to the</title>
    </sec>
    <sec id="sec-4">
      <title>Schauder basis function system for the sound recognition system for the terrain monitoring sensor network</title>
      <p>In this way, it is proposed to use the discrete Schauder transformation for the design and research of
network encoding/decoding means, which can be implemented at the algorithmic-program level or in a
hardware-technological design based on FPGA-type integrated circuits. The advantage of this method is
the use of the Schauder transformation, which has certain advantages compared to trigonometric bases,
for example, the possibility of local processing on the time and frequency interval when segmenting the
incoming sound stream, reducing the time for mathematical calculations of the expansion coeficients,
minimizing the amount of memory. All this significantly afects the speed of data delivery to the user.
A variant of the structural diagram of the application of this method in the information transmission
channel consisting of a Schauder encoder, a transmission channel and a Schauder decoder has been
developed.</p>
      <p>A method of dynamic sound recognition for a multi-sensor Internet of Things location monitoring
sensor network, characterized in that modules are placed on each device or node of the Internet of Things
that needs monitoring, which configure and receive operational data and data about the environment
of the device or node of the Internet of Things (Figure 2).</p>
      <sec id="sec-4-1">
        <title>These modules consist of a transmitter and a receiver.</title>
        <p>The transmitter contains the following modules and blocks:
• Message source: Generates an output message, which can be in the form of an analog or digital
signal;
• Encoder: Converts a message into a digital code that is convenient for transmission. This may
include data compression or error correction;
• Modulator: Uses a carrier frequency signal generated by a synthesizer to superimpose information
(modulation) on a high-frequency signal;
• Power Amplifier: Amplifies the modulation signal to a level suficient for transmission; •
Transmitting antenna: Radiates an amplified signal into space.</p>
        <p>The receiver contains the following modules and blocks:
• Receiving antenna: Receives a signal from space;
• UHF converter: The received signal is often attenuated and mixed with an intermediate frequency
(UHF) for further processing;
• Demodulator: Extracts the information signal from the modulation signal;
• Decoder: Converts the digital code back to the original form of the message, performs decoding
and error correction;
• Message Recipient: The end device receiving the recovered message.</p>
        <p>A codec using the method of Schauder basis functions performed in the encoder and decoder for the
audio recognition system works as follows:</p>
        <p>Coding (encoder):
1. Analog-to-digital conversion (ADC): The audio signal is first converted from analog to digital
using an ADC.
2. Decomposition into Schauder basis functions: The digital signal is decomposed into basic Schauder
functions. These functions form a set of orthogonal bases that can be used to represent the signal.
Decomposition is carried out by projecting the signal onto these basis functions. The result is a
set of coeficients describing the signal in the space of basis functions.
3. Quantization of coeficients: The coeficients obtained as a result of the decomposition are
quantized to reduce the amount of data. This means that the values of the coeficients are rounded
to the nearest permissible value from a limited set of levels.
4. Coding of coeficients: Quantized coeficients are encoded for further transmission or storage.</p>
        <p>This may include the use of compression algorithms such as Hufman coding or other entropy
coding techniques.</p>
        <p>Decoding (decoder):
1. Decoding coeficients: The received coded coeficients are decoded to restore the quantized
coeficients.
2. Interpolation and dequantization: The quantized coeficients are converted back to continuous
form (dequantized) to improve the accuracy of the reconstructed signal.
3. Signal reconstruction: The reconstructed signal is formed by a linear combination of Schauder
basis functions using dequantized coeficients. This process is the reverse of decomposition and
allows the signal to be restored to its original form (or with allowable losses if it is a lossy system).
4. Digital-to-analog conversion (DAC): The recovered digital signal is converted back to analog
form using a DAC if analog output is required.</p>
        <p>The working principle of the method is based on the use of Schauder basis functions, which are
orthogonal and enable the eficient representation of any signal, including audio signals, with minimal
information loss. These basis functions yield coeficients that provide a compact representation of the
signal while preserving its essential characteristics. By quantizing and encoding these coeficients,
the data volume required for transmission is significantly reduced. The decoding and reconstruction
processes then apply inverse operations to those used during encoding, allowing for the accurate
recovery of the original signal.</p>
      </sec>
    </sec>
    <sec id="sec-5">
      <title>4. Results of the study of speech signal processing</title>
      <p>Figures 3–7 illustrate implementations of a fragment of a speech signal processed using the Schauder
transformation. The signal is presented in blocks of equal size (2048 samples each). For each block, the
decomposition coeficients were calculated with a spacing of 64 samples. The number of calculated
decomposition coeficients was 64, 32, 16, 8, and 4. Accordingly, the synthesis of the output signal
was performed using the specified number of coeficients (marked as "b" in the figure). In addition, an
option for selecting informative coeficients from among the calculated ones based on the "|max|" or
"max, min" principles was investigated, as shown in Figs. c and d, respectively.</p>
      <p>By comparing the obtained signal implementations, we can conclude that the transformation based
on Schauder’s functions reproduces the speech signal well. The calculation of consecutive 64 and 32
coeficients ensures a fairly accurate reproduction of signals.</p>
      <p>When reproducing the signal by 16 coeficients, the implementation difers in the smoothing of local
peaks in the areas of the high-frequency interval, while the low-frequency areas are reproduced quite
well. If 8 coeficients out of 64 possible are used for signal synthesis, then the signalgrams show a clear
diference between the output signal and the input signal.</p>
      <p>This diference is especially clearly expressed when using 4 coeficients of the decomposition. Similar
conclusions can be drawn regarding the number of informative coeficients selected according to the
"|max|" principle. or "max, min".</p>
      <p>Comparing the images of the output signal obtained by synthesis with the same number of coeficients
but selected in diferent ways, we come to the conclusion that these images difer from each other with
a certain diference. In order to clarify these changes, it is necessary to conduct an auditory subjective
analysis in the future. Carrying out only a visual comparison of the images of the output signals, it is
dificult to determine an objective criterion in which case the output signal is of higher quality. In order
to make a qualitative assessment of the output signals, you should listen to these signals and compare
their static characteristics for diferent options.</p>
    </sec>
    <sec id="sec-6">
      <title>5. Conclusions and discussion</title>
      <p>In the theory and practice of signal processing, researchers increasingly turn to their representation by
a finite system of basis functions, orthogonal or non-orthogonal.</p>
      <p>The solution to this problem is relevant in connection with the need to improve the means of analysis,
processing and synthesis of real physical processes.</p>
      <p>
        Systems of non-orthogonal linearly independent functions have a number of advantages, for example,
currently known PL-functions and Schauder functions [
        <xref ref-type="bibr" rid="ref11 ref12">29, 11, 12, 14</xref>
        ]. So, for example, the calculation
of the decomposition coeficients in this case is reduced to the summation with the weights of the
readings of the aliasing signal on the approximation segment, which for speech signals is chosen in the
range of 8-22 ms.
      </p>
      <p>Among the advantages of non-orthogonal decompositions should also be added the simplicity of
their hardware implementations in comparison with analyzers and synthesizers built on the principle
of orthogonal transformations [29, 32, 33].</p>
      <p>The method is to include in each group of sensors sound sensors and ultrasound sensors that detect
and recognize diferent acoustic waves, while using at least three sensors of each type in such a way
that one sensor of each type is in working condition at the same time, and the other sensors were in an
inactive state, after a certain period of time, the sensor in the working state is put into the rest state,
and the other sensors in the rest state are put into the working state, and each sensor will store the
environmental and operational data that it senses.</p>
      <p>Information from the sensors is processed and transmitted to the anomaly analysis unit using a
system based on the Schauder algorithm and containing information transmission channels consisting
of a Schauder encoder, a transmission channel and a Schauder decoder, the data anomaly analysis unit
is configured to perform data anomaly analysis based on the received pre-processed data and obtaining
the result of the analysis of anomalies, as well as correcting the errors of the result of the analysis of
anomalies to obtain the final result of the analysis.</p>
    </sec>
    <sec id="sec-7">
      <title>Declaration on Generative AI</title>
      <sec id="sec-7-1">
        <title>The authors have not employed any Generative AI tools.</title>
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