RU2515089C1 - Apparatus for acoustic presentation of spatial user information - Google Patents

Abstract

FIELD: physics.

SUBSTANCE: apparatus for acoustic presentation of spatial information comprises a signal generator, a radiation channel amplifier and a transmitter, right- and left-side ultrasonic transducers, first and second analogue-to-digital converters (ADC), first and second memory units, first and second digital-to-analogue converters (DAC), first and second amplifiers, right- and left-side headphones. The apparatus also includes a clock-pulse generator, first and second frequency dividers and a delay unit, first and second switches, a counter, first and second sets of scaling-temporal filters, as well as first and second adders. Outputs of the first and second memory units are connected to data inputs of the first and second sets of scaling-temporal filters, respectively, and outputs of the first and second adders are connected to the first inputs of the first and second DAC, respectively. The apparatus includes series-connected user acoustic analyser analysis unit (UAAAU), third memory unit and multiplexer, the outputs of which are connected to control inputs of the first and second sets of scaling-temporal filters; the second output of the UAAAU is connected to the second input of the multiplexer, and the input of the UAAAU is connected to the output of the clock-pulse generator.

EFFECT: improved object recognition.

5 dwg

Description

The present invention relates to the field of special acoustics and can be used for orientation of specialists whose professional activity is related to movement in conditions of limited visibility, for example, fighters of the Ministry of Emergencies in the fire, as well as for the rehabilitation of the visually impaired.

Known technical systems and devices designed for the visually impaired, allowing them to navigate in the environment. It is customary to single out two areas for the development of technical systems for visualizing the surrounding space to help the blind.

The first direction includes instruments for indicating free path. They are the simplest and carry information only about the presence of an obstacle in the path of the blind (see China Patent CN 2907594, M.cl. A61F 9/08, published 06.06.2007. Stereo ultrasonic device for helping the blind; European patent EP 1025828, M inc. A61F 9/08, G01S 17/02, published 09.08.2000, German priority, Orientation assistance for the blind and visually impaired; German patent DE 4212163, M.cl. A61F 9/08 A61H 3/06, publ. 10/14/1993 Optical echo system for orientation of the blind; Patent for the invention of the Russian Federation No. 2040234, Mcl A61F9 / 08, publ. 07/25/1995, Ultrasonic locator for the blind; Patent for the invention of the Russian Federation No. 2359287, M .cl A61F 9/08, published June 20, 2009. An infrared locator for visually impaired people).

The most advanced device in this area is an Obstacle Detector for visually impaired people (see patent for the invention of the Russian Federation No. 2212871, Mcl A61F 9/08, published September 27, 2003), which contains a pulse generator, DAC, sound frequency generator and headset. The device emits radiation using a line of point infrared emitters and receives a signal on a line of point photoconverters, allowing you to determine the direction of the obstacle at large viewing angles.

Devices related to the first direction are the simplest, but carry information only about the presence of an obstacle in the path of the blind.

The second direction involves the use of hearing to the maximum extent possible for perception of the environment. Devices of this direction allow you to localize the object, get information about the direction and distance. In devices of this type, it is possible to organize scanning of several objects at the same time, and the sound signal carries information about the acoustic nature of objects. To ensure a wide field of view, binaural perception is used.

Close in technical essence to the present invention is an acoustic representation of spatial information (see Leslie Kay. Aerial sonars with acoustic representation of spatial information for the visually impaired. Animal Sonar Systems, edited by RG Busnel and JF Fish Plenum, New York, 1980), which refers to the second direction.

The essence of the presentation of acoustic information is as follows. An ultrasound signal with linear frequency modulation (LFM) is emitted to the medium, with a frequency range of about an octave. Binaural echo signals are received at two ultrasonic receivers. The received signals are multiplied with a reference signal to obtain a beat signal. Carry out low-pass filtering of this signal. After amplification, the signals are sent to the head phones for perception by the human auditory system. The distance to the object is represented by the frequency of the audible signal, and the direction to the object is the interaural difference in the amplitudes of the signals falling on both ears. Thus, in the device under consideration, the echo signal and the reference LFM signal are multiplied (heterodyning operation). The oscillation of the frequency beat generated as a result of low-pass filtering carries information about the distance to the object.

Electronics and battery powered in a separate unit, placed in your pocket. Thus, it provides information about the direction and distance to the object (the farther it is, the higher the sound), and also, after the user's training, within certain limits about its shape, size and texture.

The device is no longer a simple indicator of an obstacle, but to a certain extent an environmental analyzer with many important characteristics. However, the richer the information issued to the operator, the greater the psychoacoustic load he experiences. The operator processes a rather complex sound picture with the use of such parameters of the reflected signal as volume, pitch, timbre characteristics.

However, for the successful perception and classification of objects that make up the acoustic scene, a rather long learning process is required. At the same time, courses on mastering the apparatus in Germany last about a month.

In addition, in the above technical solution, the operation of heterodyning signals is selected as the procedure for matching the operating frequencies of the locator (40-120 kHz) with the frequency range of hearing, which leads to a change in the number of waves contained in the useful signal.

The closest but technical essence to the proposed invention (prototype) is an acoustic representation of spatial information for the visually impaired (see patent for the invention of the Russian Federation No. 2060028, IPC A61F 9/00, publ. 05.20.96). The device comprises left and right ultrasound transducers, a first amplifier and a right headphone connected in series, a signal generator, a radiation path amplifier and a transmitter connected in series, a second amplifier and a left headphone connected in series, the outputs of the right and left headphones being device outputs connected in series the first LCP, the first input of which is connected to the output of the right ultrasonic transducer, and the second input is connected to the output the first switch, the first memory block, the first DAC, the output of which is the input of the first amplifier, the second ADC connected in series, the first input of which is connected to the output of the left ultrasonic transducer, and the second input is connected to the output of the second switch, the second memory block, the second DAC, the output of which is the input of the second amplifier, the synchronization pulse generator connected in series with its output to the first inputs of the switches and to the first input of the counter, the first frequency divider, is connected provided by its output with the second inputs of both DACs, the second frequency divider, the output of which is the input of the signal generator and the input of the delay unit, connected by its output to the second input of the counter and the second inputs of the switches, the third inputs of which are connected to the output of the counter, the output of the signal generator is connected to the input radiation path amplifier.

The essence of the device is as follows. An ultrasonic frequency modulated pulse is emitted. Binaural reception of the echo signal by two ultrasonic microphones is performed, amplifies and converts the electrical signals into acoustic ones, they carry out coordinated filtering of the echo signals, temporarily stretch the received responses in the binaural reception channels, followed by signal perception by the human auditory system.

This device ensures the coordination of the frequency range and duration of the acoustic signal with the auditory analyzer and obtaining complete information contained in the echo signal.

At the same time, it should be noted that the main means of signal processing in this device is the human auditory analyzer. Therefore, even a minimal violation of auditory functions leads to a deterioration in the ability of the operator to navigate in space, i.e. to reduce the recognition of objects.

Achievable technical result of the invention is to improve the recognition of objects that make up the acoustic scene, by selectively increasing the noise immunity of the system.

The achievement of the technical result is ensured by the fact that in the device for the acoustic presentation of spatial information for users, comprising a series-connected signal generator, an amplifier of the radiation path and a transmitter, series-connected right ultrasonic transducer (USP), the first analog-to-digital converter (ADC), the first memory unit, the first digital-to-analog converter (PAP) connected in series, the first amplifier and the right head-phone connected in series the second USP, the second ADC, the second memory unit, the second DAC, the second amplifier and the left headphone connected in series, the outputs of the right and left headphones are respectively the first and second outputs of the device, the synchronization pulse generator, the first frequency divider, the second divider are connected in series frequencies and a delay unit, the first and second switches and a counter, the second inputs of the first and second ADCs respectively connected to the outputs of the first and second switches, the first inputs of which connected to the output of the clock generator and the first input of the counter, the second inputs of the first and second switches are connected to the second input of the counter and the output of the delay unit, and the third inputs of the first and second switches are connected to the output of the counter, the output of the second frequency divider is connected to the input of the signal generator, the second the inputs of the first and second DACs are connected to the output of the first frequency divider, in addition, a first set of time-scale filters and a first adder are sequentially connected, followed by the second set of time-scale filters and the second adder are properly connected, and the outputs of the first and second memory blocks are connected respectively to the information inputs of the first and second sets of time-scale filters, and the outputs of the first and second adders are connected to the first inputs of the first and second DACs, respectively connected to the analysis block of the auditory user analyzer (BASAP), the third memory block and multiplexer, the outputs of which are connected to the control inputs of the first and second sets of scale abno-temporal filters BASAP second output connected to the second input of the multiplexer, and BASAP input connected to the output of the generator clock pulses.

The influence of the distinctive features of the proposed device to achieve a technical result can be explained as follows.

The preliminary testing of the auditory system of the operator allows you to generate corrective time-scale gain factors for each channel, individual for each user.

When performing multi-channel time-scale filtering of responses (direct and inverse wavelet transform (VP)), the selected correction gains in each frequency channel are used, and the subsequent summation of the output signals of each channel is performed.

Consider in more detail the implementation in the proposed device direct and inverse wavelet transforms. For the input process x (t) of the acoustic receiver, a two-dimensional representation is possible - in the time-scale domain, obtained by applying the continuous wavelet transform [I. Dobeshi Ten lectures on wavelets. Izhevsk: Research Center "Regular and chaotic α theories and application examples. Advances in physical sciences. Volume 166, No. 11, 1996, p. 1145 -1170. Dremin IM, Ivanov OV, Nechitailo VA Wavelets and their Usage: Advances in Physical Sciences, Volume 171, No. 5, 2001, pp. 465-501, Dyakonov VP, Wavelets. From Theory to Practice. - M: SOLON-R, 2002, 440 pp., Kravchenko V.F. , Rvachev VA "Wavelet" - systems and their application in signal processing. Foreign radio electronics. 1996, No. 4, p.3-20].

Continuous EP can be defined as the scalar product of the process, x (t) under study, and the basis functions ψ _ατ (t) [Finish I. Ten lectures on wavelets. Izhevsk: Research Center "Regular and chaotic dynamics", 2001, 464 p.]:

$$W_{\psi }\left\{x\left(t\right)\right\}\left(\alpha ,\tau \right)={\displaystyle \underset{-\infty }{\overset{\infty }{\int }}x\left(t\right)}\overline{{\psi }_{\alpha \tau }}\left(t\right)dt=\frac{\mathrm{one}}{\sqrt{{\alpha }_{-\infty }}}{\displaystyle \underset{-\infty }{\overset{\infty }{\int }}x\left(t\right)\overline{\psi }}\left(\frac{t-\tau }{\alpha }\right)dt,(\mathrm{one})$$

where the bar above indicates the operation of complex pairing.

The general principle of constructing the basis of the VP consists in the use of large-scale transformations with the compression parameter α and displacements with the shift parameter τ of the original wavelet function ψ (t) (the so-called mother wavelet):

(f) должна удовлетворять еще одному условию:To be a wavelet, the basis functions ψ _ατ (t) ∈ L ² (R) must have a number of necessary properties [6-10]. They should be: quadratically integrable, alternating (have a zero mean), while the wavelets should tend to zero by ± ∞, and for practical purposes - the faster, the better (and the wavelet should be well localized both in time and in frequency ) In order to make the reverse EP possible, the wavelet spectral function

(f) must satisfy one more condition:

The formula for continuous inverse wavelet transform is:

$$x\left(t\right)=\frac{\mathrm{one}}{C_{\psi }}{\displaystyle \underset{-\infty }{\overset{\infty }{\int }}{\displaystyle \underset{0}{\overset{\infty }{\int }}\left[W_{\psi }x\right]\left(\alpha ,\tau \right){\alpha }^{-\mathrm{one}/2}\psi \left(\frac{t-\tau }{\alpha }\right)\frac{d\alpha d\tau }{{\mathrm{<figure-callout\; id="2"\; label="\alpha "\; filenames="00000018.png"\; state="\{\{state\}\}">\alpha </figure-callout>}}^2}.(\mathrm{four})}}$$

As can be seen from (4), the initial signal x (t) can be reconstructed through the integral sum of the same basis functions ψ _ατ (t) with weights in the form of the wavelet spectrum of the signal [W _ψ x] (α, τ). Here the constant C _ψ (3) acts as a normalizing coefficient similar to the coefficient (2π) ^1/2 normalizing the Fourier transform. The adjustable amplification factors selected in the preliminary testing in each frequency channel are weighting factors that determine the contribution of each of the scales to the total response received by the user's hearing aid. Then the operation is weighted summation in accordance with the ratio:

$$S{S}_m={\displaystyle \sum _{a=0}^{M-\mathrm{one}}{W}_{a,m}}\cdot {\gamma }_a$$

where W is the result of the time-scale transformation, γ _a are the correction factors, and is the scale number, m = 0 ... N-1, N is the number of scale samples.

In this case, the selection of correction factors is carried out by the analysis unit of the auditory analyzer, and the processing of the selected correction coefficients for feeding to the time-scale filters is provided by the introduction of the third memory block and multiplexer.

As a result, the proposed signal processing makes it possible to take into account the user's actual perception of acoustic signals and to transmit more fully environmental information to his auditory analyzer, which improves recognition of objects.

The invention is illustrated by drawings, in which Fig. 1 shows an example implementation of the proposed device, Fig. 2 shows a block diagram of a BASAP operation algorithm, Fig. 3 shows a block diagram of a direct wavelet transform algorithm using a Haar wavelet implemented in a Mathcad environment , figure 4 shows a block diagram of the inverse wavelet transform algorithm performed in the MathCAD environment, figure 5 presents the probabilities of the correct detection of the presented implementations against the background of natural noise.

The structural diagram of the proposed device, shown in figure 1, contains:

1 - signal generator,

2 - radiation path amplifier,

3 - transmitter,

4, 5 - right and left ultrasonic transducers, 6, 7 - the first and second analog-to-digital converters, 8, 9 - the first and second memory blocks,

10, 11 - the first and second digital-to-analog converters,

12, 13 - the first and second amplifiers,

14, 15 - right and left head phones,

16 is a clock generator,

17 - the first frequency divider,

18 is a second frequency divider,

19, 20 - the first and second switches,

21 is a counter

22 - delay unit

23, 24 - the first and second sets of time-scale filters,

25, 26 - the first and second adders,

27 - block analysis of the auditory analyzer of the user (BASAP),

28 - the third block of memory,

29 - multiplexer.

At the same time, in the acoustic representation of spatial information for users, a signal generator 1, an amplifier 2 of the radiation path and a transmitter 3 are connected in series, the right ultrasonic transducer 4 (USP), the first analog-to-digital converter (ADC) 6, the first memory unit 8 are connected in series the first digital-to-analog converter (DAC) 10 is connected, the first amplifier 12 and the right headphone 14 are connected in series to the second USP 5, the second ADC 7, the second memory unit 9, in series the second DAC 11, the second amplifier 13 and the left headphone 15 are connected, while the outputs of the right and left headphones 14 and 15 are respectively the first and second outputs of the device, the synchronizing pulse generator 16, the first frequency divider 17, the second frequency divider 18 are connected in series, and delay unit 22, the first and second switches 19 (20) and the counter 21, and the second inputs of the first and second ADCs 6 (7) are connected respectively to the outputs of the first and second switches 19 (20), the first inputs of which are connected to the output of the generator 16 synchronizing pulses and the first input of the counter 21, the second inputs of the first and second switches 19 (20) are connected to the second input of the counter 21 and the output of the delay unit 22, and the third inputs of the first and second switches 19 (20) are connected to the output of the counter 21, the output of the second the frequency divider 18 is connected to the input of the signal generator 1, the second inputs of the DAC 10 (11) are connected to the output of the first frequency divider 17, the first set of 23 time-scale filters and the adder 25 are connected in series, the second set of 24 time-scale filters are connected in series filters and the adder 26, and the outputs of the first and second memory units 8 (9) are connected to the information inputs of the first and second sets of time-scale filters 23 (24), and the outputs of the adders 25 (26) are the first inputs of the first and second DACs 10 ( 11), the auditory user analyzer analysis unit 27 (BASAP), the third memory unit 28 and the multiplexer 29, the outputs of which are connected to the control inputs of the first and second sets of time-frequency filters 23, are connected in series, the second output of the BASAP 27 is connected to the second inputultipleksora 29 and BASAP input 27 connected to the generator 16, clock pulses.

Consider the operation of the proposed device.

Using a signal generator 1, an amplifier of the radiation path 2, transmitter 3, a very short (ultrashort) ultrasonic pulse is emitted into the medium, which is close to the δ-function. In this case, taking into account the spatial coordinate, the input signal, in fact, represents the impulse response (IC) of the scattering object

$$X\left(x,t\right)=h\left(x,t\right)*\delta \left(x,t\right)\approx h\left(x,t\right)(\mathrm{one})$$

where x is the spatial coordinate; h (x, t) - THEIR scattering object,

* - a convolution symbol.

The implementation of the reflected signal goes to the ADC 6, 7, where it is converted to digital form and entered into memory blocks 8, 9.

Using DACs 10, 11, samples from memory blocks are read out at times slower than they were entered, which ensures the conversion of the input implementation according to the law

$$X\left(x,t\right)\to X\left(x/\alpha ,t/\alpha \right),(5)$$

where α = f1 / f2,

f1 is the carrier frequency of the probe signal,

f2 is the average frequency of the range perceived by the human auditory analyzer. Or considering (1)

h (x, t) → h (x / α, t / α),

The read frequency is set by the first frequency divider 17. The radiation frequency of the probe pulse is set by the second frequency divider 18. Implementations are sent to sets 23 (24) of time-scale filters that implement the wavelet transform, and their subsequent summation is performed in blocks 25 (26), taking into account correction factors coming from the multiplexer 29.

The operation of direct wavelet transform using the Haar wavelet is carried out in accordance with the expression:

, $$D_{n,m}^{j+\mathrm{one}}=S_{2n+\mathrm{1,2}m+\mathrm{one}}+S_{2n+\mathrm{1,2}m}-S_{2n\mathrm{,\; 2}m+\mathrm{one}}+S_{2n\mathrm{,\; 2}m}$$

,

- вейвлет-коэффициенты, соответствующие горизонтальному и вертикальному направлениям, для j-го масштаба; j=0,…,log₂N-1 - количество масштабов; $$n=\mathrm{0,}\dots ,\frac{N}{2^{j+1}}-1$$

; $$m=\mathrm{0,}\dots ,\frac{N}{2^{j+1}}-1$$

- количество строк и столбцов результирующей матрицы вейвлет-коэффициентов для j-го масштаба. Вейвлет Хаара реализован в среде Mathcad в качестве рабочего встроенного модуля (фиг.3).Where $$D_{n,m}^{j+\mathrm{one}}$$

- wavelet coefficients corresponding to horizontal and vertical directions for the j-th scale; j = 0, ..., log ₂ N-1 is the number of scales; $$n=0...,\frac{N}{2^{j+\mathrm{one}}}-\mathrm{one}$$

; $$m=0...,\frac{N}{2^{j+\mathrm{one}}}-\mathrm{one}$$

- the number of rows and columns of the resulting matrix of wavelet coefficients for the j-th scale. The Haar wavelet is implemented in the Mathcad environment as a working built-in module (Fig. 3).

In this case, the forecasting procedure for each of the scales of the time-scale plane is performed.

The operation of the inverse wavelet transform is performed in the MathCAD environment and is presented in the form of an algorithm (Fig. 4) in accordance with the expression:

$$S\left(t\right)={\displaystyle \sum _{n=0}^{N-\mathrm{one}}{\displaystyle \sum _{m=0}^{N-\mathrm{one}}{D}_{n,m}}}{\psi }_{n,m},$$

where ψ _{n, m} are basis functions.

BASAP 27 is designed to obtain corrective gain in each processing channel. Adjustable gains in each frequency channel are weighting factors that determine the contribution of each of the scales to the total response received by the user's auditory analyzer (SAP). In order to take into account the peculiarities of the SAP, it is necessary to initially take the values of the correction factors equal to unity. When conducting tests in the work of BASAP 27, the forced choice technique was used [Green D. Application of the theory of detection in psychophysics, TIIER 5, vol. 58, 1970], the essence of which is the choice of the test subject for a certain time interval from several hypotheses. In our implementation example, there are 2 hypotheses: is there a signal or not a signal.

, где x - соответствует правильному обнаружению, y - общее количество предъявляемых стимулов. Полученные значения используются в качестве корректирующих коэффициентов. Для набора статистики оператору предъявляются сигналы с разными частотами на фоне естественных шумов, программным методом фиксируется статистика правильного обнаружения, пропуска сигналов и ложного обнаружения. Для набора репрезентативной выборки проводится необходимое количество испытаний. Пример кривых представлен на Фиг.5 (верхняя кривая - вероятность правильного обнаружения в предлагаемом устройстве, нижняя кривая - вероятность правильного обнаружения в прототипе). Введение коэффициентов позволяет повысить статистические показатели правильного обнаружения сигналов на фоне естественных шумов. Корректирующие коэффициенты берутся как величины, обратные к вероятности правильного обнаружения для каждого масштаба, полученные в результате предварительного тестирования. Полученные значения корректирующих коэффициентов хранятся в буферном ЗУ 28 и с помощью мультиплексора 29 по команде из БАСАП 27 подаются в соответствующий канал масштабно-временных фильтров 23 (24).As a result of the tests, detection statistics are collected. $$\raisebox{1ex}{$x$}\!\left/ \!\raisebox{-1ex}{$y$}\right.$$

, where x - corresponds to the correct detection, y - the total number of presented stimuli. The obtained values are used as correction factors. For statistics, the operator is presented with signals with different frequencies against a background of natural noise, the software records the statistics of correct detection, skipping signals and false detection. For a set of representative samples, the required number of tests is performed. An example of the curves is presented in Figure 5 (the upper curve is the probability of correct detection in the proposed device, the lower curve is the probability of correct detection in the prototype). The introduction of coefficients allows to increase the statistical indicators of the correct detection of signals against the background of natural noise. Correction factors are taken as the reciprocal of the probability of correct detection for each scale, obtained as a result of preliminary testing. The obtained values of the correction factors are stored in the buffer memory 28 and, with the help of the multiplexer 29, are supplied to the corresponding channel of the time-scale filters 23 (24) by the command from the BASAP 27.

The received signals are converted into analog form using the DAC 10 (11) and, after amplification in blocks 12 (13), are transmitted to the head phones 14 (15).

The introduction of new compared to the prototype blocks: the first and second.

A set of 23, 24 time-scale filters, the first and second adders 25, 26, the analysis block 27 of the auditory analyzer of the user (BASAP), the third block 28 of the memory, the multiplexer 29 allows you to get a new positive effect - more fully transmit to the hearing analyzer the operator about the environment environment (increase noise immunity).

The authors conducted a preliminary assessment of the effectiveness of the proposed method. A group of operators in the amount of three people was randomly presented with signals against a background of interference.

Before listening, a time-scale conversion operation was performed with the introduction of time-scale coefficients at each scale, followed by weight summation. Used signal frequencies: 100, 200, 400, 800, 1600, 3200, 6400 Hz. The upper graph (figure 4) corresponds to the probability of correct detection of the presented implementations using the proposed operations. The lower graph is based on the processing implemented in the prototype method. It can be seen that the gain lies in the range of 0.5-2 dB.

We show an example of the execution of device blocks.

Signal generator 1, radiation path amplifier 2, transmitter 3,

ultrasonic transducers 4 (5), first and second memory units 8 (9), digital-analog transducers 10 (11), amplifiers 12 (13), headphones 14 (15), clock generator 16, frequency dividers 17 (18), switches 19 (20), the counter 21, the delay unit 22 can be implemented similarly to the prototype.

Multiplexer 29 (multiplex channel) is used for exchange

information between the memory block 28 and the time-scale filters 23 (24) and is known in digital technology [Drozdov EA, Komarnitsky VA, Pyatibratov AP Multiprogramming digital computers, M., Ed. USSR Ministry of Defense, p. 378-383].

BASAP 27 is made programmatically according to the block diagram of the algorithm (figure 5).

Adders 15, 16, the third memory block 28, which is a buffer memory, are known in digital technology (Shchegoleva L., Davydov A. Fundamentals of computer engineering and programming, L., Energoizdat, 1981, 155-201).

Type ADC 7, 8 is selected based on the required speed, determined by the frequency of the probe pulses. Since the maximum frequency used in such systems is 120 kHz, the necessary speed provides with a conversion time of less than 0.9 μs and with a clock frequency in the range of 0.4-1.5 MHz. You can use the QMbox 15-48 series ADCs with a clock frequency 1.5 MHz

Sets 23 (24) of time-scale filters that implement wavelet transform are known in modern radio engineering (see RF patent for invention No. 2246132, M.cl. G06F 17/14, published on July 20, 2004).

Claims (1)

A device for acoustic representation of spatial information for users, comprising a signal generator, a radiation path amplifier and a transmitter, a right-side ultrasonic transducer (USP), a first analog-to-digital converter (ADC), a first memory block, a first digital-to-analog converter (DAC), connected in series , the first amplifier and the right head-phone, connected in series to the second USP, the second ADC, the second memory unit, in series the second DAC, the second amplifier and the left headphone, the outputs of the right and left headphones are respectively the first and second outputs of the device, serially connected clock generator, first frequency divider, second frequency divider and delay unit, first and second switches and counter moreover, the second inputs of the first and second ADCs are connected respectively to the outputs of the first and second switches, the first inputs of which are connected to the output of the clock generator and the first counter input, the second inputs of the first and second switches are connected to the second counter input and the output of the delay unit, and the third inputs of the first and second switches are connected to the counter output, the output of the second frequency divider is connected to the input of the signal generator, the second inputs of the first and second DACs are connected to the output of the first frequency divider, characterized in that the first set of scale-time filters and the first adder are connected in series, and the second set of scales is connected in series nio-temporal filters and a second adder, the outputs of the first and second memory blocks respectively connected to the information inputs of the first and second sets of time-scale filters, and the outputs of the first and second adders connected to the first inputs of the first and second DACs, connected in series to the auditory analysis unit user analyzer (BASAP), the third memory block and multiplexer, the outputs of which are connected to the control inputs of the first and second sets of time-scale filters, the second output B SAP is coupled to a second input of the multiplexer, and BASAP input connected to the output of the generator clock pulses.

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