EA040542B1 - METHOD FOR NEUROSENSOROUS AUDIO STIMULATION AND DEVICE FOR ITS IMPLEMENTATION - Google Patents

Description

The invention relates to the processing of acoustic signals used in methods and means intended for the correction of hearing and speech, based on the principle of sensory sound stimulation of adults and children.

There is a known method of teaching languages, according to which, on the basis of the saying of the student in the language to be studied, the first signal containing the specified saying is formed, the received signal is processed, the processed signals are applied to the head and ears of the student (patent for invention No. 2101770, publ. 10.01.1998 ) At the stage of signal processing, a second signal is generated based on the first signal, the second signal is changed with respect to the first signal depending on the bandwidth and envelope of the speech signal corresponding to the language to be studied, the third signal is generated based on the first signal, and the specified third signal change with respect to the first signal depending on the frequency spectrum of the utterance and the characteristics of the language being studied - the bandwidth and envelope of the speech signal, produce a selective supply to the trainee of the second or third signal, depending on whether the amplitude of the first signal is respectively higher or lower than the specified th threshold value.

After the utterance is uttered, the learner selectively receives in response either the first sound message, which corresponds to his own utterance, modified according to predetermined parameters of the language being studied, in particular the frequencies of this language, or the second corrective sound message, which corresponds to his own utterance, but which modified in this case according to the coefficient of harmonics of the saying itself and the characteristics of the language being studied.

These two sound messages are transmitted to a person through the environment of his own bone structure and through the air environment.

To implement the proposed method, a device is proposed that contains a microphone for receiving a person's utterance in the language being studied and creating the first signal; a first processing circuit for the first signal to produce a second signal modified with respect to the first signal according to an envelope curve derived from preselected bandwidths of the target language, a second processing circuit for the first signal that generates a third signal modified in relation to the first signal according to the frequency spectrum of the utterance, and according to the characteristics of the language; means for selectively supplying the second or third signal to the trainee, depending on whether the amplitude of the first signal is lower or higher than the specified threshold. In addition, the device contains a control means in the form of buttons and adjustment knobs, a bone sound vibrator located on the human skin, preferably on the head, headphones.

The acoustic signal is processed by two multi-frequency equalizer processing circuits, which are controlled by controlled parameters, in particular using successive filters tuned to different frequencies in order to obtain an envelope curve that represents the language being studied, and to obtain an envelope curve that corresponds to the actual utterance of the student.

The known method is based on the contrast created between the voice output signal and the response audio signal that corresponds to this audio output signal.

To correct a person's hearing and speech impairment, different sound effects on different parts of the inner ear are necessary, i.e. exposure to an acoustic signal with different amplitude-frequency characteristics (hereinafter AFC). At the same time, it is effective to use sudden transitions from low frequencies to high frequencies to create a unique sound contrast. To do this, the acoustic signal is formed by a special set of frequency settings. The known device does not provide a system of settings and the formation of an acoustic signal that can provide the necessary impact on the muscles of the inner ear and, therefore, effective correction of hearing and speech impairment.

There is a known method for the treatment of sensory hearing impairments, accompanied by a deterioration in speech perception, which accelerates the treatment of sensory hearing impairments due to additional acoustic exposure with simultaneous electrotactile stimulation of the tongue (RF patent No. 2618164, publ. 04/29/2016). The method consists in the fact that the patient after improving operations on the ear, at the stage of rehabilitation, undergoes auditory training simultaneously with electrotactile stimulation of the tongue. Auditory training is listening to an audio signal against the background of interference. Listening is carried out through the headphones of a 2-channel clinical audiometer. Electrotactile stimulation of the tongue is performed using the intraoral display of the apparatus for vestibular rehabilitation. An audio signal is a fragment of speech, which is pre-read by the speaker and after digital processing by means of a personal computer software is recorded for further playback. The modified audio material in terms of frequency-spectral or other parameters is used as a useful signal (for example, a phonetic fragment of monosyllabic words) and as speech noise. The useful audio signal and the interference are sent through two different channels of the audiometer to one earphone for the possibility of their ipsilateral and simultaneous presentation. Adjustment of the level of the incoming audio signal for each of the channels is carried out simultaneously. The audio signal, like speech interference, is delivered at a level of 20 dB above the speech perception threshold, which is determined in advance at the examination stage. Before the procedure, the patient is instructed about the need for maximum concentration in order to understand the information heard. After determining the initial level of the applied signals, the level of both useful and interference is adjusted independently of each other. The disadvantage of this method is that the method uses a direct effect on the patient's language, based on sensations that may not be comfortable, as well as additional stress with a mandatory significant concentration in order to understand the meaning of the listened speech.

The objective of the present invention is to create a highly effective method of neurosensory auditory stimulation, based on auditory training using a filtered acoustic signal of the mother's voice and Mozart's music, for both adults and children. Listening to a filtered acoustic signal affects the ear canal and causes variable contraction and relaxation of the muscles of the stapes and malleus of the middle ear, affects the vestibular apparatus, improving both fine and gross motor skills, and also improves the perception of inverted speech and physical hearing. The device for implementing the method records the mother's voice through a microphone and generates a filtered acoustic signal through headphones and a bone sound vibrator.

The technical result consists in developing an individual algorithm for auditory training by exposing the middle ear and the vestibular apparatus of a person to a filtered acoustic signal, providing variable settings for its amplitude-frequency characteristics and applying additional settings matched to the generated signal with an individual temporal algorithm of exposure depending on the required level of hearing correction. .

The technical result in the method of neurosensory auditory stimulation is achieved by diagnosing the level of hearing loss. The threshold level of bone and air sound conduction is determined: bone conduction in the frequency range of 250-4000 Hz and in the intensity range up to 80 dB; air conduction in the frequency range 125-6000 Hz and in the intensity range up to 120 dB.

Based on the results obtained, a neurosensory auditory stimulation program is compiled, which includes at least two periods of exposure to air and bone conduction by acoustic signals with different spectral characteristics for at least 30 minutes. Rest breaks are set between periods. In the first period, neurosensory auditory stimulation is performed by affecting bone conduction with an acoustic signal, the spectral characteristics of which are provided by the settings of the timbres of high and low frequencies, and when the acoustic signal reaches a predetermined level after a predetermined time delay, the effect of this signal on air conduction is added. In the second period, an acoustic signal is exposed, the spectral characteristics of which are additionally limited by a digital filter with specified amplitude-frequency characteristics and a temporal algorithm for its application, while the tone level of the acoustic signal and the delay time of the effect on bone and air sound conduction are set again. The digital filter may have a frequency response of a low-pass filter and a high-pass filter, while the digital filter is turned on according to a given time algorithm. Then the hearing level is re-diagnosed, determining the threshold level of bone conduction in the frequency range of 250-4000 Hz and in the intensity range up to 80 dB and air sound conduction in the frequency range of 125-6000 Hz and the intensity range up to 120 dB. Based on the results obtained, a decision is made on the need for a repeated course of neurosensory auditory stimulation.

The level of timbres of the upper and lower frequencies of the acoustic signal, the magnitude of the delay in the effect on bone conduction, the amplitude-frequency characteristics of the digital filter for additional limitation of the acoustic signal and the time algorithm for its use are determined depending on the results of diagnosing the level of hearing loss.

The number of periods of exposure to an acoustic signal and rest pauses, as well as their sequence, the settings of the parameters of acoustic signals depends on the level of hearing and speech impairment of each person and is determined by a specialist. The determination of the threshold level of air conduction can be carried out, for example, with an Interacoustics AD226B polyclinic audiometer.

For each recorded audio track, the following characteristics change during playback:

1) relative loudness (0-90 dB);

2) frequency components of the spectrum (0-20000 Hz).

The method is based on a sharp change in the timbre of the sound - an audio signal that is amplified at low and attenuated at high frequencies is replaced by an amplified at high and attenuated at low frequencies, and vice versa.

To implement this function, a computer is used with a software algorithm that uses two parallel sound streams, the first one generates sound for the bone environment, the second - for the air. Each audio stream also has two channels of filters C1 and C2 with independent timbre settings. The processed sound of each of the streams (bone and air) at one moment in time

- 2 040542 neither is extracted either from channel C1 or from channel C2, depending on the relative volume of the sound and the set switching threshold. The switching threshold value is set by adjusting SWITCH (from 0 to 90 dB). The SWITCH software function continuously scans the relative loudness of the audio track at the time of playback and compares it with a predetermined threshold. If the volume of the reproduced signal is below the specified threshold, then the sound in each audio stream is extracted from channel C1, if it is higher, then the sound in each audio stream is extracted from channel C2.

The frequency response of the channel C1 filter is formed by the settings BASS1/TREBLE 1, the frequency response of the channel C2 filter is formed by the settings BASS2/TREBLE2 (Fig. 1, 2). For bone and air sound streams, these settings are common. The BASS setting determines the boost/cut of all frequencies in the spectrum below the frequency set by the FREQUENCY control, the boost/cut of all frequencies in the spectrum above are adjusted with the TREBLE setting.

The maximum speed of the volume transition (filter slope) between the BASS/TREBLE settings is set by the GRADIENT setting.

In FIG. 1 shows the frequency response of the C1 filter at BASS +5.0 settings; TREBLE-5.0; FREQUENCY - 1 kHz; GRADIENT - 18 dB / octave;

in fig. 2 shows the frequency response of the C2 filter at BASS -5.0 settings; TREBLE +5.0; FREQUENCY 1 kHz; GRADIENT - 18 dB / octave;

in fig. 3 shows the function SWITCH, DELAY, PRECESSION;

in fig. 4 is a block diagram of the device;

in fig. 5 shows the control panel of the device;

in fig. 6 shows the front panel of the device case with control panel elements.

The DELAY setting delays the SWITCH function for each audio stream from 0 to 1020ms. In this case, the delay occurs only when switching the sound reading from the filter of channel C1 to the filter of channel C2 (no delay is set in the reverse transition).

The PRECESSION setting specifies an additional delay for the SWITCH function in the airstream. This function allows you to prepare the ear for a sharp change in tone in the air channel, which, in turn, softens the perception of sound load.

The sound volume for the bone channel is set by the BONE setting, for the air channel by the AIR setting. The overall volume of the track being played is set by the VOLUME setting.

The FILTER button turns on / off an additional software filter with strong frequency suppression (at least 40 dB / octave) below / above the specified one in accordance with the table._______

Selected value 0 start frequency (pos. 4) 0 Hz end frequency (pos.3) 0 Hz cycle time (pos. 2) 30 sec (pos. 1) - operating mode Rise/fall/cycle 1 500 Hz 500 Hz 1 min Rise/fall/single 2 750 Hz 750 Hz 2 minutes Fall/rise/cycle 3 1000 Hz 1000 Hz 3 min Decay/rise/single 4 1500 Hz 1500 Hz 4 min Rise Only 5 2000 Hz 2000 Hz 8 min Only recession 6 3000 Hz 3000 Hz 16 min - 7 4000 Hz 4000 Hz 32 min Continuous, HF only 8 6000 Hz 6000 Hz - Continuous, LF only 9 8000 Hz 8000 Hz - -

The FILTER function allows, when suppressing frequencies below 4000-8000 Hz, to imitate the sound spectrum of the mother's intrauterine voice.

Description of settings.

MICRO - adjustment of the amplitude of the input microphone signal from 0 to 100%. VOLUME - adjustment of the amplitude of the input linear signal from 0 to 100%.

DELAY - adjustment of the delay in the operation of the SWITCH function. The delay range is from 0 to 1020 ms.

PRECESSION - adjusts the switching delay between channels C1, C2 for the air sound stream, when the SWITCH function is triggered. The delay range is from 0 to 2550 ms.

SWITCH - adjustment of the SWITCH function activation threshold. Adjustment range from 0 to 90 dB (from 0 to 100%).

BASS Channel 1 - adjusts the amplitude of all lower frequencies of channel C1 relative to the set FREQUENCY control. Adjustment range from -25 to +25 dB.

BASS Channel 2 - adjustment of the amplitude of all the lower frequencies of channel C2 relative to the set FREQUENCY control. Adjustment range from -25 to +25 dB.

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TREBLE Channel 1 - adjustment of the amplitude of all high frequencies of channel C1 relative to the set FREQUENCY control. Adjustment range from -25 to +25 dB.

TREBLE Channel 2 - adjustment of the amplitude of all high frequencies of channel C1 relative to the set FREQUENCY control. Adjustment range from -25 to +25 dB.

LEFT+RIGHT - balance adjustment of the left and right channels of air conduction. Adjustment range from 0 to 100% for each channel.

AIR - adjustment of the amplitude of the output signal of the air sound stream. Adjustment range from 0 to 100%.

BONE - adjustment of the amplitude of the output signal of the bone sound stream. Adjustment range from 0 to 100%.

POWER - enable / disable the device.

GRADIENT - adjustment of the slope of the frequency response of the filters of channels C1 and C2. Three fixed settings - 6, 12, 18 dB/octave.

FREQUENCY - frequency adjustment, relative to which the frequency response of channels C1 and C2 is adjusted. Six fixed frequencies - 500, 750, 1000, 1500, 2000, 3000 Hz.

FILTER - enable/disable the preset program of frequency filters.

- Select a preset program. The adjustment range is from 0 to 9.

- selection of switching time. Adjustment range from 0 to 9, allows you to select the switching time in the range from 0 to 32 min.

- choice of final frequency. The adjustment range is from 0 to 9, allows you to select the end frequency in the range from 0 to 8000 Hz.

- choice of initial frequency. Adjustment range from 0 to 9. Adjustment range from 0 to 9, allows you to select the starting frequency in the range from 0 to 8000 Hz.

The method of neurosensory auditory stimulation is carried out as follows.

Example 1

An adult complained of hearing loss.

According to the diagnostic examination, on the outpatient audiometer Interacoustics AD226B, bone conduction at frequencies: 250, 500, 750, 1000, 1500, 2000, 3000 and 4000 Hz and air conduction at frequencies: 125, 250, 500, 750, 1000, 1500, 2000, 3000, 4000, 6000 Hz defined:

Threshold level of air conduction of the left ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 45 thirty 25 20 35 thirty thirty 15 10 thirty Norm 20 10 5 0 -5 -10 -15 -10 -5 0 Threshold level of air conduction of the right ear Frequency range in Hz Before the course (dB) 125 20 250 10 500 5 750 5 1000 0 1500 0 2000 0 3000 5 4000 5 6000 5 Norm 20 10 5 1 -5 -10 -15 -10 -5 0 Threshold level of bone conduction of the left ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) -5 -10 -5 0 -10 -10 -10 -10 Norm 20 15 10 5 0 0 5 5 Threshold level of bone conduction of the right ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 5 -5 -5 -10 -10 -10 -5 -5 Norm 20 15 10 5 0 0 5 5

Hearing rehabilitation started.

According to the proposed method, a course of neurosensory auditory stimulation was carried out for 12 days, lasting 54 hours, daily, with periodic exposure to acoustic signals with different amplitude-frequency characteristics.

In the first period, exposure was carried out for 30 minutes with an acoustic signal (a fragment of Mozart's music), with the level of low-frequency timbres, respectively (+5) - (-5) and high frequencies

-4040542 (-5)-(+5) at input frequency (FREQUENCY) 1000 Hz, time delay for air conduction channel switching (PRECESSION) 450 ms and channel switch threshold (SWITCH) 12 dB.____________________________________________________________

duration track Bass low Bass high Frequency Gradient Precession switch 30 minutes MNF +5/-5 -5/+5 1000 6 450 12

In the second period, for the next 30 minutes, an acoustic signal (a fragment of Mozart's music) was exposed to the same spectral characteristics with additional continuous filtering by a digital high-pass filter._____________________

duration track Basslow Bass high Frequency Gradient Precession switch ProgramFilter 30 minutes MNF +5/-5 -5/+5 1000 6 450 1.5 7007

In the third period, the exposure to the same acoustic signal was carried out for 60 minutes without additional filtration with a channel response threshold of 15.2 dB._____________

duration track Bass low Bass high Frequency Gradient Precession switch 60 minutes MNF +5/-5 -5/+5 1000 6 450 15.2

After exposure - a minimum rest pause for 120 minutes. Next, exposure to an acoustic signal was carried out according to the program of the first three periods.

At the end of the course of neurosensory auditory stimulation, a second diagnostic examination was carried out, using an Interacoustics AD226B polyclinic audiometer, for bone conduction at frequencies: 250, 500, 750, 1000, 1500, 2000, 3000 and 4000 Hz and for air conduction at frequencies: 125, 250, 500, 750, 1000, 1500, 2000, 3000, 4000, 6000 Hz.

Results:______________________________________________________________________

Threshold level of air conduction of the left ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 45 thirty 25 20 35 thirty thirty 15 10 thirty Graduation(dB) thirty 25 25 15 15 15 15 10 5 15 Norm 20 10 5 0 -5 -10 -15 -10 -5 0 Hearing improvement in dB 15 5 0 5 20 15 15 5 5 15 Threshold level of air conduction of the right ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 20 10 5 5 0 0 0 5 5 5 Graduation(dB) 15 5 15 -10 -10 -10 -10 -10 0 5 Norm 20 10 5 1 -5 -10 -15 -10 -5 0 Hearing improvement in dB 5 5 -10 15 10 10 10 15 5 0 Threshold level of bone conduction of the left ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) -5 -10 -5 0 -10 -10 -10 -10 Graduation(dB) -5 -5 0 5 -5 -5 -5 -5 Norm 20 15 10 5 0 0 5 5 Hearing improvement in dB 0 -5 -5 -5 -5 -5 -5 -5 Threshold level of bone conduction of the right ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the start of the course (dB) 5 -5 -5 -10 -10 -10 -5 -5 Graduation(dB) 10 5 0 5 -5 -5 -5 0 Norm 20 15 10 5 0 0 5 5 Hearing improvement in db -5 -10 -5 -15 -5 -5 0 -5

Example 2

A 6-year-old child with residual organic lesion of the central nervous system complained of underdeveloped fine and gross motor skills and understanding of addressed speech.

According to the diagnostic examination, on the outpatient audiometer Interacoustics

AD226B Bone Conduction at 250, 500, 750, 1000, 1500, 2000, 3000 and 4000 Hz and Air Conduction at 125, 250, 500, 750, 1000, 1500, 2000, 3000, 4000, 6000 Hz defined:

Threshold level of air conduction of the left ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 40 20 20 20 15 10 15 10 5 10 Norm 20 10 5 0 -5 -10 -15 -10 -5 0 Threshold level of air conduction of the right ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 35 thirty 25 15 15 10 10 5 5 5 Norm 20 10 5 1 -5 -10 -15 -10 -5 0 Threshold level of bone conduction of the left ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 5 5 0 10 0 5 5 0 Norm 20 15 10 5 0 0 5 5 Threshold level of bone conduction of the right ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 20 20 20 15 5 5 5 10 Norm 20 15 10 5 0 0 5 5

According to the proposed method, a course of neurosensory auditory stimulation was carried out for 12 days, lasting 54 hours, daily, periodic exposure to acoustic signals with different amplitude-frequency characteristics. Sound stimulation of phonemic and physical hearing was carried out by exposure to both high and low frequencies with an emphasis on improving the vestibular apparatus.

In the first period, exposure was carried out for 30 minutes with an acoustic signal (a fragment of Mozart's music) with the level of timbres: low frequencies (+3.5) - (-1.5) and high frequencies (-1.5) - (+3.5 ). With an input frequency (FREQUENCY) of 750 Hz and a filter slope (GRADIENT) of 6 dB, with a time delay for switching the air conduction channel (PRECESSION) of 450 ms and a channel switch threshold (SWITCH) of 12 dB. _________________________________

duration track Bass low Bass high Frequency Gradient Precession switch 30 minutes MNF +3.5/-1.5 -1.5/+3.5 750 6 450 12

In the second period, for the next 60 minutes, exposure to the VMF acoustic signal (recording of the mother's voice of the child) was carried out, with the same spectral characteristics with additional continuous filtering by a digital high-pass filter. ______

duration track Bass low Bass-high Frequency Gradient Precession switch ProgramFilter 60 minutes VMF +3.5/-1.5 -1.5/+3.5 750 6 450 12 7007

In the third period, an acoustic signal MD (a fragment of Mozart's music) was exposed to the level of low frequency timbres (-1.5) - (+3.5) and high frequencies (+3.5) - (-1.5), respectively, with incoming frequency (FREQUENCY) 1500 Hz and filter slope (GRADIENT) 12 dB, with a time delay for switching the air conduction channel (PRECESSION) 250 ms, without additional filtering, with a channel response threshold (SWITCH) 15.2 dB.

Duration min

track

MD

Bass low

-1.5/+3.5

Bass high +3.5/-1.5

Frequency

1500

Gradient

Precession

250

switch

15.2

After exposure - a minimum rest pause for 120 minutes. Next, exposure to an acoustic signal was carried out according to the program of the first three periods. At the end of the course of neurosensory auditory stimulation, a second diagnostic examination was carried out, using an Interacoustics AD226B polyclinic audiometer, for bone conduction at frequencies: 250, 500, 750, 1000, 1500, 2000, 3000 and 4000 Hz and for air conduction at frequencies: 125, 250, 500, 750, 1000, 1500, 2000, 3000, 4000,

- 6 040542

6000 Hz.

Threshold level of air conduction of the right ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 40 20 20 20 15 10 15 10 5 10 At the end of the course (dB) Norm thirty 20 10 10 15 5 15 0 5 -5 0 -10 10 -15 5 -10 5 -5 5 0 Hearing improvement in dB 10 10 5 5 10 10 5 5 0 5 Threshold level of air conduction of the right ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 35 thirty 25 15 15 10 10 5 5 5 Graduation(dB) thirty 15 15 10 5 -5 0 0 0 5 Norm 20 10 5 0 -5 -10 -15 -10 -5 0 Hearing improvement in dB 5 15 10 5 10 15 10 5 5 0 Threshold level of bone conduction of the left ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 5 5 0 10 0 5 5 0 Graduation(dB) 10 15 10 5 0 0 5 0 Norm 20 15 10 5 0 0 5 5 Hearing improvement in dB -5 -10 -10 5 0 5 0 0 Threshold level of bone conduction of the right ear Frequency range in Hz 125 250 500 750 1000 1500 2000 3000 4000 6000 Before the course (dB) 10 20 20 15 5 5 5 10 Graduation(dB) 15 15 10 5 0 0 5 0 Norm 20 15 10 5 0 0 5 5 Hearing improvement in dB -5 5 10 10 5 5 0 5

The device for implementing the method of neurosensory auditory stimulation is based on a computer and contains a digital drive 2 installed in one housing 1 (contains a digital sound signal 3, a database of filters 4, software 5), the output of which is connected to the input 6 of the processor. The processor 7 generates two streams for the audio signal (bone and air) with digital filters, using the software algorithm 5. The outputs of the processor 8 are connected to the inputs of digital-to-analog converters (hereinafter referred to as the DAC) 9 and the monitor 10. The input of the processor 6 is also connected to the microcontroller 11 installed in the control panel 12. The outputs of digital-to-analog converters 9 are connected to air sound emitters 13 and through an amplifier 14 to a bone sound emitter 15. Headphones can be used as air sound emitters, a bone sound vibrator can be used as a bone sound emitter.

The elements (knobs and buttons) of the control panel 12 are mounted on the front panel of the cabinet 1 and include the MICRO knob 16, the VOLUME knob 17, the DELAY knob 18, the PRECESSION knob 19, the SWITCH knob 20, the BASS1 knob 21, the BASS2 knob 22, the TREBLE1 knob 23 , TREBLE2 knob 24, LEFT+RIGHT knob 25, AIR knob 26, BONE knob 27, FILTER button 28, button 1-29, button 2-30, button 3-31, button 4-32, FREQUENCY button 33, GRADIENT button 34.

The microcontroller 11 reads from the elements (16-34) of the control panel 12 the values of the set audio processing parameters (frequency, filter slope, amplitude of the low and high frequency ranges) with a certain determinism, which limits the maximum number of possible combinations

- 7 040542 these parameters, but at the same time sufficient for full-fledged use in music therapy. The data read by the microcontroller 11 is transmitted to the processor 7, in which, for each combination of the set values of the audio processing of the control panel 12, strictly defined values of the filter coefficients from the database 4 of the drive 2, obtained at the stage of designing (programming) filters by the method of individual selection, are linked. The filter coefficients from the database 4 are individually selected so that the resulting filter changes the spectrum of the output signal with high accuracy and in accordance with the required shape (Fig. 1, 2). The coefficients for the filter are taken strictly from the database 4 when the processor 7 compares the values set on the control panel 12.

To identify the exact values of the filter coefficients at the design (programming) stage, audio signals were repeatedly passed through them, the spectra of which contained the entire set of audio frequencies (from 20 to 20,000 Hz) with the same amplitude and with the most accurate match of the output signal spectrum with the expected result, the values of the coefficients entered into database 4.

In the processor 7, a virtual audio stream is formed from the digital audio signal 3 based on the software algorithm 5. The stream from the input of the processor 6 goes to the VOLUME 35 block, where the volume is adjusted in accordance with the set value of element 17 of the control panel 12. From the VOLUME 35 block, the stream goes to the FILTER 36 block, in which the filter is formed in accordance with the selected program on the control panel 12 ( elements 28-32, Fig. 5). The processed sound stream from the FILTER 36 block is divided into two sound streams - bone and air.

In each audio stream, the signal is fed simultaneously to the SWITCH 37 block, the C1 channel block 38 and the C2 channel block 39. In the C1 and C2 channel blocks, filters are formed in accordance with the set values of the elements (elements 21-24, 33, 34) of the control panel 12. The SWITCH 37 block scans the relative loudness of the sound stream and compares it with the value of the set threshold of the element 20 of the control panel 12. If the volume of the sound stream is below the threshold value, then the sound is transmitted to the processor 8 output from the C1 channel block 38, otherwise from the C2 channel block 39.

When the sound volume in each stream reaches the threshold value, the reading switching from the C1 channel block 38 to the C2 channel block 39 occurs with a delay specified by the DELAY 40 block. The delay time is set by element 18 of the control panel 12. In the air channel, the PRECESSION 41 block introduces an additional switching delay reading from channel block C1 38 to channel block C2 39, the delay time is set by element 19 of the control panel 12.

After processing by filters C1/C2, the signals are sent to the output of the processor 8. The air sound stream from the output of the processor 8 is converted by means of a digital-to-analogue converter 9 into an analog form and enters the air sound emitters 13. The bone sound stream is similarly converted from the output of the processor 8 by means of digital-to-analogue converter 9 into analog form and through the amplifier 14 enters the bone sound emitter 15.

List of positions:

- body of the device;

- digital storage;

- digital sound signal;

- database of filter coefficients of the digital-to-analog converter;

- software;

- processor input;

- processor;

- processor output;

- digital-to-analog converter;

- monitor;

- microcontroller;

- control Panel;

- air sound emitter;

- amplifier;

- MICro handle;

- VOLUME knob;

- DELAY knob;

- Precession knob;

- switch handle;

- BASS 1 knob;

- BASS 2 knob;

- TREBLE 1 knob;

- TREBLE 2 handle;

- LEFT+RIGHT knob;

-

Claims (4)

26 - handle AIR; 27 - handle BONE; 28 - FILTER button; 29 - button 1; 30 - button 2; 31 - button 3; 32 - button 4; 33 - FREQUENCY button; 34 - GRADIENT button; 35 - shred VOLUME; 36 - FILTER block; 37 - block SWITCH; 38 - channel block C1; 39 - channel block C2; 40 - block DELAY; 41 - PRECESSION block. CLAIM 1. A method for neurosensory auditory stimulation, which consists in diagnosing the level of hearing impairment by determining the threshold level of bone conduction in the frequency range of 250-4000 Hz and in the intensity range up to 80 dB and air conduction in the frequency range of 125-6000 Hz and in the range intensity up to 120 dB, on the basis of the results obtained, a neurosensory auditory stimulation program is compiled, including at least two periods of exposure to air and bone conduction by acoustic signals with different spectral characteristics for at least 30 minutes, and rest pauses are set between periods; in the first period, neurosensory auditory stimulation is carried out by affecting bone conduction with an acoustic signal, the spectral characteristics of which are provided by adjusting the timbres of high and low frequencies, and when the acoustic signal reaches a predetermined level after a predetermined time delay, the effect of this signal on air conduction is added; in the second period, an acoustic signal is exposed, the spectral characteristics of which are additionally limited by a digital filter with specified amplitude-frequency characteristics and a temporal algorithm for its application, while the tone level of the acoustic signal and the delay time of the effect on bone and air conduction are set again, the digital filter has an amplitude- the frequency response of the low-pass filter, high-pass filter, while the digital filter is turned on according to a given time algorithm; re-diagnosing the level of hearing impairment, determining the threshold level of bone conduction in the intensity range up to 80 dB and air conduction in the intensity range up to 120 dB, based on the results obtained, a decision is made on the need for a repeated course of neurosensory sound stimulation. 2. The method of neurosensory auditory stimulation according to claim 1, characterized in that the level of the timbres of the upper and lower frequencies of the acoustic signal, the amount of delay in the effect on bone and air conduction, the amplitude-frequency characteristics of the digital filter for additional limitation of the acoustic signal and the time algorithm for its use are determined depending on the results of the diagnosis of the level of hearing loss. 3. The device for implementing the method of neurosensory auditory stimulation according to claim 1 contains an audio signal source installed in one housing, the output of which is connected to a processor controlled by a software algorithm that generates two parallel sound streams, one of which is intended for the bone environment, the other for the air , while each of these streams has two channels of digital filters, the outputs of which are connected to the input of a digital-to-analog converter having amplitude-frequency characteristics and switched on according to a given time algorithm, and a monitor, the input of digital filters is connected to a microcontroller connected to the control panel, and the input - output - with a storage device, while the outputs of the digital-to-analog converter are connected to an air sound emitter and through an amplifier with a bone sound emitter, digital filters have independent timbre settings, while depending on the relative sound volume and the set switching threshold The channels of the digital filters in each audio stream are designed to extract the reproduced audio signal, one of the channels is used to extract the audio signal, the volume of which is below a given threshold, the other - if the volume of the audio signal is above the threshold value. 4. A device for implementing the method according to claim 1, characterized in that headphones are used as air sound emitters, and the bone sound emitter is equipped with a contact vibrator. -