KR100671292B1 - Method for compensating a hearing impairment with multi-channel of digital a hearing aid and a hearing aid using the same - Google Patents

Abstract

Disclosed are a multi-channel deafness compensation method for finely adjusting the frequency characteristics of deafness that occur differently to a deaf person using a hearing aid for each channel, and a digital hearing aid using the same. The present invention comprises the steps of: converting an external voice signal into a digital signal; Dividing the digital signal into multiple channels of low, mid, and high frequencies based on a frequency band and outputting the digital signal to each channel; Separating the high band portion and the low band portion of the digital signal of the signal input to each channel, respectively; Receiving the separated high band portion and the low band portion, calculating a hearing loss compensation gain of the corresponding band, and accumulating and outputting them; And converting the data for which the hearing loss is compensated using the hearing loss compensation gain into an analog signal, and then converting the data into an audio signal and outputting the same.

Description

Method for compensating a hearing impairment with multi-channel of digital a hearing aid and a hearing aid using the same

1 is a block diagram showing the structure of a multi-channel digital hearing aid according to the present invention.

FIG. 2 is a block diagram illustrating a structure of a gain adjuster formed in the multichannel digital hearing aid of FIG. 1.

FIG. 3 is a block diagram illustrating a structure of a subband hearing loss compensation filter unit in FIG. 2.

FIG. 4 is a graph illustrating a waveform of a signal passing through a subband filter unit of the hearing loss compensation filter unit of FIG. 3.

FIG. 5 is a graph showing the audible threshold values of the hearing impaired and the normal in order to explain the operation of the compensation filter in the multi-channel digital hearing aid according to the present invention.

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6 is a graph for comparing compensation signals of a multi-channel digital hearing aid according to the present invention and a digital hearing aid using a conventional multiband compression scheme.

<Explanation of symbols for main parts of the drawings>

100A, 110B: Microphone 120A, 120B: A / D Converter

       130: voice input section 140A ~ 140N: channel section

       150A ~ 150N: Gain control part (AGC) 160: Signal synthesis part

       170: D / A converter 180: speaker

       152: subband feedback removal unit 154: subband deafness compensation filter unit

 201A, 201B, 211A, 211B: Frequency Spectrum

       Subband filter: 202A, 202B, 204A, 204B, 212A, 212B, 212C, 212D

       203A, 203B: Compensation Filter

       205A, 205B: Order Limit Filter

The present invention relates to a multi-channel hearing loss compensation method of a digital hearing aid and a digital hearing aid using the same, in particular a multi-channel hearing loss compensation method for finely adjusting the frequency characteristics of the hearing loss that occurs differently for the hearing loss using the hearing aid, and the same It relates to a digital hearing aid used.

In general, the human audible range is from 20Hz to 20KHz, but the conversational range is usually from 250Hz to 4000Hz. However, the high-frequency band of the hearing-sensing cells is located at the upper part of the auditory canal, and the neighboring auditory cells are observed to be denatured easily in the case of senile hearing loss or noise-induced hearing loss. Ramen is harder to hear than other bands and can be accelerated by constant exposure to high frequencies.

 Hearing aid, a tool used to compensate for hearing impairment caused by the loss of hearing ability, transforms the acoustic signal input into the hearing organ of a hearing impaired person, and as a result, the degree of recognition by the brain The goal is to be the same.

Currently commercially available hearing aids can be classified into three types: analog, digital, and mixed analog / digital types. Analog hearing aids are limited in the number of bands because the analog circuit is not flexible, reliable, or easy to adjust. It was forced to implement basic functions such as compression or amplification.

Accordingly, digital hearing aids incorporating digital circuits have been developed, and development of digital signal processing algorithms required for digital hearing aids has been continued. Digital hearing aids can easily realize complex high-performance signal processing algorithms, as well as the advantages of flexibility and reliability of circuits, and can effectively implement high-performance hearing loss compensation algorithms, such as nonlinear correction methods for patients with aural hearing loss.

Recently, multi-channel hearing aids have been developed to separately control the input sound by classifying the sound input into the hearing aid into low-, mid-, and high-pitched sounds. In the case of a multi-channel hearing aid, the input sound can be adjusted according to the characteristics of the hearing loss, so it is obvious that the technology is an advanced technology compared to the conventional hearing aid.

However, in the conventional multi-channel hearing aid, multiband compression technology, that is, the sound input from the various channels of the hearing aid is compressed into the same frequency band, and compensated for and separated from the receiver, so that interference between signals input during multiband compression occurs. In the hearing aid, even if it is below the hearing threshold, a so-called flattening characteristic that compensates by giving a constant gain over the entire band is compensated. A problem occurs.

Therefore, the multi-channel hearing aid using the conventional multi-band compression technology has a problem that the intelligibility is lowered in the reproduction of sound quality, and thus the speech discrimination power is lowered.

The present invention has been invented to solve the above problems, and in the case of the multi-channel hearing aid according to the present invention, the users of the hearing aid effectively compensate for this in consideration of the fact that gain loss values for frequencies vary according to the type and degree of hearing loss. By designing a subband deafness compensation filter to compensate for the sound input to the hearing aid for each channel to improve unnecessary compensation due to interference between input signals and frequency distortion of the input signal, the user of the hearing aid is natural and clear It is a first object of the present invention to provide a multi-channel hearing loss compensation method of a digital hearing aid for audible listening.

A second object of the present invention is to provide a digital hearing aid employing the multi-channel hearing loss compensation method of the digital hearing aid.

The present invention for achieving the first object

Converting an external voice signal into a digital signal;

Dividing the digital signal into multiple channels of low, mid, and high frequencies based on a frequency band and outputting the digital signal to each channel;

Separating the high band portion and the low band portion of the digital signal of the signal input to each channel, respectively;

Receiving the separated high band portion and the low band portion, calculating a hearing loss compensation gain of the corresponding band, and accumulating the outputs; And

Converting the data for which hearing loss is compensated using a hearing loss compensation gain into an analog signal, and then converting the data into an audio signal and outputting the same.

Digital hearing aid for performing the second purpose of the present invention,

Two microphones for receiving a voice signal;

An A / D converter for converting an analog signal input through a microphone into a digital signal;

A voice input unit for dividing and outputting each channel such as a bass, middle and treble according to the frequency of the voice signal converted into a digital signal through an A / D converter;

Subband for controlling the gain by being connected to the rear end of each channel section for receiving digital signals outputted from the voice input section respectively, separating the signals inputted into the channel section into a high band portion and a low band portion and feeding back a predetermined number of times. A gain adjuster comprising a feedback remover and a subband deafness compensation filter;

A signal synthesizing unit for synthesizing a gain adjusted signal through the gain adjusting unit; And

It is connected to the output side of the signal synthesizing unit and includes a D / A converter for converting a digital signal into an analog signal and a speaker for amplifying and outputting the analog signal.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings.

FIG. 1 is a block diagram showing the structure of a multichannel digital hearing aid according to the present invention, and FIG. 2 is a block diagram showing the structure of a gain control unit formed in the multichannel digital hearing aid of FIG.

FIG. 3 is a block diagram illustrating a structure of a subband hearing loss compensation filter unit of FIG. 2, and FIG. 4 is a graph illustrating a waveform of a signal passing through the subband filter unit of the hearing loss compensation filter unit of FIG. 3. .

5 is a graph showing the audible threshold values of the hearing impaired person and the normal person in order to explain the operation of the compensation filter in the multi-channel digital hearing aid according to the present invention, Figure 6 is a multi-channel digital hearing aid and a conventional multiple This is a graph for comparing the compensation signal of digital hearing aid using band compression method.

As shown in FIG. 1, the multi-channel digital hearing aid according to the present invention has two microphones 110A and 110B for receiving sound. The microphones 110A and 110B of the multi-channel digital hearing aid according to the present invention are configured in two dual manners in order to expand the directivity characteristic for the sound, thereby expanding the directivity characteristic for the sound from 180 degrees to 360 degrees in the related art. Improve speech recognition of hearing loss patients.

A / D converters 120A and 120B are formed at the output terminals of the two microphones 110A and 110B to convert analog signals input through the microphones 110A and 110B into digital signals. The voice signal converted into a digital signal through the A / D converters 120A and 120B is input to the voice input unit 130. The voice input unit 130 is divided into a bass, a middle, a treble, and the like according to the frequency size of the voice signal converted into a digital signal and inputs to each channel unit 140A to 140N. It is possible to form a plurality of channels in the channel parts 140A to 140N according to the level of the input frequency, such as bass, mid-range treble, fine tuning of the input voice signal, and gain for removing sound distortion. Adjusting units (AGC) 150A to 150N are formed respectively.

The gain-adjusted signal through the gain adjusting unit 150A to 150N is synthesized through the signal synthesizing unit 160, converted into an original analog signal through the D / A converter 170, and then amplified by the speaker 180. It is formed to be.

Here, the multi-channel digital hearing aid according to the present invention finely gains a hearing loss value input to each channel unit 140A to 140N, which is an ideal value of hearing loss compensation, through the gain control unit 150A to 150N. It is designed to be used by both patients with deafness and neurodeafness by remarkably eliminating negative distortion.

This effect can be achieved by the operations of the gain adjusting units 150A to 150N formed in the channel units 140A to 140N, respectively. The gain adjusting units 150A to 150N are provided with the subband feedback removing unit 152 and the sub as shown in FIG. The band deafness compensation filter unit 154 is configured.

Hereinafter, the configuration and operation of the gain control unit according to the present invention will be described with reference to FIG.

As shown in FIG. 2, the gain adjusting units 150A to 150N respectively formed at the channel parts 140A to 140N include a subband feedback removing unit 152 and a subband hearing loss compensation filter 154. 1 and 2, the analog audio signals input from the two microphones 110A and 110B are converted into digital signals 141 through the A / D converters 120A and 120B, and then through the voice input unit 130. The digital signals (D) 141 input to the channel parts 140A to 140N are input to the channel parts 140A to 140N, and the subband feedback cancellation unit 152 and the subband hearing loss compensation filter unit 154 are provided. Are input to each.

The digital signal D inputted to the subband feedback removing unit 152 includes frequency spectrums 211A and 211B and respective subband filters 212A and 212B and 212C and 212D connected to the frequency spectrums 211A and 211B. A specific frequency band, that is, an audible frequency band, is selected from the digital signals input through the signal. The selected frequency band is input to the feedback cancellers 213A, 213B, 213C, and 213D to remove components such as noise included in the inputted digital signal using an orthogonal subband adjacent projection algorithm (OSAP). That is, the subband feedback cancellation unit 152 of the present invention uses a known subband structured adaptive filter (OSAP) for fast removal of noise, etc., generated in the faulty path.

In addition, the A / D-converted digital signal D in FIG. 2 passes through the filter S and is input to the subband hearing loss compensation filter unit 154 as shown in FIG. 3. The input digital signal d (n) passes through the frequency spectrum 201A and 201B and the subband filters 202A and 202B to select only a specific frequency band, that is, an audible frequency band, from among the input digital signals. The high band portion of the input digital signal d (n) is formed to pass through the lower frequency spectrum 201B, and the low band portion of the input digital signal d (n) is the upper frequency spectrum 201A. It is formed to pass through.

Here, the frequency spectrum of the frequency bands d ₀ (n) and d ₁ (n) selected through the frequency spectrums 201A and 201B and the subband filters 202A and 202B is shown in FIG. 4. In other words, since the shape of the spectrum changes in the case of the high band as shown in FIG. 4 (b), the output signal of the compensation filters 203A and 203B is considered in the following compensation filter 203A, 203B. As shown in Equation 1, the frequency band, magnitude, and phase specification are determined.

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................식 1

Equation 1

은 주파수별로 얻고자 하는 크기에 관한 규격이며,

은 규격이 주어지는 주파수 이며,

은 주파수 별로 얻고자 하는 위상 규격을 나타낸다. 본 발명에 따른 보상필터(203A, 203B)에서 각각 주파수의 규격 및 위상을 제어하는 것은 도 5에서 보는 바와 같이 난청인과 정상인의 가청주파수의 한계값에 차이를 보이기 때문이며, 이러한 가청주파수의 한계값이 본 발명에 따른 보상필터(203A, 203B)의 주파수 대역, 크기 및 위상의 규격을 결정하는 것이다. In Equation 1, N is the specification of a given frequency band,

Is the specification of the size to be obtained for each frequency.

Is the frequency at which the specification is given,

Denotes a phase specification to be obtained for each frequency. In the compensation filters 203A and 203B according to the present invention, the specification and phase of the frequency are respectively controlled because the difference between the threshold values of the hearing loss of the hearing impaired and the normal is different as shown in FIG. 5. The specification of the frequency band, magnitude and phase of the compensation filters 203A and 203B according to the present invention is determined.

For example, since the cause of hearing loss of each deaf person is different, and the audible frequency band according to the cause of deafness is different, it is analyzed to determine the specification of the frequency band, magnitude, and phase of the compensation filters 203A and 203B according to Equation 1. will be.

Referring to FIG. 3 again, the frequency band, magnitude and phase specifications to be compensated for the frequency bands d ₀ (n) and d ₁ (n) of the signal input from the compensation filters 203A and 203B are determined and then After the compensation, the compensated signal is input to the subband filters 204A and 204B to remove the frequency band outside the audible frequency band, and the order limit filter for limiting the output frequency of the compensation values of the compensation filters 203A and 203B. When the compensation is performed more than a predetermined number of times at 205A and 206B, the accumulated compensation signal is output through the signal synthesizing unit 160 at the rear end of the order limiting filter 205A and 206B.

The overall operation of the digital hearing aid according to the present invention will be described with reference to FIG. 2. The voice signal input through the microphones 110A and 110B is converted into a digital signal through the A / D converters 120A and 120B, and the voice input unit In step 130, the sound is divided into bass, middle, and treble according to the frequency of the voice signal and inputs to the respective channel parts 140A to 140N.

The signals input to the respective channel units 140A to 140N are input to the gain adjusting units 150A to 150N to form the subband feedback cancellation unit 152 and the subband deafness compensation filter constituting the gain adjusting units 150A to 150N. Compensation is performed by the operation of the unit 154. That is, the subband hearing loss compensation filter unit 154 divides the high band portion and the low band portion of the input digital signal d (n) and performs a predetermined order (number of times) by the operation of the compensation filters 203A and 203B. Compensation is performed by converting the abnormal digital signal, and the subband feedback cancellation unit 152 performs a rapid removal of noise generated in the track path using an adaptive filter (OSAP) having a subband structure.

The digital hearing aid according to the present invention allows the conventional digital hearing aid to perform the compensation in a manner of giving a constant gain over the entire band, thereby limiting the compensation order (number of times) of the compensation filter. By using the system identification technique, the low frequency band is divided based on the frequency band, inputted to each channel, and the signal inputted to each channel is also divided into a high band and a low band to perform compensation by the above-described method. Lower order filters have equivalent performance.

Comparing the compensation signal of the multi-channel digital hearing aid according to the present invention and the digital hearing aid using the conventional multi-band compression method, as shown in FIG. 6, the digital hearing aid according to the present invention achieves desired compensation according to the power level of the input signal. It can be seen that. That is, in FIG. 6, the A portion of the hearing aid using the digital hearing aid or the conventional multi-band compression method or the appropriate compensation was made above the hearing threshold, but the compensation of the B portion below the hearing threshold. This should not be done, but in the case of the multiband compression scheme, it can be seen that compensation is achieved by giving a constant gain over the entire band, and it can be seen that compensation is not performed in the digital hearing aid according to the present invention.

In the case of the conventional multi-band compression method, the digital hearing aid performs compensation in a manner that gives a uniform uniform gain over the entire band, so that unnecessary compensation is performed even in a band below the hearing threshold (part B of FIG. 6). Distortion occurs, but the present invention is to improve this to allow the user of the hearing aid to hear the input clearly.

As described above, the present invention is a sound input to the hearing aid by designing a sub-band hearing loss compensation filter for the user of the hearing aid to effectively compensate for the fact that the gain loss value for each frequency according to the type and degree of hearing loss A system identification method is devised to have the same performance as a low order filter. As a result, unnecessary compensation due to interference between input signals and frequency distortion of the input signal are improved to allow users of hearing aids to listen to the input sound naturally and clearly.

Although the preferred embodiments of the present invention have been described in detail with reference to the accompanying drawings, the present invention is not limited thereto and may be improved or modified by those skilled in the art within the scope of the technical idea of the present invention.

Claims (5)

Iii) converting an external voice signal into a digital signal; Ii) dividing the digital signal into multiple channels of low, mid, and high frequencies based on a frequency band and outputting the digital signal to each channel; Iv) separating a high band portion and a low band portion of the digital signals of the signals input to the respective channels, respectively; Iii) receiving the separated high band portion and the low band portion, calculating the hearing loss compensation gains of the corresponding bands, and accumulating and outputting them; And Iii) converting the data for which hearing loss is compensated using the hearing loss compensation gain into an analog signal, and then converting the data into an audio signal and outputting the audio signal. delete 2. The method of claim 1, wherein the compensation signal is calculated according to the input signal in step (a).

Where N is the specification of a given frequency band,

Is the specification of the size to be obtained for each frequency,

Is the frequency given by the standard,

Indicates the phase specification to be obtained for each frequency. Two microphones 110A and 110B for receiving a voice signal; A / D converters (120A, 120B) for converting the analog signal input through the microphone (110A, 110B) into a digital signal; A voice input unit 130 for dividing and outputting each of channels such as low, middle and treble according to the frequency of the voice signal converted into a digital signal through the A / D converters 120A and 120B; It is connected to the rear end of each channel unit 140A to 140N for receiving the digital signal output to the voice input unit 130, respectively, and the high-band portion and the low-band are respectively inputted to the channel unit 140A to 140N. A gain adjusting unit 150A to 150N configured of a subband feedback removing unit 152 and a subband deafness compensation filter unit 154 for adjusting the gain by separating the portion into feedback a predetermined number of times; A signal synthesizing unit 160 for synthesizing a signal whose gain is adjusted through the gain adjusting units 150A to 150N; And And a D / A converter (170) connected to an output side of the signal synthesizing unit (160) for converting a digital signal into an analog signal and a speaker (180) for amplifying and outputting the analog signal. The method of claim 4, wherein the subband hearing loss compensation filter unit 154 separates the high band portion and the low band portion of the input digital signal d (n) and selects only the audible frequency band. 201B) and subband filters 202A and 202B; A compensation filter (203A, 203B) for outputting a compensation signal to satisfy a specification of the magnitude and phase of the compensation frequency in the signal of the selected frequency band; Order limit filters (205A, 206B) connected to the output terminals of the compensation filters (203A, 203B) to limit the number of outputs of the compensation values; And a signal synthesizer (160) connected to the output of the order limiting filter (205A, 206B) for outputting the accumulated compensation signal.

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