• 대한전기학회:학술대회논문집
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• 대한전기학회 2004년도 학술대회 논문집 정보 및 제어부문
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• pp.577-579
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• 2004

Digital hearing aids enclose $6{\sim}8$ tiny components. Those electromechanical components are individually wired by soldering which is a manual labor and sometimes causes components' damage by heating. This paper suggests a PCB design for overcome these problems. Several PCBs are designed and manufactured and circuited to produce ITE(In The Ear) type hearing aids which are inserted in the ear canal. The most optimal size of the PCB design for the ITE hearing aid is presented in this paper.

• 대한기계학회:학술대회논문집
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• 대한기계학회 2004년도 추계학술대회
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• pp.1012-1015
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• 2004

Digital hearing aids enclose $6{\sim}8$ tiny components. Those electromechanical components are individually wired by soldering which is a manual labor and sometimes causes components' damage by heating. This paper suggests a PCB design for overcome these problems. Several PCBs are designed and manufactured and circuited to produce ITE(In-the-Ear) type hearing aids which are inserted in the ear canal. The most optimal size of the PCB design for the ITE hearing aid is presented in this paper.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2006년 학술대회 논문집 정보 및 제어부문
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• pp.241-243
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• 2006

In this research, we tried to implement a PDA (Personal Digital Assistant)-based hearing aids fitting system to manipulate hearing aids-fitting parameters in the user's local environment. Due to the inherent portability of PDA system, we can consequently perform the hearing aids fitting operation without visiting a special site.

• 대한의용생체공학회:학술대회논문집
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• 대한의용생체공학회 1995년도 추계학술대회
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• pp.121-123
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• 1995

This paper presents a new method which compensates loss of loudness for digital hearing aids. Loudness grows more rapidly in frequency domain with substantial shifts of hearing threshold, so that loud sounds reach the uncomfortable sound level (UCL) at about the same physical stimulus level as with normal hearing. The result is a compression of the available dynamic range of hearing. Many techniques have been developed to compensate for hearing losses. In this paper, we propose a digital hearing aid which uses a single digital filter for reducing distortion and the fuzzy function to calculate gain factors. This function describes how much gain is needed for every frequency to restore loudness perception of a normal ear.

• 한국음향학회지
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• 제19권2호
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• pp.61-67
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• 2000

최근의 디지털 신호처리 기술과 집적 회로 설계 기술의 발달은 보청 시스템의 새로운 가능성을 제공하고 있다. 그러나, 배경 잡음은 여전히 많은 난청자가 호소하는 문제로 남아 있다. 본 논문에서는 임상 실험을 통하여 음성 대역 잡음 환경에서 감음신경성 난청자의 음성 인지 능력과 어음 변별력을 측정한 결과를 제시한다. 또한, 보다 향상된 보청 환경을 제공하기 위하여 보청 시스템의 전처리단으로써 음질 향상 기법을 이용하여 배경 잡음을 제거하였다. 음질 향상 기법은 DSP 보드를 이용하여 실시간 시스템으로 구현되었으며, 이를 이용하여 청력 검사를 실시하였다. 임상 실험을 실시한 결과, 음질 향상 기법은 배경 잡음을 제거함으로써 신호의 SNR을 개선시켜 보청 이득과 결합되어 감음신경성 난청자의 음성 변별력을 크게 향상시켰다.

• 대한의용생체공학회:의공학회지
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• 제30권2호
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• pp.169-178
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• 2009

In this study, we suggest Korean nonlinear fitting formula (KNFF) to maximize speech intelligibility for digital hearing aids based on NAL-NL1 (NAL-nonlinear, version 1). KNFF was derived from the same procedure which is used for deriving NAL-NL1. KNFF consider the long-term average speech spectrum of Korean instead of English because the frequency characteristic of Korean is different from that of English. New insertion gains of KNFF were derived using the SII (speech intelligibility index) program provided by ANSI. In addition, the insertion gains were modified to maximize the intelligibility of high frequency words. To verify effect of the new fitting gain, we performed speech discrimination test (SDT) and preference test using the hearing loss simulator from NOISH. In the SDT, a word set as test material consists of 50 1-syllable word generally used in hearing clinic. As a result of the test, in case of moderate hearing loss with severe loss on high frequency, the SDT scores of KNFF was more improved about 3.2% than NAL-NLl and about 6% in case of the sever hearing loss. Finally we have obtained the result that it was the effective way to increase gain of mid-high frequency bands and to decrease gain of low frequency bands in order to maximize speech intelligibility of Korean.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2005년도 ICCAS
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• pp.190-195
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• 2005

The hearing aid chip described in this paper is an analog & digital mixed system. The design focuses on the$\mu$DSP core. This $\mu$DSP core includes internal time delays to two inputs from front and rear microphones. The paper consists of two parts; one is the composure and signal processing algorithm of digital hearing aids and the other is Verilog HDL codes for$\mu$DSP cores. All digital modules in the design were coded and synthesized by Verilog HDL codes which were verified by Mentor Graphics and Synopsis semiconductor chip design tools.

• 한국멀티미디어학회논문지
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• 제3권5호
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• pp.495-505
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• 2000

본 논문에서는 난청환자의 청각손실로부터 디지털 보청기의 보정 파라미터를 자동적으로 추출하는 방법을 제안하고 이를 이용한 자동보정시스템을 구현한다. 디지털 보청기의 보정 파라미터는 난청환자의 청각 특성 오디오그램으로부터 자동으로 추출하였으며, 추출된 보정 파라미터는 실제 GM3036 디지털 보청기를 이용하여 보정 시스템으로 구현되었으며, 각 파라미터는 이론적인 2cc 출력에 근접하도록 테이블화 하였다. 제안한 자동 보정 시스템을 적용하여 검사기로부터 출력을 검증하였고, 50명에게 사용하게 한후 만족돌르 조사한 결과 자동 보정 시스템의 유효성을 확인할 수 있었다.

• 디지털산업정보학회논문지
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• 제8권2호
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• pp.49-58
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• 2012

This paper is on an improved orthogonal projection method which can cancel the acoustic feedback signals in the digital hearing aids. Comparing with the NLMS algorithm which is widely used for simplicity and stability, it shows that this method has the improvement of the convergence performances, and has small computational quantities, for signals with the large auto-correlation as speech signals. This uses the improved orthogonal projection algorithm which reduces the correlation of signals. To verify the convergence characteristics of the proposed algorithm, we simulated about various input signals. The acoustic feedback canceller has a 12-bit resolution with 64-tap adaptive FIR filter. And we compared the results of simulation for this algorithm with the ones for the NLMS algorithm. By these works, it is proved that the feedback canceller adopting the proposed algorithm shows about 3.5dB more high SNR than the NLMS algorithm in the colored input signals.

• 대한의용생체공학회:학술대회논문집
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• 대한의용생체공학회 1998년도 추계학술대회
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• pp.171-172
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• 1998

Recent development of digital technology has offered new possibilities for noticeable advances of hearing aids. Using the digital technology, it is possible to equip hearing aids with powerful features such as multi-channel nonlinear compression amplification and the feedback cancellation, these are often difficult to implement with analog circuits. Still, speech in noise is one of the major complaints of not only hearing impaired persons but also normal listeners. This paper describes speech intelligibility in background noise for both normal and hearing impaired listeners. Speech enhancement algorithms were implemented and compared for normal and sensorineural hearing impairment listeners.