• Journal of Sensor Science and Technology
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• v.19 no.5
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• pp.361-368
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• 2010

ACROSS device, which is composed of an implantable microphone, a signal processor, and a vibrating transducer, is a fullyimplantable middle ear hearing device(F-IMEHD) for the recovery of patients with hearing loss. And since a microphone is implanted under skin and tissue at the temporal bones, the amplitude of the sound wave is attenuated by absorption and scattering. And the vibrating transducer attached to the ossicular chain caused also the different displacement from characteristic of the stapes. For the gain control of auditory signals, most of implantable hearing devices with the digital audio signal processor still apply to fitting rules of conventional hearing aid without regard to the effect of the implanted microphone and the vibrating transducer. So it should be taken into account the effect of the implantable microphone and the vibrating transducer to use the conventional audio fitting rule. The aim of this study was to measure gain characteristics caused by the implanted microphone and the vibrating transducer attached to the ossicle chains for the gain compensation of ACROSS device. Differential floating mass transducers (DFMT) of ACROSS device were clipped on four cadaver temporal bones. And after placing the DFMT on them, displacements of the ossicle chain with the DFMT operated by 1 $mA_{peak}$ current was measured using laser Doppler vibrometer. And the sensitivity of microphones under the sampled pig skin and the skin of 3 rat back were measured by stimulus of pure tones in frequency from 0.1 to 8.9 kHz. And we confirmed that the microphone implanted under skin showed poorer frequency response in the acoustic high-frequency band than it in the low- to mid- frequency band, and the resonant frequency of the stapes vibration was changed by attaching the DFMT on the incus, the displacement of the DFMT driven with 1 $mA_{rms}$ was higher by the amount of about 20 dB than that of cadaver's stapes driven by the sound presssure of 94 dB SPL in resonance frequency range.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.3
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• pp.138-144
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• 2017

In this paper, we propose a high frequency boosting circuits compensating for age-related hearing loss. The frequency response of this hearing loss is quite similar to that of a low-pass filter of which the critical frequency get lower with age. Therefore the voltage gain of this compensation circuits increase proportionally to the frequency of signals when the frequency is higher than the critical frequency and the voltage is constant irrespective of the frequency of signals when the frequency is lower than the critical frequency. The proposed circuits consist of a differential circuit and a unity gain amplifier. Because the critical frequency of the proposed circuits is controlled simply in the shape of a volume control lever, the aged people can adjust the high frequency boosting level easily according to one's hearing loss level. The critical frequency is continuously controllable in the whole audible frequency band and the gain of this high frequency boosting circuits is above 80dB at 10kHz.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1744-1749
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• 2013

Acoustic feedback is perceived as whistling or howling, which is a major complaint of hearing-aids users. Acoustic feedback cancellation is important in hearing-aids because acoustic feedback degrades performance of the hearing aid device by reducing maximum insertion gain. Adaptive systems for estimate acoustic feedback path and feedback suppression algorithms have been proposed in order to solve this problem. A typical feedback cancellation algorithm is LMS(least mean squares) because of its computational efficiency. However it has problem of convergence performance in high correlated input signal. In this paper, we propose a new variable step-size normalized LMS(least mean squares) algorithm using VAD(voice activity detection) to overcome the limitation of the LMS algorithm. The VAD algorithm is GSAP(global speech absence probability) and the feedback cancellation algorithm is normalized LMS. The proposed algorithm applies different step-size between voice and non-voice using VAD, for high stability, fast convergence speed and low misalignment when correlated inputs, such as speech. The result of simulation with white noise mixed speech signal, the proposed algorithm shows high performance then traditional algorithm in terms of stability, convergence speed and misalignment.

• Korean Institute of Interior Design Journal
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• no.26
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• pp.112-120
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• 2001

This study investigated the extend to which the facilities satisfactorily accommodate people with different disabilities and their specific needs for improving the facilities. Systematic interviews were conducted with 90 disabled people using the non-propertionate stratified cluster random sampling method. Frequency analysis, Chi-square tests, Duncan tests, and analysis of variance(ANOVA) were employed for data analysis. The findings indicated that the respondents' assessment of accommodations and needs for improving accommodations differed by the type of disabilities. People with physical disabilities were the least satisfied with stairs, followed by seating in stadiums and/or theaters. For people with visual disabilities, the lack of guidance and information facilities was the most difficult problem. People with hearing disabilities often pointed out difficulties in using public telephones. meanwhile, people with physical disabilities asked for easy access to assistive devices and improved mobility-related circumstances. yet people with visual disabilities expected to improve way-finding relevant issues, such as detectable warnings on walking surfaces and guidance and information facilities, whereas people with hearing disabilities desired to develop hearing-aid equipments, such as TDD and assistive listening systems.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.12
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• pp.219-225
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• 2012

A low-power, low-noise pre-amplifier for digital hearing-aid application is designed. This pre-amplifier amplifies single-ended signal from an electret microphone, and produces differential output to be delivered to an ADC. It has a variable gain of 3.6, 7.2, 14.4 and 28.8 with a bandwidth between 100Hz~10kHzon. The measurement results show 85 dB of SNR, 0.05 % of harmonic distortion and $200{\mu}W$ of power consumption with 1.2V supply.

• The Korean Journal of Food & Health Convergence
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• v.5 no.5
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• pp.27-31
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• 2019

Meniere's disease (MD) is a disease that affects the inner ear. It is formed as a result of endolymphatic hydrops. Hearing loss and vertigo are important in the diagnosis of MD. There is fluctuating and progressive hearing loss. Vertigo attacks cause severe dizziness in the patient. There are many treatment options in MD. These are hearing aid, diet, medication and surgery. In this study, we will discuss the advantages and disadvantages of the different treatment options. Treatment options have been compared to find out what the appropriate treatment is. Another concern is the importance of surgery in MD. This study is combination of qualitative and quantitative studies. Much focus will be on vertigo, and appropriate treatment options of MD will be mentioned also the importance of surgery. The main question in this study is the necessity of surgery. Surgical procedures are the most doubtful treatment option because of their indications and contraindications. In this study, it has been noticed that surgical operation should be delayed as much as possible. Priority is to try other treatment options. Surgery can be considered as a last resort. When we look at the operations performed, operations are mostly done in advanced Meniere cases.

• Journal of Biomedical Engineering Research
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• v.20 no.5
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• pp.625-632
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• 1999

이식형 인공중이에 사용되는 전자트랜스듀서는 압전형 트랜스듀서에 비해 음향특성은 좋으나 부피가 크고 효율이 낮으며 수술시 정밀한 코일-자석 간격 조정이 필요하며, 최근 제안된 FMT 트랜스듀서는 외부 자장의 변화에 민감하다는 문제점이 지적되고 있다. 본 논문에서는 코일-자석 간격조정이 필요 없고 외부자장에 영향을 받지 않는 차동 플로팅메스형 전자 트랜스듀서를 제안하였다. 제안된 방식은 2개의 소형 자석을 같은 극끼리 접합함으로써 외부자장에 대해 영향을 받지 않고 효율이 높은 등의 장점을 가진다. 제안된 트랜스듀서의 진동력 및 효율에 관한 정량적인 해석을 하였으며 기존의 트랜스듀서와 비교분석을 행하였다. 그리고 공급전류에 대해 발생되는 진동력의 크기를 계산함으로서 실제 이식형 인공중이의 제작에 필요한 객관적인 설계데이타를 제시하였다. 한편, 시험 제작된 트랜스듀서의 해석결과 제안한 트랜스듀서는 기존의 FMT 트랜스듀서보다도 효율이 1.5배 정도의 향상됨을 보였으며 무부하시험 및 사체의 이소골 진동실험을 통하여 적절한 진동을 효과적으로 이소골에 전달할 수 있음을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.379-386
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• 2005

Among various kinds of hearing aids which have been developed so far. the conventional air conduction hearing aids have some problems such as the acoustic distortion, an howling effect due to acoustic feedback. Another type of hearing aid. the cochlear implant system can be applied to the profound imparied person. However. it shows the disadvantage that there is no possibility of recovery of the acoustic organ such as ossicle. On the other hand. the implantable middle ear heaving device directly vibratos the ossicular chain and has better sound qualify. good cosmetics for appearance. and wide frequency responses so that it can overcome the defects or the conventional hearing aids. In this paper, a mathematical modeling and a momentum equation derivation of the DET has been performed. For the optimization of the structure dimension generating maximal vibrating force of the DET. the computer simulation using a finite element analysis (FEA) software has been performed. Also. the vibrating transducer has been designed to make the frequency characteristics or the transducer be similar to those of the normal middle ear. Through the experimental results, the measured vibration characteristics of the DET has been evaluated to verify the performance for the application to implantable middle ear hearing devices.

• Journal of rehabilitation welfare engineering & assistive technology
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• v.11 no.2
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• pp.133-141
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• 2017

The hybrid acoustic sensor for implantable hearing aid has the structure in which a sound pressure based acoustic sensor (ECM) and a vibration based acceleration sensor are combined. This sensor combines the low frequency sensitivity of an acoustic sensor with the high frequency sensitivity of an acceleration sensor, allowing the acquisition of a wide range of sound from low to high frequency. In this paper, an acceleration sensor for use in a hybrid acoustic sensor has been proposed. The acceleration sensor captures the vibration of the tympanic membrane generated by the acoustic signal. The size of the proposed acceleration sensor was determined to diameter of 3.2 mm considering the anatomical structure of the tympanic membrane and the standard of ECM. In order to make the hybrid acoustic sensor have high sensitivity and wide bandwidth characteristics, the aim of the resonance frequency of the acceleration sensor is to be generated at about 3.5 kHz. The membrane of the acceleration sensor derives geometric structure through mathematical model and finite element analysis. Based on the analysis results, the membrane was implemented through a chemical etching process. In order to verify the frequency characteristics of the implemented membrane, vibration measurement experiment using external force was performed. The experiment results showed mechanical resonance of the membrane occurred at 3.4 kHz. Therefore, it is considered that the proposed acceleration sensor can be utilized for a hybrid acoustic sensor.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.5
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• pp.538-542
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• 2018

Echo generated in digital hearing aid vent give rise to user's discomfort. For cancelling feedback echo in vent, it is required to estimate vent impulse response exactly. The vent impulse response has time varying and sparse characteristics. The IPNLMS has been known a useful adaptive algorithm to estimate vent impulse response with these characteristics. In this paper, subband sparse adaptive filter which applying IPNLMS to subband hearing aid structure is proposed to cancel echo of vent by estimating sparse vent impulse response. In the propose method, the decomposition of input signal to subband can pre-whiten each subband signal, so adaptive filter convergence speed can be improved. And the poly phase component decomposition of adaptive filter increases sparsity of each components, and the better echo cancellation can be possible without additional computation. To derive coefficients update equation of the adaptive filter, by defining the cost function based weight NLMS is defined, and the coefficient update equation of each subband is derived. For verifying performances of the adaptive filter, convergence speed, and steady state error by white signal input, and echo cancelling results by real speech input are evaluated by comparing conventional adaptive filters.