• Proceedings of the KOR-BRONCHOESO Conference
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• 1987.05a
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• pp.12.1-12
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• 1987

1979년 McCabe가 자가면역성 감음신경성 난청이라는 용어를 처음으로 사용한 이래. 많은 학자가 면역성질환에서 감음신경성 난청이 동반될 수 있다는 보고가 있어 왔다. 자가면역성 감음신경성 난청은 보통 양측성, 비대칭적으로 점진적인 난청이 수주 혹은 수개월에 걸쳐서 진행되는 것이 특징이다. 그간 내이의 면역학적인 측면에 관한 연구가 진행되어 왔던 바, 일반적인 치료에 듣지 않으며 면역억제요법으로 효과를 볼 수 있는 감음신경성 난청의 범주를 결정할 수 있었으며, 이는 치료가 가능한 감음신경성 난청이라는 점에서 이비인후과 영역에서 관심을 가져야 할 필요가 있다고 하겠다. 저자들은 최근 자가면역성 질환인 전신성 홍반성 낭창을 가진 27세 여자환자에서 감음신경성 난청을 관찰할 수 있었으며, 면역억제요법으로 청력의 호전을 경험하였기에 문헌적 고찰과 함께 보고하는 바이다.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.41 no.1
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• pp.41-54
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• 2004

In this paper, we propose a new digital hearing aid circuit that compensates the impaired threshold level changing nonlinearly using a multi-band compensation technique. In the algorithm the hearing frequency range 8kHz is divided into 64 bands which is 125Hz resolution. Each band is controlled finely to compensate the hearing impaired proportional to personal ROM table. The multi-band is introduced using a FFT/IFFT Processor which makes to control in frequency domain. As a result, the proposed circuit is more efficient $15\%$ than a conventional ones such as FIR filter architecture in terms of the compensation gun and accuracy. The hardware size was reduced $65\%$ than a general FFT by pre-handling of the input data.

• The Journal of the Acoustical Society of Korea
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• v.19 no.2
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• pp.61-67
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• 2000

Recent progress in digital and VLSI technology has offered new possibility fer noticeable advance of hearing aids. Yet, environmental noise remains one of the major problems to hearing aid users. This paper describes results which speech recognition performance and speech discrimination performance was measured for listeners with sensorineural hearing loss, while listeners in speech-band noise. In addition, to ameliorate hearing-aided environments of hearing impaired listeners, environmental noise reduction using speech enhancement techniques are investigated as a front-end of conventional hearing aids. Speech enhancement techniques are implemented in a realtime system equipped with DSP board. The clinical test results suggest that the speech enhancement technique may work in synergy with gain functions fer the greater SNR improvement as the preprocessing algorithm of digital hearing aids.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.189-192
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• 2000

본 논문에서는 청각 장애인의 보다 향상된 보청 환경을 조성하고자 청각손실을 심리음향 모델을 적용하여 감음 신경성 난청을 보정하는 알고리즘을 제안한다. 제안한 알고리즘에서는 난청의 유형은 내이에서부터 중추 뇌에 걸친 감음계와 신경계의 장애에서 비롯되는 감음신경성 난청(sensorineural hearing loss)으로 주파수 영역상에서 MTH(minimum hearing threshold)가 균일하지 않게 상승하게되어 가청영역이 좁아지는 문제점을 해결하기 위한 방법으로 각각의 주파수 밴드마다 멀티밴드 압축 알고리즘을 적용하였다. 그러나 이 경우 각각의 주파수 밴드에 따른 서로 다른 가청 영역의 영향에 의한 변형된 스펙트럼 모양으로 인해 spectral contrast reduction과 변형된 마스킹 특성으로 인해 음성 변별력에 제한을 가하게 된다. 이것은 주변 주파수 성분들에 의한 마스킹 효과에 의한 것으로, 신호에 대한 난청인이 느끼는 지각 영역(perceptual domain)에서의 해석과 심리음향 모델 파라미터를 통한 보청기의 개발이 이루어져야 하며, 본 논문에서 그 알고리즘을 적용하였다.

• Proceedings of the KOSOMBE Conference
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• v.1996 no.11
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• pp.68-69
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• 1996

• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.12-21
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• 2000

In this paper, we propose a novel hearing aid algorithm for sensorinural hearing loss restoration with multi-channel(band) dynamic range compression and psychoacoustics. In this way, we can present a normal perception condition to the impaired listener. The proposed algorithm make loudness scaling function achieve proper loudness level, and analysis masking property for the signal will be perceived to impaired listener, and then, restore normal spectral contrast using SMR(signal-to-masking ratio) defined by distance between the level of each frequency and masking threshold.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.53 no.9
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• pp.657-662
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• 2004

The low-power design of the digital hearing aid is indispensable to achieve the compact portable device with long battery duration. In this paper, new low-power adaptive filter structure is proposed based on distributed arithmetic(DA). By modifying the DA technique, the proposed decimation filter structure can significantly reduce the power consumption and implementation area. Through Verilog-HDL coding, cell occupation of the proposed structure is reduced to 33.49% in comparison with that of the conventional multiplier structure. Since Verilog-HDL simulation processing time of the two structures are same, it is assumed that the power consumption or implementation area is proportional to the cell occupation in the simulation.

• Proceedings of the KOR-BRONCHOESO Conference
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• 1987.05a
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• pp.7.2-7
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• 1987

Waver 와 Bray (1930) 가 cochlear microphonic을 처음 발견한 이래 Derbyshire 와 Davis (1935)는 summating potential을 각각 처음 기술하여 이 세가지 전위를 electrocochleogram 이라 칭하였고 이는 감음신경성 난청의 감별진단 및 청각생리연구에 이용되어 왔다. 저자들은 정상 guinea pig 10 마리를 대상으로 DANAC 7E ERA청각계기를 사용하여 정원창에서 action potential과 summating potential을 측정하였으며 주파수에 따른 역치 자극간 간격 및 두 potential의 상호관계를 분석하여 다음과 같은 결과를 얻었다. 1) 주파수가 증가함에 따라 그 역치는 점차 감소하였다. 2) 자극음의 강도와 action potential의 $N_1$ component 진폭은 상호 비례관계를 보여 주었으며 주파수 증가에 따라 $N_1$ component 진폭은 점차 증가하였다. 3) Action potential 의 $N_1$ component latency는 주파수가 증가할수록 역비례 관계를 보여 주었다. 4) N$_1$ component의 진폭과 자극간 간격(interstimulus interval, ISI)과의 관계는 ISI가 80~160m sec사이에서 plateau를 형성하였다. 5) summating potential은 자극음의 강도가 증가함에 따라 그 진폭이 증가하였으며 action potential도 증가하였으나, SP/AP는 감소하는 경향을 보였다.

• Journal of Biomedical Engineering Research
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• v.19 no.1
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• pp.59-68
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• 1998

Digital hearing aids offer many advantages over conventional analog hearing aids. With the advent of high speed digital signal processing chips, new digital techniques have been introduced to digital hearing aids. In addition, the evaluation of new ideas in hearing aids is necessarily accompanied by intensive subject-based clinical tests which requires much time and cost. In this paper, we present an objective method to evaluate and predict the performance of hearing aid systems without the help of such subject-based tests. In the hearing impairment simulation(HIS) algorithm, a sensorineural hearing impairment medel is established from auditory test data of the impaired subject being simulated. Also, the nonlinear behavior of the loudness recruitment is defined using hearing loss functions generated from the measurements. To transform the natural input sound into the impaired one, a frequency sampling filter is designed. The filter is continuously refreshed with the level-dependent frequency response function provided by the impairment model. To assess the performance, the HIS algorithm was implemented in real-time using a floating-point DSP. Signals processed with the real-time system were presented to normal subjects and their auditory data modified by the system was measured. The sensorineural hearing impairment was simulated and tested. The threshold of hearing and the speech discrimination tests exhibited the efficiency of the system in its use for the hearing impairment simulation. Using the HIS system we evaluated three typical hearing aid algorithms.

• Proceedings of the KOSOMBE Conference
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• v.1997 no.11
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• pp.63-66
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• 1997

With the advent of high speed digital signal processing chips, new digital techniques have been introduced to the hearing instrument. This advanced hearing instrument circuitry has led to the need or and the development of new fitting approach. A number of different fitting approaches have been developed over the past few years, yet there has been little agreement on which approach is the "best" or most appropriate to use. However, when we develop not only new hearing aid, but also its fitting method, the intensive subject-based clinical tests are necessarily accompanied. In this paper, we present an objective method to evaluate and predict the performance of hearing aids without the help of such subject-based tests. In the hearing impairment simulation (HIS) algorithm, a sensorineural hearing impairment model is established from auditory test data of the impaired subject being simulated. Also, in the hearing impairment simulation system the abnormal loudness relationships created by recruitment was transposed to the normal dynamic span of hearing. The nonlinear behavior of the loudness recruitment is defined using hearing loss unctions generated from the measurements. The recruitment simulation is validated by an experiment with two impaired listeners, who compared processed speech in the normal ear with unprocessed speech in the impaired ear. To assess the performance, the HIS algorithm was implemented in real-time using a floating-point DSP.