• The Journal of the Acoustical Society of Korea
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• v.34 no.2
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• pp.91-97
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• 2015

In this paper, wireless communication from a smart phone to a hearing aid using audible frequency band sound was considered. 1kHz single channel carrier frequency was applied for amplitude shift keying (ASK) binary data modulation, and the result of the transmission and reception was tested. The overall system process was precisely explained and experimentally evaluated. The result will be applied for realizing a digital hearing aid with remote control feature by a smart phone.

• Proceedings of the KSME Conference
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• 2005.11a
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• pp.174.1-174.1
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• 2005

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.367-372
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• 2004

디지탈 보청기의 DSP 칩 파라미터들은 보청기 사용자에게 가장 최적의 파라미터들로 구성되어져야 한다. Gennum GB3211과 같은 디지탈 보청기용 DSP 칩은 전용 전자 칩으로써 약 90 여개의 제한된 파라미터들을 조합하여 약 47억여개 이상의 거의 무한한 정도의 다양한 경우 수를 대응하도록 제작되었다. 보청기 사용자의 전기-음향 효과를 극대화하기 위해 가장 최적화된 파라미터 피팅 프로그램을 개발하였다. 컴퓨터 입력이 가능한 오디오그램 청력 역치로부터 여러 다른 Formula를 사용할 뿐 만 아니라, 마이크로폰과 리시버의 보정 효과를 모두 포함하는 최적 보청기 피팅 프로그램을 개발하였으며 몇 가지 사례를 적용해 보였다.

• Journal of Biomedical Engineering Research
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• v.18 no.4
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• pp.467-476
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• 1997

This paper presents a designing of a digital hearing aid processor (DHAP) chip being operated by a dedicated DSP core. The DHAP for hearing aid devices must be feasible within a size and power consumption required. Furthermore, it should be able to compensate for wide range of hearing losses and allow sufficient flexibility for the algorithm development. In this paper, a programmable 16-bit fixed-point DSP core is employed thor the designing of the DHAP. The designed DHAP performs a nonlinear loudness correction of 8 frequency bands based on audiometric measurements of impaired subjects. By employing a programmable DSP, the DHAP provides all the flexibility needed to implement audiological algorithms. In addition, the chip has low-power feature and $5, 500\times5000$$\mu$$m^2$ dimensions that fit for wearable hearing aids.

• Smart Media Journal
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• v.7 no.3
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• pp.29-34
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• 2018

This paper presents a study on adapting optical communication technology using infrared ray for the enhancement of hearing aids in noisy environment. The transmitter module containing microphone and infrared ray-emitting diode converts audio signal to infrared optical signal and sends it out in the air. The receiver module located in a distance receives the infrared signal, converts it to electrical signal, and transfers it to an input of a digital hearing aid. Especially, the receiver module needs to be small, low voltage, and consume low power since it will be attached to hearing aids. Experiments with applying infrared communication technology of digital modulation method and analog non-modulation method show that the analog non-modulation method is adequate for infrared communication of approximately 5m distance indoor. Prototypes of transmitter module and receiver module were manufactured, and internal parameters of the digital hearing aid were adjusted to confirm normal transmit-receive operation of audio signals.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.137-140
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• 1999

본 논문에서는 디지털 보청기의 자동보정법을 제안하고 이를 이용하여 구현한 보정시스템에 대하여 기술한다. 일반적으로 난청환자의 청각 손실은 전 주파수를 걸쳐 일정하게 발생하는 것이 아니라, 고음 또는 저음 등 일부 특정 주파수 대역의 청각특성의 이득이 다른 주파수에 비해 상대적으로 낮아져서 생기는 현상이다. 그러나 현재까지의 보청기는 이와 같은 청각 손실을 가지고 있는 환자들에 대하여 전 주파수 대역의 음압을 동일 비율로 높여주는 문제가 있다. 이와 같은 보청기로 보정하게 되면, 주파수 대역별 균형이 맞지 않기 때문에 소리가 변형되어 들리게 되고, 명료도 또한 떨어지게 된다. 또한, 기존의 아날로그보청기는 생산공정 중에 잡음 피크 제거, 리미터 조정, 이득 조정(GC) 등의 작업을 수동으로 수행하게 되는데, 이는 작업공정이 복잡하고 개개인에 청각특성에 맞는 정확한 보정이 불가능한 문제점을 안고있다. 이와 같은 문제점을 해결하기 위하여 제안한 자동보정 시스템은 각 주파수 대역별 최적화된 파라미터를 자동적으로 찾아서 입력하므로 보다 정확하고 신속한 보정이 가능하게 된다. 이 시스템은 외부 IC 콘트를 장치인 Hl-PRO를 이용하여 PC 환경 하에서Visual Basic으로 구현하였으며 시스템의 출력값을 측정하여 검토한 결과 구현한 자동보정법이 유효함을 확인할 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.273-279
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• 2012

In this thesis, a nonlinear compression fitting method was studied for each frequency channel of a 64 channel digital hearing aid. Unlike conventional fitting formula method done from the result of the hearing loss test, the present fitting method uses the auditory threshold of sound pressure measured near the tympanic membrane while ITE (In-The-Ear) hearing aid is fitted into the user's ear canal. Also, the spectral distribution of the voice sound pressure was used for realizing of output sound pressure compression curves against input sound pressure level. Theoretical research results of FFT-iFFT compression algorithm has been evaluated by experimental gain measurements at each different input sound pressure level 50 dB, 70 dB, 90 dB respectively.

• Proceedings of the KSME Conference
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• 2004.11a
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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.

• Journal of IKEEE
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• v.14 no.1
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• pp.47-54
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• 2010

In this study, we analyzed the reduction of noise in a noise environment using 2, 3, 4 or 5 microphones in digital hearing aids. In order to be able to use this in actual digital hearing aids, we made the experiment microphone set similar to the behind-the-ear type (BTE) and then recorded the signal accordingly, with each situation. With the recorded signals, we reduced the noise in each signal by a noise reduction algorithm using multi-microphones. As a result, in the case of By comparing the SNR (Signal to Noise Ratio) and PESQ (Perceptual Evaluation of Speech) measurements, before and after the noise reduction, the results showed that the improvement in performance was highest when three or four microphones were used. Generally, when two or more microphones were used, we found that as the number of microphones increased there was an increase in performance.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.238-247
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• 2005

In this Paper. we Presents subband design techniques of a compensating filter and adaptive feedback canceller for the digital hearing aid. The sensorineural hearing loss has a hearing threshold that shows a nonlinear characteristic in frequency domain. and its compensation suffers from an echo that produced by an undesired time varying feedback path. Therefore. the digital hearing aid requires the compensator that can adjust gains nonlinearly in frequency bands and eliminate the echo rapidly In the Proposed digital hearing aid. the compensating filter is designed by the adaptive system identification method in subband structure, and the adaptive feedback canceller is designed by the subband affine projection algorithm. The designed compensation filter can control the nonlinear gain in each subband respectively, therefore precise compensation is possible. And the feedback canceller using the subband adaptive filter achieves fast convergence rate. The Performances of the Proposed method are verified by computer simulations as comparing with the behaviors of the previous trials.