• Journal of the Institute of Electronics Engineers of Korea SC
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• v.42 no.4 s.304
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• pp.45-50
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• 2005

In digital hearing aid system, one of the main problems is acoustic feedback which is known as howling because of miniaturization md high-gain amplification. In this paper, we proposed a feedback cancellation algorithm for hearing aid using time delay and time-varying decorrelation filter. The proposed algorithm has a kind of adaptive filter structure, which is combined with time delay and time-varying decorrelation filter to improve feedback cancellation. An all pass filter was implemented as the time-varying decorrelation filter using low frequency modulator. From the result of computer simulation, it is verified that the proposed algorithm has good ability to cancel feedback.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11C
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• pp.911-916
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• 2010

This paper proposes a new algorithm to cancel the acoustic feedback and noise signals in digital hearing aids. The proposed algorithm combines the feedback canceller to remove acoustic feedback signals and the noise canceller to reduce background noises. The feedback canceller is implemented by normal adaptive FIR filter, and the noise canceller is implemented by using the Wiener solution in frequency domain. This noise canceller has the transfer function presented by the power spectral density of signals. To verify the performances of the proposed algorithm, the simulations were carried out for the system. As the results of simulations, it was proved that we can advance 10.85dB output SNR on the average for the forward path gain of 0dB, and 11.04dB output SNR on the average for the forward path gain of 6dB, in the case of using the proposed algorithm.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.10 no.2
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• pp.294-302
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• 2009

In this paper, the trend of technology and the business strategy on 'Hearing Aids' are investigated for the industry promotion of medical devices based on IT engineering from the 316 patents of 'Hearing Aids' registered in Korean Intellectual Property Office(KIPO). The classification of technology on 'Hearing Aids' is performed according to the IPC(International Patent Classification) code to and the core technology of 'Hearing Aids' As the results of classification of IPC code, the number of patents with IPC code 'H04R', 'H04B', 'H01M', and 'A61F' are 160, 46, 40, and 19 respectively. We found that the Digital technology and the Medical Transplants technology are come to the front of 'Hearing Aids' and the foreign 'Hearing Aids' companies are filed an application with the Korean Intellectual Property Office(KIPO) before their business.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2459-2462
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• 2003

In this paper, we Propose a wavelet based adaptive algorithm which improves the convergence speed and reduces computational complexity using the fast running FIR filtering efficiently. We compared the performance of the proposed algorithm with time and frequence domain adaptive algorithm using computer simulation of adaptive noise canceler based on synthesis speech.

• Journal of Biomedical Engineering Research
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• v.25 no.4
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• pp.315-321
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• 2004

In this study, we proposed an adaptive feedback cancellation algorithm for multi-band digital healing aids. The adaptive feedback canceller (AFC) is composed of an adaptive notch filter (ANF) for feedback detection and an NLMS (normalized least mean square) adaptive filter for feedback cancellation. The proposed feedback cancellation algorithm is combined with a multi-band hearing aid algorithm which employs the MDCT (modified discrete cosine transform) filter bank for the frequency-dependent compensation of hearing losses. The proposed algorithm together with the MDCT-based multi-channel hearing aid algorithm has been evaluated via computer simulations and it has also been implemented on a commercialized DSP board for real-time verifications.

• 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 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.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.11
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• pp.149-156
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• 2012

A digital input class-D audio amplifier is presented for digital hearing aid. The class-D audio amplifier is composed of digital and analog circuits. The analog circuit converts a digital input to a analog audio signal (DAC) with noise suppression in the audio band. An interpolated digital delta-sigma modulator is used to convert data types between digital signal processor (DSP) and digital-to-analog converter (DAC). An 16-bit, 25-kbps pulse code modulated (PCM) input is interpolated to 16-bit, 50-kbps by a digital filter. The output signal of interpolation filter is noise-shaped by a third-order digital sigma-delta modulator (SDM). As a result, 1.5-bit, 3.2-Mbps signal is applied to simple digital to analog converter.

• Proceedings of the KIEE Conference
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• 2004.11c
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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.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.231-234
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• 2004

디지털 보청기를 착용한 사람의 지향성 측정 및 시뮬레이션 비교를 위해서 실제 실험 장치(무향실)와 여러 전기음향 계측기기를 이용한 실험을 수행하는 과정에서 많은 소요 시간의 문제점이 발생하였다. 그래서 실험 시간을 단축하기 위해 본 전기음향 실험기기의 자동화 및 고속화를 연구하였다 지향성 실험을 위해 간이형 무향실이 회전하도록 설계하였고, 이에 따른 자동제어를 위한 프로그래밍을 연구하였다.