• The Journal of the Acoustical Society of Korea
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• v.31 no.6
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• pp.367-372
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• 2012

This paper introduces noise masking algorithm for 64 channel digital hearing aid. 125 Hz spectral resolution is maintained for 64 channels from 125 Hz to 8000 Hz. The same spectral masking processing effects as the cochlea are considered and applied for the present hearing aid noise reduction processing algorithm. Theoretical algorithm has been ported into assembler language program software and been embedded into a DSP IC chip for the digital hearing aid. Some parts of noise masking software program code are explained, and the results of the real-time noise reduction are verified by electro-acoustic measurements.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.49 no.3
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• pp.69-74
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• 2012

This paper shows the results about field operating digital hearing aids which protect the ears of the battle field soldiers from explosive sound and minimize the difficulty of mutual communication during the battle. The essence of the hearing aid is in its signal compression technology in which soft sound is amplified while rapidly increased explosive sound is attenuated. This nonlinear compression technology can be applied for the protection of the ears of the battle field soldiers. As a core part of the hearing aid, when a new DSP IC chip is launched, the modified firmware and fitting software is developed for adaption. Ezairo 5910 which was recently launched by DSP factory in Canada was used for the development of the firmware of the hearing aid.

• Journal of Korea Multimedia Society
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• v.3 no.5
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• pp.495-505
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• 2000

In this paper, we propose an efficient auto-fitting system for digital hearing-aids which automatically adjusts the fitting parameters according to the auditory characteristics of hearing handicapped person. The fitting parameters are extracted from audiogram of hearing handicapped and are applied to digital hearing-aid purposed GM3036 chip. The characteristics of each parameter are compared with those from theoretical 2cc graph. The purposed system has applied to 50 patients and their satisfaction ratios show to the very high. As results, it shows effectiveness of proposed system.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.235-238
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• 2004

대부분의 보청기 이용자들은 자신이 장애를 겪고 있다는 사실이 알려지는 것을 꺼린다. 따라서 보청기 사용자들은 귀 바깥쪽으로 들어 나지 않는 귓속형(ITE type) 보청기를 선호한다. 하지만, 이러한 귓속형 보청기를 제작하는 것은 쉬운 일이 아니다. 보청기의 각각 부품들만을 볼 때는 그 크기가 소형이지만, 부품 모두를 귀속에 넣는다고 한다면, 공간의 확보는 꼭 필요한 사항이다. 또한 보청기는 하나의 칩(chip)에 부품이 전선(wire)을 통해 납땜이 되는 구조이다. 이는 전선의 단락을 유발할 뿐만 아니라 칩과 전선이 쉽게 떨어지며, 잦은 납Eoa 작업으로 인해 열적으로 칩의 파손까지도 일으킨다. 이를 보안하고자 여기에서는 보청기 소형화 방법으로 PCB(Printed Circuit Board)를 제시하였다. PCB 를 사용함으로써 전선의 사용을 최소화하고, 부품과 PCB 와의 직접적인 결합으로 인해 보다 견고한 보청기 제작을 목표로 하였다. 아울러 PCB 를 통한 부품의 실장으로 기존의 수작업이던 보청기 제작을 자동화와 제작 시간의 단축이라는 이점을 얻을 수 있다.

• 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.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.2
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• pp.205-212
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• 2016

In this paper, we implement the development platform of multichannel digital hearing aid algorithm using Simulink provided by Matlab. The digital hearing aids are considered medical devices designed to compensate for hearing loss, they need to be correctly selected, to help a person who has difficulty in hearing. The development platform that implemented in this paper, includes WOLA filterbank for analysis/synthesis of input signal, Wide dynamic range compression for hearing loss compensation and adaptive filter for feedback cancellation. Using the development platform, algorithm parameters for each block can be set depending on the hearing aid user. Thus it is possible to test the algorithm before the machine language. As a result, the time for algorithm development can be saved and performance and computational complexity can be optimized.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.10 no.1
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• pp.102-110
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• 2017

In this paper, we have implemented an real-time adaptive feedback cancellation(AFC) algorithm that can be applied to multi-channel digital hearing aid. Multichannel digital hearing aid typically use the FFT filterbank based Wide Dynamic Range Compression(WDRC) algorithm to compensate for hearing loss. The implemented real-time acoustic feedback cancellation algorithm has one integrated structure using the same FFT filter bank with WDRC, which can be beneficial in terms of computation affecting the hearing aid battery life. In addition, when the AFC fails to operate due to nonlinear input and output, the reduction gain is applied to improve robustness in practical environment. The implemented algorithm can be further improved by adding various signal processing algorithm such as speech enhancement.

• The Journal of the Acoustical Society of Korea
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• v.36 no.2
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• pp.123-129
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• 2017

In this paper, the realization of a directional digital hearing aid was considered. Conventional time domain time delay method was replaced with digital filters in order to make any general-purposed DSP (Digital Signal Processing) chip to produce the similar directivity pattern. Both the time delay algorithm and the digital filter algorithm were initially evaluated by Matlab (Matrix laboratory) for comparison, and it was confirmed by CSR 8675 Bluetooth DSP IC (Digital Signal Processing Integrated Circuit) chip firmware realization. Some remote control features by a smart phone was added to the smart hearing aid for user interface easiness.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.5C
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• pp.673-682
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• 2004

The adaptive beam-forming is promising approach for noise reduction in hearing aids. This approach has come in the focus of interest only recently, because of the availability of new and powerful digital signal processors. The adaptation U using usually a Least Mean Squares algorithm, updates the weight vector. In this Paper, we propose a fast wavelet based adaptive algorithm using variable step size algorithm which varies adaptive constant by the change of signal environment. We compared the performance of the proposed algorithm with the known adaptive algorithm using computer simulation of multi channel adaptive bemformer in hearing aids. As the result the proposed algorithm is suitable for adaptive signal processing area using hearing aids and has advantages reducing computational complexity. And we show the beam-forming system using proposed algorithm converges stably in a sudden change of system environment.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.175-178
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• 2004

각각 2개의 마이크로폰을 내장한 귓속형 보청기를 양쪽귀에 착용한 상태에서 소리의 지향성을 시뮬레이션하고 전기음향실험으로 비교하였다. 지향성 보청기의 신호 대 잡음 비율 향상을 고려하였다. 시뮬레이션을 위해 경계요소기법을 사용하였으며 본 논문에 의해 밝혀진 주파수 대역에서의 시간 지연을 DSP칩을 내장한 디지털 보청기의 지향성 조절 파라미터로 사용하고자 한다. 마이크로폰 사이의 간격을 10mm로 하였다. 가장 최적한 지향성이 출력 되기 위한 위상을 계산하였다.