• 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.

• Journal of Biomedical Engineering Research
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• v.32 no.3
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• pp.218-229
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• 2011

Companding algorithms have been used to enhance speech recognition in noise for cochlea implant users. The efficiency of using companding for digital hearing aid users is not yet validated. The purpose of this study is to evaluate the performance of the companding for digital hearing aid users in the various hearing loss cases. Using HeLPS, a hearing loss simulator, two different sensorinerual hearing loss conditions were simulated; mild gently sloping hearing loss(HL1) and moderate to steeply sloping hearing loss(HL2). In addition, a non-linear compression was simulated to compensate for hearing loss using national acoustic laboratories-non-linear version 1(NAL-NL1) in HeLPS. In companding, the following four different companding strategies were used changing Q values(q1, q2) of pre-filter(F filter) and post filter(G filter). Firstly, five IEEE sentences which were presented with speech-shaped noise at different SNRs(0, 5, 10, 15 dB) were processed by the companding. Secondly, the processed signals were applied to HeLPS. For comparison, signals which were not processed by companding were also applied to HeLPS. For the processed signals, log-likelihood ratio(LLR) and cepstral distance(CEP) were measured for evaluation of speech quality. Also, fourteen normal hearing listeners performed speech reception threshold(SRT) test for evaluation of speech intelligibility. As a result of this study, the processed signals with the companding and NAL-NL1 have performed better than that with only NAL-NL1 in the sensorineural hearing loss conditions. Moreover, the higher ratio of Q values showed better scores in LLR and CEP. In the SRT test, the processed signals with companding(SRT = -13.33 dB SPL) showed significantly better speech perception in noise than those processed using only NAL-NL1(SRT = -11.56 dB SPL).

• 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.

• Journal of radiological science and technology
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• v.39 no.2
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• pp.257-262
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• 2016

This study aimed to suggest a novel method of modeling a hearing aid ear shell based on Digital Imaging and Communication in Medicine (DICOM) in the hearing aid ear shell manufacturing method using a 3D printer. In the experiment, a 3D external auditory meatus was extracted by using the critical values in the DICOM volume images, and the modeling surface structures were compared in standard type STL (STereoLithography) files which could be recognized by a 3D printer. In this 3D modeling method, a conventional ear model was prepared, and the gaps between adjacent isograms produced by a 3D scanner were filled with 3D surface fragments to express the modeling structure. In this study, the same type of triangular surface structures were prepared by using the DICOM images. The result showed that the modeling surface structure based on the DICOM images provide the same environment that the conventional 3D printers may recognize, eventually enabling to print out the hearing aid ear shell shape.

• 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.

• 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.

• Journal of Biomedical Engineering Research
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• v.28 no.6
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• pp.732-743
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• 2007

Background noise is one of the major complaints of not only hearing impaired persons but also normal listeners. This paper describes the results of two experiments in which speech recognition performance was determined for listeners with normal hearing and sensorineural hearing loss in noise environment. First, we compared speech enhancement algorithms by evaluation speech recognition ability in various speech-to-noise ratios and types of noise. Next, speech enhancement algorithms by reducing background noise were presented and evaluated to improve speech intelligibility for sensorineural hearing impairment listeners. We tested three noise reduction methods using single-microphone, such as spectrum subtraction and companding, Wiener filter method, and maximum likelihood envelop estimation. Their responses in background noise were investigated and compared with those by the speech enhancement algorithm that presented in this paper. The methods improved speech recognition test score for the sensorineural hearing impaired listeners, but not for normal listeners. The results suggest the speech enhancement algorithm with the loudness compression can improve speech intelligibility for listeners with sensorineural hearing loss.

• 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.

• Journal of Biomedical Engineering Research
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• v.34 no.1
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

• Korean Journal of Audiology
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• v.18 no.3
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• pp.105-111
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• 2014

Assistive listening devices (ALDs) refer to various types of amplification equipment designed to improve the communication of individuals with hard of hearing to enhance the accessibility to speech signal when individual hearing instruments are not sufficient. There are many types of ALDs to overcome a triangle of speech to noise ratio (SNR) problems, noise, distance, and reverberation. ALDs vary in their internal electronic mechanisms ranging from simple hard-wire microphone-amplifier units to more sophisticated broadcasting systems. They usually use microphones to capture an audio source and broadcast it wirelessly over a frequency modulation (FM), infra-red, induction loop, or other transmission techniques. The seven types of ALDs are introduced including hardwire devices, FM sound system, infra-red sound system, induction loop system, telephone listening devices, television, and alert/alarm system. Further development of digital wireless technology in hearing instruments will make possible direct communication with ALDs without any accessories in the near future. There are two technology solutions for digital wireless hearing instruments improving SNR and convenience. One is near-field magnetic induction combined with Bluetooth radio frequency (RF) transmission or proprietary RF transmission and the other is proprietary RF transmission alone. Recently launched digital wireless hearing aid applying this new technology can communicate from the hearing instrument to personal computer, phones, Wi-Fi, alert systems, and ALDs via iPhone, iPad, and iPod. However, it comes with its own iOS application offering a range of features but there is no option for Android users as of this moment.