• Journal of Korea Multimedia Society
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• v.21 no.9
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• pp.1110-1118
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• 2018

Personal sound amplifiers(PSAPs) provide accessible and affordable healthcare to individuals with hearing disability. Many studies are in progress for affordable PSAP development, but still people do not have a best fitting profile for the PSAP depending on their hearing test. As a result, they do not have a personalized and profiled music and sound, which are very helpful for those who has hearing problems. In this paper, we propose a device and mobile system to provide music with an equalizer value according to the hearing condition of an individual to prevent the hearing loss. In order to overcome the limit of frequency band of the equalizer in a smart phone, we developed bluetooth controlled equalizer based on the fitting profiles.

• Journal of Biomedical Engineering Research
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• v.26 no.3
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• pp.177-184
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• 2005

In this paper, we proposed a new hearing aid algorithm to improve SNR(signal to noise ratio) of noisy speech signal and speech perception. The proposed hearing aid algorithm is a multi-band loudness compensation based independent component analysis (ICA). The proposed algorithm was compared with a conventional spectral subtraction algorithm on behind-the-ear type hearing aid. The proposed algorithm successfully separated a target speech signal from background noise and from a mixture of the speech signals. The algorithms were compared each other by means of SNR. The average improvement of SNR by ICA based algorithm was 16.64dB, whereas spectral subtraction algorithm was 8.67dB. From the clinical tests, we concluded that our proposed algorithm would help hearing aid user to hear clearly a target speech in noisy conditions.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1820-1825
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• 2005

DSP chip parameters of a digital hearing aid (HA) should be optimally selected or fitted for hearing impaired persons. The more precise parameter fitting guarantees the better compensation of the hearing loss (HL). Digital HAs adopt DSP chips for more precise fitting of various HL threshold curve patterns. A specific DSP chip such as Gennum GB3211 was designed and manufactured in order to match up to about 4.7 billion different possible HL cases with combination of 7 limited parameters. This paper deals with a digital HA fitting program which is developed for optimal fitting of GB3211 DSP chip parameters. The fitting program has completed features from audiogram input to DSP chip interface. The compensation effects of the microphone and the receiver are also included. The paper shows some application examples.

• Journal of Institute of Control, Robotics and Systems
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• v.12 no.6
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• pp.530-538
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• 2006

DSP chip parameters of a digital hearing aid (HA) should be optimally selected or fitted for hearing impaired persons. The more precise parameter fitting guarantees the better compensation of the hearing loss (HL). Digital HAs adopt DSP chips for more precise fitting of various HL threshold curve patterns. A specific DSP chip such as Gennum GB3211 was designed and manufactured in order to match up to about 4.7 billion different possible HL cases with combination of 7 limited parameters. This paper deals with a digital HA fitting program which is developed for optimal fitting of GB3211 DSP chip parameters. The fitting program has completed features from audiogram input to DSP chip interface. The compensation effects of the microphone and the receiver are also included. The paper shows some application examples.

• 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.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.415-418
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• 2005

DSP chip parameters of a digital hearing aid (HA) should be optimally selected or fitted for hearing impaired persons. The more precise parameter fitting guarantees the better compensation of the hearing loss (HL). Digital HAs adopt DSP chips for more precise fitting of various HL threshold curve patterns. A specific DSP chip such as Gennum GB3211 was designed and manufactured in order to match up to about 4.7 billion different possible HL cases with combination of 7 limited parameters. This paper deals with a digital HA fitting program which is developed for optimal fitting of GB3211 DSP chip parameters. The fitting program has completed feature from audiogram input to DSP chip interface. The compensation effects of the microphone and the receiver are also included. The paper shows some application examples.

• 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 Biomedical Engineering Research
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• v.18 no.4
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• pp.457-466
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• 1997

We developed a new hearing aid telephone which helps the hearing impaired person to improve the listening ability of telephone speech. Recently, the hearing impaired person and the elderly who has hearing loss have been continuously increased and their desire for participating society as a producer has been increased also. So they strong1y want the hearing aid devices which make compensation fortheir handicap. The hearing aid telephone is one of the basic aid devices that helps the hearing impaired to communicate well with other poeple and to acquire easily useful information through the phone. We analyze the hearing ability of the hearing impaired, design the new model of the hearing aid telephone and test the telephone in three fields-electrical, word perception, user test. Our new tolephone has lour band pass filter channels and the center frequencies of these filters are 500, 1000, 2000, 3000Hz which are considered psychoacoustic factors and telephone line characteristics. The hearing impaired can adjust the total gain characteristics of receiving sound to his hearing ability by setting four volumes in the telelphone. This procedure is called fitting which is a very important factor for the hearing impaired to take meaning of speech. The total gain of this telephone is over 20dB from 250Hz to 3200Hz range. From the results of the tests we certify that our new model is better for the hearing impaired to understand the meaning or telephone speech than the old general models. The next step of developing the hearing aid telephone is to study about compressing sidetone and noise, dividing frequency bands, selecting hearing aid pattern and compensating psychoacoustic loudness. we expect that the advanced hearing aid telephone can be developed by the research about speech perception characteristics of the hearing impaired in engineering and clinical side.

• Proceedings of the KOSOMBE Conference
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• v.1995 no.11
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• pp.121-123
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• 1995

This paper presents a new method which compensates loss of loudness for digital hearing aids. Loudness grows more rapidly in frequency domain with substantial shifts of hearing threshold, so that loud sounds reach the uncomfortable sound level (UCL) at about the same physical stimulus level as with normal hearing. The result is a compression of the available dynamic range of hearing. Many techniques have been developed to compensate for hearing losses. In this paper, we propose a digital hearing aid which uses a single digital filter for reducing distortion and the fuzzy function to calculate gain factors. This function describes how much gain is needed for every frequency to restore loudness perception of a normal ear.

• Transactions of the Korean Society of Machine Tool Engineers
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• v.17 no.3
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• pp.128-132
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• 2008

The noise levels and individual employee noise exposure levels within a factory will determine the need for hearing conservation program. The difficulty in not having an effective hearing conservation program is the risk of hearing loss that employees may sustain. In the last few years the claims for hearing loss compensation have grown due to class action litigation brought against the employer and companies that have equipment in the factory alleged to have caused hearing loss. The Blower in the factory generates the noise of 98.3dB(A) in the frequency range of 2,000Hz, which may cause occupational hearing loss. By designing close type enclosures which are made of absorption material, about 24.4dB(A) reduction has been in the factory. It is demonstrated that this kind of enclosures can be effectively used to reduce the noise in the factory.