• The Journal of the Acoustical Society of Korea
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• v.24 no.2E
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• pp.38-45
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• 2005

The noise reduction of the In-The-Ear (ITE) hearing aid (HA) can be achieved by arrays of microphones. Each of the right and the left ears was assumed to have two HA microphones. These arrays of HA microphones produce particular patterns of directivity by some time delay between two microphones. The directivity pattern geometrically increase the S/N ratio. The boundary element method (BEM) was used for the three dimensional simulation of the HA directivity pattern with the two pairs' microphone arrays. The separation between two microphones was fixed to 10 mm. The time delay between the two microphones was calculated to produce the most narrow directivity pattern in the fore front of the head. The variation of the time delay was examined in accordance with input frequencies. This numerical analysis may be then applied for the calculation of the time delay parameter of the digital hearing aid DSP chip.

• 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.

• Speech Sciences
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• v.12 no.3
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• pp.79-89
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• 2005

Acoustic feedback between microphone and receiver can be effectively cancelled adaptive feedback cancellation algorithm. Although many speech sounds have non-Gaussian distribution, most algorithms were tested with speech like sounds whose distribution were Guassian type. In this paper, we proposed an adaptive feedback cancellation algorithm based on independent component analysis (ICA) for digital hearing aid. The algorithm was tested with not only Gaussian distribution but also Laplacian distribution. We verified that the proposed algorithm has better acoustic feedback cancelling performance than conventional normalized root mean square (NLMS) algorithm, especially speech like sounds with Laplacian distribution.

• 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.

• ETRI Journal
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• v.35 no.5
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• pp.819-826
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• 2013

A class-D audio amplifier for a digital hearing aid is described. The class-D amplifier operates with a pulse-code modulated (PCM) digital input and consists of an interpolation filter, a digital sigma-delta modulator (SDM), and an analog SDM, along with an H-bridge power switch. The noise of the power switch is suppressed by feeding it back to the input of the analog SDM. The interpolation filter removes the unwanted image tones of the PCM input, improving the linearity and power efficiency. The class-D amplifier is implemented in a 0.13-${\mu}m$ CMOS process. The maximum output power delivered to the receiver (speaker) is 1.19 mW. The measured total harmonic distortion plus noise is 0.015%, and the dynamic range is 86.0 dB. The class-D amplifier consumes 304 ${\mu}W$ from a 1.2-V power supply.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.755-758
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• 2004

A hearing aid performs the role of fitting in order to provide those with a hearing problem with a better hearing ability in ordinary circumstances. The existing Fitting system in the hearing aid is consisted somewhat complex, and simply performs the role of Fitting without any supervisions of those with a hearing problem. Such is these methods makes it unable to establish an organized supervision, and shows a problem in making high efficiency. In order to solve this problem mentioned as above, the based thesis have connected the Fitting program and the patient-management database program for the easy use by the user.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.2 no.2
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• pp.37-43
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• 2009

In this study, multi-channel digital hearing aid algorithm for low power system is proposed. First, MDCT(Modified Discrete Cosine Transform) method converts time domain of input speech signal into frequency domain of it. Output signal from MDCT makes a group about each channel, and then each channel signal adjusts a gain using LCF(Loudness Compensation Function) table depending on hearing loss of an auditory person. Finally, compensation signal is composed by TDAC and IMDCT. Its all of process make progress 16-bit fixed-point operation. We use fast-MDCT instead of MDCT for reducing system complexity and previously computed tables instead of log computation for estimating a gain. This algorithm evaluate through computer simulation.

• 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.

• 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.

• Speech Sciences
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• v.11 no.3
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• pp.175-180
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• 2004

In case of the hearing impairment with tinnitus and increased sound sensitivity, it is known that the patients tend to appeal the psychologically oriented social handicap rather than communication disability. The audiologist who is responsible for such patients in aural rehabilitation should pay special attention to the counseling techniques including tinnitus retain therapy (TRT), ear protector, noise generator, or specific acoustic training based on close cooperation and rapport. And then the audiologist should try to lessen their reaction to the tinnitus by using a hearing aid. This therapies tries to focus not a. total approach but a treatment to lessen the severity of tinnitus. This paper as a case report that a unilateral sharply slopped sensorineural hearing impaired person with tinnitus and increased sound sensitivity by using four channel digital signal processing (DSP) hearing aid with programming increment at low level (PILL).